Merge "mediaex: apply seccomp filter"
diff --git a/camera/Camera.cpp b/camera/Camera.cpp
index 9bf3134..cd3b84c 100644
--- a/camera/Camera.cpp
+++ b/camera/Camera.cpp
@@ -136,6 +136,15 @@
return c->setPreviewTarget(bufferProducer);
}
+status_t Camera::setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer)
+{
+ ALOGV("setVideoTarget(%p)", bufferProducer.get());
+ sp <ICamera> c = mCamera;
+ if (c == 0) return NO_INIT;
+ ALOGD_IF(bufferProducer == 0, "app passed NULL video surface");
+ return c->setVideoTarget(bufferProducer);
+}
+
// start preview mode
status_t Camera::startPreview()
{
@@ -145,13 +154,12 @@
return c->startPreview();
}
-status_t Camera::storeMetaDataInBuffers(bool enabled)
+status_t Camera::setVideoBufferMode(int32_t videoBufferMode)
{
- ALOGV("storeMetaDataInBuffers: %s",
- enabled? "true": "false");
+ ALOGV("setVideoBufferMode: %d", videoBufferMode);
sp <ICamera> c = mCamera;
if (c == 0) return NO_INIT;
- return c->storeMetaDataInBuffers(enabled);
+ return c->setVideoBufferMode(videoBufferMode);
}
// start recording mode, must call setPreviewTarget first
diff --git a/camera/ICamera.cpp b/camera/ICamera.cpp
index 9943be6..cce5a9a 100644
--- a/camera/ICamera.cpp
+++ b/camera/ICamera.cpp
@@ -48,7 +48,8 @@
STOP_RECORDING,
RECORDING_ENABLED,
RELEASE_RECORDING_FRAME,
- STORE_META_DATA_IN_BUFFERS,
+ SET_VIDEO_BUFFER_MODE,
+ SET_VIDEO_BUFFER_TARGET,
};
class BpCamera: public BpInterface<ICamera>
@@ -151,13 +152,13 @@
remote()->transact(RELEASE_RECORDING_FRAME, data, &reply);
}
- status_t storeMetaDataInBuffers(bool enabled)
+ status_t setVideoBufferMode(int32_t videoBufferMode)
{
- ALOGV("storeMetaDataInBuffers: %s", enabled? "true": "false");
+ ALOGV("setVideoBufferMode: %d", videoBufferMode);
Parcel data, reply;
data.writeInterfaceToken(ICamera::getInterfaceDescriptor());
- data.writeInt32(enabled);
- remote()->transact(STORE_META_DATA_IN_BUFFERS, data, &reply);
+ data.writeInt32(videoBufferMode);
+ remote()->transact(SET_VIDEO_BUFFER_MODE, data, &reply);
return reply.readInt32();
}
@@ -268,6 +269,17 @@
remote()->transact(UNLOCK, data, &reply);
return reply.readInt32();
}
+
+ status_t setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer)
+ {
+ ALOGV("setVideoTarget");
+ Parcel data, reply;
+ data.writeInterfaceToken(ICamera::getInterfaceDescriptor());
+ sp<IBinder> b(IInterface::asBinder(bufferProducer));
+ data.writeStrongBinder(b);
+ remote()->transact(SET_VIDEO_BUFFER_TARGET, data, &reply);
+ return reply.readInt32();
+ }
};
IMPLEMENT_META_INTERFACE(Camera, "android.hardware.ICamera");
@@ -339,11 +351,11 @@
releaseRecordingFrame(mem);
return NO_ERROR;
} break;
- case STORE_META_DATA_IN_BUFFERS: {
- ALOGV("STORE_META_DATA_IN_BUFFERS");
+ case SET_VIDEO_BUFFER_MODE: {
+ ALOGV("SET_VIDEO_BUFFER_MODE");
CHECK_INTERFACE(ICamera, data, reply);
- bool enabled = data.readInt32();
- reply->writeInt32(storeMetaDataInBuffers(enabled));
+ int32_t mode = data.readInt32();
+ reply->writeInt32(setVideoBufferMode(mode));
return NO_ERROR;
} break;
case PREVIEW_ENABLED: {
@@ -415,6 +427,14 @@
reply->writeInt32(unlock());
return NO_ERROR;
} break;
+ case SET_VIDEO_BUFFER_TARGET: {
+ ALOGV("SET_VIDEO_BUFFER_TARGET");
+ CHECK_INTERFACE(ICamera, data, reply);
+ sp<IGraphicBufferProducer> st =
+ interface_cast<IGraphicBufferProducer>(data.readStrongBinder());
+ reply->writeInt32(setVideoTarget(st));
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/drm/mediadrm/plugins/clearkey/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/DrmPlugin.h
index ba4aefe..9095045 100644
--- a/drm/mediadrm/plugins/clearkey/DrmPlugin.h
+++ b/drm/mediadrm/plugins/clearkey/DrmPlugin.h
@@ -105,10 +105,6 @@
return android::ERROR_DRM_CANNOT_HANDLE;
}
- virtual status_t unprovisionDevice() {
- return android::ERROR_DRM_CANNOT_HANDLE;
- }
-
virtual status_t getSecureStops(List<Vector<uint8_t> >& secureStops) {
UNUSED(secureStops);
return android::ERROR_DRM_CANNOT_HANDLE;
diff --git a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
index 851ad2c..c856d7d 100644
--- a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
+++ b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
@@ -312,12 +312,6 @@
return OK;
}
- status_t MockDrmPlugin::unprovisionDevice()
- {
- ALOGD("MockDrmPlugin::unprovisionDevice()");
- return OK;
- }
-
status_t MockDrmPlugin::getSecureStop(Vector<uint8_t> const & /* ssid */,
Vector<uint8_t> & secureStop)
{
diff --git a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.h b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.h
index d0f2ddb..db266f9 100644
--- a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.h
+++ b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.h
@@ -86,8 +86,6 @@
Vector<uint8_t> &certificate,
Vector<uint8_t> &wrappedKey);
- status_t unprovisionDevice();
-
status_t getSecureStops(List<Vector<uint8_t> > &secureStops);
status_t getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop);
status_t releaseSecureStops(Vector<uint8_t> const &ssRelease);
diff --git a/include/camera/Camera.h b/include/camera/Camera.h
index 2b60842..f7bf29c 100644
--- a/include/camera/Camera.h
+++ b/include/camera/Camera.h
@@ -126,8 +126,15 @@
// send command to camera driver
status_t sendCommand(int32_t cmd, int32_t arg1, int32_t arg2);
- // tell camera hal to store meta data or real YUV in video buffers.
- status_t storeMetaDataInBuffers(bool enabled);
+ // Tell camera how to pass video buffers. videoBufferMode is one of VIDEO_BUFFER_MODE_*.
+ // Returns OK if the specified video buffer mode is supported. If videoBufferMode is
+ // VIDEO_BUFFER_MODE_BUFFER_QUEUE, setVideoTarget() must be called before starting
+ // video recording.
+ status_t setVideoBufferMode(int32_t videoBufferMode);
+
+ // Set the video buffer producer for camera to use in VIDEO_BUFFER_MODE_BUFFER_QUEUE
+ // mode.
+ status_t setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer);
void setListener(const sp<CameraListener>& listener);
void setRecordingProxyListener(const sp<ICameraRecordingProxyListener>& listener);
diff --git a/include/camera/ICamera.h b/include/camera/ICamera.h
index b025735..e35c3a4 100644
--- a/include/camera/ICamera.h
+++ b/include/camera/ICamera.h
@@ -36,6 +36,15 @@
* Keep up-to-date with ICamera.aidl in frameworks/base
*/
public:
+ enum {
+ // Pass real YUV data in video buffers through ICameraClient.dataCallbackTimestamp().
+ VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV = 0,
+ // Pass metadata in video buffers through ICameraClient.dataCallbackTimestamp().
+ VIDEO_BUFFER_MODE_DATA_CALLBACK_METADATA = 1,
+ // Pass video buffers through IGraphicBufferProducer set with setVideoTarget().
+ VIDEO_BUFFER_MODE_BUFFER_QUEUE = 2,
+ };
+
DECLARE_META_INTERFACE(Camera);
virtual void disconnect() = 0;
@@ -109,8 +118,16 @@
// send command to camera driver
virtual status_t sendCommand(int32_t cmd, int32_t arg1, int32_t arg2) = 0;
- // tell the camera hal to store meta data or real YUV data in video buffers.
- virtual status_t storeMetaDataInBuffers(bool enabled) = 0;
+
+ // Tell camera how to pass video buffers. videoBufferMode is one of VIDEO_BUFFER_MODE_*.
+ // Returns OK if the specified video buffer mode is supported. If videoBufferMode is
+ // VIDEO_BUFFER_MODE_BUFFER_QUEUE, setVideoTarget() must be called before starting video
+ // recording.
+ virtual status_t setVideoBufferMode(int32_t videoBufferMode) = 0;
+
+ // Set the video buffer producer for camera to use in VIDEO_BUFFER_MODE_BUFFER_QUEUE mode.
+ virtual status_t setVideoTarget(
+ const sp<IGraphicBufferProducer>& bufferProducer) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/IDrm.h b/include/media/IDrm.h
index 9449beb6..fd51fd0 100644
--- a/include/media/IDrm.h
+++ b/include/media/IDrm.h
@@ -71,8 +71,6 @@
Vector<uint8_t> &certificate,
Vector<uint8_t> &wrappedKey) = 0;
- virtual status_t unprovisionDevice() = 0;
-
virtual status_t getSecureStops(List<Vector<uint8_t> > &secureStops) = 0;
virtual status_t getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop) = 0;
diff --git a/include/media/stagefright/CameraSource.h b/include/media/stagefright/CameraSource.h
index 069e897..769adf8 100644
--- a/include/media/stagefright/CameraSource.h
+++ b/include/media/stagefright/CameraSource.h
@@ -23,6 +23,7 @@
#include <camera/ICamera.h>
#include <camera/ICameraRecordingProxyListener.h>
#include <camera/CameraParameters.h>
+#include <gui/BufferItemConsumer.h>
#include <utils/List.h>
#include <utils/RefBase.h>
#include <utils/String16.h>
@@ -122,6 +123,12 @@
virtual void signalBufferReturned(MediaBuffer* buffer);
protected:
+
+ /**
+ * The class for listening to BnCameraRecordingProxyListener. This is used to receive video
+ * buffers in VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV and VIDEO_BUFFER_MODE_DATA_CALLBACK_METADATA
+ * mode. When a frame is available, CameraSource::dataCallbackTimestamp() will be called.
+ */
class ProxyListener: public BnCameraRecordingProxyListener {
public:
ProxyListener(const sp<CameraSource>& source);
@@ -132,6 +139,28 @@
sp<CameraSource> mSource;
};
+ /**
+ * The class for listening to BufferQueue's onFrameAvailable. This is used to receive video
+ * buffers in VIDEO_BUFFER_MODE_BUFFER_QUEUE mode. When a frame is available,
+ * CameraSource::processBufferQueueFrame() will be called.
+ */
+ class BufferQueueListener : public Thread, public BufferItemConsumer::FrameAvailableListener {
+ public:
+ BufferQueueListener(const sp<BufferItemConsumer> &consumer,
+ const sp<CameraSource> &cameraSource);
+ virtual void onFrameAvailable(const BufferItem& item);
+ virtual bool threadLoop();
+ private:
+ static const nsecs_t kFrameAvailableTimeout = 50000000; // 50ms
+
+ sp<BufferItemConsumer> mConsumer;
+ sp<CameraSource> mCameraSource;
+
+ Mutex mLock;
+ Condition mFrameAvailableSignal;
+ bool mFrameAvailable;
+ };
+
// isBinderAlive needs linkToDeath to work.
class DeathNotifier: public IBinder::DeathRecipient {
public:
@@ -204,11 +233,29 @@
int32_t mNumGlitches;
int64_t mGlitchDurationThresholdUs;
bool mCollectStats;
- bool mIsMetaDataStoredInVideoBuffers;
+
+ // The mode video buffers are received from camera. One of VIDEO_BUFFER_MODE_*.
+ int32_t mVideoBufferMode;
+
+ /**
+ * The following variables are used in VIDEO_BUFFER_MODE_BUFFER_QUEUE mode.
+ */
+ static const size_t kConsumerBufferCount = 8;
+ // Consumer and producer of the buffer queue between this class and camera.
+ sp<BufferItemConsumer> mVideoBufferConsumer;
+ sp<IGraphicBufferProducer> mVideoBufferProducer;
+ // Memory used to send the buffers to encoder, where sp<IMemory> stores VideoNativeMetadata.
+ sp<IMemoryHeap> mMemoryHeapBase;
+ List<sp<IMemory>> mMemoryBases;
+ // A mapping from ANativeWindowBuffer sent to encoder to BufferItem received from camera.
+ // This is protected by mLock.
+ KeyedVector<ANativeWindowBuffer*, BufferItem> mReceivedBufferItemMap;
+ sp<BufferQueueListener> mBufferQueueListener;
void releaseQueuedFrames();
void releaseOneRecordingFrame(const sp<IMemory>& frame);
-
+ // Process a buffer item received in BufferQueueListener.
+ void processBufferQueueFrame(const BufferItem& buffer);
status_t init(const sp<ICamera>& camera, const sp<ICameraRecordingProxy>& proxy,
int32_t cameraId, const String16& clientName, uid_t clientUid,
@@ -219,6 +266,10 @@
int32_t cameraId, const String16& clientName, uid_t clientUid,
Size videoSize, int32_t frameRate, bool storeMetaDataInVideoBuffers);
+ // Initialize the buffer queue used in VIDEO_BUFFER_MODE_BUFFER_QUEUE mode.
+ status_t initBufferQueue(uint32_t width, uint32_t height, uint32_t format,
+ android_dataspace dataSpace, uint32_t bufferCount);
+
status_t isCameraAvailable(const sp<ICamera>& camera,
const sp<ICameraRecordingProxy>& proxy,
int32_t cameraId,
@@ -236,6 +287,10 @@
status_t checkFrameRate(const CameraParameters& params,
int32_t frameRate);
+ // Check if this frame should be skipped based on the frame's timestamp in microsecond.
+ // mLock must be locked before calling this function.
+ bool shouldSkipFrameLocked(int64_t timestampUs);
+
void stopCameraRecording();
status_t reset();
diff --git a/include/media/stagefright/MediaBuffer.h b/include/media/stagefright/MediaBuffer.h
index c8a50e8..1e0c7d4 100644
--- a/include/media/stagefright/MediaBuffer.h
+++ b/include/media/stagefright/MediaBuffer.h
@@ -22,6 +22,7 @@
#include <pthread.h>
+#include <binder/MemoryDealer.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
@@ -93,6 +94,8 @@
private:
friend class MediaBufferGroup;
friend class OMXDecoder;
+ friend class BnMediaSource;
+ friend class BpMediaSource;
// For use by OMXDecoder, reference count must be 1, drop reference
// count to 0 without signalling the observer.
@@ -118,6 +121,7 @@
MediaBuffer(const MediaBuffer &);
MediaBuffer &operator=(const MediaBuffer &);
+ sp<IMemory> mMemory;
};
} // namespace android
diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
new file mode 100644
index 0000000..324ebbb
--- /dev/null
+++ b/media/audioserver/Android.mk
@@ -0,0 +1,35 @@
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ main_audioserver.cpp
+
+LOCAL_SHARED_LIBRARIES := \
+ libaudioflinger \
+ libaudiopolicyservice \
+ libbinder \
+ liblog \
+ libmedia \
+ libradioservice \
+ libsoundtriggerservice \
+ libutils \
+
+LOCAL_C_INCLUDES := \
+ frameworks/av/services/audioflinger \
+ frameworks/av/services/audiopolicy \
+ frameworks/av/services/audiopolicy/common/managerdefinitions/include \
+ frameworks/av/services/audiopolicy/common/include \
+ frameworks/av/services/audiopolicy/engine/interface \
+ frameworks/av/services/audiopolicy/service \
+ frameworks/av/services/radio \
+ frameworks/av/services/soundtrigger \
+ $(call include-path-for, audio-utils) \
+ external/sonic \
+
+LOCAL_MODULE := audioserver
+LOCAL_32_BIT_ONLY := true
+
+LOCAL_INIT_RC := audioserver.rc
+
+include $(BUILD_EXECUTABLE)
diff --git a/media/audioserver/audioserver.rc b/media/audioserver/audioserver.rc
new file mode 100644
index 0000000..1b39c8d
--- /dev/null
+++ b/media/audioserver/audioserver.rc
@@ -0,0 +1,6 @@
+service audioserver /system/bin/audioserver
+ class main
+ user audioserver
+ # media gid needed for /dev/fm (radio) and for /data/misc/media (tee)
+ group audio camera drmrpc inet media mediadrm net_bt net_bt_admin net_bw_acct
+ ioprio rt 4
diff --git a/media/audioserver/main_audioserver.cpp b/media/audioserver/main_audioserver.cpp
new file mode 100644
index 0000000..a7123aa
--- /dev/null
+++ b/media/audioserver/main_audioserver.cpp
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audioserver"
+//#define LOG_NDEBUG 0
+
+#include <binder/IPCThreadState.h>
+#include <binder/ProcessState.h>
+#include <binder/IServiceManager.h>
+#include <utils/Log.h>
+
+// from LOCAL_C_INCLUDES
+#include "AudioFlinger.h"
+#include "AudioPolicyService.h"
+#include "RadioService.h"
+#include "SoundTriggerHwService.h"
+
+using namespace android;
+
+int main(int argc __unused, char **argv __unused)
+{
+ signal(SIGPIPE, SIG_IGN);
+
+ // TODO: add logging b/24511453#3
+
+ sp<ProcessState> proc(ProcessState::self());
+ sp<IServiceManager> sm(defaultServiceManager());
+ ALOGI("ServiceManager: %p", sm.get());
+ AudioFlinger::instantiate();
+ AudioPolicyService::instantiate();
+ RadioService::instantiate();
+ SoundTriggerHwService::instantiate();
+ ProcessState::self()->startThreadPool();
+ IPCThreadState::self()->joinThreadPool();
+}
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 666b747..9d5d996 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -240,6 +240,7 @@
errno = 0;
(void) syscall(__NR_futex, &cblk->mFutex,
mClientInServer ? FUTEX_WAIT_PRIVATE : FUTEX_WAIT, old & ~CBLK_FUTEX_WAKE, ts);
+ status_t error = errno; // clock_gettime can affect errno
// update total elapsed time spent waiting
if (measure) {
struct timespec after;
@@ -257,7 +258,7 @@
before = after;
beforeIsValid = true;
}
- switch (errno) {
+ switch (error) {
case 0: // normal wakeup by server, or by binderDied()
case EWOULDBLOCK: // benign race condition with server
case EINTR: // wait was interrupted by signal or other spurious wakeup
@@ -265,7 +266,7 @@
// FIXME these error/non-0 status are being dropped
break;
default:
- status = errno;
+ status = error;
ALOGE("%s unexpected error %s", __func__, strerror(status));
goto end;
}
diff --git a/media/libmedia/IDrm.cpp b/media/libmedia/IDrm.cpp
index b1ad0c5..5a4591c 100644
--- a/media/libmedia/IDrm.cpp
+++ b/media/libmedia/IDrm.cpp
@@ -54,7 +54,6 @@
SIGN_RSA,
VERIFY,
SET_LISTENER,
- UNPROVISION_DEVICE,
GET_SECURE_STOP,
RELEASE_ALL_SECURE_STOPS
};
@@ -277,18 +276,6 @@
return reply.readInt32();
}
- virtual status_t unprovisionDevice() {
- Parcel data, reply;
- data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
-
- status_t status = remote()->transact(UNPROVISION_DEVICE, data, &reply);
- if (status != OK) {
- return status;
- }
-
- return reply.readInt32();
- }
-
virtual status_t getSecureStops(List<Vector<uint8_t> > &secureStops) {
Parcel data, reply;
data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
@@ -749,14 +736,6 @@
return OK;
}
- case UNPROVISION_DEVICE:
- {
- CHECK_INTERFACE(IDrm, data, reply);
- status_t result = unprovisionDevice();
- reply->writeInt32(result);
- return OK;
- }
-
case GET_SECURE_STOPS:
{
CHECK_INTERFACE(IDrm, data, reply);
diff --git a/media/libmedia/IMediaExtractor.cpp b/media/libmedia/IMediaExtractor.cpp
index 85362fd..76d5648 100644
--- a/media/libmedia/IMediaExtractor.cpp
+++ b/media/libmedia/IMediaExtractor.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#define LOG_NDEBUG 0
+//#define LOG_NDEBUG 0
#define LOG_TAG "BpMediaExtractor"
#include <utils/Log.h>
diff --git a/media/libmedia/IMediaSource.cpp b/media/libmedia/IMediaSource.cpp
index aec6255..fc9a123 100644
--- a/media/libmedia/IMediaSource.cpp
+++ b/media/libmedia/IMediaSource.cpp
@@ -18,6 +18,7 @@
#define LOG_TAG "BpMediaSource"
#include <utils/Log.h>
+#include <inttypes.h>
#include <stdint.h>
#include <sys/types.h>
@@ -34,9 +35,69 @@
STOP,
PAUSE,
GETFORMAT,
- READ
+ READ,
+ RELEASE_BUFFER
};
+enum {
+ NULL_BUFFER,
+ SHARED_BUFFER,
+ INLINE_BUFFER
+};
+
+class RemoteMediaBufferReleaser : public BBinder {
+public:
+ RemoteMediaBufferReleaser(MediaBuffer *buf) {
+ mBuf = buf;
+ }
+ ~RemoteMediaBufferReleaser() {
+ if (mBuf) {
+ ALOGW("RemoteMediaBufferReleaser dtor called while still holding buffer");
+ mBuf->release();
+ }
+ }
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0) {
+ if (code == RELEASE_BUFFER) {
+ mBuf->release();
+ mBuf = NULL;
+ return OK;
+ } else {
+ return BBinder::onTransact(code, data, reply, flags);
+ }
+ }
+private:
+ MediaBuffer *mBuf;
+};
+
+
+class RemoteMediaBufferWrapper : public MediaBuffer {
+public:
+ RemoteMediaBufferWrapper(sp<IMemory> mem, sp<IBinder> source);
+protected:
+ virtual ~RemoteMediaBufferWrapper();
+private:
+ sp<IMemory> mMemory;
+ sp<IBinder> mRemoteSource;
+};
+
+RemoteMediaBufferWrapper::RemoteMediaBufferWrapper(sp<IMemory> mem, sp<IBinder> source)
+: MediaBuffer(mem->pointer(), mem->size()) {
+ mMemory = mem;
+ mRemoteSource = source;
+}
+
+RemoteMediaBufferWrapper::~RemoteMediaBufferWrapper() {
+ mMemory.clear();
+ // Explicitly ask the remote side to release the buffer. We could also just clear
+ // mRemoteSource, but that doesn't immediately release the reference on the remote side.
+ Parcel data, reply;
+ mRemoteSource->transact(RELEASE_BUFFER, data, &reply);
+ mRemoteSource.clear();
+}
+
class BpMediaSource : public BpInterface<IMediaSource> {
public:
BpMediaSource(const sp<IBinder>& impl)
@@ -94,13 +155,26 @@
return ret;
}
// wrap the returned data in a MediaBuffer
- // XXX use a group, and use shared memory for transfer
ret = reply.readInt32();
- int32_t len = reply.readInt32();
- if (len < 0) {
- ALOGV("got status %d and len %d, returning NULL buffer", ret, len);
+ int32_t buftype = reply.readInt32();
+ if (buftype == SHARED_BUFFER) {
+ sp<IBinder> remote = reply.readStrongBinder();
+ sp<IBinder> binder = reply.readStrongBinder();
+ sp<IMemory> mem = interface_cast<IMemory>(binder);
+ if (mem == NULL) {
+ ALOGE("received NULL IMemory for shared buffer");
+ }
+ size_t offset = reply.readInt32();
+ size_t length = reply.readInt32();
+ MediaBuffer *buf = new RemoteMediaBufferWrapper(mem, remote);
+ buf->set_range(offset, length);
+ buf->meta_data()->updateFromParcel(reply);
+ *buffer = buf;
+ } else if (buftype == NULL_BUFFER) {
+ ALOGV("got status %d and NULL buffer", ret);
*buffer = NULL;
} else {
+ int32_t len = reply.readInt32();
ALOGV("got status %d and len %d", ret, len);
*buffer = new MediaBuffer(len);
reply.read((*buffer)->data(), len);
@@ -183,17 +257,34 @@
} else {
ret = read(&buf, NULL);
}
- // return data inside binder for now
- // XXX return data using shared memory
+
reply->writeInt32(ret);
if (buf != NULL) {
- ALOGV("ret %d, buflen %zu", ret, buf->range_length());
- reply->writeByteArray(buf->range_length(), (uint8_t*)buf->data() + buf->range_offset());
- buf->meta_data()->writeToParcel(*reply);
- buf->release();
+ size_t usedSize = buf->range_length();
+ // even if we're using shared memory, we might not want to use it, since for small
+ // sizes it's faster to copy data through the Binder transaction
+ if (buf->mMemory != NULL && usedSize >= 64 * 1024) {
+ ALOGV("buffer is using shared memory: %zu", usedSize);
+ reply->writeInt32(SHARED_BUFFER);
+ RemoteMediaBufferReleaser *wrapper = new RemoteMediaBufferReleaser(buf);
+ reply->writeStrongBinder(wrapper);
+ reply->writeStrongBinder(IInterface::asBinder(buf->mMemory));
+ reply->writeInt32(buf->range_offset());
+ reply->writeInt32(usedSize);
+ buf->meta_data()->writeToParcel(*reply);
+ } else {
+ // buffer is not in shared memory, or is small: copy it
+ if (buf->mMemory != NULL) {
+ ALOGV("%zu shared mem available, but only %zu used", buf->mMemory->size(), buf->range_length());
+ }
+ reply->writeInt32(INLINE_BUFFER);
+ reply->writeByteArray(buf->range_length(), (uint8_t*)buf->data() + buf->range_offset());
+ buf->meta_data()->writeToParcel(*reply);
+ buf->release();
+ }
} else {
ALOGV("ret %d, buf %p", ret, buf);
- reply->writeInt32(-1);
+ reply->writeInt32(NULL_BUFFER);
}
return NO_ERROR;
}
diff --git a/media/libmediaplayerservice/Drm.cpp b/media/libmediaplayerservice/Drm.cpp
index b9cfe80..321ccbf 100644
--- a/media/libmediaplayerservice/Drm.cpp
+++ b/media/libmediaplayerservice/Drm.cpp
@@ -516,24 +516,6 @@
return mPlugin->provideProvisionResponse(response, certificate, wrappedKey);
}
-status_t Drm::unprovisionDevice() {
- Mutex::Autolock autoLock(mLock);
-
- if (mInitCheck != OK) {
- return mInitCheck;
- }
-
- if (mPlugin == NULL) {
- return -EINVAL;
- }
-
- if (!checkPermission("android.permission.REMOVE_DRM_CERTIFICATES")) {
- return -EPERM;
- }
-
- return mPlugin->unprovisionDevice();
-}
-
status_t Drm::getSecureStops(List<Vector<uint8_t> > &secureStops) {
Mutex::Autolock autoLock(mLock);
diff --git a/media/libmediaplayerservice/Drm.h b/media/libmediaplayerservice/Drm.h
index 056723c..d40019b 100644
--- a/media/libmediaplayerservice/Drm.h
+++ b/media/libmediaplayerservice/Drm.h
@@ -77,8 +77,6 @@
Vector<uint8_t> &certificate,
Vector<uint8_t> &wrappedKey);
- virtual status_t unprovisionDevice();
-
virtual status_t getSecureStops(List<Vector<uint8_t> > &secureStops);
virtual status_t getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index be59bf2..7535934 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -1071,6 +1071,7 @@
return;
}
+ bool needRepostDrainVideoQueue = false;
int64_t delayUs;
int64_t nowUs = ALooper::GetNowUs();
int64_t realTimeUs;
@@ -1091,8 +1092,14 @@
} else if (!mVideoSampleReceived) {
// Always render the first video frame.
realTimeUs = nowUs;
- } else {
+ } else if (mAudioFirstAnchorTimeMediaUs < 0
+ || mMediaClock->getRealTimeFor(mediaTimeUs, &realTimeUs) == OK) {
realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
+ } else if (mediaTimeUs - mAudioFirstAnchorTimeMediaUs >= 0) {
+ needRepostDrainVideoQueue = true;
+ realTimeUs = nowUs;
+ } else {
+ realTimeUs = nowUs;
}
}
if (!mHasAudio) {
@@ -1105,15 +1112,25 @@
// received after this buffer, repost in 10 msec. Otherwise repost
// in 500 msec.
delayUs = realTimeUs - nowUs;
+ int64_t postDelayUs = -1;
if (delayUs > 500000) {
- int64_t postDelayUs = 500000;
+ postDelayUs = 500000;
if (mHasAudio && (mLastAudioBufferDrained - entry.mBufferOrdinal) <= 0) {
postDelayUs = 10000;
}
+ } else if (needRepostDrainVideoQueue) {
+ // CHECK(mPlaybackRate > 0);
+ // CHECK(mAudioFirstAnchorTimeMediaUs >= 0);
+ // CHECK(mediaTimeUs - mAudioFirstAnchorTimeMediaUs >= 0);
+ postDelayUs = mediaTimeUs - mAudioFirstAnchorTimeMediaUs;
+ postDelayUs /= mPlaybackRate;
+ }
+
+ if (postDelayUs >= 0) {
msg->setWhat(kWhatPostDrainVideoQueue);
msg->post(postDelayUs);
mVideoScheduler->restart();
- ALOGI("possible video time jump of %dms, retrying in %dms",
+ ALOGI("possible video time jump of %dms or uninitialized media clock, retrying in %dms",
(int)(delayUs / 1000), (int)(postDelayUs / 1000));
mDrainVideoQueuePending = true;
return;
diff --git a/media/libstagefright/CallbackDataSource.cpp b/media/libstagefright/CallbackDataSource.cpp
index e17fdf8..4c0a578 100644
--- a/media/libstagefright/CallbackDataSource.cpp
+++ b/media/libstagefright/CallbackDataSource.cpp
@@ -64,7 +64,7 @@
mIDataSource->readAt(offset + totalNumRead, numToRead);
// A negative return value represents an error. Pass it on.
if (numRead < 0) {
- return numRead;
+ return numRead == ERROR_END_OF_STREAM && totalNumRead > 0 ? totalNumRead : numRead;
}
// A zero return value signals EOS. Return the bytes read so far.
if (numRead == 0) {
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index 66280da..dab623b 100644
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -22,6 +22,9 @@
#include <OMX_Component.h>
#include <binder/IPCThreadState.h>
+#include <binder/MemoryBase.h>
+#include <binder/MemoryHeapBase.h>
+#include <media/hardware/HardwareAPI.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/CameraSource.h>
#include <media/stagefright/MediaDefs.h>
@@ -503,6 +506,77 @@
return err;
}
+status_t CameraSource::initBufferQueue(uint32_t width, uint32_t height,
+ uint32_t format, android_dataspace dataSpace, uint32_t bufferCount) {
+ ALOGV("initBufferQueue");
+
+ if (mVideoBufferConsumer != nullptr || mVideoBufferProducer != nullptr) {
+ ALOGE("%s: Buffer queue already exists", __FUNCTION__);
+ return ALREADY_EXISTS;
+ }
+
+ // Create a buffer queue.
+ sp<IGraphicBufferProducer> producer;
+ sp<IGraphicBufferConsumer> consumer;
+ BufferQueue::createBufferQueue(&producer, &consumer);
+
+ uint32_t usage = GRALLOC_USAGE_SW_READ_OFTEN;
+ if (format == HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED) {
+ usage = GRALLOC_USAGE_HW_VIDEO_ENCODER;
+ }
+
+ bufferCount += kConsumerBufferCount;
+
+ mVideoBufferConsumer = new BufferItemConsumer(consumer, usage, bufferCount);
+ mVideoBufferConsumer->setName(String8::format("StageFright-CameraSource"));
+ mVideoBufferProducer = producer;
+
+ status_t res = mVideoBufferConsumer->setDefaultBufferSize(width, height);
+ if (res != OK) {
+ ALOGE("%s: Could not set buffer dimensions %dx%d: %s (%d)", __FUNCTION__, width, height,
+ strerror(-res), res);
+ return res;
+ }
+
+ res = mVideoBufferConsumer->setDefaultBufferFormat(format);
+ if (res != OK) {
+ ALOGE("%s: Could not set buffer format %d: %s (%d)", __FUNCTION__, format,
+ strerror(-res), res);
+ return res;
+ }
+
+ res = mVideoBufferConsumer->setDefaultBufferDataSpace(dataSpace);
+ if (res != OK) {
+ ALOGE("%s: Could not set data space %d: %s (%d)", __FUNCTION__, dataSpace,
+ strerror(-res), res);
+ return res;
+ }
+
+ res = mCamera->setVideoTarget(mVideoBufferProducer);
+ if (res != OK) {
+ ALOGE("%s: Failed to set video target: %s (%d)", __FUNCTION__, strerror(-res), res);
+ return res;
+ }
+
+ // Create memory heap to store buffers as VideoNativeMetadata.
+ size_t bufferSize = sizeof(VideoNativeMetadata);
+ mMemoryHeapBase = new MemoryHeapBase(bufferSize * bufferCount, 0,
+ "StageFright-CameraSource-BufferHeap");
+ for (uint32_t i = 0; i < bufferCount; i++) {
+ mMemoryBases.push_back(new MemoryBase(mMemoryHeapBase, i * bufferSize, bufferSize));
+ }
+
+ mBufferQueueListener = new BufferQueueListener(mVideoBufferConsumer, this);
+ res = mBufferQueueListener->run("CameraSource-BufferQueueListener");
+ if (res != OK) {
+ ALOGE("%s: Could not run buffer queue listener thread: %s (%d)", __FUNCTION__,
+ strerror(-res), res);
+ return res;
+ }
+
+ return OK;
+}
+
status_t CameraSource::initWithCameraAccess(
const sp<ICamera>& camera,
const sp<ICameraRecordingProxy>& proxy,
@@ -551,12 +625,23 @@
CHECK_EQ((status_t)OK, mCamera->setPreviewTarget(mSurface));
}
- // By default, do not store metadata in video buffers
- mIsMetaDataStoredInVideoBuffers = false;
- mCamera->storeMetaDataInBuffers(false);
+ // By default, store real data in video buffers.
+ mVideoBufferMode = ICamera::VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV;
if (storeMetaDataInVideoBuffers) {
- if (OK == mCamera->storeMetaDataInBuffers(true)) {
- mIsMetaDataStoredInVideoBuffers = true;
+ if (OK == mCamera->setVideoBufferMode(ICamera::VIDEO_BUFFER_MODE_BUFFER_QUEUE)) {
+ mVideoBufferMode = ICamera::VIDEO_BUFFER_MODE_BUFFER_QUEUE;
+ } else if (OK == mCamera->setVideoBufferMode(
+ ICamera::VIDEO_BUFFER_MODE_DATA_CALLBACK_METADATA)) {
+ mVideoBufferMode = ICamera::VIDEO_BUFFER_MODE_DATA_CALLBACK_METADATA;
+ }
+ }
+
+ if (mVideoBufferMode == ICamera::VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV) {
+ err = mCamera->setVideoBufferMode(ICamera::VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV);
+ if (err != OK) {
+ ALOGE("%s: Setting video buffer mode to VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV failed: "
+ "%s (err=%d)", __FUNCTION__, strerror(-err), err);
+ return err;
}
}
@@ -596,28 +681,41 @@
// will connect to the camera in ICameraRecordingProxy::startRecording.
int64_t token = IPCThreadState::self()->clearCallingIdentity();
status_t err;
- if (mNumInputBuffers > 0) {
+
+ if (mVideoBufferMode == ICamera::VIDEO_BUFFER_MODE_BUFFER_QUEUE) {
+ // Initialize buffer queue.
+ err = initBufferQueue(mVideoSize.width, mVideoSize.height, mEncoderFormat,
+ (android_dataspace_t)mEncoderDataSpace,
+ mNumInputBuffers > 0 ? mNumInputBuffers : 1);
+ if (err != OK) {
+ ALOGE("%s: Failed to initialize buffer queue: %s (err=%d)", __FUNCTION__,
+ strerror(-err), err);
+ return err;
+ }
+ } else {
+ if (mNumInputBuffers > 0) {
+ err = mCamera->sendCommand(
+ CAMERA_CMD_SET_VIDEO_BUFFER_COUNT, mNumInputBuffers, 0);
+
+ // This could happen for CameraHAL1 clients; thus the failure is
+ // not a fatal error
+ if (err != OK) {
+ ALOGW("Failed to set video buffer count to %d due to %d",
+ mNumInputBuffers, err);
+ }
+ }
+
err = mCamera->sendCommand(
- CAMERA_CMD_SET_VIDEO_BUFFER_COUNT, mNumInputBuffers, 0);
+ CAMERA_CMD_SET_VIDEO_FORMAT, mEncoderFormat, mEncoderDataSpace);
// This could happen for CameraHAL1 clients; thus the failure is
// not a fatal error
if (err != OK) {
- ALOGW("Failed to set video buffer count to %d due to %d",
- mNumInputBuffers, err);
+ ALOGW("Failed to set video encoder format/dataspace to %d, %d due to %d",
+ mEncoderFormat, mEncoderDataSpace, err);
}
}
- err = mCamera->sendCommand(
- CAMERA_CMD_SET_VIDEO_FORMAT, mEncoderFormat, mEncoderDataSpace);
-
- // This could happen for CameraHAL1 clients; thus the failure is
- // not a fatal error
- if (err != OK) {
- ALOGW("Failed to set video encoder format/dataspace to %d, %d due to %d",
- mEncoderFormat, mEncoderDataSpace, err);
- }
-
err = OK;
if (mCameraFlags & FLAGS_HOT_CAMERA) {
mCamera->unlock();
@@ -771,6 +869,14 @@
CHECK_EQ(mNumFramesReceived, mNumFramesEncoded + mNumFramesDropped);
}
+ if (mBufferQueueListener != nullptr) {
+ mBufferQueueListener->requestExit();
+ mBufferQueueListener->join();
+ mBufferQueueListener.clear();
+ }
+
+ mVideoBufferConsumer.clear();
+ mVideoBufferProducer.clear();
releaseCamera();
ALOGD("reset: X");
@@ -779,7 +885,33 @@
void CameraSource::releaseRecordingFrame(const sp<IMemory>& frame) {
ALOGV("releaseRecordingFrame");
- if (mCameraRecordingProxy != NULL) {
+
+ if (mVideoBufferMode == ICamera::VIDEO_BUFFER_MODE_BUFFER_QUEUE) {
+ // Return the buffer to buffer queue in VIDEO_BUFFER_MODE_BUFFER_QUEUE mode.
+ ssize_t offset;
+ size_t size;
+ sp<IMemoryHeap> heap = frame->getMemory(&offset, &size);
+ if (heap->getHeapID() != mMemoryHeapBase->getHeapID()) {
+ ALOGE("%s: Mismatched heap ID, ignoring release (got %x, expected %x)", __FUNCTION__,
+ heap->getHeapID(), mMemoryHeapBase->getHeapID());
+ return;
+ }
+
+ VideoNativeMetadata *payload = reinterpret_cast<VideoNativeMetadata*>(
+ (uint8_t*)heap->getBase() + offset);
+
+ // Find the corresponding buffer item for the native window buffer.
+ ssize_t index = mReceivedBufferItemMap.indexOfKey(payload->pBuffer);
+ if (index == NAME_NOT_FOUND) {
+ ALOGE("%s: Couldn't find buffer item for %p", __FUNCTION__, payload->pBuffer);
+ return;
+ }
+
+ BufferItem buffer = mReceivedBufferItemMap.valueAt(index);
+ mReceivedBufferItemMap.removeItemsAt(index);
+ mVideoBufferConsumer->releaseBuffer(buffer);
+ mMemoryBases.push_back(frame);
+ } else if (mCameraRecordingProxy != NULL) {
mCameraRecordingProxy->releaseRecordingFrame(frame);
} else if (mCamera != NULL) {
int64_t token = IPCThreadState::self()->clearCallingIdentity();
@@ -871,29 +1003,23 @@
return OK;
}
-void CameraSource::dataCallbackTimestamp(int64_t timestampUs,
- int32_t msgType __unused, const sp<IMemory> &data) {
- ALOGV("dataCallbackTimestamp: timestamp %lld us", (long long)timestampUs);
- Mutex::Autolock autoLock(mLock);
+bool CameraSource::shouldSkipFrameLocked(int64_t timestampUs) {
if (!mStarted || (mNumFramesReceived == 0 && timestampUs < mStartTimeUs)) {
ALOGV("Drop frame at %lld/%lld us", (long long)timestampUs, (long long)mStartTimeUs);
- releaseOneRecordingFrame(data);
- return;
+ return true;
}
// May need to skip frame or modify timestamp. Currently implemented
// by the subclass CameraSourceTimeLapse.
if (skipCurrentFrame(timestampUs)) {
- releaseOneRecordingFrame(data);
- return;
+ return true;
}
if (mNumFramesReceived > 0) {
if (timestampUs <= mLastFrameTimestampUs) {
ALOGW("Dropping frame with backward timestamp %lld (last %lld)",
(long long)timestampUs, (long long)mLastFrameTimestampUs);
- releaseOneRecordingFrame(data);
- return;
+ return true;
}
if (timestampUs - mLastFrameTimestampUs > mGlitchDurationThresholdUs) {
++mNumGlitches;
@@ -908,12 +1034,25 @@
if (timestampUs < mStartTimeUs) {
// Frame was captured before recording was started
// Drop it without updating the statistical data.
- releaseOneRecordingFrame(data);
- return;
+ return true;
}
mStartTimeUs = timestampUs - mStartTimeUs;
}
}
+
+ return false;
+}
+
+void CameraSource::dataCallbackTimestamp(int64_t timestampUs,
+ int32_t msgType __unused, const sp<IMemory> &data) {
+ ALOGV("dataCallbackTimestamp: timestamp %lld us", (long long)timestampUs);
+ Mutex::Autolock autoLock(mLock);
+
+ if (shouldSkipFrameLocked(timestampUs)) {
+ releaseOneRecordingFrame(data);
+ return;
+ }
+
++mNumFramesReceived;
CHECK(data != NULL && data->size() > 0);
@@ -925,9 +1064,97 @@
mFrameAvailableCondition.signal();
}
+CameraSource::BufferQueueListener::BufferQueueListener(const sp<BufferItemConsumer>& consumer,
+ const sp<CameraSource>& cameraSource) {
+ mConsumer = consumer;
+ mConsumer->setFrameAvailableListener(this);
+ mCameraSource = cameraSource;
+}
+
+void CameraSource::BufferQueueListener::onFrameAvailable(const BufferItem& /*item*/) {
+ ALOGV("%s: onFrameAvailable", __FUNCTION__);
+
+ Mutex::Autolock l(mLock);
+
+ if (!mFrameAvailable) {
+ mFrameAvailable = true;
+ mFrameAvailableSignal.signal();
+ }
+}
+
+bool CameraSource::BufferQueueListener::threadLoop() {
+ if (mConsumer == nullptr || mCameraSource == nullptr) {
+ return false;
+ }
+
+ {
+ Mutex::Autolock l(mLock);
+ while (!mFrameAvailable) {
+ if (mFrameAvailableSignal.waitRelative(mLock, kFrameAvailableTimeout) == TIMED_OUT) {
+ return true;
+ }
+ }
+ mFrameAvailable = false;
+ }
+
+ BufferItem buffer;
+ while (mConsumer->acquireBuffer(&buffer, 0) == OK) {
+ mCameraSource->processBufferQueueFrame(buffer);
+ }
+
+ return true;
+}
+
+void CameraSource::processBufferQueueFrame(const BufferItem& buffer) {
+ Mutex::Autolock autoLock(mLock);
+
+ int64_t timestampUs = buffer.mTimestamp / 1000;
+ if (shouldSkipFrameLocked(timestampUs)) {
+ mVideoBufferConsumer->releaseBuffer(buffer);
+ return;
+ }
+
+ if (mMemoryBases.empty()) {
+ ALOGW("%s: No available memory base. Dropping a recording frame.", __FUNCTION__);
+ mVideoBufferConsumer->releaseBuffer(buffer);
+ return;
+ }
+
+ ++mNumFramesReceived;
+
+ // Find a available memory slot to store the buffer as VideoNativeMetadata.
+ sp<IMemory> data = *mMemoryBases.begin();
+ mMemoryBases.erase(mMemoryBases.begin());
+
+ ssize_t offset;
+ size_t size;
+ sp<IMemoryHeap> heap = data->getMemory(&offset, &size);
+ VideoNativeMetadata *payload = reinterpret_cast<VideoNativeMetadata*>(
+ (uint8_t*)heap->getBase() + offset);
+ memset(payload, 0, sizeof(VideoNativeMetadata));
+ payload->eType = kMetadataBufferTypeANWBuffer;
+ payload->pBuffer = buffer.mGraphicBuffer->getNativeBuffer();
+ payload->nFenceFd = -1;
+
+ // Add the mapping so we can find the corresponding buffer item to release to the buffer queue
+ // when the encoder returns the native window buffer.
+ mReceivedBufferItemMap.add(payload->pBuffer, buffer);
+
+ mFramesReceived.push_back(data);
+ int64_t timeUs = mStartTimeUs + (timestampUs - mFirstFrameTimeUs);
+ mFrameTimes.push_back(timeUs);
+ ALOGV("initial delay: %" PRId64 ", current time stamp: %" PRId64,
+ mStartTimeUs, timeUs);
+ mFrameAvailableCondition.signal();
+}
+
bool CameraSource::isMetaDataStoredInVideoBuffers() const {
ALOGV("isMetaDataStoredInVideoBuffers");
- return mIsMetaDataStoredInVideoBuffers;
+
+ // Output buffers will contain metadata if camera sends us buffer in metadata mode or via
+ // buffer queue.
+ return (mVideoBufferMode == ICamera::VIDEO_BUFFER_MODE_DATA_CALLBACK_METADATA ||
+ mVideoBufferMode == ICamera::VIDEO_BUFFER_MODE_BUFFER_QUEUE);
}
CameraSource::ProxyListener::ProxyListener(const sp<CameraSource>& source) {
diff --git a/media/libstagefright/DataSource.cpp b/media/libstagefright/DataSource.cpp
index 5020c6c..163a527 100644
--- a/media/libstagefright/DataSource.cpp
+++ b/media/libstagefright/DataSource.cpp
@@ -48,6 +48,8 @@
#include <cutils/properties.h>
+#include <private/android_filesystem_config.h>
+
namespace android {
bool DataSource::getUInt16(off64_t offset, uint16_t *x) {
@@ -173,7 +175,10 @@
RegisterSniffer_l(SniffMP3);
RegisterSniffer_l(SniffAAC);
RegisterSniffer_l(SniffMPEG2PS);
- RegisterSniffer_l(SniffWVM);
+ if (getuid() == AID_MEDIA) {
+ // WVM only in the media server process
+ RegisterSniffer_l(SniffWVM);
+ }
RegisterSniffer_l(SniffMidi);
char value[PROPERTY_VALUE_MAX];
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 38eb352..8c61108 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -423,6 +423,13 @@
if (!format->findInt32("rotation-degrees", &mRotationDegrees)) {
mRotationDegrees = 0;
}
+
+ // Prevent possible integer overflow in downstream code.
+ if (mInitIsEncoder
+ && (uint64_t)mVideoWidth * mVideoHeight > (uint64_t)INT32_MAX / 4) {
+ ALOGE("buffer size is too big, width=%d, height=%d", mVideoWidth, mVideoHeight);
+ return BAD_VALUE;
+ }
}
msg->setMessage("format", format);
diff --git a/media/libstagefright/MediaExtractor.cpp b/media/libstagefright/MediaExtractor.cpp
index e1b0dd9..99e58f1 100644
--- a/media/libstagefright/MediaExtractor.cpp
+++ b/media/libstagefright/MediaExtractor.cpp
@@ -138,6 +138,17 @@
// remote extractor
ALOGV("get service manager");
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.extractor"));
+
+ // Check if it's WVM, since WVMExtractor needs to be created in the media server process,
+ // not the extractor process.
+ String8 mime8;
+ float confidence;
+ sp<AMessage> meta;
+ if (SniffWVM(source, &mime8, &confidence, &meta) &&
+ !strcasecmp(mime8, MEDIA_MIMETYPE_CONTAINER_WVM)) {
+ return new WVMExtractor(source);
+ }
+
if (binder != 0) {
sp<IMediaExtractorService> mediaExService(interface_cast<IMediaExtractorService>(binder));
sp<IMediaExtractor> ex = mediaExService->makeExtractor(RemoteDataSource::wrap(source), mime);
@@ -213,7 +224,7 @@
ret = new MatroskaExtractor(source);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2TS)) {
ret = new MPEG2TSExtractor(source);
- } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_WVM)) {
+ } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_WVM) && getuid() == AID_MEDIA) {
// Return now. WVExtractor should not have the DrmFlag set in the block below.
return new WVMExtractor(source);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC_ADTS)) {
diff --git a/media/libstagefright/codecs/amrwbenc/Android.mk b/media/libstagefright/codecs/amrwbenc/Android.mk
index fd6d007..5b90b33 100644
--- a/media/libstagefright/codecs/amrwbenc/Android.mk
+++ b/media/libstagefright/codecs/amrwbenc/Android.mk
@@ -114,7 +114,7 @@
LOCAL_CFLAGS += -Werror
LOCAL_CLANG := true
-LOCAL_SANITIZE := signed-integer-overflow
+#LOCAL_SANITIZE := signed-integer-overflow
include $(BUILD_STATIC_LIBRARY)
@@ -132,7 +132,7 @@
LOCAL_CFLAGS += -Werror
LOCAL_CLANG := true
-LOCAL_SANITIZE := signed-integer-overflow
+#LOCAL_SANITIZE := signed-integer-overflow
LOCAL_STATIC_LIBRARIES := \
libstagefright_amrwbenc
diff --git a/media/libstagefright/codecs/amrwbenc/inc/acelp.h b/media/libstagefright/codecs/amrwbenc/inc/acelp.h
index 5a1e536..97555d5 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/acelp.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/acelp.h
@@ -18,7 +18,7 @@
/*--------------------------------------------------------------------------*
* ACELP.H *
*--------------------------------------------------------------------------*
- * Function *
+ * Function *
*--------------------------------------------------------------------------*/
#ifndef __ACELP_H__
#define __ACELP_H__
@@ -33,68 +33,68 @@
Word16 median5(Word16 x[]);
void Autocorr(
- Word16 x[], /* (i) : Input signal */
- Word16 m, /* (i) : LPC order */
- Word16 r_h[], /* (o) : Autocorrelations (msb) */
- Word16 r_l[] /* (o) : Autocorrelations (lsb) */
- );
+ Word16 x[], /* (i) : Input signal */
+ Word16 m, /* (i) : LPC order */
+ Word16 r_h[], /* (o) : Autocorrelations (msb) */
+ Word16 r_l[] /* (o) : Autocorrelations (lsb) */
+ );
void Lag_window(
- Word16 r_h[], /* (i/o) : Autocorrelations (msb) */
- Word16 r_l[] /* (i/o) : Autocorrelations (lsb) */
- );
+ Word16 r_h[], /* (i/o) : Autocorrelations (msb) */
+ Word16 r_l[] /* (i/o) : Autocorrelations (lsb) */
+ );
void Init_Levinson(
- Word16 * mem /* output :static memory (18 words) */
- );
+ Word16 * mem /* output :static memory (18 words) */
+ );
void Levinson(
- Word16 Rh[], /* (i) : Rh[M+1] Vector of autocorrelations (msb) */
- Word16 Rl[], /* (i) : Rl[M+1] Vector of autocorrelations (lsb) */
- Word16 A[], /* (o) Q12 : A[M] LPC coefficients (m = 16) */
- Word16 rc[], /* (o) Q15 : rc[M] Reflection coefficients. */
- Word16 * mem /* (i/o) :static memory (18 words) */
- );
+ Word16 Rh[], /* (i) : Rh[M+1] Vector of autocorrelations (msb) */
+ Word16 Rl[], /* (i) : Rl[M+1] Vector of autocorrelations (lsb) */
+ Word16 A[], /* (o) Q12 : A[M] LPC coefficients (m = 16) */
+ Word16 rc[], /* (o) Q15 : rc[M] Reflection coefficients. */
+ Word16 * mem /* (i/o) :static memory (18 words) */
+ );
void Az_isp(
- Word16 a[], /* (i) Q12 : predictor coefficients */
- Word16 isp[], /* (o) Q15 : Immittance spectral pairs */
- Word16 old_isp[] /* (i) : old isp[] (in case not found M roots) */
- );
+ Word16 a[], /* (i) Q12 : predictor coefficients */
+ Word16 isp[], /* (o) Q15 : Immittance spectral pairs */
+ Word16 old_isp[] /* (i) : old isp[] (in case not found M roots) */
+ );
void Isp_Az(
- Word16 isp[], /* (i) Q15 : Immittance spectral pairs */
- Word16 a[], /* (o) Q12 : predictor coefficients (order = M) */
- Word16 m,
- Word16 adaptive_scaling /* (i) 0 : adaptive scaling disabled */
- /* 1 : adaptive scaling enabled */
- );
+ Word16 isp[], /* (i) Q15 : Immittance spectral pairs */
+ Word16 a[], /* (o) Q12 : predictor coefficients (order = M) */
+ Word16 m,
+ Word16 adaptive_scaling /* (i) 0 : adaptive scaling disabled */
+ /* 1 : adaptive scaling enabled */
+ );
void Isp_isf(
- Word16 isp[], /* (i) Q15 : isp[m] (range: -1<=val<1) */
- Word16 isf[], /* (o) Q15 : isf[m] normalized (range: 0.0<=val<=0.5) */
- Word16 m /* (i) : LPC order */
- );
+ Word16 isp[], /* (i) Q15 : isp[m] (range: -1<=val<1) */
+ Word16 isf[], /* (o) Q15 : isf[m] normalized (range: 0.0<=val<=0.5) */
+ Word16 m /* (i) : LPC order */
+ );
void Isf_isp(
- Word16 isf[], /* (i) Q15 : isf[m] normalized (range: 0.0<=val<=0.5) */
- Word16 isp[], /* (o) Q15 : isp[m] (range: -1<=val<1) */
- Word16 m /* (i) : LPC order */
- );
+ Word16 isf[], /* (i) Q15 : isf[m] normalized (range: 0.0<=val<=0.5) */
+ Word16 isp[], /* (o) Q15 : isp[m] (range: -1<=val<1) */
+ Word16 m /* (i) : LPC order */
+ );
void Int_isp(
- Word16 isp_old[], /* input : isps from past frame */
- Word16 isp_new[], /* input : isps from present frame */
- Word16 frac[], /* input : fraction for 3 first subfr (Q15) */
- Word16 Az[] /* output: LP coefficients in 4 subframes */
- );
+ Word16 isp_old[], /* input : isps from past frame */
+ Word16 isp_new[], /* input : isps from present frame */
+ Word16 frac[], /* input : fraction for 3 first subfr (Q15) */
+ Word16 Az[] /* output: LP coefficients in 4 subframes */
+ );
void Weight_a(
- Word16 a[], /* (i) Q12 : a[m+1] LPC coefficients */
- Word16 ap[], /* (o) Q12 : Spectral expanded LPC coefficients */
- Word16 gamma, /* (i) Q15 : Spectral expansion factor. */
- Word16 m /* (i) : LPC order. */
- );
+ Word16 a[], /* (i) Q12 : a[m+1] LPC coefficients */
+ Word16 ap[], /* (o) Q12 : Spectral expanded LPC coefficients */
+ Word16 gamma, /* (i) Q15 : Spectral expansion factor. */
+ Word16 m /* (i) : LPC order. */
+ );
/*-----------------------------------------------------------------*
@@ -102,214 +102,214 @@
*-----------------------------------------------------------------*/
void Qpisf_2s_46b(
- Word16 * isf1, /* (i) Q15 : ISF in the frequency domain (0..0.5) */
- Word16 * isf_q, /* (o) Q15 : quantized ISF (0..0.5) */
- Word16 * past_isfq, /* (io)Q15 : past ISF quantizer */
- Word16 * indice, /* (o) : quantization indices */
- Word16 nb_surv /* (i) : number of survivor (1, 2, 3 or 4) */
- );
+ Word16 * isf1, /* (i) Q15 : ISF in the frequency domain (0..0.5) */
+ Word16 * isf_q, /* (o) Q15 : quantized ISF (0..0.5) */
+ Word16 * past_isfq, /* (io)Q15 : past ISF quantizer */
+ Word16 * indice, /* (o) : quantization indices */
+ Word16 nb_surv /* (i) : number of survivor (1, 2, 3 or 4) */
+ );
void Qpisf_2s_36b(
- Word16 * isf1, /* (i) Q15 : ISF in the frequency domain (0..0.5) */
- Word16 * isf_q, /* (o) Q15 : quantized ISF (0..0.5) */
- Word16 * past_isfq, /* (io)Q15 : past ISF quantizer */
- Word16 * indice, /* (o) : quantization indices */
- Word16 nb_surv /* (i) : number of survivor (1, 2, 3 or 4) */
- );
+ Word16 * isf1, /* (i) Q15 : ISF in the frequency domain (0..0.5) */
+ Word16 * isf_q, /* (o) Q15 : quantized ISF (0..0.5) */
+ Word16 * past_isfq, /* (io)Q15 : past ISF quantizer */
+ Word16 * indice, /* (o) : quantization indices */
+ Word16 nb_surv /* (i) : number of survivor (1, 2, 3 or 4) */
+ );
void Dpisf_2s_46b(
- Word16 * indice, /* input: quantization indices */
- Word16 * isf_q, /* output: quantized ISF in frequency domain (0..0.5) */
- Word16 * past_isfq, /* i/0 : past ISF quantizer */
- Word16 * isfold, /* input : past quantized ISF */
- Word16 * isf_buf, /* input : isf buffer */
- Word16 bfi, /* input : Bad frame indicator */
- Word16 enc_dec
- );
+ Word16 * indice, /* input: quantization indices */
+ Word16 * isf_q, /* output: quantized ISF in frequency domain (0..0.5) */
+ Word16 * past_isfq, /* i/0 : past ISF quantizer */
+ Word16 * isfold, /* input : past quantized ISF */
+ Word16 * isf_buf, /* input : isf buffer */
+ Word16 bfi, /* input : Bad frame indicator */
+ Word16 enc_dec
+ );
void Dpisf_2s_36b(
- Word16 * indice, /* input: quantization indices */
- Word16 * isf_q, /* output: quantized ISF in frequency domain (0..0.5) */
- Word16 * past_isfq, /* i/0 : past ISF quantizer */
- Word16 * isfold, /* input : past quantized ISF */
- Word16 * isf_buf, /* input : isf buffer */
- Word16 bfi, /* input : Bad frame indicator */
- Word16 enc_dec
- );
+ Word16 * indice, /* input: quantization indices */
+ Word16 * isf_q, /* output: quantized ISF in frequency domain (0..0.5) */
+ Word16 * past_isfq, /* i/0 : past ISF quantizer */
+ Word16 * isfold, /* input : past quantized ISF */
+ Word16 * isf_buf, /* input : isf buffer */
+ Word16 bfi, /* input : Bad frame indicator */
+ Word16 enc_dec
+ );
void Qisf_ns(
- Word16 * isf1, /* input : ISF in the frequency domain (0..0.5) */
- Word16 * isf_q, /* output: quantized ISF */
- Word16 * indice /* output: quantization indices */
- );
+ Word16 * isf1, /* input : ISF in the frequency domain (0..0.5) */
+ Word16 * isf_q, /* output: quantized ISF */
+ Word16 * indice /* output: quantization indices */
+ );
void Disf_ns(
- Word16 * indice, /* input: quantization indices */
- Word16 * isf_q /* input : ISF in the frequency domain (0..0.5) */
- );
+ Word16 * indice, /* input: quantization indices */
+ Word16 * isf_q /* input : ISF in the frequency domain (0..0.5) */
+ );
Word16 Sub_VQ( /* output: return quantization index */
- Word16 * x, /* input : ISF residual vector */
- Word16 * dico, /* input : quantization codebook */
- Word16 dim, /* input : dimention of vector */
- Word16 dico_size, /* input : size of quantization codebook */
- Word32 * distance /* output: error of quantization */
- );
+ Word16 * x, /* input : ISF residual vector */
+ Word16 * dico, /* input : quantization codebook */
+ Word16 dim, /* input : dimention of vector */
+ Word16 dico_size, /* input : size of quantization codebook */
+ Word32 * distance /* output: error of quantization */
+ );
void Reorder_isf(
- Word16 * isf, /* (i/o) Q15: ISF in the frequency domain (0..0.5) */
- Word16 min_dist, /* (i) Q15 : minimum distance to keep */
- Word16 n /* (i) : number of ISF */
- );
+ Word16 * isf, /* (i/o) Q15: ISF in the frequency domain (0..0.5) */
+ Word16 min_dist, /* (i) Q15 : minimum distance to keep */
+ Word16 n /* (i) : number of ISF */
+ );
/*-----------------------------------------------------------------*
* filter prototypes *
*-----------------------------------------------------------------*/
void Init_Decim_12k8(
- Word16 mem[] /* output: memory (2*NB_COEF_DOWN) set to zeros */
- );
+ Word16 mem[] /* output: memory (2*NB_COEF_DOWN) set to zeros */
+ );
void Decim_12k8(
- Word16 sig16k[], /* input: signal to downsampling */
- Word16 lg, /* input: length of input */
- Word16 sig12k8[], /* output: decimated signal */
- Word16 mem[] /* in/out: memory (2*NB_COEF_DOWN) */
- );
+ Word16 sig16k[], /* input: signal to downsampling */
+ Word16 lg, /* input: length of input */
+ Word16 sig12k8[], /* output: decimated signal */
+ Word16 mem[] /* in/out: memory (2*NB_COEF_DOWN) */
+ );
void Init_HP50_12k8(Word16 mem[]);
void HP50_12k8(
- Word16 signal[], /* input/output signal */
- Word16 lg, /* lenght of signal */
- Word16 mem[] /* filter memory [6] */
- );
+ Word16 signal[], /* input/output signal */
+ Word16 lg, /* lenght of signal */
+ Word16 mem[] /* filter memory [6] */
+ );
void Init_HP400_12k8(Word16 mem[]);
void HP400_12k8(
- Word16 signal[], /* input/output signal */
- Word16 lg, /* lenght of signal */
- Word16 mem[] /* filter memory [6] */
- );
+ Word16 signal[], /* input/output signal */
+ Word16 lg, /* lenght of signal */
+ Word16 mem[] /* filter memory [6] */
+ );
void Init_Filt_6k_7k(Word16 mem[]);
void Filt_6k_7k(
- Word16 signal[], /* input: signal */
- Word16 lg, /* input: length of input */
- Word16 mem[] /* in/out: memory (size=30) */
- );
+ Word16 signal[], /* input: signal */
+ Word16 lg, /* input: length of input */
+ Word16 mem[] /* in/out: memory (size=30) */
+ );
void Filt_6k_7k_asm(
- Word16 signal[], /* input: signal */
- Word16 lg, /* input: length of input */
- Word16 mem[] /* in/out: memory (size=30) */
- );
+ Word16 signal[], /* input: signal */
+ Word16 lg, /* input: length of input */
+ Word16 mem[] /* in/out: memory (size=30) */
+ );
void LP_Decim2(
- Word16 x[], /* in/out: signal to process */
- Word16 l, /* input : size of filtering */
- Word16 mem[] /* in/out: memory (size=3) */
- );
+ Word16 x[], /* in/out: signal to process */
+ Word16 l, /* input : size of filtering */
+ Word16 mem[] /* in/out: memory (size=3) */
+ );
void Preemph(
- Word16 x[], /* (i/o) : input signal overwritten by the output */
- Word16 mu, /* (i) Q15 : preemphasis coefficient */
- Word16 lg, /* (i) : lenght of filtering */
- Word16 * mem /* (i/o) : memory (x[-1]) */
- );
+ Word16 x[], /* (i/o) : input signal overwritten by the output */
+ Word16 mu, /* (i) Q15 : preemphasis coefficient */
+ Word16 lg, /* (i) : lenght of filtering */
+ Word16 * mem /* (i/o) : memory (x[-1]) */
+ );
void Preemph2(
- Word16 x[], /* (i/o) : input signal overwritten by the output */
- Word16 mu, /* (i) Q15 : preemphasis coefficient */
- Word16 lg, /* (i) : lenght of filtering */
- Word16 * mem /* (i/o) : memory (x[-1]) */
- );
+ Word16 x[], /* (i/o) : input signal overwritten by the output */
+ Word16 mu, /* (i) Q15 : preemphasis coefficient */
+ Word16 lg, /* (i) : lenght of filtering */
+ Word16 * mem /* (i/o) : memory (x[-1]) */
+ );
void Deemph(
- Word16 x[], /* (i/o) : input signal overwritten by the output */
- Word16 mu, /* (i) Q15 : deemphasis factor */
- Word16 L, /* (i) : vector size */
- Word16 * mem /* (i/o) : memory (y[-1]) */
- );
+ Word16 x[], /* (i/o) : input signal overwritten by the output */
+ Word16 mu, /* (i) Q15 : deemphasis factor */
+ Word16 L, /* (i) : vector size */
+ Word16 * mem /* (i/o) : memory (y[-1]) */
+ );
void Deemph2(
- Word16 x[], /* (i/o) : input signal overwritten by the output */
- Word16 mu, /* (i) Q15 : deemphasis factor */
- Word16 L, /* (i) : vector size */
- Word16 * mem /* (i/o) : memory (y[-1]) */
- );
+ Word16 x[], /* (i/o) : input signal overwritten by the output */
+ Word16 mu, /* (i) Q15 : deemphasis factor */
+ Word16 L, /* (i) : vector size */
+ Word16 * mem /* (i/o) : memory (y[-1]) */
+ );
void Deemph_32(
- Word16 x_hi[], /* (i) : input signal (bit31..16) */
- Word16 x_lo[], /* (i) : input signal (bit15..4) */
- Word16 y[], /* (o) : output signal (x16) */
- Word16 mu, /* (i) Q15 : deemphasis factor */
- Word16 L, /* (i) : vector size */
- Word16 * mem /* (i/o) : memory (y[-1]) */
- );
+ Word16 x_hi[], /* (i) : input signal (bit31..16) */
+ Word16 x_lo[], /* (i) : input signal (bit15..4) */
+ Word16 y[], /* (o) : output signal (x16) */
+ Word16 mu, /* (i) Q15 : deemphasis factor */
+ Word16 L, /* (i) : vector size */
+ Word16 * mem /* (i/o) : memory (y[-1]) */
+ );
void Deemph_32_asm(
- Word16 x_hi[], /* (i) : input signal (bit31..16) */
- Word16 x_lo[], /* (i) : input signal (bit15..4) */
- Word16 y[], /* (o) : output signal (x16) */
- Word16 * mem /* (i/o) : memory (y[-1]) */
- );
+ Word16 x_hi[], /* (i) : input signal (bit31..16) */
+ Word16 x_lo[], /* (i) : input signal (bit15..4) */
+ Word16 y[], /* (o) : output signal (x16) */
+ Word16 * mem /* (i/o) : memory (y[-1]) */
+ );
void Convolve(
- Word16 x[], /* (i) : input vector */
- Word16 h[], /* (i) Q15 : impulse response */
- Word16 y[], /* (o) 12 bits: output vector */
- Word16 L /* (i) : vector size */
- );
+ Word16 x[], /* (i) : input vector */
+ Word16 h[], /* (i) Q15 : impulse response */
+ Word16 y[], /* (o) 12 bits: output vector */
+ Word16 L /* (i) : vector size */
+ );
void Convolve_asm(
- Word16 x[], /* (i) : input vector */
- Word16 h[], /* (i) Q15 : impulse response */
- Word16 y[], /* (o) 12 bits: output vector */
- Word16 L /* (i) : vector size */
- );
+ Word16 x[], /* (i) : input vector */
+ Word16 h[], /* (i) Q15 : impulse response */
+ Word16 y[], /* (o) 12 bits: output vector */
+ Word16 L /* (i) : vector size */
+ );
void Residu(
- Word16 a[], /* (i) Q12 : prediction coefficients */
- Word16 x[], /* (i) : speech (values x[-m..-1] are needed */
- Word16 y[], /* (o) : residual signal */
- Word16 lg /* (i) : size of filtering */
- );
+ Word16 a[], /* (i) Q12 : prediction coefficients */
+ Word16 x[], /* (i) : speech (values x[-m..-1] are needed */
+ Word16 y[], /* (o) : residual signal */
+ Word16 lg /* (i) : size of filtering */
+ );
void Residu_opt(
- Word16 a[], /* (i) Q12 : prediction coefficients */
- Word16 x[], /* (i) : speech (values x[-m..-1] are needed */
- Word16 y[], /* (o) : residual signal */
- Word16 lg /* (i) : size of filtering */
- );
+ Word16 a[], /* (i) Q12 : prediction coefficients */
+ Word16 x[], /* (i) : speech (values x[-m..-1] are needed */
+ Word16 y[], /* (o) : residual signal */
+ Word16 lg /* (i) : size of filtering */
+ );
void Syn_filt(
- Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
- Word16 x[], /* (i) : input signal */
- Word16 y[], /* (o) : output signal */
- Word16 lg, /* (i) : size of filtering */
- Word16 mem[], /* (i/o) : memory associated with this filtering. */
- Word16 update /* (i) : 0=no update, 1=update of memory. */
- );
+ Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
+ Word16 x[], /* (i) : input signal */
+ Word16 y[], /* (o) : output signal */
+ Word16 lg, /* (i) : size of filtering */
+ Word16 mem[], /* (i/o) : memory associated with this filtering. */
+ Word16 update /* (i) : 0=no update, 1=update of memory. */
+ );
void Syn_filt_asm(
- Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
- Word16 x[], /* (i) : input signal */
- Word16 y[], /* (o) : output signal */
- Word16 mem[] /* (i/o) : memory associated with this filtering. */
- );
+ Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
+ Word16 x[], /* (i) : input signal */
+ Word16 y[], /* (o) : output signal */
+ Word16 mem[] /* (i/o) : memory associated with this filtering. */
+ );
void Syn_filt_32(
- Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
- Word16 m, /* (i) : order of LP filter */
- Word16 exc[], /* (i) Qnew: excitation (exc[i] >> Qnew) */
- Word16 Qnew, /* (i) : exc scaling = 0(min) to 8(max) */
- Word16 sig_hi[], /* (o) /16 : synthesis high */
- Word16 sig_lo[], /* (o) /16 : synthesis low */
- Word16 lg /* (i) : size of filtering */
- );
+ Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
+ Word16 m, /* (i) : order of LP filter */
+ Word16 exc[], /* (i) Qnew: excitation (exc[i] >> Qnew) */
+ Word16 Qnew, /* (i) : exc scaling = 0(min) to 8(max) */
+ Word16 sig_hi[], /* (o) /16 : synthesis high */
+ Word16 sig_lo[], /* (o) /16 : synthesis low */
+ Word16 lg /* (i) : size of filtering */
+ );
void Syn_filt_32_asm(
- Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
- Word16 m, /* (i) : order of LP filter */
- Word16 exc[], /* (i) Qnew: excitation (exc[i] >> Qnew) */
- Word16 Qnew, /* (i) : exc scaling = 0(min) to 8(max) */
- Word16 sig_hi[], /* (o) /16 : synthesis high */
- Word16 sig_lo[], /* (o) /16 : synthesis low */
- Word16 lg /* (i) : size of filtering */
- );
+ Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
+ Word16 m, /* (i) : order of LP filter */
+ Word16 exc[], /* (i) Qnew: excitation (exc[i] >> Qnew) */
+ Word16 Qnew, /* (i) : exc scaling = 0(min) to 8(max) */
+ Word16 sig_hi[], /* (o) /16 : synthesis high */
+ Word16 sig_lo[], /* (o) /16 : synthesis low */
+ Word16 lg /* (i) : size of filtering */
+ );
/*-----------------------------------------------------------------*
* pitch prototypes *
*-----------------------------------------------------------------*/
@@ -443,12 +443,12 @@
Word16 nbbits, /* (i) : 20, 36, 44, 52, 64, 72 or 88 bits */
Word16 ser_size, /* (i) : bit rate */
Word16 _index[] /* (o) : index (20): 5+5+5+5 = 20 bits. */
- /* (o) : index (36): 9+9+9+9 = 36 bits. */
- /* (o) : index (44): 13+9+13+9 = 44 bits. */
- /* (o) : index (52): 13+13+13+13 = 52 bits. */
- /* (o) : index (64): 2+2+2+2+14+14+14+14 = 64 bits. */
- /* (o) : index (72): 10+2+10+2+10+14+10+14 = 72 bits. */
- /* (o) : index (88): 11+11+11+11+11+11+11+11 = 88 bits. */
+ /* (o) : index (36): 9+9+9+9 = 36 bits. */
+ /* (o) : index (44): 13+9+13+9 = 44 bits. */
+ /* (o) : index (52): 13+13+13+13 = 52 bits. */
+ /* (o) : index (64): 2+2+2+2+14+14+14+14 = 64 bits. */
+ /* (o) : index (72): 10+2+10+2+10+14+10+14 = 72 bits. */
+ /* (o) : index (88): 11+11+11+11+11+11+11+11 = 88 bits. */
);
void Pit_shrp(
diff --git a/media/libstagefright/codecs/amrwbenc/inc/basic_op.h b/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
index efa4a96..5808437 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
@@ -25,8 +25,8 @@
#define MAX_32 (Word32)0x7fffffffL
#define MIN_32 (Word32)0x80000000L
-#define MAX_16 (Word16)+32767 /* 0x7fff */
-#define MIN_16 (Word16)-32768 /* 0x8000 */
+#define MAX_16 (Word16)+32767 /* 0x7fff */
+#define MIN_16 (Word16)-32768 /* 0x8000 */
#define static_vo static __inline
@@ -41,22 +41,22 @@
#define L_negate(L_var1) (((L_var1) == (MIN_32)) ? (MAX_32) : (-(L_var1))) /* Long negate, 2*/
-#define extract_h(a) ((Word16)(a >> 16))
-#define extract_l(x) (Word16)((x))
-#define add1(a,b) (a + b)
-#define vo_L_msu(a,b,c) ( a - (( b * c ) << 1) )
+#define extract_h(a) ((Word16)(a >> 16))
+#define extract_l(x) (Word16)((x))
+#define add1(a,b) (a + b)
+#define vo_L_msu(a,b,c) ( a - (( b * c ) << 1) )
#define vo_mult32(a, b) ((a) * (b))
-#define vo_mult(a,b) (( a * b ) >> 15 )
-#define vo_L_mult(a,b) (((a) * (b)) << 1)
-#define vo_shr_r(var1, var2) ((var1+((Word16)(1L<<(var2-1))))>>var2)
-#define vo_sub(a,b) (a - b)
-#define vo_L_deposit_h(a) ((Word32)((a) << 16))
-#define vo_round(a) ((a + 0x00008000) >> 16)
-#define vo_extract_l(a) ((Word16)(a))
-#define vo_L_add(a,b) (a + b)
-#define vo_L_sub(a,b) (a - b)
-#define vo_mult_r(a,b) ((( a * b ) + 0x4000 ) >> 15 )
-#define vo_negate(a) (-a)
+#define vo_mult(a,b) (( a * b ) >> 15 )
+#define vo_L_mult(a,b) (((a) * (b)) << 1)
+#define vo_shr_r(var1, var2) ((var1+((Word16)(1L<<(var2-1))))>>var2)
+#define vo_sub(a,b) (a - b)
+#define vo_L_deposit_h(a) ((Word32)((a) << 16))
+#define vo_round(a) ((a + 0x00008000) >> 16)
+#define vo_extract_l(a) ((Word16)(a))
+#define vo_L_add(a,b) (a + b)
+#define vo_L_sub(a,b) (a - b)
+#define vo_mult_r(a,b) ((( a * b ) + 0x4000 ) >> 15 )
+#define vo_negate(a) (-a)
#define vo_L_shr_r(L_var1, var2) ((L_var1+((Word32)(1L<<(var2-1))))>>var2)
@@ -65,25 +65,25 @@
| Prototypes for basic arithmetic operators |
|___________________________________________________________________________|
*/
-static_vo Word16 add (Word16 var1, Word16 var2); /* Short add,1 */
-static_vo Word16 sub (Word16 var1, Word16 var2); /* Short sub,1 */
+static_vo Word16 add (Word16 var1, Word16 var2); /* Short add,1 */
+static_vo Word16 sub (Word16 var1, Word16 var2); /* Short sub,1 */
static_vo Word16 shl (Word16 var1, Word16 var2); /* Short shift left, 1 */
static_vo Word16 shr (Word16 var1, Word16 var2); /* Short shift right, 1 */
static_vo Word16 mult (Word16 var1, Word16 var2); /* Short mult, 1 */
static_vo Word32 L_mult (Word16 var1, Word16 var2); /* Long mult, 1 */
static_vo Word16 voround (Word32 L_var1); /* Round, 1 */
-static_vo Word32 L_mac (Word32 L_var3, Word16 var1, Word16 var2); /* Mac, 1 */
-static_vo Word32 L_msu (Word32 L_var3, Word16 var1, Word16 var2); /* Msu, 1 */
-static_vo Word32 L_add (Word32 L_var1, Word32 L_var2); /* Long add, 2 */
-static_vo Word32 L_sub (Word32 L_var1, Word32 L_var2); /* Long sub, 2 */
-static_vo Word16 mult_r (Word16 var1, Word16 var2); /* Mult with round, 2 */
-static_vo Word32 L_shl2(Word32 L_var1, Word16 var2); /* var2 > 0*/
-static_vo Word32 L_shl (Word32 L_var1, Word16 var2); /* Long shift left, 2 */
-static_vo Word32 L_shr (Word32 L_var1, Word16 var2); /* Long shift right, 2*/
-static_vo Word32 L_shr_r (Word32 L_var1, Word16 var2); /* Long shift right with round, 3 */
-static_vo Word16 norm_s (Word16 var1); /* Short norm, 15 */
-static_vo Word16 div_s (Word16 var1, Word16 var2); /* Short division, 18 */
-static_vo Word16 norm_l (Word32 L_var1); /* Long norm, 30 */
+static_vo Word32 L_mac (Word32 L_var3, Word16 var1, Word16 var2); /* Mac, 1 */
+static_vo Word32 L_msu (Word32 L_var3, Word16 var1, Word16 var2); /* Msu, 1 */
+static_vo Word32 L_add (Word32 L_var1, Word32 L_var2); /* Long add, 2 */
+static_vo Word32 L_sub (Word32 L_var1, Word32 L_var2); /* Long sub, 2 */
+static_vo Word16 mult_r (Word16 var1, Word16 var2); /* Mult with round, 2 */
+static_vo Word32 L_shl2(Word32 L_var1, Word16 var2); /* var2 > 0*/
+static_vo Word32 L_shl (Word32 L_var1, Word16 var2); /* Long shift left, 2 */
+static_vo Word32 L_shr (Word32 L_var1, Word16 var2); /* Long shift right, 2*/
+static_vo Word32 L_shr_r (Word32 L_var1, Word16 var2); /* Long shift right with round, 3 */
+static_vo Word16 norm_s (Word16 var1); /* Short norm, 15 */
+static_vo Word16 div_s (Word16 var1, Word16 var2); /* Short division, 18 */
+static_vo Word16 norm_l (Word32 L_var1); /* Long norm, 30 */
/*___________________________________________________________________________
| |
@@ -125,11 +125,11 @@
*/
static_vo Word16 add (Word16 var1, Word16 var2)
{
- Word16 var_out;
- Word32 L_sum;
- L_sum = (Word32) var1 + var2;
- var_out = saturate (L_sum);
- return (var_out);
+ Word16 var_out;
+ Word32 L_sum;
+ L_sum = (Word32) var1 + var2;
+ var_out = saturate (L_sum);
+ return (var_out);
}
/*___________________________________________________________________________
@@ -168,11 +168,11 @@
static_vo Word16 sub (Word16 var1, Word16 var2)
{
- Word16 var_out;
- Word32 L_diff;
- L_diff = (Word32) var1 - var2;
- var_out = saturate (L_diff);
- return (var_out);
+ Word16 var_out;
+ Word32 L_diff;
+ L_diff = (Word32) var1 - var2;
+ var_out = saturate (L_diff);
+ return (var_out);
}
/*___________________________________________________________________________
@@ -212,27 +212,27 @@
static_vo Word16 shl (Word16 var1, Word16 var2)
{
- Word16 var_out;
- Word32 result;
- if (var2 < 0)
- {
- if (var2 < -16)
- var2 = -16;
- var_out = var1 >> ((Word16)-var2);
- }
- else
- {
- result = (Word32) var1 *((Word32) 1 << var2);
- if ((var2 > 15 && var1 != 0) || (result != (Word32) ((Word16) result)))
- {
- var_out = (Word16)((var1 > 0) ? MAX_16 : MIN_16);
- }
- else
- {
- var_out = extract_l (result);
- }
- }
- return (var_out);
+ Word16 var_out;
+ Word32 result;
+ if (var2 < 0)
+ {
+ if (var2 < -16)
+ var2 = -16;
+ var_out = var1 >> ((Word16)-var2);
+ }
+ else
+ {
+ result = (Word32) var1 *((Word32) 1 << var2);
+ if ((var2 > 15 && var1 != 0) || (result != (Word32) ((Word16) result)))
+ {
+ var_out = (Word16)((var1 > 0) ? MAX_16 : MIN_16);
+ }
+ else
+ {
+ var_out = extract_l (result);
+ }
+ }
+ return (var_out);
}
/*___________________________________________________________________________
@@ -272,32 +272,32 @@
static_vo Word16 shr (Word16 var1, Word16 var2)
{
- Word16 var_out;
- if (var2 < 0)
- {
- if (var2 < -16)
- var2 = -16;
- var_out = shl(var1, (Word16)-var2);
- }
- else
- {
- if (var2 >= 15)
- {
- var_out = (Word16)((var1 < 0) ? -1 : 0);
- }
- else
- {
- if (var1 < 0)
- {
- var_out = (Word16)(~((~var1) >> var2));
- }
- else
- {
- var_out = (Word16)(var1 >> var2);
- }
- }
- }
- return (var_out);
+ Word16 var_out;
+ if (var2 < 0)
+ {
+ if (var2 < -16)
+ var2 = -16;
+ var_out = shl(var1, (Word16)-var2);
+ }
+ else
+ {
+ if (var2 >= 15)
+ {
+ var_out = (Word16)((var1 < 0) ? -1 : 0);
+ }
+ else
+ {
+ if (var1 < 0)
+ {
+ var_out = (Word16)(~((~var1) >> var2));
+ }
+ else
+ {
+ var_out = (Word16)(var1 >> var2);
+ }
+ }
+ }
+ return (var_out);
}
/*___________________________________________________________________________
@@ -337,14 +337,14 @@
static_vo Word16 mult (Word16 var1, Word16 var2)
{
- Word16 var_out;
- Word32 L_product;
- L_product = (Word32) var1 *(Word32) var2;
- L_product = (L_product & (Word32) 0xffff8000L) >> 15;
- if (L_product & (Word32) 0x00010000L)
- L_product = L_product | (Word32) 0xffff0000L;
- var_out = saturate (L_product);
- return (var_out);
+ Word16 var_out;
+ Word32 L_product;
+ L_product = (Word32) var1 *(Word32) var2;
+ L_product = (L_product & (Word32) 0xffff8000L) >> 15;
+ if (L_product & (Word32) 0x00010000L)
+ L_product = L_product | (Word32) 0xffff0000L;
+ var_out = saturate (L_product);
+ return (var_out);
}
/*___________________________________________________________________________
@@ -384,17 +384,17 @@
static_vo Word32 L_mult (Word16 var1, Word16 var2)
{
- Word32 L_var_out;
- L_var_out = (Word32) var1 *(Word32) var2;
- if (L_var_out != (Word32) 0x40000000L)
- {
- L_var_out *= 2;
- }
- else
- {
- L_var_out = MAX_32;
- }
- return (L_var_out);
+ Word32 L_var_out;
+ L_var_out = (Word32) var1 *(Word32) var2;
+ if (L_var_out != (Word32) 0x40000000L)
+ {
+ L_var_out *= 2;
+ }
+ else
+ {
+ L_var_out = MAX_32;
+ }
+ return (L_var_out);
}
/*___________________________________________________________________________
@@ -430,11 +430,11 @@
static_vo Word16 voround (Word32 L_var1)
{
- Word16 var_out;
- Word32 L_rounded;
- L_rounded = L_add (L_var1, (Word32) 0x00008000L);
- var_out = extract_h (L_rounded);
- return (var_out);
+ Word16 var_out;
+ Word32 L_rounded;
+ L_rounded = L_add (L_var1, (Word32) 0x00008000L);
+ var_out = extract_h (L_rounded);
+ return (var_out);
}
/*___________________________________________________________________________
@@ -476,11 +476,11 @@
static_vo Word32 L_mac (Word32 L_var3, Word16 var1, Word16 var2)
{
- Word32 L_var_out;
- Word32 L_product;
- L_product = ((var1 * var2) << 1);
- L_var_out = L_add (L_var3, L_product);
- return (L_var_out);
+ Word32 L_var_out;
+ Word32 L_product;
+ L_product = ((var1 * var2) << 1);
+ L_var_out = L_add (L_var3, L_product);
+ return (L_var_out);
}
/*___________________________________________________________________________
@@ -522,11 +522,11 @@
static_vo Word32 L_msu (Word32 L_var3, Word16 var1, Word16 var2)
{
- Word32 L_var_out;
- Word32 L_product;
- L_product = (var1 * var2)<<1;
- L_var_out = L_sub (L_var3, L_product);
- return (L_var_out);
+ Word32 L_var_out;
+ Word32 L_product;
+ L_product = (var1 * var2)<<1;
+ L_var_out = L_sub (L_var3, L_product);
+ return (L_var_out);
}
/*___________________________________________________________________________
@@ -564,16 +564,16 @@
__attribute__((no_sanitize("integer")))
static_vo Word32 L_add (Word32 L_var1, Word32 L_var2)
{
- Word32 L_var_out;
- L_var_out = L_var1 + L_var2;
- if (((L_var1 ^ L_var2) & MIN_32) == 0)
- {
- if ((L_var_out ^ L_var1) & MIN_32)
- {
- L_var_out = (L_var1 < 0) ? MIN_32 : MAX_32;
- }
- }
- return (L_var_out);
+ Word32 L_var_out;
+ L_var_out = L_var1 + L_var2;
+ if (((L_var1 ^ L_var2) & MIN_32) == 0)
+ {
+ if ((L_var_out ^ L_var1) & MIN_32)
+ {
+ L_var_out = (L_var1 < 0) ? MIN_32 : MAX_32;
+ }
+ }
+ return (L_var_out);
}
/*___________________________________________________________________________
@@ -608,18 +608,19 @@
|___________________________________________________________________________|
*/
+__attribute__((no_sanitize("integer")))
static_vo Word32 L_sub (Word32 L_var1, Word32 L_var2)
{
- Word32 L_var_out;
- L_var_out = L_var1 - L_var2;
- if (((L_var1 ^ L_var2) & MIN_32) != 0)
- {
- if ((L_var_out ^ L_var1) & MIN_32)
- {
- L_var_out = (L_var1 < 0L) ? MIN_32 : MAX_32;
- }
- }
- return (L_var_out);
+ Word32 L_var_out;
+ L_var_out = L_var1 - L_var2;
+ if (((L_var1 ^ L_var2) & MIN_32) != 0)
+ {
+ if ((L_var_out ^ L_var1) & MIN_32)
+ {
+ L_var_out = (L_var1 < 0L) ? MIN_32 : MAX_32;
+ }
+ }
+ return (L_var_out);
}
@@ -659,18 +660,18 @@
static_vo Word16 mult_r (Word16 var1, Word16 var2)
{
- Word16 var_out;
- Word32 L_product_arr;
- L_product_arr = (Word32) var1 *(Word32) var2; /* product */
- L_product_arr += (Word32) 0x00004000L; /* round */
- L_product_arr &= (Word32) 0xffff8000L;
- L_product_arr >>= 15; /* shift */
- if (L_product_arr & (Word32) 0x00010000L) /* sign extend when necessary */
- {
- L_product_arr |= (Word32) 0xffff0000L;
- }
- var_out = saturate (L_product_arr);
- return (var_out);
+ Word16 var_out;
+ Word32 L_product_arr;
+ L_product_arr = (Word32) var1 *(Word32) var2; /* product */
+ L_product_arr += (Word32) 0x00004000L; /* round */
+ L_product_arr &= (Word32) 0xffff8000L;
+ L_product_arr >>= 15; /* shift */
+ if (L_product_arr & (Word32) 0x00010000L) /* sign extend when necessary */
+ {
+ L_product_arr |= (Word32) 0xffff0000L;
+ }
+ var_out = saturate (L_product_arr);
+ return (var_out);
}
/*___________________________________________________________________________
@@ -709,61 +710,61 @@
static_vo Word32 L_shl (Word32 L_var1, Word16 var2)
{
- Word32 L_var_out = 0L;
- if (var2 <= 0)
- {
- if (var2 < -32)
- var2 = -32;
- L_var_out = (L_var1 >> (Word16)-var2);
- }
- else
- {
- for (; var2 > 0; var2--)
- {
- if (L_var1 > (Word32) 0X3fffffffL)
- {
- L_var_out = MAX_32;
- break;
- }
- else
- {
- if (L_var1 < (Word32) 0xc0000000L)
- {
- //Overflow = 1;
- L_var_out = MIN_32;
- break;
- }
- }
- L_var1 *= 2;
- L_var_out = L_var1;
- }
- }
- return (L_var_out);
+ Word32 L_var_out = 0L;
+ if (var2 <= 0)
+ {
+ if (var2 < -32)
+ var2 = -32;
+ L_var_out = (L_var1 >> (Word16)-var2);
+ }
+ else
+ {
+ for (; var2 > 0; var2--)
+ {
+ if (L_var1 > (Word32) 0X3fffffffL)
+ {
+ L_var_out = MAX_32;
+ break;
+ }
+ else
+ {
+ if (L_var1 < (Word32) 0xc0000000L)
+ {
+ //Overflow = 1;
+ L_var_out = MIN_32;
+ break;
+ }
+ }
+ L_var1 *= 2;
+ L_var_out = L_var1;
+ }
+ }
+ return (L_var_out);
}
static_vo Word32 L_shl2(Word32 L_var1, Word16 var2)
{
- Word32 L_var_out = 0L;
+ Word32 L_var_out = 0L;
- for (; var2 > 0; var2--)
- {
- if (L_var1 > (Word32) 0X3fffffffL)
- {
- L_var_out = MAX_32;
- break;
- }
- else
- {
- if (L_var1 < (Word32) 0xc0000000L)
- {
- L_var_out = MIN_32;
- break;
- }
- }
- L_var1 <<=1 ;
- L_var_out = L_var1;
- }
- return (L_var_out);
+ for (; var2 > 0; var2--)
+ {
+ if (L_var1 > (Word32) 0X3fffffffL)
+ {
+ L_var_out = MAX_32;
+ break;
+ }
+ else
+ {
+ if (L_var1 < (Word32) 0xc0000000L)
+ {
+ L_var_out = MIN_32;
+ break;
+ }
+ }
+ L_var1 <<=1 ;
+ L_var_out = L_var1;
+ }
+ return (L_var_out);
}
/*___________________________________________________________________________
@@ -802,32 +803,32 @@
static_vo Word32 L_shr (Word32 L_var1, Word16 var2)
{
- Word32 L_var_out;
- if (var2 < 0)
- {
- if (var2 < -32)
- var2 = -32;
- L_var_out = L_shl2(L_var1, (Word16)-var2);
- }
- else
- {
- if (var2 >= 31)
- {
- L_var_out = (L_var1 < 0L) ? -1 : 0;
- }
- else
- {
- if (L_var1 < 0)
- {
- L_var_out = ~((~L_var1) >> var2);
- }
- else
- {
- L_var_out = L_var1 >> var2;
- }
- }
- }
- return (L_var_out);
+ Word32 L_var_out;
+ if (var2 < 0)
+ {
+ if (var2 < -32)
+ var2 = -32;
+ L_var_out = L_shl2(L_var1, (Word16)-var2);
+ }
+ else
+ {
+ if (var2 >= 31)
+ {
+ L_var_out = (L_var1 < 0L) ? -1 : 0;
+ }
+ else
+ {
+ if (L_var1 < 0)
+ {
+ L_var_out = ~((~L_var1) >> var2);
+ }
+ else
+ {
+ L_var_out = L_var1 >> var2;
+ }
+ }
+ }
+ return (L_var_out);
}
/*___________________________________________________________________________
@@ -874,23 +875,23 @@
static_vo Word32 L_shr_r (Word32 L_var1, Word16 var2)
{
- Word32 L_var_out;
- if (var2 > 31)
- {
- L_var_out = 0;
- }
- else
- {
- L_var_out = L_shr (L_var1, var2);
- if (var2 > 0)
- {
- if ((L_var1 & ((Word32) 1 << (var2 - 1))) != 0)
- {
- L_var_out++;
- }
- }
- }
- return (L_var_out);
+ Word32 L_var_out;
+ if (var2 > 31)
+ {
+ L_var_out = 0;
+ }
+ else
+ {
+ L_var_out = L_shr (L_var1, var2);
+ if (var2 > 0)
+ {
+ if ((L_var1 & ((Word32) 1 << (var2 - 1))) != 0)
+ {
+ L_var_out++;
+ }
+ }
+ }
+ return (L_var_out);
}
/*___________________________________________________________________________
@@ -928,30 +929,30 @@
static_vo Word16 norm_s (Word16 var1)
{
- Word16 var_out = 0;
- if (var1 == 0)
- {
- var_out = 0;
- }
- else
- {
- if (var1 == -1)
- {
- var_out = 15;
- }
- else
- {
- if (var1 < 0)
- {
- var1 = (Word16)~var1;
- }
- for (var_out = 0; var1 < 0x4000; var_out++)
- {
- var1 <<= 1;
- }
- }
- }
- return (var_out);
+ Word16 var_out = 0;
+ if (var1 == 0)
+ {
+ var_out = 0;
+ }
+ else
+ {
+ if (var1 == -1)
+ {
+ var_out = 15;
+ }
+ else
+ {
+ if (var1 < 0)
+ {
+ var1 = (Word16)~var1;
+ }
+ for (var_out = 0; var1 < 0x4000; var_out++)
+ {
+ var1 <<= 1;
+ }
+ }
+ }
+ return (var_out);
}
/*___________________________________________________________________________
@@ -993,47 +994,47 @@
static_vo Word16 div_s (Word16 var1, Word16 var2)
{
- Word16 var_out = 0;
- Word16 iteration;
- Word32 L_num;
- Word32 L_denom;
- if ((var1 < 0) || (var2 < 0))
- {
- var_out = MAX_16;
- return var_out;
- }
- if (var2 == 0)
- {
- var_out = MAX_16;
- return var_out;
- }
- if (var1 == 0)
- {
- var_out = 0;
- }
- else
- {
- if (var1 == var2)
- {
- var_out = MAX_16;
- }
- else
- {
- L_num = L_deposit_l (var1);
- L_denom = L_deposit_l(var2);
- for (iteration = 0; iteration < 15; iteration++)
- {
- var_out <<= 1;
- L_num <<= 1;
- if (L_num >= L_denom)
- {
- L_num -= L_denom;
- var_out += 1;
- }
- }
- }
- }
- return (var_out);
+ Word16 var_out = 0;
+ Word16 iteration;
+ Word32 L_num;
+ Word32 L_denom;
+ if ((var1 < 0) || (var2 < 0))
+ {
+ var_out = MAX_16;
+ return var_out;
+ }
+ if (var2 == 0)
+ {
+ var_out = MAX_16;
+ return var_out;
+ }
+ if (var1 == 0)
+ {
+ var_out = 0;
+ }
+ else
+ {
+ if (var1 == var2)
+ {
+ var_out = MAX_16;
+ }
+ else
+ {
+ L_num = L_deposit_l (var1);
+ L_denom = L_deposit_l(var2);
+ for (iteration = 0; iteration < 15; iteration++)
+ {
+ var_out <<= 1;
+ L_num <<= 1;
+ if (L_num >= L_denom)
+ {
+ L_num -= L_denom;
+ var_out += 1;
+ }
+ }
+ }
+ }
+ return (var_out);
}
/*___________________________________________________________________________
@@ -1071,20 +1072,20 @@
static_vo Word16 norm_l (Word32 L_var1)
{
- Word16 var_out = 0;
- if (L_var1 != 0)
- {
- var_out = 31;
- if (L_var1 != (Word32) 0xffffffffL)
- {
- L_var1 ^= (L_var1 >>31);
- for (var_out = 0; L_var1 < (Word32) 0x40000000L; var_out++)
- {
- L_var1 <<= 1;
- }
- }
- }
- return (var_out);
+ Word16 var_out = 0;
+ if (L_var1 != 0)
+ {
+ var_out = 31;
+ if (L_var1 != (Word32) 0xffffffffL)
+ {
+ L_var1 ^= (L_var1 >>31);
+ for (var_out = 0; L_var1 < (Word32) 0x40000000L; var_out++)
+ {
+ L_var1 <<= 1;
+ }
+ }
+ }
+ return (var_out);
}
#endif //__BASIC_OP_H__
diff --git a/media/libstagefright/codecs/amrwbenc/inc/bits.h b/media/libstagefright/codecs/amrwbenc/inc/bits.h
index e880684..ff9c0c1 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/bits.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/bits.h
@@ -18,7 +18,7 @@
/*--------------------------------------------------------------------------*
* BITS.H *
*--------------------------------------------------------------------------*
-* Number of bits for different modes *
+* Number of bits for different modes *
*--------------------------------------------------------------------------*/
#ifndef __BITS_H__
@@ -52,16 +52,16 @@
#define RX_FRAME_TYPE (Word16)0x6b20
static const Word16 nb_of_bits[NUM_OF_MODES] = {
- NBBITS_7k,
- NBBITS_9k,
- NBBITS_12k,
- NBBITS_14k,
- NBBITS_16k,
- NBBITS_18k,
- NBBITS_20k,
- NBBITS_23k,
- NBBITS_24k,
- NBBITS_SID
+ NBBITS_7k,
+ NBBITS_9k,
+ NBBITS_12k,
+ NBBITS_14k,
+ NBBITS_16k,
+ NBBITS_18k,
+ NBBITS_20k,
+ NBBITS_23k,
+ NBBITS_24k,
+ NBBITS_SID
};
/*typedef struct
@@ -74,18 +74,18 @@
//typedef struct
//{
-// Word16 prev_ft;
-// Word16 prev_mode;
+// Word16 prev_ft;
+// Word16 prev_mode;
//} RX_State;
int PackBits(Word16 prms[], Word16 coding_mode, Word16 mode, Coder_State *st);
void Parm_serial(
- Word16 value, /* input : parameter value */
- Word16 no_of_bits, /* input : number of bits */
- Word16 ** prms
- );
+ Word16 value, /* input : parameter value */
+ Word16 no_of_bits, /* input : number of bits */
+ Word16 ** prms
+ );
#endif //__BITS_H__
diff --git a/media/libstagefright/codecs/amrwbenc/inc/cod_main.h b/media/libstagefright/codecs/amrwbenc/inc/cod_main.h
index 53ca55e..170981e 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/cod_main.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/cod_main.h
@@ -18,7 +18,7 @@
/*--------------------------------------------------------------------------*
* COD_MAIN.H *
*--------------------------------------------------------------------------*
- * Static memory in the encoder *
+ * Static memory in the encoder *
*--------------------------------------------------------------------------*/
#ifndef __COD_MAIN_H__
#define __COD_MAIN_H__
@@ -79,21 +79,21 @@
Word16 vad_hist;
Word16 gain_alpha;
/* TX_State structure */
- Word16 sid_update_counter;
+ Word16 sid_update_counter;
Word16 sid_handover_debt;
Word16 prev_ft;
- Word16 allow_dtx;
- /*some input/output buffer parameters */
- unsigned char *inputStream;
- int inputSize;
- VOAMRWBMODE mode;
- VOAMRWBFRAMETYPE frameType;
- unsigned short *outputStream;
- int outputSize;
- FrameStream *stream;
- VO_MEM_OPERATOR *pvoMemop;
- VO_MEM_OPERATOR voMemoprator;
- VO_PTR hCheck;
+ Word16 allow_dtx;
+ /*some input/output buffer parameters */
+ unsigned char *inputStream;
+ int inputSize;
+ VOAMRWBMODE mode;
+ VOAMRWBFRAMETYPE frameType;
+ unsigned short *outputStream;
+ int outputSize;
+ FrameStream *stream;
+ VO_MEM_OPERATOR *pvoMemop;
+ VO_MEM_OPERATOR voMemoprator;
+ VO_PTR hCheck;
} Coder_State;
typedef void* HAMRENC;
diff --git a/media/libstagefright/codecs/amrwbenc/inc/dtx.h b/media/libstagefright/codecs/amrwbenc/inc/dtx.h
index 0bdda67..82a9bf4 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/dtx.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/dtx.h
@@ -16,9 +16,9 @@
/*--------------------------------------------------------------------------*
- * DTX.H *
+ * DTX.H *
*--------------------------------------------------------------------------*
- * Static memory, constants and frametypes for the DTX *
+ * Static memory, constants and frametypes for the DTX *
*--------------------------------------------------------------------------*/
#ifndef __DTX_H__
diff --git a/media/libstagefright/codecs/amrwbenc/inc/log2.h b/media/libstagefright/codecs/amrwbenc/inc/log2.h
index b065eb4..3d9a6c4 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/log2.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/log2.h
@@ -45,17 +45,17 @@
********************************************************************************
*/
void Log2 (
- Word32 L_x, /* (i) : input value */
- Word16 *exponent, /* (o) : Integer part of Log2. (range: 0<=val<=30) */
- Word16 *fraction /* (o) : Fractional part of Log2. (range: 0<=val<1)*/
- );
+ Word32 L_x, /* (i) : input value */
+ Word16 *exponent, /* (o) : Integer part of Log2. (range: 0<=val<=30) */
+ Word16 *fraction /* (o) : Fractional part of Log2. (range: 0<=val<1)*/
+ );
void Log2_norm (
- Word32 L_x, /* (i) : input value (normalized) */
- Word16 exp, /* (i) : norm_l (L_x) */
- Word16 *exponent, /* (o) : Integer part of Log2. (range: 0<=val<=30) */
- Word16 *fraction /* (o) : Fractional part of Log2. (range: 0<=val<1) */
- );
+ Word32 L_x, /* (i) : input value (normalized) */
+ Word16 exp, /* (i) : norm_l (L_x) */
+ Word16 *exponent, /* (o) : Integer part of Log2. (range: 0<=val<=30) */
+ Word16 *fraction /* (o) : Fractional part of Log2. (range: 0<=val<1) */
+ );
#endif //__LOG2_H__
diff --git a/media/libstagefright/codecs/amrwbenc/inc/main.h b/media/libstagefright/codecs/amrwbenc/inc/main.h
index 3a6f963..adef2df 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/main.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/main.h
@@ -17,9 +17,9 @@
/*--------------------------------------------------------------------------*
- * MAIN.H *
+ * MAIN.H *
*--------------------------------------------------------------------------*
- * Main functions *
+ * Main functions *
*--------------------------------------------------------------------------*/
#ifndef __MAIN_H__
diff --git a/media/libstagefright/codecs/amrwbenc/inc/math_op.h b/media/libstagefright/codecs/amrwbenc/inc/math_op.h
index 7b6196b..c3c00bc 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/math_op.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/math_op.h
@@ -16,40 +16,40 @@
/*--------------------------------------------------------------------------*
- * MATH_OP.H *
+ * MATH_OP.H *
*--------------------------------------------------------------------------*
- * Mathematical operations *
+ * Mathematical operations *
*--------------------------------------------------------------------------*/
#ifndef __MATH_OP_H__
#define __MATH_OP_H__
Word32 Isqrt( /* (o) Q31 : output value (range: 0<=val<1) */
- Word32 L_x /* (i) Q0 : input value (range: 0<=val<=7fffffff) */
- );
+ Word32 L_x /* (i) Q0 : input value (range: 0<=val<=7fffffff) */
+ );
void Isqrt_n(
- Word32 * frac, /* (i/o) Q31: normalized value (1.0 < frac <= 0.5) */
- Word16 * exp /* (i/o) : exponent (value = frac x 2^exponent) */
- );
+ Word32 * frac, /* (i/o) Q31: normalized value (1.0 < frac <= 0.5) */
+ Word16 * exp /* (i/o) : exponent (value = frac x 2^exponent) */
+ );
Word32 Pow2( /* (o) Q0 : result (range: 0<=val<=0x7fffffff) */
- Word16 exponant, /* (i) Q0 : Integer part. (range: 0<=val<=30) */
- Word16 fraction /* (i) Q15 : Fractionnal part. (range: 0.0<=val<1.0) */
- );
+ Word16 exponant, /* (i) Q0 : Integer part. (range: 0<=val<=30) */
+ Word16 fraction /* (i) Q15 : Fractionnal part. (range: 0.0<=val<1.0) */
+ );
Word32 Dot_product12( /* (o) Q31: normalized result (1 < val <= -1) */
- Word16 x[], /* (i) 12bits: x vector */
- Word16 y[], /* (i) 12bits: y vector */
- Word16 lg, /* (i) : vector length */
- Word16 * exp /* (o) : exponent of result (0..+30) */
- );
+ Word16 x[], /* (i) 12bits: x vector */
+ Word16 y[], /* (i) 12bits: y vector */
+ Word16 lg, /* (i) : vector length */
+ Word16 * exp /* (o) : exponent of result (0..+30) */
+ );
Word32 Dot_product12_asm( /* (o) Q31: normalized result (1 < val <= -1) */
- Word16 x[], /* (i) 12bits: x vector */
- Word16 y[], /* (i) 12bits: y vector */
- Word16 lg, /* (i) : vector length */
- Word16 * exp /* (o) : exponent of result (0..+30) */
- );
+ Word16 x[], /* (i) 12bits: x vector */
+ Word16 y[], /* (i) 12bits: y vector */
+ Word16 lg, /* (i) : vector length */
+ Word16 * exp /* (o) : exponent of result (0..+30) */
+ );
#endif //__MATH_OP_H__
diff --git a/media/libstagefright/codecs/amrwbenc/inc/mem_align.h b/media/libstagefright/codecs/amrwbenc/inc/mem_align.h
index 442786a..2ae5a6c 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/mem_align.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/mem_align.h
@@ -14,9 +14,9 @@
** limitations under the License.
*/
/*******************************************************************************
- File: mem_align.h
+ File: mem_align.h
- Content: Memory alloc alignments functions
+ Content: Memory alloc alignments functions
*******************************************************************************/
@@ -29,7 +29,7 @@
extern void *mem_malloc(VO_MEM_OPERATOR *pMemop, unsigned int size, unsigned char alignment, unsigned int CodecID);
extern void mem_free(VO_MEM_OPERATOR *pMemop, void *mem_ptr, unsigned int CodecID);
-#endif /* __VO_MEM_ALIGN_H__ */
+#endif /* __VO_MEM_ALIGN_H__ */
diff --git a/media/libstagefright/codecs/amrwbenc/inc/p_med_o.h b/media/libstagefright/codecs/amrwbenc/inc/p_med_o.h
index 4a13f16..77487ed 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/p_med_o.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/p_med_o.h
@@ -17,36 +17,36 @@
/*--------------------------------------------------------------------------*
* P_MED_O.H *
*--------------------------------------------------------------------------*
- * Median open-loop lag search *
+ * Median open-loop lag search *
*--------------------------------------------------------------------------*/
#ifndef __P_MED_O_H__
#define __P_MED_O_H__
Word16 Pitch_med_ol( /* output: open loop pitch lag */
- Word16 wsp[], /* input : signal used to compute the open loop pitch */
- /* wsp[-pit_max] to wsp[-1] should be known */
- Word16 L_min, /* input : minimum pitch lag */
- Word16 L_max, /* input : maximum pitch lag */
- Word16 L_frame, /* input : length of frame to compute pitch */
- Word16 L_0, /* input : old_ open-loop pitch */
- Word16 * gain, /* output: normalize correlation of hp_wsp for the Lag */
- Word16 * hp_wsp_mem, /* i:o : memory of the hypass filter for hp_wsp[] (lg=9) */
- Word16 * old_hp_wsp, /* i:o : hypass wsp[] */
- Word16 wght_flg /* input : is weighting function used */
- );
+ Word16 wsp[], /* input : signal used to compute the open loop pitch */
+ /* wsp[-pit_max] to wsp[-1] should be known */
+ Word16 L_min, /* input : minimum pitch lag */
+ Word16 L_max, /* input : maximum pitch lag */
+ Word16 L_frame, /* input : length of frame to compute pitch */
+ Word16 L_0, /* input : old_ open-loop pitch */
+ Word16 * gain, /* output: normalize correlation of hp_wsp for the Lag */
+ Word16 * hp_wsp_mem, /* i:o : memory of the hypass filter for hp_wsp[] (lg=9) */
+ Word16 * old_hp_wsp, /* i:o : hypass wsp[] */
+ Word16 wght_flg /* input : is weighting function used */
+ );
Word16 Med_olag( /* output : median of 5 previous open-loop lags */
- Word16 prev_ol_lag, /* input : previous open-loop lag */
- Word16 old_ol_lag[5]
- );
+ Word16 prev_ol_lag, /* input : previous open-loop lag */
+ Word16 old_ol_lag[5]
+ );
void Hp_wsp(
- Word16 wsp[], /* i : wsp[] signal */
- Word16 hp_wsp[], /* o : hypass wsp[] */
- Word16 lg, /* i : lenght of signal */
- Word16 mem[] /* i/o : filter memory [9] */
- );
+ Word16 wsp[], /* i : wsp[] signal */
+ Word16 hp_wsp[], /* o : hypass wsp[] */
+ Word16 lg, /* i : lenght of signal */
+ Word16 mem[] /* i/o : filter memory [9] */
+ );
#endif //__P_MED_O_H__
diff --git a/media/libstagefright/codecs/amrwbenc/inc/q_pulse.h b/media/libstagefright/codecs/amrwbenc/inc/q_pulse.h
index b5d5280..67140fc 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/q_pulse.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/q_pulse.h
@@ -19,7 +19,7 @@
/*--------------------------------------------------------------------------*
* Q_PULSE.H *
*--------------------------------------------------------------------------*
- * Coding and decoding of algebraic codebook *
+ * Coding and decoding of algebraic codebook *
*--------------------------------------------------------------------------*/
#ifndef __Q_PULSE_H__
@@ -28,38 +28,38 @@
#include "typedef.h"
Word32 quant_1p_N1( /* (o) return (N+1) bits */
- Word16 pos, /* (i) position of the pulse */
- Word16 N); /* (i) number of bits for position */
+ Word16 pos, /* (i) position of the pulse */
+ Word16 N); /* (i) number of bits for position */
Word32 quant_2p_2N1( /* (o) return (2*N)+1 bits */
- Word16 pos1, /* (i) position of the pulse 1 */
- Word16 pos2, /* (i) position of the pulse 2 */
- Word16 N); /* (i) number of bits for position */
+ Word16 pos1, /* (i) position of the pulse 1 */
+ Word16 pos2, /* (i) position of the pulse 2 */
+ Word16 N); /* (i) number of bits for position */
Word32 quant_3p_3N1( /* (o) return (3*N)+1 bits */
- Word16 pos1, /* (i) position of the pulse 1 */
- Word16 pos2, /* (i) position of the pulse 2 */
- Word16 pos3, /* (i) position of the pulse 3 */
- Word16 N); /* (i) number of bits for position */
+ Word16 pos1, /* (i) position of the pulse 1 */
+ Word16 pos2, /* (i) position of the pulse 2 */
+ Word16 pos3, /* (i) position of the pulse 3 */
+ Word16 N); /* (i) number of bits for position */
Word32 quant_4p_4N1( /* (o) return (4*N)+1 bits */
- Word16 pos1, /* (i) position of the pulse 1 */
- Word16 pos2, /* (i) position of the pulse 2 */
- Word16 pos3, /* (i) position of the pulse 3 */
- Word16 pos4, /* (i) position of the pulse 4 */
- Word16 N); /* (i) number of bits for position */
+ Word16 pos1, /* (i) position of the pulse 1 */
+ Word16 pos2, /* (i) position of the pulse 2 */
+ Word16 pos3, /* (i) position of the pulse 3 */
+ Word16 pos4, /* (i) position of the pulse 4 */
+ Word16 N); /* (i) number of bits for position */
Word32 quant_4p_4N( /* (o) return 4*N bits */
- Word16 pos[], /* (i) position of the pulse 1..4 */
- Word16 N); /* (i) number of bits for position */
+ Word16 pos[], /* (i) position of the pulse 1..4 */
+ Word16 N); /* (i) number of bits for position */
Word32 quant_5p_5N( /* (o) return 5*N bits */
- Word16 pos[], /* (i) position of the pulse 1..5 */
- Word16 N); /* (i) number of bits for position */
+ Word16 pos[], /* (i) position of the pulse 1..5 */
+ Word16 N); /* (i) number of bits for position */
Word32 quant_6p_6N_2( /* (o) return (6*N)-2 bits */
- Word16 pos[], /* (i) position of the pulse 1..6 */
- Word16 N); /* (i) number of bits for position */
+ Word16 pos[], /* (i) position of the pulse 1..6 */
+ Word16 N); /* (i) number of bits for position */
#endif //__Q_PULSE_H__
diff --git a/media/libstagefright/codecs/amrwbenc/inc/stream.h b/media/libstagefright/codecs/amrwbenc/inc/stream.h
index 4c1d0f0..ec1a700 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/stream.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/stream.h
@@ -17,7 +17,7 @@
/***********************************************************************
-File: stream.h
+File: stream.h
Contains: VOME API Buffer Operator Implement Header
@@ -28,16 +28,16 @@
#include "voMem.h"
#define Frame_Maxsize 1024 * 2 //Work Buffer 10K
#define Frame_MaxByte 640 //AMR_WB Encoder one frame 320 samples = 640 Bytes
-#define MIN(a,b) ((a) < (b)? (a) : (b))
+#define MIN(a,b) ((a) < (b)? (a) : (b))
typedef struct{
- unsigned char *set_ptr;
- unsigned char *frame_ptr;
- unsigned char *frame_ptr_bk;
- int set_len;
- int framebuffer_len;
- int frame_storelen;
- int used_len;
+ unsigned char *set_ptr;
+ unsigned char *frame_ptr;
+ unsigned char *frame_ptr_bk;
+ int set_len;
+ int framebuffer_len;
+ int frame_storelen;
+ int used_len;
}FrameStream;
void voAWB_UpdateFrameBuffer(FrameStream *stream, VO_MEM_OPERATOR *pMemOP);
diff --git a/media/libstagefright/codecs/amrwbenc/inc/wb_vad.h b/media/libstagefright/codecs/amrwbenc/inc/wb_vad.h
index 6822f48..9a9af4f 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/wb_vad.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/wb_vad.h
@@ -37,28 +37,28 @@
typedef struct
{
- Word16 bckr_est[COMPLEN]; /* background noise estimate */
- Word16 ave_level[COMPLEN]; /* averaged input components for stationary */
- /* estimation */
- Word16 old_level[COMPLEN]; /* input levels of the previous frame */
- Word16 sub_level[COMPLEN]; /* input levels calculated at the end of a frame (lookahead) */
- Word16 a_data5[F_5TH_CNT][2]; /* memory for the filter bank */
- Word16 a_data3[F_3TH_CNT]; /* memory for the filter bank */
+ Word16 bckr_est[COMPLEN]; /* background noise estimate */
+ Word16 ave_level[COMPLEN]; /* averaged input components for stationary */
+ /* estimation */
+ Word16 old_level[COMPLEN]; /* input levels of the previous frame */
+ Word16 sub_level[COMPLEN]; /* input levels calculated at the end of a frame (lookahead) */
+ Word16 a_data5[F_5TH_CNT][2]; /* memory for the filter bank */
+ Word16 a_data3[F_3TH_CNT]; /* memory for the filter bank */
- Word16 burst_count; /* counts length of a speech burst */
- Word16 hang_count; /* hangover counter */
- Word16 stat_count; /* stationary counter */
+ Word16 burst_count; /* counts length of a speech burst */
+ Word16 hang_count; /* hangover counter */
+ Word16 stat_count; /* stationary counter */
- /* Note that each of the following two variables holds 15 flags. Each flag reserves 1 bit of the
- * variable. The newest flag is in the bit 15 (assuming that LSB is bit 1 and MSB is bit 16). */
- Word16 vadreg; /* flags for intermediate VAD decisions */
- Word16 tone_flag; /* tone detection flags */
+ /* Note that each of the following two variables holds 15 flags. Each flag reserves 1 bit of the
+ * variable. The newest flag is in the bit 15 (assuming that LSB is bit 1 and MSB is bit 16). */
+ Word16 vadreg; /* flags for intermediate VAD decisions */
+ Word16 tone_flag; /* tone detection flags */
- Word16 sp_est_cnt; /* counter for speech level estimation */
- Word16 sp_max; /* maximum level */
- Word16 sp_max_cnt; /* counts frames that contains speech */
- Word16 speech_level; /* estimated speech level */
- Word32 prev_pow_sum; /* power of previous frame */
+ Word16 sp_est_cnt; /* counter for speech level estimation */
+ Word16 sp_max; /* maximum level */
+ Word16 sp_max_cnt; /* counts frames that contains speech */
+ Word16 speech_level; /* estimated speech level */
+ Word32 prev_pow_sum; /* power of previous frame */
} VadVars;
diff --git a/media/libstagefright/codecs/amrwbenc/inc/wb_vad_c.h b/media/libstagefright/codecs/amrwbenc/inc/wb_vad_c.h
index 04fd318..00b1779 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/wb_vad_c.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/wb_vad_c.h
@@ -16,9 +16,9 @@
/*-------------------------------------------------------------------*
- * WB_VAD_C.H *
+ * WB_VAD_C.H *
*-------------------------------------------------------------------*
- * Constants for Voice Activity Detection. *
+ * Constants for Voice Activity Detection. *
*-------------------------------------------------------------------*/
#ifndef __WB_VAD_C_H__
diff --git a/media/libstagefright/codecs/amrwbenc/src/autocorr.c b/media/libstagefright/codecs/amrwbenc/src/autocorr.c
index 0b2ea89..3ea53f7 100644
--- a/media/libstagefright/codecs/amrwbenc/src/autocorr.c
+++ b/media/libstagefright/codecs/amrwbenc/src/autocorr.c
@@ -31,100 +31,100 @@
#define UNUSED(x) (void)(x)
void Autocorr(
- Word16 x[], /* (i) : Input signal */
- Word16 m, /* (i) : LPC order */
- Word16 r_h[], /* (o) Q15: Autocorrelations (msb) */
- Word16 r_l[] /* (o) : Autocorrelations (lsb) */
- )
+ Word16 x[], /* (i) : Input signal */
+ Word16 m, /* (i) : LPC order */
+ Word16 r_h[], /* (o) Q15: Autocorrelations (msb) */
+ Word16 r_l[] /* (o) : Autocorrelations (lsb) */
+ )
{
- Word32 i, norm, shift;
- Word16 y[L_WINDOW];
- Word32 L_sum, L_sum1, L_tmp, F_LEN;
- Word16 *p1,*p2,*p3;
- const Word16 *p4;
+ Word32 i, norm, shift;
+ Word16 y[L_WINDOW];
+ Word32 L_sum, L_sum1, L_tmp, F_LEN;
+ Word16 *p1,*p2,*p3;
+ const Word16 *p4;
UNUSED(m);
- /* Windowing of signal */
- p1 = x;
- p4 = vo_window;
- p3 = y;
+ /* Windowing of signal */
+ p1 = x;
+ p4 = vo_window;
+ p3 = y;
- for (i = 0; i < L_WINDOW; i+=4)
- {
- *p3++ = vo_mult_r((*p1++), (*p4++));
- *p3++ = vo_mult_r((*p1++), (*p4++));
- *p3++ = vo_mult_r((*p1++), (*p4++));
- *p3++ = vo_mult_r((*p1++), (*p4++));
- }
+ for (i = 0; i < L_WINDOW; i+=4)
+ {
+ *p3++ = vo_mult_r((*p1++), (*p4++));
+ *p3++ = vo_mult_r((*p1++), (*p4++));
+ *p3++ = vo_mult_r((*p1++), (*p4++));
+ *p3++ = vo_mult_r((*p1++), (*p4++));
+ }
- /* calculate energy of signal */
- L_sum = vo_L_deposit_h(16); /* sqrt(256), avoid overflow after rounding */
- for (i = 0; i < L_WINDOW; i++)
- {
- L_tmp = vo_L_mult(y[i], y[i]);
- L_tmp = (L_tmp >> 8);
- L_sum += L_tmp;
- }
+ /* calculate energy of signal */
+ L_sum = vo_L_deposit_h(16); /* sqrt(256), avoid overflow after rounding */
+ for (i = 0; i < L_WINDOW; i++)
+ {
+ L_tmp = vo_L_mult(y[i], y[i]);
+ L_tmp = (L_tmp >> 8);
+ L_sum += L_tmp;
+ }
- /* scale signal to avoid overflow in autocorrelation */
- norm = norm_l(L_sum);
- shift = 4 - (norm >> 1);
- if(shift > 0)
- {
- p1 = y;
- for (i = 0; i < L_WINDOW; i+=4)
- {
- *p1 = vo_shr_r(*p1, shift);
- p1++;
- *p1 = vo_shr_r(*p1, shift);
- p1++;
- *p1 = vo_shr_r(*p1, shift);
- p1++;
- *p1 = vo_shr_r(*p1, shift);
- p1++;
- }
- }
+ /* scale signal to avoid overflow in autocorrelation */
+ norm = norm_l(L_sum);
+ shift = 4 - (norm >> 1);
+ if(shift > 0)
+ {
+ p1 = y;
+ for (i = 0; i < L_WINDOW; i+=4)
+ {
+ *p1 = vo_shr_r(*p1, shift);
+ p1++;
+ *p1 = vo_shr_r(*p1, shift);
+ p1++;
+ *p1 = vo_shr_r(*p1, shift);
+ p1++;
+ *p1 = vo_shr_r(*p1, shift);
+ p1++;
+ }
+ }
- /* Compute and normalize r[0] */
- L_sum = 1;
- for (i = 0; i < L_WINDOW; i+=4)
- {
- L_sum += vo_L_mult(y[i], y[i]);
- L_sum += vo_L_mult(y[i+1], y[i+1]);
- L_sum += vo_L_mult(y[i+2], y[i+2]);
- L_sum += vo_L_mult(y[i+3], y[i+3]);
- }
+ /* Compute and normalize r[0] */
+ L_sum = 1;
+ for (i = 0; i < L_WINDOW; i+=4)
+ {
+ L_sum += vo_L_mult(y[i], y[i]);
+ L_sum += vo_L_mult(y[i+1], y[i+1]);
+ L_sum += vo_L_mult(y[i+2], y[i+2]);
+ L_sum += vo_L_mult(y[i+3], y[i+3]);
+ }
- norm = norm_l(L_sum);
- L_sum = (L_sum << norm);
+ norm = norm_l(L_sum);
+ L_sum = (L_sum << norm);
- r_h[0] = L_sum >> 16;
- r_l[0] = (L_sum & 0xffff)>>1;
+ r_h[0] = L_sum >> 16;
+ r_l[0] = (L_sum & 0xffff)>>1;
- /* Compute r[1] to r[m] */
- for (i = 1; i <= 8; i++)
- {
- L_sum1 = 0;
- L_sum = 0;
- F_LEN = (Word32)(L_WINDOW - 2*i);
- p1 = y;
- p2 = y + (2*i)-1;
- do{
- L_sum1 += *p1 * *p2++;
- L_sum += *p1++ * *p2;
- }while(--F_LEN!=0);
+ /* Compute r[1] to r[m] */
+ for (i = 1; i <= 8; i++)
+ {
+ L_sum1 = 0;
+ L_sum = 0;
+ F_LEN = (Word32)(L_WINDOW - 2*i);
+ p1 = y;
+ p2 = y + (2*i)-1;
+ do{
+ L_sum1 += *p1 * *p2++;
+ L_sum += *p1++ * *p2;
+ }while(--F_LEN!=0);
- L_sum1 += *p1 * *p2++;
+ L_sum1 += *p1 * *p2++;
- L_sum1 = L_sum1<<norm;
- L_sum = L_sum<<norm;
+ L_sum1 = L_sum1<<norm;
+ L_sum = L_sum<<norm;
- r_h[(2*i)-1] = L_sum1 >> 15;
- r_l[(2*i)-1] = L_sum1 & 0x00007fff;
- r_h[(2*i)] = L_sum >> 15;
- r_l[(2*i)] = L_sum & 0x00007fff;
- }
- return;
+ r_h[(2*i)-1] = L_sum1 >> 15;
+ r_l[(2*i)-1] = L_sum1 & 0x00007fff;
+ r_h[(2*i)] = L_sum >> 15;
+ r_l[(2*i)] = L_sum & 0x00007fff;
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/az_isp.c b/media/libstagefright/codecs/amrwbenc/src/az_isp.c
index 43db27a..d7074f0 100644
--- a/media/libstagefright/codecs/amrwbenc/src/az_isp.c
+++ b/media/libstagefright/codecs/amrwbenc/src/az_isp.c
@@ -58,138 +58,138 @@
static __inline Word16 Chebps2(Word16 x, Word16 f[], Word32 n);
void Az_isp(
- Word16 a[], /* (i) Q12 : predictor coefficients */
- Word16 isp[], /* (o) Q15 : Immittance spectral pairs */
- Word16 old_isp[] /* (i) : old isp[] (in case not found M roots) */
- )
+ Word16 a[], /* (i) Q12 : predictor coefficients */
+ Word16 isp[], /* (o) Q15 : Immittance spectral pairs */
+ Word16 old_isp[] /* (i) : old isp[] (in case not found M roots) */
+ )
{
- Word32 i, j, nf, ip, order;
- Word16 xlow, ylow, xhigh, yhigh, xmid, ymid, xint;
- Word16 x, y, sign, exp;
- Word16 *coef;
- Word16 f1[NC + 1], f2[NC];
- Word32 t0;
- /*-------------------------------------------------------------*
- * find the sum and diff polynomials F1(z) and F2(z) *
- * F1(z) = [A(z) + z^M A(z^-1)] *
- * F2(z) = [A(z) - z^M A(z^-1)]/(1-z^-2) *
- * *
- * for (i=0; i<NC; i++) *
- * { *
- * f1[i] = a[i] + a[M-i]; *
- * f2[i] = a[i] - a[M-i]; *
- * } *
- * f1[NC] = 2.0*a[NC]; *
- * *
- * for (i=2; i<NC; i++) Divide by (1-z^-2) *
- * f2[i] += f2[i-2]; *
- *-------------------------------------------------------------*/
- for (i = 0; i < NC; i++)
- {
- t0 = a[i] << 15;
- f1[i] = vo_round(t0 + (a[M - i] << 15)); /* =(a[i]+a[M-i])/2 */
- f2[i] = vo_round(t0 - (a[M - i] << 15)); /* =(a[i]-a[M-i])/2 */
- }
- f1[NC] = a[NC];
- for (i = 2; i < NC; i++) /* Divide by (1-z^-2) */
- f2[i] = add1(f2[i], f2[i - 2]);
+ Word32 i, j, nf, ip, order;
+ Word16 xlow, ylow, xhigh, yhigh, xmid, ymid, xint;
+ Word16 x, y, sign, exp;
+ Word16 *coef;
+ Word16 f1[NC + 1], f2[NC];
+ Word32 t0;
+ /*-------------------------------------------------------------*
+ * find the sum and diff polynomials F1(z) and F2(z) *
+ * F1(z) = [A(z) + z^M A(z^-1)] *
+ * F2(z) = [A(z) - z^M A(z^-1)]/(1-z^-2) *
+ * *
+ * for (i=0; i<NC; i++) *
+ * { *
+ * f1[i] = a[i] + a[M-i]; *
+ * f2[i] = a[i] - a[M-i]; *
+ * } *
+ * f1[NC] = 2.0*a[NC]; *
+ * *
+ * for (i=2; i<NC; i++) Divide by (1-z^-2) *
+ * f2[i] += f2[i-2]; *
+ *-------------------------------------------------------------*/
+ for (i = 0; i < NC; i++)
+ {
+ t0 = a[i] << 15;
+ f1[i] = vo_round(t0 + (a[M - i] << 15)); /* =(a[i]+a[M-i])/2 */
+ f2[i] = vo_round(t0 - (a[M - i] << 15)); /* =(a[i]-a[M-i])/2 */
+ }
+ f1[NC] = a[NC];
+ for (i = 2; i < NC; i++) /* Divide by (1-z^-2) */
+ f2[i] = add1(f2[i], f2[i - 2]);
- /*---------------------------------------------------------------------*
- * Find the ISPs (roots of F1(z) and F2(z) ) using the *
- * Chebyshev polynomial evaluation. *
- * The roots of F1(z) and F2(z) are alternatively searched. *
- * We start by finding the first root of F1(z) then we switch *
- * to F2(z) then back to F1(z) and so on until all roots are found. *
- * *
- * - Evaluate Chebyshev pol. at grid points and check for sign change.*
- * - If sign change track the root by subdividing the interval *
- * 2 times and ckecking sign change. *
- *---------------------------------------------------------------------*/
- nf = 0; /* number of found frequencies */
- ip = 0; /* indicator for f1 or f2 */
- coef = f1;
- order = NC;
- xlow = vogrid[0];
- ylow = Chebps2(xlow, coef, order);
- j = 0;
- while ((nf < M - 1) && (j < GRID_POINTS))
- {
- j ++;
- xhigh = xlow;
- yhigh = ylow;
- xlow = vogrid[j];
- ylow = Chebps2(xlow, coef, order);
- if ((ylow * yhigh) <= (Word32) 0)
- {
- /* divide 2 times the interval */
- for (i = 0; i < 2; i++)
- {
- xmid = (xlow >> 1) + (xhigh >> 1); /* xmid = (xlow + xhigh)/2 */
- ymid = Chebps2(xmid, coef, order);
- if ((ylow * ymid) <= (Word32) 0)
- {
- yhigh = ymid;
- xhigh = xmid;
- } else
- {
- ylow = ymid;
- xlow = xmid;
- }
- }
- /*-------------------------------------------------------------*
- * Linear interpolation *
- * xint = xlow - ylow*(xhigh-xlow)/(yhigh-ylow); *
- *-------------------------------------------------------------*/
- x = xhigh - xlow;
- y = yhigh - ylow;
- if (y == 0)
- {
- xint = xlow;
- } else
- {
- sign = y;
- y = abs_s(y);
- exp = norm_s(y);
- y = y << exp;
- y = div_s((Word16) 16383, y);
- t0 = x * y;
- t0 = (t0 >> (19 - exp));
- y = vo_extract_l(t0); /* y= (xhigh-xlow)/(yhigh-ylow) in Q11 */
- if (sign < 0)
- y = -y;
- t0 = ylow * y; /* result in Q26 */
- t0 = (t0 >> 10); /* result in Q15 */
- xint = vo_sub(xlow, vo_extract_l(t0)); /* xint = xlow - ylow*y */
- }
- isp[nf] = xint;
- xlow = xint;
- nf++;
- if (ip == 0)
- {
- ip = 1;
- coef = f2;
- order = NC - 1;
- } else
- {
- ip = 0;
- coef = f1;
- order = NC;
- }
- ylow = Chebps2(xlow, coef, order);
- }
- }
- /* Check if M-1 roots found */
- if(nf < M - 1)
- {
- for (i = 0; i < M; i++)
- {
- isp[i] = old_isp[i];
- }
- } else
- {
- isp[M - 1] = a[M] << 3; /* From Q12 to Q15 with saturation */
- }
- return;
+ /*---------------------------------------------------------------------*
+ * Find the ISPs (roots of F1(z) and F2(z) ) using the *
+ * Chebyshev polynomial evaluation. *
+ * The roots of F1(z) and F2(z) are alternatively searched. *
+ * We start by finding the first root of F1(z) then we switch *
+ * to F2(z) then back to F1(z) and so on until all roots are found. *
+ * *
+ * - Evaluate Chebyshev pol. at grid points and check for sign change.*
+ * - If sign change track the root by subdividing the interval *
+ * 2 times and ckecking sign change. *
+ *---------------------------------------------------------------------*/
+ nf = 0; /* number of found frequencies */
+ ip = 0; /* indicator for f1 or f2 */
+ coef = f1;
+ order = NC;
+ xlow = vogrid[0];
+ ylow = Chebps2(xlow, coef, order);
+ j = 0;
+ while ((nf < M - 1) && (j < GRID_POINTS))
+ {
+ j ++;
+ xhigh = xlow;
+ yhigh = ylow;
+ xlow = vogrid[j];
+ ylow = Chebps2(xlow, coef, order);
+ if ((ylow * yhigh) <= (Word32) 0)
+ {
+ /* divide 2 times the interval */
+ for (i = 0; i < 2; i++)
+ {
+ xmid = (xlow >> 1) + (xhigh >> 1); /* xmid = (xlow + xhigh)/2 */
+ ymid = Chebps2(xmid, coef, order);
+ if ((ylow * ymid) <= (Word32) 0)
+ {
+ yhigh = ymid;
+ xhigh = xmid;
+ } else
+ {
+ ylow = ymid;
+ xlow = xmid;
+ }
+ }
+ /*-------------------------------------------------------------*
+ * Linear interpolation *
+ * xint = xlow - ylow*(xhigh-xlow)/(yhigh-ylow); *
+ *-------------------------------------------------------------*/
+ x = xhigh - xlow;
+ y = yhigh - ylow;
+ if (y == 0)
+ {
+ xint = xlow;
+ } else
+ {
+ sign = y;
+ y = abs_s(y);
+ exp = norm_s(y);
+ y = y << exp;
+ y = div_s((Word16) 16383, y);
+ t0 = x * y;
+ t0 = (t0 >> (19 - exp));
+ y = vo_extract_l(t0); /* y= (xhigh-xlow)/(yhigh-ylow) in Q11 */
+ if (sign < 0)
+ y = -y;
+ t0 = ylow * y; /* result in Q26 */
+ t0 = (t0 >> 10); /* result in Q15 */
+ xint = vo_sub(xlow, vo_extract_l(t0)); /* xint = xlow - ylow*y */
+ }
+ isp[nf] = xint;
+ xlow = xint;
+ nf++;
+ if (ip == 0)
+ {
+ ip = 1;
+ coef = f2;
+ order = NC - 1;
+ } else
+ {
+ ip = 0;
+ coef = f1;
+ order = NC;
+ }
+ ylow = Chebps2(xlow, coef, order);
+ }
+ }
+ /* Check if M-1 roots found */
+ if(nf < M - 1)
+ {
+ for (i = 0; i < M; i++)
+ {
+ isp[i] = old_isp[i];
+ }
+ } else
+ {
+ isp[M - 1] = a[M] << 3; /* From Q12 to Q15 with saturation */
+ }
+ return;
}
/*--------------------------------------------------------------*
@@ -213,55 +213,55 @@
static __inline Word16 Chebps2(Word16 x, Word16 f[], Word32 n)
{
- Word32 i, cheb;
- Word16 b0_h, b0_l, b1_h, b1_l, b2_h, b2_l;
- Word32 t0;
+ Word32 i, cheb;
+ Word16 b0_h, b0_l, b1_h, b1_l, b2_h, b2_l;
+ Word32 t0;
- /* Note: All computation are done in Q24. */
+ /* Note: All computation are done in Q24. */
- t0 = f[0] << 13;
- b2_h = t0 >> 16;
- b2_l = (t0 & 0xffff)>>1;
+ t0 = f[0] << 13;
+ b2_h = t0 >> 16;
+ b2_l = (t0 & 0xffff)>>1;
- t0 = ((b2_h * x)<<1) + (((b2_l * x)>>15)<<1);
- t0 <<= 1;
- t0 += (f[1] << 13); /* + f[1] in Q24 */
+ t0 = ((b2_h * x)<<1) + (((b2_l * x)>>15)<<1);
+ t0 <<= 1;
+ t0 += (f[1] << 13); /* + f[1] in Q24 */
- b1_h = t0 >> 16;
- b1_l = (t0 & 0xffff) >> 1;
+ b1_h = t0 >> 16;
+ b1_l = (t0 & 0xffff) >> 1;
- for (i = 2; i < n; i++)
- {
- t0 = ((b1_h * x)<<1) + (((b1_l * x)>>15)<<1);
+ for (i = 2; i < n; i++)
+ {
+ t0 = ((b1_h * x)<<1) + (((b1_l * x)>>15)<<1);
- t0 += (b2_h * (-16384))<<1;
- t0 += (f[i] << 12);
- t0 <<= 1;
- t0 -= (b2_l << 1); /* t0 = 2.0*x*b1 - b2 + f[i]; */
+ t0 += (b2_h * (-16384))<<1;
+ t0 += (f[i] << 12);
+ t0 <<= 1;
+ t0 -= (b2_l << 1); /* t0 = 2.0*x*b1 - b2 + f[i]; */
- b0_h = t0 >> 16;
- b0_l = (t0 & 0xffff) >> 1;
+ b0_h = t0 >> 16;
+ b0_l = (t0 & 0xffff) >> 1;
- b2_l = b1_l; /* b2 = b1; */
- b2_h = b1_h;
- b1_l = b0_l; /* b1 = b0; */
- b1_h = b0_h;
- }
+ b2_l = b1_l; /* b2 = b1; */
+ b2_h = b1_h;
+ b1_l = b0_l; /* b1 = b0; */
+ b1_h = b0_h;
+ }
- t0 = ((b1_h * x)<<1) + (((b1_l * x)>>15)<<1);
- t0 += (b2_h * (-32768))<<1; /* t0 = x*b1 - b2 */
- t0 -= (b2_l << 1);
- t0 += (f[n] << 12); /* t0 = x*b1 - b2 + f[i]/2 */
+ t0 = ((b1_h * x)<<1) + (((b1_l * x)>>15)<<1);
+ t0 += (b2_h * (-32768))<<1; /* t0 = x*b1 - b2 */
+ t0 -= (b2_l << 1);
+ t0 += (f[n] << 12); /* t0 = x*b1 - b2 + f[i]/2 */
- t0 = L_shl2(t0, 6); /* Q24 to Q30 with saturation */
+ t0 = L_shl2(t0, 6); /* Q24 to Q30 with saturation */
- cheb = extract_h(t0); /* Result in Q14 */
+ cheb = extract_h(t0); /* Result in Q14 */
- if (cheb == -32768)
- {
- cheb = -32767; /* to avoid saturation in Az_isp */
- }
- return (cheb);
+ if (cheb == -32768)
+ {
+ cheb = -32767; /* to avoid saturation in Az_isp */
+ }
+ return (cheb);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/bits.c b/media/libstagefright/codecs/amrwbenc/src/bits.c
index e78dc1f..6b8bddd 100644
--- a/media/libstagefright/codecs/amrwbenc/src/bits.c
+++ b/media/libstagefright/codecs/amrwbenc/src/bits.c
@@ -17,7 +17,7 @@
/***********************************************************************
File: bits.c
- Description: Performs bit stream manipulation
+ Description: Performs bit stream manipulation
************************************************************************/
@@ -33,151 +33,151 @@
int PackBits(Word16 prms[], /* i: analysis parameters */
- Word16 coding_mode, /* i: coding bit-stream ratio mode */
- Word16 mode, /* i: coding bit-stream ratio mode*/
- Coder_State *st /*i/o: coder global parameters struct */
- )
+ Word16 coding_mode, /* i: coding bit-stream ratio mode */
+ Word16 mode, /* i: coding bit-stream ratio mode*/
+ Coder_State *st /*i/o: coder global parameters struct */
+ )
{
- Word16 i, frame_type;
- UWord8 temp;
- UWord8 *stream_ptr;
- Word16 bitstreamformat = st->frameType;
+ Word16 i, frame_type;
+ UWord8 temp;
+ UWord8 *stream_ptr;
+ Word16 bitstreamformat = st->frameType;
- unsigned short* dataOut = st->outputStream;
+ unsigned short* dataOut = st->outputStream;
- if (coding_mode == MRDTX)
- {
- st->sid_update_counter--;
+ if (coding_mode == MRDTX)
+ {
+ st->sid_update_counter--;
- if (st->prev_ft == TX_SPEECH)
- {
- frame_type = TX_SID_FIRST;
- st->sid_update_counter = 3;
- } else
- {
- if ((st->sid_handover_debt > 0) && (st->sid_update_counter > 2))
- {
- /* ensure extra updates are properly delayed after a possible SID_FIRST */
- frame_type = TX_SID_UPDATE;
- st->sid_handover_debt--;
- } else
- {
- if (st->sid_update_counter == 0)
- {
- frame_type = TX_SID_UPDATE;
- st->sid_update_counter = 8;
- } else
- {
- frame_type = TX_NO_DATA;
- }
- }
- }
- } else
- {
- st->sid_update_counter = 8;
- frame_type = TX_SPEECH;
- }
- st->prev_ft = frame_type;
+ if (st->prev_ft == TX_SPEECH)
+ {
+ frame_type = TX_SID_FIRST;
+ st->sid_update_counter = 3;
+ } else
+ {
+ if ((st->sid_handover_debt > 0) && (st->sid_update_counter > 2))
+ {
+ /* ensure extra updates are properly delayed after a possible SID_FIRST */
+ frame_type = TX_SID_UPDATE;
+ st->sid_handover_debt--;
+ } else
+ {
+ if (st->sid_update_counter == 0)
+ {
+ frame_type = TX_SID_UPDATE;
+ st->sid_update_counter = 8;
+ } else
+ {
+ frame_type = TX_NO_DATA;
+ }
+ }
+ }
+ } else
+ {
+ st->sid_update_counter = 8;
+ frame_type = TX_SPEECH;
+ }
+ st->prev_ft = frame_type;
- if(bitstreamformat == 0) /* default file format */
- {
- *(dataOut) = TX_FRAME_TYPE;
- *(dataOut + 1) = frame_type;
- *(dataOut + 2) = mode;
- for (i = 0; i < nb_of_bits[coding_mode]; i++)
- {
- *(dataOut + 3 + i) = prms[i];
- }
- return (3 + nb_of_bits[coding_mode])<<1;
- } else
- {
- if (bitstreamformat == 1) /* ITU file format */
- {
- *(dataOut) = 0x6b21;
- if(frame_type != TX_NO_DATA && frame_type != TX_SID_FIRST)
- {
- *(dataOut + 1) = nb_of_bits[coding_mode];
- for (i = 0; i < nb_of_bits[coding_mode]; i++)
- {
- if(prms[i] == BIT_0){
- *(dataOut + 2 + i) = BIT_0_ITU;
- }
- else{
- *(dataOut + 2 + i) = BIT_1_ITU;
- }
- }
- return (2 + nb_of_bits[coding_mode])<<1;
- } else
- {
- *(dataOut + 1) = 0;
- return 2<<1;
- }
- } else /* MIME/storage file format */
- {
+ if(bitstreamformat == 0) /* default file format */
+ {
+ *(dataOut) = TX_FRAME_TYPE;
+ *(dataOut + 1) = frame_type;
+ *(dataOut + 2) = mode;
+ for (i = 0; i < nb_of_bits[coding_mode]; i++)
+ {
+ *(dataOut + 3 + i) = prms[i];
+ }
+ return (3 + nb_of_bits[coding_mode])<<1;
+ } else
+ {
+ if (bitstreamformat == 1) /* ITU file format */
+ {
+ *(dataOut) = 0x6b21;
+ if(frame_type != TX_NO_DATA && frame_type != TX_SID_FIRST)
+ {
+ *(dataOut + 1) = nb_of_bits[coding_mode];
+ for (i = 0; i < nb_of_bits[coding_mode]; i++)
+ {
+ if(prms[i] == BIT_0){
+ *(dataOut + 2 + i) = BIT_0_ITU;
+ }
+ else{
+ *(dataOut + 2 + i) = BIT_1_ITU;
+ }
+ }
+ return (2 + nb_of_bits[coding_mode])<<1;
+ } else
+ {
+ *(dataOut + 1) = 0;
+ return 2<<1;
+ }
+ } else /* MIME/storage file format */
+ {
#define MRSID 9
- /* change mode index in case of SID frame */
- if (coding_mode == MRDTX)
- {
- coding_mode = MRSID;
- if (frame_type == TX_SID_FIRST)
- {
- for (i = 0; i < NBBITS_SID; i++) prms[i] = BIT_0;
- }
- }
- /* -> force NO_DATA frame */
- if (coding_mode < 0 || coding_mode > 15 || (coding_mode > MRSID && coding_mode < 14))
- {
- coding_mode = 15;
- }
- /* mark empty frames between SID updates as NO_DATA frames */
- if (coding_mode == MRSID && frame_type == TX_NO_DATA)
- {
- coding_mode = 15;
- }
- /* set pointer for packed frame, note that we handle data as bytes */
- stream_ptr = (UWord8*)dataOut;
- /* insert table of contents (ToC) byte at the beginning of the packet */
- *stream_ptr = toc_byte[coding_mode];
- stream_ptr++;
- temp = 0;
- /* sort and pack AMR-WB speech or SID bits */
- for (i = 1; i < unpacked_size[coding_mode] + 1; i++)
- {
- if (prms[sort_ptr[coding_mode][i-1]] == BIT_1)
- {
- temp++;
- }
- if (i&0x7)
- {
- temp <<= 1;
- }
- else
- {
- *stream_ptr = temp;
- stream_ptr++;
- temp = 0;
- }
- }
- /* insert SID type indication and speech mode in case of SID frame */
- if (coding_mode == MRSID)
- {
- if (frame_type == TX_SID_UPDATE)
- {
- temp++;
- }
- temp <<= 4;
- temp += mode & 0x000F;
- }
- /* insert unused bits (zeros) at the tail of the last byte */
- if (unused_size[coding_mode])
- {
- temp <<= (unused_size[coding_mode] - 1);
- }
- *stream_ptr = temp;
- /* write packed frame into file (1 byte added to cover ToC entry) */
- return (1 + packed_size[coding_mode]);
- }
- }
+ /* change mode index in case of SID frame */
+ if (coding_mode == MRDTX)
+ {
+ coding_mode = MRSID;
+ if (frame_type == TX_SID_FIRST)
+ {
+ for (i = 0; i < NBBITS_SID; i++) prms[i] = BIT_0;
+ }
+ }
+ /* -> force NO_DATA frame */
+ if (coding_mode < 0 || coding_mode > 15 || (coding_mode > MRSID && coding_mode < 14))
+ {
+ coding_mode = 15;
+ }
+ /* mark empty frames between SID updates as NO_DATA frames */
+ if (coding_mode == MRSID && frame_type == TX_NO_DATA)
+ {
+ coding_mode = 15;
+ }
+ /* set pointer for packed frame, note that we handle data as bytes */
+ stream_ptr = (UWord8*)dataOut;
+ /* insert table of contents (ToC) byte at the beginning of the packet */
+ *stream_ptr = toc_byte[coding_mode];
+ stream_ptr++;
+ temp = 0;
+ /* sort and pack AMR-WB speech or SID bits */
+ for (i = 1; i < unpacked_size[coding_mode] + 1; i++)
+ {
+ if (prms[sort_ptr[coding_mode][i-1]] == BIT_1)
+ {
+ temp++;
+ }
+ if (i&0x7)
+ {
+ temp <<= 1;
+ }
+ else
+ {
+ *stream_ptr = temp;
+ stream_ptr++;
+ temp = 0;
+ }
+ }
+ /* insert SID type indication and speech mode in case of SID frame */
+ if (coding_mode == MRSID)
+ {
+ if (frame_type == TX_SID_UPDATE)
+ {
+ temp++;
+ }
+ temp <<= 4;
+ temp += mode & 0x000F;
+ }
+ /* insert unused bits (zeros) at the tail of the last byte */
+ if (unused_size[coding_mode])
+ {
+ temp <<= (unused_size[coding_mode] - 1);
+ }
+ *stream_ptr = temp;
+ /* write packed frame into file (1 byte added to cover ToC entry) */
+ return (1 + packed_size[coding_mode]);
+ }
+ }
}
/*-----------------------------------------------------*
@@ -185,24 +185,24 @@
*-----------------------------------------------------*/
void Parm_serial(
- Word16 value, /* input : parameter value */
- Word16 no_of_bits, /* input : number of bits */
- Word16 ** prms
- )
+ Word16 value, /* input : parameter value */
+ Word16 no_of_bits, /* input : number of bits */
+ Word16 ** prms
+ )
{
- Word16 i, bit;
- *prms += no_of_bits;
- for (i = 0; i < no_of_bits; i++)
- {
- bit = (Word16) (value & 0x0001); /* get lsb */
- if (bit == 0)
- *--(*prms) = BIT_0;
- else
- *--(*prms) = BIT_1;
- value >>= 1;
- }
- *prms += no_of_bits;
- return;
+ Word16 i, bit;
+ *prms += no_of_bits;
+ for (i = 0; i < no_of_bits; i++)
+ {
+ bit = (Word16) (value & 0x0001); /* get lsb */
+ if (bit == 0)
+ *--(*prms) = BIT_0;
+ else
+ *--(*prms) = BIT_1;
+ value >>= 1;
+ }
+ *prms += no_of_bits;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/c2t64fx.c b/media/libstagefright/codecs/amrwbenc/src/c2t64fx.c
index 768abd4..c7c9279 100644
--- a/media/libstagefright/codecs/amrwbenc/src/c2t64fx.c
+++ b/media/libstagefright/codecs/amrwbenc/src/c2t64fx.c
@@ -17,7 +17,7 @@
/************************************************************************
* File: c2t64fx.c *
* *
-* Description:Performs algebraic codebook search for 6.60kbits mode*
+* Description:Performs algebraic codebook search for 6.60kbits mode*
* *
*************************************************************************/
@@ -44,257 +44,256 @@
* Each pulse can have 32 possible positions. *
**************************************************************************/
-// There are many integer overflows in this function, as none of them appear to
-// lead to memory accesses, and performing the appropriate checks will lead
-// to considerably larger code, mark this as ignore.
-__attribute__((no_sanitize("integer")))
void ACELP_2t64_fx(
- Word16 dn[], /* (i) <12b : correlation between target x[] and H[] */
- Word16 cn[], /* (i) <12b : residual after long term prediction */
- Word16 H[], /* (i) Q12: impulse response of weighted synthesis filter */
- Word16 code[], /* (o) Q9 : algebraic (fixed) codebook excitation */
- Word16 y[], /* (o) Q9 : filtered fixed codebook excitation */
- Word16 * index /* (o) : index (12): 5+1+5+1 = 11 bits. */
- )
+ Word16 dn[], /* (i) <12b : correlation between target x[] and H[] */
+ Word16 cn[], /* (i) <12b : residual after long term prediction */
+ Word16 H[], /* (i) Q12: impulse response of weighted synthesis filter */
+ Word16 code[], /* (o) Q9 : algebraic (fixed) codebook excitation */
+ Word16 y[], /* (o) Q9 : filtered fixed codebook excitation */
+ Word16 * index /* (o) : index (12): 5+1+5+1 = 11 bits. */
+ )
{
- Word32 i, j, k, i0, i1, ix, iy, pos, pos2;
- Word16 ps, psk, ps1, ps2, alpk, alp1, alp2, sq;
- Word16 alp, val, exp, k_cn, k_dn;
- Word16 *p0, *p1, *p2, *psign;
- Word16 *h, *h_inv, *ptr_h1, *ptr_h2, *ptr_hf;
+ Word32 i, j, k, i0, i1, ix, iy, pos, pos2;
+ Word16 ps, psk, ps1, ps2, alpk, alp1, alp2, sq;
+ Word16 alp, val, exp, k_cn, k_dn;
+ Word16 *p0, *p1, *p2, *psign;
+ Word16 *h, *h_inv, *ptr_h1, *ptr_h2, *ptr_hf;
- Word16 sign[L_SUBFR], vec[L_SUBFR], dn2[L_SUBFR];
- Word16 h_buf[4 * L_SUBFR] = {0};
- Word16 rrixix[NB_TRACK][NB_POS];
- Word16 rrixiy[MSIZE];
- Word32 s, cor;
+ Word16 sign[L_SUBFR], vec[L_SUBFR], dn2[L_SUBFR];
+ Word16 h_buf[4 * L_SUBFR] = {0};
+ Word16 rrixix[NB_TRACK][NB_POS];
+ Word16 rrixiy[MSIZE];
+ Word32 s, cor;
- /*----------------------------------------------------------------*
- * Find sign for each pulse position. *
- *----------------------------------------------------------------*/
- alp = 8192; /* alp = 2.0 (Q12) */
+ /*----------------------------------------------------------------*
+ * Find sign for each pulse position. *
+ *----------------------------------------------------------------*/
+ alp = 8192; /* alp = 2.0 (Q12) */
- /* calculate energy for normalization of cn[] and dn[] */
- /* set k_cn = 32..32767 (ener_cn = 2^30..256-0) */
+ /* calculate energy for normalization of cn[] and dn[] */
+ /* set k_cn = 32..32767 (ener_cn = 2^30..256-0) */
#ifdef ASM_OPT /* asm optimization branch */
- s = Dot_product12_asm(cn, cn, L_SUBFR, &exp);
+ s = Dot_product12_asm(cn, cn, L_SUBFR, &exp);
#else
- s = Dot_product12(cn, cn, L_SUBFR, &exp);
+ s = Dot_product12(cn, cn, L_SUBFR, &exp);
#endif
- Isqrt_n(&s, &exp);
- s = L_shl(s, add1(exp, 5));
- k_cn = vo_round(s);
+ Isqrt_n(&s, &exp);
+ s = L_shl(s, add1(exp, 5));
+ if (s > INT_MAX - 0x8000) {
+ s = INT_MAX - 0x8000;
+ }
+ k_cn = vo_round(s);
- /* set k_dn = 32..512 (ener_dn = 2^30..2^22) */
+ /* set k_dn = 32..512 (ener_dn = 2^30..2^22) */
#ifdef ASM_OPT /* asm optimization branch */
- s = Dot_product12_asm(dn, dn, L_SUBFR, &exp);
+ s = Dot_product12_asm(dn, dn, L_SUBFR, &exp);
#else
- s = Dot_product12(dn, dn, L_SUBFR, &exp);
+ s = Dot_product12(dn, dn, L_SUBFR, &exp);
#endif
- Isqrt_n(&s, &exp);
- k_dn = vo_round(L_shl(s, (exp + 8))); /* k_dn = 256..4096 */
- k_dn = vo_mult_r(alp, k_dn); /* alp in Q12 */
+ Isqrt_n(&s, &exp);
+ k_dn = vo_round(L_shl(s, (exp + 8))); /* k_dn = 256..4096 */
+ k_dn = vo_mult_r(alp, k_dn); /* alp in Q12 */
- /* mix normalized cn[] and dn[] */
- p0 = cn;
- p1 = dn;
- p2 = dn2;
+ /* mix normalized cn[] and dn[] */
+ p0 = cn;
+ p1 = dn;
+ p2 = dn2;
- for (i = 0; i < L_SUBFR/4; i++)
- {
- s = (k_cn* (*p0++))+(k_dn * (*p1++));
- *p2++ = s >> 7;
- s = (k_cn* (*p0++))+(k_dn * (*p1++));
- *p2++ = s >> 7;
- s = (k_cn* (*p0++))+(k_dn * (*p1++));
- *p2++ = s >> 7;
- s = (k_cn* (*p0++))+(k_dn * (*p1++));
- *p2++ = s >> 7;
- }
+ for (i = 0; i < L_SUBFR/4; i++)
+ {
+ s = (k_cn* (*p0++))+(k_dn * (*p1++));
+ *p2++ = s >> 7;
+ s = (k_cn* (*p0++))+(k_dn * (*p1++));
+ *p2++ = s >> 7;
+ s = (k_cn* (*p0++))+(k_dn * (*p1++));
+ *p2++ = s >> 7;
+ s = (k_cn* (*p0++))+(k_dn * (*p1++));
+ *p2++ = s >> 7;
+ }
- /* set sign according to dn2[] = k_cn*cn[] + k_dn*dn[] */
- for (i = 0; i < L_SUBFR; i ++)
- {
- val = dn[i];
- ps = dn2[i];
- if (ps >= 0)
- {
- sign[i] = 32767; /* sign = +1 (Q12) */
- vec[i] = -32768;
- } else
- {
- sign[i] = -32768; /* sign = -1 (Q12) */
- vec[i] = 32767;
- dn[i] = -val;
- }
- }
- /*------------------------------------------------------------*
- * Compute h_inv[i]. *
- *------------------------------------------------------------*/
- /* impulse response buffer for fast computation */
- h = h_buf + L_SUBFR;
- h_inv = h + (L_SUBFR<<1);
+ /* set sign according to dn2[] = k_cn*cn[] + k_dn*dn[] */
+ for (i = 0; i < L_SUBFR; i ++)
+ {
+ val = dn[i];
+ ps = dn2[i];
+ if (ps >= 0)
+ {
+ sign[i] = 32767; /* sign = +1 (Q12) */
+ vec[i] = -32768;
+ } else
+ {
+ sign[i] = -32768; /* sign = -1 (Q12) */
+ vec[i] = 32767;
+ dn[i] = -val;
+ }
+ }
+ /*------------------------------------------------------------*
+ * Compute h_inv[i]. *
+ *------------------------------------------------------------*/
+ /* impulse response buffer for fast computation */
+ h = h_buf + L_SUBFR;
+ h_inv = h + (L_SUBFR<<1);
- for (i = 0; i < L_SUBFR; i++)
- {
- h[i] = H[i];
- h_inv[i] = vo_negate(h[i]);
- }
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ h[i] = H[i];
+ h_inv[i] = vo_negate(h[i]);
+ }
- /*------------------------------------------------------------*
- * Compute rrixix[][] needed for the codebook search. *
- * Result is multiplied by 0.5 *
- *------------------------------------------------------------*/
- /* Init pointers to last position of rrixix[] */
- p0 = &rrixix[0][NB_POS - 1];
- p1 = &rrixix[1][NB_POS - 1];
+ /*------------------------------------------------------------*
+ * Compute rrixix[][] needed for the codebook search. *
+ * Result is multiplied by 0.5 *
+ *------------------------------------------------------------*/
+ /* Init pointers to last position of rrixix[] */
+ p0 = &rrixix[0][NB_POS - 1];
+ p1 = &rrixix[1][NB_POS - 1];
- ptr_h1 = h;
- cor = 0x00010000L; /* for rounding */
- for (i = 0; i < NB_POS; i++)
- {
- cor += ((*ptr_h1) * (*ptr_h1) << 1);
- ptr_h1++;
- *p1-- = (extract_h(cor) >> 1);
- cor += ((*ptr_h1) * (*ptr_h1) << 1);
- ptr_h1++;
- *p0-- = (extract_h(cor) >> 1);
- }
+ ptr_h1 = h;
+ cor = 0x00010000L; /* for rounding */
+ for (i = 0; i < NB_POS; i++)
+ {
+ cor += ((*ptr_h1) * (*ptr_h1) << 1);
+ ptr_h1++;
+ *p1-- = (extract_h(cor) >> 1);
+ cor += ((*ptr_h1) * (*ptr_h1) << 1);
+ ptr_h1++;
+ *p0-- = (extract_h(cor) >> 1);
+ }
- /*------------------------------------------------------------*
- * Compute rrixiy[][] needed for the codebook search. *
- *------------------------------------------------------------*/
- pos = MSIZE - 1;
- pos2 = MSIZE - 2;
- ptr_hf = h + 1;
+ /*------------------------------------------------------------*
+ * Compute rrixiy[][] needed for the codebook search. *
+ *------------------------------------------------------------*/
+ pos = MSIZE - 1;
+ pos2 = MSIZE - 2;
+ ptr_hf = h + 1;
- for (k = 0; k < NB_POS; k++)
- {
- p1 = &rrixiy[pos];
- p0 = &rrixiy[pos2];
- cor = 0x00008000L; /* for rounding */
- ptr_h1 = h;
- ptr_h2 = ptr_hf;
+ for (k = 0; k < NB_POS; k++)
+ {
+ p1 = &rrixiy[pos];
+ p0 = &rrixiy[pos2];
+ cor = 0x00008000L; /* for rounding */
+ ptr_h1 = h;
+ ptr_h2 = ptr_hf;
- for (i = (k + 1); i < NB_POS; i++)
- {
- cor += ((*ptr_h1) * (*ptr_h2))<<1;
- ptr_h1++;
- ptr_h2++;
- *p1 = extract_h(cor);
- cor += ((*ptr_h1) * (*ptr_h2))<<1;
- ptr_h1++;
- ptr_h2++;
- *p0 = extract_h(cor);
+ for (i = (k + 1); i < NB_POS; i++)
+ {
+ cor += ((*ptr_h1) * (*ptr_h2))<<1;
+ ptr_h1++;
+ ptr_h2++;
+ *p1 = extract_h(cor);
+ cor += ((*ptr_h1) * (*ptr_h2))<<1;
+ ptr_h1++;
+ ptr_h2++;
+ *p0 = extract_h(cor);
- p1 -= (NB_POS + 1);
- p0 -= (NB_POS + 1);
- }
- cor += ((*ptr_h1) * (*ptr_h2))<<1;
- ptr_h1++;
- ptr_h2++;
- *p1 = extract_h(cor);
+ p1 -= (NB_POS + 1);
+ p0 -= (NB_POS + 1);
+ }
+ cor += ((*ptr_h1) * (*ptr_h2))<<1;
+ ptr_h1++;
+ ptr_h2++;
+ *p1 = extract_h(cor);
- pos -= NB_POS;
- pos2--;
- ptr_hf += STEP;
- }
+ pos -= NB_POS;
+ pos2--;
+ ptr_hf += STEP;
+ }
- /*------------------------------------------------------------*
- * Modification of rrixiy[][] to take signs into account. *
- *------------------------------------------------------------*/
- p0 = rrixiy;
- for (i = 0; i < L_SUBFR; i += STEP)
- {
- psign = sign;
- if (psign[i] < 0)
- {
- psign = vec;
- }
- for (j = 1; j < L_SUBFR; j += STEP)
- {
- *p0 = vo_mult(*p0, psign[j]);
- p0++;
- }
- }
- /*-------------------------------------------------------------------*
- * search 2 pulses: *
- * ~@~~~~~~~~~~~~~~ *
- * 32 pos x 32 pos = 1024 tests (all combinaisons is tested) *
- *-------------------------------------------------------------------*/
- p0 = rrixix[0];
- p1 = rrixix[1];
- p2 = rrixiy;
+ /*------------------------------------------------------------*
+ * Modification of rrixiy[][] to take signs into account. *
+ *------------------------------------------------------------*/
+ p0 = rrixiy;
+ for (i = 0; i < L_SUBFR; i += STEP)
+ {
+ psign = sign;
+ if (psign[i] < 0)
+ {
+ psign = vec;
+ }
+ for (j = 1; j < L_SUBFR; j += STEP)
+ {
+ *p0 = vo_mult(*p0, psign[j]);
+ p0++;
+ }
+ }
+ /*-------------------------------------------------------------------*
+ * search 2 pulses: *
+ * ~@~~~~~~~~~~~~~~ *
+ * 32 pos x 32 pos = 1024 tests (all combinaisons is tested) *
+ *-------------------------------------------------------------------*/
+ p0 = rrixix[0];
+ p1 = rrixix[1];
+ p2 = rrixiy;
- psk = -1;
- alpk = 1;
- ix = 0;
- iy = 1;
+ psk = -1;
+ alpk = 1;
+ ix = 0;
+ iy = 1;
- for (i0 = 0; i0 < L_SUBFR; i0 += STEP)
- {
- ps1 = dn[i0];
- alp1 = (*p0++);
- pos = -1;
- for (i1 = 1; i1 < L_SUBFR; i1 += STEP)
- {
- ps2 = add1(ps1, dn[i1]);
- alp2 = add1(alp1, add1(*p1++, *p2++));
- sq = vo_mult(ps2, ps2);
- s = vo_L_mult(alpk, sq) - ((psk * alp2)<<1);
- if (s > 0)
- {
- psk = sq;
- alpk = alp2;
- pos = i1;
- }
- }
- p1 -= NB_POS;
- if (pos >= 0)
- {
- ix = i0;
- iy = pos;
- }
- }
- /*-------------------------------------------------------------------*
- * Build the codeword, the filtered codeword and index of codevector.*
- *-------------------------------------------------------------------*/
+ for (i0 = 0; i0 < L_SUBFR; i0 += STEP)
+ {
+ ps1 = dn[i0];
+ alp1 = (*p0++);
+ pos = -1;
+ for (i1 = 1; i1 < L_SUBFR; i1 += STEP)
+ {
+ ps2 = add1(ps1, dn[i1]);
+ alp2 = add1(alp1, add1(*p1++, *p2++));
+ sq = vo_mult(ps2, ps2);
+ s = vo_L_mult(alpk, sq) - ((psk * alp2)<<1);
+ if (s > 0)
+ {
+ psk = sq;
+ alpk = alp2;
+ pos = i1;
+ }
+ }
+ p1 -= NB_POS;
+ if (pos >= 0)
+ {
+ ix = i0;
+ iy = pos;
+ }
+ }
+ /*-------------------------------------------------------------------*
+ * Build the codeword, the filtered codeword and index of codevector.*
+ *-------------------------------------------------------------------*/
- for (i = 0; i < L_SUBFR; i++)
- {
- code[i] = 0;
- }
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ code[i] = 0;
+ }
- i0 = (ix >> 1); /* pos of pulse 1 (0..31) */
- i1 = (iy >> 1); /* pos of pulse 2 (0..31) */
- if (sign[ix] > 0)
- {
- code[ix] = 512; /* codeword in Q9 format */
- p0 = h - ix;
- } else
- {
- code[ix] = -512;
- i0 += NB_POS;
- p0 = h_inv - ix;
- }
- if (sign[iy] > 0)
- {
- code[iy] = 512;
- p1 = h - iy;
- } else
- {
- code[iy] = -512;
- i1 += NB_POS;
- p1 = h_inv - iy;
- }
- *index = add1((i0 << 6), i1);
- for (i = 0; i < L_SUBFR; i++)
- {
- y[i] = vo_shr_r(add1((*p0++), (*p1++)), 3);
- }
- return;
+ i0 = (ix >> 1); /* pos of pulse 1 (0..31) */
+ i1 = (iy >> 1); /* pos of pulse 2 (0..31) */
+ if (sign[ix] > 0)
+ {
+ code[ix] = 512; /* codeword in Q9 format */
+ p0 = h - ix;
+ } else
+ {
+ code[ix] = -512;
+ i0 += NB_POS;
+ p0 = h_inv - ix;
+ }
+ if (sign[iy] > 0)
+ {
+ code[iy] = 512;
+ p1 = h - iy;
+ } else
+ {
+ code[iy] = -512;
+ i1 += NB_POS;
+ p1 = h_inv - iy;
+ }
+ *index = add1((i0 << 6), i1);
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ y[i] = vo_shr_r(add1((*p0++), (*p1++)), 3);
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/c4t64fx.c b/media/libstagefright/codecs/amrwbenc/src/c4t64fx.c
index 8704ce5..b9a9e26 100644
--- a/media/libstagefright/codecs/amrwbenc/src/c4t64fx.c
+++ b/media/libstagefright/codecs/amrwbenc/src/c4t64fx.c
@@ -17,7 +17,7 @@
/***********************************************************************
* File: c4t64fx.c *
* *
-* Description:Performs algebraic codebook search for higher modes *
+* Description:Performs algebraic codebook search for higher modes *
* *
************************************************************************/
@@ -48,15 +48,15 @@
#include "q_pulse.h"
static Word16 tipos[36] = {
- 0, 1, 2, 3, /* starting point &ipos[0], 1st iter */
- 1, 2, 3, 0, /* starting point &ipos[4], 2nd iter */
- 2, 3, 0, 1, /* starting point &ipos[8], 3rd iter */
- 3, 0, 1, 2, /* starting point &ipos[12], 4th iter */
- 0, 1, 2, 3,
- 1, 2, 3, 0,
- 2, 3, 0, 1,
- 3, 0, 1, 2,
- 0, 1, 2, 3}; /* end point for 24 pulses &ipos[35], 4th iter */
+ 0, 1, 2, 3, /* starting point &ipos[0], 1st iter */
+ 1, 2, 3, 0, /* starting point &ipos[4], 2nd iter */
+ 2, 3, 0, 1, /* starting point &ipos[8], 3rd iter */
+ 3, 0, 1, 2, /* starting point &ipos[12], 4th iter */
+ 0, 1, 2, 3,
+ 1, 2, 3, 0,
+ 2, 3, 0, 1,
+ 3, 0, 1, 2,
+ 0, 1, 2, 3}; /* end point for 24 pulses &ipos[35], 4th iter */
#define NB_PULSE_MAX 24
@@ -70,752 +70,759 @@
/* Private functions */
void cor_h_vec_012(
- Word16 h[], /* (i) scaled impulse response */
- Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
- Word16 track, /* (i) track to use */
- Word16 sign[], /* (i) sign vector */
- Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
- Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
- Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
- );
+ Word16 h[], /* (i) scaled impulse response */
+ Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
+ Word16 track, /* (i) track to use */
+ Word16 sign[], /* (i) sign vector */
+ Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
+ Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
+ Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
+ );
void cor_h_vec_012_asm(
- Word16 h[], /* (i) scaled impulse response */
- Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
- Word16 track, /* (i) track to use */
- Word16 sign[], /* (i) sign vector */
- Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
- Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
- Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
- );
+ Word16 h[], /* (i) scaled impulse response */
+ Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
+ Word16 track, /* (i) track to use */
+ Word16 sign[], /* (i) sign vector */
+ Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
+ Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
+ Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
+ );
void cor_h_vec_30(
- Word16 h[], /* (i) scaled impulse response */
- Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
- Word16 track, /* (i) track to use */
- Word16 sign[], /* (i) sign vector */
- Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
- Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
- Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
- );
+ Word16 h[], /* (i) scaled impulse response */
+ Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
+ Word16 track, /* (i) track to use */
+ Word16 sign[], /* (i) sign vector */
+ Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
+ Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
+ Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
+ );
void search_ixiy(
- Word16 nb_pos_ix, /* (i) nb of pos for pulse 1 (1..8) */
- Word16 track_x, /* (i) track of pulse 1 */
- Word16 track_y, /* (i) track of pulse 2 */
- Word16 * ps, /* (i/o) correlation of all fixed pulses */
- Word16 * alp, /* (i/o) energy of all fixed pulses */
- Word16 * ix, /* (o) position of pulse 1 */
- Word16 * iy, /* (o) position of pulse 2 */
- Word16 dn[], /* (i) corr. between target and h[] */
- Word16 dn2[], /* (i) vector of selected positions */
- Word16 cor_x[], /* (i) corr. of pulse 1 with fixed pulses */
- Word16 cor_y[], /* (i) corr. of pulse 2 with fixed pulses */
- Word16 rrixiy[][MSIZE] /* (i) corr. of pulse 1 with pulse 2 */
- );
+ Word16 nb_pos_ix, /* (i) nb of pos for pulse 1 (1..8) */
+ Word16 track_x, /* (i) track of pulse 1 */
+ Word16 track_y, /* (i) track of pulse 2 */
+ Word16 * ps, /* (i/o) correlation of all fixed pulses */
+ Word16 * alp, /* (i/o) energy of all fixed pulses */
+ Word16 * ix, /* (o) position of pulse 1 */
+ Word16 * iy, /* (o) position of pulse 2 */
+ Word16 dn[], /* (i) corr. between target and h[] */
+ Word16 dn2[], /* (i) vector of selected positions */
+ Word16 cor_x[], /* (i) corr. of pulse 1 with fixed pulses */
+ Word16 cor_y[], /* (i) corr. of pulse 2 with fixed pulses */
+ Word16 rrixiy[][MSIZE] /* (i) corr. of pulse 1 with pulse 2 */
+ );
-__attribute__((no_sanitize("integer")))
void ACELP_4t64_fx(
- Word16 dn[], /* (i) <12b : correlation between target x[] and H[] */
- Word16 cn[], /* (i) <12b : residual after long term prediction */
- Word16 H[], /* (i) Q12: impulse response of weighted synthesis filter */
- Word16 code[], /* (o) Q9 : algebraic (fixed) codebook excitation */
- Word16 y[], /* (o) Q9 : filtered fixed codebook excitation */
- Word16 nbbits, /* (i) : 20, 36, 44, 52, 64, 72 or 88 bits */
- Word16 ser_size, /* (i) : bit rate */
- Word16 _index[] /* (o) : index (20): 5+5+5+5 = 20 bits. */
- /* (o) : index (36): 9+9+9+9 = 36 bits. */
- /* (o) : index (44): 13+9+13+9 = 44 bits. */
- /* (o) : index (52): 13+13+13+13 = 52 bits. */
- /* (o) : index (64): 2+2+2+2+14+14+14+14 = 64 bits. */
- /* (o) : index (72): 10+2+10+2+10+14+10+14 = 72 bits. */
- /* (o) : index (88): 11+11+11+11+11+11+11+11 = 88 bits. */
- )
+ Word16 dn[], /* (i) <12b : correlation between target x[] and H[] */
+ Word16 cn[], /* (i) <12b : residual after long term prediction */
+ Word16 H[], /* (i) Q12: impulse response of weighted synthesis filter */
+ Word16 code[], /* (o) Q9 : algebraic (fixed) codebook excitation */
+ Word16 y[], /* (o) Q9 : filtered fixed codebook excitation */
+ Word16 nbbits, /* (i) : 20, 36, 44, 52, 64, 72 or 88 bits */
+ Word16 ser_size, /* (i) : bit rate */
+ Word16 _index[] /* (o) : index (20): 5+5+5+5 = 20 bits. */
+ /* (o) : index (36): 9+9+9+9 = 36 bits. */
+ /* (o) : index (44): 13+9+13+9 = 44 bits. */
+ /* (o) : index (52): 13+13+13+13 = 52 bits. */
+ /* (o) : index (64): 2+2+2+2+14+14+14+14 = 64 bits. */
+ /* (o) : index (72): 10+2+10+2+10+14+10+14 = 72 bits. */
+ /* (o) : index (88): 11+11+11+11+11+11+11+11 = 88 bits. */
+ )
{
- Word32 i, j, k;
- Word16 st, ix, iy, pos, index, track, nb_pulse, nbiter, j_temp;
- Word16 psk, ps, alpk, alp, val, k_cn, k_dn, exp;
- Word16 *p0, *p1, *p2, *p3, *psign;
- Word16 *h, *h_inv, *ptr_h1, *ptr_h2, *ptr_hf, h_shift;
- Word32 s, cor, L_tmp, L_index;
- Word16 dn2[L_SUBFR], sign[L_SUBFR], vec[L_SUBFR];
- Word16 ind[NPMAXPT * NB_TRACK];
- Word16 codvec[NB_PULSE_MAX], nbpos[10];
- Word16 cor_x[NB_POS], cor_y[NB_POS], pos_max[NB_TRACK];
- Word16 h_buf[4 * L_SUBFR];
- Word16 rrixix[NB_TRACK][NB_POS], rrixiy[NB_TRACK][MSIZE];
- Word16 ipos[NB_PULSE_MAX];
+ Word32 i, j, k;
+ Word16 st, ix, iy, pos, index, track, nb_pulse, nbiter, j_temp;
+ Word16 psk, ps, alpk, alp, val, k_cn, k_dn, exp;
+ Word16 *p0, *p1, *p2, *p3, *psign;
+ Word16 *h, *h_inv, *ptr_h1, *ptr_h2, *ptr_hf, h_shift;
+ Word32 s, cor, L_tmp, L_index;
+ Word16 dn2[L_SUBFR], sign[L_SUBFR], vec[L_SUBFR];
+ Word16 ind[NPMAXPT * NB_TRACK];
+ Word16 codvec[NB_PULSE_MAX], nbpos[10];
+ Word16 cor_x[NB_POS], cor_y[NB_POS], pos_max[NB_TRACK];
+ Word16 h_buf[4 * L_SUBFR];
+ Word16 rrixix[NB_TRACK][NB_POS], rrixiy[NB_TRACK][MSIZE];
+ Word16 ipos[NB_PULSE_MAX];
- switch (nbbits)
- {
- case 20: /* 20 bits, 4 pulses, 4 tracks */
- nbiter = 4; /* 4x16x16=1024 loop */
- alp = 8192; /* alp = 2.0 (Q12) */
- nb_pulse = 4;
- nbpos[0] = 4;
- nbpos[1] = 8;
- break;
- case 36: /* 36 bits, 8 pulses, 4 tracks */
- nbiter = 4; /* 4x20x16=1280 loop */
- alp = 4096; /* alp = 1.0 (Q12) */
- nb_pulse = 8;
- nbpos[0] = 4;
- nbpos[1] = 8;
- nbpos[2] = 8;
- break;
- case 44: /* 44 bits, 10 pulses, 4 tracks */
- nbiter = 4; /* 4x26x16=1664 loop */
- alp = 4096; /* alp = 1.0 (Q12) */
- nb_pulse = 10;
- nbpos[0] = 4;
- nbpos[1] = 6;
- nbpos[2] = 8;
- nbpos[3] = 8;
- break;
- case 52: /* 52 bits, 12 pulses, 4 tracks */
- nbiter = 4; /* 4x26x16=1664 loop */
- alp = 4096; /* alp = 1.0 (Q12) */
- nb_pulse = 12;
- nbpos[0] = 4;
- nbpos[1] = 6;
- nbpos[2] = 8;
- nbpos[3] = 8;
- break;
- case 64: /* 64 bits, 16 pulses, 4 tracks */
- nbiter = 3; /* 3x36x16=1728 loop */
- alp = 3277; /* alp = 0.8 (Q12) */
- nb_pulse = 16;
- nbpos[0] = 4;
- nbpos[1] = 4;
- nbpos[2] = 6;
- nbpos[3] = 6;
- nbpos[4] = 8;
- nbpos[5] = 8;
- break;
- case 72: /* 72 bits, 18 pulses, 4 tracks */
- nbiter = 3; /* 3x35x16=1680 loop */
- alp = 3072; /* alp = 0.75 (Q12) */
- nb_pulse = 18;
- nbpos[0] = 2;
- nbpos[1] = 3;
- nbpos[2] = 4;
- nbpos[3] = 5;
- nbpos[4] = 6;
- nbpos[5] = 7;
- nbpos[6] = 8;
- break;
- case 88: /* 88 bits, 24 pulses, 4 tracks */
- if(ser_size > 462)
- nbiter = 1;
- else
- nbiter = 2; /* 2x53x16=1696 loop */
+ switch (nbbits)
+ {
+ case 20: /* 20 bits, 4 pulses, 4 tracks */
+ nbiter = 4; /* 4x16x16=1024 loop */
+ alp = 8192; /* alp = 2.0 (Q12) */
+ nb_pulse = 4;
+ nbpos[0] = 4;
+ nbpos[1] = 8;
+ break;
+ case 36: /* 36 bits, 8 pulses, 4 tracks */
+ nbiter = 4; /* 4x20x16=1280 loop */
+ alp = 4096; /* alp = 1.0 (Q12) */
+ nb_pulse = 8;
+ nbpos[0] = 4;
+ nbpos[1] = 8;
+ nbpos[2] = 8;
+ break;
+ case 44: /* 44 bits, 10 pulses, 4 tracks */
+ nbiter = 4; /* 4x26x16=1664 loop */
+ alp = 4096; /* alp = 1.0 (Q12) */
+ nb_pulse = 10;
+ nbpos[0] = 4;
+ nbpos[1] = 6;
+ nbpos[2] = 8;
+ nbpos[3] = 8;
+ break;
+ case 52: /* 52 bits, 12 pulses, 4 tracks */
+ nbiter = 4; /* 4x26x16=1664 loop */
+ alp = 4096; /* alp = 1.0 (Q12) */
+ nb_pulse = 12;
+ nbpos[0] = 4;
+ nbpos[1] = 6;
+ nbpos[2] = 8;
+ nbpos[3] = 8;
+ break;
+ case 64: /* 64 bits, 16 pulses, 4 tracks */
+ nbiter = 3; /* 3x36x16=1728 loop */
+ alp = 3277; /* alp = 0.8 (Q12) */
+ nb_pulse = 16;
+ nbpos[0] = 4;
+ nbpos[1] = 4;
+ nbpos[2] = 6;
+ nbpos[3] = 6;
+ nbpos[4] = 8;
+ nbpos[5] = 8;
+ break;
+ case 72: /* 72 bits, 18 pulses, 4 tracks */
+ nbiter = 3; /* 3x35x16=1680 loop */
+ alp = 3072; /* alp = 0.75 (Q12) */
+ nb_pulse = 18;
+ nbpos[0] = 2;
+ nbpos[1] = 3;
+ nbpos[2] = 4;
+ nbpos[3] = 5;
+ nbpos[4] = 6;
+ nbpos[5] = 7;
+ nbpos[6] = 8;
+ break;
+ case 88: /* 88 bits, 24 pulses, 4 tracks */
+ if(ser_size > 462)
+ nbiter = 1;
+ else
+ nbiter = 2; /* 2x53x16=1696 loop */
- alp = 2048; /* alp = 0.5 (Q12) */
- nb_pulse = 24;
- nbpos[0] = 2;
- nbpos[1] = 2;
- nbpos[2] = 3;
- nbpos[3] = 4;
- nbpos[4] = 5;
- nbpos[5] = 6;
- nbpos[6] = 7;
- nbpos[7] = 8;
- nbpos[8] = 8;
- nbpos[9] = 8;
- break;
- default:
- nbiter = 0;
- alp = 0;
- nb_pulse = 0;
- }
+ alp = 2048; /* alp = 0.5 (Q12) */
+ nb_pulse = 24;
+ nbpos[0] = 2;
+ nbpos[1] = 2;
+ nbpos[2] = 3;
+ nbpos[3] = 4;
+ nbpos[4] = 5;
+ nbpos[5] = 6;
+ nbpos[6] = 7;
+ nbpos[7] = 8;
+ nbpos[8] = 8;
+ nbpos[9] = 8;
+ break;
+ default:
+ nbiter = 0;
+ alp = 0;
+ nb_pulse = 0;
+ }
- for (i = 0; i < nb_pulse; i++)
- {
- codvec[i] = i;
- }
+ for (i = 0; i < nb_pulse; i++)
+ {
+ codvec[i] = i;
+ }
- /*----------------------------------------------------------------*
- * Find sign for each pulse position. *
- *----------------------------------------------------------------*/
- /* calculate energy for normalization of cn[] and dn[] */
- /* set k_cn = 32..32767 (ener_cn = 2^30..256-0) */
+ /*----------------------------------------------------------------*
+ * Find sign for each pulse position. *
+ *----------------------------------------------------------------*/
+ /* calculate energy for normalization of cn[] and dn[] */
+ /* set k_cn = 32..32767 (ener_cn = 2^30..256-0) */
#ifdef ASM_OPT /* asm optimization branch */
- s = Dot_product12_asm(cn, cn, L_SUBFR, &exp);
+ s = Dot_product12_asm(cn, cn, L_SUBFR, &exp);
#else
- s = Dot_product12(cn, cn, L_SUBFR, &exp);
+ s = Dot_product12(cn, cn, L_SUBFR, &exp);
#endif
- Isqrt_n(&s, &exp);
- s = L_shl(s, (exp + 5));
- k_cn = extract_h(L_add(s, 0x8000));
+ Isqrt_n(&s, &exp);
+ s = L_shl(s, (exp + 5));
+ k_cn = extract_h(L_add(s, 0x8000));
- /* set k_dn = 32..512 (ener_dn = 2^30..2^22) */
+ /* set k_dn = 32..512 (ener_dn = 2^30..2^22) */
#ifdef ASM_OPT /* asm optimization branch */
- s = Dot_product12_asm(dn, dn, L_SUBFR, &exp);
+ s = Dot_product12_asm(dn, dn, L_SUBFR, &exp);
#else
- s = Dot_product12(dn, dn, L_SUBFR, &exp);
+ s = Dot_product12(dn, dn, L_SUBFR, &exp);
#endif
- Isqrt_n(&s, &exp);
- k_dn = (L_shl(s, (exp + 5 + 3)) + 0x8000) >> 16; /* k_dn = 256..4096 */
- k_dn = vo_mult_r(alp, k_dn); /* alp in Q12 */
+ Isqrt_n(&s, &exp);
+ k_dn = (L_shl(s, (exp + 5 + 3)) + 0x8000) >> 16; /* k_dn = 256..4096 */
+ k_dn = vo_mult_r(alp, k_dn); /* alp in Q12 */
- /* mix normalized cn[] and dn[] */
- p0 = cn;
- p1 = dn;
- p2 = dn2;
+ /* mix normalized cn[] and dn[] */
+ p0 = cn;
+ p1 = dn;
+ p2 = dn2;
- for (i = 0; i < L_SUBFR/4; i++)
- {
- s = (k_cn* (*p0++))+(k_dn * (*p1++));
- *p2++ = s >> 7;
- s = (k_cn* (*p0++))+(k_dn * (*p1++));
- *p2++ = s >> 7;
- s = (k_cn* (*p0++))+(k_dn * (*p1++));
- *p2++ = s >> 7;
- s = (k_cn* (*p0++))+(k_dn * (*p1++));
- *p2++ = s >> 7;
- }
+ for (i = 0; i < L_SUBFR/4; i++)
+ {
+ s = (k_cn* (*p0++))+(k_dn * (*p1++));
+ *p2++ = s >> 7;
+ s = (k_cn* (*p0++))+(k_dn * (*p1++));
+ *p2++ = s >> 7;
+ s = (k_cn* (*p0++))+(k_dn * (*p1++));
+ *p2++ = s >> 7;
+ s = (k_cn* (*p0++))+(k_dn * (*p1++));
+ *p2++ = s >> 7;
+ }
- /* set sign according to dn2[] = k_cn*cn[] + k_dn*dn[] */
- for(i = 0; i < L_SUBFR; i++)
- {
- val = dn[i];
- ps = dn2[i];
- if (ps >= 0)
- {
- sign[i] = 32767; /* sign = +1 (Q12) */
- vec[i] = -32768;
- } else
- {
- sign[i] = -32768; /* sign = -1 (Q12) */
- vec[i] = 32767;
- dn[i] = -val;
- dn2[i] = -ps;
- }
- }
- /*----------------------------------------------------------------*
- * Select NB_MAX position per track according to max of dn2[]. *
- *----------------------------------------------------------------*/
- pos = 0;
- for (i = 0; i < NB_TRACK; i++)
- {
- for (k = 0; k < NB_MAX; k++)
- {
- ps = -1;
- for (j = i; j < L_SUBFR; j += STEP)
- {
- if(dn2[j] > ps)
- {
- ps = dn2[j];
- pos = j;
- }
- }
- dn2[pos] = (k - NB_MAX); /* dn2 < 0 when position is selected */
- if (k == 0)
- {
- pos_max[i] = pos;
- }
- }
- }
+ /* set sign according to dn2[] = k_cn*cn[] + k_dn*dn[] */
+ for(i = 0; i < L_SUBFR; i++)
+ {
+ val = dn[i];
+ ps = dn2[i];
+ if (ps >= 0)
+ {
+ sign[i] = 32767; /* sign = +1 (Q12) */
+ vec[i] = -32768;
+ } else
+ {
+ sign[i] = -32768; /* sign = -1 (Q12) */
+ vec[i] = 32767;
+ dn[i] = -val;
+ dn2[i] = -ps;
+ }
+ }
+ /*----------------------------------------------------------------*
+ * Select NB_MAX position per track according to max of dn2[]. *
+ *----------------------------------------------------------------*/
+ pos = 0;
+ for (i = 0; i < NB_TRACK; i++)
+ {
+ for (k = 0; k < NB_MAX; k++)
+ {
+ ps = -1;
+ for (j = i; j < L_SUBFR; j += STEP)
+ {
+ if(dn2[j] > ps)
+ {
+ ps = dn2[j];
+ pos = j;
+ }
+ }
+ dn2[pos] = (k - NB_MAX); /* dn2 < 0 when position is selected */
+ if (k == 0)
+ {
+ pos_max[i] = pos;
+ }
+ }
+ }
- /*--------------------------------------------------------------*
- * Scale h[] to avoid overflow and to get maximum of precision *
- * on correlation. *
- * *
- * Maximum of h[] (h[0]) is fixed to 2048 (MAX16 / 16). *
- * ==> This allow addition of 16 pulses without saturation. *
- * *
- * Energy worst case (on resonant impulse response), *
- * - energy of h[] is approximately MAX/16. *
- * - During search, the energy is divided by 8 to avoid *
- * overflow on "alp". (energy of h[] = MAX/128). *
- * ==> "alp" worst case detected is 22854 on sinusoidal wave. *
- *--------------------------------------------------------------*/
+ /*--------------------------------------------------------------*
+ * Scale h[] to avoid overflow and to get maximum of precision *
+ * on correlation. *
+ * *
+ * Maximum of h[] (h[0]) is fixed to 2048 (MAX16 / 16). *
+ * ==> This allow addition of 16 pulses without saturation. *
+ * *
+ * Energy worst case (on resonant impulse response), *
+ * - energy of h[] is approximately MAX/16. *
+ * - During search, the energy is divided by 8 to avoid *
+ * overflow on "alp". (energy of h[] = MAX/128). *
+ * ==> "alp" worst case detected is 22854 on sinusoidal wave. *
+ *--------------------------------------------------------------*/
- /* impulse response buffer for fast computation */
+ /* impulse response buffer for fast computation */
- h = h_buf;
- h_inv = h_buf + (2 * L_SUBFR);
- L_tmp = 0;
- for (i = 0; i < L_SUBFR; i++)
- {
- *h++ = 0;
- *h_inv++ = 0;
- L_tmp += (H[i] * H[i]) << 1;
- }
- /* scale h[] down (/2) when energy of h[] is high with many pulses used */
- val = extract_h(L_tmp);
- h_shift = 0;
+ h = h_buf;
+ h_inv = h_buf + (2 * L_SUBFR);
+ L_tmp = 0;
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ *h++ = 0;
+ *h_inv++ = 0;
+ L_tmp += (H[i] * H[i]) << 1;
+ }
+ /* scale h[] down (/2) when energy of h[] is high with many pulses used */
+ val = extract_h(L_tmp);
+ h_shift = 0;
- if ((nb_pulse >= 12) && (val > 1024))
- {
- h_shift = 1;
- }
- p0 = H;
- p1 = h;
- p2 = h_inv;
+ if ((nb_pulse >= 12) && (val > 1024))
+ {
+ h_shift = 1;
+ }
+ p0 = H;
+ p1 = h;
+ p2 = h_inv;
- for (i = 0; i < L_SUBFR/4; i++)
- {
- *p1 = *p0++ >> h_shift;
- *p2++ = -(*p1++);
- *p1 = *p0++ >> h_shift;
- *p2++ = -(*p1++);
- *p1 = *p0++ >> h_shift;
- *p2++ = -(*p1++);
- *p1 = *p0++ >> h_shift;
- *p2++ = -(*p1++);
- }
+ for (i = 0; i < L_SUBFR/4; i++)
+ {
+ *p1 = *p0++ >> h_shift;
+ *p2++ = -(*p1++);
+ *p1 = *p0++ >> h_shift;
+ *p2++ = -(*p1++);
+ *p1 = *p0++ >> h_shift;
+ *p2++ = -(*p1++);
+ *p1 = *p0++ >> h_shift;
+ *p2++ = -(*p1++);
+ }
- /*------------------------------------------------------------*
- * Compute rrixix[][] needed for the codebook search. *
- * This algorithm compute impulse response energy of all *
- * positions (16) in each track (4). Total = 4x16 = 64. *
- *------------------------------------------------------------*/
+ /*------------------------------------------------------------*
+ * Compute rrixix[][] needed for the codebook search. *
+ * This algorithm compute impulse response energy of all *
+ * positions (16) in each track (4). Total = 4x16 = 64. *
+ *------------------------------------------------------------*/
- /* storage order --> i3i3, i2i2, i1i1, i0i0 */
+ /* storage order --> i3i3, i2i2, i1i1, i0i0 */
- /* Init pointers to last position of rrixix[] */
- p0 = &rrixix[0][NB_POS - 1];
- p1 = &rrixix[1][NB_POS - 1];
- p2 = &rrixix[2][NB_POS - 1];
- p3 = &rrixix[3][NB_POS - 1];
+ /* Init pointers to last position of rrixix[] */
+ p0 = &rrixix[0][NB_POS - 1];
+ p1 = &rrixix[1][NB_POS - 1];
+ p2 = &rrixix[2][NB_POS - 1];
+ p3 = &rrixix[3][NB_POS - 1];
- ptr_h1 = h;
- cor = 0x00008000L; /* for rounding */
- for (i = 0; i < NB_POS; i++)
- {
- cor += vo_L_mult((*ptr_h1), (*ptr_h1));
- ptr_h1++;
- *p3-- = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h1));
- ptr_h1++;
- *p2-- = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h1));
- ptr_h1++;
- *p1-- = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h1));
- ptr_h1++;
- *p0-- = extract_h(cor);
- }
+ ptr_h1 = h;
+ cor = 0x00008000L; /* for rounding */
+ for (i = 0; i < NB_POS; i++)
+ {
+ cor += vo_L_mult((*ptr_h1), (*ptr_h1));
+ ptr_h1++;
+ *p3-- = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h1));
+ ptr_h1++;
+ *p2-- = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h1));
+ ptr_h1++;
+ *p1-- = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h1));
+ ptr_h1++;
+ *p0-- = extract_h(cor);
+ }
- /*------------------------------------------------------------*
- * Compute rrixiy[][] needed for the codebook search. *
- * This algorithm compute correlation between 2 pulses *
- * (2 impulses responses) in 4 possible adjacents tracks. *
- * (track 0-1, 1-2, 2-3 and 3-0). Total = 4x16x16 = 1024. *
- *------------------------------------------------------------*/
+ /*------------------------------------------------------------*
+ * Compute rrixiy[][] needed for the codebook search. *
+ * This algorithm compute correlation between 2 pulses *
+ * (2 impulses responses) in 4 possible adjacents tracks. *
+ * (track 0-1, 1-2, 2-3 and 3-0). Total = 4x16x16 = 1024. *
+ *------------------------------------------------------------*/
- /* storage order --> i2i3, i1i2, i0i1, i3i0 */
+ /* storage order --> i2i3, i1i2, i0i1, i3i0 */
- pos = MSIZE - 1;
- ptr_hf = h + 1;
+ pos = MSIZE - 1;
+ ptr_hf = h + 1;
- for (k = 0; k < NB_POS; k++)
- {
- p3 = &rrixiy[2][pos];
- p2 = &rrixiy[1][pos];
- p1 = &rrixiy[0][pos];
- p0 = &rrixiy[3][pos - NB_POS];
+ for (k = 0; k < NB_POS; k++)
+ {
+ p3 = &rrixiy[2][pos];
+ p2 = &rrixiy[1][pos];
+ p1 = &rrixiy[0][pos];
+ p0 = &rrixiy[3][pos - NB_POS];
- cor = 0x00008000L; /* for rounding */
- ptr_h1 = h;
- ptr_h2 = ptr_hf;
+ cor = 0x00008000L; /* for rounding */
+ ptr_h1 = h;
+ ptr_h2 = ptr_hf;
- for (i = k + 1; i < NB_POS; i++)
- {
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p3 = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p2 = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p1 = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p0 = extract_h(cor);
+ for (i = k + 1; i < NB_POS; i++)
+ {
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p3 = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p2 = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p1 = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p0 = extract_h(cor);
- p3 -= (NB_POS + 1);
- p2 -= (NB_POS + 1);
- p1 -= (NB_POS + 1);
- p0 -= (NB_POS + 1);
- }
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p3 = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p2 = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p1 = extract_h(cor);
+ p3 -= (NB_POS + 1);
+ p2 -= (NB_POS + 1);
+ p1 -= (NB_POS + 1);
+ p0 -= (NB_POS + 1);
+ }
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p3 = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p2 = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p1 = extract_h(cor);
- pos -= NB_POS;
- ptr_hf += STEP;
- }
+ pos -= NB_POS;
+ ptr_hf += STEP;
+ }
- /* storage order --> i3i0, i2i3, i1i2, i0i1 */
+ /* storage order --> i3i0, i2i3, i1i2, i0i1 */
- pos = MSIZE - 1;
- ptr_hf = h + 3;
+ pos = MSIZE - 1;
+ ptr_hf = h + 3;
- for (k = 0; k < NB_POS; k++)
- {
- p3 = &rrixiy[3][pos];
- p2 = &rrixiy[2][pos - 1];
- p1 = &rrixiy[1][pos - 1];
- p0 = &rrixiy[0][pos - 1];
+ for (k = 0; k < NB_POS; k++)
+ {
+ p3 = &rrixiy[3][pos];
+ p2 = &rrixiy[2][pos - 1];
+ p1 = &rrixiy[1][pos - 1];
+ p0 = &rrixiy[0][pos - 1];
- cor = 0x00008000L; /* for rounding */
- ptr_h1 = h;
- ptr_h2 = ptr_hf;
+ cor = 0x00008000L; /* for rounding */
+ ptr_h1 = h;
+ ptr_h2 = ptr_hf;
- for (i = k + 1; i < NB_POS; i++)
- {
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p3 = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p2 = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p1 = extract_h(cor);
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p0 = extract_h(cor);
+ for (i = k + 1; i < NB_POS; i++)
+ {
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p3 = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p2 = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p1 = extract_h(cor);
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p0 = extract_h(cor);
- p3 -= (NB_POS + 1);
- p2 -= (NB_POS + 1);
- p1 -= (NB_POS + 1);
- p0 -= (NB_POS + 1);
- }
- cor += vo_L_mult((*ptr_h1), (*ptr_h2));
- ptr_h1++;
- ptr_h2++;
- *p3 = extract_h(cor);
+ p3 -= (NB_POS + 1);
+ p2 -= (NB_POS + 1);
+ p1 -= (NB_POS + 1);
+ p0 -= (NB_POS + 1);
+ }
+ cor += vo_L_mult((*ptr_h1), (*ptr_h2));
+ ptr_h1++;
+ ptr_h2++;
+ *p3 = extract_h(cor);
- pos--;
- ptr_hf += STEP;
- }
+ pos--;
+ ptr_hf += STEP;
+ }
- /*------------------------------------------------------------*
- * Modification of rrixiy[][] to take signs into account. *
- *------------------------------------------------------------*/
+ /*------------------------------------------------------------*
+ * Modification of rrixiy[][] to take signs into account. *
+ *------------------------------------------------------------*/
- p0 = &rrixiy[0][0];
+ p0 = &rrixiy[0][0];
- for (k = 0; k < NB_TRACK; k++)
- {
- j_temp = (k + 1)&0x03;
- for (i = k; i < L_SUBFR; i += STEP)
- {
- psign = sign;
- if (psign[i] < 0)
- {
- psign = vec;
- }
- j = j_temp;
- for (; j < L_SUBFR; j += STEP)
- {
- *p0 = vo_mult(*p0, psign[j]);
- p0++;
- }
- }
- }
+ for (k = 0; k < NB_TRACK; k++)
+ {
+ j_temp = (k + 1)&0x03;
+ for (i = k; i < L_SUBFR; i += STEP)
+ {
+ psign = sign;
+ if (psign[i] < 0)
+ {
+ psign = vec;
+ }
+ j = j_temp;
+ for (; j < L_SUBFR; j += STEP)
+ {
+ *p0 = vo_mult(*p0, psign[j]);
+ p0++;
+ }
+ }
+ }
- /*-------------------------------------------------------------------*
- * Deep first search *
- *-------------------------------------------------------------------*/
+ /*-------------------------------------------------------------------*
+ * Deep first search *
+ *-------------------------------------------------------------------*/
- psk = -1;
- alpk = 1;
+ psk = -1;
+ alpk = 1;
- for (k = 0; k < nbiter; k++)
- {
- j_temp = k<<2;
- for (i = 0; i < nb_pulse; i++)
- ipos[i] = tipos[j_temp + i];
+ for (k = 0; k < nbiter; k++)
+ {
+ j_temp = k<<2;
+ for (i = 0; i < nb_pulse; i++)
+ ipos[i] = tipos[j_temp + i];
- if(nbbits == 20)
- {
- pos = 0;
- ps = 0;
- alp = 0;
- for (i = 0; i < L_SUBFR; i++)
- {
- vec[i] = 0;
- }
- } else if ((nbbits == 36) || (nbbits == 44))
- {
- /* first stage: fix 2 pulses */
- pos = 2;
+ if(nbbits == 20)
+ {
+ pos = 0;
+ ps = 0;
+ alp = 0;
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ vec[i] = 0;
+ }
+ } else if ((nbbits == 36) || (nbbits == 44))
+ {
+ /* first stage: fix 2 pulses */
+ pos = 2;
- ix = ind[0] = pos_max[ipos[0]];
- iy = ind[1] = pos_max[ipos[1]];
- ps = dn[ix] + dn[iy];
- i = ix >> 2; /* ix / STEP */
- j = iy >> 2; /* iy / STEP */
- s = rrixix[ipos[0]][i] << 13;
- s += rrixix[ipos[1]][j] << 13;
- i = (i << 4) + j; /* (ix/STEP)*NB_POS + (iy/STEP) */
- s += rrixiy[ipos[0]][i] << 14;
- alp = (s + 0x8000) >> 16;
- if (sign[ix] < 0)
- p0 = h_inv - ix;
- else
- p0 = h - ix;
- if (sign[iy] < 0)
- p1 = h_inv - iy;
- else
- p1 = h - iy;
+ ix = ind[0] = pos_max[ipos[0]];
+ iy = ind[1] = pos_max[ipos[1]];
+ ps = dn[ix] + dn[iy];
+ i = ix >> 2; /* ix / STEP */
+ j = iy >> 2; /* iy / STEP */
+ s = rrixix[ipos[0]][i] << 13;
+ s += rrixix[ipos[1]][j] << 13;
+ i = (i << 4) + j; /* (ix/STEP)*NB_POS + (iy/STEP) */
+ s += rrixiy[ipos[0]][i] << 14;
+ alp = (s + 0x8000) >> 16;
+ if (sign[ix] < 0)
+ p0 = h_inv - ix;
+ else
+ p0 = h - ix;
+ if (sign[iy] < 0)
+ p1 = h_inv - iy;
+ else
+ p1 = h - iy;
- for (i = 0; i < L_SUBFR; i++)
- {
- vec[i] = (*p0++) + (*p1++);
- }
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ vec[i] = (*p0++) + (*p1++);
+ }
- if(nbbits == 44)
- {
- ipos[8] = 0;
- ipos[9] = 1;
- }
- } else
- {
- /* first stage: fix 4 pulses */
- pos = 4;
+ if(nbbits == 44)
+ {
+ ipos[8] = 0;
+ ipos[9] = 1;
+ }
+ } else
+ {
+ /* first stage: fix 4 pulses */
+ pos = 4;
- ix = ind[0] = pos_max[ipos[0]];
- iy = ind[1] = pos_max[ipos[1]];
- i = ind[2] = pos_max[ipos[2]];
- j = ind[3] = pos_max[ipos[3]];
- ps = add1(add1(add1(dn[ix], dn[iy]), dn[i]), dn[j]);
+ ix = ind[0] = pos_max[ipos[0]];
+ iy = ind[1] = pos_max[ipos[1]];
+ i = ind[2] = pos_max[ipos[2]];
+ j = ind[3] = pos_max[ipos[3]];
+ ps = add1(add1(add1(dn[ix], dn[iy]), dn[i]), dn[j]);
- if (sign[ix] < 0)
- p0 = h_inv - ix;
- else
- p0 = h - ix;
+ if (sign[ix] < 0)
+ p0 = h_inv - ix;
+ else
+ p0 = h - ix;
- if (sign[iy] < 0)
- p1 = h_inv - iy;
- else
- p1 = h - iy;
+ if (sign[iy] < 0)
+ p1 = h_inv - iy;
+ else
+ p1 = h - iy;
- if (sign[i] < 0)
- p2 = h_inv - i;
- else
- p2 = h - i;
+ if (sign[i] < 0)
+ p2 = h_inv - i;
+ else
+ p2 = h - i;
- if (sign[j] < 0)
- p3 = h_inv - j;
- else
- p3 = h - j;
+ if (sign[j] < 0)
+ p3 = h_inv - j;
+ else
+ p3 = h - j;
- L_tmp = 0L;
- for(i = 0; i < L_SUBFR; i++)
- {
- vec[i] = add1(add1(add1(*p0++, *p1++), *p2++), *p3++);
- L_tmp += (vec[i] * vec[i]) << 1;
- }
+ L_tmp = 0L;
+ for(i = 0; i < L_SUBFR; i++)
+ {
+ Word32 vecSq2;
+ vec[i] = add1(add1(add1(*p0++, *p1++), *p2++), *p3++);
+ vecSq2 = (vec[i] * vec[i]) << 1;
+ if (vecSq2 > 0 && L_tmp > INT_MAX - vecSq2) {
+ L_tmp = INT_MAX;
+ } else if (vecSq2 < 0 && L_tmp < INT_MIN - vecSq2) {
+ L_tmp = INT_MIN;
+ } else {
+ L_tmp += vecSq2;
+ }
+ }
- alp = ((L_tmp >> 3) + 0x8000) >> 16;
+ alp = ((L_tmp >> 3) + 0x8000) >> 16;
- if(nbbits == 72)
- {
- ipos[16] = 0;
- ipos[17] = 1;
- }
- }
+ if(nbbits == 72)
+ {
+ ipos[16] = 0;
+ ipos[17] = 1;
+ }
+ }
- /* other stages of 2 pulses */
+ /* other stages of 2 pulses */
- for (j = pos, st = 0; j < nb_pulse; j += 2, st++)
- {
- /*--------------------------------------------------*
- * Calculate correlation of all possible positions *
- * of the next 2 pulses with previous fixed pulses. *
- * Each pulse can have 16 possible positions. *
- *--------------------------------------------------*/
- if(ipos[j] == 3)
- {
- cor_h_vec_30(h, vec, ipos[j], sign, rrixix, cor_x, cor_y);
- }
- else
- {
+ for (j = pos, st = 0; j < nb_pulse; j += 2, st++)
+ {
+ /*--------------------------------------------------*
+ * Calculate correlation of all possible positions *
+ * of the next 2 pulses with previous fixed pulses. *
+ * Each pulse can have 16 possible positions. *
+ *--------------------------------------------------*/
+ if(ipos[j] == 3)
+ {
+ cor_h_vec_30(h, vec, ipos[j], sign, rrixix, cor_x, cor_y);
+ }
+ else
+ {
#ifdef ASM_OPT /* asm optimization branch */
- cor_h_vec_012_asm(h, vec, ipos[j], sign, rrixix, cor_x, cor_y);
+ cor_h_vec_012_asm(h, vec, ipos[j], sign, rrixix, cor_x, cor_y);
#else
- cor_h_vec_012(h, vec, ipos[j], sign, rrixix, cor_x, cor_y);
+ cor_h_vec_012(h, vec, ipos[j], sign, rrixix, cor_x, cor_y);
#endif
- }
- /*--------------------------------------------------*
- * Find best positions of 2 pulses. *
- *--------------------------------------------------*/
- search_ixiy(nbpos[st], ipos[j], ipos[j + 1], &ps, &alp,
- &ix, &iy, dn, dn2, cor_x, cor_y, rrixiy);
+ }
+ /*--------------------------------------------------*
+ * Find best positions of 2 pulses. *
+ *--------------------------------------------------*/
+ search_ixiy(nbpos[st], ipos[j], ipos[j + 1], &ps, &alp,
+ &ix, &iy, dn, dn2, cor_x, cor_y, rrixiy);
- ind[j] = ix;
- ind[j + 1] = iy;
+ ind[j] = ix;
+ ind[j + 1] = iy;
- if (sign[ix] < 0)
- p0 = h_inv - ix;
- else
- p0 = h - ix;
- if (sign[iy] < 0)
- p1 = h_inv - iy;
- else
- p1 = h - iy;
+ if (sign[ix] < 0)
+ p0 = h_inv - ix;
+ else
+ p0 = h - ix;
+ if (sign[iy] < 0)
+ p1 = h_inv - iy;
+ else
+ p1 = h - iy;
- for (i = 0; i < L_SUBFR; i+=4)
- {
- vec[i] += add1((*p0++), (*p1++));
- vec[i+1] += add1((*p0++), (*p1++));
- vec[i+2] += add1((*p0++), (*p1++));
- vec[i+3] += add1((*p0++), (*p1++));
- }
- }
- /* memorise the best codevector */
- ps = vo_mult(ps, ps);
- s = vo_L_msu(vo_L_mult(alpk, ps), psk, alp);
- if (s > 0)
- {
- psk = ps;
- alpk = alp;
- for (i = 0; i < nb_pulse; i++)
- {
- codvec[i] = ind[i];
- }
- for (i = 0; i < L_SUBFR; i++)
- {
- y[i] = vec[i];
- }
- }
- }
- /*-------------------------------------------------------------------*
- * Build the codeword, the filtered codeword and index of codevector.*
- *-------------------------------------------------------------------*/
- for (i = 0; i < NPMAXPT * NB_TRACK; i++)
- {
- ind[i] = -1;
- }
- for (i = 0; i < L_SUBFR; i++)
- {
- code[i] = 0;
- y[i] = vo_shr_r(y[i], 3); /* Q12 to Q9 */
- }
- val = (512 >> h_shift); /* codeword in Q9 format */
- for (k = 0; k < nb_pulse; k++)
- {
- i = codvec[k]; /* read pulse position */
- j = sign[i]; /* read sign */
- index = i >> 2; /* index = pos of pulse (0..15) */
- track = (Word16) (i & 0x03); /* track = i % NB_TRACK (0..3) */
+ for (i = 0; i < L_SUBFR; i+=4)
+ {
+ vec[i] += add1((*p0++), (*p1++));
+ vec[i+1] += add1((*p0++), (*p1++));
+ vec[i+2] += add1((*p0++), (*p1++));
+ vec[i+3] += add1((*p0++), (*p1++));
+ }
+ }
+ /* memorise the best codevector */
+ ps = vo_mult(ps, ps);
+ s = vo_L_msu(vo_L_mult(alpk, ps), psk, alp);
+ if (s > 0)
+ {
+ psk = ps;
+ alpk = alp;
+ for (i = 0; i < nb_pulse; i++)
+ {
+ codvec[i] = ind[i];
+ }
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ y[i] = vec[i];
+ }
+ }
+ }
+ /*-------------------------------------------------------------------*
+ * Build the codeword, the filtered codeword and index of codevector.*
+ *-------------------------------------------------------------------*/
+ for (i = 0; i < NPMAXPT * NB_TRACK; i++)
+ {
+ ind[i] = -1;
+ }
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ code[i] = 0;
+ y[i] = vo_shr_r(y[i], 3); /* Q12 to Q9 */
+ }
+ val = (512 >> h_shift); /* codeword in Q9 format */
+ for (k = 0; k < nb_pulse; k++)
+ {
+ i = codvec[k]; /* read pulse position */
+ j = sign[i]; /* read sign */
+ index = i >> 2; /* index = pos of pulse (0..15) */
+ track = (Word16) (i & 0x03); /* track = i % NB_TRACK (0..3) */
- if (j > 0)
- {
- code[i] += val;
- codvec[k] += 128;
- } else
- {
- code[i] -= val;
- index += NB_POS;
- }
+ if (j > 0)
+ {
+ code[i] += val;
+ codvec[k] += 128;
+ } else
+ {
+ code[i] -= val;
+ index += NB_POS;
+ }
- i = (Word16)((vo_L_mult(track, NPMAXPT) >> 1));
+ i = (Word16)((vo_L_mult(track, NPMAXPT) >> 1));
- while (ind[i] >= 0)
- {
- i += 1;
- }
- ind[i] = index;
- }
+ while (ind[i] >= 0)
+ {
+ i += 1;
+ }
+ ind[i] = index;
+ }
- k = 0;
- /* Build index of codevector */
- if(nbbits == 20)
- {
- for (track = 0; track < NB_TRACK; track++)
- {
- _index[track] = (Word16)(quant_1p_N1(ind[k], 4));
- k += NPMAXPT;
- }
- } else if(nbbits == 36)
- {
- for (track = 0; track < NB_TRACK; track++)
- {
- _index[track] = (Word16)(quant_2p_2N1(ind[k], ind[k + 1], 4));
- k += NPMAXPT;
- }
- } else if(nbbits == 44)
- {
- for (track = 0; track < NB_TRACK - 2; track++)
- {
- _index[track] = (Word16)(quant_3p_3N1(ind[k], ind[k + 1], ind[k + 2], 4));
- k += NPMAXPT;
- }
- for (track = 2; track < NB_TRACK; track++)
- {
- _index[track] = (Word16)(quant_2p_2N1(ind[k], ind[k + 1], 4));
- k += NPMAXPT;
- }
- } else if(nbbits == 52)
- {
- for (track = 0; track < NB_TRACK; track++)
- {
- _index[track] = (Word16)(quant_3p_3N1(ind[k], ind[k + 1], ind[k + 2], 4));
- k += NPMAXPT;
- }
- } else if(nbbits == 64)
- {
- for (track = 0; track < NB_TRACK; track++)
- {
- L_index = quant_4p_4N(&ind[k], 4);
- _index[track] = (Word16)((L_index >> 14) & 3);
- _index[track + NB_TRACK] = (Word16)(L_index & 0x3FFF);
- k += NPMAXPT;
- }
- } else if(nbbits == 72)
- {
- for (track = 0; track < NB_TRACK - 2; track++)
- {
- L_index = quant_5p_5N(&ind[k], 4);
- _index[track] = (Word16)((L_index >> 10) & 0x03FF);
- _index[track + NB_TRACK] = (Word16)(L_index & 0x03FF);
- k += NPMAXPT;
- }
- for (track = 2; track < NB_TRACK; track++)
- {
- L_index = quant_4p_4N(&ind[k], 4);
- _index[track] = (Word16)((L_index >> 14) & 3);
- _index[track + NB_TRACK] = (Word16)(L_index & 0x3FFF);
- k += NPMAXPT;
- }
- } else if(nbbits == 88)
- {
- for (track = 0; track < NB_TRACK; track++)
- {
- L_index = quant_6p_6N_2(&ind[k], 4);
- _index[track] = (Word16)((L_index >> 11) & 0x07FF);
- _index[track + NB_TRACK] = (Word16)(L_index & 0x07FF);
- k += NPMAXPT;
- }
- }
- return;
+ k = 0;
+ /* Build index of codevector */
+ if(nbbits == 20)
+ {
+ for (track = 0; track < NB_TRACK; track++)
+ {
+ _index[track] = (Word16)(quant_1p_N1(ind[k], 4));
+ k += NPMAXPT;
+ }
+ } else if(nbbits == 36)
+ {
+ for (track = 0; track < NB_TRACK; track++)
+ {
+ _index[track] = (Word16)(quant_2p_2N1(ind[k], ind[k + 1], 4));
+ k += NPMAXPT;
+ }
+ } else if(nbbits == 44)
+ {
+ for (track = 0; track < NB_TRACK - 2; track++)
+ {
+ _index[track] = (Word16)(quant_3p_3N1(ind[k], ind[k + 1], ind[k + 2], 4));
+ k += NPMAXPT;
+ }
+ for (track = 2; track < NB_TRACK; track++)
+ {
+ _index[track] = (Word16)(quant_2p_2N1(ind[k], ind[k + 1], 4));
+ k += NPMAXPT;
+ }
+ } else if(nbbits == 52)
+ {
+ for (track = 0; track < NB_TRACK; track++)
+ {
+ _index[track] = (Word16)(quant_3p_3N1(ind[k], ind[k + 1], ind[k + 2], 4));
+ k += NPMAXPT;
+ }
+ } else if(nbbits == 64)
+ {
+ for (track = 0; track < NB_TRACK; track++)
+ {
+ L_index = quant_4p_4N(&ind[k], 4);
+ _index[track] = (Word16)((L_index >> 14) & 3);
+ _index[track + NB_TRACK] = (Word16)(L_index & 0x3FFF);
+ k += NPMAXPT;
+ }
+ } else if(nbbits == 72)
+ {
+ for (track = 0; track < NB_TRACK - 2; track++)
+ {
+ L_index = quant_5p_5N(&ind[k], 4);
+ _index[track] = (Word16)((L_index >> 10) & 0x03FF);
+ _index[track + NB_TRACK] = (Word16)(L_index & 0x03FF);
+ k += NPMAXPT;
+ }
+ for (track = 2; track < NB_TRACK; track++)
+ {
+ L_index = quant_4p_4N(&ind[k], 4);
+ _index[track] = (Word16)((L_index >> 14) & 3);
+ _index[track + NB_TRACK] = (Word16)(L_index & 0x3FFF);
+ k += NPMAXPT;
+ }
+ } else if(nbbits == 88)
+ {
+ for (track = 0; track < NB_TRACK; track++)
+ {
+ L_index = quant_6p_6N_2(&ind[k], 4);
+ _index[track] = (Word16)((L_index >> 11) & 0x07FF);
+ _index[track + NB_TRACK] = (Word16)(L_index & 0x07FF);
+ k += NPMAXPT;
+ }
+ }
+ return;
}
@@ -825,135 +832,135 @@
* Compute correlations of h[] with vec[] for the specified track. *
*-------------------------------------------------------------------*/
void cor_h_vec_30(
- Word16 h[], /* (i) scaled impulse response */
- Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
- Word16 track, /* (i) track to use */
- Word16 sign[], /* (i) sign vector */
- Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
- Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
- Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
- )
+ Word16 h[], /* (i) scaled impulse response */
+ Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
+ Word16 track, /* (i) track to use */
+ Word16 sign[], /* (i) sign vector */
+ Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
+ Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
+ Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
+ )
{
- Word32 i, j, pos, corr;
- Word16 *p0, *p1, *p2,*p3,*cor_x,*cor_y;
- Word32 L_sum1,L_sum2;
- cor_x = cor_1;
- cor_y = cor_2;
- p0 = rrixix[track];
- p3 = rrixix[0];
- pos = track;
+ Word32 i, j, pos, corr;
+ Word16 *p0, *p1, *p2,*p3,*cor_x,*cor_y;
+ Word32 L_sum1,L_sum2;
+ cor_x = cor_1;
+ cor_y = cor_2;
+ p0 = rrixix[track];
+ p3 = rrixix[0];
+ pos = track;
- for (i = 0; i < NB_POS; i+=2)
- {
- L_sum1 = L_sum2 = 0L;
- p1 = h;
- p2 = &vec[pos];
- for (j=pos;j < L_SUBFR; j++)
- {
- L_sum1 += *p1 * *p2;
- p2-=3;
- L_sum2 += *p1++ * *p2;
- p2+=4;
- }
- p2-=3;
- L_sum2 += *p1++ * *p2++;
- L_sum2 += *p1++ * *p2++;
- L_sum2 += *p1++ * *p2++;
+ for (i = 0; i < NB_POS; i+=2)
+ {
+ L_sum1 = L_sum2 = 0L;
+ p1 = h;
+ p2 = &vec[pos];
+ for (j=pos;j < L_SUBFR; j++)
+ {
+ L_sum1 += *p1 * *p2;
+ p2-=3;
+ L_sum2 += *p1++ * *p2;
+ p2+=4;
+ }
+ p2-=3;
+ L_sum2 += *p1++ * *p2++;
+ L_sum2 += *p1++ * *p2++;
+ L_sum2 += *p1++ * *p2++;
- L_sum1 = (L_sum1 << 2);
- L_sum2 = (L_sum2 << 2);
+ L_sum1 = (L_sum1 << 2);
+ L_sum2 = (L_sum2 << 2);
- corr = vo_round(L_sum1);
- *cor_x++ = vo_mult(corr, sign[pos]) + (*p0++);
- corr = vo_round(L_sum2);
- *cor_y++ = vo_mult(corr, sign[pos-3]) + (*p3++);
- pos += STEP;
+ corr = vo_round(L_sum1);
+ *cor_x++ = vo_mult(corr, sign[pos]) + (*p0++);
+ corr = vo_round(L_sum2);
+ *cor_y++ = vo_mult(corr, sign[pos-3]) + (*p3++);
+ pos += STEP;
- L_sum1 = L_sum2 = 0L;
- p1 = h;
- p2 = &vec[pos];
- for (j=pos;j < L_SUBFR; j++)
- {
- L_sum1 += *p1 * *p2;
- p2-=3;
- L_sum2 += *p1++ * *p2;
- p2+=4;
- }
- p2-=3;
- L_sum2 += *p1++ * *p2++;
- L_sum2 += *p1++ * *p2++;
- L_sum2 += *p1++ * *p2++;
+ L_sum1 = L_sum2 = 0L;
+ p1 = h;
+ p2 = &vec[pos];
+ for (j=pos;j < L_SUBFR; j++)
+ {
+ L_sum1 += *p1 * *p2;
+ p2-=3;
+ L_sum2 += *p1++ * *p2;
+ p2+=4;
+ }
+ p2-=3;
+ L_sum2 += *p1++ * *p2++;
+ L_sum2 += *p1++ * *p2++;
+ L_sum2 += *p1++ * *p2++;
- L_sum1 = (L_sum1 << 2);
- L_sum2 = (L_sum2 << 2);
+ L_sum1 = (L_sum1 << 2);
+ L_sum2 = (L_sum2 << 2);
- corr = vo_round(L_sum1);
- *cor_x++ = vo_mult(corr, sign[pos]) + (*p0++);
- corr = vo_round(L_sum2);
- *cor_y++ = vo_mult(corr, sign[pos-3]) + (*p3++);
- pos += STEP;
- }
- return;
+ corr = vo_round(L_sum1);
+ *cor_x++ = vo_mult(corr, sign[pos]) + (*p0++);
+ corr = vo_round(L_sum2);
+ *cor_y++ = vo_mult(corr, sign[pos-3]) + (*p3++);
+ pos += STEP;
+ }
+ return;
}
void cor_h_vec_012(
- Word16 h[], /* (i) scaled impulse response */
- Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
- Word16 track, /* (i) track to use */
- Word16 sign[], /* (i) sign vector */
- Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
- Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
- Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
- )
+ Word16 h[], /* (i) scaled impulse response */
+ Word16 vec[], /* (i) scaled vector (/8) to correlate with h[] */
+ Word16 track, /* (i) track to use */
+ Word16 sign[], /* (i) sign vector */
+ Word16 rrixix[][NB_POS], /* (i) correlation of h[x] with h[x] */
+ Word16 cor_1[], /* (o) result of correlation (NB_POS elements) */
+ Word16 cor_2[] /* (o) result of correlation (NB_POS elements) */
+ )
{
- Word32 i, j, pos, corr;
- Word16 *p0, *p1, *p2,*p3,*cor_x,*cor_y;
- Word32 L_sum1,L_sum2;
- cor_x = cor_1;
- cor_y = cor_2;
- p0 = rrixix[track];
- p3 = rrixix[track+1];
- pos = track;
+ Word32 i, j, pos, corr;
+ Word16 *p0, *p1, *p2,*p3,*cor_x,*cor_y;
+ Word32 L_sum1,L_sum2;
+ cor_x = cor_1;
+ cor_y = cor_2;
+ p0 = rrixix[track];
+ p3 = rrixix[track+1];
+ pos = track;
- for (i = 0; i < NB_POS; i+=2)
- {
- L_sum1 = L_sum2 = 0L;
- p1 = h;
- p2 = &vec[pos];
- for (j=62-pos ;j >= 0; j--)
- {
- L_sum1 += *p1 * *p2++;
- L_sum2 += *p1++ * *p2;
- }
- L_sum1 += *p1 * *p2;
- L_sum1 = (L_sum1 << 2);
- L_sum2 = (L_sum2 << 2);
+ for (i = 0; i < NB_POS; i+=2)
+ {
+ L_sum1 = L_sum2 = 0L;
+ p1 = h;
+ p2 = &vec[pos];
+ for (j=62-pos ;j >= 0; j--)
+ {
+ L_sum1 += *p1 * *p2++;
+ L_sum2 += *p1++ * *p2;
+ }
+ L_sum1 += *p1 * *p2;
+ L_sum1 = (L_sum1 << 2);
+ L_sum2 = (L_sum2 << 2);
- corr = (L_sum1 + 0x8000) >> 16;
- cor_x[i] = vo_mult(corr, sign[pos]) + (*p0++);
- corr = (L_sum2 + 0x8000) >> 16;
- cor_y[i] = vo_mult(corr, sign[pos + 1]) + (*p3++);
- pos += STEP;
+ corr = (L_sum1 + 0x8000) >> 16;
+ cor_x[i] = vo_mult(corr, sign[pos]) + (*p0++);
+ corr = (L_sum2 + 0x8000) >> 16;
+ cor_y[i] = vo_mult(corr, sign[pos + 1]) + (*p3++);
+ pos += STEP;
- L_sum1 = L_sum2 = 0L;
- p1 = h;
- p2 = &vec[pos];
- for (j= 62-pos;j >= 0; j--)
- {
- L_sum1 += *p1 * *p2++;
- L_sum2 += *p1++ * *p2;
- }
- L_sum1 += *p1 * *p2;
- L_sum1 = (L_sum1 << 2);
- L_sum2 = (L_sum2 << 2);
+ L_sum1 = L_sum2 = 0L;
+ p1 = h;
+ p2 = &vec[pos];
+ for (j= 62-pos;j >= 0; j--)
+ {
+ L_sum1 += *p1 * *p2++;
+ L_sum2 += *p1++ * *p2;
+ }
+ L_sum1 += *p1 * *p2;
+ L_sum1 = (L_sum1 << 2);
+ L_sum2 = (L_sum2 << 2);
- corr = (L_sum1 + 0x8000) >> 16;
- cor_x[i+1] = vo_mult(corr, sign[pos]) + (*p0++);
- corr = (L_sum2 + 0x8000) >> 16;
- cor_y[i+1] = vo_mult(corr, sign[pos + 1]) + (*p3++);
- pos += STEP;
- }
- return;
+ corr = (L_sum1 + 0x8000) >> 16;
+ cor_x[i+1] = vo_mult(corr, sign[pos]) + (*p0++);
+ corr = (L_sum2 + 0x8000) >> 16;
+ cor_y[i+1] = vo_mult(corr, sign[pos + 1]) + (*p3++);
+ pos += STEP;
+ }
+ return;
}
/*-------------------------------------------------------------------*
@@ -963,80 +970,80 @@
*-------------------------------------------------------------------*/
void search_ixiy(
- Word16 nb_pos_ix, /* (i) nb of pos for pulse 1 (1..8) */
- Word16 track_x, /* (i) track of pulse 1 */
- Word16 track_y, /* (i) track of pulse 2 */
- Word16 * ps, /* (i/o) correlation of all fixed pulses */
- Word16 * alp, /* (i/o) energy of all fixed pulses */
- Word16 * ix, /* (o) position of pulse 1 */
- Word16 * iy, /* (o) position of pulse 2 */
- Word16 dn[], /* (i) corr. between target and h[] */
- Word16 dn2[], /* (i) vector of selected positions */
- Word16 cor_x[], /* (i) corr. of pulse 1 with fixed pulses */
- Word16 cor_y[], /* (i) corr. of pulse 2 with fixed pulses */
- Word16 rrixiy[][MSIZE] /* (i) corr. of pulse 1 with pulse 2 */
- )
+ Word16 nb_pos_ix, /* (i) nb of pos for pulse 1 (1..8) */
+ Word16 track_x, /* (i) track of pulse 1 */
+ Word16 track_y, /* (i) track of pulse 2 */
+ Word16 * ps, /* (i/o) correlation of all fixed pulses */
+ Word16 * alp, /* (i/o) energy of all fixed pulses */
+ Word16 * ix, /* (o) position of pulse 1 */
+ Word16 * iy, /* (o) position of pulse 2 */
+ Word16 dn[], /* (i) corr. between target and h[] */
+ Word16 dn2[], /* (i) vector of selected positions */
+ Word16 cor_x[], /* (i) corr. of pulse 1 with fixed pulses */
+ Word16 cor_y[], /* (i) corr. of pulse 2 with fixed pulses */
+ Word16 rrixiy[][MSIZE] /* (i) corr. of pulse 1 with pulse 2 */
+ )
{
- Word32 x, y, pos, thres_ix;
- Word16 ps1, ps2, sq, sqk;
- Word16 alp_16, alpk;
- Word16 *p0, *p1, *p2;
- Word32 s, alp0, alp1, alp2;
+ Word32 x, y, pos, thres_ix;
+ Word16 ps1, ps2, sq, sqk;
+ Word16 alp_16, alpk;
+ Word16 *p0, *p1, *p2;
+ Word32 s, alp0, alp1, alp2;
- p0 = cor_x;
- p1 = cor_y;
- p2 = rrixiy[track_x];
+ p0 = cor_x;
+ p1 = cor_y;
+ p2 = rrixiy[track_x];
- thres_ix = nb_pos_ix - NB_MAX;
+ thres_ix = nb_pos_ix - NB_MAX;
- alp0 = L_deposit_h(*alp);
- alp0 = (alp0 + 0x00008000L); /* for rounding */
+ alp0 = L_deposit_h(*alp);
+ alp0 = (alp0 + 0x00008000L); /* for rounding */
- sqk = -1;
- alpk = 1;
+ sqk = -1;
+ alpk = 1;
- for (x = track_x; x < L_SUBFR; x += STEP)
- {
- ps1 = *ps + dn[x];
- alp1 = alp0 + ((*p0++)<<13);
+ for (x = track_x; x < L_SUBFR; x += STEP)
+ {
+ ps1 = *ps + dn[x];
+ alp1 = alp0 + ((*p0++)<<13);
- if (dn2[x] < thres_ix)
- {
- pos = -1;
- for (y = track_y; y < L_SUBFR; y += STEP)
- {
- ps2 = add1(ps1, dn[y]);
+ if (dn2[x] < thres_ix)
+ {
+ pos = -1;
+ for (y = track_y; y < L_SUBFR; y += STEP)
+ {
+ ps2 = add1(ps1, dn[y]);
- alp2 = alp1 + ((*p1++)<<13);
- alp2 = alp2 + ((*p2++)<<14);
- alp_16 = extract_h(alp2);
- sq = vo_mult(ps2, ps2);
- s = vo_L_mult(alpk, sq) - ((sqk * alp_16)<<1);
+ alp2 = alp1 + ((*p1++)<<13);
+ alp2 = alp2 + ((*p2++)<<14);
+ alp_16 = extract_h(alp2);
+ sq = vo_mult(ps2, ps2);
+ s = vo_L_mult(alpk, sq) - ((sqk * alp_16)<<1);
- if (s > 0)
- {
- sqk = sq;
- alpk = alp_16;
- pos = y;
- }
- }
- p1 -= NB_POS;
+ if (s > 0)
+ {
+ sqk = sq;
+ alpk = alp_16;
+ pos = y;
+ }
+ }
+ p1 -= NB_POS;
- if (pos >= 0)
- {
- *ix = x;
- *iy = pos;
- }
- } else
- {
- p2 += NB_POS;
- }
- }
+ if (pos >= 0)
+ {
+ *ix = x;
+ *iy = pos;
+ }
+ } else
+ {
+ p2 += NB_POS;
+ }
+ }
- *ps = add1(*ps, add1(dn[*ix], dn[*iy]));
- *alp = alpk;
+ *ps = add1(*ps, add1(dn[*ix], dn[*iy]));
+ *alp = alpk;
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/convolve.c b/media/libstagefright/codecs/amrwbenc/src/convolve.c
index 4c1f7d4..9b8b3aa 100644
--- a/media/libstagefright/codecs/amrwbenc/src/convolve.c
+++ b/media/libstagefright/codecs/amrwbenc/src/convolve.c
@@ -17,8 +17,8 @@
/***********************************************************************
File: convolve.c
- Description:Perform the convolution between two vectors x[] and h[]
- and write the result in the vector y[]
+ Description:Perform the convolution between two vectors x[] and h[]
+ and write the result in the vector y[]
************************************************************************/
@@ -28,85 +28,85 @@
#define UNUSED(x) (void)(x)
void Convolve (
- Word16 x[], /* (i) : input vector */
- Word16 h[], /* (i) : impulse response */
- Word16 y[], /* (o) : output vector */
- Word16 L /* (i) : vector size */
- )
+ Word16 x[], /* (i) : input vector */
+ Word16 h[], /* (i) : impulse response */
+ Word16 y[], /* (o) : output vector */
+ Word16 L /* (i) : vector size */
+ )
{
- Word32 i, n;
- Word16 *tmpH,*tmpX;
- Word32 s;
+ Word32 i, n;
+ Word16 *tmpH,*tmpX;
+ Word32 s;
UNUSED(L);
- for (n = 0; n < 64;)
- {
- tmpH = h+n;
- tmpX = x;
- i=n+1;
- s = vo_mult32((*tmpX++), (*tmpH--));i--;
- while(i>0)
- {
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- i -= 4;
- }
- y[n] = ((s<<1) + 0x8000)>>16;
- n++;
+ for (n = 0; n < 64;)
+ {
+ tmpH = h+n;
+ tmpX = x;
+ i=n+1;
+ s = vo_mult32((*tmpX++), (*tmpH--));i--;
+ while(i>0)
+ {
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ i -= 4;
+ }
+ y[n] = ((s<<1) + 0x8000)>>16;
+ n++;
- tmpH = h+n;
- tmpX = x;
- i=n+1;
- s = vo_mult32((*tmpX++), (*tmpH--));i--;
- s += vo_mult32((*tmpX++), (*tmpH--));i--;
+ tmpH = h+n;
+ tmpX = x;
+ i=n+1;
+ s = vo_mult32((*tmpX++), (*tmpH--));i--;
+ s += vo_mult32((*tmpX++), (*tmpH--));i--;
- while(i>0)
- {
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- i -= 4;
- }
- y[n] = ((s<<1) + 0x8000)>>16;
- n++;
+ while(i>0)
+ {
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ i -= 4;
+ }
+ y[n] = ((s<<1) + 0x8000)>>16;
+ n++;
- tmpH = h+n;
- tmpX = x;
- i=n+1;
- s = vo_mult32((*tmpX++), (*tmpH--));i--;
- s += vo_mult32((*tmpX++), (*tmpH--));i--;
- s += vo_mult32((*tmpX++), (*tmpH--));i--;
+ tmpH = h+n;
+ tmpX = x;
+ i=n+1;
+ s = vo_mult32((*tmpX++), (*tmpH--));i--;
+ s += vo_mult32((*tmpX++), (*tmpH--));i--;
+ s += vo_mult32((*tmpX++), (*tmpH--));i--;
- while(i>0)
- {
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- i -= 4;
- }
- y[n] = ((s<<1) + 0x8000)>>16;
- n++;
+ while(i>0)
+ {
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ i -= 4;
+ }
+ y[n] = ((s<<1) + 0x8000)>>16;
+ n++;
- s = 0;
- tmpH = h+n;
- tmpX = x;
- i=n+1;
- while(i>0)
- {
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- s += vo_mult32((*tmpX++), (*tmpH--));
- i -= 4;
- }
- y[n] = ((s<<1) + 0x8000)>>16;
- n++;
- }
- return;
+ s = 0;
+ tmpH = h+n;
+ tmpX = x;
+ i=n+1;
+ while(i>0)
+ {
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ s += vo_mult32((*tmpX++), (*tmpH--));
+ i -= 4;
+ }
+ y[n] = ((s<<1) + 0x8000)>>16;
+ n++;
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/cor_h_x.c b/media/libstagefright/codecs/amrwbenc/src/cor_h_x.c
index d9245ed..b2aa759 100644
--- a/media/libstagefright/codecs/amrwbenc/src/cor_h_x.c
+++ b/media/libstagefright/codecs/amrwbenc/src/cor_h_x.c
@@ -17,10 +17,10 @@
/***********************************************************************
* File: cor_h_x.c *
* *
-* Description:Compute correlation between target "x[]" and "h[]" *
-* Designed for codebook search (24 pulses, 4 tracks, *
-* 4 pulses per track, 16 positions in each track) to *
-* avoid saturation. *
+* Description:Compute correlation between target "x[]" and "h[]" *
+* Designed for codebook search (24 pulses, 4 tracks, *
+* 4 pulses per track, 16 positions in each track) to *
+* avoid saturation. *
* *
************************************************************************/
@@ -33,94 +33,94 @@
#define STEP 4
void cor_h_x(
- Word16 h[], /* (i) Q12 : impulse response of weighted synthesis filter */
- Word16 x[], /* (i) Q0 : target vector */
- Word16 dn[] /* (o) <12bit : correlation between target and h[] */
- )
+ Word16 h[], /* (i) Q12 : impulse response of weighted synthesis filter */
+ Word16 x[], /* (i) Q0 : target vector */
+ Word16 dn[] /* (o) <12bit : correlation between target and h[] */
+ )
{
- Word32 i, j;
- Word32 L_tmp, y32[L_SUBFR], L_tot;
- Word16 *p1, *p2;
- Word32 *p3;
- Word32 L_max, L_max1, L_max2, L_max3;
- /* first keep the result on 32 bits and find absolute maximum */
- L_tot = 1;
- L_max = 0;
- L_max1 = 0;
- L_max2 = 0;
- L_max3 = 0;
- for (i = 0; i < L_SUBFR; i += STEP)
- {
- L_tmp = 1; /* 1 -> to avoid null dn[] */
- p1 = &x[i];
- p2 = &h[0];
- for (j = i; j < L_SUBFR; j++)
- L_tmp += vo_L_mult(*p1++, *p2++);
+ Word32 i, j;
+ Word32 L_tmp, y32[L_SUBFR], L_tot;
+ Word16 *p1, *p2;
+ Word32 *p3;
+ Word32 L_max, L_max1, L_max2, L_max3;
+ /* first keep the result on 32 bits and find absolute maximum */
+ L_tot = 1;
+ L_max = 0;
+ L_max1 = 0;
+ L_max2 = 0;
+ L_max3 = 0;
+ for (i = 0; i < L_SUBFR; i += STEP)
+ {
+ L_tmp = 1; /* 1 -> to avoid null dn[] */
+ p1 = &x[i];
+ p2 = &h[0];
+ for (j = i; j < L_SUBFR; j++)
+ L_tmp += vo_L_mult(*p1++, *p2++);
- y32[i] = L_tmp;
- L_tmp = (L_tmp > 0)? L_tmp:-L_tmp;
- if(L_tmp > L_max)
- {
- L_max = L_tmp;
- }
+ y32[i] = L_tmp;
+ L_tmp = (L_tmp > 0)? L_tmp:-L_tmp;
+ if(L_tmp > L_max)
+ {
+ L_max = L_tmp;
+ }
- L_tmp = 1L;
- p1 = &x[i+1];
- p2 = &h[0];
- for (j = i+1; j < L_SUBFR; j++)
- L_tmp += vo_L_mult(*p1++, *p2++);
+ L_tmp = 1L;
+ p1 = &x[i+1];
+ p2 = &h[0];
+ for (j = i+1; j < L_SUBFR; j++)
+ L_tmp += vo_L_mult(*p1++, *p2++);
- y32[i+1] = L_tmp;
- L_tmp = (L_tmp > 0)? L_tmp:-L_tmp;
- if(L_tmp > L_max1)
- {
- L_max1 = L_tmp;
- }
+ y32[i+1] = L_tmp;
+ L_tmp = (L_tmp > 0)? L_tmp:-L_tmp;
+ if(L_tmp > L_max1)
+ {
+ L_max1 = L_tmp;
+ }
- L_tmp = 1;
- p1 = &x[i+2];
- p2 = &h[0];
- for (j = i+2; j < L_SUBFR; j++)
- L_tmp += vo_L_mult(*p1++, *p2++);
+ L_tmp = 1;
+ p1 = &x[i+2];
+ p2 = &h[0];
+ for (j = i+2; j < L_SUBFR; j++)
+ L_tmp += vo_L_mult(*p1++, *p2++);
- y32[i+2] = L_tmp;
- L_tmp = (L_tmp > 0)? L_tmp:-L_tmp;
- if(L_tmp > L_max2)
- {
- L_max2 = L_tmp;
- }
+ y32[i+2] = L_tmp;
+ L_tmp = (L_tmp > 0)? L_tmp:-L_tmp;
+ if(L_tmp > L_max2)
+ {
+ L_max2 = L_tmp;
+ }
- L_tmp = 1;
- p1 = &x[i+3];
- p2 = &h[0];
- for (j = i+3; j < L_SUBFR; j++)
- L_tmp += vo_L_mult(*p1++, *p2++);
+ L_tmp = 1;
+ p1 = &x[i+3];
+ p2 = &h[0];
+ for (j = i+3; j < L_SUBFR; j++)
+ L_tmp += vo_L_mult(*p1++, *p2++);
- y32[i+3] = L_tmp;
- L_tmp = (L_tmp > 0)? L_tmp:-L_tmp;
- if(L_tmp > L_max3)
- {
- L_max3 = L_tmp;
- }
- }
- /* tot += 3*max / 8 */
- L_max = ((L_max + L_max1 + L_max2 + L_max3) >> 2);
- L_tot = vo_L_add(L_tot, L_max); /* +max/4 */
- L_tot = vo_L_add(L_tot, (L_max >> 1)); /* +max/8 */
+ y32[i+3] = L_tmp;
+ L_tmp = (L_tmp > 0)? L_tmp:-L_tmp;
+ if(L_tmp > L_max3)
+ {
+ L_max3 = L_tmp;
+ }
+ }
+ /* tot += 3*max / 8 */
+ L_max = ((L_max + L_max1 + L_max2 + L_max3) >> 2);
+ L_tot = vo_L_add(L_tot, L_max); /* +max/4 */
+ L_tot = vo_L_add(L_tot, (L_max >> 1)); /* +max/8 */
- /* Find the number of right shifts to do on y32[] so that */
- /* 6.0 x sumation of max of dn[] in each track not saturate. */
- j = norm_l(L_tot) - 4; /* 4 -> 16 x tot */
- p1 = dn;
- p3 = y32;
- for (i = 0; i < L_SUBFR; i+=4)
- {
- *p1++ = vo_round(L_shl(*p3++, j));
- *p1++ = vo_round(L_shl(*p3++, j));
- *p1++ = vo_round(L_shl(*p3++, j));
- *p1++ = vo_round(L_shl(*p3++, j));
- }
- return;
+ /* Find the number of right shifts to do on y32[] so that */
+ /* 6.0 x sumation of max of dn[] in each track not saturate. */
+ j = norm_l(L_tot) - 4; /* 4 -> 16 x tot */
+ p1 = dn;
+ p3 = y32;
+ for (i = 0; i < L_SUBFR; i+=4)
+ {
+ *p1++ = vo_round(L_shl(*p3++, j));
+ *p1++ = vo_round(L_shl(*p3++, j));
+ *p1++ = vo_round(L_shl(*p3++, j));
+ *p1++ = vo_round(L_shl(*p3++, j));
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/decim54.c b/media/libstagefright/codecs/amrwbenc/src/decim54.c
index 3b88514..e4c7940 100644
--- a/media/libstagefright/codecs/amrwbenc/src/decim54.c
+++ b/media/libstagefright/codecs/amrwbenc/src/decim54.c
@@ -17,7 +17,7 @@
/***********************************************************************
* File: decim54.c *
* *
-* Description:Decimation of 16kHz signal to 12.8kHz *
+* Description:Decimation of 16kHz signal to 12.8kHz *
* *
************************************************************************/
@@ -33,114 +33,114 @@
/* Local functions */
static void Down_samp(
- Word16 * sig, /* input: signal to downsampling */
- Word16 * sig_d, /* output: downsampled signal */
- Word16 L_frame_d /* input: length of output */
- );
+ Word16 * sig, /* input: signal to downsampling */
+ Word16 * sig_d, /* output: downsampled signal */
+ Word16 L_frame_d /* input: length of output */
+ );
/* 1/5 resolution interpolation filter (in Q14) */
/* -1.5dB @ 6kHz, -6dB @ 6.4kHz, -10dB @ 6.6kHz, -20dB @ 6.9kHz, -25dB @ 7kHz, -55dB @ 8kHz */
static Word16 fir_down1[4][30] =
{
- {-5, 24, -50, 54, 0, -128, 294, -408, 344, 0, -647, 1505, -2379, 3034, 13107, 3034, -2379, 1505, -647, 0, 344, -408,
- 294, -128, 0, 54, -50, 24, -5, 0},
+ {-5, 24, -50, 54, 0, -128, 294, -408, 344, 0, -647, 1505, -2379, 3034, 13107, 3034, -2379, 1505, -647, 0, 344, -408,
+ 294, -128, 0, 54, -50, 24, -5, 0},
- {-6, 19, -26, 0, 77, -188, 270, -233, 0, 434, -964, 1366, -1293, 0, 12254, 6575, -2746, 1030, 0, -507, 601, -441,
- 198, 0, -95, 99, -58, 18, 0, -1},
+ {-6, 19, -26, 0, 77, -188, 270, -233, 0, 434, -964, 1366, -1293, 0, 12254, 6575, -2746, 1030, 0, -507, 601, -441,
+ 198, 0, -95, 99, -58, 18, 0, -1},
- {-3, 9, 0, -41, 111, -170, 153, 0, -295, 649, -888, 770, 0, -1997, 9894, 9894, -1997, 0, 770, -888, 649, -295, 0,
- 153, -170, 111, -41, 0, 9, -3},
+ {-3, 9, 0, -41, 111, -170, 153, 0, -295, 649, -888, 770, 0, -1997, 9894, 9894, -1997, 0, 770, -888, 649, -295, 0,
+ 153, -170, 111, -41, 0, 9, -3},
- {-1, 0, 18, -58, 99, -95, 0, 198, -441, 601, -507, 0, 1030, -2746, 6575, 12254, 0, -1293, 1366, -964, 434, 0,
- -233, 270, -188, 77, 0, -26, 19, -6}
+ {-1, 0, 18, -58, 99, -95, 0, 198, -441, 601, -507, 0, 1030, -2746, 6575, 12254, 0, -1293, 1366, -964, 434, 0,
+ -233, 270, -188, 77, 0, -26, 19, -6}
};
void Init_Decim_12k8(
- Word16 mem[] /* output: memory (2*NB_COEF_DOWN) set to zeros */
- )
+ Word16 mem[] /* output: memory (2*NB_COEF_DOWN) set to zeros */
+ )
{
- Set_zero(mem, 2 * NB_COEF_DOWN);
- return;
+ Set_zero(mem, 2 * NB_COEF_DOWN);
+ return;
}
void Decim_12k8(
- Word16 sig16k[], /* input: signal to downsampling */
- Word16 lg, /* input: length of input */
- Word16 sig12k8[], /* output: decimated signal */
- Word16 mem[] /* in/out: memory (2*NB_COEF_DOWN) */
- )
+ Word16 sig16k[], /* input: signal to downsampling */
+ Word16 lg, /* input: length of input */
+ Word16 sig12k8[], /* output: decimated signal */
+ Word16 mem[] /* in/out: memory (2*NB_COEF_DOWN) */
+ )
{
- Word16 lg_down;
- Word16 signal[L_FRAME16k + (2 * NB_COEF_DOWN)];
+ Word16 lg_down;
+ Word16 signal[L_FRAME16k + (2 * NB_COEF_DOWN)];
- Copy(mem, signal, 2 * NB_COEF_DOWN);
+ Copy(mem, signal, 2 * NB_COEF_DOWN);
- Copy(sig16k, signal + (2 * NB_COEF_DOWN), lg);
+ Copy(sig16k, signal + (2 * NB_COEF_DOWN), lg);
- lg_down = (lg * DOWN_FAC)>>15;
+ lg_down = (lg * DOWN_FAC)>>15;
- Down_samp(signal + NB_COEF_DOWN, sig12k8, lg_down);
+ Down_samp(signal + NB_COEF_DOWN, sig12k8, lg_down);
- Copy(signal + lg, mem, 2 * NB_COEF_DOWN);
+ Copy(signal + lg, mem, 2 * NB_COEF_DOWN);
- return;
+ return;
}
static void Down_samp(
- Word16 * sig, /* input: signal to downsampling */
- Word16 * sig_d, /* output: downsampled signal */
- Word16 L_frame_d /* input: length of output */
- )
+ Word16 * sig, /* input: signal to downsampling */
+ Word16 * sig_d, /* output: downsampled signal */
+ Word16 L_frame_d /* input: length of output */
+ )
{
- Word32 i, j, frac, pos;
- Word16 *x, *y;
- Word32 L_sum;
+ Word32 i, j, frac, pos;
+ Word16 *x, *y;
+ Word32 L_sum;
- pos = 0; /* position is in Q2 -> 1/4 resolution */
- for (j = 0; j < L_frame_d; j++)
- {
- i = (pos >> 2); /* integer part */
- frac = pos & 3; /* fractional part */
- x = sig + i - NB_COEF_DOWN + 1;
- y = (Word16 *)(fir_down1 + frac);
+ pos = 0; /* position is in Q2 -> 1/4 resolution */
+ for (j = 0; j < L_frame_d; j++)
+ {
+ i = (pos >> 2); /* integer part */
+ frac = pos & 3; /* fractional part */
+ x = sig + i - NB_COEF_DOWN + 1;
+ y = (Word16 *)(fir_down1 + frac);
- L_sum = vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x++),(*y++));
- L_sum += vo_mult32((*x),(*y));
+ L_sum = vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x++),(*y++));
+ L_sum += vo_mult32((*x),(*y));
- L_sum = L_shl2(L_sum, 2);
- sig_d[j] = extract_h(L_add(L_sum, 0x8000));
- pos += FAC5; /* pos + 5/4 */
- }
- return;
+ L_sum = L_shl2(L_sum, 2);
+ sig_d[j] = extract_h(L_add(L_sum, 0x8000));
+ pos += FAC5; /* pos + 5/4 */
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/deemph.c b/media/libstagefright/codecs/amrwbenc/src/deemph.c
index 0c49d6b..cc27f6e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/deemph.c
+++ b/media/libstagefright/codecs/amrwbenc/src/deemph.c
@@ -17,9 +17,9 @@
/***********************************************************************
* File: deemph.c *
* *
-* Description:filtering through 1/(1-mu z^ -1) *
-* Deemph2 --> signal is divided by 2 *
-* Deemph_32 --> for 32 bits signal. *
+* Description:filtering through 1/(1-mu z^ -1) *
+* Deemph2 --> signal is divided by 2 *
+* Deemph_32 --> for 32 bits signal. *
* *
************************************************************************/
@@ -28,89 +28,92 @@
#include "math_op.h"
void Deemph(
- Word16 x[], /* (i/o) : input signal overwritten by the output */
- Word16 mu, /* (i) Q15 : deemphasis factor */
- Word16 L, /* (i) : vector size */
- Word16 * mem /* (i/o) : memory (y[-1]) */
- )
+ Word16 x[], /* (i/o) : input signal overwritten by the output */
+ Word16 mu, /* (i) Q15 : deemphasis factor */
+ Word16 L, /* (i) : vector size */
+ Word16 * mem /* (i/o) : memory (y[-1]) */
+ )
{
- Word32 i;
- Word32 L_tmp;
+ Word32 i;
+ Word32 L_tmp;
- L_tmp = L_deposit_h(x[0]);
- L_tmp = L_mac(L_tmp, *mem, mu);
- x[0] = vo_round(L_tmp);
+ L_tmp = L_deposit_h(x[0]);
+ L_tmp = L_mac(L_tmp, *mem, mu);
+ x[0] = vo_round(L_tmp);
- for (i = 1; i < L; i++)
- {
- L_tmp = L_deposit_h(x[i]);
- L_tmp = L_mac(L_tmp, x[i - 1], mu);
- x[i] = voround(L_tmp);
- }
+ for (i = 1; i < L; i++)
+ {
+ L_tmp = L_deposit_h(x[i]);
+ L_tmp = L_mac(L_tmp, x[i - 1], mu);
+ x[i] = voround(L_tmp);
+ }
- *mem = x[L - 1];
+ *mem = x[L - 1];
- return;
+ return;
}
void Deemph2(
- Word16 x[], /* (i/o) : input signal overwritten by the output */
- Word16 mu, /* (i) Q15 : deemphasis factor */
- Word16 L, /* (i) : vector size */
- Word16 * mem /* (i/o) : memory (y[-1]) */
- )
+ Word16 x[], /* (i/o) : input signal overwritten by the output */
+ Word16 mu, /* (i) Q15 : deemphasis factor */
+ Word16 L, /* (i) : vector size */
+ Word16 * mem /* (i/o) : memory (y[-1]) */
+ )
{
- Word32 i;
- Word32 L_tmp;
- L_tmp = x[0] << 15;
- L_tmp += ((*mem) * mu)<<1;
- x[0] = (L_tmp + 0x8000)>>16;
- for (i = 1; i < L; i++)
- {
- L_tmp = x[i] << 15;
- L_tmp += (x[i - 1] * mu)<<1;
- x[i] = (L_tmp + 0x8000)>>16;
- }
- *mem = x[L - 1];
- return;
+ Word32 i;
+ Word32 L_tmp;
+ L_tmp = x[0] << 15;
+ i = L_mult(*mem, mu);
+ L_tmp = L_add(L_tmp, i);
+ x[0] = voround(L_tmp);
+ for (i = 1; i < L; i++)
+ {
+ Word32 tmp;
+ L_tmp = x[i] << 15;
+ tmp = (x[i - 1] * mu)<<1;
+ L_tmp = L_add(L_tmp, tmp);
+ x[i] = voround(L_tmp);
+ }
+ *mem = x[L - 1];
+ return;
}
void Deemph_32(
- Word16 x_hi[], /* (i) : input signal (bit31..16) */
- Word16 x_lo[], /* (i) : input signal (bit15..4) */
- Word16 y[], /* (o) : output signal (x16) */
- Word16 mu, /* (i) Q15 : deemphasis factor */
- Word16 L, /* (i) : vector size */
- Word16 * mem /* (i/o) : memory (y[-1]) */
- )
+ Word16 x_hi[], /* (i) : input signal (bit31..16) */
+ Word16 x_lo[], /* (i) : input signal (bit15..4) */
+ Word16 y[], /* (o) : output signal (x16) */
+ Word16 mu, /* (i) Q15 : deemphasis factor */
+ Word16 L, /* (i) : vector size */
+ Word16 * mem /* (i/o) : memory (y[-1]) */
+ )
{
- Word16 fac;
- Word32 i, L_tmp;
+ Word16 fac;
+ Word32 i, L_tmp;
- fac = mu >> 1; /* Q15 --> Q14 */
+ fac = mu >> 1; /* Q15 --> Q14 */
- L_tmp = L_deposit_h(x_hi[0]);
- L_tmp += (x_lo[0] * 8)<<1;
- L_tmp = (L_tmp << 3);
- L_tmp += ((*mem) * fac)<<1;
- L_tmp = (L_tmp << 1);
- y[0] = (L_tmp + 0x8000)>>16;
+ L_tmp = L_deposit_h(x_hi[0]);
+ L_tmp += (x_lo[0] * 8)<<1;
+ L_tmp = (L_tmp << 3);
+ L_tmp += ((*mem) * fac)<<1;
+ L_tmp = (L_tmp << 1);
+ y[0] = (L_tmp + 0x8000)>>16;
- for (i = 1; i < L; i++)
- {
- L_tmp = L_deposit_h(x_hi[i]);
- L_tmp += (x_lo[i] * 8)<<1;
- L_tmp = (L_tmp << 3);
- L_tmp += (y[i - 1] * fac)<<1;
- L_tmp = (L_tmp << 1);
- y[i] = (L_tmp + 0x8000)>>16;
- }
+ for (i = 1; i < L; i++)
+ {
+ L_tmp = L_deposit_h(x_hi[i]);
+ L_tmp += (x_lo[i] * 8)<<1;
+ L_tmp = (L_tmp << 3);
+ L_tmp += (y[i - 1] * fac)<<1;
+ L_tmp = (L_tmp << 1);
+ y[i] = (L_tmp + 0x8000)>>16;
+ }
- *mem = y[L - 1];
+ *mem = y[L - 1];
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/dtx.c b/media/libstagefright/codecs/amrwbenc/src/dtx.c
index 2cfaced..6be8683 100644
--- a/media/libstagefright/codecs/amrwbenc/src/dtx.c
+++ b/media/libstagefright/codecs/amrwbenc/src/dtx.c
@@ -17,7 +17,7 @@
/***********************************************************************
* File: dtx.c *
* *
-* Description:DTX functions *
+* Description:DTX functions *
* *
************************************************************************/
@@ -35,33 +35,33 @@
#include "mem_align.h"
static void aver_isf_history(
- Word16 isf_old[],
- Word16 indices[],
- Word32 isf_aver[]
- );
+ Word16 isf_old[],
+ Word16 indices[],
+ Word32 isf_aver[]
+ );
static void find_frame_indices(
- Word16 isf_old_tx[],
- Word16 indices[],
- dtx_encState * st
- );
+ Word16 isf_old_tx[],
+ Word16 indices[],
+ dtx_encState * st
+ );
static Word16 dithering_control(
- dtx_encState * st
- );
+ dtx_encState * st
+ );
/* excitation energy adjustment depending on speech coder mode used, Q7 */
static Word16 en_adjust[9] =
{
- 230, /* mode0 = 7k : -5.4dB */
- 179, /* mode1 = 9k : -4.2dB */
- 141, /* mode2 = 12k : -3.3dB */
- 128, /* mode3 = 14k : -3.0dB */
- 122, /* mode4 = 16k : -2.85dB */
- 115, /* mode5 = 18k : -2.7dB */
- 115, /* mode6 = 20k : -2.7dB */
- 115, /* mode7 = 23k : -2.7dB */
- 115 /* mode8 = 24k : -2.7dB */
+ 230, /* mode0 = 7k : -5.4dB */
+ 179, /* mode1 = 9k : -4.2dB */
+ 141, /* mode2 = 12k : -3.3dB */
+ 128, /* mode3 = 14k : -3.0dB */
+ 122, /* mode4 = 16k : -2.85dB */
+ 115, /* mode5 = 18k : -2.7dB */
+ 115, /* mode6 = 20k : -2.7dB */
+ 115, /* mode7 = 23k : -2.7dB */
+ 115 /* mode8 = 24k : -2.7dB */
};
/**************************************************************************
@@ -71,24 +71,24 @@
**************************************************************************/
Word16 dtx_enc_init(dtx_encState ** st, Word16 isf_init[], VO_MEM_OPERATOR *pMemOP)
{
- dtx_encState *s;
+ dtx_encState *s;
- if (st == (dtx_encState **) NULL)
- {
- fprintf(stderr, "dtx_enc_init: invalid parameter\n");
- return -1;
- }
- *st = NULL;
+ if (st == (dtx_encState **) NULL)
+ {
+ fprintf(stderr, "dtx_enc_init: invalid parameter\n");
+ return -1;
+ }
+ *st = NULL;
- /* allocate memory */
- if ((s = (dtx_encState *)mem_malloc(pMemOP, sizeof(dtx_encState), 32, VO_INDEX_ENC_AMRWB)) == NULL)
- {
- fprintf(stderr, "dtx_enc_init: can not malloc state structure\n");
- return -1;
- }
- dtx_enc_reset(s, isf_init);
- *st = s;
- return 0;
+ /* allocate memory */
+ if ((s = (dtx_encState *)mem_malloc(pMemOP, sizeof(dtx_encState), 32, VO_INDEX_ENC_AMRWB)) == NULL)
+ {
+ fprintf(stderr, "dtx_enc_init: can not malloc state structure\n");
+ return -1;
+ }
+ dtx_enc_reset(s, isf_init);
+ *st = s;
+ return 0;
}
/**************************************************************************
@@ -98,40 +98,40 @@
**************************************************************************/
Word16 dtx_enc_reset(dtx_encState * st, Word16 isf_init[])
{
- Word32 i;
+ Word32 i;
- if (st == (dtx_encState *) NULL)
- {
- fprintf(stderr, "dtx_enc_reset: invalid parameter\n");
- return -1;
- }
- st->hist_ptr = 0;
- st->log_en_index = 0;
+ if (st == (dtx_encState *) NULL)
+ {
+ fprintf(stderr, "dtx_enc_reset: invalid parameter\n");
+ return -1;
+ }
+ st->hist_ptr = 0;
+ st->log_en_index = 0;
- /* Init isf_hist[] */
- for (i = 0; i < DTX_HIST_SIZE; i++)
- {
- Copy(isf_init, &st->isf_hist[i * M], M);
- }
- st->cng_seed = RANDOM_INITSEED;
+ /* Init isf_hist[] */
+ for (i = 0; i < DTX_HIST_SIZE; i++)
+ {
+ Copy(isf_init, &st->isf_hist[i * M], M);
+ }
+ st->cng_seed = RANDOM_INITSEED;
- /* Reset energy history */
- Set_zero(st->log_en_hist, DTX_HIST_SIZE);
+ /* Reset energy history */
+ Set_zero(st->log_en_hist, DTX_HIST_SIZE);
- st->dtxHangoverCount = DTX_HANG_CONST;
- st->decAnaElapsedCount = 32767;
+ st->dtxHangoverCount = DTX_HANG_CONST;
+ st->decAnaElapsedCount = 32767;
- for (i = 0; i < 28; i++)
- {
- st->D[i] = 0;
- }
+ for (i = 0; i < 28; i++)
+ {
+ st->D[i] = 0;
+ }
- for (i = 0; i < DTX_HIST_SIZE - 1; i++)
- {
- st->sumD[i] = 0;
- }
+ for (i = 0; i < DTX_HIST_SIZE - 1; i++)
+ {
+ st->sumD[i] = 0;
+ }
- return 1;
+ return 1;
}
/**************************************************************************
@@ -141,12 +141,12 @@
**************************************************************************/
void dtx_enc_exit(dtx_encState ** st, VO_MEM_OPERATOR *pMemOP)
{
- if (st == NULL || *st == NULL)
- return;
- /* deallocate memory */
- mem_free(pMemOP, *st, VO_INDEX_ENC_AMRWB);
- *st = NULL;
- return;
+ if (st == NULL || *st == NULL)
+ return;
+ /* deallocate memory */
+ mem_free(pMemOP, *st, VO_INDEX_ENC_AMRWB);
+ *st = NULL;
+ return;
}
@@ -156,133 +156,133 @@
*
**************************************************************************/
Word16 dtx_enc(
- dtx_encState * st, /* i/o : State struct */
- Word16 isf[M], /* o : CN ISF vector */
- Word16 * exc2, /* o : CN excitation */
- Word16 ** prms
- )
+ dtx_encState * st, /* i/o : State struct */
+ Word16 isf[M], /* o : CN ISF vector */
+ Word16 * exc2, /* o : CN excitation */
+ Word16 ** prms
+ )
{
- Word32 i, j;
- Word16 indice[7];
- Word16 log_en, gain, level, exp, exp0, tmp;
- Word16 log_en_int_e, log_en_int_m;
- Word32 L_isf[M], ener32, level32;
- Word16 isf_order[3];
- Word16 CN_dith;
+ Word32 i, j;
+ Word16 indice[7];
+ Word16 log_en, gain, level, exp, exp0, tmp;
+ Word16 log_en_int_e, log_en_int_m;
+ Word32 L_isf[M], ener32, level32;
+ Word16 isf_order[3];
+ Word16 CN_dith;
- /* VOX mode computation of SID parameters */
- log_en = 0;
- for (i = 0; i < M; i++)
- {
- L_isf[i] = 0;
- }
- /* average energy and isf */
- for (i = 0; i < DTX_HIST_SIZE; i++)
- {
- /* Division by DTX_HIST_SIZE = 8 has been done in dtx_buffer. log_en is in Q10 */
- log_en = add(log_en, st->log_en_hist[i]);
+ /* VOX mode computation of SID parameters */
+ log_en = 0;
+ for (i = 0; i < M; i++)
+ {
+ L_isf[i] = 0;
+ }
+ /* average energy and isf */
+ for (i = 0; i < DTX_HIST_SIZE; i++)
+ {
+ /* Division by DTX_HIST_SIZE = 8 has been done in dtx_buffer. log_en is in Q10 */
+ log_en = add(log_en, st->log_en_hist[i]);
- }
- find_frame_indices(st->isf_hist, isf_order, st);
- aver_isf_history(st->isf_hist, isf_order, L_isf);
+ }
+ find_frame_indices(st->isf_hist, isf_order, st);
+ aver_isf_history(st->isf_hist, isf_order, L_isf);
- for (j = 0; j < M; j++)
- {
- isf[j] = (Word16)(L_isf[j] >> 3); /* divide by 8 */
- }
+ for (j = 0; j < M; j++)
+ {
+ isf[j] = (Word16)(L_isf[j] >> 3); /* divide by 8 */
+ }
- /* quantize logarithmic energy to 6 bits (-6 : 66 dB) which corresponds to -2:22 in log2(E). */
- /* st->log_en_index = (short)( (log_en + 2.0) * 2.625 ); */
+ /* quantize logarithmic energy to 6 bits (-6 : 66 dB) which corresponds to -2:22 in log2(E). */
+ /* st->log_en_index = (short)( (log_en + 2.0) * 2.625 ); */
- /* increase dynamics to 7 bits (Q8) */
- log_en = (log_en >> 2);
+ /* increase dynamics to 7 bits (Q8) */
+ log_en = (log_en >> 2);
- /* Add 2 in Q8 = 512 to get log2(E) between 0:24 */
- log_en = add(log_en, 512);
+ /* Add 2 in Q8 = 512 to get log2(E) between 0:24 */
+ log_en = add(log_en, 512);
- /* Multiply by 2.625 to get full 6 bit range. 2.625 = 21504 in Q13. The result is in Q6 */
- log_en = mult(log_en, 21504);
+ /* Multiply by 2.625 to get full 6 bit range. 2.625 = 21504 in Q13. The result is in Q6 */
+ log_en = mult(log_en, 21504);
- /* Quantize Energy */
- st->log_en_index = shr(log_en, 6);
+ /* Quantize Energy */
+ st->log_en_index = shr(log_en, 6);
- if(st->log_en_index > 63)
- {
- st->log_en_index = 63;
- }
- if (st->log_en_index < 0)
- {
- st->log_en_index = 0;
- }
- /* Quantize ISFs */
- Qisf_ns(isf, isf, indice);
+ if(st->log_en_index > 63)
+ {
+ st->log_en_index = 63;
+ }
+ if (st->log_en_index < 0)
+ {
+ st->log_en_index = 0;
+ }
+ /* Quantize ISFs */
+ Qisf_ns(isf, isf, indice);
- Parm_serial(indice[0], 6, prms);
- Parm_serial(indice[1], 6, prms);
- Parm_serial(indice[2], 6, prms);
- Parm_serial(indice[3], 5, prms);
- Parm_serial(indice[4], 5, prms);
+ Parm_serial(indice[0], 6, prms);
+ Parm_serial(indice[1], 6, prms);
+ Parm_serial(indice[2], 6, prms);
+ Parm_serial(indice[3], 5, prms);
+ Parm_serial(indice[4], 5, prms);
- Parm_serial((st->log_en_index), 6, prms);
+ Parm_serial((st->log_en_index), 6, prms);
- CN_dith = dithering_control(st);
- Parm_serial(CN_dith, 1, prms);
+ CN_dith = dithering_control(st);
+ Parm_serial(CN_dith, 1, prms);
- /* level = (float)( pow( 2.0f, (float)st->log_en_index / 2.625 - 2.0 ) ); */
- /* log2(E) in Q9 (log2(E) lies in between -2:22) */
- log_en = shl(st->log_en_index, 15 - 6);
+ /* level = (float)( pow( 2.0f, (float)st->log_en_index / 2.625 - 2.0 ) ); */
+ /* log2(E) in Q9 (log2(E) lies in between -2:22) */
+ log_en = shl(st->log_en_index, 15 - 6);
- /* Divide by 2.625; log_en will be between 0:24 */
- log_en = mult(log_en, 12483);
- /* the result corresponds to log2(gain) in Q10 */
+ /* Divide by 2.625; log_en will be between 0:24 */
+ log_en = mult(log_en, 12483);
+ /* the result corresponds to log2(gain) in Q10 */
- /* Find integer part */
- log_en_int_e = (log_en >> 10);
+ /* Find integer part */
+ log_en_int_e = (log_en >> 10);
- /* Find fractional part */
- log_en_int_m = (Word16) (log_en & 0x3ff);
- log_en_int_m = shl(log_en_int_m, 5);
+ /* Find fractional part */
+ log_en_int_m = (Word16) (log_en & 0x3ff);
+ log_en_int_m = shl(log_en_int_m, 5);
- /* Subtract 2 from log_en in Q9, i.e divide the gain by 2 (energy by 4) */
- /* Add 16 in order to have the result of pow2 in Q16 */
- log_en_int_e = add(log_en_int_e, 16 - 1);
+ /* Subtract 2 from log_en in Q9, i.e divide the gain by 2 (energy by 4) */
+ /* Add 16 in order to have the result of pow2 in Q16 */
+ log_en_int_e = add(log_en_int_e, 16 - 1);
- level32 = Pow2(log_en_int_e, log_en_int_m); /* Q16 */
- exp0 = norm_l(level32);
- level32 = (level32 << exp0); /* level in Q31 */
- exp0 = (15 - exp0);
- level = extract_h(level32); /* level in Q15 */
+ level32 = Pow2(log_en_int_e, log_en_int_m); /* Q16 */
+ exp0 = norm_l(level32);
+ level32 = (level32 << exp0); /* level in Q31 */
+ exp0 = (15 - exp0);
+ level = extract_h(level32); /* level in Q15 */
- /* generate white noise vector */
- for (i = 0; i < L_FRAME; i++)
- {
- exc2[i] = (Random(&(st->cng_seed)) >> 4);
- }
+ /* generate white noise vector */
+ for (i = 0; i < L_FRAME; i++)
+ {
+ exc2[i] = (Random(&(st->cng_seed)) >> 4);
+ }
- /* gain = level / sqrt(ener) * sqrt(L_FRAME) */
+ /* gain = level / sqrt(ener) * sqrt(L_FRAME) */
- /* energy of generated excitation */
- ener32 = Dot_product12(exc2, exc2, L_FRAME, &exp);
+ /* energy of generated excitation */
+ ener32 = Dot_product12(exc2, exc2, L_FRAME, &exp);
- Isqrt_n(&ener32, &exp);
+ Isqrt_n(&ener32, &exp);
- gain = extract_h(ener32);
+ gain = extract_h(ener32);
- gain = mult(level, gain); /* gain in Q15 */
+ gain = mult(level, gain); /* gain in Q15 */
- exp = add(exp0, exp);
+ exp = add(exp0, exp);
- /* Multiply by sqrt(L_FRAME)=16, i.e. shift left by 4 */
- exp += 4;
+ /* Multiply by sqrt(L_FRAME)=16, i.e. shift left by 4 */
+ exp += 4;
- for (i = 0; i < L_FRAME; i++)
- {
- tmp = mult(exc2[i], gain); /* Q0 * Q15 */
- exc2[i] = shl(tmp, exp);
- }
+ for (i = 0; i < L_FRAME; i++)
+ {
+ tmp = mult(exc2[i], gain); /* Q0 * Q15 */
+ exc2[i] = shl(tmp, exp);
+ }
- return 0;
+ return 0;
}
/**************************************************************************
@@ -291,45 +291,45 @@
*
**************************************************************************/
Word16 dtx_buffer(
- dtx_encState * st, /* i/o : State struct */
- Word16 isf_new[], /* i : isf vector */
- Word32 enr, /* i : residual energy (in L_FRAME) */
- Word16 codec_mode
- )
+ dtx_encState * st, /* i/o : State struct */
+ Word16 isf_new[], /* i : isf vector */
+ Word32 enr, /* i : residual energy (in L_FRAME) */
+ Word16 codec_mode
+ )
{
- Word16 log_en;
+ Word16 log_en;
- Word16 log_en_e;
- Word16 log_en_m;
- st->hist_ptr = add(st->hist_ptr, 1);
- if(st->hist_ptr == DTX_HIST_SIZE)
- {
- st->hist_ptr = 0;
- }
- /* copy lsp vector into buffer */
- Copy(isf_new, &st->isf_hist[st->hist_ptr * M], M);
+ Word16 log_en_e;
+ Word16 log_en_m;
+ st->hist_ptr = add(st->hist_ptr, 1);
+ if(st->hist_ptr == DTX_HIST_SIZE)
+ {
+ st->hist_ptr = 0;
+ }
+ /* copy lsp vector into buffer */
+ Copy(isf_new, &st->isf_hist[st->hist_ptr * M], M);
- /* log_en = (float)log10(enr*0.0059322)/(float)log10(2.0f); */
- Log2(enr, &log_en_e, &log_en_m);
+ /* log_en = (float)log10(enr*0.0059322)/(float)log10(2.0f); */
+ Log2(enr, &log_en_e, &log_en_m);
- /* convert exponent and mantissa to Word16 Q7. Q7 is used to simplify averaging in dtx_enc */
- log_en = shl(log_en_e, 7); /* Q7 */
- log_en = add(log_en, shr(log_en_m, 15 - 7));
+ /* convert exponent and mantissa to Word16 Q7. Q7 is used to simplify averaging in dtx_enc */
+ log_en = shl(log_en_e, 7); /* Q7 */
+ log_en = add(log_en, shr(log_en_m, 15 - 7));
- /* Find energy per sample by multiplying with 0.0059322, i.e subtract log2(1/0.0059322) = 7.39722 The
- * constant 0.0059322 takes into account windowings and analysis length from autocorrelation
- * computations; 7.39722 in Q7 = 947 */
- /* Subtract 3 dB = 0.99658 in log2(E) = 127 in Q7. */
- /* log_en = sub( log_en, 947 + en_adjust[codec_mode] ); */
+ /* Find energy per sample by multiplying with 0.0059322, i.e subtract log2(1/0.0059322) = 7.39722 The
+ * constant 0.0059322 takes into account windowings and analysis length from autocorrelation
+ * computations; 7.39722 in Q7 = 947 */
+ /* Subtract 3 dB = 0.99658 in log2(E) = 127 in Q7. */
+ /* log_en = sub( log_en, 947 + en_adjust[codec_mode] ); */
- /* Find energy per sample (divide by L_FRAME=256), i.e subtract log2(256) = 8.0 (1024 in Q7) */
- /* Subtract 3 dB = 0.99658 in log2(E) = 127 in Q7. */
+ /* Find energy per sample (divide by L_FRAME=256), i.e subtract log2(256) = 8.0 (1024 in Q7) */
+ /* Subtract 3 dB = 0.99658 in log2(E) = 127 in Q7. */
- log_en = sub(log_en, add(1024, en_adjust[codec_mode]));
+ log_en = sub(log_en, add(1024, en_adjust[codec_mode]));
- /* Insert into the buffer */
- st->log_en_hist[st->hist_ptr] = log_en;
- return 0;
+ /* Insert into the buffer */
+ st->log_en_hist[st->hist_ptr] = log_en;
+ return 0;
}
/**************************************************************************
@@ -339,267 +339,267 @@
* the decoding side.
**************************************************************************/
void tx_dtx_handler(dtx_encState * st, /* i/o : State struct */
- Word16 vad_flag, /* i : vad decision */
- Word16 * usedMode /* i/o : mode changed or not */
- )
+ Word16 vad_flag, /* i : vad decision */
+ Word16 * usedMode /* i/o : mode changed or not */
+ )
{
- /* this state machine is in synch with the GSMEFR txDtx machine */
- st->decAnaElapsedCount = add(st->decAnaElapsedCount, 1);
+ /* this state machine is in synch with the GSMEFR txDtx machine */
+ st->decAnaElapsedCount = add(st->decAnaElapsedCount, 1);
- if (vad_flag != 0)
- {
- st->dtxHangoverCount = DTX_HANG_CONST;
- } else
- { /* non-speech */
- if (st->dtxHangoverCount == 0)
- { /* out of decoder analysis hangover */
- st->decAnaElapsedCount = 0;
- *usedMode = MRDTX;
- } else
- { /* in possible analysis hangover */
- st->dtxHangoverCount = sub(st->dtxHangoverCount, 1);
+ if (vad_flag != 0)
+ {
+ st->dtxHangoverCount = DTX_HANG_CONST;
+ } else
+ { /* non-speech */
+ if (st->dtxHangoverCount == 0)
+ { /* out of decoder analysis hangover */
+ st->decAnaElapsedCount = 0;
+ *usedMode = MRDTX;
+ } else
+ { /* in possible analysis hangover */
+ st->dtxHangoverCount = sub(st->dtxHangoverCount, 1);
- /* decAnaElapsedCount + dtxHangoverCount < DTX_ELAPSED_FRAMES_THRESH */
- if (sub(add(st->decAnaElapsedCount, st->dtxHangoverCount),
- DTX_ELAPSED_FRAMES_THRESH) < 0)
- {
- *usedMode = MRDTX;
- /* if short time since decoder update, do not add extra HO */
- }
- /* else override VAD and stay in speech mode *usedMode and add extra hangover */
- }
- }
+ /* decAnaElapsedCount + dtxHangoverCount < DTX_ELAPSED_FRAMES_THRESH */
+ if (sub(add(st->decAnaElapsedCount, st->dtxHangoverCount),
+ DTX_ELAPSED_FRAMES_THRESH) < 0)
+ {
+ *usedMode = MRDTX;
+ /* if short time since decoder update, do not add extra HO */
+ }
+ /* else override VAD and stay in speech mode *usedMode and add extra hangover */
+ }
+ }
- return;
+ return;
}
static void aver_isf_history(
- Word16 isf_old[],
- Word16 indices[],
- Word32 isf_aver[]
- )
+ Word16 isf_old[],
+ Word16 indices[],
+ Word32 isf_aver[]
+ )
{
- Word32 i, j, k;
- Word16 isf_tmp[2 * M];
- Word32 L_tmp;
+ Word32 i, j, k;
+ Word16 isf_tmp[2 * M];
+ Word32 L_tmp;
- /* Memorize in isf_tmp[][] the ISF vectors to be replaced by */
- /* the median ISF vector prior to the averaging */
- for (k = 0; k < 2; k++)
- {
- if ((indices[k] + 1) != 0)
- {
- for (i = 0; i < M; i++)
- {
- isf_tmp[k * M + i] = isf_old[indices[k] * M + i];
- isf_old[indices[k] * M + i] = isf_old[indices[2] * M + i];
- }
- }
- }
+ /* Memorize in isf_tmp[][] the ISF vectors to be replaced by */
+ /* the median ISF vector prior to the averaging */
+ for (k = 0; k < 2; k++)
+ {
+ if ((indices[k] + 1) != 0)
+ {
+ for (i = 0; i < M; i++)
+ {
+ isf_tmp[k * M + i] = isf_old[indices[k] * M + i];
+ isf_old[indices[k] * M + i] = isf_old[indices[2] * M + i];
+ }
+ }
+ }
- /* Perform the ISF averaging */
- for (j = 0; j < M; j++)
- {
- L_tmp = 0;
+ /* Perform the ISF averaging */
+ for (j = 0; j < M; j++)
+ {
+ L_tmp = 0;
- for (i = 0; i < DTX_HIST_SIZE; i++)
- {
- L_tmp = L_add(L_tmp, L_deposit_l(isf_old[i * M + j]));
- }
- isf_aver[j] = L_tmp;
- }
+ for (i = 0; i < DTX_HIST_SIZE; i++)
+ {
+ L_tmp = L_add(L_tmp, L_deposit_l(isf_old[i * M + j]));
+ }
+ isf_aver[j] = L_tmp;
+ }
- /* Retrieve from isf_tmp[][] the ISF vectors saved prior to averaging */
- for (k = 0; k < 2; k++)
- {
- if ((indices[k] + 1) != 0)
- {
- for (i = 0; i < M; i++)
- {
- isf_old[indices[k] * M + i] = isf_tmp[k * M + i];
- }
- }
- }
+ /* Retrieve from isf_tmp[][] the ISF vectors saved prior to averaging */
+ for (k = 0; k < 2; k++)
+ {
+ if ((indices[k] + 1) != 0)
+ {
+ for (i = 0; i < M; i++)
+ {
+ isf_old[indices[k] * M + i] = isf_tmp[k * M + i];
+ }
+ }
+ }
- return;
+ return;
}
static void find_frame_indices(
- Word16 isf_old_tx[],
- Word16 indices[],
- dtx_encState * st
- )
+ Word16 isf_old_tx[],
+ Word16 indices[],
+ dtx_encState * st
+ )
{
- Word32 L_tmp, summin, summax, summax2nd;
- Word16 i, j, tmp;
- Word16 ptr;
+ Word32 L_tmp, summin, summax, summax2nd;
+ Word16 i, j, tmp;
+ Word16 ptr;
- /* Remove the effect of the oldest frame from the column */
- /* sum sumD[0..DTX_HIST_SIZE-1]. sumD[DTX_HIST_SIZE] is */
- /* not updated since it will be removed later. */
+ /* Remove the effect of the oldest frame from the column */
+ /* sum sumD[0..DTX_HIST_SIZE-1]. sumD[DTX_HIST_SIZE] is */
+ /* not updated since it will be removed later. */
- tmp = DTX_HIST_SIZE_MIN_ONE;
- j = -1;
- for (i = 0; i < DTX_HIST_SIZE_MIN_ONE; i++)
- {
- j = add(j, tmp);
- st->sumD[i] = L_sub(st->sumD[i], st->D[j]);
- tmp = sub(tmp, 1);
- }
+ tmp = DTX_HIST_SIZE_MIN_ONE;
+ j = -1;
+ for (i = 0; i < DTX_HIST_SIZE_MIN_ONE; i++)
+ {
+ j = add(j, tmp);
+ st->sumD[i] = L_sub(st->sumD[i], st->D[j]);
+ tmp = sub(tmp, 1);
+ }
- /* Shift the column sum sumD. The element sumD[DTX_HIST_SIZE-1] */
- /* corresponding to the oldest frame is removed. The sum of */
- /* the distances between the latest isf and other isfs, */
- /* i.e. the element sumD[0], will be computed during this call. */
- /* Hence this element is initialized to zero. */
+ /* Shift the column sum sumD. The element sumD[DTX_HIST_SIZE-1] */
+ /* corresponding to the oldest frame is removed. The sum of */
+ /* the distances between the latest isf and other isfs, */
+ /* i.e. the element sumD[0], will be computed during this call. */
+ /* Hence this element is initialized to zero. */
- for (i = DTX_HIST_SIZE_MIN_ONE; i > 0; i--)
- {
- st->sumD[i] = st->sumD[i - 1];
- }
- st->sumD[0] = 0;
+ for (i = DTX_HIST_SIZE_MIN_ONE; i > 0; i--)
+ {
+ st->sumD[i] = st->sumD[i - 1];
+ }
+ st->sumD[0] = 0;
- /* Remove the oldest frame from the distance matrix. */
- /* Note that the distance matrix is replaced by a one- */
- /* dimensional array to save static memory. */
+ /* Remove the oldest frame from the distance matrix. */
+ /* Note that the distance matrix is replaced by a one- */
+ /* dimensional array to save static memory. */
- tmp = 0;
- for (i = 27; i >= 12; i = (Word16) (i - tmp))
- {
- tmp = add(tmp, 1);
- for (j = tmp; j > 0; j--)
- {
- st->D[i - j + 1] = st->D[i - j - tmp];
- }
- }
+ tmp = 0;
+ for (i = 27; i >= 12; i = (Word16) (i - tmp))
+ {
+ tmp = add(tmp, 1);
+ for (j = tmp; j > 0; j--)
+ {
+ st->D[i - j + 1] = st->D[i - j - tmp];
+ }
+ }
- /* Compute the first column of the distance matrix D */
- /* (squared Euclidean distances from isf1[] to isf_old_tx[][]). */
+ /* Compute the first column of the distance matrix D */
+ /* (squared Euclidean distances from isf1[] to isf_old_tx[][]). */
- ptr = st->hist_ptr;
- for (i = 1; i < DTX_HIST_SIZE; i++)
- {
- /* Compute the distance between the latest isf and the other isfs. */
- ptr = sub(ptr, 1);
- if (ptr < 0)
- {
- ptr = DTX_HIST_SIZE_MIN_ONE;
- }
- L_tmp = 0;
- for (j = 0; j < M; j++)
- {
- tmp = sub(isf_old_tx[st->hist_ptr * M + j], isf_old_tx[ptr * M + j]);
- L_tmp = L_mac(L_tmp, tmp, tmp);
- }
- st->D[i - 1] = L_tmp;
+ ptr = st->hist_ptr;
+ for (i = 1; i < DTX_HIST_SIZE; i++)
+ {
+ /* Compute the distance between the latest isf and the other isfs. */
+ ptr = sub(ptr, 1);
+ if (ptr < 0)
+ {
+ ptr = DTX_HIST_SIZE_MIN_ONE;
+ }
+ L_tmp = 0;
+ for (j = 0; j < M; j++)
+ {
+ tmp = sub(isf_old_tx[st->hist_ptr * M + j], isf_old_tx[ptr * M + j]);
+ L_tmp = L_mac(L_tmp, tmp, tmp);
+ }
+ st->D[i - 1] = L_tmp;
- /* Update also the column sums. */
- st->sumD[0] = L_add(st->sumD[0], st->D[i - 1]);
- st->sumD[i] = L_add(st->sumD[i], st->D[i - 1]);
- }
+ /* Update also the column sums. */
+ st->sumD[0] = L_add(st->sumD[0], st->D[i - 1]);
+ st->sumD[i] = L_add(st->sumD[i], st->D[i - 1]);
+ }
- /* Find the minimum and maximum distances */
- summax = st->sumD[0];
- summin = st->sumD[0];
- indices[0] = 0;
- indices[2] = 0;
- for (i = 1; i < DTX_HIST_SIZE; i++)
- {
- if (L_sub(st->sumD[i], summax) > 0)
- {
- indices[0] = i;
- summax = st->sumD[i];
- }
- if (L_sub(st->sumD[i], summin) < 0)
- {
- indices[2] = i;
- summin = st->sumD[i];
- }
- }
+ /* Find the minimum and maximum distances */
+ summax = st->sumD[0];
+ summin = st->sumD[0];
+ indices[0] = 0;
+ indices[2] = 0;
+ for (i = 1; i < DTX_HIST_SIZE; i++)
+ {
+ if (L_sub(st->sumD[i], summax) > 0)
+ {
+ indices[0] = i;
+ summax = st->sumD[i];
+ }
+ if (L_sub(st->sumD[i], summin) < 0)
+ {
+ indices[2] = i;
+ summin = st->sumD[i];
+ }
+ }
- /* Find the second largest distance */
- summax2nd = -2147483647L;
- indices[1] = -1;
- for (i = 0; i < DTX_HIST_SIZE; i++)
- {
- if ((L_sub(st->sumD[i], summax2nd) > 0) && (sub(i, indices[0]) != 0))
- {
- indices[1] = i;
- summax2nd = st->sumD[i];
- }
- }
+ /* Find the second largest distance */
+ summax2nd = -2147483647L;
+ indices[1] = -1;
+ for (i = 0; i < DTX_HIST_SIZE; i++)
+ {
+ if ((L_sub(st->sumD[i], summax2nd) > 0) && (sub(i, indices[0]) != 0))
+ {
+ indices[1] = i;
+ summax2nd = st->sumD[i];
+ }
+ }
- for (i = 0; i < 3; i++)
- {
- indices[i] = sub(st->hist_ptr, indices[i]);
- if (indices[i] < 0)
- {
- indices[i] = add(indices[i], DTX_HIST_SIZE);
- }
- }
+ for (i = 0; i < 3; i++)
+ {
+ indices[i] = sub(st->hist_ptr, indices[i]);
+ if (indices[i] < 0)
+ {
+ indices[i] = add(indices[i], DTX_HIST_SIZE);
+ }
+ }
- /* If maximum distance/MED_THRESH is smaller than minimum distance */
- /* then the median ISF vector replacement is not performed */
- tmp = norm_l(summax);
- summax = (summax << tmp);
- summin = (summin << tmp);
- L_tmp = L_mult(voround(summax), INV_MED_THRESH);
- if(L_tmp <= summin)
- {
- indices[0] = -1;
- }
- /* If second largest distance/MED_THRESH is smaller than */
- /* minimum distance then the median ISF vector replacement is */
- /* not performed */
- summax2nd = L_shl(summax2nd, tmp);
- L_tmp = L_mult(voround(summax2nd), INV_MED_THRESH);
- if(L_tmp <= summin)
- {
- indices[1] = -1;
- }
- return;
+ /* If maximum distance/MED_THRESH is smaller than minimum distance */
+ /* then the median ISF vector replacement is not performed */
+ tmp = norm_l(summax);
+ summax = (summax << tmp);
+ summin = (summin << tmp);
+ L_tmp = L_mult(voround(summax), INV_MED_THRESH);
+ if(L_tmp <= summin)
+ {
+ indices[0] = -1;
+ }
+ /* If second largest distance/MED_THRESH is smaller than */
+ /* minimum distance then the median ISF vector replacement is */
+ /* not performed */
+ summax2nd = L_shl(summax2nd, tmp);
+ L_tmp = L_mult(voround(summax2nd), INV_MED_THRESH);
+ if(L_tmp <= summin)
+ {
+ indices[1] = -1;
+ }
+ return;
}
static Word16 dithering_control(
- dtx_encState * st
- )
+ dtx_encState * st
+ )
{
- Word16 tmp, mean, CN_dith, gain_diff;
- Word32 i, ISF_diff;
+ Word16 tmp, mean, CN_dith, gain_diff;
+ Word32 i, ISF_diff;
- /* determine how stationary the spectrum of background noise is */
- ISF_diff = 0;
- for (i = 0; i < 8; i++)
- {
- ISF_diff = L_add(ISF_diff, st->sumD[i]);
- }
- if ((ISF_diff >> 26) > 0)
- {
- CN_dith = 1;
- } else
- {
- CN_dith = 0;
- }
+ /* determine how stationary the spectrum of background noise is */
+ ISF_diff = 0;
+ for (i = 0; i < 8; i++)
+ {
+ ISF_diff = L_add(ISF_diff, st->sumD[i]);
+ }
+ if ((ISF_diff >> 26) > 0)
+ {
+ CN_dith = 1;
+ } else
+ {
+ CN_dith = 0;
+ }
- /* determine how stationary the energy of background noise is */
- mean = 0;
- for (i = 0; i < DTX_HIST_SIZE; i++)
- {
- mean = add(mean, st->log_en_hist[i]);
- }
- mean = (mean >> 3);
- gain_diff = 0;
- for (i = 0; i < DTX_HIST_SIZE; i++)
- {
- tmp = abs_s(sub(st->log_en_hist[i], mean));
- gain_diff = add(gain_diff, tmp);
- }
- if (gain_diff > GAIN_THR)
- {
- CN_dith = 1;
- }
- return CN_dith;
+ /* determine how stationary the energy of background noise is */
+ mean = 0;
+ for (i = 0; i < DTX_HIST_SIZE; i++)
+ {
+ mean = add(mean, st->log_en_hist[i]);
+ }
+ mean = (mean >> 3);
+ gain_diff = 0;
+ for (i = 0; i < DTX_HIST_SIZE; i++)
+ {
+ tmp = abs_s(sub(st->log_en_hist[i], mean));
+ gain_diff = add(gain_diff, tmp);
+ }
+ if (gain_diff > GAIN_THR)
+ {
+ CN_dith = 1;
+ }
+ return CN_dith;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/g_pitch.c b/media/libstagefright/codecs/amrwbenc/src/g_pitch.c
index d681f2e..98ee87e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/g_pitch.c
+++ b/media/libstagefright/codecs/amrwbenc/src/g_pitch.c
@@ -17,9 +17,9 @@
/***********************************************************************
* File: g_pitch.c *
* *
-* Description:Compute the gain of pitch. Result in Q12 *
-* if(gain < 0) gain = 0 *
-* if(gain > 1.2) gain = 1.2 *
+* Description:Compute the gain of pitch. Result in Q12 *
+* if(gain < 0) gain = 0 *
+* if(gain > 1.2) gain = 1.2 *
************************************************************************/
#include "typedef.h"
@@ -27,52 +27,52 @@
#include "math_op.h"
Word16 G_pitch( /* (o) Q14 : Gain of pitch lag saturated to 1.2 */
- Word16 xn[], /* (i) : Pitch target. */
- Word16 y1[], /* (i) : filtered adaptive codebook. */
- Word16 g_coeff[], /* : Correlations need for gain quantization. */
- Word16 L_subfr /* : Length of subframe. */
- )
+ Word16 xn[], /* (i) : Pitch target. */
+ Word16 y1[], /* (i) : filtered adaptive codebook. */
+ Word16 g_coeff[], /* : Correlations need for gain quantization. */
+ Word16 L_subfr /* : Length of subframe. */
+ )
{
- Word32 i;
- Word16 xy, yy, exp_xy, exp_yy, gain;
- /* Compute scalar product <y1[],y1[]> */
+ Word32 i;
+ Word16 xy, yy, exp_xy, exp_yy, gain;
+ /* Compute scalar product <y1[],y1[]> */
#ifdef ASM_OPT /* asm optimization branch */
- /* Compute scalar product <xn[],y1[]> */
- xy = extract_h(Dot_product12_asm(xn, y1, L_subfr, &exp_xy));
- yy = extract_h(Dot_product12_asm(y1, y1, L_subfr, &exp_yy));
+ /* Compute scalar product <xn[],y1[]> */
+ xy = extract_h(Dot_product12_asm(xn, y1, L_subfr, &exp_xy));
+ yy = extract_h(Dot_product12_asm(y1, y1, L_subfr, &exp_yy));
#else
- /* Compute scalar product <xn[],y1[]> */
- xy = extract_h(Dot_product12(xn, y1, L_subfr, &exp_xy));
- yy = extract_h(Dot_product12(y1, y1, L_subfr, &exp_yy));
+ /* Compute scalar product <xn[],y1[]> */
+ xy = extract_h(Dot_product12(xn, y1, L_subfr, &exp_xy));
+ yy = extract_h(Dot_product12(y1, y1, L_subfr, &exp_yy));
#endif
- g_coeff[0] = yy;
- g_coeff[1] = exp_yy;
- g_coeff[2] = xy;
- g_coeff[3] = exp_xy;
+ g_coeff[0] = yy;
+ g_coeff[1] = exp_yy;
+ g_coeff[2] = xy;
+ g_coeff[3] = exp_xy;
- /* If (xy < 0) gain = 0 */
- if (xy < 0)
- return ((Word16) 0);
+ /* If (xy < 0) gain = 0 */
+ if (xy < 0)
+ return ((Word16) 0);
- /* compute gain = xy/yy */
+ /* compute gain = xy/yy */
- xy >>= 1; /* Be sure xy < yy */
- gain = div_s(xy, yy);
+ xy >>= 1; /* Be sure xy < yy */
+ gain = div_s(xy, yy);
- i = exp_xy;
- i -= exp_yy;
+ i = exp_xy;
+ i -= exp_yy;
- gain = shl(gain, i);
+ gain = shl(gain, i);
- /* if (gain > 1.2) gain = 1.2 in Q14 */
- if(gain > 19661)
- {
- gain = 19661;
- }
- return (gain);
+ /* if (gain > 1.2) gain = 1.2 in Q14 */
+ if(gain > 19661)
+ {
+ gain = 19661;
+ }
+ return (gain);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/gpclip.c b/media/libstagefright/codecs/amrwbenc/src/gpclip.c
index 800b3f9..4ce3daa 100644
--- a/media/libstagefright/codecs/amrwbenc/src/gpclip.c
+++ b/media/libstagefright/codecs/amrwbenc/src/gpclip.c
@@ -35,75 +35,75 @@
void Init_gp_clip(
- Word16 mem[] /* (o) : memory of gain of pitch clipping algorithm */
- )
+ Word16 mem[] /* (o) : memory of gain of pitch clipping algorithm */
+ )
{
- mem[0] = DIST_ISF_MAX;
- mem[1] = GAIN_PIT_MIN;
+ mem[0] = DIST_ISF_MAX;
+ mem[1] = GAIN_PIT_MIN;
}
Word16 Gp_clip(
- Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
- )
+ Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
+ )
{
- Word16 clip = 0;
- if ((mem[0] < DIST_ISF_THRES) && (mem[1] > GAIN_PIT_THRES))
- clip = 1;
+ Word16 clip = 0;
+ if ((mem[0] < DIST_ISF_THRES) && (mem[1] > GAIN_PIT_THRES))
+ clip = 1;
- return (clip);
+ return (clip);
}
void Gp_clip_test_isf(
- Word16 isf[], /* (i) : isf values (in frequency domain) */
- Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
- )
+ Word16 isf[], /* (i) : isf values (in frequency domain) */
+ Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
+ )
{
- Word16 dist, dist_min;
- Word32 i;
+ Word16 dist, dist_min;
+ Word32 i;
- dist_min = vo_sub(isf[1], isf[0]);
+ dist_min = vo_sub(isf[1], isf[0]);
- for (i = 2; i < M - 1; i++)
- {
- dist = vo_sub(isf[i], isf[i - 1]);
- if(dist < dist_min)
- {
- dist_min = dist;
- }
- }
+ for (i = 2; i < M - 1; i++)
+ {
+ dist = vo_sub(isf[i], isf[i - 1]);
+ if(dist < dist_min)
+ {
+ dist_min = dist;
+ }
+ }
- dist = extract_h(L_mac(vo_L_mult(26214, mem[0]), 6554, dist_min));
+ dist = extract_h(L_mac(vo_L_mult(26214, mem[0]), 6554, dist_min));
- if (dist > DIST_ISF_MAX)
- {
- dist = DIST_ISF_MAX;
- }
- mem[0] = dist;
+ if (dist > DIST_ISF_MAX)
+ {
+ dist = DIST_ISF_MAX;
+ }
+ mem[0] = dist;
- return;
+ return;
}
void Gp_clip_test_gain_pit(
- Word16 gain_pit, /* (i) Q14 : gain of quantized pitch */
- Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
- )
+ Word16 gain_pit, /* (i) Q14 : gain of quantized pitch */
+ Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
+ )
{
- Word16 gain;
- Word32 L_tmp;
- L_tmp = (29491 * mem[1])<<1;
- L_tmp += (3277 * gain_pit)<<1;
+ Word16 gain;
+ Word32 L_tmp;
+ L_tmp = (29491 * mem[1])<<1;
+ L_tmp += (3277 * gain_pit)<<1;
- gain = extract_h(L_tmp);
+ gain = extract_h(L_tmp);
- if(gain < GAIN_PIT_MIN)
- {
- gain = GAIN_PIT_MIN;
- }
- mem[1] = gain;
- return;
+ if(gain < GAIN_PIT_MIN)
+ {
+ gain = GAIN_PIT_MIN;
+ }
+ mem[1] = gain;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/homing.c b/media/libstagefright/codecs/amrwbenc/src/homing.c
index 565040f..a96e7db 100644
--- a/media/libstagefright/codecs/amrwbenc/src/homing.c
+++ b/media/libstagefright/codecs/amrwbenc/src/homing.c
@@ -29,18 +29,18 @@
Word16 encoder_homing_frame_test(Word16 input_frame[])
{
- Word32 i;
- Word16 j = 0;
+ Word32 i;
+ Word16 j = 0;
- /* check 320 input samples for matching EHF_MASK: defined in e_homing.h */
- for (i = 0; i < L_FRAME16k; i++)
- {
- j = (Word16) (input_frame[i] ^ EHF_MASK);
+ /* check 320 input samples for matching EHF_MASK: defined in e_homing.h */
+ for (i = 0; i < L_FRAME16k; i++)
+ {
+ j = (Word16) (input_frame[i] ^ EHF_MASK);
- if (j)
- break;
- }
+ if (j)
+ break;
+ }
- return (Word16) (!j);
+ return (Word16) (!j);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/hp400.c b/media/libstagefright/codecs/amrwbenc/src/hp400.c
index a6f9701..c658a92 100644
--- a/media/libstagefright/codecs/amrwbenc/src/hp400.c
+++ b/media/libstagefright/codecs/amrwbenc/src/hp400.c
@@ -50,56 +50,56 @@
void Init_HP400_12k8(Word16 mem[])
{
- Set_zero(mem, 6);
+ Set_zero(mem, 6);
}
void HP400_12k8(
- Word16 signal[], /* input signal / output is divided by 16 */
- Word16 lg, /* lenght of signal */
- Word16 mem[] /* filter memory [6] */
- )
+ Word16 signal[], /* input signal / output is divided by 16 */
+ Word16 lg, /* lenght of signal */
+ Word16 mem[] /* filter memory [6] */
+ )
{
- Word16 x2;
- Word16 y2_hi, y2_lo, y1_hi, y1_lo, x0, x1;
- Word32 L_tmp;
- Word32 num;
- y2_hi = *mem++;
- y2_lo = *mem++;
- y1_hi = *mem++;
- y1_lo = *mem++;
- x0 = *mem++;
- x1 = *mem;
- num = (Word32)lg;
- do
- {
- x2 = x1;
- x1 = x0;
- x0 = *signal;
- /* y[i] = b[0]*x[i] + b[1]*x[i-1] + b140[2]*x[i-2] */
- /* + a[1]*y[i-1] + a[2] * y[i-2]; */
- L_tmp = 8192L; /* rounding to maximise precision */
- L_tmp += y1_lo * a[1];
- L_tmp += y2_lo * a[2];
- L_tmp = L_tmp >> 14;
- L_tmp += (y1_hi * a[1] + y2_hi * a[2] + (x0 + x2)* b[0] + x1 * b[1]) << 1;
- L_tmp <<= 1; /* coeff Q12 --> Q13 */
- y2_hi = y1_hi;
- y2_lo = y1_lo;
- y1_hi = (Word16)(L_tmp>>16);
- y1_lo = (Word16)((L_tmp & 0xffff)>>1);
+ Word16 x2;
+ Word16 y2_hi, y2_lo, y1_hi, y1_lo, x0, x1;
+ Word32 L_tmp;
+ Word32 num;
+ y2_hi = *mem++;
+ y2_lo = *mem++;
+ y1_hi = *mem++;
+ y1_lo = *mem++;
+ x0 = *mem++;
+ x1 = *mem;
+ num = (Word32)lg;
+ do
+ {
+ x2 = x1;
+ x1 = x0;
+ x0 = *signal;
+ /* y[i] = b[0]*x[i] + b[1]*x[i-1] + b140[2]*x[i-2] */
+ /* + a[1]*y[i-1] + a[2] * y[i-2]; */
+ L_tmp = 8192L; /* rounding to maximise precision */
+ L_tmp += y1_lo * a[1];
+ L_tmp += y2_lo * a[2];
+ L_tmp = L_tmp >> 14;
+ L_tmp += (y1_hi * a[1] + y2_hi * a[2] + (x0 + x2)* b[0] + x1 * b[1]) << 1;
+ L_tmp <<= 1; /* coeff Q12 --> Q13 */
+ y2_hi = y1_hi;
+ y2_lo = y1_lo;
+ y1_hi = (Word16)(L_tmp>>16);
+ y1_lo = (Word16)((L_tmp & 0xffff)>>1);
- /* signal is divided by 16 to avoid overflow in energy computation */
- *signal++ = (L_tmp + 0x8000) >> 16;
- }while(--num !=0);
+ /* signal is divided by 16 to avoid overflow in energy computation */
+ *signal++ = (L_tmp + 0x8000) >> 16;
+ }while(--num !=0);
- *mem-- = x1;
- *mem-- = x0;
- *mem-- = y1_lo;
- *mem-- = y1_hi;
- *mem-- = y2_lo;
- *mem = y2_hi;
- return;
+ *mem-- = x1;
+ *mem-- = x0;
+ *mem-- = y1_lo;
+ *mem-- = y1_hi;
+ *mem-- = y2_lo;
+ *mem = y2_hi;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/hp50.c b/media/libstagefright/codecs/amrwbenc/src/hp50.c
index c1c7b83..807d672 100644
--- a/media/libstagefright/codecs/amrwbenc/src/hp50.c
+++ b/media/libstagefright/codecs/amrwbenc/src/hp50.c
@@ -17,7 +17,7 @@
/***********************************************************************
* File: hp50.c *
* *
-* Description: *
+* Description: *
* 2nd order high pass filter with cut off frequency at 31 Hz. *
* Designed with cheby2 function in MATLAB. *
* Optimized for fixed-point to get the following frequency response: *
@@ -51,56 +51,56 @@
void Init_HP50_12k8(Word16 mem[])
{
- Set_zero(mem, 6);
+ Set_zero(mem, 6);
}
void HP50_12k8(
- Word16 signal[], /* input/output signal */
- Word16 lg, /* lenght of signal */
- Word16 mem[] /* filter memory [6] */
- )
+ Word16 signal[], /* input/output signal */
+ Word16 lg, /* lenght of signal */
+ Word16 mem[] /* filter memory [6] */
+ )
{
- Word16 x2;
- Word16 y2_hi, y2_lo, y1_hi, y1_lo, x0, x1;
- Word32 L_tmp;
- Word32 num;
+ Word16 x2;
+ Word16 y2_hi, y2_lo, y1_hi, y1_lo, x0, x1;
+ Word32 L_tmp;
+ Word32 num;
- y2_hi = *mem++;
- y2_lo = *mem++;
- y1_hi = *mem++;
- y1_lo = *mem++;
- x0 = *mem++;
- x1 = *mem;
- num = (Word32)lg;
- do
- {
- x2 = x1;
- x1 = x0;
- x0 = *signal;
- /* y[i] = b[0]*x[i] + b[1]*x[i-1] + b140[2]*x[i-2] */
- /* + a[1]*y[i-1] + a[2] * y[i-2]; */
- L_tmp = 8192 ; /* rounding to maximise precision */
- L_tmp += y1_lo * a[1];
- L_tmp += y2_lo * a[2];
- L_tmp = L_tmp >> 14;
- L_tmp += (y1_hi * a[1] + y2_hi * a[2] + (x0 + x2) * b[0] + x1 * b[1]) << 1;
- L_tmp <<= 2; /* coeff Q12 --> Q13 */
- y2_hi = y1_hi;
- y2_lo = y1_lo;
- y1_hi = (Word16)(L_tmp>>16);
- y1_lo = (Word16)((L_tmp & 0xffff)>>1);
- *signal++ = extract_h((L_add((L_tmp<<1), 0x8000)));
- }while(--num !=0);
+ y2_hi = *mem++;
+ y2_lo = *mem++;
+ y1_hi = *mem++;
+ y1_lo = *mem++;
+ x0 = *mem++;
+ x1 = *mem;
+ num = (Word32)lg;
+ do
+ {
+ x2 = x1;
+ x1 = x0;
+ x0 = *signal;
+ /* y[i] = b[0]*x[i] + b[1]*x[i-1] + b140[2]*x[i-2] */
+ /* + a[1]*y[i-1] + a[2] * y[i-2]; */
+ L_tmp = 8192 ; /* rounding to maximise precision */
+ L_tmp += y1_lo * a[1];
+ L_tmp += y2_lo * a[2];
+ L_tmp = L_tmp >> 14;
+ L_tmp += (y1_hi * a[1] + y2_hi * a[2] + (x0 + x2) * b[0] + x1 * b[1]) << 1;
+ L_tmp <<= 2; /* coeff Q12 --> Q13 */
+ y2_hi = y1_hi;
+ y2_lo = y1_lo;
+ y1_hi = (Word16)(L_tmp>>16);
+ y1_lo = (Word16)((L_tmp & 0xffff)>>1);
+ *signal++ = extract_h((L_add((L_tmp<<1), 0x8000)));
+ }while(--num !=0);
- *mem-- = x1;
- *mem-- = x0;
- *mem-- = y1_lo;
- *mem-- = y1_hi;
- *mem-- = y2_lo;
- *mem-- = y2_hi;
+ *mem-- = x1;
+ *mem-- = x0;
+ *mem-- = y1_lo;
+ *mem-- = y1_hi;
+ *mem-- = y2_lo;
+ *mem-- = y2_hi;
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/hp6k.c b/media/libstagefright/codecs/amrwbenc/src/hp6k.c
index 8e66eb0..0baf612 100644
--- a/media/libstagefright/codecs/amrwbenc/src/hp6k.c
+++ b/media/libstagefright/codecs/amrwbenc/src/hp6k.c
@@ -17,10 +17,10 @@
/***********************************************************************
* File: hp6k.c *
* *
-* Description:15th order band pass 6kHz to 7kHz FIR filter *
+* Description:15th order band pass 6kHz to 7kHz FIR filter *
* frequency: 4kHz 5kHz 5.5kHz 6kHz 6.5kHz 7kHz 7.5kHz 8kHz *
-* dB loss: -60dB -45dB -13dB -3dB 0dB -3dB -13dB -45dB *
-* *
+* dB loss: -60dB -45dB -13dB -3dB 0dB -3dB -13dB -45dB *
+* *
************************************************************************/
#include "typedef.h"
@@ -34,58 +34,58 @@
Word16 fir_6k_7k[L_FIR] =
{
- -32, 47, 32, -27, -369,
- 1122, -1421, 0, 3798, -8880,
- 12349, -10984, 3548, 7766, -18001,
- 22118, -18001, 7766, 3548, -10984,
- 12349, -8880, 3798, 0, -1421,
- 1122, -369, -27, 32, 47,
- -32
+ -32, 47, 32, -27, -369,
+ 1122, -1421, 0, 3798, -8880,
+ 12349, -10984, 3548, 7766, -18001,
+ 22118, -18001, 7766, 3548, -10984,
+ 12349, -8880, 3798, 0, -1421,
+ 1122, -369, -27, 32, 47,
+ -32
};
void Init_Filt_6k_7k(Word16 mem[]) /* mem[30] */
{
- Set_zero(mem, L_FIR - 1);
- return;
+ Set_zero(mem, L_FIR - 1);
+ return;
}
void Filt_6k_7k(
- Word16 signal[], /* input: signal */
- Word16 lg, /* input: length of input */
- Word16 mem[] /* in/out: memory (size=30) */
- )
+ Word16 signal[], /* input: signal */
+ Word16 lg, /* input: length of input */
+ Word16 mem[] /* in/out: memory (size=30) */
+ )
{
- Word16 x[L_SUBFR16k + (L_FIR - 1)];
- Word32 i, L_tmp;
+ Word16 x[L_SUBFR16k + (L_FIR - 1)];
+ Word32 i, L_tmp;
- Copy(mem, x, L_FIR - 1);
- for (i = lg - 1; i >= 0; i--)
- {
- x[i + L_FIR - 1] = signal[i] >> 2; /* gain of filter = 4 */
- }
- for (i = 0; i < lg; i++)
- {
- L_tmp = (x[i] + x[i+ 30]) * fir_6k_7k[0];
- L_tmp += (x[i+1] + x[i + 29]) * fir_6k_7k[1];
- L_tmp += (x[i+2] + x[i + 28]) * fir_6k_7k[2];
- L_tmp += (x[i+3] + x[i + 27]) * fir_6k_7k[3];
- L_tmp += (x[i+4] + x[i + 26]) * fir_6k_7k[4];
- L_tmp += (x[i+5] + x[i + 25]) * fir_6k_7k[5];
- L_tmp += (x[i+6] + x[i + 24]) * fir_6k_7k[6];
- L_tmp += (x[i+7] + x[i + 23]) * fir_6k_7k[7];
- L_tmp += (x[i+8] + x[i + 22]) * fir_6k_7k[8];
- L_tmp += (x[i+9] + x[i + 21]) * fir_6k_7k[9];
- L_tmp += (x[i+10] + x[i + 20]) * fir_6k_7k[10];
- L_tmp += (x[i+11] + x[i + 19]) * fir_6k_7k[11];
- L_tmp += (x[i+12] + x[i + 18]) * fir_6k_7k[12];
- L_tmp += (x[i+13] + x[i + 17]) * fir_6k_7k[13];
- L_tmp += (x[i+14] + x[i + 16]) * fir_6k_7k[14];
- L_tmp += (x[i+15]) * fir_6k_7k[15];
- signal[i] = (L_tmp + 0x4000) >> 15;
- }
+ Copy(mem, x, L_FIR - 1);
+ for (i = lg - 1; i >= 0; i--)
+ {
+ x[i + L_FIR - 1] = signal[i] >> 2; /* gain of filter = 4 */
+ }
+ for (i = 0; i < lg; i++)
+ {
+ L_tmp = (x[i] + x[i+ 30]) * fir_6k_7k[0];
+ L_tmp += (x[i+1] + x[i + 29]) * fir_6k_7k[1];
+ L_tmp += (x[i+2] + x[i + 28]) * fir_6k_7k[2];
+ L_tmp += (x[i+3] + x[i + 27]) * fir_6k_7k[3];
+ L_tmp += (x[i+4] + x[i + 26]) * fir_6k_7k[4];
+ L_tmp += (x[i+5] + x[i + 25]) * fir_6k_7k[5];
+ L_tmp += (x[i+6] + x[i + 24]) * fir_6k_7k[6];
+ L_tmp += (x[i+7] + x[i + 23]) * fir_6k_7k[7];
+ L_tmp += (x[i+8] + x[i + 22]) * fir_6k_7k[8];
+ L_tmp += (x[i+9] + x[i + 21]) * fir_6k_7k[9];
+ L_tmp += (x[i+10] + x[i + 20]) * fir_6k_7k[10];
+ L_tmp += (x[i+11] + x[i + 19]) * fir_6k_7k[11];
+ L_tmp += (x[i+12] + x[i + 18]) * fir_6k_7k[12];
+ L_tmp += (x[i+13] + x[i + 17]) * fir_6k_7k[13];
+ L_tmp += (x[i+14] + x[i + 16]) * fir_6k_7k[14];
+ L_tmp += (x[i+15]) * fir_6k_7k[15];
+ signal[i] = (L_tmp + 0x4000) >> 15;
+ }
- Copy(x + lg, mem, L_FIR - 1);
+ Copy(x + lg, mem, L_FIR - 1);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/hp_wsp.c b/media/libstagefright/codecs/amrwbenc/src/hp_wsp.c
index bc1ec49..f0347cb 100644
--- a/media/libstagefright/codecs/amrwbenc/src/hp_wsp.c
+++ b/media/libstagefright/codecs/amrwbenc/src/hp_wsp.c
@@ -48,101 +48,101 @@
/* Initialization of static values */
void Init_Hp_wsp(Word16 mem[])
{
- Set_zero(mem, 9);
+ Set_zero(mem, 9);
- return;
+ return;
}
void scale_mem_Hp_wsp(Word16 mem[], Word16 exp)
{
- Word32 i;
- Word32 L_tmp;
+ Word32 i;
+ Word32 L_tmp;
- for (i = 0; i < 6; i += 2)
- {
- L_tmp = ((mem[i] << 16) + (mem[i + 1]<<1));
- L_tmp = L_shl(L_tmp, exp);
- mem[i] = L_tmp >> 16;
- mem[i + 1] = (L_tmp & 0xffff)>>1;
- }
+ for (i = 0; i < 6; i += 2)
+ {
+ L_tmp = ((mem[i] << 16) + (mem[i + 1]<<1));
+ L_tmp = L_shl(L_tmp, exp);
+ mem[i] = L_tmp >> 16;
+ mem[i + 1] = (L_tmp & 0xffff)>>1;
+ }
- for (i = 6; i < 9; i++)
- {
- L_tmp = L_deposit_h(mem[i]); /* x[i] */
- L_tmp = L_shl(L_tmp, exp);
- mem[i] = vo_round(L_tmp);
- }
+ for (i = 6; i < 9; i++)
+ {
+ L_tmp = L_deposit_h(mem[i]); /* x[i] */
+ L_tmp = L_shl(L_tmp, exp);
+ mem[i] = vo_round(L_tmp);
+ }
- return;
+ return;
}
void Hp_wsp(
- Word16 wsp[], /* i : wsp[] signal */
- Word16 hp_wsp[], /* o : hypass wsp[] */
- Word16 lg, /* i : lenght of signal */
- Word16 mem[] /* i/o : filter memory [9] */
- )
+ Word16 wsp[], /* i : wsp[] signal */
+ Word16 hp_wsp[], /* o : hypass wsp[] */
+ Word16 lg, /* i : lenght of signal */
+ Word16 mem[] /* i/o : filter memory [9] */
+ )
{
- Word16 x0, x1, x2, x3;
- Word16 y3_hi, y3_lo, y2_hi, y2_lo, y1_hi, y1_lo;
- Word32 i, L_tmp;
+ Word16 x0, x1, x2, x3;
+ Word16 y3_hi, y3_lo, y2_hi, y2_lo, y1_hi, y1_lo;
+ Word32 i, L_tmp;
- y3_hi = mem[0];
- y3_lo = mem[1];
- y2_hi = mem[2];
- y2_lo = mem[3];
- y1_hi = mem[4];
- y1_lo = mem[5];
- x0 = mem[6];
- x1 = mem[7];
- x2 = mem[8];
+ y3_hi = mem[0];
+ y3_lo = mem[1];
+ y2_hi = mem[2];
+ y2_lo = mem[3];
+ y1_hi = mem[4];
+ y1_lo = mem[5];
+ x0 = mem[6];
+ x1 = mem[7];
+ x2 = mem[8];
- for (i = 0; i < lg; i++)
- {
- x3 = x2;
- x2 = x1;
- x1 = x0;
- x0 = wsp[i];
- /* y[i] = b[0]*x[i] + b[1]*x[i-1] + b140[2]*x[i-2] + b[3]*x[i-3] */
- /* + a[1]*y[i-1] + a[2] * y[i-2] + a[3]*y[i-3] */
+ for (i = 0; i < lg; i++)
+ {
+ x3 = x2;
+ x2 = x1;
+ x1 = x0;
+ x0 = wsp[i];
+ /* y[i] = b[0]*x[i] + b[1]*x[i-1] + b140[2]*x[i-2] + b[3]*x[i-3] */
+ /* + a[1]*y[i-1] + a[2] * y[i-2] + a[3]*y[i-3] */
- L_tmp = 16384L; /* rounding to maximise precision */
- L_tmp += (y1_lo * a[1])<<1;
- L_tmp += (y2_lo * a[2])<<1;
- L_tmp += (y3_lo * a[3])<<1;
- L_tmp = L_tmp >> 15;
- L_tmp += (y1_hi * a[1])<<1;
- L_tmp += (y2_hi * a[2])<<1;
- L_tmp += (y3_hi * a[3])<<1;
- L_tmp += (x0 * b[0])<<1;
- L_tmp += (x1 * b[1])<<1;
- L_tmp += (x2 * b[2])<<1;
- L_tmp += (x3 * b[3])<<1;
+ L_tmp = 16384L; /* rounding to maximise precision */
+ L_tmp += (y1_lo * a[1])<<1;
+ L_tmp += (y2_lo * a[2])<<1;
+ L_tmp += (y3_lo * a[3])<<1;
+ L_tmp = L_tmp >> 15;
+ L_tmp += (y1_hi * a[1])<<1;
+ L_tmp += (y2_hi * a[2])<<1;
+ L_tmp += (y3_hi * a[3])<<1;
+ L_tmp += (x0 * b[0])<<1;
+ L_tmp += (x1 * b[1])<<1;
+ L_tmp += (x2 * b[2])<<1;
+ L_tmp += (x3 * b[3])<<1;
- L_tmp = L_tmp << 2;
+ L_tmp = L_tmp << 2;
- y3_hi = y2_hi;
- y3_lo = y2_lo;
- y2_hi = y1_hi;
- y2_lo = y1_lo;
- y1_hi = L_tmp >> 16;
- y1_lo = (L_tmp & 0xffff) >>1;
+ y3_hi = y2_hi;
+ y3_lo = y2_lo;
+ y2_hi = y1_hi;
+ y2_lo = y1_lo;
+ y1_hi = L_tmp >> 16;
+ y1_lo = (L_tmp & 0xffff) >>1;
- hp_wsp[i] = (L_tmp + 0x4000)>>15;
- }
+ hp_wsp[i] = (L_tmp + 0x4000)>>15;
+ }
- mem[0] = y3_hi;
- mem[1] = y3_lo;
- mem[2] = y2_hi;
- mem[3] = y2_lo;
- mem[4] = y1_hi;
- mem[5] = y1_lo;
- mem[6] = x0;
- mem[7] = x1;
- mem[8] = x2;
+ mem[0] = y3_hi;
+ mem[1] = y3_lo;
+ mem[2] = y2_hi;
+ mem[3] = y2_lo;
+ mem[4] = y1_hi;
+ mem[5] = y1_lo;
+ mem[6] = x0;
+ mem[7] = x1;
+ mem[8] = x2;
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/int_lpc.c b/media/libstagefright/codecs/amrwbenc/src/int_lpc.c
index 1119bc7..3d8b8cb 100644
--- a/media/libstagefright/codecs/amrwbenc/src/int_lpc.c
+++ b/media/libstagefright/codecs/amrwbenc/src/int_lpc.c
@@ -30,36 +30,36 @@
void Int_isp(
- Word16 isp_old[], /* input : isps from past frame */
- Word16 isp_new[], /* input : isps from present frame */
- Word16 frac[], /* input : fraction for 3 first subfr (Q15) */
- Word16 Az[] /* output: LP coefficients in 4 subframes */
- )
+ Word16 isp_old[], /* input : isps from past frame */
+ Word16 isp_new[], /* input : isps from present frame */
+ Word16 frac[], /* input : fraction for 3 first subfr (Q15) */
+ Word16 Az[] /* output: LP coefficients in 4 subframes */
+ )
{
- Word32 i, k;
- Word16 fac_old, fac_new;
- Word16 isp[M];
- Word32 L_tmp;
+ Word32 i, k;
+ Word16 fac_old, fac_new;
+ Word16 isp[M];
+ Word32 L_tmp;
- for (k = 0; k < 3; k++)
- {
- fac_new = frac[k];
- fac_old = (32767 - fac_new) + 1; /* 1.0 - fac_new */
+ for (k = 0; k < 3; k++)
+ {
+ fac_new = frac[k];
+ fac_old = (32767 - fac_new) + 1; /* 1.0 - fac_new */
- for (i = 0; i < M; i++)
- {
- L_tmp = (isp_old[i] * fac_old)<<1;
- L_tmp += (isp_new[i] * fac_new)<<1;
- isp[i] = (L_tmp + 0x8000)>>16;
- }
- Isp_Az(isp, Az, M, 0);
- Az += MP1;
- }
+ for (i = 0; i < M; i++)
+ {
+ L_tmp = (isp_old[i] * fac_old)<<1;
+ L_tmp += (isp_new[i] * fac_new)<<1;
+ isp[i] = (L_tmp + 0x8000)>>16;
+ }
+ Isp_Az(isp, Az, M, 0);
+ Az += MP1;
+ }
- /* 4th subframe: isp_new (frac=1.0) */
- Isp_Az(isp_new, Az, M, 0);
+ /* 4th subframe: isp_new (frac=1.0) */
+ Isp_Az(isp_new, Az, M, 0);
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/isp_az.c b/media/libstagefright/codecs/amrwbenc/src/isp_az.c
index 30a8bbd..62e29e7 100644
--- a/media/libstagefright/codecs/amrwbenc/src/isp_az.c
+++ b/media/libstagefright/codecs/amrwbenc/src/isp_az.c
@@ -35,132 +35,132 @@
static void Get_isp_pol_16kHz(Word16 * isp, Word32 * f, Word16 n);
void Isp_Az(
- Word16 isp[], /* (i) Q15 : Immittance spectral pairs */
- Word16 a[], /* (o) Q12 : predictor coefficients (order = M) */
- Word16 m,
- Word16 adaptive_scaling /* (i) 0 : adaptive scaling disabled */
- /* 1 : adaptive scaling enabled */
- )
+ Word16 isp[], /* (i) Q15 : Immittance spectral pairs */
+ Word16 a[], /* (o) Q12 : predictor coefficients (order = M) */
+ Word16 m,
+ Word16 adaptive_scaling /* (i) 0 : adaptive scaling disabled */
+ /* 1 : adaptive scaling enabled */
+ )
{
- Word32 i, j;
- Word16 hi, lo;
- Word32 f1[NC16k + 1], f2[NC16k];
- Word16 nc;
- Word32 t0;
- Word16 q, q_sug;
- Word32 tmax;
+ Word32 i, j;
+ Word16 hi, lo;
+ Word32 f1[NC16k + 1], f2[NC16k];
+ Word16 nc;
+ Word32 t0;
+ Word16 q, q_sug;
+ Word32 tmax;
- nc = (m >> 1);
- if(nc > 8)
- {
- Get_isp_pol_16kHz(&isp[0], f1, nc);
- for (i = 0; i <= nc; i++)
- {
- f1[i] = f1[i] << 2;
- }
- } else
- Get_isp_pol(&isp[0], f1, nc);
+ nc = (m >> 1);
+ if(nc > 8)
+ {
+ Get_isp_pol_16kHz(&isp[0], f1, nc);
+ for (i = 0; i <= nc; i++)
+ {
+ f1[i] = f1[i] << 2;
+ }
+ } else
+ Get_isp_pol(&isp[0], f1, nc);
- if (nc > 8)
- {
- Get_isp_pol_16kHz(&isp[1], f2, (nc - 1));
- for (i = 0; i <= nc - 1; i++)
- {
- f2[i] = f2[i] << 2;
- }
- } else
- Get_isp_pol(&isp[1], f2, (nc - 1));
+ if (nc > 8)
+ {
+ Get_isp_pol_16kHz(&isp[1], f2, (nc - 1));
+ for (i = 0; i <= nc - 1; i++)
+ {
+ f2[i] = f2[i] << 2;
+ }
+ } else
+ Get_isp_pol(&isp[1], f2, (nc - 1));
- /*-----------------------------------------------------*
- * Multiply F2(z) by (1 - z^-2) *
- *-----------------------------------------------------*/
+ /*-----------------------------------------------------*
+ * Multiply F2(z) by (1 - z^-2) *
+ *-----------------------------------------------------*/
- for (i = (nc - 1); i > 1; i--)
- {
- f2[i] = vo_L_sub(f2[i], f2[i - 2]); /* f2[i] -= f2[i-2]; */
- }
+ for (i = (nc - 1); i > 1; i--)
+ {
+ f2[i] = vo_L_sub(f2[i], f2[i - 2]); /* f2[i] -= f2[i-2]; */
+ }
- /*----------------------------------------------------------*
- * Scale F1(z) by (1+isp[m-1]) and F2(z) by (1-isp[m-1]) *
- *----------------------------------------------------------*/
+ /*----------------------------------------------------------*
+ * Scale F1(z) by (1+isp[m-1]) and F2(z) by (1-isp[m-1]) *
+ *----------------------------------------------------------*/
- for (i = 0; i < nc; i++)
- {
- /* f1[i] *= (1.0 + isp[M-1]); */
+ for (i = 0; i < nc; i++)
+ {
+ /* f1[i] *= (1.0 + isp[M-1]); */
- hi = f1[i] >> 16;
- lo = (f1[i] & 0xffff)>>1;
+ hi = f1[i] >> 16;
+ lo = (f1[i] & 0xffff)>>1;
- t0 = Mpy_32_16(hi, lo, isp[m - 1]);
- f1[i] = vo_L_add(f1[i], t0);
+ t0 = Mpy_32_16(hi, lo, isp[m - 1]);
+ f1[i] = vo_L_add(f1[i], t0);
- /* f2[i] *= (1.0 - isp[M-1]); */
+ /* f2[i] *= (1.0 - isp[M-1]); */
- hi = f2[i] >> 16;
- lo = (f2[i] & 0xffff)>>1;
- t0 = Mpy_32_16(hi, lo, isp[m - 1]);
- f2[i] = vo_L_sub(f2[i], t0);
- }
+ hi = f2[i] >> 16;
+ lo = (f2[i] & 0xffff)>>1;
+ t0 = Mpy_32_16(hi, lo, isp[m - 1]);
+ f2[i] = vo_L_sub(f2[i], t0);
+ }
- /*-----------------------------------------------------*
- * A(z) = (F1(z)+F2(z))/2 *
- * F1(z) is symmetric and F2(z) is antisymmetric *
- *-----------------------------------------------------*/
+ /*-----------------------------------------------------*
+ * A(z) = (F1(z)+F2(z))/2 *
+ * F1(z) is symmetric and F2(z) is antisymmetric *
+ *-----------------------------------------------------*/
- /* a[0] = 1.0; */
- a[0] = 4096;
- tmax = 1;
- for (i = 1, j = m - 1; i < nc; i++, j--)
- {
- /* a[i] = 0.5*(f1[i] + f2[i]); */
+ /* a[0] = 1.0; */
+ a[0] = 4096;
+ tmax = 1;
+ for (i = 1, j = m - 1; i < nc; i++, j--)
+ {
+ /* a[i] = 0.5*(f1[i] + f2[i]); */
- t0 = vo_L_add(f1[i], f2[i]); /* f1[i] + f2[i] */
- tmax |= L_abs(t0);
- a[i] = (Word16)(vo_L_shr_r(t0, 12)); /* from Q23 to Q12 and * 0.5 */
+ t0 = vo_L_add(f1[i], f2[i]); /* f1[i] + f2[i] */
+ tmax |= L_abs(t0);
+ a[i] = (Word16)(vo_L_shr_r(t0, 12)); /* from Q23 to Q12 and * 0.5 */
- /* a[j] = 0.5*(f1[i] - f2[i]); */
+ /* a[j] = 0.5*(f1[i] - f2[i]); */
- t0 = vo_L_sub(f1[i], f2[i]); /* f1[i] - f2[i] */
- tmax |= L_abs(t0);
- a[j] = (Word16)(vo_L_shr_r(t0, 12)); /* from Q23 to Q12 and * 0.5 */
- }
+ t0 = vo_L_sub(f1[i], f2[i]); /* f1[i] - f2[i] */
+ tmax |= L_abs(t0);
+ a[j] = (Word16)(vo_L_shr_r(t0, 12)); /* from Q23 to Q12 and * 0.5 */
+ }
- /* rescale data if overflow has occured and reprocess the loop */
- if(adaptive_scaling == 1)
- q = 4 - norm_l(tmax); /* adaptive scaling enabled */
- else
- q = 0; /* adaptive scaling disabled */
+ /* rescale data if overflow has occured and reprocess the loop */
+ if(adaptive_scaling == 1)
+ q = 4 - norm_l(tmax); /* adaptive scaling enabled */
+ else
+ q = 0; /* adaptive scaling disabled */
- if (q > 0)
- {
- q_sug = (12 + q);
- for (i = 1, j = m - 1; i < nc; i++, j--)
- {
- /* a[i] = 0.5*(f1[i] + f2[i]); */
- t0 = vo_L_add(f1[i], f2[i]); /* f1[i] + f2[i] */
- a[i] = (Word16)(vo_L_shr_r(t0, q_sug)); /* from Q23 to Q12 and * 0.5 */
+ if (q > 0)
+ {
+ q_sug = (12 + q);
+ for (i = 1, j = m - 1; i < nc; i++, j--)
+ {
+ /* a[i] = 0.5*(f1[i] + f2[i]); */
+ t0 = vo_L_add(f1[i], f2[i]); /* f1[i] + f2[i] */
+ a[i] = (Word16)(vo_L_shr_r(t0, q_sug)); /* from Q23 to Q12 and * 0.5 */
- /* a[j] = 0.5*(f1[i] - f2[i]); */
- t0 = vo_L_sub(f1[i], f2[i]); /* f1[i] - f2[i] */
- a[j] = (Word16)(vo_L_shr_r(t0, q_sug)); /* from Q23 to Q12 and * 0.5 */
- }
- a[0] = shr(a[0], q);
- }
- else
- {
- q_sug = 12;
- q = 0;
- }
- /* a[NC] = 0.5*f1[NC]*(1.0 + isp[M-1]); */
- hi = f1[nc] >> 16;
- lo = (f1[nc] & 0xffff)>>1;
- t0 = Mpy_32_16(hi, lo, isp[m - 1]);
- t0 = vo_L_add(f1[nc], t0);
- a[nc] = (Word16)(L_shr_r(t0, q_sug)); /* from Q23 to Q12 and * 0.5 */
- /* a[m] = isp[m-1]; */
+ /* a[j] = 0.5*(f1[i] - f2[i]); */
+ t0 = vo_L_sub(f1[i], f2[i]); /* f1[i] - f2[i] */
+ a[j] = (Word16)(vo_L_shr_r(t0, q_sug)); /* from Q23 to Q12 and * 0.5 */
+ }
+ a[0] = shr(a[0], q);
+ }
+ else
+ {
+ q_sug = 12;
+ q = 0;
+ }
+ /* a[NC] = 0.5*f1[NC]*(1.0 + isp[M-1]); */
+ hi = f1[nc] >> 16;
+ lo = (f1[nc] & 0xffff)>>1;
+ t0 = Mpy_32_16(hi, lo, isp[m - 1]);
+ t0 = vo_L_add(f1[nc], t0);
+ a[nc] = (Word16)(L_shr_r(t0, q_sug)); /* from Q23 to Q12 and * 0.5 */
+ /* a[m] = isp[m-1]; */
- a[m] = vo_shr_r(isp[m - 1], (3 + q)); /* from Q15 to Q12 */
- return;
+ a[m] = vo_shr_r(isp[m - 1], (3 + q)); /* from Q15 to Q12 */
+ return;
}
/*-----------------------------------------------------------*
@@ -185,63 +185,63 @@
static void Get_isp_pol(Word16 * isp, Word32 * f, Word16 n)
{
- Word16 hi, lo;
- Word32 i, j, t0;
- /* All computation in Q23 */
+ Word16 hi, lo;
+ Word32 i, j, t0;
+ /* All computation in Q23 */
- f[0] = vo_L_mult(4096, 1024); /* f[0] = 1.0; in Q23 */
- f[1] = vo_L_mult(isp[0], -256); /* f[1] = -2.0*isp[0] in Q23 */
+ f[0] = vo_L_mult(4096, 1024); /* f[0] = 1.0; in Q23 */
+ f[1] = vo_L_mult(isp[0], -256); /* f[1] = -2.0*isp[0] in Q23 */
- f += 2; /* Advance f pointer */
- isp += 2; /* Advance isp pointer */
- for (i = 2; i <= n; i++)
- {
- *f = f[-2];
- for (j = 1; j < i; j++, f--)
- {
- hi = f[-1]>>16;
- lo = (f[-1] & 0xffff)>>1;
+ f += 2; /* Advance f pointer */
+ isp += 2; /* Advance isp pointer */
+ for (i = 2; i <= n; i++)
+ {
+ *f = f[-2];
+ for (j = 1; j < i; j++, f--)
+ {
+ hi = f[-1]>>16;
+ lo = (f[-1] & 0xffff)>>1;
- t0 = Mpy_32_16(hi, lo, *isp); /* t0 = f[-1] * isp */
- t0 = t0 << 1;
- *f = vo_L_sub(*f, t0); /* *f -= t0 */
- *f = vo_L_add(*f, f[-2]); /* *f += f[-2] */
- }
- *f -= (*isp << 9); /* *f -= isp<<8 */
- f += i; /* Advance f pointer */
- isp += 2; /* Advance isp pointer */
- }
- return;
+ t0 = Mpy_32_16(hi, lo, *isp); /* t0 = f[-1] * isp */
+ t0 = t0 << 1;
+ *f = vo_L_sub(*f, t0); /* *f -= t0 */
+ *f = vo_L_add(*f, f[-2]); /* *f += f[-2] */
+ }
+ *f -= (*isp << 9); /* *f -= isp<<8 */
+ f += i; /* Advance f pointer */
+ isp += 2; /* Advance isp pointer */
+ }
+ return;
}
static void Get_isp_pol_16kHz(Word16 * isp, Word32 * f, Word16 n)
{
- Word16 hi, lo;
- Word32 i, j, t0;
+ Word16 hi, lo;
+ Word32 i, j, t0;
- /* All computation in Q23 */
- f[0] = L_mult(4096, 256); /* f[0] = 1.0; in Q23 */
- f[1] = L_mult(isp[0], -64); /* f[1] = -2.0*isp[0] in Q23 */
+ /* All computation in Q23 */
+ f[0] = L_mult(4096, 256); /* f[0] = 1.0; in Q23 */
+ f[1] = L_mult(isp[0], -64); /* f[1] = -2.0*isp[0] in Q23 */
- f += 2; /* Advance f pointer */
- isp += 2; /* Advance isp pointer */
+ f += 2; /* Advance f pointer */
+ isp += 2; /* Advance isp pointer */
- for (i = 2; i <= n; i++)
- {
- *f = f[-2];
- for (j = 1; j < i; j++, f--)
- {
- VO_L_Extract(f[-1], &hi, &lo);
- t0 = Mpy_32_16(hi, lo, *isp); /* t0 = f[-1] * isp */
- t0 = L_shl2(t0, 1);
- *f = L_sub(*f, t0); /* *f -= t0 */
- *f = L_add(*f, f[-2]); /* *f += f[-2] */
- }
- *f = L_msu(*f, *isp, 64); /* *f -= isp<<8 */
- f += i; /* Advance f pointer */
- isp += 2; /* Advance isp pointer */
- }
- return;
+ for (i = 2; i <= n; i++)
+ {
+ *f = f[-2];
+ for (j = 1; j < i; j++, f--)
+ {
+ VO_L_Extract(f[-1], &hi, &lo);
+ t0 = Mpy_32_16(hi, lo, *isp); /* t0 = f[-1] * isp */
+ t0 = L_shl2(t0, 1);
+ *f = L_sub(*f, t0); /* *f -= t0 */
+ *f = L_add(*f, f[-2]); /* *f += f[-2] */
+ }
+ *f = L_msu(*f, *isp, 64); /* *f -= isp<<8 */
+ f += i; /* Advance f pointer */
+ isp += 2; /* Advance isp pointer */
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/isp_isf.c b/media/libstagefright/codecs/amrwbenc/src/isp_isf.c
index b4ba408..56798e0 100644
--- a/media/libstagefright/codecs/amrwbenc/src/isp_isf.c
+++ b/media/libstagefright/codecs/amrwbenc/src/isp_isf.c
@@ -18,11 +18,11 @@
* File: isp_isf.c *
* *
* Description: *
-* Isp_isf Transformation isp to isf *
-* Isf_isp Transformation isf to isp *
+* Isp_isf Transformation isp to isf *
+* Isf_isp Transformation isf to isp *
* *
-* The transformation from isp[i] to isf[i] and isf[i] to isp[i] *
-* are approximated by a look-up table and interpolation *
+* The transformation from isp[i] to isf[i] and isf[i] to isp[i] *
+* are approximated by a look-up table and interpolation *
* *
************************************************************************/
@@ -31,59 +31,59 @@
#include "isp_isf.tab" /* Look-up table for transformations */
void Isp_isf(
- Word16 isp[], /* (i) Q15 : isp[m] (range: -1<=val<1) */
- Word16 isf[], /* (o) Q15 : isf[m] normalized (range: 0.0<=val<=0.5) */
- Word16 m /* (i) : LPC order */
- )
+ Word16 isp[], /* (i) Q15 : isp[m] (range: -1<=val<1) */
+ Word16 isf[], /* (o) Q15 : isf[m] normalized (range: 0.0<=val<=0.5) */
+ Word16 m /* (i) : LPC order */
+ )
{
- Word32 i, ind;
- Word32 L_tmp;
- ind = 127; /* beging at end of table -1 */
- for (i = (m - 1); i >= 0; i--)
- {
- if (i >= (m - 2))
- { /* m-2 is a constant */
- ind = 127; /* beging at end of table -1 */
- }
- /* find value in table that is just greater than isp[i] */
- while (table[ind] < isp[i])
- ind--;
- /* acos(isp[i])= ind*128 + ( ( isp[i]-table[ind] ) * slope[ind] )/2048 */
- L_tmp = vo_L_mult(vo_sub(isp[i], table[ind]), slope[ind]);
- isf[i] = vo_round((L_tmp << 4)); /* (isp[i]-table[ind])*slope[ind])>>11 */
- isf[i] = add1(isf[i], (ind << 7));
- }
- isf[m - 1] = (isf[m - 1] >> 1);
- return;
+ Word32 i, ind;
+ Word32 L_tmp;
+ ind = 127; /* beging at end of table -1 */
+ for (i = (m - 1); i >= 0; i--)
+ {
+ if (i >= (m - 2))
+ { /* m-2 is a constant */
+ ind = 127; /* beging at end of table -1 */
+ }
+ /* find value in table that is just greater than isp[i] */
+ while (table[ind] < isp[i])
+ ind--;
+ /* acos(isp[i])= ind*128 + ( ( isp[i]-table[ind] ) * slope[ind] )/2048 */
+ L_tmp = vo_L_mult(vo_sub(isp[i], table[ind]), slope[ind]);
+ isf[i] = vo_round((L_tmp << 4)); /* (isp[i]-table[ind])*slope[ind])>>11 */
+ isf[i] = add1(isf[i], (ind << 7));
+ }
+ isf[m - 1] = (isf[m - 1] >> 1);
+ return;
}
void Isf_isp(
- Word16 isf[], /* (i) Q15 : isf[m] normalized (range: 0.0<=val<=0.5) */
- Word16 isp[], /* (o) Q15 : isp[m] (range: -1<=val<1) */
- Word16 m /* (i) : LPC order */
- )
+ Word16 isf[], /* (i) Q15 : isf[m] normalized (range: 0.0<=val<=0.5) */
+ Word16 isp[], /* (o) Q15 : isp[m] (range: -1<=val<1) */
+ Word16 m /* (i) : LPC order */
+ )
{
- Word16 offset;
- Word32 i, ind, L_tmp;
+ Word16 offset;
+ Word32 i, ind, L_tmp;
- for (i = 0; i < m - 1; i++)
- {
- isp[i] = isf[i];
- }
- isp[m - 1] = (isf[m - 1] << 1);
+ for (i = 0; i < m - 1; i++)
+ {
+ isp[i] = isf[i];
+ }
+ isp[m - 1] = (isf[m - 1] << 1);
- for (i = 0; i < m; i++)
- {
- ind = (isp[i] >> 7); /* ind = b7-b15 of isf[i] */
- offset = (Word16) (isp[i] & 0x007f); /* offset = b0-b6 of isf[i] */
+ for (i = 0; i < m; i++)
+ {
+ ind = (isp[i] >> 7); /* ind = b7-b15 of isf[i] */
+ offset = (Word16) (isp[i] & 0x007f); /* offset = b0-b6 of isf[i] */
- /* isp[i] = table[ind]+ ((table[ind+1]-table[ind])*offset) / 128 */
- L_tmp = vo_L_mult(vo_sub(table[ind + 1], table[ind]), offset);
- isp[i] = add1(table[ind], (Word16)((L_tmp >> 8)));
- }
+ /* isp[i] = table[ind]+ ((table[ind+1]-table[ind])*offset) / 128 */
+ L_tmp = vo_L_mult(vo_sub(table[ind + 1], table[ind]), offset);
+ isp[i] = add1(table[ind], (Word16)((L_tmp >> 8)));
+ }
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/lag_wind.c b/media/libstagefright/codecs/amrwbenc/src/lag_wind.c
index 49c622c..527430b 100644
--- a/media/libstagefright/codecs/amrwbenc/src/lag_wind.c
+++ b/media/libstagefright/codecs/amrwbenc/src/lag_wind.c
@@ -17,8 +17,8 @@
/***********************************************************************
* File: lag_wind.c *
* *
-* Description: Lag_windows on autocorrelations *
-* r[i] *= lag_wind[i] *
+* Description: Lag_windows on autocorrelations *
+* r[i] *= lag_wind[i] *
* *
************************************************************************/
@@ -29,20 +29,20 @@
void Lag_window(
- Word16 r_h[], /* (i/o) : Autocorrelations (msb) */
- Word16 r_l[] /* (i/o) : Autocorrelations (lsb) */
- )
+ Word16 r_h[], /* (i/o) : Autocorrelations (msb) */
+ Word16 r_l[] /* (i/o) : Autocorrelations (lsb) */
+ )
{
- Word32 i;
- Word32 x;
+ Word32 i;
+ Word32 x;
- for (i = 1; i <= M; i++)
- {
- x = Mpy_32(r_h[i], r_l[i], volag_h[i - 1], volag_l[i - 1]);
- r_h[i] = x >> 16;
- r_l[i] = (x & 0xffff)>>1;
- }
- return;
+ for (i = 1; i <= M; i++)
+ {
+ x = Mpy_32(r_h[i], r_l[i], volag_h[i - 1], volag_l[i - 1]);
+ r_h[i] = x >> 16;
+ r_l[i] = (x & 0xffff)>>1;
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/levinson.c b/media/libstagefright/codecs/amrwbenc/src/levinson.c
index 4b2f8ed..9d5a3bd 100644
--- a/media/libstagefright/codecs/amrwbenc/src/levinson.c
+++ b/media/libstagefright/codecs/amrwbenc/src/levinson.c
@@ -21,7 +21,7 @@
* *
************************************************************************/
/*---------------------------------------------------------------------------*
- * LEVINSON.C *
+ * LEVINSON.C *
*---------------------------------------------------------------------------*
* *
* LEVINSON-DURBIN algorithm in double precision *
@@ -96,154 +96,154 @@
#define NC (M/2)
void Init_Levinson(
- Word16 * mem /* output :static memory (18 words) */
- )
+ Word16 * mem /* output :static memory (18 words) */
+ )
{
- Set_zero(mem, 18); /* old_A[0..M-1] = 0, old_rc[0..1] = 0 */
- return;
+ Set_zero(mem, 18); /* old_A[0..M-1] = 0, old_rc[0..1] = 0 */
+ return;
}
void Levinson(
- Word16 Rh[], /* (i) : Rh[M+1] Vector of autocorrelations (msb) */
- Word16 Rl[], /* (i) : Rl[M+1] Vector of autocorrelations (lsb) */
- Word16 A[], /* (o) Q12 : A[M] LPC coefficients (m = 16) */
- Word16 rc[], /* (o) Q15 : rc[M] Reflection coefficients. */
- Word16 * mem /* (i/o) :static memory (18 words) */
- )
+ Word16 Rh[], /* (i) : Rh[M+1] Vector of autocorrelations (msb) */
+ Word16 Rl[], /* (i) : Rl[M+1] Vector of autocorrelations (lsb) */
+ Word16 A[], /* (o) Q12 : A[M] LPC coefficients (m = 16) */
+ Word16 rc[], /* (o) Q15 : rc[M] Reflection coefficients. */
+ Word16 * mem /* (i/o) :static memory (18 words) */
+ )
{
- Word32 i, j;
- Word16 hi, lo;
- Word16 Kh, Kl; /* reflection coefficient; hi and lo */
- Word16 alp_h, alp_l, alp_exp; /* Prediction gain; hi lo and exponent */
- Word16 Ah[M + 1], Al[M + 1]; /* LPC coef. in double prec. */
- Word16 Anh[M + 1], Anl[M + 1]; /* LPC coef.for next iteration in double prec. */
- Word32 t0, t1, t2; /* temporary variable */
- Word16 *old_A, *old_rc;
+ Word32 i, j;
+ Word16 hi, lo;
+ Word16 Kh, Kl; /* reflection coefficient; hi and lo */
+ Word16 alp_h, alp_l, alp_exp; /* Prediction gain; hi lo and exponent */
+ Word16 Ah[M + 1], Al[M + 1]; /* LPC coef. in double prec. */
+ Word16 Anh[M + 1], Anl[M + 1]; /* LPC coef.for next iteration in double prec. */
+ Word32 t0, t1, t2; /* temporary variable */
+ Word16 *old_A, *old_rc;
- /* Last A(z) for case of unstable filter */
- old_A = mem;
- old_rc = mem + M;
+ /* Last A(z) for case of unstable filter */
+ old_A = mem;
+ old_rc = mem + M;
- /* K = A[1] = -R[1] / R[0] */
+ /* K = A[1] = -R[1] / R[0] */
- t1 = ((Rh[1] << 16) + (Rl[1] << 1)); /* R[1] in Q31 */
- t2 = L_abs(t1); /* abs R[1] */
- t0 = Div_32(t2, Rh[0], Rl[0]); /* R[1]/R[0] in Q31 */
- if (t1 > 0)
- t0 = -t0; /* -R[1]/R[0] */
+ t1 = ((Rh[1] << 16) + (Rl[1] << 1)); /* R[1] in Q31 */
+ t2 = L_abs(t1); /* abs R[1] */
+ t0 = Div_32(t2, Rh[0], Rl[0]); /* R[1]/R[0] in Q31 */
+ if (t1 > 0)
+ t0 = -t0; /* -R[1]/R[0] */
- Kh = t0 >> 16;
- Kl = (t0 & 0xffff)>>1;
- rc[0] = Kh;
- t0 = (t0 >> 4); /* A[1] in Q27 */
+ Kh = t0 >> 16;
+ Kl = (t0 & 0xffff)>>1;
+ rc[0] = Kh;
+ t0 = (t0 >> 4); /* A[1] in Q27 */
- Ah[1] = t0 >> 16;
- Al[1] = (t0 & 0xffff)>>1;
+ Ah[1] = t0 >> 16;
+ Al[1] = (t0 & 0xffff)>>1;
- /* Alpha = R[0] * (1-K**2) */
- t0 = Mpy_32(Kh, Kl, Kh, Kl); /* K*K in Q31 */
- t0 = L_abs(t0); /* Some case <0 !! */
- t0 = vo_L_sub((Word32) 0x7fffffffL, t0); /* 1 - K*K in Q31 */
+ /* Alpha = R[0] * (1-K**2) */
+ t0 = Mpy_32(Kh, Kl, Kh, Kl); /* K*K in Q31 */
+ t0 = L_abs(t0); /* Some case <0 !! */
+ t0 = vo_L_sub((Word32) 0x7fffffffL, t0); /* 1 - K*K in Q31 */
- hi = t0 >> 16;
- lo = (t0 & 0xffff)>>1;
+ hi = t0 >> 16;
+ lo = (t0 & 0xffff)>>1;
- t0 = Mpy_32(Rh[0], Rl[0], hi, lo); /* Alpha in Q31 */
+ t0 = Mpy_32(Rh[0], Rl[0], hi, lo); /* Alpha in Q31 */
- /* Normalize Alpha */
- alp_exp = norm_l(t0);
- t0 = (t0 << alp_exp);
+ /* Normalize Alpha */
+ alp_exp = norm_l(t0);
+ t0 = (t0 << alp_exp);
- alp_h = t0 >> 16;
- alp_l = (t0 & 0xffff)>>1;
- /*--------------------------------------*
- * ITERATIONS I=2 to M *
- *--------------------------------------*/
- for (i = 2; i <= M; i++)
- {
- /* t0 = SUM ( R[j]*A[i-j] ,j=1,i-1 ) + R[i] */
- t0 = 0;
- for (j = 1; j < i; j++)
- t0 = vo_L_add(t0, Mpy_32(Rh[j], Rl[j], Ah[i - j], Al[i - j]));
+ alp_h = t0 >> 16;
+ alp_l = (t0 & 0xffff)>>1;
+ /*--------------------------------------*
+ * ITERATIONS I=2 to M *
+ *--------------------------------------*/
+ for (i = 2; i <= M; i++)
+ {
+ /* t0 = SUM ( R[j]*A[i-j] ,j=1,i-1 ) + R[i] */
+ t0 = 0;
+ for (j = 1; j < i; j++)
+ t0 = vo_L_add(t0, Mpy_32(Rh[j], Rl[j], Ah[i - j], Al[i - j]));
- t0 = t0 << 4; /* result in Q27 -> convert to Q31 */
- /* No overflow possible */
- t1 = ((Rh[i] << 16) + (Rl[i] << 1));
- t0 = vo_L_add(t0, t1); /* add R[i] in Q31 */
+ t0 = t0 << 4; /* result in Q27 -> convert to Q31 */
+ /* No overflow possible */
+ t1 = ((Rh[i] << 16) + (Rl[i] << 1));
+ t0 = vo_L_add(t0, t1); /* add R[i] in Q31 */
- /* K = -t0 / Alpha */
- t1 = L_abs(t0);
- t2 = Div_32(t1, alp_h, alp_l); /* abs(t0)/Alpha */
- if (t0 > 0)
- t2 = -t2; /* K =-t0/Alpha */
- t2 = (t2 << alp_exp); /* denormalize; compare to Alpha */
+ /* K = -t0 / Alpha */
+ t1 = L_abs(t0);
+ t2 = Div_32(t1, alp_h, alp_l); /* abs(t0)/Alpha */
+ if (t0 > 0)
+ t2 = -t2; /* K =-t0/Alpha */
+ t2 = (t2 << alp_exp); /* denormalize; compare to Alpha */
- Kh = t2 >> 16;
- Kl = (t2 & 0xffff)>>1;
+ Kh = t2 >> 16;
+ Kl = (t2 & 0xffff)>>1;
- rc[i - 1] = Kh;
- /* Test for unstable filter. If unstable keep old A(z) */
- if (abs_s(Kh) > 32750)
- {
- A[0] = 4096; /* Ai[0] not stored (always 1.0) */
- for (j = 0; j < M; j++)
- {
- A[j + 1] = old_A[j];
- }
- rc[0] = old_rc[0]; /* only two rc coefficients are needed */
- rc[1] = old_rc[1];
- return;
- }
- /*------------------------------------------*
- * Compute new LPC coeff. -> An[i] *
- * An[j]= A[j] + K*A[i-j] , j=1 to i-1 *
- * An[i]= K *
- *------------------------------------------*/
- for (j = 1; j < i; j++)
- {
- t0 = Mpy_32(Kh, Kl, Ah[i - j], Al[i - j]);
- t0 = vo_L_add(t0, ((Ah[j] << 16) + (Al[j] << 1)));
- Anh[j] = t0 >> 16;
- Anl[j] = (t0 & 0xffff)>>1;
- }
- t2 = (t2 >> 4); /* t2 = K in Q31 ->convert to Q27 */
+ rc[i - 1] = Kh;
+ /* Test for unstable filter. If unstable keep old A(z) */
+ if (abs_s(Kh) > 32750)
+ {
+ A[0] = 4096; /* Ai[0] not stored (always 1.0) */
+ for (j = 0; j < M; j++)
+ {
+ A[j + 1] = old_A[j];
+ }
+ rc[0] = old_rc[0]; /* only two rc coefficients are needed */
+ rc[1] = old_rc[1];
+ return;
+ }
+ /*------------------------------------------*
+ * Compute new LPC coeff. -> An[i] *
+ * An[j]= A[j] + K*A[i-j] , j=1 to i-1 *
+ * An[i]= K *
+ *------------------------------------------*/
+ for (j = 1; j < i; j++)
+ {
+ t0 = Mpy_32(Kh, Kl, Ah[i - j], Al[i - j]);
+ t0 = vo_L_add(t0, ((Ah[j] << 16) + (Al[j] << 1)));
+ Anh[j] = t0 >> 16;
+ Anl[j] = (t0 & 0xffff)>>1;
+ }
+ t2 = (t2 >> 4); /* t2 = K in Q31 ->convert to Q27 */
- VO_L_Extract(t2, &Anh[i], &Anl[i]); /* An[i] in Q27 */
+ VO_L_Extract(t2, &Anh[i], &Anl[i]); /* An[i] in Q27 */
- /* Alpha = Alpha * (1-K**2) */
- t0 = Mpy_32(Kh, Kl, Kh, Kl); /* K*K in Q31 */
- t0 = L_abs(t0); /* Some case <0 !! */
- t0 = vo_L_sub((Word32) 0x7fffffffL, t0); /* 1 - K*K in Q31 */
- hi = t0 >> 16;
- lo = (t0 & 0xffff)>>1;
- t0 = Mpy_32(alp_h, alp_l, hi, lo); /* Alpha in Q31 */
+ /* Alpha = Alpha * (1-K**2) */
+ t0 = Mpy_32(Kh, Kl, Kh, Kl); /* K*K in Q31 */
+ t0 = L_abs(t0); /* Some case <0 !! */
+ t0 = vo_L_sub((Word32) 0x7fffffffL, t0); /* 1 - K*K in Q31 */
+ hi = t0 >> 16;
+ lo = (t0 & 0xffff)>>1;
+ t0 = Mpy_32(alp_h, alp_l, hi, lo); /* Alpha in Q31 */
- /* Normalize Alpha */
- j = norm_l(t0);
- t0 = (t0 << j);
- alp_h = t0 >> 16;
- alp_l = (t0 & 0xffff)>>1;
- alp_exp += j; /* Add normalization to alp_exp */
+ /* Normalize Alpha */
+ j = norm_l(t0);
+ t0 = (t0 << j);
+ alp_h = t0 >> 16;
+ alp_l = (t0 & 0xffff)>>1;
+ alp_exp += j; /* Add normalization to alp_exp */
- /* A[j] = An[j] */
- for (j = 1; j <= i; j++)
- {
- Ah[j] = Anh[j];
- Al[j] = Anl[j];
- }
- }
- /* Truncate A[i] in Q27 to Q12 with rounding */
- A[0] = 4096;
- for (i = 1; i <= M; i++)
- {
- t0 = (Ah[i] << 16) + (Al[i] << 1);
- old_A[i - 1] = A[i] = vo_round((t0 << 1));
- }
- old_rc[0] = rc[0];
- old_rc[1] = rc[1];
+ /* A[j] = An[j] */
+ for (j = 1; j <= i; j++)
+ {
+ Ah[j] = Anh[j];
+ Al[j] = Anl[j];
+ }
+ }
+ /* Truncate A[i] in Q27 to Q12 with rounding */
+ A[0] = 4096;
+ for (i = 1; i <= M; i++)
+ {
+ t0 = (Ah[i] << 16) + (Al[i] << 1);
+ old_A[i - 1] = A[i] = vo_round((t0 << 1));
+ }
+ old_rc[0] = rc[0];
+ old_rc[1] = rc[1];
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/log2.c b/media/libstagefright/codecs/amrwbenc/src/log2.c
index 0f65541..f14058e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/log2.c
+++ b/media/libstagefright/codecs/amrwbenc/src/log2.c
@@ -54,33 +54,33 @@
*************************************************************************/
void Log2_norm (
- Word32 L_x, /* (i) : input value (normalized) */
- Word16 exp, /* (i) : norm_l (L_x) */
- Word16 *exponent, /* (o) : Integer part of Log2. (range: 0<=val<=30) */
- Word16 *fraction /* (o) : Fractional part of Log2. (range: 0<=val<1) */
- )
+ Word32 L_x, /* (i) : input value (normalized) */
+ Word16 exp, /* (i) : norm_l (L_x) */
+ Word16 *exponent, /* (o) : Integer part of Log2. (range: 0<=val<=30) */
+ Word16 *fraction /* (o) : Fractional part of Log2. (range: 0<=val<1) */
+ )
{
- Word16 i, a, tmp;
- Word32 L_y;
- if (L_x <= (Word32) 0)
- {
- *exponent = 0;
- *fraction = 0;
- return;
- }
- *exponent = (30 - exp);
- L_x = (L_x >> 9);
- i = extract_h (L_x); /* Extract b25-b31 */
- L_x = (L_x >> 1);
- a = (Word16)(L_x); /* Extract b10-b24 of fraction */
- a = (Word16)(a & (Word16)0x7fff);
- i -= 32;
- L_y = L_deposit_h (table[i]); /* table[i] << 16 */
- tmp = vo_sub(table[i], table[i + 1]); /* table[i] - table[i+1] */
- L_y = vo_L_msu (L_y, tmp, a); /* L_y -= tmp*a*2 */
- *fraction = extract_h (L_y);
+ Word16 i, a, tmp;
+ Word32 L_y;
+ if (L_x <= (Word32) 0)
+ {
+ *exponent = 0;
+ *fraction = 0;
+ return;
+ }
+ *exponent = (30 - exp);
+ L_x = (L_x >> 9);
+ i = extract_h (L_x); /* Extract b25-b31 */
+ L_x = (L_x >> 1);
+ a = (Word16)(L_x); /* Extract b10-b24 of fraction */
+ a = (Word16)(a & (Word16)0x7fff);
+ i -= 32;
+ L_y = L_deposit_h (table[i]); /* table[i] << 16 */
+ tmp = vo_sub(table[i], table[i + 1]); /* table[i] - table[i+1] */
+ L_y = vo_L_msu (L_y, tmp, a); /* L_y -= tmp*a*2 */
+ *fraction = extract_h (L_y);
- return;
+ return;
}
/*************************************************************************
@@ -96,15 +96,15 @@
*************************************************************************/
void Log2 (
- Word32 L_x, /* (i) : input value */
- Word16 *exponent, /* (o) : Integer part of Log2. (range: 0<=val<=30) */
- Word16 *fraction /* (o) : Fractional part of Log2. (range: 0<=val<1) */
- )
+ Word32 L_x, /* (i) : input value */
+ Word16 *exponent, /* (o) : Integer part of Log2. (range: 0<=val<=30) */
+ Word16 *fraction /* (o) : Fractional part of Log2. (range: 0<=val<1) */
+ )
{
- Word16 exp;
+ Word16 exp;
- exp = norm_l(L_x);
- Log2_norm ((L_x << exp), exp, exponent, fraction);
+ exp = norm_l(L_x);
+ Log2_norm ((L_x << exp), exp, exponent, fraction);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/lp_dec2.c b/media/libstagefright/codecs/amrwbenc/src/lp_dec2.c
index 1d5d076..9a9dd34 100644
--- a/media/libstagefright/codecs/amrwbenc/src/lp_dec2.c
+++ b/media/libstagefright/codecs/amrwbenc/src/lp_dec2.c
@@ -17,7 +17,7 @@
/***********************************************************************
* File: lp_dec2.c *
* *
-* Description:Decimate a vector by 2 with 2nd order fir filter *
+* Description:Decimate a vector by 2 with 2nd order fir filter *
* *
************************************************************************/
@@ -33,36 +33,36 @@
static Word16 h_fir[L_FIR] = {4260, 7536, 9175, 7536, 4260};
void LP_Decim2(
- Word16 x[], /* in/out: signal to process */
- Word16 l, /* input : size of filtering */
- Word16 mem[] /* in/out: memory (size=3) */
- )
+ Word16 x[], /* in/out: signal to process */
+ Word16 l, /* input : size of filtering */
+ Word16 mem[] /* in/out: memory (size=3) */
+ )
{
- Word16 *p_x, x_buf[L_FRAME + L_MEM];
- Word32 i, j;
- Word32 L_tmp;
- /* copy initial filter states into buffer */
- p_x = x_buf;
- for (i = 0; i < L_MEM; i++)
- {
- *p_x++ = mem[i];
- mem[i] = x[l - L_MEM + i];
- }
- for (i = 0; i < l; i++)
- {
- *p_x++ = x[i];
- }
- for (i = 0, j = 0; i < l; i += 2, j++)
- {
- p_x = &x_buf[i];
- L_tmp = ((*p_x++) * h_fir[0]);
- L_tmp += ((*p_x++) * h_fir[1]);
- L_tmp += ((*p_x++) * h_fir[2]);
- L_tmp += ((*p_x++) * h_fir[3]);
- L_tmp += ((*p_x++) * h_fir[4]);
- x[j] = (L_tmp + 0x4000)>>15;
- }
- return;
+ Word16 *p_x, x_buf[L_FRAME + L_MEM];
+ Word32 i, j;
+ Word32 L_tmp;
+ /* copy initial filter states into buffer */
+ p_x = x_buf;
+ for (i = 0; i < L_MEM; i++)
+ {
+ *p_x++ = mem[i];
+ mem[i] = x[l - L_MEM + i];
+ }
+ for (i = 0; i < l; i++)
+ {
+ *p_x++ = x[i];
+ }
+ for (i = 0, j = 0; i < l; i += 2, j++)
+ {
+ p_x = &x_buf[i];
+ L_tmp = ((*p_x++) * h_fir[0]);
+ L_tmp += ((*p_x++) * h_fir[1]);
+ L_tmp += ((*p_x++) * h_fir[2]);
+ L_tmp += ((*p_x++) * h_fir[3]);
+ L_tmp += ((*p_x++) * h_fir[4]);
+ x[j] = (L_tmp + 0x4000)>>15;
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/math_op.c b/media/libstagefright/codecs/amrwbenc/src/math_op.c
index 7affbb2..9d7c74e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/math_op.c
+++ b/media/libstagefright/codecs/amrwbenc/src/math_op.c
@@ -55,17 +55,17 @@
|___________________________________________________________________________|
*/
Word32 Isqrt( /* (o) Q31 : output value (range: 0<=val<1) */
- Word32 L_x /* (i) Q0 : input value (range: 0<=val<=7fffffff) */
- )
+ Word32 L_x /* (i) Q0 : input value (range: 0<=val<=7fffffff) */
+ )
{
- Word16 exp;
- Word32 L_y;
- exp = norm_l(L_x);
- L_x = (L_x << exp); /* L_x is normalized */
- exp = (31 - exp);
- Isqrt_n(&L_x, &exp);
- L_y = (L_x << exp); /* denormalization */
- return (L_y);
+ Word16 exp;
+ Word32 L_y;
+ exp = norm_l(L_x);
+ L_x = (L_x << exp); /* L_x is normalized */
+ exp = (31 - exp);
+ Isqrt_n(&L_x, &exp);
+ L_y = (L_x << exp); /* denormalization */
+ return (L_y);
}
/*___________________________________________________________________________
@@ -90,43 +90,43 @@
*/
static Word16 table_isqrt[49] =
{
- 32767, 31790, 30894, 30070, 29309, 28602, 27945, 27330, 26755, 26214,
- 25705, 25225, 24770, 24339, 23930, 23541, 23170, 22817, 22479, 22155,
- 21845, 21548, 21263, 20988, 20724, 20470, 20225, 19988, 19760, 19539,
- 19326, 19119, 18919, 18725, 18536, 18354, 18176, 18004, 17837, 17674,
- 17515, 17361, 17211, 17064, 16921, 16782, 16646, 16514, 16384
+ 32767, 31790, 30894, 30070, 29309, 28602, 27945, 27330, 26755, 26214,
+ 25705, 25225, 24770, 24339, 23930, 23541, 23170, 22817, 22479, 22155,
+ 21845, 21548, 21263, 20988, 20724, 20470, 20225, 19988, 19760, 19539,
+ 19326, 19119, 18919, 18725, 18536, 18354, 18176, 18004, 17837, 17674,
+ 17515, 17361, 17211, 17064, 16921, 16782, 16646, 16514, 16384
};
void Isqrt_n(
- Word32 * frac, /* (i/o) Q31: normalized value (1.0 < frac <= 0.5) */
- Word16 * exp /* (i/o) : exponent (value = frac x 2^exponent) */
- )
+ Word32 * frac, /* (i/o) Q31: normalized value (1.0 < frac <= 0.5) */
+ Word16 * exp /* (i/o) : exponent (value = frac x 2^exponent) */
+ )
{
- Word16 i, a, tmp;
+ Word16 i, a, tmp;
- if (*frac <= (Word32) 0)
- {
- *exp = 0;
- *frac = 0x7fffffffL;
- return;
- }
+ if (*frac <= (Word32) 0)
+ {
+ *exp = 0;
+ *frac = 0x7fffffffL;
+ return;
+ }
- if((*exp & 1) == 1) /*If exponant odd -> shift right */
- *frac = (*frac) >> 1;
+ if((*exp & 1) == 1) /*If exponant odd -> shift right */
+ *frac = (*frac) >> 1;
- *exp = negate((*exp - 1) >> 1);
+ *exp = negate((*exp - 1) >> 1);
- *frac = (*frac >> 9);
- i = extract_h(*frac); /* Extract b25-b31 */
- *frac = (*frac >> 1);
- a = (Word16)(*frac); /* Extract b10-b24 */
- a = (Word16) (a & (Word16) 0x7fff);
- i -= 16;
- *frac = L_deposit_h(table_isqrt[i]); /* table[i] << 16 */
- tmp = vo_sub(table_isqrt[i], table_isqrt[i + 1]); /* table[i] - table[i+1]) */
- *frac = vo_L_msu(*frac, tmp, a); /* frac -= tmp*a*2 */
+ *frac = (*frac >> 9);
+ i = extract_h(*frac); /* Extract b25-b31 */
+ *frac = (*frac >> 1);
+ a = (Word16)(*frac); /* Extract b10-b24 */
+ a = (Word16) (a & (Word16) 0x7fff);
+ i -= 16;
+ *frac = L_deposit_h(table_isqrt[i]); /* table[i] << 16 */
+ tmp = vo_sub(table_isqrt[i], table_isqrt[i + 1]); /* table[i] - table[i+1]) */
+ *frac = vo_L_msu(*frac, tmp, a); /* frac -= tmp*a*2 */
- return;
+ return;
}
/*___________________________________________________________________________
@@ -149,34 +149,34 @@
*/
static Word16 table_pow2[33] =
{
- 16384, 16743, 17109, 17484, 17867, 18258, 18658, 19066, 19484, 19911,
- 20347, 20792, 21247, 21713, 22188, 22674, 23170, 23678, 24196, 24726,
- 25268, 25821, 26386, 26964, 27554, 28158, 28774, 29405, 30048, 30706,
- 31379, 32066, 32767
+ 16384, 16743, 17109, 17484, 17867, 18258, 18658, 19066, 19484, 19911,
+ 20347, 20792, 21247, 21713, 22188, 22674, 23170, 23678, 24196, 24726,
+ 25268, 25821, 26386, 26964, 27554, 28158, 28774, 29405, 30048, 30706,
+ 31379, 32066, 32767
};
Word32 Pow2( /* (o) Q0 : result (range: 0<=val<=0x7fffffff) */
- Word16 exponant, /* (i) Q0 : Integer part. (range: 0<=val<=30) */
- Word16 fraction /* (i) Q15 : Fractionnal part. (range: 0.0<=val<1.0) */
- )
+ Word16 exponant, /* (i) Q0 : Integer part. (range: 0<=val<=30) */
+ Word16 fraction /* (i) Q15 : Fractionnal part. (range: 0.0<=val<1.0) */
+ )
{
- Word16 exp, i, a, tmp;
- Word32 L_x;
+ Word16 exp, i, a, tmp;
+ Word32 L_x;
- L_x = vo_L_mult(fraction, 32); /* L_x = fraction<<6 */
- i = extract_h(L_x); /* Extract b10-b16 of fraction */
- L_x =L_x >> 1;
- a = (Word16)(L_x); /* Extract b0-b9 of fraction */
- a = (Word16) (a & (Word16) 0x7fff);
+ L_x = vo_L_mult(fraction, 32); /* L_x = fraction<<6 */
+ i = extract_h(L_x); /* Extract b10-b16 of fraction */
+ L_x =L_x >> 1;
+ a = (Word16)(L_x); /* Extract b0-b9 of fraction */
+ a = (Word16) (a & (Word16) 0x7fff);
- L_x = L_deposit_h(table_pow2[i]); /* table[i] << 16 */
- tmp = vo_sub(table_pow2[i], table_pow2[i + 1]); /* table[i] - table[i+1] */
- L_x -= (tmp * a)<<1; /* L_x -= tmp*a*2 */
+ L_x = L_deposit_h(table_pow2[i]); /* table[i] << 16 */
+ tmp = vo_sub(table_pow2[i], table_pow2[i + 1]); /* table[i] - table[i+1] */
+ L_x -= (tmp * a)<<1; /* L_x -= tmp*a*2 */
- exp = vo_sub(30, exponant);
- L_x = vo_L_shr_r(L_x, exp);
+ exp = vo_sub(30, exponant);
+ L_x = vo_L_shr_r(L_x, exp);
- return (L_x);
+ return (L_x);
}
/*___________________________________________________________________________
@@ -194,25 +194,30 @@
*/
Word32 Dot_product12( /* (o) Q31: normalized result (1 < val <= -1) */
- Word16 x[], /* (i) 12bits: x vector */
- Word16 y[], /* (i) 12bits: y vector */
- Word16 lg, /* (i) : vector length */
- Word16 * exp /* (o) : exponent of result (0..+30) */
- )
+ Word16 x[], /* (i) 12bits: x vector */
+ Word16 y[], /* (i) 12bits: y vector */
+ Word16 lg, /* (i) : vector length */
+ Word16 * exp /* (o) : exponent of result (0..+30) */
+ )
{
- Word16 sft;
- Word32 i, L_sum;
- L_sum = 0;
- for (i = 0; i < lg; i++)
- {
- L_sum += x[i] * y[i];
- }
- L_sum = (L_sum << 1) + 1;
- /* Normalize acc in Q31 */
- sft = norm_l(L_sum);
- L_sum = L_sum << sft;
- *exp = 30 - sft; /* exponent = 0..30 */
- return (L_sum);
+ Word16 sft;
+ Word32 i, L_sum;
+ L_sum = 0;
+ for (i = 0; i < lg; i++)
+ {
+ Word32 tmp = (Word32) x[i] * (Word32) y[i];
+ if (tmp == (Word32) 0x40000000L) {
+ tmp = MAX_32;
+ }
+ L_sum = L_add(L_sum, tmp);
+ }
+ L_sum = L_shl2(L_sum, 1);
+ L_sum = L_add(L_sum, 1);
+ /* Normalize acc in Q31 */
+ sft = norm_l(L_sum);
+ L_sum = L_sum << sft;
+ *exp = 30 - sft; /* exponent = 0..30 */
+ return (L_sum);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/mem_align.c b/media/libstagefright/codecs/amrwbenc/src/mem_align.c
index 3b7853f..04e5976 100644
--- a/media/libstagefright/codecs/amrwbenc/src/mem_align.c
+++ b/media/libstagefright/codecs/amrwbenc/src/mem_align.c
@@ -15,18 +15,18 @@
*/
/*******************************************************************************
- File: mem_align.c
+ File: mem_align.c
- Content: Memory alloc alignments functions
+ Content: Memory alloc alignments functions
*******************************************************************************/
-#include "mem_align.h"
+#include "mem_align.h"
#ifdef _MSC_VER
-#include <stddef.h>
+#include <stddef.h>
#else
-#include <stdint.h>
+#include <stdint.h>
#endif
/*****************************************************************************
@@ -39,50 +39,50 @@
void *
mem_malloc(VO_MEM_OPERATOR *pMemop, unsigned int size, unsigned char alignment, unsigned int CodecID)
{
- int ret;
- unsigned char *mem_ptr;
- VO_MEM_INFO MemInfo;
+ int ret;
+ unsigned char *mem_ptr;
+ VO_MEM_INFO MemInfo;
- if (!alignment) {
+ if (!alignment) {
- MemInfo.Flag = 0;
- MemInfo.Size = size + 1;
- ret = pMemop->Alloc(CodecID, &MemInfo);
- if(ret != 0)
- return 0;
- mem_ptr = (unsigned char *)MemInfo.VBuffer;
+ MemInfo.Flag = 0;
+ MemInfo.Size = size + 1;
+ ret = pMemop->Alloc(CodecID, &MemInfo);
+ if(ret != 0)
+ return 0;
+ mem_ptr = (unsigned char *)MemInfo.VBuffer;
- pMemop->Set(CodecID, mem_ptr, 0, size + 1);
+ pMemop->Set(CodecID, mem_ptr, 0, size + 1);
- *mem_ptr = (unsigned char)1;
+ *mem_ptr = (unsigned char)1;
- return ((void *)(mem_ptr+1));
- } else {
- unsigned char *tmp;
+ return ((void *)(mem_ptr+1));
+ } else {
+ unsigned char *tmp;
- MemInfo.Flag = 0;
- MemInfo.Size = size + alignment;
- ret = pMemop->Alloc(CodecID, &MemInfo);
- if(ret != 0)
- return 0;
+ MemInfo.Flag = 0;
+ MemInfo.Size = size + alignment;
+ ret = pMemop->Alloc(CodecID, &MemInfo);
+ if(ret != 0)
+ return 0;
- tmp = (unsigned char *)MemInfo.VBuffer;
+ tmp = (unsigned char *)MemInfo.VBuffer;
- pMemop->Set(CodecID, tmp, 0, size + alignment);
+ pMemop->Set(CodecID, tmp, 0, size + alignment);
- mem_ptr =
- (unsigned char *) ((intptr_t) (tmp + alignment - 1) &
- (~((intptr_t) (alignment - 1))));
+ mem_ptr =
+ (unsigned char *) ((intptr_t) (tmp + alignment - 1) &
+ (~((intptr_t) (alignment - 1))));
- if (mem_ptr == tmp)
- mem_ptr += alignment;
+ if (mem_ptr == tmp)
+ mem_ptr += alignment;
- *(mem_ptr - 1) = (unsigned char) (mem_ptr - tmp);
+ *(mem_ptr - 1) = (unsigned char) (mem_ptr - tmp);
- return ((void *)mem_ptr);
- }
+ return ((void *)mem_ptr);
+ }
- return(0);
+ return(0);
}
@@ -96,16 +96,16 @@
mem_free(VO_MEM_OPERATOR *pMemop, void *mem_ptr, unsigned int CodecID)
{
- unsigned char *ptr;
+ unsigned char *ptr;
- if (mem_ptr == 0)
- return;
+ if (mem_ptr == 0)
+ return;
- ptr = mem_ptr;
+ ptr = mem_ptr;
- ptr -= *(ptr - 1);
+ ptr -= *(ptr - 1);
- pMemop->Free(CodecID, ptr);
+ pMemop->Free(CodecID, ptr);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/oper_32b.c b/media/libstagefright/codecs/amrwbenc/src/oper_32b.c
index 27cad76..e6f80d0 100644
--- a/media/libstagefright/codecs/amrwbenc/src/oper_32b.c
+++ b/media/libstagefright/codecs/amrwbenc/src/oper_32b.c
@@ -56,9 +56,9 @@
__inline void VO_L_Extract (Word32 L_32, Word16 *hi, Word16 *lo)
{
- *hi = (Word16)(L_32 >> 16);
- *lo = (Word16)((L_32 & 0xffff) >> 1);
- return;
+ *hi = (Word16)(L_32 >> 16);
+ *lo = (Word16)((L_32 & 0xffff) >> 1);
+ return;
}
/*****************************************************************************
@@ -84,11 +84,11 @@
Word32 L_Comp (Word16 hi, Word16 lo)
{
- Word32 L_32;
+ Word32 L_32;
- L_32 = L_deposit_h (hi);
+ L_32 = L_deposit_h (hi);
- return (L_mac (L_32, lo, 1)); /* = hi<<16 + lo<<1 */
+ return (L_mac (L_32, lo, 1)); /* = hi<<16 + lo<<1 */
}
/*****************************************************************************
@@ -113,13 +113,13 @@
__inline Word32 Mpy_32 (Word16 hi1, Word16 lo1, Word16 hi2, Word16 lo2)
{
- Word32 L_32;
- L_32 = (hi1 * hi2);
- L_32 += (hi1 * lo2) >> 15;
- L_32 += (lo1 * hi2) >> 15;
- L_32 <<= 1;
+ Word32 L_32;
+ L_32 = (hi1 * hi2);
+ L_32 += (hi1 * lo2) >> 15;
+ L_32 += (lo1 * hi2) >> 15;
+ L_32 <<= 1;
- return (L_32);
+ return (L_32);
}
/*****************************************************************************
@@ -142,12 +142,12 @@
__inline Word32 Mpy_32_16 (Word16 hi, Word16 lo, Word16 n)
{
- Word32 L_32;
+ Word32 L_32;
- L_32 = (hi * n)<<1;
- L_32 += (((lo * n)>>15)<<1);
+ L_32 = (hi * n)<<1;
+ L_32 += (((lo * n)>>15)<<1);
- return (L_32);
+ return (L_32);
}
/*****************************************************************************
@@ -194,30 +194,30 @@
Word32 Div_32 (Word32 L_num, Word16 denom_hi, Word16 denom_lo)
{
- Word16 approx, hi, lo, n_hi, n_lo;
- Word32 L_32;
+ Word16 approx, hi, lo, n_hi, n_lo;
+ Word32 L_32;
- /* First approximation: 1 / L_denom = 1/denom_hi */
+ /* First approximation: 1 / L_denom = 1/denom_hi */
- approx = div_s ((Word16) 0x3fff, denom_hi);
+ approx = div_s ((Word16) 0x3fff, denom_hi);
- /* 1/L_denom = approx * (2.0 - L_denom * approx) */
+ /* 1/L_denom = approx * (2.0 - L_denom * approx) */
- L_32 = Mpy_32_16 (denom_hi, denom_lo, approx);
+ L_32 = Mpy_32_16 (denom_hi, denom_lo, approx);
- L_32 = L_sub ((Word32) 0x7fffffffL, L_32);
- hi = L_32 >> 16;
- lo = (L_32 & 0xffff) >> 1;
+ L_32 = L_sub ((Word32) 0x7fffffffL, L_32);
+ hi = L_32 >> 16;
+ lo = (L_32 & 0xffff) >> 1;
- L_32 = Mpy_32_16 (hi, lo, approx);
+ L_32 = Mpy_32_16 (hi, lo, approx);
- /* L_num * (1/L_denom) */
- hi = L_32 >> 16;
- lo = (L_32 & 0xffff) >> 1;
- VO_L_Extract (L_num, &n_hi, &n_lo);
- L_32 = Mpy_32 (n_hi, n_lo, hi, lo);
- L_32 = L_shl2(L_32, 2);
+ /* L_num * (1/L_denom) */
+ hi = L_32 >> 16;
+ lo = (L_32 & 0xffff) >> 1;
+ VO_L_Extract (L_num, &n_hi, &n_lo);
+ L_32 = Mpy_32 (n_hi, n_lo, hi, lo);
+ L_32 = L_shl2(L_32, 2);
- return (L_32);
+ return (L_32);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/p_med_ol.c b/media/libstagefright/codecs/amrwbenc/src/p_med_ol.c
index b8174b9..5d2b4bd 100644
--- a/media/libstagefright/codecs/amrwbenc/src/p_med_ol.c
+++ b/media/libstagefright/codecs/amrwbenc/src/p_med_ol.c
@@ -18,7 +18,7 @@
* File: p_med_ol.c *
* *
* Description: Compute the open loop pitch lag *
-* output: open loop pitch lag *
+* output: open loop pitch lag *
************************************************************************/
#include "typedef.h"
@@ -29,131 +29,131 @@
#include "p_med_ol.tab"
Word16 Pitch_med_ol(
- Word16 wsp[], /* i: signal used to compute the open loop pitch*/
+ Word16 wsp[], /* i: signal used to compute the open loop pitch*/
/* wsp[-pit_max] to wsp[-1] should be known */
- Coder_State *st, /* i/o: codec global structure */
- Word16 L_frame /* i: length of frame to compute pitch */
- )
+ Coder_State *st, /* i/o: codec global structure */
+ Word16 L_frame /* i: length of frame to compute pitch */
+ )
{
- Word16 Tm;
- Word16 hi, lo;
- Word16 *ww, *we, *hp_wsp;
- Word16 exp_R0, exp_R1, exp_R2;
- Word32 i, j, max, R0, R1, R2;
- Word16 *p1, *p2;
- Word16 L_min = 17; /* minimum pitch lag: PIT_MIN / OPL_DECIM */
- Word16 L_max = 115; /* maximum pitch lag: PIT_MAX / OPL_DECIM */
- Word16 L_0 = st->old_T0_med; /* old open-loop pitch */
- Word16 *gain = &(st->ol_gain); /* normalize correlation of hp_wsp for the lag */
- Word16 *hp_wsp_mem = st->hp_wsp_mem; /* memory of the hypass filter for hp_wsp[] (lg = 9)*/
- Word16 *old_hp_wsp = st->old_hp_wsp; /* hypass wsp[] */
- Word16 wght_flg = st->ol_wght_flg; /* is weighting function used */
+ Word16 Tm;
+ Word16 hi, lo;
+ Word16 *ww, *we, *hp_wsp;
+ Word16 exp_R0, exp_R1, exp_R2;
+ Word32 i, j, max, R0, R1, R2;
+ Word16 *p1, *p2;
+ Word16 L_min = 17; /* minimum pitch lag: PIT_MIN / OPL_DECIM */
+ Word16 L_max = 115; /* maximum pitch lag: PIT_MAX / OPL_DECIM */
+ Word16 L_0 = st->old_T0_med; /* old open-loop pitch */
+ Word16 *gain = &(st->ol_gain); /* normalize correlation of hp_wsp for the lag */
+ Word16 *hp_wsp_mem = st->hp_wsp_mem; /* memory of the hypass filter for hp_wsp[] (lg = 9)*/
+ Word16 *old_hp_wsp = st->old_hp_wsp; /* hypass wsp[] */
+ Word16 wght_flg = st->ol_wght_flg; /* is weighting function used */
- ww = &corrweight[198];
- we = &corrweight[98 + L_max - L_0];
+ ww = &corrweight[198];
+ we = &corrweight[98 + L_max - L_0];
- max = MIN_32;
- Tm = 0;
- for (i = L_max; i > L_min; i--)
- {
- /* Compute the correlation */
- R0 = 0;
- p1 = wsp;
- p2 = &wsp[-i];
- for (j = 0; j < L_frame; j+=4)
- {
- R0 += vo_L_mult((*p1++), (*p2++));
- R0 += vo_L_mult((*p1++), (*p2++));
- R0 += vo_L_mult((*p1++), (*p2++));
- R0 += vo_L_mult((*p1++), (*p2++));
- }
- /* Weighting of the correlation function. */
- hi = R0>>16;
- lo = (R0 & 0xffff)>>1;
+ max = MIN_32;
+ Tm = 0;
+ for (i = L_max; i > L_min; i--)
+ {
+ /* Compute the correlation */
+ R0 = 0;
+ p1 = wsp;
+ p2 = &wsp[-i];
+ for (j = 0; j < L_frame; j+=4)
+ {
+ R0 += vo_L_mult((*p1++), (*p2++));
+ R0 += vo_L_mult((*p1++), (*p2++));
+ R0 += vo_L_mult((*p1++), (*p2++));
+ R0 += vo_L_mult((*p1++), (*p2++));
+ }
+ /* Weighting of the correlation function. */
+ hi = R0>>16;
+ lo = (R0 & 0xffff)>>1;
- R0 = Mpy_32_16(hi, lo, *ww);
- ww--;
+ R0 = Mpy_32_16(hi, lo, *ww);
+ ww--;
- if ((L_0 > 0) && (wght_flg > 0))
- {
- /* Weight the neighbourhood of the old lag. */
- hi = R0>>16;
- lo = (R0 & 0xffff)>>1;
- R0 = Mpy_32_16(hi, lo, *we);
- we--;
- }
- if(R0 >= max)
- {
- max = R0;
- Tm = i;
- }
- }
+ if ((L_0 > 0) && (wght_flg > 0))
+ {
+ /* Weight the neighbourhood of the old lag. */
+ hi = R0>>16;
+ lo = (R0 & 0xffff)>>1;
+ R0 = Mpy_32_16(hi, lo, *we);
+ we--;
+ }
+ if(R0 >= max)
+ {
+ max = R0;
+ Tm = i;
+ }
+ }
- /* Hypass the wsp[] vector */
- hp_wsp = old_hp_wsp + L_max;
- Hp_wsp(wsp, hp_wsp, L_frame, hp_wsp_mem);
+ /* Hypass the wsp[] vector */
+ hp_wsp = old_hp_wsp + L_max;
+ Hp_wsp(wsp, hp_wsp, L_frame, hp_wsp_mem);
- /* Compute normalize correlation at delay Tm */
- R0 = 0;
- R1 = 0;
- R2 = 0;
- p1 = hp_wsp;
- p2 = hp_wsp - Tm;
- for (j = 0; j < L_frame; j+=4)
- {
- R2 += vo_mult32(*p1, *p1);
- R1 += vo_mult32(*p2, *p2);
- R0 += vo_mult32(*p1++, *p2++);
- R2 += vo_mult32(*p1, *p1);
- R1 += vo_mult32(*p2, *p2);
- R0 += vo_mult32(*p1++, *p2++);
- R2 += vo_mult32(*p1, *p1);
- R1 += vo_mult32(*p2, *p2);
- R0 += vo_mult32(*p1++, *p2++);
- R2 += vo_mult32(*p1, *p1);
- R1 += vo_mult32(*p2, *p2);
- R0 += vo_mult32(*p1++, *p2++);
- }
- R0 = R0 <<1;
- R1 = (R1 <<1) + 1L;
- R2 = (R2 <<1) + 1L;
- /* gain = R0/ sqrt(R1*R2) */
+ /* Compute normalize correlation at delay Tm */
+ R0 = 0;
+ R1 = 0;
+ R2 = 0;
+ p1 = hp_wsp;
+ p2 = hp_wsp - Tm;
+ for (j = 0; j < L_frame; j+=4)
+ {
+ R2 += vo_mult32(*p1, *p1);
+ R1 += vo_mult32(*p2, *p2);
+ R0 += vo_mult32(*p1++, *p2++);
+ R2 += vo_mult32(*p1, *p1);
+ R1 += vo_mult32(*p2, *p2);
+ R0 += vo_mult32(*p1++, *p2++);
+ R2 += vo_mult32(*p1, *p1);
+ R1 += vo_mult32(*p2, *p2);
+ R0 += vo_mult32(*p1++, *p2++);
+ R2 += vo_mult32(*p1, *p1);
+ R1 += vo_mult32(*p2, *p2);
+ R0 += vo_mult32(*p1++, *p2++);
+ }
+ R0 = R0 <<1;
+ R1 = (R1 <<1) + 1L;
+ R2 = (R2 <<1) + 1L;
+ /* gain = R0/ sqrt(R1*R2) */
- exp_R0 = norm_l(R0);
- R0 = (R0 << exp_R0);
+ exp_R0 = norm_l(R0);
+ R0 = (R0 << exp_R0);
- exp_R1 = norm_l(R1);
- R1 = (R1 << exp_R1);
+ exp_R1 = norm_l(R1);
+ R1 = (R1 << exp_R1);
- exp_R2 = norm_l(R2);
- R2 = (R2 << exp_R2);
+ exp_R2 = norm_l(R2);
+ R2 = (R2 << exp_R2);
- R1 = vo_L_mult(vo_round(R1), vo_round(R2));
+ R1 = vo_L_mult(voround(R1), voround(R2));
- i = norm_l(R1);
- R1 = (R1 << i);
+ i = norm_l(R1);
+ R1 = (R1 << i);
- exp_R1 += exp_R2;
- exp_R1 += i;
- exp_R1 = 62 - exp_R1;
+ exp_R1 += exp_R2;
+ exp_R1 += i;
+ exp_R1 = 62 - exp_R1;
- Isqrt_n(&R1, &exp_R1);
+ Isqrt_n(&R1, &exp_R1);
- R0 = vo_L_mult(voround(R0), voround(R1));
- exp_R0 = 31 - exp_R0;
- exp_R0 += exp_R1;
+ R0 = vo_L_mult(voround(R0), voround(R1));
+ exp_R0 = 31 - exp_R0;
+ exp_R0 += exp_R1;
- *gain = vo_round(L_shl(R0, exp_R0));
+ *gain = vo_round(L_shl(R0, exp_R0));
- /* Shitf hp_wsp[] for next frame */
+ /* Shitf hp_wsp[] for next frame */
- for (i = 0; i < L_max; i++)
- {
- old_hp_wsp[i] = old_hp_wsp[i + L_frame];
- }
+ for (i = 0; i < L_max; i++)
+ {
+ old_hp_wsp[i] = old_hp_wsp[i + L_frame];
+ }
- return (Tm);
+ return (Tm);
}
/************************************************************************
@@ -171,84 +171,84 @@
Word16 median5(Word16 x[])
{
- Word16 x1, x2, x3, x4, x5;
- Word16 tmp;
+ Word16 x1, x2, x3, x4, x5;
+ Word16 tmp;
- x1 = x[-2];
- x2 = x[-1];
- x3 = x[0];
- x4 = x[1];
- x5 = x[2];
+ x1 = x[-2];
+ x2 = x[-1];
+ x3 = x[0];
+ x4 = x[1];
+ x5 = x[2];
- if (x2 < x1)
- {
- tmp = x1;
- x1 = x2;
- x2 = tmp;
- }
- if (x3 < x1)
- {
- tmp = x1;
- x1 = x3;
- x3 = tmp;
- }
- if (x4 < x1)
- {
- tmp = x1;
- x1 = x4;
- x4 = tmp;
- }
- if (x5 < x1)
- {
- x5 = x1;
- }
- if (x3 < x2)
- {
- tmp = x2;
- x2 = x3;
- x3 = tmp;
- }
- if (x4 < x2)
- {
- tmp = x2;
- x2 = x4;
- x4 = tmp;
- }
- if (x5 < x2)
- {
- x5 = x2;
- }
- if (x4 < x3)
- {
- x3 = x4;
- }
- if (x5 < x3)
- {
- x3 = x5;
- }
- return (x3);
+ if (x2 < x1)
+ {
+ tmp = x1;
+ x1 = x2;
+ x2 = tmp;
+ }
+ if (x3 < x1)
+ {
+ tmp = x1;
+ x1 = x3;
+ x3 = tmp;
+ }
+ if (x4 < x1)
+ {
+ tmp = x1;
+ x1 = x4;
+ x4 = tmp;
+ }
+ if (x5 < x1)
+ {
+ x5 = x1;
+ }
+ if (x3 < x2)
+ {
+ tmp = x2;
+ x2 = x3;
+ x3 = tmp;
+ }
+ if (x4 < x2)
+ {
+ tmp = x2;
+ x2 = x4;
+ x4 = tmp;
+ }
+ if (x5 < x2)
+ {
+ x5 = x2;
+ }
+ if (x4 < x3)
+ {
+ x3 = x4;
+ }
+ if (x5 < x3)
+ {
+ x3 = x5;
+ }
+ return (x3);
}
Word16 Med_olag( /* output : median of 5 previous open-loop lags */
- Word16 prev_ol_lag, /* input : previous open-loop lag */
- Word16 old_ol_lag[5]
- )
+ Word16 prev_ol_lag, /* input : previous open-loop lag */
+ Word16 old_ol_lag[5]
+ )
{
- Word32 i;
+ Word32 i;
- /* Use median of 5 previous open-loop lags as old lag */
+ /* Use median of 5 previous open-loop lags as old lag */
- for (i = 4; i > 0; i--)
- {
- old_ol_lag[i] = old_ol_lag[i - 1];
- }
+ for (i = 4; i > 0; i--)
+ {
+ old_ol_lag[i] = old_ol_lag[i - 1];
+ }
- old_ol_lag[0] = prev_ol_lag;
+ old_ol_lag[0] = prev_ol_lag;
- i = median5(&old_ol_lag[2]);
+ i = median5(&old_ol_lag[2]);
- return i;
+ return i;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/pit_shrp.c b/media/libstagefright/codecs/amrwbenc/src/pit_shrp.c
index 6f55b8f..f100253 100644
--- a/media/libstagefright/codecs/amrwbenc/src/pit_shrp.c
+++ b/media/libstagefright/codecs/amrwbenc/src/pit_shrp.c
@@ -25,24 +25,24 @@
#include "basic_op.h"
void Pit_shrp(
- Word16 * x, /* in/out: impulse response (or algebraic code) */
- Word16 pit_lag, /* input : pitch lag */
- Word16 sharp, /* input : pitch sharpening factor (Q15) */
- Word16 L_subfr /* input : subframe size */
- )
+ Word16 * x, /* in/out: impulse response (or algebraic code) */
+ Word16 pit_lag, /* input : pitch lag */
+ Word16 sharp, /* input : pitch sharpening factor (Q15) */
+ Word16 L_subfr /* input : subframe size */
+ )
{
- Word32 i;
- Word32 L_tmp;
- Word16 *x_ptr = x + pit_lag;
+ Word32 i;
+ Word32 L_tmp;
+ Word16 *x_ptr = x + pit_lag;
- for (i = pit_lag; i < L_subfr; i++)
- {
- L_tmp = (*x_ptr << 15);
- L_tmp += *x++ * sharp;
- *x_ptr++ = ((L_tmp + 0x4000)>>15);
- }
+ for (i = pit_lag; i < L_subfr; i++)
+ {
+ L_tmp = (*x_ptr << 15);
+ L_tmp += *x++ * sharp;
+ *x_ptr++ = ((L_tmp + 0x4000)>>15);
+ }
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c b/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
index b66b55e..de2a221 100644
--- a/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
+++ b/media/libstagefright/codecs/amrwbenc/src/pitch_f4.c
@@ -18,7 +18,7 @@
* File: pitch_f4.c *
* *
* Description: Find the closed loop pitch period with *
-* 1/4 subsample resolution. *
+* 1/4 subsample resolution. *
* *
************************************************************************/
@@ -37,117 +37,117 @@
#ifdef ASM_OPT
void Norm_corr_asm(
- Word16 exc[], /* (i) : excitation buffer */
- Word16 xn[], /* (i) : target vector */
- Word16 h[], /* (i) Q15 : impulse response of synth/wgt filters */
- Word16 L_subfr,
- Word16 t_min, /* (i) : minimum value of pitch lag. */
- Word16 t_max, /* (i) : maximum value of pitch lag. */
- Word16 corr_norm[] /* (o) Q15 : normalized correlation */
- );
+ Word16 exc[], /* (i) : excitation buffer */
+ Word16 xn[], /* (i) : target vector */
+ Word16 h[], /* (i) Q15 : impulse response of synth/wgt filters */
+ Word16 L_subfr,
+ Word16 t_min, /* (i) : minimum value of pitch lag. */
+ Word16 t_max, /* (i) : maximum value of pitch lag. */
+ Word16 corr_norm[] /* (o) Q15 : normalized correlation */
+ );
#else
static void Norm_Corr(
- Word16 exc[], /* (i) : excitation buffer */
- Word16 xn[], /* (i) : target vector */
- Word16 h[], /* (i) Q15 : impulse response of synth/wgt filters */
- Word16 L_subfr,
- Word16 t_min, /* (i) : minimum value of pitch lag. */
- Word16 t_max, /* (i) : maximum value of pitch lag. */
- Word16 corr_norm[] /* (o) Q15 : normalized correlation */
- );
+ Word16 exc[], /* (i) : excitation buffer */
+ Word16 xn[], /* (i) : target vector */
+ Word16 h[], /* (i) Q15 : impulse response of synth/wgt filters */
+ Word16 L_subfr,
+ Word16 t_min, /* (i) : minimum value of pitch lag. */
+ Word16 t_max, /* (i) : maximum value of pitch lag. */
+ Word16 corr_norm[] /* (o) Q15 : normalized correlation */
+ );
#endif
static Word16 Interpol_4( /* (o) : interpolated value */
- Word16 * x, /* (i) : input vector */
- Word32 frac /* (i) : fraction (-4..+3) */
- );
+ Word16 * x, /* (i) : input vector */
+ Word32 frac /* (i) : fraction (-4..+3) */
+ );
Word16 Pitch_fr4( /* (o) : pitch period. */
- Word16 exc[], /* (i) : excitation buffer */
- Word16 xn[], /* (i) : target vector */
- Word16 h[], /* (i) Q15 : impulse response of synth/wgt filters */
- Word16 t0_min, /* (i) : minimum value in the searched range. */
- Word16 t0_max, /* (i) : maximum value in the searched range. */
- Word16 * pit_frac, /* (o) : chosen fraction (0, 1, 2 or 3). */
- Word16 i_subfr, /* (i) : indicator for first subframe. */
- Word16 t0_fr2, /* (i) : minimum value for resolution 1/2 */
- Word16 t0_fr1, /* (i) : minimum value for resolution 1 */
- Word16 L_subfr /* (i) : Length of subframe */
- )
+ Word16 exc[], /* (i) : excitation buffer */
+ Word16 xn[], /* (i) : target vector */
+ Word16 h[], /* (i) Q15 : impulse response of synth/wgt filters */
+ Word16 t0_min, /* (i) : minimum value in the searched range. */
+ Word16 t0_max, /* (i) : maximum value in the searched range. */
+ Word16 * pit_frac, /* (o) : chosen fraction (0, 1, 2 or 3). */
+ Word16 i_subfr, /* (i) : indicator for first subframe. */
+ Word16 t0_fr2, /* (i) : minimum value for resolution 1/2 */
+ Word16 t0_fr1, /* (i) : minimum value for resolution 1 */
+ Word16 L_subfr /* (i) : Length of subframe */
+ )
{
- Word32 fraction, i;
- Word16 t_min, t_max;
- Word16 max, t0, step, temp;
- Word16 *corr;
- Word16 corr_v[40]; /* Total length = t0_max-t0_min+1+2*L_inter */
+ Word32 fraction, i;
+ Word16 t_min, t_max;
+ Word16 max, t0, step, temp;
+ Word16 *corr;
+ Word16 corr_v[40]; /* Total length = t0_max-t0_min+1+2*L_inter */
- /* Find interval to compute normalized correlation */
+ /* Find interval to compute normalized correlation */
- t_min = t0_min - L_INTERPOL1;
- t_max = t0_max + L_INTERPOL1;
- corr = &corr_v[-t_min];
- /* Compute normalized correlation between target and filtered excitation */
+ t_min = t0_min - L_INTERPOL1;
+ t_max = t0_max + L_INTERPOL1;
+ corr = &corr_v[-t_min];
+ /* Compute normalized correlation between target and filtered excitation */
#ifdef ASM_OPT /* asm optimization branch */
Norm_corr_asm(exc, xn, h, L_subfr, t_min, t_max, corr);
#else
- Norm_Corr(exc, xn, h, L_subfr, t_min, t_max, corr);
+ Norm_Corr(exc, xn, h, L_subfr, t_min, t_max, corr);
#endif
- /* Find integer pitch */
+ /* Find integer pitch */
- max = corr[t0_min];
- t0 = t0_min;
- for (i = t0_min + 1; i <= t0_max; i++)
- {
- if (corr[i] >= max)
- {
- max = corr[i];
- t0 = i;
- }
- }
- /* If first subframe and t0 >= t0_fr1, do not search fractionnal pitch */
- if ((i_subfr == 0) && (t0 >= t0_fr1))
- {
- *pit_frac = 0;
- return (t0);
- }
- /*------------------------------------------------------------------*
- * Search fractionnal pitch with 1/4 subsample resolution. *
- * Test the fractions around t0 and choose the one which maximizes *
- * the interpolated normalized correlation. *
- *------------------------------------------------------------------*/
+ max = corr[t0_min];
+ t0 = t0_min;
+ for (i = t0_min + 1; i <= t0_max; i++)
+ {
+ if (corr[i] >= max)
+ {
+ max = corr[i];
+ t0 = i;
+ }
+ }
+ /* If first subframe and t0 >= t0_fr1, do not search fractionnal pitch */
+ if ((i_subfr == 0) && (t0 >= t0_fr1))
+ {
+ *pit_frac = 0;
+ return (t0);
+ }
+ /*------------------------------------------------------------------*
+ * Search fractionnal pitch with 1/4 subsample resolution. *
+ * Test the fractions around t0 and choose the one which maximizes *
+ * the interpolated normalized correlation. *
+ *------------------------------------------------------------------*/
- step = 1; /* 1/4 subsample resolution */
- fraction = -3;
- if ((t0_fr2 == PIT_MIN)||((i_subfr == 0) && (t0 >= t0_fr2)))
- {
- step = 2; /* 1/2 subsample resolution */
- fraction = -2;
- }
- if(t0 == t0_min)
- {
- fraction = 0;
- }
- max = Interpol_4(&corr[t0], fraction);
+ step = 1; /* 1/4 subsample resolution */
+ fraction = -3;
+ if ((t0_fr2 == PIT_MIN)||((i_subfr == 0) && (t0 >= t0_fr2)))
+ {
+ step = 2; /* 1/2 subsample resolution */
+ fraction = -2;
+ }
+ if(t0 == t0_min)
+ {
+ fraction = 0;
+ }
+ max = Interpol_4(&corr[t0], fraction);
- for (i = fraction + step; i <= 3; i += step)
- {
- temp = Interpol_4(&corr[t0], i);
- if(temp > max)
- {
- max = temp;
- fraction = i;
- }
- }
- /* limit the fraction value in the interval [0,1,2,3] */
- if (fraction < 0)
- {
- fraction += UP_SAMP;
- t0 -= 1;
- }
- *pit_frac = fraction;
- return (t0);
+ for (i = fraction + step; i <= 3; i += step)
+ {
+ temp = Interpol_4(&corr[t0], i);
+ if(temp > max)
+ {
+ max = temp;
+ fraction = i;
+ }
+ }
+ /* limit the fraction value in the interval [0,1,2,3] */
+ if (fraction < 0)
+ {
+ fraction += UP_SAMP;
+ t0 -= 1;
+ }
+ *pit_frac = fraction;
+ return (t0);
}
@@ -161,109 +161,109 @@
************************************************************************************/
#ifndef ASM_OPT
static void Norm_Corr(
- Word16 exc[], /* (i) : excitation buffer */
- Word16 xn[], /* (i) : target vector */
- Word16 h[], /* (i) Q15 : impulse response of synth/wgt filters */
- Word16 L_subfr,
- Word16 t_min, /* (i) : minimum value of pitch lag. */
- Word16 t_max, /* (i) : maximum value of pitch lag. */
- Word16 corr_norm[]) /* (o) Q15 : normalized correlation */
+ Word16 exc[], /* (i) : excitation buffer */
+ Word16 xn[], /* (i) : target vector */
+ Word16 h[], /* (i) Q15 : impulse response of synth/wgt filters */
+ Word16 L_subfr,
+ Word16 t_min, /* (i) : minimum value of pitch lag. */
+ Word16 t_max, /* (i) : maximum value of pitch lag. */
+ Word16 corr_norm[]) /* (o) Q15 : normalized correlation */
{
- Word32 i, k, t;
- Word32 corr, exp_corr, norm, exp, scale;
- Word16 exp_norm, excf[L_SUBFR], tmp;
- Word32 L_tmp, L_tmp1, L_tmp2;
+ Word32 i, k, t;
+ Word32 corr, exp_corr, norm, exp, scale;
+ Word16 exp_norm, excf[L_SUBFR], tmp;
+ Word32 L_tmp, L_tmp1, L_tmp2;
UNUSED(L_subfr);
- /* compute the filtered excitation for the first delay t_min */
- k = -t_min;
+ /* compute the filtered excitation for the first delay t_min */
+ k = -t_min;
#ifdef ASM_OPT /* asm optimization branch */
- Convolve_asm(&exc[k], h, excf, 64);
+ Convolve_asm(&exc[k], h, excf, 64);
#else
- Convolve(&exc[k], h, excf, 64);
+ Convolve(&exc[k], h, excf, 64);
#endif
- /* Compute rounded down 1/sqrt(energy of xn[]) */
- L_tmp = 0;
- for (i = 0; i < 64; i+=4)
- {
- L_tmp += (xn[i] * xn[i]);
- L_tmp += (xn[i+1] * xn[i+1]);
- L_tmp += (xn[i+2] * xn[i+2]);
- L_tmp += (xn[i+3] * xn[i+3]);
- }
+ /* Compute rounded down 1/sqrt(energy of xn[]) */
+ L_tmp = 0;
+ for (i = 0; i < 64; i+=4)
+ {
+ L_tmp += (xn[i] * xn[i]);
+ L_tmp += (xn[i+1] * xn[i+1]);
+ L_tmp += (xn[i+2] * xn[i+2]);
+ L_tmp += (xn[i+3] * xn[i+3]);
+ }
- L_tmp = (L_tmp << 1) + 1;
- exp = norm_l(L_tmp);
- exp = (32 - exp);
- //exp = exp + 2; /* energy of xn[] x 2 + rounded up */
- scale = -(exp >> 1); /* (1<<scale) < 1/sqrt(energy rounded) */
+ L_tmp = (L_tmp << 1) + 1;
+ exp = norm_l(L_tmp);
+ exp = (32 - exp);
+ //exp = exp + 2; /* energy of xn[] x 2 + rounded up */
+ scale = -(exp >> 1); /* (1<<scale) < 1/sqrt(energy rounded) */
- /* loop for every possible period */
+ /* loop for every possible period */
- for (t = t_min; t <= t_max; t++)
- {
- /* Compute correlation between xn[] and excf[] */
- L_tmp = 0;
- L_tmp1 = 0;
- for (i = 0; i < 64; i+=4)
- {
- L_tmp += (xn[i] * excf[i]);
- L_tmp1 += (excf[i] * excf[i]);
- L_tmp += (xn[i+1] * excf[i+1]);
- L_tmp1 += (excf[i+1] * excf[i+1]);
- L_tmp += (xn[i+2] * excf[i+2]);
- L_tmp1 += (excf[i+2] * excf[i+2]);
- L_tmp += (xn[i+3] * excf[i+3]);
- L_tmp1 += (excf[i+3] * excf[i+3]);
- }
+ for (t = t_min; t <= t_max; t++)
+ {
+ /* Compute correlation between xn[] and excf[] */
+ L_tmp = 0;
+ L_tmp1 = 0;
+ for (i = 0; i < 64; i+=4)
+ {
+ L_tmp += (xn[i] * excf[i]);
+ L_tmp1 += (excf[i] * excf[i]);
+ L_tmp += (xn[i+1] * excf[i+1]);
+ L_tmp1 += (excf[i+1] * excf[i+1]);
+ L_tmp += (xn[i+2] * excf[i+2]);
+ L_tmp1 += (excf[i+2] * excf[i+2]);
+ L_tmp += (xn[i+3] * excf[i+3]);
+ L_tmp1 += (excf[i+3] * excf[i+3]);
+ }
- L_tmp = (L_tmp << 1) + 1;
- L_tmp1 = (L_tmp1 << 1) + 1;
+ L_tmp = (L_tmp << 1) + 1;
+ L_tmp1 = (L_tmp1 << 1) + 1;
- exp = norm_l(L_tmp);
- L_tmp = (L_tmp << exp);
- exp_corr = (30 - exp);
- corr = extract_h(L_tmp);
+ exp = norm_l(L_tmp);
+ L_tmp = (L_tmp << exp);
+ exp_corr = (30 - exp);
+ corr = extract_h(L_tmp);
- exp = norm_l(L_tmp1);
- L_tmp = (L_tmp1 << exp);
- exp_norm = (30 - exp);
+ exp = norm_l(L_tmp1);
+ L_tmp = (L_tmp1 << exp);
+ exp_norm = (30 - exp);
- Isqrt_n(&L_tmp, &exp_norm);
- norm = extract_h(L_tmp);
+ Isqrt_n(&L_tmp, &exp_norm);
+ norm = extract_h(L_tmp);
- /* Normalize correlation = correlation * (1/sqrt(energy)) */
+ /* Normalize correlation = correlation * (1/sqrt(energy)) */
- L_tmp = vo_L_mult(corr, norm);
+ L_tmp = vo_L_mult(corr, norm);
- L_tmp2 = exp_corr + exp_norm + scale;
- if(L_tmp2 < 0)
- {
- L_tmp2 = -L_tmp2;
- L_tmp = L_tmp >> L_tmp2;
- }
- else
- {
- L_tmp = L_tmp << L_tmp2;
- }
+ L_tmp2 = exp_corr + exp_norm + scale;
+ if(L_tmp2 < 0)
+ {
+ L_tmp2 = -L_tmp2;
+ L_tmp = L_tmp >> L_tmp2;
+ }
+ else
+ {
+ L_tmp = L_tmp << L_tmp2;
+ }
- corr_norm[t] = vo_round(L_tmp);
- /* modify the filtered excitation excf[] for the next iteration */
+ corr_norm[t] = vo_round(L_tmp);
+ /* modify the filtered excitation excf[] for the next iteration */
- if(t != t_max)
- {
- k = -(t + 1);
- tmp = exc[k];
- for (i = 63; i > 0; i--)
- {
- excf[i] = add1(vo_mult(tmp, h[i]), excf[i - 1]);
- }
- excf[0] = vo_mult(tmp, h[0]);
- }
- }
- return;
+ if(t != t_max)
+ {
+ k = -(t + 1);
+ tmp = exc[k];
+ for (i = 63; i > 0; i--)
+ {
+ excf[i] = add1(vo_mult(tmp, h[i]), excf[i - 1]);
+ }
+ excf[0] = vo_mult(tmp, h[0]);
+ }
+ }
+ return;
}
#endif
@@ -276,10 +276,10 @@
/* 1/4 resolution interpolation filter (-3 dB at 0.791*fs/2) in Q14 */
static Word16 inter4_1[4][8] =
{
- {-12, 420, -1732, 5429, 13418, -1242, 73, 32},
- {-26, 455, -2142, 9910, 9910, -2142, 455, -26},
- {32, 73, -1242, 13418, 5429, -1732, 420, -12},
- {206, -766, 1376, 14746, 1376, -766, 206, 0}
+ {-12, 420, -1732, 5429, 13418, -1242, 73, 32},
+ {-26, 455, -2142, 9910, 9910, -2142, 455, -26},
+ {32, 73, -1242, 13418, 5429, -1732, 420, -12},
+ {206, -766, 1376, 14746, 1376, -766, 206, 0}
};
/*** Coefficients in floating point
@@ -292,34 +292,34 @@
***/
static Word16 Interpol_4( /* (o) : interpolated value */
- Word16 * x, /* (i) : input vector */
- Word32 frac /* (i) : fraction (-4..+3) */
- )
+ Word16 * x, /* (i) : input vector */
+ Word32 frac /* (i) : fraction (-4..+3) */
+ )
{
- Word16 sum;
- Word32 k, L_sum;
- Word16 *ptr;
+ Word16 sum;
+ Word32 k, L_sum;
+ Word16 *ptr;
- if (frac < 0)
- {
- frac += UP_SAMP;
- x--;
- }
- x = x - L_INTERPOL1 + 1;
- k = UP_SAMP - 1 - frac;
- ptr = &(inter4_1[k][0]);
+ if (frac < 0)
+ {
+ frac += UP_SAMP;
+ x--;
+ }
+ x = x - L_INTERPOL1 + 1;
+ k = UP_SAMP - 1 - frac;
+ ptr = &(inter4_1[k][0]);
- L_sum = vo_mult32(x[0], (*ptr++));
- L_sum += vo_mult32(x[1], (*ptr++));
- L_sum += vo_mult32(x[2], (*ptr++));
- L_sum += vo_mult32(x[3], (*ptr++));
- L_sum += vo_mult32(x[4], (*ptr++));
- L_sum += vo_mult32(x[5], (*ptr++));
- L_sum += vo_mult32(x[6], (*ptr++));
- L_sum += vo_mult32(x[7], (*ptr++));
+ L_sum = vo_mult32(x[0], (*ptr++));
+ L_sum += vo_mult32(x[1], (*ptr++));
+ L_sum += vo_mult32(x[2], (*ptr++));
+ L_sum += vo_mult32(x[3], (*ptr++));
+ L_sum += vo_mult32(x[4], (*ptr++));
+ L_sum += vo_mult32(x[5], (*ptr++));
+ L_sum += vo_mult32(x[6], (*ptr++));
+ L_sum += vo_mult32(x[7], (*ptr++));
- sum = extract_h(L_add(L_shl2(L_sum, 2), 0x8000));
- return (sum);
+ sum = extract_h(L_add(L_shl2(L_sum, 2), 0x8000));
+ return (sum);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/pred_lt4.c b/media/libstagefright/codecs/amrwbenc/src/pred_lt4.c
index 8404cf9..386cab3 100644
--- a/media/libstagefright/codecs/amrwbenc/src/pred_lt4.c
+++ b/media/libstagefright/codecs/amrwbenc/src/pred_lt4.c
@@ -34,86 +34,86 @@
Word16 inter4_2[4][32] =
{
- {0,-2,4,-2,-10,38,-88,165,-275,424,-619,871,-1207,1699,-2598,5531,14031,-2147,780,-249,
- -16,153,-213,226,-209,175,-133,91,-55,28,-10,2},
+ {0,-2,4,-2,-10,38,-88,165,-275,424,-619,871,-1207,1699,-2598,5531,14031,-2147,780,-249,
+ -16,153,-213,226,-209,175,-133,91,-55,28,-10,2},
- {1,-7,19,-33,47,-52,43,-9,-60,175,-355,626,-1044,1749,-3267,10359,10359,-3267,1749,-1044,
- 626,-355,175,-60,-9,43,-52,47,-33,19, -7, 1},
+ {1,-7,19,-33,47,-52,43,-9,-60,175,-355,626,-1044,1749,-3267,10359,10359,-3267,1749,-1044,
+ 626,-355,175,-60,-9,43,-52,47,-33,19, -7, 1},
- {2,-10,28,-55,91,-133,175,-209,226,-213,153,-16,-249,780,-2147,14031,5531,-2598,1699,-1207,
- 871,-619,424,-275,165,-88,38,-10,-2,4,-2,0},
+ {2,-10,28,-55,91,-133,175,-209,226,-213,153,-16,-249,780,-2147,14031,5531,-2598,1699,-1207,
+ 871,-619,424,-275,165,-88,38,-10,-2,4,-2,0},
- {1,-7,22,-49,92,-153,231,-325,431,-544,656,-762,853,-923,968,15401,968,-923,853,-762,
- 656,-544,431,-325,231,-153,92,-49,22,-7, 1, 0}
+ {1,-7,22,-49,92,-153,231,-325,431,-544,656,-762,853,-923,968,15401,968,-923,853,-762,
+ 656,-544,431,-325,231,-153,92,-49,22,-7, 1, 0}
};
void Pred_lt4(
- Word16 exc[], /* in/out: excitation buffer */
- Word16 T0, /* input : integer pitch lag */
- Word16 frac, /* input : fraction of lag */
- Word16 L_subfr /* input : subframe size */
- )
+ Word16 exc[], /* in/out: excitation buffer */
+ Word16 T0, /* input : integer pitch lag */
+ Word16 frac, /* input : fraction of lag */
+ Word16 L_subfr /* input : subframe size */
+ )
{
- Word16 j, k, *x;
- Word32 L_sum;
- Word16 *ptr, *ptr1;
- Word16 *ptr2;
+ Word16 j, k, *x;
+ Word32 L_sum;
+ Word16 *ptr, *ptr1;
+ Word16 *ptr2;
- x = exc - T0;
- frac = -frac;
- if (frac < 0)
- {
- frac += UP_SAMP;
- x--;
- }
- x -= 15; /* x = L_INTERPOL2 - 1 */
- k = 3 - frac; /* k = UP_SAMP - 1 - frac */
+ x = exc - T0;
+ frac = -frac;
+ if (frac < 0)
+ {
+ frac += UP_SAMP;
+ x--;
+ }
+ x -= 15; /* x = L_INTERPOL2 - 1 */
+ k = 3 - frac; /* k = UP_SAMP - 1 - frac */
- ptr2 = &(inter4_2[k][0]);
- for (j = 0; j < L_subfr; j++)
- {
- ptr = ptr2;
- ptr1 = x;
- L_sum = vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum += vo_mult32((*ptr1++), (*ptr++));
+ ptr2 = &(inter4_2[k][0]);
+ for (j = 0; j < L_subfr; j++)
+ {
+ ptr = ptr2;
+ ptr1 = x;
+ L_sum = vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
+ L_sum += vo_mult32((*ptr1++), (*ptr++));
- L_sum = L_shl2(L_sum, 2);
- exc[j] = extract_h(L_add(L_sum, 0x8000));
- x++;
- }
+ L_sum = L_shl2(L_sum, 2);
+ exc[j] = extract_h(L_add(L_sum, 0x8000));
+ x++;
+ }
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/preemph.c b/media/libstagefright/codecs/amrwbenc/src/preemph.c
index d113288..70c8650 100644
--- a/media/libstagefright/codecs/amrwbenc/src/preemph.c
+++ b/media/libstagefright/codecs/amrwbenc/src/preemph.c
@@ -18,7 +18,7 @@
* File: preemph.c *
* *
* Description: Preemphasis: filtering through 1 - g z^-1 *
-* Preemph2 --> signal is multiplied by 2 *
+* Preemph2 --> signal is multiplied by 2 *
* *
************************************************************************/
@@ -26,65 +26,74 @@
#include "basic_op.h"
void Preemph(
- Word16 x[], /* (i/o) : input signal overwritten by the output */
- Word16 mu, /* (i) Q15 : preemphasis coefficient */
- Word16 lg, /* (i) : lenght of filtering */
- Word16 * mem /* (i/o) : memory (x[-1]) */
- )
+ Word16 x[], /* (i/o) : input signal overwritten by the output */
+ Word16 mu, /* (i) Q15 : preemphasis coefficient */
+ Word16 lg, /* (i) : lenght of filtering */
+ Word16 * mem /* (i/o) : memory (x[-1]) */
+ )
{
- Word16 temp;
- Word32 i, L_tmp;
+ Word16 temp;
+ Word32 i, L_tmp;
- temp = x[lg - 1];
+ temp = x[lg - 1];
- for (i = lg - 1; i > 0; i--)
- {
- L_tmp = L_deposit_h(x[i]);
- L_tmp -= (x[i - 1] * mu)<<1;
- x[i] = (L_tmp + 0x8000)>>16;
- }
+ for (i = lg - 1; i > 0; i--)
+ {
+ L_tmp = L_deposit_h(x[i]);
+ L_tmp -= (x[i - 1] * mu)<<1;
+ x[i] = (L_tmp + 0x8000)>>16;
+ }
- L_tmp = L_deposit_h(x[0]);
- L_tmp -= ((*mem) * mu)<<1;
- x[0] = (L_tmp + 0x8000)>>16;
+ L_tmp = L_deposit_h(x[0]);
+ L_tmp -= ((*mem) * mu)<<1;
+ x[0] = (L_tmp + 0x8000)>>16;
- *mem = temp;
+ *mem = temp;
- return;
+ return;
}
void Preemph2(
- Word16 x[], /* (i/o) : input signal overwritten by the output */
- Word16 mu, /* (i) Q15 : preemphasis coefficient */
- Word16 lg, /* (i) : lenght of filtering */
- Word16 * mem /* (i/o) : memory (x[-1]) */
- )
+ Word16 x[], /* (i/o) : input signal overwritten by the output */
+ Word16 mu, /* (i) Q15 : preemphasis coefficient */
+ Word16 lg, /* (i) : lenght of filtering */
+ Word16 * mem /* (i/o) : memory (x[-1]) */
+ )
{
- Word16 temp;
- Word32 i, L_tmp;
+ Word16 temp;
+ Word32 i, L_tmp;
- temp = x[lg - 1];
+ temp = x[lg - 1];
- for (i = (Word16) (lg - 1); i > 0; i--)
- {
- L_tmp = L_deposit_h(x[i]);
- L_tmp -= (x[i - 1] * mu)<<1;
- L_tmp = (L_tmp << 1);
- if (L_tmp > INT32_MAX - 0x8000) {
- L_tmp = INT32_MAX - 0x8000;
- }
- x[i] = (L_tmp + 0x8000)>>16;
- }
+ for (i = (Word16) (lg - 1); i > 0; i--)
+ {
+ L_tmp = L_deposit_h(x[i]);
+ L_tmp -= (x[i - 1] * mu)<<1; // only called with mu == 22282, so this won't overflow
+ if (L_tmp > INT32_MAX / 2) {
+ L_tmp = INT32_MAX / 2;
+ }
+ L_tmp = (L_tmp << 1);
+ if (L_tmp > INT32_MAX - 0x8000) {
+ L_tmp = INT32_MAX - 0x8000;
+ }
+ x[i] = (L_tmp + 0x8000)>>16;
+ }
- L_tmp = L_deposit_h(x[0]);
- L_tmp -= ((*mem) * mu)<<1;
- L_tmp = (L_tmp << 1);
- x[0] = (L_tmp + 0x8000)>>16;
+ L_tmp = L_deposit_h(x[0]);
+ L_tmp -= ((*mem) * mu)<<1;
+ if (L_tmp > INT32_MAX / 2) {
+ L_tmp = INT32_MAX / 2;
+ }
+ L_tmp = (L_tmp << 1);
+ if (L_tmp > INT32_MAX - 0x8000) {
+ L_tmp = INT32_MAX - 0x8000;
+ }
+ x[0] = (L_tmp + 0x8000)>>16;
- *mem = temp;
+ *mem = temp;
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/q_gain2.c b/media/libstagefright/codecs/amrwbenc/src/q_gain2.c
index e8ca043..bb797d8 100644
--- a/media/libstagefright/codecs/amrwbenc/src/q_gain2.c
+++ b/media/libstagefright/codecs/amrwbenc/src/q_gain2.c
@@ -45,300 +45,300 @@
void Init_Q_gain2(
- Word16 * mem /* output :static memory (2 words) */
- )
+ Word16 * mem /* output :static memory (2 words) */
+ )
{
- Word32 i;
+ Word32 i;
- /* 4nd order quantizer energy predictor (init to -14.0 in Q10) */
- for (i = 0; i < PRED_ORDER; i++)
- {
- mem[i] = -14336; /* past_qua_en[i] */
- }
+ /* 4nd order quantizer energy predictor (init to -14.0 in Q10) */
+ for (i = 0; i < PRED_ORDER; i++)
+ {
+ mem[i] = -14336; /* past_qua_en[i] */
+ }
- return;
+ return;
}
Word16 Q_gain2( /* Return index of quantization. */
- Word16 xn[], /* (i) Q_xn: Target vector. */
- Word16 y1[], /* (i) Q_xn: Adaptive codebook. */
- Word16 Q_xn, /* (i) : xn and y1 format */
- Word16 y2[], /* (i) Q9 : Filtered innovative vector. */
- Word16 code[], /* (i) Q9 : Innovative vector. */
- Word16 g_coeff[], /* (i) : Correlations <xn y1> <y1 y1> */
- /* Compute in G_pitch(). */
- Word16 L_subfr, /* (i) : Subframe lenght. */
- Word16 nbits, /* (i) : number of bits (6 or 7) */
- Word16 * gain_pit, /* (i/o)Q14: Pitch gain. */
- Word32 * gain_cod, /* (o) Q16 : Code gain. */
- Word16 gp_clip, /* (i) : Gp Clipping flag */
- Word16 * mem /* (i/o) : static memory (2 words) */
- )
+ Word16 xn[], /* (i) Q_xn: Target vector. */
+ Word16 y1[], /* (i) Q_xn: Adaptive codebook. */
+ Word16 Q_xn, /* (i) : xn and y1 format */
+ Word16 y2[], /* (i) Q9 : Filtered innovative vector. */
+ Word16 code[], /* (i) Q9 : Innovative vector. */
+ Word16 g_coeff[], /* (i) : Correlations <xn y1> <y1 y1> */
+ /* Compute in G_pitch(). */
+ Word16 L_subfr, /* (i) : Subframe lenght. */
+ Word16 nbits, /* (i) : number of bits (6 or 7) */
+ Word16 * gain_pit, /* (i/o)Q14: Pitch gain. */
+ Word32 * gain_cod, /* (o) Q16 : Code gain. */
+ Word16 gp_clip, /* (i) : Gp Clipping flag */
+ Word16 * mem /* (i/o) : static memory (2 words) */
+ )
{
- Word16 index, *p, min_ind, size;
- Word16 exp, frac, gcode0, exp_gcode0, e_max, exp_code, qua_ener;
- Word16 g_pitch, g2_pitch, g_code, g_pit_cod, g2_code, g2_code_lo;
- Word16 coeff[5], coeff_lo[5], exp_coeff[5];
- Word16 exp_max[5];
- Word32 i, j, L_tmp, dist_min;
- Word16 *past_qua_en, *t_qua_gain;
+ Word16 index, *p, min_ind, size;
+ Word16 exp, frac, gcode0, exp_gcode0, e_max, exp_code, qua_ener;
+ Word16 g_pitch, g2_pitch, g_code, g_pit_cod, g2_code, g2_code_lo;
+ Word16 coeff[5], coeff_lo[5], exp_coeff[5];
+ Word16 exp_max[5];
+ Word32 i, j, L_tmp, dist_min;
+ Word16 *past_qua_en, *t_qua_gain;
- past_qua_en = mem;
+ past_qua_en = mem;
- /*-----------------------------------------------------------------*
- * - Find the initial quantization pitch index *
- * - Set gains search range *
- *-----------------------------------------------------------------*/
- if (nbits == 6)
- {
- t_qua_gain = t_qua_gain6b;
- min_ind = 0;
- size = RANGE;
+ /*-----------------------------------------------------------------*
+ * - Find the initial quantization pitch index *
+ * - Set gains search range *
+ *-----------------------------------------------------------------*/
+ if (nbits == 6)
+ {
+ t_qua_gain = t_qua_gain6b;
+ min_ind = 0;
+ size = RANGE;
- if(gp_clip == 1)
- {
- size = size - 16; /* limit gain pitch to 1.0 */
- }
- } else
- {
- t_qua_gain = t_qua_gain7b;
+ if(gp_clip == 1)
+ {
+ size = size - 16; /* limit gain pitch to 1.0 */
+ }
+ } else
+ {
+ t_qua_gain = t_qua_gain7b;
- p = t_qua_gain7b + RANGE; /* pt at 1/4th of table */
+ p = t_qua_gain7b + RANGE; /* pt at 1/4th of table */
- j = nb_qua_gain7b - RANGE;
+ j = nb_qua_gain7b - RANGE;
- if (gp_clip == 1)
- {
- j = j - 27; /* limit gain pitch to 1.0 */
- }
- min_ind = 0;
- g_pitch = *gain_pit;
+ if (gp_clip == 1)
+ {
+ j = j - 27; /* limit gain pitch to 1.0 */
+ }
+ min_ind = 0;
+ g_pitch = *gain_pit;
- for (i = 0; i < j; i++, p += 2)
- {
- if (g_pitch > *p)
- {
- min_ind = min_ind + 1;
- }
- }
- size = RANGE;
- }
+ for (i = 0; i < j; i++, p += 2)
+ {
+ if (g_pitch > *p)
+ {
+ min_ind = min_ind + 1;
+ }
+ }
+ size = RANGE;
+ }
- /*------------------------------------------------------------------*
- * Compute coefficient need for the quantization. *
- * *
- * coeff[0] = y1 y1 *
- * coeff[1] = -2 xn y1 *
- * coeff[2] = y2 y2 *
- * coeff[3] = -2 xn y2 *
- * coeff[4] = 2 y1 y2 *
- * *
- * Product <y1 y1> and <xn y1> have been compute in G_pitch() and *
- * are in vector g_coeff[]. *
- *------------------------------------------------------------------*/
+ /*------------------------------------------------------------------*
+ * Compute coefficient need for the quantization. *
+ * *
+ * coeff[0] = y1 y1 *
+ * coeff[1] = -2 xn y1 *
+ * coeff[2] = y2 y2 *
+ * coeff[3] = -2 xn y2 *
+ * coeff[4] = 2 y1 y2 *
+ * *
+ * Product <y1 y1> and <xn y1> have been compute in G_pitch() and *
+ * are in vector g_coeff[]. *
+ *------------------------------------------------------------------*/
- coeff[0] = g_coeff[0];
- exp_coeff[0] = g_coeff[1];
- coeff[1] = negate(g_coeff[2]); /* coeff[1] = -2 xn y1 */
- exp_coeff[1] = g_coeff[3] + 1;
+ coeff[0] = g_coeff[0];
+ exp_coeff[0] = g_coeff[1];
+ coeff[1] = negate(g_coeff[2]); /* coeff[1] = -2 xn y1 */
+ exp_coeff[1] = g_coeff[3] + 1;
- /* Compute scalar product <y2[],y2[]> */
+ /* Compute scalar product <y2[],y2[]> */
#ifdef ASM_OPT /* asm optimization branch */
- coeff[2] = extract_h(Dot_product12_asm(y2, y2, L_subfr, &exp));
+ coeff[2] = extract_h(Dot_product12_asm(y2, y2, L_subfr, &exp));
#else
- coeff[2] = extract_h(Dot_product12(y2, y2, L_subfr, &exp));
+ coeff[2] = extract_h(Dot_product12(y2, y2, L_subfr, &exp));
#endif
- exp_coeff[2] = (exp - 18) + (Q_xn << 1); /* -18 (y2 Q9) */
+ exp_coeff[2] = (exp - 18) + (Q_xn << 1); /* -18 (y2 Q9) */
- /* Compute scalar product -2*<xn[],y2[]> */
+ /* Compute scalar product -2*<xn[],y2[]> */
#ifdef ASM_OPT /* asm optimization branch */
- coeff[3] = extract_h(L_negate(Dot_product12_asm(xn, y2, L_subfr, &exp)));
+ coeff[3] = extract_h(L_negate(Dot_product12_asm(xn, y2, L_subfr, &exp)));
#else
- coeff[3] = extract_h(L_negate(Dot_product12(xn, y2, L_subfr, &exp)));
+ coeff[3] = extract_h(L_negate(Dot_product12(xn, y2, L_subfr, &exp)));
#endif
- exp_coeff[3] = (exp - 8) + Q_xn; /* -9 (y2 Q9), +1 (2 xn y2) */
+ exp_coeff[3] = (exp - 8) + Q_xn; /* -9 (y2 Q9), +1 (2 xn y2) */
- /* Compute scalar product 2*<y1[],y2[]> */
+ /* Compute scalar product 2*<y1[],y2[]> */
#ifdef ASM_OPT /* asm optimization branch */
- coeff[4] = extract_h(Dot_product12_asm(y1, y2, L_subfr, &exp));
+ coeff[4] = extract_h(Dot_product12_asm(y1, y2, L_subfr, &exp));
#else
- coeff[4] = extract_h(Dot_product12(y1, y2, L_subfr, &exp));
+ coeff[4] = extract_h(Dot_product12(y1, y2, L_subfr, &exp));
#endif
- exp_coeff[4] = (exp - 8) + Q_xn; /* -9 (y2 Q9), +1 (2 y1 y2) */
+ exp_coeff[4] = (exp - 8) + Q_xn; /* -9 (y2 Q9), +1 (2 y1 y2) */
- /*-----------------------------------------------------------------*
- * Find energy of code and compute: *
- * *
- * L_tmp = MEAN_ENER - 10log10(energy of code/ L_subfr) *
- * = MEAN_ENER - 3.0103*log2(energy of code/ L_subfr) *
- *-----------------------------------------------------------------*/
+ /*-----------------------------------------------------------------*
+ * Find energy of code and compute: *
+ * *
+ * L_tmp = MEAN_ENER - 10log10(energy of code/ L_subfr) *
+ * = MEAN_ENER - 3.0103*log2(energy of code/ L_subfr) *
+ *-----------------------------------------------------------------*/
#ifdef ASM_OPT /* asm optimization branch */
- L_tmp = Dot_product12_asm(code, code, L_subfr, &exp_code);
+ L_tmp = Dot_product12_asm(code, code, L_subfr, &exp_code);
#else
- L_tmp = Dot_product12(code, code, L_subfr, &exp_code);
+ L_tmp = Dot_product12(code, code, L_subfr, &exp_code);
#endif
- /* exp_code: -18 (code in Q9), -6 (/L_subfr), -31 (L_tmp Q31->Q0) */
- exp_code = (exp_code - (18 + 6 + 31));
+ /* exp_code: -18 (code in Q9), -6 (/L_subfr), -31 (L_tmp Q31->Q0) */
+ exp_code = (exp_code - (18 + 6 + 31));
- Log2(L_tmp, &exp, &frac);
- exp += exp_code;
- L_tmp = Mpy_32_16(exp, frac, -24660); /* x -3.0103(Q13) -> Q14 */
+ Log2(L_tmp, &exp, &frac);
+ exp += exp_code;
+ L_tmp = Mpy_32_16(exp, frac, -24660); /* x -3.0103(Q13) -> Q14 */
- L_tmp += (MEAN_ENER * 8192)<<1; /* + MEAN_ENER in Q14 */
+ L_tmp += (MEAN_ENER * 8192)<<1; /* + MEAN_ENER in Q14 */
- /*-----------------------------------------------------------------*
- * Compute gcode0. *
- * = Sum(i=0,1) pred[i]*past_qua_en[i] + mean_ener - ener_code *
- *-----------------------------------------------------------------*/
- L_tmp = (L_tmp << 10); /* From Q14 to Q24 */
- L_tmp += (pred[0] * past_qua_en[0])<<1; /* Q13*Q10 -> Q24 */
- L_tmp += (pred[1] * past_qua_en[1])<<1; /* Q13*Q10 -> Q24 */
- L_tmp += (pred[2] * past_qua_en[2])<<1; /* Q13*Q10 -> Q24 */
- L_tmp += (pred[3] * past_qua_en[3])<<1; /* Q13*Q10 -> Q24 */
+ /*-----------------------------------------------------------------*
+ * Compute gcode0. *
+ * = Sum(i=0,1) pred[i]*past_qua_en[i] + mean_ener - ener_code *
+ *-----------------------------------------------------------------*/
+ L_tmp = (L_tmp << 10); /* From Q14 to Q24 */
+ L_tmp += (pred[0] * past_qua_en[0])<<1; /* Q13*Q10 -> Q24 */
+ L_tmp += (pred[1] * past_qua_en[1])<<1; /* Q13*Q10 -> Q24 */
+ L_tmp += (pred[2] * past_qua_en[2])<<1; /* Q13*Q10 -> Q24 */
+ L_tmp += (pred[3] * past_qua_en[3])<<1; /* Q13*Q10 -> Q24 */
- gcode0 = extract_h(L_tmp); /* From Q24 to Q8 */
+ gcode0 = extract_h(L_tmp); /* From Q24 to Q8 */
- /*-----------------------------------------------------------------*
- * gcode0 = pow(10.0, gcode0/20) *
- * = pow(2, 3.321928*gcode0/20) *
- * = pow(2, 0.166096*gcode0) *
- *-----------------------------------------------------------------*/
+ /*-----------------------------------------------------------------*
+ * gcode0 = pow(10.0, gcode0/20) *
+ * = pow(2, 3.321928*gcode0/20) *
+ * = pow(2, 0.166096*gcode0) *
+ *-----------------------------------------------------------------*/
- L_tmp = vo_L_mult(gcode0, 5443); /* *0.166096 in Q15 -> Q24 */
- L_tmp = L_tmp >> 8; /* From Q24 to Q16 */
- VO_L_Extract(L_tmp, &exp_gcode0, &frac); /* Extract exponent of gcode0 */
+ L_tmp = vo_L_mult(gcode0, 5443); /* *0.166096 in Q15 -> Q24 */
+ L_tmp = L_tmp >> 8; /* From Q24 to Q16 */
+ VO_L_Extract(L_tmp, &exp_gcode0, &frac); /* Extract exponent of gcode0 */
- gcode0 = (Word16)(Pow2(14, frac)); /* Put 14 as exponent so that */
- /* output of Pow2() will be: */
- /* 16384 < Pow2() <= 32767 */
- exp_gcode0 -= 14;
+ gcode0 = (Word16)(Pow2(14, frac)); /* Put 14 as exponent so that */
+ /* output of Pow2() will be: */
+ /* 16384 < Pow2() <= 32767 */
+ exp_gcode0 -= 14;
- /*-------------------------------------------------------------------------*
- * Find the best quantizer *
- * ~~~~~~~~~~~~~~~~~~~~~~~ *
- * Before doing the computation we need to aling exponents of coeff[] *
- * to be sure to have the maximum precision. *
- * *
- * In the table the pitch gains are in Q14, the code gains are in Q11 and *
- * are multiply by gcode0 which have been multiply by 2^exp_gcode0. *
- * Also when we compute g_pitch*g_pitch, g_code*g_code and g_pitch*g_code *
- * we divide by 2^15. *
- * Considering all the scaling above we have: *
- * *
- * exp_code = exp_gcode0-11+15 = exp_gcode0+4 *
- * *
- * g_pitch*g_pitch = -14-14+15 *
- * g_pitch = -14 *
- * g_code*g_code = (2*exp_code)+15 *
- * g_code = exp_code *
- * g_pitch*g_code = -14 + exp_code +15 *
- * *
- * g_pitch*g_pitch * coeff[0] ;exp_max0 = exp_coeff[0] - 13 *
- * g_pitch * coeff[1] ;exp_max1 = exp_coeff[1] - 14 *
- * g_code*g_code * coeff[2] ;exp_max2 = exp_coeff[2] +15+(2*exp_code) *
- * g_code * coeff[3] ;exp_max3 = exp_coeff[3] + exp_code *
- * g_pitch*g_code * coeff[4] ;exp_max4 = exp_coeff[4] + 1 + exp_code *
- *-------------------------------------------------------------------------*/
+ /*-------------------------------------------------------------------------*
+ * Find the best quantizer *
+ * ~~~~~~~~~~~~~~~~~~~~~~~ *
+ * Before doing the computation we need to aling exponents of coeff[] *
+ * to be sure to have the maximum precision. *
+ * *
+ * In the table the pitch gains are in Q14, the code gains are in Q11 and *
+ * are multiply by gcode0 which have been multiply by 2^exp_gcode0. *
+ * Also when we compute g_pitch*g_pitch, g_code*g_code and g_pitch*g_code *
+ * we divide by 2^15. *
+ * Considering all the scaling above we have: *
+ * *
+ * exp_code = exp_gcode0-11+15 = exp_gcode0+4 *
+ * *
+ * g_pitch*g_pitch = -14-14+15 *
+ * g_pitch = -14 *
+ * g_code*g_code = (2*exp_code)+15 *
+ * g_code = exp_code *
+ * g_pitch*g_code = -14 + exp_code +15 *
+ * *
+ * g_pitch*g_pitch * coeff[0] ;exp_max0 = exp_coeff[0] - 13 *
+ * g_pitch * coeff[1] ;exp_max1 = exp_coeff[1] - 14 *
+ * g_code*g_code * coeff[2] ;exp_max2 = exp_coeff[2] +15+(2*exp_code) *
+ * g_code * coeff[3] ;exp_max3 = exp_coeff[3] + exp_code *
+ * g_pitch*g_code * coeff[4] ;exp_max4 = exp_coeff[4] + 1 + exp_code *
+ *-------------------------------------------------------------------------*/
- exp_code = (exp_gcode0 + 4);
- exp_max[0] = (exp_coeff[0] - 13);
- exp_max[1] = (exp_coeff[1] - 14);
- exp_max[2] = (exp_coeff[2] + (15 + (exp_code << 1)));
- exp_max[3] = (exp_coeff[3] + exp_code);
- exp_max[4] = (exp_coeff[4] + (1 + exp_code));
+ exp_code = (exp_gcode0 + 4);
+ exp_max[0] = (exp_coeff[0] - 13);
+ exp_max[1] = (exp_coeff[1] - 14);
+ exp_max[2] = (exp_coeff[2] + (15 + (exp_code << 1)));
+ exp_max[3] = (exp_coeff[3] + exp_code);
+ exp_max[4] = (exp_coeff[4] + (1 + exp_code));
- /* Find maximum exponant */
+ /* Find maximum exponant */
- e_max = exp_max[0];
- for (i = 1; i < 5; i++)
- {
- if(exp_max[i] > e_max)
- {
- e_max = exp_max[i];
- }
- }
+ e_max = exp_max[0];
+ for (i = 1; i < 5; i++)
+ {
+ if(exp_max[i] > e_max)
+ {
+ e_max = exp_max[i];
+ }
+ }
- /* align coeff[] and save in special 32 bit double precision */
+ /* align coeff[] and save in special 32 bit double precision */
- for (i = 0; i < 5; i++)
- {
- j = add1(vo_sub(e_max, exp_max[i]), 2);/* /4 to avoid overflow */
- L_tmp = L_deposit_h(coeff[i]);
- L_tmp = L_shr(L_tmp, j);
- VO_L_Extract(L_tmp, &coeff[i], &coeff_lo[i]);
- coeff_lo[i] = (coeff_lo[i] >> 3); /* lo >> 3 */
- }
+ for (i = 0; i < 5; i++)
+ {
+ j = add1(vo_sub(e_max, exp_max[i]), 2);/* /4 to avoid overflow */
+ L_tmp = L_deposit_h(coeff[i]);
+ L_tmp = L_shr(L_tmp, j);
+ VO_L_Extract(L_tmp, &coeff[i], &coeff_lo[i]);
+ coeff_lo[i] = (coeff_lo[i] >> 3); /* lo >> 3 */
+ }
- /* Codebook search */
- dist_min = MAX_32;
- p = &t_qua_gain[min_ind << 1];
+ /* Codebook search */
+ dist_min = MAX_32;
+ p = &t_qua_gain[min_ind << 1];
- index = 0;
- for (i = 0; i < size; i++)
- {
- g_pitch = *p++;
- g_code = *p++;
+ index = 0;
+ for (i = 0; i < size; i++)
+ {
+ g_pitch = *p++;
+ g_code = *p++;
- g_code = ((g_code * gcode0) + 0x4000)>>15;
- g2_pitch = ((g_pitch * g_pitch) + 0x4000)>>15;
- g_pit_cod = ((g_code * g_pitch) + 0x4000)>>15;
- L_tmp = (g_code * g_code)<<1;
- VO_L_Extract(L_tmp, &g2_code, &g2_code_lo);
+ g_code = ((g_code * gcode0) + 0x4000)>>15;
+ g2_pitch = ((g_pitch * g_pitch) + 0x4000)>>15;
+ g_pit_cod = ((g_code * g_pitch) + 0x4000)>>15;
+ L_tmp = (g_code * g_code)<<1;
+ VO_L_Extract(L_tmp, &g2_code, &g2_code_lo);
- L_tmp = (coeff[2] * g2_code_lo)<<1;
- L_tmp = (L_tmp >> 3);
- L_tmp += (coeff_lo[0] * g2_pitch)<<1;
- L_tmp += (coeff_lo[1] * g_pitch)<<1;
- L_tmp += (coeff_lo[2] * g2_code)<<1;
- L_tmp += (coeff_lo[3] * g_code)<<1;
- L_tmp += (coeff_lo[4] * g_pit_cod)<<1;
- L_tmp = (L_tmp >> 12);
- L_tmp += (coeff[0] * g2_pitch)<<1;
- L_tmp += (coeff[1] * g_pitch)<<1;
- L_tmp += (coeff[2] * g2_code)<<1;
- L_tmp += (coeff[3] * g_code)<<1;
- L_tmp += (coeff[4] * g_pit_cod)<<1;
+ L_tmp = (coeff[2] * g2_code_lo)<<1;
+ L_tmp = (L_tmp >> 3);
+ L_tmp += (coeff_lo[0] * g2_pitch)<<1;
+ L_tmp += (coeff_lo[1] * g_pitch)<<1;
+ L_tmp += (coeff_lo[2] * g2_code)<<1;
+ L_tmp += (coeff_lo[3] * g_code)<<1;
+ L_tmp += (coeff_lo[4] * g_pit_cod)<<1;
+ L_tmp = (L_tmp >> 12);
+ L_tmp += (coeff[0] * g2_pitch)<<1;
+ L_tmp += (coeff[1] * g_pitch)<<1;
+ L_tmp += (coeff[2] * g2_code)<<1;
+ L_tmp += (coeff[3] * g_code)<<1;
+ L_tmp += (coeff[4] * g_pit_cod)<<1;
- if(L_tmp < dist_min)
- {
- dist_min = L_tmp;
- index = i;
- }
- }
+ if(L_tmp < dist_min)
+ {
+ dist_min = L_tmp;
+ index = i;
+ }
+ }
- /* Read the quantized gains */
- index = index + min_ind;
- p = &t_qua_gain[(index + index)];
- *gain_pit = *p++; /* selected pitch gain in Q14 */
- g_code = *p++; /* selected code gain in Q11 */
+ /* Read the quantized gains */
+ index = index + min_ind;
+ p = &t_qua_gain[(index + index)];
+ *gain_pit = *p++; /* selected pitch gain in Q14 */
+ g_code = *p++; /* selected code gain in Q11 */
- L_tmp = vo_L_mult(g_code, gcode0); /* Q11*Q0 -> Q12 */
- L_tmp = L_shl(L_tmp, (exp_gcode0 + 4)); /* Q12 -> Q16 */
+ L_tmp = vo_L_mult(g_code, gcode0); /* Q11*Q0 -> Q12 */
+ L_tmp = L_shl(L_tmp, (exp_gcode0 + 4)); /* Q12 -> Q16 */
- *gain_cod = L_tmp; /* gain of code in Q16 */
+ *gain_cod = L_tmp; /* gain of code in Q16 */
- /*---------------------------------------------------*
- * qua_ener = 20*log10(g_code) *
- * = 6.0206*log2(g_code) *
- * = 6.0206*(log2(g_codeQ11) - 11) *
- *---------------------------------------------------*/
+ /*---------------------------------------------------*
+ * qua_ener = 20*log10(g_code) *
+ * = 6.0206*log2(g_code) *
+ * = 6.0206*(log2(g_codeQ11) - 11) *
+ *---------------------------------------------------*/
- L_tmp = L_deposit_l(g_code);
- Log2(L_tmp, &exp, &frac);
- exp -= 11;
- L_tmp = Mpy_32_16(exp, frac, 24660); /* x 6.0206 in Q12 */
+ L_tmp = L_deposit_l(g_code);
+ Log2(L_tmp, &exp, &frac);
+ exp -= 11;
+ L_tmp = Mpy_32_16(exp, frac, 24660); /* x 6.0206 in Q12 */
- qua_ener = (Word16)(L_tmp >> 3); /* result in Q10 */
+ qua_ener = (Word16)(L_tmp >> 3); /* result in Q10 */
- /* update table of past quantized energies */
+ /* update table of past quantized energies */
- past_qua_en[3] = past_qua_en[2];
- past_qua_en[2] = past_qua_en[1];
- past_qua_en[1] = past_qua_en[0];
- past_qua_en[0] = qua_ener;
+ past_qua_en[3] = past_qua_en[2];
+ past_qua_en[2] = past_qua_en[1];
+ past_qua_en[1] = past_qua_en[0];
+ past_qua_en[0] = qua_ener;
- return (index);
+ return (index);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/q_pulse.c b/media/libstagefright/codecs/amrwbenc/src/q_pulse.c
index d658602..fe0bdda 100644
--- a/media/libstagefright/codecs/amrwbenc/src/q_pulse.c
+++ b/media/libstagefright/codecs/amrwbenc/src/q_pulse.c
@@ -29,372 +29,372 @@
#define NB_POS 16 /* pos in track, mask for sign bit */
Word32 quant_1p_N1( /* (o) return N+1 bits */
- Word16 pos, /* (i) position of the pulse */
- Word16 N) /* (i) number of bits for position */
+ Word16 pos, /* (i) position of the pulse */
+ Word16 N) /* (i) number of bits for position */
{
- Word16 mask;
- Word32 index;
+ Word16 mask;
+ Word32 index;
- mask = (1 << N) - 1; /* mask = ((1<<N)-1); */
- /*-------------------------------------------------------*
- * Quantization of 1 pulse with N+1 bits: *
- *-------------------------------------------------------*/
- index = L_deposit_l((Word16) (pos & mask));
- if ((pos & NB_POS) != 0)
- {
- index = vo_L_add(index, L_deposit_l(1 << N)); /* index += 1 << N; */
- }
- return (index);
+ mask = (1 << N) - 1; /* mask = ((1<<N)-1); */
+ /*-------------------------------------------------------*
+ * Quantization of 1 pulse with N+1 bits: *
+ *-------------------------------------------------------*/
+ index = L_deposit_l((Word16) (pos & mask));
+ if ((pos & NB_POS) != 0)
+ {
+ index = vo_L_add(index, L_deposit_l(1 << N)); /* index += 1 << N; */
+ }
+ return (index);
}
Word32 quant_2p_2N1( /* (o) return (2*N)+1 bits */
- Word16 pos1, /* (i) position of the pulse 1 */
- Word16 pos2, /* (i) position of the pulse 2 */
- Word16 N) /* (i) number of bits for position */
+ Word16 pos1, /* (i) position of the pulse 1 */
+ Word16 pos2, /* (i) position of the pulse 2 */
+ Word16 N) /* (i) number of bits for position */
{
- Word16 mask, tmp;
- Word32 index;
- mask = (1 << N) - 1; /* mask = ((1<<N)-1); */
- /*-------------------------------------------------------*
- * Quantization of 2 pulses with 2*N+1 bits: *
- *-------------------------------------------------------*/
- if (((pos2 ^ pos1) & NB_POS) == 0)
- {
- /* sign of 1st pulse == sign of 2th pulse */
- if(pos1 <= pos2) /* ((pos1 - pos2) <= 0) */
- {
- /* index = ((pos1 & mask) << N) + (pos2 & mask); */
- index = L_deposit_l(add1((((Word16) (pos1 & mask)) << N), ((Word16) (pos2 & mask))));
- } else
- {
- /* ((pos2 & mask) << N) + (pos1 & mask); */
- index = L_deposit_l(add1((((Word16) (pos2 & mask)) << N), ((Word16) (pos1 & mask))));
- }
- if ((pos1 & NB_POS) != 0)
- {
- tmp = (N << 1);
- index = vo_L_add(index, (1L << tmp)); /* index += 1 << (2*N); */
- }
- } else
- {
- /* sign of 1st pulse != sign of 2th pulse */
- if (vo_sub((Word16) (pos1 & mask), (Word16) (pos2 & mask)) <= 0)
- {
- /* index = ((pos2 & mask) << N) + (pos1 & mask); */
- index = L_deposit_l(add1((((Word16) (pos2 & mask)) << N), ((Word16) (pos1 & mask))));
- if ((pos2 & NB_POS) != 0)
- {
- tmp = (N << 1); /* index += 1 << (2*N); */
- index = vo_L_add(index, (1L << tmp));
- }
- } else
- {
- /* index = ((pos1 & mask) << N) + (pos2 & mask); */
- index = L_deposit_l(add1((((Word16) (pos1 & mask)) << N), ((Word16) (pos2 & mask))));
- if ((pos1 & NB_POS) != 0)
- {
- tmp = (N << 1);
- index = vo_L_add(index, (1 << tmp)); /* index += 1 << (2*N); */
- }
- }
- }
- return (index);
+ Word16 mask, tmp;
+ Word32 index;
+ mask = (1 << N) - 1; /* mask = ((1<<N)-1); */
+ /*-------------------------------------------------------*
+ * Quantization of 2 pulses with 2*N+1 bits: *
+ *-------------------------------------------------------*/
+ if (((pos2 ^ pos1) & NB_POS) == 0)
+ {
+ /* sign of 1st pulse == sign of 2th pulse */
+ if(pos1 <= pos2) /* ((pos1 - pos2) <= 0) */
+ {
+ /* index = ((pos1 & mask) << N) + (pos2 & mask); */
+ index = L_deposit_l(add1((((Word16) (pos1 & mask)) << N), ((Word16) (pos2 & mask))));
+ } else
+ {
+ /* ((pos2 & mask) << N) + (pos1 & mask); */
+ index = L_deposit_l(add1((((Word16) (pos2 & mask)) << N), ((Word16) (pos1 & mask))));
+ }
+ if ((pos1 & NB_POS) != 0)
+ {
+ tmp = (N << 1);
+ index = vo_L_add(index, (1L << tmp)); /* index += 1 << (2*N); */
+ }
+ } else
+ {
+ /* sign of 1st pulse != sign of 2th pulse */
+ if (vo_sub((Word16) (pos1 & mask), (Word16) (pos2 & mask)) <= 0)
+ {
+ /* index = ((pos2 & mask) << N) + (pos1 & mask); */
+ index = L_deposit_l(add1((((Word16) (pos2 & mask)) << N), ((Word16) (pos1 & mask))));
+ if ((pos2 & NB_POS) != 0)
+ {
+ tmp = (N << 1); /* index += 1 << (2*N); */
+ index = vo_L_add(index, (1L << tmp));
+ }
+ } else
+ {
+ /* index = ((pos1 & mask) << N) + (pos2 & mask); */
+ index = L_deposit_l(add1((((Word16) (pos1 & mask)) << N), ((Word16) (pos2 & mask))));
+ if ((pos1 & NB_POS) != 0)
+ {
+ tmp = (N << 1);
+ index = vo_L_add(index, (1 << tmp)); /* index += 1 << (2*N); */
+ }
+ }
+ }
+ return (index);
}
Word32 quant_3p_3N1( /* (o) return (3*N)+1 bits */
- Word16 pos1, /* (i) position of the pulse 1 */
- Word16 pos2, /* (i) position of the pulse 2 */
- Word16 pos3, /* (i) position of the pulse 3 */
- Word16 N) /* (i) number of bits for position */
+ Word16 pos1, /* (i) position of the pulse 1 */
+ Word16 pos2, /* (i) position of the pulse 2 */
+ Word16 pos3, /* (i) position of the pulse 3 */
+ Word16 N) /* (i) number of bits for position */
{
- Word16 nb_pos;
- Word32 index;
+ Word16 nb_pos;
+ Word32 index;
- nb_pos =(1 <<(N - 1)); /* nb_pos = (1<<(N-1)); */
- /*-------------------------------------------------------*
- * Quantization of 3 pulses with 3*N+1 bits: *
- *-------------------------------------------------------*/
- if (((pos1 ^ pos2) & nb_pos) == 0)
- {
- index = quant_2p_2N1(pos1, pos2, sub(N, 1)); /* index = quant_2p_2N1(pos1, pos2, (N-1)); */
- /* index += (pos1 & nb_pos) << N; */
- index = vo_L_add(index, (L_deposit_l((Word16) (pos1 & nb_pos)) << N));
- /* index += quant_1p_N1(pos3, N) << (2*N); */
- index = vo_L_add(index, (quant_1p_N1(pos3, N)<<(N << 1)));
+ nb_pos =(1 <<(N - 1)); /* nb_pos = (1<<(N-1)); */
+ /*-------------------------------------------------------*
+ * Quantization of 3 pulses with 3*N+1 bits: *
+ *-------------------------------------------------------*/
+ if (((pos1 ^ pos2) & nb_pos) == 0)
+ {
+ index = quant_2p_2N1(pos1, pos2, sub(N, 1)); /* index = quant_2p_2N1(pos1, pos2, (N-1)); */
+ /* index += (pos1 & nb_pos) << N; */
+ index = vo_L_add(index, (L_deposit_l((Word16) (pos1 & nb_pos)) << N));
+ /* index += quant_1p_N1(pos3, N) << (2*N); */
+ index = vo_L_add(index, (quant_1p_N1(pos3, N)<<(N << 1)));
- } else if (((pos1 ^ pos3) & nb_pos) == 0)
- {
- index = quant_2p_2N1(pos1, pos3, sub(N, 1)); /* index = quant_2p_2N1(pos1, pos3, (N-1)); */
- index = vo_L_add(index, (L_deposit_l((Word16) (pos1 & nb_pos)) << N));
- /* index += (pos1 & nb_pos) << N; */
- index = vo_L_add(index, (quant_1p_N1(pos2, N) << (N << 1)));
- /* index += quant_1p_N1(pos2, N) <<
- * (2*N); */
- } else
- {
- index = quant_2p_2N1(pos2, pos3, (N - 1)); /* index = quant_2p_2N1(pos2, pos3, (N-1)); */
- /* index += (pos2 & nb_pos) << N; */
- index = vo_L_add(index, (L_deposit_l((Word16) (pos2 & nb_pos)) << N));
- /* index += quant_1p_N1(pos1, N) << (2*N); */
- index = vo_L_add(index, (quant_1p_N1(pos1, N) << (N << 1)));
- }
- return (index);
+ } else if (((pos1 ^ pos3) & nb_pos) == 0)
+ {
+ index = quant_2p_2N1(pos1, pos3, sub(N, 1)); /* index = quant_2p_2N1(pos1, pos3, (N-1)); */
+ index = vo_L_add(index, (L_deposit_l((Word16) (pos1 & nb_pos)) << N));
+ /* index += (pos1 & nb_pos) << N; */
+ index = vo_L_add(index, (quant_1p_N1(pos2, N) << (N << 1)));
+ /* index += quant_1p_N1(pos2, N) <<
+ * (2*N); */
+ } else
+ {
+ index = quant_2p_2N1(pos2, pos3, (N - 1)); /* index = quant_2p_2N1(pos2, pos3, (N-1)); */
+ /* index += (pos2 & nb_pos) << N; */
+ index = vo_L_add(index, (L_deposit_l((Word16) (pos2 & nb_pos)) << N));
+ /* index += quant_1p_N1(pos1, N) << (2*N); */
+ index = vo_L_add(index, (quant_1p_N1(pos1, N) << (N << 1)));
+ }
+ return (index);
}
Word32 quant_4p_4N1( /* (o) return (4*N)+1 bits */
- Word16 pos1, /* (i) position of the pulse 1 */
- Word16 pos2, /* (i) position of the pulse 2 */
- Word16 pos3, /* (i) position of the pulse 3 */
- Word16 pos4, /* (i) position of the pulse 4 */
- Word16 N) /* (i) number of bits for position */
+ Word16 pos1, /* (i) position of the pulse 1 */
+ Word16 pos2, /* (i) position of the pulse 2 */
+ Word16 pos3, /* (i) position of the pulse 3 */
+ Word16 pos4, /* (i) position of the pulse 4 */
+ Word16 N) /* (i) number of bits for position */
{
- Word16 nb_pos;
- Word32 index;
+ Word16 nb_pos;
+ Word32 index;
- nb_pos = 1 << (N - 1); /* nb_pos = (1<<(N-1)); */
- /*-------------------------------------------------------*
- * Quantization of 4 pulses with 4*N+1 bits: *
- *-------------------------------------------------------*/
- if (((pos1 ^ pos2) & nb_pos) == 0)
- {
- index = quant_2p_2N1(pos1, pos2, sub(N, 1)); /* index = quant_2p_2N1(pos1, pos2, (N-1)); */
- /* index += (pos1 & nb_pos) << N; */
- index = vo_L_add(index, (L_deposit_l((Word16) (pos1 & nb_pos)) << N));
- /* index += quant_2p_2N1(pos3, pos4, N) << (2*N); */
- index = vo_L_add(index, (quant_2p_2N1(pos3, pos4, N) << (N << 1)));
- } else if (((pos1 ^ pos3) & nb_pos) == 0)
- {
- index = quant_2p_2N1(pos1, pos3, (N - 1));
- /* index += (pos1 & nb_pos) << N; */
- index = vo_L_add(index, (L_deposit_l((Word16) (pos1 & nb_pos)) << N));
- /* index += quant_2p_2N1(pos2, pos4, N) << (2*N); */
- index = vo_L_add(index, (quant_2p_2N1(pos2, pos4, N) << (N << 1)));
- } else
- {
- index = quant_2p_2N1(pos2, pos3, (N - 1));
- /* index += (pos2 & nb_pos) << N; */
- index = vo_L_add(index, (L_deposit_l((Word16) (pos2 & nb_pos)) << N));
- /* index += quant_2p_2N1(pos1, pos4, N) << (2*N); */
- index = vo_L_add(index, (quant_2p_2N1(pos1, pos4, N) << (N << 1)));
- }
- return (index);
+ nb_pos = 1 << (N - 1); /* nb_pos = (1<<(N-1)); */
+ /*-------------------------------------------------------*
+ * Quantization of 4 pulses with 4*N+1 bits: *
+ *-------------------------------------------------------*/
+ if (((pos1 ^ pos2) & nb_pos) == 0)
+ {
+ index = quant_2p_2N1(pos1, pos2, sub(N, 1)); /* index = quant_2p_2N1(pos1, pos2, (N-1)); */
+ /* index += (pos1 & nb_pos) << N; */
+ index = vo_L_add(index, (L_deposit_l((Word16) (pos1 & nb_pos)) << N));
+ /* index += quant_2p_2N1(pos3, pos4, N) << (2*N); */
+ index = vo_L_add(index, (quant_2p_2N1(pos3, pos4, N) << (N << 1)));
+ } else if (((pos1 ^ pos3) & nb_pos) == 0)
+ {
+ index = quant_2p_2N1(pos1, pos3, (N - 1));
+ /* index += (pos1 & nb_pos) << N; */
+ index = vo_L_add(index, (L_deposit_l((Word16) (pos1 & nb_pos)) << N));
+ /* index += quant_2p_2N1(pos2, pos4, N) << (2*N); */
+ index = vo_L_add(index, (quant_2p_2N1(pos2, pos4, N) << (N << 1)));
+ } else
+ {
+ index = quant_2p_2N1(pos2, pos3, (N - 1));
+ /* index += (pos2 & nb_pos) << N; */
+ index = vo_L_add(index, (L_deposit_l((Word16) (pos2 & nb_pos)) << N));
+ /* index += quant_2p_2N1(pos1, pos4, N) << (2*N); */
+ index = vo_L_add(index, (quant_2p_2N1(pos1, pos4, N) << (N << 1)));
+ }
+ return (index);
}
Word32 quant_4p_4N( /* (o) return 4*N bits */
- Word16 pos[], /* (i) position of the pulse 1..4 */
- Word16 N) /* (i) number of bits for position */
+ Word16 pos[], /* (i) position of the pulse 1..4 */
+ Word16 N) /* (i) number of bits for position */
{
- Word16 nb_pos, mask __unused, n_1, tmp;
- Word16 posA[4], posB[4];
- Word32 i, j, k, index;
+ Word16 nb_pos, mask __unused, n_1, tmp;
+ Word16 posA[4], posB[4];
+ Word32 i, j, k, index;
- n_1 = (Word16) (N - 1);
- nb_pos = (1 << n_1); /* nb_pos = (1<<n_1); */
- mask = vo_sub((1 << N), 1); /* mask = ((1<<N)-1); */
+ n_1 = (Word16) (N - 1);
+ nb_pos = (1 << n_1); /* nb_pos = (1<<n_1); */
+ mask = vo_sub((1 << N), 1); /* mask = ((1<<N)-1); */
- i = 0;
- j = 0;
- for (k = 0; k < 4; k++)
- {
- if ((pos[k] & nb_pos) == 0)
- {
- posA[i++] = pos[k];
- } else
- {
- posB[j++] = pos[k];
- }
- }
+ i = 0;
+ j = 0;
+ for (k = 0; k < 4; k++)
+ {
+ if ((pos[k] & nb_pos) == 0)
+ {
+ posA[i++] = pos[k];
+ } else
+ {
+ posB[j++] = pos[k];
+ }
+ }
- switch (i)
- {
- case 0:
- tmp = vo_sub((N << 2), 3); /* index = 1 << ((4*N)-3); */
- index = (1L << tmp);
- /* index += quant_4p_4N1(posB[0], posB[1], posB[2], posB[3], n_1); */
- index = vo_L_add(index, quant_4p_4N1(posB[0], posB[1], posB[2], posB[3], n_1));
- break;
- case 1:
- /* index = quant_1p_N1(posA[0], n_1) << ((3*n_1)+1); */
- tmp = add1((Word16)((vo_L_mult(3, n_1) >> 1)), 1);
- index = L_shl(quant_1p_N1(posA[0], n_1), tmp);
- /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1); */
- index = vo_L_add(index, quant_3p_3N1(posB[0], posB[1], posB[2], n_1));
- break;
- case 2:
- tmp = ((n_1 << 1) + 1); /* index = quant_2p_2N1(posA[0], posA[1], n_1) << ((2*n_1)+1); */
- index = L_shl(quant_2p_2N1(posA[0], posA[1], n_1), tmp);
- /* index += quant_2p_2N1(posB[0], posB[1], n_1); */
- index = vo_L_add(index, quant_2p_2N1(posB[0], posB[1], n_1));
- break;
- case 3:
- /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << N; */
- index = L_shl(quant_3p_3N1(posA[0], posA[1], posA[2], n_1), N);
- index = vo_L_add(index, quant_1p_N1(posB[0], n_1)); /* index += quant_1p_N1(posB[0], n_1); */
- break;
- case 4:
- index = quant_4p_4N1(posA[0], posA[1], posA[2], posA[3], n_1);
- break;
- default:
- index = 0;
- fprintf(stderr, "Error in function quant_4p_4N\n");
- }
- tmp = ((N << 2) - 2); /* index += (i & 3) << ((4*N)-2); */
- index = vo_L_add(index, L_shl((L_deposit_l(i) & (3L)), tmp));
+ switch (i)
+ {
+ case 0:
+ tmp = vo_sub((N << 2), 3); /* index = 1 << ((4*N)-3); */
+ index = (1L << tmp);
+ /* index += quant_4p_4N1(posB[0], posB[1], posB[2], posB[3], n_1); */
+ index = vo_L_add(index, quant_4p_4N1(posB[0], posB[1], posB[2], posB[3], n_1));
+ break;
+ case 1:
+ /* index = quant_1p_N1(posA[0], n_1) << ((3*n_1)+1); */
+ tmp = add1((Word16)((vo_L_mult(3, n_1) >> 1)), 1);
+ index = L_shl(quant_1p_N1(posA[0], n_1), tmp);
+ /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1); */
+ index = vo_L_add(index, quant_3p_3N1(posB[0], posB[1], posB[2], n_1));
+ break;
+ case 2:
+ tmp = ((n_1 << 1) + 1); /* index = quant_2p_2N1(posA[0], posA[1], n_1) << ((2*n_1)+1); */
+ index = L_shl(quant_2p_2N1(posA[0], posA[1], n_1), tmp);
+ /* index += quant_2p_2N1(posB[0], posB[1], n_1); */
+ index = vo_L_add(index, quant_2p_2N1(posB[0], posB[1], n_1));
+ break;
+ case 3:
+ /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << N; */
+ index = L_shl(quant_3p_3N1(posA[0], posA[1], posA[2], n_1), N);
+ index = vo_L_add(index, quant_1p_N1(posB[0], n_1)); /* index += quant_1p_N1(posB[0], n_1); */
+ break;
+ case 4:
+ index = quant_4p_4N1(posA[0], posA[1], posA[2], posA[3], n_1);
+ break;
+ default:
+ index = 0;
+ fprintf(stderr, "Error in function quant_4p_4N\n");
+ }
+ tmp = ((N << 2) - 2); /* index += (i & 3) << ((4*N)-2); */
+ index = vo_L_add(index, L_shl((L_deposit_l(i) & (3L)), tmp));
- return (index);
+ return (index);
}
Word32 quant_5p_5N( /* (o) return 5*N bits */
- Word16 pos[], /* (i) position of the pulse 1..5 */
- Word16 N) /* (i) number of bits for position */
+ Word16 pos[], /* (i) position of the pulse 1..5 */
+ Word16 N) /* (i) number of bits for position */
{
- Word16 nb_pos, n_1, tmp;
- Word16 posA[5], posB[5];
- Word32 i, j, k, index, tmp2;
+ Word16 nb_pos, n_1, tmp;
+ Word16 posA[5], posB[5];
+ Word32 i, j, k, index, tmp2;
- n_1 = (Word16) (N - 1);
- nb_pos = (1 << n_1); /* nb_pos = (1<<n_1); */
+ n_1 = (Word16) (N - 1);
+ nb_pos = (1 << n_1); /* nb_pos = (1<<n_1); */
- i = 0;
- j = 0;
- for (k = 0; k < 5; k++)
- {
- if ((pos[k] & nb_pos) == 0)
- {
- posA[i++] = pos[k];
- } else
- {
- posB[j++] = pos[k];
- }
- }
+ i = 0;
+ j = 0;
+ for (k = 0; k < 5; k++)
+ {
+ if ((pos[k] & nb_pos) == 0)
+ {
+ posA[i++] = pos[k];
+ } else
+ {
+ posB[j++] = pos[k];
+ }
+ }
- switch (i)
- {
- case 0:
- tmp = vo_sub((Word16)((vo_L_mult(5, N) >> 1)), 1); /* ((5*N)-1)) */
- index = L_shl(1L, tmp); /* index = 1 << ((5*N)-1); */
- tmp = add1((N << 1), 1); /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1) << ((2*N)+1);*/
- tmp2 = L_shl(quant_3p_3N1(posB[0], posB[1], posB[2], n_1), tmp);
- index = vo_L_add(index, tmp2);
- index = vo_L_add(index, quant_2p_2N1(posB[3], posB[4], N)); /* index += quant_2p_2N1(posB[3], posB[4], N); */
- break;
- case 1:
- tmp = vo_sub((Word16)((vo_L_mult(5, N) >> 1)), 1); /* index = 1 << ((5*N)-1); */
- index = L_shl(1L, tmp);
- tmp = add1((N << 1), 1); /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1) <<((2*N)+1); */
- tmp2 = L_shl(quant_3p_3N1(posB[0], posB[1], posB[2], n_1), tmp);
- index = vo_L_add(index, tmp2);
- index = vo_L_add(index, quant_2p_2N1(posB[3], posA[0], N)); /* index += quant_2p_2N1(posB[3], posA[0], N); */
- break;
- case 2:
- tmp = vo_sub((Word16)((vo_L_mult(5, N) >> 1)), 1); /* ((5*N)-1)) */
- index = L_shl(1L, tmp); /* index = 1 << ((5*N)-1); */
- tmp = add1((N << 1), 1); /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1) << ((2*N)+1); */
- tmp2 = L_shl(quant_3p_3N1(posB[0], posB[1], posB[2], n_1), tmp);
- index = vo_L_add(index, tmp2);
- index = vo_L_add(index, quant_2p_2N1(posA[0], posA[1], N)); /* index += quant_2p_2N1(posA[0], posA[1], N); */
- break;
- case 3:
- tmp = add1((N << 1), 1); /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << ((2*N)+1); */
- index = L_shl(quant_3p_3N1(posA[0], posA[1], posA[2], n_1), tmp);
- index = vo_L_add(index, quant_2p_2N1(posB[0], posB[1], N)); /* index += quant_2p_2N1(posB[0], posB[1], N); */
- break;
- case 4:
- tmp = add1((N << 1), 1); /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << ((2*N)+1); */
- index = L_shl(quant_3p_3N1(posA[0], posA[1], posA[2], n_1), tmp);
- index = vo_L_add(index, quant_2p_2N1(posA[3], posB[0], N)); /* index += quant_2p_2N1(posA[3], posB[0], N); */
- break;
- case 5:
- tmp = add1((N << 1), 1); /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << ((2*N)+1); */
- index = L_shl(quant_3p_3N1(posA[0], posA[1], posA[2], n_1), tmp);
- index = vo_L_add(index, quant_2p_2N1(posA[3], posA[4], N)); /* index += quant_2p_2N1(posA[3], posA[4], N); */
- break;
- default:
- index = 0;
- fprintf(stderr, "Error in function quant_5p_5N\n");
- }
+ switch (i)
+ {
+ case 0:
+ tmp = vo_sub((Word16)((vo_L_mult(5, N) >> 1)), 1); /* ((5*N)-1)) */
+ index = L_shl(1L, tmp); /* index = 1 << ((5*N)-1); */
+ tmp = add1((N << 1), 1); /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1) << ((2*N)+1);*/
+ tmp2 = L_shl(quant_3p_3N1(posB[0], posB[1], posB[2], n_1), tmp);
+ index = vo_L_add(index, tmp2);
+ index = vo_L_add(index, quant_2p_2N1(posB[3], posB[4], N)); /* index += quant_2p_2N1(posB[3], posB[4], N); */
+ break;
+ case 1:
+ tmp = vo_sub((Word16)((vo_L_mult(5, N) >> 1)), 1); /* index = 1 << ((5*N)-1); */
+ index = L_shl(1L, tmp);
+ tmp = add1((N << 1), 1); /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1) <<((2*N)+1); */
+ tmp2 = L_shl(quant_3p_3N1(posB[0], posB[1], posB[2], n_1), tmp);
+ index = vo_L_add(index, tmp2);
+ index = vo_L_add(index, quant_2p_2N1(posB[3], posA[0], N)); /* index += quant_2p_2N1(posB[3], posA[0], N); */
+ break;
+ case 2:
+ tmp = vo_sub((Word16)((vo_L_mult(5, N) >> 1)), 1); /* ((5*N)-1)) */
+ index = L_shl(1L, tmp); /* index = 1 << ((5*N)-1); */
+ tmp = add1((N << 1), 1); /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1) << ((2*N)+1); */
+ tmp2 = L_shl(quant_3p_3N1(posB[0], posB[1], posB[2], n_1), tmp);
+ index = vo_L_add(index, tmp2);
+ index = vo_L_add(index, quant_2p_2N1(posA[0], posA[1], N)); /* index += quant_2p_2N1(posA[0], posA[1], N); */
+ break;
+ case 3:
+ tmp = add1((N << 1), 1); /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << ((2*N)+1); */
+ index = L_shl(quant_3p_3N1(posA[0], posA[1], posA[2], n_1), tmp);
+ index = vo_L_add(index, quant_2p_2N1(posB[0], posB[1], N)); /* index += quant_2p_2N1(posB[0], posB[1], N); */
+ break;
+ case 4:
+ tmp = add1((N << 1), 1); /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << ((2*N)+1); */
+ index = L_shl(quant_3p_3N1(posA[0], posA[1], posA[2], n_1), tmp);
+ index = vo_L_add(index, quant_2p_2N1(posA[3], posB[0], N)); /* index += quant_2p_2N1(posA[3], posB[0], N); */
+ break;
+ case 5:
+ tmp = add1((N << 1), 1); /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << ((2*N)+1); */
+ index = L_shl(quant_3p_3N1(posA[0], posA[1], posA[2], n_1), tmp);
+ index = vo_L_add(index, quant_2p_2N1(posA[3], posA[4], N)); /* index += quant_2p_2N1(posA[3], posA[4], N); */
+ break;
+ default:
+ index = 0;
+ fprintf(stderr, "Error in function quant_5p_5N\n");
+ }
- return (index);
+ return (index);
}
Word32 quant_6p_6N_2( /* (o) return (6*N)-2 bits */
- Word16 pos[], /* (i) position of the pulse 1..6 */
- Word16 N) /* (i) number of bits for position */
+ Word16 pos[], /* (i) position of the pulse 1..6 */
+ Word16 N) /* (i) number of bits for position */
{
- Word16 nb_pos, n_1;
- Word16 posA[6], posB[6];
- Word32 i, j, k, index;
+ Word16 nb_pos, n_1;
+ Word16 posA[6], posB[6];
+ Word32 i, j, k, index;
- /* !! N and n_1 are constants -> it doesn't need to be operated by Basic Operators */
- n_1 = (Word16) (N - 1);
- nb_pos = (1 << n_1); /* nb_pos = (1<<n_1); */
+ /* !! N and n_1 are constants -> it doesn't need to be operated by Basic Operators */
+ n_1 = (Word16) (N - 1);
+ nb_pos = (1 << n_1); /* nb_pos = (1<<n_1); */
- i = 0;
- j = 0;
- for (k = 0; k < 6; k++)
- {
- if ((pos[k] & nb_pos) == 0)
- {
- posA[i++] = pos[k];
- } else
- {
- posB[j++] = pos[k];
- }
- }
+ i = 0;
+ j = 0;
+ for (k = 0; k < 6; k++)
+ {
+ if ((pos[k] & nb_pos) == 0)
+ {
+ posA[i++] = pos[k];
+ } else
+ {
+ posB[j++] = pos[k];
+ }
+ }
- switch (i)
- {
- case 0:
- index = (1 << (Word16) (6 * N - 5)); /* index = 1 << ((6*N)-5); */
- index = vo_L_add(index, (quant_5p_5N(posB, n_1) << N)); /* index += quant_5p_5N(posB, n_1) << N; */
- index = vo_L_add(index, quant_1p_N1(posB[5], n_1)); /* index += quant_1p_N1(posB[5], n_1); */
- break;
- case 1:
- index = (1L << (Word16) (6 * N - 5)); /* index = 1 << ((6*N)-5); */
- index = vo_L_add(index, (quant_5p_5N(posB, n_1) << N)); /* index += quant_5p_5N(posB, n_1) << N; */
- index = vo_L_add(index, quant_1p_N1(posA[0], n_1)); /* index += quant_1p_N1(posA[0], n_1); */
- break;
- case 2:
- index = (1L << (Word16) (6 * N - 5)); /* index = 1 << ((6*N)-5); */
- /* index += quant_4p_4N(posB, n_1) << ((2*n_1)+1); */
- index = vo_L_add(index, (quant_4p_4N(posB, n_1) << (Word16) (2 * n_1 + 1)));
- index = vo_L_add(index, quant_2p_2N1(posA[0], posA[1], n_1)); /* index += quant_2p_2N1(posA[0], posA[1], n_1); */
- break;
- case 3:
- index = (quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << (Word16) (3 * n_1 + 1));
- /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << ((3*n_1)+1); */
- index =vo_L_add(index, quant_3p_3N1(posB[0], posB[1], posB[2], n_1));
- /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1); */
- break;
- case 4:
- i = 2;
- index = (quant_4p_4N(posA, n_1) << (Word16) (2 * n_1 + 1)); /* index = quant_4p_4N(posA, n_1) << ((2*n_1)+1); */
- index = vo_L_add(index, quant_2p_2N1(posB[0], posB[1], n_1)); /* index += quant_2p_2N1(posB[0], posB[1], n_1); */
- break;
- case 5:
- i = 1;
- index = (quant_5p_5N(posA, n_1) << N); /* index = quant_5p_5N(posA, n_1) << N; */
- index = vo_L_add(index, quant_1p_N1(posB[0], n_1)); /* index += quant_1p_N1(posB[0], n_1); */
- break;
- case 6:
- i = 0;
- index = (quant_5p_5N(posA, n_1) << N); /* index = quant_5p_5N(posA, n_1) << N; */
- index = vo_L_add(index, quant_1p_N1(posA[5], n_1)); /* index += quant_1p_N1(posA[5], n_1); */
- break;
- default:
- index = 0;
- fprintf(stderr, "Error in function quant_6p_6N_2\n");
- }
- index = vo_L_add(index, ((L_deposit_l(i) & 3L) << (Word16) (6 * N - 4))); /* index += (i & 3) << ((6*N)-4); */
+ switch (i)
+ {
+ case 0:
+ index = (1 << (Word16) (6 * N - 5)); /* index = 1 << ((6*N)-5); */
+ index = vo_L_add(index, (quant_5p_5N(posB, n_1) << N)); /* index += quant_5p_5N(posB, n_1) << N; */
+ index = vo_L_add(index, quant_1p_N1(posB[5], n_1)); /* index += quant_1p_N1(posB[5], n_1); */
+ break;
+ case 1:
+ index = (1L << (Word16) (6 * N - 5)); /* index = 1 << ((6*N)-5); */
+ index = vo_L_add(index, (quant_5p_5N(posB, n_1) << N)); /* index += quant_5p_5N(posB, n_1) << N; */
+ index = vo_L_add(index, quant_1p_N1(posA[0], n_1)); /* index += quant_1p_N1(posA[0], n_1); */
+ break;
+ case 2:
+ index = (1L << (Word16) (6 * N - 5)); /* index = 1 << ((6*N)-5); */
+ /* index += quant_4p_4N(posB, n_1) << ((2*n_1)+1); */
+ index = vo_L_add(index, (quant_4p_4N(posB, n_1) << (Word16) (2 * n_1 + 1)));
+ index = vo_L_add(index, quant_2p_2N1(posA[0], posA[1], n_1)); /* index += quant_2p_2N1(posA[0], posA[1], n_1); */
+ break;
+ case 3:
+ index = (quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << (Word16) (3 * n_1 + 1));
+ /* index = quant_3p_3N1(posA[0], posA[1], posA[2], n_1) << ((3*n_1)+1); */
+ index =vo_L_add(index, quant_3p_3N1(posB[0], posB[1], posB[2], n_1));
+ /* index += quant_3p_3N1(posB[0], posB[1], posB[2], n_1); */
+ break;
+ case 4:
+ i = 2;
+ index = (quant_4p_4N(posA, n_1) << (Word16) (2 * n_1 + 1)); /* index = quant_4p_4N(posA, n_1) << ((2*n_1)+1); */
+ index = vo_L_add(index, quant_2p_2N1(posB[0], posB[1], n_1)); /* index += quant_2p_2N1(posB[0], posB[1], n_1); */
+ break;
+ case 5:
+ i = 1;
+ index = (quant_5p_5N(posA, n_1) << N); /* index = quant_5p_5N(posA, n_1) << N; */
+ index = vo_L_add(index, quant_1p_N1(posB[0], n_1)); /* index += quant_1p_N1(posB[0], n_1); */
+ break;
+ case 6:
+ i = 0;
+ index = (quant_5p_5N(posA, n_1) << N); /* index = quant_5p_5N(posA, n_1) << N; */
+ index = vo_L_add(index, quant_1p_N1(posA[5], n_1)); /* index += quant_1p_N1(posA[5], n_1); */
+ break;
+ default:
+ index = 0;
+ fprintf(stderr, "Error in function quant_6p_6N_2\n");
+ }
+ index = vo_L_add(index, ((L_deposit_l(i) & 3L) << (Word16) (6 * N - 4))); /* index += (i & 3) << ((6*N)-4); */
- return (index);
+ return (index);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/qisf_ns.c b/media/libstagefright/codecs/amrwbenc/src/qisf_ns.c
index fc2f00d..eac98e2 100644
--- a/media/libstagefright/codecs/amrwbenc/src/qisf_ns.c
+++ b/media/libstagefright/codecs/amrwbenc/src/qisf_ns.c
@@ -33,30 +33,30 @@
*------------------------------------------------------------------*/
void Qisf_ns(
- Word16 * isf1, /* input : ISF in the frequency domain (0..0.5) */
- Word16 * isf_q, /* output: quantized ISF */
- Word16 * indice /* output: quantization indices */
- )
+ Word16 * isf1, /* input : ISF in the frequency domain (0..0.5) */
+ Word16 * isf_q, /* output: quantized ISF */
+ Word16 * indice /* output: quantization indices */
+ )
{
- Word16 i;
- Word32 tmp;
+ Word16 i;
+ Word32 tmp;
- for (i = 0; i < ORDER; i++)
- {
- isf_q[i] = sub(isf1[i], mean_isf_noise[i]);
- }
+ for (i = 0; i < ORDER; i++)
+ {
+ isf_q[i] = sub(isf1[i], mean_isf_noise[i]);
+ }
- indice[0] = Sub_VQ(&isf_q[0], dico1_isf_noise, 2, SIZE_BK_NOISE1, &tmp);
- indice[1] = Sub_VQ(&isf_q[2], dico2_isf_noise, 3, SIZE_BK_NOISE2, &tmp);
- indice[2] = Sub_VQ(&isf_q[5], dico3_isf_noise, 3, SIZE_BK_NOISE3, &tmp);
- indice[3] = Sub_VQ(&isf_q[8], dico4_isf_noise, 4, SIZE_BK_NOISE4, &tmp);
- indice[4] = Sub_VQ(&isf_q[12], dico5_isf_noise, 4, SIZE_BK_NOISE5, &tmp);
+ indice[0] = Sub_VQ(&isf_q[0], dico1_isf_noise, 2, SIZE_BK_NOISE1, &tmp);
+ indice[1] = Sub_VQ(&isf_q[2], dico2_isf_noise, 3, SIZE_BK_NOISE2, &tmp);
+ indice[2] = Sub_VQ(&isf_q[5], dico3_isf_noise, 3, SIZE_BK_NOISE3, &tmp);
+ indice[3] = Sub_VQ(&isf_q[8], dico4_isf_noise, 4, SIZE_BK_NOISE4, &tmp);
+ indice[4] = Sub_VQ(&isf_q[12], dico5_isf_noise, 4, SIZE_BK_NOISE5, &tmp);
- /* decoding the ISFs */
+ /* decoding the ISFs */
- Disf_ns(indice, isf_q);
+ Disf_ns(indice, isf_q);
- return;
+ return;
}
/********************************************************************
@@ -70,41 +70,41 @@
*********************************************************************/
void Disf_ns(
- Word16 * indice, /* input: quantization indices */
- Word16 * isf_q /* input : ISF in the frequency domain (0..0.5) */
- )
+ Word16 * indice, /* input: quantization indices */
+ Word16 * isf_q /* input : ISF in the frequency domain (0..0.5) */
+ )
{
- Word16 i;
+ Word16 i;
- for (i = 0; i < 2; i++)
- {
- isf_q[i] = dico1_isf_noise[indice[0] * 2 + i];
- }
- for (i = 0; i < 3; i++)
- {
- isf_q[i + 2] = dico2_isf_noise[indice[1] * 3 + i];
- }
- for (i = 0; i < 3; i++)
- {
- isf_q[i + 5] = dico3_isf_noise[indice[2] * 3 + i];
- }
- for (i = 0; i < 4; i++)
- {
- isf_q[i + 8] = dico4_isf_noise[indice[3] * 4 + i];
- }
- for (i = 0; i < 4; i++)
- {
- isf_q[i + 12] = dico5_isf_noise[indice[4] * 4 + i];
- }
+ for (i = 0; i < 2; i++)
+ {
+ isf_q[i] = dico1_isf_noise[indice[0] * 2 + i];
+ }
+ for (i = 0; i < 3; i++)
+ {
+ isf_q[i + 2] = dico2_isf_noise[indice[1] * 3 + i];
+ }
+ for (i = 0; i < 3; i++)
+ {
+ isf_q[i + 5] = dico3_isf_noise[indice[2] * 3 + i];
+ }
+ for (i = 0; i < 4; i++)
+ {
+ isf_q[i + 8] = dico4_isf_noise[indice[3] * 4 + i];
+ }
+ for (i = 0; i < 4; i++)
+ {
+ isf_q[i + 12] = dico5_isf_noise[indice[4] * 4 + i];
+ }
- for (i = 0; i < ORDER; i++)
- {
- isf_q[i] = add(isf_q[i], mean_isf_noise[i]);
- }
+ for (i = 0; i < ORDER; i++)
+ {
+ isf_q[i] = add(isf_q[i], mean_isf_noise[i]);
+ }
- Reorder_isf(isf_q, ISF_GAP, ORDER);
+ Reorder_isf(isf_q, ISF_GAP, ORDER);
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/qpisf_2s.c b/media/libstagefright/codecs/amrwbenc/src/qpisf_2s.c
index c711cd0..bec334e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/qpisf_2s.c
+++ b/media/libstagefright/codecs/amrwbenc/src/qpisf_2s.c
@@ -36,13 +36,13 @@
/* private functions */
static void VQ_stage1(
- Word16 * x, /* input : ISF residual vector */
- Word16 * dico, /* input : quantization codebook */
- Word16 dim, /* input : dimention of vector */
- Word16 dico_size, /* input : size of quantization codebook */
- Word16 * index, /* output: indices of survivors */
- Word16 surv /* input : number of survivor */
- );
+ Word16 * x, /* input : ISF residual vector */
+ Word16 * dico, /* input : quantization codebook */
+ Word16 dim, /* input : dimention of vector */
+ Word16 dico_size, /* input : size of quantization codebook */
+ Word16 * index, /* output: indices of survivors */
+ Word16 surv /* input : number of survivor */
+ );
/**************************************************************************
* Function: Qpisf_2s_46B() *
@@ -54,84 +54,84 @@
***************************************************************************/
void Qpisf_2s_46b(
- Word16 * isf1, /* (i) Q15 : ISF in the frequency domain (0..0.5) */
- Word16 * isf_q, /* (o) Q15 : quantized ISF (0..0.5) */
- Word16 * past_isfq, /* (io)Q15 : past ISF quantizer */
- Word16 * indice, /* (o) : quantization indices */
- Word16 nb_surv /* (i) : number of survivor (1, 2, 3 or 4) */
- )
+ Word16 * isf1, /* (i) Q15 : ISF in the frequency domain (0..0.5) */
+ Word16 * isf_q, /* (o) Q15 : quantized ISF (0..0.5) */
+ Word16 * past_isfq, /* (io)Q15 : past ISF quantizer */
+ Word16 * indice, /* (o) : quantization indices */
+ Word16 nb_surv /* (i) : number of survivor (1, 2, 3 or 4) */
+ )
{
- Word16 tmp_ind[5];
- Word16 surv1[N_SURV_MAX]; /* indices of survivors from 1st stage */
- Word32 i, k, temp, min_err, distance;
- Word16 isf[ORDER];
- Word16 isf_stage2[ORDER];
+ Word16 tmp_ind[5];
+ Word16 surv1[N_SURV_MAX]; /* indices of survivors from 1st stage */
+ Word32 i, k, temp, min_err, distance;
+ Word16 isf[ORDER];
+ Word16 isf_stage2[ORDER];
- for (i = 0; i < ORDER; i++)
- {
- isf[i] = vo_sub(isf1[i], mean_isf[i]);
- isf[i] = vo_sub(isf[i], vo_mult(MU, past_isfq[i]));
- }
+ for (i = 0; i < ORDER; i++)
+ {
+ isf[i] = vo_sub(isf1[i], mean_isf[i]);
+ isf[i] = vo_sub(isf[i], vo_mult(MU, past_isfq[i]));
+ }
- VQ_stage1(&isf[0], dico1_isf, 9, SIZE_BK1, surv1, nb_surv);
+ VQ_stage1(&isf[0], dico1_isf, 9, SIZE_BK1, surv1, nb_surv);
- distance = MAX_32;
+ distance = MAX_32;
- for (k = 0; k < nb_surv; k++)
- {
- for (i = 0; i < 9; i++)
- {
- isf_stage2[i] = vo_sub(isf[i], dico1_isf[i + surv1[k] * 9]);
- }
- tmp_ind[0] = Sub_VQ(&isf_stage2[0], dico21_isf, 3, SIZE_BK21, &min_err);
- temp = min_err;
- tmp_ind[1] = Sub_VQ(&isf_stage2[3], dico22_isf, 3, SIZE_BK22, &min_err);
- temp = vo_L_add(temp, min_err);
- tmp_ind[2] = Sub_VQ(&isf_stage2[6], dico23_isf, 3, SIZE_BK23, &min_err);
- temp = vo_L_add(temp, min_err);
+ for (k = 0; k < nb_surv; k++)
+ {
+ for (i = 0; i < 9; i++)
+ {
+ isf_stage2[i] = vo_sub(isf[i], dico1_isf[i + surv1[k] * 9]);
+ }
+ tmp_ind[0] = Sub_VQ(&isf_stage2[0], dico21_isf, 3, SIZE_BK21, &min_err);
+ temp = min_err;
+ tmp_ind[1] = Sub_VQ(&isf_stage2[3], dico22_isf, 3, SIZE_BK22, &min_err);
+ temp = vo_L_add(temp, min_err);
+ tmp_ind[2] = Sub_VQ(&isf_stage2[6], dico23_isf, 3, SIZE_BK23, &min_err);
+ temp = vo_L_add(temp, min_err);
- if(temp < distance)
- {
- distance = temp;
- indice[0] = surv1[k];
- for (i = 0; i < 3; i++)
- {
- indice[i + 2] = tmp_ind[i];
- }
- }
- }
+ if(temp < distance)
+ {
+ distance = temp;
+ indice[0] = surv1[k];
+ for (i = 0; i < 3; i++)
+ {
+ indice[i + 2] = tmp_ind[i];
+ }
+ }
+ }
- VQ_stage1(&isf[9], dico2_isf, 7, SIZE_BK2, surv1, nb_surv);
+ VQ_stage1(&isf[9], dico2_isf, 7, SIZE_BK2, surv1, nb_surv);
- distance = MAX_32;
+ distance = MAX_32;
- for (k = 0; k < nb_surv; k++)
- {
- for (i = 0; i < 7; i++)
- {
- isf_stage2[i] = vo_sub(isf[9 + i], dico2_isf[i + surv1[k] * 7]);
- }
+ for (k = 0; k < nb_surv; k++)
+ {
+ for (i = 0; i < 7; i++)
+ {
+ isf_stage2[i] = vo_sub(isf[9 + i], dico2_isf[i + surv1[k] * 7]);
+ }
- tmp_ind[0] = Sub_VQ(&isf_stage2[0], dico24_isf, 3, SIZE_BK24, &min_err);
- temp = min_err;
- tmp_ind[1] = Sub_VQ(&isf_stage2[3], dico25_isf, 4, SIZE_BK25, &min_err);
- temp = vo_L_add(temp, min_err);
+ tmp_ind[0] = Sub_VQ(&isf_stage2[0], dico24_isf, 3, SIZE_BK24, &min_err);
+ temp = min_err;
+ tmp_ind[1] = Sub_VQ(&isf_stage2[3], dico25_isf, 4, SIZE_BK25, &min_err);
+ temp = vo_L_add(temp, min_err);
- if(temp < distance)
- {
- distance = temp;
- indice[1] = surv1[k];
- for (i = 0; i < 2; i++)
- {
- indice[i + 5] = tmp_ind[i];
- }
- }
- }
+ if(temp < distance)
+ {
+ distance = temp;
+ indice[1] = surv1[k];
+ for (i = 0; i < 2; i++)
+ {
+ indice[i + 5] = tmp_ind[i];
+ }
+ }
+ }
- Dpisf_2s_46b(indice, isf_q, past_isfq, isf_q, isf_q, 0, 0);
+ Dpisf_2s_46b(indice, isf_q, past_isfq, isf_q, isf_q, 0, 0);
- return;
+ return;
}
/*****************************************************************************
@@ -144,76 +144,76 @@
******************************************************************************/
void Qpisf_2s_36b(
- Word16 * isf1, /* (i) Q15 : ISF in the frequency domain (0..0.5) */
- Word16 * isf_q, /* (o) Q15 : quantized ISF (0..0.5) */
- Word16 * past_isfq, /* (io)Q15 : past ISF quantizer */
- Word16 * indice, /* (o) : quantization indices */
- Word16 nb_surv /* (i) : number of survivor (1, 2, 3 or 4) */
- )
+ Word16 * isf1, /* (i) Q15 : ISF in the frequency domain (0..0.5) */
+ Word16 * isf_q, /* (o) Q15 : quantized ISF (0..0.5) */
+ Word16 * past_isfq, /* (io)Q15 : past ISF quantizer */
+ Word16 * indice, /* (o) : quantization indices */
+ Word16 nb_surv /* (i) : number of survivor (1, 2, 3 or 4) */
+ )
{
- Word16 i, k, tmp_ind[5];
- Word16 surv1[N_SURV_MAX]; /* indices of survivors from 1st stage */
- Word32 temp, min_err, distance;
- Word16 isf[ORDER];
- Word16 isf_stage2[ORDER];
+ Word16 i, k, tmp_ind[5];
+ Word16 surv1[N_SURV_MAX]; /* indices of survivors from 1st stage */
+ Word32 temp, min_err, distance;
+ Word16 isf[ORDER];
+ Word16 isf_stage2[ORDER];
- for (i = 0; i < ORDER; i++)
- {
- isf[i] = vo_sub(isf1[i], mean_isf[i]);
- isf[i] = vo_sub(isf[i], vo_mult(MU, past_isfq[i]));
- }
+ for (i = 0; i < ORDER; i++)
+ {
+ isf[i] = vo_sub(isf1[i], mean_isf[i]);
+ isf[i] = vo_sub(isf[i], vo_mult(MU, past_isfq[i]));
+ }
- VQ_stage1(&isf[0], dico1_isf, 9, SIZE_BK1, surv1, nb_surv);
+ VQ_stage1(&isf[0], dico1_isf, 9, SIZE_BK1, surv1, nb_surv);
- distance = MAX_32;
+ distance = MAX_32;
- for (k = 0; k < nb_surv; k++)
- {
- for (i = 0; i < 9; i++)
- {
- isf_stage2[i] = vo_sub(isf[i], dico1_isf[i + surv1[k] * 9]);
- }
+ for (k = 0; k < nb_surv; k++)
+ {
+ for (i = 0; i < 9; i++)
+ {
+ isf_stage2[i] = vo_sub(isf[i], dico1_isf[i + surv1[k] * 9]);
+ }
- tmp_ind[0] = Sub_VQ(&isf_stage2[0], dico21_isf_36b, 5, SIZE_BK21_36b, &min_err);
- temp = min_err;
- tmp_ind[1] = Sub_VQ(&isf_stage2[5], dico22_isf_36b, 4, SIZE_BK22_36b, &min_err);
- temp = vo_L_add(temp, min_err);
+ tmp_ind[0] = Sub_VQ(&isf_stage2[0], dico21_isf_36b, 5, SIZE_BK21_36b, &min_err);
+ temp = min_err;
+ tmp_ind[1] = Sub_VQ(&isf_stage2[5], dico22_isf_36b, 4, SIZE_BK22_36b, &min_err);
+ temp = vo_L_add(temp, min_err);
- if(temp < distance)
- {
- distance = temp;
- indice[0] = surv1[k];
- for (i = 0; i < 2; i++)
- {
- indice[i + 2] = tmp_ind[i];
- }
- }
- }
+ if(temp < distance)
+ {
+ distance = temp;
+ indice[0] = surv1[k];
+ for (i = 0; i < 2; i++)
+ {
+ indice[i + 2] = tmp_ind[i];
+ }
+ }
+ }
- VQ_stage1(&isf[9], dico2_isf, 7, SIZE_BK2, surv1, nb_surv);
- distance = MAX_32;
+ VQ_stage1(&isf[9], dico2_isf, 7, SIZE_BK2, surv1, nb_surv);
+ distance = MAX_32;
- for (k = 0; k < nb_surv; k++)
- {
- for (i = 0; i < 7; i++)
- {
- isf_stage2[i] = vo_sub(isf[9 + i], dico2_isf[i + surv1[k] * 7]);
- }
+ for (k = 0; k < nb_surv; k++)
+ {
+ for (i = 0; i < 7; i++)
+ {
+ isf_stage2[i] = vo_sub(isf[9 + i], dico2_isf[i + surv1[k] * 7]);
+ }
- tmp_ind[0] = Sub_VQ(&isf_stage2[0], dico23_isf_36b, 7, SIZE_BK23_36b, &min_err);
- temp = min_err;
+ tmp_ind[0] = Sub_VQ(&isf_stage2[0], dico23_isf_36b, 7, SIZE_BK23_36b, &min_err);
+ temp = min_err;
- if(temp < distance)
- {
- distance = temp;
- indice[1] = surv1[k];
- indice[4] = tmp_ind[0];
- }
- }
+ if(temp < distance)
+ {
+ distance = temp;
+ indice[1] = surv1[k];
+ indice[4] = tmp_ind[0];
+ }
+ }
- Dpisf_2s_36b(indice, isf_q, past_isfq, isf_q, isf_q, 0, 0);
+ Dpisf_2s_36b(indice, isf_q, past_isfq, isf_q, isf_q, 0, 0);
- return;
+ return;
}
/*********************************************************************
@@ -223,90 +223,90 @@
**********************************************************************/
void Dpisf_2s_46b(
- Word16 * indice, /* input: quantization indices */
- Word16 * isf_q, /* output: quantized ISF in frequency domain (0..0.5) */
- Word16 * past_isfq, /* i/0 : past ISF quantizer */
- Word16 * isfold, /* input : past quantized ISF */
- Word16 * isf_buf, /* input : isf buffer */
- Word16 bfi, /* input : Bad frame indicator */
- Word16 enc_dec
- )
+ Word16 * indice, /* input: quantization indices */
+ Word16 * isf_q, /* output: quantized ISF in frequency domain (0..0.5) */
+ Word16 * past_isfq, /* i/0 : past ISF quantizer */
+ Word16 * isfold, /* input : past quantized ISF */
+ Word16 * isf_buf, /* input : isf buffer */
+ Word16 bfi, /* input : Bad frame indicator */
+ Word16 enc_dec
+ )
{
- Word16 ref_isf[M], tmp;
- Word32 i, j, L_tmp;
+ Word16 ref_isf[M], tmp;
+ Word32 i, j, L_tmp;
- if (bfi == 0) /* Good frame */
- {
- for (i = 0; i < 9; i++)
- {
- isf_q[i] = dico1_isf[indice[0] * 9 + i];
- }
- for (i = 0; i < 7; i++)
- {
- isf_q[i + 9] = dico2_isf[indice[1] * 7 + i];
- }
+ if (bfi == 0) /* Good frame */
+ {
+ for (i = 0; i < 9; i++)
+ {
+ isf_q[i] = dico1_isf[indice[0] * 9 + i];
+ }
+ for (i = 0; i < 7; i++)
+ {
+ isf_q[i + 9] = dico2_isf[indice[1] * 7 + i];
+ }
- for (i = 0; i < 3; i++)
- {
- isf_q[i] = add1(isf_q[i], dico21_isf[indice[2] * 3 + i]);
- isf_q[i + 3] = add1(isf_q[i + 3], dico22_isf[indice[3] * 3 + i]);
- isf_q[i + 6] = add1(isf_q[i + 6], dico23_isf[indice[4] * 3 + i]);
- isf_q[i + 9] = add1(isf_q[i + 9], dico24_isf[indice[5] * 3 + i]);
- }
+ for (i = 0; i < 3; i++)
+ {
+ isf_q[i] = add1(isf_q[i], dico21_isf[indice[2] * 3 + i]);
+ isf_q[i + 3] = add1(isf_q[i + 3], dico22_isf[indice[3] * 3 + i]);
+ isf_q[i + 6] = add1(isf_q[i + 6], dico23_isf[indice[4] * 3 + i]);
+ isf_q[i + 9] = add1(isf_q[i + 9], dico24_isf[indice[5] * 3 + i]);
+ }
- for (i = 0; i < 4; i++)
- {
- isf_q[i + 12] = add1(isf_q[i + 12], dico25_isf[indice[6] * 4 + i]);
- }
+ for (i = 0; i < 4; i++)
+ {
+ isf_q[i + 12] = add1(isf_q[i + 12], dico25_isf[indice[6] * 4 + i]);
+ }
- for (i = 0; i < ORDER; i++)
- {
- tmp = isf_q[i];
- isf_q[i] = add1(tmp, mean_isf[i]);
- isf_q[i] = add1(isf_q[i], vo_mult(MU, past_isfq[i]));
- past_isfq[i] = tmp;
- }
+ for (i = 0; i < ORDER; i++)
+ {
+ tmp = isf_q[i];
+ isf_q[i] = add1(tmp, mean_isf[i]);
+ isf_q[i] = add1(isf_q[i], vo_mult(MU, past_isfq[i]));
+ past_isfq[i] = tmp;
+ }
- if (enc_dec)
- {
- for (i = 0; i < M; i++)
- {
- for (j = (L_MEANBUF - 1); j > 0; j--)
- {
- isf_buf[j * M + i] = isf_buf[(j - 1) * M + i];
- }
- isf_buf[i] = isf_q[i];
- }
- }
- } else
- { /* bad frame */
- for (i = 0; i < M; i++)
- {
- L_tmp = mean_isf[i] << 14;
- for (j = 0; j < L_MEANBUF; j++)
- {
- L_tmp += (isf_buf[j * M + i] << 14);
- }
- ref_isf[i] = vo_round(L_tmp);
- }
+ if (enc_dec)
+ {
+ for (i = 0; i < M; i++)
+ {
+ for (j = (L_MEANBUF - 1); j > 0; j--)
+ {
+ isf_buf[j * M + i] = isf_buf[(j - 1) * M + i];
+ }
+ isf_buf[i] = isf_q[i];
+ }
+ }
+ } else
+ { /* bad frame */
+ for (i = 0; i < M; i++)
+ {
+ L_tmp = mean_isf[i] << 14;
+ for (j = 0; j < L_MEANBUF; j++)
+ {
+ L_tmp += (isf_buf[j * M + i] << 14);
+ }
+ ref_isf[i] = vo_round(L_tmp);
+ }
- /* use the past ISFs slightly shifted towards their mean */
- for (i = 0; i < ORDER; i++)
- {
- isf_q[i] = add1(vo_mult(ALPHA, isfold[i]), vo_mult(ONE_ALPHA, ref_isf[i]));
- }
+ /* use the past ISFs slightly shifted towards their mean */
+ for (i = 0; i < ORDER; i++)
+ {
+ isf_q[i] = add1(vo_mult(ALPHA, isfold[i]), vo_mult(ONE_ALPHA, ref_isf[i]));
+ }
- /* estimate past quantized residual to be used in next frame */
- for (i = 0; i < ORDER; i++)
- {
- tmp = add1(ref_isf[i], vo_mult(past_isfq[i], MU)); /* predicted ISF */
- past_isfq[i] = vo_sub(isf_q[i], tmp);
- past_isfq[i] = (past_isfq[i] >> 1); /* past_isfq[i] *= 0.5 */
- }
- }
+ /* estimate past quantized residual to be used in next frame */
+ for (i = 0; i < ORDER; i++)
+ {
+ tmp = add1(ref_isf[i], vo_mult(past_isfq[i], MU)); /* predicted ISF */
+ past_isfq[i] = vo_sub(isf_q[i], tmp);
+ past_isfq[i] = (past_isfq[i] >> 1); /* past_isfq[i] *= 0.5 */
+ }
+ }
- Reorder_isf(isf_q, ISF_GAP, ORDER);
- return;
+ Reorder_isf(isf_q, ISF_GAP, ORDER);
+ return;
}
/*********************************************************************
@@ -316,92 +316,92 @@
*********************************************************************/
void Dpisf_2s_36b(
- Word16 * indice, /* input: quantization indices */
- Word16 * isf_q, /* output: quantized ISF in frequency domain (0..0.5) */
- Word16 * past_isfq, /* i/0 : past ISF quantizer */
- Word16 * isfold, /* input : past quantized ISF */
- Word16 * isf_buf, /* input : isf buffer */
- Word16 bfi, /* input : Bad frame indicator */
- Word16 enc_dec
- )
+ Word16 * indice, /* input: quantization indices */
+ Word16 * isf_q, /* output: quantized ISF in frequency domain (0..0.5) */
+ Word16 * past_isfq, /* i/0 : past ISF quantizer */
+ Word16 * isfold, /* input : past quantized ISF */
+ Word16 * isf_buf, /* input : isf buffer */
+ Word16 bfi, /* input : Bad frame indicator */
+ Word16 enc_dec
+ )
{
- Word16 ref_isf[M], tmp;
- Word32 i, j, L_tmp;
+ Word16 ref_isf[M], tmp;
+ Word32 i, j, L_tmp;
- if (bfi == 0) /* Good frame */
- {
- for (i = 0; i < 9; i++)
- {
- isf_q[i] = dico1_isf[indice[0] * 9 + i];
- }
- for (i = 0; i < 7; i++)
- {
- isf_q[i + 9] = dico2_isf[indice[1] * 7 + i];
- }
+ if (bfi == 0) /* Good frame */
+ {
+ for (i = 0; i < 9; i++)
+ {
+ isf_q[i] = dico1_isf[indice[0] * 9 + i];
+ }
+ for (i = 0; i < 7; i++)
+ {
+ isf_q[i + 9] = dico2_isf[indice[1] * 7 + i];
+ }
- for (i = 0; i < 5; i++)
- {
- isf_q[i] = add1(isf_q[i], dico21_isf_36b[indice[2] * 5 + i]);
- }
- for (i = 0; i < 4; i++)
- {
- isf_q[i + 5] = add1(isf_q[i + 5], dico22_isf_36b[indice[3] * 4 + i]);
- }
- for (i = 0; i < 7; i++)
- {
- isf_q[i + 9] = add1(isf_q[i + 9], dico23_isf_36b[indice[4] * 7 + i]);
- }
+ for (i = 0; i < 5; i++)
+ {
+ isf_q[i] = add1(isf_q[i], dico21_isf_36b[indice[2] * 5 + i]);
+ }
+ for (i = 0; i < 4; i++)
+ {
+ isf_q[i + 5] = add1(isf_q[i + 5], dico22_isf_36b[indice[3] * 4 + i]);
+ }
+ for (i = 0; i < 7; i++)
+ {
+ isf_q[i + 9] = add1(isf_q[i + 9], dico23_isf_36b[indice[4] * 7 + i]);
+ }
- for (i = 0; i < ORDER; i++)
- {
- tmp = isf_q[i];
- isf_q[i] = add1(tmp, mean_isf[i]);
- isf_q[i] = add1(isf_q[i], vo_mult(MU, past_isfq[i]));
- past_isfq[i] = tmp;
- }
+ for (i = 0; i < ORDER; i++)
+ {
+ tmp = isf_q[i];
+ isf_q[i] = add1(tmp, mean_isf[i]);
+ isf_q[i] = add1(isf_q[i], vo_mult(MU, past_isfq[i]));
+ past_isfq[i] = tmp;
+ }
- if (enc_dec)
- {
- for (i = 0; i < M; i++)
- {
- for (j = (L_MEANBUF - 1); j > 0; j--)
- {
- isf_buf[j * M + i] = isf_buf[(j - 1) * M + i];
- }
- isf_buf[i] = isf_q[i];
- }
- }
- } else
- { /* bad frame */
- for (i = 0; i < M; i++)
- {
- L_tmp = (mean_isf[i] << 14);
- for (j = 0; j < L_MEANBUF; j++)
- {
- L_tmp += (isf_buf[j * M + i] << 14);
- }
- ref_isf[i] = vo_round(L_tmp);
- }
+ if (enc_dec)
+ {
+ for (i = 0; i < M; i++)
+ {
+ for (j = (L_MEANBUF - 1); j > 0; j--)
+ {
+ isf_buf[j * M + i] = isf_buf[(j - 1) * M + i];
+ }
+ isf_buf[i] = isf_q[i];
+ }
+ }
+ } else
+ { /* bad frame */
+ for (i = 0; i < M; i++)
+ {
+ L_tmp = (mean_isf[i] << 14);
+ for (j = 0; j < L_MEANBUF; j++)
+ {
+ L_tmp += (isf_buf[j * M + i] << 14);
+ }
+ ref_isf[i] = vo_round(L_tmp);
+ }
- /* use the past ISFs slightly shifted towards their mean */
- for (i = 0; i < ORDER; i++)
- {
- isf_q[i] = add1(vo_mult(ALPHA, isfold[i]), vo_mult(ONE_ALPHA, ref_isf[i]));
- }
+ /* use the past ISFs slightly shifted towards their mean */
+ for (i = 0; i < ORDER; i++)
+ {
+ isf_q[i] = add1(vo_mult(ALPHA, isfold[i]), vo_mult(ONE_ALPHA, ref_isf[i]));
+ }
- /* estimate past quantized residual to be used in next frame */
- for (i = 0; i < ORDER; i++)
- {
- tmp = add1(ref_isf[i], vo_mult(past_isfq[i], MU)); /* predicted ISF */
- past_isfq[i] = vo_sub(isf_q[i], tmp);
- past_isfq[i] = past_isfq[i] >> 1; /* past_isfq[i] *= 0.5 */
- }
- }
+ /* estimate past quantized residual to be used in next frame */
+ for (i = 0; i < ORDER; i++)
+ {
+ tmp = add1(ref_isf[i], vo_mult(past_isfq[i], MU)); /* predicted ISF */
+ past_isfq[i] = vo_sub(isf_q[i], tmp);
+ past_isfq[i] = past_isfq[i] >> 1; /* past_isfq[i] *= 0.5 */
+ }
+ }
- Reorder_isf(isf_q, ISF_GAP, ORDER);
+ Reorder_isf(isf_q, ISF_GAP, ORDER);
- return;
+ return;
}
@@ -419,122 +419,122 @@
****************************************************************************/
void Reorder_isf(
- Word16 * isf, /* (i/o) Q15: ISF in the frequency domain (0..0.5) */
- Word16 min_dist, /* (i) Q15 : minimum distance to keep */
- Word16 n /* (i) : number of ISF */
- )
+ Word16 * isf, /* (i/o) Q15: ISF in the frequency domain (0..0.5) */
+ Word16 min_dist, /* (i) Q15 : minimum distance to keep */
+ Word16 n /* (i) : number of ISF */
+ )
{
- Word32 i;
- Word16 isf_min;
+ Word32 i;
+ Word16 isf_min;
- isf_min = min_dist;
- for (i = 0; i < n - 1; i++)
- {
- if(isf[i] < isf_min)
- {
- isf[i] = isf_min;
- }
- isf_min = (isf[i] + min_dist);
- }
- return;
+ isf_min = min_dist;
+ for (i = 0; i < n - 1; i++)
+ {
+ if(isf[i] < isf_min)
+ {
+ isf[i] = isf_min;
+ }
+ isf_min = (isf[i] + min_dist);
+ }
+ return;
}
Word16 Sub_VQ( /* output: return quantization index */
- Word16 * x, /* input : ISF residual vector */
- Word16 * dico, /* input : quantization codebook */
- Word16 dim, /* input : dimention of vector */
- Word16 dico_size, /* input : size of quantization codebook */
- Word32 * distance /* output: error of quantization */
- )
+ Word16 * x, /* input : ISF residual vector */
+ Word16 * dico, /* input : quantization codebook */
+ Word16 dim, /* input : dimention of vector */
+ Word16 dico_size, /* input : size of quantization codebook */
+ Word32 * distance /* output: error of quantization */
+ )
{
- Word16 temp, *p_dico;
- Word32 i, j, index;
- Word32 dist_min, dist;
+ Word16 temp, *p_dico;
+ Word32 i, j, index;
+ Word32 dist_min, dist;
- dist_min = MAX_32;
- p_dico = dico;
+ dist_min = MAX_32;
+ p_dico = dico;
- index = 0;
- for (i = 0; i < dico_size; i++)
- {
- dist = 0;
+ index = 0;
+ for (i = 0; i < dico_size; i++)
+ {
+ dist = 0;
- for (j = 0; j < dim; j++)
- {
- temp = x[j] - (*p_dico++);
- dist += (temp * temp)<<1;
- }
+ for (j = 0; j < dim; j++)
+ {
+ temp = x[j] - (*p_dico++);
+ dist += (temp * temp)<<1;
+ }
- if(dist < dist_min)
- {
- dist_min = dist;
- index = i;
- }
- }
+ if(dist < dist_min)
+ {
+ dist_min = dist;
+ index = i;
+ }
+ }
- *distance = dist_min;
+ *distance = dist_min;
- /* Reading the selected vector */
- p_dico = &dico[index * dim];
- for (j = 0; j < dim; j++)
- {
- x[j] = *p_dico++;
- }
+ /* Reading the selected vector */
+ p_dico = &dico[index * dim];
+ for (j = 0; j < dim; j++)
+ {
+ x[j] = *p_dico++;
+ }
- return index;
+ return index;
}
static void VQ_stage1(
- Word16 * x, /* input : ISF residual vector */
- Word16 * dico, /* input : quantization codebook */
- Word16 dim, /* input : dimention of vector */
- Word16 dico_size, /* input : size of quantization codebook */
- Word16 * index, /* output: indices of survivors */
- Word16 surv /* input : number of survivor */
- )
+ Word16 * x, /* input : ISF residual vector */
+ Word16 * dico, /* input : quantization codebook */
+ Word16 dim, /* input : dimention of vector */
+ Word16 dico_size, /* input : size of quantization codebook */
+ Word16 * index, /* output: indices of survivors */
+ Word16 surv /* input : number of survivor */
+ )
{
- Word16 temp, *p_dico;
- Word32 i, j, k, l;
- Word32 dist_min[N_SURV_MAX], dist;
+ Word16 temp, *p_dico;
+ Word32 i, j, k, l;
+ Word32 dist_min[N_SURV_MAX], dist;
- dist_min[0] = MAX_32;
- dist_min[1] = MAX_32;
- dist_min[2] = MAX_32;
- dist_min[3] = MAX_32;
- index[0] = 0;
- index[1] = 1;
- index[2] = 2;
- index[3] = 3;
+ dist_min[0] = MAX_32;
+ dist_min[1] = MAX_32;
+ dist_min[2] = MAX_32;
+ dist_min[3] = MAX_32;
+ index[0] = 0;
+ index[1] = 1;
+ index[2] = 2;
+ index[3] = 3;
- p_dico = dico;
+ p_dico = dico;
- for (i = 0; i < dico_size; i++)
- {
- dist = 0;
- for (j = 0; j < dim; j++)
- {
- temp = x[j] - (*p_dico++);
- dist += (temp * temp)<<1;
- }
+ for (i = 0; i < dico_size; i++)
+ {
+ dist = 0;
+ for (j = 0; j < dim; j++)
+ {
+ temp = x[j] - (*p_dico++);
+ dist += (temp * temp)<<1;
+ }
- for (k = 0; k < surv; k++)
- {
- if(dist < dist_min[k])
- {
- for (l = surv - 1; l > k; l--)
- {
- dist_min[l] = dist_min[l - 1];
- index[l] = index[l - 1];
- }
- dist_min[k] = dist;
- index[k] = i;
- break;
- }
- }
- }
- return;
+ for (k = 0; k < surv; k++)
+ {
+ if(dist < dist_min[k])
+ {
+ for (l = surv - 1; l > k; l--)
+ {
+ dist_min[l] = dist_min[l - 1];
+ index[l] = index[l - 1];
+ }
+ dist_min[k] = dist;
+ index[k] = i;
+ break;
+ }
+ }
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/random.c b/media/libstagefright/codecs/amrwbenc/src/random.c
index b896863..758343c 100644
--- a/media/libstagefright/codecs/amrwbenc/src/random.c
+++ b/media/libstagefright/codecs/amrwbenc/src/random.c
@@ -26,8 +26,8 @@
Word16 Random(Word16 * seed)
{
- /* static Word16 seed = 21845; */
- *seed = (Word16)(L_add((L_mult(*seed, 31821) >> 1), 13849L));
- return (*seed);
+ /* static Word16 seed = 21845; */
+ *seed = (Word16)(L_add((L_mult(*seed, 31821) >> 1), 13849L));
+ return (*seed);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/residu.c b/media/libstagefright/codecs/amrwbenc/src/residu.c
index b0c04b5..76d0e41 100644
--- a/media/libstagefright/codecs/amrwbenc/src/residu.c
+++ b/media/libstagefright/codecs/amrwbenc/src/residu.c
@@ -26,41 +26,41 @@
#include "basic_op.h"
void Residu(
- Word16 a[], /* (i) Q12 : prediction coefficients */
- Word16 x[], /* (i) : speech (values x[-m..-1] are needed */
- Word16 y[], /* (o) x2 : residual signal */
- Word16 lg /* (i) : size of filtering */
- )
+ Word16 a[], /* (i) Q12 : prediction coefficients */
+ Word16 x[], /* (i) : speech (values x[-m..-1] are needed */
+ Word16 y[], /* (o) x2 : residual signal */
+ Word16 lg /* (i) : size of filtering */
+ )
{
- Word16 i,*p1, *p2;
- Word32 s;
- for (i = 0; i < lg; i++)
- {
- p1 = a;
- p2 = &x[i];
- s = vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1++), (*p2--));
- s += vo_mult32((*p1), (*p2));
+ Word16 i,*p1, *p2;
+ Word32 s;
+ for (i = 0; i < lg; i++)
+ {
+ p1 = a;
+ p2 = &x[i];
+ s = vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1++), (*p2--));
+ s += vo_mult32((*p1), (*p2));
- s = L_shl2(s, 5);
- y[i] = extract_h(L_add(s, 0x8000));
- }
+ s = L_shl2(s, 5);
+ y[i] = extract_h(L_add(s, 0x8000));
+ }
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/scale.c b/media/libstagefright/codecs/amrwbenc/src/scale.c
index 418cc06..21458c8 100644
--- a/media/libstagefright/codecs/amrwbenc/src/scale.c
+++ b/media/libstagefright/codecs/amrwbenc/src/scale.c
@@ -25,32 +25,32 @@
#include "basic_op.h"
void Scale_sig(
- Word16 x[], /* (i/o) : signal to scale */
- Word16 lg, /* (i) : size of x[] */
- Word16 exp /* (i) : exponent: x = round(x << exp) */
- )
+ Word16 x[], /* (i/o) : signal to scale */
+ Word16 lg, /* (i) : size of x[] */
+ Word16 exp /* (i) : exponent: x = round(x << exp) */
+ )
{
- Word32 i;
- Word32 L_tmp;
- if(exp > 0)
- {
- for (i = lg - 1 ; i >= 0; i--)
- {
- L_tmp = L_shl2(x[i], 16 + exp);
- x[i] = extract_h(L_add(L_tmp, 0x8000));
- }
- }
- else
- {
- exp = -exp;
- for (i = lg - 1; i >= 0; i--)
- {
- L_tmp = x[i] << 16;
- L_tmp >>= exp;
- x[i] = (L_tmp + 0x8000)>>16;
- }
- }
- return;
+ Word32 i;
+ Word32 L_tmp;
+ if(exp > 0)
+ {
+ for (i = lg - 1 ; i >= 0; i--)
+ {
+ L_tmp = L_shl2(x[i], 16 + exp);
+ x[i] = extract_h(L_add(L_tmp, 0x8000));
+ }
+ }
+ else
+ {
+ exp = -exp;
+ for (i = lg - 1; i >= 0; i--)
+ {
+ L_tmp = x[i] << 16;
+ L_tmp >>= exp;
+ x[i] = (L_tmp + 0x8000)>>16;
+ }
+ }
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/stream.c b/media/libstagefright/codecs/amrwbenc/src/stream.c
index 780f009..a39149e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/stream.c
+++ b/media/libstagefright/codecs/amrwbenc/src/stream.c
@@ -25,34 +25,34 @@
void voAWB_InitFrameBuffer(FrameStream *stream)
{
- stream->set_ptr = NULL;
- stream->frame_ptr_bk = stream->frame_ptr;
- stream->set_len = 0;
- stream->framebuffer_len = 0;
- stream->frame_storelen = 0;
+ stream->set_ptr = NULL;
+ stream->frame_ptr_bk = stream->frame_ptr;
+ stream->set_len = 0;
+ stream->framebuffer_len = 0;
+ stream->frame_storelen = 0;
}
void voAWB_UpdateFrameBuffer(
- FrameStream *stream,
- VO_MEM_OPERATOR *pMemOP
- )
+ FrameStream *stream,
+ VO_MEM_OPERATOR *pMemOP
+ )
{
- int len;
- len = MIN(Frame_Maxsize - stream->frame_storelen, stream->set_len);
- pMemOP->Copy(VO_INDEX_ENC_AMRWB, stream->frame_ptr_bk + stream->frame_storelen , stream->set_ptr, len);
- stream->set_len -= len;
- stream->set_ptr += len;
- stream->framebuffer_len = stream->frame_storelen + len;
- stream->frame_ptr = stream->frame_ptr_bk;
- stream->used_len += len;
+ int len;
+ len = MIN(Frame_Maxsize - stream->frame_storelen, stream->set_len);
+ pMemOP->Copy(VO_INDEX_ENC_AMRWB, stream->frame_ptr_bk + stream->frame_storelen , stream->set_ptr, len);
+ stream->set_len -= len;
+ stream->set_ptr += len;
+ stream->framebuffer_len = stream->frame_storelen + len;
+ stream->frame_ptr = stream->frame_ptr_bk;
+ stream->used_len += len;
}
void voAWB_FlushFrameBuffer(FrameStream *stream)
{
- stream->set_ptr = NULL;
- stream->frame_ptr_bk = stream->frame_ptr;
- stream->set_len = 0;
- stream->framebuffer_len = 0;
- stream->frame_storelen = 0;
+ stream->set_ptr = NULL;
+ stream->frame_ptr_bk = stream->frame_ptr;
+ stream->set_len = 0;
+ stream->framebuffer_len = 0;
+ stream->frame_storelen = 0;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/syn_filt.c b/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
index 961aadc..7eba12f 100644
--- a/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
+++ b/media/libstagefright/codecs/amrwbenc/src/syn_filt.c
@@ -29,134 +29,134 @@
#define UNUSED(x) (void)(x)
void Syn_filt(
- Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
- Word16 x[], /* (i) : input signal */
- Word16 y[], /* (o) : output signal */
- Word16 lg, /* (i) : size of filtering */
- Word16 mem[], /* (i/o) : memory associated with this filtering. */
- Word16 update /* (i) : 0=no update, 1=update of memory. */
- )
+ Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
+ Word16 x[], /* (i) : input signal */
+ Word16 y[], /* (o) : output signal */
+ Word16 lg, /* (i) : size of filtering */
+ Word16 mem[], /* (i/o) : memory associated with this filtering. */
+ Word16 update /* (i) : 0=no update, 1=update of memory. */
+ )
{
- Word32 i, a0;
- Word16 y_buf[L_SUBFR16k + M16k];
- Word32 L_tmp;
- Word16 *yy, *p1, *p2;
- yy = &y_buf[0];
- /* copy initial filter states into synthesis buffer */
- for (i = 0; i < 16; i++)
- {
- *yy++ = mem[i];
- }
- a0 = (a[0] >> 1); /* input / 2 */
- /* Do the filtering. */
- for (i = 0; i < lg; i++)
- {
- p1 = &a[1];
- p2 = &yy[i-1];
- L_tmp = vo_mult32(a0, x[i]);
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1++), (*p2--));
- L_tmp -= vo_mult32((*p1), (*p2));
+ Word32 i, a0;
+ Word16 y_buf[L_SUBFR16k + M16k];
+ Word32 L_tmp;
+ Word16 *yy, *p1, *p2;
+ yy = &y_buf[0];
+ /* copy initial filter states into synthesis buffer */
+ for (i = 0; i < 16; i++)
+ {
+ *yy++ = mem[i];
+ }
+ a0 = (a[0] >> 1); /* input / 2 */
+ /* Do the filtering. */
+ for (i = 0; i < lg; i++)
+ {
+ p1 = &a[1];
+ p2 = &yy[i-1];
+ L_tmp = vo_mult32(a0, x[i]);
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1++), (*p2--));
+ L_tmp -= vo_mult32((*p1), (*p2));
- L_tmp = L_shl2(L_tmp, 4);
- y[i] = yy[i] = extract_h(L_add(L_tmp, 0x8000));
- }
- /* Update memory if required */
- if (update)
- for (i = 0; i < 16; i++)
- {
- mem[i] = yy[lg - 16 + i];
- }
- return;
+ L_tmp = L_shl2(L_tmp, 4);
+ y[i] = yy[i] = extract_h(L_add(L_tmp, 0x8000));
+ }
+ /* Update memory if required */
+ if (update)
+ for (i = 0; i < 16; i++)
+ {
+ mem[i] = yy[lg - 16 + i];
+ }
+ return;
}
void Syn_filt_32(
- Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
- Word16 m, /* (i) : order of LP filter */
- Word16 exc[], /* (i) Qnew: excitation (exc[i] >> Qnew) */
- Word16 Qnew, /* (i) : exc scaling = 0(min) to 8(max) */
- Word16 sig_hi[], /* (o) /16 : synthesis high */
- Word16 sig_lo[], /* (o) /16 : synthesis low */
- Word16 lg /* (i) : size of filtering */
- )
+ Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients */
+ Word16 m, /* (i) : order of LP filter */
+ Word16 exc[], /* (i) Qnew: excitation (exc[i] >> Qnew) */
+ Word16 Qnew, /* (i) : exc scaling = 0(min) to 8(max) */
+ Word16 sig_hi[], /* (o) /16 : synthesis high */
+ Word16 sig_lo[], /* (o) /16 : synthesis low */
+ Word16 lg /* (i) : size of filtering */
+ )
{
- Word32 i,a0;
- Word32 L_tmp, L_tmp1;
- Word16 *p1, *p2, *p3;
+ Word32 i,a0;
+ Word32 L_tmp, L_tmp1;
+ Word16 *p1, *p2, *p3;
UNUSED(m);
- a0 = a[0] >> (4 + Qnew); /* input / 16 and >>Qnew */
- /* Do the filtering. */
- for (i = 0; i < lg; i++)
- {
- L_tmp = 0;
- L_tmp1 = 0;
- p1 = a;
- p2 = &sig_lo[i - 1];
- p3 = &sig_hi[i - 1];
+ a0 = a[0] >> (4 + Qnew); /* input / 16 and >>Qnew */
+ /* Do the filtering. */
+ for (i = 0; i < lg; i++)
+ {
+ L_tmp = 0;
+ L_tmp1 = 0;
+ p1 = a;
+ p2 = &sig_lo[i - 1];
+ p3 = &sig_hi[i - 1];
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp -= vo_mult32((*p2--), (*p1));
- L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
+ L_tmp -= vo_mult32((*p2--), (*p1));
+ L_tmp1 -= vo_mult32((*p3--), (*p1++));
- L_tmp = L_tmp >> 11;
- L_tmp += vo_L_mult(exc[i], a0);
+ L_tmp = L_tmp >> 11;
+ L_tmp += vo_L_mult(exc[i], a0);
- /* sig_hi = bit16 to bit31 of synthesis */
- L_tmp = L_tmp - (L_tmp1<<1);
+ /* sig_hi = bit16 to bit31 of synthesis */
+ L_tmp = L_tmp - (L_tmp1<<1);
- L_tmp = L_tmp >> 3; /* ai in Q12 */
- sig_hi[i] = extract_h(L_tmp);
+ L_tmp = L_tmp >> 3; /* ai in Q12 */
+ sig_hi[i] = extract_h(L_tmp);
- /* sig_lo = bit4 to bit15 of synthesis */
- L_tmp >>= 4; /* 4 : sig_lo[i] >> 4 */
- sig_lo[i] = (Word16)((L_tmp - (sig_hi[i] << 13)));
- }
+ /* sig_lo = bit4 to bit15 of synthesis */
+ L_tmp >>= 4; /* 4 : sig_lo[i] >> 4 */
+ sig_lo[i] = (Word16)((L_tmp - (sig_hi[i] << 13)));
+ }
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/updt_tar.c b/media/libstagefright/codecs/amrwbenc/src/updt_tar.c
index 96779fd..ba7c2ff 100644
--- a/media/libstagefright/codecs/amrwbenc/src/updt_tar.c
+++ b/media/libstagefright/codecs/amrwbenc/src/updt_tar.c
@@ -25,24 +25,25 @@
#include "basic_op.h"
void Updt_tar(
- Word16 * x, /* (i) Q0 : old target (for pitch search) */
- Word16 * x2, /* (o) Q0 : new target (for codebook search) */
- Word16 * y, /* (i) Q0 : filtered adaptive codebook vector */
- Word16 gain, /* (i) Q14 : adaptive codebook gain */
- Word16 L /* (i) : subframe size */
- )
+ Word16 * x, /* (i) Q0 : old target (for pitch search) */
+ Word16 * x2, /* (o) Q0 : new target (for codebook search) */
+ Word16 * y, /* (i) Q0 : filtered adaptive codebook vector */
+ Word16 gain, /* (i) Q14 : adaptive codebook gain */
+ Word16 L /* (i) : subframe size */
+ )
{
- Word32 i;
- Word32 L_tmp;
+ Word32 i;
+ Word32 L_tmp, L_tmp2;
- for (i = 0; i < L; i++)
- {
- L_tmp = x[i] << 15;
- L_tmp -= (y[i] * gain)<<1;
- x2[i] = extract_h(L_shl2(L_tmp, 1));
- }
+ for (i = 0; i < L; i++)
+ {
+ L_tmp = x[i] << 15;
+ L_tmp2 = L_mult(y[i], gain);
+ L_tmp = L_sub(L_tmp, L_tmp2);
+ x2[i] = extract_h(L_shl2(L_tmp, 1));
+ }
- return;
+ return;
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/util.c b/media/libstagefright/codecs/amrwbenc/src/util.c
index 333140d..374245f 100644
--- a/media/libstagefright/codecs/amrwbenc/src/util.c
+++ b/media/libstagefright/codecs/amrwbenc/src/util.c
@@ -30,15 +30,15 @@
************************************************************************/
void Set_zero(
- Word16 x[], /* (o) : vector to clear */
- Word16 L /* (i) : length of vector */
- )
+ Word16 x[], /* (o) : vector to clear */
+ Word16 L /* (i) : length of vector */
+ )
{
- Word32 num = (Word32)L;
- while (num > 0) {
- *x++ = 0;
+ Word32 num = (Word32)L;
+ while (num > 0) {
+ *x++ = 0;
--num;
- }
+ }
}
@@ -49,28 +49,28 @@
*********************************************************************/
void Copy(
- Word16 x[], /* (i) : input vector */
- Word16 y[], /* (o) : output vector */
- Word16 L /* (i) : vector length */
- )
+ Word16 x[], /* (i) : input vector */
+ Word16 y[], /* (o) : output vector */
+ Word16 L /* (i) : vector length */
+ )
{
- Word32 temp1,temp2,num;
+ Word32 temp1,temp2,num;
if (L <= 0) {
return;
}
- if(L&1)
- {
- temp1 = *x++;
- *y++ = temp1;
- }
- num = (Word32)(L>>1);
- while (num > 0) {
- temp1 = *x++;
- temp2 = *x++;
- *y++ = temp1;
- *y++ = temp2;
+ if(L&1)
+ {
+ temp1 = *x++;
+ *y++ = temp1;
+ }
+ num = (Word32)(L>>1);
+ while (num > 0) {
+ temp1 = *x++;
+ temp2 = *x++;
+ *y++ = temp1;
+ *y++ = temp2;
--num;
- }
+ }
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c b/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
index df7b9b3..4cafb01 100644
--- a/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
+++ b/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c
@@ -19,8 +19,8 @@
* *
* Description: Performs the main encoder routine *
* Fixed-point C simulation of AMR WB ACELP coding *
-* algorithm with 20 msspeech frames for *
-* wideband speech signals. *
+* algorithm with 20 msspeech frames for *
+* wideband speech signals. *
* *
************************************************************************/
@@ -51,95 +51,95 @@
/* isp tables for initialization */
static Word16 isp_init[M] =
{
- 32138, 30274, 27246, 23170, 18205, 12540, 6393, 0,
- -6393, -12540, -18205, -23170, -27246, -30274, -32138, 1475
+ 32138, 30274, 27246, 23170, 18205, 12540, 6393, 0,
+ -6393, -12540, -18205, -23170, -27246, -30274, -32138, 1475
};
static Word16 isf_init[M] =
{
- 1024, 2048, 3072, 4096, 5120, 6144, 7168, 8192,
- 9216, 10240, 11264, 12288, 13312, 14336, 15360, 3840
+ 1024, 2048, 3072, 4096, 5120, 6144, 7168, 8192,
+ 9216, 10240, 11264, 12288, 13312, 14336, 15360, 3840
};
/* High Band encoding */
static const Word16 HP_gain[16] =
{
- 3624, 4673, 5597, 6479, 7425, 8378, 9324, 10264,
- 11210, 12206, 13391, 14844, 16770, 19655, 24289, 32728
+ 3624, 4673, 5597, 6479, 7425, 8378, 9324, 10264,
+ 11210, 12206, 13391, 14844, 16770, 19655, 24289, 32728
};
/* Private function declaration */
static Word16 synthesis(
- Word16 Aq[], /* A(z) : quantized Az */
- Word16 exc[], /* (i) : excitation at 12kHz */
- Word16 Q_new, /* (i) : scaling performed on exc */
- Word16 synth16k[], /* (o) : 16kHz synthesis signal */
- Coder_State * st /* (i/o) : State structure */
- );
+ Word16 Aq[], /* A(z) : quantized Az */
+ Word16 exc[], /* (i) : excitation at 12kHz */
+ Word16 Q_new, /* (i) : scaling performed on exc */
+ Word16 synth16k[], /* (o) : 16kHz synthesis signal */
+ Coder_State * st /* (i/o) : State structure */
+ );
/* Codec some parameters initialization */
void Reset_encoder(void *st, Word16 reset_all)
{
- Word16 i;
- Coder_State *cod_state;
- cod_state = (Coder_State *) st;
- Set_zero(cod_state->old_exc, PIT_MAX + L_INTERPOL);
- Set_zero(cod_state->mem_syn, M);
- Set_zero(cod_state->past_isfq, M);
- cod_state->mem_w0 = 0;
- cod_state->tilt_code = 0;
- cod_state->first_frame = 1;
- Init_gp_clip(cod_state->gp_clip);
- cod_state->L_gc_thres = 0;
- if (reset_all != 0)
- {
- /* Static vectors to zero */
- Set_zero(cod_state->old_speech, L_TOTAL - L_FRAME);
- Set_zero(cod_state->old_wsp, (PIT_MAX / OPL_DECIM));
- Set_zero(cod_state->mem_decim2, 3);
- /* routines initialization */
- Init_Decim_12k8(cod_state->mem_decim);
- Init_HP50_12k8(cod_state->mem_sig_in);
- Init_Levinson(cod_state->mem_levinson);
- Init_Q_gain2(cod_state->qua_gain);
- Init_Hp_wsp(cod_state->hp_wsp_mem);
- /* isp initialization */
- Copy(isp_init, cod_state->ispold, M);
- Copy(isp_init, cod_state->ispold_q, M);
- /* variable initialization */
- cod_state->mem_preemph = 0;
- cod_state->mem_wsp = 0;
- cod_state->Q_old = 15;
- cod_state->Q_max[0] = 15;
- cod_state->Q_max[1] = 15;
- cod_state->old_wsp_max = 0;
- cod_state->old_wsp_shift = 0;
- /* pitch ol initialization */
- cod_state->old_T0_med = 40;
- cod_state->ol_gain = 0;
- cod_state->ada_w = 0;
- cod_state->ol_wght_flg = 0;
- for (i = 0; i < 5; i++)
- {
- cod_state->old_ol_lag[i] = 40;
- }
- Set_zero(cod_state->old_hp_wsp, (L_FRAME / 2) / OPL_DECIM + (PIT_MAX / OPL_DECIM));
- Set_zero(cod_state->mem_syn_hf, M);
- Set_zero(cod_state->mem_syn_hi, M);
- Set_zero(cod_state->mem_syn_lo, M);
- Init_HP50_12k8(cod_state->mem_sig_out);
- Init_Filt_6k_7k(cod_state->mem_hf);
- Init_HP400_12k8(cod_state->mem_hp400);
- Copy(isf_init, cod_state->isfold, M);
- cod_state->mem_deemph = 0;
- cod_state->seed2 = 21845;
- Init_Filt_6k_7k(cod_state->mem_hf2);
- cod_state->gain_alpha = 32767;
- cod_state->vad_hist = 0;
- wb_vad_reset(cod_state->vadSt);
- dtx_enc_reset(cod_state->dtx_encSt, isf_init);
- }
- return;
+ Word16 i;
+ Coder_State *cod_state;
+ cod_state = (Coder_State *) st;
+ Set_zero(cod_state->old_exc, PIT_MAX + L_INTERPOL);
+ Set_zero(cod_state->mem_syn, M);
+ Set_zero(cod_state->past_isfq, M);
+ cod_state->mem_w0 = 0;
+ cod_state->tilt_code = 0;
+ cod_state->first_frame = 1;
+ Init_gp_clip(cod_state->gp_clip);
+ cod_state->L_gc_thres = 0;
+ if (reset_all != 0)
+ {
+ /* Static vectors to zero */
+ Set_zero(cod_state->old_speech, L_TOTAL - L_FRAME);
+ Set_zero(cod_state->old_wsp, (PIT_MAX / OPL_DECIM));
+ Set_zero(cod_state->mem_decim2, 3);
+ /* routines initialization */
+ Init_Decim_12k8(cod_state->mem_decim);
+ Init_HP50_12k8(cod_state->mem_sig_in);
+ Init_Levinson(cod_state->mem_levinson);
+ Init_Q_gain2(cod_state->qua_gain);
+ Init_Hp_wsp(cod_state->hp_wsp_mem);
+ /* isp initialization */
+ Copy(isp_init, cod_state->ispold, M);
+ Copy(isp_init, cod_state->ispold_q, M);
+ /* variable initialization */
+ cod_state->mem_preemph = 0;
+ cod_state->mem_wsp = 0;
+ cod_state->Q_old = 15;
+ cod_state->Q_max[0] = 15;
+ cod_state->Q_max[1] = 15;
+ cod_state->old_wsp_max = 0;
+ cod_state->old_wsp_shift = 0;
+ /* pitch ol initialization */
+ cod_state->old_T0_med = 40;
+ cod_state->ol_gain = 0;
+ cod_state->ada_w = 0;
+ cod_state->ol_wght_flg = 0;
+ for (i = 0; i < 5; i++)
+ {
+ cod_state->old_ol_lag[i] = 40;
+ }
+ Set_zero(cod_state->old_hp_wsp, (L_FRAME / 2) / OPL_DECIM + (PIT_MAX / OPL_DECIM));
+ Set_zero(cod_state->mem_syn_hf, M);
+ Set_zero(cod_state->mem_syn_hi, M);
+ Set_zero(cod_state->mem_syn_lo, M);
+ Init_HP50_12k8(cod_state->mem_sig_out);
+ Init_Filt_6k_7k(cod_state->mem_hf);
+ Init_HP400_12k8(cod_state->mem_hp400);
+ Copy(isf_init, cod_state->isfold, M);
+ cod_state->mem_deemph = 0;
+ cod_state->seed2 = 21845;
+ Init_Filt_6k_7k(cod_state->mem_hf2);
+ cod_state->gain_alpha = 32767;
+ cod_state->vad_hist = 0;
+ wb_vad_reset(cod_state->vadSt);
+ dtx_enc_reset(cod_state->dtx_encSt, isf_init);
+ }
+ return;
}
/*-----------------------------------------------------------------*
@@ -149,1176 +149,1180 @@
* *
*-----------------------------------------------------------------*/
void coder(
- Word16 * mode, /* input : used mode */
- Word16 speech16k[], /* input : 320 new speech samples (at 16 kHz) */
- Word16 prms[], /* output: output parameters */
- Word16 * ser_size, /* output: bit rate of the used mode */
- void *spe_state, /* i/o : State structure */
- Word16 allow_dtx /* input : DTX ON/OFF */
- )
+ Word16 * mode, /* input : used mode */
+ Word16 speech16k[], /* input : 320 new speech samples (at 16 kHz) */
+ Word16 prms[], /* output: output parameters */
+ Word16 * ser_size, /* output: bit rate of the used mode */
+ void *spe_state, /* i/o : State structure */
+ Word16 allow_dtx /* input : DTX ON/OFF */
+ )
{
- /* Coder states */
- Coder_State *st;
- /* Speech vector */
- Word16 old_speech[L_TOTAL];
- Word16 *new_speech, *speech, *p_window;
+ /* Coder states */
+ Coder_State *st;
+ /* Speech vector */
+ Word16 old_speech[L_TOTAL];
+ Word16 *new_speech, *speech, *p_window;
- /* Weighted speech vector */
- Word16 old_wsp[L_FRAME + (PIT_MAX / OPL_DECIM)];
- Word16 *wsp;
+ /* Weighted speech vector */
+ Word16 old_wsp[L_FRAME + (PIT_MAX / OPL_DECIM)];
+ Word16 *wsp;
- /* Excitation vector */
- Word16 old_exc[(L_FRAME + 1) + PIT_MAX + L_INTERPOL];
- Word16 *exc;
+ /* Excitation vector */
+ Word16 old_exc[(L_FRAME + 1) + PIT_MAX + L_INTERPOL];
+ Word16 *exc;
- /* LPC coefficients */
- Word16 r_h[M + 1], r_l[M + 1]; /* Autocorrelations of windowed speech */
- Word16 rc[M]; /* Reflection coefficients. */
- Word16 Ap[M + 1]; /* A(z) with spectral expansion */
- Word16 ispnew[M]; /* immittance spectral pairs at 4nd sfr */
- Word16 ispnew_q[M]; /* quantized ISPs at 4nd subframe */
- Word16 isf[M]; /* ISF (frequency domain) at 4nd sfr */
- Word16 *p_A, *p_Aq; /* ptr to A(z) for the 4 subframes */
- Word16 A[NB_SUBFR * (M + 1)]; /* A(z) unquantized for the 4 subframes */
- Word16 Aq[NB_SUBFR * (M + 1)]; /* A(z) quantized for the 4 subframes */
+ /* LPC coefficients */
+ Word16 r_h[M + 1], r_l[M + 1]; /* Autocorrelations of windowed speech */
+ Word16 rc[M]; /* Reflection coefficients. */
+ Word16 Ap[M + 1]; /* A(z) with spectral expansion */
+ Word16 ispnew[M]; /* immittance spectral pairs at 4nd sfr */
+ Word16 ispnew_q[M]; /* quantized ISPs at 4nd subframe */
+ Word16 isf[M]; /* ISF (frequency domain) at 4nd sfr */
+ Word16 *p_A, *p_Aq; /* ptr to A(z) for the 4 subframes */
+ Word16 A[NB_SUBFR * (M + 1)]; /* A(z) unquantized for the 4 subframes */
+ Word16 Aq[NB_SUBFR * (M + 1)]; /* A(z) quantized for the 4 subframes */
- /* Other vectors */
- Word16 xn[L_SUBFR]; /* Target vector for pitch search */
- Word16 xn2[L_SUBFR]; /* Target vector for codebook search */
- Word16 dn[L_SUBFR]; /* Correlation between xn2 and h1 */
- Word16 cn[L_SUBFR]; /* Target vector in residual domain */
- Word16 h1[L_SUBFR]; /* Impulse response vector */
- Word16 h2[L_SUBFR]; /* Impulse response vector */
- Word16 code[L_SUBFR]; /* Fixed codebook excitation */
- Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */
- Word16 y2[L_SUBFR]; /* Filtered adaptive excitation */
- Word16 error[M + L_SUBFR]; /* error of quantization */
- Word16 synth[L_SUBFR]; /* 12.8kHz synthesis vector */
- Word16 exc2[L_FRAME]; /* excitation vector */
- Word16 buf[L_FRAME]; /* VAD buffer */
+ /* Other vectors */
+ Word16 xn[L_SUBFR]; /* Target vector for pitch search */
+ Word16 xn2[L_SUBFR]; /* Target vector for codebook search */
+ Word16 dn[L_SUBFR]; /* Correlation between xn2 and h1 */
+ Word16 cn[L_SUBFR]; /* Target vector in residual domain */
+ Word16 h1[L_SUBFR]; /* Impulse response vector */
+ Word16 h2[L_SUBFR]; /* Impulse response vector */
+ Word16 code[L_SUBFR]; /* Fixed codebook excitation */
+ Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */
+ Word16 y2[L_SUBFR]; /* Filtered adaptive excitation */
+ Word16 error[M + L_SUBFR]; /* error of quantization */
+ Word16 synth[L_SUBFR]; /* 12.8kHz synthesis vector */
+ Word16 exc2[L_FRAME]; /* excitation vector */
+ Word16 buf[L_FRAME]; /* VAD buffer */
- /* Scalars */
- Word32 i, j, i_subfr, select, pit_flag, clip_gain, vad_flag;
- Word16 codec_mode;
- Word16 T_op, T_op2, T0, T0_min, T0_max, T0_frac, index;
- Word16 gain_pit, gain_code, g_coeff[4], g_coeff2[4];
- Word16 tmp, gain1, gain2, exp, Q_new, mu, shift, max;
- Word16 voice_fac;
- Word16 indice[8];
- Word32 L_tmp, L_gain_code, L_max, L_tmp1;
- Word16 code2[L_SUBFR]; /* Fixed codebook excitation */
- Word16 stab_fac, fac, gain_code_lo;
+ /* Scalars */
+ Word32 i, j, i_subfr, select, pit_flag, clip_gain, vad_flag;
+ Word16 codec_mode;
+ Word16 T_op, T_op2, T0, T0_min, T0_max, T0_frac, index;
+ Word16 gain_pit, gain_code, g_coeff[4], g_coeff2[4];
+ Word16 tmp, gain1, gain2, exp, Q_new, mu, shift, max;
+ Word16 voice_fac;
+ Word16 indice[8];
+ Word32 L_tmp, L_gain_code, L_max, L_tmp1;
+ Word16 code2[L_SUBFR]; /* Fixed codebook excitation */
+ Word16 stab_fac, fac, gain_code_lo;
- Word16 corr_gain;
- Word16 *vo_p0, *vo_p1, *vo_p2, *vo_p3;
+ Word16 corr_gain;
+ Word16 *vo_p0, *vo_p1, *vo_p2, *vo_p3;
- st = (Coder_State *) spe_state;
+ st = (Coder_State *) spe_state;
- *ser_size = nb_of_bits[*mode];
- codec_mode = *mode;
+ *ser_size = nb_of_bits[*mode];
+ codec_mode = *mode;
- /*--------------------------------------------------------------------------*
- * Initialize pointers to speech vector. *
- * *
- * *
- * |-------|-------|-------|-------|-------|-------| *
- * past sp sf1 sf2 sf3 sf4 L_NEXT *
- * <------- Total speech buffer (L_TOTAL) ------> *
- * old_speech *
- * <------- LPC analysis window (L_WINDOW) ------> *
- * | <-- present frame (L_FRAME) ----> *
- * p_window | <----- new speech (L_FRAME) ----> *
- * | | *
- * speech | *
- * new_speech *
- *--------------------------------------------------------------------------*/
+ /*--------------------------------------------------------------------------*
+ * Initialize pointers to speech vector. *
+ * *
+ * *
+ * |-------|-------|-------|-------|-------|-------| *
+ * past sp sf1 sf2 sf3 sf4 L_NEXT *
+ * <------- Total speech buffer (L_TOTAL) ------> *
+ * old_speech *
+ * <------- LPC analysis window (L_WINDOW) ------> *
+ * | <-- present frame (L_FRAME) ----> *
+ * p_window | <----- new speech (L_FRAME) ----> *
+ * | | *
+ * speech | *
+ * new_speech *
+ *--------------------------------------------------------------------------*/
- new_speech = old_speech + L_TOTAL - L_FRAME - L_FILT; /* New speech */
- speech = old_speech + L_TOTAL - L_FRAME - L_NEXT; /* Present frame */
- p_window = old_speech + L_TOTAL - L_WINDOW;
+ new_speech = old_speech + L_TOTAL - L_FRAME - L_FILT; /* New speech */
+ speech = old_speech + L_TOTAL - L_FRAME - L_NEXT; /* Present frame */
+ p_window = old_speech + L_TOTAL - L_WINDOW;
- exc = old_exc + PIT_MAX + L_INTERPOL;
- wsp = old_wsp + (PIT_MAX / OPL_DECIM);
+ exc = old_exc + PIT_MAX + L_INTERPOL;
+ wsp = old_wsp + (PIT_MAX / OPL_DECIM);
- /* copy coder memory state into working space */
- Copy(st->old_speech, old_speech, L_TOTAL - L_FRAME);
- Copy(st->old_wsp, old_wsp, PIT_MAX / OPL_DECIM);
- Copy(st->old_exc, old_exc, PIT_MAX + L_INTERPOL);
+ /* copy coder memory state into working space */
+ Copy(st->old_speech, old_speech, L_TOTAL - L_FRAME);
+ Copy(st->old_wsp, old_wsp, PIT_MAX / OPL_DECIM);
+ Copy(st->old_exc, old_exc, PIT_MAX + L_INTERPOL);
- /*---------------------------------------------------------------*
- * Down sampling signal from 16kHz to 12.8kHz *
- * -> The signal is extended by L_FILT samples (padded to zero) *
- * to avoid additional delay (L_FILT samples) in the coder. *
- * The last L_FILT samples are approximated after decimation and *
- * are used (and windowed) only in autocorrelations. *
- *---------------------------------------------------------------*/
+ /*---------------------------------------------------------------*
+ * Down sampling signal from 16kHz to 12.8kHz *
+ * -> The signal is extended by L_FILT samples (padded to zero) *
+ * to avoid additional delay (L_FILT samples) in the coder. *
+ * The last L_FILT samples are approximated after decimation and *
+ * are used (and windowed) only in autocorrelations. *
+ *---------------------------------------------------------------*/
- Decim_12k8(speech16k, L_FRAME16k, new_speech, st->mem_decim);
+ Decim_12k8(speech16k, L_FRAME16k, new_speech, st->mem_decim);
- /* last L_FILT samples for autocorrelation window */
- Copy(st->mem_decim, code, 2 * L_FILT16k);
- Set_zero(error, L_FILT16k); /* set next sample to zero */
- Decim_12k8(error, L_FILT16k, new_speech + L_FRAME, code);
+ /* last L_FILT samples for autocorrelation window */
+ Copy(st->mem_decim, code, 2 * L_FILT16k);
+ Set_zero(error, L_FILT16k); /* set next sample to zero */
+ Decim_12k8(error, L_FILT16k, new_speech + L_FRAME, code);
- /*---------------------------------------------------------------*
- * Perform 50Hz HP filtering of input signal. *
- *---------------------------------------------------------------*/
+ /*---------------------------------------------------------------*
+ * Perform 50Hz HP filtering of input signal. *
+ *---------------------------------------------------------------*/
- HP50_12k8(new_speech, L_FRAME, st->mem_sig_in);
+ HP50_12k8(new_speech, L_FRAME, st->mem_sig_in);
- /* last L_FILT samples for autocorrelation window */
- Copy(st->mem_sig_in, code, 6);
- HP50_12k8(new_speech + L_FRAME, L_FILT, code);
+ /* last L_FILT samples for autocorrelation window */
+ Copy(st->mem_sig_in, code, 6);
+ HP50_12k8(new_speech + L_FRAME, L_FILT, code);
- /*---------------------------------------------------------------*
- * Perform fixed preemphasis through 1 - g z^-1 *
- * Scale signal to get maximum of precision in filtering *
- *---------------------------------------------------------------*/
+ /*---------------------------------------------------------------*
+ * Perform fixed preemphasis through 1 - g z^-1 *
+ * Scale signal to get maximum of precision in filtering *
+ *---------------------------------------------------------------*/
- mu = PREEMPH_FAC >> 1; /* Q15 --> Q14 */
+ mu = PREEMPH_FAC >> 1; /* Q15 --> Q14 */
- /* get max of new preemphased samples (L_FRAME+L_FILT) */
- L_tmp = new_speech[0] << 15;
- L_tmp -= (st->mem_preemph * mu)<<1;
- L_max = L_abs(L_tmp);
+ /* get max of new preemphased samples (L_FRAME+L_FILT) */
+ L_tmp = new_speech[0] << 15;
+ L_tmp -= (st->mem_preemph * mu)<<1;
+ L_max = L_abs(L_tmp);
- for (i = 1; i < L_FRAME + L_FILT; i++)
- {
- L_tmp = new_speech[i] << 15;
- L_tmp -= (new_speech[i - 1] * mu)<<1;
- L_tmp = L_abs(L_tmp);
- if(L_tmp > L_max)
- {
- L_max = L_tmp;
- }
- }
+ for (i = 1; i < L_FRAME + L_FILT; i++)
+ {
+ L_tmp = new_speech[i] << 15;
+ L_tmp -= (new_speech[i - 1] * mu)<<1;
+ L_tmp = L_abs(L_tmp);
+ if(L_tmp > L_max)
+ {
+ L_max = L_tmp;
+ }
+ }
- /* get scaling factor for new and previous samples */
- /* limit scaling to Q_MAX to keep dynamic for ringing in low signal */
- /* limit scaling to Q_MAX also to avoid a[0]<1 in syn_filt_32 */
- tmp = extract_h(L_max);
- if (tmp == 0)
- {
- shift = Q_MAX;
- } else
- {
- shift = norm_s(tmp) - 1;
- if (shift < 0)
- {
- shift = 0;
- }
- if (shift > Q_MAX)
- {
- shift = Q_MAX;
- }
- }
- Q_new = shift;
- if (Q_new > st->Q_max[0])
- {
- Q_new = st->Q_max[0];
- }
- if (Q_new > st->Q_max[1])
- {
- Q_new = st->Q_max[1];
- }
- exp = (Q_new - st->Q_old);
- st->Q_old = Q_new;
- st->Q_max[1] = st->Q_max[0];
- st->Q_max[0] = shift;
+ /* get scaling factor for new and previous samples */
+ /* limit scaling to Q_MAX to keep dynamic for ringing in low signal */
+ /* limit scaling to Q_MAX also to avoid a[0]<1 in syn_filt_32 */
+ tmp = extract_h(L_max);
+ if (tmp == 0)
+ {
+ shift = Q_MAX;
+ } else
+ {
+ shift = norm_s(tmp) - 1;
+ if (shift < 0)
+ {
+ shift = 0;
+ }
+ if (shift > Q_MAX)
+ {
+ shift = Q_MAX;
+ }
+ }
+ Q_new = shift;
+ if (Q_new > st->Q_max[0])
+ {
+ Q_new = st->Q_max[0];
+ }
+ if (Q_new > st->Q_max[1])
+ {
+ Q_new = st->Q_max[1];
+ }
+ exp = (Q_new - st->Q_old);
+ st->Q_old = Q_new;
+ st->Q_max[1] = st->Q_max[0];
+ st->Q_max[0] = shift;
- /* preemphasis with scaling (L_FRAME+L_FILT) */
- tmp = new_speech[L_FRAME - 1];
+ /* preemphasis with scaling (L_FRAME+L_FILT) */
+ tmp = new_speech[L_FRAME - 1];
- for (i = L_FRAME + L_FILT - 1; i > 0; i--)
- {
- L_tmp = new_speech[i] << 15;
- L_tmp -= (new_speech[i - 1] * mu)<<1;
- L_tmp = (L_tmp << Q_new);
- new_speech[i] = vo_round(L_tmp);
- }
+ for (i = L_FRAME + L_FILT - 1; i > 0; i--)
+ {
+ L_tmp = new_speech[i] << 15;
+ L_tmp -= (new_speech[i - 1] * mu)<<1;
+ L_tmp = (L_tmp << Q_new);
+ new_speech[i] = vo_round(L_tmp);
+ }
- L_tmp = new_speech[0] << 15;
- L_tmp -= (st->mem_preemph * mu)<<1;
- L_tmp = (L_tmp << Q_new);
- new_speech[0] = vo_round(L_tmp);
+ L_tmp = new_speech[0] << 15;
+ L_tmp -= (st->mem_preemph * mu)<<1;
+ L_tmp = (L_tmp << Q_new);
+ new_speech[0] = vo_round(L_tmp);
- st->mem_preemph = tmp;
+ st->mem_preemph = tmp;
- /* scale previous samples and memory */
+ /* scale previous samples and memory */
- Scale_sig(old_speech, L_TOTAL - L_FRAME - L_FILT, exp);
- Scale_sig(old_exc, PIT_MAX + L_INTERPOL, exp);
- Scale_sig(st->mem_syn, M, exp);
- Scale_sig(st->mem_decim2, 3, exp);
- Scale_sig(&(st->mem_wsp), 1, exp);
- Scale_sig(&(st->mem_w0), 1, exp);
+ Scale_sig(old_speech, L_TOTAL - L_FRAME - L_FILT, exp);
+ Scale_sig(old_exc, PIT_MAX + L_INTERPOL, exp);
+ Scale_sig(st->mem_syn, M, exp);
+ Scale_sig(st->mem_decim2, 3, exp);
+ Scale_sig(&(st->mem_wsp), 1, exp);
+ Scale_sig(&(st->mem_w0), 1, exp);
- /*------------------------------------------------------------------------*
- * Call VAD *
- * Preemphesis scale down signal in low frequency and keep dynamic in HF.*
- * Vad work slightly in futur (new_speech = speech + L_NEXT - L_FILT). *
- *------------------------------------------------------------------------*/
- Copy(new_speech, buf, L_FRAME);
+ /*------------------------------------------------------------------------*
+ * Call VAD *
+ * Preemphesis scale down signal in low frequency and keep dynamic in HF.*
+ * Vad work slightly in futur (new_speech = speech + L_NEXT - L_FILT). *
+ *------------------------------------------------------------------------*/
+ Copy(new_speech, buf, L_FRAME);
#ifdef ASM_OPT /* asm optimization branch */
- Scale_sig_opt(buf, L_FRAME, 1 - Q_new);
+ Scale_sig_opt(buf, L_FRAME, 1 - Q_new);
#else
- Scale_sig(buf, L_FRAME, 1 - Q_new);
+ Scale_sig(buf, L_FRAME, 1 - Q_new);
#endif
- vad_flag = wb_vad(st->vadSt, buf); /* Voice Activity Detection */
- if (vad_flag == 0)
- {
- st->vad_hist = (st->vad_hist + 1);
- } else
- {
- st->vad_hist = 0;
- }
+ vad_flag = wb_vad(st->vadSt, buf); /* Voice Activity Detection */
+ if (vad_flag == 0)
+ {
+ st->vad_hist = (st->vad_hist + 1);
+ } else
+ {
+ st->vad_hist = 0;
+ }
- /* DTX processing */
- if (allow_dtx != 0)
- {
- /* Note that mode may change here */
- tx_dtx_handler(st->dtx_encSt, vad_flag, mode);
- *ser_size = nb_of_bits[*mode];
- }
+ /* DTX processing */
+ if (allow_dtx != 0)
+ {
+ /* Note that mode may change here */
+ tx_dtx_handler(st->dtx_encSt, vad_flag, mode);
+ *ser_size = nb_of_bits[*mode];
+ }
- if(*mode != MRDTX)
- {
- Parm_serial(vad_flag, 1, &prms);
- }
- /*------------------------------------------------------------------------*
- * Perform LPC analysis *
- * ~~~~~~~~~~~~~~~~~~~~ *
- * - autocorrelation + lag windowing *
- * - Levinson-durbin algorithm to find a[] *
- * - convert a[] to isp[] *
- * - convert isp[] to isf[] for quantization *
- * - quantize and code the isf[] *
- * - convert isf[] to isp[] for interpolation *
- * - find the interpolated ISPs and convert to a[] for the 4 subframes *
- *------------------------------------------------------------------------*/
+ if(*mode != MRDTX)
+ {
+ Parm_serial(vad_flag, 1, &prms);
+ }
+ /*------------------------------------------------------------------------*
+ * Perform LPC analysis *
+ * ~~~~~~~~~~~~~~~~~~~~ *
+ * - autocorrelation + lag windowing *
+ * - Levinson-durbin algorithm to find a[] *
+ * - convert a[] to isp[] *
+ * - convert isp[] to isf[] for quantization *
+ * - quantize and code the isf[] *
+ * - convert isf[] to isp[] for interpolation *
+ * - find the interpolated ISPs and convert to a[] for the 4 subframes *
+ *------------------------------------------------------------------------*/
- /* LP analysis centered at 4nd subframe */
- Autocorr(p_window, M, r_h, r_l); /* Autocorrelations */
- Lag_window(r_h, r_l); /* Lag windowing */
- Levinson(r_h, r_l, A, rc, st->mem_levinson); /* Levinson Durbin */
- Az_isp(A, ispnew, st->ispold); /* From A(z) to ISP */
+ /* LP analysis centered at 4nd subframe */
+ Autocorr(p_window, M, r_h, r_l); /* Autocorrelations */
+ Lag_window(r_h, r_l); /* Lag windowing */
+ Levinson(r_h, r_l, A, rc, st->mem_levinson); /* Levinson Durbin */
+ Az_isp(A, ispnew, st->ispold); /* From A(z) to ISP */
- /* Find the interpolated ISPs and convert to a[] for all subframes */
- Int_isp(st->ispold, ispnew, interpol_frac, A);
+ /* Find the interpolated ISPs and convert to a[] for all subframes */
+ Int_isp(st->ispold, ispnew, interpol_frac, A);
- /* update ispold[] for the next frame */
- Copy(ispnew, st->ispold, M);
+ /* update ispold[] for the next frame */
+ Copy(ispnew, st->ispold, M);
- /* Convert ISPs to frequency domain 0..6400 */
- Isp_isf(ispnew, isf, M);
+ /* Convert ISPs to frequency domain 0..6400 */
+ Isp_isf(ispnew, isf, M);
- /* check resonance for pitch clipping algorithm */
- Gp_clip_test_isf(isf, st->gp_clip);
+ /* check resonance for pitch clipping algorithm */
+ Gp_clip_test_isf(isf, st->gp_clip);
- /*----------------------------------------------------------------------*
- * Perform PITCH_OL analysis *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~ *
- * - Find the residual res[] for the whole speech frame *
- * - Find the weighted input speech wsp[] for the whole speech frame *
- * - scale wsp[] to avoid overflow in pitch estimation *
- * - Find open loop pitch lag for whole speech frame *
- *----------------------------------------------------------------------*/
- p_A = A;
- for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
- {
- /* Weighting of LPC coefficients */
- Weight_a(p_A, Ap, GAMMA1, M);
+ /*----------------------------------------------------------------------*
+ * Perform PITCH_OL analysis *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~ *
+ * - Find the residual res[] for the whole speech frame *
+ * - Find the weighted input speech wsp[] for the whole speech frame *
+ * - scale wsp[] to avoid overflow in pitch estimation *
+ * - Find open loop pitch lag for whole speech frame *
+ *----------------------------------------------------------------------*/
+ p_A = A;
+ for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
+ {
+ /* Weighting of LPC coefficients */
+ Weight_a(p_A, Ap, GAMMA1, M);
#ifdef ASM_OPT /* asm optimization branch */
- Residu_opt(Ap, &speech[i_subfr], &wsp[i_subfr], L_SUBFR);
+ Residu_opt(Ap, &speech[i_subfr], &wsp[i_subfr], L_SUBFR);
#else
- Residu(Ap, &speech[i_subfr], &wsp[i_subfr], L_SUBFR);
+ Residu(Ap, &speech[i_subfr], &wsp[i_subfr], L_SUBFR);
#endif
- p_A += (M + 1);
- }
+ p_A += (M + 1);
+ }
- Deemph2(wsp, TILT_FAC, L_FRAME, &(st->mem_wsp));
+ Deemph2(wsp, TILT_FAC, L_FRAME, &(st->mem_wsp));
- /* find maximum value on wsp[] for 12 bits scaling */
- max = 0;
- for (i = 0; i < L_FRAME; i++)
- {
- tmp = abs_s(wsp[i]);
- if(tmp > max)
- {
- max = tmp;
- }
- }
- tmp = st->old_wsp_max;
- if(max > tmp)
- {
- tmp = max; /* tmp = max(wsp_max, old_wsp_max) */
- }
- st->old_wsp_max = max;
+ /* find maximum value on wsp[] for 12 bits scaling */
+ max = 0;
+ for (i = 0; i < L_FRAME; i++)
+ {
+ tmp = abs_s(wsp[i]);
+ if(tmp > max)
+ {
+ max = tmp;
+ }
+ }
+ tmp = st->old_wsp_max;
+ if(max > tmp)
+ {
+ tmp = max; /* tmp = max(wsp_max, old_wsp_max) */
+ }
+ st->old_wsp_max = max;
- shift = norm_s(tmp) - 3;
- if (shift > 0)
- {
- shift = 0; /* shift = 0..-3 */
- }
- /* decimation of wsp[] to search pitch in LF and to reduce complexity */
- LP_Decim2(wsp, L_FRAME, st->mem_decim2);
+ shift = norm_s(tmp) - 3;
+ if (shift > 0)
+ {
+ shift = 0; /* shift = 0..-3 */
+ }
+ /* decimation of wsp[] to search pitch in LF and to reduce complexity */
+ LP_Decim2(wsp, L_FRAME, st->mem_decim2);
- /* scale wsp[] in 12 bits to avoid overflow */
+ /* scale wsp[] in 12 bits to avoid overflow */
#ifdef ASM_OPT /* asm optimization branch */
- Scale_sig_opt(wsp, L_FRAME / OPL_DECIM, shift);
+ Scale_sig_opt(wsp, L_FRAME / OPL_DECIM, shift);
#else
- Scale_sig(wsp, L_FRAME / OPL_DECIM, shift);
+ Scale_sig(wsp, L_FRAME / OPL_DECIM, shift);
#endif
- /* scale old_wsp (warning: exp must be Q_new-Q_old) */
- exp = exp + (shift - st->old_wsp_shift);
- st->old_wsp_shift = shift;
+ /* scale old_wsp (warning: exp must be Q_new-Q_old) */
+ exp = exp + (shift - st->old_wsp_shift);
+ st->old_wsp_shift = shift;
- Scale_sig(old_wsp, PIT_MAX / OPL_DECIM, exp);
- Scale_sig(st->old_hp_wsp, PIT_MAX / OPL_DECIM, exp);
+ Scale_sig(old_wsp, PIT_MAX / OPL_DECIM, exp);
+ Scale_sig(st->old_hp_wsp, PIT_MAX / OPL_DECIM, exp);
- scale_mem_Hp_wsp(st->hp_wsp_mem, exp);
+ scale_mem_Hp_wsp(st->hp_wsp_mem, exp);
- /* Find open loop pitch lag for whole speech frame */
+ /* Find open loop pitch lag for whole speech frame */
- if(*ser_size == NBBITS_7k)
- {
- /* Find open loop pitch lag for whole speech frame */
- T_op = Pitch_med_ol(wsp, st, L_FRAME / OPL_DECIM);
- } else
- {
- /* Find open loop pitch lag for first 1/2 frame */
- T_op = Pitch_med_ol(wsp, st, (L_FRAME/2) / OPL_DECIM);
- }
+ if(*ser_size == NBBITS_7k)
+ {
+ /* Find open loop pitch lag for whole speech frame */
+ T_op = Pitch_med_ol(wsp, st, L_FRAME / OPL_DECIM);
+ } else
+ {
+ /* Find open loop pitch lag for first 1/2 frame */
+ T_op = Pitch_med_ol(wsp, st, (L_FRAME/2) / OPL_DECIM);
+ }
- if(st->ol_gain > 19661) /* 0.6 in Q15 */
- {
- st->old_T0_med = Med_olag(T_op, st->old_ol_lag);
- st->ada_w = 32767;
- } else
- {
- st->ada_w = vo_mult(st->ada_w, 29491);
- }
+ if(st->ol_gain > 19661) /* 0.6 in Q15 */
+ {
+ st->old_T0_med = Med_olag(T_op, st->old_ol_lag);
+ st->ada_w = 32767;
+ } else
+ {
+ st->ada_w = vo_mult(st->ada_w, 29491);
+ }
- if(st->ada_w < 26214)
- st->ol_wght_flg = 0;
- else
- st->ol_wght_flg = 1;
+ if(st->ada_w < 26214)
+ st->ol_wght_flg = 0;
+ else
+ st->ol_wght_flg = 1;
- wb_vad_tone_detection(st->vadSt, st->ol_gain);
- T_op *= OPL_DECIM;
+ wb_vad_tone_detection(st->vadSt, st->ol_gain);
+ T_op *= OPL_DECIM;
- if(*ser_size != NBBITS_7k)
- {
- /* Find open loop pitch lag for second 1/2 frame */
- T_op2 = Pitch_med_ol(wsp + ((L_FRAME / 2) / OPL_DECIM), st, (L_FRAME/2) / OPL_DECIM);
+ if(*ser_size != NBBITS_7k)
+ {
+ /* Find open loop pitch lag for second 1/2 frame */
+ T_op2 = Pitch_med_ol(wsp + ((L_FRAME / 2) / OPL_DECIM), st, (L_FRAME/2) / OPL_DECIM);
- if(st->ol_gain > 19661) /* 0.6 in Q15 */
- {
- st->old_T0_med = Med_olag(T_op2, st->old_ol_lag);
- st->ada_w = 32767;
- } else
- {
- st->ada_w = mult(st->ada_w, 29491);
- }
+ if(st->ol_gain > 19661) /* 0.6 in Q15 */
+ {
+ st->old_T0_med = Med_olag(T_op2, st->old_ol_lag);
+ st->ada_w = 32767;
+ } else
+ {
+ st->ada_w = mult(st->ada_w, 29491);
+ }
- if(st->ada_w < 26214)
- st->ol_wght_flg = 0;
- else
- st->ol_wght_flg = 1;
+ if(st->ada_w < 26214)
+ st->ol_wght_flg = 0;
+ else
+ st->ol_wght_flg = 1;
- wb_vad_tone_detection(st->vadSt, st->ol_gain);
+ wb_vad_tone_detection(st->vadSt, st->ol_gain);
- T_op2 *= OPL_DECIM;
+ T_op2 *= OPL_DECIM;
- } else
- {
- T_op2 = T_op;
- }
- /*----------------------------------------------------------------------*
- * DTX-CNG *
- *----------------------------------------------------------------------*/
- if(*mode == MRDTX) /* CNG mode */
- {
- /* Buffer isf's and energy */
+ } else
+ {
+ T_op2 = T_op;
+ }
+ /*----------------------------------------------------------------------*
+ * DTX-CNG *
+ *----------------------------------------------------------------------*/
+ if(*mode == MRDTX) /* CNG mode */
+ {
+ /* Buffer isf's and energy */
#ifdef ASM_OPT /* asm optimization branch */
- Residu_opt(&A[3 * (M + 1)], speech, exc, L_FRAME);
+ Residu_opt(&A[3 * (M + 1)], speech, exc, L_FRAME);
#else
- Residu(&A[3 * (M + 1)], speech, exc, L_FRAME);
+ Residu(&A[3 * (M + 1)], speech, exc, L_FRAME);
#endif
- for (i = 0; i < L_FRAME; i++)
- {
- exc2[i] = shr(exc[i], Q_new);
- }
+ for (i = 0; i < L_FRAME; i++)
+ {
+ exc2[i] = shr(exc[i], Q_new);
+ }
- L_tmp = 0;
- for (i = 0; i < L_FRAME; i++)
- L_tmp += (exc2[i] * exc2[i])<<1;
+ L_tmp = 0;
+ for (i = 0; i < L_FRAME; i++)
+ L_tmp += (exc2[i] * exc2[i])<<1;
- L_tmp >>= 1;
+ L_tmp >>= 1;
- dtx_buffer(st->dtx_encSt, isf, L_tmp, codec_mode);
+ dtx_buffer(st->dtx_encSt, isf, L_tmp, codec_mode);
- /* Quantize and code the ISFs */
- dtx_enc(st->dtx_encSt, isf, exc2, &prms);
+ /* Quantize and code the ISFs */
+ dtx_enc(st->dtx_encSt, isf, exc2, &prms);
- /* Convert ISFs to the cosine domain */
- Isf_isp(isf, ispnew_q, M);
- Isp_Az(ispnew_q, Aq, M, 0);
+ /* Convert ISFs to the cosine domain */
+ Isf_isp(isf, ispnew_q, M);
+ Isp_Az(ispnew_q, Aq, M, 0);
- for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
- {
- corr_gain = synthesis(Aq, &exc2[i_subfr], 0, &speech16k[i_subfr * 5 / 4], st);
- }
- Copy(isf, st->isfold, M);
+ for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
+ {
+ corr_gain = synthesis(Aq, &exc2[i_subfr], 0, &speech16k[i_subfr * 5 / 4], st);
+ }
+ Copy(isf, st->isfold, M);
- /* reset speech coder memories */
- Reset_encoder(st, 0);
+ /* reset speech coder memories */
+ Reset_encoder(st, 0);
- /*--------------------------------------------------*
- * Update signal for next frame. *
- * -> save past of speech[] and wsp[]. *
- *--------------------------------------------------*/
+ /*--------------------------------------------------*
+ * Update signal for next frame. *
+ * -> save past of speech[] and wsp[]. *
+ *--------------------------------------------------*/
- Copy(&old_speech[L_FRAME], st->old_speech, L_TOTAL - L_FRAME);
- Copy(&old_wsp[L_FRAME / OPL_DECIM], st->old_wsp, PIT_MAX / OPL_DECIM);
+ Copy(&old_speech[L_FRAME], st->old_speech, L_TOTAL - L_FRAME);
+ Copy(&old_wsp[L_FRAME / OPL_DECIM], st->old_wsp, PIT_MAX / OPL_DECIM);
- return;
- }
- /*----------------------------------------------------------------------*
- * ACELP *
- *----------------------------------------------------------------------*/
+ return;
+ }
+ /*----------------------------------------------------------------------*
+ * ACELP *
+ *----------------------------------------------------------------------*/
- /* Quantize and code the ISFs */
+ /* Quantize and code the ISFs */
- if (*ser_size <= NBBITS_7k)
- {
- Qpisf_2s_36b(isf, isf, st->past_isfq, indice, 4);
+ if (*ser_size <= NBBITS_7k)
+ {
+ Qpisf_2s_36b(isf, isf, st->past_isfq, indice, 4);
- Parm_serial(indice[0], 8, &prms);
- Parm_serial(indice[1], 8, &prms);
- Parm_serial(indice[2], 7, &prms);
- Parm_serial(indice[3], 7, &prms);
- Parm_serial(indice[4], 6, &prms);
- } else
- {
- Qpisf_2s_46b(isf, isf, st->past_isfq, indice, 4);
+ Parm_serial(indice[0], 8, &prms);
+ Parm_serial(indice[1], 8, &prms);
+ Parm_serial(indice[2], 7, &prms);
+ Parm_serial(indice[3], 7, &prms);
+ Parm_serial(indice[4], 6, &prms);
+ } else
+ {
+ Qpisf_2s_46b(isf, isf, st->past_isfq, indice, 4);
- Parm_serial(indice[0], 8, &prms);
- Parm_serial(indice[1], 8, &prms);
- Parm_serial(indice[2], 6, &prms);
- Parm_serial(indice[3], 7, &prms);
- Parm_serial(indice[4], 7, &prms);
- Parm_serial(indice[5], 5, &prms);
- Parm_serial(indice[6], 5, &prms);
- }
+ Parm_serial(indice[0], 8, &prms);
+ Parm_serial(indice[1], 8, &prms);
+ Parm_serial(indice[2], 6, &prms);
+ Parm_serial(indice[3], 7, &prms);
+ Parm_serial(indice[4], 7, &prms);
+ Parm_serial(indice[5], 5, &prms);
+ Parm_serial(indice[6], 5, &prms);
+ }
- /* Check stability on isf : distance between old isf and current isf */
+ /* Check stability on isf : distance between old isf and current isf */
- L_tmp = 0;
- for (i = 0; i < M - 1; i++)
- {
- tmp = vo_sub(isf[i], st->isfold[i]);
- L_tmp += (tmp * tmp)<<1;
- }
+ L_tmp = 0;
+ for (i = 0; i < M - 1; i++)
+ {
+ tmp = vo_sub(isf[i], st->isfold[i]);
+ L_tmp += (tmp * tmp)<<1;
+ }
- tmp = extract_h(L_shl2(L_tmp, 8));
+ tmp = extract_h(L_shl2(L_tmp, 8));
- tmp = vo_mult(tmp, 26214); /* tmp = L_tmp*0.8/256 */
- tmp = vo_sub(20480, tmp); /* 1.25 - tmp (in Q14) */
+ tmp = vo_mult(tmp, 26214); /* tmp = L_tmp*0.8/256 */
+ tmp = vo_sub(20480, tmp); /* 1.25 - tmp (in Q14) */
- stab_fac = shl(tmp, 1);
+ stab_fac = shl(tmp, 1);
- if (stab_fac < 0)
- {
- stab_fac = 0;
- }
- Copy(isf, st->isfold, M);
+ if (stab_fac < 0)
+ {
+ stab_fac = 0;
+ }
+ Copy(isf, st->isfold, M);
- /* Convert ISFs to the cosine domain */
- Isf_isp(isf, ispnew_q, M);
+ /* Convert ISFs to the cosine domain */
+ Isf_isp(isf, ispnew_q, M);
- if (st->first_frame != 0)
- {
- st->first_frame = 0;
- Copy(ispnew_q, st->ispold_q, M);
- }
- /* Find the interpolated ISPs and convert to a[] for all subframes */
+ if (st->first_frame != 0)
+ {
+ st->first_frame = 0;
+ Copy(ispnew_q, st->ispold_q, M);
+ }
+ /* Find the interpolated ISPs and convert to a[] for all subframes */
- Int_isp(st->ispold_q, ispnew_q, interpol_frac, Aq);
+ Int_isp(st->ispold_q, ispnew_q, interpol_frac, Aq);
- /* update ispold[] for the next frame */
- Copy(ispnew_q, st->ispold_q, M);
+ /* update ispold[] for the next frame */
+ Copy(ispnew_q, st->ispold_q, M);
- p_Aq = Aq;
- for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
- {
+ p_Aq = Aq;
+ for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
+ {
#ifdef ASM_OPT /* asm optimization branch */
- Residu_opt(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
+ Residu_opt(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
#else
- Residu(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
+ Residu(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
#endif
- p_Aq += (M + 1);
- }
+ p_Aq += (M + 1);
+ }
- /* Buffer isf's and energy for dtx on non-speech frame */
- if (vad_flag == 0)
- {
- for (i = 0; i < L_FRAME; i++)
- {
- exc2[i] = exc[i] >> Q_new;
- }
- L_tmp = 0;
- for (i = 0; i < L_FRAME; i++)
- L_tmp += (exc2[i] * exc2[i])<<1;
- L_tmp >>= 1;
+ /* Buffer isf's and energy for dtx on non-speech frame */
+ if (vad_flag == 0)
+ {
+ for (i = 0; i < L_FRAME; i++)
+ {
+ exc2[i] = exc[i] >> Q_new;
+ }
+ L_tmp = 0;
+ for (i = 0; i < L_FRAME; i++) {
+ Word32 tmp = L_mult(exc2[i], exc2[i]); // (exc2[i] * exc2[i])<<1;
+ L_tmp = L_add(L_tmp, tmp);
+ }
+ L_tmp >>= 1;
- dtx_buffer(st->dtx_encSt, isf, L_tmp, codec_mode);
- }
- /* range for closed loop pitch search in 1st subframe */
+ dtx_buffer(st->dtx_encSt, isf, L_tmp, codec_mode);
+ }
+ /* range for closed loop pitch search in 1st subframe */
- T0_min = T_op - 8;
- if (T0_min < PIT_MIN)
- {
- T0_min = PIT_MIN;
- }
- T0_max = (T0_min + 15);
+ T0_min = T_op - 8;
+ if (T0_min < PIT_MIN)
+ {
+ T0_min = PIT_MIN;
+ }
+ T0_max = (T0_min + 15);
- if(T0_max > PIT_MAX)
- {
- T0_max = PIT_MAX;
- T0_min = T0_max - 15;
- }
- /*------------------------------------------------------------------------*
- * Loop for every subframe in the analysis frame *
- *------------------------------------------------------------------------*
- * To find the pitch and innovation parameters. The subframe size is *
- * L_SUBFR and the loop is repeated L_FRAME/L_SUBFR times. *
- * - compute the target signal for pitch search *
- * - compute impulse response of weighted synthesis filter (h1[]) *
- * - find the closed-loop pitch parameters *
- * - encode the pitch dealy *
- * - find 2 lt prediction (with / without LP filter for lt pred) *
- * - find 2 pitch gains and choose the best lt prediction. *
- * - find target vector for codebook search *
- * - update the impulse response h1[] for codebook search *
- * - correlation between target vector and impulse response *
- * - codebook search and encoding *
- * - VQ of pitch and codebook gains *
- * - find voicing factor and tilt of code for next subframe. *
- * - update states of weighting filter *
- * - find excitation and synthesis speech *
- *------------------------------------------------------------------------*/
- p_A = A;
- p_Aq = Aq;
- for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
- {
- pit_flag = i_subfr;
- if ((i_subfr == 2 * L_SUBFR) && (*ser_size > NBBITS_7k))
- {
- pit_flag = 0;
- /* range for closed loop pitch search in 3rd subframe */
- T0_min = (T_op2 - 8);
+ if(T0_max > PIT_MAX)
+ {
+ T0_max = PIT_MAX;
+ T0_min = T0_max - 15;
+ }
+ /*------------------------------------------------------------------------*
+ * Loop for every subframe in the analysis frame *
+ *------------------------------------------------------------------------*
+ * To find the pitch and innovation parameters. The subframe size is *
+ * L_SUBFR and the loop is repeated L_FRAME/L_SUBFR times. *
+ * - compute the target signal for pitch search *
+ * - compute impulse response of weighted synthesis filter (h1[]) *
+ * - find the closed-loop pitch parameters *
+ * - encode the pitch dealy *
+ * - find 2 lt prediction (with / without LP filter for lt pred) *
+ * - find 2 pitch gains and choose the best lt prediction. *
+ * - find target vector for codebook search *
+ * - update the impulse response h1[] for codebook search *
+ * - correlation between target vector and impulse response *
+ * - codebook search and encoding *
+ * - VQ of pitch and codebook gains *
+ * - find voicing factor and tilt of code for next subframe. *
+ * - update states of weighting filter *
+ * - find excitation and synthesis speech *
+ *------------------------------------------------------------------------*/
+ p_A = A;
+ p_Aq = Aq;
+ for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
+ {
+ pit_flag = i_subfr;
+ if ((i_subfr == 2 * L_SUBFR) && (*ser_size > NBBITS_7k))
+ {
+ pit_flag = 0;
+ /* range for closed loop pitch search in 3rd subframe */
+ T0_min = (T_op2 - 8);
- if (T0_min < PIT_MIN)
- {
- T0_min = PIT_MIN;
- }
- T0_max = (T0_min + 15);
- if (T0_max > PIT_MAX)
- {
- T0_max = PIT_MAX;
- T0_min = (T0_max - 15);
- }
- }
- /*-----------------------------------------------------------------------*
- * *
- * Find the target vector for pitch search: *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ *
- * *
- * |------| res[n] *
- * speech[n]---| A(z) |-------- *
- * |------| | |--------| error[n] |------| *
- * zero -- (-)--| 1/A(z) |-----------| W(z) |-- target *
- * exc |--------| |------| *
- * *
- * Instead of subtracting the zero-input response of filters from *
- * the weighted input speech, the above configuration is used to *
- * compute the target vector. *
- * *
- *-----------------------------------------------------------------------*/
+ if (T0_min < PIT_MIN)
+ {
+ T0_min = PIT_MIN;
+ }
+ T0_max = (T0_min + 15);
+ if (T0_max > PIT_MAX)
+ {
+ T0_max = PIT_MAX;
+ T0_min = (T0_max - 15);
+ }
+ }
+ /*-----------------------------------------------------------------------*
+ * *
+ * Find the target vector for pitch search: *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ *
+ * *
+ * |------| res[n] *
+ * speech[n]---| A(z) |-------- *
+ * |------| | |--------| error[n] |------| *
+ * zero -- (-)--| 1/A(z) |-----------| W(z) |-- target *
+ * exc |--------| |------| *
+ * *
+ * Instead of subtracting the zero-input response of filters from *
+ * the weighted input speech, the above configuration is used to *
+ * compute the target vector. *
+ * *
+ *-----------------------------------------------------------------------*/
- for (i = 0; i < M; i++)
- {
- error[i] = vo_sub(speech[i + i_subfr - M], st->mem_syn[i]);
- }
+ for (i = 0; i < M; i++)
+ {
+ error[i] = vo_sub(speech[i + i_subfr - M], st->mem_syn[i]);
+ }
#ifdef ASM_OPT /* asm optimization branch */
- Residu_opt(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
+ Residu_opt(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
#else
- Residu(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
+ Residu(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
#endif
- Syn_filt(p_Aq, &exc[i_subfr], error + M, L_SUBFR, error, 0);
- Weight_a(p_A, Ap, GAMMA1, M);
+ Syn_filt(p_Aq, &exc[i_subfr], error + M, L_SUBFR, error, 0);
+ Weight_a(p_A, Ap, GAMMA1, M);
#ifdef ASM_OPT /* asm optimization branch */
- Residu_opt(Ap, error + M, xn, L_SUBFR);
+ Residu_opt(Ap, error + M, xn, L_SUBFR);
#else
- Residu(Ap, error + M, xn, L_SUBFR);
+ Residu(Ap, error + M, xn, L_SUBFR);
#endif
- Deemph2(xn, TILT_FAC, L_SUBFR, &(st->mem_w0));
+ Deemph2(xn, TILT_FAC, L_SUBFR, &(st->mem_w0));
- /*----------------------------------------------------------------------*
- * Find approx. target in residual domain "cn[]" for inovation search. *
- *----------------------------------------------------------------------*/
- /* first half: xn[] --> cn[] */
- Set_zero(code, M);
- Copy(xn, code + M, L_SUBFR / 2);
- tmp = 0;
- Preemph2(code + M, TILT_FAC, L_SUBFR / 2, &tmp);
- Weight_a(p_A, Ap, GAMMA1, M);
- Syn_filt(Ap,code + M, code + M, L_SUBFR / 2, code, 0);
+ /*----------------------------------------------------------------------*
+ * Find approx. target in residual domain "cn[]" for inovation search. *
+ *----------------------------------------------------------------------*/
+ /* first half: xn[] --> cn[] */
+ Set_zero(code, M);
+ Copy(xn, code + M, L_SUBFR / 2);
+ tmp = 0;
+ Preemph2(code + M, TILT_FAC, L_SUBFR / 2, &tmp);
+ Weight_a(p_A, Ap, GAMMA1, M);
+ Syn_filt(Ap,code + M, code + M, L_SUBFR / 2, code, 0);
#ifdef ASM_OPT /* asm optimization branch */
- Residu_opt(p_Aq,code + M, cn, L_SUBFR / 2);
+ Residu_opt(p_Aq,code + M, cn, L_SUBFR / 2);
#else
- Residu(p_Aq,code + M, cn, L_SUBFR / 2);
+ Residu(p_Aq,code + M, cn, L_SUBFR / 2);
#endif
- /* second half: res[] --> cn[] (approximated and faster) */
- Copy(&exc[i_subfr + (L_SUBFR / 2)], cn + (L_SUBFR / 2), L_SUBFR / 2);
+ /* second half: res[] --> cn[] (approximated and faster) */
+ Copy(&exc[i_subfr + (L_SUBFR / 2)], cn + (L_SUBFR / 2), L_SUBFR / 2);
- /*---------------------------------------------------------------*
- * Compute impulse response, h1[], of weighted synthesis filter *
- *---------------------------------------------------------------*/
+ /*---------------------------------------------------------------*
+ * Compute impulse response, h1[], of weighted synthesis filter *
+ *---------------------------------------------------------------*/
- Set_zero(error, M + L_SUBFR);
- Weight_a(p_A, error + M, GAMMA1, M);
+ Set_zero(error, M + L_SUBFR);
+ Weight_a(p_A, error + M, GAMMA1, M);
- vo_p0 = error+M;
- vo_p3 = h1;
- for (i = 0; i < L_SUBFR; i++)
- {
- L_tmp = *vo_p0 << 14; /* x4 (Q12 to Q14) */
- vo_p1 = p_Aq + 1;
- vo_p2 = vo_p0-1;
- for (j = 1; j <= M/4; j++)
- {
- L_tmp -= *vo_p1++ * *vo_p2--;
- L_tmp -= *vo_p1++ * *vo_p2--;
- L_tmp -= *vo_p1++ * *vo_p2--;
- L_tmp -= *vo_p1++ * *vo_p2--;
- }
- *vo_p3++ = *vo_p0++ = vo_round((L_tmp <<4));
- }
- /* deemph without division by 2 -> Q14 to Q15 */
- tmp = 0;
- Deemph2(h1, TILT_FAC, L_SUBFR, &tmp); /* h1 in Q14 */
+ vo_p0 = error+M;
+ vo_p3 = h1;
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ L_tmp = *vo_p0 << 14; /* x4 (Q12 to Q14) */
+ vo_p1 = p_Aq + 1;
+ vo_p2 = vo_p0-1;
+ for (j = 1; j <= M/4; j++)
+ {
+ L_tmp -= *vo_p1++ * *vo_p2--;
+ L_tmp -= *vo_p1++ * *vo_p2--;
+ L_tmp -= *vo_p1++ * *vo_p2--;
+ L_tmp -= *vo_p1++ * *vo_p2--;
+ }
+ *vo_p3++ = *vo_p0++ = vo_round((L_tmp <<4));
+ }
+ /* deemph without division by 2 -> Q14 to Q15 */
+ tmp = 0;
+ Deemph2(h1, TILT_FAC, L_SUBFR, &tmp); /* h1 in Q14 */
- /* h2 in Q12 for codebook search */
- Copy(h1, h2, L_SUBFR);
+ /* h2 in Q12 for codebook search */
+ Copy(h1, h2, L_SUBFR);
- /*---------------------------------------------------------------*
- * scale xn[] and h1[] to avoid overflow in dot_product12() *
- *---------------------------------------------------------------*/
+ /*---------------------------------------------------------------*
+ * scale xn[] and h1[] to avoid overflow in dot_product12() *
+ *---------------------------------------------------------------*/
#ifdef ASM_OPT /* asm optimization branch */
- Scale_sig_opt(h2, L_SUBFR, -2);
- Scale_sig_opt(xn, L_SUBFR, shift); /* scaling of xn[] to limit dynamic at 12 bits */
- Scale_sig_opt(h1, L_SUBFR, 1 + shift); /* set h1[] in Q15 with scaling for convolution */
+ Scale_sig_opt(h2, L_SUBFR, -2);
+ Scale_sig_opt(xn, L_SUBFR, shift); /* scaling of xn[] to limit dynamic at 12 bits */
+ Scale_sig_opt(h1, L_SUBFR, 1 + shift); /* set h1[] in Q15 with scaling for convolution */
#else
- Scale_sig(h2, L_SUBFR, -2);
- Scale_sig(xn, L_SUBFR, shift); /* scaling of xn[] to limit dynamic at 12 bits */
- Scale_sig(h1, L_SUBFR, 1 + shift); /* set h1[] in Q15 with scaling for convolution */
+ Scale_sig(h2, L_SUBFR, -2);
+ Scale_sig(xn, L_SUBFR, shift); /* scaling of xn[] to limit dynamic at 12 bits */
+ Scale_sig(h1, L_SUBFR, 1 + shift); /* set h1[] in Q15 with scaling for convolution */
#endif
- /*----------------------------------------------------------------------*
- * Closed-loop fractional pitch search *
- *----------------------------------------------------------------------*/
- /* find closed loop fractional pitch lag */
- if(*ser_size <= NBBITS_9k)
- {
- T0 = Pitch_fr4(&exc[i_subfr], xn, h1, T0_min, T0_max, &T0_frac,
- pit_flag, PIT_MIN, PIT_FR1_8b, L_SUBFR);
+ /*----------------------------------------------------------------------*
+ * Closed-loop fractional pitch search *
+ *----------------------------------------------------------------------*/
+ /* find closed loop fractional pitch lag */
+ if(*ser_size <= NBBITS_9k)
+ {
+ T0 = Pitch_fr4(&exc[i_subfr], xn, h1, T0_min, T0_max, &T0_frac,
+ pit_flag, PIT_MIN, PIT_FR1_8b, L_SUBFR);
- /* encode pitch lag */
- if (pit_flag == 0) /* if 1st/3rd subframe */
- {
- /*--------------------------------------------------------------*
- * The pitch range for the 1st/3rd subframe is encoded with *
- * 8 bits and is divided as follows: *
- * PIT_MIN to PIT_FR1-1 resolution 1/2 (frac = 0 or 2) *
- * PIT_FR1 to PIT_MAX resolution 1 (frac = 0) *
- *--------------------------------------------------------------*/
- if (T0 < PIT_FR1_8b)
- {
- index = ((T0 << 1) + (T0_frac >> 1) - (PIT_MIN<<1));
- } else
- {
- index = ((T0 - PIT_FR1_8b) + ((PIT_FR1_8b - PIT_MIN)*2));
- }
+ /* encode pitch lag */
+ if (pit_flag == 0) /* if 1st/3rd subframe */
+ {
+ /*--------------------------------------------------------------*
+ * The pitch range for the 1st/3rd subframe is encoded with *
+ * 8 bits and is divided as follows: *
+ * PIT_MIN to PIT_FR1-1 resolution 1/2 (frac = 0 or 2) *
+ * PIT_FR1 to PIT_MAX resolution 1 (frac = 0) *
+ *--------------------------------------------------------------*/
+ if (T0 < PIT_FR1_8b)
+ {
+ index = ((T0 << 1) + (T0_frac >> 1) - (PIT_MIN<<1));
+ } else
+ {
+ index = ((T0 - PIT_FR1_8b) + ((PIT_FR1_8b - PIT_MIN)*2));
+ }
- Parm_serial(index, 8, &prms);
+ Parm_serial(index, 8, &prms);
- /* find T0_min and T0_max for subframe 2 and 4 */
- T0_min = (T0 - 8);
- if (T0_min < PIT_MIN)
- {
- T0_min = PIT_MIN;
- }
- T0_max = T0_min + 15;
- if (T0_max > PIT_MAX)
- {
- T0_max = PIT_MAX;
- T0_min = (T0_max - 15);
- }
- } else
- { /* if subframe 2 or 4 */
- /*--------------------------------------------------------------*
- * The pitch range for subframe 2 or 4 is encoded with 5 bits: *
- * T0_min to T0_max resolution 1/2 (frac = 0 or 2) *
- *--------------------------------------------------------------*/
- i = (T0 - T0_min);
- index = (i << 1) + (T0_frac >> 1);
+ /* find T0_min and T0_max for subframe 2 and 4 */
+ T0_min = (T0 - 8);
+ if (T0_min < PIT_MIN)
+ {
+ T0_min = PIT_MIN;
+ }
+ T0_max = T0_min + 15;
+ if (T0_max > PIT_MAX)
+ {
+ T0_max = PIT_MAX;
+ T0_min = (T0_max - 15);
+ }
+ } else
+ { /* if subframe 2 or 4 */
+ /*--------------------------------------------------------------*
+ * The pitch range for subframe 2 or 4 is encoded with 5 bits: *
+ * T0_min to T0_max resolution 1/2 (frac = 0 or 2) *
+ *--------------------------------------------------------------*/
+ i = (T0 - T0_min);
+ index = (i << 1) + (T0_frac >> 1);
- Parm_serial(index, 5, &prms);
- }
- } else
- {
- T0 = Pitch_fr4(&exc[i_subfr], xn, h1, T0_min, T0_max, &T0_frac,
- pit_flag, PIT_FR2, PIT_FR1_9b, L_SUBFR);
+ Parm_serial(index, 5, &prms);
+ }
+ } else
+ {
+ T0 = Pitch_fr4(&exc[i_subfr], xn, h1, T0_min, T0_max, &T0_frac,
+ pit_flag, PIT_FR2, PIT_FR1_9b, L_SUBFR);
- /* encode pitch lag */
- if (pit_flag == 0) /* if 1st/3rd subframe */
- {
- /*--------------------------------------------------------------*
- * The pitch range for the 1st/3rd subframe is encoded with *
- * 9 bits and is divided as follows: *
- * PIT_MIN to PIT_FR2-1 resolution 1/4 (frac = 0,1,2 or 3) *
- * PIT_FR2 to PIT_FR1-1 resolution 1/2 (frac = 0 or 1) *
- * PIT_FR1 to PIT_MAX resolution 1 (frac = 0) *
- *--------------------------------------------------------------*/
+ /* encode pitch lag */
+ if (pit_flag == 0) /* if 1st/3rd subframe */
+ {
+ /*--------------------------------------------------------------*
+ * The pitch range for the 1st/3rd subframe is encoded with *
+ * 9 bits and is divided as follows: *
+ * PIT_MIN to PIT_FR2-1 resolution 1/4 (frac = 0,1,2 or 3) *
+ * PIT_FR2 to PIT_FR1-1 resolution 1/2 (frac = 0 or 1) *
+ * PIT_FR1 to PIT_MAX resolution 1 (frac = 0) *
+ *--------------------------------------------------------------*/
- if (T0 < PIT_FR2)
- {
- index = ((T0 << 2) + T0_frac) - (PIT_MIN << 2);
- } else if(T0 < PIT_FR1_9b)
- {
- index = ((((T0 << 1) + (T0_frac >> 1)) - (PIT_FR2<<1)) + ((PIT_FR2 - PIT_MIN)<<2));
- } else
- {
- index = (((T0 - PIT_FR1_9b) + ((PIT_FR2 - PIT_MIN)<<2)) + ((PIT_FR1_9b - PIT_FR2)<<1));
- }
+ if (T0 < PIT_FR2)
+ {
+ index = ((T0 << 2) + T0_frac) - (PIT_MIN << 2);
+ } else if(T0 < PIT_FR1_9b)
+ {
+ index = ((((T0 << 1) + (T0_frac >> 1)) - (PIT_FR2<<1)) + ((PIT_FR2 - PIT_MIN)<<2));
+ } else
+ {
+ index = (((T0 - PIT_FR1_9b) + ((PIT_FR2 - PIT_MIN)<<2)) + ((PIT_FR1_9b - PIT_FR2)<<1));
+ }
- Parm_serial(index, 9, &prms);
+ Parm_serial(index, 9, &prms);
- /* find T0_min and T0_max for subframe 2 and 4 */
+ /* find T0_min and T0_max for subframe 2 and 4 */
- T0_min = (T0 - 8);
- if (T0_min < PIT_MIN)
- {
- T0_min = PIT_MIN;
- }
- T0_max = T0_min + 15;
+ T0_min = (T0 - 8);
+ if (T0_min < PIT_MIN)
+ {
+ T0_min = PIT_MIN;
+ }
+ T0_max = T0_min + 15;
- if (T0_max > PIT_MAX)
- {
- T0_max = PIT_MAX;
- T0_min = (T0_max - 15);
- }
- } else
- { /* if subframe 2 or 4 */
- /*--------------------------------------------------------------*
- * The pitch range for subframe 2 or 4 is encoded with 6 bits: *
- * T0_min to T0_max resolution 1/4 (frac = 0,1,2 or 3) *
- *--------------------------------------------------------------*/
- i = (T0 - T0_min);
- index = (i << 2) + T0_frac;
- Parm_serial(index, 6, &prms);
- }
- }
+ if (T0_max > PIT_MAX)
+ {
+ T0_max = PIT_MAX;
+ T0_min = (T0_max - 15);
+ }
+ } else
+ { /* if subframe 2 or 4 */
+ /*--------------------------------------------------------------*
+ * The pitch range for subframe 2 or 4 is encoded with 6 bits: *
+ * T0_min to T0_max resolution 1/4 (frac = 0,1,2 or 3) *
+ *--------------------------------------------------------------*/
+ i = (T0 - T0_min);
+ index = (i << 2) + T0_frac;
+ Parm_serial(index, 6, &prms);
+ }
+ }
- /*-----------------------------------------------------------------*
- * Gain clipping test to avoid unstable synthesis on frame erasure *
- *-----------------------------------------------------------------*/
+ /*-----------------------------------------------------------------*
+ * Gain clipping test to avoid unstable synthesis on frame erasure *
+ *-----------------------------------------------------------------*/
- clip_gain = 0;
- if((st->gp_clip[0] < 154) && (st->gp_clip[1] > 14746))
- clip_gain = 1;
+ clip_gain = 0;
+ if((st->gp_clip[0] < 154) && (st->gp_clip[1] > 14746))
+ clip_gain = 1;
- /*-----------------------------------------------------------------*
- * - find unity gain pitch excitation (adaptive codebook entry) *
- * with fractional interpolation. *
- * - find filtered pitch exc. y1[]=exc[] convolved with h1[]) *
- * - compute pitch gain1 *
- *-----------------------------------------------------------------*/
- /* find pitch exitation */
+ /*-----------------------------------------------------------------*
+ * - find unity gain pitch excitation (adaptive codebook entry) *
+ * with fractional interpolation. *
+ * - find filtered pitch exc. y1[]=exc[] convolved with h1[]) *
+ * - compute pitch gain1 *
+ *-----------------------------------------------------------------*/
+ /* find pitch exitation */
#ifdef ASM_OPT /* asm optimization branch */
- pred_lt4_asm(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1);
+ pred_lt4_asm(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1);
#else
- Pred_lt4(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1);
+ Pred_lt4(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1);
#endif
- if (*ser_size > NBBITS_9k)
- {
+ if (*ser_size > NBBITS_9k)
+ {
#ifdef ASM_OPT /* asm optimization branch */
- Convolve_asm(&exc[i_subfr], h1, y1, L_SUBFR);
+ Convolve_asm(&exc[i_subfr], h1, y1, L_SUBFR);
#else
- Convolve(&exc[i_subfr], h1, y1, L_SUBFR);
+ Convolve(&exc[i_subfr], h1, y1, L_SUBFR);
#endif
- gain1 = G_pitch(xn, y1, g_coeff, L_SUBFR);
- /* clip gain if necessary to avoid problem at decoder */
- if ((clip_gain != 0) && (gain1 > GP_CLIP))
- {
- gain1 = GP_CLIP;
- }
- /* find energy of new target xn2[] */
- Updt_tar(xn, dn, y1, gain1, L_SUBFR); /* dn used temporary */
- } else
- {
- gain1 = 0;
- }
- /*-----------------------------------------------------------------*
- * - find pitch excitation filtered by 1st order LP filter. *
- * - find filtered pitch exc. y2[]=exc[] convolved with h1[]) *
- * - compute pitch gain2 *
- *-----------------------------------------------------------------*/
- /* find pitch excitation with lp filter */
- vo_p0 = exc + i_subfr-1;
- vo_p1 = code;
- /* find pitch excitation with lp filter */
- for (i = 0; i < L_SUBFR/2; i++)
- {
- L_tmp = 5898 * *vo_p0++;
- L_tmp1 = 5898 * *vo_p0;
- L_tmp += 20972 * *vo_p0++;
- L_tmp1 += 20972 * *vo_p0++;
- L_tmp1 += 5898 * *vo_p0--;
- L_tmp += 5898 * *vo_p0;
- *vo_p1++ = (L_tmp + 0x4000)>>15;
- *vo_p1++ = (L_tmp1 + 0x4000)>>15;
- }
+ gain1 = G_pitch(xn, y1, g_coeff, L_SUBFR);
+ /* clip gain if necessary to avoid problem at decoder */
+ if ((clip_gain != 0) && (gain1 > GP_CLIP))
+ {
+ gain1 = GP_CLIP;
+ }
+ /* find energy of new target xn2[] */
+ Updt_tar(xn, dn, y1, gain1, L_SUBFR); /* dn used temporary */
+ } else
+ {
+ gain1 = 0;
+ }
+ /*-----------------------------------------------------------------*
+ * - find pitch excitation filtered by 1st order LP filter. *
+ * - find filtered pitch exc. y2[]=exc[] convolved with h1[]) *
+ * - compute pitch gain2 *
+ *-----------------------------------------------------------------*/
+ /* find pitch excitation with lp filter */
+ vo_p0 = exc + i_subfr-1;
+ vo_p1 = code;
+ /* find pitch excitation with lp filter */
+ for (i = 0; i < L_SUBFR/2; i++)
+ {
+ L_tmp = 5898 * *vo_p0++;
+ L_tmp1 = 5898 * *vo_p0;
+ L_tmp += 20972 * *vo_p0++;
+ L_tmp1 += 20972 * *vo_p0++;
+ L_tmp1 += 5898 * *vo_p0--;
+ L_tmp += 5898 * *vo_p0;
+ *vo_p1++ = (L_tmp + 0x4000)>>15;
+ *vo_p1++ = (L_tmp1 + 0x4000)>>15;
+ }
#ifdef ASM_OPT /* asm optimization branch */
- Convolve_asm(code, h1, y2, L_SUBFR);
+ Convolve_asm(code, h1, y2, L_SUBFR);
#else
- Convolve(code, h1, y2, L_SUBFR);
+ Convolve(code, h1, y2, L_SUBFR);
#endif
- gain2 = G_pitch(xn, y2, g_coeff2, L_SUBFR);
+ gain2 = G_pitch(xn, y2, g_coeff2, L_SUBFR);
- /* clip gain if necessary to avoid problem at decoder */
- if ((clip_gain != 0) && (gain2 > GP_CLIP))
- {
- gain2 = GP_CLIP;
- }
- /* find energy of new target xn2[] */
- Updt_tar(xn, xn2, y2, gain2, L_SUBFR);
- /*-----------------------------------------------------------------*
- * use the best prediction (minimise quadratic error). *
- *-----------------------------------------------------------------*/
- select = 0;
- if(*ser_size > NBBITS_9k)
- {
- L_tmp = 0L;
- vo_p0 = dn;
- vo_p1 = xn2;
- for (i = 0; i < L_SUBFR/2; i++)
- {
- L_tmp += *vo_p0 * *vo_p0;
- vo_p0++;
- L_tmp -= *vo_p1 * *vo_p1;
- vo_p1++;
- L_tmp += *vo_p0 * *vo_p0;
- vo_p0++;
- L_tmp -= *vo_p1 * *vo_p1;
- vo_p1++;
- }
+ /* clip gain if necessary to avoid problem at decoder */
+ if ((clip_gain != 0) && (gain2 > GP_CLIP))
+ {
+ gain2 = GP_CLIP;
+ }
+ /* find energy of new target xn2[] */
+ Updt_tar(xn, xn2, y2, gain2, L_SUBFR);
+ /*-----------------------------------------------------------------*
+ * use the best prediction (minimise quadratic error). *
+ *-----------------------------------------------------------------*/
+ select = 0;
+ if(*ser_size > NBBITS_9k)
+ {
+ L_tmp = 0L;
+ vo_p0 = dn;
+ vo_p1 = xn2;
+ for (i = 0; i < L_SUBFR/2; i++)
+ {
+ L_tmp += *vo_p0 * *vo_p0;
+ vo_p0++;
+ L_tmp -= *vo_p1 * *vo_p1;
+ vo_p1++;
+ L_tmp += *vo_p0 * *vo_p0;
+ vo_p0++;
+ L_tmp -= *vo_p1 * *vo_p1;
+ vo_p1++;
+ }
- if (L_tmp <= 0)
- {
- select = 1;
- }
- Parm_serial(select, 1, &prms);
- }
- if (select == 0)
- {
- /* use the lp filter for pitch excitation prediction */
- gain_pit = gain2;
- Copy(code, &exc[i_subfr], L_SUBFR);
- Copy(y2, y1, L_SUBFR);
- Copy(g_coeff2, g_coeff, 4);
- } else
- {
- /* no filter used for pitch excitation prediction */
- gain_pit = gain1;
- Copy(dn, xn2, L_SUBFR); /* target vector for codebook search */
- }
- /*-----------------------------------------------------------------*
- * - update cn[] for codebook search *
- *-----------------------------------------------------------------*/
- Updt_tar(cn, cn, &exc[i_subfr], gain_pit, L_SUBFR);
+ if (L_tmp <= 0)
+ {
+ select = 1;
+ }
+ Parm_serial(select, 1, &prms);
+ }
+ if (select == 0)
+ {
+ /* use the lp filter for pitch excitation prediction */
+ gain_pit = gain2;
+ Copy(code, &exc[i_subfr], L_SUBFR);
+ Copy(y2, y1, L_SUBFR);
+ Copy(g_coeff2, g_coeff, 4);
+ } else
+ {
+ /* no filter used for pitch excitation prediction */
+ gain_pit = gain1;
+ Copy(dn, xn2, L_SUBFR); /* target vector for codebook search */
+ }
+ /*-----------------------------------------------------------------*
+ * - update cn[] for codebook search *
+ *-----------------------------------------------------------------*/
+ Updt_tar(cn, cn, &exc[i_subfr], gain_pit, L_SUBFR);
#ifdef ASM_OPT /* asm optimization branch */
- Scale_sig_opt(cn, L_SUBFR, shift); /* scaling of cn[] to limit dynamic at 12 bits */
+ Scale_sig_opt(cn, L_SUBFR, shift); /* scaling of cn[] to limit dynamic at 12 bits */
#else
- Scale_sig(cn, L_SUBFR, shift); /* scaling of cn[] to limit dynamic at 12 bits */
+ Scale_sig(cn, L_SUBFR, shift); /* scaling of cn[] to limit dynamic at 12 bits */
#endif
- /*-----------------------------------------------------------------*
- * - include fixed-gain pitch contribution into impulse resp. h1[] *
- *-----------------------------------------------------------------*/
- tmp = 0;
- Preemph(h2, st->tilt_code, L_SUBFR, &tmp);
+ /*-----------------------------------------------------------------*
+ * - include fixed-gain pitch contribution into impulse resp. h1[] *
+ *-----------------------------------------------------------------*/
+ tmp = 0;
+ Preemph(h2, st->tilt_code, L_SUBFR, &tmp);
- if (T0_frac > 2)
- T0 = (T0 + 1);
- Pit_shrp(h2, T0, PIT_SHARP, L_SUBFR);
- /*-----------------------------------------------------------------*
- * - Correlation between target xn2[] and impulse response h1[] *
- * - Innovative codebook search *
- *-----------------------------------------------------------------*/
- cor_h_x(h2, xn2, dn);
- if (*ser_size <= NBBITS_7k)
- {
- ACELP_2t64_fx(dn, cn, h2, code, y2, indice);
+ if (T0_frac > 2)
+ T0 = (T0 + 1);
+ Pit_shrp(h2, T0, PIT_SHARP, L_SUBFR);
+ /*-----------------------------------------------------------------*
+ * - Correlation between target xn2[] and impulse response h1[] *
+ * - Innovative codebook search *
+ *-----------------------------------------------------------------*/
+ cor_h_x(h2, xn2, dn);
+ if (*ser_size <= NBBITS_7k)
+ {
+ ACELP_2t64_fx(dn, cn, h2, code, y2, indice);
- Parm_serial(indice[0], 12, &prms);
- } else if(*ser_size <= NBBITS_9k)
- {
- ACELP_4t64_fx(dn, cn, h2, code, y2, 20, *ser_size, indice);
+ Parm_serial(indice[0], 12, &prms);
+ } else if(*ser_size <= NBBITS_9k)
+ {
+ ACELP_4t64_fx(dn, cn, h2, code, y2, 20, *ser_size, indice);
- Parm_serial(indice[0], 5, &prms);
- Parm_serial(indice[1], 5, &prms);
- Parm_serial(indice[2], 5, &prms);
- Parm_serial(indice[3], 5, &prms);
- } else if(*ser_size <= NBBITS_12k)
- {
- ACELP_4t64_fx(dn, cn, h2, code, y2, 36, *ser_size, indice);
+ Parm_serial(indice[0], 5, &prms);
+ Parm_serial(indice[1], 5, &prms);
+ Parm_serial(indice[2], 5, &prms);
+ Parm_serial(indice[3], 5, &prms);
+ } else if(*ser_size <= NBBITS_12k)
+ {
+ ACELP_4t64_fx(dn, cn, h2, code, y2, 36, *ser_size, indice);
- Parm_serial(indice[0], 9, &prms);
- Parm_serial(indice[1], 9, &prms);
- Parm_serial(indice[2], 9, &prms);
- Parm_serial(indice[3], 9, &prms);
- } else if(*ser_size <= NBBITS_14k)
- {
- ACELP_4t64_fx(dn, cn, h2, code, y2, 44, *ser_size, indice);
+ Parm_serial(indice[0], 9, &prms);
+ Parm_serial(indice[1], 9, &prms);
+ Parm_serial(indice[2], 9, &prms);
+ Parm_serial(indice[3], 9, &prms);
+ } else if(*ser_size <= NBBITS_14k)
+ {
+ ACELP_4t64_fx(dn, cn, h2, code, y2, 44, *ser_size, indice);
- Parm_serial(indice[0], 13, &prms);
- Parm_serial(indice[1], 13, &prms);
- Parm_serial(indice[2], 9, &prms);
- Parm_serial(indice[3], 9, &prms);
- } else if(*ser_size <= NBBITS_16k)
- {
- ACELP_4t64_fx(dn, cn, h2, code, y2, 52, *ser_size, indice);
+ Parm_serial(indice[0], 13, &prms);
+ Parm_serial(indice[1], 13, &prms);
+ Parm_serial(indice[2], 9, &prms);
+ Parm_serial(indice[3], 9, &prms);
+ } else if(*ser_size <= NBBITS_16k)
+ {
+ ACELP_4t64_fx(dn, cn, h2, code, y2, 52, *ser_size, indice);
- Parm_serial(indice[0], 13, &prms);
- Parm_serial(indice[1], 13, &prms);
- Parm_serial(indice[2], 13, &prms);
- Parm_serial(indice[3], 13, &prms);
- } else if(*ser_size <= NBBITS_18k)
- {
- ACELP_4t64_fx(dn, cn, h2, code, y2, 64, *ser_size, indice);
+ Parm_serial(indice[0], 13, &prms);
+ Parm_serial(indice[1], 13, &prms);
+ Parm_serial(indice[2], 13, &prms);
+ Parm_serial(indice[3], 13, &prms);
+ } else if(*ser_size <= NBBITS_18k)
+ {
+ ACELP_4t64_fx(dn, cn, h2, code, y2, 64, *ser_size, indice);
- Parm_serial(indice[0], 2, &prms);
- Parm_serial(indice[1], 2, &prms);
- Parm_serial(indice[2], 2, &prms);
- Parm_serial(indice[3], 2, &prms);
- Parm_serial(indice[4], 14, &prms);
- Parm_serial(indice[5], 14, &prms);
- Parm_serial(indice[6], 14, &prms);
- Parm_serial(indice[7], 14, &prms);
- } else if(*ser_size <= NBBITS_20k)
- {
- ACELP_4t64_fx(dn, cn, h2, code, y2, 72, *ser_size, indice);
+ Parm_serial(indice[0], 2, &prms);
+ Parm_serial(indice[1], 2, &prms);
+ Parm_serial(indice[2], 2, &prms);
+ Parm_serial(indice[3], 2, &prms);
+ Parm_serial(indice[4], 14, &prms);
+ Parm_serial(indice[5], 14, &prms);
+ Parm_serial(indice[6], 14, &prms);
+ Parm_serial(indice[7], 14, &prms);
+ } else if(*ser_size <= NBBITS_20k)
+ {
+ ACELP_4t64_fx(dn, cn, h2, code, y2, 72, *ser_size, indice);
- Parm_serial(indice[0], 10, &prms);
- Parm_serial(indice[1], 10, &prms);
- Parm_serial(indice[2], 2, &prms);
- Parm_serial(indice[3], 2, &prms);
- Parm_serial(indice[4], 10, &prms);
- Parm_serial(indice[5], 10, &prms);
- Parm_serial(indice[6], 14, &prms);
- Parm_serial(indice[7], 14, &prms);
- } else
- {
- ACELP_4t64_fx(dn, cn, h2, code, y2, 88, *ser_size, indice);
+ Parm_serial(indice[0], 10, &prms);
+ Parm_serial(indice[1], 10, &prms);
+ Parm_serial(indice[2], 2, &prms);
+ Parm_serial(indice[3], 2, &prms);
+ Parm_serial(indice[4], 10, &prms);
+ Parm_serial(indice[5], 10, &prms);
+ Parm_serial(indice[6], 14, &prms);
+ Parm_serial(indice[7], 14, &prms);
+ } else
+ {
+ ACELP_4t64_fx(dn, cn, h2, code, y2, 88, *ser_size, indice);
- Parm_serial(indice[0], 11, &prms);
- Parm_serial(indice[1], 11, &prms);
- Parm_serial(indice[2], 11, &prms);
- Parm_serial(indice[3], 11, &prms);
- Parm_serial(indice[4], 11, &prms);
- Parm_serial(indice[5], 11, &prms);
- Parm_serial(indice[6], 11, &prms);
- Parm_serial(indice[7], 11, &prms);
- }
- /*-------------------------------------------------------*
- * - Add the fixed-gain pitch contribution to code[]. *
- *-------------------------------------------------------*/
- tmp = 0;
- Preemph(code, st->tilt_code, L_SUBFR, &tmp);
- Pit_shrp(code, T0, PIT_SHARP, L_SUBFR);
- /*----------------------------------------------------------*
- * - Compute the fixed codebook gain *
- * - quantize fixed codebook gain *
- *----------------------------------------------------------*/
- if(*ser_size <= NBBITS_9k)
- {
- index = Q_gain2(xn, y1, Q_new + shift, y2, code, g_coeff, L_SUBFR, 6,
- &gain_pit, &L_gain_code, clip_gain, st->qua_gain);
- Parm_serial(index, 6, &prms);
- } else
- {
- index = Q_gain2(xn, y1, Q_new + shift, y2, code, g_coeff, L_SUBFR, 7,
- &gain_pit, &L_gain_code, clip_gain, st->qua_gain);
- Parm_serial(index, 7, &prms);
- }
- /* test quantized gain of pitch for pitch clipping algorithm */
- Gp_clip_test_gain_pit(gain_pit, st->gp_clip);
+ Parm_serial(indice[0], 11, &prms);
+ Parm_serial(indice[1], 11, &prms);
+ Parm_serial(indice[2], 11, &prms);
+ Parm_serial(indice[3], 11, &prms);
+ Parm_serial(indice[4], 11, &prms);
+ Parm_serial(indice[5], 11, &prms);
+ Parm_serial(indice[6], 11, &prms);
+ Parm_serial(indice[7], 11, &prms);
+ }
+ /*-------------------------------------------------------*
+ * - Add the fixed-gain pitch contribution to code[]. *
+ *-------------------------------------------------------*/
+ tmp = 0;
+ Preemph(code, st->tilt_code, L_SUBFR, &tmp);
+ Pit_shrp(code, T0, PIT_SHARP, L_SUBFR);
+ /*----------------------------------------------------------*
+ * - Compute the fixed codebook gain *
+ * - quantize fixed codebook gain *
+ *----------------------------------------------------------*/
+ if(*ser_size <= NBBITS_9k)
+ {
+ index = Q_gain2(xn, y1, Q_new + shift, y2, code, g_coeff, L_SUBFR, 6,
+ &gain_pit, &L_gain_code, clip_gain, st->qua_gain);
+ Parm_serial(index, 6, &prms);
+ } else
+ {
+ index = Q_gain2(xn, y1, Q_new + shift, y2, code, g_coeff, L_SUBFR, 7,
+ &gain_pit, &L_gain_code, clip_gain, st->qua_gain);
+ Parm_serial(index, 7, &prms);
+ }
+ /* test quantized gain of pitch for pitch clipping algorithm */
+ Gp_clip_test_gain_pit(gain_pit, st->gp_clip);
- L_tmp = L_shl(L_gain_code, Q_new);
- gain_code = extract_h(L_add(L_tmp, 0x8000));
+ L_tmp = L_shl(L_gain_code, Q_new);
+ gain_code = extract_h(L_add(L_tmp, 0x8000));
- /*----------------------------------------------------------*
- * Update parameters for the next subframe. *
- * - tilt of code: 0.0 (unvoiced) to 0.5 (voiced) *
- *----------------------------------------------------------*/
- /* find voice factor in Q15 (1=voiced, -1=unvoiced) */
- Copy(&exc[i_subfr], exc2, L_SUBFR);
+ /*----------------------------------------------------------*
+ * Update parameters for the next subframe. *
+ * - tilt of code: 0.0 (unvoiced) to 0.5 (voiced) *
+ *----------------------------------------------------------*/
+ /* find voice factor in Q15 (1=voiced, -1=unvoiced) */
+ Copy(&exc[i_subfr], exc2, L_SUBFR);
#ifdef ASM_OPT /* asm optimization branch */
- Scale_sig_opt(exc2, L_SUBFR, shift);
+ Scale_sig_opt(exc2, L_SUBFR, shift);
#else
- Scale_sig(exc2, L_SUBFR, shift);
+ Scale_sig(exc2, L_SUBFR, shift);
#endif
- voice_fac = voice_factor(exc2, shift, gain_pit, code, gain_code, L_SUBFR);
- /* tilt of code for next subframe: 0.5=voiced, 0=unvoiced */
- st->tilt_code = ((voice_fac >> 2) + 8192);
- /*------------------------------------------------------*
- * - Update filter's memory "mem_w0" for finding the *
- * target vector in the next subframe. *
- * - Find the total excitation *
- * - Find synthesis speech to update mem_syn[]. *
- *------------------------------------------------------*/
+ voice_fac = voice_factor(exc2, shift, gain_pit, code, gain_code, L_SUBFR);
+ /* tilt of code for next subframe: 0.5=voiced, 0=unvoiced */
+ st->tilt_code = ((voice_fac >> 2) + 8192);
+ /*------------------------------------------------------*
+ * - Update filter's memory "mem_w0" for finding the *
+ * target vector in the next subframe. *
+ * - Find the total excitation *
+ * - Find synthesis speech to update mem_syn[]. *
+ *------------------------------------------------------*/
- /* y2 in Q9, gain_pit in Q14 */
- L_tmp = (gain_code * y2[L_SUBFR - 1])<<1;
- L_tmp = L_shl(L_tmp, (5 + shift));
- L_tmp = L_negate(L_tmp);
- L_tmp += (xn[L_SUBFR - 1] * 16384)<<1;
- L_tmp -= (y1[L_SUBFR - 1] * gain_pit)<<1;
- L_tmp = L_shl(L_tmp, (1 - shift));
- st->mem_w0 = extract_h(L_add(L_tmp, 0x8000));
+ /* y2 in Q9, gain_pit in Q14 */
+ L_tmp = (gain_code * y2[L_SUBFR - 1])<<1;
+ L_tmp = L_shl(L_tmp, (5 + shift));
+ L_tmp = L_negate(L_tmp);
+ L_tmp += (xn[L_SUBFR - 1] * 16384)<<1;
+ L_tmp -= (y1[L_SUBFR - 1] * gain_pit)<<1;
+ L_tmp = L_shl(L_tmp, (1 - shift));
+ st->mem_w0 = extract_h(L_add(L_tmp, 0x8000));
- if (*ser_size >= NBBITS_24k)
- Copy(&exc[i_subfr], exc2, L_SUBFR);
+ if (*ser_size >= NBBITS_24k)
+ Copy(&exc[i_subfr], exc2, L_SUBFR);
- for (i = 0; i < L_SUBFR; i++)
- {
- /* code in Q9, gain_pit in Q14 */
- L_tmp = (gain_code * code[i])<<1;
- L_tmp = (L_tmp << 5);
- L_tmp += (exc[i + i_subfr] * gain_pit)<<1;
- L_tmp = L_shl2(L_tmp, 1);
- exc[i + i_subfr] = extract_h(L_add(L_tmp, 0x8000));
- }
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ Word32 tmp;
+ /* code in Q9, gain_pit in Q14 */
+ L_tmp = (gain_code * code[i])<<1;
+ L_tmp = (L_tmp << 5);
+ tmp = L_mult(exc[i + i_subfr], gain_pit); // (exc[i + i_subfr] * gain_pit)<<1
+ L_tmp = L_add(L_tmp, tmp);
+ L_tmp = L_shl2(L_tmp, 1);
+ exc[i + i_subfr] = extract_h(L_add(L_tmp, 0x8000));
+ }
- Syn_filt(p_Aq,&exc[i_subfr], synth, L_SUBFR, st->mem_syn, 1);
+ Syn_filt(p_Aq,&exc[i_subfr], synth, L_SUBFR, st->mem_syn, 1);
- if(*ser_size >= NBBITS_24k)
- {
- /*------------------------------------------------------------*
- * phase dispersion to enhance noise in low bit rate *
- *------------------------------------------------------------*/
- /* L_gain_code in Q16 */
- VO_L_Extract(L_gain_code, &gain_code, &gain_code_lo);
+ if(*ser_size >= NBBITS_24k)
+ {
+ /*------------------------------------------------------------*
+ * phase dispersion to enhance noise in low bit rate *
+ *------------------------------------------------------------*/
+ /* L_gain_code in Q16 */
+ VO_L_Extract(L_gain_code, &gain_code, &gain_code_lo);
- /*------------------------------------------------------------*
- * noise enhancer *
- * ~~~~~~~~~~~~~~ *
- * - Enhance excitation on noise. (modify gain of code) *
- * If signal is noisy and LPC filter is stable, move gain *
- * of code 1.5 dB toward gain of code threshold. *
- * This decrease by 3 dB noise energy variation. *
- *------------------------------------------------------------*/
- tmp = (16384 - (voice_fac >> 1)); /* 1=unvoiced, 0=voiced */
- fac = vo_mult(stab_fac, tmp);
- L_tmp = L_gain_code;
- if(L_tmp < st->L_gc_thres)
- {
- L_tmp = vo_L_add(L_tmp, Mpy_32_16(gain_code, gain_code_lo, 6226));
- if(L_tmp > st->L_gc_thres)
- {
- L_tmp = st->L_gc_thres;
- }
- } else
- {
- L_tmp = Mpy_32_16(gain_code, gain_code_lo, 27536);
- if(L_tmp < st->L_gc_thres)
- {
- L_tmp = st->L_gc_thres;
- }
- }
- st->L_gc_thres = L_tmp;
+ /*------------------------------------------------------------*
+ * noise enhancer *
+ * ~~~~~~~~~~~~~~ *
+ * - Enhance excitation on noise. (modify gain of code) *
+ * If signal is noisy and LPC filter is stable, move gain *
+ * of code 1.5 dB toward gain of code threshold. *
+ * This decrease by 3 dB noise energy variation. *
+ *------------------------------------------------------------*/
+ tmp = (16384 - (voice_fac >> 1)); /* 1=unvoiced, 0=voiced */
+ fac = vo_mult(stab_fac, tmp);
+ L_tmp = L_gain_code;
+ if(L_tmp < st->L_gc_thres)
+ {
+ L_tmp = vo_L_add(L_tmp, Mpy_32_16(gain_code, gain_code_lo, 6226));
+ if(L_tmp > st->L_gc_thres)
+ {
+ L_tmp = st->L_gc_thres;
+ }
+ } else
+ {
+ L_tmp = Mpy_32_16(gain_code, gain_code_lo, 27536);
+ if(L_tmp < st->L_gc_thres)
+ {
+ L_tmp = st->L_gc_thres;
+ }
+ }
+ st->L_gc_thres = L_tmp;
- L_gain_code = Mpy_32_16(gain_code, gain_code_lo, (32767 - fac));
- VO_L_Extract(L_tmp, &gain_code, &gain_code_lo);
- L_gain_code = vo_L_add(L_gain_code, Mpy_32_16(gain_code, gain_code_lo, fac));
+ L_gain_code = Mpy_32_16(gain_code, gain_code_lo, (32767 - fac));
+ VO_L_Extract(L_tmp, &gain_code, &gain_code_lo);
+ L_gain_code = vo_L_add(L_gain_code, Mpy_32_16(gain_code, gain_code_lo, fac));
- /*------------------------------------------------------------*
- * pitch enhancer *
- * ~~~~~~~~~~~~~~ *
- * - Enhance excitation on voice. (HP filtering of code) *
- * On voiced signal, filtering of code by a smooth fir HP *
- * filter to decrease energy of code in low frequency. *
- *------------------------------------------------------------*/
+ /*------------------------------------------------------------*
+ * pitch enhancer *
+ * ~~~~~~~~~~~~~~ *
+ * - Enhance excitation on voice. (HP filtering of code) *
+ * On voiced signal, filtering of code by a smooth fir HP *
+ * filter to decrease energy of code in low frequency. *
+ *------------------------------------------------------------*/
- tmp = ((voice_fac >> 3) + 4096); /* 0.25=voiced, 0=unvoiced */
+ tmp = ((voice_fac >> 3) + 4096); /* 0.25=voiced, 0=unvoiced */
- L_tmp = L_deposit_h(code[0]);
- L_tmp -= (code[1] * tmp)<<1;
- code2[0] = vo_round(L_tmp);
+ L_tmp = L_deposit_h(code[0]);
+ L_tmp -= (code[1] * tmp)<<1;
+ code2[0] = vo_round(L_tmp);
- for (i = 1; i < L_SUBFR - 1; i++)
- {
- L_tmp = L_deposit_h(code[i]);
- L_tmp -= (code[i + 1] * tmp)<<1;
- L_tmp -= (code[i - 1] * tmp)<<1;
- code2[i] = vo_round(L_tmp);
- }
+ for (i = 1; i < L_SUBFR - 1; i++)
+ {
+ L_tmp = L_deposit_h(code[i]);
+ L_tmp -= (code[i + 1] * tmp)<<1;
+ L_tmp -= (code[i - 1] * tmp)<<1;
+ code2[i] = vo_round(L_tmp);
+ }
- L_tmp = L_deposit_h(code[L_SUBFR - 1]);
- L_tmp -= (code[L_SUBFR - 2] * tmp)<<1;
- code2[L_SUBFR - 1] = vo_round(L_tmp);
+ L_tmp = L_deposit_h(code[L_SUBFR - 1]);
+ L_tmp -= (code[L_SUBFR - 2] * tmp)<<1;
+ code2[L_SUBFR - 1] = vo_round(L_tmp);
- /* build excitation */
- gain_code = vo_round(L_shl(L_gain_code, Q_new));
+ /* build excitation */
+ gain_code = vo_round(L_shl(L_gain_code, Q_new));
- for (i = 0; i < L_SUBFR; i++)
- {
- L_tmp = (code2[i] * gain_code)<<1;
- L_tmp = (L_tmp << 5);
- L_tmp += (exc2[i] * gain_pit)<<1;
- L_tmp = (L_tmp << 1);
- exc2[i] = vo_round(L_tmp);
- }
+ for (i = 0; i < L_SUBFR; i++)
+ {
+ L_tmp = (code2[i] * gain_code)<<1;
+ L_tmp = (L_tmp << 5);
+ L_tmp += (exc2[i] * gain_pit)<<1;
+ L_tmp = (L_tmp << 1);
+ exc2[i] = voround(L_tmp);
+ }
- corr_gain = synthesis(p_Aq, exc2, Q_new, &speech16k[i_subfr * 5 / 4], st);
- Parm_serial(corr_gain, 4, &prms);
- }
- p_A += (M + 1);
- p_Aq += (M + 1);
- } /* end of subframe loop */
+ corr_gain = synthesis(p_Aq, exc2, Q_new, &speech16k[i_subfr * 5 / 4], st);
+ Parm_serial(corr_gain, 4, &prms);
+ }
+ p_A += (M + 1);
+ p_Aq += (M + 1);
+ } /* end of subframe loop */
- /*--------------------------------------------------*
- * Update signal for next frame. *
- * -> save past of speech[], wsp[] and exc[]. *
- *--------------------------------------------------*/
- Copy(&old_speech[L_FRAME], st->old_speech, L_TOTAL - L_FRAME);
- Copy(&old_wsp[L_FRAME / OPL_DECIM], st->old_wsp, PIT_MAX / OPL_DECIM);
- Copy(&old_exc[L_FRAME], st->old_exc, PIT_MAX + L_INTERPOL);
- return;
+ /*--------------------------------------------------*
+ * Update signal for next frame. *
+ * -> save past of speech[], wsp[] and exc[]. *
+ *--------------------------------------------------*/
+ Copy(&old_speech[L_FRAME], st->old_speech, L_TOTAL - L_FRAME);
+ Copy(&old_wsp[L_FRAME / OPL_DECIM], st->old_wsp, PIT_MAX / OPL_DECIM);
+ Copy(&old_exc[L_FRAME], st->old_exc, PIT_MAX + L_INTERPOL);
+ return;
}
/*-----------------------------------------------------*
@@ -1329,225 +1333,225 @@
*-----------------------------------------------------*/
static Word16 synthesis(
- Word16 Aq[], /* A(z) : quantized Az */
- Word16 exc[], /* (i) : excitation at 12kHz */
- Word16 Q_new, /* (i) : scaling performed on exc */
- Word16 synth16k[], /* (o) : 16kHz synthesis signal */
- Coder_State * st /* (i/o) : State structure */
- )
+ Word16 Aq[], /* A(z) : quantized Az */
+ Word16 exc[], /* (i) : excitation at 12kHz */
+ Word16 Q_new, /* (i) : scaling performed on exc */
+ Word16 synth16k[], /* (o) : 16kHz synthesis signal */
+ Coder_State * st /* (i/o) : State structure */
+ )
{
- Word16 fac, tmp, exp;
- Word16 ener, exp_ener;
- Word32 L_tmp, i;
+ Word16 fac, tmp, exp;
+ Word16 ener, exp_ener;
+ Word32 L_tmp, i;
- Word16 synth_hi[M + L_SUBFR], synth_lo[M + L_SUBFR];
- Word16 synth[L_SUBFR];
- Word16 HF[L_SUBFR16k]; /* High Frequency vector */
- Word16 Ap[M + 1];
+ Word16 synth_hi[M + L_SUBFR], synth_lo[M + L_SUBFR];
+ Word16 synth[L_SUBFR];
+ Word16 HF[L_SUBFR16k]; /* High Frequency vector */
+ Word16 Ap[M + 1];
- Word16 HF_SP[L_SUBFR16k]; /* High Frequency vector (from original signal) */
+ Word16 HF_SP[L_SUBFR16k]; /* High Frequency vector (from original signal) */
- Word16 HP_est_gain, HP_calc_gain, HP_corr_gain;
- Word16 dist_min, dist;
- Word16 HP_gain_ind = 0;
- Word16 gain1, gain2;
- Word16 weight1, weight2;
+ Word16 HP_est_gain, HP_calc_gain, HP_corr_gain;
+ Word16 dist_min, dist;
+ Word16 HP_gain_ind = 0;
+ Word16 gain1, gain2;
+ Word16 weight1, weight2;
- /*------------------------------------------------------------*
- * speech synthesis *
- * ~~~~~~~~~~~~~~~~ *
- * - Find synthesis speech corresponding to exc2[]. *
- * - Perform fixed deemphasis and hp 50hz filtering. *
- * - Oversampling from 12.8kHz to 16kHz. *
- *------------------------------------------------------------*/
- Copy(st->mem_syn_hi, synth_hi, M);
- Copy(st->mem_syn_lo, synth_lo, M);
+ /*------------------------------------------------------------*
+ * speech synthesis *
+ * ~~~~~~~~~~~~~~~~ *
+ * - Find synthesis speech corresponding to exc2[]. *
+ * - Perform fixed deemphasis and hp 50hz filtering. *
+ * - Oversampling from 12.8kHz to 16kHz. *
+ *------------------------------------------------------------*/
+ Copy(st->mem_syn_hi, synth_hi, M);
+ Copy(st->mem_syn_lo, synth_lo, M);
#ifdef ASM_OPT /* asm optimization branch */
- Syn_filt_32_asm(Aq, M, exc, Q_new, synth_hi + M, synth_lo + M, L_SUBFR);
+ Syn_filt_32_asm(Aq, M, exc, Q_new, synth_hi + M, synth_lo + M, L_SUBFR);
#else
- Syn_filt_32(Aq, M, exc, Q_new, synth_hi + M, synth_lo + M, L_SUBFR);
+ Syn_filt_32(Aq, M, exc, Q_new, synth_hi + M, synth_lo + M, L_SUBFR);
#endif
- Copy(synth_hi + L_SUBFR, st->mem_syn_hi, M);
- Copy(synth_lo + L_SUBFR, st->mem_syn_lo, M);
+ Copy(synth_hi + L_SUBFR, st->mem_syn_hi, M);
+ Copy(synth_lo + L_SUBFR, st->mem_syn_lo, M);
#ifdef ASM_OPT /* asm optimization branch */
- Deemph_32_asm(synth_hi + M, synth_lo + M, synth, &(st->mem_deemph));
+ Deemph_32_asm(synth_hi + M, synth_lo + M, synth, &(st->mem_deemph));
#else
- Deemph_32(synth_hi + M, synth_lo + M, synth, PREEMPH_FAC, L_SUBFR, &(st->mem_deemph));
+ Deemph_32(synth_hi + M, synth_lo + M, synth, PREEMPH_FAC, L_SUBFR, &(st->mem_deemph));
#endif
- HP50_12k8(synth, L_SUBFR, st->mem_sig_out);
+ HP50_12k8(synth, L_SUBFR, st->mem_sig_out);
- /* Original speech signal as reference for high band gain quantisation */
- for (i = 0; i < L_SUBFR16k; i++)
- {
- HF_SP[i] = synth16k[i];
- }
+ /* Original speech signal as reference for high band gain quantisation */
+ for (i = 0; i < L_SUBFR16k; i++)
+ {
+ HF_SP[i] = synth16k[i];
+ }
- /*------------------------------------------------------*
- * HF noise synthesis *
- * ~~~~~~~~~~~~~~~~~~ *
- * - Generate HF noise between 5.5 and 7.5 kHz. *
- * - Set energy of noise according to synthesis tilt. *
- * tilt > 0.8 ==> - 14 dB (voiced) *
- * tilt 0.5 ==> - 6 dB (voiced or noise) *
- * tilt < 0.0 ==> 0 dB (noise) *
- *------------------------------------------------------*/
- /* generate white noise vector */
- for (i = 0; i < L_SUBFR16k; i++)
- {
- HF[i] = Random(&(st->seed2))>>3;
- }
- /* energy of excitation */
+ /*------------------------------------------------------*
+ * HF noise synthesis *
+ * ~~~~~~~~~~~~~~~~~~ *
+ * - Generate HF noise between 5.5 and 7.5 kHz. *
+ * - Set energy of noise according to synthesis tilt. *
+ * tilt > 0.8 ==> - 14 dB (voiced) *
+ * tilt 0.5 ==> - 6 dB (voiced or noise) *
+ * tilt < 0.0 ==> 0 dB (noise) *
+ *------------------------------------------------------*/
+ /* generate white noise vector */
+ for (i = 0; i < L_SUBFR16k; i++)
+ {
+ HF[i] = Random(&(st->seed2))>>3;
+ }
+ /* energy of excitation */
#ifdef ASM_OPT /* asm optimization branch */
- Scale_sig_opt(exc, L_SUBFR, -3);
- Q_new = Q_new - 3;
- ener = extract_h(Dot_product12_asm(exc, exc, L_SUBFR, &exp_ener));
+ Scale_sig_opt(exc, L_SUBFR, -3);
+ Q_new = Q_new - 3;
+ ener = extract_h(Dot_product12_asm(exc, exc, L_SUBFR, &exp_ener));
#else
- Scale_sig(exc, L_SUBFR, -3);
- Q_new = Q_new - 3;
- ener = extract_h(Dot_product12(exc, exc, L_SUBFR, &exp_ener));
+ Scale_sig(exc, L_SUBFR, -3);
+ Q_new = Q_new - 3;
+ ener = extract_h(Dot_product12(exc, exc, L_SUBFR, &exp_ener));
#endif
- exp_ener = exp_ener - (Q_new + Q_new);
- /* set energy of white noise to energy of excitation */
+ exp_ener = exp_ener - (Q_new + Q_new);
+ /* set energy of white noise to energy of excitation */
#ifdef ASM_OPT /* asm optimization branch */
- tmp = extract_h(Dot_product12_asm(HF, HF, L_SUBFR16k, &exp));
+ tmp = extract_h(Dot_product12_asm(HF, HF, L_SUBFR16k, &exp));
#else
- tmp = extract_h(Dot_product12(HF, HF, L_SUBFR16k, &exp));
+ tmp = extract_h(Dot_product12(HF, HF, L_SUBFR16k, &exp));
#endif
- if(tmp > ener)
- {
- tmp = (tmp >> 1); /* Be sure tmp < ener */
- exp = (exp + 1);
- }
- L_tmp = L_deposit_h(div_s(tmp, ener)); /* result is normalized */
- exp = (exp - exp_ener);
- Isqrt_n(&L_tmp, &exp);
- L_tmp = L_shl(L_tmp, (exp + 1)); /* L_tmp x 2, L_tmp in Q31 */
- tmp = extract_h(L_tmp); /* tmp = 2 x sqrt(ener_exc/ener_hf) */
+ if(tmp > ener)
+ {
+ tmp = (tmp >> 1); /* Be sure tmp < ener */
+ exp = (exp + 1);
+ }
+ L_tmp = L_deposit_h(div_s(tmp, ener)); /* result is normalized */
+ exp = (exp - exp_ener);
+ Isqrt_n(&L_tmp, &exp);
+ L_tmp = L_shl(L_tmp, (exp + 1)); /* L_tmp x 2, L_tmp in Q31 */
+ tmp = extract_h(L_tmp); /* tmp = 2 x sqrt(ener_exc/ener_hf) */
- for (i = 0; i < L_SUBFR16k; i++)
- {
- HF[i] = vo_mult(HF[i], tmp);
- }
+ for (i = 0; i < L_SUBFR16k; i++)
+ {
+ HF[i] = vo_mult(HF[i], tmp);
+ }
- /* find tilt of synthesis speech (tilt: 1=voiced, -1=unvoiced) */
- HP400_12k8(synth, L_SUBFR, st->mem_hp400);
+ /* find tilt of synthesis speech (tilt: 1=voiced, -1=unvoiced) */
+ HP400_12k8(synth, L_SUBFR, st->mem_hp400);
- L_tmp = 1L;
- for (i = 0; i < L_SUBFR; i++)
- L_tmp += (synth[i] * synth[i])<<1;
+ L_tmp = 1L;
+ for (i = 0; i < L_SUBFR; i++)
+ L_tmp += (synth[i] * synth[i])<<1;
- exp = norm_l(L_tmp);
- ener = extract_h(L_tmp << exp); /* ener = r[0] */
+ exp = norm_l(L_tmp);
+ ener = extract_h(L_tmp << exp); /* ener = r[0] */
- L_tmp = 1L;
- for (i = 1; i < L_SUBFR; i++)
- L_tmp +=(synth[i] * synth[i - 1])<<1;
+ L_tmp = 1L;
+ for (i = 1; i < L_SUBFR; i++)
+ L_tmp +=(synth[i] * synth[i - 1])<<1;
- tmp = extract_h(L_tmp << exp); /* tmp = r[1] */
+ tmp = extract_h(L_tmp << exp); /* tmp = r[1] */
- if (tmp > 0)
- {
- fac = div_s(tmp, ener);
- } else
- {
- fac = 0;
- }
+ if (tmp > 0)
+ {
+ fac = div_s(tmp, ener);
+ } else
+ {
+ fac = 0;
+ }
- /* modify energy of white noise according to synthesis tilt */
- gain1 = 32767 - fac;
- gain2 = vo_mult(gain1, 20480);
- gain2 = shl(gain2, 1);
+ /* modify energy of white noise according to synthesis tilt */
+ gain1 = 32767 - fac;
+ gain2 = vo_mult(gain1, 20480);
+ gain2 = shl(gain2, 1);
- if (st->vad_hist > 0)
- {
- weight1 = 0;
- weight2 = 32767;
- } else
- {
- weight1 = 32767;
- weight2 = 0;
- }
- tmp = vo_mult(weight1, gain1);
- tmp = add1(tmp, vo_mult(weight2, gain2));
+ if (st->vad_hist > 0)
+ {
+ weight1 = 0;
+ weight2 = 32767;
+ } else
+ {
+ weight1 = 32767;
+ weight2 = 0;
+ }
+ tmp = vo_mult(weight1, gain1);
+ tmp = add1(tmp, vo_mult(weight2, gain2));
- if (tmp != 0)
- {
- tmp = (tmp + 1);
- }
- HP_est_gain = tmp;
+ if (tmp != 0)
+ {
+ tmp = (tmp + 1);
+ }
+ HP_est_gain = tmp;
- if(HP_est_gain < 3277)
- {
- HP_est_gain = 3277; /* 0.1 in Q15 */
- }
- /* synthesis of noise: 4.8kHz..5.6kHz --> 6kHz..7kHz */
- Weight_a(Aq, Ap, 19661, M); /* fac=0.6 */
+ if(HP_est_gain < 3277)
+ {
+ HP_est_gain = 3277; /* 0.1 in Q15 */
+ }
+ /* synthesis of noise: 4.8kHz..5.6kHz --> 6kHz..7kHz */
+ Weight_a(Aq, Ap, 19661, M); /* fac=0.6 */
#ifdef ASM_OPT /* asm optimization branch */
- Syn_filt_asm(Ap, HF, HF, st->mem_syn_hf);
- /* noise High Pass filtering (1ms of delay) */
- Filt_6k_7k_asm(HF, L_SUBFR16k, st->mem_hf);
- /* filtering of the original signal */
- Filt_6k_7k_asm(HF_SP, L_SUBFR16k, st->mem_hf2);
+ Syn_filt_asm(Ap, HF, HF, st->mem_syn_hf);
+ /* noise High Pass filtering (1ms of delay) */
+ Filt_6k_7k_asm(HF, L_SUBFR16k, st->mem_hf);
+ /* filtering of the original signal */
+ Filt_6k_7k_asm(HF_SP, L_SUBFR16k, st->mem_hf2);
- /* check the gain difference */
- Scale_sig_opt(HF_SP, L_SUBFR16k, -1);
- ener = extract_h(Dot_product12_asm(HF_SP, HF_SP, L_SUBFR16k, &exp_ener));
- /* set energy of white noise to energy of excitation */
- tmp = extract_h(Dot_product12_asm(HF, HF, L_SUBFR16k, &exp));
+ /* check the gain difference */
+ Scale_sig_opt(HF_SP, L_SUBFR16k, -1);
+ ener = extract_h(Dot_product12_asm(HF_SP, HF_SP, L_SUBFR16k, &exp_ener));
+ /* set energy of white noise to energy of excitation */
+ tmp = extract_h(Dot_product12_asm(HF, HF, L_SUBFR16k, &exp));
#else
- Syn_filt(Ap, HF, HF, L_SUBFR16k, st->mem_syn_hf, 1);
- /* noise High Pass filtering (1ms of delay) */
- Filt_6k_7k(HF, L_SUBFR16k, st->mem_hf);
- /* filtering of the original signal */
- Filt_6k_7k(HF_SP, L_SUBFR16k, st->mem_hf2);
- /* check the gain difference */
- Scale_sig(HF_SP, L_SUBFR16k, -1);
- ener = extract_h(Dot_product12(HF_SP, HF_SP, L_SUBFR16k, &exp_ener));
- /* set energy of white noise to energy of excitation */
- tmp = extract_h(Dot_product12(HF, HF, L_SUBFR16k, &exp));
+ Syn_filt(Ap, HF, HF, L_SUBFR16k, st->mem_syn_hf, 1);
+ /* noise High Pass filtering (1ms of delay) */
+ Filt_6k_7k(HF, L_SUBFR16k, st->mem_hf);
+ /* filtering of the original signal */
+ Filt_6k_7k(HF_SP, L_SUBFR16k, st->mem_hf2);
+ /* check the gain difference */
+ Scale_sig(HF_SP, L_SUBFR16k, -1);
+ ener = extract_h(Dot_product12(HF_SP, HF_SP, L_SUBFR16k, &exp_ener));
+ /* set energy of white noise to energy of excitation */
+ tmp = extract_h(Dot_product12(HF, HF, L_SUBFR16k, &exp));
#endif
- if (tmp > ener)
- {
- tmp = (tmp >> 1); /* Be sure tmp < ener */
- exp = (exp + 1);
- }
- L_tmp = L_deposit_h(div_s(tmp, ener)); /* result is normalized */
- exp = vo_sub(exp, exp_ener);
- Isqrt_n(&L_tmp, &exp);
- L_tmp = L_shl(L_tmp, exp); /* L_tmp, L_tmp in Q31 */
- HP_calc_gain = extract_h(L_tmp); /* tmp = sqrt(ener_input/ener_hf) */
+ if (tmp > ener)
+ {
+ tmp = (tmp >> 1); /* Be sure tmp < ener */
+ exp = (exp + 1);
+ }
+ L_tmp = L_deposit_h(div_s(tmp, ener)); /* result is normalized */
+ exp = vo_sub(exp, exp_ener);
+ Isqrt_n(&L_tmp, &exp);
+ L_tmp = L_shl(L_tmp, exp); /* L_tmp, L_tmp in Q31 */
+ HP_calc_gain = extract_h(L_tmp); /* tmp = sqrt(ener_input/ener_hf) */
- /* st->gain_alpha *= st->dtx_encSt->dtxHangoverCount/7 */
- L_tmp = (vo_L_mult(st->dtx_encSt->dtxHangoverCount, 4681) << 15);
- st->gain_alpha = vo_mult(st->gain_alpha, extract_h(L_tmp));
+ /* st->gain_alpha *= st->dtx_encSt->dtxHangoverCount/7 */
+ L_tmp = (vo_L_mult(st->dtx_encSt->dtxHangoverCount, 4681) << 15);
+ st->gain_alpha = vo_mult(st->gain_alpha, extract_h(L_tmp));
- if(st->dtx_encSt->dtxHangoverCount > 6)
- st->gain_alpha = 32767;
- HP_est_gain = HP_est_gain >> 1; /* From Q15 to Q14 */
- HP_corr_gain = add1(vo_mult(HP_calc_gain, st->gain_alpha), vo_mult((32767 - st->gain_alpha), HP_est_gain));
+ if(st->dtx_encSt->dtxHangoverCount > 6)
+ st->gain_alpha = 32767;
+ HP_est_gain = HP_est_gain >> 1; /* From Q15 to Q14 */
+ HP_corr_gain = add1(vo_mult(HP_calc_gain, st->gain_alpha), vo_mult((32767 - st->gain_alpha), HP_est_gain));
- /* Quantise the correction gain */
- dist_min = 32767;
- for (i = 0; i < 16; i++)
- {
- dist = vo_mult((HP_corr_gain - HP_gain[i]), (HP_corr_gain - HP_gain[i]));
- if (dist_min > dist)
- {
- dist_min = dist;
- HP_gain_ind = i;
- }
- }
- HP_corr_gain = HP_gain[HP_gain_ind];
- /* return the quantised gain index when using the highest mode, otherwise zero */
- return (HP_gain_ind);
+ /* Quantise the correction gain */
+ dist_min = 32767;
+ for (i = 0; i < 16; i++)
+ {
+ dist = vo_mult((HP_corr_gain - HP_gain[i]), (HP_corr_gain - HP_gain[i]));
+ if (dist_min > dist)
+ {
+ dist_min = dist;
+ HP_gain_ind = i;
+ }
+ }
+ HP_corr_gain = HP_gain[HP_gain_ind];
+ /* return the quantised gain index when using the highest mode, otherwise zero */
+ return (HP_gain_ind);
}
/*************************************************
@@ -1558,33 +1562,33 @@
int AMR_Enc_Encode(HAMRENC hCodec)
{
- Word32 i;
- Coder_State *gData = (Coder_State*)hCodec;
- Word16 *signal;
- Word16 packed_size = 0;
- Word16 prms[NB_BITS_MAX];
- Word16 coding_mode = 0, nb_bits, allow_dtx, mode, reset_flag;
- mode = gData->mode;
- coding_mode = gData->mode;
- nb_bits = nb_of_bits[mode];
- signal = (Word16 *)gData->inputStream;
- allow_dtx = gData->allow_dtx;
+ Word32 i;
+ Coder_State *gData = (Coder_State*)hCodec;
+ Word16 *signal;
+ Word16 packed_size = 0;
+ Word16 prms[NB_BITS_MAX];
+ Word16 coding_mode = 0, nb_bits, allow_dtx, mode, reset_flag;
+ mode = gData->mode;
+ coding_mode = gData->mode;
+ nb_bits = nb_of_bits[mode];
+ signal = (Word16 *)gData->inputStream;
+ allow_dtx = gData->allow_dtx;
- /* check for homing frame */
- reset_flag = encoder_homing_frame_test(signal);
+ /* check for homing frame */
+ reset_flag = encoder_homing_frame_test(signal);
- for (i = 0; i < L_FRAME16k; i++) /* Delete the 2 LSBs (14-bit input) */
- {
- *(signal + i) = (Word16) (*(signal + i) & 0xfffC);
- }
+ for (i = 0; i < L_FRAME16k; i++) /* Delete the 2 LSBs (14-bit input) */
+ {
+ *(signal + i) = (Word16) (*(signal + i) & 0xfffC);
+ }
- coder(&coding_mode, signal, prms, &nb_bits, gData, allow_dtx);
- packed_size = PackBits(prms, coding_mode, mode, gData);
- if (reset_flag != 0)
- {
- Reset_encoder(gData, 1);
- }
- return packed_size;
+ coder(&coding_mode, signal, prms, &nb_bits, gData, allow_dtx);
+ packed_size = PackBits(prms, coding_mode, mode, gData);
+ if (reset_flag != 0)
+ {
+ Reset_encoder(gData, 1);
+ }
+ return packed_size;
}
/***************************************************************************
@@ -1594,94 +1598,94 @@
***************************************************************************/
VO_U32 VO_API voAMRWB_Init(VO_HANDLE * phCodec, /* o: the audio codec handle */
- VO_AUDIO_CODINGTYPE vType, /* i: Codec Type ID */
- VO_CODEC_INIT_USERDATA * pUserData /* i: init Parameters */
- )
+ VO_AUDIO_CODINGTYPE vType, /* i: Codec Type ID */
+ VO_CODEC_INIT_USERDATA * pUserData /* i: init Parameters */
+ )
{
- Coder_State *st;
- FrameStream *stream;
+ Coder_State *st;
+ FrameStream *stream;
#ifdef USE_DEAULT_MEM
- VO_MEM_OPERATOR voMemoprator;
+ VO_MEM_OPERATOR voMemoprator;
#endif
- VO_MEM_OPERATOR *pMemOP;
+ VO_MEM_OPERATOR *pMemOP;
UNUSED(vType);
- int interMem = 0;
+ int interMem = 0;
- if(pUserData == NULL || pUserData->memflag != VO_IMF_USERMEMOPERATOR || pUserData->memData == NULL )
- {
+ if(pUserData == NULL || pUserData->memflag != VO_IMF_USERMEMOPERATOR || pUserData->memData == NULL )
+ {
#ifdef USE_DEAULT_MEM
- voMemoprator.Alloc = cmnMemAlloc;
- voMemoprator.Copy = cmnMemCopy;
- voMemoprator.Free = cmnMemFree;
- voMemoprator.Set = cmnMemSet;
- voMemoprator.Check = cmnMemCheck;
- interMem = 1;
- pMemOP = &voMemoprator;
+ voMemoprator.Alloc = cmnMemAlloc;
+ voMemoprator.Copy = cmnMemCopy;
+ voMemoprator.Free = cmnMemFree;
+ voMemoprator.Set = cmnMemSet;
+ voMemoprator.Check = cmnMemCheck;
+ interMem = 1;
+ pMemOP = &voMemoprator;
#else
- *phCodec = NULL;
- return VO_ERR_INVALID_ARG;
+ *phCodec = NULL;
+ return VO_ERR_INVALID_ARG;
#endif
- }
- else
- {
- pMemOP = (VO_MEM_OPERATOR *)pUserData->memData;
- }
- /*-------------------------------------------------------------------------*
- * Memory allocation for coder state. *
- *-------------------------------------------------------------------------*/
- if ((st = (Coder_State *)mem_malloc(pMemOP, sizeof(Coder_State), 32, VO_INDEX_ENC_AMRWB)) == NULL)
- {
- return VO_ERR_OUTOF_MEMORY;
- }
+ }
+ else
+ {
+ pMemOP = (VO_MEM_OPERATOR *)pUserData->memData;
+ }
+ /*-------------------------------------------------------------------------*
+ * Memory allocation for coder state. *
+ *-------------------------------------------------------------------------*/
+ if ((st = (Coder_State *)mem_malloc(pMemOP, sizeof(Coder_State), 32, VO_INDEX_ENC_AMRWB)) == NULL)
+ {
+ return VO_ERR_OUTOF_MEMORY;
+ }
- st->vadSt = NULL;
- st->dtx_encSt = NULL;
- st->sid_update_counter = 3;
- st->sid_handover_debt = 0;
- st->prev_ft = TX_SPEECH;
- st->inputStream = NULL;
- st->inputSize = 0;
+ st->vadSt = NULL;
+ st->dtx_encSt = NULL;
+ st->sid_update_counter = 3;
+ st->sid_handover_debt = 0;
+ st->prev_ft = TX_SPEECH;
+ st->inputStream = NULL;
+ st->inputSize = 0;
- /* Default setting */
- st->mode = VOAMRWB_MD2385; /* bit rate 23.85kbps */
- st->frameType = VOAMRWB_RFC3267; /* frame type: RFC3267 */
- st->allow_dtx = 0; /* disable DTX mode */
+ /* Default setting */
+ st->mode = VOAMRWB_MD2385; /* bit rate 23.85kbps */
+ st->frameType = VOAMRWB_RFC3267; /* frame type: RFC3267 */
+ st->allow_dtx = 0; /* disable DTX mode */
- st->outputStream = NULL;
- st->outputSize = 0;
+ st->outputStream = NULL;
+ st->outputSize = 0;
- st->stream = (FrameStream *)mem_malloc(pMemOP, sizeof(FrameStream), 32, VO_INDEX_ENC_AMRWB);
- if(st->stream == NULL)
- return VO_ERR_OUTOF_MEMORY;
+ st->stream = (FrameStream *)mem_malloc(pMemOP, sizeof(FrameStream), 32, VO_INDEX_ENC_AMRWB);
+ if(st->stream == NULL)
+ return VO_ERR_OUTOF_MEMORY;
- st->stream->frame_ptr = (unsigned char *)mem_malloc(pMemOP, Frame_Maxsize, 32, VO_INDEX_ENC_AMRWB);
- if(st->stream->frame_ptr == NULL)
- return VO_ERR_OUTOF_MEMORY;
+ st->stream->frame_ptr = (unsigned char *)mem_malloc(pMemOP, Frame_Maxsize, 32, VO_INDEX_ENC_AMRWB);
+ if(st->stream->frame_ptr == NULL)
+ return VO_ERR_OUTOF_MEMORY;
- stream = st->stream;
- voAWB_InitFrameBuffer(stream);
+ stream = st->stream;
+ voAWB_InitFrameBuffer(stream);
- wb_vad_init(&(st->vadSt), pMemOP);
- dtx_enc_init(&(st->dtx_encSt), isf_init, pMemOP);
+ wb_vad_init(&(st->vadSt), pMemOP);
+ dtx_enc_init(&(st->dtx_encSt), isf_init, pMemOP);
- Reset_encoder((void *) st, 1);
+ Reset_encoder((void *) st, 1);
- if(interMem)
- {
- st->voMemoprator.Alloc = cmnMemAlloc;
- st->voMemoprator.Copy = cmnMemCopy;
- st->voMemoprator.Free = cmnMemFree;
- st->voMemoprator.Set = cmnMemSet;
- st->voMemoprator.Check = cmnMemCheck;
- pMemOP = &st->voMemoprator;
- }
+ if(interMem)
+ {
+ st->voMemoprator.Alloc = cmnMemAlloc;
+ st->voMemoprator.Copy = cmnMemCopy;
+ st->voMemoprator.Free = cmnMemFree;
+ st->voMemoprator.Set = cmnMemSet;
+ st->voMemoprator.Check = cmnMemCheck;
+ pMemOP = &st->voMemoprator;
+ }
- st->pvoMemop = pMemOP;
+ st->pvoMemop = pMemOP;
- *phCodec = (void *) st;
+ *phCodec = (void *) st;
- return VO_ERR_NONE;
+ return VO_ERR_NONE;
}
/**********************************************************************************
@@ -1691,32 +1695,32 @@
***********************************************************************************/
VO_U32 VO_API voAMRWB_SetInputData(
- VO_HANDLE hCodec, /* i/o: The codec handle which was created by Init function */
- VO_CODECBUFFER * pInput /* i: The input buffer parameter */
- )
+ VO_HANDLE hCodec, /* i/o: The codec handle which was created by Init function */
+ VO_CODECBUFFER * pInput /* i: The input buffer parameter */
+ )
{
- Coder_State *gData;
- FrameStream *stream;
+ Coder_State *gData;
+ FrameStream *stream;
- if(NULL == hCodec)
- {
- return VO_ERR_INVALID_ARG;
- }
+ if(NULL == hCodec)
+ {
+ return VO_ERR_INVALID_ARG;
+ }
- gData = (Coder_State *)hCodec;
- stream = gData->stream;
+ gData = (Coder_State *)hCodec;
+ stream = gData->stream;
- if(NULL == pInput || NULL == pInput->Buffer)
- {
- return VO_ERR_INVALID_ARG;
- }
+ if(NULL == pInput || NULL == pInput->Buffer)
+ {
+ return VO_ERR_INVALID_ARG;
+ }
- stream->set_ptr = pInput->Buffer;
- stream->set_len = pInput->Length;
- stream->frame_ptr = stream->frame_ptr_bk;
- stream->used_len = 0;
+ stream->set_ptr = pInput->Buffer;
+ stream->set_len = pInput->Length;
+ stream->frame_ptr = stream->frame_ptr_bk;
+ stream->used_len = 0;
- return VO_ERR_NONE;
+ return VO_ERR_NONE;
}
/**************************************************************************************
@@ -1726,52 +1730,52 @@
***************************************************************************************/
VO_U32 VO_API voAMRWB_GetOutputData(
- VO_HANDLE hCodec, /* i: The Codec Handle which was created by Init function*/
- VO_CODECBUFFER * pOutput, /* o: The output audio data */
- VO_AUDIO_OUTPUTINFO * pAudioFormat /* o: The encoder module filled audio format and used the input size*/
- )
+ VO_HANDLE hCodec, /* i: The Codec Handle which was created by Init function*/
+ VO_CODECBUFFER * pOutput, /* o: The output audio data */
+ VO_AUDIO_OUTPUTINFO * pAudioFormat /* o: The encoder module filled audio format and used the input size*/
+ )
{
- Coder_State* gData = (Coder_State*)hCodec;
- VO_MEM_OPERATOR *pMemOP;
- FrameStream *stream = (FrameStream *)gData->stream;
- pMemOP = (VO_MEM_OPERATOR *)gData->pvoMemop;
+ Coder_State* gData = (Coder_State*)hCodec;
+ VO_MEM_OPERATOR *pMemOP;
+ FrameStream *stream = (FrameStream *)gData->stream;
+ pMemOP = (VO_MEM_OPERATOR *)gData->pvoMemop;
- if(stream->framebuffer_len < Frame_MaxByte) /* check the work buffer len */
- {
- stream->frame_storelen = stream->framebuffer_len;
- if(stream->frame_storelen)
- {
- pMemOP->Copy(VO_INDEX_ENC_AMRWB, stream->frame_ptr_bk , stream->frame_ptr , stream->frame_storelen);
- }
- if(stream->set_len > 0)
- {
- voAWB_UpdateFrameBuffer(stream, pMemOP);
- }
- if(stream->framebuffer_len < Frame_MaxByte)
- {
- if(pAudioFormat)
- pAudioFormat->InputUsed = stream->used_len;
- return VO_ERR_INPUT_BUFFER_SMALL;
- }
- }
+ if(stream->framebuffer_len < Frame_MaxByte) /* check the work buffer len */
+ {
+ stream->frame_storelen = stream->framebuffer_len;
+ if(stream->frame_storelen)
+ {
+ pMemOP->Copy(VO_INDEX_ENC_AMRWB, stream->frame_ptr_bk , stream->frame_ptr , stream->frame_storelen);
+ }
+ if(stream->set_len > 0)
+ {
+ voAWB_UpdateFrameBuffer(stream, pMemOP);
+ }
+ if(stream->framebuffer_len < Frame_MaxByte)
+ {
+ if(pAudioFormat)
+ pAudioFormat->InputUsed = stream->used_len;
+ return VO_ERR_INPUT_BUFFER_SMALL;
+ }
+ }
- gData->inputStream = stream->frame_ptr;
- gData->outputStream = (unsigned short*)pOutput->Buffer;
+ gData->inputStream = stream->frame_ptr;
+ gData->outputStream = (unsigned short*)pOutput->Buffer;
- gData->outputSize = AMR_Enc_Encode(gData); /* encoder main function */
+ gData->outputSize = AMR_Enc_Encode(gData); /* encoder main function */
- pOutput->Length = gData->outputSize; /* get the output buffer length */
- stream->frame_ptr += 640; /* update the work buffer ptr */
- stream->framebuffer_len -= 640;
+ pOutput->Length = gData->outputSize; /* get the output buffer length */
+ stream->frame_ptr += 640; /* update the work buffer ptr */
+ stream->framebuffer_len -= 640;
- if(pAudioFormat) /* return output audio information */
- {
- pAudioFormat->Format.Channels = 1;
- pAudioFormat->Format.SampleRate = 8000;
- pAudioFormat->Format.SampleBits = 16;
- pAudioFormat->InputUsed = stream->used_len;
- }
- return VO_ERR_NONE;
+ if(pAudioFormat) /* return output audio information */
+ {
+ pAudioFormat->Format.Channels = 1;
+ pAudioFormat->Format.SampleRate = 8000;
+ pAudioFormat->Format.SampleBits = 16;
+ pAudioFormat->InputUsed = stream->used_len;
+ }
+ return VO_ERR_NONE;
}
/*************************************************************************
@@ -1782,50 +1786,50 @@
VO_U32 VO_API voAMRWB_SetParam(
- VO_HANDLE hCodec, /* i/o: The Codec Handle which was created by Init function */
- VO_S32 uParamID, /* i: The param ID */
- VO_PTR pData /* i: The param value depend on the ID */
- )
+ VO_HANDLE hCodec, /* i/o: The Codec Handle which was created by Init function */
+ VO_S32 uParamID, /* i: The param ID */
+ VO_PTR pData /* i: The param value depend on the ID */
+ )
{
- Coder_State* gData = (Coder_State*)hCodec;
- FrameStream *stream = (FrameStream *)(gData->stream);
- int *lValue = (int*)pData;
+ Coder_State* gData = (Coder_State*)hCodec;
+ FrameStream *stream = (FrameStream *)(gData->stream);
+ int *lValue = (int*)pData;
- switch(uParamID)
- {
- /* setting AMR-WB frame type*/
- case VO_PID_AMRWB_FRAMETYPE:
- if(*lValue < VOAMRWB_DEFAULT || *lValue > VOAMRWB_RFC3267)
- return VO_ERR_WRONG_PARAM_ID;
- gData->frameType = *lValue;
- break;
- /* setting AMR-WB bit rate */
- case VO_PID_AMRWB_MODE:
- {
- if(*lValue < VOAMRWB_MD66 || *lValue > VOAMRWB_MD2385)
- return VO_ERR_WRONG_PARAM_ID;
- gData->mode = *lValue;
- }
- break;
- /* enable or disable DTX mode */
- case VO_PID_AMRWB_DTX:
- gData->allow_dtx = (Word16)(*lValue);
- break;
+ switch(uParamID)
+ {
+ /* setting AMR-WB frame type*/
+ case VO_PID_AMRWB_FRAMETYPE:
+ if(*lValue < VOAMRWB_DEFAULT || *lValue > VOAMRWB_RFC3267)
+ return VO_ERR_WRONG_PARAM_ID;
+ gData->frameType = *lValue;
+ break;
+ /* setting AMR-WB bit rate */
+ case VO_PID_AMRWB_MODE:
+ {
+ if(*lValue < VOAMRWB_MD66 || *lValue > VOAMRWB_MD2385)
+ return VO_ERR_WRONG_PARAM_ID;
+ gData->mode = *lValue;
+ }
+ break;
+ /* enable or disable DTX mode */
+ case VO_PID_AMRWB_DTX:
+ gData->allow_dtx = (Word16)(*lValue);
+ break;
- case VO_PID_COMMON_HEADDATA:
- break;
+ case VO_PID_COMMON_HEADDATA:
+ break;
/* flush the work buffer */
- case VO_PID_COMMON_FLUSH:
- stream->set_ptr = NULL;
- stream->frame_storelen = 0;
- stream->framebuffer_len = 0;
- stream->set_len = 0;
- break;
+ case VO_PID_COMMON_FLUSH:
+ stream->set_ptr = NULL;
+ stream->frame_storelen = 0;
+ stream->framebuffer_len = 0;
+ stream->set_len = 0;
+ break;
- default:
- return VO_ERR_WRONG_PARAM_ID;
- }
- return VO_ERR_NONE;
+ default:
+ return VO_ERR_WRONG_PARAM_ID;
+ }
+ return VO_ERR_NONE;
}
/**************************************************************************
@@ -1835,52 +1839,52 @@
***************************************************************************/
VO_U32 VO_API voAMRWB_GetParam(
- VO_HANDLE hCodec, /* i: The Codec Handle which was created by Init function */
- VO_S32 uParamID, /* i: The param ID */
- VO_PTR pData /* o: The param value depend on the ID */
- )
+ VO_HANDLE hCodec, /* i: The Codec Handle which was created by Init function */
+ VO_S32 uParamID, /* i: The param ID */
+ VO_PTR pData /* o: The param value depend on the ID */
+ )
{
- int temp;
- Coder_State* gData = (Coder_State*)hCodec;
+ int temp;
+ Coder_State* gData = (Coder_State*)hCodec;
- if (gData==NULL)
- return VO_ERR_INVALID_ARG;
- switch(uParamID)
- {
- /* output audio format */
- case VO_PID_AMRWB_FORMAT:
- {
- VO_AUDIO_FORMAT* fmt = (VO_AUDIO_FORMAT*)pData;
- fmt->Channels = 1;
- fmt->SampleRate = 16000;
- fmt->SampleBits = 16;
- break;
- }
+ if (gData==NULL)
+ return VO_ERR_INVALID_ARG;
+ switch(uParamID)
+ {
+ /* output audio format */
+ case VO_PID_AMRWB_FORMAT:
+ {
+ VO_AUDIO_FORMAT* fmt = (VO_AUDIO_FORMAT*)pData;
+ fmt->Channels = 1;
+ fmt->SampleRate = 16000;
+ fmt->SampleBits = 16;
+ break;
+ }
/* output audio channel number */
- case VO_PID_AMRWB_CHANNELS:
- temp = 1;
- pData = (void *)(&temp);
- break;
+ case VO_PID_AMRWB_CHANNELS:
+ temp = 1;
+ pData = (void *)(&temp);
+ break;
/* output audio sample rate */
- case VO_PID_AMRWB_SAMPLERATE:
- temp = 16000;
- pData = (void *)(&temp);
- break;
- /* output audio frame type */
- case VO_PID_AMRWB_FRAMETYPE:
- temp = gData->frameType;
- pData = (void *)(&temp);
- break;
- /* output audio bit rate */
- case VO_PID_AMRWB_MODE:
- temp = gData->mode;
- pData = (void *)(&temp);
- break;
- default:
- return VO_ERR_WRONG_PARAM_ID;
- }
+ case VO_PID_AMRWB_SAMPLERATE:
+ temp = 16000;
+ pData = (void *)(&temp);
+ break;
+ /* output audio frame type */
+ case VO_PID_AMRWB_FRAMETYPE:
+ temp = gData->frameType;
+ pData = (void *)(&temp);
+ break;
+ /* output audio bit rate */
+ case VO_PID_AMRWB_MODE:
+ temp = gData->mode;
+ pData = (void *)(&temp);
+ break;
+ default:
+ return VO_ERR_WRONG_PARAM_ID;
+ }
- return VO_ERR_NONE;
+ return VO_ERR_NONE;
}
/***********************************************************************************
@@ -1890,32 +1894,32 @@
*************************************************************************************/
VO_U32 VO_API voAMRWB_Uninit(VO_HANDLE hCodec /* i/o: Codec handle pointer */
- )
+ )
{
- Coder_State* gData = (Coder_State*)hCodec;
- VO_MEM_OPERATOR *pMemOP;
- pMemOP = gData->pvoMemop;
+ Coder_State* gData = (Coder_State*)hCodec;
+ VO_MEM_OPERATOR *pMemOP;
+ pMemOP = gData->pvoMemop;
- if(hCodec)
- {
- if(gData->stream)
- {
- if(gData->stream->frame_ptr_bk)
- {
- mem_free(pMemOP, gData->stream->frame_ptr_bk, VO_INDEX_ENC_AMRWB);
- gData->stream->frame_ptr_bk = NULL;
- }
- mem_free(pMemOP, gData->stream, VO_INDEX_ENC_AMRWB);
- gData->stream = NULL;
- }
- wb_vad_exit(&(((Coder_State *) gData)->vadSt), pMemOP);
- dtx_enc_exit(&(((Coder_State *) gData)->dtx_encSt), pMemOP);
+ if(hCodec)
+ {
+ if(gData->stream)
+ {
+ if(gData->stream->frame_ptr_bk)
+ {
+ mem_free(pMemOP, gData->stream->frame_ptr_bk, VO_INDEX_ENC_AMRWB);
+ gData->stream->frame_ptr_bk = NULL;
+ }
+ mem_free(pMemOP, gData->stream, VO_INDEX_ENC_AMRWB);
+ gData->stream = NULL;
+ }
+ wb_vad_exit(&(((Coder_State *) gData)->vadSt), pMemOP);
+ dtx_enc_exit(&(((Coder_State *) gData)->dtx_encSt), pMemOP);
- mem_free(pMemOP, hCodec, VO_INDEX_ENC_AMRWB);
- hCodec = NULL;
- }
+ mem_free(pMemOP, hCodec, VO_INDEX_ENC_AMRWB);
+ hCodec = NULL;
+ }
- return VO_ERR_NONE;
+ return VO_ERR_NONE;
}
/********************************************************************************
@@ -1925,19 +1929,19 @@
********************************************************************************/
VO_S32 VO_API voGetAMRWBEncAPI(
- VO_AUDIO_CODECAPI * pEncHandle /* i/o: Codec handle pointer */
- )
+ VO_AUDIO_CODECAPI * pEncHandle /* i/o: Codec handle pointer */
+ )
{
- if(NULL == pEncHandle)
- return VO_ERR_INVALID_ARG;
- pEncHandle->Init = voAMRWB_Init;
- pEncHandle->SetInputData = voAMRWB_SetInputData;
- pEncHandle->GetOutputData = voAMRWB_GetOutputData;
- pEncHandle->SetParam = voAMRWB_SetParam;
- pEncHandle->GetParam = voAMRWB_GetParam;
- pEncHandle->Uninit = voAMRWB_Uninit;
+ if(NULL == pEncHandle)
+ return VO_ERR_INVALID_ARG;
+ pEncHandle->Init = voAMRWB_Init;
+ pEncHandle->SetInputData = voAMRWB_SetInputData;
+ pEncHandle->GetOutputData = voAMRWB_GetOutputData;
+ pEncHandle->SetParam = voAMRWB_SetParam;
+ pEncHandle->GetParam = voAMRWB_GetParam;
+ pEncHandle->Uninit = voAMRWB_Uninit;
- return VO_ERR_NONE;
+ return VO_ERR_NONE;
}
#ifdef __cplusplus
diff --git a/media/libstagefright/codecs/amrwbenc/src/voicefac.c b/media/libstagefright/codecs/amrwbenc/src/voicefac.c
index d890044..c9f48c2 100644
--- a/media/libstagefright/codecs/amrwbenc/src/voicefac.c
+++ b/media/libstagefright/codecs/amrwbenc/src/voicefac.c
@@ -26,65 +26,65 @@
#include "math_op.h"
Word16 voice_factor( /* (o) Q15 : factor (-1=unvoiced to 1=voiced) */
- Word16 exc[], /* (i) Q_exc : pitch excitation */
- Word16 Q_exc, /* (i) : exc format */
- Word16 gain_pit, /* (i) Q14 : gain of pitch */
- Word16 code[], /* (i) Q9 : Fixed codebook excitation */
- Word16 gain_code, /* (i) Q0 : gain of code */
- Word16 L_subfr /* (i) : subframe length */
- )
+ Word16 exc[], /* (i) Q_exc : pitch excitation */
+ Word16 Q_exc, /* (i) : exc format */
+ Word16 gain_pit, /* (i) Q14 : gain of pitch */
+ Word16 code[], /* (i) Q9 : Fixed codebook excitation */
+ Word16 gain_code, /* (i) Q0 : gain of code */
+ Word16 L_subfr /* (i) : subframe length */
+ )
{
- Word16 tmp, exp, ener1, exp1, ener2, exp2;
- Word32 i, L_tmp;
+ Word16 tmp, exp, ener1, exp1, ener2, exp2;
+ Word32 i, L_tmp;
#ifdef ASM_OPT /* asm optimization branch */
- ener1 = extract_h(Dot_product12_asm(exc, exc, L_subfr, &exp1));
+ ener1 = extract_h(Dot_product12_asm(exc, exc, L_subfr, &exp1));
#else
- ener1 = extract_h(Dot_product12(exc, exc, L_subfr, &exp1));
+ ener1 = extract_h(Dot_product12(exc, exc, L_subfr, &exp1));
#endif
- exp1 = exp1 - (Q_exc + Q_exc);
- L_tmp = vo_L_mult(gain_pit, gain_pit);
- exp = norm_l(L_tmp);
- tmp = extract_h(L_tmp << exp);
- ener1 = vo_mult(ener1, tmp);
- exp1 = exp1 - exp - 10; /* 10 -> gain_pit Q14 to Q9 */
+ exp1 = exp1 - (Q_exc + Q_exc);
+ L_tmp = vo_L_mult(gain_pit, gain_pit);
+ exp = norm_l(L_tmp);
+ tmp = extract_h(L_tmp << exp);
+ ener1 = vo_mult(ener1, tmp);
+ exp1 = exp1 - exp - 10; /* 10 -> gain_pit Q14 to Q9 */
#ifdef ASM_OPT /* asm optimization branch */
- ener2 = extract_h(Dot_product12_asm(code, code, L_subfr, &exp2));
+ ener2 = extract_h(Dot_product12_asm(code, code, L_subfr, &exp2));
#else
- ener2 = extract_h(Dot_product12(code, code, L_subfr, &exp2));
+ ener2 = extract_h(Dot_product12(code, code, L_subfr, &exp2));
#endif
- exp = norm_s(gain_code);
- tmp = gain_code << exp;
- tmp = vo_mult(tmp, tmp);
- ener2 = vo_mult(ener2, tmp);
- exp2 = exp2 - (exp + exp);
+ exp = norm_s(gain_code);
+ tmp = gain_code << exp;
+ tmp = vo_mult(tmp, tmp);
+ ener2 = vo_mult(ener2, tmp);
+ exp2 = exp2 - (exp + exp);
- i = exp1 - exp2;
+ i = exp1 - exp2;
- if (i >= 0)
- {
- ener1 = ener1 >> 1;
- ener2 = ener2 >> (i + 1);
- } else
- {
- ener1 = ener1 >> (1 - i);
- ener2 = ener2 >> 1;
- }
+ if (i >= 0)
+ {
+ ener1 = ener1 >> 1;
+ ener2 = ener2 >> (i + 1);
+ } else
+ {
+ ener1 = ener1 >> (1 - i);
+ ener2 = ener2 >> 1;
+ }
- tmp = vo_sub(ener1, ener2);
- ener1 = add1(add1(ener1, ener2), 1);
+ tmp = vo_sub(ener1, ener2);
+ ener1 = add1(add1(ener1, ener2), 1);
- if (tmp >= 0)
- {
- tmp = div_s(tmp, ener1);
- } else
- {
- tmp = vo_negate(div_s(vo_negate(tmp), ener1));
- }
+ if (tmp >= 0)
+ {
+ tmp = div_s(tmp, ener1);
+ } else
+ {
+ tmp = vo_negate(div_s(vo_negate(tmp), ener1));
+ }
- return (tmp);
+ return (tmp);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/wb_vad.c b/media/libstagefright/codecs/amrwbenc/src/wb_vad.c
index 2beaefd..866a69c 100644
--- a/media/libstagefright/codecs/amrwbenc/src/wb_vad.c
+++ b/media/libstagefright/codecs/amrwbenc/src/wb_vad.c
@@ -44,30 +44,30 @@
*********************************************************************************/
static Word16 ilog2( /* return: output value of the log2 */
- Word16 mant /* i: value to be converted */
- )
+ Word16 mant /* i: value to be converted */
+ )
{
- Word16 ex, ex2, res;
- Word32 i, l_temp;
+ Word16 ex, ex2, res;
+ Word32 i, l_temp;
- if (mant <= 0)
- {
- mant = 1;
- }
- ex = norm_s(mant);
- mant = mant << ex;
+ if (mant <= 0)
+ {
+ mant = 1;
+ }
+ ex = norm_s(mant);
+ mant = mant << ex;
- for (i = 0; i < 3; i++)
- mant = vo_mult(mant, mant);
- l_temp = vo_L_mult(mant, mant);
+ for (i = 0; i < 3; i++)
+ mant = vo_mult(mant, mant);
+ l_temp = vo_L_mult(mant, mant);
- ex2 = norm_l(l_temp);
- mant = extract_h(l_temp << ex2);
+ ex2 = norm_l(l_temp);
+ mant = extract_h(l_temp << ex2);
- res = (ex + 16) << 10;
- res = add1(res, (ex2 << 6));
- res = vo_sub(add1(res, 127), (mant >> 8));
- return (res);
+ res = (ex + 16) << 10;
+ res = add1(res, (ex2 << 6));
+ res = vo_sub(add1(res, 127), (mant >> 8));
+ return (res);
}
/******************************************************************************
@@ -79,23 +79,23 @@
*******************************************************************************/
static void filter5(
- Word16 * in0, /* i/o : input values; output low-pass part */
- Word16 * in1, /* i/o : input values; output high-pass part */
- Word16 data[] /* i/o : filter memory */
- )
+ Word16 * in0, /* i/o : input values; output low-pass part */
+ Word16 * in1, /* i/o : input values; output high-pass part */
+ Word16 data[] /* i/o : filter memory */
+ )
{
- Word16 temp0, temp1, temp2;
+ Word16 temp0, temp1, temp2;
- temp0 = vo_sub(*in0, vo_mult(COEFF5_1, data[0]));
- temp1 = add1(data[0], vo_mult(COEFF5_1, temp0));
- data[0] = temp0;
+ temp0 = vo_sub(*in0, vo_mult(COEFF5_1, data[0]));
+ temp1 = add1(data[0], vo_mult(COEFF5_1, temp0));
+ data[0] = temp0;
- temp0 = vo_sub(*in1, vo_mult(COEFF5_2, data[1]));
- temp2 = add1(data[1], vo_mult(COEFF5_2, temp0));
- data[1] = temp0;
+ temp0 = vo_sub(*in1, vo_mult(COEFF5_2, data[1]));
+ temp2 = add1(data[1], vo_mult(COEFF5_2, temp0));
+ data[1] = temp0;
- *in0 = extract_h((vo_L_add(temp1, temp2) << 15));
- *in1 = extract_h((vo_L_sub(temp1, temp2) << 15));
+ *in0 = extract_h((vo_L_add(temp1, temp2) << 15));
+ *in1 = extract_h((vo_L_sub(temp1, temp2) << 15));
}
/******************************************************************************
@@ -107,19 +107,19 @@
*******************************************************************************/
static void filter3(
- Word16 * in0, /* i/o : input values; output low-pass part */
- Word16 * in1, /* i/o : input values; output high-pass part */
- Word16 * data /* i/o : filter memory */
- )
+ Word16 * in0, /* i/o : input values; output low-pass part */
+ Word16 * in1, /* i/o : input values; output high-pass part */
+ Word16 * data /* i/o : filter memory */
+ )
{
- Word16 temp1, temp2;
+ Word16 temp1, temp2;
- temp1 = vo_sub(*in1, vo_mult(COEFF3, *data));
- temp2 = add1(*data, vo_mult(COEFF3, temp1));
- *data = temp1;
+ temp1 = vo_sub(*in1, vo_mult(COEFF3, *data));
+ temp2 = add1(*data, vo_mult(COEFF3, temp1));
+ *data = temp1;
- *in1 = extract_h((vo_L_sub(*in0, temp2) << 15));
- *in0 = extract_h((vo_L_add(*in0, temp2) << 15));
+ *in1 = extract_h((vo_L_sub(*in0, temp2) << 15));
+ *in0 = extract_h((vo_L_add(*in0, temp2) << 15));
}
/******************************************************************************
@@ -135,36 +135,36 @@
******************************************************************************/
static Word16 level_calculation( /* return: signal level */
- Word16 data[], /* i : signal buffer */
- Word16 * sub_level, /* i : level calculated at the end of the previous frame*/
- /* o : level of signal calculated from the last */
- /* (count2 - count1) samples */
- Word16 count1, /* i : number of samples to be counted */
- Word16 count2, /* i : number of samples to be counted */
- Word16 ind_m, /* i : step size for the index of the data buffer */
- Word16 ind_a, /* i : starting index of the data buffer */
- Word16 scale /* i : scaling for the level calculation */
- )
+ Word16 data[], /* i : signal buffer */
+ Word16 * sub_level, /* i : level calculated at the end of the previous frame*/
+ /* o : level of signal calculated from the last */
+ /* (count2 - count1) samples */
+ Word16 count1, /* i : number of samples to be counted */
+ Word16 count2, /* i : number of samples to be counted */
+ Word16 ind_m, /* i : step size for the index of the data buffer */
+ Word16 ind_a, /* i : starting index of the data buffer */
+ Word16 scale /* i : scaling for the level calculation */
+ )
{
- Word32 i, l_temp1, l_temp2;
- Word16 level;
+ Word32 i, l_temp1, l_temp2;
+ Word16 level;
- l_temp1 = 0L;
- for (i = count1; i < count2; i++)
- {
- l_temp1 += (abs_s(data[ind_m * i + ind_a])<<1);
- }
+ l_temp1 = 0L;
+ for (i = count1; i < count2; i++)
+ {
+ l_temp1 += (abs_s(data[ind_m * i + ind_a])<<1);
+ }
- l_temp2 = vo_L_add(l_temp1, L_shl(*sub_level, 16 - scale));
- *sub_level = extract_h(L_shl(l_temp1, scale));
+ l_temp2 = vo_L_add(l_temp1, L_shl(*sub_level, 16 - scale));
+ *sub_level = extract_h(L_shl(l_temp1, scale));
- for (i = 0; i < count1; i++)
- {
- l_temp2 += (abs_s(data[ind_m * i + ind_a])<<1);
- }
- level = extract_h(L_shl2(l_temp2, scale));
+ for (i = 0; i < count1; i++)
+ {
+ l_temp2 += (abs_s(data[ind_m * i + ind_a])<<1);
+ }
+ level = extract_h(L_shl2(l_temp2, scale));
- return level;
+ return level;
}
/******************************************************************************
@@ -176,75 +176,75 @@
*******************************************************************************/
static void filter_bank(
- VadVars * st, /* i/o : State struct */
- Word16 in[], /* i : input frame */
- Word16 level[] /* o : signal levels at each band */
- )
+ VadVars * st, /* i/o : State struct */
+ Word16 in[], /* i : input frame */
+ Word16 level[] /* o : signal levels at each band */
+ )
{
- Word32 i;
- Word16 tmp_buf[FRAME_LEN];
+ Word32 i;
+ Word16 tmp_buf[FRAME_LEN];
- /* shift input 1 bit down for safe scaling */
- for (i = 0; i < FRAME_LEN; i++)
- {
- tmp_buf[i] = in[i] >> 1;
- }
+ /* shift input 1 bit down for safe scaling */
+ for (i = 0; i < FRAME_LEN; i++)
+ {
+ tmp_buf[i] = in[i] >> 1;
+ }
- /* run the filter bank */
- for (i = 0; i < 128; i++)
- {
- filter5(&tmp_buf[2 * i], &tmp_buf[2 * i + 1], st->a_data5[0]);
- }
- for (i = 0; i < 64; i++)
- {
- filter5(&tmp_buf[4 * i], &tmp_buf[4 * i + 2], st->a_data5[1]);
- filter5(&tmp_buf[4 * i + 1], &tmp_buf[4 * i + 3], st->a_data5[2]);
- }
- for (i = 0; i < 32; i++)
- {
- filter5(&tmp_buf[8 * i], &tmp_buf[8 * i + 4], st->a_data5[3]);
- filter5(&tmp_buf[8 * i + 2], &tmp_buf[8 * i + 6], st->a_data5[4]);
- filter3(&tmp_buf[8 * i + 3], &tmp_buf[8 * i + 7], &st->a_data3[0]);
- }
- for (i = 0; i < 16; i++)
- {
- filter3(&tmp_buf[16 * i + 0], &tmp_buf[16 * i + 8], &st->a_data3[1]);
- filter3(&tmp_buf[16 * i + 4], &tmp_buf[16 * i + 12], &st->a_data3[2]);
- filter3(&tmp_buf[16 * i + 6], &tmp_buf[16 * i + 14], &st->a_data3[3]);
- }
+ /* run the filter bank */
+ for (i = 0; i < 128; i++)
+ {
+ filter5(&tmp_buf[2 * i], &tmp_buf[2 * i + 1], st->a_data5[0]);
+ }
+ for (i = 0; i < 64; i++)
+ {
+ filter5(&tmp_buf[4 * i], &tmp_buf[4 * i + 2], st->a_data5[1]);
+ filter5(&tmp_buf[4 * i + 1], &tmp_buf[4 * i + 3], st->a_data5[2]);
+ }
+ for (i = 0; i < 32; i++)
+ {
+ filter5(&tmp_buf[8 * i], &tmp_buf[8 * i + 4], st->a_data5[3]);
+ filter5(&tmp_buf[8 * i + 2], &tmp_buf[8 * i + 6], st->a_data5[4]);
+ filter3(&tmp_buf[8 * i + 3], &tmp_buf[8 * i + 7], &st->a_data3[0]);
+ }
+ for (i = 0; i < 16; i++)
+ {
+ filter3(&tmp_buf[16 * i + 0], &tmp_buf[16 * i + 8], &st->a_data3[1]);
+ filter3(&tmp_buf[16 * i + 4], &tmp_buf[16 * i + 12], &st->a_data3[2]);
+ filter3(&tmp_buf[16 * i + 6], &tmp_buf[16 * i + 14], &st->a_data3[3]);
+ }
- for (i = 0; i < 8; i++)
- {
- filter3(&tmp_buf[32 * i + 0], &tmp_buf[32 * i + 16], &st->a_data3[4]);
- filter3(&tmp_buf[32 * i + 8], &tmp_buf[32 * i + 24], &st->a_data3[5]);
- }
+ for (i = 0; i < 8; i++)
+ {
+ filter3(&tmp_buf[32 * i + 0], &tmp_buf[32 * i + 16], &st->a_data3[4]);
+ filter3(&tmp_buf[32 * i + 8], &tmp_buf[32 * i + 24], &st->a_data3[5]);
+ }
- /* calculate levels in each frequency band */
+ /* calculate levels in each frequency band */
- /* 4800 - 6400 Hz */
- level[11] = level_calculation(tmp_buf, &st->sub_level[11], 16, 64, 4, 1, 14);
- /* 4000 - 4800 Hz */
- level[10] = level_calculation(tmp_buf, &st->sub_level[10], 8, 32, 8, 7, 15);
- /* 3200 - 4000 Hz */
- level[9] = level_calculation(tmp_buf, &st->sub_level[9],8, 32, 8, 3, 15);
- /* 2400 - 3200 Hz */
- level[8] = level_calculation(tmp_buf, &st->sub_level[8],8, 32, 8, 2, 15);
- /* 2000 - 2400 Hz */
- level[7] = level_calculation(tmp_buf, &st->sub_level[7],4, 16, 16, 14, 16);
- /* 1600 - 2000 Hz */
- level[6] = level_calculation(tmp_buf, &st->sub_level[6],4, 16, 16, 6, 16);
- /* 1200 - 1600 Hz */
- level[5] = level_calculation(tmp_buf, &st->sub_level[5],4, 16, 16, 4, 16);
- /* 800 - 1200 Hz */
- level[4] = level_calculation(tmp_buf, &st->sub_level[4],4, 16, 16, 12, 16);
- /* 600 - 800 Hz */
- level[3] = level_calculation(tmp_buf, &st->sub_level[3],2, 8, 32, 8, 17);
- /* 400 - 600 Hz */
- level[2] = level_calculation(tmp_buf, &st->sub_level[2],2, 8, 32, 24, 17);
- /* 200 - 400 Hz */
- level[1] = level_calculation(tmp_buf, &st->sub_level[1],2, 8, 32, 16, 17);
- /* 0 - 200 Hz */
- level[0] = level_calculation(tmp_buf, &st->sub_level[0],2, 8, 32, 0, 17);
+ /* 4800 - 6400 Hz */
+ level[11] = level_calculation(tmp_buf, &st->sub_level[11], 16, 64, 4, 1, 14);
+ /* 4000 - 4800 Hz */
+ level[10] = level_calculation(tmp_buf, &st->sub_level[10], 8, 32, 8, 7, 15);
+ /* 3200 - 4000 Hz */
+ level[9] = level_calculation(tmp_buf, &st->sub_level[9],8, 32, 8, 3, 15);
+ /* 2400 - 3200 Hz */
+ level[8] = level_calculation(tmp_buf, &st->sub_level[8],8, 32, 8, 2, 15);
+ /* 2000 - 2400 Hz */
+ level[7] = level_calculation(tmp_buf, &st->sub_level[7],4, 16, 16, 14, 16);
+ /* 1600 - 2000 Hz */
+ level[6] = level_calculation(tmp_buf, &st->sub_level[6],4, 16, 16, 6, 16);
+ /* 1200 - 1600 Hz */
+ level[5] = level_calculation(tmp_buf, &st->sub_level[5],4, 16, 16, 4, 16);
+ /* 800 - 1200 Hz */
+ level[4] = level_calculation(tmp_buf, &st->sub_level[4],4, 16, 16, 12, 16);
+ /* 600 - 800 Hz */
+ level[3] = level_calculation(tmp_buf, &st->sub_level[3],2, 8, 32, 8, 17);
+ /* 400 - 600 Hz */
+ level[2] = level_calculation(tmp_buf, &st->sub_level[2],2, 8, 32, 24, 17);
+ /* 200 - 400 Hz */
+ level[1] = level_calculation(tmp_buf, &st->sub_level[1],2, 8, 32, 16, 17);
+ /* 0 - 200 Hz */
+ level[0] = level_calculation(tmp_buf, &st->sub_level[0],2, 8, 32, 0, 17);
}
/******************************************************************************
@@ -255,86 +255,86 @@
*******************************************************************************/
static void update_cntrl(
- VadVars * st, /* i/o : State structure */
- Word16 level[] /* i : sub-band levels of the input frame */
- )
+ VadVars * st, /* i/o : State structure */
+ Word16 level[] /* i : sub-band levels of the input frame */
+ )
{
- Word32 i;
- Word16 num, temp, stat_rat, exp, denom;
- Word16 alpha;
+ Word32 i;
+ Word16 num, temp, stat_rat, exp, denom;
+ Word16 alpha;
- /* if a tone has been detected for a while, initialize stat_count */
- if (sub((Word16) (st->tone_flag & 0x7c00), 0x7c00) == 0)
- {
- st->stat_count = STAT_COUNT;
- } else
- {
- /* if 8 last vad-decisions have been "0", reinitialize stat_count */
- if ((st->vadreg & 0x7f80) == 0)
- {
- st->stat_count = STAT_COUNT;
- } else
- {
- stat_rat = 0;
- for (i = 0; i < COMPLEN; i++)
- {
- if(level[i] > st->ave_level[i])
- {
- num = level[i];
- denom = st->ave_level[i];
- } else
- {
- num = st->ave_level[i];
- denom = level[i];
- }
- /* Limit nimimum value of num and denom to STAT_THR_LEVEL */
- if(num < STAT_THR_LEVEL)
- {
- num = STAT_THR_LEVEL;
- }
- if(denom < STAT_THR_LEVEL)
- {
- denom = STAT_THR_LEVEL;
- }
- exp = norm_s(denom);
- denom = denom << exp;
+ /* if a tone has been detected for a while, initialize stat_count */
+ if (sub((Word16) (st->tone_flag & 0x7c00), 0x7c00) == 0)
+ {
+ st->stat_count = STAT_COUNT;
+ } else
+ {
+ /* if 8 last vad-decisions have been "0", reinitialize stat_count */
+ if ((st->vadreg & 0x7f80) == 0)
+ {
+ st->stat_count = STAT_COUNT;
+ } else
+ {
+ stat_rat = 0;
+ for (i = 0; i < COMPLEN; i++)
+ {
+ if(level[i] > st->ave_level[i])
+ {
+ num = level[i];
+ denom = st->ave_level[i];
+ } else
+ {
+ num = st->ave_level[i];
+ denom = level[i];
+ }
+ /* Limit nimimum value of num and denom to STAT_THR_LEVEL */
+ if(num < STAT_THR_LEVEL)
+ {
+ num = STAT_THR_LEVEL;
+ }
+ if(denom < STAT_THR_LEVEL)
+ {
+ denom = STAT_THR_LEVEL;
+ }
+ exp = norm_s(denom);
+ denom = denom << exp;
- /* stat_rat = num/denom * 64 */
- temp = div_s(num >> 1, denom);
- stat_rat = add1(stat_rat, shr(temp, (8 - exp)));
- }
+ /* stat_rat = num/denom * 64 */
+ temp = div_s(num >> 1, denom);
+ stat_rat = add1(stat_rat, shr(temp, (8 - exp)));
+ }
- /* compare stat_rat with a threshold and update stat_count */
- if(stat_rat > STAT_THR)
- {
- st->stat_count = STAT_COUNT;
- } else
- {
- if ((st->vadreg & 0x4000) != 0)
- {
+ /* compare stat_rat with a threshold and update stat_count */
+ if(stat_rat > STAT_THR)
+ {
+ st->stat_count = STAT_COUNT;
+ } else
+ {
+ if ((st->vadreg & 0x4000) != 0)
+ {
- if (st->stat_count != 0)
- {
- st->stat_count = st->stat_count - 1;
- }
- }
- }
- }
- }
+ if (st->stat_count != 0)
+ {
+ st->stat_count = st->stat_count - 1;
+ }
+ }
+ }
+ }
+ }
- /* Update average amplitude estimate for stationarity estimation */
- alpha = ALPHA4;
- if(st->stat_count == STAT_COUNT)
- {
- alpha = 32767;
- } else if ((st->vadreg & 0x4000) == 0)
- {
- alpha = ALPHA5;
- }
- for (i = 0; i < COMPLEN; i++)
- {
- st->ave_level[i] = add1(st->ave_level[i], vo_mult_r(alpha, vo_sub(level[i], st->ave_level[i])));
- }
+ /* Update average amplitude estimate for stationarity estimation */
+ alpha = ALPHA4;
+ if(st->stat_count == STAT_COUNT)
+ {
+ alpha = 32767;
+ } else if ((st->vadreg & 0x4000) == 0)
+ {
+ alpha = ALPHA5;
+ }
+ for (i = 0; i < COMPLEN; i++)
+ {
+ st->ave_level[i] = add1(st->ave_level[i], vo_mult_r(alpha, vo_sub(level[i], st->ave_level[i])));
+ }
}
/******************************************************************************
@@ -345,38 +345,38 @@
*******************************************************************************/
static Word16 hangover_addition( /* return: VAD_flag indicating final VAD decision */
- VadVars * st, /* i/o : State structure */
- Word16 low_power, /* i : flag power of the input frame */
- Word16 hang_len, /* i : hangover length */
- Word16 burst_len /* i : minimum burst length for hangover addition */
- )
+ VadVars * st, /* i/o : State structure */
+ Word16 low_power, /* i : flag power of the input frame */
+ Word16 hang_len, /* i : hangover length */
+ Word16 burst_len /* i : minimum burst length for hangover addition */
+ )
{
- /* if the input power (pow_sum) is lower than a threshold, clear counters and set VAD_flag to "0" */
- if (low_power != 0)
- {
- st->burst_count = 0;
- st->hang_count = 0;
- return 0;
- }
- /* update the counters (hang_count, burst_count) */
- if ((st->vadreg & 0x4000) != 0)
- {
- st->burst_count = st->burst_count + 1;
- if(st->burst_count >= burst_len)
- {
- st->hang_count = hang_len;
- }
- return 1;
- } else
- {
- st->burst_count = 0;
- if (st->hang_count > 0)
- {
- st->hang_count = st->hang_count - 1;
- return 1;
- }
- }
- return 0;
+ /* if the input power (pow_sum) is lower than a threshold, clear counters and set VAD_flag to "0" */
+ if (low_power != 0)
+ {
+ st->burst_count = 0;
+ st->hang_count = 0;
+ return 0;
+ }
+ /* update the counters (hang_count, burst_count) */
+ if ((st->vadreg & 0x4000) != 0)
+ {
+ st->burst_count = st->burst_count + 1;
+ if(st->burst_count >= burst_len)
+ {
+ st->hang_count = hang_len;
+ }
+ return 1;
+ } else
+ {
+ st->burst_count = 0;
+ if (st->hang_count > 0)
+ {
+ st->hang_count = st->hang_count - 1;
+ return 1;
+ }
+ }
+ return 0;
}
/******************************************************************************
@@ -387,66 +387,66 @@
*******************************************************************************/
static void noise_estimate_update(
- VadVars * st, /* i/o : State structure */
- Word16 level[] /* i : sub-band levels of the input frame */
- )
+ VadVars * st, /* i/o : State structure */
+ Word16 level[] /* i : sub-band levels of the input frame */
+ )
{
- Word32 i;
- Word16 alpha_up, alpha_down, bckr_add = 2;
+ Word32 i;
+ Word16 alpha_up, alpha_down, bckr_add = 2;
- /* Control update of bckr_est[] */
- update_cntrl(st, level);
+ /* Control update of bckr_est[] */
+ update_cntrl(st, level);
- /* Choose update speed */
- if ((0x7800 & st->vadreg) == 0)
- {
- alpha_up = ALPHA_UP1;
- alpha_down = ALPHA_DOWN1;
- } else
- {
- if (st->stat_count == 0)
- {
- alpha_up = ALPHA_UP2;
- alpha_down = ALPHA_DOWN2;
- } else
- {
- alpha_up = 0;
- alpha_down = ALPHA3;
- bckr_add = 0;
- }
- }
+ /* Choose update speed */
+ if ((0x7800 & st->vadreg) == 0)
+ {
+ alpha_up = ALPHA_UP1;
+ alpha_down = ALPHA_DOWN1;
+ } else
+ {
+ if (st->stat_count == 0)
+ {
+ alpha_up = ALPHA_UP2;
+ alpha_down = ALPHA_DOWN2;
+ } else
+ {
+ alpha_up = 0;
+ alpha_down = ALPHA3;
+ bckr_add = 0;
+ }
+ }
- /* Update noise estimate (bckr_est) */
- for (i = 0; i < COMPLEN; i++)
- {
- Word16 temp;
- temp = (st->old_level[i] - st->bckr_est[i]);
+ /* Update noise estimate (bckr_est) */
+ for (i = 0; i < COMPLEN; i++)
+ {
+ Word16 temp;
+ temp = (st->old_level[i] - st->bckr_est[i]);
- if (temp < 0)
- { /* update downwards */
- st->bckr_est[i] = add1(-2, add(st->bckr_est[i],vo_mult_r(alpha_down, temp)));
- /* limit minimum value of the noise estimate to NOISE_MIN */
- if(st->bckr_est[i] < NOISE_MIN)
- {
- st->bckr_est[i] = NOISE_MIN;
- }
- } else
- { /* update upwards */
- st->bckr_est[i] = add1(bckr_add, add1(st->bckr_est[i],vo_mult_r(alpha_up, temp)));
+ if (temp < 0)
+ { /* update downwards */
+ st->bckr_est[i] = add1(-2, add(st->bckr_est[i],vo_mult_r(alpha_down, temp)));
+ /* limit minimum value of the noise estimate to NOISE_MIN */
+ if(st->bckr_est[i] < NOISE_MIN)
+ {
+ st->bckr_est[i] = NOISE_MIN;
+ }
+ } else
+ { /* update upwards */
+ st->bckr_est[i] = add1(bckr_add, add1(st->bckr_est[i],vo_mult_r(alpha_up, temp)));
- /* limit maximum value of the noise estimate to NOISE_MAX */
- if(st->bckr_est[i] > NOISE_MAX)
- {
- st->bckr_est[i] = NOISE_MAX;
- }
- }
- }
+ /* limit maximum value of the noise estimate to NOISE_MAX */
+ if(st->bckr_est[i] > NOISE_MAX)
+ {
+ st->bckr_est[i] = NOISE_MAX;
+ }
+ }
+ }
- /* Update signal levels of the previous frame (old_level) */
- for (i = 0; i < COMPLEN; i++)
- {
- st->old_level[i] = level[i];
- }
+ /* Update signal levels of the previous frame (old_level) */
+ for (i = 0; i < COMPLEN; i++)
+ {
+ st->old_level[i] = level[i];
+ }
}
/******************************************************************************
@@ -457,100 +457,100 @@
*******************************************************************************/
static Word16 vad_decision( /* return value : VAD_flag */
- VadVars * st, /* i/o : State structure */
- Word16 level[COMPLEN], /* i : sub-band levels of the input frame */
- Word32 pow_sum /* i : power of the input frame */
- )
+ VadVars * st, /* i/o : State structure */
+ Word16 level[COMPLEN], /* i : sub-band levels of the input frame */
+ Word32 pow_sum /* i : power of the input frame */
+ )
{
- Word32 i;
- Word32 L_snr_sum;
- Word32 L_temp;
- Word16 vad_thr, temp, noise_level;
- Word16 low_power_flag;
- Word16 hang_len, burst_len;
- Word16 ilog2_speech_level, ilog2_noise_level;
- Word16 temp2;
+ Word32 i;
+ Word32 L_snr_sum;
+ Word32 L_temp;
+ Word16 vad_thr, temp, noise_level;
+ Word16 low_power_flag;
+ Word16 hang_len, burst_len;
+ Word16 ilog2_speech_level, ilog2_noise_level;
+ Word16 temp2;
- /* Calculate squared sum of the input levels (level) divided by the background noise components
- * (bckr_est). */
- L_snr_sum = 0;
- for (i = 0; i < COMPLEN; i++)
- {
- Word16 exp;
+ /* Calculate squared sum of the input levels (level) divided by the background noise components
+ * (bckr_est). */
+ L_snr_sum = 0;
+ for (i = 0; i < COMPLEN; i++)
+ {
+ Word16 exp;
- exp = norm_s(st->bckr_est[i]);
- temp = (st->bckr_est[i] << exp);
- temp = div_s((level[i] >> 1), temp);
- temp = shl(temp, (exp - (UNIRSHFT - 1)));
- L_snr_sum = L_mac(L_snr_sum, temp, temp);
- }
+ exp = norm_s(st->bckr_est[i]);
+ temp = (st->bckr_est[i] << exp);
+ temp = div_s((level[i] >> 1), temp);
+ temp = shl(temp, (exp - (UNIRSHFT - 1)));
+ L_snr_sum = L_mac(L_snr_sum, temp, temp);
+ }
- /* Calculate average level of estimated background noise */
- L_temp = 0;
- for (i = 1; i < COMPLEN; i++) /* ignore lowest band */
- {
- L_temp = vo_L_add(L_temp, st->bckr_est[i]);
- }
+ /* Calculate average level of estimated background noise */
+ L_temp = 0;
+ for (i = 1; i < COMPLEN; i++) /* ignore lowest band */
+ {
+ L_temp = vo_L_add(L_temp, st->bckr_est[i]);
+ }
- noise_level = extract_h((L_temp << 12));
- /* if SNR is lower than a threshold (MIN_SPEECH_SNR), and increase speech_level */
- temp = vo_mult(noise_level, MIN_SPEECH_SNR) << 3;
+ noise_level = extract_h((L_temp << 12));
+ /* if SNR is lower than a threshold (MIN_SPEECH_SNR), and increase speech_level */
+ temp = vo_mult(noise_level, MIN_SPEECH_SNR) << 3;
- if(st->speech_level < temp)
- {
- st->speech_level = temp;
- }
- ilog2_noise_level = ilog2(noise_level);
+ if(st->speech_level < temp)
+ {
+ st->speech_level = temp;
+ }
+ ilog2_noise_level = ilog2(noise_level);
- /* If SNR is very poor, speech_level is probably corrupted by noise level. This is correctred by
- * subtracting MIN_SPEECH_SNR*noise_level from speech level */
- ilog2_speech_level = ilog2(st->speech_level - temp);
+ /* If SNR is very poor, speech_level is probably corrupted by noise level. This is correctred by
+ * subtracting MIN_SPEECH_SNR*noise_level from speech level */
+ ilog2_speech_level = ilog2(st->speech_level - temp);
- temp = add1(vo_mult(NO_SLOPE, (ilog2_noise_level - NO_P1)), THR_HIGH);
+ temp = add1(vo_mult(NO_SLOPE, (ilog2_noise_level - NO_P1)), THR_HIGH);
- temp2 = add1(SP_CH_MIN, vo_mult(SP_SLOPE, (ilog2_speech_level - SP_P1)));
- if (temp2 < SP_CH_MIN)
- {
- temp2 = SP_CH_MIN;
- }
- if (temp2 > SP_CH_MAX)
- {
- temp2 = SP_CH_MAX;
- }
- vad_thr = temp + temp2;
+ temp2 = add1(SP_CH_MIN, vo_mult(SP_SLOPE, (ilog2_speech_level - SP_P1)));
+ if (temp2 < SP_CH_MIN)
+ {
+ temp2 = SP_CH_MIN;
+ }
+ if (temp2 > SP_CH_MAX)
+ {
+ temp2 = SP_CH_MAX;
+ }
+ vad_thr = temp + temp2;
- if(vad_thr < THR_MIN)
- {
- vad_thr = THR_MIN;
- }
- /* Shift VAD decision register */
- st->vadreg = (st->vadreg >> 1);
+ if(vad_thr < THR_MIN)
+ {
+ vad_thr = THR_MIN;
+ }
+ /* Shift VAD decision register */
+ st->vadreg = (st->vadreg >> 1);
- /* Make intermediate VAD decision */
- if(L_snr_sum > vo_L_mult(vad_thr, (512 * COMPLEN)))
- {
- st->vadreg = (Word16) (st->vadreg | 0x4000);
- }
- /* check if the input power (pow_sum) is lower than a threshold" */
- if(pow_sum < VAD_POW_LOW)
- {
- low_power_flag = 1;
- } else
- {
- low_power_flag = 0;
- }
- /* Update background noise estimates */
- noise_estimate_update(st, level);
+ /* Make intermediate VAD decision */
+ if(L_snr_sum > vo_L_mult(vad_thr, (512 * COMPLEN)))
+ {
+ st->vadreg = (Word16) (st->vadreg | 0x4000);
+ }
+ /* check if the input power (pow_sum) is lower than a threshold" */
+ if(pow_sum < VAD_POW_LOW)
+ {
+ low_power_flag = 1;
+ } else
+ {
+ low_power_flag = 0;
+ }
+ /* Update background noise estimates */
+ noise_estimate_update(st, level);
- /* Calculate values for hang_len and burst_len based on vad_thr */
- hang_len = add1(vo_mult(HANG_SLOPE, (vad_thr - HANG_P1)), HANG_HIGH);
- if(hang_len < HANG_LOW)
- {
- hang_len = HANG_LOW;
- }
- burst_len = add1(vo_mult(BURST_SLOPE, (vad_thr - BURST_P1)), BURST_HIGH);
+ /* Calculate values for hang_len and burst_len based on vad_thr */
+ hang_len = add1(vo_mult(HANG_SLOPE, (vad_thr - HANG_P1)), HANG_HIGH);
+ if(hang_len < HANG_LOW)
+ {
+ hang_len = HANG_LOW;
+ }
+ burst_len = add1(vo_mult(BURST_SLOPE, (vad_thr - BURST_P1)), BURST_HIGH);
- return (hangover_addition(st, low_power_flag, hang_len, burst_len));
+ return (hangover_addition(st, low_power_flag, hang_len, burst_len));
}
/******************************************************************************
@@ -566,54 +566,54 @@
*******************************************************************************/
static void Estimate_Speech(
- VadVars * st, /* i/o : State structure */
- Word16 in_level /* level of the input frame */
- )
+ VadVars * st, /* i/o : State structure */
+ Word16 in_level /* level of the input frame */
+ )
{
- Word16 alpha;
+ Word16 alpha;
- /* if the required activity count cannot be achieved, reset counters */
- if((st->sp_est_cnt - st->sp_max_cnt) > (SP_EST_COUNT - SP_ACTIVITY_COUNT))
- {
- st->sp_est_cnt = 0;
- st->sp_max = 0;
- st->sp_max_cnt = 0;
- }
- st->sp_est_cnt += 1;
+ /* if the required activity count cannot be achieved, reset counters */
+ if((st->sp_est_cnt - st->sp_max_cnt) > (SP_EST_COUNT - SP_ACTIVITY_COUNT))
+ {
+ st->sp_est_cnt = 0;
+ st->sp_max = 0;
+ st->sp_max_cnt = 0;
+ }
+ st->sp_est_cnt += 1;
- if (((st->vadreg & 0x4000)||(in_level > st->speech_level)) && (in_level > MIN_SPEECH_LEVEL1))
- {
- /* update sp_max */
- if(in_level > st->sp_max)
- {
- st->sp_max = in_level;
- }
- st->sp_max_cnt += 1;
+ if (((st->vadreg & 0x4000)||(in_level > st->speech_level)) && (in_level > MIN_SPEECH_LEVEL1))
+ {
+ /* update sp_max */
+ if(in_level > st->sp_max)
+ {
+ st->sp_max = in_level;
+ }
+ st->sp_max_cnt += 1;
- if(st->sp_max_cnt >= SP_ACTIVITY_COUNT)
- {
- Word16 tmp;
- /* update speech estimate */
- tmp = (st->sp_max >> 1); /* scale to get "average" speech level */
+ if(st->sp_max_cnt >= SP_ACTIVITY_COUNT)
+ {
+ Word16 tmp;
+ /* update speech estimate */
+ tmp = (st->sp_max >> 1); /* scale to get "average" speech level */
- /* select update speed */
- if(tmp > st->speech_level)
- {
- alpha = ALPHA_SP_UP;
- } else
- {
- alpha = ALPHA_SP_DOWN;
- }
- if(tmp > MIN_SPEECH_LEVEL2)
- {
- st->speech_level = add1(st->speech_level, vo_mult_r(alpha, vo_sub(tmp, st->speech_level)));
- }
- /* clear all counters used for speech estimation */
- st->sp_max = 0;
- st->sp_max_cnt = 0;
- st->sp_est_cnt = 0;
- }
- }
+ /* select update speed */
+ if(tmp > st->speech_level)
+ {
+ alpha = ALPHA_SP_UP;
+ } else
+ {
+ alpha = ALPHA_SP_DOWN;
+ }
+ if(tmp > MIN_SPEECH_LEVEL2)
+ {
+ st->speech_level = add1(st->speech_level, vo_mult_r(alpha, vo_sub(tmp, st->speech_level)));
+ }
+ /* clear all counters used for speech estimation */
+ st->sp_max = 0;
+ st->sp_max_cnt = 0;
+ st->sp_est_cnt = 0;
+ }
+ }
}
/******************************************************************************
@@ -624,30 +624,30 @@
*******************************************************************************/
Word16 wb_vad_init( /* return: non-zero with error, zero for ok. */
- VadVars ** state, /* i/o : State structure */
- VO_MEM_OPERATOR *pMemOP
- )
+ VadVars ** state, /* i/o : State structure */
+ VO_MEM_OPERATOR *pMemOP
+ )
{
- VadVars *s;
+ VadVars *s;
- if (state == (VadVars **) NULL)
- {
- fprintf(stderr, "vad_init: invalid parameter\n");
- return -1;
- }
- *state = NULL;
+ if (state == (VadVars **) NULL)
+ {
+ fprintf(stderr, "vad_init: invalid parameter\n");
+ return -1;
+ }
+ *state = NULL;
- /* allocate memory */
- if ((s = (VadVars *) mem_malloc(pMemOP, sizeof(VadVars), 32, VO_INDEX_ENC_AMRWB)) == NULL)
- {
- fprintf(stderr, "vad_init: can not malloc state structure\n");
- return -1;
- }
- wb_vad_reset(s);
+ /* allocate memory */
+ if ((s = (VadVars *) mem_malloc(pMemOP, sizeof(VadVars), 32, VO_INDEX_ENC_AMRWB)) == NULL)
+ {
+ fprintf(stderr, "vad_init: can not malloc state structure\n");
+ return -1;
+ }
+ wb_vad_reset(s);
- *state = s;
+ *state = s;
- return 0;
+ return 0;
}
/******************************************************************************
@@ -658,51 +658,51 @@
*******************************************************************************/
Word16 wb_vad_reset( /* return: non-zero with error, zero for ok. */
- VadVars * state /* i/o : State structure */
- )
+ VadVars * state /* i/o : State structure */
+ )
{
- Word32 i, j;
+ Word32 i, j;
- if (state == (VadVars *) NULL)
- {
- fprintf(stderr, "vad_reset: invalid parameter\n");
- return -1;
- }
- state->tone_flag = 0;
- state->vadreg = 0;
- state->hang_count = 0;
- state->burst_count = 0;
- state->hang_count = 0;
+ if (state == (VadVars *) NULL)
+ {
+ fprintf(stderr, "vad_reset: invalid parameter\n");
+ return -1;
+ }
+ state->tone_flag = 0;
+ state->vadreg = 0;
+ state->hang_count = 0;
+ state->burst_count = 0;
+ state->hang_count = 0;
- /* initialize memory used by the filter bank */
- for (i = 0; i < F_5TH_CNT; i++)
- {
- for (j = 0; j < 2; j++)
- {
- state->a_data5[i][j] = 0;
- }
- }
+ /* initialize memory used by the filter bank */
+ for (i = 0; i < F_5TH_CNT; i++)
+ {
+ for (j = 0; j < 2; j++)
+ {
+ state->a_data5[i][j] = 0;
+ }
+ }
- for (i = 0; i < F_3TH_CNT; i++)
- {
- state->a_data3[i] = 0;
- }
+ for (i = 0; i < F_3TH_CNT; i++)
+ {
+ state->a_data3[i] = 0;
+ }
- /* initialize the rest of the memory */
- for (i = 0; i < COMPLEN; i++)
- {
- state->bckr_est[i] = NOISE_INIT;
- state->old_level[i] = NOISE_INIT;
- state->ave_level[i] = NOISE_INIT;
- state->sub_level[i] = 0;
- }
+ /* initialize the rest of the memory */
+ for (i = 0; i < COMPLEN; i++)
+ {
+ state->bckr_est[i] = NOISE_INIT;
+ state->old_level[i] = NOISE_INIT;
+ state->ave_level[i] = NOISE_INIT;
+ state->sub_level[i] = 0;
+ }
- state->sp_est_cnt = 0;
- state->sp_max = 0;
- state->sp_max_cnt = 0;
- state->speech_level = SPEECH_LEVEL_INIT;
- state->prev_pow_sum = 0;
- return 0;
+ state->sp_est_cnt = 0;
+ state->sp_max = 0;
+ state->sp_max_cnt = 0;
+ state->speech_level = SPEECH_LEVEL_INIT;
+ state->prev_pow_sum = 0;
+ return 0;
}
/******************************************************************************
@@ -713,16 +713,16 @@
*******************************************************************************/
void wb_vad_exit(
- VadVars ** state, /* i/o : State structure */
- VO_MEM_OPERATOR *pMemOP
- )
+ VadVars ** state, /* i/o : State structure */
+ VO_MEM_OPERATOR *pMemOP
+ )
{
- if (state == NULL || *state == NULL)
- return;
- /* deallocate memory */
- mem_free(pMemOP, *state, VO_INDEX_ENC_AMRWB);
- *state = NULL;
- return;
+ if (state == NULL || *state == NULL)
+ return;
+ /* deallocate memory */
+ mem_free(pMemOP, *state, VO_INDEX_ENC_AMRWB);
+ *state = NULL;
+ return;
}
/******************************************************************************
@@ -735,18 +735,18 @@
*******************************************************************************/
void wb_vad_tone_detection(
- VadVars * st, /* i/o : State struct */
- Word16 p_gain /* pitch gain */
- )
+ VadVars * st, /* i/o : State struct */
+ Word16 p_gain /* pitch gain */
+ )
{
- /* update tone flag */
- st->tone_flag = (st->tone_flag >> 1);
+ /* update tone flag */
+ st->tone_flag = (st->tone_flag >> 1);
- /* if (pitch_gain > TONE_THR) set tone flag */
- if (p_gain > TONE_THR)
- {
- st->tone_flag = (Word16) (st->tone_flag | 0x4000);
- }
+ /* if (pitch_gain > TONE_THR) set tone flag */
+ if (p_gain > TONE_THR)
+ {
+ st->tone_flag = (Word16) (st->tone_flag | 0x4000);
+ }
}
/******************************************************************************
@@ -757,50 +757,50 @@
*******************************************************************************/
Word16 wb_vad( /* Return value : VAD Decision, 1 = speech, 0 = noise */
- VadVars * st, /* i/o : State structure */
- Word16 in_buf[] /* i : samples of the input frame */
- )
+ VadVars * st, /* i/o : State structure */
+ Word16 in_buf[] /* i : samples of the input frame */
+ )
{
- Word16 level[COMPLEN];
- Word32 i;
- Word16 VAD_flag, temp;
- Word32 L_temp, pow_sum;
+ Word16 level[COMPLEN];
+ Word32 i;
+ Word16 VAD_flag, temp;
+ Word32 L_temp, pow_sum;
- /* Calculate power of the input frame. */
- L_temp = 0L;
- for (i = 0; i < FRAME_LEN; i++)
- {
- L_temp = L_mac(L_temp, in_buf[i], in_buf[i]);
- }
+ /* Calculate power of the input frame. */
+ L_temp = 0L;
+ for (i = 0; i < FRAME_LEN; i++)
+ {
+ L_temp = L_mac(L_temp, in_buf[i], in_buf[i]);
+ }
- /* pow_sum = power of current frame and previous frame */
- pow_sum = L_add(L_temp, st->prev_pow_sum);
+ /* pow_sum = power of current frame and previous frame */
+ pow_sum = L_add(L_temp, st->prev_pow_sum);
- /* save power of current frame for next call */
- st->prev_pow_sum = L_temp;
+ /* save power of current frame for next call */
+ st->prev_pow_sum = L_temp;
- /* If input power is very low, clear tone flag */
- if (pow_sum < POW_TONE_THR)
- {
- st->tone_flag = (Word16) (st->tone_flag & 0x1fff);
- }
- /* Run the filter bank and calculate signal levels at each band */
- filter_bank(st, in_buf, level);
+ /* If input power is very low, clear tone flag */
+ if (pow_sum < POW_TONE_THR)
+ {
+ st->tone_flag = (Word16) (st->tone_flag & 0x1fff);
+ }
+ /* Run the filter bank and calculate signal levels at each band */
+ filter_bank(st, in_buf, level);
- /* compute VAD decision */
- VAD_flag = vad_decision(st, level, pow_sum);
+ /* compute VAD decision */
+ VAD_flag = vad_decision(st, level, pow_sum);
- /* Calculate input level */
- L_temp = 0;
- for (i = 1; i < COMPLEN; i++) /* ignore lowest band */
- {
- L_temp = vo_L_add(L_temp, level[i]);
- }
+ /* Calculate input level */
+ L_temp = 0;
+ for (i = 1; i < COMPLEN; i++) /* ignore lowest band */
+ {
+ L_temp = vo_L_add(L_temp, level[i]);
+ }
- temp = extract_h(L_temp << 12);
+ temp = extract_h(L_temp << 12);
- Estimate_Speech(st, temp); /* Estimate speech level */
- return (VAD_flag);
+ Estimate_Speech(st, temp); /* Estimate speech level */
+ return (VAD_flag);
}
diff --git a/media/libstagefright/codecs/amrwbenc/src/weight_a.c b/media/libstagefright/codecs/amrwbenc/src/weight_a.c
index a02b48d..23b774e 100644
--- a/media/libstagefright/codecs/amrwbenc/src/weight_a.c
+++ b/media/libstagefright/codecs/amrwbenc/src/weight_a.c
@@ -18,7 +18,7 @@
* File: weight_a.c *
* *
* Description:Weighting of LPC coefficients *
-* ap[i] = a[i] * (gamma ** i) *
+* ap[i] = a[i] * (gamma ** i) *
* *
************************************************************************/
@@ -26,22 +26,22 @@
#include "basic_op.h"
void Weight_a(
- Word16 a[], /* (i) Q12 : a[m+1] LPC coefficients */
- Word16 ap[], /* (o) Q12 : Spectral expanded LPC coefficients */
- Word16 gamma, /* (i) Q15 : Spectral expansion factor. */
- Word16 m /* (i) : LPC order. */
- )
+ Word16 a[], /* (i) Q12 : a[m+1] LPC coefficients */
+ Word16 ap[], /* (o) Q12 : Spectral expanded LPC coefficients */
+ Word16 gamma, /* (i) Q15 : Spectral expansion factor. */
+ Word16 m /* (i) : LPC order. */
+ )
{
- Word32 num = m - 1, fac;
- *ap++ = *a++;
- fac = gamma;
- do{
- *ap++ =(Word16)(((vo_L_mult((*a++), fac)) + 0x8000) >> 16);
- fac = (vo_L_mult(fac, gamma) + 0x8000) >> 16;
- }while(--num != 0);
+ Word32 num = m - 1, fac;
+ *ap++ = *a++;
+ fac = gamma;
+ do{
+ *ap++ =(Word16)(((vo_L_mult((*a++), fac)) + 0x8000) >> 16);
+ fac = (vo_L_mult(fac, gamma) + 0x8000) >> 16;
+ }while(--num != 0);
- *ap++ = (Word16)(((vo_L_mult((*a++), fac)) + 0x8000) >> 16);
- return;
+ *ap++ = (Word16)(((vo_L_mult((*a++), fac)) + 0x8000) >> 16);
+ return;
}
diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
index 410f9d0..0f28e8d 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
@@ -106,24 +106,24 @@
status_t SoftVPXEncoder::initEncoder() {
vpx_codec_err_t codec_return;
+ status_t result = UNKNOWN_ERROR;
- mCodecContext = new vpx_codec_ctx_t;
- mCodecConfiguration = new vpx_codec_enc_cfg_t;
mCodecInterface = vpx_codec_vp8_cx();
-
if (mCodecInterface == NULL) {
- return UNKNOWN_ERROR;
+ goto CLEAN_UP;
}
ALOGD("VP8: initEncoder. BRMode: %u. TSLayers: %zu. KF: %u. QP: %u - %u",
(uint32_t)mBitrateControlMode, mTemporalLayers, mKeyFrameInterval,
mMinQuantizer, mMaxQuantizer);
+
+ mCodecConfiguration = new vpx_codec_enc_cfg_t;
codec_return = vpx_codec_enc_config_default(mCodecInterface,
mCodecConfiguration,
0); // Codec specific flags
if (codec_return != VPX_CODEC_OK) {
ALOGE("Error populating default configuration for vpx encoder.");
- return UNKNOWN_ERROR;
+ goto CLEAN_UP;
}
mCodecConfiguration->g_w = mWidth;
@@ -250,7 +250,7 @@
default:
{
ALOGE("Wrong number of temporal layers %zu", mTemporalLayers);
- return UNKNOWN_ERROR;
+ goto CLEAN_UP;
}
}
@@ -272,6 +272,7 @@
mCodecConfiguration->rc_max_quantizer = mMaxQuantizer;
}
+ mCodecContext = new vpx_codec_ctx_t;
codec_return = vpx_codec_enc_init(mCodecContext,
mCodecInterface,
mCodecConfiguration,
@@ -279,7 +280,7 @@
if (codec_return != VPX_CODEC_OK) {
ALOGE("Error initializing vpx encoder");
- return UNKNOWN_ERROR;
+ goto CLEAN_UP;
}
codec_return = vpx_codec_control(mCodecContext,
@@ -287,7 +288,7 @@
mDCTPartitions);
if (codec_return != VPX_CODEC_OK) {
ALOGE("Error setting dct partitions for vpx encoder.");
- return UNKNOWN_ERROR;
+ goto CLEAN_UP;
}
// Extra CBR settings
@@ -313,7 +314,7 @@
}
if (codec_return != VPX_CODEC_OK) {
ALOGE("Error setting cbr parameters for vpx encoder.");
- return UNKNOWN_ERROR;
+ goto CLEAN_UP;
}
}
@@ -321,16 +322,20 @@
free(mConversionBuffer);
mConversionBuffer = NULL;
if (((uint64_t)mWidth * mHeight) > ((uint64_t)INT32_MAX / 3)) {
- ALOGE("b/25812794, Buffer size is too big.");
- return UNKNOWN_ERROR;
+ ALOGE("b/25812794, Buffer size is too big, width=%d, height=%d.", mWidth, mHeight);
+ goto CLEAN_UP;
}
mConversionBuffer = (uint8_t *)malloc(mWidth * mHeight * 3 / 2);
if (mConversionBuffer == NULL) {
ALOGE("Allocating conversion buffer failed.");
- return UNKNOWN_ERROR;
+ goto CLEAN_UP;
}
}
return OK;
+
+CLEAN_UP:
+ releaseEncoder();
+ return result;
}
diff --git a/media/libstagefright/foundation/MediaBuffer.cpp b/media/libstagefright/foundation/MediaBuffer.cpp
index 1f80a47..d83a351 100644
--- a/media/libstagefright/foundation/MediaBuffer.cpp
+++ b/media/libstagefright/foundation/MediaBuffer.cpp
@@ -30,6 +30,9 @@
namespace android {
+// allocations larger than this will use shared memory
+static const size_t kSharedMemThreshold = 64 * 1024;
+
MediaBuffer::MediaBuffer(void *data, size_t size)
: mObserver(NULL),
mNextBuffer(NULL),
@@ -47,13 +50,29 @@
: mObserver(NULL),
mNextBuffer(NULL),
mRefCount(0),
- mData(malloc(size)),
+ mData(NULL),
mSize(size),
mRangeOffset(0),
mRangeLength(size),
mOwnsData(true),
mMetaData(new MetaData),
mOriginal(NULL) {
+ if (size < kSharedMemThreshold) {
+ mData = malloc(size);
+ } else {
+ sp<MemoryDealer> memoryDealer = new MemoryDealer(size, "MediaBuffer");
+ mMemory = memoryDealer->allocate(size);
+ if (mMemory == NULL) {
+ ALOGW("Failed to allocate shared memory, trying regular allocation!");
+ mData = malloc(size);
+ if (mData == NULL) {
+ ALOGE("Out of memory");
+ }
+ } else {
+ mData = mMemory->pointer();
+ ALOGV("Allocated shared mem buffer of size %zu @ %p", size, mData);
+ }
+ }
}
MediaBuffer::MediaBuffer(const sp<GraphicBuffer>& graphicBuffer)
@@ -158,7 +177,7 @@
MediaBuffer::~MediaBuffer() {
CHECK(mObserver == NULL);
- if (mOwnsData && mData != NULL) {
+ if (mOwnsData && mData != NULL && mMemory == NULL) {
free(mData);
mData = NULL;
}
diff --git a/media/libstagefright/rtsp/ARTPSource.cpp b/media/libstagefright/rtsp/ARTPSource.cpp
index d7c3bd6..576a0a4 100644
--- a/media/libstagefright/rtsp/ARTPSource.cpp
+++ b/media/libstagefright/rtsp/ARTPSource.cpp
@@ -116,8 +116,15 @@
// to the highest sequence number (extended to 32 bits) received so far.
uint32_t seq1 = seqNum | (mHighestSeqNumber & 0xffff0000);
- uint32_t seq2 = seqNum | ((mHighestSeqNumber & 0xffff0000) + 0x10000);
- uint32_t seq3 = seqNum | ((mHighestSeqNumber & 0xffff0000) - 0x10000);
+
+ // non-overflowing version of:
+ // uint32_t seq2 = seqNum | ((mHighestSeqNumber & 0xffff0000) + 0x10000);
+ uint32_t seq2 = seqNum | (((mHighestSeqNumber >> 16) + 1) << 16);
+
+ // non-underflowing version of:
+ // uint32_t seq2 = seqNum | ((mHighestSeqNumber & 0xffff0000) - 0x10000);
+ uint32_t seq3 = seqNum | ((((mHighestSeqNumber >> 16) | 0x10000) - 1) << 16);
+
uint32_t diff1 = AbsDiff(seq1, mHighestSeqNumber);
uint32_t diff2 = AbsDiff(seq2, mHighestSeqNumber);
uint32_t diff3 = AbsDiff(seq3, mHighestSeqNumber);
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index 580d8c1..ee99a26 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -14,22 +14,15 @@
main_mediaserver.cpp
LOCAL_SHARED_LIBRARIES := \
- libaudioflinger \
- libaudiopolicyservice \
libcamera_metadata\
libcameraservice \
- libicuuc \
libmedialogservice \
libresourcemanagerservice \
libcutils \
- libnbaio \
libmedia \
libmediaplayerservice \
libutils \
- liblog \
libbinder \
- libsoundtriggerservice \
- libradioservice
LOCAL_STATIC_LIBRARIES := \
libicuandroid_utils \
@@ -37,18 +30,8 @@
LOCAL_C_INCLUDES := \
frameworks/av/media/libmediaplayerservice \
- frameworks/av/services/medialog \
- frameworks/av/services/audioflinger \
- frameworks/av/services/audiopolicy \
- frameworks/av/services/audiopolicy/common/managerdefinitions/include \
- frameworks/av/services/audiopolicy/common/include \
- frameworks/av/services/audiopolicy/engine/interface \
frameworks/av/services/camera/libcameraservice \
frameworks/av/services/mediaresourcemanager \
- $(call include-path-for, audio-utils) \
- frameworks/av/services/soundtrigger \
- frameworks/av/services/radio \
- external/sonic
LOCAL_MODULE:= mediaserver
LOCAL_32_BIT_ONLY := true
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index 4a485ed..7e3041b 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -18,126 +18,30 @@
#define LOG_TAG "mediaserver"
//#define LOG_NDEBUG 0
-#include <fcntl.h>
-#include <sys/prctl.h>
-#include <sys/wait.h>
#include <binder/IPCThreadState.h>
#include <binder/ProcessState.h>
#include <binder/IServiceManager.h>
-#include <cutils/properties.h>
#include <utils/Log.h>
#include "RegisterExtensions.h"
// from LOCAL_C_INCLUDES
-#include "AudioFlinger.h"
#include "CameraService.h"
-#include "IcuUtils.h"
-#include "MediaLogService.h"
#include "MediaPlayerService.h"
#include "ResourceManagerService.h"
-#include "service/AudioPolicyService.h"
-#include "SoundTriggerHwService.h"
-#include "RadioService.h"
using namespace android;
-int main(int argc __unused, char** argv)
+int main(int argc __unused, char **argv __unused)
{
signal(SIGPIPE, SIG_IGN);
- char value[PROPERTY_VALUE_MAX];
- bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
- pid_t childPid;
- // FIXME The advantage of making the process containing media.log service the parent process of
- // the process that contains all the other real services, is that it allows us to collect more
- // detailed information such as signal numbers, stop and continue, resource usage, etc.
- // But it is also more complex. Consider replacing this by independent processes, and using
- // binder on death notification instead.
- if (doLog && (childPid = fork()) != 0) {
- // media.log service
- //prctl(PR_SET_NAME, (unsigned long) "media.log", 0, 0, 0);
- // unfortunately ps ignores PR_SET_NAME for the main thread, so use this ugly hack
- strcpy(argv[0], "media.log");
- sp<ProcessState> proc(ProcessState::self());
- MediaLogService::instantiate();
- ProcessState::self()->startThreadPool();
- for (;;) {
- siginfo_t info;
- int ret = waitid(P_PID, childPid, &info, WEXITED | WSTOPPED | WCONTINUED);
- if (ret == EINTR) {
- continue;
- }
- if (ret < 0) {
- break;
- }
- char buffer[32];
- const char *code;
- switch (info.si_code) {
- case CLD_EXITED:
- code = "CLD_EXITED";
- break;
- case CLD_KILLED:
- code = "CLD_KILLED";
- break;
- case CLD_DUMPED:
- code = "CLD_DUMPED";
- break;
- case CLD_STOPPED:
- code = "CLD_STOPPED";
- break;
- case CLD_TRAPPED:
- code = "CLD_TRAPPED";
- break;
- case CLD_CONTINUED:
- code = "CLD_CONTINUED";
- break;
- default:
- snprintf(buffer, sizeof(buffer), "unknown (%d)", info.si_code);
- code = buffer;
- break;
- }
- struct rusage usage;
- getrusage(RUSAGE_CHILDREN, &usage);
- ALOG(LOG_ERROR, "media.log", "pid %d status %d code %s user %ld.%03lds sys %ld.%03lds",
- info.si_pid, info.si_status, code,
- usage.ru_utime.tv_sec, usage.ru_utime.tv_usec / 1000,
- usage.ru_stime.tv_sec, usage.ru_stime.tv_usec / 1000);
- sp<IServiceManager> sm = defaultServiceManager();
- sp<IBinder> binder = sm->getService(String16("media.log"));
- if (binder != 0) {
- Vector<String16> args;
- binder->dump(-1, args);
- }
- switch (info.si_code) {
- case CLD_EXITED:
- case CLD_KILLED:
- case CLD_DUMPED: {
- ALOG(LOG_INFO, "media.log", "exiting");
- _exit(0);
- // not reached
- }
- default:
- break;
- }
- }
- } else {
- // all other services
- if (doLog) {
- prctl(PR_SET_PDEATHSIG, SIGKILL); // if parent media.log dies before me, kill me also
- setpgid(0, 0); // but if I die first, don't kill my parent
- }
- InitializeIcuOrDie();
- sp<ProcessState> proc(ProcessState::self());
- sp<IServiceManager> sm = defaultServiceManager();
- ALOGI("ServiceManager: %p", sm.get());
- AudioFlinger::instantiate();
- MediaPlayerService::instantiate();
- ResourceManagerService::instantiate();
- CameraService::instantiate();
- AudioPolicyService::instantiate();
- SoundTriggerHwService::instantiate();
- RadioService::instantiate();
- registerExtensions();
- ProcessState::self()->startThreadPool();
- IPCThreadState::self()->joinThreadPool();
- }
+
+ sp<ProcessState> proc(ProcessState::self());
+ sp<IServiceManager> sm(defaultServiceManager());
+ ALOGI("ServiceManager: %p", sm.get());
+ MediaPlayerService::instantiate();
+ ResourceManagerService::instantiate();
+ CameraService::instantiate();
+ registerExtensions();
+ ProcessState::self()->startThreadPool();
+ IPCThreadState::self()->joinThreadPool();
}
diff --git a/media/mtp/MtpDevice.cpp b/media/mtp/MtpDevice.cpp
index f8b913a..23e432e 100644
--- a/media/mtp/MtpDevice.cpp
+++ b/media/mtp/MtpDevice.cpp
@@ -659,6 +659,13 @@
return false;
}
+ // If object size 0 byte, the remote device can reply response packet
+ // without sending any data packets.
+ if (mData.getContainerType() == MTP_CONTAINER_TYPE_RESPONSE) {
+ mResponse.copyFrom(mData);
+ return mResponse.getResponseCode() == MTP_RESPONSE_OK;
+ }
+
const uint32_t fullLength = mData.getContainerLength();
if ((!expectedLength && fullLength < MTP_CONTAINER_HEADER_SIZE) ||
(expectedLength && *expectedLength + MTP_CONTAINER_HEADER_SIZE != fullLength)) {
diff --git a/media/mtp/MtpUtils.cpp b/media/mtp/MtpUtils.cpp
index 0667bdd..ebf3601 100644
--- a/media/mtp/MtpUtils.cpp
+++ b/media/mtp/MtpUtils.cpp
@@ -19,8 +19,6 @@
#include <stdio.h>
#include <time.h>
-#include <../private/bionic_time.h> /* TODO: switch this code to icu4c! */
-
#include "MtpUtils.h"
namespace android {
@@ -32,38 +30,40 @@
DD replaced by the day (01-31), T is a constant character 'T' delimiting time from date,
hh is replaced by the hour (00-23), mm is replaced by the minute (00-59), and ss by the
second (00-59). The ".s" is optional, and represents tenths of a second.
+This is followed by a UTC offset given as "[+-]zzzz" or the literal "Z", meaning UTC.
*/
bool parseDateTime(const char* dateTime, time_t& outSeconds) {
int year, month, day, hour, minute, second;
- struct tm tm;
-
if (sscanf(dateTime, "%04d%02d%02dT%02d%02d%02d",
- &year, &month, &day, &hour, &minute, &second) != 6)
+ &year, &month, &day, &hour, &minute, &second) != 6)
return false;
- const char* tail = dateTime + 15;
+
// skip optional tenth of second
- if (tail[0] == '.' && tail[1])
- tail += 2;
- //FIXME - support +/-hhmm
+ const char* tail = dateTime + 15;
+ if (tail[0] == '.' && tail[1]) tail += 2;
+
+ // FIXME: "Z" means UTC, but non-"Z" doesn't mean local time.
+ // It might be that you're in Asia/Seoul on vacation and your Android
+ // device has noticed this via the network, but your camera was set to
+ // America/Los_Angeles once when you bought it and doesn't know where
+ // it is right now, so the camera says "20160106T081700-0800" but we
+ // just ignore the "-0800" and assume local time which is actually "+0900".
+ // I think to support this (without switching to Java or using icu4c)
+ // you'd want to always use timegm(3) and then manually add/subtract
+ // the UTC offset parsed from the string (taking care of wrapping).
+ // mktime(3) ignores the tm_gmtoff field, so you can't let it do the work.
bool useUTC = (tail[0] == 'Z');
- // hack to compute timezone
- time_t dummy;
- localtime_r(&dummy, &tm);
-
+ struct tm tm = {};
tm.tm_sec = second;
tm.tm_min = minute;
tm.tm_hour = hour;
tm.tm_mday = day;
tm.tm_mon = month - 1; // mktime uses months in 0 - 11 range
tm.tm_year = year - 1900;
- tm.tm_wday = 0;
tm.tm_isdst = -1;
- if (useUTC)
- outSeconds = mktime(&tm);
- else
- outSeconds = mktime_tz(&tm, tm.tm_zone);
+ outSeconds = useUTC ? timegm(&tm) : mktime(&tm);
return true;
}
@@ -73,7 +73,7 @@
localtime_r(&seconds, &tm);
snprintf(buffer, bufferLength, "%04d%02d%02dT%02d%02d%02d",
- tm.tm_year + 1900,
+ tm.tm_year + 1900,
tm.tm_mon + 1, // localtime_r uses months in 0 - 11 range
tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec);
}
diff --git a/media/utils/BatteryNotifier.cpp b/media/utils/BatteryNotifier.cpp
index 7f9cd7a..341d391 100644
--- a/media/utils/BatteryNotifier.cpp
+++ b/media/utils/BatteryNotifier.cpp
@@ -14,6 +14,9 @@
* limitations under the License.
*/
+#define LOG_TAG "BatteryNotifier"
+//#define LOG_NDEBUG 0
+
#include "include/mediautils/BatteryNotifier.h"
#include <binder/IServiceManager.h>
@@ -64,7 +67,7 @@
sp<IBatteryStats> batteryService = getBatteryService_l();
mVideoRefCount = 0;
if (batteryService != nullptr) {
- batteryService->noteResetAudio();
+ batteryService->noteResetVideo();
}
}
@@ -72,7 +75,7 @@
Mutex::Autolock _l(mLock);
sp<IBatteryStats> batteryService = getBatteryService_l();
if (mAudioRefCount == 0 && batteryService != nullptr) {
- batteryService->noteStartAudio(AID_MEDIA);
+ batteryService->noteStartAudio(AID_AUDIOSERVER);
}
mAudioRefCount++;
}
@@ -88,7 +91,7 @@
mAudioRefCount--;
if (mAudioRefCount == 0 && batteryService != nullptr) {
- batteryService->noteStopAudio(AID_MEDIA);
+ batteryService->noteStopAudio(AID_AUDIOSERVER);
}
}
@@ -190,20 +193,25 @@
const String16 name("batterystats");
mBatteryStatService = interface_cast<IBatteryStats>(sm->checkService(name));
if (mBatteryStatService == nullptr) {
- ALOGE("batterystats service unavailable!");
+ // this may occur normally during the init sequence as mediaserver
+ // and audioserver start before the batterystats service is available.
+ ALOGW("batterystats service unavailable!");
return nullptr;
}
mDeathNotifier = new DeathNotifier();
IInterface::asBinder(mBatteryStatService)->linkToDeath(mDeathNotifier);
- // Notify start now if media already started
+ // Notify start now if mediaserver or audioserver is already started.
+ // 1) mediaserver and audioserver is started before batterystats service
+ // 2) batterystats server may have crashed.
if (mVideoRefCount > 0) {
mBatteryStatService->noteStartVideo(AID_MEDIA);
}
if (mAudioRefCount > 0) {
- mBatteryStatService->noteStartAudio(AID_MEDIA);
+ mBatteryStatService->noteStartAudio(AID_AUDIOSERVER);
}
+ // TODO: Notify for camera and flashlight state as well?
}
return mBatteryStatService;
}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 385e0b9..d2786b9 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -185,8 +185,8 @@
mSystemReady(false)
{
getpid_cached = getpid();
- char value[PROPERTY_VALUE_MAX];
- bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
+ // disable media.log until the service is reenabled, see b/26306954
+ const bool doLog = false; // property_get_bool("ro.test_harness", false);
if (doLog) {
mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
MemoryHeapBase::READ_ONLY);
@@ -261,16 +261,17 @@
}
// Tell media.log service about any old writers that still need to be unregistered
- sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
- if (binder != 0) {
- sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
- for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
- sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
- mUnregisteredWriters.pop();
- mediaLogService->unregisterWriter(iMemory);
+ if (mLogMemoryDealer != 0) {
+ sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
+ if (binder != 0) {
+ sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
+ for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
+ sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
+ mUnregisteredWriters.pop();
+ mediaLogService->unregisterWriter(iMemory);
+ }
}
}
-
}
static const char * const audio_interfaces[] = {
@@ -1457,8 +1458,15 @@
cblk.clear();
buffers.clear();
+ const uid_t callingUid = IPCThreadState::self()->getCallingUid();
+ if (!isTrustedCallingUid(callingUid)) {
+ ALOGW_IF((uid_t)clientUid != callingUid,
+ "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
+ clientUid = callingUid;
+ }
+
// check calling permissions
- if (!recordingAllowed(opPackageName)) {
+ if (!recordingAllowed(opPackageName, tid, clientUid)) {
ALOGE("openRecord() permission denied: recording not allowed");
lStatus = PERMISSION_DENIED;
goto Exit;
@@ -1508,7 +1516,6 @@
}
ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input);
- // TODO: the uid should be passed in as a parameter to openRecord
recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
frameCount, lSessionId, notificationFrames,
clientUid, flags, tid, &lStatus);
@@ -2591,7 +2598,7 @@
// check recording permission for visualizer
if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
- !recordingAllowed(opPackageName)) {
+ !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
diff --git a/services/audioflinger/ServiceUtilities.cpp b/services/audioflinger/ServiceUtilities.cpp
index 2e68dad..afc2440 100644
--- a/services/audioflinger/ServiceUtilities.cpp
+++ b/services/audioflinger/ServiceUtilities.cpp
@@ -32,29 +32,37 @@
// Not valid until initialized by AudioFlinger constructor. It would have to be
// re-initialized if the process containing AudioFlinger service forks (which it doesn't).
+// This is often used to validate binder interface calls within audioserver
+// (e.g. AudioPolicyManager to AudioFlinger).
pid_t getpid_cached;
-bool recordingAllowed(const String16& opPackageName) {
- // Note: We are getting the UID from the calling IPC thread state because all
- // clients that perform recording create AudioRecord in their own processes
- // and the system does not create AudioRecord objects on behalf of apps. This
- // differs from playback where in some situations the system recreates AudioTrack
- // instances associated with a client's MediaPlayer on behalf of this client.
- // In the latter case we have to store the client UID and pass in along for
- // security checks.
+// A trusted calling UID may specify the client UID as part of a binder interface call.
+// otherwise the calling UID must be equal to the client UID.
+bool isTrustedCallingUid(uid_t uid) {
+ switch (uid) {
+ case AID_MEDIA:
+ case AID_AUDIOSERVER:
+ return true;
+ default:
+ return false;
+ }
+}
+bool recordingAllowed(const String16& opPackageName, pid_t pid, uid_t uid) {
+ // we're always OK.
if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+
static const String16 sRecordAudio("android.permission.RECORD_AUDIO");
+ // We specify a pid and uid here as mediaserver (aka MediaRecorder or StageFrightRecorder)
+ // may open a record track on behalf of a client. Note that pid may be a tid.
// IMPORTANT: Don't use PermissionCache - a runtime permission and may change.
- const bool ok = checkCallingPermission(sRecordAudio);
+ const bool ok = checkPermission(sRecordAudio, pid, uid);
if (!ok) {
ALOGE("Request requires android.permission.RECORD_AUDIO");
return false;
}
- const uid_t uid = IPCThreadState::self()->getCallingUid();
-
// To permit command-line native tests
if (uid == AID_ROOT) return true;
diff --git a/services/audioflinger/ServiceUtilities.h b/services/audioflinger/ServiceUtilities.h
index fba6dce..1e79553 100644
--- a/services/audioflinger/ServiceUtilities.h
+++ b/services/audioflinger/ServiceUtilities.h
@@ -19,8 +19,8 @@
namespace android {
extern pid_t getpid_cached;
-
-bool recordingAllowed(const String16& opPackageName);
+bool isTrustedCallingUid(uid_t uid);
+bool recordingAllowed(const String16& opPackageName, pid_t pid, uid_t uid);
bool captureAudioOutputAllowed();
bool captureHotwordAllowed();
bool settingsAllowed();
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 553970b..7c9b81e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -914,14 +914,14 @@
status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK,
binder,
getWakeLockTag(),
- String16("media"),
+ String16("audioserver"),
uid,
true /* FIXME force oneway contrary to .aidl */);
} else {
status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
binder,
getWakeLockTag(),
- String16("media"),
+ String16("audioserver"),
true /* FIXME force oneway contrary to .aidl */);
}
if (status == NO_ERROR) {
@@ -980,8 +980,12 @@
void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
getPowerManager_l();
- if (mWakeLockToken == NULL) {
- ALOGE("no wake lock to update!");
+ if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called.
+ if (mSystemReady) {
+ ALOGE("no wake lock to update, but system ready!");
+ } else {
+ ALOGW("no wake lock to update, system not ready yet");
+ }
return;
}
if (mPowerManager != 0) {
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 0e24b52..b1638ea 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -100,13 +100,11 @@
mType(type),
mThreadIoHandle(thread->id())
{
- // if the caller is us, trust the specified uid
- if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
- int newclientUid = IPCThreadState::self()->getCallingUid();
- if (clientUid != -1 && clientUid != newclientUid) {
- ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
- }
- clientUid = newclientUid;
+ const uid_t callingUid = IPCThreadState::self()->getCallingUid();
+ if (!isTrustedCallingUid(callingUid) || clientUid == -1) {
+ ALOGW_IF(clientUid != -1 && clientUid != (int)callingUid,
+ "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
+ clientUid = (int)callingUid;
}
// clientUid contains the uid of the app that is responsible for this track, so we can blame
// battery usage on it.
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 1a1cbb2..34984f9 100755
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -18,9 +18,7 @@
#include <system/audio.h>
-static const uint32_t gDynamicRate = 0;
static const audio_format_t gDynamicFormat = AUDIO_FORMAT_DEFAULT;
-static const audio_channel_mask_t gDynamicChannelMask = AUDIO_CHANNEL_NONE;
// For mixed output and inputs, the policy will use max mixer sampling rates.
// Do not limit sampling rate otherwise
diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk
index dc7eff7..dffeb7e 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.mk
+++ b/services/audiopolicy/common/managerdefinitions/Android.mk
@@ -9,6 +9,7 @@
src/HwModule.cpp \
src/IOProfile.cpp \
src/AudioPort.cpp \
+ src/AudioProfile.cpp \
src/AudioPolicyMix.cpp \
src/AudioPatch.cpp \
src/AudioInputDescriptor.cpp \
@@ -18,7 +19,8 @@
src/SoundTriggerSession.cpp \
src/SessionRoute.cpp \
src/AudioSourceDescriptor.cpp \
- src/TypeConverter.cpp
+ src/TypeConverter.cpp \
+ src/AudioSession.cpp
LOCAL_SHARED_LIBRARIES := \
libcutils \
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
index 6d04811..7e5ef5d 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -17,6 +17,7 @@
#pragma once
#include "AudioPort.h"
+#include "AudioSession.h"
#include <utils/Errors.h>
#include <system/audio.h>
#include <utils/SortedVector.h>
@@ -36,7 +37,6 @@
void setIoHandle(audio_io_handle_t ioHandle);
audio_port_handle_t getId() const;
audio_module_handle_t getModuleHandle() const;
- void changeOpenRefCount(int delta);
uint32_t getOpenRefCount() const;
status_t dump(int fd);
@@ -45,13 +45,7 @@
audio_devices_t mDevice; // current device this input is routed to
AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount; // number of AudioRecord clients active on
- // this input
- audio_source_t mInputSource; // input source selected by application
- //(mediarecorder.h)
const sp<IOProfile> mProfile; // I/O profile this output derives from
- SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
- bool mIsSoundTrigger; // used by a soundtrigger capture
virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig = NULL) const;
@@ -61,10 +55,20 @@
SortedVector<audio_session_t> getPreemptedSessions() const;
bool hasPreemptedSession(audio_session_t session) const;
void clearPreemptedSessions();
+ bool isActive() const;
+ bool isSourceActive(audio_source_t source) const;
+ audio_source_t inputSource() const;
+ bool isSoundTrigger() const;
+ status_t addAudioSession(audio_session_t session,
+ const sp<AudioSession>& audioSession);
+ status_t removeAudioSession(audio_session_t session);
+ sp<AudioSession> getAudioSession(audio_session_t session) const;
+ AudioSessionCollection getActiveAudioSessions() const;
private:
audio_port_handle_t mId;
- uint32_t mOpenRefCount;
+ // audio sessions attached to this input
+ AudioSessionCollection mSessions;
// Because a preemtible capture session can preempt another one, we end up in an endless loop
// situation were each session is allowed to restart after being preempted,
// thus preempting the other one which restarts and so on.
@@ -72,7 +76,6 @@
// a particular input started and prevent preemption of this active input by this session.
// We also inherit sessions from the preempted input to avoid a 3 way preemption loop etc...
SortedVector<audio_session_t> mPreemptedSessions;
-
};
class AudioInputCollection :
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index 5ff0396..20c0e36 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -99,9 +99,8 @@
sp<OutputProfile> outProfile;
outProfile = new OutputProfile(String8("primary"));
outProfile->attach(module);
- outProfile->mSamplingRates.add(44100);
- outProfile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
- outProfile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
+ outProfile->addAudioProfile(
+ new AudioProfile(AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 44100));
outProfile->addSupportedDevice(mDefaultOutputDevices);
outProfile->setFlags(AUDIO_OUTPUT_FLAG_PRIMARY);
module->mOutputProfiles.add(outProfile);
@@ -109,9 +108,8 @@
sp<InputProfile> inProfile;
inProfile = new InputProfile(String8("primary"));
inProfile->attach(module);
- inProfile->mSamplingRates.add(8000);
- inProfile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
- inProfile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
+ inProfile->addAudioProfile(
+ new AudioProfile(AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_MONO, 8000));
inProfile->addSupportedDevice(defaultInputDevice);
module->mInputProfiles.add(inProfile);
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
index 5666920..9aaaddf 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -16,6 +16,7 @@
#pragma once
+#include "AudioProfile.h"
#include <utils/String8.h>
#include <utils/Vector.h>
#include <utils/RefBase.h>
@@ -57,42 +58,44 @@
virtual void toAudioPort(struct audio_port *port) const;
virtual void importAudioPort(const sp<AudioPort> port);
- void clearCapabilities();
+ void clearCapabilities() { mProfiles.clearProfiles(); }
- void setSupportedFormats(const Vector <audio_format_t> &formats);
- void setSupportedSamplingRates(const Vector <uint32_t> &sampleRates)
- {
- mSamplingRates = sampleRates;
- }
- void setSupportedChannelMasks(const Vector <audio_channel_mask_t> &channelMasks)
- {
- mChannelMasks = channelMasks;
- }
+ void addAudioProfile(const sp<AudioProfile> &profile) { mProfiles.add(profile); }
+
+ void setAudioProfiles(const AudioProfileVector &profiles) { mProfiles = profiles; }
+ AudioProfileVector &getAudioProfiles() { return mProfiles; }
+
+ bool hasValidAudioProfile() const { return mProfiles.hasValidProfile(); }
+
+ bool hasDynamicAudioProfile() const { return mProfiles.hasDynamicProfile(); }
// searches for an exact match
- status_t checkExactSamplingRate(uint32_t samplingRate) const;
- // searches for a compatible match, and returns the best match via updatedSamplingRate
- status_t checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const;
- // searches for an exact match
- status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
- // searches for a compatible match, currently implemented for input channel masks only
- status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask,
- audio_channel_mask_t *updatedChannelMask) const;
+ status_t checkExactAudioProfile(uint32_t samplingRate,
+ audio_channel_mask_t channelMask,
+ audio_format_t format) const
+ {
+ return mProfiles.checkExactProfile(samplingRate, channelMask, format);
+ }
- status_t checkExactFormat(audio_format_t format) const;
- // searches for a compatible match, currently implemented for input formats only
- status_t checkCompatibleFormat(audio_format_t format, audio_format_t *updatedFormat) const;
+ // searches for a compatible match, currently implemented for input
+ // parameters are input|output, returned value is the best match.
+ status_t checkCompatibleAudioProfile(uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format) const
+ {
+ return mProfiles.checkCompatibleProfile(samplingRate, channelMask, format, mType, mRole);
+ }
+
+ void clearAudioProfiles() { return mProfiles.clearProfiles(); }
+
status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
- uint32_t pickSamplingRate() const;
- audio_channel_mask_t pickChannelMask() const;
- audio_format_t pickFormat() const;
+ void pickAudioProfile(uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format) const;
static const audio_format_t sPcmFormatCompareTable[];
- static int compareFormats(const audio_format_t *format1, const audio_format_t *format2) {
- return compareFormats(*format1, *format2);
- }
+
static int compareFormats(audio_format_t format1, audio_format_t format2);
audio_module_handle_t getModuleHandle() const;
@@ -105,23 +108,27 @@
((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SINK));
}
- void dump(int fd, int spaces) const;
+ inline bool isDirectOutput() const
+ {
+ return (mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD));
+ }
+
+ void dump(int fd, int spaces, bool verbose = true) const;
void log(const char* indent) const;
- // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
- // indicates the supported parameters should be read from the output stream
- // after it is opened for the first time
- Vector <uint32_t> mSamplingRates; // supported sampling rates
- Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
- Vector <audio_format_t> mFormats; // supported audio formats
AudioGainCollection mGains; // gain controllers
sp<HwModule> mModule; // audio HW module exposing this I/O stream
private:
+ void pickChannelMask(audio_channel_mask_t &channelMask, const ChannelsVector &channelMasks) const;
+ void pickSamplingRate(uint32_t &rate,const SampleRateVector &samplingRates) const;
+
String8 mName;
audio_port_type_t mType;
audio_port_role_t mRole;
uint32_t mFlags; // attribute flags mask (e.g primary output, direct output...).
+ AudioProfileVector mProfiles; // AudioProfiles supported by this port (format, Rates, Channels)
static volatile int32_t mNextUniqueId;
};
@@ -132,9 +139,9 @@
virtual ~AudioPortConfig() {}
status_t applyAudioPortConfig(const struct audio_port_config *config,
- struct audio_port_config *backupConfig = NULL);
+ struct audio_port_config *backupConfig = NULL);
virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const = 0;
+ const struct audio_port_config *srcConfig = NULL) const = 0;
virtual sp<AudioPort> getAudioPort() const = 0;
uint32_t mSamplingRate;
audio_format_t mFormat;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
new file mode 100644
index 0000000..9780dc6
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
@@ -0,0 +1,353 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include "policy.h"
+#include <utils/String8.h>
+#include <utils/SortedVector.h>
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <system/audio.h>
+#include <cutils/config_utils.h>
+
+namespace android {
+
+typedef SortedVector<uint32_t> SampleRateVector;
+typedef SortedVector<audio_channel_mask_t> ChannelsVector;
+typedef Vector<audio_format_t> FormatVector;
+
+template <typename T>
+bool operator == (const SortedVector<T> &left, const SortedVector<T> &right);
+
+class AudioProfile : public virtual RefBase
+{
+public:
+ AudioProfile(audio_format_t format,
+ audio_channel_mask_t channelMasks,
+ uint32_t samplingRate) :
+ mName(String8("")),
+ mFormat(format)
+ {
+ mChannelMasks.add(channelMasks);
+ mSamplingRates.add(samplingRate);
+ }
+
+ AudioProfile(audio_format_t format,
+ const ChannelsVector &channelMasks,
+ const SampleRateVector &samplingRateCollection) :
+ mName(String8("")),
+ mFormat(format),
+ mChannelMasks(channelMasks),
+ mSamplingRates(samplingRateCollection)
+ {}
+
+ audio_format_t getFormat() const { return mFormat; }
+
+ void setChannels(const ChannelsVector &channelMasks)
+ {
+ if (mIsDynamicChannels) {
+ mChannelMasks = channelMasks;
+ }
+ }
+ const ChannelsVector &getChannels() const { return mChannelMasks; }
+
+ void setSampleRates(const SampleRateVector &sampleRates)
+ {
+ if (mIsDynamicRate) {
+ mSamplingRates = sampleRates;
+ }
+ }
+ const SampleRateVector &getSampleRates() const { return mSamplingRates; }
+
+ bool isValid() const { return hasValidFormat() && hasValidRates() && hasValidChannels(); }
+
+ void clear()
+ {
+ if (mIsDynamicChannels) {
+ mChannelMasks.clear();
+ }
+ if (mIsDynamicRate) {
+ mSamplingRates.clear();
+ }
+ }
+
+ inline bool supportsChannels(audio_channel_mask_t channels) const
+ {
+ return mChannelMasks.indexOf(channels) >= 0;
+ }
+ inline bool supportsRate(uint32_t rate) const
+ {
+ return mSamplingRates.indexOf(rate) >= 0;
+ }
+
+ status_t checkExact(uint32_t rate, audio_channel_mask_t channels, audio_format_t format) const;
+
+ status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask,
+ audio_channel_mask_t &updatedChannelMask,
+ audio_port_type_t portType,
+ audio_port_role_t portRole) const;
+
+ status_t checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t &updatedSamplingRate) const;
+
+ bool hasValidFormat() const { return mFormat != AUDIO_FORMAT_DEFAULT; }
+ bool hasValidRates() const { return !mSamplingRates.isEmpty(); }
+ bool hasValidChannels() const { return !mChannelMasks.isEmpty(); }
+
+ void setDynamicChannels(bool dynamic) { mIsDynamicChannels = dynamic; }
+ bool isDynamicChannels() const { return mIsDynamicChannels; }
+
+ void setDynamicRate(bool dynamic) { mIsDynamicRate = dynamic; }
+ bool isDynamicRate() const { return mIsDynamicRate; }
+
+ void setDynamicFormat(bool dynamic) { mIsDynamicFormat = dynamic; }
+ bool isDynamicFormat() const { return mIsDynamicFormat; }
+
+ bool isDynamic() { return mIsDynamicFormat || mIsDynamicChannels || mIsDynamicRate; }
+
+ void dump(int fd, int spaces) const;
+
+private:
+ String8 mName;
+ audio_format_t mFormat;
+ ChannelsVector mChannelMasks;
+ SampleRateVector mSamplingRates;
+
+ bool mIsDynamicFormat = false;
+ bool mIsDynamicChannels = false;
+ bool mIsDynamicRate = false;
+};
+
+
+class AudioProfileVector : public Vector<sp<AudioProfile> >
+{
+public:
+ ssize_t add(const sp<AudioProfile> &profile)
+ {
+ ssize_t index = Vector::add(profile);
+ // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry.
+ // TODO: compareFormats could be a lambda to convert between pointer-to-format to format:
+ // [](const audio_format_t *format1, const audio_format_t *format2) {
+ // return compareFormats(*format1, *format2);
+ // }
+ sort(compareFormats);
+ return index;
+ }
+
+ // This API is intended to be used by the policy manager once retrieving capabilities
+ // for a profile with dynamic format, rate and channels attributes
+ ssize_t addProfileFromHal(const sp<AudioProfile> &profileToAdd)
+ {
+ // Check valid profile to add:
+ if (!profileToAdd->hasValidFormat()) {
+ return -1;
+ }
+ if (!profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
+ FormatVector formats;
+ formats.add(profileToAdd->getFormat());
+ setFormats(FormatVector(formats));
+ return 0;
+ }
+ if (!profileToAdd->hasValidChannels() && profileToAdd->hasValidRates()) {
+ setSampleRatesFor(profileToAdd->getSampleRates(), profileToAdd->getFormat());
+ return 0;
+ }
+ if (profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
+ setChannelsFor(profileToAdd->getChannels(), profileToAdd->getFormat());
+ return 0;
+ }
+ // Go through the list of profile to avoid duplicates
+ for (size_t profileIndex = 0; profileIndex < size(); profileIndex++) {
+ const sp<AudioProfile> &profile = itemAt(profileIndex);
+ if (profile->isValid() && profile == profileToAdd) {
+ // Nothing to do
+ return profileIndex;
+ }
+ }
+ profileToAdd->setDynamicFormat(true); // set the format as dynamic to allow removal
+ return add(profileToAdd);
+ }
+
+ sp<AudioProfile> getFirstValidProfile() const
+ {
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->isValid()) {
+ return itemAt(i);
+ }
+ }
+ return 0;
+ }
+
+ bool hasValidProfile() const { return getFirstValidProfile() != 0; }
+
+ status_t checkExactProfile(uint32_t samplingRate, audio_channel_mask_t channelMask,
+ audio_format_t format) const;
+
+ status_t checkCompatibleProfile(uint32_t &samplingRate, audio_channel_mask_t &channelMask,
+ audio_format_t &format,
+ audio_port_type_t portType,
+ audio_port_role_t portRole) const;
+
+ FormatVector getSupportedFormats() const
+ {
+ FormatVector supportedFormats;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->hasValidFormat()) {
+ supportedFormats.add(itemAt(i)->getFormat());
+ }
+ }
+ return supportedFormats;
+ }
+
+ bool hasDynamicProfile() const
+ {
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->isDynamic()) {
+ return true;
+ }
+ }
+ return false;
+ }
+
+ bool hasDynamicFormat() const
+ {
+ return getProfileFor(gDynamicFormat) != 0;
+ }
+
+ bool hasDynamicChannelsFor(audio_format_t format) const
+ {
+ for (size_t i = 0; i < size(); i++) {
+ sp<AudioProfile> profile = itemAt(i);
+ if (profile->getFormat() == format && profile->isDynamicChannels()) {
+ return true;
+ }
+ }
+ return false;
+ }
+
+ bool hasDynamicRateFor(audio_format_t format) const
+ {
+ for (size_t i = 0; i < size(); i++) {
+ sp<AudioProfile> profile = itemAt(i);
+ if (profile->getFormat() == format && profile->isDynamicRate()) {
+ return true;
+ }
+ }
+ return false;
+ }
+
+ // One audio profile will be added for each format supported by Audio HAL
+ void setFormats(const FormatVector &formats)
+ {
+ // Only allow to change the format of dynamic profile
+ sp<AudioProfile> dynamicFormatProfile = getProfileFor(gDynamicFormat);
+ if (dynamicFormatProfile == 0) {
+ return;
+ }
+ clearProfiles();
+ for (size_t i = 0; i < formats.size(); i++) {
+ sp<AudioProfile> profile = new AudioProfile(formats[i],
+ dynamicFormatProfile->getChannels(),
+ dynamicFormatProfile->getSampleRates());
+ profile->setDynamicFormat(true);
+ profile->setDynamicChannels(dynamicFormatProfile->isDynamicChannels());
+ profile->setDynamicRate(dynamicFormatProfile->isDynamicRate());
+ add(profile);
+ }
+ }
+
+ void setSampleRatesFor(const SampleRateVector &sampleRates, audio_format_t format)
+ {
+ for (size_t i = 0; i < size(); i++) {
+ sp<AudioProfile> profile = itemAt(i);
+ if (profile->getFormat() == format && profile->isDynamicRate()) {
+ if (profile->hasValidRates()) {
+ // Need to create a new profile with same format
+ sp<AudioProfile> profileToAdd = new AudioProfile(format, profile->getChannels(),
+ sampleRates);
+ profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
+ add(profileToAdd);
+ } else {
+ profile->setSampleRates(sampleRates);
+ }
+ return;
+ }
+ }
+ }
+
+ void setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format)
+ {
+ for (size_t i = 0; i < size(); i++) {
+ sp<AudioProfile> profile = itemAt(i);
+ if (profile->getFormat() == format && profile->isDynamicChannels()) {
+ if (profile->hasValidChannels()) {
+ // Need to create a new profile with same format
+ sp<AudioProfile> profileToAdd = new AudioProfile(format, channelMasks,
+ profile->getSampleRates());
+ profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
+ add(profileToAdd);
+ } else {
+ profile->setChannels(channelMasks);
+ }
+ return;
+ }
+ }
+ }
+
+ void clearProfiles()
+ {
+ for (size_t i = size(); i != 0; ) {
+ sp<AudioProfile> profile = itemAt(--i);
+ if (profile->isDynamicFormat() && profile->hasValidFormat()) {
+ removeAt(i);
+ continue;
+ }
+ profile->clear();
+ }
+ }
+
+ void dump(int fd, int spaces) const
+ {
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "%*s- Profiles:\n", spaces, "");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ snprintf(buffer, SIZE, "%*sProfile %zu:", spaces + 4, "", i);
+ write(fd, buffer, strlen(buffer));
+ itemAt(i)->dump(fd, spaces + 8);
+ }
+ }
+
+private:
+ sp<AudioProfile> getProfileFor(audio_format_t format) const
+ {
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->getFormat() == format) {
+ return itemAt(i);
+ }
+ }
+ return 0;
+ }
+
+ static int compareFormats(const sp<AudioProfile> *profile1, const sp<AudioProfile> *profile2);
+};
+
+bool operator == (const AudioProfile &left, const AudioProfile &right);
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioSession.h b/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
new file mode 100644
index 0000000..6feef80
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
@@ -0,0 +1,87 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <system/audio.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+class AudioSession : public RefBase
+{
+public:
+ AudioSession(audio_session_t session,
+ audio_source_t inputSource,
+ audio_format_t format,
+ uint32_t sampleRate,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ uid_t uid,
+ bool isSoundTrigger);
+
+ status_t dump(int fd, int spaces, int index) const;
+
+ audio_session_t session() const { return mSession; }
+ audio_source_t inputSource()const { return mInputSource; }
+ audio_format_t format() const { return mFormat; }
+ uint32_t sampleRate() const { return mSampleRate; }
+ audio_channel_mask_t channelMask() const { return mChannelMask; }
+ audio_input_flags_t flags() const { return mFlags; }
+ uid_t uid() const { return mUid; }
+ bool matches(const sp<AudioSession> &other) const;
+ bool isSoundTrigger() const { return mIsSoundTrigger; }
+ uint32_t openCount() const { return mOpenCount; } ;
+ uint32_t activeCount() const { return mActiveCount; } ;
+
+ uint32_t changeOpenCount(int delta);
+ uint32_t changeActiveCount(int delta);
+
+private:
+ const audio_session_t mSession;
+ const audio_source_t mInputSource;
+ const audio_format_t mFormat;
+ const uint32_t mSampleRate;
+ const audio_channel_mask_t mChannelMask;
+ const audio_input_flags_t mFlags;
+ const uid_t mUid;
+ bool mIsSoundTrigger;
+ uint32_t mOpenCount;
+ uint32_t mActiveCount;
+};
+
+class AudioSessionCollection :
+ public DefaultKeyedVector<audio_session_t, sp<AudioSession> >
+{
+public:
+ status_t addSession(audio_session_t session,
+ const sp<AudioSession>& audioSession);
+
+ status_t removeSession(audio_session_t session);
+
+ uint32_t getOpenCount() const;
+
+ AudioSessionCollection getActiveSessions() const;
+ bool hasActiveSession() const;
+ bool isSourceActive(audio_source_t source) const;
+
+ status_t dump(int fd, int spaces) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h
index 1eddab9..ab23105 100644
--- a/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h
+++ b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h
@@ -45,8 +45,7 @@
static void loadAudioPortGains(cnode *root, AudioPort &audioPort);
static void loadDeviceDescriptorGains(cnode *root, sp<DeviceDescriptor> &deviceDesc);
static status_t loadHwModuleDevice(cnode *root, DeviceVector &devices);
- static status_t loadHwModuleInput(cnode *root, sp<HwModule> &module);
- static status_t loadHwModuleOutput(cnode *root, sp<HwModule> &module);
+ static status_t loadHwModuleProfile(cnode *root, sp<HwModule> &module, audio_port_role_t role);
static void loadDevicesFromTag(const char *tag, DeviceVector &devices,
const DeviceVector &declaredDevices);
static void loadHwModules(cnode *root, HwModuleCollection &hwModules,
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 3f43963..8ece51b 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -50,7 +50,7 @@
virtual void importAudioPort(const sp<AudioPort> port);
audio_port_handle_t getId() const;
- status_t dump(int fd, int spaces, int index) const;
+ status_t dump(int fd, int spaces, int index, bool verbose = true) const;
void log() const;
String8 mAddress;
@@ -85,7 +85,7 @@
audio_policy_dev_state_t getDeviceConnectionState(const sp<DeviceDescriptor> &devDesc) const;
- status_t dump(int fd, const String8 &direction) const;
+ status_t dump(int fd, const String8 &tag, int spaces = 0, bool verbose = true) const;
private:
void refreshTypes();
diff --git a/services/audiopolicy/common/managerdefinitions/include/TypeConverter.h b/services/audiopolicy/common/managerdefinitions/include/TypeConverter.h
index 9b96770..14e2ecc 100644
--- a/services/audiopolicy/common/managerdefinitions/include/TypeConverter.h
+++ b/services/audiopolicy/common/managerdefinitions/include/TypeConverter.h
@@ -27,14 +27,10 @@
namespace android {
-#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and
- // "formats" in outputs descriptors indicating that supported
- // values should be queried after opening the output.
-
struct SampleRateTraits
{
typedef uint32_t Type;
- typedef Vector<Type> Collection;
+ typedef SortedVector<Type> Collection;
};
struct DeviceTraits
{
@@ -59,7 +55,7 @@
struct ChannelTraits
{
typedef audio_channel_mask_t Type;
- typedef Vector<Type> Collection;
+ typedef SortedVector<Type> Collection;
};
struct OutputChannelTraits : public ChannelTraits {};
struct InputChannelTraits : public ChannelTraits {};
@@ -139,50 +135,27 @@
const char *del = "|")
{
SampleRateTraits::Collection samplingRateCollection;
- // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
- // rates should be read from the output stream after it is opened for the first time
- if (samplingRates == DYNAMIC_VALUE_TAG) {
- samplingRateCollection.add(gDynamicRate);
- } else {
- collectionFromString<SampleRateTraits>(samplingRates, samplingRateCollection, del);
- }
+ collectionFromString<SampleRateTraits>(samplingRates, samplingRateCollection, del);
return samplingRateCollection;
}
static FormatTraits::Collection formatsFromString(const std::string &formats, const char *del = "|")
{
FormatTraits::Collection formatCollection;
- // by convention, "0' in the first entry in mFormats indicates the supported formats
- // should be read from the output stream after it is opened for the first time
- if (formats == DYNAMIC_VALUE_TAG) {
- formatCollection.add(gDynamicFormat);
- } else {
- FormatConverter::collectionFromString(formats, formatCollection, del);
- }
+ FormatConverter::collectionFromString(formats, formatCollection, del);
return formatCollection;
}
static audio_format_t formatFromString(const std::string &literalFormat)
{
audio_format_t format;
- // by convention, "0' in the first entry in literalFormat indicates the supported formats
- // should be read from the output stream after it is opened for the first time
- if (literalFormat == DYNAMIC_VALUE_TAG) {
- return gDynamicFormat;
- } else {
- FormatConverter::fromString(literalFormat, format);
- }
+ FormatConverter::fromString(literalFormat, format);
return format;
}
static audio_channel_mask_t channelMaskFromString(const std::string &literalChannels)
{
audio_channel_mask_t channels;
- // by convention, "0' in the first entry in literalChannels indicates the supported channels
- // should be read from the output stream after it is opened for the first time
- if (literalChannels == DYNAMIC_VALUE_TAG) {
- return gDynamicChannelMask;
- }
if (!OutputChannelConverter::fromString(literalChannels, channels) ||
!InputChannelConverter::fromString(literalChannels, channels)) {
return AUDIO_CHANNEL_INVALID;
@@ -194,13 +167,9 @@
const char *del = "|")
{
ChannelTraits::Collection channelMaskCollection;
- if (channels == DYNAMIC_VALUE_TAG) {
- channelMaskCollection.add(gDynamicChannelMask);
- } else {
- OutputChannelConverter::collectionFromString(channels, channelMaskCollection, del);
- InputChannelConverter::collectionFromString(channels, channelMaskCollection, del);
- ChannelIndexConverter::collectionFromString(channels, channelMaskCollection, del);
- }
+ OutputChannelConverter::collectionFromString(channels, channelMaskCollection, del);
+ InputChannelConverter::collectionFromString(channels, channelMaskCollection, del);
+ ChannelIndexConverter::collectionFromString(channels, channelMaskCollection, del);
return channelMaskCollection;
}
@@ -208,12 +177,8 @@
const char *del = "|")
{
InputChannelTraits::Collection inputChannelMaskCollection;
- if (inChannels == DYNAMIC_VALUE_TAG) {
- inputChannelMaskCollection.add(gDynamicChannelMask);
- } else {
- InputChannelConverter::collectionFromString(inChannels, inputChannelMaskCollection, del);
- ChannelIndexConverter::collectionFromString(inChannels, inputChannelMaskCollection, del);
- }
+ InputChannelConverter::collectionFromString(inChannels, inputChannelMaskCollection, del);
+ ChannelIndexConverter::collectionFromString(inChannels, inputChannelMaskCollection, del);
return inputChannelMaskCollection;
}
@@ -221,14 +186,8 @@
const char *del = "|")
{
OutputChannelTraits::Collection outputChannelMaskCollection;
- // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
- // masks should be read from the output stream after it is opened for the first time
- if (outChannels == DYNAMIC_VALUE_TAG) {
- outputChannelMaskCollection.add(gDynamicChannelMask);
- } else {
- OutputChannelConverter::collectionFromString(outChannels, outputChannelMaskCollection, del);
- ChannelIndexConverter::collectionFromString(outChannels, outputChannelMaskCollection, del);
- }
+ OutputChannelConverter::collectionFromString(outChannels, outputChannelMaskCollection, del);
+ ChannelIndexConverter::collectionFromString(outChannels, outputChannelMaskCollection, del);
return outputChannelMaskCollection;
}
diff --git a/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
index aac9e4d..0a27947 100644
--- a/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
+++ b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
@@ -65,3 +65,7 @@
#define GAIN_STEP_VALUE "step_value_mB"
#define GAIN_MIN_RAMP_MS "min_ramp_ms"
#define GAIN_MAX_RAMP_MS "max_ramp_ms"
+
+#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and
+ // "formats" in outputs descriptors indicating that supported
+ // values should be queried after opening the output.
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
index e828cc0..9b6469c 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -28,14 +28,11 @@
AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
: mIoHandle(0),
- mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
- mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false), mId(0),
- mOpenRefCount(0)
+ mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0),
+ mProfile(profile), mId(0)
{
if (profile != NULL) {
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
+ profile->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
if (profile->mGains.size() > 0) {
profile->mGains[0]->getDefaultConfig(&mGain);
}
@@ -56,20 +53,9 @@
return mProfile->getModuleHandle();
}
-void AudioInputDescriptor::changeOpenRefCount(int delta)
-{
- if ((delta + (int)mOpenRefCount) < 0) {
- ALOGW("changeOpenRefCount() invalid delta %d, refCount %d", delta, mOpenRefCount);
- mOpenRefCount = 0;
- return;
- }
- mOpenRefCount += delta;
- ALOGV("changeOpenRefCount() count %d", mOpenRefCount);
-}
-
uint32_t AudioInputDescriptor::getOpenRefCount() const
{
- return mOpenRefCount;
+ return mSessions.getOpenCount();
}
audio_port_handle_t AudioInputDescriptor::getId() const
@@ -77,6 +63,13 @@
return mId;
}
+audio_source_t AudioInputDescriptor::inputSource() const
+{
+ // TODO: return highest priority input source
+ return mSessions.size() > 0 ? mSessions.valueAt(0)->inputSource() :
+ AUDIO_SOURCE_DEFAULT;
+}
+
void AudioInputDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
@@ -95,7 +88,7 @@
dstConfig->type = AUDIO_PORT_TYPE_MIX;
dstConfig->ext.mix.hw_module = getModuleHandle();
dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.source = mInputSource;
+ dstConfig->ext.mix.usecase.source = inputSource();
}
void AudioInputDescriptor::toAudioPort(struct audio_port *port) const
@@ -130,6 +123,40 @@
mPreemptedSessions.clear();
}
+bool AudioInputDescriptor::isActive() const {
+ return mSessions.hasActiveSession();
+}
+
+bool AudioInputDescriptor::isSourceActive(audio_source_t source) const
+{
+ return mSessions.isSourceActive(source);
+}
+
+bool AudioInputDescriptor::isSoundTrigger() const {
+ // sound trigger and non sound trigger sessions are not mixed
+ // on a given input
+ return mSessions.valueAt(0)->isSoundTrigger();
+}
+
+sp<AudioSession> AudioInputDescriptor::getAudioSession(
+ audio_session_t session) const {
+ return mSessions.valueFor(session);
+}
+
+AudioSessionCollection AudioInputDescriptor::getActiveAudioSessions() const
+{
+ return mSessions.getActiveSessions();
+}
+
+status_t AudioInputDescriptor::addAudioSession(audio_session_t session,
+ const sp<AudioSession>& audioSession) {
+ return mSessions.addSession(session, audioSession);
+}
+
+status_t AudioInputDescriptor::removeAudioSession(audio_session_t session) {
+ return mSessions.removeSession(session);
+}
+
status_t AudioInputDescriptor::dump(int fd)
{
const size_t SIZE = 256;
@@ -146,13 +173,11 @@
result.append(buffer);
snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
result.append(buffer);
- snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
- result.append(buffer);
- snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
- result.append(buffer);
write(fd, result.string(), result.size());
+ mSessions.dump(fd, 1);
+
return NO_ERROR;
}
@@ -160,10 +185,7 @@
{
for (size_t i = 0; i < size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = valueAt(i);
- if (inputDescriptor->mRefCount == 0) {
- continue;
- }
- if (inputDescriptor->mInputSource == (int)source) {
+ if (inputDescriptor->isSourceActive(source)) {
return true;
}
}
@@ -186,8 +208,8 @@
{
uint32_t count = 0;
for (size_t i = 0; i < size(); i++) {
- const sp<AudioInputDescriptor> desc = valueAt(i);
- if (desc->mRefCount > 0) {
+ const sp<AudioInputDescriptor> inputDescriptor = valueAt(i);
+ if (inputDescriptor->isActive()) {
count++;
}
}
@@ -197,9 +219,10 @@
audio_io_handle_t AudioInputCollection::getActiveInput(bool ignoreVirtualInputs)
{
for (size_t i = 0; i < size(); i++) {
- const sp<AudioInputDescriptor> input_descriptor = valueAt(i);
- if ((input_descriptor->mRefCount > 0)
- && (!ignoreVirtualInputs || !is_virtual_input_device(input_descriptor->mDevice))) {
+ const sp<AudioInputDescriptor> inputDescriptor = valueAt(i);
+ if ((inputDescriptor->isActive())
+ && (!ignoreVirtualInputs ||
+ !is_virtual_input_device(inputDescriptor->mDevice))) {
return keyAt(i);
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 223fe80..5d0f03f 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -47,9 +47,7 @@
mStrategyMutedByDevice[i] = false;
}
if (port != NULL) {
- mSamplingRate = port->pickSamplingRate();
- mFormat = port->pickFormat();
- mChannelMask = port->pickChannelMask();
+ port->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
if (port->mGains.size() > 0) {
port->mGains[0]->getDefaultConfig(&mGain);
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index 1713095..cde0923 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -16,7 +16,6 @@
#define LOG_TAG "APM::AudioPort"
//#define LOG_NDEBUG 0
-#include <media/AudioResamplerPublic.h>
#include "TypeConverter.h"
#include "AudioPort.h"
#include "HwModule.h"
@@ -68,420 +67,165 @@
void AudioPort::toAudioPort(struct audio_port *port) const
{
+ // TODO: update this function once audio_port structure reflects the new profile definition.
+ // For compatibility reason: flatening the AudioProfile into audio_port structure.
+ SortedVector<audio_format_t> flatenedFormats;
+ SampleRateVector flatenedRates;
+ ChannelsVector flatenedChannels;
+ for (size_t profileIndex = 0; profileIndex < mProfiles.size(); profileIndex++) {
+ if (mProfiles[profileIndex]->isValid()) {
+ audio_format_t formatToExport = mProfiles[profileIndex]->getFormat();
+ const SampleRateVector &ratesToExport = mProfiles[profileIndex]->getSampleRates();
+ const ChannelsVector &channelsToExport = mProfiles[profileIndex]->getChannels();
+
+ if (flatenedFormats.indexOf(formatToExport) < 0) {
+ flatenedFormats.add(formatToExport);
+ }
+ for (size_t rateIndex = 0; rateIndex < ratesToExport.size(); rateIndex++) {
+ uint32_t rate = ratesToExport[rateIndex];
+ if (flatenedRates.indexOf(rate) < 0) {
+ flatenedRates.add(rate);
+ }
+ }
+ for (size_t chanIndex = 0; chanIndex < channelsToExport.size(); chanIndex++) {
+ audio_channel_mask_t channels = channelsToExport[chanIndex];
+ if (flatenedChannels.indexOf(channels) < 0) {
+ flatenedChannels.add(channels);
+ }
+ }
+ if (flatenedRates.size() > AUDIO_PORT_MAX_SAMPLING_RATES ||
+ flatenedChannels.size() > AUDIO_PORT_MAX_CHANNEL_MASKS ||
+ flatenedFormats.size() > AUDIO_PORT_MAX_FORMATS) {
+ ALOGE("%s: bailing out: cannot export profiles to port config", __FUNCTION__);
+ return;
+ }
+ }
+ }
port->role = mRole;
port->type = mType;
strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
- unsigned int i;
- for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
- if (mSamplingRates[i] != 0) {
- port->sample_rates[i] = mSamplingRates[i];
- }
+ port->num_sample_rates = flatenedRates.size();
+ port->num_channel_masks = flatenedChannels.size();
+ port->num_formats = flatenedFormats.size();
+ for (size_t i = 0; i < flatenedRates.size(); i++) {
+ port->sample_rates[i] = flatenedRates[i];
}
- port->num_sample_rates = i;
- for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
- if (mChannelMasks[i] != 0) {
- port->channel_masks[i] = mChannelMasks[i];
- }
+ for (size_t i = 0; i < flatenedChannels.size(); i++) {
+ port->channel_masks[i] = flatenedChannels[i];
}
- port->num_channel_masks = i;
- for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
- if (mFormats[i] != 0) {
- port->formats[i] = mFormats[i];
- }
+ for (size_t i = 0; i < flatenedFormats.size(); i++) {
+ port->formats[i] = flatenedFormats[i];
}
- port->num_formats = i;
ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+ uint32_t i;
for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
port->gains[i] = mGains[i]->getGain();
}
port->num_gains = i;
}
-void AudioPort::importAudioPort(const sp<AudioPort> port) {
- for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
- const uint32_t rate = port->mSamplingRates.itemAt(k);
- if (rate != 0) { // skip "dynamic" rates
- bool hasRate = false;
- for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
- if (rate == mSamplingRates.itemAt(l)) {
- hasRate = true;
+void AudioPort::importAudioPort(const sp<AudioPort> port)
+{
+ size_t indexToImport;
+ for (indexToImport = 0; indexToImport < port->mProfiles.size(); indexToImport++) {
+ const sp<AudioProfile> &profileToImport = port->mProfiles[indexToImport];
+ if (profileToImport->isValid()) {
+ // Import only valid port, i.e. valid format, non empty rates and channels masks
+ bool hasSameProfile = false;
+ for (size_t profileIndex = 0; profileIndex < mProfiles.size(); profileIndex++) {
+ if (*mProfiles[profileIndex] == *profileToImport) {
+ // never import a profile twice
+ hasSameProfile = true;
break;
}
}
- if (!hasRate) { // never import a sampling rate twice
- mSamplingRates.add(rate);
+ if (hasSameProfile) { // never import a same profile twice
+ continue;
}
- }
- }
- for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
- const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
- if (mask != 0) { // skip "dynamic" masks
- bool hasMask = false;
- for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
- if (mask == mChannelMasks.itemAt(l)) {
- hasMask = true;
- break;
- }
- }
- if (!hasMask) { // never import a channel mask twice
- mChannelMasks.add(mask);
- }
- }
- }
- for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
- const audio_format_t format = port->mFormats.itemAt(k);
- if (format != 0) { // skip "dynamic" formats
- bool hasFormat = false;
- for (size_t l = 0 ; l < mFormats.size() ; l++) {
- if (format == mFormats.itemAt(l)) {
- hasFormat = true;
- break;
- }
- }
- if (!hasFormat) { // never import a format twice
- mFormats.add(format);
- }
+ addAudioProfile(profileToImport);
}
}
}
-void AudioPort::clearCapabilities() {
- mChannelMasks.clear();
- mFormats.clear();
- mSamplingRates.clear();
-}
-
-void AudioPort::setSupportedFormats(const Vector <audio_format_t> &formats)
+void AudioPort::pickSamplingRate(uint32_t &pickedRate,const SampleRateVector &samplingRates) const
{
- mFormats = formats;
- // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry.
- // TODO: compareFormats could be a lambda to convert between pointer-to-format to format:
- // [](const audio_format_t *format1, const audio_format_t *format2) {
- // return compareFormats(*format1, *format2);
- // }
- mFormats.sort(compareFormats);
-}
-
-status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if (mSamplingRates[i] == samplingRate) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = samplingRate;
- }
- return NO_ERROR;
- }
-
- // Search for the closest supported sampling rate that is above (preferred)
- // or below (acceptable) the desired sampling rate, within a permitted ratio.
- // The sampling rates do not need to be sorted in ascending order.
- ssize_t maxBelow = -1;
- ssize_t minAbove = -1;
- uint32_t candidate;
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- candidate = mSamplingRates[i];
- if (candidate == samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- // candidate < desired
- if (candidate < samplingRate) {
- if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
- maxBelow = i;
- }
- // candidate > desired
- } else {
- if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
- minAbove = i;
- }
- }
- }
-
- // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
- if (minAbove >= 0) {
- candidate = mSamplingRates[minAbove];
- if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // But if we have to up-sample from a lower sampling rate, that's OK.
- if (maxBelow >= 0) {
- candidate = mSamplingRates[maxBelow];
- if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // leave updatedSamplingRate unmodified
- return BAD_VALUE;
-}
-
-status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- if (mChannelMasks[i] == channelMask) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask,
- audio_channel_mask_t *updatedChannelMask) const
-{
- if (mChannelMasks.isEmpty()) {
- if (updatedChannelMask != NULL) {
- *updatedChannelMask = channelMask;
- }
- return NO_ERROR;
- }
-
- const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
- const bool isIndex = audio_channel_mask_get_representation(channelMask)
- == AUDIO_CHANNEL_REPRESENTATION_INDEX;
- int bestMatch = 0;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- audio_channel_mask_t supported = mChannelMasks[i];
- if (supported == channelMask) {
- // Exact matches always taken.
- if (updatedChannelMask != NULL) {
- *updatedChannelMask = channelMask;
- }
- return NO_ERROR;
- }
-
- // AUDIO_CHANNEL_NONE (value: 0) is used for dynamic channel support
- if (isRecordThread && supported != AUDIO_CHANNEL_NONE) {
- // Approximate (best) match:
- // The match score measures how well the supported channel mask matches the
- // desired mask, where increasing-is-better.
- //
- // TODO: Some tweaks may be needed.
- // Should be a static function of the data processing library.
- //
- // In priority:
- // match score = 1000 if legacy channel conversion equivalent (always prefer this)
- // OR
- // match score += 100 if the channel mask representations match
- // match score += number of channels matched.
- //
- // If there are no matched channels, the mask may still be accepted
- // but the playback or record will be silent.
- const bool isSupportedIndex = (audio_channel_mask_get_representation(supported)
- == AUDIO_CHANNEL_REPRESENTATION_INDEX);
- int match;
- if (isIndex && isSupportedIndex) {
- // index equivalence
- match = 100 + __builtin_popcount(
- audio_channel_mask_get_bits(channelMask)
- & audio_channel_mask_get_bits(supported));
- } else if (isIndex && !isSupportedIndex) {
- const uint32_t equivalentBits =
- (1 << audio_channel_count_from_in_mask(supported)) - 1 ;
- match = __builtin_popcount(
- audio_channel_mask_get_bits(channelMask) & equivalentBits);
- } else if (!isIndex && isSupportedIndex) {
- const uint32_t equivalentBits =
- (1 << audio_channel_count_from_in_mask(channelMask)) - 1;
- match = __builtin_popcount(
- equivalentBits & audio_channel_mask_get_bits(supported));
- } else {
- // positional equivalence
- match = 100 + __builtin_popcount(
- audio_channel_mask_get_bits(channelMask)
- & audio_channel_mask_get_bits(supported));
- switch (supported) {
- case AUDIO_CHANNEL_IN_FRONT_BACK:
- case AUDIO_CHANNEL_IN_STEREO:
- if (channelMask == AUDIO_CHANNEL_IN_MONO) {
- match = 1000;
- }
- break;
- case AUDIO_CHANNEL_IN_MONO:
- if (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
- || channelMask == AUDIO_CHANNEL_IN_STEREO) {
- match = 1000;
- }
- break;
- default:
- break;
- }
- }
- if (match > bestMatch) {
- bestMatch = match;
- if (updatedChannelMask != NULL) {
- *updatedChannelMask = supported;
- } else {
- return NO_ERROR; // any match will do in this case.
- }
- }
- }
- }
- return bestMatch > 0 ? NO_ERROR : BAD_VALUE;
-}
-
-status_t AudioPort::checkExactFormat(audio_format_t format) const
-{
- if (mFormats.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if (mFormats[i] == format) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPort::checkCompatibleFormat(audio_format_t format, audio_format_t *updatedFormat)
- const
-{
- if (mFormats.isEmpty()) {
- if (updatedFormat != NULL) {
- *updatedFormat = format;
- }
- return NO_ERROR;
- }
-
- const bool checkInexact = // when port is input and format is linear pcm
- mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK
- && audio_is_linear_pcm(format);
-
- // iterate from best format to worst format (reverse order)
- for (ssize_t i = mFormats.size() - 1; i >= 0 ; --i) {
- if (mFormats[i] == format ||
- (checkInexact
- && mFormats[i] != AUDIO_FORMAT_DEFAULT
- && audio_is_linear_pcm(mFormats[i]))) {
- // for inexact checks we take the first linear pcm format due to sorting.
- if (updatedFormat != NULL) {
- *updatedFormat = mFormats[i];
- }
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-uint32_t AudioPort::pickSamplingRate() const
-{
- // special case for uninitialized dynamic profile
- if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
- return 0;
- }
-
+ pickedRate = 0;
// For direct outputs, pick minimum sampling rate: this helps ensuring that the
// channel count / sampling rate combination chosen will be supported by the connected
// sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ if (isDirectOutput()) {
uint32_t samplingRate = UINT_MAX;
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
- samplingRate = mSamplingRates[i];
+ for (size_t i = 0; i < samplingRates.size(); i ++) {
+ if ((samplingRates[i] < samplingRate) && (samplingRates[i] > 0)) {
+ samplingRate = samplingRates[i];
}
}
- return (samplingRate == UINT_MAX) ? 0 : samplingRate;
- }
+ pickedRate = (samplingRate == UINT_MAX) ? 0 : samplingRate;
+ } else {
+ uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
- uint32_t samplingRate = 0;
- uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
-
- // For mixed output and inputs, use max mixer sampling rates. Do not
- // limit sampling rate otherwise
- // For inputs, also see checkCompatibleSamplingRate().
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxRate = UINT_MAX;
- }
- // TODO: should mSamplingRates[] be ordered in terms of our preference
- // and we return the first (and hence most preferred) match? This is of concern if
- // we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
- samplingRate = mSamplingRates[i];
+ // For mixed output and inputs, use max mixer sampling rates. Do not
+ // limit sampling rate otherwise
+ // For inputs, also see checkCompatibleSamplingRate().
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxRate = UINT_MAX;
+ }
+ // TODO: should mSamplingRates[] be ordered in terms of our preference
+ // and we return the first (and hence most preferred) match? This is of concern if
+ // we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
+ for (size_t i = 0; i < samplingRates.size(); i ++) {
+ if ((samplingRates[i] > pickedRate) && (samplingRates[i] <= maxRate)) {
+ pickedRate = samplingRates[i];
+ }
}
}
- return samplingRate;
}
-audio_channel_mask_t AudioPort::pickChannelMask() const
+void AudioPort::pickChannelMask(audio_channel_mask_t &pickedChannelMask,
+ const ChannelsVector &channelMasks) const
{
- // special case for uninitialized dynamic profile
- if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
- return AUDIO_CHANNEL_NONE;
- }
- audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
-
+ pickedChannelMask = AUDIO_CHANNEL_NONE;
// For direct outputs, pick minimum channel count: this helps ensuring that the
// channel count / sampling rate combination chosen will be supported by the connected
// sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ if (isDirectOutput()) {
uint32_t channelCount = UINT_MAX;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ for (size_t i = 0; i < channelMasks.size(); i ++) {
uint32_t cnlCount;
if (useInputChannelMask()) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
} else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
}
if ((cnlCount < channelCount) && (cnlCount > 0)) {
- channelMask = mChannelMasks[i];
+ pickedChannelMask = channelMasks[i];
channelCount = cnlCount;
}
}
- return channelMask;
- }
+ } else {
+ uint32_t channelCount = 0;
+ uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
- uint32_t channelCount = 0;
- uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
-
- // For mixed output and inputs, use max mixer channel count. Do not
- // limit channel count otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxCount = UINT_MAX;
- }
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (useInputChannelMask()) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ // For mixed output and inputs, use max mixer channel count. Do not
+ // limit channel count otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxCount = UINT_MAX;
}
- if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
+ for (size_t i = 0; i < channelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (useInputChannelMask()) {
+ cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
+ }
+ if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+ pickedChannelMask = channelMasks[i];
+ channelCount = cnlCount;
+ }
}
}
- return channelMask;
}
/* format in order of increasing preference */
@@ -494,8 +238,7 @@
AUDIO_FORMAT_PCM_FLOAT,
};
-int AudioPort::compareFormats(audio_format_t format1,
- audio_format_t format2)
+int AudioPort::compareFormats(audio_format_t format1, audio_format_t format2)
{
// NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
// compressed format and better than any PCM format. This is by design of pickFormat()
@@ -525,36 +268,49 @@
return index1 - index2;
}
-audio_format_t AudioPort::pickFormat() const
+void AudioPort::pickAudioProfile(uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format) const
{
- // special case for uninitialized dynamic profile
- if (mFormats.size() == 1 && mFormats[0] == 0) {
- return AUDIO_FORMAT_DEFAULT;
- }
+ format = AUDIO_FORMAT_DEFAULT;
+ samplingRate = 0;
+ channelMask = AUDIO_CHANNEL_NONE;
- audio_format_t format = AUDIO_FORMAT_DEFAULT;
- audio_format_t bestFormat =
- AudioPort::sPcmFormatCompareTable[
- ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1];
- // For mixed output and inputs, use best mixer output format. Do not
- // limit format otherwise
- if ((mType != AUDIO_PORT_TYPE_MIX) ||
- ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
+ // special case for uninitialized dynamic profile
+ if (!mProfiles.hasValidProfile()) {
+ return;
+ }
+ audio_format_t bestFormat = sPcmFormatCompareTable[ARRAY_SIZE(sPcmFormatCompareTable) - 1];
+ // For mixed output and inputs, use best mixer output format.
+ // Do not limit format otherwise
+ if ((mType != AUDIO_PORT_TYPE_MIX) || isDirectOutput()) {
bestFormat = AUDIO_FORMAT_INVALID;
}
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if ((compareFormats(mFormats[i], format) > 0) &&
- (compareFormats(mFormats[i], bestFormat) <= 0)) {
- format = mFormats[i];
+ for (size_t i = 0; i < mProfiles.size(); i ++) {
+ if (!mProfiles[i]->isValid()) {
+ continue;
+ }
+ audio_format_t formatToCompare = mProfiles[i]->getFormat();
+ if ((compareFormats(formatToCompare, format) > 0) &&
+ (compareFormats(formatToCompare, bestFormat) <= 0)) {
+ uint32_t pickedSamplingRate = 0;
+ audio_channel_mask_t pickedChannelMask = AUDIO_CHANNEL_NONE;
+ pickChannelMask(pickedChannelMask, mProfiles[i]->getChannels());
+ pickSamplingRate(pickedSamplingRate, mProfiles[i]->getSampleRates());
+
+ if (formatToCompare != AUDIO_FORMAT_DEFAULT && pickedChannelMask != AUDIO_CHANNEL_NONE
+ && pickedSamplingRate != 0) {
+ format = formatToCompare;
+ channelMask = pickedChannelMask;
+ samplingRate = pickedSamplingRate;
+ // TODO: shall we return on the first one or still trying to pick a better Profile?
+ }
}
}
- return format;
}
-status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig,
- int index) const
+status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig, int index) const
{
if (index < 0 || (size_t)index >= mGains.size()) {
return BAD_VALUE;
@@ -562,7 +318,7 @@
return mGains[index]->checkConfig(gainConfig);
}
-void AudioPort::dump(int fd, int spaces) const
+void AudioPort::dump(int fd, int spaces, bool verbose) const
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -571,66 +327,17 @@
if (mName.length() != 0) {
snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
result.append(buffer);
+ write(fd, result.string(), result.size());
}
+ if (verbose) {
+ mProfiles.dump(fd, spaces);
- if (mSamplingRates.size() != 0) {
- snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- if (i == 0 && mSamplingRates[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ if (mGains.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
+ write(fd, buffer, strlen(buffer) + 1);
+ for (size_t i = 0; i < mGains.size(); i++) {
+ mGains[i]->dump(fd, spaces + 2, i);
}
- result.append(buffer);
- result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mChannelMasks.size() != 0) {
- snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
-
- if (i == 0 && mChannelMasks[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
- }
- result.append(buffer);
- result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mFormats.size() != 0) {
- snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mFormats.size(); i++) {
- std::string formatLiteral;
- bool success = FormatConverter::toString(mFormats[i], formatLiteral);
- if (i == 0 && !success) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- if (!success) {
- snprintf(buffer, SIZE, "%#x", mFormats[i]);
- } else {
- snprintf(buffer, SIZE, "%s", formatLiteral.c_str());
- }
- }
- result.append(buffer);
- result.append(i == (mFormats.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
- write(fd, result.string(), result.size());
- if (mGains.size() != 0) {
- snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
- write(fd, buffer, strlen(buffer) + 1);
- for (size_t i = 0; i < mGains.size(); i++) {
- mGains[i]->dump(fd, spaces + 2, i);
}
}
}
@@ -650,9 +357,8 @@
mGain.index = -1;
}
-status_t AudioPortConfig::applyAudioPortConfig(
- const struct audio_port_config *config,
- struct audio_port_config *backupConfig)
+status_t AudioPortConfig::applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig)
{
struct audio_port_config localBackupConfig;
status_t status = NO_ERROR;
@@ -665,25 +371,19 @@
status = NO_INIT;
goto exit;
}
+ status = audioport->checkExactAudioProfile(config->sample_rate,
+ config->channel_mask,
+ config->format);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- status = audioport->checkExactSamplingRate(config->sample_rate);
- if (status != NO_ERROR) {
- goto exit;
- }
mSamplingRate = config->sample_rate;
}
if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- status = audioport->checkExactChannelMask(config->channel_mask);
- if (status != NO_ERROR) {
- goto exit;
- }
mChannelMask = config->channel_mask;
}
if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- status = audioport->checkExactFormat(config->format);
- if (status != NO_ERROR) {
- goto exit;
- }
mFormat = config->format;
}
if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
new file mode 100644
index 0000000..c599665
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
@@ -0,0 +1,288 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioProfile"
+//#define LOG_NDEBUG 0
+
+#include "AudioProfile.h"
+#include "AudioPort.h"
+#include "HwModule.h"
+#include "AudioGain.h"
+#include <utils/SortedVector.h>
+#include "TypeConverter.h"
+#include <media/AudioResamplerPublic.h>
+#include <algorithm>
+
+namespace android {
+
+status_t AudioProfile::checkExact(uint32_t samplingRate, audio_channel_mask_t channelMask,
+ audio_format_t format) const
+{
+ if (format == mFormat &&
+ (mChannelMasks.isEmpty() || supportsChannels(channelMask)) &&
+ (mSamplingRates.isEmpty() || supportsRate(samplingRate))) {
+ return NO_ERROR;
+ }
+ return BAD_VALUE;
+}
+
+template <typename T>
+bool operator == (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+ if (left.size() != right.size()) {
+ return false;
+ }
+ for(size_t index = 0; index < right.size(); index++) {
+ if (left[index] != right[index]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+bool operator == (const AudioProfile &left, const AudioProfile &compareTo)
+{
+ return (left.getFormat() == compareTo.getFormat()) &&
+ (left.getChannels() == compareTo.getChannels()) &&
+ (left.getSampleRates() == compareTo.getSampleRates());
+}
+
+status_t AudioProfile::checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t &updatedSamplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ updatedSamplingRate = samplingRate;
+ return NO_ERROR;
+ }
+ // Search for the closest supported sampling rate that is above (preferred)
+ // or below (acceptable) the desired sampling rate, within a permitted ratio.
+ // The sampling rates are sorted in ascending order.
+ size_t orderOfDesiredRate = mSamplingRates.orderOf(samplingRate);
+
+ // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
+ if (orderOfDesiredRate < mSamplingRates.size()) {
+ uint32_t candidate = mSamplingRates[orderOfDesiredRate];
+ if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
+ updatedSamplingRate = candidate;
+ return NO_ERROR;
+ }
+ }
+ // But if we have to up-sample from a lower sampling rate, that's OK.
+ if (orderOfDesiredRate != 0) {
+ uint32_t candidate = mSamplingRates[orderOfDesiredRate - 1];
+ if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) {
+ updatedSamplingRate = candidate;
+ return NO_ERROR;
+ }
+ }
+ // leave updatedSamplingRate unmodified
+ return BAD_VALUE;
+}
+
+status_t AudioProfile::checkCompatibleChannelMask(audio_channel_mask_t channelMask,
+ audio_channel_mask_t &updatedChannelMask,
+ audio_port_type_t portType,
+ audio_port_role_t portRole) const
+{
+ if (mChannelMasks.isEmpty()) {
+ updatedChannelMask = channelMask;
+ return NO_ERROR;
+ }
+ const bool isRecordThread = portType == AUDIO_PORT_TYPE_MIX && portRole == AUDIO_PORT_ROLE_SINK;
+ const bool isIndex = audio_channel_mask_get_representation(channelMask)
+ == AUDIO_CHANNEL_REPRESENTATION_INDEX;
+ int bestMatch = 0;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ audio_channel_mask_t supported = mChannelMasks[i];
+ if (supported == channelMask) {
+ // Exact matches always taken.
+ updatedChannelMask = channelMask;
+ return NO_ERROR;
+ }
+
+ // AUDIO_CHANNEL_NONE (value: 0) is used for dynamic channel support
+ if (isRecordThread && supported != AUDIO_CHANNEL_NONE) {
+ // Approximate (best) match:
+ // The match score measures how well the supported channel mask matches the
+ // desired mask, where increasing-is-better.
+ //
+ // TODO: Some tweaks may be needed.
+ // Should be a static function of the data processing library.
+ //
+ // In priority:
+ // match score = 1000 if legacy channel conversion equivalent (always prefer this)
+ // OR
+ // match score += 100 if the channel mask representations match
+ // match score += number of channels matched.
+ //
+ // If there are no matched channels, the mask may still be accepted
+ // but the playback or record will be silent.
+ const bool isSupportedIndex = (audio_channel_mask_get_representation(supported)
+ == AUDIO_CHANNEL_REPRESENTATION_INDEX);
+ int match;
+ if (isIndex && isSupportedIndex) {
+ // index equivalence
+ match = 100 + __builtin_popcount(
+ audio_channel_mask_get_bits(channelMask)
+ & audio_channel_mask_get_bits(supported));
+ } else if (isIndex && !isSupportedIndex) {
+ const uint32_t equivalentBits =
+ (1 << audio_channel_count_from_in_mask(supported)) - 1 ;
+ match = __builtin_popcount(
+ audio_channel_mask_get_bits(channelMask) & equivalentBits);
+ } else if (!isIndex && isSupportedIndex) {
+ const uint32_t equivalentBits =
+ (1 << audio_channel_count_from_in_mask(channelMask)) - 1;
+ match = __builtin_popcount(
+ equivalentBits & audio_channel_mask_get_bits(supported));
+ } else {
+ // positional equivalence
+ match = 100 + __builtin_popcount(
+ audio_channel_mask_get_bits(channelMask)
+ & audio_channel_mask_get_bits(supported));
+ switch (supported) {
+ case AUDIO_CHANNEL_IN_FRONT_BACK:
+ case AUDIO_CHANNEL_IN_STEREO:
+ if (channelMask == AUDIO_CHANNEL_IN_MONO) {
+ match = 1000;
+ }
+ break;
+ case AUDIO_CHANNEL_IN_MONO:
+ if (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
+ || channelMask == AUDIO_CHANNEL_IN_STEREO) {
+ match = 1000;
+ }
+ break;
+ default:
+ break;
+ }
+ }
+ if (match > bestMatch) {
+ bestMatch = match;
+ updatedChannelMask = supported;
+ }
+ }
+ }
+ return bestMatch > 0 ? NO_ERROR : BAD_VALUE;
+}
+
+void AudioProfile::dump(int fd, int spaces) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%s%s%s\n", mIsDynamicFormat ? "[dynamic format]" : "",
+ mIsDynamicChannels ? "[dynamic channels]" : "",
+ mIsDynamicRate ? "[dynamic rates]" : "");
+ result.append(buffer);
+ if (mName.length() != 0) {
+ snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
+ result.append(buffer);
+ }
+ std::string formatLiteral;
+ if (FormatConverter::toString(mFormat, formatLiteral)) {
+ snprintf(buffer, SIZE, "%*s- format: %s\n", spaces, "", formatLiteral.c_str());
+ result.append(buffer);
+ }
+ if (!mSamplingRates.isEmpty()) {
+ snprintf(buffer, SIZE, "%*s- sampling rates:", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ result.append(buffer);
+ result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (!mChannelMasks.isEmpty()) {
+ snprintf(buffer, SIZE, "%*s- channel masks:", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+ result.append(buffer);
+ result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+ write(fd, result.string(), result.size());
+}
+
+status_t AudioProfileVector::checkExactProfile(uint32_t samplingRate,
+ audio_channel_mask_t channelMask,
+ audio_format_t format) const
+{
+ if (isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < size(); i++) {
+ const sp<AudioProfile> profile = itemAt(i);
+ if (profile->checkExact(samplingRate, channelMask, format) == NO_ERROR) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioProfileVector::checkCompatibleProfile(uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format,
+ audio_port_type_t portType,
+ audio_port_role_t portRole) const
+{
+ if (isEmpty()) {
+ return NO_ERROR;
+ }
+
+ const bool checkInexact = // when port is input and format is linear pcm
+ portType == AUDIO_PORT_TYPE_MIX && portRole == AUDIO_PORT_ROLE_SINK
+ && audio_is_linear_pcm(format);
+
+ // iterate from best format to worst format (reverse order)
+ for (ssize_t i = size() - 1; i >= 0 ; --i) {
+ const sp<AudioProfile> profile = itemAt(i);
+ audio_format_t formatToCompare = profile->getFormat();
+ if (formatToCompare == format ||
+ (checkInexact
+ && formatToCompare != AUDIO_FORMAT_DEFAULT
+ && audio_is_linear_pcm(formatToCompare))) {
+ // Compatible profile has been found, checks if this profile has compatible
+ // rate and channels as well
+ audio_channel_mask_t updatedChannels;
+ uint32_t updatedRate;
+ if (profile->checkCompatibleChannelMask(channelMask, updatedChannels,
+ portType, portRole) == NO_ERROR &&
+ profile->checkCompatibleSamplingRate(samplingRate, updatedRate) == NO_ERROR) {
+ // for inexact checks we take the first linear pcm format due to sorting.
+ format = formatToCompare;
+ channelMask = updatedChannels;
+ samplingRate = updatedRate;
+ return NO_ERROR;
+ }
+ }
+ }
+ return BAD_VALUE;
+}
+
+int AudioProfileVector::compareFormats(const sp<AudioProfile> *profile1,
+ const sp<AudioProfile> *profile2)
+{
+ return AudioPort::compareFormats((*profile1)->getFormat(), (*profile2)->getFormat());
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
new file mode 100644
index 0000000..2cec951
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
@@ -0,0 +1,198 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioSession"
+//#define LOG_NDEBUG 0
+
+#include "AudioSession.h"
+#include "AudioGain.h"
+#include "TypeConverter.h"
+#include <cutils/log.h>
+#include <utils/String8.h>
+
+namespace android {
+
+AudioSession::AudioSession(audio_session_t session,
+ audio_source_t inputSource,
+ audio_format_t format,
+ uint32_t sampleRate,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ uid_t uid,
+ bool isSoundTrigger) :
+ mSession(session), mInputSource(inputSource),
+ mFormat(format), mSampleRate(sampleRate), mChannelMask(channelMask),
+ mFlags(flags), mUid(uid), mIsSoundTrigger(isSoundTrigger),
+ mOpenCount(1), mActiveCount(0)
+{
+}
+
+uint32_t AudioSession::changeOpenCount(int delta)
+{
+ if ((delta + (int)mOpenCount) < 0) {
+ ALOGW("%s invalid delta %d, open count %d",
+ __FUNCTION__, delta, mOpenCount);
+ mOpenCount = (uint32_t)(-delta);
+ }
+ mOpenCount += delta;
+ ALOGV("%s open count %d", __FUNCTION__, mOpenCount);
+ return mOpenCount;
+}
+
+uint32_t AudioSession::changeActiveCount(int delta)
+{
+ if ((delta + (int)mActiveCount) < 0) {
+ ALOGW("%s invalid delta %d, active count %d",
+ __FUNCTION__, delta, mActiveCount);
+ mActiveCount = (uint32_t)(-delta);
+ }
+ mActiveCount += delta;
+ ALOGV("%s active count %d", __FUNCTION__, mActiveCount);
+ return mActiveCount;
+}
+
+bool AudioSession::matches(const sp<AudioSession> &other) const
+{
+ if (other->session() == mSession &&
+ other->inputSource() == mInputSource &&
+ other->format() == mFormat &&
+ other->sampleRate() == mSampleRate &&
+ other->channelMask() == mChannelMask &&
+ other->flags() == mFlags &&
+ other->uid() == mUid) {
+ return true;
+ }
+ return false;
+}
+
+
+status_t AudioSession::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sAudio session %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- session: %2d\n", spaces, "", mSession);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- input source: %d\n", spaces, "", mInputSource);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- format: %08x\n", spaces, "", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- sample: %d\n", spaces, "", mSampleRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- channel mask: %08x\n",
+ spaces, "", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- is soundtrigger: %s\n",
+ spaces, "", mIsSoundTrigger ? "true" : "false");
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- open count: %d\n", spaces, "", mOpenCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- active count: %d\n", spaces, "", mActiveCount);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioSessionCollection::addSession(audio_session_t session,
+ const sp<AudioSession>& audioSession)
+{
+ ssize_t index = indexOfKey(session);
+
+ if (index >= 0) {
+ ALOGW("addSession() session %d already in", session);
+ return ALREADY_EXISTS;
+ }
+ add(session, audioSession);
+ ALOGV("addSession() session %d client %d source %d",
+ session, audioSession->uid(), audioSession->inputSource());
+ return NO_ERROR;
+}
+
+status_t AudioSessionCollection::removeSession(audio_session_t session)
+{
+ ssize_t index = indexOfKey(session);
+
+ if (index < 0) {
+ ALOGW("removeSession() session %d not in", session);
+ return ALREADY_EXISTS;
+ }
+ ALOGV("removeSession() session %d", session);
+ removeItemsAt(index);
+ return NO_ERROR;
+}
+
+uint32_t AudioSessionCollection::getOpenCount() const
+{
+ uint32_t openCount = 0;
+ for (size_t i = 0; i < size(); i++) {
+ openCount += valueAt(i)->openCount();
+ }
+ return openCount;
+}
+
+AudioSessionCollection AudioSessionCollection::getActiveSessions() const
+{
+ AudioSessionCollection activeSessions;
+ for (size_t i = 0; i < size(); i++) {
+ if (valueAt(i)->activeCount() != 0) {
+ activeSessions.add(valueAt(i)->session(), valueAt(i));
+ }
+ }
+ return activeSessions;
+}
+
+bool AudioSessionCollection::hasActiveSession() const
+{
+ return getActiveSessions().size() != 0;
+}
+
+bool AudioSessionCollection::isSourceActive(audio_source_t source) const
+{
+ for (size_t i = 0; i < size(); i++) {
+ const sp<AudioSession> audioSession = valueAt(i);
+ // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it
+ // corresponds to an active capture triggered by a hardware hotword recognition
+ if (audioSession->activeCount() > 0 &&
+ ((audioSession->inputSource() == source) ||
+ ((source == AUDIO_SOURCE_VOICE_RECOGNITION) &&
+ (audioSession->inputSource() == AUDIO_SOURCE_HOTWORD) &&
+ audioSession->isSoundTrigger()))) {
+ return true;
+ }
+ }
+ return false;
+}
+
+
+status_t AudioSessionCollection::dump(int fd, int spaces) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ snprintf(buffer, SIZE, "%*sAudio Sessions:\n", spaces, "");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ valueAt(i)->dump(fd, spaces + 2, i);
+ }
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
index 786c4e8..b187857 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
@@ -135,9 +135,15 @@
deviceDesc->mAddress = String8((char *)node->value);
} else if (strcmp(node->name, CHANNELS_TAG) == 0) {
if (audio_is_input_device(type)) {
- deviceDesc->setSupportedChannelMasks(inputChannelMasksFromString(node->value));
+ deviceDesc->addAudioProfile(
+ new AudioProfile(gDynamicFormat,
+ inputChannelMasksFromString(node->value),
+ SampleRateVector()));
} else {
- deviceDesc->setSupportedChannelMasks(outputChannelMasksFromString(node->value));
+ deviceDesc->addAudioProfile(
+ new AudioProfile(gDynamicFormat,
+ outputChannelMasksFromString(node->value),
+ SampleRateVector()));
}
} else if (strcmp(node->name, GAINS_TAG) == 0) {
loadDeviceDescriptorGains(node, deviceDesc);
@@ -153,95 +159,74 @@
}
//static
-status_t ConfigParsingUtils::loadHwModuleInput(cnode *root, sp<HwModule> &module)
+status_t ConfigParsingUtils::loadHwModuleProfile(cnode *root, sp<HwModule> &module,
+ audio_port_role_t role)
{
cnode *node = root->first_child;
- sp<InputProfile> profile = new InputProfile(String8(root->name));
+ sp<IOProfile> profile = new IOProfile(String8(root->name), role);
+
+ AudioProfileVector audioProfiles;
+ SampleRateVector sampleRates;
+ ChannelsVector channels;
+ FormatVector formats;
while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->setSupportedSamplingRates(samplingRatesFromString(node->value));
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->setSupportedFormats(formatsFromString(node->value));
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->setSupportedChannelMasks(inputChannelMasksFromString(node->value));
+ if (strcmp(node->name, FORMATS_TAG) == 0 &&
+ strcmp(node->value, DYNAMIC_VALUE_TAG) != 0) {
+ formats = formatsFromString(node->value);
+ } else if (strcmp(node->name, SAMPLING_RATES_TAG) == 0 &&
+ strcmp(node->value, DYNAMIC_VALUE_TAG) != 0) {
+ collectionFromString<SampleRateTraits>(node->value, sampleRates);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0 &&
+ strcmp(node->value, DYNAMIC_VALUE_TAG) != 0) {
+ if (role == AUDIO_PORT_ROLE_SINK) {
+ channels = inputChannelMasksFromString(node->value);
+ } else {
+ channels = outputChannelMasksFromString(node->value);
+ }
} else if (strcmp(node->name, DEVICES_TAG) == 0) {
DeviceVector devices;
loadDevicesFromTag(node->value, devices, module->getDeclaredDevices());
profile->setSupportedDevices(devices);
} else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->setFlags(InputFlagConverter::maskFromString(node->value));
+ if (role == AUDIO_PORT_ROLE_SINK) {
+ profile->setFlags(InputFlagConverter::maskFromString(node->value));
+ } else {
+ profile->setFlags(ConfigParsingUtils::parseOutputFlagNames(node->value));
+ }
} else if (strcmp(node->name, GAINS_TAG) == 0) {
loadAudioPortGains(node, *profile);
}
node = node->next;
}
- ALOGW_IF(profile->getSupportedDevices().isEmpty(),
- "loadInput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadInput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadInput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadInput() invalid supported formats");
- if (!profile->getSupportedDevices().isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadInput() adding input Supported Devices %04x",
- profile->getSupportedDevices().types());
- return module->addInputProfile(profile);
+ if (formats.isEmpty()) {
+ sp<AudioProfile> profileToAdd = new AudioProfile(gDynamicFormat, channels, sampleRates);
+ profileToAdd->setDynamicFormat(true);
+ profileToAdd->setDynamicChannels(channels.isEmpty());
+ profileToAdd->setDynamicRate(sampleRates.isEmpty());
+ audioProfiles.add(profileToAdd);
} else {
- return BAD_VALUE;
- }
-}
-
-//static
-status_t ConfigParsingUtils::loadHwModuleOutput(cnode *root, sp<HwModule> &module)
-{
- cnode *node = root->first_child;
-
- sp<OutputProfile> profile = new OutputProfile(String8(root->name));
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->setSupportedSamplingRates(samplingRatesFromString(node->value));
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->setSupportedFormats(formatsFromString(node->value));
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->setSupportedChannelMasks(outputChannelMasksFromString(node->value));
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- DeviceVector devices;
- loadDevicesFromTag(node->value, devices, module->getDeclaredDevices());
- profile->setSupportedDevices(devices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->setFlags(ConfigParsingUtils::parseOutputFlagNames(node->value));
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- loadAudioPortGains(node, *profile);
+ for (size_t i = 0; i < formats.size(); i++) {
+ // For compatibility reason, for each format, creates a profile with the same
+ // collection of rate and channels.
+ sp<AudioProfile> profileToAdd = new AudioProfile(formats[i], channels, sampleRates);
+ profileToAdd->setDynamicFormat(formats[i] == gDynamicFormat);
+ profileToAdd->setDynamicChannels(channels.isEmpty());
+ profileToAdd->setDynamicRate(sampleRates.isEmpty());
+ audioProfiles.add(profileToAdd);
}
- node = node->next;
}
- ALOGW_IF(profile->getSupportedDevices().isEmpty(),
- "loadOutput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadOutput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadOutput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadOutput() invalid supported formats");
- if (!profile->getSupportedDevices().isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
- profile->getSupportedDevices().types(), profile->getFlags());
- return module->addOutputProfile(profile);
- } else {
- return BAD_VALUE;
+ profile->setAudioProfiles(audioProfiles);
+ ALOGW_IF(!profile->hasSupportedDevices(), "load%s() invalid supported devices",
+ role == AUDIO_PORT_ROLE_SINK ? "Input" : "Output");
+ if (profile->hasSupportedDevices()) {
+ ALOGV("load%s() adding Supported Devices %04x, mFlags %04x",
+ role == AUDIO_PORT_ROLE_SINK ? "Input" : "Output",
+ profile->getSupportedDevicesType(), profile->getFlags());
+ return module->addProfile(profile);
}
+ return BAD_VALUE;
}
//static
@@ -268,7 +253,7 @@
node = node->first_child;
while (node) {
ALOGV("loadHwModule() loading output %s", node->name);
- status_t tmpStatus = loadHwModuleOutput(node, module);
+ status_t tmpStatus = loadHwModuleProfile(node, module, AUDIO_PORT_ROLE_SOURCE);
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
status = tmpStatus;
}
@@ -280,7 +265,7 @@
node = node->first_child;
while (node) {
ALOGV("loadHwModule() loading input %s", node->name);
- status_t tmpStatus = loadHwModuleInput(node, module);
+ status_t tmpStatus = loadHwModuleProfile(node, module, AUDIO_PORT_ROLE_SINK);
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
status = tmpStatus;
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index f7ddb35..d752485 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -207,15 +207,18 @@
return device;
}
-status_t DeviceVector::dump(int fd, const String8 &direction) const
+status_t DeviceVector::dump(int fd, const String8 &tag, int spaces, bool verbose) const
{
+ if (isEmpty()) {
+ return NO_ERROR;
+ }
const size_t SIZE = 256;
char buffer[SIZE];
- snprintf(buffer, SIZE, "\n Available %s devices:\n", direction.string());
+ snprintf(buffer, SIZE, "%*s %s devices:\n", spaces, "", tag.string());
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < size(); i++) {
- itemAt(i)->dump(fd, 2, i);
+ itemAt(i)->dump(fd, spaces + 4, i, verbose);
}
return NO_ERROR;
}
@@ -263,12 +266,10 @@
void DeviceDescriptor::importAudioPort(const sp<AudioPort> port) {
AudioPort::importAudioPort(port);
- mSamplingRate = port->pickSamplingRate();
- mFormat = port->pickFormat();
- mChannelMask = port->pickChannelMask();
+ port->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
}
-status_t DeviceDescriptor::dump(int fd, int spaces, int index) const
+status_t DeviceDescriptor::dump(int fd, int spaces, int index, bool verbose) const
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -290,7 +291,7 @@
result.append(buffer);
}
write(fd, result.string(), result.size());
- AudioPort::dump(fd, spaces);
+ AudioPort::dump(fd, spaces, verbose);
return NO_ERROR;
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index b02eb6a..dce0890 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -43,13 +43,12 @@
}
status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address)
+ audio_devices_t device, String8 address)
{
sp<IOProfile> profile = new OutputProfile(name);
- profile->mSamplingRates.add(config->sample_rate);
- profile->mChannelMasks.add(config->channel_mask);
- profile->mFormats.add(config->format);
+ profile->addAudioProfile(new AudioProfile(config->format, config->channel_mask,
+ config->sample_rate));
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
devDesc->mAddress = address;
@@ -105,19 +104,18 @@
}
status_t HwModule::addInputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address)
+ audio_devices_t device, String8 address)
{
sp<IOProfile> profile = new InputProfile(name);
-
- profile->mSamplingRates.add(config->sample_rate);
- profile->mChannelMasks.add(config->channel_mask);
- profile->mFormats.add(config->format);
+ profile->addAudioProfile(new AudioProfile(config->format, config->channel_mask,
+ config->sample_rate));
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
devDesc->mAddress = address;
profile->addSupportedDevice(devDesc);
- ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
+ ALOGV("addInputProfile() name %s rate %d mask 0x%08x",
+ name.string(), config->sample_rate, config->channel_mask);
return addInputProfile(profile);
}
@@ -164,12 +162,7 @@
mInputProfiles[i]->dump(fd);
}
}
- if (mDeclaredDevices.size()) {
- write(fd, " - devices:\n", strlen(" - devices:\n"));
- for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
- mDeclaredDevices[i]->dump(fd, 4, i);
- }
- }
+ mDeclaredDevices.dump(fd, String8("Declared"), 2, true);
}
sp <HwModule> HwModuleCollection::getModuleFromName(const char *name) const
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index d41d239..fe38f54 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -20,6 +20,7 @@
#include "IOProfile.h"
#include "HwModule.h"
#include "AudioGain.h"
+#include "TypeConverter.h"
namespace android {
@@ -63,37 +64,25 @@
}
}
- if (samplingRate == 0) {
- return false;
- }
- uint32_t myUpdatedSamplingRate = samplingRate;
- if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
- return false;
- }
- if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
- NO_ERROR) {
+ if (samplingRate == 0 || !audio_is_valid_format(format) ||
+ (isPlaybackThread && (!audio_is_output_channel(channelMask))) ||
+ (isRecordThread && (!audio_is_input_channel(channelMask)))) {
return false;
}
- if (!audio_is_valid_format(format)) {
- return false;
- }
- if (isPlaybackThread && checkExactFormat(format) != NO_ERROR) {
- return false;
- }
audio_format_t myUpdatedFormat = format;
- if (isRecordThread && checkCompatibleFormat(format, &myUpdatedFormat) != NO_ERROR) {
- return false;
- }
-
- if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
- checkExactChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
audio_channel_mask_t myUpdatedChannelMask = channelMask;
- if (isRecordThread && (!audio_is_input_channel(channelMask) ||
- checkCompatibleChannelMask(channelMask, &myUpdatedChannelMask) != NO_ERROR)) {
- return false;
+ uint32_t myUpdatedSamplingRate = samplingRate;
+ if (isRecordThread)
+ {
+ if (checkCompatibleAudioProfile(
+ myUpdatedSamplingRate, myUpdatedChannelMask, myUpdatedFormat) != NO_ERROR) {
+ return false;
+ }
+ } else {
+ if (checkExactAudioProfile(samplingRate, channelMask, format) != NO_ERROR) {
+ return false;
+ }
}
if (isPlaybackThread && (getFlags() & flags) != flags) {
@@ -130,37 +119,13 @@
snprintf(buffer, SIZE, " - flags: 0x%04x\n", getFlags());
result.append(buffer);
- snprintf(buffer, SIZE, " - devices:\n");
- result.append(buffer);
write(fd, result.string(), result.size());
- for (size_t i = 0; i < mSupportedDevices.size(); i++) {
- mSupportedDevices[i]->dump(fd, 6, i);
- }
+ mSupportedDevices.dump(fd, String8("- Supported"), 2, false);
}
void IOProfile::log()
{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- ALOGV(" - sampling rates: ");
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- ALOGV(" %d", mSamplingRates[i]);
- }
-
- ALOGV(" - channel masks: ");
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV(" 0x%04x", mChannelMasks[i]);
- }
-
- ALOGV(" - formats: ");
- for (size_t i = 0; i < mFormats.size(); i++) {
- ALOGV(" 0x%08x", mFormats[i]);
- }
-
- ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
- ALOGV(" - flags: 0x%04x\n", getFlags());
+ // @TODO: forward log to AudioPort
}
}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/TypeConverter.cpp b/services/audiopolicy/common/managerdefinitions/src/TypeConverter.cpp
index 0397f97..74d8809 100644
--- a/services/audiopolicy/common/managerdefinitions/src/TypeConverter.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/TypeConverter.cpp
@@ -168,14 +168,14 @@
template <>
const ChannelIndexConverter::Table ChannelIndexConverter::mTable[] = {
- "AUDIO_CHANNEL_INDEX_MASK_1", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_1),
- "AUDIO_CHANNEL_INDEX_MASK_2", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_2),
- "AUDIO_CHANNEL_INDEX_MASK_3", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_3),
- "AUDIO_CHANNEL_INDEX_MASK_4", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_4),
- "AUDIO_CHANNEL_INDEX_MASK_5", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_5),
- "AUDIO_CHANNEL_INDEX_MASK_6", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_6),
- "AUDIO_CHANNEL_INDEX_MASK_7", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_7),
- "AUDIO_CHANNEL_INDEX_MASK_8", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_8),
+ {"AUDIO_CHANNEL_INDEX_MASK_1", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_1)},
+ {"AUDIO_CHANNEL_INDEX_MASK_2", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_2)},
+ {"AUDIO_CHANNEL_INDEX_MASK_3", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_3)},
+ {"AUDIO_CHANNEL_INDEX_MASK_4", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_4)},
+ {"AUDIO_CHANNEL_INDEX_MASK_5", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_5)},
+ {"AUDIO_CHANNEL_INDEX_MASK_6", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_6)},
+ {"AUDIO_CHANNEL_INDEX_MASK_7", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_7)},
+ {"AUDIO_CHANNEL_INDEX_MASK_8", static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_INDEX_MASK_8)},
};
template<>
const size_t ChannelIndexConverter::mSize = sizeof(ChannelIndexConverter::mTable) /
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/example/Android.mk b/services/audiopolicy/engineconfigurable/parameter-framework/example/Android.mk
index e9b1902..55da35e 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/example/Android.mk
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/example/Android.mk
@@ -7,6 +7,7 @@
#
################################################################################################
+ifeq (1, 0)
LOCAL_PATH := $(call my-dir)
@@ -103,3 +104,5 @@
LOCAL_SRC_FILES := Settings/$(LOCAL_MODULE_STEM)
include $(BUILD_PREBUILT)
endif # pfw_rebuild_settings
+
+endif # ifeq (1, 0)
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 6ae2aff..11dabae 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -418,7 +418,9 @@
if (activeInput != 0) {
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
if (activeDesc->getModuleHandle() == txSourceDeviceDesc->getModuleHandle()) {
- audio_session_t activeSession = activeDesc->mSessions.itemAt(0);
+ //FIXME: consider all active sessions
+ AudioSessionCollection activeSessions = activeDesc->getActiveAudioSessions();
+ audio_session_t activeSession = activeSessions.keyAt(0);
stopInput(activeInput, activeSession);
releaseInput(activeInput, activeSession);
}
@@ -1329,7 +1331,6 @@
audio_devices_t device;
// handle legacy remote submix case where the address was not always specified
String8 address = String8("");
- bool isSoundTrigger = false;
audio_source_t inputSource = attr->source;
audio_source_t halInputSource;
AudioMix *policyMix = NULL;
@@ -1385,16 +1386,44 @@
*inputType = API_INPUT_LEGACY;
}
- if (inputSource == AUDIO_SOURCE_HOTWORD) {
- ssize_t index = mSoundTriggerSessions.indexOfKey(session);
- if (index >= 0) {
- *input = mSoundTriggerSessions.valueFor(session);
- isSoundTrigger = true;
- flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
- ALOGV("SoundTrigger capture on session %d input %d", session, *input);
- } else {
- halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
- }
+ }
+
+ *input = getInputForDevice(device, address, session, uid, inputSource,
+ samplingRate, format, channelMask, flags,
+ policyMix);
+ if (*input == AUDIO_IO_HANDLE_NONE) {
+ mInputRoutes.removeRoute(session);
+ return INVALID_OPERATION;
+ }
+ ALOGV("getInputForAttr() returns input type = %d", *inputType);
+ return NO_ERROR;
+}
+
+
+audio_io_handle_t AudioPolicyManager::getInputForDevice(audio_devices_t device,
+ String8 address,
+ audio_session_t session,
+ uid_t uid,
+ audio_source_t inputSource,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ AudioMix *policyMix)
+{
+ audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ audio_source_t halInputSource = inputSource;
+ bool isSoundTrigger = false;
+
+ if (inputSource == AUDIO_SOURCE_HOTWORD) {
+ ssize_t index = mSoundTriggerSessions.indexOfKey(session);
+ if (index >= 0) {
+ input = mSoundTriggerSessions.valueFor(session);
+ isSoundTrigger = true;
+ flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
+ ALOGV("SoundTrigger capture on session %d input %d", session, input);
+ } else {
+ halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
}
}
@@ -1414,25 +1443,62 @@
} else if (profileFlags != AUDIO_INPUT_FLAG_NONE) {
profileFlags = AUDIO_INPUT_FLAG_NONE; // retry
} else { // fail
- ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u,"
- "format %#x, channelMask 0x%X, flags %#x",
+ ALOGW("getInputForDevice() could not find profile for device 0x%X,"
+ "samplingRate %u, format %#x, channelMask 0x%X, flags %#x",
device, samplingRate, format, channelMask, flags);
- return BAD_VALUE;
+ return input;
}
}
if (profile->getModuleHandle() == 0) {
ALOGE("getInputForAttr(): HW module %s not opened", profile->getModuleName());
- return NO_INIT;
+ return input;
}
+ sp<AudioSession> audioSession = new AudioSession(session,
+ inputSource,
+ format,
+ samplingRate,
+ channelMask,
+ flags,
+ uid,
+ isSoundTrigger);
+
+// TODO enable input reuse
+#if 0
+ // reuse an open input if possible
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ sp<AudioInputDescriptor> desc = mInputs.valueAt(i);
+ // reuse input if it shares the same profile and same sound trigger attribute
+ if (profile == desc->mProfile &&
+ isSoundTrigger == desc->isSoundTrigger()) {
+
+ sp<AudioSession> as = desc->getAudioSession(session);
+ if (as != 0) {
+ // do not allow unmatching properties on same session
+ if (as->matches(audioSession)) {
+ as->changeOpenCount(1);
+ } else {
+ ALOGW("getInputForDevice() record with different attributes"
+ " exists for session %d", session);
+ return input;
+ }
+ } else {
+ desc->addAudioSession(session, audioSession);
+ }
+ ALOGV("getInputForDevice() reusing input %d", mInputs.keyAt(i));
+ return mInputs.keyAt(i);
+ }
+ }
+#endif
+
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = profileSamplingRate;
config.channel_mask = profileChannelMask;
config.format = profileFormat;
status_t status = mpClientInterface->openInput(profile->getModuleHandle(),
- input,
+ &input,
&config,
&device,
address,
@@ -1440,37 +1506,31 @@
profileFlags);
// only accept input with the exact requested set of parameters
- if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE ||
+ if (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE ||
(profileSamplingRate != config.sample_rate) ||
(profileFormat != config.format) ||
(profileChannelMask != config.channel_mask)) {
- ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d,"
- " channelMask %x",
+ ALOGW("getInputForAttr() failed opening input: samplingRate %d"
+ ", format %d, channelMask %x",
samplingRate, format, channelMask);
- if (*input != AUDIO_IO_HANDLE_NONE) {
- mpClientInterface->closeInput(*input);
+ if (input != AUDIO_IO_HANDLE_NONE) {
+ mpClientInterface->closeInput(input);
}
- return BAD_VALUE;
+ return AUDIO_IO_HANDLE_NONE;
}
sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
- inputDesc->mInputSource = inputSource;
- inputDesc->mRefCount = 0;
inputDesc->mSamplingRate = profileSamplingRate;
inputDesc->mFormat = profileFormat;
inputDesc->mChannelMask = profileChannelMask;
inputDesc->mDevice = device;
- inputDesc->mSessions.add(session);
- inputDesc->mIsSoundTrigger = isSoundTrigger;
inputDesc->mPolicyMix = policyMix;
- inputDesc->changeOpenRefCount(1);
+ inputDesc->addAudioSession(session, audioSession);
- ALOGV("getInputForAttr() returns input type = %d", *inputType);
-
- addInput(*input, inputDesc);
+ addInput(input, inputDesc);
mpClientInterface->onAudioPortListUpdate();
- return NO_ERROR;
+ return input;
}
status_t AudioPolicyManager::startInput(audio_io_handle_t input,
@@ -1484,8 +1544,8 @@
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
- index = inputDesc->mSessions.indexOf(session);
- if (index < 0) {
+ sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
+ if (audioSession == 0) {
ALOGW("startInput() unknown session %d on input %d", session, input);
return BAD_VALUE;
}
@@ -1500,11 +1560,14 @@
// If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
// otherwise the active input continues and the new input cannot be started.
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
- if ((activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) &&
+ if ((activeDesc->inputSource() == AUDIO_SOURCE_HOTWORD) &&
!activeDesc->hasPreemptedSession(session)) {
ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
- audio_session_t activeSession = activeDesc->mSessions.itemAt(0);
- SortedVector<audio_session_t> sessions = activeDesc->getPreemptedSessions();
+ //FIXME: consider all active sessions
+ AudioSessionCollection activeSessions = activeDesc->getActiveAudioSessions();
+ audio_session_t activeSession = activeSessions.keyAt(0);
+ SortedVector<audio_session_t> sessions =
+ activeDesc->getPreemptedSessions();
sessions.add(activeSession);
inputDesc->setPreemptedSessions(sessions);
stopInput(activeInput, activeSession);
@@ -1528,7 +1591,7 @@
// Routing?
mInputRoutes.incRouteActivity(session);
- if (inputDesc->mRefCount == 0 || mInputRoutes.hasRouteChanged(session)) {
+ if (!inputDesc->isActive() || mInputRoutes.hasRouteChanged(session)) {
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
&& ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
@@ -1559,9 +1622,9 @@
}
}
- ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
+ ALOGV("AudioPolicyManager::startInput() input source = %d", audioSession->inputSource());
- inputDesc->mRefCount++;
+ audioSession->changeActiveCount(1);
return NO_ERROR;
}
@@ -1576,23 +1639,23 @@
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
- index = inputDesc->mSessions.indexOf(session);
+ sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
if (index < 0) {
ALOGW("stopInput() unknown session %d on input %d", session, input);
return BAD_VALUE;
}
- if (inputDesc->mRefCount == 0) {
+ if (audioSession->activeCount() == 0) {
ALOGW("stopInput() input %d already stopped", input);
return INVALID_OPERATION;
}
- inputDesc->mRefCount--;
+ audioSession->changeActiveCount(-1);
// Routing?
mInputRoutes.decRouteActivity(session);
- if (inputDesc->mRefCount == 0) {
+ if (!inputDesc->isActive()) {
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
&& ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
@@ -1642,17 +1705,22 @@
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
ALOG_ASSERT(inputDesc != 0);
- index = inputDesc->mSessions.indexOf(session);
+ sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
if (index < 0) {
ALOGW("releaseInput() unknown session %d on input %d", session, input);
return;
}
- inputDesc->mSessions.remove(session);
- if (inputDesc->getOpenRefCount() == 0) {
- ALOGW("releaseInput() invalid open ref count %d", inputDesc->getOpenRefCount());
+
+ if (audioSession->openCount() == 0) {
+ ALOGW("releaseInput() invalid open count %d on session %d",
+ audioSession->openCount(), session);
return;
}
- inputDesc->changeOpenRefCount(-1);
+
+ if (audioSession->changeOpenCount(-1) == 0) {
+ inputDesc->removeAudioSession(session);
+ }
+
if (inputDesc->getOpenRefCount() > 0) {
ALOGV("releaseInput() exit > 0");
return;
@@ -1867,23 +1935,9 @@
{
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
- if (inputDescriptor->mRefCount == 0) {
- continue;
- }
- if (inputDescriptor->mInputSource == (int)source) {
+ if (inputDescriptor->isSourceActive(source)) {
return true;
}
- // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it
- // corresponds to an active capture triggered by a hardware hotword recognition
- if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) &&
- (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) {
- // FIXME: we should not assume that the first session is the active one and keep
- // activity count per session. Same in startInput().
- ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0));
- if (index >= 0) {
- return true;
- }
- }
}
return false;
}
@@ -1926,7 +1980,7 @@
return INVALID_OPERATION;
}
- ALOGV("registerPolicyMixes() num mixes %d", mixes.size());
+ ALOGV("registerPolicyMixes() num mixes %zu", mixes.size());
for (size_t i = 0; i < mixes.size(); i++) {
String8 address = mixes[i].mRegistrationId;
@@ -1973,7 +2027,7 @@
return INVALID_OPERATION;
}
- ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size());
+ ALOGV("unregisterPolicyMixes() num mixes %zu", mixes.size());
for (size_t i = 0; i < mixes.size(); i++) {
String8 address = mixes[i].mRegistrationId;
@@ -2037,8 +2091,8 @@
write(fd, result.string(), result.size());
- mAvailableOutputDevices.dump(fd, String8("output"));
- mAvailableInputDevices.dump(fd, String8("input"));
+ mAvailableOutputDevices.dump(fd, String8("Available output"));
+ mAvailableInputDevices.dump(fd, String8("Available input"));
mHwModules.dump(fd);
mOutputs.dump(fd);
mInputs.dump(fd);
@@ -2218,7 +2272,7 @@
patch->sources[0].type);
#if LOG_NDEBUG == 0
for (size_t i = 0; i < patch->num_sinks; i++) {
- ALOGV("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id,
+ ALOGV("createAudioPatch sink %zu: id %d role %d type %d", i, patch->sinks[i].id,
patch->sinks[i].role,
patch->sinks[i].type);
}
@@ -2663,7 +2717,7 @@
SortedVector<audio_io_handle_t> inputsToClose;
for (size_t i = 0; i < mInputs.size(); i++) {
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(i);
- if (affectedSources.indexOf(inputDesc->mInputSource) >= 0) {
+ if (affectedSources.indexOf(inputDesc->inputSource()) >= 0) {
inputsToClose.add(inputDesc->mIoHandle);
}
}
@@ -3026,7 +3080,6 @@
}
sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(inProfile);
- inputDesc->mInputSource = AUDIO_SOURCE_MIC;
inputDesc->mDevice = profileType;
// find the address
@@ -3390,7 +3443,7 @@
}
}
- ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size());
+ ALOGV(" found %zu profiles, %zu outputs", profiles.size(), outputs.size());
if (profiles.isEmpty() && outputs.isEmpty()) {
ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
@@ -3449,55 +3502,14 @@
mpClientInterface->setParameters(output, String8(param));
free(param);
}
-
- // Here is where we step through and resolve any "dynamic" fields
- String8 reply;
- char *value;
- if (profile->mSamplingRates[0] == 0) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
- ALOGV("checkOutputsForDevice() supported sampling rates %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->setSupportedSamplingRates(samplingRatesFromString(value + 1));
- }
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
- ALOGV("checkOutputsForDevice() supported formats %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->setSupportedFormats(formatsFromString(value + 1));
- }
- }
- if (profile->mChannelMasks[0] == 0) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
- ALOGV("checkOutputsForDevice() supported channel masks %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->setSupportedChannelMasks(outputChannelMasksFromString(value + 1));
- }
- }
- if (((profile->mSamplingRates[0] == 0) &&
- (profile->mSamplingRates.size() < 2)) ||
- ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
- (profile->mFormats.size() < 2)) ||
- ((profile->mChannelMasks[0] == 0) &&
- (profile->mChannelMasks.size() < 2))) {
+ updateAudioProfiles(output, profile->getAudioProfiles());
+ if (!profile->hasValidAudioProfile()) {
ALOGW("checkOutputsForDevice() missing param");
mpClientInterface->closeOutput(output);
output = AUDIO_IO_HANDLE_NONE;
- } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 ||
- profile->mChannelMasks[0] == 0) {
+ } else if (profile->hasDynamicAudioProfile()) {
mpClientInterface->closeOutput(output);
- config.sample_rate = profile->pickSamplingRate();
- config.channel_mask = profile->pickChannelMask();
- config.format = profile->pickFormat();
+ profile->pickAudioProfile(config.sample_rate, config.channel_mask, config.format);
config.offload_info.sample_rate = config.sample_rate;
config.offload_info.channel_mask = config.channel_mask;
config.offload_info.format = config.format;
@@ -3621,18 +3633,7 @@
if (profile->supportDevice(device)) {
ALOGV("checkOutputsForDevice(): "
"clearing direct output profile %zu on module %zu", j, i);
- if (profile->mSamplingRates[0] == 0) {
- profile->mSamplingRates.clear();
- profile->mSamplingRates.add(0);
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.clear();
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- }
- if (profile->mChannelMasks[0] == 0) {
- profile->mChannelMasks.clear();
- profile->mChannelMasks.add(0);
- }
+ profile->clearAudioProfiles();
}
}
}
@@ -3738,42 +3739,8 @@
mpClientInterface->setParameters(input, String8(param));
free(param);
}
-
- // Here is where we step through and resolve any "dynamic" fields
- String8 reply;
- char *value;
- if (profile->mSamplingRates[0] == 0) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
- ALOGV("checkInputsForDevice() direct input sup sampling rates %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->setSupportedSamplingRates(samplingRatesFromString(value + 1));
- }
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
- ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->setSupportedFormats(formatsFromString(value + 1));
- }
- }
- if (profile->mChannelMasks[0] == 0) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
- ALOGV("checkInputsForDevice() direct input sup channel masks %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->setSupportedChannelMasks(inputChannelMasksFromString(value + 1));
- }
- }
- if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
- ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) ||
- ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) {
+ updateAudioProfiles(input, profile->getAudioProfiles());
+ if (!profile->hasValidAudioProfile()) {
ALOGW("checkInputsForDevice() direct input missing param");
mpClientInterface->closeInput(input);
input = AUDIO_IO_HANDLE_NONE;
@@ -3824,18 +3791,7 @@
if (profile->supportDevice(device)) {
ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
profile_index, module_index);
- if (profile->mSamplingRates[0] == 0) {
- profile->mSamplingRates.clear();
- profile->mSamplingRates.add(0);
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.clear();
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- }
- if (profile->mChannelMasks[0] == 0) {
- profile->mChannelMasks.clear();
- profile->mChannelMasks.add(0);
- }
+ profile->clearAudioProfiles();
}
}
}
@@ -4171,7 +4127,7 @@
}
}
- audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->mInputSource);
+ audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->inputSource());
return device;
}
@@ -4583,7 +4539,7 @@
// AUDIO_SOURCE_HOTWORD is for internal use only:
// handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
- !inputDesc->mIsSoundTrigger) {
+ !inputDesc->isSoundTrigger()) {
patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
}
patch.num_sinks = 1;
@@ -5095,5 +5051,51 @@
}
}
+void AudioPolicyManager::updateAudioProfiles(audio_io_handle_t ioHandle,
+ AudioProfileVector &profiles)
+{
+ String8 reply;
+ char *value;
+ // Format MUST be checked first to update the list of AudioProfile
+ if (profiles.hasDynamicFormat()) {
+ reply = mpClientInterface->getParameters(ioHandle,
+ String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+ ALOGV("%s: supported formats %s", __FUNCTION__, reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value == NULL) {
+ ALOGE("%s: failed to retrieve format, bailing out", __FUNCTION__);
+ return;
+ }
+ profiles.setFormats(formatsFromString(value + 1));
+ }
+ const FormatVector &supportedFormats = profiles.getSupportedFormats();
+
+ for(size_t formatIndex = 0; formatIndex < supportedFormats.size(); formatIndex++) {
+ audio_format_t format = supportedFormats[formatIndex];
+ AudioParameter requestedParameters;
+ requestedParameters.addInt(String8(AUDIO_PARAMETER_STREAM_FORMAT), format);
+
+ if (profiles.hasDynamicRateFor(format)) {
+ reply = mpClientInterface->getParameters(ioHandle,
+ requestedParameters.toString() + ";" +
+ AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES);
+ ALOGV("%s: supported sampling rates %s", __FUNCTION__, reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profiles.setSampleRatesFor(samplingRatesFromString(value + 1), format);
+ }
+ }
+ if (profiles.hasDynamicChannelsFor(format)) {
+ reply = mpClientInterface->getParameters(ioHandle,
+ requestedParameters.toString() + ";" +
+ AUDIO_PARAMETER_STREAM_SUP_CHANNELS);
+ ALOGV("%s: supported channel masks %s", __FUNCTION__, reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profiles.setChannelsFor(channelMasksFromString(value + 1), format);
+ }
+ }
+ }
+}
}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index bb71090..59163ca 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -587,6 +587,9 @@
// Audio Policy Engine Interface.
AudioPolicyManagerInterface *mEngine;
private:
+ // If any, resolve any "dynamic" fields of an Audio Profiles collection
+ void updateAudioProfiles(audio_io_handle_t ioHandle, AudioProfileVector &profiles);
+
// updates device caching and output for streams that can influence the
// routing of notifications
void handleNotificationRoutingForStream(audio_stream_type_t stream);
@@ -610,6 +613,18 @@
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo);
+ // internal method to return the input handle for the given device and format
+ audio_io_handle_t getInputForDevice(audio_devices_t device,
+ String8 address,
+ audio_session_t session,
+ uid_t uid,
+ audio_source_t inputSource,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ AudioMix *policyMix);
+
// internal function to derive a stream type value from audio attributes
audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
// event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 41a3fcb..8c976aa 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -164,13 +164,11 @@
ALOGV("getOutput()");
Mutex::Autolock _l(mLock);
- // if the caller is us, trust the specified uid
- if (IPCThreadState::self()->getCallingPid() != getpid_cached || uid == (uid_t)-1) {
- uid_t newclientUid = IPCThreadState::self()->getCallingUid();
- if (uid != (uid_t)-1 && uid != newclientUid) {
- ALOGW("%s uid %d tried to pass itself off as %d", __FUNCTION__, newclientUid, uid);
- }
- uid = newclientUid;
+ const uid_t callingUid = IPCThreadState::self()->getCallingUid();
+ if (!isTrustedCallingUid(callingUid) || uid == (uid_t)-1) {
+ ALOGW_IF(uid != (uid_t)-1 && uid != callingUid,
+ "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, uid);
+ uid = callingUid;
}
return mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, uid, samplingRate,
format, channelMask, flags, selectedDeviceId, offloadInfo);
@@ -284,13 +282,11 @@
sp<AudioPolicyEffects>audioPolicyEffects;
status_t status;
AudioPolicyInterface::input_type_t inputType;
- // if the caller is us, trust the specified uid
- if (IPCThreadState::self()->getCallingPid() != getpid_cached || uid == (uid_t)-1) {
- uid_t newclientUid = IPCThreadState::self()->getCallingUid();
- if (uid != (uid_t)-1 && uid != newclientUid) {
- ALOGW("%s uid %d tried to pass itself off as %d", __FUNCTION__, newclientUid, uid);
- }
- uid = newclientUid;
+ const uid_t callingUid = IPCThreadState::self()->getCallingUid();
+ if (!isTrustedCallingUid(callingUid) || uid == (uid_t)-1) {
+ ALOGW_IF(uid != (uid_t)-1 && uid != callingUid,
+ "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, uid);
+ uid = callingUid;
}
{
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index c5fe69f..18b97a3 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -293,7 +293,7 @@
virtual status_t startPreview() = 0;
virtual void stopPreview() = 0;
virtual bool previewEnabled() = 0;
- virtual status_t storeMetaDataInBuffers(bool enabled) = 0;
+ virtual status_t setVideoBufferMode(int32_t videoBufferMode) = 0;
virtual status_t startRecording() = 0;
virtual void stopRecording() = 0;
virtual bool recordingEnabled() = 0;
@@ -304,6 +304,7 @@
virtual status_t setParameters(const String8& params) = 0;
virtual String8 getParameters() const = 0;
virtual status_t sendCommand(int32_t cmd, int32_t arg1, int32_t arg2) = 0;
+ virtual status_t setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer) = 0;
// Interface used by CameraService
Client(const sp<CameraService>& cameraService,
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 175920f..6722512 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -93,7 +93,6 @@
mStreamingProcessor = new StreamingProcessor(this);
threadName = String8::format("C2-%d-StreamProc",
mCameraId);
- mStreamingProcessor->run(threadName.string());
mFrameProcessor = new FrameProcessor(mDevice, this);
threadName = String8::format("C2-%d-FrameProc",
@@ -390,7 +389,6 @@
l.mParameters.state = Parameters::DISCONNECTED;
}
- mStreamingProcessor->requestExit();
mFrameProcessor->requestExit();
mCaptureSequencer->requestExit();
mJpegProcessor->requestExit();
@@ -404,7 +402,6 @@
// complete callbacks that re-enter Camera2Client
mBinderSerializationLock.unlock();
- mStreamingProcessor->join();
mFrameProcessor->join();
mCaptureSequencer->join();
mJpegProcessor->join();
@@ -944,7 +941,7 @@
return l.mParameters.state == Parameters::PREVIEW;
}
-status_t Camera2Client::storeMetaDataInBuffers(bool enabled) {
+status_t Camera2Client::setVideoBufferMode(int32_t videoBufferMode) {
ATRACE_CALL();
Mutex::Autolock icl(mBinderSerializationLock);
status_t res;
@@ -963,7 +960,12 @@
break;
}
- l.mParameters.storeMetadataInBuffers = enabled;
+ if (videoBufferMode != VIDEO_BUFFER_MODE_BUFFER_QUEUE) {
+ ALOGE("%s: %d: Only video buffer queue is supported", __FUNCTION__, __LINE__);
+ return BAD_VALUE;
+ }
+
+ l.mParameters.videoBufferMode = videoBufferMode;
return OK;
}
@@ -1009,10 +1011,14 @@
return INVALID_OPERATION;
};
- if (!params.storeMetadataInBuffers) {
- ALOGE("%s: Camera %d: Recording only supported in metadata mode, but "
- "non-metadata recording mode requested!", __FUNCTION__,
- mCameraId);
+ if (params.videoBufferMode != VIDEO_BUFFER_MODE_BUFFER_QUEUE) {
+ ALOGE("%s: Camera %d: Recording only supported buffer queue mode, but "
+ "mode %d is requested!", __FUNCTION__, mCameraId, params.videoBufferMode);
+ return INVALID_OPERATION;
+ }
+
+ if (!mStreamingProcessor->haveValidRecordingWindow()) {
+ ALOGE("%s: No valid recording window", __FUNCTION__);
return INVALID_OPERATION;
}
@@ -1176,28 +1182,28 @@
mCameraService->playSound(CameraService::SOUND_RECORDING_STOP);
- // Remove recording stream to prevent it from slowing down takePicture later
- if (!l.mParameters.recordingHint && l.mParameters.isJpegSizeOverridden()) {
- res = stopStream();
- if (res != OK) {
- ALOGE("%s: Camera %d: Can't stop streaming: %s (%d)",
- __FUNCTION__, mCameraId, strerror(-res), res);
- }
- res = mDevice->waitUntilDrained();
- if (res != OK) {
- ALOGE("%s: Camera %d: Waiting to stop streaming failed: %s (%d)",
- __FUNCTION__, mCameraId, strerror(-res), res);
- }
- // Clean up recording stream
- res = mStreamingProcessor->deleteRecordingStream();
- if (res != OK) {
- ALOGE("%s: Camera %d: Unable to delete recording stream before "
- "stop preview: %s (%d)",
- __FUNCTION__, mCameraId, strerror(-res), res);
- }
- l.mParameters.recoverOverriddenJpegSize();
+ // Remove recording stream because the video target may be abandoned soon.
+ res = stopStream();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Can't stop streaming: %s (%d)",
+ __FUNCTION__, mCameraId, strerror(-res), res);
}
+ res = mDevice->waitUntilDrained();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Waiting to stop streaming failed: %s (%d)",
+ __FUNCTION__, mCameraId, strerror(-res), res);
+ }
+ // Clean up recording stream
+ res = mStreamingProcessor->deleteRecordingStream();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to delete recording stream before "
+ "stop preview: %s (%d)",
+ __FUNCTION__, mCameraId, strerror(-res), res);
+ }
+ l.mParameters.recoverOverriddenJpegSize();
+
+ // Restart preview
res = startPreviewL(l.mParameters, true);
if (res != OK) {
ALOGE("%s: Camera %d: Unable to return to preview",
@@ -1224,10 +1230,7 @@
void Camera2Client::releaseRecordingFrame(const sp<IMemory>& mem) {
ATRACE_CALL();
- Mutex::Autolock icl(mBinderSerializationLock);
- if ( checkPid(__FUNCTION__) != OK) return;
-
- mStreamingProcessor->releaseRecordingFrame(mem);
+ ALOGW("%s: Not supported in buffer queue mode.", __FUNCTION__);
}
status_t Camera2Client::autoFocus() {
@@ -1529,10 +1532,10 @@
case CAMERA_CMD_PING:
return commandPingL();
case CAMERA_CMD_SET_VIDEO_BUFFER_COUNT:
- return commandSetVideoBufferCountL(arg1);
case CAMERA_CMD_SET_VIDEO_FORMAT:
- return commandSetVideoFormatL(arg1,
- static_cast<android_dataspace>(arg2));
+ ALOGE("%s: command %d (arguments %d, %d) is not supported.",
+ __FUNCTION__, cmd, arg1, arg2);
+ return BAD_VALUE;
default:
ALOGE("%s: Unknown command %d (arguments %d, %d)",
__FUNCTION__, cmd, arg1, arg2);
@@ -1674,27 +1677,6 @@
}
}
-status_t Camera2Client::commandSetVideoBufferCountL(size_t count) {
- if (recordingEnabledL()) {
- ALOGE("%s: Camera %d: Error setting video buffer count after "
- "recording was started", __FUNCTION__, mCameraId);
- return INVALID_OPERATION;
- }
-
- return mStreamingProcessor->setRecordingBufferCount(count);
-}
-
-status_t Camera2Client::commandSetVideoFormatL(int format,
- android_dataspace dataspace) {
- if (recordingEnabledL()) {
- ALOGE("%s: Camera %d: Error setting video format after "
- "recording was started", __FUNCTION__, mCameraId);
- return INVALID_OPERATION;
- }
-
- return mStreamingProcessor->setRecordingFormat(format, dataspace);
-}
-
void Camera2Client::notifyError(ICameraDeviceCallbacks::CameraErrorCode errorCode,
const CaptureResultExtras& resultExtras) {
int32_t err = CAMERA_ERROR_UNKNOWN;
@@ -2118,6 +2100,84 @@
return res;
}
+status_t Camera2Client::setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer) {
+ ATRACE_CALL();
+ ALOGV("%s: E", __FUNCTION__);
+ Mutex::Autolock icl(mBinderSerializationLock);
+ status_t res;
+ if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
+
+ sp<IBinder> binder = IInterface::asBinder(bufferProducer);
+ if (binder == mVideoSurface) {
+ ALOGV("%s: Camera %d: New video window is same as old video window",
+ __FUNCTION__, mCameraId);
+ return NO_ERROR;
+ }
+
+ sp<Surface> window;
+ int format;
+ android_dataspace dataSpace;
+
+ if (bufferProducer != nullptr) {
+ // Using controlledByApp flag to ensure that the buffer queue remains in
+ // async mode for the old camera API, where many applications depend
+ // on that behavior.
+ window = new Surface(bufferProducer, /*controlledByApp*/ true);
+
+ ANativeWindow *anw = window.get();
+
+ if ((res = anw->query(anw, NATIVE_WINDOW_FORMAT, &format)) != OK) {
+ ALOGE("%s: Failed to query Surface format", __FUNCTION__);
+ return res;
+ }
+
+ if ((res = anw->query(anw, NATIVE_WINDOW_DEFAULT_DATASPACE,
+ reinterpret_cast<int*>(&dataSpace))) != OK) {
+ ALOGE("%s: Failed to query Surface dataSpace", __FUNCTION__);
+ return res;
+ }
+ }
+
+ Parameters::State state;
+ {
+ SharedParameters::Lock l(mParameters);
+ state = l.mParameters.state;
+ }
+
+ switch (state) {
+ case Parameters::STOPPED:
+ case Parameters::WAITING_FOR_PREVIEW_WINDOW:
+ case Parameters::PREVIEW:
+ // OK
+ break;
+ case Parameters::DISCONNECTED:
+ case Parameters::RECORD:
+ case Parameters::STILL_CAPTURE:
+ case Parameters::VIDEO_SNAPSHOT:
+ default:
+ ALOGE("%s: Camera %d: Cannot set video target while in state %s",
+ __FUNCTION__, mCameraId,
+ Parameters::getStateName(state));
+ return INVALID_OPERATION;
+ }
+
+ mVideoSurface = binder;
+ res = mStreamingProcessor->setRecordingWindow(window);
+ if (res != OK) {
+ ALOGE("%s: Unable to set new recording window: %s (%d)",
+ __FUNCTION__, strerror(-res), res);
+ return res;
+ }
+
+ {
+ SharedParameters::Lock l(mParameters);
+ l.mParameters.videoFormat = format;
+ l.mParameters.videoDataSpace = dataSpace;
+ }
+
+ return OK;
+}
+
const char* Camera2Client::kAutofocusLabel = "autofocus";
const char* Camera2Client::kTakepictureLabel = "take_picture";
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index e1e18c9..428dca1 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -66,7 +66,7 @@
virtual status_t startPreview();
virtual void stopPreview();
virtual bool previewEnabled();
- virtual status_t storeMetaDataInBuffers(bool enabled);
+ virtual status_t setVideoBufferMode(int32_t videoBufferMode);
virtual status_t startRecording();
virtual void stopRecording();
virtual bool recordingEnabled();
@@ -79,6 +79,7 @@
virtual status_t sendCommand(int32_t cmd, int32_t arg1, int32_t arg2);
virtual void notifyError(ICameraDeviceCallbacks::CameraErrorCode errorCode,
const CaptureResultExtras& resultExtras);
+ virtual status_t setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer);
/**
* Interface used by CameraService
@@ -194,6 +195,7 @@
/* Preview/Recording related members */
sp<IBinder> mPreviewSurface;
+ sp<IBinder> mVideoSurface;
sp<camera2::StreamingProcessor> mStreamingProcessor;
/** Preview callback related members */
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index 30b462b..ced9d8c 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -482,14 +482,24 @@
mHardware->releaseRecordingFrame(mem);
}
-status_t CameraClient::storeMetaDataInBuffers(bool enabled)
-{
- LOG1("storeMetaDataInBuffers: %s", enabled? "true": "false");
+status_t CameraClient::setVideoBufferMode(int32_t videoBufferMode) {
+ LOG1("setVideoBufferMode: %d", videoBufferMode);
+ bool enableMetadataInBuffers = false;
+
+ if (videoBufferMode == VIDEO_BUFFER_MODE_DATA_CALLBACK_METADATA) {
+ enableMetadataInBuffers = true;
+ } else if (videoBufferMode != VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV) {
+ ALOGE("%s: %d: videoBufferMode %d is not supported.", __FUNCTION__, __LINE__,
+ videoBufferMode);
+ return BAD_VALUE;
+ }
+
Mutex::Autolock lock(mLock);
if (checkPidAndHardware() != NO_ERROR) {
return UNKNOWN_ERROR;
}
- return mHardware->storeMetaDataInBuffers(enabled);
+
+ return mHardware->storeMetaDataInBuffers(enableMetadataInBuffers);
}
bool CameraClient::previewEnabled() {
@@ -991,4 +1001,9 @@
return -1;
}
+status_t CameraClient::setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer) {
+ ALOGE("%s: %d: CameraClient doesn't support setting a video target.", __FUNCTION__, __LINE__);
+ return INVALID_OPERATION;
+}
+
}; // namespace android
diff --git a/services/camera/libcameraservice/api1/CameraClient.h b/services/camera/libcameraservice/api1/CameraClient.h
index 95616b2..66e57d5 100644
--- a/services/camera/libcameraservice/api1/CameraClient.h
+++ b/services/camera/libcameraservice/api1/CameraClient.h
@@ -44,7 +44,7 @@
virtual status_t startPreview();
virtual void stopPreview();
virtual bool previewEnabled();
- virtual status_t storeMetaDataInBuffers(bool enabled);
+ virtual status_t setVideoBufferMode(int32_t videoBufferMode);
virtual status_t startRecording();
virtual void stopRecording();
virtual bool recordingEnabled();
@@ -55,6 +55,7 @@
virtual status_t setParameters(const String8& params);
virtual String8 getParameters() const;
virtual status_t sendCommand(int32_t cmd, int32_t arg1, int32_t arg2);
+ virtual status_t setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer);
// Interface used by CameraService
CameraClient(const sp<CameraService>& cameraService,
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index f901dda..7a97396 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -30,6 +30,7 @@
#include "Parameters.h"
#include "system/camera.h"
#include "hardware/camera_common.h"
+#include <camera/ICamera.h>
#include <media/MediaProfiles.h>
#include <media/mediarecorder.h>
@@ -870,8 +871,9 @@
}
// Set up initial state for non-Camera.Parameters state variables
-
- storeMetadataInBuffers = true;
+ videoFormat = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
+ videoDataSpace = HAL_DATASPACE_BT709;
+ videoBufferMode = ICamera::VIDEO_BUFFER_MODE_DATA_CALLBACK_YUV;
playShutterSound = true;
enableFaceDetect = false;
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index c5bbf63..c437722 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -131,7 +131,8 @@
int zoom;
- int videoWidth, videoHeight;
+ int videoWidth, videoHeight, videoFormat;
+ android_dataspace videoDataSpace;
bool recordingHint;
bool videoStabilization;
@@ -141,7 +142,8 @@
// These parameters are also part of the camera API-visible state, but not
// directly listed in Camera.Parameters
- bool storeMetadataInBuffers;
+ // One of ICamera::VIDEO_BUFFER_MODE_*
+ int32_t videoBufferMode;
bool playShutterSound;
bool enableFaceDetect;
diff --git a/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp b/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
index 88a0f50..211bdae 100644
--- a/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
@@ -49,13 +49,7 @@
mPreviewRequestId(Camera2Client::kPreviewRequestIdStart),
mPreviewStreamId(NO_STREAM),
mRecordingRequestId(Camera2Client::kRecordingRequestIdStart),
- mRecordingStreamId(NO_STREAM),
- mRecordingFrameAvailable(false),
- mRecordingHeapCount(kDefaultRecordingHeapCount),
- mRecordingHeapFree(kDefaultRecordingHeapCount),
- mRecordingFormat(kDefaultRecordingFormat),
- mRecordingDataSpace(kDefaultRecordingDataSpace),
- mRecordingGrallocUsage(kDefaultRecordingGrallocUsage)
+ mRecordingStreamId(NO_STREAM)
{
}
@@ -78,11 +72,30 @@
return OK;
}
+status_t StreamingProcessor::setRecordingWindow(sp<Surface> window) {
+ ATRACE_CALL();
+ status_t res;
+
+ res = deleteRecordingStream();
+ if (res != OK) return res;
+
+ Mutex::Autolock m(mMutex);
+
+ mRecordingWindow = window;
+
+ return OK;
+}
+
bool StreamingProcessor::haveValidPreviewWindow() const {
Mutex::Autolock m(mMutex);
return mPreviewWindow != 0;
}
+bool StreamingProcessor::haveValidRecordingWindow() const {
+ Mutex::Autolock m(mMutex);
+ return mRecordingWindow != nullptr;
+}
+
status_t StreamingProcessor::updatePreviewRequest(const Parameters ¶ms) {
ATRACE_CALL();
status_t res;
@@ -244,86 +257,6 @@
return mPreviewStreamId;
}
-status_t StreamingProcessor::setRecordingBufferCount(size_t count) {
- ATRACE_CALL();
- // Make sure we can support this many buffer slots
- if (count > BufferQueue::NUM_BUFFER_SLOTS) {
- ALOGE("%s: Camera %d: Too many recording buffers requested: %zu, max %d",
- __FUNCTION__, mId, count, BufferQueue::NUM_BUFFER_SLOTS);
- return BAD_VALUE;
- }
-
- Mutex::Autolock m(mMutex);
-
- ALOGV("%s: Camera %d: New recording buffer count from encoder: %zu",
- __FUNCTION__, mId, count);
-
- // Need to re-size consumer and heap
- if (mRecordingHeapCount != count) {
- ALOGV("%s: Camera %d: Resetting recording heap and consumer",
- __FUNCTION__, mId);
-
- if (isStreamActive(mActiveStreamIds, mRecordingStreamId)) {
- ALOGE("%s: Camera %d: Setting recording buffer count when "
- "recording stream is already active!", __FUNCTION__,
- mId);
- return INVALID_OPERATION;
- }
-
- releaseAllRecordingFramesLocked();
-
- if (mRecordingHeap != 0) {
- mRecordingHeap.clear();
- }
- mRecordingHeapCount = count;
- mRecordingHeapFree = count;
-
- mRecordingConsumer.clear();
- }
-
- return OK;
-}
-
-status_t StreamingProcessor::setRecordingFormat(int format,
- android_dataspace dataSpace) {
- ATRACE_CALL();
-
- Mutex::Autolock m(mMutex);
-
- ALOGV("%s: Camera %d: New recording format/dataspace from encoder: %X, %X",
- __FUNCTION__, mId, format, dataSpace);
-
- mRecordingFormat = format;
- mRecordingDataSpace = dataSpace;
- int prevGrallocUsage = mRecordingGrallocUsage;
- if (mRecordingFormat == HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED) {
- mRecordingGrallocUsage = GRALLOC_USAGE_HW_VIDEO_ENCODER;
- } else {
- mRecordingGrallocUsage = GRALLOC_USAGE_SW_READ_OFTEN;
- }
-
- ALOGV("%s: Camera %d: New recording gralloc usage: %08X", __FUNCTION__, mId,
- mRecordingGrallocUsage);
-
- if (prevGrallocUsage != mRecordingGrallocUsage) {
- ALOGV("%s: Camera %d: Resetting recording consumer for new usage",
- __FUNCTION__, mId);
-
- if (isStreamActive(mActiveStreamIds, mRecordingStreamId)) {
- ALOGE("%s: Camera %d: Changing recording format when "
- "recording stream is already active!", __FUNCTION__,
- mId);
- return INVALID_OPERATION;
- }
-
- releaseAllRecordingFramesLocked();
-
- mRecordingConsumer.clear();
- }
-
- return OK;
-}
-
status_t StreamingProcessor::updateRecordingRequest(const Parameters ¶ms) {
ATRACE_CALL();
status_t res;
@@ -395,11 +328,11 @@
return res;
}
- if (mRecordingConsumer == 0 ||
+ if (mRecordingWindow == nullptr ||
currentWidth != (uint32_t)params.videoWidth ||
currentHeight != (uint32_t)params.videoHeight ||
- currentFormat != (uint32_t)mRecordingFormat ||
- currentDataSpace != mRecordingDataSpace) {
+ currentFormat != (uint32_t)params.videoFormat ||
+ currentDataSpace != params.videoDataSpace) {
*needsUpdate = true;
}
*needsUpdate = false;
@@ -417,26 +350,6 @@
return INVALID_OPERATION;
}
- bool newConsumer = false;
- if (mRecordingConsumer == 0) {
- ALOGV("%s: Camera %d: Creating recording consumer with %zu + 1 "
- "consumer-side buffers", __FUNCTION__, mId, mRecordingHeapCount);
- // Create CPU buffer queue endpoint. We need one more buffer here so that we can
- // always acquire and free a buffer when the heap is full; otherwise the consumer
- // will have buffers in flight we'll never clear out.
- sp<IGraphicBufferProducer> producer;
- sp<IGraphicBufferConsumer> consumer;
- BufferQueue::createBufferQueue(&producer, &consumer);
- mRecordingConsumer = new BufferItemConsumer(consumer,
- mRecordingGrallocUsage,
- mRecordingHeapCount + 1);
- mRecordingConsumer->setFrameAvailableListener(this);
- mRecordingConsumer->setName(String8("Camera2-RecordingConsumer"));
- mRecordingWindow = new Surface(producer);
- newConsumer = true;
- // Allocate memory later, since we don't know buffer size until receipt
- }
-
if (mRecordingStreamId != NO_STREAM) {
// Check if stream parameters have to change
uint32_t currentWidth, currentHeight;
@@ -453,9 +366,8 @@
}
if (currentWidth != (uint32_t)params.videoWidth ||
currentHeight != (uint32_t)params.videoHeight ||
- currentFormat != (uint32_t)mRecordingFormat ||
- currentDataSpace != mRecordingDataSpace ||
- newConsumer) {
+ currentFormat != (uint32_t)params.videoFormat ||
+ currentDataSpace != params.videoDataSpace) {
// TODO: Should wait to be sure previous recording has finished
res = device->deleteStream(mRecordingStreamId);
@@ -475,10 +387,9 @@
}
if (mRecordingStreamId == NO_STREAM) {
- mRecordingFrameCount = 0;
res = device->createStream(mRecordingWindow,
params.videoWidth, params.videoHeight,
- mRecordingFormat, mRecordingDataSpace,
+ params.videoFormat, params.videoDataSpace,
CAMERA3_STREAM_ROTATION_0, &mRecordingStreamId);
if (res != OK) {
ALOGE("%s: Camera %d: Can't create output stream for recording: "
@@ -542,20 +453,6 @@
Mutex::Autolock m(mMutex);
- // If a recording stream is being started up and no recording
- // stream is active yet, free up any outstanding buffers left
- // from the previous recording session. There should never be
- // any, so if there are, warn about it.
- bool isRecordingStreamIdle = !isStreamActive(mActiveStreamIds, mRecordingStreamId);
- bool startRecordingStream = isStreamActive(outputStreams, mRecordingStreamId);
- if (startRecordingStream && isRecordingStreamIdle) {
- releaseAllRecordingFramesLocked();
- }
-
- ALOGV("%s: Camera %d: %s started, recording heap has %zu free of %zu",
- __FUNCTION__, mId, (type == PREVIEW) ? "preview" : "recording",
- mRecordingHeapFree, mRecordingHeapCount);
-
CameraMetadata &request = (type == PREVIEW) ?
mPreviewRequest : mRecordingRequest;
@@ -692,272 +589,6 @@
return OK;
}
-void StreamingProcessor::onFrameAvailable(const BufferItem& /*item*/) {
- ATRACE_CALL();
- Mutex::Autolock l(mMutex);
- if (!mRecordingFrameAvailable) {
- mRecordingFrameAvailable = true;
- mRecordingFrameAvailableSignal.signal();
- }
-
-}
-
-bool StreamingProcessor::threadLoop() {
- status_t res;
-
- {
- Mutex::Autolock l(mMutex);
- while (!mRecordingFrameAvailable) {
- res = mRecordingFrameAvailableSignal.waitRelative(
- mMutex, kWaitDuration);
- if (res == TIMED_OUT) return true;
- }
- mRecordingFrameAvailable = false;
- }
-
- do {
- res = processRecordingFrame();
- } while (res == OK);
-
- return true;
-}
-
-status_t StreamingProcessor::processRecordingFrame() {
- ATRACE_CALL();
- status_t res;
- sp<Camera2Heap> recordingHeap;
- size_t heapIdx = 0;
- nsecs_t timestamp;
-
- sp<Camera2Client> client = mClient.promote();
- if (client == 0) {
- // Discard frames during shutdown
- BufferItem imgBuffer;
- res = mRecordingConsumer->acquireBuffer(&imgBuffer, 0);
- if (res != OK) {
- if (res != BufferItemConsumer::NO_BUFFER_AVAILABLE) {
- ALOGE("%s: Camera %d: Can't acquire recording buffer: %s (%d)",
- __FUNCTION__, mId, strerror(-res), res);
- }
- return res;
- }
- mRecordingConsumer->releaseBuffer(imgBuffer);
- return OK;
- }
-
- {
- /* acquire SharedParameters before mMutex so we don't dead lock
- with Camera2Client code calling into StreamingProcessor */
- SharedParameters::Lock l(client->getParameters());
- Mutex::Autolock m(mMutex);
- BufferItem imgBuffer;
- res = mRecordingConsumer->acquireBuffer(&imgBuffer, 0);
- if (res != OK) {
- if (res != BufferItemConsumer::NO_BUFFER_AVAILABLE) {
- ALOGE("%s: Camera %d: Can't acquire recording buffer: %s (%d)",
- __FUNCTION__, mId, strerror(-res), res);
- }
- return res;
- }
- timestamp = imgBuffer.mTimestamp;
-
- mRecordingFrameCount++;
- ALOGVV("OnRecordingFrame: Frame %d", mRecordingFrameCount);
-
- if (l.mParameters.state != Parameters::RECORD &&
- l.mParameters.state != Parameters::VIDEO_SNAPSHOT) {
- ALOGV("%s: Camera %d: Discarding recording image buffers "
- "received after recording done", __FUNCTION__,
- mId);
- mRecordingConsumer->releaseBuffer(imgBuffer);
- return INVALID_OPERATION;
- }
-
- if (mRecordingHeap == 0) {
- size_t payloadSize = sizeof(VideoNativeMetadata);
- ALOGV("%s: Camera %d: Creating recording heap with %zu buffers of "
- "size %zu bytes", __FUNCTION__, mId,
- mRecordingHeapCount, payloadSize);
-
- mRecordingHeap = new Camera2Heap(payloadSize, mRecordingHeapCount,
- "Camera2Client::RecordingHeap");
- if (mRecordingHeap->mHeap->getSize() == 0) {
- ALOGE("%s: Camera %d: Unable to allocate memory for recording",
- __FUNCTION__, mId);
- mRecordingConsumer->releaseBuffer(imgBuffer);
- return NO_MEMORY;
- }
- for (size_t i = 0; i < mRecordingBuffers.size(); i++) {
- if (mRecordingBuffers[i].mSlot !=
- BufferItemConsumer::INVALID_BUFFER_SLOT) {
- ALOGE("%s: Camera %d: Non-empty recording buffers list!",
- __FUNCTION__, mId);
- }
- }
- mRecordingBuffers.clear();
- mRecordingBuffers.setCapacity(mRecordingHeapCount);
- mRecordingBuffers.insertAt(0, mRecordingHeapCount);
-
- mRecordingHeapHead = 0;
- mRecordingHeapFree = mRecordingHeapCount;
- }
-
- if (mRecordingHeapFree == 0) {
- ALOGE("%s: Camera %d: No free recording buffers, dropping frame",
- __FUNCTION__, mId);
- mRecordingConsumer->releaseBuffer(imgBuffer);
- return NO_MEMORY;
- }
-
- heapIdx = mRecordingHeapHead;
- mRecordingHeapHead = (mRecordingHeapHead + 1) % mRecordingHeapCount;
- mRecordingHeapFree--;
-
- ALOGVV("%s: Camera %d: Timestamp %lld",
- __FUNCTION__, mId, timestamp);
-
- ssize_t offset;
- size_t size;
- sp<IMemoryHeap> heap =
- mRecordingHeap->mBuffers[heapIdx]->getMemory(&offset,
- &size);
-
- VideoNativeMetadata *payload = reinterpret_cast<VideoNativeMetadata*>(
- (uint8_t*)heap->getBase() + offset);
- payload->eType = kMetadataBufferTypeANWBuffer;
- payload->pBuffer = imgBuffer.mGraphicBuffer->getNativeBuffer();
- payload->nFenceFd = -1;
-
- ALOGVV("%s: Camera %d: Sending out ANWBuffer %p",
- __FUNCTION__, mId, payload->pBuffer);
-
- mRecordingBuffers.replaceAt(imgBuffer, heapIdx);
- recordingHeap = mRecordingHeap;
- }
-
- // Call outside locked parameters to allow re-entrancy from notification
- Camera2Client::SharedCameraCallbacks::Lock l(client->mSharedCameraCallbacks);
- if (l.mRemoteCallback != 0) {
- l.mRemoteCallback->dataCallbackTimestamp(timestamp,
- CAMERA_MSG_VIDEO_FRAME,
- recordingHeap->mBuffers[heapIdx]);
- } else {
- ALOGW("%s: Camera %d: Remote callback gone", __FUNCTION__, mId);
- }
-
- return OK;
-}
-
-void StreamingProcessor::releaseRecordingFrame(const sp<IMemory>& mem) {
- ATRACE_CALL();
- status_t res;
-
- Mutex::Autolock m(mMutex);
- // Make sure this is for the current heap
- ssize_t offset;
- size_t size;
- sp<IMemoryHeap> heap = mem->getMemory(&offset, &size);
- if (heap->getHeapID() != mRecordingHeap->mHeap->getHeapID()) {
- ALOGW("%s: Camera %d: Mismatched heap ID, ignoring release "
- "(got %x, expected %x)", __FUNCTION__, mId,
- heap->getHeapID(), mRecordingHeap->mHeap->getHeapID());
- return;
- }
-
- VideoNativeMetadata *payload = reinterpret_cast<VideoNativeMetadata*>(
- (uint8_t*)heap->getBase() + offset);
-
- if (payload->eType != kMetadataBufferTypeANWBuffer) {
- ALOGE("%s: Camera %d: Recording frame type invalid (got %x, expected %x)",
- __FUNCTION__, mId, payload->eType,
- kMetadataBufferTypeANWBuffer);
- return;
- }
-
- // Release the buffer back to the recording queue
- size_t itemIndex;
- for (itemIndex = 0; itemIndex < mRecordingBuffers.size(); itemIndex++) {
- const BufferItem item = mRecordingBuffers[itemIndex];
- if (item.mSlot != BufferItemConsumer::INVALID_BUFFER_SLOT &&
- item.mGraphicBuffer->getNativeBuffer() == payload->pBuffer) {
- break;
- }
- }
-
- if (itemIndex == mRecordingBuffers.size()) {
- ALOGE("%s: Camera %d: Can't find returned ANW Buffer %p in list of "
- "outstanding buffers", __FUNCTION__, mId,
- payload->pBuffer);
- return;
- }
-
- ALOGVV("%s: Camera %d: Freeing returned ANW buffer %p index %d", __FUNCTION__,
- mId, payload->pBuffer, itemIndex);
-
- res = mRecordingConsumer->releaseBuffer(mRecordingBuffers[itemIndex]);
- if (res != OK) {
- ALOGE("%s: Camera %d: Unable to free recording frame "
- "(Returned ANW buffer: %p): %s (%d)", __FUNCTION__,
- mId, payload->pBuffer, strerror(-res), res);
- return;
- }
- mRecordingBuffers.replaceAt(itemIndex);
-
- mRecordingHeapFree++;
- ALOGV_IF(mRecordingHeapFree == mRecordingHeapCount,
- "%s: Camera %d: All %d recording buffers returned",
- __FUNCTION__, mId, mRecordingHeapCount);
-}
-
-void StreamingProcessor::releaseAllRecordingFramesLocked() {
- ATRACE_CALL();
- status_t res;
-
- if (mRecordingConsumer == 0) {
- return;
- }
-
- ALOGV("%s: Camera %d: Releasing all recording buffers", __FUNCTION__,
- mId);
-
- size_t releasedCount = 0;
- for (size_t itemIndex = 0; itemIndex < mRecordingBuffers.size(); itemIndex++) {
- const BufferItem item = mRecordingBuffers[itemIndex];
- if (item.mSlot != BufferItemConsumer::INVALID_BUFFER_SLOT) {
- res = mRecordingConsumer->releaseBuffer(mRecordingBuffers[itemIndex]);
- if (res != OK) {
- ALOGE("%s: Camera %d: Unable to free recording frame "
- "(buffer_handle_t: %p): %s (%d)", __FUNCTION__,
- mId, item.mGraphicBuffer->handle, strerror(-res), res);
- }
- mRecordingBuffers.replaceAt(itemIndex);
- releasedCount++;
- }
- }
-
- if (releasedCount > 0) {
- ALOGW("%s: Camera %d: Force-freed %zu outstanding buffers "
- "from previous recording session", __FUNCTION__, mId, releasedCount);
- ALOGE_IF(releasedCount != mRecordingHeapCount - mRecordingHeapFree,
- "%s: Camera %d: Force-freed %zu buffers, but expected %zu",
- __FUNCTION__, mId, releasedCount, mRecordingHeapCount - mRecordingHeapFree);
- }
-
- mRecordingHeapHead = 0;
- mRecordingHeapFree = mRecordingHeapCount;
-}
-
-bool StreamingProcessor::isStreamActive(const Vector<int32_t> &streams,
- int32_t recordingStreamId) {
- for (size_t i = 0; i < streams.size(); i++) {
- if (streams[i] == recordingStreamId) {
- return true;
- }
- }
- return false;
-}
-
-
status_t StreamingProcessor::dump(int fd, const Vector<String16>& /*args*/) {
String8 result;
diff --git a/services/camera/libcameraservice/api1/client2/StreamingProcessor.h b/services/camera/libcameraservice/api1/client2/StreamingProcessor.h
index 0b17eae..57e6389 100644
--- a/services/camera/libcameraservice/api1/client2/StreamingProcessor.h
+++ b/services/camera/libcameraservice/api1/client2/StreamingProcessor.h
@@ -37,24 +37,22 @@
/**
* Management and processing for preview and recording streams
*/
-class StreamingProcessor:
- public Thread, public BufferItemConsumer::FrameAvailableListener {
+class StreamingProcessor : public virtual VirtualLightRefBase {
public:
StreamingProcessor(sp<Camera2Client> client);
~StreamingProcessor();
status_t setPreviewWindow(sp<Surface> window);
+ status_t setRecordingWindow(sp<Surface> window);
bool haveValidPreviewWindow() const;
+ bool haveValidRecordingWindow() const;
status_t updatePreviewRequest(const Parameters ¶ms);
status_t updatePreviewStream(const Parameters ¶ms);
status_t deletePreviewStream();
int getPreviewStreamId() const;
- status_t setRecordingBufferCount(size_t count);
- status_t setRecordingFormat(int format, android_dataspace_t dataspace);
-
status_t updateRecordingRequest(const Parameters ¶ms);
// If needsUpdate is set to true, a updateRecordingStream call with params will recreate
// recording stream
@@ -81,11 +79,6 @@
status_t getActiveRequestId() const;
status_t incrementStreamingIds();
- // Callback for new recording frames from HAL
- virtual void onFrameAvailable(const BufferItem& item);
- // Callback from stagefright which returns used recording frames
- void releaseRecordingFrame(const sp<IMemory>& mem);
-
status_t dump(int fd, const Vector<String16>& args);
private:
@@ -110,47 +103,10 @@
CameraMetadata mPreviewRequest;
sp<Surface> mPreviewWindow;
- // Recording-related members
- static const nsecs_t kWaitDuration = 50000000; // 50 ms
-
int32_t mRecordingRequestId;
int mRecordingStreamId;
- int mRecordingFrameCount;
- sp<BufferItemConsumer> mRecordingConsumer;
sp<Surface> mRecordingWindow;
CameraMetadata mRecordingRequest;
- sp<camera2::Camera2Heap> mRecordingHeap;
-
- bool mRecordingFrameAvailable;
- Condition mRecordingFrameAvailableSignal;
-
- static const size_t kDefaultRecordingHeapCount = 8;
- size_t mRecordingHeapCount;
- Vector<BufferItem> mRecordingBuffers;
- size_t mRecordingHeapHead, mRecordingHeapFree;
-
- static const int kDefaultRecordingFormat =
- HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
- int mRecordingFormat;
-
- static const android_dataspace kDefaultRecordingDataSpace =
- HAL_DATASPACE_BT709;
- android_dataspace mRecordingDataSpace;
-
- static const int kDefaultRecordingGrallocUsage =
- GRALLOC_USAGE_HW_VIDEO_ENCODER;
- int mRecordingGrallocUsage;
-
- virtual bool threadLoop();
-
- status_t processRecordingFrame();
-
- // Unilaterally free any buffers still outstanding to stagefright
- void releaseAllRecordingFramesLocked();
-
- // Determine if the specified stream is currently in use
- static bool isStreamActive(const Vector<int32_t> &streams,
- int32_t recordingStreamId);
};