Merge "Add frame dump output"
diff --git a/drm/mediadrm/plugins/mock/Android.mk b/drm/mediadrm/plugins/mock/Android.mk
index ada23a2..26c245b 100644
--- a/drm/mediadrm/plugins/mock/Android.mk
+++ b/drm/mediadrm/plugins/mock/Android.mk
@@ -21,7 +21,8 @@
LOCAL_MODULE := libmockdrmcryptoplugin
-LOCAL_MODULE_PATH := $(TARGET_OUT_VENDOR_SHARED_LIBRARIES)/mediadrm
+LOCAL_PROPRIETARY_MODULE := true
+LOCAL_MODULE_RELATIVE_PATH := mediadrm
LOCAL_SHARED_LIBRARIES := \
libutils liblog
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index fd86737..28fdfd4 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -118,6 +118,7 @@
static bool routedToA2dpOutput(audio_stream_type_t streamType);
+ // return status NO_ERROR implies *buffSize > 0
static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask, size_t* buffSize);
diff --git a/include/media/IMediaHTTPConnection.h b/include/media/IMediaHTTPConnection.h
index e048b64..2a63eb7 100644
--- a/include/media/IMediaHTTPConnection.h
+++ b/include/media/IMediaHTTPConnection.h
@@ -38,6 +38,7 @@
virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
virtual off64_t getSize() = 0;
virtual status_t getMIMEType(String8 *mimeType) = 0;
+ virtual status_t getUri(String8 *uri) = 0;
private:
DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPConnection);
diff --git a/libvideoeditor/lvpp/Android.mk b/libvideoeditor/lvpp/Android.mk
index 2286827..860d351 100755
--- a/libvideoeditor/lvpp/Android.mk
+++ b/libvideoeditor/lvpp/Android.mk
@@ -71,7 +71,6 @@
$(TOP)/frameworks/av/media/libstagefright \
$(TOP)/frameworks/av/media/libstagefright/include \
$(TOP)/frameworks/av/media/libstagefright/rtsp \
- $(call include-path-for, corecg graphics) \
$(TOP)/frameworks/av/libvideoeditor/osal/inc \
$(TOP)/frameworks/av/libvideoeditor/vss/common/inc \
$(TOP)/frameworks/av/libvideoeditor/vss/mcs/inc \
diff --git a/libvideoeditor/vss/stagefrightshells/src/Android.mk b/libvideoeditor/vss/stagefrightshells/src/Android.mk
index e30b85d..9188942 100755
--- a/libvideoeditor/vss/stagefrightshells/src/Android.mk
+++ b/libvideoeditor/vss/stagefrightshells/src/Android.mk
@@ -33,7 +33,6 @@
$(TOP)/frameworks/av/media/libstagefright \
$(TOP)/frameworks/av/media/libstagefright/include \
$(TOP)/frameworks/av/media/libstagefright/rtsp \
- $(call include-path-for, corecg graphics) \
$(TOP)/frameworks/av/libvideoeditor/lvpp \
$(TOP)/frameworks/av/libvideoeditor/osal/inc \
$(TOP)/frameworks/av/libvideoeditor/vss/inc \
diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c
index 4ee05f2..d25dc9b 100644
--- a/media/libeffects/downmix/EffectDownmix.c
+++ b/media/libeffects/downmix/EffectDownmix.c
@@ -629,7 +629,9 @@
return -EINVAL;
}
- memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
+ if (&pDwmModule->config != pConfig) {
+ memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
+ }
if (init) {
pDownmixer->type = DOWNMIX_TYPE_FOLD;
diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk
index dd2d306..c92c543 100644
--- a/media/libeffects/visualizer/Android.mk
+++ b/media/libeffects/visualizer/Android.mk
@@ -17,7 +17,6 @@
LOCAL_MODULE:= libvisualizer
LOCAL_C_INCLUDES := \
- $(call include-path-for, graphics corecg) \
$(call include-path-for, audio-effects)
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index e0acae6..f3770e4 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -72,7 +72,6 @@
LOCAL_MODULE:= libmedia
LOCAL_C_INCLUDES := \
- $(call include-path-for, graphics corecg) \
$(TOP)/frameworks/native/include/media/openmax \
external/icu4c/common \
external/icu4c/i18n \
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 700718d..4438dfd 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -41,30 +41,22 @@
return BAD_VALUE;
}
- // default to 0 in case of error
- *frameCount = 0;
-
- size_t size = 0;
+ size_t size;
status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
if (status != NO_ERROR) {
- ALOGE("AudioSystem could not query the input buffer size; status %d", status);
- return NO_INIT;
+ ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, "
+ "channelMask %#x; status %d", sampleRate, format, channelMask, status);
+ return status;
}
- if (size == 0) {
+ // We double the size of input buffer for ping pong use of record buffer.
+ // Assumes audio_is_linear_pcm(format)
+ if ((*frameCount = (size * 2) / (popcount(channelMask) * audio_bytes_per_sample(format))) == 0) {
ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
- // We double the size of input buffer for ping pong use of record buffer.
- size <<= 1;
-
- // Assumes audio_is_linear_pcm(format)
- uint32_t channelCount = popcount(channelMask);
- size /= channelCount * audio_bytes_per_sample(format);
-
- *frameCount = size;
return NO_ERROR;
}
@@ -110,10 +102,8 @@
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
- if (mAudioRecord != 0) {
- mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
- mAudioRecord.clear();
- }
+ mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
+ mAudioRecord.clear();
IPCThreadState::self()->flushCommands();
AudioSystem::releaseAudioSessionId(mSessionId, -1);
}
@@ -133,6 +123,11 @@
transfer_type transferType,
audio_input_flags_t flags)
{
+ ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %d, "
+ "notificationFrames %d, sessionId %d, transferType %d, flags %#x",
+ inputSource, sampleRate, format, channelMask, frameCountInt, notificationFrames,
+ sessionId, transferType, flags);
+
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
@@ -163,16 +158,15 @@
}
size_t frameCount = frameCountInt;
- ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
- frameCount);
-
AutoMutex lock(mLock);
+ // invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
+ // handle default values first.
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
@@ -209,15 +203,19 @@
uint32_t channelCount = popcount(channelMask);
mChannelCount = channelCount;
- // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t)
- mFrameSize = channelCount * audio_bytes_per_sample(format);
+ if (audio_is_linear_pcm(format)) {
+ mFrameSize = channelCount * audio_bytes_per_sample(format);
+ } else {
+ mFrameSize = sizeof(uint8_t);
+ }
// validate framecount
- size_t minFrameCount = 0;
+ size_t minFrameCount;
status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
sampleRate, format, channelMask);
if (status != NO_ERROR) {
- ALOGE("getMinFrameCount() failed; status %d", status);
+ ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d",
+ sampleRate, format, channelMask, status);
return status;
}
ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
@@ -258,7 +256,6 @@
mActive = false;
mCbf = cbf;
- mRefreshRemaining = true;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
@@ -485,10 +482,12 @@
ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
"session ID changed from %d to %d", originalSessionId, mSessionId);
- if (record == 0 || status != NO_ERROR) {
+ if (status != NO_ERROR) {
ALOGE("AudioFlinger could not create record track, status: %d", status);
goto release;
}
+ ALOG_ASSERT(record != 0);
+
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
@@ -502,27 +501,21 @@
ALOGE("Could not get control block pointer");
return NO_INIT;
}
+ // invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
-
- // We retain a copy of the I/O handle, but don't own the reference
- mInput = input;
mAudioRecord = record;
+
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
- // note that temp is the (possibly revised) value of mFrameCount
+ // note that temp is the (possibly revised) value of frameCount
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
}
frameCount = temp;
- // If IAudioRecord is re-created, don't let the requested frameCount
- // decrease. This can confuse clients that cache frameCount().
- if (frameCount > mReqFrameCount) {
- mReqFrameCount = frameCount;
- }
// FIXME missing fast track frameCount logic
mAwaitBoost = false;
@@ -544,10 +537,21 @@
}
}
- // starting address of buffers in shared memory
+ // We retain a copy of the I/O handle, but don't own the reference
+ mInput = input;
+ mRefreshRemaining = true;
+
+ // Starting address of buffers in shared memory, immediately after the control block. This
+ // address is for the mapping within client address space. AudioFlinger::TrackBase::mBuffer
+ // is for the server address space.
void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
mFrameCount = frameCount;
+ // If IAudioRecord is re-created, don't let the requested frameCount
+ // decrease. This can confuse clients that cache frameCount().
+ if (frameCount > mReqFrameCount) {
+ mReqFrameCount = frameCount;
+ }
// update proxy
mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 5c62260..adf3847 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -195,6 +195,11 @@
int uid,
pid_t pid)
{
+ ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %d, "
+ "flags #%x, notificationFrames %d, sessionId %d, transferType %d",
+ streamType, sampleRate, format, channelMask, frameCountInt, flags, notificationFrames,
+ sessionId, transferType);
+
switch (transferType) {
case TRANSFER_DEFAULT:
if (sharedBuffer != 0) {
@@ -288,6 +293,9 @@
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
+ mChannelMask = channelMask;
+ uint32_t channelCount = popcount(channelMask);
+ mChannelCount = channelCount;
// AudioFlinger does not currently support 8-bit data in shared memory
if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
@@ -311,10 +319,6 @@
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
- mChannelMask = channelMask;
- uint32_t channelCount = popcount(channelMask);
- mChannelCount = channelCount;
-
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
mFrameSizeAF = channelCount * sizeof(int16_t);
@@ -1012,10 +1016,12 @@
mClientUid,
&status);
- if (track == 0) {
+ if (status != NO_ERROR) {
ALOGE("AudioFlinger could not create track, status: %d", status);
goto release;
}
+ ALOG_ASSERT(track != 0);
+
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
@@ -1035,6 +1041,7 @@
mDeathNotifier.clear();
}
mAudioTrack = track;
+
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
@@ -1046,6 +1053,7 @@
ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
}
frameCount = temp;
+
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
@@ -1099,6 +1107,7 @@
mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME don't believe this lie
mLatency = afLatency + (1000*frameCount) / mSampleRate;
+
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
diff --git a/media/libmedia/IMediaHTTPConnection.cpp b/media/libmedia/IMediaHTTPConnection.cpp
index 622d9cf..22c470a 100644
--- a/media/libmedia/IMediaHTTPConnection.cpp
+++ b/media/libmedia/IMediaHTTPConnection.cpp
@@ -33,6 +33,7 @@
READ_AT,
GET_SIZE,
GET_MIME_TYPE,
+ GET_URI
};
struct BpMediaHTTPConnection : public BpInterface<IMediaHTTPConnection> {
@@ -147,6 +148,26 @@
return OK;
}
+ virtual status_t getUri(String8 *uri) {
+ *uri = String8("");
+
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ IMediaHTTPConnection::getInterfaceDescriptor());
+
+ remote()->transact(GET_URI, data, &reply);
+
+ int32_t exceptionCode = reply.readExceptionCode();
+
+ if (exceptionCode) {
+ return UNKNOWN_ERROR;
+ }
+
+ *uri = String8(reply.readString16());
+
+ return OK;
+ }
+
private:
sp<IMemory> mMemory;
};
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index 8f21632..4189a5e 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -45,7 +45,6 @@
libstagefright_rtsp \
LOCAL_C_INCLUDES := \
- $(call include-path-for, graphics corecg) \
$(TOP)/frameworks/av/media/libstagefright/include \
$(TOP)/frameworks/av/media/libstagefright/rtsp \
$(TOP)/frameworks/av/media/libstagefright/wifi-display \
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 845a589..5b7a236 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -748,7 +748,7 @@
return OK;
}
-status_t StagefrightRecorder::prepare() {
+status_t StagefrightRecorder::prepareInternal() {
ALOGV("prepare");
if (mOutputFd < 0) {
ALOGE("Output file descriptor is invalid");
@@ -794,6 +794,13 @@
return status;
}
+status_t StagefrightRecorder::prepare() {
+ if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+ return prepareInternal();
+ }
+ return OK;
+}
+
status_t StagefrightRecorder::start() {
ALOGV("start");
if (mOutputFd < 0) {
@@ -801,15 +808,20 @@
return INVALID_OPERATION;
}
- // Get UID here for permission checking
- mClientUid = IPCThreadState::self()->getCallingUid();
+ status_t status = OK;
+
+ if (mVideoSource != VIDEO_SOURCE_SURFACE) {
+ status = prepareInternal();
+ if (status != OK) {
+ return status;
+ }
+ }
+
if (mWriter == NULL) {
ALOGE("File writer is not avaialble");
return UNKNOWN_ERROR;
}
- status_t status = OK;
-
switch (mOutputFormat) {
case OUTPUT_FORMAT_DEFAULT:
case OUTPUT_FORMAT_THREE_GPP:
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 7d6abd3..377d168 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -127,6 +127,7 @@
sp<IGraphicBufferProducer> mGraphicBufferProducer;
sp<ALooper> mLooper;
+ status_t prepareInternal();
status_t setupMPEG4Recording();
void setupMPEG4MetaData(sp<MetaData> *meta);
status_t setupAMRRecording();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index d47ac98..a750ad0 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -1006,7 +1006,14 @@
&NuPlayer::performScanSources));
}
- flushDecoder(audio, formatChange);
+ sp<AMessage> newFormat = mSource->getFormat(audio);
+ sp<Decoder> &decoder = audio ? mAudioDecoder : mVideoDecoder;
+ if (formatChange && !decoder->supportsSeamlessFormatChange(newFormat)) {
+ flushDecoder(audio, /* needShutdown = */ true);
+ } else {
+ flushDecoder(audio, /* needShutdown = */ false);
+ err = OK;
+ }
} else {
// This stream is unaffected by the discontinuity
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 22f699e..2423fd5 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -67,6 +67,7 @@
// queue.
bool needDedicatedLooper = !strncasecmp(mime.c_str(), "video/", 6);
+ mFormat = format;
mCodec = new ACodec;
if (needDedicatedLooper && mCodecLooper == NULL) {
@@ -147,5 +148,65 @@
}
}
+bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const {
+ if (targetFormat == NULL) {
+ return true;
+ }
+
+ AString mime;
+ if (!targetFormat->findString("mime", &mime)) {
+ return false;
+ }
+
+ if (!strcasecmp(mime.c_str(), MEDIA_MIMETYPE_AUDIO_AAC)) {
+ // field-by-field comparison
+ const char * keys[] = { "channel-count", "sample-rate", "is-adts" };
+ for (unsigned int i = 0; i < sizeof(keys) / sizeof(keys[0]); i++) {
+ int32_t oldVal, newVal;
+ if (!mFormat->findInt32(keys[i], &oldVal) || !targetFormat->findInt32(keys[i], &newVal)
+ || oldVal != newVal) {
+ return false;
+ }
+ }
+
+ sp<ABuffer> oldBuf, newBuf;
+ if (mFormat->findBuffer("csd-0", &oldBuf) && targetFormat->findBuffer("csd-0", &newBuf)) {
+ if (oldBuf->size() != newBuf->size()) {
+ return false;
+ }
+ return !memcmp(oldBuf->data(), newBuf->data(), oldBuf->size());
+ }
+ }
+ return false;
+}
+
+bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetFormat) const {
+ if (mFormat == NULL) {
+ return false;
+ }
+
+ if (targetFormat == NULL) {
+ return true;
+ }
+
+ AString oldMime, newMime;
+ if (!mFormat->findString("mime", &oldMime)
+ || !targetFormat->findString("mime", &newMime)
+ || !(oldMime == newMime)) {
+ return false;
+ }
+
+ bool audio = !strncasecmp(oldMime.c_str(), "audio/", strlen("audio/"));
+ bool seamless;
+ if (audio) {
+ seamless = supportsSeamlessAudioFormatChange(targetFormat);
+ } else {
+ seamless = mCodec != NULL && mCodec->isConfiguredForAdaptivePlayback();
+ }
+
+ ALOGV("%s seamless support for %s", seamless ? "yes" : "no", oldMime.c_str());
+ return seamless;
+}
+
} // namespace android
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
index a876148..78ea74a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
@@ -36,6 +36,8 @@
void signalResume();
void initiateShutdown();
+ bool supportsSeamlessFormatChange(const sp<AMessage> &to) const;
+
protected:
virtual ~Decoder();
@@ -49,6 +51,7 @@
sp<AMessage> mNotify;
sp<NativeWindowWrapper> mNativeWindow;
+ sp<AMessage> mFormat;
sp<ACodec> mCodec;
sp<ALooper> mCodecLooper;
@@ -59,6 +62,8 @@
void onFillThisBuffer(const sp<AMessage> &msg);
+ bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const;
+
DISALLOW_EVIL_CONSTRUCTORS(Decoder);
};
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index f80772a..2a3fa04 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -913,6 +913,8 @@
case FOURCC('e', 'l', 's', 't'):
{
+ *offset += chunk_size;
+
// See 14496-12 8.6.6
uint8_t version;
if (mDataSource->readAt(data_offset, &version, 1) < 1) {
@@ -975,12 +977,13 @@
mLastTrack->meta->setInt32(kKeyEncoderPadding, paddingsamples);
}
}
- *offset += chunk_size;
break;
}
case FOURCC('f', 'r', 'm', 'a'):
{
+ *offset += chunk_size;
+
uint32_t original_fourcc;
if (mDataSource->readAt(data_offset, &original_fourcc, 4) < 4) {
return ERROR_IO;
@@ -994,12 +997,13 @@
mLastTrack->meta->setInt32(kKeyChannelCount, num_channels);
mLastTrack->meta->setInt32(kKeySampleRate, sample_rate);
}
- *offset += chunk_size;
break;
}
case FOURCC('t', 'e', 'n', 'c'):
{
+ *offset += chunk_size;
+
if (chunk_size < 32) {
return ERROR_MALFORMED;
}
@@ -1044,23 +1048,25 @@
mLastTrack->meta->setInt32(kKeyCryptoMode, defaultAlgorithmId);
mLastTrack->meta->setInt32(kKeyCryptoDefaultIVSize, defaultIVSize);
mLastTrack->meta->setData(kKeyCryptoKey, 'tenc', defaultKeyId, 16);
- *offset += chunk_size;
break;
}
case FOURCC('t', 'k', 'h', 'd'):
{
+ *offset += chunk_size;
+
status_t err;
if ((err = parseTrackHeader(data_offset, chunk_data_size)) != OK) {
return err;
}
- *offset += chunk_size;
break;
}
case FOURCC('p', 's', 's', 'h'):
{
+ *offset += chunk_size;
+
PsshInfo pssh;
if (mDataSource->readAt(data_offset + 4, &pssh.uuid, 16) < 16) {
@@ -1086,12 +1092,13 @@
}
mPssh.push_back(pssh);
- *offset += chunk_size;
break;
}
case FOURCC('m', 'd', 'h', 'd'):
{
+ *offset += chunk_size;
+
if (chunk_data_size < 4) {
return ERROR_MALFORMED;
}
@@ -1172,7 +1179,6 @@
mLastTrack->meta->setCString(
kKeyMediaLanguage, lang_code);
- *offset += chunk_size;
break;
}
@@ -1339,11 +1345,12 @@
mLastTrack->sampleTable->setChunkOffsetParams(
chunk_type, data_offset, chunk_data_size);
+ *offset += chunk_size;
+
if (err != OK) {
return err;
}
- *offset += chunk_size;
break;
}
@@ -1353,11 +1360,12 @@
mLastTrack->sampleTable->setSampleToChunkParams(
data_offset, chunk_data_size);
+ *offset += chunk_size;
+
if (err != OK) {
return err;
}
- *offset += chunk_size;
break;
}
@@ -1368,6 +1376,8 @@
mLastTrack->sampleTable->setSampleSizeParams(
chunk_type, data_offset, chunk_data_size);
+ *offset += chunk_size;
+
if (err != OK) {
return err;
}
@@ -1408,7 +1418,6 @@
}
mLastTrack->meta->setInt32(kKeyMaxInputSize, max_size);
}
- *offset += chunk_size;
// NOTE: setting another piece of metadata invalidates any pointers (such as the
// mimetype) previously obtained, so don't cache them.
@@ -1432,6 +1441,8 @@
case FOURCC('s', 't', 't', 's'):
{
+ *offset += chunk_size;
+
status_t err =
mLastTrack->sampleTable->setTimeToSampleParams(
data_offset, chunk_data_size);
@@ -1440,12 +1451,13 @@
return err;
}
- *offset += chunk_size;
break;
}
case FOURCC('c', 't', 't', 's'):
{
+ *offset += chunk_size;
+
status_t err =
mLastTrack->sampleTable->setCompositionTimeToSampleParams(
data_offset, chunk_data_size);
@@ -1454,12 +1466,13 @@
return err;
}
- *offset += chunk_size;
break;
}
case FOURCC('s', 't', 's', 's'):
{
+ *offset += chunk_size;
+
status_t err =
mLastTrack->sampleTable->setSyncSampleParams(
data_offset, chunk_data_size);
@@ -1468,13 +1481,14 @@
return err;
}
- *offset += chunk_size;
break;
}
// @xyz
case FOURCC('\xA9', 'x', 'y', 'z'):
{
+ *offset += chunk_size;
+
// Best case the total data length inside "@xyz" box
// would be 8, for instance "@xyz" + "\x00\x04\x15\xc7" + "0+0/",
// where "\x00\x04" is the text string length with value = 4,
@@ -1503,12 +1517,13 @@
buffer[location_length] = '\0';
mFileMetaData->setCString(kKeyLocation, buffer);
- *offset += chunk_size;
break;
}
case FOURCC('e', 's', 'd', 's'):
{
+ *offset += chunk_size;
+
if (chunk_data_size < 4) {
return ERROR_MALFORMED;
}
@@ -1546,12 +1561,13 @@
}
}
- *offset += chunk_size;
break;
}
case FOURCC('a', 'v', 'c', 'C'):
{
+ *offset += chunk_size;
+
sp<ABuffer> buffer = new ABuffer(chunk_data_size);
if (mDataSource->readAt(
@@ -1562,12 +1578,12 @@
mLastTrack->meta->setData(
kKeyAVCC, kTypeAVCC, buffer->data(), chunk_data_size);
- *offset += chunk_size;
break;
}
case FOURCC('d', '2', '6', '3'):
{
+ *offset += chunk_size;
/*
* d263 contains a fixed 7 bytes part:
* vendor - 4 bytes
@@ -1593,7 +1609,6 @@
mLastTrack->meta->setData(kKeyD263, kTypeD263, buffer, chunk_data_size);
- *offset += chunk_size;
break;
}
@@ -1601,11 +1616,13 @@
{
uint8_t buffer[4];
if (chunk_data_size < (off64_t)sizeof(buffer)) {
+ *offset += chunk_size;
return ERROR_MALFORMED;
}
if (mDataSource->readAt(
data_offset, buffer, 4) < 4) {
+ *offset += chunk_size;
return ERROR_IO;
}
@@ -1639,6 +1656,8 @@
case FOURCC('n', 'a', 'm', 'e'):
case FOURCC('d', 'a', 't', 'a'):
{
+ *offset += chunk_size;
+
if (mPath.size() == 6 && underMetaDataPath(mPath)) {
status_t err = parseITunesMetaData(data_offset, chunk_data_size);
@@ -1647,12 +1666,13 @@
}
}
- *offset += chunk_size;
break;
}
case FOURCC('m', 'v', 'h', 'd'):
{
+ *offset += chunk_size;
+
if (chunk_data_size < 24) {
return ERROR_MALFORMED;
}
@@ -1680,7 +1700,6 @@
mFileMetaData->setCString(kKeyDate, s.string());
- *offset += chunk_size;
break;
}
@@ -1701,6 +1720,8 @@
case FOURCC('h', 'd', 'l', 'r'):
{
+ *offset += chunk_size;
+
uint32_t buffer;
if (mDataSource->readAt(
data_offset + 8, &buffer, 4) < 4) {
@@ -1715,7 +1736,6 @@
mLastTrack->meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_TEXT_3GPP);
}
- *offset += chunk_size;
break;
}
@@ -1740,6 +1760,8 @@
delete[] buffer;
buffer = NULL;
+ // advance read pointer so we don't end up reading this again
+ *offset += chunk_size;
return ERROR_IO;
}
@@ -1754,6 +1776,8 @@
case FOURCC('c', 'o', 'v', 'r'):
{
+ *offset += chunk_size;
+
if (mFileMetaData != NULL) {
ALOGV("chunk_data_size = %lld and data_offset = %lld",
chunk_data_size, data_offset);
@@ -1768,7 +1792,6 @@
buffer->data() + kSkipBytesOfDataBox, chunk_data_size - kSkipBytesOfDataBox);
}
- *offset += chunk_size;
break;
}
@@ -1779,25 +1802,27 @@
case FOURCC('a', 'l', 'b', 'm'):
case FOURCC('y', 'r', 'r', 'c'):
{
+ *offset += chunk_size;
+
status_t err = parse3GPPMetaData(data_offset, chunk_data_size, depth);
if (err != OK) {
return err;
}
- *offset += chunk_size;
break;
}
case FOURCC('I', 'D', '3', '2'):
{
+ *offset += chunk_size;
+
if (chunk_data_size < 6) {
return ERROR_MALFORMED;
}
parseID3v2MetaData(data_offset + 6);
- *offset += chunk_size;
break;
}
diff --git a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
index 15eabfb..52c85e5 100755
--- a/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
+++ b/media/libstagefright/codecs/on2/h264dec/source/h264bsd_intra_prediction.c
@@ -1110,7 +1110,7 @@
/* Variables */
- u32 i, j;
+ i32 i, j;
i32 a, b, c;
i32 tmp;
@@ -1123,20 +1123,20 @@
a = 16 * (above[15] + left[15]);
for (i = 0, b = 0; i < 8; i++)
- b += ((i32)i + 1) * (above[8+i] - above[6-i]);
+ b += (i + 1) * (above[8+i] - above[6-i]);
b = (5 * b + 32) >> 6;
for (i = 0, c = 0; i < 7; i++)
- c += ((i32)i + 1) * (left[8+i] - left[6-i]);
+ c += (i + 1) * (left[8+i] - left[6-i]);
/* p[-1,-1] has to be accessed through above pointer */
- c += ((i32)i + 1) * (left[8+i] - above[-1]);
+ c += (i + 1) * (left[8+i] - above[-1]);
c = (5 * c + 32) >> 6;
for (i = 0; i < 16; i++)
{
for (j = 0; j < 16; j++)
{
- tmp = (a + b * ((i32)j - 7) + c * ((i32)i - 7) + 16) >> 5;
+ tmp = (a + b * (j - 7) + c * (i - 7) + 16) >> 5;
data[i*16+j] = (u8)CLIP1(tmp);
}
}
diff --git a/media/libstagefright/http/MediaHTTP.cpp b/media/libstagefright/http/MediaHTTP.cpp
index 157d967..2d29913 100644
--- a/media/libstagefright/http/MediaHTTP.cpp
+++ b/media/libstagefright/http/MediaHTTP.cpp
@@ -171,6 +171,10 @@
}
String8 MediaHTTP::getUri() {
+ String8 uri;
+ if (OK == mHTTPConnection->getUri(&uri)) {
+ return uri;
+ }
return String8(mLastURI.c_str());
}
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index bc26de1..95779c4 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -1007,8 +1007,7 @@
uint32_t resumeMask = 0;
sp<AnotherPacketSource> sources[kMaxStreams];
- // TRICKY: looping from i as earlier streams are already removed from streamMask
- for (size_t j = i; j < kMaxStreams; ++j) {
+ for (size_t j = 0; j < kMaxStreams; ++j) {
if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) {
sources[j] = mPacketSources.valueFor(indexToType(j));
resumeMask |= indexToType(j);
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 4dc5db0..587a6d5 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -24,6 +24,7 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/Utils.h>
#include <media/mediaplayer.h>
namespace android {
@@ -352,9 +353,27 @@
if (!meta->findString(key, &groupID)) {
*uri = mItems.itemAt(index).mURI;
- // Assume media without any more specific attribute contains
- // audio and video, but no subtitles.
- return !strcmp("audio", key) || !strcmp("video", key);
+ AString codecs;
+ if (!meta->findString("codecs", &codecs)) {
+ // Assume media without any more specific attribute contains
+ // audio and video, but no subtitles.
+ return !strcmp("audio", key) || !strcmp("video", key);
+ } else {
+ // Split the comma separated list of codecs.
+ size_t offset = 0;
+ ssize_t commaPos = -1;
+ codecs.append(',');
+ while ((commaPos = codecs.find(",", offset)) >= 0) {
+ AString codec(codecs, offset, commaPos - offset);
+ // return true only if a codec of type `key` ("audio"/"video")
+ // is found.
+ if (codecIsType(codec, key)) {
+ return true;
+ }
+ offset = commaPos + 1;
+ }
+ return false;
+ }
}
sp<MediaGroup> group = mMediaGroups.valueFor(groupID);
@@ -682,12 +701,22 @@
*meta = new AMessage;
}
(*meta)->setInt32("bandwidth", x);
+ } else if (!strcasecmp("codecs", key.c_str())) {
+ if (!isQuotedString(val)) {
+ ALOGE("Expected quoted string for %s attribute, "
+ "got '%s' instead.",
+ key.c_str(), val.c_str());;
+
+ return ERROR_MALFORMED;
+ }
+
+ key.tolower();
+ const AString &codecs = unquoteString(val);
+ (*meta)->setString(key.c_str(), codecs.c_str());
} else if (!strcasecmp("audio", key.c_str())
|| !strcasecmp("video", key.c_str())
|| !strcasecmp("subtitles", key.c_str())) {
- if (val.size() < 2
- || val.c_str()[0] != '"'
- || val.c_str()[val.size() - 1] != '"') {
+ if (!isQuotedString(val)) {
ALOGE("Expected quoted string for %s attribute, "
"got '%s' instead.",
key.c_str(), val.c_str());
@@ -695,7 +724,7 @@
return ERROR_MALFORMED;
}
- AString groupID(val, 1, val.size() - 2);
+ const AString &groupID = unquoteString(val);
ssize_t groupIndex = mMediaGroups.indexOfKey(groupID);
if (groupIndex < 0) {
@@ -1084,4 +1113,121 @@
return OK;
}
+// static
+bool M3UParser::isQuotedString(const AString &str) {
+ if (str.size() < 2
+ || str.c_str()[0] != '"'
+ || str.c_str()[str.size() - 1] != '"') {
+ return false;
+ }
+ return true;
+}
+
+// static
+AString M3UParser::unquoteString(const AString &str) {
+ if (!isQuotedString(str)) {
+ return str;
+ }
+ return AString(str, 1, str.size() - 2);
+}
+
+// static
+bool M3UParser::codecIsType(const AString &codec, const char *type) {
+ if (codec.size() < 4) {
+ return false;
+ }
+ const char *c = codec.c_str();
+ switch (FOURCC(c[0], c[1], c[2], c[3])) {
+ // List extracted from http://www.mp4ra.org/codecs.html
+ case 'ac-3':
+ case 'alac':
+ case 'dra1':
+ case 'dtsc':
+ case 'dtse':
+ case 'dtsh':
+ case 'dtsl':
+ case 'ec-3':
+ case 'enca':
+ case 'g719':
+ case 'g726':
+ case 'm4ae':
+ case 'mlpa':
+ case 'mp4a':
+ case 'raw ':
+ case 'samr':
+ case 'sawb':
+ case 'sawp':
+ case 'sevc':
+ case 'sqcp':
+ case 'ssmv':
+ case 'twos':
+ case 'agsm':
+ case 'alaw':
+ case 'dvi ':
+ case 'fl32':
+ case 'fl64':
+ case 'ima4':
+ case 'in24':
+ case 'in32':
+ case 'lpcm':
+ case 'Qclp':
+ case 'QDM2':
+ case 'QDMC':
+ case 'ulaw':
+ case 'vdva':
+ return !strcmp("audio", type);
+
+ case 'avc1':
+ case 'avc2':
+ case 'avcp':
+ case 'drac':
+ case 'encv':
+ case 'mjp2':
+ case 'mp4v':
+ case 'mvc1':
+ case 'mvc2':
+ case 'resv':
+ case 's263':
+ case 'svc1':
+ case 'vc-1':
+ case 'CFHD':
+ case 'civd':
+ case 'DV10':
+ case 'dvh5':
+ case 'dvh6':
+ case 'dvhp':
+ case 'DVOO':
+ case 'DVOR':
+ case 'DVTV':
+ case 'DVVT':
+ case 'flic':
+ case 'gif ':
+ case 'h261':
+ case 'h263':
+ case 'HD10':
+ case 'jpeg':
+ case 'M105':
+ case 'mjpa':
+ case 'mjpb':
+ case 'png ':
+ case 'PNTG':
+ case 'rle ':
+ case 'rpza':
+ case 'Shr0':
+ case 'Shr1':
+ case 'Shr2':
+ case 'Shr3':
+ case 'Shr4':
+ case 'SVQ1':
+ case 'SVQ3':
+ case 'tga ':
+ case 'tiff':
+ case 'WRLE':
+ return !strcmp("video", type);
+
+ default:
+ return false;
+ }
+}
+
} // namespace android
diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h
index 2051e41..ccd6556 100644
--- a/media/libstagefright/httplive/M3UParser.h
+++ b/media/libstagefright/httplive/M3UParser.h
@@ -96,6 +96,10 @@
static status_t ParseInt32(const char *s, int32_t *x);
static status_t ParseDouble(const char *s, double *x);
+ static bool isQuotedString(const AString &str);
+ static AString unquoteString(const AString &str);
+ static bool codecIsType(const AString &codec, const char *type);
+
DISALLOW_EVIL_CONSTRUCTORS(M3UParser);
};
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 67d83b1..f00b82a 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -193,6 +193,7 @@
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t->downmixerBufferProvider = NULL;
+ t->mSinkFormat = AUDIO_FORMAT_PCM_16_BIT;
status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
if (status == OK) {
@@ -440,6 +441,13 @@
// for a specific track? or per mixer?
/* case DOWNMIX_TYPE:
break */
+ case SINK_FORMAT: {
+ audio_format_t format = static_cast<audio_format_t>(valueInt);
+ if (track.mSinkFormat != format) {
+ track.mSinkFormat = format;
+ ALOGV("setParameter(TRACK, SINK_FORMAT, %#x)", format);
+ }
+ } break;
default:
LOG_FATAL("bad param");
}
@@ -1043,7 +1051,7 @@
void AudioMixer::process__nop(state_t* state, int64_t pts)
{
uint32_t e0 = state->enabledTracks;
- size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
+ size_t sampleCount = state->frameCount * MAX_NUM_CHANNELS;
while (e0) {
// process by group of tracks with same output buffer to
// avoid multiple memset() on same buffer
@@ -1062,7 +1070,8 @@
}
e0 &= ~(e1);
- memset(t1.mainBuffer, 0, bufSize);
+ memset(t1.mainBuffer, 0, sampleCount
+ * audio_bytes_per_sample(t1.mSinkFormat));
}
while (e1) {
@@ -1170,8 +1179,18 @@
}
}
}
- ditherAndClamp(out, outTemp, BLOCKSIZE);
- out += BLOCKSIZE;
+ switch (t1.mSinkFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_float_from_q19_12(reinterpret_cast<float *>(out), outTemp, BLOCKSIZE * 2);
+ out += BLOCKSIZE * 2; // output is 2 floats/frame.
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ ditherAndClamp(out, outTemp, BLOCKSIZE);
+ out += BLOCKSIZE; // output is 1 int32_t (2 int16_t samples)/frame
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
+ }
numFrames += BLOCKSIZE;
} while (numFrames < state->frameCount);
}
@@ -1253,7 +1272,16 @@
}
}
}
- ditherAndClamp(out, outTemp, numFrames);
+ switch (t1.mSinkFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_float_from_q19_12(reinterpret_cast<float*>(out), outTemp, numFrames*2);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ ditherAndClamp(out, outTemp, numFrames);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
+ }
}
}
@@ -1294,27 +1322,45 @@
}
size_t outFrames = b.frameCount;
- if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
- // volume is boosted, so we might need to clamp even though
- // we process only one track.
+ switch (t.mSinkFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT: {
+ float *fout = reinterpret_cast<float*>(out);
+ static float scale = 1. / (32768. * 4096.); // exact when inverted
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
- int32_t l = mulRL(1, rl, vrl) >> 12;
- int32_t r = mulRL(0, rl, vrl) >> 12;
- // clamping...
- l = clamp16(l);
- r = clamp16(r);
- *out++ = (r<<16) | (l & 0xFFFF);
+ int32_t l = mulRL(1, rl, vrl);
+ int32_t r = mulRL(0, rl, vrl);
+ *fout++ = static_cast<float>(l) * scale;
+ *fout++ = static_cast<float>(r) * scale;
} while (--outFrames);
- } else {
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl) >> 12;
- int32_t r = mulRL(0, rl, vrl) >> 12;
- *out++ = (r<<16) | (l & 0xFFFF);
- } while (--outFrames);
+ } break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+ // volume is boosted, so we might need to clamp even though
+ // we process only one track.
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl) >> 12;
+ int32_t r = mulRL(0, rl, vrl) >> 12;
+ // clamping...
+ l = clamp16(l);
+ r = clamp16(r);
+ *out++ = (r<<16) | (l & 0xFFFF);
+ } while (--outFrames);
+ } else {
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl) >> 12;
+ int32_t r = mulRL(0, rl, vrl) >> 12;
+ *out++ = (r<<16) | (l & 0xFFFF);
+ } while (--outFrames);
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad sink format: %d", t.mSinkFormat);
}
numFrames -= b.frameCount;
t.bufferProvider->releaseBuffer(&b);
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index d286986..3355db4 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -77,6 +77,7 @@
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
DOWNMIX_TYPE = 0X4004,
+ SINK_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
// for target RESAMPLE
SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
// parameter 'value' is the new sample rate in Hz.
@@ -193,7 +194,9 @@
int32_t sessionId;
- int32_t padding[2];
+ audio_format_t mSinkFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+
+ int32_t padding[1];
// 16-byte boundary
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 939b128..7e4ca0c 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -165,6 +165,10 @@
mCoefBuffer(NULL)
{
mVolumeSimd[0] = mVolumeSimd[1] = 0;
+ // The AudioResampler base class assumes we are always ready for 1:1 resampling.
+ // We reset mInSampleRate to 0, so setSampleRate() will calculate filters for
+ // setSampleRate() for 1:1. (May be removed if precalculated filters are used.)
+ mInSampleRate = 0;
mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better
}
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 74d5702..3ab3ba9 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -27,6 +27,7 @@
#include <time.h>
#include <math.h>
#include <audio_utils/sndfile.h>
+#include <utils/Vector.h>
using namespace android;
@@ -34,7 +35,7 @@
static int usage(const char* name) {
fprintf(stderr,"Usage: %s [-p] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
- " [-i input-sample-rate] [-o output-sample-rate] [-O #] [<input-file>]"
+ " [-i input-sample-rate] [-o output-sample-rate] [-O csv] [-P csv] [<input-file>]"
" <output-file>\n", name);
fprintf(stderr," -p enable profiling\n");
fprintf(stderr," -h create wav file\n");
@@ -51,10 +52,50 @@
fprintf(stderr," dhq : dynamic high quality\n");
fprintf(stderr," -i input file sample rate (ignored if input file is specified)\n");
fprintf(stderr," -o output file sample rate\n");
- fprintf(stderr," -O # frames output per call to resample()\n");
+ fprintf(stderr," -O # frames output per call to resample() in CSV format\n");
+ fprintf(stderr," -P # frames provided per call to resample() in CSV format\n");
return -1;
}
+// Convert a list of integers in CSV format to a Vector of those values.
+// Returns the number of elements in the list, or -1 on error.
+int parseCSV(const char *string, Vector<int>& values)
+{
+ // pass 1: count the number of values and do syntax check
+ size_t numValues = 0;
+ bool hadDigit = false;
+ for (const char *p = string; ; ) {
+ switch (*p++) {
+ case '0': case '1': case '2': case '3': case '4':
+ case '5': case '6': case '7': case '8': case '9':
+ hadDigit = true;
+ break;
+ case '\0':
+ if (hadDigit) {
+ // pass 2: allocate and initialize vector of values
+ values.resize(++numValues);
+ values.editItemAt(0) = atoi(p = optarg);
+ for (size_t i = 1; i < numValues; ) {
+ if (*p++ == ',') {
+ values.editItemAt(i++) = atoi(p);
+ }
+ }
+ return numValues;
+ }
+ // fall through
+ case ',':
+ if (hadDigit) {
+ hadDigit = false;
+ numValues++;
+ break;
+ }
+ // fall through
+ default:
+ return -1;
+ }
+ }
+}
+
int main(int argc, char* argv[]) {
const char* const progname = argv[0];
@@ -65,10 +106,11 @@
int input_freq = 0;
int output_freq = 0;
AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
- size_t framesPerCall = 0;
+ Vector<int> Ovalues;
+ Vector<int> Pvalues;
int ch;
- while ((ch = getopt(argc, argv, "pfhvsq:i:o:O:")) != -1) {
+ while ((ch = getopt(argc, argv, "pfhvsq:i:o:O:P:")) != -1) {
switch (ch) {
case 'p':
profileResample = true;
@@ -114,7 +156,16 @@
output_freq = atoi(optarg);
break;
case 'O':
- framesPerCall = atoi(optarg);
+ if (parseCSV(optarg, Ovalues) < 0) {
+ fprintf(stderr, "incorrect syntax for -O option\n");
+ return -1;
+ }
+ break;
+ case 'P':
+ if (parseCSV(optarg, Pvalues) < 0) {
+ fprintf(stderr, "incorrect syntax for -P option\n");
+ return -1;
+ }
break;
case '?':
default:
@@ -182,12 +233,14 @@
const int mChannels;
size_t mNextFrame; // index of next frame to provide
size_t mUnrel; // number of frames not yet released
+ const Vector<int> mPvalues; // number of frames provided per call
+ size_t mNextPidx; // index of next entry in mPvalues to use
public:
- Provider(const void* addr, size_t size, int channels)
+ Provider(const void* addr, size_t size, int channels, const Vector<int>& Pvalues)
: mAddr((int16_t*) addr),
mNumFrames(size / (channels*sizeof(int16_t))),
mChannels(channels),
- mNextFrame(0), mUnrel(0) {
+ mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) {
}
virtual status_t getNextBuffer(Buffer* buffer,
int64_t pts = kInvalidPTS) {
@@ -196,6 +249,16 @@
if (requestedFrames > mNumFrames - mNextFrame) {
buffer->frameCount = mNumFrames - mNextFrame;
}
+ if (!mPvalues.isEmpty()) {
+ size_t provided = mPvalues[mNextPidx++];
+ printf("mPvalue[%d]=%u not %u\n", mNextPidx-1, provided, buffer->frameCount);
+ if (provided < buffer->frameCount) {
+ buffer->frameCount = provided;
+ }
+ if (mNextPidx >= mPvalues.size()) {
+ mNextPidx = 0;
+ }
+ }
if (gVerbose) {
printf("getNextBuffer() requested %u frames out of %u frames available,"
" and returned %u frames\n",
@@ -230,7 +293,7 @@
void reset() {
mNextFrame = 0;
}
- } provider(input_vaddr, input_size, channels);
+ } provider(input_vaddr, input_size, channels, Pvalues);
size_t input_frames = input_size / (channels * sizeof(int16_t));
if (gVerbose) {
@@ -348,11 +411,17 @@
if (gVerbose) {
printf("resample() %u output frames\n", out_frames);
}
- if (framesPerCall == 0 || framesPerCall > out_frames) {
- framesPerCall = out_frames;
+ if (Ovalues.isEmpty()) {
+ Ovalues.push(out_frames);
}
- for (size_t i = 0; i < out_frames; ) {
- size_t thisFrames = framesPerCall <= out_frames - i ? framesPerCall : out_frames - i;
+ for (size_t i = 0, j = 0; i < out_frames; ) {
+ size_t thisFrames = Ovalues[j++];
+ if (j >= Ovalues.size()) {
+ j = 0;
+ }
+ if (thisFrames == 0 || thisFrames > out_frames - i) {
+ thisFrames = out_frames - i;
+ }
resampler->resample((int*) output_vaddr + 2*i, thisFrames, &provider);
i += thisFrames;
}
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 0a88a75..80b7cd4 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -407,12 +407,6 @@
l.mParameters.state = Parameters::DISCONNECTED;
}
- mStreamingProcessor->deletePreviewStream();
- mStreamingProcessor->deleteRecordingStream();
- mJpegProcessor->deleteStream();
- mCallbackProcessor->deleteStream();
- mZslProcessor->deleteStream();
-
mStreamingProcessor->requestExit();
mFrameProcessor->requestExit();
mCaptureSequencer->requestExit();
@@ -429,6 +423,14 @@
mZslProcessorThread->join();
mCallbackProcessor->join();
+ ALOGV("Camera %d: Deleting streams", mCameraId);
+
+ mStreamingProcessor->deletePreviewStream();
+ mStreamingProcessor->deleteRecordingStream();
+ mJpegProcessor->deleteStream();
+ mCallbackProcessor->deleteStream();
+ mZslProcessor->deleteStream();
+
ALOGV("Camera %d: Disconnecting device", mCameraId);
mDevice->disconnect();