Merge "Enable Scudo for mediaswcodec"
diff --git a/apex/Android.bp b/apex/Android.bp
index eee26ae..991696c 100644
--- a/apex/Android.bp
+++ b/apex/Android.bp
@@ -21,7 +21,10 @@
"libamrextractor",
"libflacextractor",
"libmidiextractor",
+ "libmkvextractor",
"libmp3extractor",
+ "libmp4extractor",
+ "liboggextractor",
"libwavextractor",
],
key: "com.android.media.key",
diff --git a/camera/ndk/impl/ACameraDevice.cpp b/camera/ndk/impl/ACameraDevice.cpp
index 00da54e..657d41f 100644
--- a/camera/ndk/impl/ACameraDevice.cpp
+++ b/camera/ndk/impl/ACameraDevice.cpp
@@ -633,7 +633,7 @@
}
std::set<std::pair<ANativeWindow*, OutputConfiguration>> outputSet;
- for (auto outConfig : outputs->mOutputs) {
+ for (const auto& outConfig : outputs->mOutputs) {
ANativeWindow* anw = outConfig.mWindow;
sp<IGraphicBufferProducer> iGBP(nullptr);
ret = getIGBPfromAnw(anw, iGBP);
@@ -706,7 +706,7 @@
}
// add new streams
- for (auto outputPair : addSet) {
+ for (const auto& outputPair : addSet) {
int streamId;
remoteRet = mRemote->createStream(outputPair.second, &streamId);
if (!remoteRet.isOk()) {
@@ -839,7 +839,7 @@
const auto& gbps = outputPairIt->second.second.getGraphicBufferProducers();
for (const auto& outGbp : gbps) {
- for (auto surface : request->mSurfaceList) {
+ for (const auto& surface : request->mSurfaceList) {
if (surface->getIGraphicBufferProducer() == outGbp) {
ANativeWindow* anw = static_cast<ANativeWindow*>(surface.get());
ALOGV("Camera %s Lost output buffer for ANW %p frame %" PRId64,
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index c7619af..7a10302 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -191,7 +191,6 @@
LOCAL_MODULE:= mediafilter
LOCAL_SANITIZE := cfi
-LOCAL_SANITIZE_DIAG := cfi
include $(BUILD_EXECUTABLE)
diff --git a/include/media/MediaExtractorPluginHelper.h b/include/media/MediaExtractorPluginHelper.h
index 292ec93..705aa81 100644
--- a/include/media/MediaExtractorPluginHelper.h
+++ b/include/media/MediaExtractorPluginHelper.h
@@ -183,32 +183,32 @@
mBuffer = buf;
}
- ~MediaBufferHelperV3() {}
+ virtual ~MediaBufferHelperV3() {}
- void release() {
+ virtual void release() {
mBuffer->release(mBuffer->handle);
}
- void* data() {
+ virtual void* data() {
return mBuffer->data(mBuffer->handle);
}
- size_t size() {
+ virtual size_t size() {
return mBuffer->size(mBuffer->handle);
}
- size_t range_offset() {
+ virtual size_t range_offset() {
return mBuffer->range_offset(mBuffer->handle);
}
- size_t range_length() {
+ virtual size_t range_length() {
return mBuffer->range_length(mBuffer->handle);
}
- void set_range(size_t offset, size_t length) {
+ virtual void set_range(size_t offset, size_t length) {
mBuffer->set_range(mBuffer->handle, offset, length);
}
- AMediaFormat *meta_data() {
+ virtual AMediaFormat *meta_data() {
return mBuffer->meta_data(mBuffer->handle);
}
};
diff --git a/media/extractors/ogg/Android.bp b/media/extractors/ogg/Android.bp
index b28877d..604ec59 100644
--- a/media/extractors/ogg/Android.bp
+++ b/media/extractors/ogg/Android.bp
@@ -13,7 +13,6 @@
shared_libs: [
"liblog",
- "libmediaextractor",
"libmediandk",
],
diff --git a/media/extractors/ogg/OggExtractor.cpp b/media/extractors/ogg/OggExtractor.cpp
index cc2c792..29fe2b1 100644
--- a/media/extractors/ogg/OggExtractor.cpp
+++ b/media/extractors/ogg/OggExtractor.cpp
@@ -28,11 +28,8 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/base64.h>
#include <media/stagefright/foundation/ByteUtils.h>
-#include <media/stagefright/MediaBufferBase.h>
-#include <media/stagefright/MediaBufferGroup.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MetaDataBase.h>
#include <media/stagefright/MetaDataUtils.h>
#include <system/audio.h>
#include <utils/String8.h>
@@ -48,7 +45,7 @@
namespace android {
-struct OggSource : public MediaTrackHelperV2 {
+struct OggSource : public MediaTrackHelperV3 {
explicit OggSource(OggExtractor *extractor);
virtual media_status_t getFormat(AMediaFormat *);
@@ -57,7 +54,7 @@
virtual media_status_t stop();
virtual media_status_t read(
- MediaBufferBase **buffer, const ReadOptions *options = NULL);
+ MediaBufferHelperV3 **buffer, const ReadOptions *options = NULL);
protected:
virtual ~OggSource();
@@ -85,7 +82,7 @@
status_t seekToTime(int64_t timeUs);
status_t seekToOffset(off64_t offset);
- virtual media_status_t readNextPacket(MediaBufferBase **buffer) = 0;
+ virtual media_status_t readNextPacket(MediaBufferHelperV3 **buffer) = 0;
status_t init();
@@ -93,6 +90,9 @@
return AMediaFormat_copy(meta, mFileMeta);
}
+ void setBufferGroup(MediaBufferGroupHelperV3 *group) {
+ mBufferGroup = group;
+ }
protected:
struct Page {
uint64_t mGranulePosition;
@@ -110,6 +110,7 @@
int64_t mTimeUs;
};
+ MediaBufferGroupHelperV3 *mBufferGroup;
DataSourceHelper *mSource;
off64_t mOffset;
Page mCurrentPage;
@@ -148,7 +149,7 @@
// 1 - bitstream identification header
// 3 - comment header
// 5 - codec setup header (Vorbis only)
- virtual media_status_t verifyHeader(MediaBufferBase *buffer, uint8_t type) = 0;
+ virtual media_status_t verifyHeader(MediaBufferHelperV3 *buffer, uint8_t type) = 0;
// Read the next ogg packet from the underlying data source; optionally
// calculate the timestamp for the output packet whilst pretending
@@ -156,9 +157,9 @@
//
// *buffer is NULL'ed out immediately upon entry, and if successful a new buffer is allocated;
// clients are responsible for releasing the original buffer.
- media_status_t _readNextPacket(MediaBufferBase **buffer, bool calcVorbisTimestamp);
+ media_status_t _readNextPacket(MediaBufferHelperV3 **buffer, bool calcVorbisTimestamp);
- int32_t getPacketBlockSize(MediaBufferBase *buffer);
+ int32_t getPacketBlockSize(MediaBufferHelperV3 *buffer);
void parseFileMetaData();
@@ -182,7 +183,7 @@
virtual uint64_t approxBitrate() const;
- virtual media_status_t readNextPacket(MediaBufferBase **buffer) {
+ virtual media_status_t readNextPacket(MediaBufferHelperV3 **buffer) {
return _readNextPacket(buffer, /* calcVorbisTimestamp = */ true);
}
@@ -194,7 +195,7 @@
return granulePos * 1000000ll / mVi.rate;
}
- virtual media_status_t verifyHeader(MediaBufferBase *buffer, uint8_t type);
+ virtual media_status_t verifyHeader(MediaBufferHelperV3 *buffer, uint8_t type);
};
struct MyOpusExtractor : public MyOggExtractor {
@@ -212,16 +213,16 @@
return 0;
}
- virtual media_status_t readNextPacket(MediaBufferBase **buffer);
+ virtual media_status_t readNextPacket(MediaBufferHelperV3 **buffer);
protected:
virtual int64_t getTimeUsOfGranule(uint64_t granulePos) const;
- virtual media_status_t verifyHeader(MediaBufferBase *buffer, uint8_t type);
+ virtual media_status_t verifyHeader(MediaBufferHelperV3 *buffer, uint8_t type);
private:
- media_status_t verifyOpusHeader(MediaBufferBase *buffer);
- media_status_t verifyOpusComments(MediaBufferBase *buffer);
- uint32_t getNumSamplesInPacket(MediaBufferBase *buffer) const;
+ media_status_t verifyOpusHeader(MediaBufferHelperV3 *buffer);
+ media_status_t verifyOpusComments(MediaBufferHelperV3 *buffer);
+ uint32_t getNumSamplesInPacket(MediaBufferHelperV3 *buffer) const;
uint8_t mChannelCount;
uint16_t mCodecDelay;
@@ -249,7 +250,9 @@
if (mStarted) {
return AMEDIA_ERROR_INVALID_OPERATION;
}
-
+ // initialize buffer group with a single small buffer, but a generous upper limit
+ mBufferGroup->init(1 /* number of buffers */, 128 /* size */, 64 /* max number of buffers */);
+ mExtractor->mImpl->setBufferGroup(mBufferGroup);
mStarted = true;
return AMEDIA_OK;
@@ -262,7 +265,7 @@
}
media_status_t OggSource::read(
- MediaBufferBase **out, const ReadOptions *options) {
+ MediaBufferHelperV3 **out, const ReadOptions *options) {
*out = NULL;
int64_t seekTimeUs;
@@ -274,26 +277,27 @@
}
}
- MediaBufferBase *packet;
+ MediaBufferHelperV3 *packet;
media_status_t err = mExtractor->mImpl->readNextPacket(&packet);
if (err != AMEDIA_OK) {
return err;
}
+ AMediaFormat *meta = packet->meta_data();
#if 0
int64_t timeUs;
- if (packet->meta_data().findInt64(kKeyTime, &timeUs)) {
+ if (AMediaFormat_findInt64(meta, AMEDIAFORMAT_KEY_TIME_US, timeStampUs)) {
ALOGI("found time = %lld us", timeUs);
} else {
ALOGI("NO time");
}
#endif
- packet->meta_data().setInt32(kKeyIsSyncFrame, 1);
+ AMediaFormat_setInt32(meta, AMEDIAFORMAT_KEY_IS_SYNC_FRAME, 1);
*out = packet;
-
+ ALOGV("returning buffer %p", packet);
return AMEDIA_OK;
}
@@ -304,7 +308,8 @@
const char *mimeType,
size_t numHeaders,
int64_t seekPreRollUs)
- : mSource(source),
+ : mBufferGroup(NULL),
+ mSource(source),
mOffset(0),
mCurGranulePosition(0),
mPrevGranulePosition(0),
@@ -573,13 +578,13 @@
return sizeof(header) + page->mNumSegments + totalSize;
}
-media_status_t MyOpusExtractor::readNextPacket(MediaBufferBase **out) {
+media_status_t MyOpusExtractor::readNextPacket(MediaBufferHelperV3 **out) {
if (mOffset <= mFirstDataOffset && mStartGranulePosition < 0) {
// The first sample might not start at time 0; find out where by subtracting
// the number of samples on the first page from the granule position
// (position of last complete sample) of the first page. This happens
// the first time before we attempt to read a packet from the first page.
- MediaBufferBase *mBuf;
+ MediaBufferHelperV3 *mBuf;
uint32_t numSamples = 0;
uint64_t curGranulePosition = 0;
while (true) {
@@ -617,24 +622,25 @@
int32_t currentPageSamples;
// Calculate timestamps by accumulating durations starting from the first sample of a page;
// We assume that we only seek to page boundaries.
- if ((*out)->meta_data().findInt32(kKeyValidSamples, ¤tPageSamples)) {
+ AMediaFormat *meta = (*out)->meta_data();
+ if (AMediaFormat_getInt32(meta, AMEDIAFORMAT_KEY_VALID_SAMPLES, ¤tPageSamples)) {
// first packet in page
if (mOffset == mFirstDataOffset) {
currentPageSamples -= mStartGranulePosition;
- (*out)->meta_data().setInt32(kKeyValidSamples, currentPageSamples);
+ AMediaFormat_setInt32(meta, AMEDIAFORMAT_KEY_VALID_SAMPLES, currentPageSamples);
}
mCurGranulePosition = mCurrentPage.mGranulePosition - currentPageSamples;
}
int64_t timeUs = getTimeUsOfGranule(mCurGranulePosition);
- (*out)->meta_data().setInt64(kKeyTime, timeUs);
+ AMediaFormat_setInt64(meta, AMEDIAFORMAT_KEY_TIME_US, timeUs);
uint32_t frames = getNumSamplesInPacket(*out);
mCurGranulePosition += frames;
return AMEDIA_OK;
}
-uint32_t MyOpusExtractor::getNumSamplesInPacket(MediaBufferBase *buffer) const {
+uint32_t MyOpusExtractor::getNumSamplesInPacket(MediaBufferHelperV3 *buffer) const {
if (buffer == NULL || buffer->range_length() < 1) {
return 0;
}
@@ -680,10 +686,66 @@
return numSamples;
}
-media_status_t MyOggExtractor::_readNextPacket(MediaBufferBase **out, bool calcVorbisTimestamp) {
+/*
+ * basic mediabuffer implementation used during initial parsing of the
+ * header packets, which happens before we have a buffer group
+ */
+class StandAloneMediaBuffer : public MediaBufferHelperV3 {
+private:
+ void *mData;
+ size_t mSize;
+ size_t mOffset;
+ size_t mLength;
+ AMediaFormat *mFormat;
+public:
+ StandAloneMediaBuffer(size_t size) : MediaBufferHelperV3(NULL) {
+ mSize = size;
+ mData = malloc(mSize);
+ mOffset = 0;
+ mLength = mSize;
+ mFormat = AMediaFormat_new();
+ ALOGV("created standalone media buffer %p of size %zu", this, mSize);
+ }
+
+ ~StandAloneMediaBuffer() override {
+ free(mData);
+ AMediaFormat_delete(mFormat);
+ ALOGV("deleted standalone media buffer %p of size %zu", this, mSize);
+ }
+
+ void release() override {
+ delete this;
+ }
+
+ void* data() override {
+ return mData;
+ }
+
+ size_t size() override {
+ return mSize;
+ }
+
+ size_t range_offset() override {
+ return mOffset;
+ }
+
+ size_t range_length() override {
+ return mLength;
+ }
+
+ void set_range(size_t offset, size_t length) override {
+ mOffset = offset;
+ mLength = length;
+ }
+ AMediaFormat *meta_data() override {
+ return mFormat;
+ }
+};
+
+media_status_t MyOggExtractor::_readNextPacket(MediaBufferHelperV3 **out, bool calcVorbisTimestamp) {
*out = NULL;
- MediaBufferBase *buffer = NULL;
+ MediaBufferHelperV3 *buffer = NULL;
int64_t timeUs = -1;
for (;;) {
@@ -719,7 +781,13 @@
ALOGE("b/36592202");
return AMEDIA_ERROR_MALFORMED;
}
- MediaBufferBase *tmp = MediaBufferBase::Create(fullSize);
+ MediaBufferHelperV3 *tmp;
+ if (mBufferGroup) {
+ mBufferGroup->acquire_buffer(&tmp, false, fullSize);
+ ALOGV("acquired buffer %p from group", tmp);
+ } else {
+ tmp = new StandAloneMediaBuffer(fullSize);
+ }
if (tmp == NULL) {
if (buffer != NULL) {
buffer->release();
@@ -727,6 +795,7 @@
ALOGE("b/36592202");
return AMEDIA_ERROR_MALFORMED;
}
+ AMediaFormat_clear(tmp->meta_data());
if (buffer != NULL) {
memcpy(tmp->data(), buffer->data(), buffer->range_length());
tmp->set_range(0, buffer->range_length());
@@ -756,8 +825,9 @@
// We've just read the entire packet.
if (mFirstPacketInPage) {
- buffer->meta_data().setInt32(
- kKeyValidSamples, mCurrentPageSamples);
+ AMediaFormat *meta = buffer->meta_data();
+ AMediaFormat_setInt32(
+ meta, AMEDIAFORMAT_KEY_VALID_SAMPLES, mCurrentPageSamples);
mFirstPacketInPage = false;
}
@@ -778,7 +848,8 @@
mCurrentPage.mPrevPacketPos += actualBlockSize / 2;
mCurrentPage.mPrevPacketSize = curBlockSize;
}
- buffer->meta_data().setInt64(kKeyTime, timeUs);
+ AMediaFormat *meta = buffer->meta_data();
+ AMediaFormat_setInt64(meta, AMEDIAFORMAT_KEY_TIME_US, timeUs);
}
*out = buffer;
@@ -824,11 +895,13 @@
// is already complete.
if (timeUs >= 0) {
- buffer->meta_data().setInt64(kKeyTime, timeUs);
+ AMediaFormat *meta = buffer->meta_data();
+ AMediaFormat_setInt64(meta, AMEDIAFORMAT_KEY_TIME_US, timeUs);
}
- buffer->meta_data().setInt32(
- kKeyValidSamples, mCurrentPageSamples);
+ AMediaFormat *meta = buffer->meta_data();
+ AMediaFormat_setInt32(
+ meta, AMEDIAFORMAT_KEY_VALID_SAMPLES, mCurrentPageSamples);
mFirstPacketInPage = false;
*out = buffer;
@@ -843,7 +916,7 @@
AMediaFormat_setString(mMeta, AMEDIAFORMAT_KEY_MIME, mMimeType);
media_status_t err;
- MediaBufferBase *packet;
+ MediaBufferHelperV3 *packet;
for (size_t i = 0; i < mNumHeaders; ++i) {
// ignore timestamp for configuration packets
if ((err = _readNextPacket(&packet, /* calcVorbisTimestamp = */ false)) != AMEDIA_OK) {
@@ -920,7 +993,7 @@
}
}
-int32_t MyOggExtractor::getPacketBlockSize(MediaBufferBase *buffer) {
+int32_t MyOggExtractor::getPacketBlockSize(MediaBufferHelperV3 *buffer) {
const uint8_t *data =
(const uint8_t *)buffer->data() + buffer->range_offset();
@@ -960,7 +1033,7 @@
return pcmSamplePosition * 1000000ll / kOpusSampleRate;
}
-media_status_t MyOpusExtractor::verifyHeader(MediaBufferBase *buffer, uint8_t type) {
+media_status_t MyOpusExtractor::verifyHeader(MediaBufferHelperV3 *buffer, uint8_t type) {
switch (type) {
// there are actually no header types defined in the Opus spec; we choose 1 and 3 to mean
// header and comments such that we can share code with MyVorbisExtractor.
@@ -973,7 +1046,7 @@
}
}
-media_status_t MyOpusExtractor::verifyOpusHeader(MediaBufferBase *buffer) {
+media_status_t MyOpusExtractor::verifyOpusHeader(MediaBufferHelperV3 *buffer) {
const size_t kOpusHeaderSize = 19;
const uint8_t *data =
(const uint8_t *)buffer->data() + buffer->range_offset();
@@ -1001,7 +1074,7 @@
return AMEDIA_OK;
}
-media_status_t MyOpusExtractor::verifyOpusComments(MediaBufferBase *buffer) {
+media_status_t MyOpusExtractor::verifyOpusComments(MediaBufferHelperV3 *buffer) {
// add artificial framing bit so we can reuse _vorbis_unpack_comment
int32_t commentSize = buffer->range_length() + 1;
auto tmp = heapbuffer<uint8_t>(commentSize);
@@ -1094,7 +1167,7 @@
}
media_status_t MyVorbisExtractor::verifyHeader(
- MediaBufferBase *buffer, uint8_t type) {
+ MediaBufferHelperV3 *buffer, uint8_t type) {
const uint8_t *data =
(const uint8_t *)buffer->data() + buffer->range_offset();
@@ -1262,7 +1335,7 @@
return mInitCheck != OK ? 0 : 1;
}
-MediaTrackHelperV2 *OggExtractor::getTrack(size_t index) {
+MediaTrackHelperV3 *OggExtractor::getTrack(size_t index) {
if (index >= 1) {
return NULL;
}
@@ -1284,13 +1357,13 @@
return mImpl->getFileMetaData(meta);
}
-static CMediaExtractorV2* CreateExtractor(
+static CMediaExtractorV3* CreateExtractor(
CDataSource *source,
void *) {
- return wrapV2(new OggExtractor(new DataSourceHelper(source)));
+ return wrapV3(new OggExtractor(new DataSourceHelper(source)));
}
-static CreatorFuncV2 Sniff(
+static CreatorFuncV3 Sniff(
CDataSource *source,
float *confidence,
void **,
@@ -1311,11 +1384,11 @@
__attribute__ ((visibility ("default")))
ExtractorDef GETEXTRACTORDEF() {
return {
- EXTRACTORDEF_VERSION_CURRENT,
+ EXTRACTORDEF_VERSION_CURRENT + 1,
UUID("8cc5cd06-f772-495e-8a62-cba9649374e9"),
1, // version
"Ogg Extractor",
- { .v2 = Sniff }
+ { .v3 = Sniff }
};
}
diff --git a/media/extractors/ogg/OggExtractor.h b/media/extractors/ogg/OggExtractor.h
index cd674f3..97506ad 100644
--- a/media/extractors/ogg/OggExtractor.h
+++ b/media/extractors/ogg/OggExtractor.h
@@ -31,11 +31,11 @@
struct MyOggExtractor;
struct OggSource;
-struct OggExtractor : public MediaExtractorPluginHelperV2 {
+struct OggExtractor : public MediaExtractorPluginHelperV3 {
explicit OggExtractor(DataSourceHelper *source);
virtual size_t countTracks();
- virtual MediaTrackHelperV2 *getTrack(size_t index);
+ virtual MediaTrackHelperV3 *getTrack(size_t index);
virtual media_status_t getTrackMetaData(AMediaFormat *meta, size_t index, uint32_t flags);
virtual media_status_t getMetaData(AMediaFormat *meta);
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 4a0e6da..58ef7b1 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -258,7 +258,7 @@
callbackResult = maybeCallDataCallback(mCallbackBuffer, mCallbackFrames);
if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
- ALOGD("callback returned AAUDIO_CALLBACK_RESULT_STOP");
+ ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
break;
}
}
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 2ae37a5..9af47b2 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -293,7 +293,7 @@
break;
}
} else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
- ALOGV("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
+ ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
break;
}
}
diff --git a/media/libaaudio/tests/Android.bp b/media/libaaudio/tests/Android.bp
index 319467e..cb243a0 100644
--- a/media/libaaudio/tests/Android.bp
+++ b/media/libaaudio/tests/Android.bp
@@ -184,3 +184,15 @@
"libutils",
],
}
+
+cc_test {
+ name: "test_return_stop",
+ defaults: ["libaaudio_tests_defaults"],
+ srcs: ["test_return_stop.cpp"],
+ shared_libs: [
+ "libaaudio",
+ "libbinder",
+ "libcutils",
+ "libutils",
+ ],
+}
diff --git a/media/libaaudio/tests/test_return_stop.cpp b/media/libaaudio/tests/test_return_stop.cpp
new file mode 100644
index 0000000..f34c3c8
--- /dev/null
+++ b/media/libaaudio/tests/test_return_stop.cpp
@@ -0,0 +1,284 @@
+/*
+ * Copyright (C) 2018 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.
+ */
+
+/**
+ * Return stop from the callback.
+ * Expect the callback to cease.
+ * Check the logcat for bad behavior.
+ */
+
+#include <stdio.h>
+#include <thread>
+#include <unistd.h>
+
+#include <aaudio/AAudio.h>
+
+#define DEFAULT_TIMEOUT_NANOS ((int64_t)1000000000)
+#define STOP_AT_MSEC 1000
+#define LOOP_DURATION_MSEC 4000
+#define SLEEP_DURATION_MSEC 200
+
+static void s_myErrorCallbackProc(
+ AAudioStream *stream,
+ void *userData,
+ aaudio_result_t error);
+
+struct AudioEngine {
+ AAudioStreamBuilder *builder = nullptr;
+ AAudioStream *stream = nullptr;
+ std::thread *thread = nullptr;
+ int32_t stopAtFrame = 0;
+ bool stopped = false;
+ // These counters are read and written by the callback and the main thread.
+ std::atomic<int32_t> framesRead{};
+ std::atomic<int32_t> startingFramesRead{};
+ std::atomic<int32_t> framesCalled{};
+ std::atomic<int32_t> callbackCount{};
+ std::atomic<int32_t> callbackCountAfterStop{};
+
+ void reset() {
+ framesRead.store(0);
+ startingFramesRead.store(0);
+ framesCalled.store(0);
+ callbackCount.store(0);
+ callbackCountAfterStop.store(0);
+ stopped = false;
+ }
+};
+
+// Callback function that fills the audio output buffer.
+static aaudio_data_callback_result_t s_myDataCallbackProc(
+ AAudioStream *stream,
+ void *userData,
+ void *audioData,
+ int32_t numFrames
+) {
+ (void) audioData;
+ (void) numFrames;
+ AudioEngine *engine = (struct AudioEngine *)userData;
+ engine->callbackCount++;
+ if (engine->stopped) {
+ engine->callbackCountAfterStop++;
+ }
+
+ engine->framesRead = (int32_t)AAudioStream_getFramesRead(stream);
+ if (engine->startingFramesRead == 0) {
+ engine->startingFramesRead.store(engine->framesRead.load());
+ }
+ engine->framesCalled += numFrames;
+ if (engine->framesCalled >= engine->stopAtFrame) {
+ engine->stopped = true;
+ return AAUDIO_CALLBACK_RESULT_STOP;
+ } else {
+ return AAUDIO_CALLBACK_RESULT_CONTINUE;
+ }
+}
+
+static aaudio_result_t s_OpenAudioStream(struct AudioEngine *engine,
+ aaudio_direction_t direction,
+ aaudio_sharing_mode_t sharingMode,
+ aaudio_performance_mode_t perfMode) {
+ // Use an AAudioStreamBuilder to contain requested parameters.
+ aaudio_result_t result = AAudio_createStreamBuilder(&engine->builder);
+ if (result != AAUDIO_OK) {
+ printf("AAudio_createStreamBuilder returned %s",
+ AAudio_convertResultToText(result));
+ return result;
+ }
+
+ // Request stream properties.
+ AAudioStreamBuilder_setFormat(engine->builder, AAUDIO_FORMAT_PCM_FLOAT);
+ AAudioStreamBuilder_setPerformanceMode(engine->builder, perfMode);
+ AAudioStreamBuilder_setSharingMode(engine->builder, sharingMode);
+ AAudioStreamBuilder_setDirection(engine->builder, direction);
+ AAudioStreamBuilder_setDataCallback(engine->builder, s_myDataCallbackProc, engine);
+ AAudioStreamBuilder_setErrorCallback(engine->builder, s_myErrorCallbackProc, engine);
+
+ // Create an AAudioStream using the Builder.
+ result = AAudioStreamBuilder_openStream(engine->builder, &engine->stream);
+ if (result != AAUDIO_OK) {
+ printf("AAudioStreamBuilder_openStream returned %s",
+ AAudio_convertResultToText(result));
+ return result;
+ }
+
+ return result;
+}
+
+static aaudio_result_t s_CloseAudioStream(struct AudioEngine *engine) {
+ aaudio_result_t result = AAUDIO_OK;
+ if (engine->stream != nullptr) {
+ result = AAudioStream_close(engine->stream);
+ if (result != AAUDIO_OK) {
+ printf("AAudioStream_close returned %s\n",
+ AAudio_convertResultToText(result));
+ }
+ engine->stream = nullptr;
+ }
+ AAudioStreamBuilder_delete(engine->builder);
+ engine->builder = nullptr;
+ return result;
+}
+
+static void s_myErrorCallbackProc(
+ AAudioStream *stream __unused,
+ void *userData __unused,
+ aaudio_result_t error) {
+ printf("%s() - error = %d\n", __func__, error);
+}
+
+void usage() {
+ printf("test_return_stop [-i] [-x] [-n] [-c]\n");
+ printf(" -i direction INPUT, otherwise OUTPUT\n");
+ printf(" -x sharing mode EXCLUSIVE, otherwise SHARED\n");
+ printf(" -n performance mode NONE, otherwise LOW_LATENCY\n");
+ printf(" -c always return CONTINUE from callback, not STOP\n");
+}
+
+int main(int argc, char **argv) {
+ (void) argc;
+ (void) argv;
+ struct AudioEngine engine;
+ aaudio_sharing_mode_t sharingMode = AAUDIO_SHARING_MODE_SHARED;
+ aaudio_performance_mode_t perfMode = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
+ aaudio_direction_t direction = AAUDIO_DIRECTION_OUTPUT;
+ aaudio_result_t result = AAUDIO_OK;
+ bool alwaysContinue = false;
+ int errorCount = 0;
+ int callbackResult = EXIT_SUCCESS;
+
+ // Make printf print immediately so that debug info is not stuck
+ // in a buffer if we hang or crash.
+ setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
+
+ printf("Test Return Stop V1.0\n");
+ printf("Wait for a few seconds.\n");
+ printf("You should see callbackCount and framesRead stop advancing\n");
+ printf("when callbackCount reaches %d msec\n", STOP_AT_MSEC);
+ printf("\n");
+
+ for (int i = 1; i < argc; i++) {
+ const char *arg = argv[i];
+ if (arg[0] == '-') {
+ char option = arg[1];
+ switch (option) {
+ case 'c':
+ alwaysContinue = true;
+ break;
+ case 'i':
+ direction = AAUDIO_DIRECTION_INPUT;
+ break;
+ case 'n':
+ perfMode = AAUDIO_PERFORMANCE_MODE_NONE;
+ break;
+ case 'x':
+ sharingMode = AAUDIO_SHARING_MODE_EXCLUSIVE;
+ break;
+ default:
+ usage();
+ exit(EXIT_FAILURE);
+ break;
+ }
+ } else {
+ usage();
+ exit(EXIT_FAILURE);
+ break;
+ }
+ }
+
+ result = s_OpenAudioStream(&engine, direction, sharingMode, perfMode);
+ if (result != AAUDIO_OK) {
+ printf("s_OpenAudioStream returned %s",
+ AAudio_convertResultToText(result));
+ errorCount++;
+ }
+
+ int32_t framesPerBurst = AAudioStream_getFramesPerBurst(engine.stream);
+ // Check to see what kind of stream we actually got.
+ int32_t deviceId = AAudioStream_getDeviceId(engine.stream);
+ aaudio_performance_mode_t actualPerfMode = AAudioStream_getPerformanceMode(engine.stream);
+ printf("-------- opened: deviceId = %3d, framesPerBurst = %3d, perfMode = %d\n",
+ deviceId, framesPerBurst, actualPerfMode);
+
+ // Calculate how many callbacks needed.
+ if (alwaysContinue) {
+ engine.stopAtFrame = INT32_MAX;
+ } else {
+ int32_t sampleRate = AAudioStream_getSampleRate(engine.stream);
+ engine.stopAtFrame = STOP_AT_MSEC * sampleRate / 1000;
+ }
+
+ for (int loops = 0; loops < 2 && result == AAUDIO_OK; loops++) {
+ engine.reset();
+
+ // Start stream.
+ result = AAudioStream_requestStart(engine.stream);
+ printf("AAudioStream_requestStart() returned %d >>>>>>>>>>>>>>>>>>>>>>\n", result);
+ if (result != AAUDIO_OK) {
+ printf("ERROR - AAudioStream_requestStart returned %s",
+ AAudio_convertResultToText(result));
+ errorCount++;
+ break;
+ }
+
+ if (result == AAUDIO_OK) {
+ const int watchLoops = LOOP_DURATION_MSEC / SLEEP_DURATION_MSEC;
+ for (int i = watchLoops; i > 0; i--) {
+ printf("playing silence #%02d, framesRead = %7d, framesWritten = %7d,"
+ " framesCalled = %6d, callbackCount = %4d\n",
+ i,
+ (int32_t) AAudioStream_getFramesRead(engine.stream),
+ (int32_t) AAudioStream_getFramesWritten(engine.stream),
+ engine.framesCalled.load(),
+ engine.callbackCount.load()
+ );
+ usleep(SLEEP_DURATION_MSEC * 1000);
+ }
+ }
+
+ if (engine.stopAtFrame != INT32_MAX) {
+ callbackResult = (engine.callbackCountAfterStop == 0) ? EXIT_SUCCESS
+ : EXIT_FAILURE;
+ if (callbackResult) {
+ printf("ERROR - Callback count after STOP = %d\n",
+ engine.callbackCountAfterStop.load());
+ errorCount++;
+ }
+ }
+
+ if (engine.startingFramesRead.load() == engine.framesRead.load()) {
+ printf("ERROR - framesRead did not advance across callbacks\n");
+ errorCount++;
+ }
+
+ result = AAudioStream_requestStop(engine.stream);
+ printf("AAudioStream_requestStop() returned %d <<<<<<<<<<<<<<<<<<<<<\n", result);
+ if (result != AAUDIO_OK) {
+ errorCount++;
+ }
+ usleep(SLEEP_DURATION_MSEC * 1000);
+ printf("getFramesRead() = %d, getFramesWritten() = %d\n",
+ (int32_t) AAudioStream_getFramesRead(engine.stream),
+ (int32_t) AAudioStream_getFramesWritten(engine.stream));
+ }
+
+ s_CloseAudioStream(&engine);
+
+ printf("aaudio result = %d = %s\n", result, AAudio_convertResultToText(result));
+ printf("test %s\n", errorCount ? "FAILED" : "PASSED");
+
+ return errorCount ? EXIT_FAILURE : EXIT_SUCCESS;
+}
diff --git a/media/libaudiohal/EffectsFactoryHalInterface.cpp b/media/libaudiohal/EffectsFactoryHalInterface.cpp
index e21c235..bd3ef61 100644
--- a/media/libaudiohal/EffectsFactoryHalInterface.cpp
+++ b/media/libaudiohal/EffectsFactoryHalInterface.cpp
@@ -25,13 +25,13 @@
// static
sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
if (hardware::audio::effect::V5_0::IEffectsFactory::getService() != nullptr) {
- return V5_0::createEffectsFactoryHal();
+ return effect::V5_0::createEffectsFactoryHal();
}
if (hardware::audio::effect::V4_0::IEffectsFactory::getService() != nullptr) {
- return V4_0::createEffectsFactoryHal();
+ return effect::V4_0::createEffectsFactoryHal();
}
if (hardware::audio::effect::V2_0::IEffectsFactory::getService() != nullptr) {
- return V2_0::createEffectsFactoryHal();
+ return effect::V2_0::createEffectsFactoryHal();
}
return nullptr;
}
diff --git a/media/libaudiohal/HalDeathHandlerHidl.cpp b/media/libaudiohal/HalDeathHandlerHidl.cpp
index 1e3ab58..6e33523 100644
--- a/media/libaudiohal/HalDeathHandlerHidl.cpp
+++ b/media/libaudiohal/HalDeathHandlerHidl.cpp
@@ -54,7 +54,7 @@
handler.second();
}
ALOGE("HAL server crashed, audio server is restarting");
- exit(1);
+ _exit(1); // Avoid calling atexit handlers, as this code runs on a thread from RPC threadpool.
}
} // namespace android
diff --git a/media/libaudiohal/impl/ConversionHelperHidl.cpp b/media/libaudiohal/impl/ConversionHelperHidl.cpp
index 9747859..9f8a520 100644
--- a/media/libaudiohal/impl/ConversionHelperHidl.cpp
+++ b/media/libaudiohal/impl/ConversionHelperHidl.cpp
@@ -22,19 +22,12 @@
#include "ConversionHelperHidl.h"
-using ::android::hardware::audio::CPP_VERSION::Result;
-
-#if MAJOR_VERSION >= 4
-using ::android::hardware::audio::CPP_VERSION::AudioMicrophoneChannelMapping;
-using ::android::hardware::audio::CPP_VERSION::AudioMicrophoneDirectionality;
-using ::android::hardware::audio::CPP_VERSION::AudioMicrophoneLocation;
-using ::android::hardware::audio::CPP_VERSION::DeviceAddress;
-using ::android::hardware::audio::CPP_VERSION::MicrophoneInfo;
-#endif
-
namespace android {
namespace CPP_VERSION {
+using namespace ::android::hardware::audio::common::CPP_VERSION;
+using namespace ::android::hardware::audio::CPP_VERSION;
+
// static
status_t ConversionHelperHidl::keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys) {
AudioParameter halKeys(keys);
diff --git a/media/libaudiohal/impl/EffectBufferHalHidl.cpp b/media/libaudiohal/impl/EffectBufferHalHidl.cpp
index 6ef4e8a..5367972 100644
--- a/media/libaudiohal/impl/EffectBufferHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectBufferHalHidl.cpp
@@ -30,6 +30,7 @@
using ::android::hidl::allocator::V1_0::IAllocator;
namespace android {
+namespace effect {
namespace CPP_VERSION {
// static
@@ -142,5 +143,6 @@
memcpy(mExternalData, mAudioBuffer.raw, size);
}
+} // namespace effect
} // namespace CPP_VERSION
} // namespace android
diff --git a/media/libaudiohal/impl/EffectBufferHalHidl.h b/media/libaudiohal/impl/EffectBufferHalHidl.h
index 0c99a02..4826813 100644
--- a/media/libaudiohal/impl/EffectBufferHalHidl.h
+++ b/media/libaudiohal/impl/EffectBufferHalHidl.h
@@ -23,13 +23,15 @@
#include <media/audiohal/EffectBufferHalInterface.h>
#include <system/audio_effect.h>
-using android::hardware::audio::effect::CPP_VERSION::AudioBuffer;
using android::hardware::hidl_memory;
using android::hidl::memory::V1_0::IMemory;
namespace android {
+namespace effect {
namespace CPP_VERSION {
+using namespace ::android::hardware::audio::effect::CPP_VERSION;
+
class EffectBufferHalHidl : public EffectBufferHalInterface
{
public:
@@ -73,6 +75,7 @@
};
} // namespace CPP_VERSION
+} // namespace effect
} // namespace android
#endif // ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H
diff --git a/media/libaudiohal/impl/EffectHalHidl.cpp b/media/libaudiohal/impl/EffectHalHidl.cpp
index df79b95..b0597b3 100644
--- a/media/libaudiohal/impl/EffectHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectHalHidl.cpp
@@ -26,11 +26,6 @@
#include "EffectHalHidl.h"
#include "HidlUtils.h"
-using ::android::hardware::audio::effect::CPP_VERSION::AudioBuffer;
-using ::android::hardware::audio::effect::CPP_VERSION::EffectBufferAccess;
-using ::android::hardware::audio::effect::CPP_VERSION::EffectConfigParameters;
-using ::android::hardware::audio::effect::CPP_VERSION::MessageQueueFlagBits;
-using ::android::hardware::audio::effect::CPP_VERSION::Result;
using ::android::hardware::audio::common::CPP_VERSION::implementation::HidlUtils;
using ::android::hardware::audio::common::utils::EnumBitfield;
using ::android::hardware::hidl_vec;
@@ -38,9 +33,11 @@
using ::android::hardware::Return;
namespace android {
+namespace effect {
namespace CPP_VERSION {
using namespace ::android::hardware::audio::common::CPP_VERSION;
+using namespace ::android::hardware::audio::effect::CPP_VERSION;
EffectHalHidl::EffectHalHidl(const sp<IEffect>& effect, uint64_t effectId)
: mEffect(effect), mEffectId(effectId), mBuffersChanged(true), mEfGroup(nullptr) {
@@ -338,4 +335,5 @@
}
} // namespace CPP_VERSION
+} // namespace effect
} // namespace android
diff --git a/media/libaudiohal/impl/EffectHalHidl.h b/media/libaudiohal/impl/EffectHalHidl.h
index cd447ff..9d9f707 100644
--- a/media/libaudiohal/impl/EffectHalHidl.h
+++ b/media/libaudiohal/impl/EffectHalHidl.h
@@ -23,17 +23,15 @@
#include <fmq/MessageQueue.h>
#include <system/audio_effect.h>
-using ::android::hardware::audio::effect::CPP_VERSION::EffectBufferConfig;
-using ::android::hardware::audio::effect::CPP_VERSION::EffectConfig;
-using ::android::hardware::audio::effect::CPP_VERSION::EffectDescriptor;
-using ::android::hardware::audio::effect::CPP_VERSION::IEffect;
-using EffectResult = ::android::hardware::audio::effect::CPP_VERSION::Result;
using ::android::hardware::EventFlag;
using ::android::hardware::MessageQueue;
namespace android {
+namespace effect {
namespace CPP_VERSION {
+using namespace ::android::hardware::audio::effect::CPP_VERSION;
+
class EffectHalHidl : public EffectHalInterface
{
public:
@@ -70,7 +68,7 @@
private:
friend class EffectsFactoryHalHidl;
- typedef MessageQueue<EffectResult, hardware::kSynchronizedReadWrite> StatusMQ;
+ typedef MessageQueue<Result, hardware::kSynchronizedReadWrite> StatusMQ;
sp<IEffect> mEffect;
const uint64_t mEffectId;
@@ -80,7 +78,7 @@
std::unique_ptr<StatusMQ> mStatusMQ;
EventFlag* mEfGroup;
- static status_t analyzeResult(const EffectResult& result);
+ static status_t analyzeResult(const Result& result);
static void effectBufferConfigFromHal(
const buffer_config_t& halConfig, EffectBufferConfig* config);
static void effectBufferConfigToHal(
@@ -105,6 +103,7 @@
};
} // namespace CPP_VERSION
+} // namespace effect
} // namespace android
#endif // ANDROID_HARDWARE_EFFECT_HAL_HIDL_H
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index 7fea466..7fd6bde 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -26,14 +26,14 @@
#include "HidlUtils.h"
using ::android::hardware::audio::common::CPP_VERSION::implementation::HidlUtils;
-using ::android::hardware::audio::effect::CPP_VERSION::IEffect;
-using ::android::hardware::audio::effect::CPP_VERSION::Result;
using ::android::hardware::Return;
namespace android {
+namespace effect {
namespace CPP_VERSION {
using namespace ::android::hardware::audio::common::CPP_VERSION;
+using namespace ::android::hardware::audio::effect::CPP_VERSION;
EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
mEffectsFactory = IEffectsFactory::getService();
@@ -145,6 +145,6 @@
return EffectBufferHalHidl::mirror(external, size, buffer);
}
-
} // namespace CPP_VERSION
+} // namespace effect
} // namespace android
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
index 7027153..01178ff 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -24,11 +24,12 @@
#include "ConversionHelperHidl.h"
namespace android {
+namespace effect {
namespace CPP_VERSION {
-using ::android::hardware::audio::effect::CPP_VERSION::EffectDescriptor;
-using ::android::hardware::audio::effect::CPP_VERSION::IEffectsFactory;
using ::android::hardware::hidl_vec;
+using ::android::CPP_VERSION::ConversionHelperHidl;
+using namespace ::android::hardware::audio::effect::CPP_VERSION;
class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
{
@@ -70,6 +71,7 @@
}
} // namespace CPP_VERSION
+} // namespace effect
} // namespace android
#endif // ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index 9765f1e..c12b362 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -35,6 +35,7 @@
namespace android {
namespace CPP_VERSION {
+using EffectHalHidl = ::android::effect::CPP_VERSION::EffectHalHidl;
using ReadCommand = ::android::hardware::audio::CPP_VERSION::IStreamIn::ReadCommand;
using namespace ::android::hardware::audio::common::CPP_VERSION;
diff --git a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
index 1d912a0..c7319d0 100644
--- a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
+++ b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
@@ -25,18 +25,29 @@
namespace android {
+namespace effect {
namespace V2_0 {
sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
} // namespace V2_0
namespace V4_0 {
sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
} // namespace V4_0
namespace V5_0 {
sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
+} // namespace V5_0
+} // namespace effect
+
+namespace V2_0 {
+sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
+} // namespace V2_0
+
+namespace V4_0 {
+sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
+} // namespace V4_0
+
+namespace V5_0 {
sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
} // namespace V5_0
diff --git a/media/libaudioprocessing/Android.bp b/media/libaudioprocessing/Android.bp
new file mode 100644
index 0000000..817fb0b
--- /dev/null
+++ b/media/libaudioprocessing/Android.bp
@@ -0,0 +1,54 @@
+cc_defaults {
+ name: "libaudioprocessing_defaults",
+
+ export_include_dirs: ["include"],
+
+ shared_libs: [
+ "libaudiohal",
+ "libaudioutils",
+ "libcutils",
+ "liblog",
+ "libnbaio",
+ "libnblog",
+ "libsonic",
+ "libutils",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+
+ // uncomment to disable NEON on architectures that actually do support NEON, for benchmarking
+ // "-DUSE_NEON=false",
+ ],
+}
+
+cc_library_shared {
+ name: "libaudioprocessing",
+ defaults: ["libaudioprocessing_defaults"],
+
+ srcs: [
+ "BufferProviders.cpp",
+ "RecordBufferConverter.cpp",
+ ],
+ whole_static_libs: ["libaudioprocessing_arm"],
+}
+
+cc_library_static {
+ name: "libaudioprocessing_arm",
+ defaults: ["libaudioprocessing_defaults"],
+
+ srcs: [
+ "AudioMixer.cpp",
+ "AudioResampler.cpp",
+ "AudioResamplerCubic.cpp",
+ "AudioResamplerSinc.cpp",
+ "AudioResamplerDyn.cpp",
+ ],
+
+ arch: {
+ arm: {
+ instruction_set: "arm",
+ },
+ },
+}
diff --git a/media/libaudioprocessing/Android.mk b/media/libaudioprocessing/Android.mk
deleted file mode 100644
index da1ecc2..0000000
--- a/media/libaudioprocessing/Android.mk
+++ /dev/null
@@ -1,40 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- AudioMixer.cpp.arm \
- AudioResampler.cpp.arm \
- AudioResamplerCubic.cpp.arm \
- AudioResamplerSinc.cpp.arm \
- AudioResamplerDyn.cpp.arm \
- BufferProviders.cpp \
- RecordBufferConverter.cpp \
-
-LOCAL_C_INCLUDES := \
- $(TOP) \
- $(call include-path-for, audio-utils) \
- $(LOCAL_PATH)/include \
-
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/include
-
-LOCAL_SHARED_LIBRARIES := \
- libaudiohal \
- libaudioutils \
- libcutils \
- liblog \
- libnbaio \
- libnblog \
- libsonic \
- libutils \
-
-LOCAL_MODULE := libaudioprocessing
-
-LOCAL_CFLAGS := -Werror -Wall
-
-# uncomment to disable NEON on architectures that actually do support NEON, for benchmarking
-#LOCAL_CFLAGS += -DUSE_NEON=false
-
-include $(BUILD_SHARED_LIBRARY)
-
-include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/media/libaudioprocessing/BufferProviders.cpp b/media/libaudioprocessing/BufferProviders.cpp
index a1a1a0d..fe92d43 100644
--- a/media/libaudioprocessing/BufferProviders.cpp
+++ b/media/libaudioprocessing/BufferProviders.cpp
@@ -20,7 +20,7 @@
#include <audio_utils/primitives.h>
#include <audio_utils/format.h>
#include <audio_utils/channels.h>
-#include <external/sonic/sonic.h>
+#include <sonic.h>
#include <media/audiohal/EffectBufferHalInterface.h>
#include <media/audiohal/EffectHalInterface.h>
#include <media/audiohal/EffectsFactoryHalInterface.h>
diff --git a/media/libaudioprocessing/audio-resampler/Android.bp b/media/libaudioprocessing/audio-resampler/Android.bp
new file mode 100644
index 0000000..dc70310
--- /dev/null
+++ b/media/libaudioprocessing/audio-resampler/Android.bp
@@ -0,0 +1,15 @@
+cc_library_shared {
+ name: "libaudio-resampler",
+
+ srcs: ["AudioResamplerCoefficients.cpp"],
+
+ shared_libs: [
+ "libutils",
+ "liblog",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+}
diff --git a/media/libaudioprocessing/audio-resampler/Android.mk b/media/libaudioprocessing/audio-resampler/Android.mk
deleted file mode 100644
index bb2807c..0000000
--- a/media/libaudioprocessing/audio-resampler/Android.mk
+++ /dev/null
@@ -1,16 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- AudioResamplerCoefficients.cpp
-
-LOCAL_MODULE := libaudio-resampler
-
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_SHARED_LIBRARIES := libutils liblog
-
-LOCAL_CFLAGS += -Werror -Wall
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libaudioprocessing/tests/Android.bp b/media/libaudioprocessing/tests/Android.bp
new file mode 100644
index 0000000..811c16b
--- /dev/null
+++ b/media/libaudioprocessing/tests/Android.bp
@@ -0,0 +1,51 @@
+// Build the unit tests for libaudioprocessing
+
+cc_defaults {
+ name: "libaudioprocessing_test_defaults",
+
+ header_libs: ["libbase_headers"],
+ shared_libs: [
+ "libaudioutils",
+ "libaudioprocessing",
+ "libcutils",
+ "liblog",
+ "libutils",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+}
+
+//
+// resampler unit test
+//
+cc_test {
+ name: "resampler_tests",
+ defaults: ["libaudioprocessing_test_defaults"],
+
+ srcs: ["resampler_tests.cpp"],
+}
+
+//
+// audio mixer test tool
+//
+cc_binary {
+ name: "test-mixer",
+ defaults: ["libaudioprocessing_test_defaults"],
+
+ srcs: ["test-mixer.cpp"],
+ static_libs: ["libsndfile"],
+}
+
+//
+// build audio resampler test tool
+//
+cc_binary {
+ name: "test-resampler",
+ defaults: ["libaudioprocessing_test_defaults"],
+
+ srcs: ["test-resampler.cpp"],
+ static_libs: ["libsndfile"],
+}
diff --git a/media/libaudioprocessing/tests/Android.mk b/media/libaudioprocessing/tests/Android.mk
deleted file mode 100644
index 31ffbdc..0000000
--- a/media/libaudioprocessing/tests/Android.mk
+++ /dev/null
@@ -1,93 +0,0 @@
-# Build the unit tests for libaudioprocessing
-
-LOCAL_PATH := $(call my-dir)
-
-#
-# resampler unit test
-#
-include $(CLEAR_VARS)
-
-LOCAL_SHARED_LIBRARIES := \
- libaudioutils \
- libaudioprocessing \
- libcutils \
- liblog \
- libutils \
-
-LOCAL_C_INCLUDES := \
- $(call include-path-for, audio-utils) \
-
-LOCAL_SRC_FILES := \
- resampler_tests.cpp
-
-LOCAL_HEADER_LIBRARIES := libbase_headers
-
-LOCAL_MODULE := resampler_tests
-
-LOCAL_MODULE_TAGS := tests
-
-LOCAL_CFLAGS := -Werror -Wall
-
-include $(BUILD_NATIVE_TEST)
-
-#
-# audio mixer test tool
-#
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- test-mixer.cpp \
-
-LOCAL_C_INCLUDES := \
- $(call include-path-for, audio-utils) \
-
-LOCAL_STATIC_LIBRARIES := \
- libsndfile \
-
-LOCAL_SHARED_LIBRARIES := \
- libaudioprocessing \
- libaudioutils \
- libcutils \
- liblog \
- libutils \
-
-LOCAL_HEADER_LIBRARIES := libbase_headers
-
-LOCAL_MODULE := test-mixer
-
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_CFLAGS := -Werror -Wall
-
-include $(BUILD_EXECUTABLE)
-
-#
-# build audio resampler test tool
-#
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- test-resampler.cpp \
-
-LOCAL_C_INCLUDES := \
- $(call include-path-for, audio-utils) \
-
-LOCAL_STATIC_LIBRARIES := \
- libsndfile \
-
-LOCAL_SHARED_LIBRARIES := \
- libaudioprocessing \
- libaudioutils \
- libcutils \
- liblog \
- libutils \
-
-LOCAL_HEADER_LIBRARIES := libbase_headers
-
-LOCAL_MODULE := test-resampler
-
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_CFLAGS := -Werror -Wall
-
-include $(BUILD_EXECUTABLE)
diff --git a/media/libeffects/config/src/EffectsConfig.cpp b/media/libeffects/config/src/EffectsConfig.cpp
index 351b1ee..76b4adc 100644
--- a/media/libeffects/config/src/EffectsConfig.cpp
+++ b/media/libeffects/config/src/EffectsConfig.cpp
@@ -305,7 +305,7 @@
return parseWithPath(path);
}
- for (std::string location : DEFAULT_LOCATIONS) {
+ for (const std::string& location : DEFAULT_LOCATIONS) {
std::string defaultPath = location + '/' + DEFAULT_NAME;
if (access(defaultPath.c_str(), R_OK) != 0) {
continue;
diff --git a/media/libeffects/lvm/lib/Bass/lib/LVDBE.h b/media/libeffects/lvm/lib/Bass/lib/LVDBE.h
index a1fa79a..cc066b0 100644
--- a/media/libeffects/lvm/lib/Bass/lib/LVDBE.h
+++ b/media/libeffects/lvm/lib/Bass/lib/LVDBE.h
@@ -199,8 +199,10 @@
#define LVDBE_CAP_FS_44100 128
#define LVDBE_CAP_FS_48000 256
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
-#define LVDBE_CAP_FS_96000 512
-#define LVDBE_CAP_FS_192000 1024
+#define LVDBE_CAP_FS_88200 512
+#define LVDBE_CAP_FS_96000 1024
+#define LVDBE_CAP_FS_176400 2048
+#define LVDBE_CAP_FS_192000 4096
#endif
typedef enum
@@ -215,8 +217,10 @@
LVDBE_FS_44100 = 7,
LVDBE_FS_48000 = 8,
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
- LVDBE_FS_96000 = 9,
- LVDBE_FS_192000 = 10,
+ LVDBE_FS_88200 = 9,
+ LVDBE_FS_96000 = 10,
+ LVDBE_FS_176400 = 11,
+ LVDBE_FS_192000 = 12,
#endif
LVDBE_FS_MAX = LVM_MAXINT_32
} LVDBE_Fs_en;
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h b/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h
index 4ecaf14..8f058e8 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h
@@ -580,12 +580,24 @@
#define HPF_Fs48000_Fc55_B2 0.989882f
#ifdef HIGHER_FS
+#define HPF_Fs88200_Fc55_A0 0.985818f
+#define HPF_Fs88200_Fc55_A1 (-1.971636f)
+#define HPF_Fs88200_Fc55_A2 0.985818f
+#define HPF_Fs88200_Fc55_B1 (-1.994466f)
+#define HPF_Fs88200_Fc55_B2 0.994481f
+
#define HPF_Fs96000_Fc55_A0 0.986040f
#define HPF_Fs96000_Fc55_A1 (-1.972080f)
#define HPF_Fs96000_Fc55_A2 0.986040f
#define HPF_Fs96000_Fc55_B1 (-1.994915f)
#define HPF_Fs96000_Fc55_B2 0.994928f
+#define HPF_Fs176400_Fc55_A0 0.987183f
+#define HPF_Fs176400_Fc55_A1 (-1.974366f)
+#define HPF_Fs176400_Fc55_A2 0.987183f
+#define HPF_Fs176400_Fc55_B1 (-1.997233f)
+#define HPF_Fs176400_Fc55_B2 0.997237f
+
#define HPF_Fs192000_Fc55_A0 0.987294f
#define HPF_Fs192000_Fc55_A1 (-1.974588f)
#define HPF_Fs192000_Fc55_A2 0.987294f
@@ -642,12 +654,24 @@
#define HPF_Fs48000_Fc66_B2 0.987871f
#ifdef HIGHER_FS
+#define HPF_Fs88200_Fc66_A0 0.985273f
+#define HPF_Fs88200_Fc66_A1 (-1.970546f)
+#define HPF_Fs88200_Fc66_A2 0.985273f
+#define HPF_Fs88200_Fc66_B1 (-1.993359f)
+#define HPF_Fs88200_Fc66_B2 0.993381f
+
#define HPF_Fs96000_Fc66_A0 0.985539f
#define HPF_Fs96000_Fc66_A1 (-1.971077f)
#define HPF_Fs96000_Fc66_A2 0.985539f
#define HPF_Fs96000_Fc66_B1 (-1.993898f)
#define HPF_Fs96000_Fc66_B2 0.993917f
+#define HPF_Fs176400_Fc66_A0 0.986910f
+#define HPF_Fs176400_Fc66_A1 (-1.973820f)
+#define HPF_Fs176400_Fc66_A2 0.986910f
+#define HPF_Fs176400_Fc66_B1 (-1.996679f)
+#define HPF_Fs176400_Fc66_B2 0.996685f
+
#define HPF_Fs192000_Fc66_A0 0.987043f
#define HPF_Fs192000_Fc66_A1 (-1.974086f)
#define HPF_Fs192000_Fc66_A2 0.987043f
@@ -703,12 +727,24 @@
#define HPF_Fs48000_Fc78_B2 0.985681f
#ifdef HIGHER_FS
+#define HPF_Fs88200_Fc78_A0 0.984678f
+#define HPF_Fs88200_Fc78_A1 (-1.969356f)
+#define HPF_Fs88200_Fc78_A2 0.984678f
+#define HPF_Fs88200_Fc78_B1 (-1.992151f)
+#define HPF_Fs88200_Fc78_B2 0.992182f
+
#define HPF_Fs96000_Fc78_A0 0.984992f
#define HPF_Fs96000_Fc78_A1 (-1.969984f)
#define HPF_Fs96000_Fc78_A2 0.984992f
#define HPF_Fs96000_Fc78_B1 (-1.992789f)
#define HPF_Fs96000_Fc78_B2 0.992815f
+#define HPF_Fs176400_Fc78_A0 0.986612f
+#define HPF_Fs176400_Fc78_A1 (-1.973224f)
+#define HPF_Fs176400_Fc78_A2 0.986612f
+#define HPF_Fs176400_Fc78_B1 (-1.996076f)
+#define HPF_Fs176400_Fc78_B2 0.996083f
+
#define HPF_Fs192000_Fc78_A0 0.986769f
#define HPF_Fs192000_Fc78_A1 (-1.973539f)
#define HPF_Fs192000_Fc78_A2 0.986769f
@@ -764,12 +800,24 @@
#define HPF_Fs48000_Fc90_B2 0.983497f
#ifdef HIGHER_FS
+#define HPF_Fs88200_Fc90_A0 0.984084f
+#define HPF_Fs88200_Fc90_A1 (-1.968168f)
+#define HPF_Fs88200_Fc90_A2 0.984084f
+#define HPF_Fs88200_Fc90_B1 (-1.990944f)
+#define HPF_Fs88200_Fc90_B2 0.990985f
+
#define HPF_Fs96000_Fc90_A0 0.984446f
#define HPF_Fs96000_Fc90_A1 (-1.968892f)
#define HPF_Fs96000_Fc90_A2 0.984446f
#define HPF_Fs96000_Fc90_B1 (-1.991680f)
#define HPF_Fs96000_Fc90_B2 0.991714f
+#define HPF_Fs176400_Fc90_A0 0.986314f
+#define HPF_Fs176400_Fc90_A1 (-1.972629f)
+#define HPF_Fs176400_Fc90_A2 0.986314f
+#define HPF_Fs176400_Fc90_B1 (-1.995472f)
+#define HPF_Fs176400_Fc90_B2 0.995482f
+
#define HPF_Fs192000_Fc90_A0 0.986496f
#define HPF_Fs192000_Fc90_A1 (-1.972992f)
#define HPF_Fs192000_Fc90_A2 0.986496f
@@ -831,12 +879,24 @@
#define BPF_Fs48000_Fc55_B2 0.996875f
#ifdef HIGHER_FS
+#define BPF_Fs88200_Fc55_A0 0.000831f
+#define BPF_Fs88200_Fc55_A1 0.000000f
+#define BPF_Fs88200_Fc55_A2 (-0.000831f)
+#define BPF_Fs88200_Fc55_B1 (-1.998321f)
+#define BPF_Fs88200_Fc55_B2 0.998338f
+
#define BPF_Fs96000_Fc55_A0 0.000762f
#define BPF_Fs96000_Fc55_A1 0.000000f
#define BPF_Fs96000_Fc55_A2 (-0.000762f)
#define BPF_Fs96000_Fc55_B1 (-1.998461f)
#define BPF_Fs96000_Fc55_B2 0.998477f
+#define BPF_Fs176400_Fc55_A0 0.000416f
+#define BPF_Fs176400_Fc55_A1 0.000000f
+#define BPF_Fs176400_Fc55_A2 (-0.000416f)
+#define BPF_Fs176400_Fc55_B1 (-1.999164f)
+#define BPF_Fs176400_Fc55_B2 0.999169f
+
#define BPF_Fs192000_Fc55_A0 0.000381f
#define BPF_Fs192000_Fc55_A1 0.000000f
#define BPF_Fs192000_Fc55_A2 (-0.000381f)
@@ -892,12 +952,24 @@
#define BPF_Fs48000_Fc66_B2 0.995690f
#ifdef HIGHER_FS
+#define BPF_Fs88200_Fc66_A0 0.001146f
+#define BPF_Fs88200_Fc66_A1 0.000000f
+#define BPF_Fs88200_Fc66_A2 (-0.001146f)
+#define BPF_Fs88200_Fc66_B1 (-1.997684f)
+#define BPF_Fs88200_Fc66_B2 0.997708f
+
#define BPF_Fs96000_Fc66_A0 0.001055f
#define BPF_Fs96000_Fc66_A1 0.000000f
#define BPF_Fs96000_Fc66_A2 (-0.001055f)
#define BPF_Fs96000_Fc66_B1 (-1.997868f)
#define BPF_Fs96000_Fc66_B2 0.997891f
+#define BPF_Fs176400_Fc66_A0 0.000573f
+#define BPF_Fs176400_Fc66_A1 0.000000f
+#define BPF_Fs176400_Fc66_A2 (-0.000573f)
+#define BPF_Fs176400_Fc66_B1 (-1.998847f)
+#define BPF_Fs176400_Fc66_B2 0.998853f
+
#define BPF_Fs192000_Fc66_A0 0.000528f
#define BPF_Fs192000_Fc66_A1 0.000000f
#define BPF_Fs192000_Fc66_A2 (-0.000528f)
@@ -953,12 +1025,24 @@
#define BPF_Fs48000_Fc78_B2 0.993639f
#ifdef HIGHER_FS
+#define BPF_Fs88200_Fc78_A0 0.001693f
+#define BPF_Fs88200_Fc78_A1 0.000000f
+#define BPF_Fs88200_Fc78_A2 (-0.001693f)
+#define BPF_Fs88200_Fc78_B1 (-1.996582f)
+#define BPF_Fs88200_Fc78_B2 0.996615f
+
#define BPF_Fs96000_Fc78_A0 0.001555f
#define BPF_Fs96000_Fc78_A1 0.000000f
#define BPF_Fs96000_Fc78_A2 (-0.0015555f)
#define BPF_Fs96000_Fc78_B1 (-1.996860f)
#define BPF_Fs96000_Fc78_B2 0.996891f
+#define BPF_Fs176400_Fc78_A0 0.000847f
+#define BPF_Fs176400_Fc78_A1 0.000000f
+#define BPF_Fs176400_Fc78_A2 (-0.000847f)
+#define BPF_Fs176400_Fc78_B1 (-1.998298f)
+#define BPF_Fs176400_Fc78_B2 0.998306f
+
#define BPF_Fs192000_Fc78_A0 0.000778f
#define BPF_Fs192000_Fc78_A1 0.000000f
#define BPF_Fs192000_Fc78_A2 (-0.000778f)
@@ -1014,12 +1098,24 @@
#define BPF_Fs48000_Fc90_B2 0.992177f
#ifdef HIGHER_FS
+#define BPF_Fs88200_Fc90_A0 0.002083f
+#define BPF_Fs88200_Fc90_A1 0.000000f
+#define BPF_Fs88200_Fc90_A2 (-0.002083f)
+#define BPF_Fs88200_Fc90_B1 (-1.995791f)
+#define BPF_Fs88200_Fc90_B2 0.995835f
+
#define BPF_Fs96000_Fc90_A0 0.001913f
#define BPF_Fs96000_Fc90_A1 0.000000f
#define BPF_Fs96000_Fc90_A2 (-0.001913f)
#define BPF_Fs96000_Fc90_B1 (-1.996134f)
#define BPF_Fs96000_Fc90_B2 0.996174f
+#define BPF_Fs176400_Fc90_A0 0.001042f
+#define BPF_Fs176400_Fc90_A1 0.000000f
+#define BPF_Fs176400_Fc90_A2 (-0.001042f)
+#define BPF_Fs176400_Fc90_B1 (-1.997904f)
+#define BPF_Fs176400_Fc90_B2 0.997915f
+
#define BPF_Fs192000_Fc90_A0 0.000958f
#define BPF_Fs192000_Fc90_A1 0.000000f
#define BPF_Fs192000_Fc90_A2 (-0.000958f)
@@ -1045,7 +1141,9 @@
#define AGC_ATTACK_Fs48000 0.971628f
#ifdef HIGHER_FS
+#define AGC_ATTACK_Fs88200 0.984458f
#define AGC_ATTACK_Fs96000 0.985712f
+#define AGC_ATTACK_Fs176400 0.992199f
#define AGC_ATTACK_Fs192000 0.992830f
#endif
@@ -1062,7 +1160,9 @@
#define AGC_DECAY_Fs48000 0.000007f
#ifdef HIGHER_FS
+#define AGC_DECAY_Fs88200 0.0000038f
#define AGC_DECAY_FS96000 0.0000035f
+#define AGC_DECAY_Fs176400 0.00000188f
#define AGC_DECAY_FS192000 0.00000175f
#endif
@@ -1125,7 +1225,9 @@
#define VOL_TC_Fs44100 0.004525f
#define VOL_TC_Fs48000 0.004158f
#ifdef HIGHER_FS
+#define VOL_TC_Fs88200 0.002263f
#define VOL_TC_Fs96000 0.002079f
+#define VOL_TC_Fs176400 0.001131f
#define VOL_TC_Fs192000 0.001039f
#endif
#define MIX_TC_Fs8000 29365 /* Floating point value 0.896151 */
@@ -1138,9 +1240,13 @@
#define MIX_TC_Fs44100 32097 /* Floating point value 0.979515 */
#define MIX_TC_Fs48000 32150 /* Floating point value 0.981150 */
#ifdef HIGHER_FS
+/* Floating point value 0.989704 */
+#define MIX_TC_Fs88200 32430
#define MIX_TC_Fs96000 32456 /* Floating point value 0.990530 */
+/* Floating point value 0.994838 */
+#define MIX_TC_Fs176400 32598
#define MIX_TC_Fs192000 32611 /* Floating point value 0.992524 */
#endif
#endif /*BUILD_FLOAT*/
-#endif
\ No newline at end of file
+#endif
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.c b/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.c
index c4a9b14..a2ce404 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.c
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.c
@@ -88,11 +88,21 @@
-HPF_Fs48000_Fc55_B2,
-HPF_Fs48000_Fc55_B1},
#ifdef HIGHER_FS
+ {HPF_Fs88200_Fc55_A2, /* 88kS/s coefficients */
+ HPF_Fs88200_Fc55_A1,
+ HPF_Fs88200_Fc55_A0,
+ -HPF_Fs88200_Fc55_B2,
+ -HPF_Fs88200_Fc55_B1},
{HPF_Fs96000_Fc55_A2, /* 96kS/s coefficients */
HPF_Fs96000_Fc55_A1,
HPF_Fs96000_Fc55_A0,
-HPF_Fs96000_Fc55_B2,
-HPF_Fs96000_Fc55_B1},
+ {HPF_Fs176400_Fc55_A2, /* 176kS/s coefficients */
+ HPF_Fs176400_Fc55_A1,
+ HPF_Fs176400_Fc55_A0,
+ -HPF_Fs176400_Fc55_B2,
+ -HPF_Fs176400_Fc55_B1},
{HPF_Fs192000_Fc55_A2, /* 192kS/s coefficients */
HPF_Fs192000_Fc55_A1,
HPF_Fs192000_Fc55_A0,
@@ -147,11 +157,21 @@
-HPF_Fs48000_Fc66_B2,
-HPF_Fs48000_Fc66_B1},
#ifdef HIGHER_FS
+ {HPF_Fs88200_Fc66_A2, /* 88kS/s coefficients */
+ HPF_Fs88200_Fc66_A1,
+ HPF_Fs88200_Fc66_A0,
+ -HPF_Fs88200_Fc66_B2,
+ -HPF_Fs88200_Fc66_B1},
{HPF_Fs96000_Fc66_A2, /* 96kS/s coefficients */
HPF_Fs96000_Fc66_A1,
HPF_Fs96000_Fc66_A0,
-HPF_Fs96000_Fc66_B2,
-HPF_Fs96000_Fc66_B1},
+ {HPF_Fs176400_Fc66_A2, /* 176kS/s coefficients */
+ HPF_Fs176400_Fc66_A1,
+ HPF_Fs176400_Fc66_A0,
+ -HPF_Fs176400_Fc66_B2,
+ -HPF_Fs176400_Fc66_B1},
{HPF_Fs192000_Fc66_A2, /* 192kS/s coefficients */
HPF_Fs192000_Fc66_A1,
HPF_Fs192000_Fc66_A0,
@@ -207,11 +227,21 @@
-HPF_Fs48000_Fc78_B2,
-HPF_Fs48000_Fc78_B1},
#ifdef HIGHER_FS
+ {HPF_Fs88200_Fc78_A2, /* 88kS/s coefficients */
+ HPF_Fs88200_Fc78_A1,
+ HPF_Fs88200_Fc78_A0,
+ -HPF_Fs88200_Fc78_B2,
+ -HPF_Fs88200_Fc78_B1},
{HPF_Fs96000_Fc78_A2, /* 96kS/s coefficients */
HPF_Fs96000_Fc78_A1,
HPF_Fs96000_Fc78_A0,
-HPF_Fs96000_Fc78_B2,
-HPF_Fs96000_Fc78_B1},
+ {HPF_Fs176400_Fc78_A2, /* 176kS/s coefficients */
+ HPF_Fs176400_Fc78_A1,
+ HPF_Fs176400_Fc78_A0,
+ -HPF_Fs176400_Fc78_B2,
+ -HPF_Fs176400_Fc78_B1},
{HPF_Fs192000_Fc78_A2, /* 192kS/s coefficients */
HPF_Fs192000_Fc78_A1,
HPF_Fs192000_Fc78_A0,
@@ -269,11 +299,21 @@
#ifdef HIGHER_FS
,
+ {HPF_Fs88200_Fc90_A2, /* 88kS/s coefficients */
+ HPF_Fs88200_Fc90_A1,
+ HPF_Fs88200_Fc90_A0,
+ -HPF_Fs88200_Fc90_B2,
+ -HPF_Fs88200_Fc90_B1},
{HPF_Fs96000_Fc90_A2, /* 96kS/s coefficients */
HPF_Fs96000_Fc90_A1,
HPF_Fs96000_Fc90_A0,
-HPF_Fs96000_Fc90_B2,
-HPF_Fs96000_Fc90_B1},
+ {HPF_Fs176400_Fc90_A2, /* 176kS/s coefficients */
+ HPF_Fs176400_Fc90_A1,
+ HPF_Fs176400_Fc90_A0,
+ -HPF_Fs176400_Fc90_B2,
+ -HPF_Fs176400_Fc90_B1},
{HPF_Fs192000_Fc90_A2, /* 192kS/s coefficients */
HPF_Fs192000_Fc90_A1,
HPF_Fs192000_Fc90_A0,
@@ -320,9 +360,15 @@
-BPF_Fs48000_Fc55_B2,
-BPF_Fs48000_Fc55_B1},
#ifdef HIGHER_FS
+ {BPF_Fs88200_Fc55_A0, /* 88kS/s coefficients */
+ -BPF_Fs88200_Fc55_B2,
+ -BPF_Fs88200_Fc55_B1},
{BPF_Fs96000_Fc55_A0, /* 96kS/s coefficients */
-BPF_Fs96000_Fc55_B2,
-BPF_Fs96000_Fc55_B1},
+ {BPF_Fs176400_Fc55_A0, /* 176kS/s coefficients */
+ -BPF_Fs176400_Fc55_B2,
+ -BPF_Fs176400_Fc55_B1},
{BPF_Fs192000_Fc55_A0, /* 192kS/s coefficients */
-BPF_Fs192000_Fc55_B2,
-BPF_Fs192000_Fc55_B1},
@@ -357,9 +403,15 @@
-BPF_Fs48000_Fc66_B2,
-BPF_Fs48000_Fc66_B1},
#ifdef HIGHER_FS
+ {BPF_Fs88200_Fc66_A0, /* 88kS/s coefficients */
+ -BPF_Fs88200_Fc66_B2,
+ -BPF_Fs88200_Fc66_B1},
{BPF_Fs96000_Fc66_A0, /* 96kS/s coefficients */
-BPF_Fs96000_Fc66_B2,
-BPF_Fs96000_Fc66_B1},
+ {BPF_Fs176400_Fc66_A0, /* 176kS/s coefficients */
+ -BPF_Fs176400_Fc66_B2,
+ -BPF_Fs176400_Fc66_B1},
{BPF_Fs192000_Fc66_A0, /* 192kS/s coefficients */
-BPF_Fs192000_Fc66_B2,
-BPF_Fs192000_Fc66_B1},
@@ -394,9 +446,15 @@
-BPF_Fs48000_Fc78_B2,
-BPF_Fs48000_Fc78_B1},
#ifdef HIGHER_FS
+ {BPF_Fs88200_Fc66_A0, /* 88kS/s coefficients */
+ -BPF_Fs88200_Fc66_B2,
+ -BPF_Fs88200_Fc66_B1},
{BPF_Fs96000_Fc78_A0, /* 96kS/s coefficients */
-BPF_Fs96000_Fc78_B2,
-BPF_Fs96000_Fc78_B1},
+ {BPF_Fs176400_Fc66_A0, /* 176kS/s coefficients */
+ -BPF_Fs176400_Fc66_B2,
+ -BPF_Fs176400_Fc66_B1},
{BPF_Fs192000_Fc78_A0, /* 192kS/s coefficients */
-BPF_Fs192000_Fc78_B2,
-BPF_Fs192000_Fc78_B1},
@@ -432,9 +490,15 @@
-BPF_Fs48000_Fc90_B1}
#ifdef HIGHER_FS
,
+ {BPF_Fs88200_Fc90_A0, /* 88kS/s coefficients */
+ -BPF_Fs88200_Fc90_B2,
+ -BPF_Fs88200_Fc90_B1},
{BPF_Fs96000_Fc90_A0, /* 96kS/s coefficients */
-BPF_Fs96000_Fc90_B2,
-BPF_Fs96000_Fc90_B1},
+ {BPF_Fs176400_Fc90_A0, /* 176kS/s coefficients */
+ -BPF_Fs176400_Fc90_B2,
+ -BPF_Fs176400_Fc90_B1},
{BPF_Fs192000_Fc90_A0, /* 192kS/s coefficients */
-BPF_Fs192000_Fc90_B2,
-BPF_Fs192000_Fc90_B1}
@@ -466,7 +530,9 @@
AGC_ATTACK_Fs44100,
AGC_ATTACK_Fs48000
#ifdef HIGHER_FS
+ ,AGC_ATTACK_Fs88200
,AGC_ATTACK_Fs96000
+ ,AGC_ATTACK_Fs176400
,AGC_ATTACK_Fs192000
#endif
@@ -488,7 +554,9 @@
AGC_DECAY_Fs44100,
AGC_DECAY_Fs48000
#ifdef HIGHER_FS
+ ,AGC_DECAY_Fs88200
,AGC_DECAY_FS96000
+ ,AGC_DECAY_Fs176400
,AGC_DECAY_FS192000
#endif
@@ -583,7 +651,9 @@
VOL_TC_Fs44100,
VOL_TC_Fs48000
#ifdef HIGHER_FS
+ ,VOL_TC_Fs88200
,VOL_TC_Fs96000
+ ,VOL_TC_Fs176400
,VOL_TC_Fs192000
#endif
};
@@ -602,7 +672,9 @@
MIX_TC_Fs44100,
MIX_TC_Fs48000
#ifdef HIGHER_FS
+ ,MIX_TC_Fs88200
,MIX_TC_Fs96000
+ ,MIX_TC_Fs176400
,MIX_TC_Fs192000
#endif
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h b/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h
index 8c04847..bab4049 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h
@@ -487,6 +487,97 @@
#define HPF_Fs48000_Gain15_B2 0.000000
#ifdef HIGHER_FS
+/* Coefficients for sample rate 88200 */
+/* Gain = 1.000000 dB */
+#define HPF_Fs88200_Gain1_A0 1.094374f
+#define HPF_Fs88200_Gain1_A1 (-0.641256f)
+#define HPF_Fs88200_Gain1_A2 0.000000f
+#define HPF_Fs88200_Gain1_B1 (-0.546882f)
+#define HPF_Fs88200_Gain1_B2 0.000000f
+/* Gain = 2.000000 dB */
+#define HPF_Fs88200_Gain2_A0 1.200264f
+#define HPF_Fs88200_Gain2_A1 (-0.747146f)
+#define HPF_Fs88200_Gain2_A2 0.000000f
+#define HPF_Fs88200_Gain2_B1 (-0.546882f)
+#define HPF_Fs88200_Gain2_B2 0.000000f
+/* Gain = 3.000000 dB */
+#define HPF_Fs88200_Gain3_A0 1.319074f
+#define HPF_Fs88200_Gain3_A1 (-0.865956f)
+#define HPF_Fs88200_Gain3_A2 0.000000f
+#define HPF_Fs88200_Gain3_B1 (-0.546882f)
+#define HPF_Fs88200_Gain3_B2 0.000000f
+/* Gain = 4.000000 dB */
+#define HPF_Fs88200_Gain4_A0 1.452380f
+#define HPF_Fs88200_Gain4_A1 (-0.999263f)
+#define HPF_Fs88200_Gain4_A2 0.000000f
+#define HPF_Fs88200_Gain4_B1 (-0.546882f)
+#define HPF_Fs88200_Gain4_B2 0.000000f
+/* Gain = 5.000000 dB */
+#define HPF_Fs88200_Gain5_A0 1.601953f
+#define HPF_Fs88200_Gain5_A1 (-1.148836f)
+#define HPF_Fs88200_Gain5_A2 0.000000f
+#define HPF_Fs88200_Gain5_B1 (-0.546882f)
+#define HPF_Fs88200_Gain5_B2 0.000000f
+/* Gain = 6.000000 dB */
+#define HPF_Fs88200_Gain6_A0 1.769777f
+#define HPF_Fs88200_Gain6_A1 (-1.316659f)
+#define HPF_Fs88200_Gain6_A2 0.000000f
+#define HPF_Fs88200_Gain6_B1 (-0.546882f)
+#define HPF_Fs88200_Gain6_B2 0.000000f
+/* Gain = 7.000000 dB */
+#define HPF_Fs88200_Gain7_A0 1.958078f
+#define HPF_Fs88200_Gain7_A1 (-1.504960f)
+#define HPF_Fs88200_Gain7_A2 0.000000f
+#define HPF_Fs88200_Gain7_B1 (-0.546882f)
+#define HPF_Fs88200_Gain7_B2 0.000000f
+/* Gain = 8.000000 dB */
+#define HPF_Fs88200_Gain8_A0 2.169355f
+#define HPF_Fs88200_Gain8_A1 (-1.716238f)
+#define HPF_Fs88200_Gain8_A2 0.000000f
+#define HPF_Fs88200_Gain8_B1 (-0.546882f)
+#define HPF_Fs88200_Gain8_B2 0.000000f
+/* Gain = 9.000000 dB */
+#define HPF_Fs88200_Gain9_A0 2.406412f
+#define HPF_Fs88200_Gain9_A1 (-1.953295f)
+#define HPF_Fs88200_Gain9_A2 0.000000f
+#define HPF_Fs88200_Gain9_B1 (-0.546882f)
+#define HPF_Fs88200_Gain9_B2 0.000000f
+/* Gain = 10.000000 dB */
+#define HPF_Fs88200_Gain10_A0 2.672395f
+#define HPF_Fs88200_Gain10_A1 (-2.219277f)
+#define HPF_Fs88200_Gain10_A2 0.000000f
+#define HPF_Fs88200_Gain10_B1 (-0.546882f)
+#define HPF_Fs88200_Gain10_B2 0.000000f
+/* Gain = 11.000000 dB */
+#define HPF_Fs88200_Gain11_A0 2.970832f
+#define HPF_Fs88200_Gain11_A1 (-2.517714f)
+#define HPF_Fs88200_Gain11_A2 0.000000f
+#define HPF_Fs88200_Gain11_B1 (-0.546882f)
+#define HPF_Fs88200_Gain11_B2 0.000000f
+/* Gain = 12.000000 dB */
+#define HPF_Fs88200_Gain12_A0 3.305684f
+#define HPF_Fs88200_Gain12_A1 (-2.852566f)
+#define HPF_Fs88200_Gain12_A2 0.000000f
+#define HPF_Fs88200_Gain12_B1 (-0.546882f)
+#define HPF_Fs88200_Gain12_B2 0.000000f
+/* Gain = 13.000000 dB */
+#define HPF_Fs88200_Gain13_A0 3.681394f
+#define HPF_Fs88200_Gain13_A1 (-3.228276f)
+#define HPF_Fs88200_Gain13_A2 0.000000f
+#define HPF_Fs88200_Gain13_B1 (-0.546882f)
+#define HPF_Fs88200_Gain13_B2 0.000000f
+/* Gain = 14.000000 dB */
+#define HPF_Fs88200_Gain14_A0 4.102947f
+#define HPF_Fs88200_Gain14_A1 (-3.649830f)
+#define HPF_Fs88200_Gain14_A2 0.000000f
+#define HPF_Fs88200_Gain14_B1 (-0.546882f)
+#define HPF_Fs88200_Gain14_B2 0.000000f
+/* Gain = 15.000000 dB */
+#define HPF_Fs88200_Gain15_A0 4.575938f
+#define HPF_Fs88200_Gain15_A1 (-4.122820f)
+#define HPF_Fs88200_Gain15_A2 0.000000f
+#define HPF_Fs88200_Gain15_B1 (-0.546882f)
+#define HPF_Fs88200_Gain15_B2 0.000000f
/* Coefficients for sample rate 96000Hz */
/* Gain = 1.000000 dB */
@@ -580,6 +671,98 @@
#define HPF_Fs96000_Gain15_B1 (-0.577350)
#define HPF_Fs96000_Gain15_B2 0.000000
+/* Coefficients for sample rate 176400 */
+/* Gain = 1.000000 dB */
+#define HPF_Fs176400_Gain1_A0 1.106711f
+#define HPF_Fs176400_Gain1_A1 (-0.855807f)
+#define HPF_Fs176400_Gain1_A2 0.000000f
+#define HPF_Fs176400_Gain1_B1 (-0.749096f)
+#define HPF_Fs176400_Gain1_B2 0.000000f
+/* Gain = 2.000000 dB */
+#define HPF_Fs176400_Gain2_A0 1.226443f
+#define HPF_Fs176400_Gain2_A1 (-0.975539f)
+#define HPF_Fs176400_Gain2_A2 0.000000f
+#define HPF_Fs176400_Gain2_B1 (-0.749096f)
+#define HPF_Fs176400_Gain2_B2 0.000000f
+/* Gain = 3.000000 dB */
+#define HPF_Fs176400_Gain3_A0 1.360784f
+#define HPF_Fs176400_Gain3_A1 (-1.109880f)
+#define HPF_Fs176400_Gain3_A2 0.000000f
+#define HPF_Fs176400_Gain3_B1 (-0.749096f)
+#define HPF_Fs176400_Gain3_B2 0.000000f
+/* Gain = 4.000000 dB */
+#define HPF_Fs176400_Gain4_A0 1.511517f
+#define HPF_Fs176400_Gain4_A1 (-1.260613f)
+#define HPF_Fs176400_Gain4_A2 0.000000f
+#define HPF_Fs176400_Gain4_B1 (-0.749096f)
+#define HPF_Fs176400_Gain4_B2 0.000000f
+/* Gain = 5.000000 dB */
+#define HPF_Fs176400_Gain5_A0 1.680643f
+#define HPF_Fs176400_Gain5_A1 (-1.429739f)
+#define HPF_Fs176400_Gain5_A2 0.000000f
+#define HPF_Fs176400_Gain5_B1 (-0.749096f)
+#define HPF_Fs176400_Gain5_B2 0.000000f
+/* Gain = 6.000000 dB */
+#define HPF_Fs176400_Gain6_A0 1.870405f
+#define HPF_Fs176400_Gain6_A1 (-1.619501f)
+#define HPF_Fs176400_Gain6_A2 0.000000f
+#define HPF_Fs176400_Gain6_B1 (-0.749096f)
+#define HPF_Fs176400_Gain6_B2 0.000000f
+/* Gain = 7.000000 dB */
+#define HPF_Fs176400_Gain7_A0 2.083321f
+#define HPF_Fs176400_Gain7_A1 (-1.832417f)
+#define HPF_Fs176400_Gain7_A2 0.000000f
+#define HPF_Fs176400_Gain7_B1 (-0.749096f)
+#define HPF_Fs176400_Gain7_B2 0.000000f
+/* Gain = 8.000000 dB */
+#define HPF_Fs176400_Gain8_A0 2.322217f
+#define HPF_Fs176400_Gain8_A1 (-2.071313f)
+#define HPF_Fs176400_Gain8_A2 0.000000f
+#define HPF_Fs176400_Gain8_B1 (-0.749096f)
+#define HPF_Fs176400_Gain8_B2 0.000000f
+/* Gain = 9.000000 dB */
+#define HPF_Fs176400_Gain9_A0 2.590263f
+#define HPF_Fs176400_Gain9_A1 (-2.339359f)
+#define HPF_Fs176400_Gain9_A2 0.000000f
+#define HPF_Fs176400_Gain9_B1 (-0.749096f)
+#define HPF_Fs176400_Gain9_B2 0.000000f
+/* Gain = 10.000000 dB */
+#define HPF_Fs176400_Gain10_A0 2.891016f
+#define HPF_Fs176400_Gain10_A1 (-2.640112f)
+#define HPF_Fs176400_Gain10_A2 0.000000f
+#define HPF_Fs176400_Gain10_B1 (-0.749096f)
+#define HPF_Fs176400_Gain10_B2 0.000000f
+/* Gain = 11.000000 dB */
+#define HPF_Fs176400_Gain11_A0 3.228465f
+#define HPF_Fs176400_Gain11_A1 (-2.977561f)
+#define HPF_Fs176400_Gain11_A2 0.000000f
+#define HPF_Fs176400_Gain11_B1 (-0.749096f)
+#define HPF_Fs176400_Gain11_B2 0.000000f
+/* Gain = 12.000000 dB */
+#define HPF_Fs176400_Gain12_A0 3.607090f
+#define HPF_Fs176400_Gain12_A1 (-3.356186f)
+#define HPF_Fs176400_Gain12_A2 0.000000f
+#define HPF_Fs176400_Gain12_B1 (-0.749096f)
+#define HPF_Fs176400_Gain12_B2 0.000000f
+/* Gain = 13.000000 dB */
+#define HPF_Fs176400_Gain13_A0 4.031914f
+#define HPF_Fs176400_Gain13_A1 (-3.781010f)
+#define HPF_Fs176400_Gain13_A2 0.000000f
+#define HPF_Fs176400_Gain13_B1 (-0.749096f)
+#define HPF_Fs176400_Gain13_B2 0.000000f
+/* Gain = 14.000000 dB */
+#define HPF_Fs176400_Gain14_A0 4.508575f
+#define HPF_Fs176400_Gain14_A1 (-4.257671f)
+#define HPF_Fs176400_Gain14_A2 0.000000f
+#define HPF_Fs176400_Gain14_B1 (-0.749096f)
+#define HPF_Fs176400_Gain14_B2 0.000000f
+/* Gain = 15.000000 dB */
+#define HPF_Fs176400_Gain15_A0 5.043397f
+#define HPF_Fs176400_Gain15_A1 (-4.792493f)
+#define HPF_Fs176400_Gain15_A2 0.000000f
+#define HPF_Fs176400_Gain15_B1 (-0.749096f)
+#define HPF_Fs176400_Gain15_B2 0.000000f
+
/* Coefficients for sample rate 192000Hz */
/* Gain = 1.000000 dB */
#define HPF_Fs192000_Gain1_A0 1.107823
@@ -1216,4 +1399,4 @@
#endif
-#endif
\ No newline at end of file
+#endif
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c
index 7b85f23..62b4c73 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c
@@ -71,7 +71,8 @@
((pParams->SampleRate != LVM_FS_8000) && (pParams->SampleRate != LVM_FS_11025) && (pParams->SampleRate != LVM_FS_12000) &&
(pParams->SampleRate != LVM_FS_16000) && (pParams->SampleRate != LVM_FS_22050) && (pParams->SampleRate != LVM_FS_24000) &&
(pParams->SampleRate != LVM_FS_32000) && (pParams->SampleRate != LVM_FS_44100) && (pParams->SampleRate != LVM_FS_48000) &&
- (pParams->SampleRate != LVM_FS_96000) && (pParams->SampleRate != LVM_FS_192000)) ||
+ (pParams->SampleRate != LVM_FS_88200) && (pParams->SampleRate != LVM_FS_96000) &&
+ (pParams->SampleRate != LVM_FS_176400) && (pParams->SampleRate != LVM_FS_192000)) ||
#else
((pParams->SampleRate != LVM_FS_8000) && (pParams->SampleRate != LVM_FS_11025) && (pParams->SampleRate != LVM_FS_12000) &&
(pParams->SampleRate != LVM_FS_16000) && (pParams->SampleRate != LVM_FS_22050) && (pParams->SampleRate != LVM_FS_24000) &&
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c
index ade329b..0669a81 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c
@@ -233,7 +233,13 @@
* Set the capabilities
*/
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
- DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 | LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 | LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 | LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 | LVDBE_CAP_FS_48000 | LVDBE_CAP_FS_96000 | LVDBE_CAP_FS_192000;
+ DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 |
+ LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 |
+ LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 |
+ LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 |
+ LVDBE_CAP_FS_48000 | LVDBE_CAP_FS_88200 |
+ LVDBE_CAP_FS_96000 | LVDBE_CAP_FS_176400 |
+ LVDBE_CAP_FS_192000;
#else
DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 | LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 | LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 | LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 | LVDBE_CAP_FS_48000;
#endif
@@ -270,7 +276,13 @@
* Set the capabilities
*/
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
- EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 | LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 | LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 | LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 | LVEQNB_CAP_FS_48000 | LVEQNB_CAP_FS_96000 | LVEQNB_CAP_FS_192000;
+ EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 |
+ LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 |
+ LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 |
+ LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 |
+ LVEQNB_CAP_FS_48000 | LVEQNB_CAP_FS_88200 |
+ LVEQNB_CAP_FS_96000 | LVEQNB_CAP_FS_176400 |
+ LVEQNB_CAP_FS_192000;
#else
EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 | LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 | LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 | LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 | LVEQNB_CAP_FS_48000;
#endif
@@ -747,7 +759,13 @@
* Set the initialisation capabilities
*/
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
- DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 | LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 | LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 | LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 | LVDBE_CAP_FS_48000 | LVDBE_CAP_FS_96000 | LVDBE_CAP_FS_192000;
+ DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 |
+ LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 |
+ LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 |
+ LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 |
+ LVDBE_CAP_FS_48000 | LVDBE_CAP_FS_88200 |
+ LVDBE_CAP_FS_96000 | LVDBE_CAP_FS_176400 |
+ LVDBE_CAP_FS_192000;
#else
DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 | LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 | LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 | LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 | LVDBE_CAP_FS_48000;
#endif
@@ -805,7 +823,13 @@
* Set the initialisation capabilities
*/
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
- EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 | LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 | LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 | LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 | LVEQNB_CAP_FS_48000 | LVEQNB_CAP_FS_96000 | LVEQNB_CAP_FS_192000;
+ EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 |
+ LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 |
+ LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 |
+ LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 |
+ LVEQNB_CAP_FS_48000 | LVEQNB_CAP_FS_88200 |
+ LVEQNB_CAP_FS_96000 | LVEQNB_CAP_FS_176400 |
+ LVEQNB_CAP_FS_192000;
#else
EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 | LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 | LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 | LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 | LVEQNB_CAP_FS_48000;
#endif
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.c
index 199ddde..a5356d2 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.c
@@ -269,6 +269,53 @@
-HPF_Fs48000_Gain15_B1}
#ifdef HIGHER_FS
,
+ /* 88kHz Sampling rate */
+ {HPF_Fs88200_Gain1_A1, /* Gain Setting 1 */
+ HPF_Fs88200_Gain1_A0,
+ -HPF_Fs88200_Gain1_B1},
+ {HPF_Fs88200_Gain2_A1, /* Gain Setting 2 */
+ HPF_Fs88200_Gain2_A0,
+ -HPF_Fs88200_Gain2_B1},
+ {HPF_Fs88200_Gain3_A1, /* Gain Setting 3 */
+ HPF_Fs88200_Gain3_A0,
+ -HPF_Fs88200_Gain3_B1},
+ {HPF_Fs88200_Gain4_A1, /* Gain Setting 4 */
+ HPF_Fs88200_Gain4_A0,
+ -HPF_Fs88200_Gain4_B1},
+ {HPF_Fs88200_Gain5_A1, /* Gain Setting 5 */
+ HPF_Fs88200_Gain5_A0,
+ -HPF_Fs88200_Gain5_B1},
+ {HPF_Fs88200_Gain6_A1, /* Gain Setting 6 */
+ HPF_Fs88200_Gain6_A0,
+ -HPF_Fs88200_Gain6_B1},
+ {HPF_Fs88200_Gain7_A1, /* Gain Setting 7 */
+ HPF_Fs88200_Gain7_A0,
+ -HPF_Fs88200_Gain7_B1},
+ {HPF_Fs88200_Gain8_A1, /* Gain Setting 8 */
+ HPF_Fs88200_Gain8_A0,
+ -HPF_Fs88200_Gain8_B1},
+ {HPF_Fs88200_Gain9_A1, /* Gain Setting 9 */
+ HPF_Fs88200_Gain9_A0,
+ -HPF_Fs88200_Gain9_B1},
+ {HPF_Fs88200_Gain10_A1, /* Gain Setting 10 */
+ HPF_Fs88200_Gain10_A0,
+ -HPF_Fs88200_Gain10_B1},
+ {HPF_Fs88200_Gain11_A1, /* Gain Setting 11 */
+ HPF_Fs88200_Gain11_A0,
+ -HPF_Fs88200_Gain11_B1},
+ {HPF_Fs88200_Gain12_A1, /* Gain Setting 12 */
+ HPF_Fs88200_Gain12_A0,
+ -HPF_Fs88200_Gain12_B1},
+ {HPF_Fs88200_Gain13_A1, /* Gain Setting 13 */
+ HPF_Fs88200_Gain13_A0,
+ -HPF_Fs88200_Gain13_B1},
+ {HPF_Fs88200_Gain14_A1, /* Gain Setting 14 */
+ HPF_Fs88200_Gain14_A0,
+ -HPF_Fs88200_Gain14_B1},
+ {HPF_Fs88200_Gain15_A1, /* Gain Setting 15 */
+ HPF_Fs88200_Gain15_A0,
+ -HPF_Fs88200_Gain15_B1},
+
/* 96kHz sampling rate */
{HPF_Fs96000_Gain1_A1, /* Gain setting 1 */
HPF_Fs96000_Gain1_A0,
@@ -316,6 +363,53 @@
HPF_Fs96000_Gain15_A0,
-HPF_Fs96000_Gain15_B1},
+ /* 176kHz Sampling rate */
+ {HPF_Fs176400_Gain1_A1, /* Gain Setting 1 */
+ HPF_Fs176400_Gain1_A0,
+ -HPF_Fs176400_Gain1_B1},
+ {HPF_Fs176400_Gain2_A1, /* Gain Setting 2 */
+ HPF_Fs176400_Gain2_A0,
+ -HPF_Fs176400_Gain2_B1},
+ {HPF_Fs176400_Gain3_A1, /* Gain Setting 3 */
+ HPF_Fs176400_Gain3_A0,
+ -HPF_Fs176400_Gain3_B1},
+ {HPF_Fs176400_Gain4_A1, /* Gain Setting 4 */
+ HPF_Fs176400_Gain4_A0,
+ -HPF_Fs176400_Gain4_B1},
+ {HPF_Fs176400_Gain5_A1, /* Gain Setting 5 */
+ HPF_Fs176400_Gain5_A0,
+ -HPF_Fs176400_Gain5_B1},
+ {HPF_Fs176400_Gain6_A1, /* Gain Setting 6 */
+ HPF_Fs176400_Gain6_A0,
+ -HPF_Fs176400_Gain6_B1},
+ {HPF_Fs176400_Gain7_A1, /* Gain Setting 7 */
+ HPF_Fs176400_Gain7_A0,
+ -HPF_Fs176400_Gain7_B1},
+ {HPF_Fs176400_Gain8_A1, /* Gain Setting 8 */
+ HPF_Fs176400_Gain8_A0,
+ -HPF_Fs176400_Gain8_B1},
+ {HPF_Fs176400_Gain9_A1, /* Gain Setting 9 */
+ HPF_Fs176400_Gain9_A0,
+ -HPF_Fs176400_Gain9_B1},
+ {HPF_Fs176400_Gain10_A1, /* Gain Setting 10 */
+ HPF_Fs176400_Gain10_A0,
+ -HPF_Fs176400_Gain10_B1},
+ {HPF_Fs176400_Gain11_A1, /* Gain Setting 11 */
+ HPF_Fs176400_Gain11_A0,
+ -HPF_Fs176400_Gain11_B1},
+ {HPF_Fs176400_Gain12_A1, /* Gain Setting 12 */
+ HPF_Fs176400_Gain12_A0,
+ -HPF_Fs176400_Gain12_B1},
+ {HPF_Fs176400_Gain13_A1, /* Gain Setting 13 */
+ HPF_Fs176400_Gain13_A0,
+ -HPF_Fs176400_Gain13_B1},
+ {HPF_Fs176400_Gain14_A1, /* Gain Setting 14 */
+ HPF_Fs176400_Gain14_A0,
+ -HPF_Fs176400_Gain14_B1},
+ {HPF_Fs176400_Gain15_A1, /* Gain Setting 15 */
+ HPF_Fs176400_Gain15_A0,
+ -HPF_Fs176400_Gain15_B1},
+
/* 192kHz sampling rate */
{HPF_Fs192000_Gain1_A1, /* Gain setting 1 */
HPF_Fs192000_Gain1_A0,
diff --git a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
index 303b62d..59586e0 100644
--- a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
+++ b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
@@ -169,8 +169,10 @@
LVM_FS_44100 = 7,
LVM_FS_48000 = 8,
#ifdef HIGHER_FS
- LVM_FS_96000 = 9,
- LVM_FS_192000 = 10,
+ LVM_FS_88200 = 9,
+ LVM_FS_96000 = 10,
+ LVM_FS_176400 = 11,
+ LVM_FS_192000 = 12,
#endif
LVM_FS_INVALID = LVM_MAXENUM-1,
LVM_FS_DUMMY = LVM_MAXENUM
diff --git a/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h b/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
index e7fdbf6..385dbcf 100644
--- a/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
+++ b/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
@@ -201,8 +201,10 @@
#define LVEQNB_CAP_FS_44100 128
#define LVEQNB_CAP_FS_48000 256
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
-#define LVEQNB_CAP_FS_96000 512
-#define LVEQNB_CAP_FS_192000 1024
+#define LVEQNB_CAP_FS_88200 512
+#define LVEQNB_CAP_FS_96000 1024
+#define LVEQNB_CAP_FS_176400 2048
+#define LVEQNB_CAP_FS_192000 4096
#endif
typedef enum
@@ -217,8 +219,10 @@
LVEQNB_FS_44100 = 7,
LVEQNB_FS_48000 = 8,
#ifdef HIGHER_FS
- LVEQNB_FS_96000 = 9,
- LVEQNB_FS_192000 = 10,
+ LVEQNB_FS_88200 = 9,
+ LVEQNB_FS_96000 = 10,
+ LVEQNB_FS_176400 = 11,
+ LVEQNB_FS_192000 = 12,
#endif
LVEQNB_FS_MAX = LVM_MAXINT_32
} LVEQNB_Fs_en;
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h
index 42ea46f..755141e 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h
@@ -109,7 +109,9 @@
#define LVEQNB_2PiOn_48000 0.000131f
#ifdef HIGHER_FS
+#define LVEQNB_2PiOn_88200 0.000071f
#define LVEQNB_2PiOn_96000 0.000065f
+#define LVEQNB_2PiOn_176400 0.000036f
#define LVEQNB_2PiOn_192000 0.000033f
#endif
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.c
index 563181c..453c42d 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.c
@@ -46,7 +46,9 @@
32000,
44100,
48000,
+ 88200,
96000,
+ 176400,
192000
};
#else
@@ -82,7 +84,9 @@
LVEQNB_2PiOn_44100,
LVEQNB_2PiOn_48000
#ifdef HIGHER_FS
+ ,LVEQNB_2PiOn_88200
,LVEQNB_2PiOn_96000
+ ,LVEQNB_2PiOn_176400
,LVEQNB_2PiOn_192000
#endif
};
@@ -249,30 +253,4 @@
16586, /* a2 */
-44}; /* a3 */
-/************************************************************************************/
-/* */
-/* Bypass mixer time constants (100ms) */
-/* */
-/************************************************************************************/
-#define LVEQNB_MIX_TC_Fs8000 32580 /* Floating point value 0.994262695 */
-#define LVEQNB_MIX_TC_Fs11025 32632 /* Floating point value 0.995849609 */
-#define LVEQNB_MIX_TC_Fs12000 32643 /* Floating point value 0.996185303 */
-#define LVEQNB_MIX_TC_Fs16000 32674 /* Floating point value 0.997131348 */
-#define LVEQNB_MIX_TC_Fs22050 32700 /* Floating point value 0.997924805 */
-#define LVEQNB_MIX_TC_Fs24000 32705 /* Floating point value 0.998077393 */
-#define LVEQNB_MIX_TC_Fs32000 32721 /* Floating point value 0.998565674 */
-#define LVEQNB_MIX_TC_Fs44100 32734 /* Floating point value 0.998962402 */
-#define LVEQNB_MIX_TC_Fs48000 32737 /* Floating point value 0.999053955 */
-
-
-const LVM_INT16 LVEQNB_MixerTCTable[] = {
- LVEQNB_MIX_TC_Fs8000,
- LVEQNB_MIX_TC_Fs11025,
- LVEQNB_MIX_TC_Fs12000,
- LVEQNB_MIX_TC_Fs16000,
- LVEQNB_MIX_TC_Fs22050,
- LVEQNB_MIX_TC_Fs24000,
- LVEQNB_MIX_TC_Fs32000,
- LVEQNB_MIX_TC_Fs44100,
- LVEQNB_MIX_TC_Fs48000};
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_Private.h b/media/libeffects/lvm/lib/Reverb/src/LVREV_Private.h
index ff7475e..c915ac0 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_Private.h
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_Private.h
@@ -123,7 +123,7 @@
#ifndef HIGHER_FS
#define LVREV_NUM_FS 9 /* Number of supported sample rates */
#else
-#define LVREV_NUM_FS 11 /* Number of supported sample rates */
+#define LVREV_NUM_FS 13 /* Number of supported sample rates */
#endif
#define LVREV_MAXBLKSIZE_LIMIT 64 /* Maximum block size low limit */
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.c
index 8c7807f..dfed28e 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.c
@@ -68,7 +68,8 @@
(pNewParams->SampleRate != LVM_FS_44100) &&
(pNewParams->SampleRate != LVM_FS_48000)
#ifdef HIGHER_FS
- && (pNewParams->SampleRate != LVM_FS_96000) && (pNewParams->SampleRate != LVM_FS_192000)
+ && (pNewParams->SampleRate != LVM_FS_88200) && (pNewParams->SampleRate != LVM_FS_96000)
+ && (pNewParams->SampleRate != LVM_FS_176400) && (pNewParams->SampleRate != LVM_FS_192000)
#endif
)
#ifdef SUPPORT_MC
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.c
index b3edc60..1058740 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.c
@@ -52,7 +52,9 @@
32000,
44100,
48000,
+ 88200,
96000,
+ 176400,
192000
};
#endif
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Private.h b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Private.h
index a750bb0..ee07e2e 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Private.h
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Private.h
@@ -46,7 +46,7 @@
#ifndef HIGHER_FS
#define LVPSA_NR_SUPPORTED_RATE 9 /* From 8000Hz to 48000Hz*/
#else
-#define LVPSA_NR_SUPPORTED_RATE 11 /* From 8000Hz to 192000Hz*/
+#define LVPSA_NR_SUPPORTED_RATE 13 /* From 8000Hz to 192000Hz*/
#endif
#define LVPSA_NR_SUPPORTED_SPEED 3 /* LOW, MEDIUM, HIGH */
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.c
index 1287503..f8af496 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.c
@@ -54,7 +54,9 @@
32000,
44100,
48000,
+ 88200,
96000,
+ 176400,
192000}; /* 192kS/s */
#endif
@@ -78,7 +80,9 @@
48696,
44739
#ifdef HIGHER_FS
+ ,24348
,22369
+ ,12174
,11185 /* 192kS/s */
#endif
};
@@ -105,7 +109,9 @@
882,
960
#ifdef HIGHER_FS
+ ,1764
,1920
+ ,3528
,3840 /* 192kS/s */
#endif
};
@@ -128,7 +134,9 @@
30, /* 44100 S/s */
32 /* 48000 S/s */
#ifdef HIGHER_FS
+ ,60 /* 88200 S/s */
,64 /* 96000 S/s */
+ ,120 /* 176400 S/s */
,128 /*192000 S/s */
#endif
};
@@ -153,7 +161,9 @@
4781,
4392
#ifdef HIGHER_FS
+ ,2390
,2196
+ ,1195
,1098 /* 192kS/s */
#endif
};
@@ -169,7 +179,9 @@
0.1459089f,
0.1340372f
#ifdef HIGHER_FS
+ ,0.0729476f
,0.0670186f
+ ,0.0364738f
,0.0335093f /* 192kS/s */
#endif
};
@@ -352,7 +364,9 @@
/* 48kS/s */
{-0.9932638457976282f,0.0066249934025109f},
#ifdef HIGHER_FS
+ {-0.9931269618682563f,0.0067592649720609f},
{-0.9932638457976282f,0.0066249934025109f},
+ {-0.9931269618682563f,0.0067592649720609f},
{-0.9932638457976282f,0.0066249934025109f},
#endif
/* 8kS/s */ /* LVPSA_SPEED_MEDIUM */
@@ -368,7 +382,9 @@
/* 48kS/s */
{-0.9540119562298059f,0.0445343819446862f},
#ifdef HIGHER_FS
+ {-0.9531011912040412f,0.0453995238058269f},
{-0.9540119562298059f,0.0445343819446862f},
+ {-0.9531011912040412f,0.0453995238058269f},
{-0.9540119562298059f,0.0445343819446862f},
#endif
/* 8kS/s */ /* LVPSA_SPEED_HIGH */
@@ -383,7 +399,9 @@
/* 48kS/s */
{-0.7274807319045067f,0.2356666540727019f}
#ifdef HIGHER_FS
+ ,{-0.7229706319049001f,0.2388987224549055f}
,{-0.7274807319045067f,0.2356666540727019f}
+ ,{-0.7229706319049001f,0.2388987224549055f}
,{-0.7274807319045067f,0.2356666540727019f}
#endif
};
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
index 0c2fe53..277d95c 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
@@ -152,6 +152,24 @@
#define CS_SIDE_48000_SCALE 14
#ifdef HIGHER_FS
+/* Coefficients for 88200Hz sample rate.
+ * The filter coefficients are obtained by carrying out
+ * state-space analysis using the coefficients available
+ * for 44100Hz.
+ */
+#define CS_MIDDLE_88200_A0 0.233846f
+#define CS_MIDDLE_88200_A1 (-0.232657f)
+#define CS_MIDDLE_88200_A2 0.000000f
+#define CS_MIDDLE_88200_B1 (-0.992747f)
+#define CS_MIDDLE_88200_B2 0.000000f
+#define CS_MIDDLE_88200_SCALE 15
+#define CS_SIDE_88200_A0 0.231541f
+#define CS_SIDE_88200_A1 (-0.289586f)
+#define CS_SIDE_88200_A2 0.058045f
+#define CS_SIDE_88200_B1 (-1.765300f)
+#define CS_SIDE_88200_B2 0.769816f
+#define CS_SIDE_88200_SCALE 14
+
/* Stereo Enhancer coefficients for 96000Hz sample rate, scaled with 0.165*/
/* high pass filter with cutoff frequency 102.18 Hz*/
#define CS_MIDDLE_96000_A0 0.235532
@@ -168,6 +186,24 @@
#define CS_SIDE_96000_B2 0.797236
#define CS_SIDE_96000_SCALE 14
+/* Stereo Enhancer coefficients for 176400Hz sample rate.
+ * The filter coefficients are obtained by carrying out
+ * state-space analysis using the coefficients available
+ * for 44100Hz.
+ */
+#define CS_MIDDLE_176400_A0 0.233973f
+#define CS_MIDDLE_176400_A1 (-0.233378f)
+#define CS_MIDDLE_176400_A2 0.000000f
+#define CS_MIDDLE_176400_B1 (-0.996367f)
+#define CS_MIDDLE_176400_B2 0.000000f
+#define CS_MIDDLE_176400_SCALE 15
+#define CS_SIDE_176400_A0 0.199836f
+#define CS_SIDE_176400_A1 (-0.307544f)
+#define CS_SIDE_176400_A2 0.107708f
+#define CS_SIDE_176400_B1 (-1.876572f)
+#define CS_SIDE_176400_B2 0.877771f
+#define CS_SIDE_176400_SCALE 14
+
/* Stereo Enhancer coefficients for 192000Hz sample rate, scaled with 0.1689*/
#define CS_MIDDLE_192000_A0 0.241219
#define CS_MIDDLE_192000_A1 (-0.240656)
@@ -199,7 +235,13 @@
#define LVCS_STEREODELAY_CS_24KHZ 279 /* Sample rate 24kS/s */
#define LVCS_STEREODELAY_CS_32KHZ 372 /* Sample rate 32kS/s */
#define LVCS_STEREODELAY_CS_44KHZ 512 /* Sample rate 44kS/s */
+// TODO: this should linearly scale by frequency but is limited to 512 frames until
+// we ensure enough buffer size has been allocated.
#define LVCS_STEREODELAY_CS_48KHZ 512 /* Sample rate 48kS/s */
+#define LVCS_STEREODELAY_CS_88KHZ 512 /* Sample rate 88.2kS/s */
+#define LVCS_STEREODELAY_CS_96KHZ 512 /* Sample rate 96kS/s */
+#define LVCS_STEREODELAY_CS_176KHZ 512 /* Sample rate 176.4kS/s */
+#define LVCS_STEREODELAY_CS_192KHZ 512 /* Sample rate 196kS/s */
/* Reverb coefficients for 8000 Hz sample rate, scaled with 1.038030 */
#define CS_REVERB_8000_A0 0.667271
@@ -275,6 +317,14 @@
#define CS_REVERB_48000_SCALE 14
#ifdef HIGHER_FS
+/* Reverb coefficients for 88200Hz sample rate, scaled with 0.8 */
+/* Band pass filter with fc1=500 and fc2=8000 */
+#define CS_REVERB_88200_A0 0.171901f
+#define CS_REVERB_88200_A1 0.000000f
+#define CS_REVERB_88200_A2 (-0.171901f)
+#define CS_REVERB_88200_B1 (-1.553948f)
+#define CS_REVERB_88200_B2 (0.570248f)
+#define CS_REVERB_88200_SCALE 14
/* Reverb coefficients for 96000Hz sample rate, scaled with 0.8 */
/* Band pass filter with fc1=500 and fc2=8000*/
#define CS_REVERB_96000_A0 0.1602488
@@ -284,6 +334,14 @@
#define CS_REVERB_96000_B2 0.599377
#define CS_REVERB_96000_SCALE 14
+/* Reverb coefficients for 176400Hz sample rate, scaled with 0.8 */
+/* Band pass filter with fc1=500 and fc2=8000 */
+#define CS_REVERB_176400_A0 0.094763f
+#define CS_REVERB_176400_A1 0.000000f
+#define CS_REVERB_176400_A2 (-0.094763f)
+#define CS_REVERB_176400_B1 (-1.758593f)
+#define CS_REVERB_176400_B2 (0.763091f)
+#define CS_REVERB_176400_SCALE 14
/* Reverb coefficients for 192000Hz sample rate, scaled with 0.8 */
/* Band pass filter with fc1=500 and fc2=8000*/
#define CS_REVERB_192000_A0 0.0878369
@@ -446,6 +504,24 @@
#ifdef HIGHER_FS
+/* Equaliser coefficients for 88200Hz sample rate.
+ * The filter coefficients are obtained by carrying out
+ * state-space analysis using the coefficients available
+ * for 44100Hz.
+ */
+#define CS_EQUALISER_88200_A0 1.771899f
+#define CS_EQUALISER_88200_A1 (-2.930762f)
+#define CS_EQUALISER_88200_A2 1.172175f
+#define CS_EQUALISER_88200_B1 (-1.438349f)
+#define CS_EQUALISER_88200_B2 0.442520f
+#define CS_EQUALISER_88200_SCALE 13
+#define CSEX_EQUALISER_88200_A0 2.675241f
+#define CSEX_EQUALISER_88200_A1 (-4.466154f)
+#define CSEX_EQUALISER_88200_A2 1.810305f
+#define CSEX_EQUALISER_88200_B1 (-0.925350f)
+#define CSEX_EQUALISER_88200_B2 (-0.066616f)
+#define CSEX_EQUALISER_88200_SCALE 13
+
#define CS_EQUALISER_96000_A0 1.784497
#define CS_EQUALISER_96000_A1 (-3.001435)
#define CS_EQUALISER_96000_A2 1.228422
@@ -458,6 +534,23 @@
#define CSEX_EQUALISER_96000_B1 (-0.971718)
#define CSEX_EQUALISER_96000_B2 (-0.021216)
#define CSEX_EQUALISER_96000_SCALE 13
+/* Equaliser coefficients for 176400Hz sample rate.
+ * The filter coefficients are obtained by carrying out
+ * state-space analysis using the coefficients available
+ * for 44100Hz.
+ */
+#define CS_EQUALISER_176400_A0 1.883440f
+#define CS_EQUALISER_176400_A1 (-3.414272f)
+#define CS_EQUALISER_176400_A2 1.534702f
+#define CS_EQUALISER_176400_B1 (-1.674614f)
+#define CS_EQUALISER_176400_B2 0.675827f
+#define CS_EQUALISER_176400_SCALE 13
+#define CSEX_EQUALISER_176400_A0 3.355068f
+#define CSEX_EQUALISER_176400_A1 (-6.112578f)
+#define CSEX_EQUALISER_176400_A2 2.764135f
+#define CSEX_EQUALISER_176400_B1 (-1.268533f)
+#define CSEX_EQUALISER_176400_B2 0.271277f
+#define CSEX_EQUALISER_176400_SCALE 13
#define CS_EQUALISER_192000_A0 1.889582
#define CS_EQUALISER_192000_A1 (-3.456140)
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.c
index 0765764..a1fb48f 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.c
@@ -74,10 +74,18 @@
(LVM_UINT16 )CS_MIDDLE_48000_SCALE}
#ifdef HIGHER_FS
,
+ {CS_MIDDLE_88200_A0, /* 88kS/s coefficients */
+ CS_MIDDLE_88200_A1,
+ CS_MIDDLE_88200_B1,
+ (LVM_UINT16)CS_MIDDLE_88200_SCALE},
{CS_MIDDLE_96000_A0, /* 96kS/s coefficients */
CS_MIDDLE_96000_A1,
CS_MIDDLE_96000_B1,
(LVM_UINT16 )CS_MIDDLE_96000_SCALE},
+ {CS_MIDDLE_176400_A0, /* 176kS/s coefficients */
+ CS_MIDDLE_176400_A1,
+ CS_MIDDLE_176400_B1,
+ (LVM_UINT16)CS_MIDDLE_176400_SCALE},
{CS_MIDDLE_192000_A0, /* 192kS/s coefficients */
CS_MIDDLE_192000_A1,
CS_MIDDLE_192000_B1,
@@ -144,12 +152,24 @@
(LVM_UINT16 )CS_SIDE_48000_SCALE}
#ifdef HIGHER_FS
,
+ {CS_SIDE_88200_A0, /* 88kS/s coefficients */
+ CS_SIDE_88200_A1,
+ CS_SIDE_88200_A2,
+ CS_SIDE_88200_B1,
+ CS_SIDE_88200_B2,
+ (LVM_UINT16)CS_SIDE_88200_SCALE},
{CS_SIDE_96000_A0, /* 96kS/s coefficients */
CS_SIDE_96000_A1,
CS_SIDE_96000_A2,
CS_SIDE_96000_B1,
CS_SIDE_96000_B2,
(LVM_UINT16 )CS_SIDE_96000_SCALE},
+ {CS_SIDE_176400_A0, /*176kS/s coefficients */
+ CS_SIDE_176400_A1,
+ CS_SIDE_176400_A2,
+ CS_SIDE_176400_B1,
+ CS_SIDE_176400_B2,
+ (LVM_UINT16)CS_SIDE_176400_SCALE},
{CS_SIDE_192000_A0, /* 192kS/s coefficients */
CS_SIDE_192000_A1,
CS_SIDE_192000_A2,
@@ -223,12 +243,24 @@
CS_EQUALISER_48000_B2,
(LVM_UINT16 )CS_EQUALISER_48000_SCALE},
#ifdef HIGHER_FS
+ {CS_EQUALISER_88200_A0, /* 88kS/s coeffieients */
+ CS_EQUALISER_88200_A1,
+ CS_EQUALISER_88200_A2,
+ CS_EQUALISER_88200_B1,
+ CS_EQUALISER_88200_B2,
+ (LVM_UINT16)CS_EQUALISER_88200_SCALE},
{CS_EQUALISER_96000_A0, /* 96kS/s coefficients */
CS_EQUALISER_96000_A1,
CS_EQUALISER_96000_A2,
CS_EQUALISER_96000_B1,
CS_EQUALISER_96000_B2,
(LVM_UINT16 )CS_EQUALISER_96000_SCALE},
+ {CS_EQUALISER_176400_A0, /* 176kS/s coefficients */
+ CS_EQUALISER_176400_A1,
+ CS_EQUALISER_176400_A2,
+ CS_EQUALISER_176400_B1,
+ CS_EQUALISER_176400_B2,
+ (LVM_UINT16)CS_EQUALISER_176400_SCALE},
{CS_EQUALISER_192000_A0, /* 192kS/s coefficients */
CS_EQUALISER_192000_A1,
CS_EQUALISER_192000_A2,
@@ -294,12 +326,24 @@
(LVM_UINT16 )CSEX_EQUALISER_48000_SCALE}
#ifdef HIGHER_FS
,
+ {CSEX_EQUALISER_88200_A0, /* 88kS/s coefficients */
+ CSEX_EQUALISER_88200_A1,
+ CSEX_EQUALISER_88200_A2,
+ CSEX_EQUALISER_88200_B1,
+ CSEX_EQUALISER_88200_B2,
+ (LVM_UINT16)CSEX_EQUALISER_88200_SCALE},
{CSEX_EQUALISER_96000_A0, /* 96kS/s coefficients */
CSEX_EQUALISER_96000_A1,
CSEX_EQUALISER_96000_A2,
CSEX_EQUALISER_96000_B1,
CSEX_EQUALISER_96000_B2,
(LVM_UINT16 )CSEX_EQUALISER_96000_SCALE},
+ {CSEX_EQUALISER_176400_A0, /* 176kS/s coefficients */
+ CSEX_EQUALISER_176400_A1,
+ CSEX_EQUALISER_176400_A2,
+ CSEX_EQUALISER_176400_B1,
+ CSEX_EQUALISER_176400_B2,
+ (LVM_UINT16)CSEX_EQUALISER_176400_SCALE},
{CSEX_EQUALISER_192000_A0, /* 192kS/s coefficients */
CSEX_EQUALISER_192000_A1,
CSEX_EQUALISER_192000_A2,
@@ -326,7 +370,12 @@
LVCS_STEREODELAY_CS_24KHZ,
LVCS_STEREODELAY_CS_32KHZ,
LVCS_STEREODELAY_CS_44KHZ,
- LVCS_STEREODELAY_CS_48KHZ};
+ LVCS_STEREODELAY_CS_48KHZ,
+ LVCS_STEREODELAY_CS_88KHZ,
+ LVCS_STEREODELAY_CS_96KHZ,
+ LVCS_STEREODELAY_CS_176KHZ,
+ LVCS_STEREODELAY_CS_192KHZ,
+};
/************************************************************************************/
/* */
@@ -392,12 +441,24 @@
(LVM_UINT16 )CS_REVERB_48000_SCALE}
#ifdef HIGHER_FS
,
+ {CS_REVERB_88200_A0, /* 88kS/s coefficients */
+ CS_REVERB_88200_A1,
+ CS_REVERB_88200_A2,
+ CS_REVERB_88200_B1,
+ CS_REVERB_88200_B2,
+ (LVM_UINT16)CS_REVERB_88200_SCALE},
{CS_REVERB_96000_A0, /* 96kS/s coefficients */
CS_REVERB_96000_A1,
CS_REVERB_96000_A2,
CS_REVERB_96000_B1,
CS_REVERB_96000_B2,
(LVM_UINT16 )CS_REVERB_96000_SCALE},
+ {CS_REVERB_176400_A0, /* 176kS/s coefficients */
+ CS_REVERB_176400_A1,
+ CS_REVERB_176400_A2,
+ CS_REVERB_176400_B1,
+ CS_REVERB_176400_B2,
+ (LVM_UINT16)CS_REVERB_176400_SCALE},
{CS_REVERB_192000_A0, /* 192kS/s coefficients */
CS_REVERB_192000_A1,
CS_REVERB_192000_A2,
@@ -509,12 +570,14 @@
#define LVCS_VOL_TC_Fs44100 32734 /* Floating point value 0.998962402 */
#define LVCS_VOL_TC_Fs48000 32737 /* Floating point value 0.999053955 */
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
+#define LVCS_VOL_TC_Fs88200 32751 /* Floating point value 0.999481066 */
#define LVCS_VOL_TC_Fs96000 32751 /* Floating point value 0.999511703 */ /* Todo @ need to re check this value*/
+#define LVCS_VOL_TC_Fs176400 32759 /* Floating point value 0.999740499 */
#define LVCS_VOL_TC_Fs192000 32763 /* Floating point value 0.999877925 */ /* Todo @ need to re check this value*/
#endif
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
-const LVM_INT16 LVCS_VolumeTCTable[11] = {LVCS_VOL_TC_Fs8000,
+const LVM_INT16 LVCS_VolumeTCTable[13] = {LVCS_VOL_TC_Fs8000,
LVCS_VOL_TC_Fs11025,
LVCS_VOL_TC_Fs12000,
LVCS_VOL_TC_Fs16000,
@@ -523,7 +586,9 @@
LVCS_VOL_TC_Fs32000,
LVCS_VOL_TC_Fs44100,
LVCS_VOL_TC_Fs48000,
+ LVCS_VOL_TC_Fs88200,
LVCS_VOL_TC_Fs96000,
+ LVCS_VOL_TC_Fs176400,
LVCS_VOL_TC_Fs192000
};
#else
@@ -545,7 +610,7 @@
/* */
/************************************************************************************/
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
-const LVM_INT32 LVCS_SampleRateTable[11] = {8000,
+const LVM_INT32 LVCS_SampleRateTable[13] = {8000,
11025,
12000,
16000,
@@ -554,7 +619,9 @@
32000,
44100,
48000,
+ 88200,
96000,
+ 176400,
192000
};
#else
diff --git a/media/libeffects/lvm/tests/lvmtest.cpp b/media/libeffects/lvm/tests/lvmtest.cpp
index 01c5955..99551cc 100644
--- a/media/libeffects/lvm/tests/lvmtest.cpp
+++ b/media/libeffects/lvm/tests/lvmtest.cpp
@@ -447,19 +447,69 @@
lvmConfigParams_t *plvmConfigParams,
LVM_ControlParams_t *params) {
LVM_ReturnStatus_en LvmStatus = LVM_SUCCESS; /* Function call status */
- LVM_EQNB_BandDef_t BandDefs[MAX_NUM_BANDS]; /* Equaliser band definitions */
- int eqPresetLevel = plvmConfigParams->eqPresetLevel;
- int nrChannels = plvmConfigParams->nrChannels;
- params->NrChannels = nrChannels;
/* Set the initial process parameters */
/* General parameters */
params->OperatingMode = LVM_MODE_ON;
- params->SampleRate = LVM_FS_44100;
- params->SourceFormat = LVM_STEREO;
params->SpeakerType = LVM_HEADPHONES;
- pContext->pBundledContext->SampleRate = LVM_FS_44100;
+ const int nrChannels = plvmConfigParams->nrChannels;
+ params->NrChannels = nrChannels;
+ if (nrChannels == 1) {
+ params->SourceFormat = LVM_MONO;
+ } else if (nrChannels == 2) {
+ params->SourceFormat = LVM_STEREO;
+ } else if (nrChannels > 2 && nrChannels <= 8) { // FCC_2 FCC_8
+ params->SourceFormat = LVM_MULTICHANNEL;
+ } else {
+ return -EINVAL;
+ }
+
+ LVM_Fs_en sampleRate;
+ switch (plvmConfigParams->samplingFreq) {
+ case 8000:
+ sampleRate = LVM_FS_8000;
+ break;
+ case 11025:
+ sampleRate = LVM_FS_11025;
+ break;
+ case 12000:
+ sampleRate = LVM_FS_12000;
+ break;
+ case 16000:
+ sampleRate = LVM_FS_16000;
+ break;
+ case 22050:
+ sampleRate = LVM_FS_22050;
+ break;
+ case 24000:
+ sampleRate = LVM_FS_24000;
+ break;
+ case 32000:
+ sampleRate = LVM_FS_32000;
+ break;
+ case 44100:
+ sampleRate = LVM_FS_44100;
+ break;
+ case 48000:
+ sampleRate = LVM_FS_48000;
+ break;
+ case 88200:
+ sampleRate = LVM_FS_88200;
+ break;
+ case 96000:
+ sampleRate = LVM_FS_96000;
+ break;
+ case 176400:
+ sampleRate = LVM_FS_176400;
+ break;
+ case 192000:
+ sampleRate = LVM_FS_192000;
+ break;
+ default:
+ return -EINVAL;
+ }
+ params->SampleRate = sampleRate;
/* Concert Sound parameters */
params->VirtualizerOperatingMode = plvmConfigParams->csEnable;
@@ -468,14 +518,17 @@
params->CS_EffectLevel = LVM_CS_EFFECT_NONE;
/* N-Band Equaliser parameters */
- params->EQNB_OperatingMode = plvmConfigParams->eqEnable;
- params->pEQNB_BandDefinition = &BandDefs[0];
+ const int eqPresetLevel = plvmConfigParams->eqPresetLevel;
+ LVM_EQNB_BandDef_t BandDefs[MAX_NUM_BANDS]; /* Equaliser band definitions */
for (int i = 0; i < FIVEBAND_NUMBANDS; i++) {
BandDefs[i].Frequency = EQNB_5BandPresetsFrequencies[i];
BandDefs[i].QFactor = EQNB_5BandPresetsQFactors[i];
BandDefs[i].Gain =
EQNB_5BandSoftPresets[(FIVEBAND_NUMBANDS * eqPresetLevel) + i];
}
+ params->EQNB_OperatingMode = plvmConfigParams->eqEnable;
+ // Caution: raw pointer to stack data, stored in instance by LVM_SetControlParameters.
+ params->pEQNB_BandDefinition = &BandDefs[0];
/* Volume Control parameters */
params->VC_EffectLevel = 0;
@@ -490,16 +543,6 @@
/* Bass Enhancement parameters */
params->BE_OperatingMode = plvmConfigParams->bassEnable;
- if (nrChannels == 1) {
- params->SourceFormat = LVM_MONO;
- }
- if (nrChannels == 2) {
- params->SourceFormat = LVM_STEREO;
- }
- if ((nrChannels > 2) && (nrChannels <= 8)) {
- params->SourceFormat = LVM_MULTICHANNEL;
- }
-
/* Activate the initial settings */
LvmStatus =
LVM_SetControlParameters(pContext->pBundledContext->hInstance, params);
@@ -613,7 +656,9 @@
samplingFreq != 12000 && samplingFreq != 16000 &&
samplingFreq != 22050 && samplingFreq != 24000 &&
samplingFreq != 32000 && samplingFreq != 44100 &&
- samplingFreq != 48000 && samplingFreq != 96000) {
+ samplingFreq != 48000 && samplingFreq != 88200 &&
+ samplingFreq != 96000 && samplingFreq != 176400 &&
+ samplingFreq != 192000) {
ALOGE("\nError: Unsupported Sampling Frequency : %d\n", samplingFreq);
return -1;
}
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 09e9964..b5860de 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -1275,10 +1275,18 @@
pContext->pBundledContext->SamplesPerSecond = 48000 * NrChannels;
break;
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
+ case 88200:
+ SampleRate = LVM_FS_88200;
+ pContext->pBundledContext->SamplesPerSecond = 88200 * NrChannels;
+ break;
case 96000:
SampleRate = LVM_FS_96000;
pContext->pBundledContext->SamplesPerSecond = 96000 * NrChannels;
break;
+ case 176400:
+ SampleRate = LVM_FS_176400;
+ pContext->pBundledContext->SamplesPerSecond = 176400 * NrChannels;
+ break;
case 192000:
SampleRate = LVM_FS_192000;
pContext->pBundledContext->SamplesPerSecond = 192000 * NrChannels;
diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
index d558169..602f607 100644
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
@@ -675,9 +675,15 @@
SampleRate = LVM_FS_48000;
break;
#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
+ case 88200:
+ SampleRate = LVM_FS_88200;
+ break;
case 96000:
SampleRate = LVM_FS_96000;
break;
+ case 176400:
+ SampleRate = LVM_FS_176400;
+ break;
case 192000:
SampleRate = LVM_FS_192000;
break;
diff --git a/media/libmedia/TypeConverter.cpp b/media/libmedia/TypeConverter.cpp
index fb861d7..b5a7172 100644
--- a/media/libmedia/TypeConverter.cpp
+++ b/media/libmedia/TypeConverter.cpp
@@ -285,6 +285,7 @@
template <>
const StreamTypeConverter::Table StreamTypeConverter::mTable[] = {
+ MAKE_STRING_FROM_ENUM(AUDIO_STREAM_DEFAULT),
MAKE_STRING_FROM_ENUM(AUDIO_STREAM_VOICE_CALL),
MAKE_STRING_FROM_ENUM(AUDIO_STREAM_SYSTEM),
MAKE_STRING_FROM_ENUM(AUDIO_STREAM_RING),
@@ -361,6 +362,22 @@
TERMINATOR
};
+template <>
+const AudioFlagConverter::Table AudioFlagConverter::mTable[] = {
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_NONE),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_AUDIBILITY_ENFORCED),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_SECURE),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_SCO),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_BEACON),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_HW_AV_SYNC),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_HW_HOTWORD),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_BYPASS_MUTE),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_LOW_LATENCY),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_DEEP_BUFFER),
+ TERMINATOR
+};
+
template class TypeConverter<OutputDeviceTraits>;
template class TypeConverter<InputDeviceTraits>;
template class TypeConverter<OutputFlagTraits>;
@@ -374,6 +391,7 @@
template class TypeConverter<AudioModeTraits>;
template class TypeConverter<UsageTraits>;
template class TypeConverter<SourceTraits>;
+template class TypeConverter<AudioFlagTraits>;
bool deviceFromString(const std::string& literalDevice, audio_devices_t& device) {
return InputDeviceConverter::fromString(literalDevice, device) ||
diff --git a/media/libmedia/include/media/TypeConverter.h b/media/libmedia/include/media/TypeConverter.h
index 86f0d4c..418e09c 100644
--- a/media/libmedia/include/media/TypeConverter.h
+++ b/media/libmedia/include/media/TypeConverter.h
@@ -20,6 +20,7 @@
#include <string>
#include <string.h>
+#include <vector>
#include <system/audio.h>
#include <utils/Log.h>
#include <utils/Vector.h>
@@ -30,77 +31,55 @@
namespace android {
-struct SampleRateTraits
-{
- typedef uint32_t Type;
- typedef SortedVector<Type> Collection;
-};
-struct DeviceTraits
-{
- typedef audio_devices_t Type;
- typedef Vector<Type> Collection;
-};
-struct OutputDeviceTraits : public DeviceTraits {};
-struct InputDeviceTraits : public DeviceTraits {};
-struct OutputFlagTraits
-{
- typedef audio_output_flags_t Type;
- typedef Vector<Type> Collection;
-};
-struct InputFlagTraits
-{
- typedef audio_input_flags_t Type;
- typedef Vector<Type> Collection;
-};
-struct FormatTraits
-{
- typedef audio_format_t Type;
- typedef Vector<Type> Collection;
-};
-struct ChannelTraits
-{
- typedef audio_channel_mask_t Type;
- typedef SortedVector<Type> Collection;
-};
-struct OutputChannelTraits : public ChannelTraits {};
-struct InputChannelTraits : public ChannelTraits {};
-struct ChannelIndexTraits : public ChannelTraits {};
-struct GainModeTraits
-{
- typedef audio_gain_mode_t Type;
- typedef Vector<Type> Collection;
-};
-struct StreamTraits
-{
- typedef audio_stream_type_t Type;
- typedef Vector<Type> Collection;
-};
-struct AudioModeTraits
-{
- typedef audio_mode_t Type;
- typedef Vector<Type> Collection;
-};
-struct AudioContentTraits
-{
- typedef audio_content_type_t Type;
- typedef Vector<Type> Collection;
-};
-struct UsageTraits
-{
- typedef audio_usage_t Type;
- typedef Vector<Type> Collection;
-};
-struct SourceTraits
-{
- typedef audio_source_t Type;
- typedef Vector<Type> Collection;
-};
template <typename T>
struct DefaultTraits
{
typedef T Type;
- typedef Vector<Type> Collection;
+ typedef std::vector<Type> Collection;
+ static void add(Collection &collection, Type value)
+ {
+ collection.push_back(value);
+ }
};
+template <typename T>
+struct VectorTraits
+{
+ typedef T Type;
+ typedef Vector<Type> Collection;
+ static void add(Collection &collection, Type value)
+ {
+ collection.add(value);
+ }
+};
+template <typename T>
+struct SortedVectorTraits
+{
+ typedef T Type;
+ typedef SortedVector<Type> Collection;
+ static void add(Collection &collection, Type value)
+ {
+ collection.add(value);
+ }
+};
+
+using SampleRateTraits = SortedVectorTraits<uint32_t>;
+using DeviceTraits = DefaultTraits<audio_devices_t>;
+struct OutputDeviceTraits : public DeviceTraits {};
+struct InputDeviceTraits : public DeviceTraits {};
+using ChannelTraits = SortedVectorTraits<audio_channel_mask_t>;
+struct OutputChannelTraits : public ChannelTraits {};
+struct InputChannelTraits : public ChannelTraits {};
+struct ChannelIndexTraits : public ChannelTraits {};
+using InputFlagTraits = DefaultTraits<audio_input_flags_t>;
+using OutputFlagTraits = DefaultTraits<audio_output_flags_t>;
+using FormatTraits = VectorTraits<audio_format_t>;
+using GainModeTraits = DefaultTraits<audio_gain_mode_t>;
+using StreamTraits = DefaultTraits<audio_stream_type_t>;
+using AudioModeTraits = DefaultTraits<audio_mode_t>;
+using AudioContentTraits = DefaultTraits<audio_content_type_t>;
+using UsageTraits = DefaultTraits<audio_usage_t>;
+using SourceTraits = DefaultTraits<audio_source_t>;
+struct AudioFlagTraits : public DefaultTraits<audio_flags_mask_t> {};
template <class Traits>
static void collectionFromString(const std::string &str, typename Traits::Collection &collection,
@@ -110,7 +89,7 @@
for (const char *cstr = strtok(literal, del); cstr != NULL; cstr = strtok(NULL, del)) {
typename Traits::Type value;
if (utilities::convertTo<std::string, typename Traits::Type >(cstr, value)) {
- collection.add(value);
+ Traits::add(collection, value);
}
}
free(literal);
@@ -181,7 +160,7 @@
for (const char *cstr = strtok(literal, del); cstr != NULL; cstr = strtok(NULL, del)) {
typename Traits::Type value;
if (fromString(cstr, value)) {
- collection.add(value);
+ Traits::add(collection, value);
}
}
free(literal);
@@ -234,6 +213,7 @@
typedef TypeConverter<AudioContentTraits> AudioContentTypeConverter;
typedef TypeConverter<UsageTraits> UsageTypeConverter;
typedef TypeConverter<SourceTraits> SourceTypeConverter;
+typedef TypeConverter<AudioFlagTraits> AudioFlagConverter;
template<> const OutputDeviceConverter::Table OutputDeviceConverter::mTable[];
template<> const InputDeviceConverter::Table InputDeviceConverter::mTable[];
@@ -249,6 +229,7 @@
template<> const AudioContentTypeConverter::Table AudioContentTypeConverter::mTable[];
template<> const UsageTypeConverter::Table UsageTypeConverter::mTable[];
template<> const SourceTypeConverter::Table SourceTypeConverter::mTable[];
+template<> const AudioFlagConverter::Table AudioFlagConverter::mTable[];
bool deviceFromString(const std::string& literalDevice, audio_devices_t& device);
@@ -274,6 +255,69 @@
OutputChannelTraits::Collection outputChannelMasksFromString(
const std::string &outChannels, const char *del = AudioParameter::valueListSeparator);
+static inline std::string toString(audio_usage_t usage)
+{
+ std::string usageLiteral;
+ if (!android::UsageTypeConverter::toString(usage, usageLiteral)) {
+ ALOGV("failed to convert usage: %d", usage);
+ return "AUDIO_USAGE_UNKNOWN";
+ }
+ return usageLiteral;
+}
+
+static inline std::string toString(audio_content_type_t content)
+{
+ std::string contentLiteral;
+ if (!android::AudioContentTypeConverter::toString(content, contentLiteral)) {
+ ALOGV("failed to convert content type: %d", content);
+ return "AUDIO_CONTENT_TYPE_UNKNOWN";
+ }
+ return contentLiteral;
+}
+
+static inline std::string toString(audio_stream_type_t stream)
+{
+ std::string streamLiteral;
+ if (!android::StreamTypeConverter::toString(stream, streamLiteral)) {
+ ALOGV("failed to convert stream: %d", stream);
+ return "AUDIO_STREAM_DEFAULT";
+ }
+ return streamLiteral;
+}
+
+static inline std::string toString(audio_source_t source)
+{
+ std::string sourceLiteral;
+ if (!android::SourceTypeConverter::toString(source, sourceLiteral)) {
+ ALOGV("failed to convert source: %d", source);
+ return "AUDIO_SOURCE_DEFAULT";
+ }
+ return sourceLiteral;
+}
+
+static inline std::string toString(const audio_attributes_t &attributes)
+{
+ std::ostringstream result;
+ result << "{ Content type: " << toString(attributes.content_type)
+ << " Usage: " << toString(attributes.usage)
+ << " Source: " << toString(attributes.source)
+ << " Flags: " << attributes.flags
+ << " Tags: " << attributes.tags
+ << " }";
+
+ return result.str();
+}
+
+static inline std::string toString(audio_mode_t mode)
+{
+ std::string modeLiteral;
+ if (!android::AudioModeConverter::toString(mode, modeLiteral)) {
+ ALOGV("failed to convert mode: %d", mode);
+ return "AUDIO_MODE_INVALID";
+ }
+ return modeLiteral;
+}
+
}; // namespace android
#endif /*ANDROID_TYPE_CONVERTER_H_*/
diff --git a/media/libstagefright/MediaTrack.cpp b/media/libstagefright/MediaTrack.cpp
index 5f2e601..ef252f4 100644
--- a/media/libstagefright/MediaTrack.cpp
+++ b/media/libstagefright/MediaTrack.cpp
@@ -245,6 +245,9 @@
if (format->mFormat->findInt32("crypto-skip-byte-block", &val32)) {
meta.setInt32(kKeySkipByteBlock, val32);
}
+ if (format->mFormat->findInt32("valid-samples", &val32)) {
+ meta.setInt32(kKeyValidSamples, val32);
+ }
sp<ABuffer> valbuf;
if (format->mFormat->findBuffer("crypto-plain-sizes", &valbuf)) {
meta.setData(kKeyPlainSizes,
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index cfa9fd9..5e8d173 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -613,13 +613,14 @@
{ "crypto-default-iv-size", kKeyCryptoDefaultIVSize },
{ "crypto-encrypted-byte-block", kKeyEncryptedByteBlock },
{ "crypto-skip-byte-block", kKeySkipByteBlock },
+ { "frame-count", kKeyFrameCount },
{ "max-bitrate", kKeyMaxBitRate },
{ "pcm-big-endian", kKeyPcmBigEndian },
{ "temporal-layer-count", kKeyTemporalLayerCount },
{ "temporal-layer-id", kKeyTemporalLayerId },
{ "thumbnail-width", kKeyThumbnailWidth },
{ "thumbnail-height", kKeyThumbnailHeight },
- { "frame-count", kKeyFrameCount },
+ { "valid-samples", kKeyValidSamples },
}
};
diff --git a/media/ndk/NdkMediaFormat.cpp b/media/ndk/NdkMediaFormat.cpp
index 8f3a9f3..92d3aef 100644
--- a/media/ndk/NdkMediaFormat.cpp
+++ b/media/ndk/NdkMediaFormat.cpp
@@ -377,6 +377,7 @@
EXPORT const char* AMEDIAFORMAT_KEY_TITLE = "title";
EXPORT const char* AMEDIAFORMAT_KEY_TRACK_ID = "track-id";
EXPORT const char* AMEDIAFORMAT_KEY_TRACK_INDEX = "track-index";
+EXPORT const char* AMEDIAFORMAT_KEY_VALID_SAMPLES = "valid-samples";
EXPORT const char* AMEDIAFORMAT_KEY_WIDTH = "width";
EXPORT const char* AMEDIAFORMAT_KEY_YEAR = "year";
diff --git a/media/ndk/include/media/NdkMediaFormat.h b/media/ndk/include/media/NdkMediaFormat.h
index 6e7e0f9..2551228 100644
--- a/media/ndk/include/media/NdkMediaFormat.h
+++ b/media/ndk/include/media/NdkMediaFormat.h
@@ -226,6 +226,7 @@
extern const char* AMEDIAFORMAT_KEY_THUMBNAIL_TIME __INTRODUCED_IN(29);
extern const char* AMEDIAFORMAT_KEY_THUMBNAIL_WIDTH __INTRODUCED_IN(29);
extern const char* AMEDIAFORMAT_KEY_TITLE __INTRODUCED_IN(29);
+extern const char* AMEDIAFORMAT_KEY_VALID_SAMPLES __INTRODUCED_IN(29);
extern const char* AMEDIAFORMAT_KEY_YEAR __INTRODUCED_IN(29);
#endif /* __ANDROID_API__ >= 29 */
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index d24cc9b..c50084e 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -140,6 +140,7 @@
AMEDIAFORMAT_KEY_TIME_US; # var introduced=28
AMEDIAFORMAT_KEY_TRACK_INDEX; # var introduced=28
AMEDIAFORMAT_KEY_TRACK_ID; # var introduced=28
+ AMEDIAFORMAT_KEY_VALID_SAMPLES; # var introduced=29
AMEDIAFORMAT_KEY_WIDTH; # var introduced=21
AMEDIAFORMAT_KEY_YEAR; # var introduced=29
AMediaCodecActionCode_isRecoverable; # introduced=28
diff --git a/packages/MediaComponents/apex/java/android/media/MediaMetadata.java b/packages/MediaComponents/apex/java/android/media/MediaMetadata.java
index 33e6916..adfd20b 100644
--- a/packages/MediaComponents/apex/java/android/media/MediaMetadata.java
+++ b/packages/MediaComponents/apex/java/android/media/MediaMetadata.java
@@ -422,9 +422,7 @@
}
private MediaMetadata(Parcel in) {
- //TODO(b/119789387): Resolve hidden API usage: Bundle#setDefusable
- //mBundle = Bundle.setDefusable(in.readBundle(), true);
- mBundle = new Bundle(); //TODO:remove this.
+ mBundle = in.readBundle();
}
/**
diff --git a/packages/MediaComponents/apex/java/android/media/session/ISession.aidl b/packages/MediaComponents/apex/java/android/media/session/ISession.aidl
index cbd93cb..6363ed0 100644
--- a/packages/MediaComponents/apex/java/android/media/session/ISession.aidl
+++ b/packages/MediaComponents/apex/java/android/media/session/ISession.aidl
@@ -39,7 +39,7 @@
void destroy();
// These commands are for the TransportPerformer
- void setMetadata(in MediaMetadata metadata);
+ void setMetadata(in MediaMetadata metadata, long duration, String metadataDescription);
void setPlaybackState(in PlaybackState state);
//TODO(b/119750807): Resolve hidden API usage ParceledListSlice.
//void setQueue(in ParceledListSlice queue);
diff --git a/packages/MediaComponents/apex/java/android/media/session/MediaSession.java b/packages/MediaComponents/apex/java/android/media/session/MediaSession.java
index 1ae1d2c..4ebfb8e 100644
--- a/packages/MediaComponents/apex/java/android/media/session/MediaSession.java
+++ b/packages/MediaComponents/apex/java/android/media/session/MediaSession.java
@@ -30,6 +30,7 @@
import android.media.MediaMetadata;
import android.media.Rating;
import android.media.VolumeProvider;
+import android.media.session.MediaSessionManager.RemoteUserInfo;
import android.net.Uri;
import android.os.Bundle;
import android.os.Handler;
@@ -40,7 +41,6 @@
import android.os.RemoteException;
import android.os.ResultReceiver;
import android.os.UserHandle;
-import android.media.session.MediaSessionManager.RemoteUserInfo;
import android.service.media.MediaBrowserService;
import android.text.TextUtils;
import android.util.Log;
@@ -439,11 +439,21 @@
* @see android.media.MediaMetadata.Builder#putBitmap
*/
public void setMetadata(@Nullable MediaMetadata metadata) {
+ long duration = -1;
+ int fields = 0;
+ MediaDescription description = null;
if (metadata != null) {
metadata = (new MediaMetadata.Builder(metadata, mMaxBitmapSize)).build();
+ if (metadata.containsKey(MediaMetadata.METADATA_KEY_DURATION)) {
+ duration = metadata.getLong(MediaMetadata.METADATA_KEY_DURATION);
+ }
+ fields = metadata.size();
+ description = metadata.getDescription();
}
+ String metadataDescription = "size=" + fields + ", description=" + description;
+
try {
- mBinder.setMetadata(metadata);
+ mBinder.setMetadata(metadata, duration, metadataDescription);
} catch (RemoteException e) {
Log.wtf(TAG, "Dead object in setPlaybackState.", e);
}
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
index ebb9352..bb9cad8 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -116,6 +116,7 @@
audio_module_handle_t getModuleHandle() const;
uint32_t getModuleVersionMajor() const;
const char *getModuleName() const;
+ sp<HwModule> getModule() const { return mModule; }
bool useInputChannelMask() const
{
@@ -137,12 +138,12 @@
void log(const char* indent) const;
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;
+ sp<HwModule> mModule; // audio HW module exposing this I/O stream
String8 mName;
audio_port_type_t mType;
audio_port_role_t mRole;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioRoute.h b/services/audiopolicy/common/managerdefinitions/include/AudioRoute.h
index 330f1d4..0357ff4 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioRoute.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioRoute.h
@@ -46,6 +46,19 @@
audio_route_type_t getType() const { return mType; }
+ /**
+ * @brief supportsPatch checks if an audio patch is supported by a Route declared in
+ * the audio_policy_configuration.xml file.
+ * If the patch is supported natively by an AudioHAL (which supports of course Routing API 3.0),
+ * audiopolicy will not request AudioFlinger to use a software bridge to realize a patch
+ * between 2 ports.
+ * @param srcPort (aka the source) to be considered
+ * @param dstPort (aka the sink) to be considered
+ * @return true if the audio route supports the connection between the sink and the source,
+ * false otherwise
+ */
+ bool supportsPatch(const sp<AudioPort> &srcPort, const sp<AudioPort> &dstPort) const;
+
void dump(String8 *dst, int spaces) const;
private:
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 6f99bf3..d02123c 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -39,6 +39,8 @@
virtual const String8 getTagName() const { return mTagName; }
audio_devices_t type() const { return mDeviceType; }
+ String8 address() const { return mAddress; }
+ void setAddress(const String8 &address) { mAddress = address; }
const FormatVector& encodedFormats() const { return mEncodedFormats; }
@@ -57,39 +59,113 @@
audio_port_handle_t getId() const;
void dump(String8 *dst, int spaces, int index, bool verbose = true) const;
void log() const;
-
- String8 mAddress;
+ std::string toString() const;
private:
+ String8 mAddress{""};
String8 mTagName; // Unique human readable identifier for a device port found in conf file.
audio_devices_t mDeviceType;
FormatVector mEncodedFormats;
- audio_port_handle_t mId;
-
-friend class DeviceVector;
+ audio_port_handle_t mId = AUDIO_PORT_HANDLE_NONE;
};
class DeviceVector : public SortedVector<sp<DeviceDescriptor> >
{
public:
DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
+ explicit DeviceVector(const sp<DeviceDescriptor>& item) : DeviceVector()
+ {
+ add(item);
+ }
ssize_t add(const sp<DeviceDescriptor>& item);
void add(const DeviceVector &devices);
ssize_t remove(const sp<DeviceDescriptor>& item);
+ void remove(const DeviceVector &devices);
ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
audio_devices_t types() const { return mDeviceTypes; }
// If 'address' is empty, a device with a non-empty address may be returned
// if there is no device with the specified 'type' and empty address.
- sp<DeviceDescriptor> getDevice(audio_devices_t type, const String8 &address) const;
+ sp<DeviceDescriptor> getDevice(audio_devices_t type, const String8 &address = {}) const;
DeviceVector getDevicesFromTypeMask(audio_devices_t types) const;
+
+ /**
+ * @brief getDeviceFromId
+ * @param id of the DeviceDescriptor to seach (aka Port handle).
+ * @return DeviceDescriptor associated to port id if found, nullptr otherwise. If the id is
+ * equal to AUDIO_PORT_HANDLE_NONE, it also returns a nullptr.
+ */
sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
sp<DeviceDescriptor> getDeviceFromTagName(const String8 &tagName) const;
DeviceVector getDevicesFromHwModule(audio_module_handle_t moduleHandle) const;
audio_devices_t getDeviceTypesFromHwModule(audio_module_handle_t moduleHandle) const;
+ bool contains(const sp<DeviceDescriptor>& item) const { return indexOf(item) >= 0; }
+
+ /**
+ * @brief containsAtLeastOne
+ * @param devices vector of devices to check against.
+ * @return true if the DeviceVector contains at list one of the devices from the given vector.
+ */
+ bool containsAtLeastOne(const DeviceVector &devices) const;
+
+ /**
+ * @brief containsAllDevices
+ * @param devices vector of devices to check against.
+ * @return true if the DeviceVector contains all the devices from the given vector
+ */
+ bool containsAllDevices(const DeviceVector &devices) const;
+
+ /**
+ * @brief filter the devices supported by this collection against another collection
+ * @param devices to filter against
+ * @return
+ */
+ DeviceVector filter(const DeviceVector &devices) const;
+
+ /**
+ * @brief merge two vectors. As SortedVector Implementation is buggy (it does not check the size
+ * of the destination vector, only of the source, it provides a safe implementation
+ * @param devices source device vector to merge with
+ * @return size of the merged vector.
+ */
+ ssize_t merge(const DeviceVector &devices)
+ {
+ if (isEmpty()) {
+ add(devices);
+ return size();
+ }
+ return SortedVector::merge(devices);
+ }
+
+ /**
+ * @brief operator == DeviceVector are equals if all the DeviceDescriptor can be found (aka
+ * DeviceDescriptor with same type and address) and the vector has same size.
+ * @param right DeviceVector to compare to.
+ * @return true if right contains the same device and has the same size.
+ */
+ bool operator==(const DeviceVector &right) const
+ {
+ if (size() != right.size()) {
+ return false;
+ }
+ for (const auto &device : *this) {
+ if (right.indexOf(device) < 0) {
+ return false;
+ }
+ }
+ return true;
+ }
+
+ bool operator!=(const DeviceVector &right) const
+ {
+ return !operator==(right);
+ }
+
+ std::string toString() const;
+
void dump(String8 *dst, const String8 &tag, int spaces = 0, bool verbose = true) const;
private:
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index 6560431..2b57fa9 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -81,6 +81,17 @@
return mPorts.findByTagName(tagName);
}
+ /**
+ * @brief supportsPatch checks if an audio patch between 2 ports beloging to this HwModule
+ * is supported by a HwModule. The ports and the route shall be declared in the
+ * audio_policy_configuration.xml file.
+ * @param srcPort (aka the source) to be considered
+ * @param dstPort (aka the sink) to be considered
+ * @return true if the HwModule supports the connection between the sink and the source,
+ * false otherwise
+ */
+ bool supportsPatch(const sp<AudioPort> &srcPort, const sp<AudioPort> &dstPort) const;
+
// TODO remove from here (split serialization)
void dump(String8 *dst) const;
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index 8ff8238..ca6ca56 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -88,7 +88,7 @@
bool supportDeviceAddress(const String8 &address) const
{
- return mSupportedDevices[0]->mAddress == address;
+ return mSupportedDevices[0]->address() == address;
}
// chose first device present in mSupportedDevices also part of deviceType
diff --git a/services/audiopolicy/common/managerdefinitions/include/TypeConverter.h b/services/audiopolicy/common/managerdefinitions/include/TypeConverter.h
index 63c19d1..6b0476c 100644
--- a/services/audiopolicy/common/managerdefinitions/include/TypeConverter.h
+++ b/services/audiopolicy/common/managerdefinitions/include/TypeConverter.h
@@ -23,26 +23,10 @@
namespace android {
-struct DeviceCategoryTraits
-{
- typedef device_category Type;
- typedef Vector<Type> Collection;
-};
-struct MixTypeTraits
-{
- typedef int32_t Type;
- typedef Vector<Type> Collection;
-};
-struct RouteFlagTraits
-{
- typedef uint32_t Type;
- typedef Vector<Type> Collection;
-};
-struct RuleTraits
-{
- typedef uint32_t Type;
- typedef Vector<Type> Collection;
-};
+struct RuleTraits : public DefaultTraits<uint32_t> {};
+using DeviceCategoryTraits = DefaultTraits<device_category>;
+struct MixTypeTraits : public DefaultTraits<int32_t> {};
+struct RouteFlagTraits : public DefaultTraits<uint32_t> {};
typedef TypeConverter<DeviceCategoryTraits> DeviceCategoryConverter;
typedef TypeConverter<MixTypeTraits> MixTypeConverter;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 4ce6b08..97504ab 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -698,8 +698,8 @@
sp<SwAudioOutputDescriptor> primaryOutput = getPrimaryOutput();
if ((primaryOutput != NULL) && (primaryOutput->mProfile != NULL)
- && (primaryOutput->mProfile->mModule != NULL)) {
- sp<HwModule> primaryHwModule = primaryOutput->mProfile->mModule;
+ && (primaryOutput->mProfile->getModule() != NULL)) {
+ sp<HwModule> primaryHwModule = primaryOutput->mProfile->getModule();
Vector <sp<IOProfile>> primaryHwModuleOutputProfiles =
primaryHwModule->getOutputProfiles();
for (size_t i = 0; i < primaryHwModuleOutputProfiles.size(); i++) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
index c1fe5b0..79f0919 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
@@ -37,4 +37,19 @@
dst->append("\n");
}
+bool AudioRoute::supportsPatch(const sp<AudioPort> &srcPort, const sp<AudioPort> &dstPort) const
+{
+ if (mSink == 0 || dstPort == 0 || dstPort != mSink) {
+ return false;
+ }
+ ALOGV("%s: sinks %s matching", __FUNCTION__, mSink->getTagName().string());
+ for (const auto &sourcePort : mSources) {
+ if (sourcePort == srcPort) {
+ ALOGV("%s: sources %s matching", __FUNCTION__, sourcePort->getTagName().string());
+ return true;
+ }
+ }
+ return false;
+}
+
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 9e5f944..04cbcd1 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -35,7 +35,7 @@
AudioPort(String8(""), AUDIO_PORT_TYPE_DEVICE,
audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
AUDIO_PORT_ROLE_SOURCE),
- mAddress(""), mTagName(tagName), mDeviceType(type), mEncodedFormats(encodedFormats), mId(0)
+ mTagName(tagName), mDeviceType(type), mEncodedFormats(encodedFormats)
{
if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX || type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
mAddress = String8("0");
@@ -132,6 +132,13 @@
return ret;
}
+void DeviceVector::remove(const DeviceVector &devices)
+{
+ for (const auto& device : devices) {
+ remove(device);
+ }
+}
+
DeviceVector DeviceVector::getDevicesFromHwModule(audio_module_handle_t moduleHandle) const
{
DeviceVector devices;
@@ -159,9 +166,9 @@
sp<DeviceDescriptor> device;
for (size_t i = 0; i < size(); i++) {
if (itemAt(i)->type() == type) {
- if (address == "" || itemAt(i)->mAddress == address) {
+ if (address == "" || itemAt(i)->address() == address) {
device = itemAt(i);
- if (itemAt(i)->mAddress == address) {
+ if (itemAt(i)->address() == address) {
break;
}
}
@@ -174,9 +181,11 @@
sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const
{
- for (const auto& device : *this) {
- if (device->getId() == id) {
- return device;
+ if (id != AUDIO_PORT_HANDLE_NONE) {
+ for (const auto& device : *this) {
+ if (device->getId() == id) {
+ return device;
+ }
}
}
return nullptr;
@@ -188,8 +197,8 @@
bool isOutput = audio_is_output_devices(type);
type &= ~AUDIO_DEVICE_BIT_IN;
for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
- bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType);
- audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN;
+ bool curIsOutput = audio_is_output_devices(itemAt(i)->type());
+ audio_devices_t curType = itemAt(i)->type() & ~AUDIO_DEVICE_BIT_IN;
if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
devices.add(itemAt(i));
type &= ~curType;
@@ -251,8 +260,7 @@
// without the test?
// This has been demonstrated to NOT be true (at start up)
// ALOG_ASSERT(mModule != NULL);
- dstConfig->ext.device.hw_module =
- mModule != 0 ? mModule->getHandle() : AUDIO_MODULE_HANDLE_NONE;
+ dstConfig->ext.device.hw_module = getModuleHandle();
(void)audio_utils_strlcpy_zerofill(dstConfig->ext.device.address, mAddress.string());
}
@@ -263,7 +271,7 @@
port->id = mId;
toAudioPortConfig(&port->active_config);
port->ext.device.type = mDeviceType;
- port->ext.device.hw_module = mModule->getHandle();
+ port->ext.device.hw_module = getModuleHandle();
(void)audio_utils_strlcpy_zerofill(port->ext.device.address, mAddress.string());
}
@@ -294,6 +302,49 @@
AudioPort::dump(dst, spaces, verbose);
}
+std::string DeviceDescriptor::toString() const
+{
+ std::stringstream sstream;
+ sstream << "type:0x" << std::hex << type() << ",@:" << mAddress;
+ return sstream.str();
+}
+
+std::string DeviceVector::toString() const
+{
+ if (isEmpty()) {
+ return {"AUDIO_DEVICE_NONE"};
+ }
+ std::string result = {"{"};
+ for (const auto &device : *this) {
+ if (device != *begin()) {
+ result += ";";
+ }
+ result += device->toString();
+ }
+ return result + "}";
+}
+
+DeviceVector DeviceVector::filter(const DeviceVector &devices) const
+{
+ DeviceVector filteredDevices;
+ for (const auto &device : *this) {
+ if (devices.contains(device)) {
+ filteredDevices.add(device);
+ }
+ }
+ return filteredDevices;
+}
+
+bool DeviceVector::containsAtLeastOne(const DeviceVector &devices) const
+{
+ return !filter(devices).isEmpty();
+}
+
+bool DeviceVector::containsAllDevices(const DeviceVector &devices) const
+{
+ return filter(devices).size() == devices.size();
+}
+
void DeviceDescriptor::log() const
{
std::string device;
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 92bc595..80af88d 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -51,7 +51,7 @@
config->sample_rate));
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
- devDesc->mAddress = address;
+ devDesc->setAddress(address);
profile->addSupportedDevice(devDesc);
return addOutputProfile(profile);
@@ -113,7 +113,7 @@
config->sample_rate));
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
- devDesc->mAddress = address;
+ devDesc->setAddress(address);
profile->addSupportedDevice(devDesc);
ALOGV("addInputProfile() name %s rate %d mask 0x%08x",
@@ -218,6 +218,15 @@
mHandle = handle;
}
+bool HwModule::supportsPatch(const sp<AudioPort> &srcPort, const sp<AudioPort> &dstPort) const {
+ for (const auto &route : mRoutes) {
+ if (route->supportsPatch(srcPort, dstPort)) {
+ return true;
+ }
+ }
+ return false;
+}
+
void HwModule::dump(String8 *dst) const
{
dst->appendFormat(" - name: %s\n", getName());
@@ -287,7 +296,7 @@
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
devDesc->setName(String8(device_name));
- devDesc->mAddress = address;
+ devDesc->setAddress(address);
return devDesc;
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index 179a678..1154654 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -516,7 +516,7 @@
std::string address = getXmlAttribute(cur, Attributes::address);
if (!address.empty()) {
ALOGV("%s: address=%s for %s", __func__, address.c_str(), name.c_str());
- deviceDesc->mAddress = String8(address.c_str());
+ deviceDesc->setAddress(String8(address.c_str()));
}
AudioProfileTraits::Collection profiles;
@@ -535,7 +535,7 @@
return Status::fromStatusT(status);
}
ALOGV("%s: adding device tag %s type %08x address %s", __func__,
- deviceDesc->getName().string(), type, deviceDesc->mAddress.string());
+ deviceDesc->getName().string(), type, deviceDesc->address().string());
return deviceDesc;
}
@@ -742,7 +742,7 @@
}
ALOGV("%s: %s=%s",
__func__, tag, reinterpret_cast<const char*>(pointDefinition.get()));
- Vector<int32_t> point;
+ std::vector<int32_t> point;
collectionFromString<DefaultTraits<int32_t>>(
reinterpret_cast<const char*>(pointDefinition.get()), point, ",");
if (point.size() != 2) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f07b797..64a2b8a 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -149,13 +149,13 @@
// Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
// parameters on newly connected devices (instead of opening the outputs...)
- broadcastDeviceConnectionState(device, state, devDesc->mAddress);
+ broadcastDeviceConnectionState(device, state, devDesc->address());
- if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
+ if (checkOutputsForDevice(devDesc, state, outputs, devDesc->address()) != NO_ERROR) {
mAvailableOutputDevices.remove(devDesc);
broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- devDesc->mAddress);
+ devDesc->address());
return INVALID_OPERATION;
}
// Propagate device availability to Engine
@@ -178,12 +178,12 @@
ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
// Send Disconnect to HALs
- broadcastDeviceConnectionState(device, state, devDesc->mAddress);
+ broadcastDeviceConnectionState(device, state, devDesc->address());
// remove device from available output devices
mAvailableOutputDevices.remove(devDesc);
- checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
+ checkOutputsForDevice(devDesc, state, outputs, devDesc->address());
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
@@ -265,11 +265,11 @@
// Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
// parameters on newly connected devices (instead of opening the inputs...)
- broadcastDeviceConnectionState(device, state, devDesc->mAddress);
+ broadcastDeviceConnectionState(device, state, devDesc->address());
- if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) {
+ if (checkInputsForDevice(devDesc, state, inputs, devDesc->address()) != NO_ERROR) {
broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- devDesc->mAddress);
+ devDesc->address());
return INVALID_OPERATION;
}
@@ -294,9 +294,9 @@
ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
// Set Disconnect to HALs
- broadcastDeviceConnectionState(device, state, devDesc->mAddress);
+ broadcastDeviceConnectionState(device, state, devDesc->address());
- checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress);
+ checkInputsForDevice(devDesc, state, inputs, devDesc->address());
mAvailableInputDevices.remove(devDesc);
// Propagate device availability to Engine
@@ -780,17 +780,39 @@
return output;
}
-status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *output,
- audio_session_t session,
- audio_stream_type_t *stream,
- uid_t uid,
- const audio_config_t *config,
- audio_output_flags_t *flags,
- audio_port_handle_t *selectedDeviceId,
- audio_port_handle_t *portId)
+status_t AudioPolicyManager::getAudioAttributes(audio_attributes_t *dstAttr,
+ const audio_attributes_t *srcAttr,
+ audio_stream_type_t srcStream)
{
- audio_attributes_t attributes;
+ if (srcAttr != NULL) {
+ if (!isValidAttributes(srcAttr)) {
+ ALOGE("%s invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+ __func__,
+ srcAttr->usage, srcAttr->content_type, srcAttr->flags,
+ srcAttr->tags);
+ return BAD_VALUE;
+ }
+ *dstAttr = *srcAttr;
+ } else {
+ if (srcStream < AUDIO_STREAM_MIN || srcStream >= AUDIO_STREAM_PUBLIC_CNT) {
+ ALOGE("%s: invalid stream type", __func__);
+ return BAD_VALUE;
+ }
+ stream_type_to_audio_attributes(srcStream, dstAttr);
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::getOutputForAttrInt(audio_attributes_t *resultAttr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ const audio_attributes_t *attr,
+ audio_stream_type_t *stream,
+ uid_t uid,
+ const audio_config_t *config,
+ audio_output_flags_t *flags,
+ audio_port_handle_t *selectedDeviceId)
+{
DeviceVector outputDevices;
routing_strategy strategy;
audio_devices_t device;
@@ -798,35 +820,20 @@
audio_devices_t msdDevice =
getModuleDeviceTypes(mAvailableOutputDevices, AUDIO_HARDWARE_MODULE_ID_MSD);
- // The supplied portId must be AUDIO_PORT_HANDLE_NONE
- if (*portId != AUDIO_PORT_HANDLE_NONE) {
- return INVALID_OPERATION;
+ status_t status = getAudioAttributes(resultAttr, attr, *stream);
+ if (status != NO_ERROR) {
+ return status;
}
- if (attr != NULL) {
- if (!isValidAttributes(attr)) {
- ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
- attr->usage, attr->content_type, attr->flags,
- attr->tags);
- return BAD_VALUE;
- }
- attributes = *attr;
- } else {
- if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) {
- ALOGE("getOutputForAttr(): invalid stream type");
- return BAD_VALUE;
- }
- stream_type_to_audio_attributes(*stream, &attributes);
- }
-
- ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x"
+ ALOGV("%s usage=%d, content=%d, tag=%s flags=%08x"
" session %d selectedDeviceId %d",
- attributes.usage, attributes.content_type, attributes.tags, attributes.flags,
+ __func__,
+ resultAttr->usage, resultAttr->content_type, resultAttr->tags, resultAttr->flags,
session, requestedDeviceId);
- *stream = streamTypefromAttributesInt(&attributes);
+ *stream = streamTypefromAttributesInt(resultAttr);
- strategy = getStrategyForAttr(&attributes);
+ strategy = getStrategyForAttr(resultAttr);
// First check for explicit routing (eg. setPreferredDevice)
if (requestedDeviceId != AUDIO_PORT_HANDLE_NONE) {
@@ -836,30 +843,30 @@
} else {
// If no explict route, is there a matching dynamic policy that applies?
sp<SwAudioOutputDescriptor> desc;
- if (mPolicyMixes.getOutputForAttr(attributes, uid, desc) == NO_ERROR) {
+ if (mPolicyMixes.getOutputForAttr(*resultAttr, uid, desc) == NO_ERROR) {
ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr");
if (!audio_has_proportional_frames(config->format)) {
return BAD_VALUE;
}
- *stream = streamTypefromAttributesInt(&attributes);
+ *stream = streamTypefromAttributesInt(resultAttr);
*output = desc->mIoHandle;
AudioMix *mix = desc->mPolicyMix;
sp<DeviceDescriptor> deviceDesc =
mAvailableOutputDevices.getDevice(mix->mDeviceType, mix->mDeviceAddress);
*selectedDeviceId = deviceDesc != 0 ? deviceDesc->getId() : AUDIO_PORT_HANDLE_NONE;
- ALOGV("getOutputForAttr() returns output %d", *output);
- goto exit;
+ ALOGV("%s returns output %d", __func__, *output);
+ return NO_ERROR;
}
// Virtual sources must always be dynamicaly or explicitly routed
- if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
- ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
+ if (resultAttr->usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+ ALOGW("%s no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE", __func__);
return BAD_VALUE;
}
device = getDeviceForStrategy(strategy, false /*fromCache*/);
}
- if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
+ if ((resultAttr->flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
*flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
}
@@ -869,7 +876,7 @@
// to getOutputForDevice.
// TODO: Remove check of AUDIO_STREAM_MUSIC once migration is completed on the app side.
if (device == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
- (*stream == AUDIO_STREAM_MUSIC || attributes.usage == AUDIO_USAGE_VOICE_COMMUNICATION) &&
+ (*stream == AUDIO_STREAM_MUSIC || resultAttr->usage == AUDIO_USAGE_VOICE_COMMUNICATION) &&
audio_is_linear_pcm(config->format) &&
isInCall()) {
if (requestedDeviceId != AUDIO_PORT_HANDLE_NONE) {
@@ -880,9 +887,9 @@
}
}
- ALOGV("getOutputForAttr() device 0x%x, sampling rate %d, format %#x, channel mask %#x, "
+ ALOGV("%s device 0x%x, sampling rate %d, format %#x, channel mask %#x, "
"flags %#x",
- device, config->sample_rate, config->format, config->channel_mask, *flags);
+ __func__, device, config->sample_rate, config->format, config->channel_mask, *flags);
*output = AUDIO_IO_HANDLE_NONE;
if (msdDevice != AUDIO_DEVICE_NONE) {
@@ -903,25 +910,50 @@
}
outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
- *selectedDeviceId = outputDevices.size() > 0 ? outputDevices.itemAt(0)->getId()
- : AUDIO_PORT_HANDLE_NONE;
+ *selectedDeviceId = getFirstDeviceId(outputDevices);
-exit:
+ ALOGV("%s returns output %d selectedDeviceId %d", __func__, *output, *selectedDeviceId);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uid_t uid,
+ const audio_config_t *config,
+ audio_output_flags_t *flags,
+ audio_port_handle_t *selectedDeviceId,
+ audio_port_handle_t *portId)
+{
+ // The supplied portId must be AUDIO_PORT_HANDLE_NONE
+ if (*portId != AUDIO_PORT_HANDLE_NONE) {
+ return INVALID_OPERATION;
+ }
+ const audio_port_handle_t requestedDeviceId = *selectedDeviceId;
+ audio_attributes_t resultAttr;
+ status_t status = getOutputForAttrInt(&resultAttr, output, session, attr, stream, uid,
+ config, flags, selectedDeviceId);
+ if (status != NO_ERROR) {
+ return status;
+ }
+
audio_config_base_t clientConfig = {.sample_rate = config->sample_rate,
.format = config->format,
.channel_mask = config->channel_mask };
*portId = AudioPort::getNextUniqueId();
sp<TrackClientDescriptor> clientDesc =
- new TrackClientDescriptor(*portId, uid, session, attributes, clientConfig,
+ new TrackClientDescriptor(*portId, uid, session, resultAttr, clientConfig,
requestedDeviceId, *stream,
- getStrategyForAttr(&attributes),
+ getStrategyForAttr(&resultAttr),
*flags);
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(*output);
outputDesc->addClient(clientDesc);
- ALOGV(" getOutputForAttr() returns output %d selectedDeviceId %d for port ID %d",
- *output, *selectedDeviceId, *portId);
+ ALOGV("%s returns output %d selectedDeviceId %d for port ID %d",
+ __func__, *output, requestedDeviceId, *portId);
return NO_ERROR;
}
@@ -1020,8 +1052,7 @@
new SwAudioOutputDescriptor(profile, mpClientInterface);
DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
- String8 address = outputDevices.size() > 0 ? outputDevices.itemAt(0)->mAddress
- : String8("");
+ String8 address = getFirstDeviceAddress(outputDevices);
// MSD patch may be using the only output stream that can service this request. Release
// MSD patch to prioritize this request over any active output on MSD.
@@ -1722,10 +1753,7 @@
}
// Explicit routing?
- sp<DeviceDescriptor> deviceDesc;
- if (*selectedDeviceId != AUDIO_PORT_HANDLE_NONE) {
- deviceDesc = mAvailableInputDevices.getDeviceFromId(*selectedDeviceId);
- }
+ sp<DeviceDescriptor> deviceDesc = mAvailableInputDevices.getDeviceFromId(*selectedDeviceId);
// special case for mmap capture: if an input IO handle is specified, we reuse this input if
// possible
@@ -1831,8 +1859,7 @@
exit:
inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
- *selectedDeviceId = inputDevices.size() > 0 ? inputDevices.itemAt(0)->getId()
- : AUDIO_PORT_HANDLE_NONE;
+ *selectedDeviceId = getFirstDeviceId(inputDevices);
isSoundTrigger = inputSource == AUDIO_SOURCE_HOTWORD &&
mSoundTriggerSessions.indexOfKey(session) > 0;
@@ -1963,7 +1990,7 @@
if (address == "") {
DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
// the inputs vector must be of size >= 1, but we don't want to crash here
- address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress : String8("");
+ address = getFirstDeviceAddress(inputDevices);
}
status_t status = inputDesc->open(&lConfig, device, address,
@@ -2930,7 +2957,7 @@
}
if (!outputDesc->mProfile->isCompatibleProfile(devDesc->type(),
- devDesc->mAddress,
+ devDesc->address(),
patch->sources[0].sample_rate,
NULL, // updatedSamplingRate
patch->sources[0].format,
@@ -2987,7 +3014,7 @@
}
if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(),
- devDesc->mAddress,
+ devDesc->address(),
patch->sinks[0].sample_rate,
NULL, /*updatedSampleRate*/
patch->sinks[0].format,
@@ -3050,8 +3077,10 @@
// create a software bridge in PatchPanel if:
// - source and sink devices are on different HW modules OR
// - audio HAL version is < 3.0
+ // - audio HAL version is >= 3.0 but no route has been declared between devices
if (!srcDeviceDesc->hasSameHwModuleAs(sinkDeviceDesc) ||
- (srcDeviceDesc->mModule->getHalVersionMajor() < 3)) {
+ (srcDeviceDesc->getModuleVersionMajor() < 3) ||
+ !srcDeviceDesc->getModule()->supportsPatch(srcDeviceDesc, sinkDeviceDesc)) {
// support only one sink device for now to simplify output selection logic
if (patch->num_sinks > 1) {
return INVALID_OPERATION;
@@ -3395,16 +3424,25 @@
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- if (srcDeviceDesc->getAudioPort()->mModule->getHandle() ==
- sinkDeviceDesc->getAudioPort()->mModule->getHandle() &&
- srcDeviceDesc->getAudioPort()->mModule->getHalVersionMajor() >= 3 &&
+ if (srcDeviceDesc->hasSameHwModuleAs(sinkDeviceDesc) &&
+ srcDeviceDesc->getModuleVersionMajor() >= 3 &&
+ sinkDeviceDesc->getModule()->supportsPatch(srcDeviceDesc, sinkDeviceDesc) &&
srcDeviceDesc->getAudioPort()->mGains.size() > 0) {
- ALOGV("%s AUDIO_DEVICE_API_VERSION_3_0", __FUNCTION__);
+ ALOGV("%s Device to Device route supported by >=3.0 HAL", __FUNCTION__);
+ // TODO: may explicitly specify whether we should use HW or SW patch
// create patch between src device and output device
// create Hwoutput and add to mHwOutputs
} else {
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(sinkDevice, mOutputs);
- audio_io_handle_t output = selectOutput(outputs);
+ audio_attributes_t resultAttr;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = sourceDesc->config().sample_rate;
+ config.channel_mask = sourceDesc->config().channel_mask;
+ config.format = sourceDesc->config().format;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE,
+ &attributes, &stream, sourceDesc->uid(), &config, &flags, &selectedDeviceId);
if (output == AUDIO_IO_HANDLE_NONE) {
ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevice);
return INVALID_OPERATION;
@@ -3437,6 +3475,13 @@
__FUNCTION__, status);
return INVALID_OPERATION;
}
+
+ if (outputDesc->getClient(sourceDesc->portId()) != nullptr) {
+ ALOGW("%s source portId has already been attached to outputDesc", __func__);
+ return INVALID_OPERATION;
+ }
+ outputDesc->addClient(sourceDesc);
+
uint32_t delayMs = 0;
status = startSource(outputDesc, sourceDesc, &delayMs);
@@ -3615,7 +3660,7 @@
AUDIO_DEVICE_OUT_HDMI);
for (size_t i = 0; i < hdmiOutputDevices.size(); i++) {
// Simulate reconnection to update enabled surround sound formats.
- String8 address = hdmiOutputDevices[i]->mAddress;
+ String8 address = hdmiOutputDevices[i]->address();
String8 name = hdmiOutputDevices[i]->getName();
status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
@@ -3635,7 +3680,7 @@
AUDIO_DEVICE_IN_HDMI);
for (size_t i = 0; i < hdmiInputDevices.size(); i++) {
// Simulate reconnection to update enabled surround sound formats.
- String8 address = hdmiInputDevices[i]->mAddress;
+ String8 address = hdmiInputDevices[i]->address();
String8 name = hdmiInputDevices[i]->getName();
status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
@@ -3893,8 +3938,7 @@
const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
const DeviceVector &devicesForType = supportedDevices.getDevicesFromTypeMask(
profileType);
- String8 address = devicesForType.size() > 0 ? devicesForType.itemAt(0)->mAddress
- : String8("");
+ String8 address = getFirstDeviceAddress(devicesForType);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = outputDesc->open(nullptr, profileType, address,
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
@@ -3948,8 +3992,7 @@
DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(profileType);
// the inputs vector must be of size >= 1, but we don't want to crash here
- String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress
- : String8("");
+ String8 address = getFirstDeviceAddress(inputDevices);
ALOGV(" for input device 0x%x using address %s", profileType, address.string());
ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
@@ -4011,11 +4054,11 @@
}
// If microphones address is empty, set it according to device type
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
- if (mAvailableInputDevices[i]->mAddress.isEmpty()) {
+ if (mAvailableInputDevices[i]->address().isEmpty()) {
if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BUILTIN_MIC) {
- mAvailableInputDevices[i]->mAddress = String8(AUDIO_BOTTOM_MICROPHONE_ADDRESS);
+ mAvailableInputDevices[i]->address() = String8(AUDIO_BOTTOM_MICROPHONE_ADDRESS);
} else if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BACK_MIC) {
- mAvailableInputDevices[i]->mAddress = String8(AUDIO_BACK_MICROPHONE_ADDRESS);
+ mAvailableInputDevices[i]->address() = String8(AUDIO_BACK_MICROPHONE_ADDRESS);
}
}
}
@@ -5184,8 +5227,9 @@
if (!deviceList.isEmpty()) {
PatchBuilder patchBuilder;
patchBuilder.addSource(outputDesc);
- for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
- patchBuilder.addSink(deviceList.itemAt(i));
+ ALOG_ASSERT(deviceList.size() <= AUDIO_PATCH_PORTS_MAX, "Too many sink ports");
+ for (const auto &device : deviceList) {
+ patchBuilder.addSink(device);
}
installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), delayMs);
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index d0708b8..86993d4 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -519,6 +519,19 @@
return mAvailableInputDevices.getDeviceTypesFromHwModule(
mPrimaryOutput->getModuleHandle());
}
+ /**
+ * @brief getFirstDeviceId of the Device Vector
+ * @return if the collection is not empty, it returns the first device Id,
+ * otherwise AUDIO_PORT_HANDLE_NONE
+ */
+ audio_port_handle_t getFirstDeviceId(const DeviceVector &devices) const
+ {
+ return (devices.size() > 0) ? devices.itemAt(0)->getId() : AUDIO_PORT_HANDLE_NONE;
+ }
+ String8 getFirstDeviceAddress(const DeviceVector &devices) const
+ {
+ return (devices.size() > 0) ? devices.itemAt(0)->address() : String8("");
+ }
uint32_t updateCallRouting(audio_devices_t rxDevice, uint32_t delayMs = 0);
sp<AudioPatch> createTelephonyPatch(bool isRx, audio_devices_t device, uint32_t delayMs);
@@ -661,6 +674,21 @@
const String8& address /*in*/,
SortedVector<audio_io_handle_t>& outputs /*out*/);
uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
+ // internal method, get audio_attributes_t from either a source audio_attributes_t
+ // or audio_stream_type_t, respectively.
+ status_t getAudioAttributes(audio_attributes_t *dstAttr,
+ const audio_attributes_t *srcAttr,
+ audio_stream_type_t srcStream);
+ // internal method, called by getOutputForAttr() and connectAudioSource.
+ status_t getOutputForAttrInt(audio_attributes_t *resultAttr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ const audio_attributes_t *attr,
+ audio_stream_type_t *stream,
+ uid_t uid,
+ const audio_config_t *config,
+ audio_output_flags_t *flags,
+ audio_port_handle_t *selectedDeviceId);
// internal method to return the output handle for the given device and format
audio_io_handle_t getOutputForDevice(
audio_devices_t device,
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index c1a4c11..46fbc3e 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -256,7 +256,7 @@
Vector<int32_t> outputStreamIds;
std::vector<std::string> requestedPhysicalIds;
if (request.mSurfaceList.size() > 0) {
- for (sp<Surface> surface : request.mSurfaceList) {
+ for (const sp<Surface>& surface : request.mSurfaceList) {
if (surface == 0) continue;
int32_t streamId;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 856af13..12fbf82 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -175,7 +175,7 @@
session->interfaceChain([](
::android::hardware::hidl_vec<::android::hardware::hidl_string> interfaceChain) {
ALOGV("Session interface chain:");
- for (auto iface : interfaceChain) {
+ for (const auto& iface : interfaceChain) {
ALOGV(" %s", iface.c_str());
}
});
diff --git a/services/mediaextractor/Android.mk b/services/mediaextractor/Android.mk
index 7c9c727..336bbe8 100644
--- a/services/mediaextractor/Android.mk
+++ b/services/mediaextractor/Android.mk
@@ -40,10 +40,7 @@
# extractor libraries
LOCAL_REQUIRED_MODULES += \
- libmkvextractor \
- libmp4extractor \
libmpeg2extractor \
- liboggextractor \
LOCAL_SRC_FILES := main_extractorservice.cpp
LOCAL_SHARED_LIBRARIES := libmedia libmediaextractorservice libbinder libutils \
diff --git a/services/oboeservice/AAudioEndpointManager.cpp b/services/oboeservice/AAudioEndpointManager.cpp
index cca1895..a1fc0ea 100644
--- a/services/oboeservice/AAudioEndpointManager.cpp
+++ b/services/oboeservice/AAudioEndpointManager.cpp
@@ -108,7 +108,7 @@
const AAudioStreamConfiguration &configuration) {
sp<AAudioServiceEndpoint> endpoint;
mExclusiveSearchCount++;
- for (const auto ep : mExclusiveStreams) {
+ for (const auto& ep : mExclusiveStreams) {
if (ep->matches(configuration)) {
mExclusiveFoundCount++;
endpoint = ep;
@@ -126,7 +126,7 @@
const AAudioStreamConfiguration &configuration) {
sp<AAudioServiceEndpointShared> endpoint;
mSharedSearchCount++;
- for (const auto ep : mSharedStreams) {
+ for (const auto& ep : mSharedStreams) {
if (ep->matches(configuration)) {
mSharedFoundCount++;
endpoint = ep;
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index 539735a..4dfb62a 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -65,7 +65,7 @@
result << " Connected: " << mConnected.load() << "\n";
result << " Registered Streams:" << "\n";
result << AAudioServiceStreamShared::dumpHeader() << "\n";
- for (const auto stream : mRegisteredStreams) {
+ for (const auto& stream : mRegisteredStreams) {
result << stream->dump() << "\n";
}
@@ -78,7 +78,7 @@
// @return true if stream found
bool AAudioServiceEndpoint::isStreamRegistered(audio_port_handle_t portHandle) {
std::lock_guard<std::mutex> lock(mLockStreams);
- for (const auto stream : mRegisteredStreams) {
+ for (const auto& stream : mRegisteredStreams) {
if (stream->getPortHandle() == portHandle) {
return true;
}
@@ -89,7 +89,7 @@
void AAudioServiceEndpoint::disconnectRegisteredStreams() {
std::lock_guard<std::mutex> lock(mLockStreams);
mConnected.store(false);
- for (const auto stream : mRegisteredStreams) {
+ for (const auto& stream : mRegisteredStreams) {
ALOGD("disconnectRegisteredStreams() stop and disconnect %p", stream.get());
stream->stop();
stream->disconnect();
diff --git a/services/oboeservice/AAudioServiceEndpointCapture.cpp b/services/oboeservice/AAudioServiceEndpointCapture.cpp
index 7ae7f1b..37d105b 100644
--- a/services/oboeservice/AAudioServiceEndpointCapture.cpp
+++ b/services/oboeservice/AAudioServiceEndpointCapture.cpp
@@ -81,9 +81,10 @@
{ // brackets are for lock_guard
std::lock_guard <std::mutex> lock(mLockStreams);
- for (const auto clientStream : mRegisteredStreams) {
- if (clientStream->isRunning()) {
+ for (const auto& clientStream : mRegisteredStreams) {
+ if (clientStream->isRunning() && !clientStream->isSuspended()) {
int64_t clientFramesWritten = 0;
+
sp<AAudioServiceStreamShared> streamShared =
static_cast<AAudioServiceStreamShared *>(clientStream.get());
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index e4dbee1..6c28083 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -371,7 +371,7 @@
float volume = values[0];
ALOGD("%s(%p) volume[0] = %f", __func__, this, volume);
std::lock_guard<std::mutex> lock(mLockStreams);
- for(const auto stream : mRegisteredStreams) {
+ for(const auto& stream : mRegisteredStreams) {
stream->onVolumeChanged(volume);
}
};
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.cpp b/services/oboeservice/AAudioServiceEndpointPlay.cpp
index 923a1a4..1e1c552 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.cpp
+++ b/services/oboeservice/AAudioServiceEndpointPlay.cpp
@@ -80,10 +80,14 @@
int64_t mmapFramesWritten = getStreamInternal()->getFramesWritten();
std::lock_guard <std::mutex> lock(mLockStreams);
- for (const auto clientStream : mRegisteredStreams) {
+ for (const auto& clientStream : mRegisteredStreams) {
int64_t clientFramesRead = 0;
bool allowUnderflow = true;
+ if (clientStream->isSuspended()) {
+ continue; // dead stream
+ }
+
aaudio_stream_state_t state = clientStream->getState();
if (state == AAUDIO_STREAM_STATE_STOPPING) {
allowUnderflow = false; // just read what is already in the FIFO
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 354b36a..defbb7b 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -179,6 +179,7 @@
}
setFlowing(false);
+ setSuspended(false);
// Start with fresh presentation timestamps.
mAtomicTimestamp.clear();
@@ -345,7 +346,9 @@
}
int32_t count = mUpMessageQueue->getFifoBuffer()->write(command, 1);
if (count != 1) {
- ALOGE("%s(): Queue full. Did client die? %s", __func__, getTypeText());
+ ALOGW("%s(): Queue full. Did client stop? Suspending stream. what = %u, %s",
+ __func__, command->what, getTypeText());
+ setSuspended(true);
return AAUDIO_ERROR_WOULD_BLOCK;
} else {
return AAUDIO_OK;
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index a1815d0..7904b25 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -204,6 +204,20 @@
}
/**
+ * Set false when the stream should not longer be processed.
+ * This may be caused by a message queue overflow.
+ * Set true when stream is started.
+ * @param suspended
+ */
+ void setSuspended(bool suspended) {
+ mSuspended = suspended;
+ }
+
+ bool isSuspended() const {
+ return mSuspended;
+ }
+
+ /**
* Atomically increment the number of active references to the stream by AAudioService.
*
* This is called under a global lock in AAudioStreamTracker.
@@ -304,7 +318,12 @@
// This is modified under a global lock in AAudioStreamTracker.
int32_t mCallingCount = 0;
+ // This indicates that a stream that is being referenced by a binder call needs to closed.
std::atomic<bool> mCloseNeeded{false};
+
+ // This indicate that a running stream should not be processed because of an error,
+ // for example a full message queue. Note that this atomic is unrelated to mCloseNeeded.
+ std::atomic<bool> mSuspended{false};
};
} /* namespace aaudio */