Merge "libeffects: Add multichannel processing support"
diff --git a/include/media/VolumeShaper.h b/include/media/VolumeShaper.h
index a3aaece..79afd6c 100644
--- a/include/media/VolumeShaper.h
+++ b/include/media/VolumeShaper.h
@@ -551,7 +551,7 @@
static int64_t convertTimespecToUs(const struct timespec &tv)
{
- return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000;
+ return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
}
// current monotonic time in microseconds.
diff --git a/media/codec2/components/base/SimpleC2Component.cpp b/media/codec2/components/base/SimpleC2Component.cpp
index 7990ee5..50b4d20 100644
--- a/media/codec2/components/base/SimpleC2Component.cpp
+++ b/media/codec2/components/base/SimpleC2Component.cpp
@@ -489,6 +489,13 @@
}
ALOGV("start processing frame #%" PRIu64, work->input.ordinal.frameIndex.peeku());
+ // If input buffer list is not empty, it means we have some input to process on.
+ // However, input could be a null buffer. In such case, clear the buffer list
+ // before making call to process().
+ if (!work->input.buffers.empty() && !work->input.buffers[0]) {
+ ALOGD("Encountered null input buffer. Clearing the input buffer");
+ work->input.buffers.clear();
+ }
process(work, mOutputBlockPool);
ALOGV("processed frame #%" PRIu64, work->input.ordinal.frameIndex.peeku());
{
diff --git a/media/codec2/components/raw/C2SoftRawDec.cpp b/media/codec2/components/raw/C2SoftRawDec.cpp
index 8d2a652..5c83481 100644
--- a/media/codec2/components/raw/C2SoftRawDec.cpp
+++ b/media/codec2/components/raw/C2SoftRawDec.cpp
@@ -83,6 +83,18 @@
DefineParam(mInputMaxBufSize, C2_PARAMKEY_INPUT_MAX_BUFFER_SIZE)
.withConstValue(new C2StreamMaxBufferSizeInfo::input(0u, 64 * 1024))
.build());
+
+ addParameter(
+ DefineParam(mPcmEncodingInfo, C2_PARAMKEY_PCM_ENCODING)
+ .withDefault(new C2StreamPcmEncodingInfo::output(0u, C2Config::PCM_16))
+ .withFields({C2F(mPcmEncodingInfo, value).oneOf({
+ C2Config::PCM_16,
+ C2Config::PCM_8,
+ C2Config::PCM_FLOAT})
+ })
+ .withSetter((Setter<decltype(*mPcmEncodingInfo)>::StrictValueWithNoDeps))
+ .build());
+
}
private:
@@ -94,6 +106,7 @@
std::shared_ptr<C2StreamChannelCountInfo::output> mChannelCount;
std::shared_ptr<C2BitrateTuning::input> mBitrate;
std::shared_ptr<C2StreamMaxBufferSizeInfo::input> mInputMaxBufSize;
+ std::shared_ptr<C2StreamPcmEncodingInfo::output> mPcmEncodingInfo;
};
C2SoftRawDec::C2SoftRawDec(
diff --git a/media/codec2/components/vpx/C2SoftVpxDec.cpp b/media/codec2/components/vpx/C2SoftVpxDec.cpp
index 01de681..8ecbf5d 100644
--- a/media/codec2/components/vpx/C2SoftVpxDec.cpp
+++ b/media/codec2/components/vpx/C2SoftVpxDec.cpp
@@ -97,6 +97,26 @@
.withSetter(ProfileLevelSetter, mSize)
.build());
+ mHdr10PlusInfoInput = C2StreamHdr10PlusInfo::input::AllocShared(0);
+ addParameter(
+ DefineParam(mHdr10PlusInfoInput, C2_PARAMKEY_INPUT_HDR10_PLUS_INFO)
+ .withDefault(mHdr10PlusInfoInput)
+ .withFields({
+ C2F(mHdr10PlusInfoInput, m.value).any(),
+ })
+ .withSetter(Hdr10PlusInfoInputSetter)
+ .build());
+
+ mHdr10PlusInfoOutput = C2StreamHdr10PlusInfo::output::AllocShared(0);
+ addParameter(
+ DefineParam(mHdr10PlusInfoOutput, C2_PARAMKEY_OUTPUT_HDR10_PLUS_INFO)
+ .withDefault(mHdr10PlusInfoOutput)
+ .withFields({
+ C2F(mHdr10PlusInfoOutput, m.value).any(),
+ })
+ .withSetter(Hdr10PlusInfoOutputSetter)
+ .build());
+
#if 0
// sample BT.2020 static info
mHdrStaticInfo = std::make_shared<C2StreamHdrStaticInfo::output>();
@@ -217,6 +237,18 @@
return C2R::Ok();
}
+ static C2R Hdr10PlusInfoInputSetter(bool mayBlock, C2P<C2StreamHdr10PlusInfo::input> &me) {
+ (void)mayBlock;
+ (void)me; // TODO: validate
+ return C2R::Ok();
+ }
+
+ static C2R Hdr10PlusInfoOutputSetter(bool mayBlock, C2P<C2StreamHdr10PlusInfo::output> &me) {
+ (void)mayBlock;
+ (void)me; // TODO: validate
+ return C2R::Ok();
+ }
+
private:
std::shared_ptr<C2StreamProfileLevelInfo::input> mProfileLevel;
std::shared_ptr<C2StreamPictureSizeInfo::output> mSize;
@@ -228,6 +260,8 @@
#if 0
std::shared_ptr<C2StreamHdrStaticInfo::output> mHdrStaticInfo;
#endif
+ std::shared_ptr<C2StreamHdr10PlusInfo::input> mHdr10PlusInfoInput;
+ std::shared_ptr<C2StreamHdr10PlusInfo::output> mHdr10PlusInfoOutput;
#endif
};
@@ -370,7 +404,8 @@
const std::shared_ptr<C2GraphicBlock> &block) {
std::shared_ptr<C2Buffer> buffer = createGraphicBuffer(block,
C2Rect(mWidth, mHeight));
- auto fillWork = [buffer, index](const std::unique_ptr<C2Work> &work) {
+ auto fillWork = [buffer, index, intf = this->mIntf](
+ const std::unique_ptr<C2Work> &work) {
uint32_t flags = 0;
if ((work->input.flags & C2FrameData::FLAG_END_OF_STREAM) &&
(c2_cntr64_t(index) == work->input.ordinal.frameIndex)) {
@@ -382,6 +417,28 @@
work->worklets.front()->output.buffers.push_back(buffer);
work->worklets.front()->output.ordinal = work->input.ordinal;
work->workletsProcessed = 1u;
+
+ for (const std::unique_ptr<C2Param> ¶m: work->input.configUpdate) {
+ if (param) {
+ C2StreamHdr10PlusInfo::input *hdr10PlusInfo =
+ C2StreamHdr10PlusInfo::input::From(param.get());
+
+ if (hdr10PlusInfo != nullptr) {
+ std::vector<std::unique_ptr<C2SettingResult>> failures;
+ std::unique_ptr<C2Param> outParam = C2Param::CopyAsStream(
+ *param.get(), true /*output*/, param->stream());
+ c2_status_t err = intf->config(
+ { outParam.get() }, C2_MAY_BLOCK, &failures);
+ if (err == C2_OK) {
+ work->worklets.front()->output.configUpdate.push_back(
+ C2Param::Copy(*outParam.get()));
+ } else {
+ ALOGE("finishWork: Config update size failed");
+ }
+ break;
+ }
+ }
+ }
};
if (work && c2_cntr64_t(index) == work->input.ordinal.frameIndex) {
fillWork(work);
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index f903bbb..852d6d6 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -1348,9 +1348,7 @@
}
void CCodec::signalSetParameters(const sp<AMessage> ¶ms) {
- sp<AMessage> msg = new AMessage(kWhatSetParameters, this);
- msg->setMessage("params", params);
- msg->post();
+ setParameters(params);
}
void CCodec::setParameters(const sp<AMessage> ¶ms) {
@@ -1515,13 +1513,6 @@
setInputSurface(surface);
break;
}
- case kWhatSetParameters: {
- setDeadline(now, 50ms, "setParameters");
- sp<AMessage> params;
- CHECK(msg->findMessage("params", ¶ms));
- setParameters(params);
- break;
- }
case kWhatWorkDone: {
std::unique_ptr<C2Work> work;
size_t numDiscardedInputBuffers;
@@ -1594,6 +1585,7 @@
C2StreamColorAspectsInfo::output::PARAM_TYPE,
C2StreamDataSpaceInfo::output::PARAM_TYPE,
C2StreamHdrStaticInfo::output::PARAM_TYPE,
+ C2StreamHdr10PlusInfo::output::PARAM_TYPE,
C2StreamPixelAspectRatioInfo::output::PARAM_TYPE,
C2StreamSurfaceScalingInfo::output::PARAM_TYPE
};
@@ -1677,7 +1669,7 @@
deadline->set(std::chrono::steady_clock::now() + 3s, "eos");
}
// TODO: query and use input/pipeline/output delay combined
- if (count >= 8) {
+ if (count >= 4) {
CCodecWatchdog::getInstance()->watch(this);
Mutexed<NamedTimePoint>::Locked deadline(mQueueDeadline);
deadline->set(std::chrono::steady_clock::now() + 3s, "queue");
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 01b9c1e..55a97d8 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -787,8 +787,13 @@
std::unique_ptr<CCodecBufferChannel::InputBuffers> toArrayMode(
size_t size) final {
int32_t capacity = kLinearBufferSize;
- (void)mFormat->findInt32(C2_NAME_STREAM_MAX_BUFFER_SIZE_SETTING, &capacity);
-
+ (void)mFormat->findInt32(KEY_MAX_INPUT_SIZE, &capacity);
+ if ((size_t)capacity > kMaxLinearBufferSize) {
+ ALOGD("client requested %d, capped to %zu", capacity, kMaxLinearBufferSize);
+ capacity = kMaxLinearBufferSize;
+ }
+ // TODO: proper max input size
+ // TODO: read usage from intf
std::unique_ptr<InputBuffersArray> array(
new InputBuffersArray(mComponentName.c_str(), "1D-Input[N]"));
array->setPool(mPool);
@@ -1807,17 +1812,29 @@
status_t CCodecBufferChannel::renderOutputBuffer(
const sp<MediaCodecBuffer> &buffer, int64_t timestampNs) {
+ ALOGV("[%s] renderOutputBuffer: %p", mName, buffer.get());
std::shared_ptr<C2Buffer> c2Buffer;
+ bool released = false;
{
Mutexed<std::unique_ptr<OutputBuffers>>::Locked buffers(mOutputBuffers);
if (*buffers) {
- (*buffers)->releaseBuffer(buffer, &c2Buffer);
+ released = (*buffers)->releaseBuffer(buffer, &c2Buffer);
}
}
+ // NOTE: some apps try to releaseOutputBuffer() with timestamp and/or render
+ // set to true.
+ sendOutputBuffers();
+ // input buffer feeding may have been gated by pending output buffers
+ feedInputBufferIfAvailable();
if (!c2Buffer) {
+ if (released) {
+ ALOGD("[%s] The app is calling releaseOutputBuffer() with "
+ "timestamp or render=true with non-video buffers. Apps should "
+ "call releaseOutputBuffer() with render=false for those.",
+ mName);
+ }
return INVALID_OPERATION;
}
- sendOutputBuffers();
#if 0
const std::vector<std::shared_ptr<const C2Info>> infoParams = c2Buffer->info();
@@ -1871,6 +1888,11 @@
std::static_pointer_cast<const C2StreamHdrStaticInfo::output>(
c2Buffer->getInfo(C2StreamHdrStaticInfo::output::PARAM_TYPE));
+ // HDR10 plus info
+ std::shared_ptr<const C2StreamHdr10PlusInfo::output> hdr10PlusInfo =
+ std::static_pointer_cast<const C2StreamHdr10PlusInfo::output>(
+ c2Buffer->getInfo(C2StreamHdr10PlusInfo::output::PARAM_TYPE));
+
{
Mutexed<OutputSurface>::Locked output(mOutputSurface);
if (output->surface == nullptr) {
@@ -1898,35 +1920,45 @@
videoScalingMode,
transform,
Fence::NO_FENCE, 0);
- if (hdrStaticInfo) {
- struct android_smpte2086_metadata smpte2086_meta = {
- .displayPrimaryRed = {
- hdrStaticInfo->mastering.red.x, hdrStaticInfo->mastering.red.y
- },
- .displayPrimaryGreen = {
- hdrStaticInfo->mastering.green.x, hdrStaticInfo->mastering.green.y
- },
- .displayPrimaryBlue = {
- hdrStaticInfo->mastering.blue.x, hdrStaticInfo->mastering.blue.y
- },
- .whitePoint = {
- hdrStaticInfo->mastering.white.x, hdrStaticInfo->mastering.white.y
- },
- .maxLuminance = hdrStaticInfo->mastering.maxLuminance,
- .minLuminance = hdrStaticInfo->mastering.minLuminance,
- };
-
- struct android_cta861_3_metadata cta861_meta = {
- .maxContentLightLevel = hdrStaticInfo->maxCll,
- .maxFrameAverageLightLevel = hdrStaticInfo->maxFall,
- };
-
+ if (hdrStaticInfo || hdr10PlusInfo) {
HdrMetadata hdr;
- hdr.validTypes = HdrMetadata::SMPTE2086 | HdrMetadata::CTA861_3;
- hdr.smpte2086 = smpte2086_meta;
- hdr.cta8613 = cta861_meta;
+ if (hdrStaticInfo) {
+ struct android_smpte2086_metadata smpte2086_meta = {
+ .displayPrimaryRed = {
+ hdrStaticInfo->mastering.red.x, hdrStaticInfo->mastering.red.y
+ },
+ .displayPrimaryGreen = {
+ hdrStaticInfo->mastering.green.x, hdrStaticInfo->mastering.green.y
+ },
+ .displayPrimaryBlue = {
+ hdrStaticInfo->mastering.blue.x, hdrStaticInfo->mastering.blue.y
+ },
+ .whitePoint = {
+ hdrStaticInfo->mastering.white.x, hdrStaticInfo->mastering.white.y
+ },
+ .maxLuminance = hdrStaticInfo->mastering.maxLuminance,
+ .minLuminance = hdrStaticInfo->mastering.minLuminance,
+ };
+
+ struct android_cta861_3_metadata cta861_meta = {
+ .maxContentLightLevel = hdrStaticInfo->maxCll,
+ .maxFrameAverageLightLevel = hdrStaticInfo->maxFall,
+ };
+
+ hdr.validTypes = HdrMetadata::SMPTE2086 | HdrMetadata::CTA861_3;
+ hdr.smpte2086 = smpte2086_meta;
+ hdr.cta8613 = cta861_meta;
+ }
+ if (hdr10PlusInfo) {
+ hdr.validTypes |= HdrMetadata::HDR10PLUS;
+ hdr.hdr10plus.assign(
+ hdr10PlusInfo->m.value,
+ hdr10PlusInfo->m.value + hdr10PlusInfo->flexCount());
+ }
qbi.setHdrMetadata(hdr);
}
+ // we don't have dirty regions
+ qbi.setSurfaceDamage(Region::INVALID_REGION);
android::IGraphicBufferProducer::QueueBufferOutput qbo;
status_t result = mComponent->queueToOutputSurface(block, qbi, &qbo);
if (result != OK) {
@@ -1961,8 +1993,8 @@
}
}
if (released) {
- feedInputBufferIfAvailable();
sendOutputBuffers();
+ feedInputBufferIfAvailable();
} else {
ALOGD("[%s] MediaCodec discarded an unknown buffer", mName);
}
diff --git a/media/codec2/sfplugin/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index 8dbfd0e..ef02e74 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -570,6 +570,12 @@
add(ConfigMapper("csd-0", C2_PARAMKEY_INIT_DATA, "value")
.limitTo(D::OUTPUT & D::READ));
+ add(ConfigMapper(KEY_HDR10_PLUS_INFO, C2_PARAMKEY_INPUT_HDR10_PLUS_INFO, "value")
+ .limitTo(D::VIDEO & D::PARAM & D::INPUT));
+
+ add(ConfigMapper(KEY_HDR10_PLUS_INFO, C2_PARAMKEY_OUTPUT_HDR10_PLUS_INFO, "value")
+ .limitTo(D::VIDEO & D::OUTPUT));
+
add(ConfigMapper(C2_PARAMKEY_TEMPORAL_LAYERING, C2_PARAMKEY_TEMPORAL_LAYERING, "")
.limitTo(D::ENCODER & D::VIDEO & D::OUTPUT));
@@ -624,7 +630,23 @@
.limitTo(D::AUDIO & D::CODED));
add(ConfigMapper(KEY_PCM_ENCODING, C2_PARAMKEY_PCM_ENCODING, "value")
- .limitTo(D::AUDIO));
+ .limitTo(D::AUDIO)
+ .withMappers([](C2Value v) -> C2Value {
+ int32_t value;
+ C2Config::pcm_encoding_t to;
+ if (v.get(&value) && C2Mapper::map(value, &to)) {
+ return to;
+ }
+ return C2Value();
+ }, [](C2Value v) -> C2Value {
+ C2Config::pcm_encoding_t value;
+ int32_t to;
+ using C2ValueType=typename _c2_reduce_enum_to_underlying_type<decltype(value)>::type;
+ if (v.get((C2ValueType*)&value) && C2Mapper::map(value, &to)) {
+ return to;
+ }
+ return C2Value();
+ }));
add(ConfigMapper(KEY_IS_ADTS, C2_PARAMKEY_AAC_PACKAGING, "value")
.limitTo(D::AUDIO & D::CODED)
diff --git a/media/codec2/sfplugin/Codec2Buffer.cpp b/media/codec2/sfplugin/Codec2Buffer.cpp
index bf6062e..1113ae8 100644
--- a/media/codec2/sfplugin/Codec2Buffer.cpp
+++ b/media/codec2/sfplugin/Codec2Buffer.cpp
@@ -109,9 +109,11 @@
// DummyContainerBuffer
+static uint8_t sDummyByte[1] = { 0 };
+
DummyContainerBuffer::DummyContainerBuffer(
const sp<AMessage> &format, const std::shared_ptr<C2Buffer> &buffer)
- : Codec2Buffer(format, new ABuffer(nullptr, 1)),
+ : Codec2Buffer(format, new ABuffer(sDummyByte, 1)),
mBufferRef(buffer) {
setRange(0, buffer ? 1 : 0);
}
diff --git a/media/codec2/sfplugin/utils/Codec2BufferUtils.cpp b/media/codec2/sfplugin/utils/Codec2BufferUtils.cpp
index b7519da..84d22a3 100644
--- a/media/codec2/sfplugin/utils/Codec2BufferUtils.cpp
+++ b/media/codec2/sfplugin/utils/Codec2BufferUtils.cpp
@@ -88,16 +88,30 @@
uint32_t planeW = img->mWidth / plane.colSampling;
uint32_t planeH = img->mHeight / plane.rowSampling;
- for (uint32_t row = 0; row < planeH; ++row) {
- decltype(imgRow) imgPtr = imgRow;
- decltype(viewRow) viewPtr = viewRow;
- for (uint32_t col = 0; col < planeW; ++col) {
- MemCopier<ToMediaImage, 0>::copy(imgPtr, viewPtr, bpp);
- imgPtr += img->mPlane[i].mColInc;
- viewPtr += plane.colInc;
+
+ bool canCopyByRow = (plane.colInc == 1) && (img->mPlane[i].mColInc == 1);
+ bool canCopyByPlane = canCopyByRow && (plane.rowInc == img->mPlane[i].mRowInc);
+ if (canCopyByPlane) {
+ MemCopier<ToMediaImage, 0>::copy(imgRow, viewRow, plane.rowInc * planeH);
+ } else if (canCopyByRow) {
+ for (uint32_t row = 0; row < planeH; ++row) {
+ MemCopier<ToMediaImage, 0>::copy(
+ imgRow, viewRow, std::min(plane.rowInc, img->mPlane[i].mRowInc));
+ imgRow += img->mPlane[i].mRowInc;
+ viewRow += plane.rowInc;
}
- imgRow += img->mPlane[i].mRowInc;
- viewRow += plane.rowInc;
+ } else {
+ for (uint32_t row = 0; row < planeH; ++row) {
+ decltype(imgRow) imgPtr = imgRow;
+ decltype(viewRow) viewPtr = viewRow;
+ for (uint32_t col = 0; col < planeW; ++col) {
+ MemCopier<ToMediaImage, 0>::copy(imgPtr, viewPtr, bpp);
+ imgPtr += img->mPlane[i].mColInc;
+ viewPtr += plane.colInc;
+ }
+ imgRow += img->mPlane[i].mRowInc;
+ viewRow += plane.rowInc;
+ }
}
}
return OK;
diff --git a/media/extractors/flac/FLACExtractor.cpp b/media/extractors/flac/FLACExtractor.cpp
index 22b96e5..4e04605 100644
--- a/media/extractors/flac/FLACExtractor.cpp
+++ b/media/extractors/flac/FLACExtractor.cpp
@@ -586,9 +586,6 @@
void FLACParser::releaseBuffers()
{
- CHECK(mGroup != NULL);
- delete mGroup;
- mGroup = NULL;
}
MediaBufferHelperV3 *FLACParser::readBuffer(bool doSeek, FLAC__uint64 sample)
diff --git a/media/extractors/ogg/Android.bp b/media/extractors/ogg/Android.bp
index 01acb2c..b28877d 100644
--- a/media/extractors/ogg/Android.bp
+++ b/media/extractors/ogg/Android.bp
@@ -7,6 +7,10 @@
"external/tremolo",
],
+ header_libs: [
+ "libaudio_system_headers",
+ ],
+
shared_libs: [
"liblog",
"libmediaextractor",
diff --git a/media/extractors/ogg/OggExtractor.cpp b/media/extractors/ogg/OggExtractor.cpp
index a52ccb1..cc2c792 100644
--- a/media/extractors/ogg/OggExtractor.cpp
+++ b/media/extractors/ogg/OggExtractor.cpp
@@ -34,6 +34,7 @@
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaDataBase.h>
#include <media/stagefright/MetaDataUtils.h>
+#include <system/audio.h>
#include <utils/String8.h>
extern "C" {
@@ -133,6 +134,8 @@
Vector<TOCEntry> mTableOfContents;
+ int32_t mHapticChannelCount;
+
ssize_t readPage(off64_t offset, Page *page);
status_t findNextPage(off64_t startOffset, off64_t *pageOffset);
@@ -163,6 +166,8 @@
void buildTableOfContents();
+ void setChannelMask(int channelCount);
+
MyOggExtractor(const MyOggExtractor &);
MyOggExtractor &operator=(const MyOggExtractor &);
};
@@ -310,7 +315,8 @@
mMimeType(mimeType),
mNumHeaders(numHeaders),
mSeekPreRollUs(seekPreRollUs),
- mFirstDataOffset(-1) {
+ mFirstDataOffset(-1),
+ mHapticChannelCount(0) {
mCurrentPage.mNumSegments = 0;
vorbis_info_init(&mVi);
@@ -1083,6 +1089,7 @@
}
parseFileMetaData();
+ setChannelMask(mChannelCount);
return AMEDIA_OK;
}
@@ -1157,6 +1164,7 @@
}
parseFileMetaData();
+ setChannelMask(mVi.channels);
break;
}
@@ -1192,6 +1200,29 @@
parseVorbisComment(mFileMeta, comment, commentLength);
//ALOGI("comment #%d: '%s'", i + 1, mVc.user_comments[i]);
}
+
+ AMediaFormat_getInt32(mFileMeta, "haptic", &mHapticChannelCount);
+}
+
+void MyOggExtractor::setChannelMask(int channelCount) {
+ // Set channel mask according to channel count. When haptic channel count is found in
+ // file meta, set haptic channel mask to try haptic playback.
+ if (mHapticChannelCount > 0) {
+ const audio_channel_mask_t hapticChannelMask =
+ haptic_channel_mask_from_count(mHapticChannelCount);
+ const int32_t audioChannelCount = channelCount - mHapticChannelCount;
+ if (hapticChannelMask == AUDIO_CHANNEL_INVALID
+ || audioChannelCount <= 0 || audioChannelCount > FCC_8) {
+ ALOGE("Invalid haptic channel count found in metadata: %d", mHapticChannelCount);
+ } else {
+ const audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(
+ audioChannelCount) | hapticChannelMask;
+ AMediaFormat_setInt32(mMeta, AMEDIAFORMAT_KEY_CHANNEL_MASK, channelMask);
+ }
+ } else {
+ AMediaFormat_setInt32(mMeta, AMEDIAFORMAT_KEY_CHANNEL_MASK,
+ audio_channel_out_mask_from_count(channelCount));
+ }
}
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index efe65bb..3e91717 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -1315,6 +1315,13 @@
return aps->setA11yServicesUids(uids);
}
+bool AudioSystem::isHapticPlaybackSupported()
+{
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return false;
+ return aps->isHapticPlaybackSupported();
+}
+
// ---------------------------------------------------------------------------
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 8f06ee7..1f6dd60 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -66,7 +66,7 @@
static int64_t convertTimespecToUs(const struct timespec &tv)
{
- return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000;
+ return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
}
// TODO move to audio_utils.
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 86e417a..0ce8b16 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -89,6 +89,7 @@
REMOVE_SOURCE_DEFAULT_EFFECT,
SET_ASSISTANT_UID,
SET_A11Y_SERVICES_UIDS,
+ IS_HAPTIC_PLAYBACK_SUPPORTED,
};
#define MAX_ITEMS_PER_LIST 1024
@@ -978,6 +979,17 @@
return static_cast <status_t> (reply.readInt32());
}
+ virtual bool isHapticPlaybackSupported()
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ status_t status = remote()->transact(IS_HAPTIC_PLAYBACK_SUPPORTED, data, &reply);
+ if (status != NO_ERROR) {
+ return false;
+ }
+ return reply.readBool();
+ }
+
};
IMPLEMENT_META_INTERFACE(AudioPolicyService, "android.media.IAudioPolicyService");
@@ -1795,6 +1807,13 @@
return NO_ERROR;
}
+ case IS_HAPTIC_PLAYBACK_SUPPORTED: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ bool isSupported = isHapticPlaybackSupported();
+ reply->writeBool(isSupported);
+ return NO_ERROR;
+ }
+
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index ca1879f..74156ca 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -345,6 +345,8 @@
static status_t setAssistantUid(uid_t uid);
static status_t setA11yServicesUids(const std::vector<uid_t>& uids);
+ static bool isHapticPlaybackSupported();
+
// ----------------------------------------------------------------------------
class AudioPortCallback : public RefBase
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index e3386a4..61f3b27 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -185,6 +185,8 @@
virtual status_t setAssistantUid(uid_t uid) = 0;
virtual status_t setA11yServicesUids(const std::vector<uid_t>& uids) = 0;
+
+ virtual bool isHapticPlaybackSupported() = 0;
};
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index e0f5a40..08c6a50 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -48,7 +48,8 @@
{"amrwb", AUDIO_ENCODER_AMR_WB},
{"aac", AUDIO_ENCODER_AAC},
{"heaac", AUDIO_ENCODER_HE_AAC},
- {"aaceld", AUDIO_ENCODER_AAC_ELD}
+ {"aaceld", AUDIO_ENCODER_AAC_ELD},
+ {"opus", AUDIO_ENCODER_OPUS}
};
const MediaProfiles::NameToTagMap MediaProfiles::sFileFormatMap[] = {
diff --git a/media/libmedia/TypeConverter.cpp b/media/libmedia/TypeConverter.cpp
index 514c795..fb861d7 100644
--- a/media/libmedia/TypeConverter.cpp
+++ b/media/libmedia/TypeConverter.cpp
@@ -233,6 +233,12 @@
MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_7POINT1POINT2),
MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_7POINT1POINT4),
+ MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_HAPTIC_A),
+ MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_MONO_HAPTIC_A),
+ MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_STEREO_HAPTIC_A),
+ MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_HAPTIC_AB),
+ MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_MONO_HAPTIC_AB),
+ MAKE_STRING_FROM_ENUM(AUDIO_CHANNEL_OUT_STEREO_HAPTIC_AB),
TERMINATOR
};
diff --git a/media/libmedia/include/media/mediarecorder.h b/media/libmedia/include/media/mediarecorder.h
index d8b0fe7..bdf1aae 100644
--- a/media/libmedia/include/media/mediarecorder.h
+++ b/media/libmedia/include/media/mediarecorder.h
@@ -67,7 +67,7 @@
OUTPUT_FORMAT_AAC_ADTS = 6,
OUTPUT_FORMAT_AUDIO_ONLY_END = 7, // Used in validating the output format. Should be the
- // at the end of the audio only output formats.
+ // at the end of the audio only output formats.
/* Stream over a socket, limited to a single stream */
OUTPUT_FORMAT_RTP_AVP = 7,
@@ -81,6 +81,9 @@
/* HEIC data in a HEIF container */
OUTPUT_FORMAT_HEIF = 10,
+ /* Opus data in a OGG container */
+ OUTPUT_FORMAT_OGG = 11,
+
OUTPUT_FORMAT_LIST_END // must be last - used to validate format type
};
@@ -92,6 +95,7 @@
AUDIO_ENCODER_HE_AAC = 4,
AUDIO_ENCODER_AAC_ELD = 5,
AUDIO_ENCODER_VORBIS = 6,
+ AUDIO_ENCODER_OPUS = 7,
AUDIO_ENCODER_LIST_END // must be the last - used to validate the audio encoder type
};
diff --git a/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.cpp b/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.cpp
index 2ea55f6..e53900b 100644
--- a/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.cpp
+++ b/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.cpp
@@ -273,10 +273,10 @@
if (fetchType == LiveSession::STREAMTYPE_SUBTITLES) {
notify->post();
- msg->post(delayUs > 0ll ? delayUs : 0ll);
+ msg->post(delayUs > 0LL ? delayUs : 0LL);
return;
} else if (fetchType == LiveSession::STREAMTYPE_METADATA) {
- if (delayUs < -1000000ll) { // 1 second
+ if (delayUs < -1000000LL) { // 1 second
continue;
}
notify->post();
@@ -288,7 +288,7 @@
}
// try again in 1 second
- msg->post(1000000ll);
+ msg->post(1000000LL);
}
void NuPlayer2::HTTPLiveSource2::onMessageReceived(const sp<AMessage> &msg) {
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp
index 1561850..81ffbc7 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp
@@ -926,7 +926,7 @@
}
}
- msg->post(1000000ll); // poll again in a second.
+ msg->post(1000000LL); // poll again in a second.
break;
}
@@ -1194,7 +1194,7 @@
}
if (rescan) {
- msg->post(100000ll);
+ msg->post(100000LL);
mScanSourcesPending = true;
}
break;
@@ -2870,7 +2870,7 @@
int64_t posMs;
int64_t timeUs, posUs;
driver->getCurrentPosition(&posMs);
- posUs = posMs * 1000ll;
+ posUs = posMs * 1000LL;
CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
if (posUs < timeUs) {
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.cpp
index a9f2104..98c3403 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.cpp
@@ -567,7 +567,7 @@
ccBuf->meta()->setInt32(AMEDIAFORMAT_KEY_TRACK_INDEX, mSelectedTrack);
ccBuf->meta()->setInt64("timeUs", timeUs);
- ccBuf->meta()->setInt64("durationUs", 0ll);
+ ccBuf->meta()->setInt64("durationUs", 0LL);
sp<AMessage> msg = mNotify->dup();
msg->setInt32("what", kWhatClosedCaptionData);
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
index 931b86e..49e3e3b 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
@@ -71,10 +71,10 @@
mCCDecoder(ccDecoder),
mPid(pid),
mUid(uid),
- mSkipRenderingUntilMediaTimeUs(-1ll),
- mNumFramesTotal(0ll),
- mNumInputFramesDropped(0ll),
- mNumOutputFramesDropped(0ll),
+ mSkipRenderingUntilMediaTimeUs(-1LL),
+ mNumFramesTotal(0LL),
+ mNumInputFramesDropped(0LL),
+ mNumOutputFramesDropped(0LL),
mVideoWidth(0),
mVideoHeight(0),
mIsAudio(true),
@@ -428,10 +428,10 @@
// TODO: For now, layer fps is calculated for some specific architectures.
// But it really should be extracted from the stream.
mVideoTemporalLayerAggregateFps[0] =
- mFrameRateTotal / (float)(1ll << (mNumVideoTemporalLayerTotal - 1));
+ mFrameRateTotal / (float)(1LL << (mNumVideoTemporalLayerTotal - 1));
for (int32_t i = 1; i < mNumVideoTemporalLayerTotal; ++i) {
mVideoTemporalLayerAggregateFps[i] =
- mFrameRateTotal / (float)(1ll << (mNumVideoTemporalLayerTotal - i))
+ mFrameRateTotal / (float)(1LL << (mNumVideoTemporalLayerTotal - i))
+ mVideoTemporalLayerAggregateFps[i - 1];
}
}
@@ -952,7 +952,7 @@
int32_t layerId = 0;
bool haveLayerId = accessUnit->meta()->findInt32("temporal-layer-id", &layerId);
- if (mRenderer->getVideoLateByUs() > 100000ll
+ if (mRenderer->getVideoLateByUs() > 100000LL
&& mIsVideoAVC
&& !IsAVCReferenceFrame(accessUnit)) {
dropAccessUnit = true;
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.cpp
index 1f1b69e..914f29f 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.cpp
@@ -122,7 +122,7 @@
mRequestInputBuffersPending = true;
sp<AMessage> msg = new AMessage(kWhatRequestInputBuffers, this);
- msg->post(10 * 1000ll);
+ msg->post(10 * 1000LL);
}
}
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.cpp
index 0e0c1d8..0514e88 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.cpp
@@ -46,7 +46,7 @@
: DecoderBase(notify),
mSource(source),
mRenderer(renderer),
- mSkipRenderingUntilMediaTimeUs(-1ll),
+ mSkipRenderingUntilMediaTimeUs(-1LL),
mReachedEOS(true),
mPendingAudioErr(OK),
mPendingBuffersToDrain(0),
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp
index eff8866..56d708a 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp
@@ -372,7 +372,7 @@
ALOGD("seekTo(%p) (%lld ms, %d) at state %d", this, (long long)msec, mode, mState);
Mutex::Autolock autoLock(mLock);
- int64_t seekTimeUs = msec * 1000ll;
+ int64_t seekTimeUs = msec * 1000LL;
switch (mState) {
case STATE_PREPARED:
@@ -426,7 +426,7 @@
return UNKNOWN_ERROR;
}
- *msec = (mDurationUs + 500ll) / 1000;
+ *msec = (mDurationUs + 500LL) / 1000;
return OK;
}
@@ -612,7 +612,7 @@
int64_t msec = 0;
// getCurrentPosition should always return OK
getCurrentPosition(&msec);
- return mPlayer->selectTrack(trackIndex, true /* select */, msec * 1000ll);
+ return mPlayer->selectTrack(trackIndex, true /* select */, msec * 1000LL);
}
case MEDIA_PLAYER2_INVOKE_ID_UNSELECT_TRACK:
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.cpp
index d800412..9d9e179 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.cpp
@@ -67,10 +67,10 @@
// Maximum time in paused state when offloading audio decompression. When elapsed, the AudioSink
// is closed to allow the audio DSP to power down.
-static const int64_t kOffloadPauseMaxUs = 10000000ll;
+static const int64_t kOffloadPauseMaxUs = 10000000LL;
// Maximum allowed delay from AudioSink, 1.5 seconds.
-static const int64_t kMaxAllowedAudioSinkDelayUs = 1500000ll;
+static const int64_t kMaxAllowedAudioSinkDelayUs = 1500000LL;
static const int64_t kMinimumAudioClockUpdatePeriodUs = 20 /* msec */ * 1000;
@@ -84,7 +84,7 @@
};
// static
-const int64_t NuPlayer2::Renderer::kMinPositionUpdateDelayUs = 100000ll;
+const int64_t NuPlayer2::Renderer::kMinPositionUpdateDelayUs = 100000LL;
NuPlayer2::Renderer::Renderer(
const sp<MediaPlayer2Interface::AudioSink> &sink,
@@ -108,7 +108,7 @@
mAudioFirstAnchorTimeMediaUs(-1),
mAnchorTimeMediaUs(-1),
mAnchorNumFramesWritten(-1),
- mVideoLateByUs(0ll),
+ mVideoLateByUs(0LL),
mNextVideoTimeMediaUs(-1),
mHasAudio(false),
mHasVideo(false),
@@ -1142,7 +1142,7 @@
int64_t nowUs = ALooper::GetNowUs();
int64_t mediaUs;
if (mMediaClock->getMediaTime(nowUs, &mediaUs) != OK) {
- return 0ll;
+ return 0LL;
} else {
return writtenAudioDurationUs - (mediaUs - mAudioFirstAnchorTimeMediaUs);
}
@@ -1357,7 +1357,7 @@
tooLate = false;
}
- entry->mNotifyConsumed->setInt64("timestampNs", realTimeUs * 1000ll);
+ entry->mNotifyConsumed->setInt64("timestampNs", realTimeUs * 1000LL);
entry->mNotifyConsumed->setInt32("render", !tooLate);
entry->mNotifyConsumed->post();
mVideoQueue.erase(mVideoQueue.begin());
@@ -1489,7 +1489,7 @@
ALOGV("queueDiff = %.2f secs", diff / 1E6);
- if (diff > 100000ll) {
+ if (diff > 100000LL) {
// Audio data starts More than 0.1 secs before video.
// Drop some audio.
diff --git a/media/libmediaplayer2/nuplayer2/RTSPSource2.cpp b/media/libmediaplayer2/nuplayer2/RTSPSource2.cpp
index aed925b..a70269e 100644
--- a/media/libmediaplayer2/nuplayer2/RTSPSource2.cpp
+++ b/media/libmediaplayer2/nuplayer2/RTSPSource2.cpp
@@ -30,7 +30,7 @@
namespace android {
-const int64_t kNearEOSTimeoutUs = 2000000ll; // 2 secs
+const int64_t kNearEOSTimeoutUs = 2000000LL; // 2 secs
// Default Buffer Underflow/Prepare/StartServer/Overflow Marks
static const int kUnderflowMarkMs = 1000; // 1 second
@@ -168,7 +168,7 @@
// We're going to buffer at least 2 secs worth data on all tracks before
// starting playback (both at startup and after a seek).
- static const int64_t kMinDurationUs = 2000000ll;
+ static const int64_t kMinDurationUs = 2000000LL;
int64_t mediaDurationUs = 0;
getDuration(&mediaDurationUs);
@@ -272,7 +272,7 @@
}
status_t NuPlayer2::RTSPSource2::getDuration(int64_t *durationUs) {
- *durationUs = -1ll;
+ *durationUs = -1LL;
int64_t audioDurationUs;
if (mAudioTrack != NULL
@@ -321,7 +321,7 @@
void NuPlayer2::RTSPSource2::schedulePollBuffering() {
sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
- msg->post(1000000ll); // 1 second intervals
+ msg->post(1000000LL); // 1 second intervals
}
void NuPlayer2::RTSPSource2::checkBuffering(
@@ -345,10 +345,10 @@
int64_t maxRebufferingMarkUs;
{
Mutex::Autolock _l(mBufferingSettingsLock);
- initialMarkUs = mBufferingSettings.mInitialMarkMs * 1000ll;
+ initialMarkUs = mBufferingSettings.mInitialMarkMs * 1000LL;
// TODO: maxRebufferingMarkUs could be larger than
// mBufferingSettings.mResumePlaybackMarkMs * 1000ll.
- maxRebufferingMarkUs = mBufferingSettings.mResumePlaybackMarkMs * 1000ll;
+ maxRebufferingMarkUs = mBufferingSettings.mResumePlaybackMarkMs * 1000LL;
}
// isFinished when duration is 0 checks for EOS result only
if (bufferedDurationUs > initialMarkUs
@@ -368,7 +368,7 @@
++overflowCount;
}
int64_t startServerMarkUs =
- (kUnderflowMarkMs * 1000ll + maxRebufferingMarkUs) / 2;
+ (kUnderflowMarkMs * 1000LL + maxRebufferingMarkUs) / 2;
if (bufferedDurationUs < startServerMarkUs) {
++startCount;
}
@@ -639,7 +639,7 @@
int64_t nptUs =
((double)rtpTime - (double)info->mRTPTime)
/ info->mTimeScale
- * 1000000ll
+ * 1000000LL
+ info->mNormalPlaytimeUs;
accessUnit->meta()->setInt64("timeUs", nptUs);
@@ -747,7 +747,7 @@
TrackInfo info;
info.mTimeScale = timeScale;
info.mRTPTime = 0;
- info.mNormalPlaytimeUs = 0ll;
+ info.mNormalPlaytimeUs = 0LL;
info.mNPTMappingValid = false;
if ((isAudio && mAudioTrack == NULL)
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index e3ae02e..eae52c2 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -46,6 +46,7 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaCodecSource.h>
+#include <media/stagefright/OggWriter.h>
#include <media/stagefright/PersistentSurface.h>
#include <media/MediaProfiles.h>
#include <camera/CameraParameters.h>
@@ -948,6 +949,10 @@
status = setupMPEG2TSRecording();
break;
+ case OUTPUT_FORMAT_OGG:
+ status = setupOggRecording();
+ break;
+
default:
ALOGE("Unsupported output file format: %d", mOutputFormat);
status = UNKNOWN_ERROR;
@@ -1013,6 +1018,7 @@
case OUTPUT_FORMAT_AAC_ADTS:
case OUTPUT_FORMAT_RTP_AVP:
case OUTPUT_FORMAT_MPEG2TS:
+ case OUTPUT_FORMAT_OGG:
{
sp<MetaData> meta = new MetaData;
int64_t startTimeUs = systemTime() / 1000;
@@ -1113,6 +1119,9 @@
format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
format->setInt32("aac-profile", OMX_AUDIO_AACObjectELD);
break;
+ case AUDIO_ENCODER_OPUS:
+ format->setString("mime", MEDIA_MIMETYPE_AUDIO_OPUS);
+ break;
default:
ALOGE("Unknown audio encoder: %d", mAudioEncoder);
@@ -1169,6 +1178,13 @@
return setupRawAudioRecording();
}
+status_t StagefrightRecorder::setupOggRecording() {
+ CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_OGG);
+
+ mWriter = new OggWriter(mOutputFd);
+ return setupRawAudioRecording();
+}
+
status_t StagefrightRecorder::setupAMRRecording() {
CHECK(mOutputFormat == OUTPUT_FORMAT_AMR_NB ||
mOutputFormat == OUTPUT_FORMAT_AMR_WB);
@@ -1813,6 +1829,7 @@
case AUDIO_ENCODER_AAC:
case AUDIO_ENCODER_HE_AAC:
case AUDIO_ENCODER_AAC_ELD:
+ case AUDIO_ENCODER_OPUS:
break;
default:
@@ -1863,19 +1880,18 @@
mTotalBitRate += mVideoBitRate;
}
- if (mOutputFormat != OUTPUT_FORMAT_WEBM) {
- // Audio source is added at the end if it exists.
- // This help make sure that the "recoding" sound is suppressed for
- // camcorder applications in the recorded files.
- // TODO Audio source is currently unsupported for webm output; vorbis encoder needed.
- // disable audio for time lapse recording
- bool disableAudio = mCaptureFpsEnable && mCaptureFps < mFrameRate;
- if (!disableAudio && mAudioSource != AUDIO_SOURCE_CNT) {
- err = setupAudioEncoder(writer);
- if (err != OK) return err;
- mTotalBitRate += mAudioBitRate;
- }
+ // Audio source is added at the end if it exists.
+ // This help make sure that the "recoding" sound is suppressed for
+ // camcorder applications in the recorded files.
+ // disable audio for time lapse recording
+ const bool disableAudio = mCaptureFpsEnable && mCaptureFps < mFrameRate;
+ if (!disableAudio && mAudioSource != AUDIO_SOURCE_CNT) {
+ err = setupAudioEncoder(writer);
+ if (err != OK) return err;
+ mTotalBitRate += mAudioBitRate;
+ }
+ if (mOutputFormat != OUTPUT_FORMAT_WEBM) {
if (mCaptureFpsEnable) {
mp4writer->setCaptureRate(mCaptureFps);
}
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index faa2e59..2ada301 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -166,6 +166,7 @@
void setupMPEG4orWEBMMetaData(sp<MetaData> *meta);
status_t setupAMRRecording();
status_t setupAACRecording();
+ status_t setupOggRecording();
status_t setupRawAudioRecording();
status_t setupRTPRecording();
status_t setupMPEG2TSRecording();
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index f3b69d6..e2aa8f8 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -68,7 +68,7 @@
mVideoDataGeneration(0),
mFetchSubtitleDataGeneration(0),
mFetchTimedTextDataGeneration(0),
- mDurationUs(-1ll),
+ mDurationUs(-1LL),
mAudioIsVorbis(false),
mIsSecure(false),
mIsStreaming(false),
@@ -76,7 +76,7 @@
mUID(uid),
mMediaClock(mediaClock),
mFd(-1),
- mBitrate(-1ll),
+ mBitrate(-1LL),
mPendingReadBufferTypes(0) {
ALOGV("GenericSource");
CHECK(mediaClock != NULL);
@@ -727,7 +727,7 @@
}
if (msg->what() == kWhatFetchSubtitleData) {
- subTimeUs -= 1000000ll; // send subtile data one second earlier
+ subTimeUs -= 1000000LL; // send subtile data one second earlier
}
sp<AMessage> msg2 = new AMessage(sendWhat, this);
msg2->setInt32("generation", msgGeneration);
@@ -764,7 +764,7 @@
notify->post();
if (msg->what() == kWhatSendSubtitleData) {
- nextSubTimeUs -= 1000000ll; // send subtile data one second earlier
+ nextSubTimeUs -= 1000000LL; // send subtile data one second earlier
}
mMediaClock->addTimer(msg, nextSubTimeUs);
}
@@ -855,7 +855,7 @@
// TODO: maxRebufferingMarkMs could be larger than
// mBufferingSettings.mResumePlaybackMarkMs
int64_t restartBufferingMarkUs =
- mBufferingSettings.mResumePlaybackMarkMs * 1000ll / 2;
+ mBufferingSettings.mResumePlaybackMarkMs * 1000LL / 2;
if (finalResult == OK) {
if (durationUs < restartBufferingMarkUs) {
postReadBuffer(audio? MEDIA_TRACK_TYPE_AUDIO : MEDIA_TRACK_TYPE_VIDEO);
@@ -1446,7 +1446,7 @@
// TODO: maxRebufferingMarkMs could be larger than
// mBufferingSettings.mResumePlaybackMarkMs
int64_t markUs = (mPreparing ? mBufferingSettings.mInitialMarkMs
- : mBufferingSettings.mResumePlaybackMarkMs) * 1000ll;
+ : mBufferingSettings.mResumePlaybackMarkMs) * 1000LL;
if (finalResult == ERROR_END_OF_STREAM || durationUs >= markUs) {
if (mPreparing || mSentPauseOnBuffering) {
Track *counterTrack =
@@ -1514,12 +1514,12 @@
sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
msg->setInt32("generation", mPollBufferingGeneration);
// Enquires buffering status every second.
- msg->post(1000000ll);
+ msg->post(1000000LL);
}
void NuPlayer::GenericSource::onPollBuffering() {
status_t finalStatus = UNKNOWN_ERROR;
- int64_t cachedDurationUs = -1ll;
+ int64_t cachedDurationUs = -1LL;
ssize_t cachedDataRemaining = -1;
if (mCachedSource != NULL) {
@@ -1527,15 +1527,15 @@
if (finalStatus == OK) {
off64_t size;
- int64_t bitrate = 0ll;
+ int64_t bitrate = 0LL;
if (mDurationUs > 0 && mCachedSource->getSize(&size) == OK) {
// |bitrate| uses bits/second unit, while size is number of bytes.
- bitrate = size * 8000000ll / mDurationUs;
+ bitrate = size * 8000000LL / mDurationUs;
} else if (mBitrate > 0) {
bitrate = mBitrate;
}
if (bitrate > 0) {
- cachedDurationUs = cachedDataRemaining * 8000000ll / bitrate;
+ cachedDurationUs = cachedDataRemaining * 8000000LL / bitrate;
}
}
}
@@ -1560,8 +1560,8 @@
return;
}
- if (cachedDurationUs >= 0ll) {
- if (mDurationUs > 0ll) {
+ if (cachedDurationUs >= 0LL) {
+ if (mDurationUs > 0LL) {
int64_t cachedPosUs = getLastReadPosition() + cachedDurationUs;
int percentage = 100.0 * cachedPosUs / mDurationUs;
if (percentage > 100) {
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
index 11f1bfd..77e7885 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
@@ -271,10 +271,10 @@
if (fetchType == LiveSession::STREAMTYPE_SUBTITLES) {
notify->post();
- msg->post(delayUs > 0ll ? delayUs : 0ll);
+ msg->post(delayUs > 0LL ? delayUs : 0LL);
return;
} else if (fetchType == LiveSession::STREAMTYPE_METADATA) {
- if (delayUs < -1000000ll) { // 1 second
+ if (delayUs < -1000000LL) { // 1 second
continue;
}
notify->post();
@@ -286,7 +286,7 @@
}
// try again in 1 second
- msg->post(1000000ll);
+ msg->post(1000000LL);
}
void NuPlayer::HTTPLiveSource::onMessageReceived(const sp<AMessage> &msg) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 3922767..5cf6bbd 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -761,7 +761,7 @@
}
}
- msg->post(1000000ll); // poll again in a second.
+ msg->post(1000000LL); // poll again in a second.
break;
}
@@ -1049,7 +1049,7 @@
}
if (rescan) {
- msg->post(100000ll);
+ msg->post(100000LL);
mScanSourcesPending = true;
}
break;
@@ -2670,7 +2670,7 @@
int posMs;
int64_t timeUs, posUs;
driver->getCurrentPosition(&posMs);
- posUs = (int64_t) posMs * 1000ll;
+ posUs = (int64_t) posMs * 1000LL;
CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
if (posUs < timeUs) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
index ec30d0c..0156ad2 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
@@ -554,7 +554,7 @@
ccBuf->meta()->setInt32("track-index", mSelectedTrack);
ccBuf->meta()->setInt64("timeUs", timeUs);
- ccBuf->meta()->setInt64("durationUs", 0ll);
+ ccBuf->meta()->setInt64("durationUs", 0LL);
sp<AMessage> msg = mNotify->dup();
msg->setInt32("what", kWhatClosedCaptionData);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index a2ec699..df1ffde 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -71,10 +71,10 @@
mCCDecoder(ccDecoder),
mPid(pid),
mUid(uid),
- mSkipRenderingUntilMediaTimeUs(-1ll),
- mNumFramesTotal(0ll),
- mNumInputFramesDropped(0ll),
- mNumOutputFramesDropped(0ll),
+ mSkipRenderingUntilMediaTimeUs(-1LL),
+ mNumFramesTotal(0LL),
+ mNumInputFramesDropped(0LL),
+ mNumOutputFramesDropped(0LL),
mVideoWidth(0),
mVideoHeight(0),
mIsAudio(true),
@@ -409,10 +409,10 @@
// TODO: For now, layer fps is calculated for some specific architectures.
// But it really should be extracted from the stream.
mVideoTemporalLayerAggregateFps[0] =
- mFrameRateTotal / (float)(1ll << (mNumVideoTemporalLayerTotal - 1));
+ mFrameRateTotal / (float)(1LL << (mNumVideoTemporalLayerTotal - 1));
for (int32_t i = 1; i < mNumVideoTemporalLayerTotal; ++i) {
mVideoTemporalLayerAggregateFps[i] =
- mFrameRateTotal / (float)(1ll << (mNumVideoTemporalLayerTotal - i))
+ mFrameRateTotal / (float)(1LL << (mNumVideoTemporalLayerTotal - i))
+ mVideoTemporalLayerAggregateFps[i - 1];
}
}
@@ -934,7 +934,7 @@
int32_t layerId = 0;
bool haveLayerId = accessUnit->meta()->findInt32("temporal-layer-id", &layerId);
- if (mRenderer->getVideoLateByUs() > 100000ll
+ if (mRenderer->getVideoLateByUs() > 100000LL
&& mIsVideoAVC
&& !IsAVCReferenceFrame(accessUnit)) {
dropAccessUnit = true;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.cpp
index d0de7b0..3e96d27 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.cpp
@@ -120,7 +120,7 @@
mRequestInputBuffersPending = true;
sp<AMessage> msg = new AMessage(kWhatRequestInputBuffers, this);
- msg->post(10 * 1000ll);
+ msg->post(10 * 1000LL);
}
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
index 6b05b53..0997e7d 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
@@ -47,7 +47,7 @@
: DecoderBase(notify),
mSource(source),
mRenderer(renderer),
- mSkipRenderingUntilMediaTimeUs(-1ll),
+ mSkipRenderingUntilMediaTimeUs(-1LL),
mReachedEOS(true),
mPendingAudioErr(OK),
mPendingBuffersToDrain(0),
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 44f223d..ba3ebaa 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -474,7 +474,7 @@
ALOGD("seekTo(%p) (%d ms, %d) at state %d", this, msec, mode, mState);
Mutex::Autolock autoLock(mLock);
- int64_t seekTimeUs = msec * 1000ll;
+ int64_t seekTimeUs = msec * 1000LL;
switch (mState) {
case STATE_PREPARED:
@@ -531,7 +531,7 @@
return UNKNOWN_ERROR;
}
- *msec = (mDurationUs + 500ll) / 1000;
+ *msec = (mDurationUs + 500LL) / 1000;
return OK;
}
@@ -744,7 +744,7 @@
int msec = 0;
// getCurrentPosition should always return OK
getCurrentPosition(&msec);
- return mPlayer->selectTrack(trackIndex, true /* select */, msec * 1000ll);
+ return mPlayer->selectTrack(trackIndex, true /* select */, msec * 1000LL);
}
case INVOKE_ID_UNSELECT_TRACK:
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index b258332..c8f6738 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -70,10 +70,10 @@
// Maximum time in paused state when offloading audio decompression. When elapsed, the AudioSink
// is closed to allow the audio DSP to power down.
-static const int64_t kOffloadPauseMaxUs = 10000000ll;
+static const int64_t kOffloadPauseMaxUs = 10000000LL;
// Maximum allowed delay from AudioSink, 1.5 seconds.
-static const int64_t kMaxAllowedAudioSinkDelayUs = 1500000ll;
+static const int64_t kMaxAllowedAudioSinkDelayUs = 1500000LL;
static const int64_t kMinimumAudioClockUpdatePeriodUs = 20 /* msec */ * 1000;
@@ -125,7 +125,7 @@
mAudioFirstAnchorTimeMediaUs(-1),
mAnchorTimeMediaUs(-1),
mAnchorNumFramesWritten(-1),
- mVideoLateByUs(0ll),
+ mVideoLateByUs(0LL),
mNextVideoTimeMediaUs(-1),
mHasAudio(false),
mHasVideo(false),
@@ -580,7 +580,7 @@
// play back.
int64_t delayUs =
mAudioSink->msecsPerFrame()
- * numFramesPendingPlayout * 1000ll;
+ * numFramesPendingPlayout * 1000LL;
if (mPlaybackRate > 1.0f) {
delayUs /= mPlaybackRate;
}
@@ -1172,7 +1172,7 @@
int64_t nowUs = ALooper::GetNowUs();
int64_t mediaUs;
if (mMediaClock->getMediaTime(nowUs, &mediaUs) != OK) {
- return 0ll;
+ return 0LL;
} else {
return writtenAudioDurationUs - (mediaUs - mAudioFirstAnchorTimeMediaUs);
}
@@ -1387,7 +1387,7 @@
tooLate = false;
}
- entry->mNotifyConsumed->setInt64("timestampNs", realTimeUs * 1000ll);
+ entry->mNotifyConsumed->setInt64("timestampNs", realTimeUs * 1000LL);
entry->mNotifyConsumed->setInt32("render", !tooLate);
entry->mNotifyConsumed->post();
mVideoQueue.erase(mVideoQueue.begin());
@@ -1519,7 +1519,7 @@
ALOGV("queueDiff = %.2f secs", diff / 1E6);
- if (diff > 100000ll) {
+ if (diff > 100000LL) {
// Audio data starts More than 0.1 secs before video.
// Drop some audio.
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index 851217b..bf14ec2 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -30,7 +30,7 @@
namespace android {
-const int64_t kNearEOSTimeoutUs = 2000000ll; // 2 secs
+const int64_t kNearEOSTimeoutUs = 2000000LL; // 2 secs
// Default Buffer Underflow/Prepare/StartServer/Overflow Marks
static const int kUnderflowMarkMs = 1000; // 1 second
@@ -169,7 +169,7 @@
// We're going to buffer at least 2 secs worth data on all tracks before
// starting playback (both at startup and after a seek).
- static const int64_t kMinDurationUs = 2000000ll;
+ static const int64_t kMinDurationUs = 2000000LL;
int64_t mediaDurationUs = 0;
getDuration(&mediaDurationUs);
@@ -273,7 +273,7 @@
}
status_t NuPlayer::RTSPSource::getDuration(int64_t *durationUs) {
- *durationUs = -1ll;
+ *durationUs = -1LL;
int64_t audioDurationUs;
if (mAudioTrack != NULL
@@ -322,7 +322,7 @@
void NuPlayer::RTSPSource::schedulePollBuffering() {
sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
- msg->post(1000000ll); // 1 second intervals
+ msg->post(1000000LL); // 1 second intervals
}
void NuPlayer::RTSPSource::checkBuffering(
@@ -346,10 +346,10 @@
int64_t maxRebufferingMarkUs;
{
Mutex::Autolock _l(mBufferingSettingsLock);
- initialMarkUs = mBufferingSettings.mInitialMarkMs * 1000ll;
+ initialMarkUs = mBufferingSettings.mInitialMarkMs * 1000LL;
// TODO: maxRebufferingMarkUs could be larger than
// mBufferingSettings.mResumePlaybackMarkMs * 1000ll.
- maxRebufferingMarkUs = mBufferingSettings.mResumePlaybackMarkMs * 1000ll;
+ maxRebufferingMarkUs = mBufferingSettings.mResumePlaybackMarkMs * 1000LL;
}
// isFinished when duration is 0 checks for EOS result only
if (bufferedDurationUs > initialMarkUs
@@ -369,7 +369,7 @@
++overflowCount;
}
int64_t startServerMarkUs =
- (kUnderflowMarkMs * 1000ll + maxRebufferingMarkUs) / 2;
+ (kUnderflowMarkMs * 1000LL + maxRebufferingMarkUs) / 2;
if (bufferedDurationUs < startServerMarkUs) {
++startCount;
}
@@ -640,7 +640,7 @@
int64_t nptUs =
((double)rtpTime - (double)info->mRTPTime)
/ info->mTimeScale
- * 1000000ll
+ * 1000000LL
+ info->mNormalPlaytimeUs;
accessUnit->meta()->setInt64("timeUs", nptUs);
@@ -748,7 +748,7 @@
TrackInfo info;
info.mTimeScale = timeScale;
info.mRTPTime = 0;
- info.mNormalPlaytimeUs = 0ll;
+ info.mNormalPlaytimeUs = 0LL;
info.mNPTMappingValid = false;
if ((isAudio && mAudioTrack == NULL)
diff --git a/media/libmediaplayerservice/nuplayer/StreamingSource.cpp b/media/libmediaplayerservice/nuplayer/StreamingSource.cpp
index b3da53f..afdcd37 100644
--- a/media/libmediaplayerservice/nuplayer/StreamingSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/StreamingSource.cpp
@@ -186,7 +186,7 @@
// We're going to buffer at least 2 secs worth data on all tracks before
// starting playback (both at startup and after a seek).
- static const int64_t kMinDurationUs = 2000000ll;
+ static const int64_t kMinDurationUs = 2000000LL;
sp<AnotherPacketSource> audioTrack = getSource(true /*audio*/);
sp<AnotherPacketSource> videoTrack = getSource(false /*audio*/);
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index f4b5600..114f492 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2357,6 +2357,17 @@
return err;
}
+status_t ACodec::setAudioPresentation(int32_t presentationId, int32_t programId) {
+ OMX_AUDIO_CONFIG_ANDROID_AUDIOPRESENTATION config;
+ InitOMXParams(&config);
+ config.nPresentationId = (OMX_S32)presentationId;
+ config.nProgramId = (OMX_S32)programId;
+ status_t err = mOMXNode->setConfig(
+ (OMX_INDEXTYPE)OMX_IndexConfigAudioPresentation,
+ &config, sizeof(config));
+ return err;
+}
+
status_t ACodec::setPriority(int32_t priority) {
if (priority < 0) {
return BAD_VALUE;
@@ -7452,6 +7463,18 @@
}
}
+ int32_t presentationId = -1;
+ if (params->findInt32("audio-presentation-presentation-id", &presentationId)) {
+ int32_t programId = -1;
+ params->findInt32("audio-presentation-program-id", &programId);
+ status_t err = setAudioPresentation(presentationId, programId);
+ if (err != OK) {
+ ALOGI("[%s] failed setAudioPresentation. Failure is fine since this key is optional",
+ mComponentName.c_str());
+ err = OK;
+ }
+ }
+
// Ignore errors as failure is expected for codecs that aren't video encoders.
(void)configureTemporalLayers(params, false /* inConfigure */, mOutputFormat);
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index 02bb4e0..9aea88a 100644
--- a/media/libstagefright/Android.bp
+++ b/media/libstagefright/Android.bp
@@ -120,6 +120,7 @@
"MediaMuxer.cpp",
"NuCachedSource2.cpp",
"NuMediaExtractor.cpp",
+ "OggWriter.cpp",
"OMXClient.cpp",
"OmxInfoBuilder.cpp",
"RemoteMediaExtractor.cpp",
@@ -159,6 +160,7 @@
"libstagefright_codecbase",
"libstagefright_foundation",
"libstagefright_omx_utils",
+ "libstagefright_opus_common",
"libstagefright_xmlparser",
"libRScpp",
"libhidlallocatorutils",
@@ -179,6 +181,7 @@
"libstagefright_webm",
"libstagefright_timedtext",
"libvpx",
+ "libogg",
"libwebm",
"libstagefright_esds",
"libstagefright_id3",
diff --git a/media/libstagefright/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp
index 98f59b5..9ba2add 100644
--- a/media/libstagefright/MediaMuxer.cpp
+++ b/media/libstagefright/MediaMuxer.cpp
@@ -35,6 +35,7 @@
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MPEG4Writer.h>
+#include <media/stagefright/OggWriter.h>
#include <media/stagefright/Utils.h>
namespace android {
@@ -52,6 +53,8 @@
mWriter = new MPEG4Writer(fd);
} else if (format == OUTPUT_FORMAT_WEBM) {
mWriter = new WebmWriter(fd);
+ } else if (format == OUTPUT_FORMAT_OGG) {
+ mWriter = new OggWriter(fd);
}
if (mWriter != NULL) {
@@ -59,6 +62,8 @@
if (format == OUTPUT_FORMAT_HEIF) {
// Note that the key uses recorder file types.
mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_HEIF);
+ } else if (format == OUTPUT_FORMAT_OGG) {
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_OGG);
}
mState = INITIALIZED;
}
diff --git a/media/libstagefright/MetaDataUtils.cpp b/media/libstagefright/MetaDataUtils.cpp
index a3259fd..dbc287e 100644
--- a/media/libstagefright/MetaDataUtils.cpp
+++ b/media/libstagefright/MetaDataUtils.cpp
@@ -308,6 +308,8 @@
void parseVorbisComment(
AMediaFormat *fileMeta, const char *comment, size_t commentLength) {
+ // Haptic tag is only kept here as it will only be used in extractor to generate channel mask.
+ const char* const haptic = "haptic";
struct {
const char *const mTag;
const char *mKey;
@@ -328,6 +330,7 @@
{ "LYRICIST", AMEDIAFORMAT_KEY_LYRICIST },
{ "METADATA_BLOCK_PICTURE", AMEDIAFORMAT_KEY_ALBUMART },
{ "ANDROID_LOOP", AMEDIAFORMAT_KEY_LOOP },
+ { "ANDROID_HAPTIC", haptic },
};
for (size_t j = 0; j < sizeof(kMap) / sizeof(kMap[0]); ++j) {
@@ -343,6 +346,15 @@
if (!strcasecmp(&comment[tagLen + 1], "true")) {
AMediaFormat_setInt32(fileMeta, AMEDIAFORMAT_KEY_LOOP, 1);
}
+ } else if (kMap[j].mKey == haptic) {
+ char *end;
+ errno = 0;
+ const int hapticChannelCount = strtol(&comment[tagLen + 1], &end, 10);
+ if (errno == 0) {
+ AMediaFormat_setInt32(fileMeta, haptic, hapticChannelCount);
+ } else {
+ ALOGE("Error(%d) when parsing haptic channel count", errno);
+ }
} else {
AMediaFormat_setString(fileMeta, kMap[j].mKey, &comment[tagLen + 1]);
}
diff --git a/media/libstagefright/OggWriter.cpp b/media/libstagefright/OggWriter.cpp
new file mode 100644
index 0000000..ad55c56
--- /dev/null
+++ b/media/libstagefright/OggWriter.cpp
@@ -0,0 +1,397 @@
+/*
+ * 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.
+ */
+
+#define LOG_TAG "OggWriter"
+
+#include <fcntl.h>
+#include <inttypes.h>
+#include <sys/prctl.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+
+#include <media/MediaSource.h>
+#include <media/mediarecorder.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OggWriter.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include "OpusHeader.h"
+
+extern "C" {
+#include <ogg/ogg.h>
+}
+
+// store the int32 value in little-endian order.
+static inline void writeint(char *buf, int base, int32_t val) {
+ buf[base + 3] = ((val) >> 24) & 0xff;
+ buf[base + 2] = ((val) >> 16) & 0xff;
+ buf[base + 1] = ((val) >> 8) & 0xff;
+ buf[base] = (val)&0xff;
+}
+
+// linkage between our header OggStreamState and the underlying ogg_stream_state
+// so that consumers of our interface do not require the ogg headers themselves.
+struct OggStreamState : public ogg_stream_state {};
+
+namespace android {
+
+OggWriter::OggWriter(int fd)
+ : mFd(dup(fd)),
+ mInitCheck(mFd < 0 ? NO_INIT : OK) {
+ // empty
+}
+
+OggWriter::~OggWriter() {
+ if (mStarted) {
+ reset();
+ }
+
+ if (mFd != -1) {
+ close(mFd);
+ mFd = -1;
+ }
+
+ free(mOs);
+}
+
+status_t OggWriter::initCheck() const {
+ return mInitCheck;
+}
+
+status_t OggWriter::addSource(const sp<MediaSource>& source) {
+ if (mInitCheck != OK) {
+ return mInitCheck;
+ }
+
+ if (mSource != NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ // Support is limited to single track of Opus audio.
+ const char* mime;
+ source->getFormat()->findCString(kKeyMIMEType, &mime);
+ const char* opus = MEDIA_MIMETYPE_AUDIO_OPUS;
+ if (strncasecmp(mime, opus, strlen(opus))) {
+ ALOGE("Track (%s) other than %s is not supported", mime, opus);
+ return ERROR_UNSUPPORTED;
+ }
+
+ mOs = (OggStreamState*) malloc(sizeof(ogg_stream_state));
+ if (ogg_stream_init((ogg_stream_state*)mOs, rand()) == -1) {
+ ALOGE("ogg stream init failed");
+ return UNKNOWN_ERROR;
+ }
+
+ // Write Ogg headers.
+ int32_t nChannels = 0;
+ if (!source->getFormat()->findInt32(kKeyChannelCount, &nChannels)) {
+ ALOGE("Missing format keys for audio track");
+ source->getFormat()->dumpToLog();
+ return BAD_VALUE;
+ }
+ source->getFormat()->dumpToLog();
+
+ int32_t sampleRate = 0;
+ if (!source->getFormat()->findInt32(kKeySampleRate, &sampleRate)) {
+ ALOGE("Missing format key for sample rate");
+ source->getFormat()->dumpToLog();
+ return UNKNOWN_ERROR;
+ }
+
+ mSampleRate = sampleRate;
+
+ OpusHeader header;
+ header.channels = nChannels;
+ header.num_streams = nChannels;
+ header.num_coupled = 0;
+ header.channel_mapping = ((nChannels > 8) ? 255 : (nChannels > 2));
+ header.gain_db = 0;
+ header.skip_samples = 0;
+
+ // headers are 21-bytes + something driven by channel count
+ // expect numbers in the low 30's here. WriteOpusHeader() will tell us
+ // if things are bad.
+ unsigned char header_data[100];
+ ogg_packet op;
+ ogg_page og;
+
+ const int packet_size = WriteOpusHeader(header, mSampleRate, (uint8_t*)header_data,
+ sizeof(header_data));
+
+ if (packet_size < 0) {
+ ALOGE("opus header writing failed");
+ return UNKNOWN_ERROR;
+ }
+ op.packet = header_data;
+ op.bytes = packet_size;
+ op.b_o_s = 1;
+ op.e_o_s = 0;
+ op.granulepos = 0;
+ op.packetno = 0;
+ ogg_stream_packetin((ogg_stream_state*)mOs, &op);
+
+ int ret;
+ while ((ret = ogg_stream_flush((ogg_stream_state*)mOs, &og))) {
+ if (!ret) break;
+ write(mFd, og.header, og.header_len);
+ write(mFd, og.body, og.body_len);
+ }
+
+
+ const char* vendor_string = "libopus";
+ const int vendor_length = strlen(vendor_string);
+ int user_comment_list_length = 0;
+
+ const int comments_length = 8 + 4 + vendor_length + 4 + user_comment_list_length;
+ char* comments = (char*)malloc(comments_length);
+ if (comments == NULL) {
+ ALOGE("failed to allocate ogg comment buffer");
+ return UNKNOWN_ERROR;
+ }
+ memcpy(comments, "OpusTags", 8);
+ writeint(comments, 8, vendor_length);
+ memcpy(comments + 12, vendor_string, vendor_length);
+ writeint(comments, 12 + vendor_length, user_comment_list_length);
+
+ op.packet = (unsigned char*)comments;
+ op.bytes = comments_length;
+ op.b_o_s = 0;
+ op.e_o_s = 0;
+ op.granulepos = 0;
+ op.packetno = 1;
+ ogg_stream_packetin((ogg_stream_state*)mOs, &op);
+
+ while ((ret = ogg_stream_flush((ogg_stream_state*)mOs, &og))) {
+ if (!ret) break;
+ write(mFd, og.header, og.header_len);
+ write(mFd, og.body, og.body_len);
+ }
+
+ mSource = source;
+ free(comments);
+ return OK;
+}
+
+status_t OggWriter::start(MetaData* /* params */) {
+ if (mInitCheck != OK) {
+ return mInitCheck;
+ }
+
+ if (mSource == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ if (mStarted && mPaused) {
+ mPaused = false;
+ mResumed = true;
+ return OK;
+ } else if (mStarted) {
+ // Already started, does nothing
+ return OK;
+ }
+
+ status_t err = mSource->start();
+
+ if (err != OK) {
+ return err;
+ }
+
+ pthread_attr_t attr;
+ pthread_attr_init(&attr);
+ pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
+
+ mReachedEOS = false;
+ mDone = false;
+
+ pthread_create(&mThread, &attr, ThreadWrapper, this);
+ pthread_attr_destroy(&attr);
+
+ mStarted = true;
+
+ return OK;
+}
+
+status_t OggWriter::pause() {
+ if (!mStarted) {
+ return OK;
+ }
+ mPaused = true;
+ return OK;
+}
+
+status_t OggWriter::reset() {
+ if (!mStarted) {
+ return OK;
+ }
+
+ mDone = true;
+
+ void* dummy;
+ pthread_join(mThread, &dummy);
+
+ status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
+ {
+ status_t status = mSource->stop();
+ if (err == OK && (status != OK && status != ERROR_END_OF_STREAM)) {
+ err = status;
+ }
+ }
+
+ mStarted = false;
+ return err;
+}
+
+bool OggWriter::exceedsFileSizeLimit() {
+ if (mMaxFileSizeLimitBytes == 0) {
+ return false;
+ }
+ return mEstimatedSizeBytes > mMaxFileSizeLimitBytes;
+}
+
+bool OggWriter::exceedsFileDurationLimit() {
+ if (mMaxFileDurationLimitUs == 0) {
+ return false;
+ }
+ return mEstimatedDurationUs > mMaxFileDurationLimitUs;
+}
+
+// static
+void* OggWriter::ThreadWrapper(void* me) {
+ return (void*)(uintptr_t) static_cast<OggWriter*>(me)->threadFunc();
+}
+
+status_t OggWriter::threadFunc() {
+ mEstimatedDurationUs = 0;
+ mEstimatedSizeBytes = 0;
+ bool stoppedPrematurely = true;
+ int64_t previousPausedDurationUs = 0;
+ int64_t maxTimestampUs = 0;
+ status_t err = OK;
+
+ prctl(PR_SET_NAME, (unsigned long)"OggWriter", 0, 0, 0);
+
+ while (!mDone) {
+ MediaBufferBase* buffer = nullptr;
+ err = mSource->read(&buffer);
+
+ if (err != OK) {
+ ALOGW("failed to read next buffer");
+ break;
+ }
+
+ if (mPaused) {
+ buffer->release();
+ buffer = nullptr;
+ continue;
+ }
+ mEstimatedSizeBytes += buffer->range_length();
+ if (exceedsFileSizeLimit()) {
+ buffer->release();
+ buffer = nullptr;
+ notify(MEDIA_RECORDER_EVENT_INFO, MEDIA_RECORDER_INFO_MAX_FILESIZE_REACHED, 0);
+ ALOGW("estimated size(%" PRId64 ") exceeds limit (%" PRId64 ")",
+ mEstimatedSizeBytes, mMaxFileSizeLimitBytes);
+ break;
+ }
+ int64_t timestampUs;
+ CHECK(buffer->meta_data().findInt64(kKeyTime, ×tampUs));
+ if (timestampUs > mEstimatedDurationUs) {
+ mEstimatedDurationUs = timestampUs;
+ }
+ if (mResumed) {
+ previousPausedDurationUs += (timestampUs - maxTimestampUs - 20000);
+ mResumed = false;
+ }
+
+ timestampUs -= previousPausedDurationUs;
+
+ ALOGV("time stamp: %" PRId64 ", previous paused duration: %" PRId64, timestampUs,
+ previousPausedDurationUs);
+ if (timestampUs > maxTimestampUs) {
+ maxTimestampUs = timestampUs;
+ }
+
+ if (exceedsFileDurationLimit()) {
+ buffer->release();
+ buffer = nullptr;
+ notify(MEDIA_RECORDER_EVENT_INFO, MEDIA_RECORDER_INFO_MAX_DURATION_REACHED, 0);
+ ALOGW("estimated duration(%" PRId64 " us) exceeds limit(%" PRId64 " us)",
+ mEstimatedDurationUs, mMaxFileDurationLimitUs);
+ break;
+ }
+
+ ogg_packet op;
+ ogg_page og;
+ op.packet = (uint8_t*)buffer->data() + buffer->range_offset();
+ op.bytes = (long)buffer->range_length();
+ op.b_o_s = 0;
+ op.e_o_s = mReachedEOS ? 1 : 0;
+ // granulepos is the total number of PCM audio samples @ 48 kHz, up to and
+ // including the current packet.
+ ogg_int64_t granulepos = (48000 * mEstimatedDurationUs) / 1000000;
+ op.granulepos = granulepos;
+
+ // Headers are at packets 0 and 1.
+ op.packetno = 2 + (ogg_int32_t)mCurrentPacketId++;
+ ogg_stream_packetin((ogg_stream_state*)mOs, &op);
+ size_t n = 0;
+
+ while (ogg_stream_flush((ogg_stream_state*)mOs, &og) > 0) {
+ write(mFd, og.header, og.header_len);
+ write(mFd, og.body, og.body_len);
+ n = n + og.header_len + og.body_len;
+ }
+
+ if (n < buffer->range_length()) {
+ buffer->release();
+ buffer = nullptr;
+ err = ERROR_IO;
+ break;
+ }
+
+ if (err != OK) {
+ break;
+ }
+
+ stoppedPrematurely = false;
+
+ buffer->release();
+ buffer = nullptr;
+ }
+
+ // end of stream is an ok thing
+ if (err == ERROR_END_OF_STREAM) {
+ err = OK;
+ }
+
+ if (err == OK && stoppedPrematurely) {
+ err = ERROR_MALFORMED;
+ }
+
+ close(mFd);
+ mFd = -1;
+ mReachedEOS = true;
+
+ return err;
+}
+
+bool OggWriter::reachedEOS() {
+ return mReachedEOS;
+}
+
+} // namespace android
diff --git a/media/libstagefright/StagefrightPluginLoader.cpp b/media/libstagefright/StagefrightPluginLoader.cpp
index 26d7dff..b90649c 100644
--- a/media/libstagefright/StagefrightPluginLoader.cpp
+++ b/media/libstagefright/StagefrightPluginLoader.cpp
@@ -34,9 +34,7 @@
} // unnamed namespace
-StagefrightPluginLoader::StagefrightPluginLoader(const char *libPath)
- : mCreateCodec(nullptr),
- mCreateBuilder(nullptr) {
+StagefrightPluginLoader::StagefrightPluginLoader(const char *libPath) {
if (android::base::GetIntProperty("debug.media.codec2", 0) == 0) {
ALOGD("CCodec is disabled.");
return;
diff --git a/media/libstagefright/StagefrightPluginLoader.h b/media/libstagefright/StagefrightPluginLoader.h
index 999d30c..78effbf 100644
--- a/media/libstagefright/StagefrightPluginLoader.h
+++ b/media/libstagefright/StagefrightPluginLoader.h
@@ -40,10 +40,10 @@
static Mutex sMutex;
static std::unique_ptr<StagefrightPluginLoader> sInstance;
- void *mLibHandle;
- CodecBase::CreateCodecFunc mCreateCodec;
- MediaCodecListBuilderBase::CreateBuilderFunc mCreateBuilder;
- CodecBase::CreateInputSurfaceFunc mCreateInputSurface;
+ void *mLibHandle{nullptr};
+ CodecBase::CreateCodecFunc mCreateCodec{nullptr};
+ MediaCodecListBuilderBase::CreateBuilderFunc mCreateBuilder{nullptr};
+ CodecBase::CreateInputSurfaceFunc mCreateInputSurface{nullptr};
};
} // namespace android
diff --git a/media/libstagefright/foundation/AString.cpp b/media/libstagefright/foundation/AString.cpp
index a8adff5..fb51cc5 100644
--- a/media/libstagefright/foundation/AString.cpp
+++ b/media/libstagefright/foundation/AString.cpp
@@ -365,6 +365,8 @@
// static
AString AString::FromParcel(const Parcel &parcel) {
size_t size = static_cast<size_t>(parcel.readInt32());
+ // The static analyzer incorrectly reports a false-positive here in c++17.
+ // https://bugs.llvm.org/show_bug.cgi?id=38176 . NOLINTNEXTLINE
return AString(static_cast<const char *>(parcel.readInplace(size)), size);
}
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index 5dd1966..80125d4 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -479,6 +479,7 @@
status_t setPriority(int32_t priority);
status_t setLatency(uint32_t latency);
status_t getLatency(uint32_t *latency);
+ status_t setAudioPresentation(int32_t presentationId, int32_t programId);
status_t setOperatingRate(float rateFloat, bool isVideo);
status_t getIntraRefreshPeriod(uint32_t *intraRefreshPeriod);
status_t setIntraRefreshPeriod(uint32_t intraRefreshPeriod, bool inConfigure);
diff --git a/media/libstagefright/include/media/stagefright/MediaMuxer.h b/media/libstagefright/include/media/stagefright/MediaMuxer.h
index 66f4d72..69d6cde 100644
--- a/media/libstagefright/include/media/stagefright/MediaMuxer.h
+++ b/media/libstagefright/include/media/stagefright/MediaMuxer.h
@@ -49,6 +49,7 @@
OUTPUT_FORMAT_WEBM = 1,
OUTPUT_FORMAT_THREE_GPP = 2,
OUTPUT_FORMAT_HEIF = 3,
+ OUTPUT_FORMAT_OGG = 4,
OUTPUT_FORMAT_LIST_END // must be last - used to validate format type
};
diff --git a/media/libstagefright/include/media/stagefright/OggWriter.h b/media/libstagefright/include/media/stagefright/OggWriter.h
new file mode 100644
index 0000000..e3837cd
--- /dev/null
+++ b/media/libstagefright/include/media/stagefright/OggWriter.h
@@ -0,0 +1,73 @@
+/*
+ * 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.
+ */
+
+#ifndef OGG_WRITER_H_
+
+#define OGG_WRITER_H_
+
+#include <stdio.h>
+
+#include <media/stagefright/MediaWriter.h>
+#include <utils/threads.h>
+
+struct OggStreamState;
+
+namespace android {
+
+struct OggWriter : public MediaWriter {
+ OggWriter(int fd);
+
+ status_t initCheck() const;
+
+ virtual status_t addSource(const sp<MediaSource>& source);
+ virtual bool reachedEOS();
+ virtual status_t start(MetaData* params = NULL);
+ virtual status_t stop() { return reset(); }
+ virtual status_t pause();
+
+protected:
+ ~OggWriter();
+
+private:
+ int mFd;
+ status_t mInitCheck;
+ sp<MediaSource> mSource;
+ bool mStarted = false;
+ volatile bool mPaused = false;
+ volatile bool mResumed = false;
+ volatile bool mDone;
+ volatile bool mReachedEOS;
+ pthread_t mThread;
+ int64_t mSampleRate;
+ int64_t mEstimatedSizeBytes;
+ int64_t mEstimatedDurationUs;
+
+ static void* ThreadWrapper(void*);
+ status_t threadFunc();
+ bool exceedsFileSizeLimit();
+ bool exceedsFileDurationLimit();
+ status_t reset();
+
+ int32_t mCurrentPacketId;
+ OggStreamState* mOs = nullptr;
+
+ OggWriter(const OggWriter&);
+ OggWriter& operator=(const OggWriter&);
+};
+
+} // namespace android
+
+#endif // OGG_WRITER_H_
diff --git a/media/libstagefright/opus/Android.bp b/media/libstagefright/opus/Android.bp
new file mode 100644
index 0000000..c5086ec
--- /dev/null
+++ b/media/libstagefright/opus/Android.bp
@@ -0,0 +1,21 @@
+cc_library_shared {
+ name: "libstagefright_opus_common",
+ vendor_available: true,
+
+ export_include_dirs: ["include"],
+
+ srcs: ["OpusHeader.cpp"],
+
+ shared_libs: ["liblog"],
+
+ cflags: ["-Werror"],
+
+ sanitize: {
+ integer_overflow: true,
+ cfi: true,
+ diag: {
+ integer_overflow: true,
+ cfi: true,
+ },
+ },
+}
\ No newline at end of file
diff --git a/media/libstagefright/opus/OpusHeader.cpp b/media/libstagefright/opus/OpusHeader.cpp
new file mode 100644
index 0000000..e4a460c
--- /dev/null
+++ b/media/libstagefright/opus/OpusHeader.cpp
@@ -0,0 +1,185 @@
+/*
+ * 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftOpus"
+
+#include <cstring>
+#include <stdint.h>
+
+#include <log/log.h>
+
+#include "OpusHeader.h"
+
+namespace android {
+
+// Opus uses Vorbis channel mapping, and Vorbis channel mapping specifies
+// mappings for up to 8 channels. This information is part of the Vorbis I
+// Specification:
+// http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html
+constexpr int kMaxChannels = 8;
+
+constexpr uint8_t kOpusChannelMap[kMaxChannels][kMaxChannels] = {
+ {0},
+ {0, 1},
+ {0, 2, 1},
+ {0, 1, 2, 3},
+ {0, 4, 1, 2, 3},
+ {0, 4, 1, 2, 3, 5},
+ {0, 4, 1, 2, 3, 5, 6},
+ {0, 6, 1, 2, 3, 4, 5, 7},
+};
+
+// Opus always has a 48kHz output rate. This is true for all Opus, not just this
+// implementation.
+constexpr int kRate = 48000;
+// Size of the Opus header excluding optional mapping information.
+constexpr size_t kOpusHeaderSize = 19;
+// Offset to magic string that starts Opus header.
+constexpr size_t kOpusHeaderLabelOffset = 0;
+// Offset to Opus version in the Opus header.
+constexpr size_t kOpusHeaderVersionOffset = 8;
+// Offset to the channel count byte in the Opus header.
+constexpr size_t kOpusHeaderChannelsOffset = 9;
+// Offset to the pre-skip value in the Opus header.
+constexpr size_t kOpusHeaderSkipSamplesOffset = 10;
+// Offset to sample rate in the Opus header.
+constexpr size_t kOpusHeaderSampleRateOffset = 12;
+// Offset to the gain value in the Opus header.
+constexpr size_t kOpusHeaderGainOffset = 16;
+// Offset to the channel mapping byte in the Opus header.
+constexpr size_t kOpusHeaderChannelMappingOffset = 18;
+// Opus Header contains a stream map. The mapping values are in the header
+// beyond the always present |kOpusHeaderSize| bytes of data. The mapping
+// data contains stream count, coupling information, and per channel mapping
+// values:
+// - Byte 0: Number of streams.
+// - Byte 1: Number coupled.
+// - Byte 2: Starting at byte 2 are |header->channels| uint8 mapping
+// values.
+// Offset to the number of streams in the Opus header.
+constexpr size_t kOpusHeaderNumStreamsOffset = 19;
+// Offset to the number of streams that are coupled in the Opus header.
+constexpr size_t kOpusHeaderNumCoupledStreamsOffset = 20;
+// Offset to the stream to channel mapping in the Opus header.
+constexpr size_t kOpusHeaderStreamMapOffset = 21;
+// Maximum packet size used in Xiph's opusdec.
+constexpr int kMaxOpusOutputPacketSizeSamples = 960 * 6;
+
+// Default audio output channel layout. Used to initialize |stream_map| in
+// OpusHeader, and passed to opus_multistream_decoder_create() when the header
+// does not contain mapping information. The values are valid only for mono and
+// stereo output: Opus streams with more than 2 channels require a stream map.
+constexpr int kMaxChannelsWithDefaultLayout = 2;
+constexpr uint8_t kDefaultOpusChannelLayout[kMaxChannelsWithDefaultLayout] = {0, 1};
+
+static uint16_t ReadLE16(const uint8_t* data, size_t data_size, uint32_t read_offset) {
+ // check whether the 2nd byte is within the buffer
+ if (read_offset + 1 >= data_size) return 0;
+ uint16_t val;
+ val = data[read_offset];
+ val |= data[read_offset + 1] << 8;
+ return val;
+}
+
+// Parses Opus Header. Header spec: http://wiki.xiph.org/OggOpus#ID_Header
+bool ParseOpusHeader(const uint8_t* data, size_t data_size, OpusHeader* header) {
+ if (data_size < kOpusHeaderSize) {
+ ALOGV("Header size is too small.");
+ return false;
+ }
+ header->channels = data[kOpusHeaderChannelsOffset];
+
+ if (header->channels < 1 || header->channels > kMaxChannels) {
+ ALOGV("Invalid Header, bad channel count: %d", header->channels);
+ return false;
+ }
+ header->skip_samples = ReadLE16(data, data_size, kOpusHeaderSkipSamplesOffset);
+ header->gain_db = static_cast<int16_t>(ReadLE16(data, data_size, kOpusHeaderGainOffset));
+ header->channel_mapping = data[kOpusHeaderChannelMappingOffset];
+ if (!header->channel_mapping) {
+ if (header->channels > kMaxChannelsWithDefaultLayout) {
+ ALOGV("Invalid Header, missing stream map.");
+ return false;
+ }
+ header->num_streams = 1;
+ header->num_coupled = header->channels > 1;
+ header->stream_map[0] = 0;
+ header->stream_map[1] = 1;
+ return true;
+ }
+ if (data_size < kOpusHeaderStreamMapOffset + header->channels) {
+ ALOGV("Invalid stream map; insufficient data for current channel "
+ "count: %d",
+ header->channels);
+ return false;
+ }
+ header->num_streams = data[kOpusHeaderNumStreamsOffset];
+ header->num_coupled = data[kOpusHeaderNumCoupledStreamsOffset];
+ if (header->num_streams + header->num_coupled != header->channels) {
+ ALOGV("Inconsistent channel mapping.");
+ return false;
+ }
+ for (int i = 0; i < header->channels; ++i)
+ header->stream_map[i] = data[kOpusHeaderStreamMapOffset + i];
+ return true;
+}
+
+int WriteOpusHeader(const OpusHeader &header, int input_sample_rate,
+ uint8_t* output, size_t output_size) {
+ // See https://wiki.xiph.org/OggOpus#ID_Header.
+ const size_t total_size = kOpusHeaderStreamMapOffset + header.channels;
+ if (output_size < total_size) {
+ ALOGE("Output buffer too small for header.");
+ return -1;
+ }
+
+ // ensure entire header is cleared, even though we overwrite much of it below
+ memset(output, 0, output_size);
+
+ // Set magic signature.
+ memcpy(output + kOpusHeaderLabelOffset, "OpusHead", 8);
+ // Set Opus version.
+ output[kOpusHeaderVersionOffset] = 1;
+ // Set channel count.
+ output[kOpusHeaderChannelsOffset] = (uint8_t)header.channels;
+ // Set pre-skip
+ memcpy(output + kOpusHeaderSkipSamplesOffset, &header.skip_samples, sizeof(uint16_t));
+ // Set original input sample rate in Hz.
+ memcpy(output + kOpusHeaderSampleRateOffset, &input_sample_rate, sizeof(uint32_t));
+ // Set output gain in dB.
+ memcpy(output + kOpusHeaderGainOffset, &header.gain_db, sizeof(uint16_t));
+
+ if (header.channels > 2) {
+ // Set channel mapping
+ output[kOpusHeaderChannelMappingOffset] = 1;
+ // Assuming no coupled streams. This should actually be
+ // channels() - |coupled_streams|.
+ output[kOpusHeaderNumStreamsOffset] = header.channels;
+ output[kOpusHeaderNumCoupledStreamsOffset] = 0;
+
+ // Set the actual stream map.
+ for (int i = 0; i < header.channels; ++i) {
+ output[kOpusHeaderStreamMapOffset + i] = kOpusChannelMap[header.channels - 1][i];
+ }
+ return kOpusHeaderStreamMapOffset + header.channels + 1;
+ } else {
+ output[kOpusHeaderChannelMappingOffset] = 0;
+ return kOpusHeaderChannelMappingOffset + 1;
+ }
+}
+
+} // namespace android
diff --git a/media/libstagefright/opus/include/OpusHeader.h b/media/libstagefright/opus/include/OpusHeader.h
new file mode 100644
index 0000000..f9f79cd
--- /dev/null
+++ b/media/libstagefright/opus/include/OpusHeader.h
@@ -0,0 +1,41 @@
+/*
+ * 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.
+ */
+
+/*
+ * The Opus specification is part of IETF RFC 6716:
+ * http://tools.ietf.org/html/rfc6716
+ */
+
+#ifndef OPUS_HEADER_H_
+#define OPUS_HEADER_H_
+
+namespace android {
+
+struct OpusHeader {
+ int channels;
+ int channel_mapping;
+ int num_streams;
+ int num_coupled;
+ int16_t gain_db;
+ int skip_samples;
+ uint8_t stream_map[8];
+};
+
+bool ParseOpusHeader(const uint8_t* data, size_t data_size, OpusHeader* header);
+int WriteOpusHeader(const OpusHeader &header, int input_sample_rate, uint8_t* output, size_t output_size);
+} // namespace android
+
+#endif // OPUS_HEADER_H_
diff --git a/media/libstagefright/webm/Android.bp b/media/libstagefright/webm/Android.bp
index 64ecc2d..1f840b7 100644
--- a/media/libstagefright/webm/Android.bp
+++ b/media/libstagefright/webm/Android.bp
@@ -28,6 +28,7 @@
shared_libs: [
"libstagefright_foundation",
+ "libstagefright_opus_common",
"libutils",
"liblog",
],
diff --git a/media/libstagefright/webm/WebmElement.cpp b/media/libstagefright/webm/WebmElement.cpp
index a5120b9..4d504e0 100644
--- a/media/libstagefright/webm/WebmElement.cpp
+++ b/media/libstagefright/webm/WebmElement.cpp
@@ -305,6 +305,7 @@
}
sp<WebmElement> WebmElement::AudioTrackEntry(
+ const char *codec,
int chans,
double rate,
const sp<ABuffer> &buf,
@@ -322,7 +323,7 @@
uid,
lacing,
lang,
- "A_VORBIS",
+ codec,
kAudioType,
trackEntryFields);
diff --git a/media/libstagefright/webm/WebmElement.h b/media/libstagefright/webm/WebmElement.h
index ffbba1b..a94c23f 100644
--- a/media/libstagefright/webm/WebmElement.h
+++ b/media/libstagefright/webm/WebmElement.h
@@ -50,6 +50,7 @@
static sp<WebmElement> SegmentInfo(uint64_t scale = 1000000, double dur = 0);
static sp<WebmElement> AudioTrackEntry(
+ const char *codec,
int chans,
double rate,
const sp<ABuffer> &buf,
diff --git a/media/libstagefright/webm/WebmWriter.cpp b/media/libstagefright/webm/WebmWriter.cpp
index 4d73eb8..7b4b23a 100644
--- a/media/libstagefright/webm/WebmWriter.cpp
+++ b/media/libstagefright/webm/WebmWriter.cpp
@@ -23,6 +23,8 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <OpusHeader.h>
#include <utils/Errors.h>
@@ -112,46 +114,102 @@
// static
sp<WebmElement> WebmWriter::audioTrack(const sp<MetaData>& md) {
int32_t nChannels, samplerate;
- uint32_t type;
- const void *headerData1;
- const char headerData2[] = { 3, 'v', 'o', 'r', 'b', 'i', 's', 7, 0, 0, 0,
- 'a', 'n', 'd', 'r', 'o', 'i', 'd', 0, 0, 0, 0, 1 };
- const void *headerData3;
- size_t headerSize1, headerSize2 = sizeof(headerData2), headerSize3;
+ const char* mimeType;
if (!md->findInt32(kKeyChannelCount, &nChannels)
- || !md->findInt32(kKeySampleRate, &samplerate)
- || !md->findData(kKeyVorbisInfo, &type, &headerData1, &headerSize1)
- || !md->findData(kKeyVorbisBooks, &type, &headerData3, &headerSize3)) {
+ || !md->findInt32(kKeySampleRate, &samplerate)
+ || !md->findCString(kKeyMIMEType, &mimeType)) {
ALOGE("Missing format keys for audio track");
md->dumpToLog();
return NULL;
}
- size_t codecPrivateSize = 1;
- codecPrivateSize += XiphLaceCodeLen(headerSize1);
- codecPrivateSize += XiphLaceCodeLen(headerSize2);
- codecPrivateSize += headerSize1 + headerSize2 + headerSize3;
+ if (!strncasecmp(mimeType, MEDIA_MIMETYPE_AUDIO_OPUS, strlen(MEDIA_MIMETYPE_AUDIO_OPUS))) {
+ // Opus in WebM is a well-known, yet under-documented, format. The codec private data
+ // of the track is an Opus Ogg header (https://tools.ietf.org/html/rfc7845#section-5.1)
+ // The name of the track isn't standardized, its value should be "A_OPUS".
+ OpusHeader header;
+ header.channels = nChannels;
+ header.num_streams = nChannels;
+ header.num_coupled = 0;
+ // - Channel mapping family (8 bits unsigned)
+ // -- 0 = one stream: mono or L,R stereo
+ // -- 1 = channels in vorbis spec order: mono or L,R stereo or ... or FL,C,FR,RL,RR,LFE, ...
+ // -- 2..254 = reserved (treat as 255)
+ // -- 255 = no defined channel meaning
+ //
+ // our implementation encodes: 0, 1, or 255
+ header.channel_mapping = ((nChannels > 8) ? 255 : (nChannels > 2));
+ header.gain_db = 0;
+ header.skip_samples = 0;
- off_t off = 0;
- sp<ABuffer> codecPrivateBuf = new ABuffer(codecPrivateSize);
- uint8_t *codecPrivateData = codecPrivateBuf->data();
- codecPrivateData[off++] = 2;
+ // headers are 21-bytes + something driven by channel count
+ // expect numbers in the low 30's here. WriteOpusHeader() will tell us
+ // if things are bad.
+ unsigned char header_data[100];
+ int headerSize = WriteOpusHeader(header, samplerate, (uint8_t*)header_data,
+ sizeof(header_data));
- off += XiphLaceEnc(codecPrivateData + off, headerSize1);
- off += XiphLaceEnc(codecPrivateData + off, headerSize2);
+ if (headerSize < 0) {
+ // didn't fill out that header for some reason
+ ALOGE("failed to generate OPUS header");
+ return NULL;
+ }
- memcpy(codecPrivateData + off, headerData1, headerSize1);
- off += headerSize1;
- memcpy(codecPrivateData + off, headerData2, headerSize2);
- off += headerSize2;
- memcpy(codecPrivateData + off, headerData3, headerSize3);
+ size_t codecPrivateSize = 0;
+ codecPrivateSize += headerSize;
- sp<WebmElement> entry = WebmElement::AudioTrackEntry(
- nChannels,
- samplerate,
- codecPrivateBuf);
- return entry;
+ off_t off = 0;
+ sp<ABuffer> codecPrivateBuf = new ABuffer(codecPrivateSize);
+ uint8_t* codecPrivateData = codecPrivateBuf->data();
+
+ memcpy(codecPrivateData + off, (uint8_t*)header_data, headerSize);
+ sp<WebmElement> entry =
+ WebmElement::AudioTrackEntry("A_OPUS", nChannels, samplerate, codecPrivateBuf);
+ return entry;
+ } else if (!strncasecmp(mimeType,
+ MEDIA_MIMETYPE_AUDIO_VORBIS,
+ strlen(MEDIA_MIMETYPE_AUDIO_VORBIS))) {
+ uint32_t type;
+ const void *headerData1;
+ const char headerData2[] = { 3, 'v', 'o', 'r', 'b', 'i', 's', 7, 0, 0, 0,
+ 'a', 'n', 'd', 'r', 'o', 'i', 'd', 0, 0, 0, 0, 1 };
+ const void *headerData3;
+ size_t headerSize1, headerSize2 = sizeof(headerData2), headerSize3;
+
+ if (!md->findData(kKeyVorbisInfo, &type, &headerData1, &headerSize1)
+ || !md->findData(kKeyVorbisBooks, &type, &headerData3, &headerSize3)) {
+ ALOGE("Missing header format keys for vorbis track");
+ md->dumpToLog();
+ return NULL;
+ }
+
+ size_t codecPrivateSize = 1;
+ codecPrivateSize += XiphLaceCodeLen(headerSize1);
+ codecPrivateSize += XiphLaceCodeLen(headerSize2);
+ codecPrivateSize += headerSize1 + headerSize2 + headerSize3;
+
+ off_t off = 0;
+ sp<ABuffer> codecPrivateBuf = new ABuffer(codecPrivateSize);
+ uint8_t *codecPrivateData = codecPrivateBuf->data();
+ codecPrivateData[off++] = 2;
+
+ off += XiphLaceEnc(codecPrivateData + off, headerSize1);
+ off += XiphLaceEnc(codecPrivateData + off, headerSize2);
+
+ memcpy(codecPrivateData + off, headerData1, headerSize1);
+ off += headerSize1;
+ memcpy(codecPrivateData + off, headerData2, headerSize2);
+ off += headerSize2;
+ memcpy(codecPrivateData + off, headerData3, headerSize3);
+
+ sp<WebmElement> entry =
+ WebmElement::AudioTrackEntry("A_VORBIS", nChannels, samplerate, codecPrivateBuf);
+ return entry;
+ } else {
+ ALOGE("Track (%s) is not a supported audio format", mimeType);
+ return NULL;
+ }
}
size_t WebmWriter::numTracks() {
@@ -382,16 +440,18 @@
const char *vp8 = MEDIA_MIMETYPE_VIDEO_VP8;
const char *vp9 = MEDIA_MIMETYPE_VIDEO_VP9;
const char *vorbis = MEDIA_MIMETYPE_AUDIO_VORBIS;
+ const char* opus = MEDIA_MIMETYPE_AUDIO_OPUS;
size_t streamIndex;
if (!strncasecmp(mime, vp8, strlen(vp8)) ||
!strncasecmp(mime, vp9, strlen(vp9))) {
streamIndex = kVideoIndex;
- } else if (!strncasecmp(mime, vorbis, strlen(vorbis))) {
+ } else if (!strncasecmp(mime, vorbis, strlen(vorbis)) ||
+ !strncasecmp(mime, opus, strlen(opus))) {
streamIndex = kAudioIndex;
} else {
- ALOGE("Track (%s) other than %s, %s or %s is not supported",
- mime, vp8, vp9, vorbis);
+ ALOGE("Track (%s) other than %s, %s, %s, or %s is not supported",
+ mime, vp8, vp9, vorbis, opus);
return ERROR_UNSUPPORTED;
}
diff --git a/media/ndk/NdkMediaFormat.cpp b/media/ndk/NdkMediaFormat.cpp
index 200439c..e0af80d 100644
--- a/media/ndk/NdkMediaFormat.cpp
+++ b/media/ndk/NdkMediaFormat.cpp
@@ -273,6 +273,9 @@
EXPORT const char* AMEDIAFORMAT_KEY_ALBUMARTIST = "albumartist";
EXPORT const char* AMEDIAFORMAT_KEY_ARTIST = "artist";
EXPORT const char* AMEDIAFORMAT_KEY_AUDIO_PRESENTATION_INFO = "audio-presentation-info";
+EXPORT const char* AMEDIAFORMAT_KEY_AUDIO_PRESENTATION_PRESENTATION_ID =
+ "audio-presentation-presentation-id";
+EXPORT const char* AMEDIAFORMAT_KEY_AUDIO_PRESENTATION_PROGRAM_ID = "audio-presentation-program-id";
EXPORT const char* AMEDIAFORMAT_KEY_AUDIO_SESSION_ID = "audio-session-id";
EXPORT const char* AMEDIAFORMAT_KEY_AUTHOR = "author";
EXPORT const char* AMEDIAFORMAT_KEY_BITRATE_MODE = "bitrate-mode";
diff --git a/media/ndk/include/media/NdkMediaCodec.h b/media/ndk/include/media/NdkMediaCodec.h
index 9dc120d..b3ee853 100644
--- a/media/ndk/include/media/NdkMediaCodec.h
+++ b/media/ndk/include/media/NdkMediaCodec.h
@@ -241,12 +241,6 @@
AMediaFormat* AMediaCodec_getOutputFormat(AMediaCodec*) __INTRODUCED_IN(21);
/**
- * Get format of the buffer. The specified buffer index must have been previously obtained from
- * dequeueOutputBuffer.
- */
-AMediaFormat* AMediaCodec_getBufferFormat(AMediaCodec*, size_t index) __INTRODUCED_IN(21);
-
-/**
* If you are done with a buffer, use this call to return the buffer to
* the codec. If you previously specified a surface when configuring this
* video decoder you can optionally render the buffer.
@@ -353,6 +347,12 @@
#if __ANDROID_API__ >= 28
/**
+ * Get format of the buffer. The specified buffer index must have been previously obtained from
+ * dequeueOutputBuffer.
+ */
+AMediaFormat* AMediaCodec_getBufferFormat(AMediaCodec*, size_t index) __INTRODUCED_IN(28);
+
+/**
* Get the component name. If the codec was created by createDecoderByType
* or createEncoderByType, what component is chosen is not known beforehand.
* Caller shall call AMediaCodec_releaseName to free the returned pointer.
diff --git a/media/ndk/include/media/NdkMediaFormat.h b/media/ndk/include/media/NdkMediaFormat.h
index 13d9135..2cd1d04 100644
--- a/media/ndk/include/media/NdkMediaFormat.h
+++ b/media/ndk/include/media/NdkMediaFormat.h
@@ -181,6 +181,8 @@
extern const char* AMEDIAFORMAT_KEY_ALBUMARTIST __INTRODUCED_IN(29);
extern const char* AMEDIAFORMAT_KEY_ARTIST __INTRODUCED_IN(29);
extern const char* AMEDIAFORMAT_KEY_AUDIO_PRESENTATION_INFO __INTRODUCED_IN(29);
+extern const char* AMEDIAFORMAT_KEY_AUDIO_PRESENTATION_PRESENTATION_ID __INTRODUCED_IN(29);
+extern const char* AMEDIAFORMAT_KEY_AUDIO_PRESENTATION_PROGRAM_ID __INTRODUCED_IN(29);
extern const char* AMEDIAFORMAT_KEY_AUTHOR __INTRODUCED_IN(29);
extern const char* AMEDIAFORMAT_KEY_BITS_PER_SAMPLE __INTRODUCED_IN(29);
extern const char* AMEDIAFORMAT_KEY_CDTRACKNUMBER __INTRODUCED_IN(29);
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index 88736ab..3567899 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -158,7 +158,7 @@
AMediaCodec_dequeueInputBuffer;
AMediaCodec_dequeueOutputBuffer;
AMediaCodec_flush;
- AMediaCodec_getBufferFormat; # introduced=21
+ AMediaCodec_getBufferFormat; # introduced=28
AMediaCodec_getInputBuffer;
AMediaCodec_getInputFormat; # introduced=28
AMediaCodec_getName; # introduced=28
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 1b20693..6c698f6 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -365,16 +365,17 @@
static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) {
switch (audio_channel_mask_get_representation(channelMask)) {
case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
- uint32_t channelCount = FCC_2; // stereo is default
- if (kEnableExtendedChannels) {
- channelCount = audio_channel_count_from_out_mask(channelMask);
- if (channelCount < FCC_2 // mono is not supported at this time
- || channelCount > AudioMixer::MAX_NUM_CHANNELS) {
- return false;
- }
+ // Haptic channel mask is only applicable for channel position mask.
+ const uint32_t channelCount = audio_channel_count_from_out_mask(
+ channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL);
+ const uint32_t maxChannelCount = kEnableExtendedChannels
+ ? AudioMixer::MAX_NUM_CHANNELS : FCC_2;
+ if (channelCount < FCC_2 // mono is not supported at this time
+ || channelCount > maxChannelCount) {
+ return false;
}
// check that channelMask is the "canonical" one we expect for the channelCount.
- return channelMask == audio_channel_out_mask_from_count(channelCount);
+ return audio_channel_position_mask_is_out_canonical(channelMask);
}
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
if (kEnableExtendedChannels) {
diff --git a/services/audioflinger/BufLog.cpp b/services/audioflinger/BufLog.cpp
index ae96036..5f6aca0 100644
--- a/services/audioflinger/BufLog.cpp
+++ b/services/audioflinger/BufLog.cpp
@@ -115,7 +115,7 @@
unsigned int samplingRate,
size_t maxBytes = 0) : mId(id), mFormat(format), mChannels(channels),
mSamplingRate(samplingRate), mMaxBytes(maxBytes) {
- mByteCount = 0l;
+ mByteCount = 0;
mPaused = false;
if (tag != NULL) {
(void)audio_utils_strlcpy(mTag, tag);
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index c1db78b..ea6389c 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -228,6 +228,8 @@
bool reported) = 0;
virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled) = 0;
+ virtual bool isHapticPlaybackSupported() = 0;
+
virtual void setAppState(uid_t uid, app_state_t state);
};
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index fb0e436..f07b797 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -484,7 +484,7 @@
audio_devices_t outputDevice = isRx ? device : AUDIO_DEVICE_OUT_TELEPHONY_TX;
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(outputDevice, mOutputs);
- audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID);
+ audio_io_handle_t output = selectOutput(outputs);
// request to reuse existing output stream if one is already opened to reach the target device
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
@@ -774,7 +774,7 @@
// and AudioSystem::getOutputSamplingRate().
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
- audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID);
+ audio_io_handle_t output = selectOutput(outputs);
ALOGV("getOutput() stream %d selected device %08x, output %d", stream, device, output);
return output;
@@ -1091,7 +1091,8 @@
// at this stage we should ignore the DIRECT flag as no direct output could be found earlier
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
- output = selectOutput(outputs, *flags, config->format);
+ output = selectOutput(outputs, *flags, config->format,
+ config->channel_mask, config->sample_rate);
}
ALOGW_IF((output == 0), "getOutputForDevice() could not find output for stream %d, "
"sampling rate %d, format %#x, channels %#x, flags %#x",
@@ -1252,15 +1253,18 @@
audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
audio_output_flags_t flags,
- audio_format_t format)
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t samplingRate)
{
// select one output among several that provide a path to a particular device or set of
// devices (the list was previously build by getOutputsForDevice()).
// The priority is as follows:
- // 1: the output with the highest number of requested policy flags
- // 2: the output with the bit depth the closest to the requested one
- // 3: the primary output
- // 4: the first output in the list
+ // 1: the output supporting haptic playback when requesting haptic playback
+ // 2: the output with the highest number of requested policy flags
+ // 3: the output with the bit depth the closest to the requested one
+ // 4: the primary output
+ // 5: the first output in the list
if (outputs.size() == 0) {
return AUDIO_IO_HANDLE_NONE;
@@ -1270,6 +1274,8 @@
}
int maxCommonFlags = 0;
+ const size_t hapticChannelCount = audio_channel_count_from_out_mask(
+ channelMask & AUDIO_CHANNEL_HAPTIC_ALL);
audio_io_handle_t outputForFlags = AUDIO_IO_HANDLE_NONE;
audio_io_handle_t outputForPrimary = AUDIO_IO_HANDLE_NONE;
audio_io_handle_t outputForFormat = AUDIO_IO_HANDLE_NONE;
@@ -1282,6 +1288,24 @@
if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
continue;
}
+ // If haptic channel is specified, use the haptic output if present.
+ // When using haptic output, same audio format and sample rate are required.
+ if (hapticChannelCount > 0) {
+ // If haptic channel is specified, use the first output that
+ // support haptic playback.
+ if (audio_channel_count_from_out_mask(
+ outputDesc->mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL) >= hapticChannelCount
+ && format == outputDesc->mFormat
+ && samplingRate == outputDesc->mSamplingRate) {
+ return output;
+ }
+ } else {
+ // When haptic channel is not specified, skip haptic output.
+ if (outputDesc->mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL) {
+ continue;
+ }
+ }
+
// if a valid format is specified, skip output if not compatible
if (format != AUDIO_FORMAT_INVALID) {
if (!audio_is_linear_pcm(format)) {
@@ -3036,9 +3060,7 @@
getOutputsForDevice(sinkDeviceDesc->type(), mOutputs);
// if the sink device is reachable via an opened output stream, request to go via
// this output stream by adding a second source to the patch description
- audio_io_handle_t output = selectOutput(outputs,
- AUDIO_OUTPUT_FLAG_NONE,
- AUDIO_FORMAT_INVALID);
+ audio_io_handle_t output = selectOutput(outputs);
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc->isDuplicated()) {
@@ -3382,8 +3404,7 @@
// create Hwoutput and add to mHwOutputs
} else {
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(sinkDevice, mOutputs);
- audio_io_handle_t output =
- selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID);
+ audio_io_handle_t output = selectOutput(outputs);
if (output == AUDIO_IO_HANDLE_NONE) {
ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevice);
return INVALID_OPERATION;
@@ -3655,6 +3676,23 @@
}
}
+bool AudioPolicyManager::isHapticPlaybackSupported()
+{
+ for (const auto& hwModule : mHwModules) {
+ const OutputProfileCollection &outputProfiles = hwModule->getOutputProfiles();
+ for (const auto &outProfile : outputProfiles) {
+ struct audio_port audioPort;
+ outProfile->toAudioPort(&audioPort);
+ for (size_t i = 0; i < audioPort.num_channel_masks; i++) {
+ if (audioPort.channel_masks[i] & AUDIO_CHANNEL_HAPTIC_ALL) {
+ return true;
+ }
+ }
+ }
+ }
+ return false;
+}
+
status_t AudioPolicyManager::disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
{
ALOGV("%s port Id %d", __FUNCTION__, sourceDesc->portId());
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 709cce0..d0708b8 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -247,6 +247,8 @@
virtual void setAppState(uid_t uid, app_state_t state);
+ virtual bool isHapticPlaybackSupported();
+
protected:
// A constructor that allows more fine-grained control over initialization process,
// used in automatic tests.
@@ -473,8 +475,10 @@
uint32_t delayMs);
audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
- audio_output_flags_t flags,
- audio_format_t format);
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ audio_format_t format = AUDIO_FORMAT_INVALID,
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE,
+ uint32_t samplingRate = 0);
// samplingRate, format, channelMask are in/out and so may be modified
sp<IOProfile> getInputProfile(audio_devices_t device,
const String8& address,
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index ae92ae5..439764b 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -1133,4 +1133,15 @@
return NO_ERROR;
}
+bool AudioPolicyService::isHapticPlaybackSupported()
+{
+ if (mAudioPolicyManager == NULL) {
+ ALOGW("%s, mAudioPolicyManager == NULL", __func__);
+ return false;
+ }
+ Mutex::Autolock _l(mLock);
+ AutoCallerClear acc;
+ return mAudioPolicyManager->isHapticPlaybackSupported();
+}
+
} // namespace android
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index f490fb9..23c3daa 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -218,6 +218,8 @@
virtual status_t setAssistantUid(uid_t uid);
virtual status_t setA11yServicesUids(const std::vector<uid_t>& uids);
+ virtual bool isHapticPlaybackSupported();
+
status_t doStopOutput(audio_port_handle_t portId);
void doReleaseOutput(audio_port_handle_t portId);