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gecko/dom/media/omx/OMXCodecWrapper.cpp

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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "OMXCodecWrapper.h"
#include "OMXCodecDescriptorUtil.h"
#include "TrackEncoder.h"
#include <binder/ProcessState.h>
#include <cutils/properties.h>
#include <media/ICrypto.h>
#include <media/IOMX.h>
#include <OMX_Component.h>
#include <stagefright/MediaDefs.h>
#include <stagefright/MediaErrors.h>
#include "AudioChannelFormat.h"
#include <mozilla/Monitor.h>
#include "mozilla/layers/GrallocTextureClient.h"
using namespace mozilla;
using namespace mozilla::gfx;
using namespace mozilla::layers;
#define INPUT_BUFFER_TIMEOUT_US (5 * 1000ll)
// AMR NB kbps
#define AMRNB_BITRATE 12200
#define CODEC_ERROR(args...) \
do { \
__android_log_print(ANDROID_LOG_ERROR, "OMXCodecWrapper", ##args); \
} while (0)
namespace android {
enum BufferState
{
BUFFER_OK,
BUFFER_FAIL,
WAIT_FOR_NEW_BUFFER
};
bool
OMXCodecReservation::ReserveOMXCodec()
{
if (!mManagerService.get()) {
sp<MediaResourceManagerClient::EventListener> listener = this;
mClient = new MediaResourceManagerClient(listener);
mManagerService = mClient->getMediaResourceManagerService();
if (!mManagerService.get()) {
mClient = nullptr;
return true; // not really in use, but not usable
}
}
return (mManagerService->requestMediaResource(mClient, mType, false) == OK); // don't wait
}
void
OMXCodecReservation::ReleaseOMXCodec()
{
if (!mManagerService.get() || !mClient.get()) {
return;
}
mManagerService->cancelClient(mClient, mType);
}
OMXAudioEncoder*
OMXCodecWrapper::CreateAACEncoder()
{
nsAutoPtr<OMXAudioEncoder> aac(new OMXAudioEncoder(CodecType::AAC_ENC));
// Return the object only when media codec is valid.
NS_ENSURE_TRUE(aac->IsValid(), nullptr);
return aac.forget();
}
OMXAudioEncoder*
OMXCodecWrapper::CreateAMRNBEncoder()
{
nsAutoPtr<OMXAudioEncoder> amr(new OMXAudioEncoder(CodecType::AMR_NB_ENC));
// Return the object only when media codec is valid.
NS_ENSURE_TRUE(amr->IsValid(), nullptr);
return amr.forget();
}
OMXVideoEncoder*
OMXCodecWrapper::CreateAVCEncoder()
{
nsAutoPtr<OMXVideoEncoder> avc(new OMXVideoEncoder(CodecType::AVC_ENC));
// Return the object only when media codec is valid.
NS_ENSURE_TRUE(avc->IsValid(), nullptr);
return avc.forget();
}
OMXCodecWrapper::OMXCodecWrapper(CodecType aCodecType)
: mCodecType(aCodecType)
, mStarted(false)
, mAMRCSDProvided(false)
{
ProcessState::self()->startThreadPool();
mLooper = new ALooper();
mLooper->start();
if (aCodecType == CodecType::AVC_ENC) {
mCodec = MediaCodec::CreateByType(mLooper, MEDIA_MIMETYPE_VIDEO_AVC, true);
} else if (aCodecType == CodecType::AMR_NB_ENC) {
mCodec = MediaCodec::CreateByType(mLooper, MEDIA_MIMETYPE_AUDIO_AMR_NB, true);
} else if (aCodecType == CodecType::AAC_ENC) {
mCodec = MediaCodec::CreateByType(mLooper, MEDIA_MIMETYPE_AUDIO_AAC, true);
} else {
NS_ERROR("Unknown codec type.");
}
}
OMXCodecWrapper::~OMXCodecWrapper()
{
if (mCodec.get()) {
Stop();
mCodec->release();
}
mLooper->stop();
}
status_t
OMXCodecWrapper::Start()
{
// Already started.
NS_ENSURE_FALSE(mStarted, OK);
status_t result = mCodec->start();
mStarted = (result == OK);
// Get references to MediaCodec buffers.
if (result == OK) {
mCodec->getInputBuffers(&mInputBufs);
mCodec->getOutputBuffers(&mOutputBufs);
}
return result;
}
status_t
OMXCodecWrapper::Stop()
{
// Already stopped.
NS_ENSURE_TRUE(mStarted, OK);
status_t result = mCodec->stop();
mStarted = !(result == OK);
return result;
}
// Check system property to see if we're running on emulator.
static bool
IsRunningOnEmulator()
{
char qemu[PROPERTY_VALUE_MAX];
property_get("ro.kernel.qemu", qemu, "");
return strncmp(qemu, "1", 1) == 0;
}
#define ENCODER_CONFIG_BITRATE 2000000 // bps
// How many seconds between I-frames.
#define ENCODER_CONFIG_I_FRAME_INTERVAL 1
// Wait up to 5ms for input buffers.
nsresult
OMXVideoEncoder::Configure(int aWidth, int aHeight, int aFrameRate,
BlobFormat aBlobFormat)
{
NS_ENSURE_TRUE(aWidth > 0 && aHeight > 0 && aFrameRate > 0,
NS_ERROR_INVALID_ARG);
OMX_VIDEO_AVCLEVELTYPE level = OMX_VIDEO_AVCLevel3;
OMX_VIDEO_CONTROLRATETYPE bitrateMode = OMX_Video_ControlRateConstant;
// Set up configuration parameters for AVC/H.264 encoder.
sp<AMessage> format = new AMessage;
// Fixed values
format->setString("mime", MEDIA_MIMETYPE_VIDEO_AVC);
format->setInt32("bitrate", ENCODER_CONFIG_BITRATE);
format->setInt32("i-frame-interval", ENCODER_CONFIG_I_FRAME_INTERVAL);
// See mozilla::layers::GrallocImage, supports YUV 4:2:0, CbCr width and
// height is half that of Y
format->setInt32("color-format", OMX_COLOR_FormatYUV420SemiPlanar);
format->setInt32("profile", OMX_VIDEO_AVCProfileBaseline);
format->setInt32("level", level);
format->setInt32("bitrate-mode", bitrateMode);
format->setInt32("store-metadata-in-buffers", 0);
format->setInt32("prepend-sps-pps-to-idr-frames", 0);
// Input values.
format->setInt32("width", aWidth);
format->setInt32("height", aHeight);
format->setInt32("stride", aWidth);
format->setInt32("slice-height", aHeight);
format->setInt32("frame-rate", aFrameRate);
return ConfigureDirect(format, aBlobFormat);
}
nsresult
OMXVideoEncoder::ConfigureDirect(sp<AMessage>& aFormat,
BlobFormat aBlobFormat)
{
// We now allow re-configuration to handle resolution/framerate/etc changes
if (mStarted) {
Stop();
}
MOZ_ASSERT(!mStarted, "OMX Stop() failed?");
int width = 0;
int height = 0;
int frameRate = 0;
aFormat->findInt32("width", &width);
aFormat->findInt32("height", &height);
aFormat->findInt32("frame-rate", &frameRate);
NS_ENSURE_TRUE(width > 0 && height > 0 && frameRate > 0,
NS_ERROR_INVALID_ARG);
// Limitation of soft AVC/H.264 encoder running on emulator in stagefright.
static bool emu = IsRunningOnEmulator();
if (emu) {
if (width > 352 || height > 288) {
CODEC_ERROR("SoftAVCEncoder doesn't support resolution larger than CIF");
return NS_ERROR_INVALID_ARG;
}
aFormat->setInt32("level", OMX_VIDEO_AVCLevel2);
aFormat->setInt32("bitrate-mode", OMX_Video_ControlRateVariable);
}
status_t result = mCodec->configure(aFormat, nullptr, nullptr,
MediaCodec::CONFIGURE_FLAG_ENCODE);
NS_ENSURE_TRUE(result == OK, NS_ERROR_FAILURE);
mWidth = width;
mHeight = height;
mBlobFormat = aBlobFormat;
result = Start();
return result == OK ? NS_OK : NS_ERROR_FAILURE;
}
// Copy pixels from planar YUV (4:4:4/4:2:2/4:2:0) or NV21 (semi-planar 4:2:0)
// format to NV12 (semi-planar 4:2:0) format for QCOM HW encoder.
// Planar YUV: YYY...UUU...VVV...
// NV21: YYY...VUVU...
// NV12: YYY...UVUV...
// For 4:4:4/4:2:2 -> 4:2:0, subsample using odd row/column without
// interpolation.
// aSource contains info about source image data, and the result will be stored
// in aDestination, whose size needs to be >= Y plane size * 3 / 2.
static void
ConvertPlanarYCbCrToNV12(const PlanarYCbCrData* aSource, uint8_t* aDestination)
{
// Fill Y plane.
uint8_t* y = aSource->mYChannel;
IntSize ySize = aSource->mYSize;
// Y plane.
for (int i = 0; i < ySize.height; i++) {
memcpy(aDestination, y, ySize.width);
aDestination += ySize.width;
y += aSource->mYStride;
}
// Fill interleaved UV plane.
uint8_t* u = aSource->mCbChannel;
uint8_t* v = aSource->mCrChannel;
IntSize uvSize = aSource->mCbCrSize;
// Subsample to 4:2:0 if source is 4:4:4 or 4:2:2.
// Y plane width & height should be multiple of U/V plane width & height.
MOZ_ASSERT(ySize.width % uvSize.width == 0 &&
ySize.height % uvSize.height == 0);
size_t uvWidth = ySize.width / 2;
size_t uvHeight = ySize.height / 2;
size_t horiSubsample = uvSize.width / uvWidth;
size_t uPixStride = horiSubsample * (1 + aSource->mCbSkip);
size_t vPixStride = horiSubsample * (1 + aSource->mCrSkip);
size_t lineStride = uvSize.height / uvHeight * aSource->mCbCrStride;
for (int i = 0; i < uvHeight; i++) {
// 1st pixel per line.
uint8_t* uSrc = u;
uint8_t* vSrc = v;
for (int j = 0; j < uvWidth; j++) {
*aDestination++ = *uSrc;
*aDestination++ = *vSrc;
// Pick next source pixel.
uSrc += uPixStride;
vSrc += vPixStride;
}
// Pick next source line.
u += lineStride;
v += lineStride;
}
}
// Convert pixels in graphic buffer to NV12 format. aSource is the layer image
// containing source graphic buffer, and aDestination is the destination of
// conversion. Currently 3 source format are supported:
// - NV21/HAL_PIXEL_FORMAT_YCrCb_420_SP (from camera preview window).
// - YV12/HAL_PIXEL_FORMAT_YV12 (from video decoder).
// - QCOM proprietary/HAL_PIXEL_FORMAT_YCbCr_420_SP_VENUS (from Flame HW video decoder)
static void
ConvertGrallocImageToNV12(GrallocImage* aSource, uint8_t* aDestination)
{
// Get graphic buffer.
sp<GraphicBuffer> graphicBuffer = aSource->GetGraphicBuffer();
int pixelFormat = graphicBuffer->getPixelFormat();
void* imgPtr = nullptr;
graphicBuffer->lock(GraphicBuffer::USAGE_SW_READ_MASK, &imgPtr);
// Build PlanarYCbCrData for NV21 or YV12 buffer.
PlanarYCbCrData yuv;
switch (pixelFormat) {
case HAL_PIXEL_FORMAT_YCrCb_420_SP: // From camera.
yuv.mYChannel = static_cast<uint8_t*>(imgPtr);
yuv.mYSkip = 0;
yuv.mYSize.width = graphicBuffer->getWidth();
yuv.mYSize.height = graphicBuffer->getHeight();
yuv.mYStride = graphicBuffer->getStride();
// 4:2:0.
yuv.mCbCrSize.width = yuv.mYSize.width / 2;
yuv.mCbCrSize.height = yuv.mYSize.height / 2;
// Interleaved VU plane.
yuv.mCrChannel = yuv.mYChannel + (yuv.mYStride * yuv.mYSize.height);
yuv.mCrSkip = 1;
yuv.mCbChannel = yuv.mCrChannel + 1;
yuv.mCbSkip = 1;
yuv.mCbCrStride = yuv.mYStride;
ConvertPlanarYCbCrToNV12(&yuv, aDestination);
break;
case HAL_PIXEL_FORMAT_YV12: // From video decoder.
// Android YV12 format is defined in system/core/include/system/graphics.h
yuv.mYChannel = static_cast<uint8_t*>(imgPtr);
yuv.mYSkip = 0;
yuv.mYSize.width = graphicBuffer->getWidth();
yuv.mYSize.height = graphicBuffer->getHeight();
yuv.mYStride = graphicBuffer->getStride();
// 4:2:0.
yuv.mCbCrSize.width = yuv.mYSize.width / 2;
yuv.mCbCrSize.height = yuv.mYSize.height / 2;
yuv.mCrChannel = yuv.mYChannel + (yuv.mYStride * yuv.mYSize.height);
// Aligned to 16 bytes boundary.
yuv.mCbCrStride = (yuv.mYStride / 2 + 15) & ~0x0F;
yuv.mCrSkip = 0;
yuv.mCbChannel = yuv.mCrChannel + (yuv.mCbCrStride * yuv.mCbCrSize.height);
yuv.mCbSkip = 0;
ConvertPlanarYCbCrToNV12(&yuv, aDestination);
break;
// From QCOM video decoder on Flame. See bug 997593.
case GrallocImage::HAL_PIXEL_FORMAT_YCbCr_420_SP_VENUS:
// Venus formats are doucmented in kernel/include/media/msm_media_info.h:
yuv.mYChannel = static_cast<uint8_t*>(imgPtr);
yuv.mYSkip = 0;
yuv.mYSize.width = graphicBuffer->getWidth();
yuv.mYSize.height = graphicBuffer->getHeight();
// - Y & UV Width aligned to 128
yuv.mYStride = (yuv.mYSize.width + 127) & ~127;
yuv.mCbCrSize.width = yuv.mYSize.width / 2;
yuv.mCbCrSize.height = yuv.mYSize.height / 2;
// - Y height aligned to 32
yuv.mCbChannel = yuv.mYChannel + (yuv.mYStride * ((yuv.mYSize.height + 31) & ~31));
// Interleaved VU plane.
yuv.mCbSkip = 1;
yuv.mCrChannel = yuv.mCbChannel + 1;
yuv.mCrSkip = 1;
yuv.mCbCrStride = yuv.mYStride;
ConvertPlanarYCbCrToNV12(&yuv, aDestination);
break;
default:
NS_ERROR("Unsupported input gralloc image type. Should never be here.");
}
graphicBuffer->unlock();
}
nsresult
OMXVideoEncoder::Encode(const Image* aImage, int aWidth, int aHeight,
int64_t aTimestamp, int aInputFlags)
{
MOZ_ASSERT(mStarted, "Configure() should be called before Encode().");
NS_ENSURE_TRUE(aWidth == mWidth && aHeight == mHeight && aTimestamp >= 0,
NS_ERROR_INVALID_ARG);
Image* img = const_cast<Image*>(aImage);
ImageFormat format = ImageFormat::PLANAR_YCBCR;
if (img) {
format = img->GetFormat();
gfx::IntSize size = img->GetSize();
// Validate input image.
NS_ENSURE_TRUE(aWidth == size.width, NS_ERROR_INVALID_ARG);
NS_ENSURE_TRUE(aHeight == size.height, NS_ERROR_INVALID_ARG);
if (format == ImageFormat::PLANAR_YCBCR) {
// Test for data, allowing SetDataNoCopy() on an image without an mBuffer
// (as used from WebrtcOMXH264VideoCodec, and a few other places) - bug 1067442
const PlanarYCbCrData* yuv = static_cast<PlanarYCbCrImage*>(img)->GetData();
NS_ENSURE_TRUE(yuv->mYChannel, NS_ERROR_INVALID_ARG);
} else if (format == ImageFormat::GRALLOC_PLANAR_YCBCR) {
// Reject unsupported gralloc-ed buffers.
int halFormat = static_cast<GrallocImage*>(img)->GetGraphicBuffer()->getPixelFormat();
NS_ENSURE_TRUE(halFormat == HAL_PIXEL_FORMAT_YCrCb_420_SP ||
halFormat == HAL_PIXEL_FORMAT_YV12 ||
halFormat == GrallocImage::HAL_PIXEL_FORMAT_YCbCr_420_SP_VENUS,
NS_ERROR_INVALID_ARG);
} else {
// TODO: support RGB to YUV color conversion.
NS_ERROR("Unsupported input image type.");
return NS_ERROR_INVALID_ARG;
}
}
status_t result;
// Dequeue an input buffer.
uint32_t index;
result = mCodec->dequeueInputBuffer(&index, INPUT_BUFFER_TIMEOUT_US);
if (result == -EAGAIN) {
// Drop the frame when out of input buffer.
return NS_OK;
}
NS_ENSURE_TRUE(result == OK, NS_ERROR_FAILURE);
const sp<ABuffer>& inBuf = mInputBufs.itemAt(index);
uint8_t* dst = inBuf->data();
size_t dstSize = inBuf->capacity();
size_t yLen = aWidth * aHeight;
size_t uvLen = yLen / 2;
// Buffer should be large enough to hold input image data.
MOZ_ASSERT(dstSize >= yLen + uvLen);
dstSize = yLen + uvLen;
inBuf->setRange(0, dstSize);
if (!img) {
// Generate muted/black image directly in buffer.
// Fill Y plane.
memset(dst, 0x10, yLen);
// Fill UV plane.
memset(dst + yLen, 0x80, uvLen);
} else {
if (format == ImageFormat::GRALLOC_PLANAR_YCBCR) {
ConvertGrallocImageToNV12(static_cast<GrallocImage*>(img), dst);
} else if (format == ImageFormat::PLANAR_YCBCR) {
ConvertPlanarYCbCrToNV12(static_cast<PlanarYCbCrImage*>(img)->GetData(),
dst);
}
}
// Queue this input buffer.
result = mCodec->queueInputBuffer(index, 0, dstSize, aTimestamp, aInputFlags);
return result == OK ? NS_OK : NS_ERROR_FAILURE;
}
status_t
OMXVideoEncoder::AppendDecoderConfig(nsTArray<uint8_t>* aOutputBuf,
ABuffer* aData)
{
// Codec already parsed aData. Using its result makes generating config blob
// much easier.
sp<AMessage> format;
mCodec->getOutputFormat(&format);
// NAL unit format is needed by WebRTC for RTP packets; AVC/H.264 decoder
// config descriptor is needed to construct MP4 'avcC' box.
status_t result = GenerateAVCDescriptorBlob(format, aOutputBuf, mBlobFormat);
return result;
}
// Override to replace NAL unit start code with 4-bytes unit length.
// See ISO/IEC 14496-15 5.2.3.
void
OMXVideoEncoder::AppendFrame(nsTArray<uint8_t>* aOutputBuf,
const uint8_t* aData, size_t aSize)
{
aOutputBuf->SetCapacity(aSize);
if (mBlobFormat == BlobFormat::AVC_NAL) {
// Append NAL format data without modification.
aOutputBuf->AppendElements(aData, aSize);
return;
}
// Replace start code with data length.
uint8_t length[] = {
(aSize >> 24) & 0xFF,
(aSize >> 16) & 0xFF,
(aSize >> 8) & 0xFF,
aSize & 0xFF,
};
aOutputBuf->AppendElements(length, sizeof(length));
aOutputBuf->AppendElements(aData + sizeof(length), aSize);
}
// MediaCodec::setParameters() is available only after API level 18.
#if ANDROID_VERSION >= 18
nsresult
OMXVideoEncoder::SetBitrate(int32_t aKbps)
{
sp<AMessage> msg = new AMessage();
#if ANDROID_VERSION >= 19
// XXX Do we need a runtime check here?
msg->setInt32("video-bitrate", aKbps * 1000 /* kbps -> bps */);
#else
msg->setInt32("videoBitrate", aKbps * 1000 /* kbps -> bps */);
#endif
status_t result = mCodec->setParameters(msg);
MOZ_ASSERT(result == OK);
return result == OK ? NS_OK : NS_ERROR_FAILURE;
}
#endif
nsresult
OMXVideoEncoder::RequestIDRFrame()
{
MOZ_ASSERT(mStarted, "Configure() should be called before RequestIDRFrame().");
return mCodec->requestIDRFrame() == OK ? NS_OK : NS_ERROR_FAILURE;
}
nsresult
OMXAudioEncoder::Configure(int aChannels, int aInputSampleRate,
int aEncodedSampleRate)
{
MOZ_ASSERT(!mStarted);
NS_ENSURE_TRUE(aChannels > 0 && aInputSampleRate > 0 && aEncodedSampleRate >= 0,
NS_ERROR_INVALID_ARG);
if (aInputSampleRate != aEncodedSampleRate) {
int error;
mResampler = speex_resampler_init(aChannels,
aInputSampleRate,
aEncodedSampleRate,
SPEEX_RESAMPLER_QUALITY_DEFAULT,
&error);
if (error != RESAMPLER_ERR_SUCCESS) {
return NS_ERROR_FAILURE;
}
speex_resampler_skip_zeros(mResampler);
}
// Set up configuration parameters for AAC encoder.
sp<AMessage> format = new AMessage;
// Fixed values.
if (mCodecType == AAC_ENC) {
format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
format->setInt32("aac-profile", OMX_AUDIO_AACObjectLC);
format->setInt32("bitrate", kAACBitrate);
format->setInt32("sample-rate", aInputSampleRate);
} else if (mCodecType == AMR_NB_ENC) {
format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_NB);
format->setInt32("bitrate", AMRNB_BITRATE);
format->setInt32("sample-rate", aEncodedSampleRate);
} else {
MOZ_ASSERT(false, "Can't support this codec type!!");
}
// Input values.
format->setInt32("channel-count", aChannels);
status_t result = mCodec->configure(format, nullptr, nullptr,
MediaCodec::CONFIGURE_FLAG_ENCODE);
NS_ENSURE_TRUE(result == OK, NS_ERROR_FAILURE);
mChannels = aChannels;
mSampleDuration = 1000000 / aInputSampleRate;
mResamplingRatio = aEncodedSampleRate > 0 ? 1.0 *
aEncodedSampleRate / aInputSampleRate : 1.0;
result = Start();
return result == OK ? NS_OK : NS_ERROR_FAILURE;
}
class InputBufferHelper MOZ_FINAL {
public:
InputBufferHelper(sp<MediaCodec>& aCodec, Vector<sp<ABuffer> >& aBuffers,
OMXAudioEncoder& aEncoder, int aInputFlags)
: mCodec(aCodec)
, mBuffers(aBuffers)
, mOMXAEncoder(aEncoder)
, mInputFlags(aInputFlags)
, mIndex(0)
, mData(nullptr)
, mOffset(0)
, mCapicity(0)
{}
~InputBufferHelper()
{
// Unflushed data in buffer.
MOZ_ASSERT(!mData);
}
status_t Dequeue()
{
// Shouldn't have dequeued buffer.
MOZ_ASSERT(!mData);
status_t result = mCodec->dequeueInputBuffer(&mIndex,
INPUT_BUFFER_TIMEOUT_US);
NS_ENSURE_TRUE(result == OK, result);
sp<ABuffer> inBuf = mBuffers.itemAt(mIndex);
mData = inBuf->data();
mCapicity = inBuf->capacity();
mOffset = 0;
return OK;
}
status_t Enqueue(int64_t aTimestamp, int aFlags)
{
// Should have dequeued buffer.
MOZ_ASSERT(mData);
// Queue this buffer.
status_t result = mCodec->queueInputBuffer(mIndex, 0, mOffset, aTimestamp,
aFlags);
NS_ENSURE_TRUE(result == OK, result);
mData = nullptr;
return OK;
}
// Read audio data in aChunk, resample them if needed,
// and then send the result to OMX input buffer (or buffers if one buffer is not enough).
// aSamplesRead will be the number of samples that have been read from aChunk.
BufferState ReadChunk(AudioChunk& aChunk, size_t* aSamplesRead)
{
size_t chunkSamples = aChunk.GetDuration();
size_t bytesToCopy = chunkSamples * mOMXAEncoder.mResamplingRatio
* mOMXAEncoder.mChannels * sizeof(AudioDataValue);
size_t bytesCopied = 0;
if (bytesToCopy <= AvailableSize()) {
if (aChunk.IsNull()) {
bytesCopied = SendSilenceToBuffer(chunkSamples);
} else {
bytesCopied = SendChunkToBuffer(aChunk, chunkSamples);
}
UpdateAfterSendChunk(chunkSamples, bytesCopied, aSamplesRead);
} else {
// Interleave data to a temporary buffer.
nsAutoTArray<AudioDataValue, 9600> pcm;
pcm.SetLength(bytesToCopy);
AudioDataValue* interleavedSource = pcm.Elements();
AudioTrackEncoder::InterleaveTrackData(aChunk, chunkSamples,
mOMXAEncoder.mChannels,
interleavedSource);
// When the data size of chunk is larger than the buffer capacity,
// we split it into sub-chunks to fill up buffers.
size_t subChunkSamples = 0;
while(GetNextSubChunk(bytesToCopy, subChunkSamples)) {
// To avoid enqueueing an empty buffer, we follow the order that
// clear up buffer first, then create one, send data to it in the end.
if (!IsEmpty()) {
// Submit the filled-up buffer and request a new buffer.
status_t result = Enqueue(mOMXAEncoder.mTimestamp,
mInputFlags & ~OMXCodecWrapper::BUFFER_EOS);
if (result != OK) {
return BUFFER_FAIL;
}
result = Dequeue();
if (result == -EAGAIN) {
return WAIT_FOR_NEW_BUFFER;
}
if (result != OK) {
return BUFFER_FAIL;
}
}
if (aChunk.IsNull()) {
bytesCopied = SendSilenceToBuffer(subChunkSamples);
} else {
bytesCopied = SendInterleavedSubChunkToBuffer(interleavedSource, subChunkSamples);
}
UpdateAfterSendChunk(subChunkSamples, bytesCopied, aSamplesRead);
// Move to the position where samples are not yet send to the buffer.
interleavedSource += subChunkSamples * mOMXAEncoder.mChannels;
}
}
return BUFFER_OK;
}
// No audio data left in segment but we still have to feed something to
// MediaCodec in order to notify EOS.
void SendEOSToBuffer(size_t* aSamplesRead)
{
size_t bytesToCopy = SendSilenceToBuffer(1);
IncreaseOffset(bytesToCopy);
*aSamplesRead = 1;
}
private:
uint8_t* GetPointer() { return mData + mOffset; }
const size_t AvailableSize() { return mCapicity - mOffset; }
void IncreaseOffset(size_t aValue)
{
// Should never out of bound.
MOZ_ASSERT(mOffset + aValue <= mCapicity);
mOffset += aValue;
}
bool IsEmpty()
{
return (mOffset == 0);
}
const size_t GetCapacity()
{
return mCapicity;
}
// Update buffer offset, timestamp and the total number of copied samples.
void UpdateAfterSendChunk(size_t aSamplesNum, size_t aBytesToCopy,
size_t* aSourceSamplesCopied)
{
*aSourceSamplesCopied += aSamplesNum;
mOMXAEncoder.mTimestamp += aSamplesNum * mOMXAEncoder.mSampleDuration;
IncreaseOffset(aBytesToCopy);
}
// Send slince auido data when the chunk is null,
// and return the copied bytes number of audio data.
size_t SendSilenceToBuffer(size_t aSamplesNum)
{
AudioDataValue* dst = reinterpret_cast<AudioDataValue*>(GetPointer());
size_t bytesToCopy = aSamplesNum * mOMXAEncoder.mResamplingRatio
* mOMXAEncoder.mChannels * sizeof(AudioDataValue);
memset(dst, 0, bytesToCopy);
return bytesToCopy;
}
// Interleave chunk data and send it to buffer,
// and return the copied bytes number of audio data.
size_t SendChunkToBuffer(AudioChunk& aSource, size_t aSamplesNum)
{
AudioDataValue* dst = reinterpret_cast<AudioDataValue*>(GetPointer());
size_t bytesToCopy = aSamplesNum * mOMXAEncoder.mResamplingRatio
* mOMXAEncoder.mChannels * sizeof(AudioDataValue);
uint32_t dstSamplesCopied = aSamplesNum;
if (mOMXAEncoder.mResampler) {
nsAutoTArray<AudioDataValue, 9600> pcm;
pcm.SetLength(bytesToCopy);
AudioTrackEncoder::InterleaveTrackData(aSource, aSamplesNum,
mOMXAEncoder.mChannels,
pcm.Elements());
int16_t* tempSource = reinterpret_cast<int16_t*>(pcm.Elements());
speex_resampler_process_interleaved_int(mOMXAEncoder.mResampler, tempSource,
&aSamplesNum, dst,
&dstSamplesCopied);
} else {
AudioTrackEncoder::InterleaveTrackData(aSource, aSamplesNum,
mOMXAEncoder.mChannels, dst);
}
return dstSamplesCopied * mOMXAEncoder.mChannels * sizeof(AudioDataValue);
}
// Send the interleaved data of the sub chunk to buffer,
// and return the copied bytes number of audio data.
size_t SendInterleavedSubChunkToBuffer(AudioDataValue* aSource,
size_t aSamplesNum)
{
AudioDataValue* dst = reinterpret_cast<AudioDataValue*>(GetPointer());
uint32_t dstSamplesCopied = aSamplesNum;
if (mOMXAEncoder.mResampler) {
int16_t* tempSource = reinterpret_cast<int16_t*>(aSource);
speex_resampler_process_interleaved_int(mOMXAEncoder.mResampler,
tempSource, &aSamplesNum,
dst, &dstSamplesCopied);
} else {
// Directly copy interleaved data into buffer
memcpy(dst, aSource,
aSamplesNum * mOMXAEncoder.mChannels * sizeof(AudioDataValue));
}
return dstSamplesCopied * mOMXAEncoder.mChannels * sizeof(AudioDataValue);
}
// Determine the size of sub-chunk (aSamplesToCopy) according to buffer capacity.
// For subsequent call, the number of bytes remain to be copied will also be updated in this function.
bool GetNextSubChunk(size_t& aBytesToCopy, size_t& aSamplesToCopy)
{
size_t bufferCapabity = GetCapacity();
size_t sampleBytes = mOMXAEncoder.mChannels * mOMXAEncoder.mResamplingRatio
* sizeof(AudioDataValue);
if (aBytesToCopy) {
if (aBytesToCopy > bufferCapabity) {
aSamplesToCopy = bufferCapabity / sampleBytes;
aBytesToCopy -= aSamplesToCopy * sampleBytes;
} else {
aSamplesToCopy = aBytesToCopy / sampleBytes;
aBytesToCopy = 0;
}
return true;
}
return false;
}
sp<MediaCodec>& mCodec;
Vector<sp<ABuffer> >& mBuffers;
OMXAudioEncoder& mOMXAEncoder;
int mInputFlags;
size_t mIndex;
uint8_t* mData;
size_t mCapicity;
size_t mOffset;
};
OMXAudioEncoder::~OMXAudioEncoder()
{
if (mResampler) {
speex_resampler_destroy(mResampler);
mResampler = nullptr;
}
}
nsresult
OMXAudioEncoder::Encode(AudioSegment& aSegment, int aInputFlags)
{
#ifndef MOZ_SAMPLE_TYPE_S16
#error MediaCodec accepts only 16-bit PCM data.
#endif
MOZ_ASSERT(mStarted, "Configure() should be called before Encode().");
size_t numSamples = aSegment.GetDuration();
// Get input buffer.
InputBufferHelper buffer(mCodec, mInputBufs, *this, aInputFlags);
status_t result = buffer.Dequeue();
if (result == -EAGAIN) {
// All input buffers are full. Caller can try again later after consuming
// some output buffers.
return NS_OK;
}
NS_ENSURE_TRUE(result == OK, NS_ERROR_FAILURE);
size_t sourceSamplesCopied = 0; // Number of copied samples.
if (numSamples > 0) {
// Copy input PCM data to input buffer until queue is empty.
AudioSegment::ChunkIterator iter(const_cast<AudioSegment&>(aSegment));
while (!iter.IsEnded()) {
BufferState result = buffer.ReadChunk(*iter, &sourceSamplesCopied);
if (result == WAIT_FOR_NEW_BUFFER) {
// All input buffers are full. Caller can try again later after
// consuming some output buffers.
aSegment.RemoveLeading(sourceSamplesCopied);
return NS_OK;
} else if (result == BUFFER_FAIL) {
return NS_ERROR_FAILURE;
} else {
iter.Next();
}
}
// Remove the samples already been copied into buffer
if (sourceSamplesCopied > 0) {
aSegment.RemoveLeading(sourceSamplesCopied);
}
} else if (aInputFlags & BUFFER_EOS) {
buffer.SendEOSToBuffer(&sourceSamplesCopied);
}
// Enqueue the remaining data to buffer
MOZ_ASSERT(sourceSamplesCopied > 0, "No data needs to be enqueued!");
int flags = aInputFlags;
if (aSegment.GetDuration() > 0) {
// Don't signal EOS until source segment is empty.
flags &= ~BUFFER_EOS;
}
result = buffer.Enqueue(mTimestamp, flags);
NS_ENSURE_TRUE(result == OK, NS_ERROR_FAILURE);
return NS_OK;
}
// Generate decoder config descriptor (defined in ISO/IEC 14496-1 8.3.4.1) for
// AAC. The hard-coded bytes are copied from
// MPEG4Writer::Track::writeMp4aEsdsBox() implementation in libstagefright.
status_t
OMXAudioEncoder::AppendDecoderConfig(nsTArray<uint8_t>* aOutputBuf,
ABuffer* aData)
{
MOZ_ASSERT(aData);
const size_t csdSize = aData->size();
// See
// http://wiki.multimedia.cx/index.php?title=Understanding_AAC#Packaging.2FEncapsulation_And_Setup_Data
// AAC decoder specific descriptor contains 2 bytes.
NS_ENSURE_TRUE(csdSize == 2, ERROR_MALFORMED);
// Encoder output must be consistent with kAACFrameDuration:
// 14th bit (frame length flag) == 0 => 1024 (kAACFrameDuration) samples.
NS_ENSURE_TRUE((aData->data()[1] & 0x04) == 0, ERROR_MALFORMED);
// Decoder config descriptor
const uint8_t decConfig[] = {
0x04, // Decoder config descriptor tag.
15 + csdSize, // Size: following bytes + csd size.
0x40, // Object type: MPEG-4 audio.
0x15, // Stream type: audio, reserved: 1.
0x00, 0x03, 0x00, // Buffer size: 768 (kAACFrameSize).
0x00, 0x01, 0x77, 0x00, // Max bitrate: 96000 (kAACBitrate).
0x00, 0x01, 0x77, 0x00, // Avg bitrate: 96000 (kAACBitrate).
0x05, // Decoder specific descriptor tag.
csdSize, // Data size.
};
// SL config descriptor.
const uint8_t slConfig[] = {
0x06, // SL config descriptor tag.
0x01, // Size.
0x02, // Fixed value.
};
aOutputBuf->SetCapacity(sizeof(decConfig) + csdSize + sizeof(slConfig));
aOutputBuf->AppendElements(decConfig, sizeof(decConfig));
aOutputBuf->AppendElements(aData->data(), csdSize);
aOutputBuf->AppendElements(slConfig, sizeof(slConfig));
return OK;
}
nsresult
OMXCodecWrapper::GetNextEncodedFrame(nsTArray<uint8_t>* aOutputBuf,
int64_t* aOutputTimestamp,
int* aOutputFlags, int64_t aTimeOut)
{
MOZ_ASSERT(mStarted,
"Configure() should be called before GetNextEncodedFrame().");
// Dequeue a buffer from output buffers.
size_t index = 0;
size_t outOffset = 0;
size_t outSize = 0;
int64_t outTimeUs = 0;
uint32_t outFlags = 0;
bool retry = false;
do {
status_t result = mCodec->dequeueOutputBuffer(&index, &outOffset, &outSize,
&outTimeUs, &outFlags,
aTimeOut);
switch (result) {
case OK:
break;
case INFO_OUTPUT_BUFFERS_CHANGED:
// Update our references to new buffers.
result = mCodec->getOutputBuffers(&mOutputBufs);
// Get output from a new buffer.
retry = true;
break;
case INFO_FORMAT_CHANGED:
// It's okay: for encoder, MediaCodec reports this only to inform caller
// that there will be a codec config buffer next.
return NS_OK;
case -EAGAIN:
// Output buffer not available. Caller can try again later.
return NS_OK;
default:
CODEC_ERROR("MediaCodec error:%d", result);
MOZ_ASSERT(false, "MediaCodec error.");
return NS_ERROR_FAILURE;
}
} while (retry);
if (aOutputBuf) {
aOutputBuf->Clear();
const sp<ABuffer> omxBuf = mOutputBufs.itemAt(index);
if (outFlags & MediaCodec::BUFFER_FLAG_CODECCONFIG) {
// Codec specific data.
if (AppendDecoderConfig(aOutputBuf, omxBuf.get()) != OK) {
mCodec->releaseOutputBuffer(index);
return NS_ERROR_FAILURE;
}
} else if ((mCodecType == AMR_NB_ENC) && !mAMRCSDProvided){
// OMX AMR codec won't provide csd data, need to generate a fake one.
nsRefPtr<EncodedFrame> audiodata = new EncodedFrame();
// Decoder config descriptor
const uint8_t decConfig[] = {
0x0, 0x0, 0x0, 0x0, // vendor: 4 bytes
0x0, // decoder version
0x83, 0xFF, // mode set: all enabled
0x00, // mode change period
0x01, // frames per sample
};
aOutputBuf->AppendElements(decConfig, sizeof(decConfig));
outFlags |= MediaCodec::BUFFER_FLAG_CODECCONFIG;
mAMRCSDProvided = true;
} else {
AppendFrame(aOutputBuf, omxBuf->data(), omxBuf->size());
}
}
mCodec->releaseOutputBuffer(index);
if (aOutputTimestamp) {
*aOutputTimestamp = outTimeUs;
}
if (aOutputFlags) {
*aOutputFlags = outFlags;
}
return NS_OK;
}
}
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