gecko/dom/media/fmp4/apple/AppleATDecoder.cpp

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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* 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 "AppleUtils.h"
#include "MP4Reader.h"
#include "MP4Decoder.h"
#include "mp4_demuxer/Adts.h"
#include "mp4_demuxer/DecoderData.h"
#include "AppleATDecoder.h"
#include "prlog.h"
#ifdef PR_LOGGING
PRLogModuleInfo* GetAppleMediaLog();
#define LOG(...) PR_LOG(GetAppleMediaLog(), PR_LOG_DEBUG, (__VA_ARGS__))
#else
#define LOG(...)
#endif
#define FourCC2Str(n) ((char[5]){(char)(n >> 24), (char)(n >> 16), (char)(n >> 8), (char)(n), 0})
namespace mozilla {
AppleATDecoder::AppleATDecoder(const mp4_demuxer::AudioDecoderConfig& aConfig,
MediaTaskQueue* aAudioTaskQueue,
MediaDataDecoderCallback* aCallback)
: mConfig(aConfig)
, mFileStreamError(false)
, mTaskQueue(aAudioTaskQueue)
, mCallback(aCallback)
, mConverter(nullptr)
, mStream(nullptr)
{
MOZ_COUNT_CTOR(AppleATDecoder);
LOG("Creating Apple AudioToolbox decoder");
LOG("Audio Decoder configuration: %s %d Hz %d channels %d bits per channel",
mConfig.mime_type,
mConfig.samples_per_second,
mConfig.channel_count,
mConfig.bits_per_sample);
if (!strcmp(mConfig.mime_type, "audio/mpeg")) {
mFormatID = kAudioFormatMPEGLayer3;
} else if (!strcmp(mConfig.mime_type, "audio/mp4a-latm")) {
mFormatID = kAudioFormatMPEG4AAC;
} else {
mFormatID = 0;
}
}
AppleATDecoder::~AppleATDecoder()
{
MOZ_COUNT_DTOR(AppleATDecoder);
MOZ_ASSERT(!mConverter);
}
nsresult
AppleATDecoder::Init()
{
if (!mFormatID) {
NS_ERROR("Non recognised format");
return NS_ERROR_FAILURE;
}
return NS_OK;
}
nsresult
AppleATDecoder::Input(mp4_demuxer::MP4Sample* aSample)
{
LOG("mp4 input sample %p %lld us %lld pts%s %llu bytes audio",
aSample,
aSample->duration,
aSample->composition_timestamp,
aSample->is_sync_point ? " keyframe" : "",
(unsigned long long)aSample->size);
// Queue a task to perform the actual decoding on a separate thread.
mTaskQueue->Dispatch(
NS_NewRunnableMethodWithArg<nsAutoPtr<mp4_demuxer::MP4Sample>>(
this,
&AppleATDecoder::SubmitSample,
nsAutoPtr<mp4_demuxer::MP4Sample>(aSample)));
return NS_OK;
}
nsresult
AppleATDecoder::Flush()
{
LOG("Flushing AudioToolbox AAC decoder");
mTaskQueue->Flush();
mQueuedSamples.Clear();
OSStatus rv = AudioConverterReset(mConverter);
if (rv) {
LOG("Error %d resetting AudioConverter", rv);
return NS_ERROR_FAILURE;
}
return NS_OK;
}
nsresult
AppleATDecoder::Drain()
{
LOG("Draining AudioToolbox AAC decoder");
mTaskQueue->AwaitIdle();
mCallback->DrainComplete();
return Flush();
}
nsresult
AppleATDecoder::Shutdown()
{
LOG("Shutdown: Apple AudioToolbox AAC decoder");
mQueuedSamples.Clear();
OSStatus rv = AudioConverterDispose(mConverter);
if (rv) {
LOG("error %d disposing of AudioConverter", rv);
return NS_ERROR_FAILURE;
}
mConverter = nullptr;
if (mStream) {
rv = AudioFileStreamClose(mStream);
if (rv) {
LOG("error %d disposing of AudioFileStream", rv);
return NS_ERROR_FAILURE;
}
mStream = nullptr;
}
return NS_OK;
}
struct PassthroughUserData {
UInt32 mChannels;
UInt32 mDataSize;
const void* mData;
AudioStreamPacketDescription mPacket;
};
// Error value we pass through the decoder to signal that nothing
// has gone wrong during decoding and we're done processing the packet.
const uint32_t kNoMoreDataErr = 'MOAR';
static OSStatus
_PassthroughInputDataCallback(AudioConverterRef aAudioConverter,
UInt32* aNumDataPackets /* in/out */,
AudioBufferList* aData /* in/out */,
AudioStreamPacketDescription** aPacketDesc,
void* aUserData)
{
PassthroughUserData* userData = (PassthroughUserData*)aUserData;
if (!userData->mDataSize) {
*aNumDataPackets = 0;
return kNoMoreDataErr;
}
LOG("AudioConverter wants %u packets of audio data\n", *aNumDataPackets);
if (aPacketDesc) {
userData->mPacket.mStartOffset = 0;
userData->mPacket.mVariableFramesInPacket = 0;
userData->mPacket.mDataByteSize = userData->mDataSize;
*aPacketDesc = &userData->mPacket;
}
aData->mBuffers[0].mNumberChannels = userData->mChannels;
aData->mBuffers[0].mDataByteSize = userData->mDataSize;
aData->mBuffers[0].mData = const_cast<void*>(userData->mData);
// No more data to provide following this run.
userData->mDataSize = 0;
return noErr;
}
void
AppleATDecoder::SubmitSample(nsAutoPtr<mp4_demuxer::MP4Sample> aSample)
{
nsresult rv = NS_OK;
if (!mConverter) {
rv = SetupDecoder(aSample);
if (rv != NS_OK && rv != NS_ERROR_NOT_INITIALIZED) {
mCallback->Error();
return;
}
}
mQueuedSamples.AppendElement(aSample);
if (rv == NS_OK) {
for (size_t i = 0; i < mQueuedSamples.Length(); i++) {
if (NS_FAILED(DecodeSample(mQueuedSamples[i]))) {
mQueuedSamples.Clear();
mCallback->Error();
return;
}
}
mQueuedSamples.Clear();
}
if (mTaskQueue->IsEmpty()) {
mCallback->InputExhausted();
}
}
nsresult
AppleATDecoder::DecodeSample(mp4_demuxer::MP4Sample* aSample)
{
// Array containing the queued decoded audio frames, about to be output.
nsTArray<AudioDataValue> outputData;
UInt32 channels = mOutputFormat.mChannelsPerFrame;
// Pick a multiple of the frame size close to a power of two
// for efficient allocation.
const uint32_t MAX_AUDIO_FRAMES = 128;
const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * channels;
// Descriptions for _decompressed_ audio packets. ignored.
nsAutoArrayPtr<AudioStreamPacketDescription>
packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
// This API insists on having packets spoon-fed to it from a callback.
// This structure exists only to pass our state.
PassthroughUserData userData =
{ channels, (UInt32)aSample->size, aSample->data };
// Decompressed audio buffer
nsAutoArrayPtr<AudioDataValue> decoded(new AudioDataValue[maxDecodedSamples]);
do {
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = channels;
decBuffer.mBuffers[0].mDataByteSize =
maxDecodedSamples * sizeof(AudioDataValue);
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
_PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNoMoreDataErr) {
LOG("Error decoding audio stream: %d\n", rv);
return NS_ERROR_FAILURE;
}
if (numFrames) {
outputData.AppendElements(decoded.get(), numFrames * channels);
LOG("%d frames decoded", numFrames);
}
if (rv == kNoMoreDataErr) {
LOG("done processing compressed packet");
break;
}
} while (true);
if (outputData.IsEmpty()) {
return NS_OK;
}
size_t numFrames = outputData.Length() / channels;
int rate = mOutputFormat.mSampleRate;
CheckedInt<Microseconds> duration = FramesToUsecs(numFrames, rate);
if (!duration.isValid()) {
NS_WARNING("Invalid count of accumulated audio samples");
return NS_ERROR_FAILURE;
}
LOG("pushed audio at time %lfs; duration %lfs\n",
(double)aSample->composition_timestamp / USECS_PER_S,
(double)duration.value() / USECS_PER_S);
nsAutoArrayPtr<AudioDataValue> data(new AudioDataValue[outputData.Length()]);
PodCopy(data.get(), &outputData[0], outputData.Length());
nsRefPtr<AudioData> audio = new AudioData(aSample->byte_offset,
aSample->composition_timestamp,
duration.value(),
numFrames,
data.forget(),
channels,
rate);
mCallback->Output(audio);
return NS_OK;
}
nsresult
AppleATDecoder::GetInputAudioDescription(AudioStreamBasicDescription& aDesc,
const Vector<uint8_t>& aExtraData)
{
// Request the properties from CoreAudio using the codec magic cookie
AudioFormatInfo formatInfo;
PodZero(&formatInfo.mASBD);
formatInfo.mASBD.mFormatID = mFormatID;
if (mFormatID == kAudioFormatMPEG4AAC) {
formatInfo.mASBD.mFormatFlags = mConfig.extended_profile;
}
formatInfo.mMagicCookieSize = aExtraData.length();
formatInfo.mMagicCookie = aExtraData.begin();
UInt32 formatListSize;
// Attempt to retrieve the default format using
// kAudioFormatProperty_FormatInfo method.
// This method only retrieves the FramesPerPacket information required
// by the decoder, which depends on the codec type and profile.
aDesc.mFormatID = mFormatID;
aDesc.mChannelsPerFrame = mConfig.channel_count;
aDesc.mSampleRate = mConfig.samples_per_second;
UInt32 inputFormatSize = sizeof(aDesc);
OSStatus rv = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
0,
NULL,
&inputFormatSize,
&aDesc);
if (NS_WARN_IF(rv)) {
return NS_ERROR_FAILURE;
}
// If any of the methods below fail, we will return the default format as
// created using kAudioFormatProperty_FormatInfo above.
rv = AudioFormatGetPropertyInfo(kAudioFormatProperty_FormatList,
sizeof(formatInfo),
&formatInfo,
&formatListSize);
if (rv || (formatListSize % sizeof(AudioFormatListItem))) {
return NS_OK;
}
size_t listCount = formatListSize / sizeof(AudioFormatListItem);
nsAutoArrayPtr<AudioFormatListItem> formatList(
new AudioFormatListItem[listCount]);
rv = AudioFormatGetProperty(kAudioFormatProperty_FormatList,
sizeof(formatInfo),
&formatInfo,
&formatListSize,
formatList);
if (rv) {
return NS_OK;
}
LOG("found %u available audio stream(s)",
formatListSize / sizeof(AudioFormatListItem));
// Get the index number of the first playable format.
// This index number will be for the highest quality layer the platform
// is capable of playing.
UInt32 itemIndex;
UInt32 indexSize = sizeof(itemIndex);
rv = AudioFormatGetProperty(kAudioFormatProperty_FirstPlayableFormatFromList,
formatListSize,
formatList,
&indexSize,
&itemIndex);
if (rv) {
return NS_OK;
}
aDesc = formatList[itemIndex].mASBD;
return NS_OK;
}
nsresult
AppleATDecoder::SetupDecoder(mp4_demuxer::MP4Sample* aSample)
{
if (mFormatID == kAudioFormatMPEG4AAC &&
mConfig.extended_profile == 2) {
// Check for implicit SBR signalling if stream is AAC-LC
// This will provide us with an updated magic cookie for use with
// GetInputAudioDescription.
if (NS_SUCCEEDED(GetImplicitAACMagicCookie(aSample)) &&
!mMagicCookie.length()) {
// nothing found yet, will try again later
return NS_ERROR_NOT_INITIALIZED;
}
// An error occurred, fallback to using default stream description
}
LOG("Initializing Apple AudioToolbox decoder");
AudioStreamBasicDescription inputFormat;
PodZero(&inputFormat);
nsresult rv =
GetInputAudioDescription(inputFormat,
mMagicCookie.length() ?
mMagicCookie : mConfig.extra_data);
if (NS_FAILED(rv)) {
return rv;
}
// Fill in the output format manually.
PodZero(&mOutputFormat);
mOutputFormat.mFormatID = kAudioFormatLinearPCM;
mOutputFormat.mSampleRate = inputFormat.mSampleRate;
mOutputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
#if defined(MOZ_SAMPLE_TYPE_FLOAT32)
mOutputFormat.mBitsPerChannel = 32;
mOutputFormat.mFormatFlags =
kLinearPCMFormatFlagIsFloat |
0;
#else
# error Unknown audio sample type
#endif
// Set up the decoder so it gives us one sample per frame
mOutputFormat.mFramesPerPacket = 1;
mOutputFormat.mBytesPerPacket = mOutputFormat.mBytesPerFrame
= mOutputFormat.mChannelsPerFrame * mOutputFormat.mBitsPerChannel / 8;
OSStatus status = AudioConverterNew(&inputFormat, &mOutputFormat, &mConverter);
if (status) {
LOG("Error %d constructing AudioConverter", status);
mConverter = nullptr;
return NS_ERROR_FAILURE;
}
return NS_OK;
}
static void
_MetadataCallback(void* aAppleATDecoder,
AudioFileStreamID aStream,
AudioFileStreamPropertyID aProperty,
UInt32* aFlags)
{
AppleATDecoder* decoder = static_cast<AppleATDecoder*>(aAppleATDecoder);
LOG("MetadataCallback receiving: '%s'", FourCC2Str(aProperty));
if (aProperty == kAudioFileStreamProperty_MagicCookieData) {
UInt32 size;
Boolean writeable;
OSStatus rv = AudioFileStreamGetPropertyInfo(aStream,
aProperty,
&size,
&writeable);
if (rv) {
LOG("Couldn't get property info for '%s' (%s)",
FourCC2Str(aProperty), FourCC2Str(rv));
decoder->mFileStreamError = true;
return;
}
nsAutoArrayPtr<uint8_t> data(new uint8_t[size]);
rv = AudioFileStreamGetProperty(aStream, aProperty,
&size, data);
if (rv) {
LOG("Couldn't get property '%s' (%s)",
FourCC2Str(aProperty), FourCC2Str(rv));
decoder->mFileStreamError = true;
return;
}
decoder->mMagicCookie.append(data.get(), size);
}
}
static void
_SampleCallback(void* aSBR,
UInt32 aNumBytes,
UInt32 aNumPackets,
const void* aData,
AudioStreamPacketDescription* aPackets)
{
}
nsresult
AppleATDecoder::GetImplicitAACMagicCookie(const mp4_demuxer::MP4Sample* aSample)
{
// Prepend ADTS header to AAC audio.
mp4_demuxer::MP4Sample adtssample(*aSample);
bool rv = mp4_demuxer::Adts::ConvertSample(mConfig.channel_count,
mConfig.frequency_index,
mConfig.aac_profile,
&adtssample);
if (!rv) {
NS_WARNING("Failed to apply ADTS header");
return NS_ERROR_FAILURE;
}
if (!mStream) {
OSStatus rv = AudioFileStreamOpen(this,
_MetadataCallback,
_SampleCallback,
kAudioFileAAC_ADTSType,
&mStream);
if (rv) {
NS_WARNING("Couldn't open AudioFileStream");
return NS_ERROR_FAILURE;
}
}
OSStatus status = AudioFileStreamParseBytes(mStream,
adtssample.size,
adtssample.data,
0 /* discontinuity */);
if (status) {
NS_WARNING("Couldn't parse sample");
}
if (status || mFileStreamError || mMagicCookie.length()) {
// We have decoded a magic cookie or an error occurred as such
// we won't need the stream any longer.
AudioFileStreamClose(mStream);
mStream = nullptr;
}
return (mFileStreamError || status) ? NS_ERROR_FAILURE : NS_OK;
}
} // namespace mozilla