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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 "WMFAudioMFTManager.h"
#include "MediaInfo.h"
#include "TimeUnits.h"
#include "VideoUtils.h"
#include "WMFUtils.h"
#include "mozilla/AbstractThread.h"
#include "mozilla/Logging.h"
#include "mozilla/Telemetry.h"
#include "nsTArray.h"
#define LOG(...) MOZ_LOG(sPDMLog, mozilla::LogLevel::Debug, (__VA_ARGS__))
namespace mozilla {
using media::TimeUnit;
static void AACAudioSpecificConfigToUserData(uint8_t aAACProfileLevelIndication,
const uint8_t* aAudioSpecConfig,
uint32_t aConfigLength,
nsTArray<BYTE>& aOutUserData) {
MOZ_ASSERT(aOutUserData.IsEmpty());
// The MF_MT_USER_DATA for AAC is defined here:
//
// For MFAudioFormat_AAC, MF_MT_USER_DATA contains the portion of
// the HEAACWAVEINFO structure that appears after the WAVEFORMATEX
// structure (that is, after the wfx member). This is followed by
// the AudioSpecificConfig() data, as defined by ISO/IEC 14496-3.
// [...]
// The length of the AudioSpecificConfig() data is 2 bytes for AAC-LC
// or HE-AAC with implicit signaling of SBR/PS. It is more than 2 bytes
// for HE-AAC with explicit signaling of SBR/PS.
//
// The value of audioObjectType as defined in AudioSpecificConfig()
// must be 2, indicating AAC-LC. The value of extensionAudioObjectType
// must be 5 for SBR or 29 for PS.
//
// HEAACWAVEINFO structure:
// typedef struct heaacwaveinfo_tag {
// WAVEFORMATEX wfx;
// WORD wPayloadType;
// WORD wAudioProfileLevelIndication;
// WORD wStructType;
// WORD wReserved1;
// DWORD dwReserved2;
// }
const UINT32 heeInfoLen = 4 * sizeof(WORD) + sizeof(DWORD);
// The HEAACWAVEINFO must have payload and profile set,
// the rest can be all 0x00.
BYTE heeInfo[heeInfoLen] = {0};
WORD* w = (WORD*)heeInfo;
w[0] = 0x0; // Payload type raw AAC packet
w[1] = aAACProfileLevelIndication;
aOutUserData.AppendElements(heeInfo, heeInfoLen);
if (aAACProfileLevelIndication == 2 && aConfigLength > 2) {
// The AudioSpecificConfig is TTTTTFFF|FCCCCGGG
// (T=ObjectType, F=Frequency, C=Channel, G=GASpecificConfig)
// If frequency = 0xf, then the frequency is explicitly defined on 24 bits.
int8_t frequency =
(aAudioSpecConfig[0] & 0x7) << 1 | (aAudioSpecConfig[1] & 0x80) >> 7;
int8_t channels = (aAudioSpecConfig[1] & 0x78) >> 3;
int8_t gasc = aAudioSpecConfig[1] & 0x7;
if (frequency != 0xf && channels && !gasc) {
// We enter this condition if the AudioSpecificConfig should theorically
// be 2 bytes long but it's not.
// The WMF AAC decoder will error if unknown extensions are found,
// so remove them.
aConfigLength = 2;
}
}
aOutUserData.AppendElements(aAudioSpecConfig, aConfigLength);
}
WMFAudioMFTManager::WMFAudioMFTManager(const AudioInfo& aConfig)
: mAudioChannels(aConfig.mChannels),
mChannelsMap(AudioConfig::ChannelLayout::UNKNOWN_MAP),
mAudioRate(aConfig.mRate) {
MOZ_COUNT_CTOR(WMFAudioMFTManager);
if (aConfig.mMimeType.EqualsLiteral("audio/mpeg")) {
mStreamType = MP3;
} else if (aConfig.mMimeType.EqualsLiteral("audio/mp4a-latm")) {
mStreamType = AAC;
AACAudioSpecificConfigToUserData(
aConfig.mExtendedProfile, aConfig.mCodecSpecificConfig->Elements(),
aConfig.mCodecSpecificConfig->Length(), mUserData);
} else {
mStreamType = Unknown;
}
}
WMFAudioMFTManager::~WMFAudioMFTManager() {
MOZ_COUNT_DTOR(WMFAudioMFTManager);
}
const GUID& WMFAudioMFTManager::GetMFTGUID() {
MOZ_ASSERT(mStreamType != Unknown);
switch (mStreamType) {
case AAC:
return CLSID_CMSAACDecMFT;
case MP3:
return CLSID_CMP3DecMediaObject;
default:
return GUID_NULL;
};
}
const GUID& WMFAudioMFTManager::GetMediaSubtypeGUID() {
MOZ_ASSERT(mStreamType != Unknown);
switch (mStreamType) {
case AAC:
return MFAudioFormat_AAC;
case MP3:
return MFAudioFormat_MP3;
default:
return GUID_NULL;
};
}
bool WMFAudioMFTManager::Init() {
NS_ENSURE_TRUE(mStreamType != Unknown, false);
RefPtr<MFTDecoder> decoder(new MFTDecoder());
HRESULT hr = decoder->Create(GetMFTGUID());
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
// Setup input/output media types
RefPtr<IMFMediaType> inputType;
hr = wmf::MFCreateMediaType(getter_AddRefs(inputType));
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = inputType->SetGUID(MF_MT_MAJOR_TYPE, MFMediaType_Audio);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = inputType->SetGUID(MF_MT_SUBTYPE, GetMediaSubtypeGUID());
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = inputType->SetUINT32(MF_MT_AUDIO_SAMPLES_PER_SECOND, mAudioRate);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = inputType->SetUINT32(MF_MT_AUDIO_NUM_CHANNELS, mAudioChannels);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
if (mStreamType == AAC) {
hr = inputType->SetUINT32(MF_MT_AAC_PAYLOAD_TYPE, 0x0); // Raw AAC packet
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = inputType->SetBlob(MF_MT_USER_DATA, mUserData.Elements(),
mUserData.Length());
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
}
RefPtr<IMFMediaType> outputType;
hr = wmf::MFCreateMediaType(getter_AddRefs(outputType));
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = outputType->SetGUID(MF_MT_MAJOR_TYPE, MFMediaType_Audio);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = outputType->SetGUID(MF_MT_SUBTYPE, MFAudioFormat_Float);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = outputType->SetUINT32(MF_MT_AUDIO_BITS_PER_SAMPLE, 32);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
hr = decoder->SetMediaTypes(inputType, outputType);
NS_ENSURE_TRUE(SUCCEEDED(hr), false);
mDecoder = decoder;
return true;
}
HRESULT
WMFAudioMFTManager::Input(MediaRawData* aSample) {
return mDecoder->Input(aSample->Data(), uint32_t(aSample->Size()),
aSample->mTime.ToMicroseconds(),
aSample->mDuration.ToMicroseconds());
}
HRESULT
WMFAudioMFTManager::UpdateOutputType() {
HRESULT hr;
RefPtr<IMFMediaType> type;
hr = mDecoder->GetOutputMediaType(type);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
hr = type->GetUINT32(MF_MT_AUDIO_SAMPLES_PER_SECOND, &mAudioRate);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
hr = type->GetUINT32(MF_MT_AUDIO_NUM_CHANNELS, &mAudioChannels);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
uint32_t channelsMap;
hr = type->GetUINT32(MF_MT_AUDIO_CHANNEL_MASK, &channelsMap);
if (SUCCEEDED(hr)) {
mChannelsMap = channelsMap;
} else {
LOG("Unable to retrieve channel layout. Ignoring");
mChannelsMap = AudioConfig::ChannelLayout::UNKNOWN_MAP;
}
return S_OK;
}
HRESULT
WMFAudioMFTManager::Output(int64_t aStreamOffset, RefPtr<MediaData>& aOutData) {
aOutData = nullptr;
RefPtr<IMFSample> sample;
HRESULT hr;
int typeChangeCount = 0;
while (true) {
hr = mDecoder->Output(&sample);
if (hr == MF_E_TRANSFORM_NEED_MORE_INPUT) {
return hr;
}
if (hr == MF_E_TRANSFORM_STREAM_CHANGE) {
hr = mDecoder->FindDecoderOutputType();
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
hr = UpdateOutputType();
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
// Catch infinite loops, but some decoders perform at least 2 stream
// changes on consecutive calls, so be permissive.
// 100 is arbitrarily > 2.
NS_ENSURE_TRUE(typeChangeCount < 100, MF_E_TRANSFORM_STREAM_CHANGE);
++typeChangeCount;
continue;
}
break;
}
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
if (!sample) {
LOG("Audio MFTDecoder returned success but null output.");
return E_FAIL;
}
UINT32 discontinuity = false;
sample->GetUINT32(MFSampleExtension_Discontinuity, &discontinuity);
if (mFirstFrame || discontinuity) {
// Update the output type, in case this segment has a different
// rate. This also triggers on the first sample, which can have a
// different rate than is advertised in the container, and sometimes we
// don't get a MF_E_TRANSFORM_STREAM_CHANGE when the rate changes.
hr = UpdateOutputType();
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
mFirstFrame = false;
}
TimeUnit pts = GetSampleTime(sample);
NS_ENSURE_TRUE(pts.IsValid(), E_FAIL);
RefPtr<IMFMediaBuffer> buffer;
hr = sample->ConvertToContiguousBuffer(getter_AddRefs(buffer));
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
BYTE* data = nullptr; // Note: *data will be owned by the IMFMediaBuffer, we
// don't need to free it.
DWORD maxLength = 0, currentLength = 0;
hr = buffer->Lock(&data, &maxLength, &currentLength);
NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
// Output is made of floats.
int32_t numSamples = currentLength / sizeof(float);
int32_t numFrames = numSamples / mAudioChannels;
MOZ_ASSERT(numFrames >= 0);
MOZ_ASSERT(numSamples >= 0);
if (numFrames == 0) {
// All data from this chunk stripped, loop back and try to output the next
// frame, if possible.
return S_OK;
}
AlignedAudioBuffer audioData(numSamples);
if (!audioData) {
return E_OUTOFMEMORY;
}
PodCopy(audioData.Data(), reinterpret_cast<float*>(data), numSamples);
buffer->Unlock();
TimeUnit duration = FramesToTimeUnit(numFrames, mAudioRate);
NS_ENSURE_TRUE(duration.IsValid(), E_FAIL);
aOutData = new AudioData(aStreamOffset, pts, std::move(audioData),
mAudioChannels, mAudioRate, mChannelsMap);
MOZ_DIAGNOSTIC_ASSERT(duration == aOutData->mDuration, "must be equal");
#ifdef LOG_SAMPLE_DECODE
LOG("Decoded audio sample! timestamp=%lld duration=%lld currentLength=%u",
pts.ToMicroseconds(), duration.ToMicroseconds(), currentLength);
#endif
return S_OK;
}
void WMFAudioMFTManager::Shutdown() { mDecoder = nullptr; }
} // namespace mozilla
#undef LOG