Add m-esr52 at 52.6.0

1 files changed, 358 insertions, 0 deletions

diff --git a/dom/media/platforms/wmf/WMFAudioMFTManager.cpp b/dom/media/platforms/wmf/WMFAudioMFTManager.cpp
new file mode 100644
index 000000000..69b62da51
--- /dev/null
+++ b/dom/media/platforms/wmf/WMFAudioMFTManager.cpp
@@ -0,0 +1,358 @@
+/* -*- 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 "VideoUtils.h"
+#include "WMFUtils.h"
+#include "nsTArray.h"
+#include "TimeUnits.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/Logging.h"
+
+#define LOG(...) MOZ_LOG(sPDMLog, mozilla::LogLevel::Debug, (__VA_ARGS__))
+
+namespace mozilla {
+
+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:
+  // http://msdn.microsoft.com/en-us/library/windows/desktop/dd742784%28v=vs.85%29.aspx
+  //
+  // 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 profile = (aAudioSpecConfig[0] & 0xF8) >> 3;
+    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)
+  , mAudioRate(aConfig.mRate)
+  , mAudioFrameSum(0)
+  , mMustRecaptureAudioPosition(true)
+{
+  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_PCM);
+  NS_ENSURE_TRUE(SUCCEEDED(hr), false);
+
+  hr = outputType->SetUINT32(MF_MT_AUDIO_BITS_PER_SAMPLE, 16);
+  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);
+}
+
+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);
+
+  AudioConfig::ChannelLayout layout(mAudioChannels);
+  if (!layout.IsValid()) {
+    return E_FAIL;
+  }
+
+  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 = 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.");
+    nsCOMPtr<nsIRunnable> task = NS_NewRunnableFunction([]() -> void {
+      LOG("Reporting telemetry AUDIO_MFT_OUTPUT_NULL_SAMPLES");
+      Telemetry::Accumulate(Telemetry::ID::AUDIO_MFT_OUTPUT_NULL_SAMPLES, 1);
+    });
+    AbstractThread::MainThread()->Dispatch(task.forget());
+    return 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);
+
+  // Sometimes when starting decoding, the AAC decoder gives us samples
+  // with a negative timestamp. AAC does usually have preroll (or encoder
+  // delay) encoded into its bitstream, but the amount encoded to the stream
+  // is variable, and it not signalled in-bitstream. There is sometimes
+  // signalling in the MP4 container what the preroll amount, but it's
+  // inconsistent. It looks like WMF's AAC encoder may take this into
+  // account, so strip off samples with a negative timestamp to get us
+  // to a 0-timestamp start. This seems to maintain A/V sync, so we can run
+  // with this until someone complains...
+
+  // We calculate the timestamp and the duration based on the number of audio
+  // frames we've already played. We don't trust the timestamp stored on the
+  // IMFSample, as sometimes it's wrong, possibly due to buggy encoders?
+
+  // If this sample block comes after a discontinuity (i.e. a gap or seek)
+  // reset the frame counters, and capture the timestamp. Future timestamps
+  // will be offset from this block's timestamp.
+  UINT32 discontinuity = false;
+  sample->GetUINT32(MFSampleExtension_Discontinuity, &discontinuity);
+  if (mMustRecaptureAudioPosition || 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);
+
+    mAudioFrameSum = 0;
+    LONGLONG timestampHns = 0;
+    hr = sample->GetSampleTime(&timestampHns);
+    NS_ENSURE_TRUE(SUCCEEDED(hr), hr);
+    mAudioTimeOffset = media::TimeUnit::FromMicroseconds(timestampHns / 10);
+    mMustRecaptureAudioPosition = false;
+  }
+  // We can assume PCM 16 output.
+  int32_t numSamples = currentLength / 2;
+  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;
+  }
+
+  int16_t* pcm = (int16_t*)data;
+  for (int32_t i = 0; i < numSamples; ++i) {
+    audioData[i] = AudioSampleToFloat(pcm[i]);
+  }
+
+  buffer->Unlock();
+
+  media::TimeUnit timestamp =
+    mAudioTimeOffset + FramesToTimeUnit(mAudioFrameSum, mAudioRate);
+  NS_ENSURE_TRUE(timestamp.IsValid(), E_FAIL);
+
+  mAudioFrameSum += numFrames;
+
+  media::TimeUnit duration = FramesToTimeUnit(numFrames, mAudioRate);
+  NS_ENSURE_TRUE(duration.IsValid(), E_FAIL);
+
+  aOutData = new AudioData(aStreamOffset,
+                           timestamp.ToMicroseconds(),
+                           duration.ToMicroseconds(),
+                           numFrames,
+                           Move(audioData),
+                           mAudioChannels,
+                           mAudioRate);
+
+  #ifdef LOG_SAMPLE_DECODE
+  LOG("Decoded audio sample! timestamp=%lld duration=%lld currentLength=%u",
+      timestamp.ToMicroseconds(), duration.ToMicroseconds(), currentLength);
+  #endif
+
+  return S_OK;
+}
+
+void
+WMFAudioMFTManager::Shutdown()
+{
+  mDecoder = nullptr;
+}
+
+} // namespace mozilla