Backed out changeset b83be2d0614b (bug 1264199)

--HG--
extra : rebase_source : f8962f950f2fa089ca7eb297aca3b8711c5586c5

dom/media/AudioConverter.cpp

@@ -33,7 +33,19 @@ AudioConverter::AudioConverter(const AudioConfig& aIn, const AudioConfig& aOut)
   MOZ_DIAGNOSTIC_ASSERT(aOut.Interleaved(), "planar audio format not supported");
   mIn.Layout().MappingTable(mOut.Layout(), mChannelOrderMap);
   if (aIn.Rate() != aOut.Rate()) {
-    RecreateResampler();
+    int error;
+    mResampler = speex_resampler_init(aOut.Channels(),
+                                      aIn.Rate(),
+                                      aOut.Rate(),
+                                      SPEEX_RESAMPLER_QUALITY_DEFAULT,
+                                      &error);
+
+    if (error == RESAMPLER_ERR_SUCCESS) {
+      speex_resampler_skip_zeros(mResampler);
+    } else {
+      NS_WARNING("Failed to initialize resampler.");
+      mResampler = nullptr;
+    }
   }
 }
 
@@ -270,46 +282,6 @@ AudioConverter::ResampleAudio(void* aOut, const void* aIn, size_t aFrames)
   return outframes;
 }
 
-void
-AudioConverter::RecreateResampler()
-{
-  if (mResampler) {
-    speex_resampler_destroy(mResampler);
-  }
-  int error;
-  mResampler = speex_resampler_init(mOut.Channels(),
-                                    mIn.Rate(),
-                                    mOut.Rate(),
-                                    SPEEX_RESAMPLER_QUALITY_DEFAULT,
-                                    &error);
-
-  if (error == RESAMPLER_ERR_SUCCESS) {
-    speex_resampler_skip_zeros(mResampler);
-  } else {
-    NS_WARNING("Failed to initialize resampler.");
-    mResampler = nullptr;
-  }
-}
-
-size_t
-AudioConverter::DrainResampler(void* aOut)
-{
-  if (!mResampler) {
-    return 0;
-  }
-  int frames = speex_resampler_get_input_latency(mResampler);
-  AlignedByteBuffer buffer(FramesOutToSamples(frames) *
-                           AudioConfig::SampleSize(mOut.Format()));
-  if (!buffer) {
-    // OOM
-    return 0;
-  }
-  frames = ResampleAudio(aOut, buffer.Data(), frames);
-  // Tore down the resampler as it's easier than handling follow-up.
-  RecreateResampler();
-  return frames;
-}
-
 size_t
 AudioConverter::UpmixAudio(void* aOut, const void* aIn, size_t aFrames) const
 {
@@ -355,13 +327,7 @@ AudioConverter::UpmixAudio(void* aOut, const void* aIn, size_t aFrames) const
 size_t
 AudioConverter::ResampleRecipientFrames(size_t aFrames) const
 {
-  if (!aFrames && mIn.Rate() != mOut.Rate()) {
-    // The resampler will be drained, account for frames currently buffered
-    // in the resampler.
-    return speex_resampler_get_output_latency(mResampler);
-  } else {
-    return (uint64_t)aFrames * mOut.Rate() / mIn.Rate() + 1;
-  }
+  return (uint64_t)aFrames * mOut.Rate() / mIn.Rate() + 1;
 }
 
 size_t

dom/media/AudioConverter.h

@@ -123,8 +123,6 @@ public:
   // Convert the AudioDataBuffer.
   // Conversion will be done in place if possible. Otherwise a new buffer will
   // be returned.
-  // Providing an empty buffer and resampling is expected, the resampler
-  // will be drained.
   template <AudioConfig::SampleFormat Format, typename Value>
   AudioDataBuffer<Format, Value> Process(AudioDataBuffer<Format, Value>&& aBuffer)
   {
@@ -154,7 +152,7 @@ public:
       return AudioDataBuffer<Format, Value>(Move(temp1));
     }
     frames = ProcessInternal(temp1.Data(), aBuffer.Data(), frames);
-    if (mIn.Rate() == mOut.Rate()) {
+    if (!frames || mIn.Rate() == mOut.Rate()) {
       temp1.SetLength(FramesOutToSamples(frames));
       return AudioDataBuffer<Format, Value>(Move(temp1));
     }
@@ -163,17 +161,13 @@ public:
     // If we are downsampling we can re-use it.
     AlignedBuffer<Value>* outputBuffer = &temp1;
     AlignedBuffer<Value> temp2;
-    if (!frames || mOut.Rate() > mIn.Rate()) {
-      // We are upsampling or about to drain, we can't work in place.
-      // Allocate another temporary buffer where the upsampling will occur.
+    if (mOut.Rate() > mIn.Rate()) {
+      // We are upsampling, we can't work in place. Allocate another temporary
+      // buffer where the upsampling will occur.
       temp2.SetLength(FramesOutToSamples(ResampleRecipientFrames(frames)));
       outputBuffer = &temp2;
     }
-    if (!frames) {
-      frames = DrainResampler(outputBuffer->Data());
-    } else {
-      frames = ResampleAudio(outputBuffer->Data(), temp1.Data(), frames);
-    }
+    frames = ResampleAudio(outputBuffer->Data(), temp1.Data(), frames);
     outputBuffer->SetLength(FramesOutToSamples(frames));
     return AudioDataBuffer<Format, Value>(Move(*outputBuffer));
   }
@@ -229,8 +223,6 @@ private:
   SpeexResamplerState* mResampler;
   size_t ResampleAudio(void* aOut, const void* aIn, size_t aFrames);
   size_t ResampleRecipientFrames(size_t aFrames) const;
-  void RecreateResampler();
-  size_t DrainResampler(void* aOut);
 };
 
 } // namespace mozilla

dom/media/mediasink/DecodedAudioDataSink.cpp

@@ -323,12 +323,6 @@ DecodedAudioDataSink::NotifyAudioNeeded()
   MOZ_ASSERT(mOwnerThread->IsCurrentThreadIn(),
              "Not called from the owner's thread");
 
-  if (AudioQueue().IsFinished() && !AudioQueue().GetSize()) {
-    // We have reached the end of the data, drain the resampler.
-    DrainConverter();
-    return;
-  }
-
   // Always ensure we have two processed frames pending to allow for processing
   // latency.
   while (AudioQueue().GetSize() && (mProcessedQueueLength < LOW_AUDIO_USECS ||
@@ -349,8 +343,6 @@ DecodedAudioDataSink::NotifyAudioNeeded()
                  mConverter ? mConverter->InputConfig().Rate() : 0,
                  data->mChannels, data->mRate);
 
-      DrainConverter();
-
       // mFramesParsed indicates the current playtime in frames at the current
       // input sampling rate. Recalculate it per the new sampling rate.
       if (mFramesParsed) {
@@ -397,19 +389,14 @@ DecodedAudioDataSink::NotifyAudioNeeded()
       // time.
       missingFrames = std::min<int64_t>(INT32_MAX, missingFrames.value());
       mFramesParsed += missingFrames.value();
-      // We need to insert silence, first use drained frames if any.
-      missingFrames -= DrainConverter(missingFrames.value());
-      // Insert silence is still needed.
-      if (missingFrames.value()) {
-        AlignedAudioBuffer silenceData(missingFrames.value() * mOutputChannels);
-        if (!silenceData) {
-          NS_WARNING("OOM in DecodedAudioDataSink");
-          mErrored = true;
-          return;
-        }
-        RefPtr<AudioData> silence = CreateAudioFromBuffer(Move(silenceData), data);
-        PushProcessedAudio(silence);
+      AlignedAudioBuffer silenceData(missingFrames.value() * mOutputChannels);
+      if (!silenceData) {
+        NS_WARNING("OOM in DecodedAudioDataSink");
+        mErrored = true;
+        return;
       }
+      RefPtr<AudioData> silence = CreateAudioFromBuffer(Move(silenceData), data);
+      PushProcessedAudio(silence);
     }
 
     mLastEndTime = data->GetEndTime();
@@ -420,9 +407,7 @@ DecodedAudioDataSink::NotifyAudioNeeded()
         mConverter->Process(AudioSampleBuffer(Move(data->mAudioData))).Forget();
       data = CreateAudioFromBuffer(Move(convertedData), data);
     }
-    if (PushProcessedAudio(data)) {
-      mLastProcessedPacket = Some(data);
-    }
+    PushProcessedAudio(data);
   }
 }
 
@@ -462,38 +447,5 @@ DecodedAudioDataSink::CreateAudioFromBuffer(AlignedAudioBuffer&& aBuffer,
   return data.forget();
 }
 
-uint32_t
-DecodedAudioDataSink::DrainConverter(uint32_t aMaxFrames)
-{
-  MOZ_ASSERT(mOwnerThread->IsCurrentThreadIn());
-
-  if (!mConverter || !mLastProcessedPacket) {
-    // nothing to drain.
-    return 0;
-  }
-
-  RefPtr<AudioData> lastPacket = mLastProcessedPacket.ref();
-  mLastProcessedPacket.reset();
-
-  // To drain we simply provide an empty packet to the audio converter.
-  AlignedAudioBuffer convertedData =
-    mConverter->Process(AudioSampleBuffer(AlignedAudioBuffer())).Forget();
-
-  uint32_t frames = convertedData.Length() / mOutputChannels;
-  if (!convertedData.SetLength(std::min(frames, aMaxFrames) * mOutputChannels)) {
-    // This can never happen as we were reducing the length of convertData.
-    mErrored = true;
-    return 0;
-  }
-
-  RefPtr<AudioData> data =
-    CreateAudioFromBuffer(Move(convertedData), lastPacket);
-  if (!data) {
-    return 0;
-  }
-  mProcessedQueue.Push(data);
-  return data->mFrames;
-}
-
 } // namespace media
 } // namespace mozilla

dom/media/mediasink/DecodedAudioDataSink.h

@@ -114,9 +114,6 @@ private:
   void OnAudioPopped(const RefPtr<MediaData>& aSample);
   void OnAudioPushed(const RefPtr<MediaData>& aSample);
   void NotifyAudioNeeded();
-  // Drain the converter and add the output to the processed audio queue.
-  // A maximum of aMaxFrames will be added.
-  uint32_t DrainConverter(uint32_t aMaxFrames = UINT32_MAX);
   already_AddRefed<AudioData> CreateAudioFromBuffer(AlignedAudioBuffer&& aBuffer,
                                                     AudioData* aReference);
   // Add data to the processsed queue, update mProcessedQueueLength and
@@ -132,7 +129,6 @@ private:
   // the input stream. It indicates the time in frames since playback started
   // at the current input framerate.
   int64_t mFramesParsed;
-  Maybe<RefPtr<AudioData>> mLastProcessedPacket;
   int64_t mLastEndTime;
   // Never modifed after construction.
   uint32_t mOutputRate;