Bug 830707. Part 3: Don't constrain AudioSegment to a fixed number of channels. r=jesup

--HG--
extra : rebase_source : feacede00821b6673ce04c886a9c3727a4989404

content/media/AudioSegment.h

@@ -87,30 +87,12 @@ class AudioSegment : public MediaSegmentBase<AudioSegment, AudioChunk> {
 public:
   typedef mozilla::AudioSampleFormat SampleFormat;
 
-  AudioSegment() : MediaSegmentBase<AudioSegment, AudioChunk>(AUDIO), mChannels(0) {}
+  AudioSegment() : MediaSegmentBase<AudioSegment, AudioChunk>(AUDIO) {}
 
-  bool IsInitialized()
-  {
-    return mChannels > 0;
-  }
-  void Init(int32_t aChannels)
-  {
-    NS_ASSERTION(aChannels > 0, "Bad number of channels");
-    NS_ASSERTION(!IsInitialized(), "Already initialized");
-    mChannels = aChannels;
-  }
-  int32_t GetChannels()
-  {
-    NS_ASSERTION(IsInitialized(), "Not initialized");
-    return mChannels;
-  }
   void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
                     const nsTArray<const float*>& aChannelData,
                     int32_t aDuration)
   {
-    NS_ASSERTION(mChannels > 0, "Not initialized");
-    NS_ASSERTION(!aBuffer.get() || aChannelData.Length() == uint32_t(mChannels),
-                 "Wrong number of channels");
     AudioChunk* chunk = AppendChunk(aDuration);
     chunk->mBuffer = aBuffer;
     for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
@@ -123,9 +105,6 @@ public:
                     const nsTArray<const int16_t*>& aChannelData,
                     int32_t aDuration)
   {
-    NS_ASSERTION(mChannels > 0, "Not initialized");
-    NS_ASSERTION(!aBuffer.get() || aChannelData.Length() == uint32_t(mChannels),
-                 "Wrong number of channels");
     AudioChunk* chunk = AppendChunk(aDuration);
     chunk->mBuffer = aBuffer;
     for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
@@ -134,27 +113,21 @@ public:
     chunk->mVolume = 1.0f;
     chunk->mBufferFormat = AUDIO_FORMAT_S16;
   }
+  // Consumes aChunk, and returns a pointer to the persistent copy of aChunk
+  // in the segment.
+  AudioChunk* AppendAndConsumeChunk(AudioChunk* aChunk)
+  {
+    AudioChunk* chunk = AppendChunk(aChunk->mDuration);
+    chunk->mBuffer = aChunk->mBuffer.forget();
+    chunk->mChannelData.SwapElements(aChunk->mChannelData);
+    chunk->mVolume = aChunk->mVolume;
+    chunk->mBufferFormat = aChunk->mBufferFormat;
+    return chunk;
+  }
   void ApplyVolume(float aVolume);
-  /**
-   * aOutput must have a matching number of channels, but we will automatically
-   * convert sample formats.
-   */
   void WriteTo(AudioStream* aOutput);
 
-  // Segment-generic methods not in MediaSegmentBase
-  void InitFrom(const AudioSegment& aOther)
-  {
-    NS_ASSERTION(mChannels == 0, "Channels already set");
-    mChannels = aOther.mChannels;
-  }
-  void CheckCompatible(const AudioSegment& aOther) const
-  {
-    NS_ASSERTION(aOther.mChannels == mChannels, "Non-matching channels");
-  }
   static Type StaticType() { return AUDIO; }
-
-protected:
-  int32_t mChannels;
 };
 
 }

content/media/MediaDecoderStateMachine.cpp

@@ -517,9 +517,6 @@ void MediaDecoderStateMachine::SendStreamAudio(AudioData* aAudio,
   aStream->mLastAudioPacketTime = aAudio->mTime;
   aStream->mLastAudioPacketEndTime = aAudio->GetEnd();
 
-  NS_ASSERTION(aOutput->GetChannels() == int32_t(aAudio->mChannels),
-               "Wrong number of channels");
-
   // This logic has to mimic AudioLoop closely to make sure we write
   // the exact same silences
   CheckedInt64 audioWrittenOffset = UsecsToFrames(mInfo.mAudioRate,
@@ -532,7 +529,6 @@ void MediaDecoderStateMachine::SendStreamAudio(AudioData* aAudio,
     LOG(PR_LOG_DEBUG, ("%p Decoder writing %d frames of silence to MediaStream",
                        mDecoder.get(), int32_t(frameOffset.value() - audioWrittenOffset.value())));
     AudioSegment silence;
-    silence.InitFrom(*aOutput);
     silence.InsertNullDataAtStart(frameOffset.value() - audioWrittenOffset.value());
     aStream->mAudioFramesWritten += silence.GetDuration();
     aOutput->AppendFrom(&silence);
@@ -604,7 +600,6 @@ void MediaDecoderStateMachine::SendStreamData()
   if (!stream->mStreamInitialized) {
     if (mInfo.mHasAudio) {
       AudioSegment* audio = new AudioSegment();
-      audio->Init(mInfo.mAudioChannels);
       mediaStream->AddTrack(TRACK_AUDIO, mInfo.mAudioRate, 0, audio);
     }
     if (mInfo.mHasVideo) {
@@ -620,7 +615,6 @@ void MediaDecoderStateMachine::SendStreamData()
     // is captured, only the decoder thread pops from the queue (see below).
     mReader->AudioQueue().GetElementsAfter(stream->mLastAudioPacketTime, &audio);
     AudioSegment output;
-    output.Init(mInfo.mAudioChannels);
     for (uint32_t i = 0; i < audio.Length(); ++i) {
       SendStreamAudio(audio[i], stream, &output);
     }

content/media/MediaSegment.h

@@ -119,9 +119,7 @@ template <class C, class Chunk> class MediaSegmentBase : public MediaSegment {
 public:
   virtual MediaSegment* CreateEmptyClone() const
   {
-    C* s = new C();
-    s->InitFrom(*static_cast<const C*>(this));
-    return s;
+    return new C();
   }
   virtual void AppendFrom(MediaSegment* aSource)
   {
@@ -142,11 +140,6 @@ public:
   {
     AppendSliceInternal(aOther, aStart, aEnd);
   }
-  void InitToSlice(const C& aOther, TrackTicks aStart, TrackTicks aEnd)
-  {
-    static_cast<C*>(this)->InitFrom(aOther);
-    AppendSliceInternal(aOther, aStart, aEnd);
-  }
   /**
    * Replace the first aDuration ticks with null media data, because the data
    * will not be required again.
@@ -215,7 +208,6 @@ protected:
    */
   void AppendFromInternal(MediaSegmentBase<C, Chunk>* aSource)
   {
-    static_cast<C*>(this)->CheckCompatible(*static_cast<C*>(aSource));
     MOZ_ASSERT(aSource->mDuration >= 0);
     mDuration += aSource->mDuration;
     aSource->mDuration = 0;
@@ -230,7 +222,6 @@ protected:
   void AppendSliceInternal(const MediaSegmentBase<C, Chunk>& aSource,
                            TrackTicks aStart, TrackTicks aEnd)
   {
-    static_cast<C*>(this)->CheckCompatible(static_cast<const C&>(aSource));
     NS_ASSERTION(aStart <= aEnd, "Endpoints inverted");
     NS_ASSERTION(aStart >= 0 && aEnd <= aSource.mDuration,
                  "Slice out of range");

content/media/MediaStreamGraph.cpp

@@ -1158,14 +1158,14 @@ MediaStreamGraphImpl::CreateOrDestroyAudioStreams(GraphTime aAudioOutputStartTim
         // XXX allocating a AudioStream could be slow so we're going to have to do
         // something here ... preallocation, async allocation, multiplexing onto a single
         // stream ...
-        AudioSegment* audio = tracks->Get<AudioSegment>();
         MediaStream::AudioOutputStream* audioOutputStream =
           aStream->mAudioOutputStreams.AppendElement();
         audioOutputStream->mAudioPlaybackStartTime = aAudioOutputStartTime;
         audioOutputStream->mBlockedAudioTime = 0;
         audioOutputStream->mStream = AudioStream::AllocateStream();
-        audioOutputStream->mStream->Init(audio->GetChannels(),
-                                         tracks->GetRate(), AUDIO_CHANNEL_NORMAL);
+        // XXX for now, allocate stereo output. But we need to fix this to
+        // match the system's ideal channel configuration.
+        audioOutputStream->mStream->Init(2, tracks->GetRate(), AUDIO_CHANNEL_NORMAL);
         audioOutputStream->mTrackID = tracks->GetID();
       }
     }
@@ -1211,7 +1211,6 @@ MediaStreamGraphImpl::PlayAudio(MediaStream* aStream,
       end = std::min(end, aTo);
 
       AudioSegment output;
-      output.InitFrom(*audio);
       if (blocked) {
         // Track total blocked time in aStream->mBlockedAudioTime so that
         // the amount of silent samples we've inserted for blocking never gets

content/media/VideoSegment.h

@@ -104,12 +104,6 @@ public:
   }
 
   // Segment-generic methods not in MediaSegmentBase
-  void InitFrom(const VideoSegment& aOther)
-  {
-  }
-  void CheckCompatible(const VideoSegment& aOther) const
-  {
-  }
   static Type StaticType() { return VIDEO; }
 };
 

content/media/webrtc/MediaEngineDefault.cpp

@@ -17,7 +17,6 @@
 #include "nsISupportsUtils.h"
 #endif
 
-#define CHANNELS 1
 #define VIDEO_RATE USECS_PER_S
 #define AUDIO_RATE 16000
 
@@ -336,7 +335,6 @@ MediaEngineDefaultAudioSource::Start(SourceMediaStream* aStream, TrackID aID)
 
   // AddTrack will take ownership of segment
   AudioSegment* segment = new AudioSegment();
-  segment->Init(CHANNELS);
   mSource->AddTrack(aID, AUDIO_RATE, 0, segment);
 
   // We aren't going to add any more tracks
@@ -382,7 +380,6 @@ NS_IMETHODIMP
 MediaEngineDefaultAudioSource::Notify(nsITimer* aTimer)
 {
   AudioSegment segment;
-  segment.Init(CHANNELS);
   segment.InsertNullDataAtStart(AUDIO_RATE/100); // 10ms of fake data
 
   mSource->AppendToTrack(mTrackID, &segment);

content/media/webrtc/MediaEngineWebRTCAudio.cpp

@@ -149,7 +149,6 @@ MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID)
   }
 
   AudioSegment* segment = new AudioSegment();
-  segment->Init(CHANNELS);
   aStream->AddTrack(aID, SAMPLE_FREQUENCY, 0, segment);
   aStream->AdvanceKnownTracksTime(STREAM_TIME_MAX);
   LOG(("Initial audio"));
@@ -363,7 +362,6 @@ MediaEngineWebRTCAudioSource::Process(const int channel,
     memcpy(dest, audio10ms, length * sizeof(sample));
 
     AudioSegment segment;
-    segment.Init(CHANNELS);
     nsAutoTArray<const sample*,1> channels;
     channels.AppendElement(dest);
     segment.AppendFrames(buffer.forget(), channels, length);

media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp

@@ -842,7 +842,6 @@ MediaPipelineReceiveAudio::PipelineListener::PipelineListener(
       conduit_(conduit),
       played_(0) {
   mozilla::AudioSegment *segment = new mozilla::AudioSegment();
-  segment->Init(1); // 1 Channel
   source_->AddTrack(track_id_, 16000, 0, segment);
   source_->AdvanceKnownTracksTime(STREAM_TIME_MAX);
 }
@@ -875,7 +874,6 @@ NotifyPull(MediaStreamGraph* graph, StreamTime desired_time) {
     MOZ_MTLOG(PR_LOG_DEBUG, "Audio conduit returned buffer of length " << samples_length);
 
     AudioSegment segment;
-    segment.Init(1);
     nsAutoTArray<const int16_t*,1> channels;
     channels.AppendElement(samples_data);
     segment.AppendFrames(samples.forget(), channels, samples_length);

media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.h

@@ -53,7 +53,6 @@ Fake_AudioGenerator(nsDOMMediaStream* aStream) : mStream(aStream), mCount(0) {
 
     // Make a track
     mozilla::AudioSegment *segment = new mozilla::AudioSegment();
-    segment->Init(1); // 1 Channel
     mStream->GetStream()->AsSourceStream()->AddTrack(1, 16000, 0, segment);
 
     // Set the timer
@@ -71,7 +70,6 @@ Fake_AudioGenerator(nsDOMMediaStream* aStream) : mStream(aStream), mCount(0) {
     }
 
     mozilla::AudioSegment segment;
-    segment.Init(1);
     nsAutoTArray<const int16_t*,1> channelData;
     channelData.AppendElement(data);
     segment.AppendFrames(samples.forget(), channelData, 1600);

media/webrtc/signaling/test/FakeMediaStreamsImpl.h

@@ -99,7 +99,6 @@ void Fake_AudioStreamSource::Periodic() {
   }
 
   mozilla::AudioSegment segment;
-  segment.Init(1);
   nsAutoTArray<const int16_t *,1> channels;
   channels.AppendElement(data);
   segment.AppendFrames(samples.forget(), channels, AUDIO_BUFFER_SIZE);

media/webrtc/signaling/test/mediapipeline_unittest.cpp

@@ -166,7 +166,6 @@ class TestAgentReceive : public TestAgent {
     audio->SetPullEnabled(true);
 
     mozilla::AudioSegment* segment= new mozilla::AudioSegment();
-    segment->Init(1);
     audio->AddTrack(0, 100, 0, segment);
     audio->AdvanceKnownTracksTime(mozilla::STREAM_TIME_MAX);