1 files changed, 29 insertions, 28 deletions

diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.h b/dom/media/webrtc/MediaEngineWebRTCAudio.h
index e71b5ef826..6b1fbf0089 100644
--- a/dom/media/webrtc/MediaEngineWebRTCAudio.h
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.h
@@ -91,8 +91,7 @@ class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
   // Current state of the resource for this source.
   MediaEngineSourceState mState;
 
-  // The current preferences that will be forwarded to mAudioProcessingConfig
-  // below.
+  // The current preferences that will be forwarded to mInputProcessing below.
   MediaEnginePrefs mCurrentPrefs;
 
   // The AudioProcessingTrack used to inteface with the MediaTrackGraph. Set in
@@ -101,10 +100,6 @@ class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
 
   // See note at the top of this class.
   RefPtr<AudioInputProcessing> mInputProcessing;
-
-  // Copy of the config currently applied to AudioProcessing through
-  // mInputProcessing.
-  webrtc::AudioProcessing::Config mAudioProcessingConfig;
 };
 
 // This class is created on the MediaManager thread, and then exclusively used
@@ -113,15 +108,16 @@ class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
 class AudioInputProcessing : public AudioDataListener {
  public:
   explicit AudioInputProcessing(uint32_t aMaxChannelCount);
-  void Process(MediaTrackGraph* aGraph, GraphTime aFrom, GraphTime aTo,
+  void Process(AudioProcessingTrack* aTrack, GraphTime aFrom, GraphTime aTo,
                AudioSegment* aInput, AudioSegment* aOutput);
 
-  void ProcessOutputData(MediaTrackGraph* aGraph, const AudioChunk& aChunk);
+  void ProcessOutputData(AudioProcessingTrack* aTrack,
+                         const AudioChunk& aChunk);
   bool IsVoiceInput(MediaTrackGraph* aGraph) const override {
     // If we're passing data directly without AEC or any other process, this
     // means that all voice-processing has been disabled intentionaly. In this
     // case, consider that the device is not used for voice input.
-    return !PassThrough(aGraph);
+    return !IsPassThrough(aGraph);
   }
 
   void Start(MediaTrackGraph* aGraph);
@@ -135,23 +131,20 @@ class AudioInputProcessing : public AudioDataListener {
 
   void Disconnect(MediaTrackGraph* aGraph) override;
 
-  void PacketizeAndProcess(MediaTrackGraph* aGraph,
+  void PacketizeAndProcess(AudioProcessingTrack* aTrack,
                            const AudioSegment& aSegment);
 
-  void SetPassThrough(MediaTrackGraph* aGraph, bool aPassThrough);
   uint32_t GetRequestedInputChannelCount();
-  void SetRequestedInputChannelCount(MediaTrackGraph* aGraph,
-                                     CubebUtils::AudioDeviceID aDeviceId,
-                                     uint32_t aRequestedInputChannelCount);
-  // This is true when all processing is disabled, we can skip
+  // This is true when all processing is disabled, in which case we can skip
   // packetization, resampling and other processing passes.
-  bool PassThrough(MediaTrackGraph* aGraph) const;
+  bool IsPassThrough(MediaTrackGraph* aGraph) const;
 
   // This allow changing the APM options, enabling or disabling processing
-  // steps. The config gets applied the next time we're about to process input
+  // steps. The settings get applied the next time we're about to process input
   // data.
-  void ApplyConfig(MediaTrackGraph* aGraph,
-                   const webrtc::AudioProcessing::Config& aConfig);
+  void ApplySettings(MediaTrackGraph* aGraph,
+                     CubebUtils::AudioDeviceID aDeviceID,
+                     const MediaEnginePrefs& aSettings);
 
   void End();
 
@@ -164,9 +157,18 @@ class AudioInputProcessing : public AudioDataListener {
 
   bool IsEnded() const { return mEnded; }
 
+  // For testing:
+  bool HadAECAndDrift() const { return mHadAECAndDrift; }
+
  private:
   ~AudioInputProcessing() = default;
-  void EnsureAudioProcessing(MediaTrackGraph* aGraph, uint32_t aChannels);
+  webrtc::AudioProcessing::Config ConfigForPrefs(
+      const MediaEnginePrefs& aPrefs);
+  void PassThroughChanged(MediaTrackGraph* aGraph);
+  void RequestedInputChannelCountChanged(MediaTrackGraph* aGraph,
+                                         CubebUtils::AudioDeviceID aDeviceId);
+  void EnsurePacketizer(AudioProcessingTrack* aTrack);
+  void EnsureAudioProcessing(AudioProcessingTrack* aTrack);
   void ResetAudioProcessing(MediaTrackGraph* aGraph);
   PrincipalHandle GetCheckedPrincipal(const AudioSegment& aSegment);
   // This implements the processing algoritm to apply to the input (e.g. a
@@ -174,17 +176,16 @@ class AudioInputProcessing : public AudioDataListener {
   // class only accepts audio chunks of 10ms. It has two inputs and one output:
   // it is fed the speaker data and the microphone data. It outputs processed
   // input data.
-  const UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
+  UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
+  // Whether mAudioProcessing was created for AEC with clock drift.
+  // Meaningful only when mAudioProcessing is non-null;
+  bool mHadAECAndDrift = false;
   // Packetizer to be able to feed 10ms packets to the input side of
   // mAudioProcessing. Not used if the processing is bypassed.
   Maybe<AudioPacketizer<AudioDataValue, float>> mPacketizerInput;
-  // The number of channels asked for by content, after clamping to the range of
-  // legal channel count for this particular device.
-  uint32_t mRequestedInputChannelCount;
-  // mSkipProcessing is true if none of the processing passes are enabled,
-  // because of prefs or constraints. This allows simply copying the audio into
-  // the MTG, skipping resampling and the whole webrtc.org code.
-  bool mSkipProcessing;
+  // The current settings from about:config preferences and content-provided
+  // constraints.
+  MediaEnginePrefs mSettings;
   // Buffer for up to one 10ms packet of planar mixed audio output for the
   // reverse-stream (speaker data) of mAudioProcessing AEC.
   // Length is packet size * channel count, regardless of how many frames are