Set audio format before adding pepper sink to audio track.

Some audio input devices don't generate caputure data in a continuous basis, and
instead only generate when some event happens. This may be an indefinite amount
of time. However, |AudioSink::OnSetFormat| is only called whenever new samples
come in. For pepper plugins, this means that |MediaStreamAudioTrack::Configure|
can be blocked indefintely on these devices. To work around this, get the
format before attaching to the audio track.

BUG=403138

Review URL: https://codereview.chromium.org/857093002

Cr-Commit-Position: refs/heads/master@{#313458}

content/public/renderer/media_stream_audio_sink.cc

@@ -43,4 +43,17 @@ void MediaStreamAudioSink::RemoveFromAudioTrack(
   audio_track->RemoveSink(sink);
 }
 
+media::AudioParameters MediaStreamAudioSink::GetFormatFromAudioTrack(
+    const blink::WebMediaStreamTrack& track) {
+  MediaStreamTrack* native_track = MediaStreamTrack::GetTrack(track);
+  if (!native_track->is_local_track()) {
+    LOG(ERROR) << "Can't get format from a remote audio track";
+    return media::AudioParameters();
+  }
+
+  WebRtcLocalAudioTrack* audio_track =
+      static_cast<WebRtcLocalAudioTrack*>(native_track);
+  return audio_track->GetOutputFormat();
+}
+
 }  // namespace content

content/public/renderer/media_stream_audio_sink.h

@@ -37,6 +37,11 @@ class CONTENT_EXPORT MediaStreamAudioSink : public MediaStreamSink {
   static void RemoveFromAudioTrack(MediaStreamAudioSink* sink,
                                    const blink::WebMediaStreamTrack& track);
 
+  // Returns the format of the audio track.
+  // Called on the main render thread.
+  static media::AudioParameters GetFormatFromAudioTrack(
+      const blink::WebMediaStreamTrack& track);
+
   // Callback called to deliver audio data. The data in |audio_bus| respects the
   // AudioParameters passed in the last call to OnSetFormat().  Called on
   // real-time audio thread.

content/renderer/media/webrtc_audio_capturer.cc

@@ -434,6 +434,11 @@ int WebRtcAudioCapturer::MaxVolume() const {
   return WebRtcAudioDeviceImpl::kMaxVolumeLevel;
 }
 
+media::AudioParameters WebRtcAudioCapturer::GetOutputFormat() const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return audio_processor_->OutputFormat();
+}
+
 void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source,
                                   int audio_delay_milliseconds,
                                   double volume,

content/renderer/media/webrtc_audio_capturer.h

@@ -105,6 +105,10 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // call Stop()
   void Stop();
 
+  // Returns the output format.
+  // Called on the main render thread.
+  media::AudioParameters GetOutputFormat() const;
+
   // Used by the unittests to inject their own source to the capturer.
   void SetCapturerSourceForTesting(
       const scoped_refptr<media::AudioCapturerSource>& source,

content/renderer/media/webrtc_local_audio_track.cc

@@ -40,6 +40,15 @@ WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
   Stop();
 }
 
+media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const {
+  DCHECK(main_render_thread_checker_.CalledOnValidThread());
+  if (webaudio_source_.get()) {
+    return media::AudioParameters();
+  } else {
+    return capturer_->GetOutputFormat();
+  }
+}
+
 void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
                                     base::TimeTicks estimated_capture_time,
                                     bool force_report_nonzero_energy) {

content/renderer/media/webrtc_local_audio_track.h

@@ -67,6 +67,11 @@ class CONTENT_EXPORT WebRtcLocalAudioTrack
 
   webrtc::AudioTrackInterface* GetAudioAdapter() override;
 
+  // Returns the output format of the capture source. May return an invalid
+  // AudioParameters if the format is not yet available.
+  // Called on the main render thread.
+  media::AudioParameters GetOutputFormat() const;
+
   // Method called by the capturer to deliver the capture data.
   // Called on the capture audio thread.
   void Capture(const media::AudioBus& audio_bus,

content/renderer/pepper/pepper_media_stream_audio_track_host.cc

@@ -364,6 +364,13 @@ void PepperMediaStreamAudioTrackHost::OnNewBufferEnqueued() {
 
 void PepperMediaStreamAudioTrackHost::DidConnectPendingHostToResource() {
   if (!connected_) {
+    media::AudioParameters format =
+        MediaStreamAudioSink::GetFormatFromAudioTrack(track_);
+    // Although this should only be called on the audio capture thread, that
+    // can't happen until the sink is added to the audio track below.
+    if (format.IsValid())
+      audio_sink_.OnSetFormat(format);
+
     MediaStreamAudioSink::AddToAudioTrack(&audio_sink_, track_);
     connected_ = true;
   }

content/renderer/pepper/pepper_media_stream_audio_track_host.h

@@ -52,6 +52,16 @@ class PepperMediaStreamAudioTrackHost : public PepperMediaStreamTrackHostBase {
     // Send a reply to the currently pending |Configure()| request.
     void SendConfigureReply(int32_t result);
 
+    // MediaStreamAudioSink overrides:
+    // These two functions should be called on the audio thread.
+    // NOTE: For this specific instance, |OnSetFormat()| is also called on the
+    // main thread. However, the call to |OnSetFormat()| happens before this
+    // sink is added to an audio track, also on the main thread, which should
+    // avoid any potential races.
+    void OnData(const media::AudioBus& audio_bus,
+                base::TimeTicks estimated_capture_time) override;
+    void OnSetFormat(const media::AudioParameters& params) override;
+
    private:
     // Initializes buffers on the main thread.
     void SetFormatOnMainThread(int bytes_per_second, int bytes_per_frame);
@@ -62,12 +72,6 @@ class PepperMediaStreamAudioTrackHost : public PepperMediaStreamTrackHostBase {
     void SendEnqueueBufferMessageOnMainThread(int32_t index,
                                               int32_t buffers_generation);
 
-    // MediaStreamAudioSink overrides:
-    // These two functions should be called on the audio thread.
-    void OnData(const media::AudioBus& audio_bus,
-                base::TimeTicks estimated_capture_time) override;
-    void OnSetFormat(const media::AudioParameters& params) override;
-
     // Unowned host which is available during the AudioSink's lifespan.
     // It is mainly used in the main thread. But the audio thread will use
     // host_->buffer_manager() to read some buffer properties. It is safe