Fix MediaStreamAudioSourceNode custom sampleRate

Now that the AudioContext supports the optional sampleRate parameter
MediaStreamTrack data needs to be resampled to this rate and not the
hardware rate.

Bug: 945305
Change-Id: If34b72f0aee864148aa44531e87e3816ae05a799
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1541078Reviewed-by: Raymond Toy <rtoy@chromium.org>
Reviewed-by: Guido Urdaneta <guidou@chromium.org>
Reviewed-by: Daniel Cheng <dcheng@chromium.org>
Commit-Queue: Andrew MacPherson <andrew.macpherson@soundtrap.com>
Cr-Commit-Position: refs/heads/master@{#646090}

content/renderer/media/stream/media_stream_center.cc

@@ -180,7 +180,8 @@ void MediaStreamCenter::DidDisableMediaStreamTrack(
 
 blink::WebAudioSourceProvider*
 MediaStreamCenter::CreateWebAudioSourceFromMediaStreamTrack(
-    const blink::WebMediaStreamTrack& track) {
+    const blink::WebMediaStreamTrack& track,
+    int context_sample_rate) {
   DVLOG(1) << "MediaStreamCenter::createWebAudioSourceFromMediaStreamTrack";
   blink::WebPlatformMediaStreamTrack* media_stream_track =
       track.GetPlatformTrack();
@@ -195,7 +196,7 @@ MediaStreamCenter::CreateWebAudioSourceFromMediaStreamTrack(
   // TODO(tommi): Rename WebRtcLocalAudioSourceProvider to
   // WebAudioMediaStreamSink since it's not specific to any particular source.
   // https://crbug.com/577874
-  return new WebRtcLocalAudioSourceProvider(track);
+  return new WebRtcLocalAudioSourceProvider(track, context_sample_rate);
 }
 
 void MediaStreamCenter::DidStopMediaStreamSource(

content/renderer/media/stream/media_stream_center.h

@@ -40,7 +40,8 @@ class CONTENT_EXPORT MediaStreamCenter : public blink::WebMediaStreamCenter {
       const blink::WebMediaStreamTrack& track) override;
 
   blink::WebAudioSourceProvider* CreateWebAudioSourceFromMediaStreamTrack(
-      const blink::WebMediaStreamTrack& track) override;
+      const blink::WebMediaStreamTrack& track,
+      int context_sample_rate) override;
 
   void DidStopMediaStreamSource(
       const blink::WebMediaStreamSource& web_source) override;

content/renderer/media/webrtc_local_audio_source_provider.cc

@@ -28,10 +28,9 @@ namespace content {
 const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128;
 
 WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
-    const blink::WebMediaStreamTrack& track)
-    : is_enabled_(false),
-      track_(track),
-      track_stopped_(false) {
+    const blink::WebMediaStreamTrack& track,
+    int context_sample_rate)
+    : is_enabled_(false), track_(track), track_stopped_(false) {
   // Get the native audio output hardware sample-rate for the sink.
   // We need to check if there is a valid frame since the unittests
   // do not have one and they will inject their own |sink_params_| for testing.
@@ -39,13 +38,8 @@ WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
       blink::WebLocalFrame::FrameForCurrentContext();
   RenderFrame* const render_frame = RenderFrame::FromWebFrame(web_frame);
   if (render_frame) {
-    int sample_rate =
-        AudioDeviceFactory::GetOutputDeviceInfo(render_frame->GetRoutingID(),
-                                                media::AudioSinkParameters())
-            .output_params()
-            .sample_rate();
     sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                       media::CHANNEL_LAYOUT_STEREO, sample_rate,
+                       media::CHANNEL_LAYOUT_STEREO, context_sample_rate,
                        kWebAudioRenderBufferSize);
   }
   // Connect the source provider to the track as a sink.

content/renderer/media/webrtc_local_audio_source_provider.h

@@ -57,7 +57,8 @@ class CONTENT_EXPORT WebRtcLocalAudioSourceProvider
   static const size_t kWebAudioRenderBufferSize;
 
   explicit WebRtcLocalAudioSourceProvider(
-      const blink::WebMediaStreamTrack& track);
+      const blink::WebMediaStreamTrack& track,
+      int context_sample_rate);
   ~WebRtcLocalAudioSourceProvider() override;
 
   // blink::WebMediaStreamAudioSink implementation.

content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

@@ -21,9 +21,10 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test {
   void SetUp() override {
     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                          media::CHANNEL_LAYOUT_MONO, 48000, 480);
+    const int context_sample_rate = 44100;
     sink_params_.Reset(
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-        media::CHANNEL_LAYOUT_STEREO, 44100,
+        media::CHANNEL_LAYOUT_STEREO, context_sample_rate,
         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
     sink_bus_ = media::AudioBus::Create(sink_params_);
     blink::WebMediaStreamSource audio_source;
@@ -35,7 +36,8 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test {
                             audio_source);
     blink_track_.SetPlatformTrack(
         std::make_unique<blink::MediaStreamAudioTrack>(true));
-    source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
+    source_provider_.reset(
+        new WebRtcLocalAudioSourceProvider(blink_track_, context_sample_rate));
     source_provider_->SetSinkParamsForTesting(sink_params_);
     source_provider_->OnSetFormat(source_params_);
   }

third_party/blink/public/platform/web_media_stream_center.h

@@ -59,7 +59,8 @@ class WebMediaStreamCenter {
   // Caller must take the ownership of the returned |WebAudioSourceProvider|
   // object.
   virtual WebAudioSourceProvider* CreateWebAudioSourceFromMediaStreamTrack(
-      const WebMediaStreamTrack&) {
+      const WebMediaStreamTrack&,
+      int context_sample_rate) {
     return nullptr;
   }
 };

third_party/blink/renderer/modules/mediastream/media_stream_track.cc

@@ -696,9 +696,10 @@ bool MediaStreamTrack::HasPendingActivity() const {
   return !Ended() && HasEventListeners(event_type_names::kEnded);
 }
 
-std::unique_ptr<AudioSourceProvider> MediaStreamTrack::CreateWebAudioSource() {
+std::unique_ptr<AudioSourceProvider> MediaStreamTrack::CreateWebAudioSource(
+    int context_sample_rate) {
   return MediaStreamCenter::Instance().CreateWebAudioSourceFromMediaStreamTrack(
-      Component());
+      Component(), context_sample_rate);
 }
 
 void MediaStreamTrack::RegisterMediaStream(MediaStream* media_stream) {

third_party/blink/renderer/modules/mediastream/media_stream_track.h

@@ -111,7 +111,8 @@ class MODULES_EXPORT MediaStreamTrack
   // ContextLifecycleObserver
   void ContextDestroyed(ExecutionContext*) override;
 
-  std::unique_ptr<AudioSourceProvider> CreateWebAudioSource();
+  std::unique_ptr<AudioSourceProvider> CreateWebAudioSource(
+      int context_sample_rate);
 
   void Trace(blink::Visitor*) override;
 

third_party/blink/renderer/modules/webaudio/media_stream_audio_source_node.cc

@@ -146,7 +146,7 @@ MediaStreamAudioSourceNode* MediaStreamAudioSourceNode::Create(
   // Use the first audio track in the media stream.
   MediaStreamTrack* audio_track = audio_tracks[0];
   std::unique_ptr<AudioSourceProvider> provider =
-      audio_track->CreateWebAudioSource();
+      audio_track->CreateWebAudioSource(context.sampleRate());
 
   MediaStreamAudioSourceNode* node =
       MakeGarbageCollected<MediaStreamAudioSourceNode>(

third_party/blink/renderer/platform/mediastream/media_stream_center.cc

@@ -97,11 +97,13 @@ void MediaStreamCenter::DidSetContentHint(MediaStreamComponent* track) {
 
 std::unique_ptr<AudioSourceProvider>
 MediaStreamCenter::CreateWebAudioSourceFromMediaStreamTrack(
-    MediaStreamComponent* track) {
+    MediaStreamComponent* track,
+    int context_sample_rate) {
   DCHECK(track);
   if (private_) {
-    return std::make_unique<MediaStreamWebAudioSource>(base::WrapUnique(
-        private_->CreateWebAudioSourceFromMediaStreamTrack(track)));
+    return std::make_unique<MediaStreamWebAudioSource>(
+        base::WrapUnique(private_->CreateWebAudioSourceFromMediaStreamTrack(
+            track, context_sample_rate)));
   }
 
   return nullptr;

third_party/blink/renderer/platform/mediastream/media_stream_center.h

@@ -63,7 +63,8 @@ class PLATFORM_EXPORT MediaStreamCenter {
   void DidSetMediaStreamTrackEnabled(MediaStreamComponent*);
   void DidSetContentHint(MediaStreamComponent*);
   std::unique_ptr<AudioSourceProvider> CreateWebAudioSourceFromMediaStreamTrack(
-      MediaStreamComponent*);
+      MediaStreamComponent*,
+      int context_sample_rate);
 
   void DidCreateMediaStreamAndTracks(MediaStreamDescriptor*);