Switch LiveAudio to source provider solution.

content/renderer/media/media_stream_dependency_factory.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/media_stream_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 43 matching lines @@
 // WebRTC logging. It's a Google specific key, hence the "goog" prefix.
 const char kWebRtcLoggingConstraint[] = "googLog";
 
 // Constant constraint keys which disables all audio constraints.
 // Only used in combination with WebAudio sources.
 struct {
   const char* key;
   const char* value;
 } const kWebAudioConstraints[] = {
     {webrtc::MediaConstraintsInterface::kEchoCancellation,
-     webrtc::MediaConstraintsInterface::kValueTrue},
+     webrtc::MediaConstraintsInterface::kValueFalse},

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

please cc bemasc.  It does seem though that if all these are set to false we
might as well not set them and remove the ApplyFixedWebAudioConstraints
function.

[no longer working on chromium, 2013/09/10 12:43:15]

Removed the changed now, I want to completely remove the hard-coded constrains,
but lets do it in a separate CL.

[tommi (sloooow) - chröme, 2013/09/10 16:00:38]

sgtm

     {webrtc::MediaConstraintsInterface::kAutoGainControl,
-     webrtc::MediaConstraintsInterface::kValueTrue},
+     webrtc::MediaConstraintsInterface::kValueFalse},
     {webrtc::MediaConstraintsInterface::kNoiseSuppression,
-     webrtc::MediaConstraintsInterface::kValueTrue},
+     webrtc::MediaConstraintsInterface::kValueFalse},
     {webrtc::MediaConstraintsInterface::kHighpassFilter,
-     webrtc::MediaConstraintsInterface::kValueTrue},
+     webrtc::MediaConstraintsInterface::kValueFalse},
 };
 
 void ApplyFixedWebAudioConstraints(RTCMediaConstraints* constraints) {
   for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kWebAudioConstraints); ++i) {
     constraints->AddMandatory(kWebAudioConstraints[i].key,
         kWebAudioConstraints[i].value, false);
   }
 }
 
 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
@@ skipping 284 matching lines @@
     WebKit::WebMediaStream* web_stream,
     const MediaStreamExtraData::StreamStopCallback& stream_stop) {
   CreateNativeLocalMediaStream(web_stream);
 
   MediaStreamExtraData* extra_data =
      static_cast<MediaStreamExtraData*>(web_stream->extraData());
   extra_data->SetLocalStreamStopCallback(stream_stop);
 }
 
 bool MediaStreamDependencyFactory::AddNativeMediaStreamTrack(
-      const WebKit::WebMediaStream& stream,
-      const WebKit::WebMediaStreamTrack& track) {
+    const WebKit::WebMediaStream& stream,
+    const WebKit::WebMediaStreamTrack& track) {
   MediaStreamExtraData* extra_data =
      static_cast<MediaStreamExtraData*>(stream.extraData());
   webrtc::MediaStreamInterface* native_stream = extra_data->stream().get();
   DCHECK(native_stream);
 
   WebKit::WebMediaStreamSource source = track.source();
   MediaStreamSourceExtraData* source_data =
       static_cast<MediaStreamSourceExtraData*>(source.extraData());
 
+
   // In the future the constraints will belong to the track itself, but
   // right now they're on the source, so we fetch them from there.
   RTCMediaConstraints track_constraints(source.constraints());
 
-  scoped_refptr<WebRtcAudioCapturer> capturer;
+  scoped_refptr<WebAudioCapturerSource> webaudio_source;
   if (!source_data) {
     if (source.requiresAudioConsumer()) {
       // We're adding a WebAudio MediaStream.
       // Create a specific capturer for each WebAudio consumer.
-      capturer = CreateWebAudioSource(&source, &track_constraints);
+      webaudio_source = CreateWebAudioSource(&source, &track_constraints);
       source_data =
           static_cast<MediaStreamSourceExtraData*>(source.extraData());
     } else {
       // TODO(perkj): Implement support for sources from
       // remote MediaStreams.
       NOTIMPLEMENTED();
       return false;
     }
   }
 
   WebKit::WebMediaStreamSource::Type type = track.source().type();
   DCHECK(type == WebKit::WebMediaStreamSource::TypeAudio ||
          type == WebKit::WebMediaStreamSource::TypeVideo);
 
   std::string track_id = UTF16ToUTF8(track.id());
   if (source.type() == WebKit::WebMediaStreamSource::TypeAudio) {
-    if (!capturer.get() && GetWebRtcAudioDevice())
+    scoped_refptr<WebRtcAudioCapturer> capturer;
+    if (GetWebRtcAudioDevice())
       capturer = GetWebRtcAudioDevice()->GetDefaultCapturer();
 
     scoped_refptr<webrtc::AudioTrackInterface> audio_track(
         CreateLocalAudioTrack(track_id,
                               capturer,
+                              webaudio_source.get(),
                               source_data->local_audio_source(),
                               &track_constraints));
     audio_track->set_enabled(track.isEnabled());
+    if (capturer.get()) {
+      WebKit::WebMediaStreamTrack writable_track = track;
+      writable_track.setSourceProvider(capturer->AudioSourceProvider());
+    }
     return native_stream->AddTrack(audio_track.get());
   } else {
     DCHECK(source.type() == WebKit::WebMediaStreamSource::TypeVideo);
     scoped_refptr<webrtc::VideoTrackInterface> video_track(
         CreateLocalVideoTrack(track_id, source_data->video_source()));
     video_track->set_enabled(track.isEnabled());
     return native_stream->AddTrack(video_track.get());
   }
 }
 
@@ skipping 37 matching lines @@
   MediaStreamExtraData* extra_data =
       static_cast<MediaStreamExtraData*>(stream.extraData());
   webrtc::MediaStreamInterface* native_stream = extra_data->stream().get();
   DCHECK(native_stream);
 
   WebKit::WebMediaStreamSource::Type type = track.source().type();
   DCHECK(type == WebKit::WebMediaStreamSource::TypeAudio ||
          type == WebKit::WebMediaStreamSource::TypeVideo);
 
   std::string track_id = UTF16ToUTF8(track.id());
-  return type == WebKit::WebMediaStreamSource::TypeAudio ?
-      native_stream->RemoveTrack(native_stream->FindAudioTrack(track_id)) :
-      native_stream->RemoveTrack(native_stream->FindVideoTrack(track_id));
+  if (type == WebKit::WebMediaStreamSource::TypeAudio) {
+    // Remove the source provider as the track is going away.
+    WebKit::WebMediaStreamTrack writable_track = track;
+    writable_track.setSourceProvider(NULL);
+    return native_stream->RemoveTrack(native_stream->FindAudioTrack(track_id));
+  }
+
+  return native_stream->RemoveTrack(native_stream->FindVideoTrack(track_id));

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

can you add a DCHECK_EQ(type, WebKit::WebMediaStreamSource::TypeVideo) first?

[no longer working on chromium, 2013/09/10 12:43:15]

Done.

 }
 
 bool MediaStreamDependencyFactory::CreatePeerConnectionFactory() {
   DVLOG(1) << "MediaStreamDependencyFactory::CreatePeerConnectionFactory()";
   if (!pc_factory_.get()) {
     DCHECK(!audio_device_.get());
     audio_device_ = new WebRtcAudioDeviceImpl();
 
     scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
     scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
@@ skipping 96 matching lines @@
     const webrtc::MediaConstraintsInterface* constraints) {
   RtcVideoCapturer* capturer = new RtcVideoCapturer(
       video_session_id, vc_manager_.get(), is_screencast);
 
   // The video source takes ownership of |capturer|.
   scoped_refptr<webrtc::VideoSourceInterface> source =
       pc_factory_->CreateVideoSource(capturer, constraints).get();
   return source;
 }
 
-scoped_refptr<WebRtcAudioCapturer>
+scoped_refptr<WebAudioCapturerSource>
 MediaStreamDependencyFactory::CreateWebAudioSource(
     WebKit::WebMediaStreamSource* source,
     RTCMediaConstraints* constraints) {
   DVLOG(1) << "MediaStreamDependencyFactory::CreateWebAudioSource()";
   DCHECK(GetWebRtcAudioDevice());
 
-  // Set up the source and ensure that WebAudio is driving things instead of
-  // a microphone. For WebAudio, we always create a new capturer without
-  // calling initialize(), WebAudio will re-configure the capturer later on.
-  // Pass -1 as the |render_view_id| and an empty device struct to tell the
-  // capturer not to start the default source.
-  scoped_refptr<WebRtcAudioCapturer> capturer(
-      MaybeCreateAudioCapturer(-1, StreamDeviceInfo()));

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

nice to get rid of this :)

-  DCHECK(capturer.get());
-
   scoped_refptr<WebAudioCapturerSource>
-      webaudio_capturer_source(new WebAudioCapturerSource(capturer.get()));
+      webaudio_capturer_source(new WebAudioCapturerSource());
   MediaStreamSourceExtraData* source_data =
-      new content::MediaStreamSourceExtraData(webaudio_capturer_source.get());
+      new content::MediaStreamSourceExtraData();
 
   // Create a LocalAudioSource object which holds audio options.
   // Use audio constraints where all values are true, i.e., enable
   // echo cancellation, automatic gain control, noise suppression and
   // high-pass filter. SetLocalAudioSource() affects core audio parts in
   // third_party/Libjingle.
   ApplyFixedWebAudioConstraints(constraints);
   source_data->SetLocalAudioSource(CreateLocalAudioSource(constraints).get());
   source->setExtraData(source_data);
 
   // Replace the default source with WebAudio as source instead.
   source->addAudioConsumer(webaudio_capturer_source.get());
 
-  return capturer;
+  return webaudio_capturer_source;
 }
 
 scoped_refptr<webrtc::VideoTrackInterface>
 MediaStreamDependencyFactory::CreateLocalVideoTrack(
     const std::string& id,
     webrtc::VideoSourceInterface* source) {
   return pc_factory_->CreateVideoTrack(id, source).get();
 }
 
 scoped_refptr<webrtc::VideoTrackInterface>
 MediaStreamDependencyFactory::CreateLocalVideoTrack(
     const std::string& id, cricket::VideoCapturer* capturer) {
   if (!capturer) {
     LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
     return NULL;
   }
 
   // Create video source from the |capturer|.
   scoped_refptr<webrtc::VideoSourceInterface> source =
       pc_factory_->CreateVideoSource(capturer, NULL).get();
 
   // Create native track from the source.
   return pc_factory_->CreateVideoTrack(id, source.get()).get();
 }
 
 scoped_refptr<webrtc::AudioTrackInterface>
 MediaStreamDependencyFactory::CreateLocalAudioTrack(
     const std::string& id,
     const scoped_refptr<WebRtcAudioCapturer>& capturer,
-    webrtc::AudioSourceInterface* source,
+    WebAudioCapturerSource* webaudio_source,
+    webrtc::AudioSourceInterface* track_source,
     const webrtc::MediaConstraintsInterface* constraints) {
   // TODO(xians): Merge |source| to the capturer(). We can't do this today
   // because only one capturer() is supported while one |source| is created
   // for each audio track.
   scoped_refptr<WebRtcLocalAudioTrack> audio_track(
-      WebRtcLocalAudioTrack::Create(id, capturer, source, constraints));
+      WebRtcLocalAudioTrack::Create(id, capturer, webaudio_source,
+                                    track_source, constraints));
+
   // Add the WebRtcAudioDevice as the sink to the local audio track.
   audio_track->AddSink(GetWebRtcAudioDevice());
   // Start the audio track. This will hook the |audio_track| to the capturer
   // as the sink of the audio, and only start the source of the capturer if
   // it is the first audio track connecting to the capturer.
   audio_track->Start();
   return audio_track;
 }
 
 webrtc::SessionDescriptionInterface*
@@ skipping 116 matching lines @@
     } else {
       NOTREACHED() << "Worker thread not running.";
     }
   }
 }
 
 scoped_refptr<WebRtcAudioCapturer>
 MediaStreamDependencyFactory::MaybeCreateAudioCapturer(
     int render_view_id,
     const StreamDeviceInfo& device_info) {
-  scoped_refptr<WebRtcAudioCapturer> capturer;
-  if (render_view_id != -1) {
-    // From a normal getUserMedia, re-use the existing default capturer.
-    capturer = GetWebRtcAudioDevice()->GetDefaultCapturer();
-  }
+  if (render_view_id < 0)

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

Can we now simply DCHECK that render_view_id is >= 0 and skip this if()?

[no longer working on chromium, 2013/09/10 12:43:15]

I think the answer might depend on if we support gUM without a proper render
view, like extension.
I added a TODO here and changed to a DCHECK, let me know if you have another
thought.

[tommi (sloooow) - chröme, 2013/09/10 16:00:38]

sgtm

+    return NULL;
+
+  scoped_refptr<WebRtcAudioCapturer> capturer =
+      GetWebRtcAudioDevice()->GetDefaultCapturer();
+
   // If the default capturer does not exist or |render_view_id| == -1, create
   // a new capturer.
   bool is_new_capturer = false;
   if (!capturer.get()) {
     capturer = WebRtcAudioCapturer::CreateCapturer();
     is_new_capturer = true;
   }
 
   if (!capturer->Initialize(
           render_view_id,
           static_cast<media::ChannelLayout>(device_info.device.channel_layout),
           device_info.device.sample_rate, device_info.session_id,
           device_info.device.id)) {
     return NULL;
   }
 
   // Add the capturer to the WebRtcAudioDeviceImpl if it is a new capturer.
   if (is_new_capturer)
     GetWebRtcAudioDevice()->AddAudioCapturer(capturer);
 
   return capturer;
 }
 
 }  // namespace content