Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/rtc_peer_connection_handler.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/rtc_peer_connection_handler.h"
 
 #include <string>
 #include <utility>
 
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/utf_string_conversions.h"
 #include "content/renderer/media/media_stream_dependency_factory.h"
 #include "content/renderer/media/rtc_media_constraints.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaConstraints.h"
+#include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamComponent.h"
+#include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSource.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCConfiguration.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCICECandidate.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCPeerConnectionHandlerClient.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCSessionDescription.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCSessionDescriptionRequest.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCVoidRequest.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebURL.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebFrame.h"
 
 namespace content {
@@ skipping 273 matching lines @@
 
   bool return_value =
       native_peer_connection_->AddIceCandidate(native_candidate.get());
   LOG_IF(ERROR, !return_value) << "Error processing ICE candidate.";
   return return_value;
 }
 
 bool RTCPeerConnectionHandler::addStream(
     const WebKit::WebMediaStreamDescriptor& stream,
     const WebKit::WebMediaConstraints& options) {
+  // See if we're adding a WebAudio MediaStream.
+  WebKit::WebVector<WebKit::WebMediaStreamComponent> audioComponents;
+  stream.audioSources(audioComponents);
+  if (audioComponents.size() > 0) {

[perkj_chrome, 2012/11/12 10:35:11]

This seems to be in the wrong place and should be done when the underlaying
source is constructed in the media_stream_dependency_factory one way or the
other. Maybe via the MediaStreamCenter.

How does this work conceptually?
1. GetUserMedia creates a Ms with tracks where the source is a camera and a mic.


2. What is this doing? Are you replacing the mic source in the the ms from 1
with WebAudio or is this a new ms or is it a new audio track with a new source
you have added to 1? 

If you are replacing the the source in an existing audio track you should be
able to do that through the MediaStreamCenter and the
MediaStreamDependencyFactory.

If the output from WebAudio is a new MediaStream with a new audio source (the
output from the webaudio filter) - the MediaStream should be constructed through
MediaStreamCenter. Please see the existing constructor in MediaStreamCenter. 

+    if (audioComponents[0].source().isWebAudioSource()) {
+      DVLOG(1) << "RTCPeerConnectionHandler: adding WebAudio MediaStream.";
+      // TODO(crogers, xians): In reality we should be able to send a unique
+      // audio stream to each PeerConnection separately.  But currently WebRTC
+      // is only able to handle a global audio stream sent to ALL peers.
+      base::AutoLock auto_lock(audio_lock_);
+      webaudio_capturer_source_ = new WebAudioCapturerSource();

[tommi (sloooow) - chröme, 2012/11/12 09:46:18]

Should we first DCHECK that webaudioÖcapturer_source_ is NULL?

+      WebRtcAudioCapturer* capturer =
+          dependency_factory_->GetWebRtcAudioDevice()->capturer();

[tommi (sloooow) - chröme, 2012/11/12 09:46:18]

no chance that GetWebRtcAudioDevice() returns NULL or would that be a serious
error?

+      capturer->SetCapturerSource(webaudio_capturer_source_);
+      capturer->Start();
+    }
+  }
+
   RTCMediaConstraints constraints(options);
   return AddStream(stream, &constraints);
 }
 
 void RTCPeerConnectionHandler::removeStream(
     const WebKit::WebMediaStreamDescriptor& stream) {
   RemoveStream(stream);
 }
 
 void RTCPeerConnectionHandler::stop() {
   DVLOG(1) << "RTCPeerConnectionHandler::stop";
   native_peer_connection_ = NULL;
 }
 
+void RTCPeerConnectionHandler::consumeAudio(
+    const WebKit::WebVector<const float*>& audio_data,
+    size_t number_of_frames) {
+  base::AutoLock auto_lock(audio_lock_);
+  if (webaudio_capturer_source_.get())

[tommi (sloooow) - chröme, 2012/11/12 09:46:18]

nit: scoped_refptr supports checking validity without .get() and that seems to
be the norm now.

+    webaudio_capturer_source_->HandleCapture(audio_data, number_of_frames);
+}
+
 void RTCPeerConnectionHandler::OnError() {
   // TODO(perkj): Implement.
   NOTIMPLEMENTED();
 }
 
 void RTCPeerConnectionHandler::OnStateChange(StateType state_changed) {
   switch (state_changed) {
     case kReadyState: {
       WebKit::WebRTCPeerConnectionHandlerClient::ReadyState ready_state =
           GetWebKitReadyState(native_peer_connection_->ready_state());
@@ skipping 71 matching lines @@
   webrtc::SessionDescriptionInterface* native_desc =
       dependency_factory_->CreateSessionDescription(type, sdp);
 
   LOG_IF(ERROR, !native_desc) << "Failed to create native session description."
                               << " Type: " << type << " SDP: " << sdp;
 
   return native_desc;
 }
 
 }  // namespace content