MediaStream audio: Refactor 3 separate "glue" implementations into one.

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.h>
 
 #include <string>
 #include <utility>
@@ skipping 13 matching lines @@
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_audio_track.h"
 #include "content/renderer/media/media_stream_constraints_util.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #include "content/renderer/media/remote_media_stream_impl.h"
 #include "content/renderer/media/rtc_certificate.h"
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
+#include "content/renderer/media/webrtc/webrtc_audio_sink.h"
 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/base/media_switches.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebRTCAnswerOptions.h"
 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
 #include "third_party/WebKit/public/platform/WebRTCOfferOptions.h"
@@ skipping 1433 matching lines @@
 
   ++num_data_channels_created_;
 
   return new RtcDataChannelHandler(base::ThreadTaskRunnerHandle::Get(),
                                    webrtc_channel);
 }
 
 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
     const blink::WebMediaStreamTrack& track) {
   DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(!track.isNull());
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender");
   DVLOG(1) << "createDTMFSender.";
 
-  MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::From(track);
-  if (!native_track || !native_track->is_local_track() ||
-      track.source().getType() != blink::WebMediaStreamSource::TypeAudio) {
-    DLOG(ERROR) << "The DTMF sender requires a local audio track.";
+  ProcessedLocalAudioSource* const rtc_audio_source =
+      ProcessedLocalAudioSource::From(
+          MediaStreamAudioSource::From(track.source()));
+  if (!rtc_audio_source) {
+    DLOG(ERROR) << "WebRTC features are not available on this audio track.";
     return nullptr;
   }
 
-  scoped_refptr<webrtc::AudioTrackInterface> audio_track =
-      native_track->GetAudioAdapter();
+  // HACK: Create a temporary WebRtcAudioSink that can provide an instance of
+  // webrtc::AudioTrackInterface to the DtmfSender, as the interface requires.
+  //
+  // TODO(miu): The implementation only needs the track.id() string.  Thus, the
+  // interface declaring the CreateDtmfSender method should be changed to only
+  // only take the track id as an argument here.  Then, we can get rid of
+  // |dummy_sink|.
+  const std::unique_ptr<WebRtcAudioSink> dummy_sink(new WebRtcAudioSink(

[perkj_chrome, 2016/04/20 13:34:54]

Can you instead find the correct webrtc audio track from 
the local mediastreams added to the pc in local_streams_
(WebRtcMediaStreamAdapter) .

Something like:


webrtc::AudioTrackInterface* webrtc_track = nullptr;
for (auto adapter_it : local_streams_) {
webrtc_track = 
adapter_it->webrtc_media_stream()->FindAudioTrack(track.id().utf8()));
if (webrtc_track)
  break;
}

[miu, 2016/04/20 22:04:53]

Done.  Yes!  This is what I was looking for.  :)

BTW--Tested successfully using this demo:
https://webrtc.github.io/samples/src/content/peerconnection/dtmf/

+      track.id().utf8(), rtc_audio_source->rtc_source(),
+      dependency_factory_->GetWebRtcSignalingThread()));
+
   rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender(
-      native_peer_connection_->CreateDtmfSender(audio_track.get()));
+      native_peer_connection_->CreateDtmfSender(
+          dummy_sink->webrtc_audio_track()));
   if (!sender) {
     DLOG(ERROR) << "Could not create native DTMF sender.";
     return nullptr;
   }
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackCreateDTMFSender(this, track);
 
   return new RtcDtmfSenderHandler(sender);
 }
 
@@ skipping 283 matching lines @@
 }
 
 void RTCPeerConnectionHandler::ResetUMAStats() {
   DCHECK(thread_checker_.CalledOnValidThread());
   num_local_candidates_ipv6_ = 0;
   num_local_candidates_ipv4_ = 0;
   ice_connection_checking_start_ = base::TimeTicks();
   memset(ice_state_seen_, 0, sizeof(ice_state_seen_));
 }
 }  // namespace content