Update MediaStreamTrack::Stop to latest draft.

content/renderer/media/webrtc_audio_capturer.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/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
 #include "content/child/child_process.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
+#include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "content/renderer/media/webrtc_logging.h"
 #include "media/audio/sample_rates.h"
 
 namespace content {
 
 namespace {
 
 // Supported hardware sample rates for input and output sides.
@@ skipping 54 matching lines @@
     delegate_ = NULL;
   }
 
   void Stop() {
     base::AutoLock lock(lock_);
     DCHECK(delegate_);
 
     // This can be reentrant so reset |delegate_| before calling out.
     WebRtcLocalAudioTrack* temp = delegate_;
     delegate_ = NULL;
-    temp->Stop();
+    temp->StopTrack();
   }
 
   // Wrapper which allows to use std::find_if() when adding and removing
   // sinks to/from the list.
   struct TrackWrapper {
     TrackWrapper(WebRtcLocalAudioTrack* track) : track_(track) {}
     bool operator()(
         const scoped_refptr<WebRtcAudioCapturer::TrackOwner>& owner) const {
       return owner->IsEqual(track_);
     }
@@ skipping 15 matching lines @@
   WebRtcLocalAudioTrack* delegate_;
   mutable base::Lock lock_;
 
   DISALLOW_COPY_AND_ASSIGN(TrackOwner);
 };
 
 // static
 scoped_refptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer(
     int render_view_id, const StreamDeviceInfo& device_info,
     const blink::WebMediaConstraints& constraints,
-    WebRtcAudioDeviceImpl* audio_device) {
+    WebRtcAudioDeviceImpl* audio_device,
+    MediaStreamAudioSource* audio_source) {
   scoped_refptr<WebRtcAudioCapturer> capturer = new WebRtcAudioCapturer(
-      render_view_id, device_info, constraints, audio_device);
+      render_view_id, device_info, constraints, audio_device, audio_source);
   if (capturer->Initialize())
     return capturer;
 
   return NULL;
 }
 
 bool WebRtcAudioCapturer::Initialize() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
   WebRtcLogMessage(base::StringPrintf(
@@ skipping 62 matching lines @@
   if (audio_device_)
     audio_device_->AddAudioCapturer(this);
 
   return true;
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer(
     int render_view_id,
     const StreamDeviceInfo& device_info,
     const blink::WebMediaConstraints& constraints,
-    WebRtcAudioDeviceImpl* audio_device)
+    WebRtcAudioDeviceImpl* audio_device,
+    MediaStreamAudioSource* audio_source)
     : constraints_(constraints),
       audio_processor_(
           new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
               constraints, device_info.device.input.effects,
               device_info.device.type, audio_device)),
       running_(false),
       render_view_id_(render_view_id),
       device_info_(device_info),
       volume_(0),
       peer_connection_mode_(false),
       key_pressed_(false),
       need_audio_processing_(false),
-      audio_device_(audio_device) {
+      audio_device_(audio_device),
+      audio_source_(audio_source) {
   DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
 }
 
 WebRtcAudioCapturer::~WebRtcAudioCapturer() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(tracks_.IsEmpty());
   DCHECK(!running_);
   DVLOG(1) << "WebRtcAudioCapturer::~WebRtcAudioCapturer()";
 }
 
@@ skipping 13 matching lines @@
   }
 
   // Start the source if the first audio track is connected to the capturer.
   // Start() will do nothing if the capturer has already been started.
   Start();
 
 }
 
 void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) {
   DCHECK(thread_checker_.CalledOnValidThread());
-  base::AutoLock auto_lock(lock_);
+  DVLOG(1) << "WebRtcAudioCapturer::RemoveTrack()";
+  bool stop_source = false;
+  {
+    base::AutoLock auto_lock(lock_);
 
-  scoped_refptr<TrackOwner> removed_item =
-      tracks_.Remove(TrackOwner::TrackWrapper(track));
+    scoped_refptr<TrackOwner> removed_item =
+        tracks_.Remove(TrackOwner::TrackWrapper(track));
 
-  // Clear the delegate to ensure that no more capture callbacks will
-  // be sent to this sink. Also avoids a possible crash which can happen
-  // if this method is called while capturing is active.
-  if (removed_item.get())
-    removed_item->Reset();
+    // Clear the delegate to ensure that no more capture callbacks will
+    // be sent to this sink. Also avoids a possible crash which can happen
+    // if this method is called while capturing is active.
+    if (removed_item.get()) {
+      removed_item->Reset();
+      stop_source = tracks_.IsEmpty();
+    }

[no longer working on chromium, 2014/04/01 18:29:58]

you also need to set source_ to NULL to prevent starting the source again.

[perkj_chrome, 2014/04/02 13:35:49]

Please see my comment. I don't think that should be done here.

+  }
+  if (stop_source) {

[no longer working on chromium, 2014/04/01 18:29:58]

nit, remove the {}

[perkj_chrome, 2014/04/02 13:35:49]

I just added a long comment here.

+    audio_source_->StopSource();
+  }
 }
 
 void WebRtcAudioCapturer::SetCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::ChannelLayout channel_layout,
     float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
            << "sample_rate=" << sample_rate << ")";
   scoped_refptr<media::AudioCapturerSource> old_source;
@@ skipping 302 matching lines @@
   DCHECK_NE(aec_dump_file, base::kInvalidPlatformFileValue);
   audio_processor_->StartAecDump(aec_dump_file);
 }
 
 void WebRtcAudioCapturer::StopAecDump() {
   DCHECK(thread_checker_.CalledOnValidThread());
   audio_processor_->StopAecDump();
 }
 
 }  // namespace content