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 98 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_source_(audio_source),
       audio_power_monitor_(
           device_info_.device.input.sample_rate,
           base::TimeDelta::FromMilliseconds(kPowerMonitorTimeConstantMs)) {
   DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
 }
 
 WebRtcAudioCapturer::~WebRtcAudioCapturer() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(tracks_.IsEmpty());
-  DCHECK(!running_);
   DVLOG(1) << "WebRtcAudioCapturer::~WebRtcAudioCapturer()";
+  Stop();
 }
 
 void WebRtcAudioCapturer::AddTrack(WebRtcLocalAudioTrack* track) {
   DCHECK(track);
   DVLOG(1) << "WebRtcAudioCapturer::AddTrack()";
 
   {
     base::AutoLock auto_lock(lock_);
     // Verify that |track| is not already added to the list.
     DCHECK(!tracks_.Contains(TrackOwner::TrackWrapper(track)));
 
     // Add with a tag, so we remember to call OnSetFormat() on the new
     // track.
     scoped_refptr<TrackOwner> track_owner(new TrackOwner(track));
     tracks_.AddAndTag(track_owner);
   }
-
-  // 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();
+    }
+  }
+  if (stop_source) {
+    // Since WebRtcAudioCapturer does not inherit MediaStreamAudioSource,
+    // and instead MediaStreamAudioSource is composed of a WebRtcAudioCapturer,
+    // we have to call StopSource on the MediaStreamSource. This will call
+    // MediaStreamAudioSource::DoStopSource which in turn call
+    // WebRtcAudioCapturerer::Stop();
+    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;
-  bool restart_source = false;
   {
     base::AutoLock auto_lock(lock_);
     if (source_.get() == source.get())
       return;
 
     source_.swap(old_source);
     source_ = source;
 
     // Reset the flag to allow starting the new source.
-    restart_source = running_;
     running_ = false;
   }
 
   DVLOG(1) << "Switching to a new capture source.";
   if (old_source.get())
     old_source->Stop();
 
   // Dispatch the new parameters both to the sink(s) and to the new source,
   // also apply the new |constraints|.
   // The idea is to get rid of any dependency of the microphone parameters
@@ skipping 12 matching lines @@
 
     need_audio_processing_ = NeedsAudioProcessing(
         constraints_, device_info_.device.input.effects);
     // Notify all tracks about the new format.
     tracks_.TagAll();
   }
 
   if (source.get())
     source->Initialize(params, this, session_id());
 
-  if (restart_source)
-    Start();
+  Start();
 }
 
 void WebRtcAudioCapturer::EnablePeerConnectionMode() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "EnablePeerConnectionMode";
   // Do nothing if the peer connection mode has been enabled.
   if (peer_connection_mode_)
     return;
 
   peer_connection_mode_ = true;
@@ skipping 261 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