Remove dangling PlayingState pointers in WebRtcAudioRenderer.

content/renderer/media/webrtc_audio_renderer.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/non_thread_safe.h"
 #include "base/threading/thread_checker.h"
 #include "content/public/renderer/media_stream_audio_renderer.h"
@@ skipping 163 matching lines @@
   // it had already been removed or if the source isn't being rendered.
   bool RemovePlayingState(webrtc::AudioSourceInterface* source,
                           PlayingState* state);
 
   // Called whenever the Play/Pause state changes of any of the renderers
   // or if the volume of any of them is changed.
   // Here we update the shared Play state and apply volume scaling to all audio
   // sources associated with the |media_stream| based on the collective volume
   // of playing renderers.
   void OnPlayStateChanged(const blink::WebMediaStream& media_stream,
-                          PlayingState* state);
+                          PlayingState* state,
+                          bool playing_state_deleted);

[tommi (sloooow) - chröme, 2017/03/17 12:44:08]

what about including a TODO here to clean this up. I agree that it doesn't look
pretty.

[Max Morin, 2017/03/17 14:08:03]

I put a TODO in OnPlayStateRemoved.

 
   // Updates |sink_params_| and |audio_fifo_| based on |sink_|, and initializes
   // |sink_|.
   void PrepareSink();
 
   // The RenderFrame in which the audio is rendered into |sink_|.
   const int source_render_frame_id_;
   const int session_id_;
 
   const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_;
@@ skipping 49 matching lines @@
   // Stores the maximum time spent waiting for render data from the source. Used
   // for logging UMA data. Logged and reset when Stop() is called.
   base::TimeDelta max_render_time_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_