Stop source and fire MediaStreamTrack ended event if missing audio input callbacks are detected.

content/renderer/media/webrtc/processed_local_audio_source.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_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 
 #include "base/atomicops.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "content/common/media/media_stream_options.h"
 #include "content/renderer/media/media_stream_audio_level_calculator.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "media/base/audio_capturer_source.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 
+#include "base/timer/timer.h"
+
 namespace media {
 class AudioBus;
 }
 
 namespace content {
 
 class PeerConnectionDependencyFactory;
 
 // Represents a local source of audio data that is routed through the WebRTC
 // audio pipeline for post-processing (e.g., for echo cancellation during a
@@ skipping 65 matching lines @@
                int audio_delay_milliseconds,
                double volume,
                bool key_pressed) override;
   void OnCaptureError(const std::string& message) override;
 
  private:
   // Helper function to get the source buffer size based on whether audio
   // processing will take place.
   int GetBufferSize(int sample_rate) const;
 
+  void CheckIfInputStreamIsAlive();
+
   // The RenderFrame that will consume the audio data. Used when creating
   // AudioCapturerSources.
   const int consumer_render_frame_id_;
 
   PeerConnectionDependencyFactory* const pc_factory_;
 
   // In debug builds, check that all methods that could cause object graph
   // or data flow changes are being called on the main thread.
   base::ThreadChecker thread_checker_;
 
@@ skipping 15 matching lines @@
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   base::subtle::Atomic32 volume_;
 
   // Used to calculate the signal level that shows in the UI.
   MediaStreamAudioLevelCalculator level_calculator_;
 
   bool allow_invalid_render_frame_id_for_testing_;
 
+  base::RepeatingTimer check_alive_timer_;
+  base::subtle::Atomic32 input_callback_is_active_;
+  base::TimeTicks last_callback_time_;
+
   DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioSource);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_