Log error instead of CHECKing input audio sequence.

content/renderer/media/webrtc_audio_capturer.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_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
@@ skipping 111 matching lines @@
                       const blink::WebMediaConstraints& constraints,
                       WebRtcAudioDeviceImpl* audio_device,
                       MediaStreamAudioSource* audio_source);
 
   // AudioCapturerSource::CaptureCallback implementation.
   // Called on the AudioInputDevice audio thread.
   void Capture(const media::AudioBus* audio_source,
                int audio_delay_milliseconds,
                double volume,
                bool key_pressed) override;
-  void OnCaptureError() override;
+  void OnCaptureError(const std::string& message) override;
 
   // Initializes the default audio capturing source using the provided render
   // frame id and device information. Return true if success, otherwise false.
   bool Initialize();
 
   // SetCapturerSourceInternal() is called if the client on the source side
   // desires to provide their own captured audio data. Client is responsible
   // for calling Start() on its own source to get the ball rolling.
   // Called on the main render thread.
   // buffer_size is optional. Set to 0 to let it be chosen automatically.
@@ skipping 60 matching lines @@
   // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this
   // WebRtcAudioCapturer.
   MediaStreamAudioSource* const audio_source_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_