Check the sample rate before passing the audio params to webrtc audio renderer

content/renderer/media/media_stream_impl.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/media_stream_impl.h"
 
 #include <utility>
 
 #include "base/logging.h"
 #include "base/strings/string_number_conversions.h"
@@ skipping 682 matching lines @@
 
       // TODO(tommi): Storing the session id in the deviceId field doesn't
       // feel right.  Move it over to MediaStreamSourceExtraData?
       session_id_str = source.deviceId();
       content::MediaStreamSourceExtraData* extra_data =
           static_cast<content::MediaStreamSourceExtraData*>(source.extraData());
       device_info = &extra_data->device_info();
     }
   }
 
-  if (session_id_str.isEmpty() || !device_info)
+  if (session_id_str.isEmpty() || !device_info ||
+      !device_info->device.matched_output.sample_rate) {
     return false;
+  }
 
   base::StringToInt(UTF16ToUTF8(session_id_str), session_id);
   *output_sample_rate = device_info->device.matched_output.sample_rate;
   *output_frames_per_buffer =
       device_info->device.matched_output.frames_per_buffer;
 
   return true;
 }
 
 MediaStreamSourceExtraData::MediaStreamSourceExtraData(
@@ skipping 54 matching lines @@
   }
 
   for (size_t i = 0; i < video_sources.size(); ++i) {
     video_sources[i].setReadyState(
             WebKit::WebMediaStreamSource::ReadyStateEnded);
     video_sources[i].setExtraData(NULL);
   }
 }
 
 }  // namespace content