TransmitMixer: Check GetSendCodec return value.

webrtc/voice_engine/transmit_mixer.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 229 matching lines @@
 
 void TransmitMixer::GetSendCodecInfo(int* max_sample_rate,
                                      size_t* max_channels) {
   *max_sample_rate = 8000;
   *max_channels = 1;
   for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
        it.Increment()) {
     Channel* channel = it.GetChannel();
     if (channel->Sending()) {
       CodecInst codec;
-      channel->GetSendCodec(codec);
-      *max_sample_rate = std::max(*max_sample_rate, codec.plfreq);
-      *max_channels = std::max(*max_channels, codec.channels);
+      // TODO(ossu): Investigate how this could happen. b/62909493
+      if (channel->GetSendCodec(codec) == 0) {
+        *max_sample_rate = std::max(*max_sample_rate, codec.plfreq);
+        *max_channels = std::max(*max_channels, codec.channels);
+      } else {
+        LOG(LS_WARNING) << "Unable to get send codec for channel "
+                        << channel->ChannelId();
+        RTC_NOTREACHED();
+      }
     }
   }
 }
 
 int32_t
 TransmitMixer::PrepareDemux(const void* audioSamples,
                             size_t nSamples,
                             size_t nChannels,
                             uint32_t samplesPerSec,
                             uint16_t totalDelayMS,
@@ skipping 753 matching lines @@
 void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
   swap_stereo_channels_ = enable;
 }
 
 bool TransmitMixer::IsStereoChannelSwappingEnabled() {
   return swap_stereo_channels_;
 }
 
 }  // namespace voe
 }  // namespace webrtc