TransmitMixer: Check GetSendCodec return value.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 16122709c31fe976ddde3c63a477d78957f72070..4df8e67239342419447b900e9ace36b370090320 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -2759,7 +2759,8 @@ void Channel::ProcessAndEncodeAudio(const int16_t* audio_data,
     return;
   }
   CodecInst codec;
-  GetSendCodec(codec);
+  const int result = GetSendCodec(codec);

[minyue-webrtc, 2017/07/11 11:18:09]

will this be safe enough in release build?

[ossu, 2017/07/11 12:15:10]

Not sure. Added a fallback route in case this returns != 0. In that case, the
data should make it into AudioCodingModuleImpl::Add10MsDataInternal and be
rejected by HaveValidEncoder() if there's no encoder set.

+  RTC_DCHECK_EQ(result, 0);
   std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
   audio_frame->id_ = ChannelId();
   audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate);