TransmitMixer: Check GetSendCodec return value.

webrtc/voice_engine/transmit_mixer.cc

Index: webrtc/voice_engine/transmit_mixer.cc
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 6796f8457c5e5b3060920c6be9995b8f739612b2..21140ace713f9e4f65bb274c8752ba4c90f4ff3f 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -247,9 +247,13 @@ void TransmitMixer::GetSendCodecInfo(int* max_sample_rate,
     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);
+      if (channel->GetSendCodec(codec) == 0) {

[minyue-webrtc, 2017/07/10 09:30:32]

I found another usage of this, should that be taken care of as well.

https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/channel....

[ossu, 2017/07/11 10:23:11]

We could add a DCHECK there. From what I can see, it's not actually being called
- rather the other overload of ProcessAndEncodeAudio is being used.

+        *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 "

[minyue-webrtc, 2017/07/10 09:30:32]

If I understand correctly, this is not supposed to happen because Sending()
returns true.

I think we'd understand why it is possible to reach here.

Code wise, I'd add RTC_NOTREACHED() in here.

[ossu, 2017/07/11 10:23:11]

NOTREACHED after the logging, I presume? So we'll be able to spot this happening
in logs and also have it crash if run in debug mode.

+                        << channel->ChannelId();
+      }
     }
   }
 }