TransmitMixer: Check GetSendCodec return value.

webrtc/voice_engine/channel.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 2741 matching lines @@
 void Channel::ProcessAndEncodeAudio(const int16_t* audio_data,
                                     int sample_rate,
                                     size_t number_of_frames,
                                     size_t number_of_channels) {
   // Avoid posting as new task if sending was already stopped in StopSend().
   rtc::CritScope cs(&encoder_queue_lock_);
   if (!encoder_queue_is_active_) {
     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);
   audio_frame->num_channels_ = std::min(number_of_channels, codec.channels);
   RemixAndResample(audio_data, number_of_frames, number_of_channels,
                    sample_rate, &input_resampler_, audio_frame.get());
   encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
       new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
 }
 
@@ skipping 367 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc