Delete media type check in Call::NotifyBweOfReceivedPacket.

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 2693 matching lines @@
   else
     audio_coding_->DisableNack();
 }
 
 // Called when we are missing one or more packets.
 int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
   return _rtpRtcpModule->SendNACK(sequence_numbers, length);
 }
 
 void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) {
-  RTC_DCHECK(channel_state_.Get().sending);
+  // TODO(nisse): This is not quite enough to avoid race on
+  // destruction. In case we are preempted after this check, and some
+  // other thread calls ChannelManager::DestroyChannel whch calls

[henrika_webrtc, 2017/04/26 14:16:01]

nit, which

Also, in your failing test, why did you not hit the previous DCHECK?

+  // StopSend, StopSend's flush task gets posted *before* we post our
+  // task, our task can get run at the same time as the destructor.

[the sun, 2017/04/26 20:29:03]

Can we use ChannelOwner to allow the task to keep a reference to the Channel?
Seems hard to avoid ref counting in this situation.

+  if (!channel_state_.Get().sending)
+    return;
   std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
   // TODO(henrika): try to avoid copying by moving ownership of audio frame
   // either into pool of frames or into the task itself.
   audio_frame->CopyFrom(audio_input);
   audio_frame->id_ = ChannelId();
   encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
       new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
 }
 
 void Channel::ProcessAndEncodeAudio(const int16_t* audio_data,
                                     int sample_rate,
                                     size_t number_of_frames,
                                     size_t number_of_channels) {
-  RTC_DCHECK(channel_state_.Get().sending);
+  if (!channel_state_.Get().sending)
+    return;
   CodecInst codec;
   GetSendCodec(codec);
   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 369 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