Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()

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 1199 matching lines @@
   return 0;
 }
 
 int32_t Channel::StartSend() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::StartSend()");
   if (channel_state_.Get().sending) {
     return 0;
   }
   channel_state_.SetSending(true);
-
+  {
+    // It is now OK to start posting tasks to the encoder task queue.
+    rtc::CritScope cs(&encoder_queue_lock_);
+    encoder_queue_is_active_ = true;
+  }
   // Resume the previous sequence number which was reset by StopSend(). This
   // needs to be done before |sending| is set to true on the RTP/RTCP module.
   if (send_sequence_number_) {
     _rtpRtcpModule->SetSequenceNumber(send_sequence_number_);
   }
   _rtpRtcpModule->SetSendingMediaStatus(true);
   if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceError,
         "StartSend() RTP/RTCP failed to start sending");
@@ skipping 14 matching lines @@
   }
   channel_state_.SetSending(false);
 
   // Post a task to the encoder thread which sets an event when the task is
   // executed. We know that no more encoding tasks will be added to the task
   // queue for this channel since sending is now deactivated. It means that,
   // if we wait for the event to bet set, we know that no more pending tasks
   // exists and it is therfore guaranteed that the task queue will never try
   // to acccess and invalid channel object.
   RTC_DCHECK(encoder_queue_);
+
   rtc::Event flush(false, false);
-  encoder_queue_->PostTask([&flush]() { flush.Set(); });
+  {
+    // Clear |encoder_queue_is_active_| under lock to prevent any other tasks
+    // than this final "flush task" to be posted on the queue.
+    rtc::CritScope cs(&encoder_queue_lock_);
+    encoder_queue_is_active_ = false;
+    encoder_queue_->PostTask([&flush]() { flush.Set(); });
+  }
   flush.Wait(rtc::Event::kForever);
 
   // Store the sequence number to be able to pick up the same sequence for
   // the next StartSend(). This is needed for restarting device, otherwise
   // it might cause libSRTP to complain about packets being replayed.
   // TODO(xians): Remove this workaround after RtpRtcpModule's refactoring
   // CL is landed. See issue
   // https://code.google.com/p/webrtc/issues/detail?id=2111 .
   send_sequence_number_ = _rtpRtcpModule->SequenceNumber();
 
@@ skipping 1450 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);
+  // Avoid posting any new tasks if sending was already stopped in StopSend().
+  rtc::CritScope cs(&encoder_queue_lock_);
+  if (!encoder_queue_is_active_) {
+    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);
+  // 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);
   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