Change thread check to race check

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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.
  */
 
 #ifdef HAVE_WEBRTC_VOICE
 
 #include "webrtc/media/engine/webrtcvoiceengine.h"
 
 #include <algorithm>
 #include <cstdio>
 #include <functional>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/race_checker.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/engine/payload_type_mapper.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
@@ skipping 1109 matching lines @@
                         webrtc::Call* call,
                         webrtc::Transport* send_transport)
       : voe_audio_transport_(voe_audio_transport),
         call_(call),
         config_(send_transport),
         rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
     RTC_DCHECK_GE(ch, 0);
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
     RTC_DCHECK(call);
-    audio_capture_thread_checker_.DetachFromThread();
     config_.rtp.ssrc = ssrc;
     config_.rtp.c_name = c_name;
     config_.voe_channel_id = ch;
     config_.rtp.extensions = extensions;
     RecreateAudioSendStream(send_codec_spec);
   }
 
   ~WebRtcAudioSendStream() override {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     ClearSource();
@@ skipping 94 matching lines @@
     UpdateSendState();
   }
 
   // AudioSource::Sink implementation.
   // This method is called on the audio thread.
   void OnData(const void* audio_data,
               int bits_per_sample,
               int sample_rate,
               size_t number_of_channels,
               size_t number_of_frames) override {
-    RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
+    RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
     RTC_DCHECK(voe_audio_transport_);
     voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data,
                                           bits_per_sample, sample_rate,
                                           number_of_channels, number_of_frames);
   }
 
   // Callback from the |source_| when it is going away. In case Start() has
   // never been called, this callback won't be triggered.
   void OnClose() override {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 26 matching lines @@
     RTC_DCHECK(stream_);
     RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size());
     if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) {
       stream_->Start();
     } else {  // !send || source_ = nullptr
       stream_->Stop();
     }
   }
 
   rtc::ThreadChecker worker_thread_checker_;
-  rtc::ThreadChecker audio_capture_thread_checker_;
+  rtc::RaceChecker audio_capture_race_checker_;
   webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
   webrtc::Call* call_ = nullptr;
   webrtc::AudioSendStream::Config config_;
   // The stream is owned by WebRtcAudioSendStream and may be reallocated if
   // configuration changes.
   webrtc::AudioSendStream* stream_ = nullptr;
 
   // Raw pointer to AudioSource owned by LocalAudioTrackHandler.
   // PeerConnection will make sure invalidating the pointer before the object
   // goes away.
@@ skipping 1330 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE