Added thread checker to webrtc::Call.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
 #include <string.h>
 
 #include <map>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
 #include "webrtc/system_wrappers/interface/cpu_info.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
@@ skipping 58 matching lines @@
                             const PacketTime& packet_time);
 
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<ChannelGroup> channel_group_;
   volatile int next_channel_id_;
   Call::Config config_;
+  rtc::ThreadChecker configuration_thread_checker_;
 
   // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
   // ensures that we have a consistent network state signalled to all senders
   // and receivers.
   rtc::CriticalSection network_enabled_crit_;
   bool network_enabled_ GUARDED_BY(network_enabled_crit_);
 
   rtc::scoped_ptr<RWLockWrapper> receive_crit_;
   // Audio and Video receive streams are owned by the client that creates them.
   std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
@@ skipping 28 matching lines @@
 Call::Call(const Call::Config& config)
     : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
       channel_group_(new ChannelGroup(module_process_thread_.get())),
       next_channel_id_(0),
       config_(config),
       network_enabled_(true),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       event_log_(nullptr) {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   if (config.voice_engine) {
     VoECodec* voe_codec = VoECodec::GetInterface(config.voice_engine);
     if (voe_codec) {
       event_log_ = voe_codec->GetEventLog();
       voe_codec->Release();
     }
   }
 
   Trace::CreateTrace();
   module_process_thread_->Start();
 
   channel_group_->SetBweBitrates(config_.bitrate_config.min_bitrate_bps,
                                  config_.bitrate_config.start_bitrate_bps,
                                  config_.bitrate_config.max_bitrate_bps);
 }
 
 Call::~Call() {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_CHECK(audio_send_ssrcs_.empty());
   RTC_CHECK(video_send_ssrcs_.empty());
   RTC_CHECK(video_send_streams_.empty());
   RTC_CHECK(audio_receive_ssrcs_.empty());
   RTC_CHECK(video_receive_ssrcs_.empty());
   RTC_CHECK(video_receive_streams_.empty());
 
   module_process_thread_->Stop();
   Trace::ReturnTrace();
 }
 
-PacketReceiver* Call::Receiver() { return this; }
+PacketReceiver* Call::Receiver() {
+  // TODO(solenberg): Some test cases in EndToEndTest use this from a different
+  // thread. Re-enable once that is fixed.
+  // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  return this;
+}
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   AudioSendStream* send_stream = new AudioSendStream(config);
   {
     rtc::CritScope lock(&network_enabled_crit_);
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
                audio_send_ssrcs_.end());
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
 
     if (!network_enabled_)
       send_stream->SignalNetworkState(kNetworkDown);
   }
   return send_stream;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(send_stream != nullptr);
 
   send_stream->Stop();
 
   webrtc::internal::AudioSendStream* audio_send_stream =
       static_cast<webrtc::internal::AudioSendStream*>(send_stream);
   {
     WriteLockScoped write_lock(*send_crit_);
     size_t num_deleted = audio_send_ssrcs_.erase(
         audio_send_stream->config().rtp.ssrc);
     RTC_DCHECK(num_deleted == 1);
   }
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       channel_group_->GetRemoteBitrateEstimator(), config);
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                audio_receive_ssrcs_.end());
     audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     ConfigureSync(config.sync_group);
   }
   return receive_stream;
 }
 
 void Call::DestroyAudioReceiveStream(
     webrtc::AudioReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   webrtc::internal::AudioReceiveStream* audio_receive_stream =
       static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
   {
     WriteLockScoped write_lock(*receive_crit_);
     size_t num_deleted = audio_receive_ssrcs_.erase(
         audio_receive_stream->config().rtp.remote_ssrc);
     RTC_DCHECK(num_deleted == 1);
     const std::string& sync_group = audio_receive_stream->config().sync_group;
     const auto it = sync_stream_mapping_.find(sync_group);
     if (it != sync_stream_mapping_.end() &&
         it->second == audio_receive_stream) {
       sync_stream_mapping_.erase(it);
       ConfigureSync(sync_group);
     }
   }
   delete audio_receive_stream;
 }
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   VideoSendStream* send_stream = new VideoSendStream(num_cpu_cores_,
       module_process_thread_.get(), channel_group_.get(),
       rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config,
       suspended_video_send_ssrcs_);
 
   // This needs to be taken before send_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*send_crit_);
   for (uint32_t ssrc : config.rtp.ssrcs) {
     RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
     video_send_ssrcs_[ssrc] = send_stream;
   }
   video_send_streams_.insert(send_stream);
 
   if (event_log_)
     event_log_->LogVideoSendStreamConfig(config);
 
   if (!network_enabled_)
     send_stream->SignalNetworkState(kNetworkDown);
   return send_stream;
 }
 
 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
   RTC_DCHECK(send_stream != nullptr);
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   send_stream->Stop();
 
   VideoSendStream* send_stream_impl = nullptr;
   {
     WriteLockScoped write_lock(*send_crit_);
     auto it = video_send_ssrcs_.begin();
     while (it != video_send_ssrcs_.end()) {
       if (it->second == static_cast<VideoSendStream*>(send_stream)) {
         send_stream_impl = it->second;
@@ skipping 13 matching lines @@
        ++it) {
     suspended_video_send_ssrcs_[it->first] = it->second;
   }
 
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     const webrtc::VideoReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       num_cpu_cores_, channel_group_.get(),
       rtc::AtomicOps::Increment(&next_channel_id_), config,
       config_.voice_engine);
 
   // This needs to be taken before receive_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*receive_crit_);
   RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ skipping 13 matching lines @@
 
   if (event_log_)
     event_log_->LogVideoReceiveStreamConfig(config);
 
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl = nullptr;
   {
     WriteLockScoped write_lock(*receive_crit_);
     // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
     // separate SSRC there can be either one or two.
     auto it = video_receive_ssrcs_.begin();
     while (it != video_receive_ssrcs_.end()) {
       if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
         if (receive_stream_impl != nullptr)
           RTC_DCHECK(receive_stream_impl == it->second);
         receive_stream_impl = it->second;
         video_receive_ssrcs_.erase(it++);
       } else {
         ++it;
       }
     }
     video_receive_streams_.erase(receive_stream_impl);
     RTC_CHECK(receive_stream_impl != nullptr);
     ConfigureSync(receive_stream_impl->config().sync_group);
   }
   delete receive_stream_impl;
 }
 
 Call::Stats Call::GetStats() const {
+  // TODO(solenberg): Some test cases in EndToEndTest use this from a different
+  // thread. Re-enable once that is fixed.
+  // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
   channel_group_->GetBitrateController()->AvailableBandwidth(&send_bandwidth);
   std::vector<unsigned int> ssrcs;
   uint32_t recv_bandwidth = 0;
   channel_group_->GetRemoteBitrateEstimator()->LatestEstimate(&ssrcs,
                                                               &recv_bandwidth);
   stats.send_bandwidth_bps = send_bandwidth;
   stats.recv_bandwidth_bps = recv_bandwidth;
   stats.pacer_delay_ms = channel_group_->GetPacerQueuingDelayMs();
   {
     ReadLockScoped read_lock(*send_crit_);
     // TODO(solenberg): Add audio send streams.
     for (const auto& kv : video_send_ssrcs_) {
       int rtt_ms = kv.second->GetRtt();
       if (rtt_ms > 0)
         stats.rtt_ms = rtt_ms;
     }
   }
   return stats;
 }
 
 void Call::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
   if (bitrate_config.max_bitrate_bps != -1)
     RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
   if (config_.bitrate_config.min_bitrate_bps ==
           bitrate_config.min_bitrate_bps &&
       (bitrate_config.start_bitrate_bps <= 0 ||
        config_.bitrate_config.start_bitrate_bps ==
            bitrate_config.start_bitrate_bps) &&
       config_.bitrate_config.max_bitrate_bps ==
           bitrate_config.max_bitrate_bps) {
     // Nothing new to set, early abort to avoid encoder reconfigurations.
     return;
   }
   config_.bitrate_config = bitrate_config;
   channel_group_->SetBweBitrates(bitrate_config.min_bitrate_bps,
                                  bitrate_config.start_bitrate_bps,
                                  bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

[pbos-webrtc, 2015/10/16 12:58:59]

(I think this should also be on a separate network thread. This doesn't have
anything to do with configuration.)

[the sun, 2015/10/19 07:55:43]

Is there anything you want me to do about that?

   // Take crit for entire function, it needs to be held while updating streams
   // to guarantee a consistent state across streams.
   rtc::CritScope lock(&network_enabled_crit_);
   network_enabled_ = state == kNetworkUp;
   channel_group_->SignalNetworkState(state);
   {
     ReadLockScoped write_lock(*send_crit_);
     for (auto& kv : audio_send_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
     for (auto& kv : video_send_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
   }
   {
     ReadLockScoped write_lock(*receive_crit_);
     for (auto& kv : video_receive_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
   }
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());

[pbos-webrtc, 2015/10/16 12:58:59]

same here

   channel_group_->OnSentPacket(sent_packet);
 }
 
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
   if (config_.voice_engine == nullptr || sync_group.empty())
     return;
 
   AudioReceiveStream* sync_audio_stream = nullptr;
   // Find existing audio stream.
@@ skipping 103 matching lines @@
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverPacket(
     MediaType media_type,
     const uint8_t* packet,
     size_t length,
     const PacketTime& packet_time) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc