Add RTC_ prefix to (D)CHECKs and related macros.

webrtc/video/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.
  */
 
@@ skipping 126 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) {
-  DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
-  DCHECK_GE(config.bitrate_config.start_bitrate_bps,
-            config.bitrate_config.min_bitrate_bps);
+  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) {
-    DCHECK_GE(config.bitrate_config.max_bitrate_bps,
-              config.bitrate_config.start_bitrate_bps);
+    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();
 
   SetBitrateControllerConfig(config_.bitrate_config);
 }
 
 Call::~Call() {
-  CHECK_EQ(0u, video_send_ssrcs_.size());
-  CHECK_EQ(0u, video_send_streams_.size());
-  CHECK_EQ(0u, audio_receive_ssrcs_.size());
-  CHECK_EQ(0u, video_receive_ssrcs_.size());
-  CHECK_EQ(0u, video_receive_streams_.size());
+  RTC_CHECK_EQ(0u, video_send_ssrcs_.size());
+  RTC_CHECK_EQ(0u, video_send_streams_.size());
+  RTC_CHECK_EQ(0u, audio_receive_ssrcs_.size());
+  RTC_CHECK_EQ(0u, video_receive_ssrcs_.size());
+  RTC_CHECK_EQ(0u, video_receive_streams_.size());
 
   module_process_thread_->Stop();
   Trace::ReturnTrace();
 }
 
 PacketReceiver* Call::Receiver() { return this; }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   return nullptr;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   LOG(LS_INFO) << "CreateAudioReceiveStream: " << config.ToString();
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       channel_group_->GetRemoteBitrateEstimator(), config);
   {
     WriteLockScoped write_lock(*receive_crit_);
-    DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
-        audio_receive_ssrcs_.end());
+    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");
-  DCHECK(receive_stream != nullptr);
+  RTC_DCHECK(receive_stream != nullptr);
   AudioReceiveStream* audio_receive_stream =
       static_cast<AudioReceiveStream*>(receive_stream);
   {
     WriteLockScoped write_lock(*receive_crit_);
     size_t num_deleted = audio_receive_ssrcs_.erase(
         audio_receive_stream->config().rtp.remote_ssrc);
-    DCHECK(num_deleted == 1);
+    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");
   LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
-  DCHECK(!config.rtp.ssrcs.empty());
+  RTC_DCHECK(!config.rtp.ssrcs.empty());
 
   // 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) {
-    DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
+    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");
-  DCHECK(send_stream != nullptr);
+  RTC_DCHECK(send_stream != nullptr);
 
   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;
         video_send_ssrcs_.erase(it++);
       } else {
         ++it;
       }
     }
     video_send_streams_.erase(send_stream_impl);
   }
-  CHECK(send_stream_impl != nullptr);
+  RTC_CHECK(send_stream_impl != nullptr);
 
   VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates();
 
   for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin();
        it != rtp_state.end();
        ++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");
   LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
   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_);
-  DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
-      video_receive_ssrcs_.end());
+  RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
+             video_receive_ssrcs_.end());
   video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
   // TODO(pbos): Configure different RTX payloads per receive payload.
   VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
       config.rtp.rtx.begin();
   if (it != config.rtp.rtx.end())
     video_receive_ssrcs_[it->second.ssrc] = receive_stream;
   video_receive_streams_.insert(receive_stream);
 
   ConfigureSync(config.sync_group);
 
   if (!network_enabled_)
     receive_stream->SignalNetworkState(kNetworkDown);
 
   if (event_log_)
     event_log_->LogVideoReceiveStreamConfig(config);
 
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
-  DCHECK(receive_stream != nullptr);
+  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)
-          DCHECK(receive_stream_impl == it->second);
+          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);
-    CHECK(receive_stream_impl != nullptr);
+    RTC_CHECK(receive_stream_impl != nullptr);
     ConfigureSync(receive_stream_impl->config().sync_group);
   }
   delete receive_stream_impl;
 }
 
 Call::Stats Call::GetStats() const {
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
   channel_group_->GetBitrateController()->AvailableBandwidth(&send_bandwidth);
@@ skipping 11 matching lines @@
       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");
-  DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
+  RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
   if (bitrate_config.max_bitrate_bps != -1)
-    DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
+    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;
   }
@@ skipping 151 matching lines @@
     size_t length,
     const PacketTime& packet_time) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc