Replace ASSERT by RTC_DCHECK in all non-test code.

webrtc/pc/channelmanager.cc

 /*
  *  Copyright 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.
  */
 
 #include "webrtc/pc/channelmanager.h"
 
 #include <algorithm>
 
 #include "webrtc/api/mediacontroller.h"
 #include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/media/base/device.h"
 #include "webrtc/media/base/rtpdataengine.h"
 #include "webrtc/pc/srtpfilter.h"
 
 namespace cricket {
@@ skipping 120 matching lines @@
     RtpHeaderExtensions* ext) const {
   *ext = media_engine_->GetVideoCapabilities().header_extensions;
 }
 
 void ChannelManager::GetSupportedDataCodecs(
     std::vector<DataCodec>* codecs) const {
   *codecs = data_media_engine_->data_codecs();
 }
 
 bool ChannelManager::Init() {
-  ASSERT(!initialized_);
+  RTC_DCHECK(!initialized_);
   if (initialized_) {
     return false;
   }
   RTC_DCHECK(network_thread_);
   RTC_DCHECK(worker_thread_);
   if (!network_thread_->IsCurrent()) {
     // Do not allow invoking calls to other threads on the network thread.
     network_thread_->Invoke<bool>(
         RTC_FROM_HERE,
         rtc::Bind(&rtc::Thread::SetAllowBlockingCalls, network_thread_, false));
   }
 
   initialized_ = worker_thread_->Invoke<bool>(
       RTC_FROM_HERE, Bind(&ChannelManager::InitMediaEngine_w, this));
-  ASSERT(initialized_);
+  RTC_DCHECK(initialized_);
   return initialized_;
 }
 
 bool ChannelManager::InitMediaEngine_w() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   return media_engine_->Init();
 }
 
 void ChannelManager::Terminate() {
-  ASSERT(initialized_);
+  RTC_DCHECK(initialized_);
   if (!initialized_) {
     return;
   }
   worker_thread_->Invoke<void>(RTC_FROM_HERE,
                                Bind(&ChannelManager::Terminate_w, this));
   initialized_ = false;
 }
 
 void ChannelManager::DestructorDeletes_w() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   media_engine_.reset(NULL);
 }
 
 void ChannelManager::Terminate_w() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   // Need to destroy the voice/video channels
   while (!video_channels_.empty()) {
     DestroyVideoChannel_w(video_channels_.back());
   }
   while (!voice_channels_.empty()) {
     DestroyVoiceChannel_w(voice_channels_.back());
   }
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel(
@@ skipping 11 matching lines @@
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required,
     const AudioOptions& options) {
-  ASSERT(initialized_);
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-  ASSERT(nullptr != media_controller);
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(nullptr != media_controller);
   VoiceMediaChannel* media_channel = media_engine_->CreateChannel(
       media_controller->call_w(), media_controller->config(), options);
   if (!media_channel)
     return nullptr;
 
   VoiceChannel* voice_channel = new VoiceChannel(
       worker_thread_, network_thread_, media_engine_.get(), media_channel,
       transport_controller, content_name, rtcp, srtp_required);
   voice_channel->SetCryptoOptions(crypto_options_);
   if (!voice_channel->Init_w(bundle_transport_name)) {
     delete voice_channel;
     return nullptr;
   }
   voice_channels_.push_back(voice_channel);
   return voice_channel;
 }
 
 void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
   TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel");
   if (voice_channel) {
     worker_thread_->Invoke<void>(
         RTC_FROM_HERE,
         Bind(&ChannelManager::DestroyVoiceChannel_w, this, voice_channel));
   }
 }
 
 void ChannelManager::DestroyVoiceChannel_w(VoiceChannel* voice_channel) {
   TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel_w");
   // Destroy voice channel.
-  ASSERT(initialized_);
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   VoiceChannels::iterator it = std::find(voice_channels_.begin(),
       voice_channels_.end(), voice_channel);
-  ASSERT(it != voice_channels_.end());
+  RTC_DCHECK(it != voice_channels_.end());
   if (it == voice_channels_.end())
     return;
   voice_channels_.erase(it);
   delete voice_channel;
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required,
     const VideoOptions& options) {
   return worker_thread_->Invoke<VideoChannel*>(
       RTC_FROM_HERE, Bind(&ChannelManager::CreateVideoChannel_w, this,
                           media_controller, transport_controller, content_name,
                           bundle_transport_name, rtcp, srtp_required, options));
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required,
     const VideoOptions& options) {
-  ASSERT(initialized_);
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-  ASSERT(nullptr != media_controller);
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(nullptr != media_controller);
   VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
       media_controller->call_w(), media_controller->config(), options);
   if (media_channel == NULL) {
     return NULL;
   }
 
   VideoChannel* video_channel =
       new VideoChannel(worker_thread_, network_thread_, media_channel,
                        transport_controller, content_name, rtcp, srtp_required);
   video_channel->SetCryptoOptions(crypto_options_);
@@ skipping 10 matching lines @@
   if (video_channel) {
     worker_thread_->Invoke<void>(
         RTC_FROM_HERE,
         Bind(&ChannelManager::DestroyVideoChannel_w, this, video_channel));
   }
 }
 
 void ChannelManager::DestroyVideoChannel_w(VideoChannel* video_channel) {
   TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel_w");
   // Destroy video channel.
-  ASSERT(initialized_);
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   VideoChannels::iterator it = std::find(video_channels_.begin(),
       video_channels_.end(), video_channel);
-  ASSERT(it != video_channels_.end());
+  RTC_DCHECK(it != video_channels_.end());
   if (it == video_channels_.end())
     return;
 
   video_channels_.erase(it);
   delete video_channel;
 }
 
 RtpDataChannel* ChannelManager::CreateRtpDataChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required) {
   return worker_thread_->Invoke<RtpDataChannel*>(
       RTC_FROM_HERE, Bind(&ChannelManager::CreateRtpDataChannel_w, this,
                           media_controller, transport_controller, content_name,
                           bundle_transport_name, rtcp, srtp_required));
 }
 
 RtpDataChannel* ChannelManager::CreateRtpDataChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
     const std::string* bundle_transport_name,
     bool rtcp,
     bool srtp_required) {
   // This is ok to alloc from a thread other than the worker thread.
-  ASSERT(initialized_);
+  RTC_DCHECK(initialized_);
   MediaConfig config;
   if (media_controller) {
     config = media_controller->config();
   }
   DataMediaChannel* media_channel = data_media_engine_->CreateChannel(config);
   if (!media_channel) {
     LOG(LS_WARNING) << "Failed to create RTP data channel.";
     return nullptr;
   }
 
@@ skipping 15 matching lines @@
   if (data_channel) {
     worker_thread_->Invoke<void>(
         RTC_FROM_HERE,
         Bind(&ChannelManager::DestroyRtpDataChannel_w, this, data_channel));
   }
 }
 
 void ChannelManager::DestroyRtpDataChannel_w(RtpDataChannel* data_channel) {
   TRACE_EVENT0("webrtc", "ChannelManager::DestroyRtpDataChannel_w");
   // Destroy data channel.
-  ASSERT(initialized_);
+  RTC_DCHECK(initialized_);
   RtpDataChannels::iterator it =
       std::find(data_channels_.begin(), data_channels_.end(), data_channel);
-  ASSERT(it != data_channels_.end());
+  RTC_DCHECK(it != data_channels_.end());
   if (it == data_channels_.end())
     return;
 
   data_channels_.erase(it);
   delete data_channel;
 }
 
 bool ChannelManager::StartAecDump(rtc::PlatformFile file,
                                   int64_t max_size_bytes) {
   return worker_thread_->Invoke<bool>(
       RTC_FROM_HERE, Bind(&MediaEngineInterface::StartAecDump,
                           media_engine_.get(), file, max_size_bytes));
 }
 
 void ChannelManager::StopAecDump() {
   worker_thread_->Invoke<void>(
       RTC_FROM_HERE,
       Bind(&MediaEngineInterface::StopAecDump, media_engine_.get()));
 }
 
 }  // namespace cricket