/* * 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 "pc/channel_manager.h" #include #include "absl/algorithm/container.h" #include "absl/memory/memory.h" #include "absl/strings/match.h" #include "api/media_types.h" #include "api/sequence_checker.h" #include "media/base/media_constants.h" #include "pc/channel.h" #include "rtc_base/checks.h" #include "rtc_base/location.h" #include "rtc_base/trace_event.h" namespace cricket { // static std::unique_ptr ChannelManager::Create( MediaEngineInterface* media_engine, rtc::UniqueRandomIdGenerator* ssrc_generator, bool enable_rtx, rtc::Thread* worker_thread, rtc::Thread* network_thread) { RTC_DCHECK(network_thread); RTC_DCHECK(worker_thread); return absl::WrapUnique(new ChannelManager( media_engine, ssrc_generator, enable_rtx, worker_thread, network_thread)); } ChannelManager::ChannelManager(MediaEngineInterface* media_engine, rtc::UniqueRandomIdGenerator* ssrc_generator, bool enable_rtx, rtc::Thread* worker_thread, rtc::Thread* network_thread) : media_engine_(media_engine), ssrc_generator_(ssrc_generator), signaling_thread_(rtc::Thread::Current()), worker_thread_(worker_thread), network_thread_(network_thread) { RTC_DCHECK_RUN_ON(signaling_thread_); RTC_DCHECK(worker_thread_); RTC_DCHECK(network_thread_); if (media_engine_) { // TODO(tommi): Change VoiceEngine to do ctor time initialization so that // this isn't necessary. worker_thread_->Invoke(RTC_FROM_HERE, [&] { media_engine_->Init(); }); } } ChannelManager::~ChannelManager() { RTC_DCHECK_RUN_ON(signaling_thread_); } std::unique_ptr ChannelManager::CreateVoiceChannel( webrtc::Call* call, const MediaConfig& media_config, absl::string_view mid, bool srtp_required, const webrtc::CryptoOptions& crypto_options, const AudioOptions& options) { RTC_DCHECK(call); RTC_DCHECK(media_engine_); // TODO(bugs.webrtc.org/11992): Remove this workaround after updates in // PeerConnection and add the expectation that we're already on the right // thread. if (!worker_thread_->IsCurrent()) { return worker_thread_->Invoke>( RTC_FROM_HERE, [&] { return CreateVoiceChannel(call, media_config, mid, srtp_required, crypto_options, options); }); } RTC_DCHECK_RUN_ON(worker_thread_); VoiceMediaChannel* media_channel = media_engine_->voice().CreateMediaChannel( call, media_config, options, crypto_options); if (!media_channel) { return nullptr; } auto voice_channel = std::make_unique( worker_thread_, network_thread_, signaling_thread_, absl::WrapUnique(media_channel), mid, srtp_required, crypto_options, ssrc_generator_); return voice_channel; } std::unique_ptr ChannelManager::CreateVideoChannel( webrtc::Call* call, const MediaConfig& media_config, absl::string_view mid, bool srtp_required, const webrtc::CryptoOptions& crypto_options, const VideoOptions& options, webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) { RTC_DCHECK(call); RTC_DCHECK(media_engine_); // TODO(bugs.webrtc.org/11992): Remove this workaround after updates in // PeerConnection and add the expectation that we're already on the right // thread. if (!worker_thread_->IsCurrent()) { return worker_thread_->Invoke>( RTC_FROM_HERE, [&] { return CreateVideoChannel(call, media_config, mid, srtp_required, crypto_options, options, video_bitrate_allocator_factory); }); } RTC_DCHECK_RUN_ON(worker_thread_); VideoMediaChannel* media_channel = media_engine_->video().CreateMediaChannel( call, media_config, options, crypto_options, video_bitrate_allocator_factory); if (!media_channel) { return nullptr; } auto video_channel = std::make_unique( worker_thread_, network_thread_, signaling_thread_, absl::WrapUnique(media_channel), mid, srtp_required, crypto_options, ssrc_generator_); return video_channel; } } // namespace cricket