| /* |
| * 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/channelmanager.h" |
| |
| #include <algorithm> |
| #include <utility> |
| |
| #include "media/base/device.h" |
| #include "media/base/rtpdataengine.h" |
| #include "pc/srtpfilter.h" |
| #include "rtc_base/bind.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/stringencode.h" |
| #include "rtc_base/stringutils.h" |
| #include "rtc_base/trace_event.h" |
| |
| namespace cricket { |
| |
| |
| using rtc::Bind; |
| |
| ChannelManager::ChannelManager(std::unique_ptr<MediaEngineInterface> me, |
| std::unique_ptr<DataEngineInterface> dme, |
| rtc::Thread* thread) { |
| Construct(std::move(me), std::move(dme), thread, thread); |
| } |
| |
| ChannelManager::ChannelManager(std::unique_ptr<MediaEngineInterface> me, |
| rtc::Thread* worker_thread, |
| rtc::Thread* network_thread) { |
| Construct(std::move(me), |
| std::unique_ptr<DataEngineInterface>(new RtpDataEngine()), |
| worker_thread, network_thread); |
| } |
| |
| void ChannelManager::Construct(std::unique_ptr<MediaEngineInterface> me, |
| std::unique_ptr<DataEngineInterface> dme, |
| rtc::Thread* worker_thread, |
| rtc::Thread* network_thread) { |
| media_engine_ = std::move(me); |
| data_media_engine_ = std::move(dme); |
| initialized_ = false; |
| main_thread_ = rtc::Thread::Current(); |
| worker_thread_ = worker_thread; |
| network_thread_ = network_thread; |
| capturing_ = false; |
| enable_rtx_ = false; |
| } |
| |
| ChannelManager::~ChannelManager() { |
| if (initialized_) { |
| Terminate(); |
| } |
| // The media engine needs to be deleted on the worker thread for thread safe |
| // destruction, |
| worker_thread_->Invoke<void>( |
| RTC_FROM_HERE, Bind(&ChannelManager::DestructorDeletes_w, this)); |
| } |
| |
| bool ChannelManager::SetVideoRtxEnabled(bool enable) { |
| // To be safe, this call is only allowed before initialization. Apps like |
| // Flute only have a singleton ChannelManager and we don't want this flag to |
| // be toggled between calls or when there's concurrent calls. We expect apps |
| // to enable this at startup and retain that setting for the lifetime of the |
| // app. |
| if (!initialized_) { |
| enable_rtx_ = enable; |
| return true; |
| } else { |
| LOG(LS_WARNING) << "Cannot toggle rtx after initialization!"; |
| return false; |
| } |
| } |
| |
| void ChannelManager::GetSupportedAudioSendCodecs( |
| std::vector<AudioCodec>* codecs) const { |
| if (!media_engine_) { |
| return; |
| } |
| *codecs = media_engine_->audio_send_codecs(); |
| } |
| |
| void ChannelManager::GetSupportedAudioReceiveCodecs( |
| std::vector<AudioCodec>* codecs) const { |
| if (!media_engine_) { |
| return; |
| } |
| *codecs = media_engine_->audio_recv_codecs(); |
| } |
| |
| void ChannelManager::GetSupportedAudioRtpHeaderExtensions( |
| RtpHeaderExtensions* ext) const { |
| if (!media_engine_) { |
| return; |
| } |
| *ext = media_engine_->GetAudioCapabilities().header_extensions; |
| } |
| |
| void ChannelManager::GetSupportedVideoCodecs( |
| std::vector<VideoCodec>* codecs) const { |
| if (!media_engine_) { |
| return; |
| } |
| codecs->clear(); |
| |
| std::vector<VideoCodec> video_codecs = media_engine_->video_codecs(); |
| for (const auto& video_codec : video_codecs) { |
| if (!enable_rtx_ && |
| _stricmp(kRtxCodecName, video_codec.name.c_str()) == 0) { |
| continue; |
| } |
| codecs->push_back(video_codec); |
| } |
| } |
| |
| void ChannelManager::GetSupportedVideoRtpHeaderExtensions( |
| RtpHeaderExtensions* ext) const { |
| if (!media_engine_) { |
| return; |
| } |
| *ext = media_engine_->GetVideoCapabilities().header_extensions; |
| } |
| |
| void ChannelManager::GetSupportedDataCodecs( |
| std::vector<DataCodec>* codecs) const { |
| if (!data_media_engine_) { |
| return; |
| } |
| *codecs = data_media_engine_->data_codecs(); |
| } |
| |
| bool ChannelManager::Init() { |
| 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)); |
| RTC_DCHECK(initialized_); |
| return initialized_; |
| } |
| |
| bool ChannelManager::InitMediaEngine_w() { |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| if (media_engine_) { |
| return media_engine_->Init(); |
| } |
| return true; |
| } |
| |
| void ChannelManager::Terminate() { |
| RTC_DCHECK(initialized_); |
| if (!initialized_) { |
| return; |
| } |
| worker_thread_->Invoke<void>(RTC_FROM_HERE, |
| Bind(&ChannelManager::Terminate_w, this)); |
| initialized_ = false; |
| } |
| |
| void ChannelManager::DestructorDeletes_w() { |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| media_engine_.reset(nullptr); |
| } |
| |
| void ChannelManager::Terminate_w() { |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| // Need to destroy the voice/video channels |
| video_channels_.clear(); |
| voice_channels_.clear(); |
| } |
| |
| VoiceChannel* ChannelManager::CreateVoiceChannel( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_transport, |
| DtlsTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const AudioOptions& options) { |
| return worker_thread_->Invoke<VoiceChannel*>( |
| RTC_FROM_HERE, |
| Bind(&ChannelManager::CreateVoiceChannel_w, this, call, media_config, |
| rtp_transport, rtcp_transport, rtp_transport, rtcp_transport, |
| signaling_thread, content_name, srtp_required, options)); |
| } |
| |
| VoiceChannel* ChannelManager::CreateVoiceChannel( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| rtc::PacketTransportInternal* rtp_transport, |
| rtc::PacketTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const AudioOptions& options) { |
| return worker_thread_->Invoke<VoiceChannel*>( |
| RTC_FROM_HERE, |
| Bind(&ChannelManager::CreateVoiceChannel_w, this, call, media_config, |
| nullptr, nullptr, rtp_transport, rtcp_transport, signaling_thread, |
| content_name, srtp_required, options)); |
| } |
| |
| VoiceChannel* ChannelManager::CreateVoiceChannel_w( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const AudioOptions& options) { |
| RTC_DCHECK(initialized_); |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| RTC_DCHECK(nullptr != call); |
| if (!media_engine_) { |
| return nullptr; |
| } |
| |
| VoiceMediaChannel* media_channel = media_engine_->CreateChannel( |
| call, media_config, options); |
| if (!media_channel) { |
| return nullptr; |
| } |
| |
| std::unique_ptr<VoiceChannel> voice_channel( |
| new VoiceChannel(worker_thread_, network_thread_, signaling_thread, |
| media_engine_.get(), media_channel, content_name, |
| rtcp_packet_transport == nullptr, srtp_required)); |
| |
| if (!voice_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport, |
| rtp_packet_transport, rtcp_packet_transport)) { |
| return nullptr; |
| } |
| VoiceChannel* voice_channel_ptr = voice_channel.get(); |
| voice_channels_.push_back(std::move(voice_channel)); |
| return voice_channel_ptr; |
| } |
| |
| 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"); |
| RTC_DCHECK(initialized_); |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| |
| auto it = std::find_if(voice_channels_.begin(), voice_channels_.end(), |
| [&](const std::unique_ptr<VoiceChannel>& p) { |
| return p.get() == voice_channel; |
| }); |
| RTC_DCHECK(it != voice_channels_.end()); |
| if (it == voice_channels_.end()) |
| return; |
| voice_channels_.erase(it); |
| } |
| |
| VideoChannel* ChannelManager::CreateVideoChannel( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_transport, |
| DtlsTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const VideoOptions& options) { |
| return worker_thread_->Invoke<VideoChannel*>( |
| RTC_FROM_HERE, |
| Bind(&ChannelManager::CreateVideoChannel_w, this, call, media_config, |
| rtp_transport, rtcp_transport, rtp_transport, rtcp_transport, |
| signaling_thread, content_name, srtp_required, options)); |
| } |
| |
| VideoChannel* ChannelManager::CreateVideoChannel( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| rtc::PacketTransportInternal* rtp_transport, |
| rtc::PacketTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const VideoOptions& options) { |
| return worker_thread_->Invoke<VideoChannel*>( |
| RTC_FROM_HERE, |
| Bind(&ChannelManager::CreateVideoChannel_w, this, call, media_config, |
| nullptr, nullptr, rtp_transport, rtcp_transport, signaling_thread, |
| content_name, srtp_required, options)); |
| } |
| |
| VideoChannel* ChannelManager::CreateVideoChannel_w( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const VideoOptions& options) { |
| RTC_DCHECK(initialized_); |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| RTC_DCHECK(nullptr != call); |
| VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel( |
| call, media_config, options); |
| if (!media_channel) { |
| return nullptr; |
| } |
| |
| std::unique_ptr<VideoChannel> video_channel(new VideoChannel( |
| worker_thread_, network_thread_, signaling_thread, media_channel, |
| content_name, rtcp_packet_transport == nullptr, srtp_required)); |
| if (!video_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport, |
| rtp_packet_transport, rtcp_packet_transport)) { |
| return nullptr; |
| } |
| VideoChannel* video_channel_ptr = video_channel.get(); |
| video_channels_.push_back(std::move(video_channel)); |
| return video_channel_ptr; |
| } |
| |
| void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) { |
| TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel"); |
| 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"); |
| RTC_DCHECK(initialized_); |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| |
| auto it = std::find_if(video_channels_.begin(), video_channels_.end(), |
| [&](const std::unique_ptr<VideoChannel>& p) { |
| return p.get() == video_channel; |
| }); |
| RTC_DCHECK(it != video_channels_.end()); |
| if (it == video_channels_.end()) |
| return; |
| |
| video_channels_.erase(it); |
| } |
| |
| RtpDataChannel* ChannelManager::CreateRtpDataChannel( |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_transport, |
| DtlsTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required) { |
| return worker_thread_->Invoke<RtpDataChannel*>( |
| RTC_FROM_HERE, Bind(&ChannelManager::CreateRtpDataChannel_w, this, |
| media_config, rtp_transport, rtcp_transport, |
| signaling_thread, content_name, srtp_required)); |
| } |
| |
| RtpDataChannel* ChannelManager::CreateRtpDataChannel_w( |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_transport, |
| DtlsTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required) { |
| // This is ok to alloc from a thread other than the worker thread. |
| RTC_DCHECK(initialized_); |
| DataMediaChannel* media_channel |
| = data_media_engine_->CreateChannel(media_config); |
| if (!media_channel) { |
| LOG(LS_WARNING) << "Failed to create RTP data channel."; |
| return nullptr; |
| } |
| |
| std::unique_ptr<RtpDataChannel> data_channel(new RtpDataChannel( |
| worker_thread_, network_thread_, signaling_thread, media_channel, |
| content_name, rtcp_transport == nullptr, srtp_required)); |
| if (!data_channel->Init_w(rtp_transport, rtcp_transport, rtp_transport, |
| rtcp_transport)) { |
| LOG(LS_WARNING) << "Failed to init data channel."; |
| return nullptr; |
| } |
| RtpDataChannel* data_channel_ptr = data_channel.get(); |
| data_channels_.push_back(std::move(data_channel)); |
| return data_channel_ptr; |
| } |
| |
| void ChannelManager::DestroyRtpDataChannel(RtpDataChannel* data_channel) { |
| TRACE_EVENT0("webrtc", "ChannelManager::DestroyRtpDataChannel"); |
| 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"); |
| RTC_DCHECK(initialized_); |
| |
| auto it = std::find_if(data_channels_.begin(), data_channels_.end(), |
| [&](const std::unique_ptr<RtpDataChannel>& p) { |
| return p.get() == data_channel; |
| }); |
| RTC_DCHECK(it != data_channels_.end()); |
| if (it == data_channels_.end()) |
| return; |
| |
| data_channels_.erase(it); |
| } |
| |
| 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 |