/* * 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 #include #include #include #include "webrtc/call.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/video/video_receive_stream.h" #include "webrtc/video/video_send_stream.h" #include "webrtc/video_engine/include/vie_base.h" #include "webrtc/video_engine/include/vie_codec.h" #include "webrtc/video_engine/include/vie_rtp_rtcp.h" namespace webrtc { namespace internal { class Call : public webrtc::Call, public PacketReceiver { public: Call(webrtc::VideoEngine* video_engine, const Call::Config& config); virtual ~Call(); virtual PacketReceiver* Receiver() OVERRIDE; virtual std::vector GetVideoCodecs() OVERRIDE; virtual VideoSendStream::Config GetDefaultSendConfig() OVERRIDE; virtual VideoSendStream* CreateSendStream( const VideoSendStream::Config& config) OVERRIDE; virtual SendStreamState* DestroySendStream( webrtc::VideoSendStream* send_stream) OVERRIDE; virtual VideoReceiveStream::Config GetDefaultReceiveConfig() OVERRIDE; virtual VideoReceiveStream* CreateReceiveStream( const VideoReceiveStream::Config& config) OVERRIDE; virtual void DestroyReceiveStream( webrtc::VideoReceiveStream* receive_stream) OVERRIDE; virtual uint32_t SendBitrateEstimate() OVERRIDE; virtual uint32_t ReceiveBitrateEstimate() OVERRIDE; virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE; private: bool DeliverRtcp(const uint8_t* packet, size_t length); bool DeliverRtp(const RTPHeader& header, const uint8_t* packet, size_t length); Call::Config config_; std::map receive_ssrcs_; scoped_ptr receive_lock_; std::map send_ssrcs_; scoped_ptr send_lock_; scoped_ptr rtp_header_parser_; webrtc::VideoEngine* video_engine_; ViERTP_RTCP* rtp_rtcp_; ViECodec* codec_; DISALLOW_COPY_AND_ASSIGN(Call); }; } // internal class TraceDispatcher : public TraceCallback { public: TraceDispatcher() : crit_(CriticalSectionWrapper::CreateCriticalSection()), initialized_(false), filter_(kTraceNone) {} ~TraceDispatcher() { if (initialized_) { Trace::ReturnTrace(); VideoEngine::SetTraceCallback(NULL); } } virtual void Print(TraceLevel level, const char* message, int length) OVERRIDE { CriticalSectionScoped lock(crit_.get()); for (std::map::iterator it = callbacks_.begin(); it != callbacks_.end(); ++it) { if ((level & it->second->trace_filter) != kTraceNone) it->second->trace_callback->Print(level, message, length); } } void RegisterCallback(Call* call, Call::Config* config) { if (config->trace_callback == NULL) return; CriticalSectionScoped lock(crit_.get()); callbacks_[call] = config; filter_ |= config->trace_filter; if (filter_ != kTraceNone && !initialized_) { initialized_ = true; Trace::CreateTrace(); VideoEngine::SetTraceCallback(this); } VideoEngine::SetTraceFilter(filter_); } void DeregisterCallback(Call* call) { CriticalSectionScoped lock(crit_.get()); callbacks_.erase(call); filter_ = kTraceNone; for (std::map::iterator it = callbacks_.begin(); it != callbacks_.end(); ++it) { filter_ |= it->second->trace_filter; } VideoEngine::SetTraceFilter(filter_); } private: scoped_ptr crit_; bool initialized_; unsigned int filter_; std::map callbacks_; }; namespace internal { TraceDispatcher* global_trace_dispatcher = NULL; } // internal Call* Call::Create(const Call::Config& config) { if (internal::global_trace_dispatcher == NULL) { TraceDispatcher* dispatcher = new TraceDispatcher(); // TODO(pbos): Atomic compare and exchange. if (internal::global_trace_dispatcher == NULL) { internal::global_trace_dispatcher = dispatcher; } else { delete dispatcher; } } VideoEngine* video_engine = VideoEngine::Create(); assert(video_engine != NULL); return new internal::Call(video_engine, config); } namespace internal { Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config) : config_(config), receive_lock_(RWLockWrapper::CreateRWLock()), send_lock_(RWLockWrapper::CreateRWLock()), rtp_header_parser_(RtpHeaderParser::Create()), video_engine_(video_engine) { assert(video_engine != NULL); assert(config.send_transport != NULL); global_trace_dispatcher->RegisterCallback(this, &config_); rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine_); assert(rtp_rtcp_ != NULL); codec_ = ViECodec::GetInterface(video_engine_); assert(codec_ != NULL); } Call::~Call() { global_trace_dispatcher->DeregisterCallback(this); codec_->Release(); rtp_rtcp_->Release(); webrtc::VideoEngine::Delete(video_engine_); } PacketReceiver* Call::Receiver() { return this; } std::vector Call::GetVideoCodecs() { std::vector codecs; VideoCodec codec; for (size_t i = 0; i < static_cast(codec_->NumberOfCodecs()); ++i) { if (codec_->GetCodec(static_cast(i), codec) == 0) { codecs.push_back(codec); } } return codecs; } VideoSendStream::Config Call::GetDefaultSendConfig() { VideoSendStream::Config config; codec_->GetCodec(0, config.codec); return config; } VideoSendStream* Call::CreateSendStream(const VideoSendStream::Config& config) { assert(config.rtp.ssrcs.size() > 0); assert(config.codec.numberOfSimulcastStreams == 0 || config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size()); VideoSendStream* send_stream = new VideoSendStream( config_.send_transport, config_.overuse_detection, video_engine_, config); WriteLockScoped write_lock(*send_lock_); for (size_t i = 0; i < config.rtp.ssrcs.size(); ++i) { assert(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end()); send_ssrcs_[config.rtp.ssrcs[i]] = send_stream; } return send_stream; } SendStreamState* Call::DestroySendStream(webrtc::VideoSendStream* send_stream) { assert(send_stream != NULL); VideoSendStream* send_stream_impl = NULL; { WriteLockScoped write_lock(*send_lock_); for (std::map::iterator it = send_ssrcs_.begin(); it != send_ssrcs_.end(); ++it) { if (it->second == static_cast(send_stream)) { send_stream_impl = it->second; send_ssrcs_.erase(it); break; } } } assert(send_stream_impl != NULL); delete send_stream_impl; // TODO(pbos): Return its previous state return NULL; } VideoReceiveStream::Config Call::GetDefaultReceiveConfig() { return VideoReceiveStream::Config(); } VideoReceiveStream* Call::CreateReceiveStream( const VideoReceiveStream::Config& config) { VideoReceiveStream* receive_stream = new VideoReceiveStream(video_engine_, config, config_.send_transport); WriteLockScoped write_lock(*receive_lock_); assert(receive_ssrcs_.find(config.rtp.ssrc) == receive_ssrcs_.end()); receive_ssrcs_[config.rtp.ssrc] = receive_stream; return receive_stream; } void Call::DestroyReceiveStream(webrtc::VideoReceiveStream* receive_stream) { assert(receive_stream != NULL); VideoReceiveStream* receive_stream_impl = NULL; { WriteLockScoped write_lock(*receive_lock_); for (std::map::iterator it = receive_ssrcs_.begin(); it != receive_ssrcs_.end(); ++it) { if (it->second == static_cast(receive_stream)) { receive_stream_impl = it->second; receive_ssrcs_.erase(it); break; } } } assert(receive_stream_impl != NULL); delete receive_stream_impl; } uint32_t Call::SendBitrateEstimate() { // TODO(pbos): Return send-bitrate estimate return 0; } uint32_t Call::ReceiveBitrateEstimate() { // TODO(pbos): Return receive-bitrate estimate return 0; } bool Call::DeliverRtcp(const uint8_t* packet, size_t length) { // TODO(pbos): Figure out what channel needs it actually. // Do NOT broadcast! Also make sure it's a valid packet. bool rtcp_delivered = false; { ReadLockScoped read_lock(*receive_lock_); for (std::map::iterator it = receive_ssrcs_.begin(); it != receive_ssrcs_.end(); ++it) { if (it->second->DeliverRtcp(packet, length)) rtcp_delivered = true; } } { ReadLockScoped read_lock(*send_lock_); for (std::map::iterator it = send_ssrcs_.begin(); it != send_ssrcs_.end(); ++it) { if (it->second->DeliverRtcp(packet, length)) rtcp_delivered = true; } } return rtcp_delivered; } bool Call::DeliverRtp(const RTPHeader& header, const uint8_t* packet, size_t length) { VideoReceiveStream* receiver; { ReadLockScoped read_lock(*receive_lock_); std::map::iterator it = receive_ssrcs_.find(header.ssrc); if (it == receive_ssrcs_.end()) { // TODO(pbos): Log some warning, SSRC without receiver. return false; } receiver = it->second; } return receiver->DeliverRtp(static_cast(packet), length); } bool Call::DeliverPacket(const uint8_t* packet, size_t length) { // TODO(pbos): ExtensionMap if there are extensions. if (RtpHeaderParser::IsRtcp(packet, static_cast(length))) return DeliverRtcp(packet, length); RTPHeader rtp_header; if (!rtp_header_parser_->Parse(packet, static_cast(length), &rtp_header)) return false; return DeliverRtp(rtp_header, packet, length); } } // namespace internal } // namespace webrtc