/* * Copyright 2015 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. */ // This file contains classes that implement RtpSenderInterface. // An RtpSender associates a MediaStreamTrackInterface with an underlying // transport (provided by AudioProviderInterface/VideoProviderInterface) #ifndef TG_PC_RTP_SENDER_H_ #define TG_PC_RTP_SENDER_H_ #include #include #include #include "api/media_stream_interface.h" #include "api/rtp_sender_interface.h" #include "media/base/audio_source.h" #include "media/base/media_channel.h" #include "pc/dtmf_sender.h" #include "rtc_base/critical_section.h" #include "pc/rtp_sender.h" namespace webrtc { class StatsCollector; bool TgUnimplementedRtpParameterHasValue(const RtpParameters& parameters); // TgLocalAudioSinkAdapter receives data callback as a sink to the local // AudioTrack, and passes the data to the sink of AudioSource. class TgLocalAudioSinkAdapter : public AudioTrackSinkInterface, public cricket::AudioSource { public: TgLocalAudioSinkAdapter(); virtual ~TgLocalAudioSinkAdapter(); private: // AudioSinkInterface implementation. void OnData(const void* audio_data, int bits_per_sample, int sample_rate, size_t number_of_channels, size_t number_of_frames) override; // cricket::AudioSource implementation. void SetSink(cricket::AudioSource::Sink* sink) override; cricket::AudioSource::Sink* sink_; // Critical section protecting |sink_|. rtc::CriticalSection lock_; }; class TgAudioRtpSender : public DtmfProviderInterface, public RtpSenderBase { public: // Construct an RtpSender for audio with the given sender ID. // The sender is initialized with no track to send and no associated streams. // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called // at the appropriate times. // If |set_streams_observer| is not null, it is invoked when SetStreams() // is called. |set_streams_observer| is not owned by this object. If not // null, it must be valid at least until this sender becomes stopped. static rtc::scoped_refptr Create( rtc::Thread* worker_thread, const std::string& id, SetStreamsObserver* set_streams_observer); virtual ~TgAudioRtpSender(); // DtmfSenderProvider implementation. bool CanInsertDtmf() override; bool InsertDtmf(int code, int duration) override; sigslot::signal0<>* GetOnDestroyedSignal() override; // ObserverInterface implementation. void OnChanged() override; cricket::MediaType media_type() const override { return cricket::MEDIA_TYPE_AUDIO; } std::string track_kind() const override { return MediaStreamTrackInterface::kAudioKind; } rtc::scoped_refptr GetDtmfSender() const override; protected: TgAudioRtpSender(rtc::Thread* worker_thread, const std::string& id, SetStreamsObserver* set_streams_observer); void SetSend() override; void ClearSend() override; // Hooks to allow custom logic when tracks are attached and detached. void AttachTrack() override; void DetachTrack() override; void AddTrackToStats() override; void RemoveTrackFromStats() override; private: cricket::VoiceMediaChannel* voice_media_channel() { return static_cast(media_channel_); } rtc::scoped_refptr audio_track() const { return rtc::scoped_refptr( static_cast(track_.get())); } sigslot::signal0<> SignalDestroyed; rtc::scoped_refptr dtmf_sender_proxy_; bool cached_track_enabled_ = false; // Used to pass the data callback from the |track_| to the other end of // cricket::AudioSource. std::unique_ptr sink_adapter_; }; class TgVideoRtpSender : public RtpSenderBase { public: // Construct an RtpSender for video with the given sender ID. // The sender is initialized with no track to send and no associated streams. // If |set_streams_observer| is not null, it is invoked when SetStreams() // is called. |set_streams_observer| is not owned by this object. If not // null, it must be valid at least until this sender becomes stopped. static rtc::scoped_refptr Create( rtc::Thread* worker_thread, const std::string& id, SetStreamsObserver* set_streams_observer); virtual ~TgVideoRtpSender(); // ObserverInterface implementation void OnChanged() override; cricket::MediaType media_type() const override { return cricket::MEDIA_TYPE_VIDEO; } std::string track_kind() const override { return MediaStreamTrackInterface::kVideoKind; } rtc::scoped_refptr GetDtmfSender() const override; protected: TgVideoRtpSender(rtc::Thread* worker_thread, const std::string& id, SetStreamsObserver* set_streams_observer); void SetSend() override; void ClearSend() override; // Hook to allow custom logic when tracks are attached. void AttachTrack() override; private: cricket::VideoMediaChannel* video_media_channel() { return static_cast(media_channel_); } rtc::scoped_refptr video_track() const { return rtc::scoped_refptr( static_cast(track_.get())); } VideoTrackInterface::ContentHint cached_track_content_hint_ = VideoTrackInterface::ContentHint::kNone; }; } // namespace webrtc #endif // PC_RTP_SENDER_H_