| /* | 
 |  *  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 PC_RTP_SENDER_H_ | 
 | #define PC_RTP_SENDER_H_ | 
 |  | 
 | #include <stddef.h> | 
 | #include <stdint.h> | 
 |  | 
 | #include <memory> | 
 | #include <optional> | 
 | #include <string> | 
 | #include <vector> | 
 |  | 
 | #include "api/crypto/frame_encryptor_interface.h" | 
 | #include "api/dtls_transport_interface.h" | 
 | #include "api/dtmf_sender_interface.h" | 
 | #include "api/environment/environment.h" | 
 | #include "api/field_trials_view.h" | 
 | #include "api/frame_transformer_interface.h" | 
 | #include "api/media_stream_interface.h" | 
 | #include "api/media_types.h" | 
 | #include "api/rtc_error.h" | 
 | #include "api/rtp_parameters.h" | 
 | #include "api/rtp_sender_interface.h" | 
 | #include "api/scoped_refptr.h" | 
 | #include "api/sequence_checker.h" | 
 | #include "media/base/audio_source.h" | 
 | #include "media/base/media_channel.h" | 
 | #include "pc/dtmf_sender.h" | 
 | #include "pc/legacy_stats_collector_interface.h" | 
 | #include "rtc_base/checks.h" | 
 | #include "rtc_base/synchronization/mutex.h" | 
 | #include "rtc_base/thread.h" | 
 | #include "rtc_base/thread_annotations.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters); | 
 |  | 
 | // Internal interface used by PeerConnection. | 
 | class RtpSenderInternal : public RtpSenderInterface { | 
 |  public: | 
 |   // Sets the underlying MediaEngine channel associated with this RtpSender. | 
 |   // A VoiceMediaChannel should be used for audio RtpSenders and | 
 |   // a VideoMediaChannel should be used for video RtpSenders. | 
 |   // Must call SetMediaChannel(nullptr) before the media channel is destroyed. | 
 |   virtual void SetMediaChannel(MediaSendChannelInterface* media_channel) = 0; | 
 |  | 
 |   // Used to set the SSRC of the sender, once a local description has been set. | 
 |   // If `ssrc` is 0, this indiates that the sender should disconnect from the | 
 |   // underlying transport (this occurs if the sender isn't seen in a local | 
 |   // description). | 
 |   virtual void SetSsrc(uint32_t ssrc) = 0; | 
 |  | 
 |   virtual void set_stream_ids(const std::vector<std::string>& stream_ids) = 0; | 
 |   virtual void set_init_send_encodings( | 
 |       const std::vector<RtpEncodingParameters>& init_send_encodings) = 0; | 
 |   virtual void set_transport( | 
 |       scoped_refptr<DtlsTransportInterface> dtls_transport) = 0; | 
 |  | 
 |   virtual void Stop() = 0; | 
 |  | 
 |   // `GetParameters` and `SetParameters` operate with a transactional model. | 
 |   // Allow access to get/set parameters without invalidating transaction id. | 
 |   virtual RtpParameters GetParametersInternal() const = 0; | 
 |   virtual void SetParametersInternal(const RtpParameters& parameters, | 
 |                                      SetParametersCallback, | 
 |                                      bool blocking) = 0; | 
 |  | 
 |   // GetParameters and SetParameters will remove deactivated simulcast layers | 
 |   // and restore them on SetParameters. This is probably a Bad Idea, but we | 
 |   // do not know who depends on this behavior | 
 |   virtual RtpParameters GetParametersInternalWithAllLayers() const = 0; | 
 |   virtual RTCError SetParametersInternalWithAllLayers( | 
 |       const RtpParameters& parameters) = 0; | 
 |  | 
 |   // Additional checks that are specific to the current codec settings | 
 |   virtual RTCError CheckCodecParameters(const RtpParameters& parameters) = 0; | 
 |  | 
 |   // Returns an ID that changes every time SetTrack() is called, but | 
 |   // otherwise remains constant. Used to generate IDs for stats. | 
 |   // The special value zero means that no track is attached. | 
 |   virtual int AttachmentId() const = 0; | 
 |  | 
 |   // Disables the layers identified by the specified RIDs. | 
 |   // If the specified list is empty, this is a no-op. | 
 |   virtual RTCError DisableEncodingLayers( | 
 |       const std::vector<std::string>& rid) = 0; | 
 |  | 
 |   virtual void SetTransceiverAsStopped() = 0; | 
 |  | 
 |   // Used by the owning transceiver to inform the sender on the currently | 
 |   // selected codecs. | 
 |   virtual void SetSendCodecs(std::vector<Codec> send_codecs) = 0; | 
 |   virtual std::vector<Codec> GetSendCodecs() const = 0; | 
 |  | 
 |   virtual void NotifyFirstPacketSent() = 0; | 
 | }; | 
 |  | 
 | // Shared implementation for RtpSenderInternal interface. | 
 | class RtpSenderBase : public RtpSenderInternal, public ObserverInterface { | 
 |  public: | 
 |   class SetStreamsObserver { | 
 |    public: | 
 |     virtual ~SetStreamsObserver() = default; | 
 |     virtual void OnSetStreams() = 0; | 
 |   }; | 
 |  | 
 |   // Sets the underlying MediaEngine channel associated with this RtpSender. | 
 |   // A VoiceMediaChannel should be used for audio RtpSenders and | 
 |   // a VideoMediaChannel should be used for video RtpSenders. | 
 |   // Must call SetMediaChannel(nullptr) before the media channel is destroyed. | 
 |   void SetMediaChannel(MediaSendChannelInterface* media_channel) override; | 
 |  | 
 |   bool SetTrack(MediaStreamTrackInterface* track) override; | 
 |   scoped_refptr<MediaStreamTrackInterface> track() const override { | 
 |     // This method is currently called from the worker thread by | 
 |     // RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n. | 
 |     // RTC_DCHECK_RUN_ON(signaling_thread_); | 
 |     return track_; | 
 |   } | 
 |  | 
 |   RtpParameters GetParameters() const override; | 
 |   RTCError SetParameters(const RtpParameters& parameters) override; | 
 |   void SetParametersAsync(const RtpParameters& parameters, | 
 |                           SetParametersCallback callback) override; | 
 |  | 
 |   // `GetParameters` and `SetParameters` operate with a transactional model. | 
 |   // Allow access to get/set parameters without invalidating transaction id. | 
 |   RtpParameters GetParametersInternal() const override; | 
 |   void SetParametersInternal(const RtpParameters& parameters, | 
 |                              SetParametersCallback callback = nullptr, | 
 |                              bool blocking = true) override; | 
 |   RTCError CheckSetParameters(const RtpParameters& parameters); | 
 |   RTCError CheckCodecParameters(const RtpParameters& parameters) override; | 
 |   RtpParameters GetParametersInternalWithAllLayers() const override; | 
 |   RTCError SetParametersInternalWithAllLayers( | 
 |       const RtpParameters& parameters) override; | 
 |  | 
 |   // Used to set the SSRC of the sender, once a local description has been set. | 
 |   // If `ssrc` is 0, this indiates that the sender should disconnect from the | 
 |   // underlying transport (this occurs if the sender isn't seen in a local | 
 |   // description). | 
 |   void SetSsrc(uint32_t ssrc) override; | 
 |   uint32_t ssrc() const override { | 
 |     // This method is currently called from the worker thread by | 
 |     // RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n. | 
 |     // RTC_DCHECK_RUN_ON(signaling_thread_); | 
 |     return ssrc_; | 
 |   } | 
 |  | 
 |   std::vector<std::string> stream_ids() const override { | 
 |     RTC_DCHECK_RUN_ON(signaling_thread_); | 
 |     return stream_ids_; | 
 |   } | 
 |  | 
 |   // Set stream ids, eliminating duplicates in the process. | 
 |   void set_stream_ids(const std::vector<std::string>& stream_ids) override; | 
 |   void SetStreams(const std::vector<std::string>& stream_ids) override; | 
 |  | 
 |   std::string id() const override { return id_; } | 
 |  | 
 |   void set_init_send_encodings( | 
 |       const std::vector<RtpEncodingParameters>& init_send_encodings) override { | 
 |     init_parameters_.encodings = init_send_encodings; | 
 |   } | 
 |   std::vector<RtpEncodingParameters> init_send_encodings() const override { | 
 |     RTC_DCHECK_RUN_ON(signaling_thread_); | 
 |     return init_parameters_.encodings; | 
 |   } | 
 |  | 
 |   void set_transport( | 
 |       scoped_refptr<DtlsTransportInterface> dtls_transport) override { | 
 |     dtls_transport_ = dtls_transport; | 
 |   } | 
 |   scoped_refptr<DtlsTransportInterface> dtls_transport() const override { | 
 |     RTC_DCHECK_RUN_ON(signaling_thread_); | 
 |     return dtls_transport_; | 
 |   } | 
 |  | 
 |   void SetFrameEncryptor( | 
 |       scoped_refptr<FrameEncryptorInterface> frame_encryptor) override; | 
 |  | 
 |   scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const override { | 
 |     return frame_encryptor_; | 
 |   } | 
 |  | 
 |   void Stop() override; | 
 |  | 
 |   // Returns an ID that changes every time SetTrack() is called, but | 
 |   // otherwise remains constant. Used to generate IDs for stats. | 
 |   // The special value zero means that no track is attached. | 
 |   int AttachmentId() const override { return attachment_id_; } | 
 |  | 
 |   // Disables the layers identified by the specified RIDs. | 
 |   // If the specified list is empty, this is a no-op. | 
 |   RTCError DisableEncodingLayers(const std::vector<std::string>& rid) override; | 
 |  | 
 |   void SetFrameTransformer( | 
 |       scoped_refptr<FrameTransformerInterface> frame_transformer) override; | 
 |  | 
 |   void SetEncoderSelector( | 
 |       std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface> | 
 |           encoder_selector) override; | 
 |  | 
 |   void SetEncoderSelectorOnChannel(); | 
 |  | 
 |   void SetTransceiverAsStopped() override { | 
 |     RTC_DCHECK_RUN_ON(signaling_thread_); | 
 |     is_transceiver_stopped_ = true; | 
 |   } | 
 |  | 
 |   void SetSendCodecs(std::vector<Codec> send_codecs) override { | 
 |     send_codecs_ = send_codecs; | 
 |   } | 
 |   std::vector<Codec> GetSendCodecs() const override { return send_codecs_; } | 
 |  | 
 |   void NotifyFirstPacketSent() override; | 
 |   void SetObserver(RtpSenderObserverInterface* observer) override; | 
 |  | 
 |  protected: | 
 |   // 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. | 
 |   RtpSenderBase(const Environment& env, | 
 |                 Thread* worker_thread, | 
 |                 const std::string& id, | 
 |                 SetStreamsObserver* set_streams_observer); | 
 |   // TODO(bugs.webrtc.org/8694): Since SSRC == 0 is technically valid, figure | 
 |   // out some other way to test if we have a valid SSRC. | 
 |   bool can_send_track() const { return track_ && ssrc_; } | 
 |  | 
 |   virtual std::string track_kind() const = 0; | 
 |  | 
 |   // Enable sending on the media channel. | 
 |   virtual void SetSend() = 0; | 
 |   // Disable sending on the media channel. | 
 |   virtual void ClearSend() = 0; | 
 |  | 
 |   // Template method pattern to allow subclasses to add custom behavior for | 
 |   // when tracks are attached, detached, and for adding tracks to statistics. | 
 |   virtual void AttachTrack() {} | 
 |   virtual void DetachTrack() {} | 
 |   virtual void AddTrackToStats() {} | 
 |   virtual void RemoveTrackFromStats() {} | 
 |  | 
 |   const Environment env_; | 
 |   Thread* const signaling_thread_; | 
 |   Thread* const worker_thread_; | 
 |   uint32_t ssrc_ = 0; | 
 |   bool stopped_ RTC_GUARDED_BY(signaling_thread_) = false; | 
 |   bool is_transceiver_stopped_ RTC_GUARDED_BY(signaling_thread_) = false; | 
 |   int attachment_id_ = 0; | 
 |   const std::string id_; | 
 |  | 
 |   std::vector<std::string> stream_ids_; | 
 |   RtpParameters init_parameters_; | 
 |   std::vector<Codec> send_codecs_; | 
 |  | 
 |   // TODO(tommi): `media_channel_` and several other member variables in this | 
 |   // class (ssrc_, stopped_, etc) are accessed from more than one thread without | 
 |   // a guard or lock. Internally there are also several Invoke()s that we could | 
 |   // remove since the upstream code may already be performing several operations | 
 |   // on the worker thread. | 
 |   MediaSendChannelInterface* media_channel_ = nullptr; | 
 |   scoped_refptr<MediaStreamTrackInterface> track_; | 
 |  | 
 |   scoped_refptr<DtlsTransportInterface> dtls_transport_; | 
 |   scoped_refptr<FrameEncryptorInterface> frame_encryptor_; | 
 |   // `last_transaction_id_` is used to verify that `SetParameters` is receiving | 
 |   // the parameters object that was last returned from `GetParameters`. | 
 |   // As such, it is used for internal verification and is not observable by the | 
 |   // the client. It is marked as mutable to enable `GetParameters` to be a | 
 |   // const method. | 
 |   mutable std::optional<std::string> last_transaction_id_; | 
 |   std::vector<std::string> disabled_rids_; | 
 |  | 
 |   SetStreamsObserver* set_streams_observer_ = nullptr; | 
 |   RtpSenderObserverInterface* observer_ = nullptr; | 
 |   bool sent_first_packet_ = false; | 
 |  | 
 |   scoped_refptr<FrameTransformerInterface> frame_transformer_; | 
 |   std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface> | 
 |       encoder_selector_; | 
 |  | 
 |   virtual RTCError GenerateKeyFrame(const std::vector<std::string>& rids) = 0; | 
 | }; | 
 |  | 
 | // LocalAudioSinkAdapter receives data callback as a sink to the local | 
 | // AudioTrack, and passes the data to the sink of AudioSource. | 
 | class LocalAudioSinkAdapter : public AudioTrackSinkInterface, | 
 |                               public AudioSource { | 
 |  public: | 
 |   LocalAudioSinkAdapter(); | 
 |   virtual ~LocalAudioSinkAdapter(); | 
 |  | 
 |  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, | 
 |               std::optional<int64_t> absolute_capture_timestamp_ms) override; | 
 |  | 
 |   // 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 { | 
 |     OnData(audio_data, bits_per_sample, sample_rate, number_of_channels, | 
 |            number_of_frames, | 
 |            /*absolute_capture_timestamp_ms=*/std::nullopt); | 
 |   } | 
 |  | 
 |   // AudioSinkInterface implementation. | 
 |   int NumPreferredChannels() const override { return num_preferred_channels_; } | 
 |  | 
 |   // webrtc::AudioSource implementation. | 
 |   void SetSink(AudioSource::Sink* sink) override; | 
 |  | 
 |   AudioSource::Sink* sink_; | 
 |   // Critical section protecting `sink_`. | 
 |   Mutex lock_; | 
 |   int num_preferred_channels_ = -1; | 
 | }; | 
 |  | 
 | class AudioRtpSender : 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 scoped_refptr<AudioRtpSender> Create( | 
 |       const Environment& env, | 
 |       Thread* worker_thread, | 
 |       const std::string& id, | 
 |       LegacyStatsCollectorInterface* stats, | 
 |       SetStreamsObserver* set_streams_observer); | 
 |   virtual ~AudioRtpSender(); | 
 |  | 
 |   // DtmfSenderProvider implementation. | 
 |   bool CanInsertDtmf() override; | 
 |   bool InsertDtmf(int code, int duration) override; | 
 |  | 
 |   // ObserverInterface implementation. | 
 |   void OnChanged() override; | 
 |  | 
 |   webrtc::MediaType media_type() const override { | 
 |     return webrtc::MediaType::AUDIO; | 
 |   } | 
 |   std::string track_kind() const override { | 
 |     return MediaStreamTrackInterface::kAudioKind; | 
 |   } | 
 |  | 
 |   scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override; | 
 |   RTCError GenerateKeyFrame(const std::vector<std::string>& rids) override; | 
 |  | 
 |  protected: | 
 |   AudioRtpSender(const Environment& env, | 
 |                  Thread* worker_thread, | 
 |                  const std::string& id, | 
 |                  LegacyStatsCollectorInterface* legacy_stats, | 
 |                  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: | 
 |   VoiceMediaSendChannelInterface* voice_media_channel() { | 
 |     return media_channel_->AsVoiceSendChannel(); | 
 |   } | 
 |   scoped_refptr<AudioTrackInterface> audio_track() const { | 
 |     return scoped_refptr<AudioTrackInterface>( | 
 |         static_cast<AudioTrackInterface*>(track_.get())); | 
 |   } | 
 |  | 
 |   LegacyStatsCollectorInterface* legacy_stats_ = nullptr; | 
 |   scoped_refptr<DtmfSender> dtmf_sender_; | 
 |   scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_; | 
 |   bool cached_track_enabled_ = false; | 
 |  | 
 |   // Used to pass the data callback from the `track_` to the other end of | 
 |   // webrtc::AudioSource. | 
 |   std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_; | 
 | }; | 
 |  | 
 | class VideoRtpSender : 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 scoped_refptr<VideoRtpSender> Create( | 
 |       const Environment& env, | 
 |       Thread* worker_thread, | 
 |       const std::string& id, | 
 |       SetStreamsObserver* set_streams_observer); | 
 |   virtual ~VideoRtpSender(); | 
 |  | 
 |   // ObserverInterface implementation | 
 |   void OnChanged() override; | 
 |  | 
 |   webrtc::MediaType media_type() const override { | 
 |     return webrtc::MediaType::VIDEO; | 
 |   } | 
 |   std::string track_kind() const override { | 
 |     return MediaStreamTrackInterface::kVideoKind; | 
 |   } | 
 |  | 
 |   scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override; | 
 |   RTCError GenerateKeyFrame(const std::vector<std::string>& rids) override; | 
 |  | 
 |  protected: | 
 |   VideoRtpSender(const Environment& env, | 
 |                  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: | 
 |   VideoMediaSendChannelInterface* video_media_channel() { | 
 |     return media_channel_->AsVideoSendChannel(); | 
 |   } | 
 |   scoped_refptr<VideoTrackInterface> video_track() const { | 
 |     return scoped_refptr<VideoTrackInterface>( | 
 |         static_cast<VideoTrackInterface*>(track_.get())); | 
 |   } | 
 |  | 
 |   VideoTrackInterface::ContentHint cached_track_content_hint_ = | 
 |       VideoTrackInterface::ContentHint::kNone; | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // PC_RTP_SENDER_H_ |