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/*
* 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 <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/dtmf_sender_interface.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(cricket::MediaChannel* 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(
rtc::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 RTCError SetParametersInternal(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;
};
// 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(cricket::MediaChannel* media_channel) override;
bool SetTrack(MediaStreamTrackInterface* track) override;
rtc::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;
// `GetParameters` and `SetParameters` operate with a transactional model.
// Allow access to get/set parameters without invalidating transaction id.
RtpParameters GetParametersInternal() const override;
RTCError SetParametersInternal(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_;
}
void set_stream_ids(const std::vector<std::string>& stream_ids) override {
stream_ids_ = stream_ids;
}
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(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
dtls_transport_ = dtls_transport;
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
RTC_DCHECK_RUN_ON(signaling_thread_);
return dtls_transport_;
}
void SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
rtc::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 SetEncoderToPacketizerFrameTransformer(
rtc::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;
}
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(rtc::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() {}
rtc::Thread* const signaling_thread_;
rtc::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_;
// 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.
cricket::MediaChannel* media_channel_ = nullptr;
rtc::scoped_refptr<MediaStreamTrackInterface> track_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
rtc::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 absl::optional<std::string> last_transaction_id_;
std::vector<std::string> disabled_rids_;
SetStreamsObserver* set_streams_observer_ = nullptr;
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_;
std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>
encoder_selector_;
};
// 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 cricket::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,
absl::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=*/absl::nullopt);
}
// AudioSinkInterface implementation.
int NumPreferredChannels() const override { return num_preferred_channels_; }
// cricket::AudioSource implementation.
void SetSink(cricket::AudioSource::Sink* sink) override;
cricket::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 rtc::scoped_refptr<AudioRtpSender> Create(
rtc::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;
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
std::string track_kind() const override {
return MediaStreamTrackInterface::kAudioKind;
}
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
protected:
AudioRtpSender(rtc::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:
cricket::VoiceMediaChannel* voice_media_channel() {
return static_cast<cricket::VoiceMediaChannel*>(media_channel_);
}
rtc::scoped_refptr<AudioTrackInterface> audio_track() const {
return rtc::scoped_refptr<AudioTrackInterface>(
static_cast<AudioTrackInterface*>(track_.get()));
}
LegacyStatsCollectorInterface* legacy_stats_ = nullptr;
rtc::scoped_refptr<DtmfSender> dtmf_sender_;
rtc::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
// cricket::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 rtc::scoped_refptr<VideoRtpSender> Create(
rtc::Thread* worker_thread,
const std::string& id,
SetStreamsObserver* set_streams_observer);
virtual ~VideoRtpSender();
// 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<DtmfSenderInterface> GetDtmfSender() const override;
protected:
VideoRtpSender(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<cricket::VideoMediaChannel*>(media_channel_);
}
rtc::scoped_refptr<VideoTrackInterface> video_track() const {
return rtc::scoped_refptr<VideoTrackInterface>(
static_cast<VideoTrackInterface*>(track_.get()));
}
VideoTrackInterface::ContentHint cached_track_content_hint_ =
VideoTrackInterface::ContentHint::kNone;
};
} // namespace webrtc
#endif // PC_RTP_SENDER_H_