Files
platform-external-webrtc/pc/rtpsender.h
Florent Castelli 892acf01f6 Add support for send_encodings parameters in addTransceiver
This will later allow simulcast to be set up without any SDP
manipulation. Currently limited to only one layer as the SDP
generated is not spec compliant and more work is required
to support simulcast.

Initial encoding parameters are deferred and applied when the ssrc
is set on the sender. This allows parameters to be changed before
negotiation is completed.

Bug: webrtc:7600
Change-Id: I0a31cd1c2bfc72ebb61ce0fa4fa69d87e3d8b74d
Reviewed-on: https://webrtc-review.googlesource.com/95488
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24917}
2018-10-01 22:56:30 +00:00

290 lines
9.6 KiB
C++

/*
* 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_RTPSENDER_H_
#define PC_RTPSENDER_H_
#include <memory>
#include <string>
#include <vector>
#include "api/mediastreaminterface.h"
#include "api/rtpsenderinterface.h"
#include "media/base/audiosource.h"
#include "media/base/mediachannel.h"
#include "pc/dtmfsender.h"
#include "rtc_base/criticalsection.h"
namespace webrtc {
class StatsCollector;
bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters);
// Internal interface used by PeerConnection.
class RtpSenderInternal : public RtpSenderInterface {
public:
// Sets the underlying MediaEngine channel associated with this RtpSender.
// SetVoiceMediaChannel should be used for audio RtpSenders and
// SetVideoMediaChannel should be used for video RtpSenders. Must call the
// appropriate SetXxxMediaChannel(nullptr) before the media channel is
// destroyed.
virtual void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) = 0;
virtual void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_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 Stop() = 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;
};
// 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) override;
// cricket::AudioSource implementation.
void SetSink(cricket::AudioSource::Sink* sink) override;
cricket::AudioSource::Sink* sink_;
// Critical section protecting |sink_|.
rtc::CriticalSection lock_;
};
class AudioRtpSender : public DtmfProviderInterface,
public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
// StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
// at the appropriate times.
// Construct an RtpSender for audio with the given sender ID.
// The sender is initialized with no track to send and no associated streams.
AudioRtpSender(rtc::Thread* worker_thread,
const std::string& id,
StatsCollector* stats);
virtual ~AudioRtpSender();
// DtmfSenderProvider implementation.
bool CanInsertDtmf() override;
bool InsertDtmf(int code, int duration) override;
sigslot::signal0<>* GetOnDestroyedSignal() override;
// ObserverInterface implementation.
void OnChanged() override;
// RtpSenderInterface implementation.
bool SetTrack(MediaStreamTrackInterface* track) override;
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_;
}
uint32_t ssrc() const override { return ssrc_; }
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
std::string id() const override { return id_; }
std::vector<std::string> stream_ids() const override { return stream_ids_; }
RtpParameters GetParameters() override;
RTCError SetParameters(const RtpParameters& parameters) override;
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
void SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor()
const override;
// RtpSenderInternal implementation.
void SetSsrc(uint32_t ssrc) override;
void set_stream_ids(const std::vector<std::string>& stream_ids) override {
stream_ids_ = stream_ids;
}
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 {
return init_parameters_.encodings;
}
void Stop() override;
int AttachmentId() const override { return attachment_id_; }
void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) override;
void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_media_channel) override {
RTC_NOTREACHED();
}
private:
// TODO(nisse): 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_; }
// Helper function to construct options for
// AudioProviderInterface::SetAudioSend.
void SetAudioSend();
// Helper function to call SetAudioSend with "stop sending" parameters.
void ClearAudioSend();
sigslot::signal0<> SignalDestroyed;
rtc::Thread* const worker_thread_;
const std::string id_;
std::vector<std::string> stream_ids_;
RtpParameters init_parameters_;
cricket::VoiceMediaChannel* media_channel_ = nullptr;
StatsCollector* stats_ = nullptr;
rtc::scoped_refptr<AudioTrackInterface> track_;
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
absl::optional<std::string> last_transaction_id_;
uint32_t ssrc_ = 0;
bool cached_track_enabled_ = false;
bool stopped_ = false;
// Used to pass the data callback from the |track_| to the other end of
// cricket::AudioSource.
std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_;
};
class VideoRtpSender : public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
// Construct an RtpSender for video with the given sender ID.
// The sender is initialized with no track to send and no associated streams.
VideoRtpSender(rtc::Thread* worker_thread, const std::string& id);
virtual ~VideoRtpSender();
// ObserverInterface implementation
void OnChanged() override;
// RtpSenderInterface implementation
bool SetTrack(MediaStreamTrackInterface* track) override;
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_;
}
uint32_t ssrc() const override { return ssrc_; }
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
std::string id() const override { return id_; }
std::vector<std::string> stream_ids() const override { return stream_ids_; }
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 {
return init_parameters_.encodings;
}
RtpParameters GetParameters() override;
RTCError SetParameters(const RtpParameters& parameters) override;
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
void SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor()
const override;
// RtpSenderInternal implementation.
void SetSsrc(uint32_t ssrc) override;
void set_stream_ids(const std::vector<std::string>& stream_ids) override {
stream_ids_ = stream_ids;
}
void Stop() override;
int AttachmentId() const override { return attachment_id_; }
void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) override {
RTC_NOTREACHED();
}
void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_media_channel) override;
private:
bool can_send_track() const { return track_ && ssrc_; }
// Helper function to construct options for
// VideoProviderInterface::SetVideoSend.
void SetVideoSend();
// Helper function to call SetVideoSend with "stop sending" parameters.
void ClearVideoSend();
rtc::Thread* worker_thread_;
const std::string id_;
std::vector<std::string> stream_ids_;
RtpParameters init_parameters_;
cricket::VideoMediaChannel* media_channel_ = nullptr;
rtc::scoped_refptr<VideoTrackInterface> track_;
absl::optional<std::string> last_transaction_id_;
uint32_t ssrc_ = 0;
VideoTrackInterface::ContentHint cached_track_content_hint_ =
VideoTrackInterface::ContentHint::kNone;
bool stopped_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_;
};
} // namespace webrtc
#endif // PC_RTPSENDER_H_