webrtc_m130/pc/rtpsender.h
Seth Hampson 5b4f075f9c Reland "Reland "Adds support for multiple or no media stream ids.""
This is a reland of f351c3408a0c7f695447a2a9f4e6a1719a0d6a26

Reland history:
The original CL broke tests in chromium which were manually tested in
the first reland. Another small fix was added to the reland to fix a
downstream bug, which caused separate tests to fail in chromium.
These were not caught because the chromium trybot was down. These
are temporarily disabled in chrome to allow this change to roll in.

Original change's description:
> Reland "Adds support for multiple or no media stream ids."
>
> This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb
>
> Original change's description:
> > Adds support for multiple or no media stream ids.
> >
> > With Unified Plan SDP semantics, this adds support for specifying
> > either no media stream ids or multiple media stream ids for a
> > transceiver/sender/receiver. This includes serializing/deserializing
> > SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> > <appdata>" line to indicate the no stream case. Note that this does
> > not synchronize between multiple streams, this is still just supported
> > based upon the first media stream id.
> >
> > Bug: webrtc:7932, webrtc:7933
> > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> > Reviewed-on: https://webrtc-review.googlesource.com/61341
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22611}
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
> Reviewed-on: https://webrtc-review.googlesource.com/65560
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22687}

TBR=deadbeef@webrtc.org

Bug: webrtc:7932, webrtc:7933
Change-Id: Ideb30219b2f952dd51428cd4e8bd43ef49df5b17
Reviewed-on: https://webrtc-review.googlesource.com/66280
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22699}
2018-04-03 01:10:07 +00:00

267 lines
8.8 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 "rtc_base/basictypes.h"
#include "rtc_base/criticalsection.h"
#include "media/base/audiosource.h"
#include "media/base/mediachannel.h"
#include "pc/dtmfsender.h"
namespace webrtc {
class StatsCollector;
// 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 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 AudioRtpSender with a null track, a single, randomly generated
// stream id, and a randomly generated ID.
AudioRtpSender(rtc::Thread* worker_thread, StatsCollector* stats);
// Construct an AudioRtpSender with the given track and stream ids. The
// sender ID will be set to the track's ID.
AudioRtpSender(rtc::Thread* worker_thread,
rtc::scoped_refptr<AudioTrackInterface> track,
const std::vector<std::string>& stream_ids,
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() const override;
RTCError SetParameters(const RtpParameters& parameters) override;
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() 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 {
media_channel_ = voice_media_channel;
}
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_;
cricket::VoiceMediaChannel* media_channel_ = nullptr;
StatsCollector* stats_;
rtc::scoped_refptr<AudioTrackInterface> track_;
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
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;
};
class VideoRtpSender : public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
// Construct a VideoRtpSender with a null track, a single, randomly generated
// stream id, and a randomly generated ID.
explicit VideoRtpSender(rtc::Thread* worker_thread);
// Construct a VideoRtpSender with the given track and stream ids. The
// sender ID will be set to the track's ID.
VideoRtpSender(rtc::Thread* worker_thread,
rtc::scoped_refptr<VideoTrackInterface> track,
const std::vector<std::string>& stream_ids);
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_; }
RtpParameters GetParameters() const override;
RTCError SetParameters(const RtpParameters& parameters) override;
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() 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 {
media_channel_ = video_media_channel;
}
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_;
cricket::VideoMediaChannel* media_channel_ = nullptr;
rtc::scoped_refptr<VideoTrackInterface> track_;
uint32_t ssrc_ = 0;
bool cached_track_enabled_ = false;
VideoTrackInterface::ContentHint cached_track_content_hint_ =
VideoTrackInterface::ContentHint::kNone;
bool stopped_ = false;
int attachment_id_ = 0;
};
} // namespace webrtc
#endif // PC_RTPSENDER_H_