8f46c63f6f
Reason for revert: Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail. Original issue's description: > Adding the ability to create an RtpSender without a track. > > This CL also changes AddStream to immediately create a sender, rather > than waiting until the track is seen in SDP. And the PeerConnection now > builds the list of "send streams" from the list of senders, rather than > the collection of local media streams. > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5 > Cr-Commit-Position: refs/heads/master@{#10414} TBR=pthatcher@webrtc.org,pthatcher@chromium.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1426443007 Cr-Commit-Position: refs/heads/master@{#10417}
141 lines
4.6 KiB
C++
141 lines
4.6 KiB
C++
/*
|
|
* libjingle
|
|
* Copyright 2015 Google Inc.
|
|
*
|
|
* Redistribution and use in source and binary forms, with or without
|
|
* modification, are permitted provided that the following conditions are met:
|
|
*
|
|
* 1. Redistributions of source code must retain the above copyright notice,
|
|
* this list of conditions and the following disclaimer.
|
|
* 2. Redistributions in binary form must reproduce the above copyright notice,
|
|
* this list of conditions and the following disclaimer in the documentation
|
|
* and/or other materials provided with the distribution.
|
|
* 3. The name of the author may not be used to endorse or promote products
|
|
* derived from this software without specific prior written permission.
|
|
*
|
|
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
|
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
|
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
|
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
|
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
|
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
|
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
|
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
|
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
|
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
|
*/
|
|
|
|
// This file contains classes that implement RtpSenderInterface.
|
|
// An RtpSender associates a MediaStreamTrackInterface with an underlying
|
|
// transport (provided by AudioProviderInterface/VideoProviderInterface)
|
|
|
|
#ifndef TALK_APP_WEBRTC_RTPSENDER_H_
|
|
#define TALK_APP_WEBRTC_RTPSENDER_H_
|
|
|
|
#include <string>
|
|
|
|
#include "talk/app/webrtc/mediastreamprovider.h"
|
|
#include "talk/app/webrtc/rtpsenderinterface.h"
|
|
#include "talk/media/base/audiorenderer.h"
|
|
#include "webrtc/base/basictypes.h"
|
|
#include "webrtc/base/criticalsection.h"
|
|
#include "webrtc/base/scoped_ptr.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// LocalAudioSinkAdapter receives data callback as a sink to the local
|
|
// AudioTrack, and passes the data to the sink of AudioRenderer.
|
|
class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
|
|
public cricket::AudioRenderer {
|
|
public:
|
|
LocalAudioSinkAdapter();
|
|
virtual ~LocalAudioSinkAdapter();
|
|
|
|
private:
|
|
// AudioSinkInterface implementation.
|
|
void OnData(const void* audio_data,
|
|
int bits_per_sample,
|
|
int sample_rate,
|
|
int number_of_channels,
|
|
size_t number_of_frames) override;
|
|
|
|
// cricket::AudioRenderer implementation.
|
|
void SetSink(cricket::AudioRenderer::Sink* sink) override;
|
|
|
|
cricket::AudioRenderer::Sink* sink_;
|
|
// Critical section protecting |sink_|.
|
|
rtc::CriticalSection lock_;
|
|
};
|
|
|
|
class AudioRtpSender : public ObserverInterface,
|
|
public rtc::RefCountedObject<RtpSenderInterface> {
|
|
public:
|
|
AudioRtpSender(AudioTrackInterface* track,
|
|
uint32_t ssrc,
|
|
AudioProviderInterface* provider);
|
|
|
|
virtual ~AudioRtpSender();
|
|
|
|
// ObserverInterface implementation
|
|
void OnChanged() override;
|
|
|
|
// RtpSenderInterface implementation
|
|
bool SetTrack(MediaStreamTrackInterface* track) override;
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
|
|
return track_.get();
|
|
}
|
|
|
|
std::string id() const override { return id_; }
|
|
|
|
void Stop() override;
|
|
|
|
private:
|
|
void Reconfigure();
|
|
|
|
std::string id_;
|
|
rtc::scoped_refptr<AudioTrackInterface> track_;
|
|
uint32_t ssrc_;
|
|
AudioProviderInterface* provider_;
|
|
bool cached_track_enabled_;
|
|
|
|
// Used to pass the data callback from the |track_| to the other end of
|
|
// cricket::AudioRenderer.
|
|
rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
|
|
};
|
|
|
|
class VideoRtpSender : public ObserverInterface,
|
|
public rtc::RefCountedObject<RtpSenderInterface> {
|
|
public:
|
|
VideoRtpSender(VideoTrackInterface* track,
|
|
uint32_t ssrc,
|
|
VideoProviderInterface* provider);
|
|
|
|
virtual ~VideoRtpSender();
|
|
|
|
// ObserverInterface implementation
|
|
void OnChanged() override;
|
|
|
|
// RtpSenderInterface implementation
|
|
bool SetTrack(MediaStreamTrackInterface* track) override;
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
|
|
return track_.get();
|
|
}
|
|
|
|
std::string id() const override { return id_; }
|
|
|
|
void Stop() override;
|
|
|
|
private:
|
|
void Reconfigure();
|
|
|
|
std::string id_;
|
|
rtc::scoped_refptr<VideoTrackInterface> track_;
|
|
uint32_t ssrc_;
|
|
VideoProviderInterface* provider_;
|
|
bool cached_track_enabled_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // TALK_APP_WEBRTC_RTPSENDER_H_
|