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/*
* 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_
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