/*
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* libjingle
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* Copyright 2015 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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// This file contains classes that implement RtpSenderInterface.
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// An RtpSender associates a MediaStreamTrackInterface with an underlying
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// transport (provided by AudioProviderInterface/VideoProviderInterface)
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#ifndef TALK_APP_WEBRTC_RTPSENDER_H_
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#define TALK_APP_WEBRTC_RTPSENDER_H_
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#include <string>
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#include "talk/app/webrtc/mediastreamprovider.h"
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#include "talk/app/webrtc/rtpsenderinterface.h"
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#include "talk/app/webrtc/statscollector.h"
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#include "talk/media/base/audiorenderer.h"
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/scoped_ptr.h"
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namespace webrtc {
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// LocalAudioSinkAdapter receives data callback as a sink to the local
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// AudioTrack, and passes the data to the sink of AudioRenderer.
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class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
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public cricket::AudioRenderer {
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public:
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LocalAudioSinkAdapter();
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virtual ~LocalAudioSinkAdapter();
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private:
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// AudioSinkInterface implementation.
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void OnData(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) override;
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// cricket::AudioRenderer implementation.
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void SetSink(cricket::AudioRenderer::Sink* sink) override;
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cricket::AudioRenderer::Sink* sink_;
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// Critical section protecting |sink_|.
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rtc::CriticalSection lock_;
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};
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class AudioRtpSender : public ObserverInterface,
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public rtc::RefCountedObject<RtpSenderInterface> {
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public:
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// StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
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// at the appropriate times.
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AudioRtpSender(AudioTrackInterface* track,
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const std::string& stream_id,
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AudioProviderInterface* provider,
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StatsCollector* stats);
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// Randomly generates id and stream_id.
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AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
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virtual ~AudioRtpSender();
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// ObserverInterface implementation
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void OnChanged() override;
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// RtpSenderInterface implementation
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bool SetTrack(MediaStreamTrackInterface* track) override;
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rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
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return track_.get();
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}
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void SetSsrc(uint32_t ssrc) override;
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uint32_t ssrc() const override { return ssrc_; }
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_AUDIO;
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}
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std::string id() const override { return id_; }
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void set_stream_id(const std::string& stream_id) override {
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stream_id_ = stream_id;
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}
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std::string stream_id() const override { return stream_id_; }
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void Stop() override;
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private:
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bool can_send_track() const { return track_ && ssrc_; }
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// Helper function to construct options for
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// AudioProviderInterface::SetAudioSend.
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void SetAudioSend();
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std::string id_;
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std::string stream_id_;
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AudioProviderInterface* provider_;
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StatsCollector* stats_;
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rtc::scoped_refptr<AudioTrackInterface> track_;
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uint32_t ssrc_ = 0;
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bool cached_track_enabled_ = false;
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bool stopped_ = false;
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// Used to pass the data callback from the |track_| to the other end of
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// cricket::AudioRenderer.
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rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
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};
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class VideoRtpSender : public ObserverInterface,
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public rtc::RefCountedObject<RtpSenderInterface> {
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public:
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VideoRtpSender(VideoTrackInterface* track,
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const std::string& stream_id,
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VideoProviderInterface* provider);
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// Randomly generates id and stream_id.
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explicit VideoRtpSender(VideoProviderInterface* provider);
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virtual ~VideoRtpSender();
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// ObserverInterface implementation
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void OnChanged() override;
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// RtpSenderInterface implementation
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bool SetTrack(MediaStreamTrackInterface* track) override;
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rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
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return track_.get();
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}
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void SetSsrc(uint32_t ssrc) override;
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uint32_t ssrc() const override { return ssrc_; }
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_VIDEO;
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}
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std::string id() const override { return id_; }
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void set_stream_id(const std::string& stream_id) override {
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stream_id_ = stream_id;
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}
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std::string stream_id() const override { return stream_id_; }
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void Stop() override;
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private:
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bool can_send_track() const { return track_ && ssrc_; }
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// Helper function to construct options for
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// VideoProviderInterface::SetVideoSend.
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void SetVideoSend();
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std::string id_;
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std::string stream_id_;
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VideoProviderInterface* provider_;
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rtc::scoped_refptr<VideoTrackInterface> track_;
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uint32_t ssrc_ = 0;
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bool cached_track_enabled_ = false;
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bool stopped_ = false;
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};
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} // namespace webrtc
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#endif // TALK_APP_WEBRTC_RTPSENDER_H_
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