1 /* 2 * libjingle 3 * Copyright 2015 Google Inc. 4 * 5 * Redistribution and use in source and binary forms, with or without 6 * modification, are permitted provided that the following conditions are met: 7 * 8 * 1. Redistributions of source code must retain the above copyright notice, 9 * this list of conditions and the following disclaimer. 10 * 2. Redistributions in binary form must reproduce the above copyright notice, 11 * this list of conditions and the following disclaimer in the documentation 12 * and/or other materials provided with the distribution. 13 * 3. The name of the author may not be used to endorse or promote products 14 * derived from this software without specific prior written permission. 15 * 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 26 */ 27 28 // This file contains classes that implement RtpSenderInterface. 29 // An RtpSender associates a MediaStreamTrackInterface with an underlying 30 // transport (provided by AudioProviderInterface/VideoProviderInterface) 31 32 #ifndef TALK_APP_WEBRTC_RTPSENDER_H_ 33 #define TALK_APP_WEBRTC_RTPSENDER_H_ 34 35 #include <string> 36 37 #include "talk/app/webrtc/mediastreamprovider.h" 38 #include "talk/app/webrtc/rtpsenderinterface.h" 39 #include "talk/app/webrtc/statscollector.h" 40 #include "talk/media/base/audiorenderer.h" 41 #include "webrtc/base/basictypes.h" 42 #include "webrtc/base/criticalsection.h" 43 #include "webrtc/base/scoped_ptr.h" 44 45 namespace webrtc { 46 47 // LocalAudioSinkAdapter receives data callback as a sink to the local 48 // AudioTrack, and passes the data to the sink of AudioRenderer. 49 class LocalAudioSinkAdapter : public AudioTrackSinkInterface, 50 public cricket::AudioRenderer { 51 public: 52 LocalAudioSinkAdapter(); 53 virtual ~LocalAudioSinkAdapter(); 54 55 private: 56 // AudioSinkInterface implementation. 57 void OnData(const void* audio_data, 58 int bits_per_sample, 59 int sample_rate, 60 size_t number_of_channels, 61 size_t number_of_frames) override; 62 63 // cricket::AudioRenderer implementation. 64 void SetSink(cricket::AudioRenderer::Sink* sink) override; 65 66 cricket::AudioRenderer::Sink* sink_; 67 // Critical section protecting |sink_|. 68 rtc::CriticalSection lock_; 69 }; 70 71 class AudioRtpSender : public ObserverInterface, 72 public rtc::RefCountedObject<RtpSenderInterface> { 73 public: 74 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called 75 // at the appropriate times. 76 AudioRtpSender(AudioTrackInterface* track, 77 const std::string& stream_id, 78 AudioProviderInterface* provider, 79 StatsCollector* stats); 80 81 // Randomly generates id and stream_id. 82 AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats); 83 84 virtual ~AudioRtpSender(); 85 86 // ObserverInterface implementation 87 void OnChanged() override; 88 89 // RtpSenderInterface implementation 90 bool SetTrack(MediaStreamTrackInterface* track) override; track()91 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { 92 return track_.get(); 93 } 94 95 void SetSsrc(uint32_t ssrc) override; 96 ssrc()97 uint32_t ssrc() const override { return ssrc_; } 98 media_type()99 cricket::MediaType media_type() const override { 100 return cricket::MEDIA_TYPE_AUDIO; 101 } 102 id()103 std::string id() const override { return id_; } 104 set_stream_id(const std::string & stream_id)105 void set_stream_id(const std::string& stream_id) override { 106 stream_id_ = stream_id; 107 } stream_id()108 std::string stream_id() const override { return stream_id_; } 109 110 void Stop() override; 111 112 private: can_send_track()113 bool can_send_track() const { return track_ && ssrc_; } 114 // Helper function to construct options for 115 // AudioProviderInterface::SetAudioSend. 116 void SetAudioSend(); 117 118 std::string id_; 119 std::string stream_id_; 120 AudioProviderInterface* provider_; 121 StatsCollector* stats_; 122 rtc::scoped_refptr<AudioTrackInterface> track_; 123 uint32_t ssrc_ = 0; 124 bool cached_track_enabled_ = false; 125 bool stopped_ = false; 126 127 // Used to pass the data callback from the |track_| to the other end of 128 // cricket::AudioRenderer. 129 rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_; 130 }; 131 132 class VideoRtpSender : public ObserverInterface, 133 public rtc::RefCountedObject<RtpSenderInterface> { 134 public: 135 VideoRtpSender(VideoTrackInterface* track, 136 const std::string& stream_id, 137 VideoProviderInterface* provider); 138 139 // Randomly generates id and stream_id. 140 explicit VideoRtpSender(VideoProviderInterface* provider); 141 142 virtual ~VideoRtpSender(); 143 144 // ObserverInterface implementation 145 void OnChanged() override; 146 147 // RtpSenderInterface implementation 148 bool SetTrack(MediaStreamTrackInterface* track) override; track()149 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { 150 return track_.get(); 151 } 152 153 void SetSsrc(uint32_t ssrc) override; 154 ssrc()155 uint32_t ssrc() const override { return ssrc_; } 156 media_type()157 cricket::MediaType media_type() const override { 158 return cricket::MEDIA_TYPE_VIDEO; 159 } 160 id()161 std::string id() const override { return id_; } 162 set_stream_id(const std::string & stream_id)163 void set_stream_id(const std::string& stream_id) override { 164 stream_id_ = stream_id; 165 } stream_id()166 std::string stream_id() const override { return stream_id_; } 167 168 void Stop() override; 169 170 private: can_send_track()171 bool can_send_track() const { return track_ && ssrc_; } 172 // Helper function to construct options for 173 // VideoProviderInterface::SetVideoSend. 174 void SetVideoSend(); 175 176 std::string id_; 177 std::string stream_id_; 178 VideoProviderInterface* provider_; 179 rtc::scoped_refptr<VideoTrackInterface> track_; 180 uint32_t ssrc_ = 0; 181 bool cached_track_enabled_ = false; 182 bool stopped_ = false; 183 }; 184 185 } // namespace webrtc 186 187 #endif // TALK_APP_WEBRTC_RTPSENDER_H_ 188