| OLD | NEW |
| 1 /* | 1 /* |
| 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 // This file contains classes that implement RtpSenderInterface. | 11 // This file contains classes that implement RtpSenderInterface. |
| 12 // An RtpSender associates a MediaStreamTrackInterface with an underlying | 12 // An RtpSender associates a MediaStreamTrackInterface with an underlying |
| 13 // transport (provided by AudioProviderInterface/VideoProviderInterface) | 13 // transport (provided by AudioProviderInterface/VideoProviderInterface) |
| 14 | 14 |
| 15 #ifndef WEBRTC_API_RTPSENDER_H_ | 15 #ifndef WEBRTC_API_RTPSENDER_H_ |
| 16 #define WEBRTC_API_RTPSENDER_H_ | 16 #define WEBRTC_API_RTPSENDER_H_ |
| 17 | 17 |
| 18 #include <memory> | 18 #include <memory> |
| 19 #include <string> | 19 #include <string> |
| 20 | 20 |
| 21 #include "webrtc/api/mediastreamprovider.h" | 21 #include "webrtc/api/mediastreaminterface.h" |
| 22 #include "webrtc/api/rtpsenderinterface.h" | 22 #include "webrtc/api/rtpsenderinterface.h" |
| 23 #include "webrtc/api/statscollector.h" | 23 #include "webrtc/api/statscollector.h" |
| 24 #include "webrtc/base/basictypes.h" | 24 #include "webrtc/base/basictypes.h" |
| 25 #include "webrtc/base/criticalsection.h" | 25 #include "webrtc/base/criticalsection.h" |
| 26 #include "webrtc/media/base/audiosource.h" | 26 #include "webrtc/media/base/audiosource.h" |
| 27 #include "webrtc/pc/channel.h" |
| 27 | 28 |
| 28 namespace webrtc { | 29 namespace webrtc { |
| 29 | 30 |
| 30 // Internal interface used by PeerConnection. | 31 // Internal interface used by PeerConnection. |
| 31 class RtpSenderInternal : public RtpSenderInterface { | 32 class RtpSenderInternal : public RtpSenderInterface { |
| 32 public: | 33 public: |
| 33 // Used to set the SSRC of the sender, once a local description has been set. | 34 // Used to set the SSRC of the sender, once a local description has been set. |
| 34 // If |ssrc| is 0, this indiates that the sender should disconnect from the | 35 // If |ssrc| is 0, this indiates that the sender should disconnect from the |
| 35 // underlying transport (this occurs if the sender isn't seen in a local | 36 // underlying transport (this occurs if the sender isn't seen in a local |
| 36 // description). | 37 // description). |
| (...skipping 28 matching lines...) Expand all Loading... |
| 65 cricket::AudioSource::Sink* sink_; | 66 cricket::AudioSource::Sink* sink_; |
| 66 // Critical section protecting |sink_|. | 67 // Critical section protecting |sink_|. |
| 67 rtc::CriticalSection lock_; | 68 rtc::CriticalSection lock_; |
| 68 }; | 69 }; |
| 69 | 70 |
| 70 class AudioRtpSender : public ObserverInterface, | 71 class AudioRtpSender : public ObserverInterface, |
| 71 public rtc::RefCountedObject<RtpSenderInternal> { | 72 public rtc::RefCountedObject<RtpSenderInternal> { |
| 72 public: | 73 public: |
| 73 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called | 74 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called |
| 74 // at the appropriate times. | 75 // at the appropriate times. |
| 76 // |channel| can be null if one does not exist yet. |
| 75 AudioRtpSender(AudioTrackInterface* track, | 77 AudioRtpSender(AudioTrackInterface* track, |
| 76 const std::string& stream_id, | 78 const std::string& stream_id, |
| 77 AudioProviderInterface* provider, | 79 cricket::VoiceChannel* channel, |
| 78 StatsCollector* stats); | 80 StatsCollector* stats); |
| 79 | 81 |
| 80 // Randomly generates stream_id. | 82 // Randomly generates stream_id. |
| 83 // |channel| can be null if one does not exist yet. |
| 81 AudioRtpSender(AudioTrackInterface* track, | 84 AudioRtpSender(AudioTrackInterface* track, |
| 82 AudioProviderInterface* provider, | 85 cricket::VoiceChannel* channel, |
| 83 StatsCollector* stats); | 86 StatsCollector* stats); |
| 84 | 87 |
| 85 // Randomly generates id and stream_id. | 88 // Randomly generates id and stream_id. |
| 86 AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats); | 89 // |channel| can be null if one does not exist yet. |
| 90 AudioRtpSender(cricket::VoiceChannel* channel, StatsCollector* stats); |
| 87 | 91 |
| 88 virtual ~AudioRtpSender(); | 92 virtual ~AudioRtpSender(); |
| 89 | 93 |
| 90 // ObserverInterface implementation | 94 // ObserverInterface implementation |
| 91 void OnChanged() override; | 95 void OnChanged() override; |
| 92 | 96 |
| 93 // RtpSenderInterface implementation | 97 // RtpSenderInterface implementation |
| 94 bool SetTrack(MediaStreamTrackInterface* track) override; | 98 bool SetTrack(MediaStreamTrackInterface* track) override; |
| 95 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 99 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
| 96 return track_; | 100 return track_; |
| (...skipping 18 matching lines...) Expand all Loading... |
| 115 // RtpSenderInternal implementation. | 119 // RtpSenderInternal implementation. |
| 116 void SetSsrc(uint32_t ssrc) override; | 120 void SetSsrc(uint32_t ssrc) override; |
| 117 | 121 |
| 118 void set_stream_id(const std::string& stream_id) override { | 122 void set_stream_id(const std::string& stream_id) override { |
| 119 stream_id_ = stream_id; | 123 stream_id_ = stream_id; |
| 120 } | 124 } |
| 121 std::string stream_id() const override { return stream_id_; } | 125 std::string stream_id() const override { return stream_id_; } |
| 122 | 126 |
| 123 void Stop() override; | 127 void Stop() override; |
| 124 | 128 |
| 129 // Does not take ownership. |
| 130 // Should call SetChannel(nullptr) before |channel| is destroyed. |
| 131 void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; } |
| 132 |
| 125 private: | 133 private: |
| 126 // TODO(nisse): Since SSRC == 0 is technically valid, figure out | 134 // TODO(nisse): Since SSRC == 0 is technically valid, figure out |
| 127 // some other way to test if we have a valid SSRC. | 135 // some other way to test if we have a valid SSRC. |
| 128 bool can_send_track() const { return track_ && ssrc_; } | 136 bool can_send_track() const { return track_ && ssrc_; } |
| 129 // Helper function to construct options for | 137 // Helper function to construct options for |
| 130 // AudioProviderInterface::SetAudioSend. | 138 // AudioProviderInterface::SetAudioSend. |
| 131 void SetAudioSend(); | 139 void SetAudioSend(); |
| 140 // Helper function to call SetAudioSend with "stop sending" parameters. |
| 141 void ClearAudioSend(); |
| 132 | 142 |
| 133 std::string id_; | 143 std::string id_; |
| 134 std::string stream_id_; | 144 std::string stream_id_; |
| 135 AudioProviderInterface* provider_; | 145 cricket::VoiceChannel* channel_ = nullptr; |
| 136 StatsCollector* stats_; | 146 StatsCollector* stats_; |
| 137 rtc::scoped_refptr<AudioTrackInterface> track_; | 147 rtc::scoped_refptr<AudioTrackInterface> track_; |
| 138 uint32_t ssrc_ = 0; | 148 uint32_t ssrc_ = 0; |
| 139 bool cached_track_enabled_ = false; | 149 bool cached_track_enabled_ = false; |
| 140 bool stopped_ = false; | 150 bool stopped_ = false; |
| 141 | 151 |
| 142 // Used to pass the data callback from the |track_| to the other end of | 152 // Used to pass the data callback from the |track_| to the other end of |
| 143 // cricket::AudioSource. | 153 // cricket::AudioSource. |
| 144 std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_; | 154 std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_; |
| 145 }; | 155 }; |
| 146 | 156 |
| 147 class VideoRtpSender : public ObserverInterface, | 157 class VideoRtpSender : public ObserverInterface, |
| 148 public rtc::RefCountedObject<RtpSenderInternal> { | 158 public rtc::RefCountedObject<RtpSenderInternal> { |
| 149 public: | 159 public: |
| 160 // |channel| can be null if one does not exist yet. |
| 150 VideoRtpSender(VideoTrackInterface* track, | 161 VideoRtpSender(VideoTrackInterface* track, |
| 151 const std::string& stream_id, | 162 const std::string& stream_id, |
| 152 VideoProviderInterface* provider); | 163 cricket::VideoChannel* channel); |
| 153 | 164 |
| 154 // Randomly generates stream_id. | 165 // Randomly generates stream_id. |
| 155 VideoRtpSender(VideoTrackInterface* track, VideoProviderInterface* provider); | 166 // |channel| can be null if one does not exist yet. |
| 167 VideoRtpSender(VideoTrackInterface* track, cricket::VideoChannel* channel); |
| 156 | 168 |
| 157 // Randomly generates id and stream_id. | 169 // Randomly generates id and stream_id. |
| 158 explicit VideoRtpSender(VideoProviderInterface* provider); | 170 // |channel| can be null if one does not exist yet. |
| 171 explicit VideoRtpSender(cricket::VideoChannel* channel); |
| 159 | 172 |
| 160 virtual ~VideoRtpSender(); | 173 virtual ~VideoRtpSender(); |
| 161 | 174 |
| 162 // ObserverInterface implementation | 175 // ObserverInterface implementation |
| 163 void OnChanged() override; | 176 void OnChanged() override; |
| 164 | 177 |
| 165 // RtpSenderInterface implementation | 178 // RtpSenderInterface implementation |
| 166 bool SetTrack(MediaStreamTrackInterface* track) override; | 179 bool SetTrack(MediaStreamTrackInterface* track) override; |
| 167 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 180 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
| 168 return track_; | 181 return track_; |
| (...skipping 18 matching lines...) Expand all Loading... |
| 187 // RtpSenderInternal implementation. | 200 // RtpSenderInternal implementation. |
| 188 void SetSsrc(uint32_t ssrc) override; | 201 void SetSsrc(uint32_t ssrc) override; |
| 189 | 202 |
| 190 void set_stream_id(const std::string& stream_id) override { | 203 void set_stream_id(const std::string& stream_id) override { |
| 191 stream_id_ = stream_id; | 204 stream_id_ = stream_id; |
| 192 } | 205 } |
| 193 std::string stream_id() const override { return stream_id_; } | 206 std::string stream_id() const override { return stream_id_; } |
| 194 | 207 |
| 195 void Stop() override; | 208 void Stop() override; |
| 196 | 209 |
| 210 // Does not take ownership. |
| 211 // Should call SetChannel(nullptr) before |channel| is destroyed. |
| 212 void SetChannel(cricket::VideoChannel* channel) { channel_ = channel; } |
| 213 |
| 197 private: | 214 private: |
| 198 bool can_send_track() const { return track_ && ssrc_; } | 215 bool can_send_track() const { return track_ && ssrc_; } |
| 199 // Helper function to construct options for | 216 // Helper function to construct options for |
| 200 // VideoProviderInterface::SetVideoSend. | 217 // VideoProviderInterface::SetVideoSend. |
| 201 void SetVideoSend(); | 218 void SetVideoSend(); |
| 202 // Helper function to call SetVideoSend with "stop sending" parameters. | 219 // Helper function to call SetVideoSend with "stop sending" parameters. |
| 203 void ClearVideoSend(); | 220 void ClearVideoSend(); |
| 204 | 221 |
| 205 std::string id_; | 222 std::string id_; |
| 206 std::string stream_id_; | 223 std::string stream_id_; |
| 207 VideoProviderInterface* provider_; | 224 cricket::VideoChannel* channel_ = nullptr; |
| 208 rtc::scoped_refptr<VideoTrackInterface> track_; | 225 rtc::scoped_refptr<VideoTrackInterface> track_; |
| 209 uint32_t ssrc_ = 0; | 226 uint32_t ssrc_ = 0; |
| 210 bool cached_track_enabled_ = false; | 227 bool cached_track_enabled_ = false; |
| 211 bool stopped_ = false; | 228 bool stopped_ = false; |
| 212 }; | 229 }; |
| 213 | 230 |
| 214 } // namespace webrtc | 231 } // namespace webrtc |
| 215 | 232 |
| 216 #endif // WEBRTC_API_RTPSENDER_H_ | 233 #endif // WEBRTC_API_RTPSENDER_H_ |
| OLD | NEW |