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