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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/mediastreamprovider.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 | 27 |
| 28 namespace webrtc { | 28 namespace webrtc { |
| 29 | 29 |
| 30 // Internal interface used by PeerConnection. |
| 31 class RtpSenderInternal : public RtpSenderInterface { |
| 32 public: |
| 33 // 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 // underlying transport (this occurs if the sender isn't seen in a local |
| 36 // description). |
| 37 virtual void SetSsrc(uint32_t ssrc) = 0; |
| 38 |
| 39 // TODO(deadbeef): Support one sender having multiple stream ids. |
| 40 virtual void set_stream_id(const std::string& stream_id) = 0; |
| 41 virtual std::string stream_id() const = 0; |
| 42 |
| 43 virtual void Stop() = 0; |
| 44 }; |
| 45 |
| 30 // LocalAudioSinkAdapter receives data callback as a sink to the local | 46 // LocalAudioSinkAdapter receives data callback as a sink to the local |
| 31 // AudioTrack, and passes the data to the sink of AudioSource. | 47 // AudioTrack, and passes the data to the sink of AudioSource. |
| 32 class LocalAudioSinkAdapter : public AudioTrackSinkInterface, | 48 class LocalAudioSinkAdapter : public AudioTrackSinkInterface, |
| 33 public cricket::AudioSource { | 49 public cricket::AudioSource { |
| 34 public: | 50 public: |
| 35 LocalAudioSinkAdapter(); | 51 LocalAudioSinkAdapter(); |
| 36 virtual ~LocalAudioSinkAdapter(); | 52 virtual ~LocalAudioSinkAdapter(); |
| 37 | 53 |
| 38 private: | 54 private: |
| 39 // AudioSinkInterface implementation. | 55 // AudioSinkInterface implementation. |
| 40 void OnData(const void* audio_data, | 56 void OnData(const void* audio_data, |
| 41 int bits_per_sample, | 57 int bits_per_sample, |
| 42 int sample_rate, | 58 int sample_rate, |
| 43 size_t number_of_channels, | 59 size_t number_of_channels, |
| 44 size_t number_of_frames) override; | 60 size_t number_of_frames) override; |
| 45 | 61 |
| 46 // cricket::AudioSource implementation. | 62 // cricket::AudioSource implementation. |
| 47 void SetSink(cricket::AudioSource::Sink* sink) override; | 63 void SetSink(cricket::AudioSource::Sink* sink) override; |
| 48 | 64 |
| 49 cricket::AudioSource::Sink* sink_; | 65 cricket::AudioSource::Sink* sink_; |
| 50 // Critical section protecting |sink_|. | 66 // Critical section protecting |sink_|. |
| 51 rtc::CriticalSection lock_; | 67 rtc::CriticalSection lock_; |
| 52 }; | 68 }; |
| 53 | 69 |
| 54 class AudioRtpSender : public ObserverInterface, | 70 class AudioRtpSender : public ObserverInterface, |
| 55 public rtc::RefCountedObject<RtpSenderInterface> { | 71 public rtc::RefCountedObject<RtpSenderInternal> { |
| 56 public: | 72 public: |
| 57 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called | 73 // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called |
| 58 // at the appropriate times. | 74 // at the appropriate times. |
| 59 AudioRtpSender(AudioTrackInterface* track, | 75 AudioRtpSender(AudioTrackInterface* track, |
| 60 const std::string& stream_id, | 76 const std::string& stream_id, |
| 61 AudioProviderInterface* provider, | 77 AudioProviderInterface* provider, |
| 62 StatsCollector* stats); | 78 StatsCollector* stats); |
| 63 | 79 |
| 64 // Randomly generates stream_id. | 80 // Randomly generates stream_id. |
| 65 AudioRtpSender(AudioTrackInterface* track, | 81 AudioRtpSender(AudioTrackInterface* track, |
| 66 AudioProviderInterface* provider, | 82 AudioProviderInterface* provider, |
| 67 StatsCollector* stats); | 83 StatsCollector* stats); |
| 68 | 84 |
| 69 // Randomly generates id and stream_id. | 85 // Randomly generates id and stream_id. |
| 70 AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats); | 86 AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats); |
| 71 | 87 |
| 72 virtual ~AudioRtpSender(); | 88 virtual ~AudioRtpSender(); |
| 73 | 89 |
| 74 // ObserverInterface implementation | 90 // ObserverInterface implementation |
| 75 void OnChanged() override; | 91 void OnChanged() override; |
| 76 | 92 |
| 77 // RtpSenderInterface implementation | 93 // RtpSenderInterface implementation |
| 78 bool SetTrack(MediaStreamTrackInterface* track) override; | 94 bool SetTrack(MediaStreamTrackInterface* track) override; |
| 79 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 95 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
| 80 return track_.get(); | 96 return track_; |
| 81 } | 97 } |
| 82 | 98 |
| 83 void SetSsrc(uint32_t ssrc) override; | |
| 84 | |
| 85 uint32_t ssrc() const override { return ssrc_; } | 99 uint32_t ssrc() const override { return ssrc_; } |
| 86 | 100 |
| 87 cricket::MediaType media_type() const override { | 101 cricket::MediaType media_type() const override { |
| 88 return cricket::MEDIA_TYPE_AUDIO; | 102 return cricket::MEDIA_TYPE_AUDIO; |
| 89 } | 103 } |
| 90 | 104 |
| 91 std::string id() const override { return id_; } | 105 std::string id() const override { return id_; } |
| 92 | 106 |
| 107 std::vector<std::string> stream_ids() const override { |
| 108 std::vector<std::string> ret = {stream_id_}; |
| 109 return ret; |
| 110 } |
| 111 |
| 112 RtpParameters GetParameters() const override; |
| 113 bool SetParameters(const RtpParameters& parameters) override; |
| 114 |
| 115 // RtpSenderInternal implementation. |
| 116 void SetSsrc(uint32_t ssrc) override; |
| 117 |
| 93 void set_stream_id(const std::string& stream_id) override { | 118 void set_stream_id(const std::string& stream_id) override { |
| 94 stream_id_ = stream_id; | 119 stream_id_ = stream_id; |
| 95 } | 120 } |
| 96 std::string stream_id() const override { return stream_id_; } | 121 std::string stream_id() const override { return stream_id_; } |
| 97 | 122 |
| 98 void Stop() override; | 123 void Stop() override; |
| 99 | 124 |
| 100 RtpParameters GetParameters() const override; | |
| 101 bool SetParameters(const RtpParameters& parameters) override; | |
| 102 | |
| 103 private: | 125 private: |
| 104 // TODO(nisse): Since SSRC == 0 is technically valid, figure out | 126 // TODO(nisse): Since SSRC == 0 is technically valid, figure out |
| 105 // some other way to test if we have a valid SSRC. | 127 // some other way to test if we have a valid SSRC. |
| 106 bool can_send_track() const { return track_ && ssrc_; } | 128 bool can_send_track() const { return track_ && ssrc_; } |
| 107 // Helper function to construct options for | 129 // Helper function to construct options for |
| 108 // AudioProviderInterface::SetAudioSend. | 130 // AudioProviderInterface::SetAudioSend. |
| 109 void SetAudioSend(); | 131 void SetAudioSend(); |
| 110 | 132 |
| 111 std::string id_; | 133 std::string id_; |
| 112 std::string stream_id_; | 134 std::string stream_id_; |
| 113 AudioProviderInterface* provider_; | 135 AudioProviderInterface* provider_; |
| 114 StatsCollector* stats_; | 136 StatsCollector* stats_; |
| 115 rtc::scoped_refptr<AudioTrackInterface> track_; | 137 rtc::scoped_refptr<AudioTrackInterface> track_; |
| 116 uint32_t ssrc_ = 0; | 138 uint32_t ssrc_ = 0; |
| 117 bool cached_track_enabled_ = false; | 139 bool cached_track_enabled_ = false; |
| 118 bool stopped_ = false; | 140 bool stopped_ = false; |
| 119 | 141 |
| 120 // 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 |
| 121 // cricket::AudioSource. | 143 // cricket::AudioSource. |
| 122 std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_; | 144 std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_; |
| 123 }; | 145 }; |
| 124 | 146 |
| 125 class VideoRtpSender : public ObserverInterface, | 147 class VideoRtpSender : public ObserverInterface, |
| 126 public rtc::RefCountedObject<RtpSenderInterface> { | 148 public rtc::RefCountedObject<RtpSenderInternal> { |
| 127 public: | 149 public: |
| 128 VideoRtpSender(VideoTrackInterface* track, | 150 VideoRtpSender(VideoTrackInterface* track, |
| 129 const std::string& stream_id, | 151 const std::string& stream_id, |
| 130 VideoProviderInterface* provider); | 152 VideoProviderInterface* provider); |
| 131 | 153 |
| 132 // Randomly generates stream_id. | 154 // Randomly generates stream_id. |
| 133 VideoRtpSender(VideoTrackInterface* track, VideoProviderInterface* provider); | 155 VideoRtpSender(VideoTrackInterface* track, VideoProviderInterface* provider); |
| 134 | 156 |
| 135 // Randomly generates id and stream_id. | 157 // Randomly generates id and stream_id. |
| 136 explicit VideoRtpSender(VideoProviderInterface* provider); | 158 explicit VideoRtpSender(VideoProviderInterface* provider); |
| 137 | 159 |
| 138 virtual ~VideoRtpSender(); | 160 virtual ~VideoRtpSender(); |
| 139 | 161 |
| 140 // ObserverInterface implementation | 162 // ObserverInterface implementation |
| 141 void OnChanged() override; | 163 void OnChanged() override; |
| 142 | 164 |
| 143 // RtpSenderInterface implementation | 165 // RtpSenderInterface implementation |
| 144 bool SetTrack(MediaStreamTrackInterface* track) override; | 166 bool SetTrack(MediaStreamTrackInterface* track) override; |
| 145 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 167 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { |
| 146 return track_.get(); | 168 return track_; |
| 147 } | 169 } |
| 148 | 170 |
| 149 void SetSsrc(uint32_t ssrc) override; | |
| 150 | |
| 151 uint32_t ssrc() const override { return ssrc_; } | 171 uint32_t ssrc() const override { return ssrc_; } |
| 152 | 172 |
| 153 cricket::MediaType media_type() const override { | 173 cricket::MediaType media_type() const override { |
| 154 return cricket::MEDIA_TYPE_VIDEO; | 174 return cricket::MEDIA_TYPE_VIDEO; |
| 155 } | 175 } |
| 156 | 176 |
| 157 std::string id() const override { return id_; } | 177 std::string id() const override { return id_; } |
| 158 | 178 |
| 179 std::vector<std::string> stream_ids() const override { |
| 180 std::vector<std::string> ret = {stream_id_}; |
| 181 return ret; |
| 182 } |
| 183 |
| 184 RtpParameters GetParameters() const override; |
| 185 bool SetParameters(const RtpParameters& parameters) override; |
| 186 |
| 187 // RtpSenderInternal implementation. |
| 188 void SetSsrc(uint32_t ssrc) override; |
| 189 |
| 159 void set_stream_id(const std::string& stream_id) override { | 190 void set_stream_id(const std::string& stream_id) override { |
| 160 stream_id_ = stream_id; | 191 stream_id_ = stream_id; |
| 161 } | 192 } |
| 162 std::string stream_id() const override { return stream_id_; } | 193 std::string stream_id() const override { return stream_id_; } |
| 163 | 194 |
| 164 void Stop() override; | 195 void Stop() override; |
| 165 | 196 |
| 166 RtpParameters GetParameters() const override; | |
| 167 bool SetParameters(const RtpParameters& parameters) override; | |
| 168 | |
| 169 private: | 197 private: |
| 170 bool can_send_track() const { return track_ && ssrc_; } | 198 bool can_send_track() const { return track_ && ssrc_; } |
| 171 // Helper function to construct options for | 199 // Helper function to construct options for |
| 172 // VideoProviderInterface::SetVideoSend. | 200 // VideoProviderInterface::SetVideoSend. |
| 173 void SetVideoSend(); | 201 void SetVideoSend(); |
| 174 // Helper function to call SetVideoSend with "stop sending" parameters. | 202 // Helper function to call SetVideoSend with "stop sending" parameters. |
| 175 void ClearVideoSend(); | 203 void ClearVideoSend(); |
| 176 | 204 |
| 177 std::string id_; | 205 std::string id_; |
| 178 std::string stream_id_; | 206 std::string stream_id_; |
| 179 VideoProviderInterface* provider_; | 207 VideoProviderInterface* provider_; |
| 180 rtc::scoped_refptr<VideoTrackInterface> track_; | 208 rtc::scoped_refptr<VideoTrackInterface> track_; |
| 181 uint32_t ssrc_ = 0; | 209 uint32_t ssrc_ = 0; |
| 182 bool cached_track_enabled_ = false; | 210 bool cached_track_enabled_ = false; |
| 183 bool stopped_ = false; | 211 bool stopped_ = false; |
| 184 }; | 212 }; |
| 185 | 213 |
| 186 } // namespace webrtc | 214 } // namespace webrtc |
| 187 | 215 |
| 188 #endif // WEBRTC_API_RTPSENDER_H_ | 216 #endif // WEBRTC_API_RTPSENDER_H_ |
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