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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 |
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154 bool mute, | 154 bool mute, |
155 const VideoOptions* options) override; | 155 const VideoOptions* options) override; |
156 bool AddSendStream(const StreamParams& sp) override; | 156 bool AddSendStream(const StreamParams& sp) override; |
157 bool RemoveSendStream(uint32_t ssrc) override; | 157 bool RemoveSendStream(uint32_t ssrc) override; |
158 bool AddRecvStream(const StreamParams& sp) override; | 158 bool AddRecvStream(const StreamParams& sp) override; |
159 bool AddRecvStream(const StreamParams& sp, bool default_stream); | 159 bool AddRecvStream(const StreamParams& sp, bool default_stream); |
160 bool RemoveRecvStream(uint32_t ssrc) override; | 160 bool RemoveRecvStream(uint32_t ssrc) override; |
161 bool SetSink(uint32_t ssrc, | 161 bool SetSink(uint32_t ssrc, |
162 rtc::VideoSinkInterface<VideoFrame>* sink) override; | 162 rtc::VideoSinkInterface<VideoFrame>* sink) override; |
163 bool GetStats(VideoMediaInfo* info) override; | 163 bool GetStats(VideoMediaInfo* info) override; |
164 bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) override; | 164 void SetSource( |
| 165 uint32_t ssrc, |
| 166 rtc::VideoSourceInterface<cricket::VideoFrame>* source) override; |
165 | 167 |
166 void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, | 168 void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, |
167 const rtc::PacketTime& packet_time) override; | 169 const rtc::PacketTime& packet_time) override; |
168 void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, | 170 void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
169 const rtc::PacketTime& packet_time) override; | 171 const rtc::PacketTime& packet_time) override; |
170 void OnReadyToSend(bool ready) override; | 172 void OnReadyToSend(bool ready) override; |
171 void OnNetworkRouteChanged(const std::string& transport_name, | 173 void OnNetworkRouteChanged(const std::string& transport_name, |
172 const NetworkRoute& network_route) override; | 174 const NetworkRoute& network_route) override; |
173 void SetInterface(NetworkInterface* iface) override; | 175 void SetInterface(NetworkInterface* iface) override; |
174 | 176 |
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210 // These optionals are unset if not changed. | 212 // These optionals are unset if not changed. |
211 rtc::Optional<std::vector<VideoCodecSettings>> codec_settings; | 213 rtc::Optional<std::vector<VideoCodecSettings>> codec_settings; |
212 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; | 214 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; |
213 }; | 215 }; |
214 | 216 |
215 bool GetChangedSendParameters(const VideoSendParameters& params, | 217 bool GetChangedSendParameters(const VideoSendParameters& params, |
216 ChangedSendParameters* changed_params) const; | 218 ChangedSendParameters* changed_params) const; |
217 bool GetChangedRecvParameters(const VideoRecvParameters& params, | 219 bool GetChangedRecvParameters(const VideoRecvParameters& params, |
218 ChangedRecvParameters* changed_params) const; | 220 ChangedRecvParameters* changed_params) const; |
219 | 221 |
220 bool MuteStream(uint32_t ssrc, bool mute); | |
221 | |
222 void SetMaxSendBandwidth(int bps); | 222 void SetMaxSendBandwidth(int bps); |
223 void SetOptions(uint32_t ssrc, const VideoOptions& options); | 223 void SetOptions(uint32_t ssrc, const VideoOptions& options); |
224 | 224 |
225 void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config, | 225 void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config, |
226 const StreamParams& sp) const; | 226 const StreamParams& sp) const; |
227 bool CodecIsExternallySupported(const std::string& name) const; | 227 bool CodecIsExternallySupported(const std::string& name) const; |
228 bool ValidateSendSsrcAvailability(const StreamParams& sp) const | 228 bool ValidateSendSsrcAvailability(const StreamParams& sp) const |
229 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); | 229 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
230 bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const | 230 bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const |
231 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); | 231 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
232 void DeleteReceiveStream(WebRtcVideoReceiveStream* stream) | 232 void DeleteReceiveStream(WebRtcVideoReceiveStream* stream) |
233 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); | 233 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
234 | 234 |
235 static std::string CodecSettingsVectorToString( | 235 static std::string CodecSettingsVectorToString( |
236 const std::vector<VideoCodecSettings>& codecs); | 236 const std::vector<VideoCodecSettings>& codecs); |
237 | 237 |
238 // Wrapper for the sender part, this is where the capturer is connected and | 238 // Wrapper for the sender part, this is where the source is connected and |
239 // frames are then converted from cricket frames to webrtc frames. | 239 // frames are then converted from cricket frames to webrtc frames. |
240 class WebRtcVideoSendStream | 240 class WebRtcVideoSendStream |
241 : public rtc::VideoSinkInterface<cricket::VideoFrame>, | 241 : public rtc::VideoSinkInterface<cricket::VideoFrame>, |
242 public webrtc::LoadObserver { | 242 public webrtc::LoadObserver { |
243 public: | 243 public: |
244 WebRtcVideoSendStream( | 244 WebRtcVideoSendStream( |
245 webrtc::Call* call, | 245 webrtc::Call* call, |
246 const StreamParams& sp, | 246 const StreamParams& sp, |
247 const webrtc::VideoSendStream::Config& config, | 247 const webrtc::VideoSendStream::Config& config, |
248 const VideoOptions& options, | 248 const VideoOptions& options, |
249 WebRtcVideoEncoderFactory* external_encoder_factory, | 249 WebRtcVideoEncoderFactory* external_encoder_factory, |
250 bool enable_cpu_overuse_detection, | 250 bool enable_cpu_overuse_detection, |
251 int max_bitrate_bps, | 251 int max_bitrate_bps, |
252 const rtc::Optional<VideoCodecSettings>& codec_settings, | 252 const rtc::Optional<VideoCodecSettings>& codec_settings, |
253 const std::vector<webrtc::RtpExtension>& rtp_extensions, | 253 const std::vector<webrtc::RtpExtension>& rtp_extensions, |
254 const VideoSendParameters& send_params); | 254 const VideoSendParameters& send_params); |
255 virtual ~WebRtcVideoSendStream(); | 255 virtual ~WebRtcVideoSendStream(); |
256 | 256 |
257 void SetOptions(const VideoOptions& options); | 257 void SetOptions(const VideoOptions& options); |
258 // TODO(pbos): Move logic from SetOptions into this method. | 258 // TODO(pbos): Move logic from SetOptions into this method. |
259 void SetSendParameters(const ChangedSendParameters& send_params); | 259 void SetSendParameters(const ChangedSendParameters& send_params); |
260 bool SetRtpParameters(const webrtc::RtpParameters& parameters); | 260 bool SetRtpParameters(const webrtc::RtpParameters& parameters); |
261 webrtc::RtpParameters GetRtpParameters() const; | 261 webrtc::RtpParameters GetRtpParameters() const; |
262 | 262 |
263 void OnFrame(const cricket::VideoFrame& frame) override; | 263 void OnFrame(const cricket::VideoFrame& frame) override; |
264 bool SetCapturer(VideoCapturer* capturer); | 264 void SetSource(rtc::VideoSourceInterface<cricket::VideoFrame>* source); |
265 void MuteStream(bool mute); | 265 void DisconnectSource(); |
266 bool DisconnectCapturer(); | |
267 | 266 |
268 void SetSend(bool send); | 267 void SetSend(bool send); |
269 | 268 |
270 // Implements webrtc::LoadObserver. | 269 // Implements webrtc::LoadObserver. |
271 void OnLoadUpdate(Load load) override; | 270 void OnLoadUpdate(Load load) override; |
272 | 271 |
273 const std::vector<uint32_t>& GetSsrcs() const; | 272 const std::vector<uint32_t>& GetSsrcs() const; |
274 VideoSenderInfo GetVideoSenderInfo(); | 273 VideoSenderInfo GetVideoSenderInfo(); |
275 void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info); | 274 void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info); |
276 | 275 |
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358 void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_); | 357 void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_); |
359 | 358 |
360 rtc::ThreadChecker thread_checker_; | 359 rtc::ThreadChecker thread_checker_; |
361 rtc::AsyncInvoker invoker_; | 360 rtc::AsyncInvoker invoker_; |
362 rtc::Thread* worker_thread_; | 361 rtc::Thread* worker_thread_; |
363 const std::vector<uint32_t> ssrcs_; | 362 const std::vector<uint32_t> ssrcs_; |
364 const std::vector<SsrcGroup> ssrc_groups_; | 363 const std::vector<SsrcGroup> ssrc_groups_; |
365 webrtc::Call* const call_; | 364 webrtc::Call* const call_; |
366 rtc::VideoSinkWants sink_wants_; | 365 rtc::VideoSinkWants sink_wants_; |
367 // Counter used for deciding if the video resolution is currently | 366 // Counter used for deciding if the video resolution is currently |
368 // restricted by CPU usage. It is reset if |capturer_| is changed. | 367 // restricted by CPU usage. It is reset if |source_| is changed. |
369 int cpu_restricted_counter_; | 368 int cpu_restricted_counter_; |
370 // Total number of times resolution as been requested to be changed due to | 369 // Total number of times resolution as been requested to be changed due to |
371 // CPU adaptation. | 370 // CPU adaptation. |
372 int number_of_cpu_adapt_changes_; | 371 int number_of_cpu_adapt_changes_; |
373 VideoCapturer* capturer_; | 372 rtc::VideoSourceInterface<cricket::VideoFrame>* source_; |
374 WebRtcVideoEncoderFactory* const external_encoder_factory_ | 373 WebRtcVideoEncoderFactory* const external_encoder_factory_ |
375 GUARDED_BY(lock_); | 374 GUARDED_BY(lock_); |
376 | 375 |
377 rtc::CriticalSection lock_; | 376 rtc::CriticalSection lock_; |
378 webrtc::VideoSendStream* stream_ GUARDED_BY(lock_); | 377 webrtc::VideoSendStream* stream_ GUARDED_BY(lock_); |
379 // Contains settings that are the same for all streams in the MediaChannel, | 378 // Contains settings that are the same for all streams in the MediaChannel, |
380 // such as codecs, header extensions, and the global bitrate limit for the | 379 // such as codecs, header extensions, and the global bitrate limit for the |
381 // entire channel. | 380 // entire channel. |
382 VideoSendStreamParameters parameters_ GUARDED_BY(lock_); | 381 VideoSendStreamParameters parameters_ GUARDED_BY(lock_); |
383 // Contains settings that are unique for each stream, such as max_bitrate. | 382 // Contains settings that are unique for each stream, such as max_bitrate. |
384 // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_. | 383 // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_. |
385 // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only | 384 // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only |
386 // one stream per MediaChannel. | 385 // one stream per MediaChannel. |
387 webrtc::RtpParameters rtp_parameters_ GUARDED_BY(lock_); | 386 webrtc::RtpParameters rtp_parameters_ GUARDED_BY(lock_); |
388 bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); | 387 bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); |
389 VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); | 388 VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); |
390 AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); | 389 AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); |
391 Dimensions last_dimensions_ GUARDED_BY(lock_); | 390 Dimensions last_dimensions_ GUARDED_BY(lock_); |
392 webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) = | 391 webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) = |
393 webrtc::kVideoRotation_0; | 392 webrtc::kVideoRotation_0; |
394 | 393 |
395 bool sending_ GUARDED_BY(lock_); | 394 bool sending_ GUARDED_BY(lock_); |
396 bool muted_ GUARDED_BY(lock_); | |
397 | 395 |
398 // The timestamp of the first frame received | 396 // The timestamp of the first frame received |
399 // Used to generate the timestamps of subsequent frames | 397 // Used to generate the timestamps of subsequent frames |
400 int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_); | 398 int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_); |
401 | 399 |
402 // The timestamp of the last frame received | 400 // The timestamp of the last frame received |
403 // Used to generate timestamp for the black frame when capturer is removed | 401 // Used to generate timestamp for the black frame when source is removed |
404 int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_); | 402 int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_); |
405 }; | 403 }; |
406 | 404 |
407 // Wrapper for the receiver part, contains configs etc. that are needed to | 405 // Wrapper for the receiver part, contains configs etc. that are needed to |
408 // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper | 406 // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper |
409 // between rtc::VideoSinkInterface<webrtc::VideoFrame> and | 407 // between rtc::VideoSinkInterface<webrtc::VideoFrame> and |
410 // rtc::VideoSinkInterface<cricket::VideoFrame>. | 408 // rtc::VideoSinkInterface<cricket::VideoFrame>. |
411 class WebRtcVideoReceiveStream | 409 class WebRtcVideoReceiveStream |
412 : public rtc::VideoSinkInterface<webrtc::VideoFrame> { | 410 : public rtc::VideoSinkInterface<webrtc::VideoFrame> { |
413 public: | 411 public: |
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535 // TODO(deadbeef): Don't duplicate information between | 533 // TODO(deadbeef): Don't duplicate information between |
536 // send_params/recv_params, rtp_extensions, options, etc. | 534 // send_params/recv_params, rtp_extensions, options, etc. |
537 VideoSendParameters send_params_; | 535 VideoSendParameters send_params_; |
538 VideoOptions default_send_options_; | 536 VideoOptions default_send_options_; |
539 VideoRecvParameters recv_params_; | 537 VideoRecvParameters recv_params_; |
540 }; | 538 }; |
541 | 539 |
542 } // namespace cricket | 540 } // namespace cricket |
543 | 541 |
544 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ | 542 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ |
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