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Side by Side Diff: webrtc/media/engine/webrtcvideoengine2.h

Issue 1766653002: Replace SetCapturer and SetCaptureDevice by SetSource. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Make SetSource tolerate unknown ssrc and source == NULL. Created 4 years, 9 months ago
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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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151 bool mute, 151 bool mute,
152 const VideoOptions* options) override; 152 const VideoOptions* options) override;
153 bool AddSendStream(const StreamParams& sp) override; 153 bool AddSendStream(const StreamParams& sp) override;
154 bool RemoveSendStream(uint32_t ssrc) override; 154 bool RemoveSendStream(uint32_t ssrc) override;
155 bool AddRecvStream(const StreamParams& sp) override; 155 bool AddRecvStream(const StreamParams& sp) override;
156 bool AddRecvStream(const StreamParams& sp, bool default_stream); 156 bool AddRecvStream(const StreamParams& sp, bool default_stream);
157 bool RemoveRecvStream(uint32_t ssrc) override; 157 bool RemoveRecvStream(uint32_t ssrc) override;
158 bool SetSink(uint32_t ssrc, 158 bool SetSink(uint32_t ssrc,
159 rtc::VideoSinkInterface<VideoFrame>* sink) override; 159 rtc::VideoSinkInterface<VideoFrame>* sink) override;
160 bool GetStats(VideoMediaInfo* info) override; 160 bool GetStats(VideoMediaInfo* info) override;
161 bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) override; 161 void SetSource(
162 uint32_t ssrc,
163 rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
162 164
163 void OnPacketReceived(rtc::Buffer* packet, 165 void OnPacketReceived(rtc::Buffer* packet,
164 const rtc::PacketTime& packet_time) override; 166 const rtc::PacketTime& packet_time) override;
165 void OnRtcpReceived(rtc::Buffer* packet, 167 void OnRtcpReceived(rtc::Buffer* packet,
166 const rtc::PacketTime& packet_time) override; 168 const rtc::PacketTime& packet_time) override;
167 void OnReadyToSend(bool ready) override; 169 void OnReadyToSend(bool ready) override;
168 void SetInterface(NetworkInterface* iface) override; 170 void SetInterface(NetworkInterface* iface) override;
169 171
170 // Implemented for VideoMediaChannelTest. 172 // Implemented for VideoMediaChannelTest.
171 bool sending() const { return sending_; } 173 bool sending() const { return sending_; }
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217 bool ValidateSendSsrcAvailability(const StreamParams& sp) const 219 bool ValidateSendSsrcAvailability(const StreamParams& sp) const
218 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); 220 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
219 bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const 221 bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
220 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); 222 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
221 void DeleteReceiveStream(WebRtcVideoReceiveStream* stream) 223 void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
222 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); 224 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
223 225
224 static std::string CodecSettingsVectorToString( 226 static std::string CodecSettingsVectorToString(
225 const std::vector<VideoCodecSettings>& codecs); 227 const std::vector<VideoCodecSettings>& codecs);
226 228
227 // Wrapper for the sender part, this is where the capturer is connected and 229 // Wrapper for the sender part, this is where the source is connected and
228 // frames are then converted from cricket frames to webrtc frames. 230 // frames are then converted from cricket frames to webrtc frames.
229 class WebRtcVideoSendStream 231 class WebRtcVideoSendStream
230 : public rtc::VideoSinkInterface<cricket::VideoFrame>, 232 : public rtc::VideoSinkInterface<cricket::VideoFrame>,
231 public webrtc::LoadObserver { 233 public webrtc::LoadObserver {
232 public: 234 public:
233 WebRtcVideoSendStream( 235 WebRtcVideoSendStream(
234 webrtc::Call* call, 236 webrtc::Call* call,
235 const StreamParams& sp, 237 const StreamParams& sp,
236 const webrtc::VideoSendStream::Config& config, 238 const webrtc::VideoSendStream::Config& config,
237 WebRtcVideoEncoderFactory* external_encoder_factory, 239 WebRtcVideoEncoderFactory* external_encoder_factory,
238 bool enable_cpu_overuse_detection, 240 bool enable_cpu_overuse_detection,
239 int max_bitrate_bps, 241 int max_bitrate_bps,
240 const rtc::Optional<VideoCodecSettings>& codec_settings, 242 const rtc::Optional<VideoCodecSettings>& codec_settings,
241 const std::vector<webrtc::RtpExtension>& rtp_extensions, 243 const std::vector<webrtc::RtpExtension>& rtp_extensions,
242 const VideoSendParameters& send_params); 244 const VideoSendParameters& send_params);
243 virtual ~WebRtcVideoSendStream(); 245 virtual ~WebRtcVideoSendStream();
244 246
245 void SetOptions(const VideoOptions& options); 247 void SetOptions(const VideoOptions& options);
246 // TODO(pbos): Move logic from SetOptions into this method. 248 // TODO(pbos): Move logic from SetOptions into this method.
247 void SetSendParameters(const ChangedSendParameters& send_params); 249 void SetSendParameters(const ChangedSendParameters& send_params);
248 250
249 void OnFrame(const cricket::VideoFrame& frame) override; 251 void OnFrame(const cricket::VideoFrame& frame) override;
250 bool SetCapturer(VideoCapturer* capturer); 252 void SetSource(rtc::VideoSourceInterface<cricket::VideoFrame>* source);
251 void MuteStream(bool mute); 253 void MuteStream(bool mute);
252 bool DisconnectCapturer(); 254 void DisconnectSource();
253 255
254 void Start(); 256 void Start();
255 void Stop(); 257 void Stop();
256 258
257 // Implements webrtc::LoadObserver. 259 // Implements webrtc::LoadObserver.
258 void OnLoadUpdate(Load load) override; 260 void OnLoadUpdate(Load load) override;
259 261
260 const std::vector<uint32_t>& GetSsrcs() const; 262 const std::vector<uint32_t>& GetSsrcs() const;
261 VideoSenderInfo GetVideoSenderInfo(); 263 VideoSenderInfo GetVideoSenderInfo();
262 void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info); 264 void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
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340 EXCLUSIVE_LOCKS_REQUIRED(lock_); 342 EXCLUSIVE_LOCKS_REQUIRED(lock_);
341 343
342 rtc::ThreadChecker thread_checker_; 344 rtc::ThreadChecker thread_checker_;
343 rtc::AsyncInvoker invoker_; 345 rtc::AsyncInvoker invoker_;
344 rtc::Thread* worker_thread_; 346 rtc::Thread* worker_thread_;
345 const std::vector<uint32_t> ssrcs_; 347 const std::vector<uint32_t> ssrcs_;
346 const std::vector<SsrcGroup> ssrc_groups_; 348 const std::vector<SsrcGroup> ssrc_groups_;
347 webrtc::Call* const call_; 349 webrtc::Call* const call_;
348 rtc::VideoSinkWants sink_wants_; 350 rtc::VideoSinkWants sink_wants_;
349 // Counter used for deciding if the video resolution is currently 351 // Counter used for deciding if the video resolution is currently
350 // restricted by CPU usage. It is reset if |capturer_| is changed. 352 // restricted by CPU usage. It is reset if |source_| is changed.
351 int cpu_restricted_counter_; 353 int cpu_restricted_counter_;
352 // Total number of times resolution as been requested to be changed due to 354 // Total number of times resolution as been requested to be changed due to
353 // CPU adaptation. 355 // CPU adaptation.
354 int number_of_cpu_adapt_changes_; 356 int number_of_cpu_adapt_changes_;
355 VideoCapturer* capturer_; 357 rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
356 WebRtcVideoEncoderFactory* const external_encoder_factory_ 358 WebRtcVideoEncoderFactory* const external_encoder_factory_
357 GUARDED_BY(lock_); 359 GUARDED_BY(lock_);
358 360
359 rtc::CriticalSection lock_; 361 rtc::CriticalSection lock_;
360 webrtc::VideoSendStream* stream_ GUARDED_BY(lock_); 362 webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
361 VideoSendStreamParameters parameters_ GUARDED_BY(lock_); 363 VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
362 bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); 364 bool pending_encoder_reconfiguration_ GUARDED_BY(lock_);
363 VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); 365 VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_);
364 AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); 366 AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
365 Dimensions last_dimensions_ GUARDED_BY(lock_); 367 Dimensions last_dimensions_ GUARDED_BY(lock_);
366 webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) = 368 webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) =
367 webrtc::kVideoRotation_0; 369 webrtc::kVideoRotation_0;
368 370
369 bool sending_ GUARDED_BY(lock_); 371 bool sending_ GUARDED_BY(lock_);
370 bool muted_ GUARDED_BY(lock_); 372 bool muted_ GUARDED_BY(lock_);
371 373
372 // The timestamp of the first frame received 374 // The timestamp of the first frame received
373 // Used to generate the timestamps of subsequent frames 375 // Used to generate the timestamps of subsequent frames
374 int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_); 376 int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_);
375 377
376 // The timestamp of the last frame received 378 // The timestamp of the last frame received
377 // Used to generate timestamp for the black frame when capturer is removed 379 // Used to generate timestamp for the black frame when source is removed
378 int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_); 380 int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_);
379 }; 381 };
380 382
381 // Wrapper for the receiver part, contains configs etc. that are needed to 383 // Wrapper for the receiver part, contains configs etc. that are needed to
382 // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper 384 // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper
383 // between webrtc::VideoRenderer and cricket::VideoRenderer. 385 // between webrtc::VideoRenderer and cricket::VideoRenderer.
384 class WebRtcVideoReceiveStream : public webrtc::VideoRenderer { 386 class WebRtcVideoReceiveStream : public webrtc::VideoRenderer {
385 public: 387 public:
386 WebRtcVideoReceiveStream( 388 WebRtcVideoReceiveStream(
387 webrtc::Call* call, 389 webrtc::Call* call,
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515 webrtc::Call::Config::BitrateConfig bitrate_config_; 517 webrtc::Call::Config::BitrateConfig bitrate_config_;
516 // TODO(deadbeef): Don't duplicate information between 518 // TODO(deadbeef): Don't duplicate information between
517 // send_params/recv_params, rtp_extensions, options, etc. 519 // send_params/recv_params, rtp_extensions, options, etc.
518 VideoSendParameters send_params_; 520 VideoSendParameters send_params_;
519 VideoRecvParameters recv_params_; 521 VideoRecvParameters recv_params_;
520 }; 522 };
521 523
522 } // namespace cricket 524 } // namespace cricket
523 525
524 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ 526 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
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