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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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247 public: | 247 public: |
248 WebRtcVideoSendStream( | 248 WebRtcVideoSendStream( |
249 webrtc::Call* call, | 249 webrtc::Call* call, |
250 const StreamParams& sp, | 250 const StreamParams& sp, |
251 const webrtc::VideoSendStream::Config& config, | 251 const webrtc::VideoSendStream::Config& config, |
252 const VideoOptions& options, | 252 const VideoOptions& options, |
253 WebRtcVideoEncoderFactory* external_encoder_factory, | 253 WebRtcVideoEncoderFactory* external_encoder_factory, |
254 bool enable_cpu_overuse_detection, | 254 bool enable_cpu_overuse_detection, |
255 int max_bitrate_bps, | 255 int max_bitrate_bps, |
256 const rtc::Optional<VideoCodecSettings>& codec_settings, | 256 const rtc::Optional<VideoCodecSettings>& codec_settings, |
257 const std::vector<webrtc::RtpExtension>& rtp_extensions, | 257 const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions, |
258 const VideoSendParameters& send_params); | 258 const VideoSendParameters& send_params); |
259 virtual ~WebRtcVideoSendStream(); | 259 virtual ~WebRtcVideoSendStream(); |
260 | 260 |
261 void SetSendParameters(const ChangedSendParameters& send_params); | 261 void SetSendParameters(const ChangedSendParameters& send_params); |
262 bool SetRtpParameters(const webrtc::RtpParameters& parameters); | 262 bool SetRtpParameters(const webrtc::RtpParameters& parameters); |
263 webrtc::RtpParameters GetRtpParameters() const; | 263 webrtc::RtpParameters GetRtpParameters() const; |
264 | 264 |
265 void OnFrame(const cricket::VideoFrame& frame) override; | 265 void OnFrame(const cricket::VideoFrame& frame) override; |
266 bool SetVideoSend(bool mute, | 266 bool SetVideoSend(bool mute, |
267 const VideoOptions* options, | 267 const VideoOptions* options, |
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344 | 344 |
345 AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec) | 345 AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec) |
346 EXCLUSIVE_LOCKS_REQUIRED(lock_); | 346 EXCLUSIVE_LOCKS_REQUIRED(lock_); |
347 void DestroyVideoEncoder(AllocatedEncoder* encoder) | 347 void DestroyVideoEncoder(AllocatedEncoder* encoder) |
348 EXCLUSIVE_LOCKS_REQUIRED(lock_); | 348 EXCLUSIVE_LOCKS_REQUIRED(lock_); |
349 void SetCodec(const VideoCodecSettings& codec) | 349 void SetCodec(const VideoCodecSettings& codec) |
350 EXCLUSIVE_LOCKS_REQUIRED(lock_); | 350 EXCLUSIVE_LOCKS_REQUIRED(lock_); |
351 void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_); | 351 void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_); |
352 webrtc::VideoEncoderConfig CreateVideoEncoderConfig( | 352 webrtc::VideoEncoderConfig CreateVideoEncoderConfig( |
353 const Dimensions& dimensions, | 353 const Dimensions& dimensions, |
354 bool encode_from_texture, | |
354 const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_); | 355 const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_); |
355 void SetDimensions(int width, int height) | 356 void ReconfigureEncoderIfNecessary() EXCLUSIVE_LOCKS_REQUIRED(lock_); |
356 EXCLUSIVE_LOCKS_REQUIRED(lock_); | |
357 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); | 357 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); |
358 | 358 |
359 // Calls Start or Stop according to whether or not |sending_| is true, | 359 // Calls Start or Stop according to whether or not |sending_| is true, |
360 // and whether or not the encoding in |rtp_parameters_| is active. | 360 // and whether or not the encoding in |rtp_parameters_| is active. |
361 void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_); | 361 void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_); |
362 | 362 |
363 rtc::ThreadChecker thread_checker_; | 363 rtc::ThreadChecker thread_checker_; |
364 rtc::AsyncInvoker invoker_; | 364 rtc::AsyncInvoker invoker_; |
365 rtc::Thread* worker_thread_; | 365 rtc::Thread* worker_thread_; |
366 const std::vector<uint32_t> ssrcs_; | 366 const std::vector<uint32_t> ssrcs_; |
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388 // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_. | 388 // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_. |
389 // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only | 389 // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only |
390 // one stream per MediaChannel. | 390 // one stream per MediaChannel. |
391 webrtc::RtpParameters rtp_parameters_ GUARDED_BY(lock_); | 391 webrtc::RtpParameters rtp_parameters_ GUARDED_BY(lock_); |
392 bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); | 392 bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); |
393 VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); | 393 VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); |
394 AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); | 394 AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); |
395 Dimensions last_dimensions_ GUARDED_BY(lock_); | 395 Dimensions last_dimensions_ GUARDED_BY(lock_); |
396 webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) = | 396 webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) = |
397 webrtc::kVideoRotation_0; | 397 webrtc::kVideoRotation_0; |
398 bool last_frame_is_texture_ GUARDED_BY(lock_) = false; | |
pthatcher1
2016/06/15 20:40:23
We've got last_dimensions_, last_rotation_, and la
skvlad
2016/06/15 22:10:36
Replaced with VideoFrameInfo. Great suggestion, th
| |
398 | 399 |
399 bool sending_ GUARDED_BY(lock_); | 400 bool sending_ GUARDED_BY(lock_); |
400 | 401 |
401 // The timestamp of the first frame received | 402 // The timestamp of the first frame received |
402 // Used to generate the timestamps of subsequent frames | 403 // Used to generate the timestamps of subsequent frames |
403 rtc::Optional<int64_t> first_frame_timestamp_ms_ GUARDED_BY(lock_); | 404 rtc::Optional<int64_t> first_frame_timestamp_ms_ GUARDED_BY(lock_); |
404 | 405 |
405 // The timestamp of the last frame received | 406 // The timestamp of the last frame received |
406 // Used to generate timestamp for the black frame when source is removed | 407 // Used to generate timestamp for the black frame when source is removed |
407 int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_); | 408 int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_); |
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531 rtc::CriticalSection stream_crit_; | 532 rtc::CriticalSection stream_crit_; |
532 // Using primary-ssrc (first ssrc) as key. | 533 // Using primary-ssrc (first ssrc) as key. |
533 std::map<uint32_t, WebRtcVideoSendStream*> send_streams_ | 534 std::map<uint32_t, WebRtcVideoSendStream*> send_streams_ |
534 GUARDED_BY(stream_crit_); | 535 GUARDED_BY(stream_crit_); |
535 std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_ | 536 std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_ |
536 GUARDED_BY(stream_crit_); | 537 GUARDED_BY(stream_crit_); |
537 std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_); | 538 std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_); |
538 std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_); | 539 std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_); |
539 | 540 |
540 rtc::Optional<VideoCodecSettings> send_codec_; | 541 rtc::Optional<VideoCodecSettings> send_codec_; |
541 std::vector<webrtc::RtpExtension> send_rtp_extensions_; | 542 rtc::Optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_; |
542 | 543 |
543 WebRtcVideoEncoderFactory* const external_encoder_factory_; | 544 WebRtcVideoEncoderFactory* const external_encoder_factory_; |
544 WebRtcVideoDecoderFactory* const external_decoder_factory_; | 545 WebRtcVideoDecoderFactory* const external_decoder_factory_; |
545 std::vector<VideoCodecSettings> recv_codecs_; | 546 std::vector<VideoCodecSettings> recv_codecs_; |
546 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 547 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
547 webrtc::Call::Config::BitrateConfig bitrate_config_; | 548 webrtc::Call::Config::BitrateConfig bitrate_config_; |
548 // TODO(deadbeef): Don't duplicate information between | 549 // TODO(deadbeef): Don't duplicate information between |
549 // send_params/recv_params, rtp_extensions, options, etc. | 550 // send_params/recv_params, rtp_extensions, options, etc. |
550 VideoSendParameters send_params_; | 551 VideoSendParameters send_params_; |
551 VideoOptions default_send_options_; | 552 VideoOptions default_send_options_; |
552 VideoRecvParameters recv_params_; | 553 VideoRecvParameters recv_params_; |
553 bool red_disabled_by_remote_side_; | 554 bool red_disabled_by_remote_side_; |
554 }; | 555 }; |
555 | 556 |
556 } // namespace cricket | 557 } // namespace cricket |
557 | 558 |
558 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ | 559 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ |
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