Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(755)

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

Issue 1713493003: Enabling rtcp-rsize negotiation and fixing some issues with it. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing unit test and adding more TODOs. Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/media/base/mediachannel.h ('k') | webrtc/media/engine/webrtcvideoengine2.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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
(...skipping 182 matching lines...) Expand 10 before | Expand all | Expand 10 after
193 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; 193 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
194 rtc::Optional<int> max_bandwidth_bps; 194 rtc::Optional<int> max_bandwidth_bps;
195 rtc::Optional<bool> conference_mode; 195 rtc::Optional<bool> conference_mode;
196 rtc::Optional<webrtc::RtcpMode> rtcp_mode; 196 rtc::Optional<webrtc::RtcpMode> rtcp_mode;
197 }; 197 };
198 198
199 struct ChangedRecvParameters { 199 struct ChangedRecvParameters {
200 // These optionals are unset if not changed. 200 // These optionals are unset if not changed.
201 rtc::Optional<std::vector<VideoCodecSettings>> codec_settings; 201 rtc::Optional<std::vector<VideoCodecSettings>> codec_settings;
202 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; 202 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
203 rtc::Optional<webrtc::RtcpMode> rtcp_mode;
204 }; 203 };
205 204
206 bool GetChangedSendParameters(const VideoSendParameters& params, 205 bool GetChangedSendParameters(const VideoSendParameters& params,
207 ChangedSendParameters* changed_params) const; 206 ChangedSendParameters* changed_params) const;
208 bool GetChangedRecvParameters(const VideoRecvParameters& params, 207 bool GetChangedRecvParameters(const VideoRecvParameters& params,
209 ChangedRecvParameters* changed_params) const; 208 ChangedRecvParameters* changed_params) const;
210 209
211 bool MuteStream(uint32_t ssrc, bool mute); 210 bool MuteStream(uint32_t ssrc, bool mute);
212 211
213 void SetMaxSendBandwidth(int bps); 212 void SetMaxSendBandwidth(int bps);
(...skipping 190 matching lines...) Expand 10 before | Expand all | Expand 10 after
404 const webrtc::VideoReceiveStream::Config& config, 403 const webrtc::VideoReceiveStream::Config& config,
405 WebRtcVideoDecoderFactory* external_decoder_factory, 404 WebRtcVideoDecoderFactory* external_decoder_factory,
406 bool default_stream, 405 bool default_stream,
407 const std::vector<VideoCodecSettings>& recv_codecs, 406 const std::vector<VideoCodecSettings>& recv_codecs,
408 bool disable_prerenderer_smoothing); 407 bool disable_prerenderer_smoothing);
409 ~WebRtcVideoReceiveStream(); 408 ~WebRtcVideoReceiveStream();
410 409
411 const std::vector<uint32_t>& GetSsrcs() const; 410 const std::vector<uint32_t>& GetSsrcs() const;
412 411
413 void SetLocalSsrc(uint32_t local_ssrc); 412 void SetLocalSsrc(uint32_t local_ssrc);
413 // TODO(deadbeef): Move these feedback parameters into the recv parameters.
414 void SetFeedbackParameters(bool nack_enabled, 414 void SetFeedbackParameters(bool nack_enabled,
415 bool remb_enabled, 415 bool remb_enabled,
416 bool transport_cc_enabled); 416 bool transport_cc_enabled,
417 webrtc::RtcpMode rtcp_mode);
417 void SetRecvParameters(const ChangedRecvParameters& recv_params); 418 void SetRecvParameters(const ChangedRecvParameters& recv_params);
418 419
419 void RenderFrame(const webrtc::VideoFrame& frame, 420 void RenderFrame(const webrtc::VideoFrame& frame,
420 int time_to_render_ms) override; 421 int time_to_render_ms) override;
421 bool IsTextureSupported() const override; 422 bool IsTextureSupported() const override;
422 bool SmoothsRenderedFrames() const override; 423 bool SmoothsRenderedFrames() const override;
423 bool IsDefaultStream() const; 424 bool IsDefaultStream() const;
424 425
425 void SetSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink); 426 void SetSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink);
426 427
(...skipping 104 matching lines...) Expand 10 before | Expand all | Expand 10 after
531 // TODO(deadbeef): Don't duplicate information between 532 // TODO(deadbeef): Don't duplicate information between
532 // send_params/recv_params, rtp_extensions, options, etc. 533 // send_params/recv_params, rtp_extensions, options, etc.
533 VideoSendParameters send_params_; 534 VideoSendParameters send_params_;
534 VideoOptions default_send_options_; 535 VideoOptions default_send_options_;
535 VideoRecvParameters recv_params_; 536 VideoRecvParameters recv_params_;
536 }; 537 };
537 538
538 } // namespace cricket 539 } // namespace cricket
539 540
540 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ 541 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
OLDNEW
« no previous file with comments | « webrtc/media/base/mediachannel.h ('k') | webrtc/media/engine/webrtcvideoengine2.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698