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

Issue 1695663003: Drop VideoOptions from VideoSendParameters. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 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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185 VideoCodec codec; 185 VideoCodec codec;
186 webrtc::FecConfig fec; 186 webrtc::FecConfig fec;
187 int rtx_payload_type; 187 int rtx_payload_type;
188 }; 188 };
189 189
190 struct ChangedSendParameters { 190 struct ChangedSendParameters {
191 // These optionals are unset if not changed. 191 // These optionals are unset if not changed.
192 rtc::Optional<VideoCodecSettings> codec; 192 rtc::Optional<VideoCodecSettings> codec;
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<VideoOptions> options;
196 rtc::Optional<webrtc::RtcpMode> rtcp_mode; 195 rtc::Optional<webrtc::RtcpMode> rtcp_mode;
197 }; 196 };
198 197
199 struct ChangedRecvParameters { 198 struct ChangedRecvParameters {
200 // These optionals are unset if not changed. 199 // These optionals are unset if not changed.
201 rtc::Optional<std::vector<VideoCodecSettings>> codec_settings; 200 rtc::Optional<std::vector<VideoCodecSettings>> codec_settings;
202 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; 201 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
203 rtc::Optional<webrtc::RtcpMode> rtcp_mode; 202 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);
214 void SetOptions(const VideoOptions& options); 213 void SetOptions(uint32_t ssrc, const VideoOptions& options);
215 214
216 void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config, 215 void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
217 const StreamParams& sp) const; 216 const StreamParams& sp) const;
218 bool CodecIsExternallySupported(const std::string& name) const; 217 bool CodecIsExternallySupported(const std::string& name) const;
219 bool ValidateSendSsrcAvailability(const StreamParams& sp) const 218 bool ValidateSendSsrcAvailability(const StreamParams& sp) const
220 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); 219 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
221 bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const 220 bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
222 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); 221 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
223 void DeleteReceiveStream(WebRtcVideoReceiveStream* stream) 222 void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
224 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); 223 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
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517 VideoOptions options_; 516 VideoOptions options_;
518 // TODO(deadbeef): Don't duplicate information between 517 // TODO(deadbeef): Don't duplicate information between
519 // send_params/recv_params, rtp_extensions, options, etc. 518 // send_params/recv_params, rtp_extensions, options, etc.
520 VideoSendParameters send_params_; 519 VideoSendParameters send_params_;
521 VideoRecvParameters recv_params_; 520 VideoRecvParameters recv_params_;
522 }; 521 };
523 522
524 } // namespace cricket 523 } // namespace cricket
525 524
526 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ 525 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
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