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

Issue 2060403002: Add task queue to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@move_getpadding
Patch Set: Fix audio thread check when adding audio to bitrateallocator. Created 4 years, 4 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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241 241
242 // Wrapper for the sender part, this is where the source is connected and 242 // Wrapper for the sender part, this is where the source is connected and
243 // frames are then converted from cricket frames to webrtc frames. 243 // frames are then converted from cricket frames to webrtc frames.
244 class WebRtcVideoSendStream 244 class WebRtcVideoSendStream
245 : public rtc::VideoSinkInterface<cricket::VideoFrame>, 245 : public rtc::VideoSinkInterface<cricket::VideoFrame>,
246 public webrtc::LoadObserver { 246 public webrtc::LoadObserver {
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 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 rtc::Optional<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);
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277 VideoSenderInfo GetVideoSenderInfo(bool log_stats); 277 VideoSenderInfo GetVideoSenderInfo(bool log_stats);
278 void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info); 278 void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
279 279
280 private: 280 private:
281 // Parameters needed to reconstruct the underlying stream. 281 // Parameters needed to reconstruct the underlying stream.
282 // webrtc::VideoSendStream doesn't support setting a lot of options on the 282 // webrtc::VideoSendStream doesn't support setting a lot of options on the
283 // fly, so when those need to be changed we tear down and reconstruct with 283 // fly, so when those need to be changed we tear down and reconstruct with
284 // similar parameters depending on which options changed etc. 284 // similar parameters depending on which options changed etc.
285 struct VideoSendStreamParameters { 285 struct VideoSendStreamParameters {
286 VideoSendStreamParameters( 286 VideoSendStreamParameters(
287 const webrtc::VideoSendStream::Config& config, 287 webrtc::VideoSendStream::Config config,
288 const VideoOptions& options, 288 const VideoOptions& options,
289 int max_bitrate_bps, 289 int max_bitrate_bps,
290 const rtc::Optional<VideoCodecSettings>& codec_settings); 290 const rtc::Optional<VideoCodecSettings>& codec_settings);
291 webrtc::VideoSendStream::Config config; 291 webrtc::VideoSendStream::Config config;
292 VideoOptions options; 292 VideoOptions options;
293 int max_bitrate_bps; 293 int max_bitrate_bps;
294 bool conference_mode; 294 bool conference_mode;
295 rtc::Optional<VideoCodecSettings> codec_settings; 295 rtc::Optional<VideoCodecSettings> codec_settings;
296 // Sent resolutions + bitrates etc. by the underlying VideoSendStream, 296 // Sent resolutions + bitrates etc. by the underlying VideoSendStream,
297 // typically changes when setting a new resolution or reconfiguring 297 // typically changes when setting a new resolution or reconfiguring
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552 VideoSendParameters send_params_; 552 VideoSendParameters send_params_;
553 VideoOptions default_send_options_; 553 VideoOptions default_send_options_;
554 VideoRecvParameters recv_params_; 554 VideoRecvParameters recv_params_;
555 bool red_disabled_by_remote_side_; 555 bool red_disabled_by_remote_side_;
556 int64_t last_stats_log_ms_; 556 int64_t last_stats_log_ms_;
557 }; 557 };
558 558
559 } // namespace cricket 559 } // namespace cricket
560 560
561 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ 561 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
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