Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(43)

Side by Side Diff: webrtc/modules/video_coding/receiver.cc

Issue 1923713002: Remove VCMRenderBufferSizeCallback. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 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
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 268 matching lines...) Expand 10 before | Expand all | Expand 10 after
279 CriticalSectionScoped cs(crit_sect_); 279 CriticalSectionScoped cs(crit_sect_);
280 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) { 280 if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
281 return -1; 281 return -1;
282 } 282 }
283 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs; 283 max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
284 // Initializing timing to the desired delay. 284 // Initializing timing to the desired delay.
285 timing_->set_min_playout_delay(desired_delay_ms); 285 timing_->set_min_playout_delay(desired_delay_ms);
286 return 0; 286 return 0;
287 } 287 }
288 288
289 int VCMReceiver::RenderBufferSizeMs() {
290 uint32_t timestamp_start = 0u;
291 uint32_t timestamp_end = 0u;
292 // Render timestamps are computed just prior to decoding. Therefore this is
293 // only an estimate based on frames' timestamps and current timing state.
294 jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
295 if (timestamp_start == timestamp_end) {
296 return 0;
297 }
298 // Update timing.
299 const int64_t now_ms = clock_->TimeInMilliseconds();
300 timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
301 // Get render timestamps.
302 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
303 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
304 return render_end - render_start;
305 }
306
307 void VCMReceiver::RegisterStatsCallback( 289 void VCMReceiver::RegisterStatsCallback(
308 VCMReceiveStatisticsCallback* callback) { 290 VCMReceiveStatisticsCallback* callback) {
309 jitter_buffer_.RegisterStatsCallback(callback); 291 jitter_buffer_.RegisterStatsCallback(callback);
310 } 292 }
311 293
312 } // namespace webrtc 294 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/video_coding/receiver.h ('k') | webrtc/modules/video_coding/receiver_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698