Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(596)

Side by Side Diff: webrtc/media/engine/webrtcvideocapturer.cc

Issue 2019423006: Adding more detail to MessageQueue::Dispatch logging. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing one more place where RTC_FROM_HERE wasn't used. Created 4 years, 6 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/fakenetworkinterface.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) 2011 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2011 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 341 matching lines...) Expand 10 before | Expand all | Expand 10 after
352 if (start_thread_->IsCurrent()) { 352 if (start_thread_->IsCurrent()) {
353 SignalFrameCapturedOnStartThread(sample); 353 SignalFrameCapturedOnStartThread(sample);
354 } else { 354 } else {
355 // This currently happens on with at least VideoCaptureModuleV4L2 and 355 // This currently happens on with at least VideoCaptureModuleV4L2 and
356 // possibly other implementations of WebRTC's VideoCaptureModule. 356 // possibly other implementations of WebRTC's VideoCaptureModule.
357 // In order to maintain the threading contract with the upper layers and 357 // In order to maintain the threading contract with the upper layers and
358 // consistency with other capturers such as in Chrome, we need to do a 358 // consistency with other capturers such as in Chrome, we need to do a
359 // thread hop. 359 // thread hop.
360 // Note that Stop() can cause the async invoke call to be cancelled. 360 // Note that Stop() can cause the async invoke call to be cancelled.
361 async_invoker_->AsyncInvoke<void>( 361 async_invoker_->AsyncInvoke<void>(
362 start_thread_, 362 RTC_FROM_HERE, start_thread_,
363 // Note that Bind captures by value, so there's an intermediate copy 363 // Note that Bind captures by value, so there's an intermediate copy
364 // of sample. 364 // of sample.
365 rtc::Bind(&WebRtcVideoCapturer::SignalFrameCapturedOnStartThread, this, 365 rtc::Bind(&WebRtcVideoCapturer::SignalFrameCapturedOnStartThread, this,
366 sample)); 366 sample));
367 } 367 }
368 } 368 }
369 369
370 void WebRtcVideoCapturer::OnCaptureDelayChanged(const int32_t id, 370 void WebRtcVideoCapturer::OnCaptureDelayChanged(const int32_t id,
371 const int32_t delay) { 371 const int32_t delay) {
372 LOG(LS_INFO) << "Capture delay changed to " << delay << " ms"; 372 LOG(LS_INFO) << "Capture delay changed to " << delay << " ms";
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
411 pixel_width = 1; 411 pixel_width = 1;
412 pixel_height = 1; 412 pixel_height = 1;
413 // Convert units from VideoFrame RenderTimeMs to CapturedFrame (nanoseconds). 413 // Convert units from VideoFrame RenderTimeMs to CapturedFrame (nanoseconds).
414 time_stamp = sample.render_time_ms() * rtc::kNumNanosecsPerMillisec; 414 time_stamp = sample.render_time_ms() * rtc::kNumNanosecsPerMillisec;
415 data_size = rtc::checked_cast<uint32_t>(length); 415 data_size = rtc::checked_cast<uint32_t>(length);
416 data = buffer; 416 data = buffer;
417 rotation = sample.rotation(); 417 rotation = sample.rotation();
418 } 418 }
419 419
420 } // namespace cricket 420 } // namespace cricket
OLDNEW
« no previous file with comments | « webrtc/media/base/fakenetworkinterface.h ('k') | webrtc/media/engine/webrtcvideoengine2.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698