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Side by Side Diff: webrtc/api/test/fakeaudiocapturemodule.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
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 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
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620 bool FakeAudioCaptureModule::ShouldStartProcessing() { 620 bool FakeAudioCaptureModule::ShouldStartProcessing() {
621 return recording_ || playing_; 621 return recording_ || playing_;
622 } 622 }
623 623
624 void FakeAudioCaptureModule::UpdateProcessing(bool start) { 624 void FakeAudioCaptureModule::UpdateProcessing(bool start) {
625 if (start) { 625 if (start) {
626 if (!process_thread_) { 626 if (!process_thread_) {
627 process_thread_.reset(new rtc::Thread()); 627 process_thread_.reset(new rtc::Thread());
628 process_thread_->Start(); 628 process_thread_->Start();
629 } 629 }
630 process_thread_->Post(this, MSG_START_PROCESS); 630 process_thread_->Post(RTC_FROM_HERE, this, MSG_START_PROCESS);
631 } else { 631 } else {
632 if (process_thread_) { 632 if (process_thread_) {
633 process_thread_->Stop(); 633 process_thread_->Stop();
634 process_thread_.reset(nullptr); 634 process_thread_.reset(nullptr);
635 } 635 }
636 started_ = false; 636 started_ = false;
637 } 637 }
638 } 638 }
639 639
640 void FakeAudioCaptureModule::StartProcessP() { 640 void FakeAudioCaptureModule::StartProcessP() {
(...skipping 20 matching lines...) Expand all
661 } 661 }
662 if (recording_) { 662 if (recording_) {
663 SendFrameP(); 663 SendFrameP();
664 } 664 }
665 } 665 }
666 666
667 next_frame_time_ += kTimePerFrameMs; 667 next_frame_time_ += kTimePerFrameMs;
668 const int64_t current_time = rtc::TimeMillis(); 668 const int64_t current_time = rtc::TimeMillis();
669 const int64_t wait_time = 669 const int64_t wait_time =
670 (next_frame_time_ > current_time) ? next_frame_time_ - current_time : 0; 670 (next_frame_time_ > current_time) ? next_frame_time_ - current_time : 0;
671 process_thread_->PostDelayed(wait_time, this, MSG_RUN_PROCESS); 671 process_thread_->PostDelayed(RTC_FROM_HERE, wait_time, this, MSG_RUN_PROCESS);
672 } 672 }
673 673
674 void FakeAudioCaptureModule::ReceiveFrameP() { 674 void FakeAudioCaptureModule::ReceiveFrameP() {
675 ASSERT(process_thread_->IsCurrent()); 675 ASSERT(process_thread_->IsCurrent());
676 { 676 {
677 rtc::CritScope cs(&crit_callback_); 677 rtc::CritScope cs(&crit_callback_);
678 if (!audio_callback_) { 678 if (!audio_callback_) {
679 return; 679 return;
680 } 680 }
681 ResetRecBuffer(); 681 ResetRecBuffer();
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716 kNumberOfChannels, 716 kNumberOfChannels,
717 kSamplesPerSecond, kTotalDelayMs, 717 kSamplesPerSecond, kTotalDelayMs,
718 kClockDriftMs, current_mic_level, 718 kClockDriftMs, current_mic_level,
719 key_pressed, 719 key_pressed,
720 current_mic_level) != 0) { 720 current_mic_level) != 0) {
721 ASSERT(false); 721 ASSERT(false);
722 } 722 }
723 SetMicrophoneVolume(current_mic_level); 723 SetMicrophoneVolume(current_mic_level);
724 } 724 }
725 725
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