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Issue 1538463002: Revert of Forward the number of skipped frames by the OS in audio playout. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 5 years ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "content/renderer/media/webrtc_audio_renderer.h" 5 #include "content/renderer/media/webrtc_audio_renderer.h"
6 6
7 #include "base/logging.h" 7 #include "base/logging.h"
8 #include "base/metrics/histogram.h" 8 #include "base/metrics/histogram.h"
9 #include "base/strings/string_util.h" 9 #include "base/strings/string_util.h"
10 #include "base/strings/stringprintf.h" 10 #include "base/strings/stringprintf.h"
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410 410
411 media::OutputDeviceStatus WebRtcAudioRenderer::GetDeviceStatus() { 411 media::OutputDeviceStatus WebRtcAudioRenderer::GetDeviceStatus() {
412 DCHECK(thread_checker_.CalledOnValidThread()); 412 DCHECK(thread_checker_.CalledOnValidThread());
413 if (!sink_.get()) 413 if (!sink_.get())
414 return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL; 414 return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL;
415 415
416 return sink_->GetDeviceStatus(); 416 return sink_->GetDeviceStatus();
417 } 417 }
418 418
419 int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus, 419 int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
420 uint32_t audio_delay_milliseconds, 420 int audio_delay_milliseconds) {
421 uint32_t frames_skipped) {
422 DCHECK(audio_renderer_thread_checker_.CalledOnValidThread()); 421 DCHECK(audio_renderer_thread_checker_.CalledOnValidThread());
423 base::AutoLock auto_lock(lock_); 422 base::AutoLock auto_lock(lock_);
424 if (!source_) 423 if (!source_)
425 return 0; 424 return 0;
426 425
427 DVLOG(2) << "WebRtcAudioRenderer::Render()"; 426 DVLOG(2) << "WebRtcAudioRenderer::Render()";
428 DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds; 427 DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds;
429 428
430 DCHECK_LE(audio_delay_milliseconds, static_cast<uint32_t>(INT_MAX)); 429 audio_delay_milliseconds_ = audio_delay_milliseconds;
431 audio_delay_milliseconds_ = static_cast<int>(audio_delay_milliseconds);
432 430
433 if (audio_fifo_) 431 if (audio_fifo_)
434 audio_fifo_->Consume(audio_bus, audio_bus->frames()); 432 audio_fifo_->Consume(audio_bus, audio_bus->frames());
435 else 433 else
436 SourceCallback(0, audio_bus); 434 SourceCallback(0, audio_bus);
437 435
438 return (state_ == PLAYING) ? audio_bus->frames() : 0; 436 return (state_ == PLAYING) ? audio_bus->frames() : 0;
439 } 437 }
440 438
441 void WebRtcAudioRenderer::OnRenderError() { 439 void WebRtcAudioRenderer::OnRenderError() {
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650 sink_params_ = new_sink_params; 648 sink_params_ = new_sink_params;
651 fifo_delay_milliseconds_ = new_fifo_delay_milliseconds; 649 fifo_delay_milliseconds_ = new_fifo_delay_milliseconds;
652 if (new_audio_fifo.get()) 650 if (new_audio_fifo.get())
653 audio_fifo_ = new_audio_fifo.Pass(); 651 audio_fifo_ = new_audio_fifo.Pass();
654 } 652 }
655 653
656 sink_->Initialize(new_sink_params, this); 654 sink_->Initialize(new_sink_params, this);
657 } 655 }
658 656
659 } // namespace content 657 } // namespace content
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