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Side by Side Diff: content/renderer/media/media_stream_audio_processor.cc

Issue 717203002: Revert of Reland 597923002: Fix the way how we create webrtc::AudioProcessing in Chrome (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: fix build/android/pylib/utils/isolator.py Created 6 years, 1 month ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 1 // Copyright 2013 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/media_stream_audio_processor.h" 5 #include "content/renderer/media/media_stream_audio_processor.h"
6 6
7 #include "base/command_line.h" 7 #include "base/command_line.h"
8 #include "base/debug/trace_event.h" 8 #include "base/debug/trace_event.h"
9 #if defined(OS_MACOSX) 9 #if defined(OS_MACOSX)
10 #include "base/metrics/field_trial.h" 10 #include "base/metrics/field_trial.h"
11 #endif 11 #endif
12 #include "base/metrics/histogram.h" 12 #include "base/metrics/histogram.h"
13 #include "content/public/common/content_switches.h" 13 #include "content/public/common/content_switches.h"
14 #include "content/renderer/media/media_stream_audio_processor_options.h" 14 #include "content/renderer/media/media_stream_audio_processor_options.h"
15 #include "content/renderer/media/rtc_media_constraints.h" 15 #include "content/renderer/media/rtc_media_constraints.h"
16 #include "content/renderer/media/webrtc_audio_device_impl.h" 16 #include "content/renderer/media/webrtc_audio_device_impl.h"
17 #include "media/audio/audio_parameters.h" 17 #include "media/audio/audio_parameters.h"
18 #include "media/base/audio_converter.h" 18 #include "media/base/audio_converter.h"
19 #include "media/base/audio_fifo.h" 19 #include "media/base/audio_fifo.h"
20 #include "media/base/channel_layout.h" 20 #include "media/base/channel_layout.h"
21 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" 21 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
22 #include "third_party/libjingle/overrides/init_webrtc.h"
23 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" 22 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h"
24 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" 23 #include "third_party/webrtc/modules/audio_processing/typing_detection.h"
25 24
26 namespace content { 25 namespace content {
27 26
28 namespace { 27 namespace {
29 28
30 using webrtc::AudioProcessing; 29 using webrtc::AudioProcessing;
31 30
32 #if defined(OS_ANDROID) 31 #if defined(OS_ANDROID)
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448 if (goog_experimental_aec) 447 if (goog_experimental_aec)
449 config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); 448 config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true));
450 if (goog_experimental_ns) 449 if (goog_experimental_ns)
451 config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(true)); 450 config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(true));
452 #if defined(OS_MACOSX) 451 #if defined(OS_MACOSX)
453 if (base::FieldTrialList::FindFullName("NoReportedDelayOnMac") == "Enabled") 452 if (base::FieldTrialList::FindFullName("NoReportedDelayOnMac") == "Enabled")
454 config.Set<webrtc::ReportedDelay>(new webrtc::ReportedDelay(false)); 453 config.Set<webrtc::ReportedDelay>(new webrtc::ReportedDelay(false));
455 #endif 454 #endif
456 455
457 // Create and configure the webrtc::AudioProcessing. 456 // Create and configure the webrtc::AudioProcessing.
458 audio_processing_.reset(CreateWebRtcAudioProcessing(config)); 457 audio_processing_.reset(webrtc::AudioProcessing::Create(config));
459 458
460 // Enable the audio processing components. 459 // Enable the audio processing components.
461 if (echo_cancellation) { 460 if (echo_cancellation) {
462 EnableEchoCancellation(audio_processing_.get()); 461 EnableEchoCancellation(audio_processing_.get());
463 462
464 if (playout_data_source_) 463 if (playout_data_source_)
465 playout_data_source_->AddPlayoutSink(this); 464 playout_data_source_->AddPlayoutSink(this);
466 465
467 // Prepare for logging echo information. If there are data remaining in 466 // Prepare for logging echo information. If there are data remaining in
468 // |echo_information_| we simply discard it. 467 // |echo_information_| we simply discard it.
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631 vad->stream_has_voice()); 630 vad->stream_has_voice());
632 base::subtle::Release_Store(&typing_detected_, detected); 631 base::subtle::Release_Store(&typing_detected_, detected);
633 } 632 }
634 633
635 // Return 0 if the volume hasn't been changed, and otherwise the new volume. 634 // Return 0 if the volume hasn't been changed, and otherwise the new volume.
636 return (agc->stream_analog_level() == volume) ? 635 return (agc->stream_analog_level() == volume) ?
637 0 : agc->stream_analog_level(); 636 0 : agc->stream_analog_level();
638 } 637 }
639 638
640 } // namespace content 639 } // namespace content
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