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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 <stddef.h> | 7 #include <stddef.h> |
8 #include <stdint.h> | 8 #include <stdint.h> |
9 #include <utility> | 9 #include <utility> |
10 | 10 |
11 #include "base/command_line.h" | 11 #include "base/command_line.h" |
12 #include "base/metrics/field_trial.h" | 12 #include "base/metrics/field_trial.h" |
13 #include "base/metrics/histogram.h" | 13 #include "base/metrics/histogram.h" |
14 #include "base/strings/string_number_conversions.h" | 14 #include "base/strings/string_number_conversions.h" |
15 #include "base/trace_event/trace_event.h" | 15 #include "base/trace_event/trace_event.h" |
16 #include "build/build_config.h" | 16 #include "build/build_config.h" |
17 #include "content/public/common/content_switches.h" | 17 #include "content/public/common/content_switches.h" |
18 #include "content/renderer/media/media_stream_audio_processor_options.h" | 18 #include "content/renderer/media/media_stream_audio_processor_options.h" |
19 #include "content/renderer/media/rtc_media_constraints.h" | 19 #include "content/renderer/media/rtc_media_constraints.h" |
20 #include "content/renderer/media/webrtc_audio_device_impl.h" | 20 #include "content/renderer/media/webrtc_audio_device_impl.h" |
21 #include "media/audio/audio_parameters.h" | 21 #include "media/audio/audio_parameters.h" |
22 #include "media/base/audio_converter.h" | 22 #include "media/base/audio_converter.h" |
23 #include "media/base/audio_fifo.h" | 23 #include "media/base/audio_fifo.h" |
24 #include "media/base/channel_layout.h" | 24 #include "media/base/channel_layout.h" |
25 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | 25 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
26 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface
.h" | 26 #include "third_party/webrtc/api/mediaconstraintsinterface.h" |
27 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" | 27 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" |
28 | 28 |
29 namespace content { | 29 namespace content { |
30 | 30 |
31 namespace { | 31 namespace { |
32 | 32 |
33 using webrtc::AudioProcessing; | 33 using webrtc::AudioProcessing; |
34 using webrtc::NoiseSuppression; | 34 using webrtc::NoiseSuppression; |
35 | 35 |
36 const int kAudioProcessingNumberOfChannels = 1; | 36 const int kAudioProcessingNumberOfChannels = 1; |
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747 if (echo_information_) { | 747 if (echo_information_) { |
748 echo_information_.get()->UpdateAecDelayStats(ap->echo_cancellation()); | 748 echo_information_.get()->UpdateAecDelayStats(ap->echo_cancellation()); |
749 } | 749 } |
750 | 750 |
751 // Return 0 if the volume hasn't been changed, and otherwise the new volume. | 751 // Return 0 if the volume hasn't been changed, and otherwise the new volume. |
752 return (agc->stream_analog_level() == volume) ? | 752 return (agc->stream_analog_level() == volume) ? |
753 0 : agc->stream_analog_level(); | 753 0 : agc->stream_analog_level(); |
754 } | 754 } |
755 | 755 |
756 } // namespace content | 756 } // namespace content |
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