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

Issue 155863003: Add basic support for "googDucking" to getUserMedia on Windows. (Closed) Base URL: http://git.chromium.org/chromium/src.git@master
Patch Set: Initialize enumeration flag in constructor Created 6 years, 10 months 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_options.h" 5 #include "content/renderer/media/media_stream_audio_processor_options.h"
6 6
7 #include "base/files/file_path.h" 7 #include "base/files/file_path.h"
8 #include "base/logging.h" 8 #include "base/logging.h"
9 #include "base/path_service.h" 9 #include "base/path_service.h"
10 #include "base/strings/utf_string_conversions.h" 10 #include "base/strings/utf_string_conversions.h"
11 #include "content/common/media/media_stream_options.h"
11 #include "content/renderer/media/rtc_media_constraints.h" 12 #include "content/renderer/media/rtc_media_constraints.h"
12 #include "media/audio/audio_parameters.h" 13 #include "media/audio/audio_parameters.h"
13 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" 14 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
14 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" 15 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h"
15 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h " 16 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h "
16 17
17 namespace content { 18 namespace content {
18 19
19 namespace { 20 namespace {
20 21
(...skipping 13 matching lines...) Expand all
34 { webrtc::MediaConstraintsInterface::kAutoGainControl, 35 { webrtc::MediaConstraintsInterface::kAutoGainControl,
35 webrtc::MediaConstraintsInterface::kValueTrue }, 36 webrtc::MediaConstraintsInterface::kValueTrue },
36 { webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl, 37 { webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl,
37 webrtc::MediaConstraintsInterface::kValueTrue }, 38 webrtc::MediaConstraintsInterface::kValueTrue },
38 { webrtc::MediaConstraintsInterface::kNoiseSuppression, 39 { webrtc::MediaConstraintsInterface::kNoiseSuppression,
39 webrtc::MediaConstraintsInterface::kValueTrue }, 40 webrtc::MediaConstraintsInterface::kValueTrue },
40 { webrtc::MediaConstraintsInterface::kHighpassFilter, 41 { webrtc::MediaConstraintsInterface::kHighpassFilter,
41 webrtc::MediaConstraintsInterface::kValueTrue }, 42 webrtc::MediaConstraintsInterface::kValueTrue },
42 { webrtc::MediaConstraintsInterface::kTypingNoiseDetection, 43 { webrtc::MediaConstraintsInterface::kTypingNoiseDetection,
43 webrtc::MediaConstraintsInterface::kValueTrue }, 44 webrtc::MediaConstraintsInterface::kValueTrue },
45 #if defined(OS_WIN)
46 { content::kMediaStreamAudioDucking,
47 webrtc::MediaConstraintsInterface::kValueTrue },
48 #endif
44 }; 49 };
45 50
46 } // namespace 51 } // namespace
47 52
48 void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) { 53 void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) {
49 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { 54 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) {
50 bool already_set_value; 55 bool already_set_value;
51 if (!webrtc::FindConstraint(constraints, kDefaultAudioConstraints[i].key, 56 if (!webrtc::FindConstraint(constraints, kDefaultAudioConstraints[i].key,
52 &already_set_value, NULL)) { 57 &already_set_value, NULL)) {
53 constraints->AddOptional(kDefaultAudioConstraints[i].key, 58 constraints->AddOptional(kDefaultAudioConstraints[i].key,
(...skipping 108 matching lines...) Expand 10 before | Expand all | Expand 10 after
162 const webrtc::GainControl::Mode mode = webrtc::GainControl::kFixedDigital; 167 const webrtc::GainControl::Mode mode = webrtc::GainControl::kFixedDigital;
163 #else 168 #else
164 const webrtc::GainControl::Mode mode = webrtc::GainControl::kAdaptiveAnalog; 169 const webrtc::GainControl::Mode mode = webrtc::GainControl::kAdaptiveAnalog;
165 #endif 170 #endif
166 int err = audio_processing->gain_control()->set_mode(mode); 171 int err = audio_processing->gain_control()->set_mode(mode);
167 err |= audio_processing->gain_control()->Enable(true); 172 err |= audio_processing->gain_control()->Enable(true);
168 CHECK_EQ(err, 0); 173 CHECK_EQ(err, 0);
169 } 174 }
170 175
171 } // namespace content 176 } // namespace content
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