Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(145)

Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 1952123003: Surface the IntelligibilityEnhancer on MediaConstraints (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebasing Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/media/base/mediachannel.h ('k') | webrtc/media/engine/webrtcvoiceengine.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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
(...skipping 103 matching lines...) Expand 10 before | Expand all | Expand 10 after
114 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; 114 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
115 // The primary instance of WebRtc VoiceEngine. 115 // The primary instance of WebRtc VoiceEngine.
116 std::unique_ptr<VoEWrapper> voe_wrapper_; 116 std::unique_ptr<VoEWrapper> voe_wrapper_;
117 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 117 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
118 std::vector<AudioCodec> codecs_; 118 std::vector<AudioCodec> codecs_;
119 std::vector<WebRtcVoiceMediaChannel*> channels_; 119 std::vector<WebRtcVoiceMediaChannel*> channels_;
120 webrtc::Config voe_config_; 120 webrtc::Config voe_config_;
121 bool is_dumping_aec_ = false; 121 bool is_dumping_aec_ = false;
122 122
123 webrtc::AgcConfig default_agc_config_; 123 webrtc::AgcConfig default_agc_config_;
124 // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns 124 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns and
125 // values, and apply them in case they are missing in the audio options. We 125 // intelligibility_enhancer values, and apply them in case they are missing
126 // need to do this because SetExtraOptions() will revert to defaults for 126 // in the audio options. We need to do this because SetExtraOptions() will
127 // options which are not provided. 127 // revert to defaults for options which are not provided.
128 rtc::Optional<bool> extended_filter_aec_; 128 rtc::Optional<bool> extended_filter_aec_;
129 rtc::Optional<bool> delay_agnostic_aec_; 129 rtc::Optional<bool> delay_agnostic_aec_;
130 rtc::Optional<bool> experimental_ns_; 130 rtc::Optional<bool> experimental_ns_;
131 rtc::Optional<bool> intelligibility_enhancer_;
131 132
132 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine); 133 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine);
133 }; 134 };
134 135
135 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses 136 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
136 // WebRtc Voice Engine. 137 // WebRtc Voice Engine.
137 class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, 138 class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
138 public webrtc::Transport { 139 public webrtc::Transport {
139 public: 140 public:
140 WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, 141 WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
(...skipping 150 matching lines...) Expand 10 before | Expand all | Expand 10 after
291 int cng_payload_type = -1; 292 int cng_payload_type = -1;
292 int cng_plfreq = -1; 293 int cng_plfreq = -1;
293 webrtc::CodecInst codec_inst; 294 webrtc::CodecInst codec_inst;
294 } send_codec_spec_; 295 } send_codec_spec_;
295 296
296 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 297 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
297 }; 298 };
298 } // namespace cricket 299 } // namespace cricket
299 300
300 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 301 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
OLDNEW
« no previous file with comments | « webrtc/media/base/mediachannel.h ('k') | webrtc/media/engine/webrtcvoiceengine.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698