Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(103)

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

Issue 2072753002: WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed tommi's comments. Created 4 years, 5 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/engine/webrtcmediaengine.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 105 matching lines...) Expand 10 before | Expand all | Expand 10 after
116 bool ApplyOptions(const AudioOptions& options); 116 bool ApplyOptions(const AudioOptions& options);
117 void SetDefaultDevices(); 117 void SetDefaultDevices();
118 118
119 // webrtc::TraceCallback: 119 // webrtc::TraceCallback:
120 void Print(webrtc::TraceLevel level, const char* trace, int length) override; 120 void Print(webrtc::TraceLevel level, const char* trace, int length) override;
121 121
122 void StartAecDump(const std::string& filename); 122 void StartAecDump(const std::string& filename);
123 int CreateVoEChannel(); 123 int CreateVoEChannel();
124 webrtc::AudioDeviceModule* adm(); 124 webrtc::AudioDeviceModule* adm();
125 125
126 AudioCodecs CollectRecvCodecs() const;
127
126 rtc::ThreadChecker signal_thread_checker_; 128 rtc::ThreadChecker signal_thread_checker_;
127 rtc::ThreadChecker worker_thread_checker_; 129 rtc::ThreadChecker worker_thread_checker_;
128 130
129 // The audio device manager. 131 // The audio device manager.
130 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; 132 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
131 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; 133 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
132 // The primary instance of WebRtc VoiceEngine. 134 // The primary instance of WebRtc VoiceEngine.
133 std::unique_ptr<VoEWrapper> voe_wrapper_; 135 std::unique_ptr<VoEWrapper> voe_wrapper_;
134 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 136 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
135 std::vector<AudioCodec> codecs_; 137 std::vector<AudioCodec> send_codecs_;
138 std::vector<AudioCodec> recv_codecs_;
136 std::vector<WebRtcVoiceMediaChannel*> channels_; 139 std::vector<WebRtcVoiceMediaChannel*> channels_;
137 webrtc::Config voe_config_; 140 webrtc::Config voe_config_;
138 bool is_dumping_aec_ = false; 141 bool is_dumping_aec_ = false;
139 142
140 webrtc::AgcConfig default_agc_config_; 143 webrtc::AgcConfig default_agc_config_;
141 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns 144 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns
142 // level controller, and intelligibility_enhancer values, and apply them 145 // level controller, and intelligibility_enhancer values, and apply them
143 // in case they are missing in the audio options. We need to do this because 146 // in case they are missing in the audio options. We need to do this because
144 // SetExtraOptions() will revert to defaults for options which are not 147 // SetExtraOptions() will revert to defaults for options which are not
145 // provided. 148 // provided.
(...skipping 150 matching lines...) Expand 10 before | Expand all | Expand 10 after
296 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; 299 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
297 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 300 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
298 301
299 SendCodecSpec send_codec_spec_; 302 SendCodecSpec send_codec_spec_;
300 303
301 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 304 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
302 }; 305 };
303 } // namespace cricket 306 } // namespace cricket
304 307
305 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 308 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
OLDNEW
« no previous file with comments | « webrtc/media/engine/webrtcmediaengine.h ('k') | webrtc/media/engine/webrtcvoiceengine.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698