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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 |
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94 // easily at any time. | 94 // easily at any time. |
95 bool ApplyOptions(const AudioOptions& options); | 95 bool ApplyOptions(const AudioOptions& options); |
96 void SetDefaultDevices(); | 96 void SetDefaultDevices(); |
97 | 97 |
98 // webrtc::TraceCallback: | 98 // webrtc::TraceCallback: |
99 void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 99 void Print(webrtc::TraceLevel level, const char* trace, int length) override; |
100 | 100 |
101 void StartAecDump(const std::string& filename); | 101 void StartAecDump(const std::string& filename); |
102 int CreateVoEChannel(); | 102 int CreateVoEChannel(); |
103 webrtc::AudioDeviceModule* adm(); | 103 webrtc::AudioDeviceModule* adm(); |
| 104 webrtc::AudioProcessing* apm(); |
104 | 105 |
105 AudioCodecs CollectRecvCodecs() const; | 106 AudioCodecs CollectRecvCodecs() const; |
106 | 107 |
107 rtc::ThreadChecker signal_thread_checker_; | 108 rtc::ThreadChecker signal_thread_checker_; |
108 rtc::ThreadChecker worker_thread_checker_; | 109 rtc::ThreadChecker worker_thread_checker_; |
109 | 110 |
110 // The audio device manager. | 111 // The audio device manager. |
111 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; | 112 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; |
112 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; | 113 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; |
| 114 // Reference to the APM, owned by VoE. |
| 115 webrtc::AudioProcessing* apm_ = nullptr; |
113 // The primary instance of WebRtc VoiceEngine. | 116 // The primary instance of WebRtc VoiceEngine. |
114 std::unique_ptr<VoEWrapper> voe_wrapper_; | 117 std::unique_ptr<VoEWrapper> voe_wrapper_; |
115 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 118 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
116 std::vector<AudioCodec> send_codecs_; | 119 std::vector<AudioCodec> send_codecs_; |
117 std::vector<AudioCodec> recv_codecs_; | 120 std::vector<AudioCodec> recv_codecs_; |
118 std::vector<WebRtcVoiceMediaChannel*> channels_; | 121 std::vector<WebRtcVoiceMediaChannel*> channels_; |
119 webrtc::VoEBase::ChannelConfig channel_config_; | 122 webrtc::VoEBase::ChannelConfig channel_config_; |
120 bool is_dumping_aec_ = false; | 123 bool is_dumping_aec_ = false; |
121 | 124 |
122 webrtc::AgcConfig default_agc_config_; | 125 webrtc::AgcConfig default_agc_config_; |
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265 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 268 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
266 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 269 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
267 | 270 |
268 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; | 271 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; |
269 | 272 |
270 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 273 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
271 }; | 274 }; |
272 } // namespace cricket | 275 } // namespace cricket |
273 | 276 |
274 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 277 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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