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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 |
| 11 #ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 11 #ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
| 12 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 12 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
| 13 | 13 |
| 14 #include <map> | 14 #include <map> |
| 15 #include <memory> | 15 #include <memory> |
| 16 #include <string> | 16 #include <string> |
| 17 #include <vector> | 17 #include <vector> |
| 18 | 18 |
| 19 #include "webrtc/api/call/audio_state.h" | 19 #include "webrtc/api/call/audio_state.h" |
| 20 #include "webrtc/base/buffer.h" | 20 #include "webrtc/base/buffer.h" |
| 21 #include "webrtc/base/constructormagic.h" | 21 #include "webrtc/base/constructormagic.h" |
| 22 #include "webrtc/base/networkroute.h" | 22 #include "webrtc/base/networkroute.h" |
| 23 #include "webrtc/base/scoped_ref_ptr.h" | 23 #include "webrtc/base/scoped_ref_ptr.h" |
| 24 #include "webrtc/base/stream.h" | 24 #include "webrtc/base/stream.h" |
| 25 #include "webrtc/base/thread_checker.h" | 25 #include "webrtc/base/thread_checker.h" |
| 26 #include "webrtc/call.h" | 26 #include "webrtc/call.h" |
| 27 #include "webrtc/common.h" | |
| 28 #include "webrtc/config.h" | 27 #include "webrtc/config.h" |
| 29 #include "webrtc/media/base/rtputils.h" | 28 #include "webrtc/media/base/rtputils.h" |
| 30 #include "webrtc/media/engine/webrtccommon.h" | 29 #include "webrtc/media/engine/webrtccommon.h" |
| 31 #include "webrtc/media/engine/webrtcvoe.h" | 30 #include "webrtc/media/engine/webrtcvoe.h" |
| 31 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
| 32 #include "webrtc/pc/channel.h" | 32 #include "webrtc/pc/channel.h" |
| 33 | 33 |
| 34 namespace cricket { | 34 namespace cricket { |
| 35 | 35 |
| 36 class AudioDeviceModule; | 36 class AudioDeviceModule; |
| 37 class AudioSource; | 37 class AudioSource; |
| 38 class VoEWrapper; | 38 class VoEWrapper; |
| 39 class WebRtcVoiceMediaChannel; | 39 class WebRtcVoiceMediaChannel; |
| 40 | 40 |
| 41 struct SendCodecSpec { | 41 struct SendCodecSpec { |
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| 130 | 130 |
| 131 // The audio device manager. | 131 // The audio device manager. |
| 132 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; | 132 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; |
| 133 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; | 133 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; |
| 134 // The primary instance of WebRtc VoiceEngine. | 134 // The primary instance of WebRtc VoiceEngine. |
| 135 std::unique_ptr<VoEWrapper> voe_wrapper_; | 135 std::unique_ptr<VoEWrapper> voe_wrapper_; |
| 136 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 136 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
| 137 std::vector<AudioCodec> send_codecs_; | 137 std::vector<AudioCodec> send_codecs_; |
| 138 std::vector<AudioCodec> recv_codecs_; | 138 std::vector<AudioCodec> recv_codecs_; |
| 139 std::vector<WebRtcVoiceMediaChannel*> channels_; | 139 std::vector<WebRtcVoiceMediaChannel*> channels_; |
| 140 webrtc::Config voe_config_; | 140 webrtc::AudioCodingModule::Config acm_config_; |
| 141 bool is_dumping_aec_ = false; | 141 bool is_dumping_aec_ = false; |
| 142 | 142 |
| 143 webrtc::AgcConfig default_agc_config_; | 143 webrtc::AgcConfig default_agc_config_; |
| 144 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns | 144 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns |
| 145 // level controller, and intelligibility_enhancer values, and apply them | 145 // level controller, and intelligibility_enhancer values, and apply them |
| 146 // 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 |
| 147 // SetExtraOptions() will revert to defaults for options which are not | 147 // SetExtraOptions() will revert to defaults for options which are not |
| 148 // provided. | 148 // provided. |
| 149 rtc::Optional<bool> extended_filter_aec_; | 149 rtc::Optional<bool> extended_filter_aec_; |
| 150 rtc::Optional<bool> delay_agnostic_aec_; | 150 rtc::Optional<bool> delay_agnostic_aec_; |
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| 296 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 296 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
| 297 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 297 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| 298 | 298 |
| 299 SendCodecSpec send_codec_spec_; | 299 SendCodecSpec send_codec_spec_; |
| 300 | 300 |
| 301 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 301 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
| 302 }; | 302 }; |
| 303 } // namespace cricket | 303 } // namespace cricket |
| 304 | 304 |
| 305 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 305 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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