OLD | NEW |
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 39 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
50 | 50 |
51 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. | 51 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. |
52 // It uses the WebRtc VoiceEngine library for audio handling. | 52 // It uses the WebRtc VoiceEngine library for audio handling. |
53 class WebRtcVoiceEngine final : public webrtc::TraceCallback { | 53 class WebRtcVoiceEngine final : public webrtc::TraceCallback { |
54 friend class WebRtcVoiceMediaChannel; | 54 friend class WebRtcVoiceMediaChannel; |
55 public: | 55 public: |
56 WebRtcVoiceEngine( | 56 WebRtcVoiceEngine( |
57 webrtc::AudioDeviceModule* adm, | 57 webrtc::AudioDeviceModule* adm, |
58 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, | 58 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, |
59 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, | 59 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
60 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer); | 60 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| 61 rtc::TaskQueue* low_priority_worker_queue); |
61 // Dependency injection for testing. | 62 // Dependency injection for testing. |
62 WebRtcVoiceEngine( | 63 WebRtcVoiceEngine( |
63 webrtc::AudioDeviceModule* adm, | 64 webrtc::AudioDeviceModule* adm, |
64 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, | 65 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, |
65 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, | 66 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
66 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, | 67 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| 68 rtc::TaskQueue* low_priority_worker_queue, |
67 VoEWrapper* voe_wrapper); | 69 VoEWrapper* voe_wrapper); |
68 ~WebRtcVoiceEngine() override; | 70 ~WebRtcVoiceEngine() override; |
69 | 71 |
70 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const; | 72 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const; |
71 VoiceMediaChannel* CreateChannel(webrtc::Call* call, | 73 VoiceMediaChannel* CreateChannel(webrtc::Call* call, |
72 const MediaConfig& config, | 74 const MediaConfig& config, |
73 const AudioOptions& options); | 75 const AudioOptions& options); |
74 | 76 |
75 int GetInputLevel(); | 77 int GetInputLevel(); |
76 | 78 |
(...skipping 28 matching lines...) Expand all Loading... |
105 // ignored. This allows us to selectively turn on and off different options | 107 // ignored. This allows us to selectively turn on and off different options |
106 // easily at any time. | 108 // easily at any time. |
107 bool ApplyOptions(const AudioOptions& options); | 109 bool ApplyOptions(const AudioOptions& options); |
108 | 110 |
109 // webrtc::TraceCallback: | 111 // webrtc::TraceCallback: |
110 void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 112 void Print(webrtc::TraceLevel level, const char* trace, int length) override; |
111 | 113 |
112 void StartAecDump(const std::string& filename); | 114 void StartAecDump(const std::string& filename); |
113 int CreateVoEChannel(); | 115 int CreateVoEChannel(); |
114 | 116 |
115 rtc::TaskQueue low_priority_worker_queue_; | |
116 | |
117 webrtc::AudioDeviceModule* adm(); | 117 webrtc::AudioDeviceModule* adm(); |
118 webrtc::AudioProcessing* apm(); | 118 webrtc::AudioProcessing* apm(); |
119 webrtc::voe::TransmitMixer* transmit_mixer(); | 119 webrtc::voe::TransmitMixer* transmit_mixer(); |
120 | 120 |
121 AudioCodecs CollectCodecs( | 121 AudioCodecs CollectCodecs( |
122 const std::vector<webrtc::AudioCodecSpec>& specs) const; | 122 const std::vector<webrtc::AudioCodecSpec>& specs) const; |
123 | 123 |
124 rtc::ThreadChecker signal_thread_checker_; | 124 rtc::ThreadChecker signal_thread_checker_; |
125 rtc::ThreadChecker worker_thread_checker_; | 125 rtc::ThreadChecker worker_thread_checker_; |
126 | 126 |
127 // The audio device manager. | 127 // The audio device manager. |
128 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; | 128 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; |
129 rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_; | 129 rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_; |
130 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; | 130 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; |
131 // Reference to the APM, owned by VoE. | 131 // Reference to the APM, owned by VoE. |
132 webrtc::AudioProcessing* apm_ = nullptr; | 132 webrtc::AudioProcessing* apm_ = nullptr; |
133 // Reference to the TransmitMixer, owned by VoE. | 133 // Reference to the TransmitMixer, owned by VoE. |
134 webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr; | 134 webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr; |
135 // The primary instance of WebRtc VoiceEngine. | 135 // The primary instance of WebRtc VoiceEngine. |
136 std::unique_ptr<VoEWrapper> voe_wrapper_; | 136 std::unique_ptr<VoEWrapper> voe_wrapper_; |
137 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 137 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
138 std::vector<AudioCodec> send_codecs_; | 138 std::vector<AudioCodec> send_codecs_; |
139 std::vector<AudioCodec> recv_codecs_; | 139 std::vector<AudioCodec> recv_codecs_; |
140 std::vector<WebRtcVoiceMediaChannel*> channels_; | 140 std::vector<WebRtcVoiceMediaChannel*> channels_; |
141 webrtc::VoEBase::ChannelConfig channel_config_; | 141 webrtc::VoEBase::ChannelConfig channel_config_; |
142 bool is_dumping_aec_ = false; | |
143 | 142 |
144 webrtc::AgcConfig default_agc_config_; | 143 webrtc::AgcConfig default_agc_config_; |
145 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns | 144 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns |
146 // level controller, and intelligibility_enhancer values, and apply them | 145 // level controller, and intelligibility_enhancer values, and apply them |
147 // 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 |
148 // SetExtraOptions() will revert to defaults for options which are not | 147 // SetExtraOptions() will revert to defaults for options which are not |
149 // provided. | 148 // provided. |
150 rtc::Optional<bool> extended_filter_aec_; | 149 rtc::Optional<bool> extended_filter_aec_; |
151 rtc::Optional<bool> delay_agnostic_aec_; | 150 rtc::Optional<bool> delay_agnostic_aec_; |
152 rtc::Optional<bool> experimental_ns_; | 151 rtc::Optional<bool> experimental_ns_; |
153 rtc::Optional<bool> intelligibility_enhancer_; | 152 rtc::Optional<bool> intelligibility_enhancer_; |
154 rtc::Optional<bool> level_control_; | 153 rtc::Optional<bool> level_control_; |
155 | 154 |
156 webrtc::AudioProcessing::Config apm_config_; | 155 webrtc::AudioProcessing::Config apm_config_; |
157 | 156 |
| 157 rtc::TaskQueue* low_priority_worker_queue_; |
| 158 |
158 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine); | 159 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine); |
159 }; | 160 }; |
160 | 161 |
161 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses | 162 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses |
162 // WebRtc Voice Engine. | 163 // WebRtc Voice Engine. |
163 class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, | 164 class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, |
164 public webrtc::Transport { | 165 public webrtc::Transport { |
165 public: | 166 public: |
166 WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, | 167 WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, |
167 const MediaConfig& config, | 168 const MediaConfig& config, |
(...skipping 130 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
298 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 299 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
299 | 300 |
300 rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> | 301 rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> |
301 send_codec_spec_; | 302 send_codec_spec_; |
302 | 303 |
303 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 304 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
304 }; | 305 }; |
305 } // namespace cricket | 306 } // namespace cricket |
306 | 307 |
307 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 308 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
OLD | NEW |