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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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54 webrtc::AudioDeviceModule* adm, | 54 webrtc::AudioDeviceModule* adm, |
55 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, | 55 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
56 VoEWrapper* voe_wrapper); | 56 VoEWrapper* voe_wrapper); |
57 ~WebRtcVoiceEngine() override; | 57 ~WebRtcVoiceEngine() override; |
58 | 58 |
59 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const; | 59 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const; |
60 VoiceMediaChannel* CreateChannel(webrtc::Call* call, | 60 VoiceMediaChannel* CreateChannel(webrtc::Call* call, |
61 const MediaConfig& config, | 61 const MediaConfig& config, |
62 const AudioOptions& options); | 62 const AudioOptions& options); |
63 | 63 |
64 bool GetOutputVolume(int* level); | |
65 bool SetOutputVolume(int level); | |
66 int GetInputLevel(); | 64 int GetInputLevel(); |
67 | 65 |
68 const std::vector<AudioCodec>& codecs(); | 66 const std::vector<AudioCodec>& codecs(); |
69 RtpCapabilities GetCapabilities() const; | 67 RtpCapabilities GetCapabilities() const; |
70 | 68 |
71 // For tracking WebRtc channels. Needed because we have to pause them | 69 // For tracking WebRtc channels. Needed because we have to pause them |
72 // all when switching devices. | 70 // all when switching devices. |
73 // May only be called by WebRtcVoiceMediaChannel. | 71 // May only be called by WebRtcVoiceMediaChannel. |
74 void RegisterChannel(WebRtcVoiceMediaChannel* channel); | 72 void RegisterChannel(WebRtcVoiceMediaChannel* channel); |
75 void UnregisterChannel(WebRtcVoiceMediaChannel* channel); | 73 void UnregisterChannel(WebRtcVoiceMediaChannel* channel); |
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298 int cng_payload_type = -1; | 296 int cng_payload_type = -1; |
299 int cng_plfreq = -1; | 297 int cng_plfreq = -1; |
300 webrtc::CodecInst codec_inst; | 298 webrtc::CodecInst codec_inst; |
301 } send_codec_spec_; | 299 } send_codec_spec_; |
302 | 300 |
303 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 301 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
304 }; | 302 }; |
305 } // namespace cricket | 303 } // namespace cricket |
306 | 304 |
307 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 305 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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