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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2004 Google Inc. | 3 * Copyright 2004 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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87 // Called by WebRtcVoiceMediaChannel to set a gain offset from | 87 // Called by WebRtcVoiceMediaChannel to set a gain offset from |
88 // the default AGC target level. | 88 // the default AGC target level. |
89 bool AdjustAgcLevel(int delta); | 89 bool AdjustAgcLevel(int delta); |
90 | 90 |
91 VoEWrapper* voe() { return voe_wrapper_.get(); } | 91 VoEWrapper* voe() { return voe_wrapper_.get(); } |
92 int GetLastEngineError(); | 92 int GetLastEngineError(); |
93 | 93 |
94 // Set the external ADM. This can only be called before Init. | 94 // Set the external ADM. This can only be called before Init. |
95 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm); | 95 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm); |
96 | 96 |
97 // Starts AEC dump using an existing file. A maximum file size in bytes can be | 97 // Starts AEC dump using existing file. |
98 // specified. When the maximum file size is reached, logging is stopped and | 98 bool StartAecDump(rtc::PlatformFile file); |
99 // the file is closed. If max_size_bytes is set to <= 0, no limit will be | |
100 // used. | |
101 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); | |
102 | 99 |
103 // Stops AEC dump. | 100 // Stops AEC dump. |
104 void StopAecDump(); | 101 void StopAecDump(); |
105 | 102 |
106 // Starts recording an RtcEventLog using an existing file until 10 minutes | 103 // Starts recording an RtcEventLog using an existing file until 10 minutes |
107 // pass or the StopRtcEventLog function is called. | 104 // pass or the StopRtcEventLog function is called. |
108 bool StartRtcEventLog(rtc::PlatformFile file); | 105 bool StartRtcEventLog(rtc::PlatformFile file); |
109 | 106 |
110 // Stops recording the RtcEventLog. | 107 // Stops recording the RtcEventLog. |
111 void StopRtcEventLog(); | 108 void StopRtcEventLog(); |
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283 | 280 |
284 class WebRtcAudioReceiveStream; | 281 class WebRtcAudioReceiveStream; |
285 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 282 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
286 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 283 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
287 | 284 |
288 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 285 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
289 }; | 286 }; |
290 } // namespace cricket | 287 } // namespace cricket |
291 | 288 |
292 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 289 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
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