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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 existing file. | 97 // Starts AEC dump using an existing file. A maximum file size in bytes can be |
98 bool StartAecDump(rtc::PlatformFile file); | 98 // specified. When the maximum file size is reached, logging is stopped and |
| 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); |
99 | 102 |
100 // Stops AEC dump. | 103 // Stops AEC dump. |
101 void StopAecDump(); | 104 void StopAecDump(); |
102 | 105 |
103 // Starts recording an RtcEventLog using an existing file until 10 minutes | 106 // Starts recording an RtcEventLog using an existing file until 10 minutes |
104 // pass or the StopRtcEventLog function is called. | 107 // pass or the StopRtcEventLog function is called. |
105 bool StartRtcEventLog(rtc::PlatformFile file); | 108 bool StartRtcEventLog(rtc::PlatformFile file); |
106 | 109 |
107 // Stops recording the RtcEventLog. | 110 // Stops recording the RtcEventLog. |
108 void StopRtcEventLog(); | 111 void StopRtcEventLog(); |
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280 | 283 |
281 class WebRtcAudioReceiveStream; | 284 class WebRtcAudioReceiveStream; |
282 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 285 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
283 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 286 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
284 | 287 |
285 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 288 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
286 }; | 289 }; |
287 } // namespace cricket | 290 } // namespace cricket |
288 | 291 |
289 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 292 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
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