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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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78 | 78 |
79 // Starts AEC dump using an existing file. A maximum file size in bytes can be | 79 // Starts AEC dump using an existing file. A maximum file size in bytes can be |
80 // specified. When the maximum file size is reached, logging is stopped and | 80 // specified. When the maximum file size is reached, logging is stopped and |
81 // the file is closed. If max_size_bytes is set to <= 0, no limit will be | 81 // the file is closed. If max_size_bytes is set to <= 0, no limit will be |
82 // used. | 82 // used. |
83 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); | 83 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); |
84 | 84 |
85 // Stops AEC dump. | 85 // Stops AEC dump. |
86 void StopAecDump(); | 86 void StopAecDump(); |
87 | 87 |
88 // Starts recording an RtcEventLog using an existing file until 10 minutes | 88 // Starts recording an RtcEventLog using an existing file until the log file |
89 // pass or the StopRtcEventLog function is called. | 89 // reaches the maximum filesize or the StopRtcEventLog function is called. |
90 bool StartRtcEventLog(rtc::PlatformFile file); | 90 // If the value of max_size_bytes is <= 0, no limit is used. |
| 91 bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes); |
91 | 92 |
92 // Stops recording the RtcEventLog. | 93 // Stops recording the RtcEventLog. |
93 void StopRtcEventLog(); | 94 void StopRtcEventLog(); |
94 | 95 |
95 private: | 96 private: |
96 // Every option that is "set" will be applied. Every option not "set" will be | 97 // Every option that is "set" will be applied. Every option not "set" will be |
97 // ignored. This allows us to selectively turn on and off different options | 98 // ignored. This allows us to selectively turn on and off different options |
98 // easily at any time. | 99 // easily at any time. |
99 bool ApplyOptions(const AudioOptions& options); | 100 bool ApplyOptions(const AudioOptions& options); |
100 void SetDefaultDevices(); | 101 void SetDefaultDevices(); |
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286 int cng_payload_type = -1; | 287 int cng_payload_type = -1; |
287 int cng_plfreq = -1; | 288 int cng_plfreq = -1; |
288 webrtc::CodecInst codec_inst; | 289 webrtc::CodecInst codec_inst; |
289 } send_codec_spec_; | 290 } send_codec_spec_; |
290 | 291 |
291 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 292 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
292 }; | 293 }; |
293 } // namespace cricket | 294 } // namespace cricket |
294 | 295 |
295 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 296 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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