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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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102 | 102 |
103 // Starts AEC dump using an existing file. A maximum file size in bytes can be | 103 // Starts AEC dump using an existing file. A maximum file size in bytes can be |
104 // specified. When the maximum file size is reached, logging is stopped and | 104 // specified. When the maximum file size is reached, logging is stopped and |
105 // the file is closed. If max_size_bytes is set to <= 0, no limit will be | 105 // the file is closed. If max_size_bytes is set to <= 0, no limit will be |
106 // used. | 106 // used. |
107 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); | 107 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); |
108 | 108 |
109 // Stops AEC dump. | 109 // Stops AEC dump. |
110 void StopAecDump(); | 110 void StopAecDump(); |
111 | 111 |
| 112 // Starts recording an RtcEventLog using an existing file until the log file |
| 113 // reaches the maximum filesize or the StopRtcEventLog function is called. |
| 114 // If the value of max_size_bytes is <= 0, no limit is used. |
| 115 bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes); |
| 116 |
| 117 // Stops recording the RtcEventLog. |
| 118 void StopRtcEventLog(); |
| 119 |
112 private: | 120 private: |
113 // Every option that is "set" will be applied. Every option not "set" will be | 121 // Every option that is "set" will be applied. Every option not "set" will be |
114 // ignored. This allows us to selectively turn on and off different options | 122 // ignored. This allows us to selectively turn on and off different options |
115 // easily at any time. | 123 // easily at any time. |
116 bool ApplyOptions(const AudioOptions& options); | 124 bool ApplyOptions(const AudioOptions& options); |
117 void SetDefaultDevices(); | 125 void SetDefaultDevices(); |
118 | 126 |
119 // webrtc::TraceCallback: | 127 // webrtc::TraceCallback: |
120 void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 128 void Print(webrtc::TraceLevel level, const char* trace, int length) override; |
121 | 129 |
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294 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 302 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
295 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 303 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
296 | 304 |
297 SendCodecSpec send_codec_spec_; | 305 SendCodecSpec send_codec_spec_; |
298 | 306 |
299 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 307 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
300 }; | 308 }; |
301 } // namespace cricket | 309 } // namespace cricket |
302 | 310 |
303 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 311 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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