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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 |
| (...skipping 31 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 42 | 42 |
| 43 // Returns the timestamp of the next audio output event, in milliseconds. The | 43 // Returns the timestamp of the next audio output event, in milliseconds. The |
| 44 // maximum value of int64_t is returned if there are no more audio output | 44 // maximum value of int64_t is returned if there are no more audio output |
| 45 // events available. | 45 // events available. |
| 46 int64_t NextAudioOutputEventMs(); | 46 int64_t NextAudioOutputEventMs(); |
| 47 | 47 |
| 48 private: | 48 private: |
| 49 RtcEventLogSource(); | 49 RtcEventLogSource(); |
| 50 | 50 |
| 51 bool OpenFile(const std::string& file_name); | 51 bool OpenFile(const std::string& file_name); |
| 52 bool IsAudioSsrc(uint32_t ssrc) const; | |
| 53 | |
| 54 // SSRCs of incoming Audio streams found in the event log. | |
|
hlundin-webrtc
2017/05/23 13:00:31
Nit: Audio -> audio
perkj_webrtc
2017/05/24 12:32:21
Done.
| |
| 55 std::vector<uint32_t> streams_; | |
| 52 | 56 |
| 53 size_t rtp_packet_index_ = 0; | 57 size_t rtp_packet_index_ = 0; |
| 54 size_t audio_output_index_ = 0; | 58 size_t audio_output_index_ = 0; |
| 55 | 59 |
| 56 ParsedRtcEventLog parsed_stream_; | 60 ParsedRtcEventLog parsed_stream_; |
| 57 std::unique_ptr<RtpHeaderParser> parser_; | 61 std::unique_ptr<RtpHeaderParser> parser_; |
| 58 | 62 |
| 59 RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource); | 63 RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource); |
| 60 }; | 64 }; |
| 61 | 65 |
| 62 } // namespace test | 66 } // namespace test |
| 63 } // namespace webrtc | 67 } // namespace webrtc |
| 64 | 68 |
| 65 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ | 69 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |
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