| Artem Titov | b1f2d60 | 2019-07-10 14:40:58 +0200 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 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 |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | // #ifndef AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_FLAGS_H_ |
| 11 | // #define AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_FLAGS_H_ |
| 12 | |
| 13 | #include "rtc_base/flags.h" |
| 14 | |
| 15 | WEBRTC_DEFINE_int(sample_rate_hz, |
| 16 | 16000, |
| 17 | "Sample rate (Hz) of the produced audio files."); |
| 18 | |
| 19 | WEBRTC_DEFINE_bool( |
| 20 | quick, |
| 21 | false, |
| 22 | "Don't do the full audio recording. " |
| 23 | "Used to quickly check that the test runs without crashing."); |
| 24 | |
| 25 | WEBRTC_DEFINE_string(test_case_prefix, "", "Test case prefix."); |
| 26 | |
| 27 | // #endif // AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_FLAGS_H_ |