From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- .../audio/test/low_bandwidth_audio_test_flags.cc | 28 ++++++++++++++++++++++ 1 file changed, 28 insertions(+) create mode 100644 third_party/libwebrtc/audio/test/low_bandwidth_audio_test_flags.cc (limited to 'third_party/libwebrtc/audio/test/low_bandwidth_audio_test_flags.cc') diff --git a/third_party/libwebrtc/audio/test/low_bandwidth_audio_test_flags.cc b/third_party/libwebrtc/audio/test/low_bandwidth_audio_test_flags.cc new file mode 100644 index 0000000000..9d93790d3d --- /dev/null +++ b/third_party/libwebrtc/audio/test/low_bandwidth_audio_test_flags.cc @@ -0,0 +1,28 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +// #ifndef AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_FLAGS_H_ +// #define AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_FLAGS_H_ + +#include "absl/flags/flag.h" + +ABSL_FLAG(int, + sample_rate_hz, + 16000, + "Sample rate (Hz) of the produced audio files."); + +ABSL_FLAG(bool, + quick, + false, + "Don't do the full audio recording. " + "Used to quickly check that the test runs without crashing."); + +ABSL_FLAG(std::string, test_case_prefix, "", "Test case prefix."); + +// #endif // AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_FLAGS_H_ -- cgit v1.2.3