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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
59 lines
1.8 KiB
C++
59 lines
1.8 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Unit tests for test InputAudioFile class.
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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TEST(TestInputAudioFile, DuplicateInterleaveSeparateSrcDst) {
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static const size_t kSamples = 10;
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static const size_t kChannels = 2;
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int16_t input[kSamples];
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for (size_t i = 0; i < kSamples; ++i) {
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input[i] = rtc::checked_cast<int16_t>(i);
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}
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int16_t output[kSamples * kChannels];
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InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, output);
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// Verify output
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int16_t* output_ptr = output;
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for (size_t i = 0; i < kSamples; ++i) {
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for (size_t j = 0; j < kChannels; ++j) {
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EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
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}
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}
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}
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TEST(TestInputAudioFile, DuplicateInterleaveSameSrcDst) {
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static const size_t kSamples = 10;
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static const size_t kChannels = 5;
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int16_t input[kSamples * kChannels];
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for (size_t i = 0; i < kSamples; ++i) {
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input[i] = rtc::checked_cast<int16_t>(i);
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}
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InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, input);
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// Verify output
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int16_t* output_ptr = input;
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for (size_t i = 0; i < kSamples; ++i) {
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for (size_t j = 0; j < kChannels; ++j) {
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EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
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}
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}
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}
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} // namespace test
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} // namespace webrtc
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