webrtc/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

58 lines
1.7 KiB
C++

/*
* Copyright (c) 2014 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.
*/
// Unit tests for test InputAudioFile class.
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
TEST(TestInputAudioFile, DuplicateInterleaveSeparateSrcDst) {
static const size_t kSamples = 10;
static const size_t kChannels = 2;
int16_t input[kSamples];
for (size_t i = 0; i < kSamples; ++i) {
input[i] = i;
}
int16_t output[kSamples * kChannels];
InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, output);
// Verify output
int16_t* output_ptr = output;
for (size_t i = 0; i < kSamples; ++i) {
for (size_t j = 0; j < kChannels; ++j) {
EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
}
}
}
TEST(TestInputAudioFile, DuplicateInterleaveSameSrcDst) {
static const size_t kSamples = 10;
static const size_t kChannels = 5;
int16_t input[kSamples * kChannels];
for (size_t i = 0; i < kSamples; ++i) {
input[i] = i;
}
InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, input);
// Verify output
int16_t* output_ptr = input;
for (size_t i = 0; i < kSamples; ++i) {
for (size_t j = 0; j < kChannels; ++j) {
EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
}
}
}
} // namespace test
} // namespace webrtc