webrtc/modules/audio_coding/neteq/merge_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

39 lines
1.2 KiB
C++

/*
* Copyright (c) 2012 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 Merge class.
#include "modules/audio_coding/neteq/merge.h"
#include <vector>
#include "modules/audio_coding/neteq/background_noise.h"
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/random_vector.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "test/gtest.h"
namespace webrtc {
TEST(Merge, CreateAndDestroy) {
int fs = 8000;
size_t channels = 1;
BackgroundNoise bgn(channels);
SyncBuffer sync_buffer(1, 1000);
RandomVector random_vector;
StatisticsCalculator statistics;
Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
Merge merge(fs, channels, &expand, &sync_buffer);
}
// TODO(hlundin): Write more tests.
} // namespace webrtc