blob: b8847999dcb38f31a152194c6a93c394a8808a2a [file] [log] [blame]
/*
* Copyright (c) 2016 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.
*/
#include "modules/audio_processing/audio_buffer.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
const size_t kNumFrames = 480u;
const size_t kStereo = 2u;
const size_t kMono = 1u;
void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) {
EXPECT_EQ(ab.num_channels(), num_channels);
}
} // namespace
TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
ExpectNumChannels(ab, kStereo);
ab.set_num_channels(kMono);
ExpectNumChannels(ab, kMono);
ab.InitForNewData();
ExpectNumChannels(ab, kStereo);
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(AudioBufferTest, SetNumChannelsDeathTest) {
AudioBuffer ab(kNumFrames, kMono, kNumFrames, kMono, kNumFrames);
EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
}
#endif
} // namespace webrtc