| /* |
| * 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 kSampleRateHz = 48000u; |
| 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(kSampleRateHz, kStereo, kSampleRateHz, kStereo, kSampleRateHz, |
| kStereo); |
| ExpectNumChannels(ab, kStereo); |
| ab.set_num_channels(1); |
| ExpectNumChannels(ab, kMono); |
| ab.RestoreNumChannels(); |
| ExpectNumChannels(ab, kStereo); |
| } |
| |
| #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) |
| TEST(AudioBufferTest, SetNumChannelsDeathTest) { |
| AudioBuffer ab(kSampleRateHz, kMono, kSampleRateHz, kMono, kSampleRateHz, |
| kMono); |
| EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels"); |
| } |
| #endif |
| |
| } // namespace webrtc |