webrtc/common_audio/channel_buffer_unittest.cc
Jonas Olsson a4d873786f Format almost everything.
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}
2019-07-08 13:45:15 +00:00

66 lines
2 KiB
C++

/*
* 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 "common_audio/channel_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 IFChannelBuffer& ifchb, size_t num_channels) {
EXPECT_EQ(ifchb.ibuf_const()->num_channels(), num_channels);
EXPECT_EQ(ifchb.fbuf_const()->num_channels(), num_channels);
EXPECT_EQ(ifchb.num_channels(), num_channels);
}
} // namespace
TEST(ChannelBufferTest, SetNumChannelsSetsNumChannels) {
ChannelBuffer<float> chb(kNumFrames, kStereo);
EXPECT_EQ(chb.num_channels(), kStereo);
chb.set_num_channels(kMono);
EXPECT_EQ(chb.num_channels(), kMono);
}
TEST(IFChannelBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
IFChannelBuffer ifchb(kNumFrames, kStereo);
ExpectNumChannels(ifchb, kStereo);
ifchb.set_num_channels(kMono);
ExpectNumChannels(ifchb, kMono);
}
TEST(IFChannelBufferTest, SettingNumChannelsOfOneChannelBufferSetsTheOther) {
IFChannelBuffer ifchb(kNumFrames, kStereo);
ExpectNumChannels(ifchb, kStereo);
ifchb.ibuf()->set_num_channels(kMono);
ExpectNumChannels(ifchb, kMono);
ifchb.fbuf()->set_num_channels(kStereo);
ExpectNumChannels(ifchb, kStereo);
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(ChannelBufferTest, SetNumChannelsDeathTest) {
ChannelBuffer<float> chb(kNumFrames, kMono);
EXPECT_DEATH(chb.set_num_channels(kStereo), "num_channels");
}
TEST(IFChannelBufferTest, SetNumChannelsDeathTest) {
IFChannelBuffer ifchb(kNumFrames, kMono);
EXPECT_DEATH(ifchb.ibuf()->set_num_channels(kStereo), "num_channels");
}
#endif
} // namespace webrtc