webrtc_m130/modules/audio_processing/audio_buffer_unittest.cc
Per Åhgren 81c0cf287c Simplification and refactoring of the AudioBuffer code
This CL performs a major refactoring and simplification
of the AudioBuffer code that.
-Removes 7 of the 9 internal buffers of the AudioBuffer.
-Avoids the implicit copying required to keep the
 internal buffers in sync.
-Removes all code relating to handling of fixed-point
 sample data in the AudioBuffer.
-Changes the naming of the class methods to reflect
 that only floating point is handled.
-Corrects some bugs in the code.
-Extends the handling of internal downmixing to be
 more generic.

Bug: webrtc:10882
Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28928}
2019-08-21 13:40:59 +00:00

46 lines
1.3 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 "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);
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);
EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
}
#endif
} // namespace webrtc