mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
d47941e018
This is a reland of 81c0cf287c
Original change's description:
> 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}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
47 lines
1.4 KiB
C++
47 lines
1.4 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,
|
|
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
|