f254e9e9e5
This reverts commit 81c0cf287c8514cb1cd6f3baca484d668c6eb128. Reason for revert: internal test failures 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} TBR=gustaf@webrtc.org,peah@webrtc.org Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10882 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28931}
46 lines
1.3 KiB
C++
46 lines
1.3 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/audio_buffer.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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const size_t kNumFrames = 480u;
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const size_t kStereo = 2u;
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const size_t kMono = 1u;
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void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) {
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EXPECT_EQ(ab.num_channels(), num_channels);
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}
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} // namespace
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TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
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AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
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ExpectNumChannels(ab, kStereo);
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ab.set_num_channels(kMono);
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ExpectNumChannels(ab, kMono);
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ab.InitForNewData();
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ExpectNumChannels(ab, kStereo);
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}
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#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
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TEST(AudioBufferTest, SetNumChannelsDeathTest) {
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AudioBuffer ab(kNumFrames, kMono, kNumFrames, kMono, kNumFrames);
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EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
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}
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#endif
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} // namespace webrtc
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