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Revert "Simplification and refactoring of the AudioBuffer code"

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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}
This commit is contained in:
Steve Anton
2019-08-21 17:52:28 +00:00
committed by Commit Bot
parent f5815fa6bb
commit f254e9e9e5
32 changed files with 449 additions and 538 deletions
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10
modules/audio_processing/audio_buffer_unittest.cc
View File

@ -16,7 +16,7 @@ namespace webrtc {
namespace {
const size_t kSampleRateHz = 48000u;
const size_t kNumFrames = 480u;
const size_t kStereo = 2u;
const size_t kMono = 1u;
@ -27,17 +27,17 @@ void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) {
} // namespace
TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
AudioBuffer ab(kSampleRateHz, kStereo, kSampleRateHz, kStereo, kSampleRateHz);
AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
ExpectNumChannels(ab, kStereo);
ab.set_num_channels(1);
ab.set_num_channels(kMono);
ExpectNumChannels(ab, kMono);
ab.RestoreNumChannels();
ab.InitForNewData();
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);
AudioBuffer ab(kNumFrames, kMono, kNumFrames, kMono, kNumFrames);
EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
}
#endif