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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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11
modules/audio_processing/aec3/block_delay_buffer_unittest.cc
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@ -53,6 +53,7 @@ TEST(BlockDelayBuffer, CorrectDelayApplied) {
for (auto rate : {8000, 16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate, delay));
size_t num_bands = NumBandsForRate(rate);
size_t fullband_frame_length = rate / 100;
size_t subband_frame_length = rate == 8000 ? 80 : 160;
BlockDelayBuffer delay_buffer(num_bands, subband_frame_length, delay);
@ -60,23 +61,25 @@ TEST(BlockDelayBuffer, CorrectDelayApplied) {
static constexpr size_t kNumFramesToProcess = 20;
for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
++frame_index) {
AudioBuffer audio_buffer(rate, 1, rate, 1, rate);
AudioBuffer audio_buffer(fullband_frame_length, 1,
fullband_frame_length, 1,
fullband_frame_length);
if (rate > 16000) {
audio_buffer.SplitIntoFrequencyBands();
}
size_t first_sample_index = frame_index * subband_frame_length;
PopulateInputFrame(subband_frame_length, num_bands, first_sample_index,
&audio_buffer.split_bands(0)[0]);
&audio_buffer.split_bands_f(0)[0]);
delay_buffer.DelaySignal(&audio_buffer);
for (size_t k = 0; k < num_bands; ++k) {
size_t sample_index = first_sample_index;
for (size_t i = 0; i < subband_frame_length; ++i, ++sample_index) {
if (sample_index < delay) {
EXPECT_EQ(0.f, audio_buffer.split_bands(0)[k][i]);
EXPECT_EQ(0.f, audio_buffer.split_bands_f(0)[k][i]);
} else {
EXPECT_EQ(SampleValue(sample_index - delay),
audio_buffer.split_bands(0)[k][i]);
audio_buffer.split_bands_f(0)[k][i]);
}
}
}