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Simplification and refactoring of the AudioBuffer code

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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}
This commit is contained in:
Per Åhgren
2019-08-21 15:02:37 +02:00
committed by Commit Bot
parent f69bd5f184
commit 81c0cf287c
32 changed files with 541 additions and 452 deletions
Download Patch File Download Diff File Expand all files Collapse all files

11
modules/audio_processing/aec3/block_delay_buffer_unittest.cc
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@ -53,7 +53,6 @@ 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);
@ -61,25 +60,23 @@ TEST(BlockDelayBuffer, CorrectDelayApplied) {
static constexpr size_t kNumFramesToProcess = 20;
for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
++frame_index) {
AudioBuffer audio_buffer(fullband_frame_length, 1,
fullband_frame_length, 1,
fullband_frame_length);
AudioBuffer audio_buffer(rate, 1, rate, 1, rate);
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_f(0)[0]);
&audio_buffer.split_bands(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_f(0)[k][i]);
EXPECT_EQ(0.f, audio_buffer.split_bands(0)[k][i]);
} else {
EXPECT_EQ(SampleValue(sample_index - delay),
audio_buffer.split_bands_f(0)[k][i]);
audio_buffer.split_bands(0)[k][i]);
}
}
}