AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M131 release

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Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
This commit is contained in:
Arun Raghavan
2024-12-24 19:32:07 -05:00
parent 8bdb53d91c
commit b5c48b97f6
263 changed files with 4628 additions and 20416 deletions
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16
webrtc/modules/audio_processing/audio_buffer.cc
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@ -15,7 +15,6 @@
#include <cstdint>
#include "common_audio/channel_buffer.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "modules/audio_processing/splitting_filter.h"
#include "rtc_base/checks.h"
@ -25,7 +24,6 @@ namespace {
constexpr size_t kSamplesPer32kHzChannel = 320;
constexpr size_t kSamplesPer48kHzChannel = 480;
constexpr size_t kMaxSamplesPerChannel = AudioBuffer::kMaxSampleRate / 100;
size_t NumBandsFromFramesPerChannel(size_t num_frames) {
if (num_frames == kSamplesPer32kHzChannel) {
@ -110,9 +108,9 @@ void AudioBuffer::CopyFrom(const float* const* stacked_data,
const bool resampling_needed = input_num_frames_ != buffer_num_frames_;
if (downmix_needed) {
RTC_DCHECK_GE(kMaxSamplesPerChannel, input_num_frames_);
RTC_DCHECK_GE(kMaxSamplesPerChannel10ms, input_num_frames_);
std::array<float, kMaxSamplesPerChannel> downmix;
std::array<float, kMaxSamplesPerChannel10ms> downmix;
if (downmix_by_averaging_) {
const float kOneByNumChannels = 1.f / input_num_channels_;
for (size_t i = 0; i < input_num_frames_; ++i) {
@ -230,7 +228,7 @@ void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
if (num_channels_ == 1) {
if (input_num_channels_ == 1) {
if (resampling_required) {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
S16ToFloatS16(interleaved, input_num_frames_, float_buffer.data());
input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_,
data_->channels()[0],
@ -239,7 +237,7 @@ void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
S16ToFloatS16(interleaved, input_num_frames_, data_->channels()[0]);
}
} else {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
float* downmixed_data =
resampling_required ? float_buffer.data() : data_->channels()[0];
if (downmix_by_averaging_) {
@ -274,7 +272,7 @@ void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
};
if (resampling_required) {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
for (size_t i = 0; i < num_channels_; ++i) {
deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
float_buffer.data());
@ -302,7 +300,7 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
int16_t* interleaved = interleaved_data;
if (num_channels_ == 1) {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
if (resampling_required) {
output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_,
@ -335,7 +333,7 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
if (resampling_required) {
for (size_t i = 0; i < num_channels_; ++i) {
std::array<float, kMaxSamplesPerChannel> float_buffer;
std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
output_resamplers_[i]->Resample(data_->channels()[i],
buffer_num_frames_, float_buffer.data(),
output_num_frames_);