AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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153
webrtc/common_audio/audio_converter.cc
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@ -8,32 +8,36 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/audio_converter.h"
#include "common_audio/audio_converter.h"
#include <cstring>
#include <memory>
#include <utility>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/system_wrappers/include/scoped_vector.h"
using rtc::checked_cast;
#include "common_audio/channel_buffer.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
class CopyConverter : public AudioConverter {
public:
CopyConverter(int src_channels, size_t src_frames, int dst_channels,
CopyConverter(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~CopyConverter() override {};
~CopyConverter() override {}
void Convert(const float* const* src, size_t src_size, float* const* dst,
void Convert(const float* const* src,
size_t src_size,
float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
if (src != dst) {
for (int i = 0; i < src_channels(); ++i)
for (size_t i = 0; i < src_channels(); ++i)
std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
}
}
@ -41,17 +45,21 @@ class CopyConverter : public AudioConverter {
class UpmixConverter : public AudioConverter {
public:
UpmixConverter(int src_channels, size_t src_frames, int dst_channels,
UpmixConverter(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~UpmixConverter() override {};
~UpmixConverter() override {}
void Convert(const float* const* src, size_t src_size, float* const* dst,
void Convert(const float* const* src,
size_t src_size,
float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
for (size_t i = 0; i < dst_frames(); ++i) {
const float value = src[0][i];
for (int j = 0; j < dst_channels(); ++j)
for (size_t j = 0; j < dst_channels(); ++j)
dst[j][i] = value;
}
}
@ -59,19 +67,22 @@ class UpmixConverter : public AudioConverter {
class DownmixConverter : public AudioConverter {
public:
DownmixConverter(int src_channels, size_t src_frames, int dst_channels,
DownmixConverter(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
}
~DownmixConverter() override {};
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~DownmixConverter() override {}
void Convert(const float* const* src, size_t src_size, float* const* dst,
void Convert(const float* const* src,
size_t src_size,
float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
float* dst_mono = dst[0];
for (size_t i = 0; i < src_frames(); ++i) {
float sum = 0;
for (int j = 0; j < src_channels(); ++j)
for (size_t j = 0; j < src_channels(); ++j)
sum += src[j][i];
dst_mono[i] = sum / src_channels();
}
@ -80,16 +91,21 @@ class DownmixConverter : public AudioConverter {
class ResampleConverter : public AudioConverter {
public:
ResampleConverter(int src_channels, size_t src_frames, int dst_channels,
ResampleConverter(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
resamplers_.reserve(src_channels);
for (int i = 0; i < src_channels; ++i)
resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
for (size_t i = 0; i < src_channels; ++i)
resamplers_.push_back(std::unique_ptr<PushSincResampler>(
new PushSincResampler(src_frames, dst_frames)));
}
~ResampleConverter() override {};
~ResampleConverter() override {}
void Convert(const float* const* src, size_t src_size, float* const* dst,
void Convert(const float* const* src,
size_t src_size,
float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
for (size_t i = 0; i < resamplers_.size(); ++i)
@ -97,69 +113,73 @@ class ResampleConverter : public AudioConverter {
}
private:
ScopedVector<PushSincResampler> resamplers_;
std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
};
// Apply a vector of converters in serial, in the order given. At least two
// converters must be provided.
class CompositionConverter : public AudioConverter {
public:
CompositionConverter(ScopedVector<AudioConverter> converters)
: converters_(converters.Pass()) {
RTC_CHECK_GE(converters_.size(), 2u);
explicit CompositionConverter(
std::vector<std::unique_ptr<AudioConverter>> converters)
: converters_(std::move(converters)) {
RTC_CHECK_GE(converters_.size(), 2);
// We need an intermediate buffer after every converter.
for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
buffers_.push_back(new ChannelBuffer<float>((*it)->dst_frames(),
(*it)->dst_channels()));
buffers_.push_back(
std::unique_ptr<ChannelBuffer<float>>(new ChannelBuffer<float>(
(*it)->dst_frames(), (*it)->dst_channels())));
}
~CompositionConverter() override {};
~CompositionConverter() override {}
void Convert(const float* const* src, size_t src_size, float* const* dst,
void Convert(const float* const* src,
size_t src_size,
float* const* dst,
size_t dst_capacity) override {
converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
buffers_.front()->size());
for (size_t i = 2; i < converters_.size(); ++i) {
auto src_buffer = buffers_[i - 2];
auto dst_buffer = buffers_[i - 1];
converters_[i]->Convert(src_buffer->channels(),
src_buffer->size(),
dst_buffer->channels(),
dst_buffer->size());
auto& src_buffer = buffers_[i - 2];
auto& dst_buffer = buffers_[i - 1];
converters_[i]->Convert(src_buffer->channels(), src_buffer->size(),
dst_buffer->channels(), dst_buffer->size());
}
converters_.back()->Convert(buffers_.back()->channels(),
buffers_.back()->size(), dst, dst_capacity);
}
private:
ScopedVector<AudioConverter> converters_;
ScopedVector<ChannelBuffer<float>> buffers_;
std::vector<std::unique_ptr<AudioConverter>> converters_;
std::vector<std::unique_ptr<ChannelBuffer<float>>> buffers_;
};
rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
std::unique_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
size_t src_frames,
int dst_channels,
size_t dst_channels,
size_t dst_frames) {
rtc::scoped_ptr<AudioConverter> sp;
std::unique_ptr<AudioConverter> sp;
if (src_channels > dst_channels) {
if (src_frames != dst_frames) {
ScopedVector<AudioConverter> converters;
converters.push_back(new DownmixConverter(src_channels, src_frames,
dst_channels, src_frames));
converters.push_back(new ResampleConverter(dst_channels, src_frames,
dst_channels, dst_frames));
sp.reset(new CompositionConverter(converters.Pass()));
std::vector<std::unique_ptr<AudioConverter>> converters;
converters.push_back(std::unique_ptr<AudioConverter>(new DownmixConverter(
src_channels, src_frames, dst_channels, src_frames)));
converters.push_back(
std::unique_ptr<AudioConverter>(new ResampleConverter(
dst_channels, src_frames, dst_channels, dst_frames)));
sp.reset(new CompositionConverter(std::move(converters)));
} else {
sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
dst_frames));
}
} else if (src_channels < dst_channels) {
if (src_frames != dst_frames) {
ScopedVector<AudioConverter> converters;
converters.push_back(new ResampleConverter(src_channels, src_frames,
src_channels, dst_frames));
converters.push_back(new UpmixConverter(src_channels, dst_frames,
dst_channels, dst_frames));
sp.reset(new CompositionConverter(converters.Pass()));
std::vector<std::unique_ptr<AudioConverter>> converters;
converters.push_back(
std::unique_ptr<AudioConverter>(new ResampleConverter(
src_channels, src_frames, src_channels, dst_frames)));
converters.push_back(std::unique_ptr<AudioConverter>(new UpmixConverter(
src_channels, dst_frames, dst_channels, dst_frames)));
sp.reset(new CompositionConverter(std::move(converters)));
} else {
sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
dst_frames));
@ -168,22 +188,21 @@ rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
dst_frames));
} else {
sp.reset(new CopyConverter(src_channels, src_frames, dst_channels,
dst_frames));
sp.reset(
new CopyConverter(src_channels, src_frames, dst_channels, dst_frames));
}
return sp.Pass();
return sp;
}
// For CompositionConverter.
AudioConverter::AudioConverter()
: src_channels_(0),
src_frames_(0),
dst_channels_(0),
dst_frames_(0) {}
: src_channels_(0), src_frames_(0), dst_channels_(0), dst_frames_(0) {}
AudioConverter::AudioConverter(int src_channels, size_t src_frames,
int dst_channels, size_t dst_frames)
AudioConverter::AudioConverter(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames)
: src_channels_(src_channels),
src_frames_(src_frames),
dst_channels_(dst_channels),