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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135
webrtc/common_audio/resampler/push_resampler.cc
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@ -8,40 +8,78 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "common_audio/resampler/include/push_resampler.h"
#include <stdint.h>
#include <string.h>
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include <memory>
#include "common_audio/include/audio_util.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
// These checks were factored out into a non-templatized function
// due to problems with clang on Windows in debug builds.
// For some reason having the DCHECKs inline in the template code
// caused the compiler to generate code that threw off the linker.
// TODO(tommi): Re-enable when we've figured out what the problem is.
// http://crbug.com/615050
void CheckValidInitParams(int src_sample_rate_hz,
int dst_sample_rate_hz,
size_t num_channels) {
// The below checks are temporarily disabled on WEBRTC_WIN due to problems
// with clang debug builds.
#if !defined(WEBRTC_WIN) && defined(__clang__)
RTC_DCHECK_GT(src_sample_rate_hz, 0);
RTC_DCHECK_GT(dst_sample_rate_hz, 0);
RTC_DCHECK_GT(num_channels, 0);
#endif
}
void CheckExpectedBufferSizes(size_t src_length,
size_t dst_capacity,
size_t num_channels,
int src_sample_rate,
int dst_sample_rate) {
// The below checks are temporarily disabled on WEBRTC_WIN due to problems
// with clang debug builds.
// TODO(tommi): Re-enable when we've figured out what the problem is.
// http://crbug.com/615050
#if !defined(WEBRTC_WIN) && defined(__clang__)
const size_t src_size_10ms = src_sample_rate * num_channels / 100;
const size_t dst_size_10ms = dst_sample_rate * num_channels / 100;
RTC_DCHECK_EQ(src_length, src_size_10ms);
RTC_DCHECK_GE(dst_capacity, dst_size_10ms);
#endif
}
} // namespace
template <typename T>
PushResampler<T>::PushResampler()
: src_sample_rate_hz_(0),
dst_sample_rate_hz_(0),
num_channels_(0) {
}
: src_sample_rate_hz_(0), dst_sample_rate_hz_(0), num_channels_(0) {}
template <typename T>
PushResampler<T>::~PushResampler() {
}
PushResampler<T>::~PushResampler() {}
template <typename T>
int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
int num_channels) {
size_t num_channels) {
CheckValidInitParams(src_sample_rate_hz, dst_sample_rate_hz, num_channels);
if (src_sample_rate_hz == src_sample_rate_hz_ &&
dst_sample_rate_hz == dst_sample_rate_hz_ &&
num_channels == num_channels_)
num_channels == num_channels_) {
// No-op if settings haven't changed.
return 0;
}
if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
num_channels <= 0 || num_channels > 2)
if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 || num_channels <= 0) {
return -1;
}
src_sample_rate_hz_ = src_sample_rate_hz;
dst_sample_rate_hz_ = dst_sample_rate_hz;
@ -51,29 +89,28 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
static_cast<size_t>(src_sample_rate_hz / 100);
const size_t dst_size_10ms_mono =
static_cast<size_t>(dst_sample_rate_hz / 100);
sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
dst_size_10ms_mono));
if (num_channels_ == 2) {
src_left_.reset(new T[src_size_10ms_mono]);
src_right_.reset(new T[src_size_10ms_mono]);
dst_left_.reset(new T[dst_size_10ms_mono]);
dst_right_.reset(new T[dst_size_10ms_mono]);
sinc_resampler_right_.reset(new PushSincResampler(src_size_10ms_mono,
dst_size_10ms_mono));
channel_resamplers_.clear();
for (size_t i = 0; i < num_channels; ++i) {
channel_resamplers_.push_back(ChannelResampler());
auto channel_resampler = channel_resamplers_.rbegin();
channel_resampler->resampler = std::make_unique<PushSincResampler>(
src_size_10ms_mono, dst_size_10ms_mono);
channel_resampler->source.resize(src_size_10ms_mono);
channel_resampler->destination.resize(dst_size_10ms_mono);
}
channel_data_array_.resize(num_channels_);
return 0;
}
template <typename T>
int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
int PushResampler<T>::Resample(const T* src,
size_t src_length,
T* dst,
size_t dst_capacity) {
const size_t src_size_10ms =
static_cast<size_t>(src_sample_rate_hz_ * num_channels_ / 100);
const size_t dst_size_10ms =
static_cast<size_t>(dst_sample_rate_hz_ * num_channels_ / 100);
if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
return -1;
CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_,
src_sample_rate_hz_, dst_sample_rate_hz_);
if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
// The old resampler provides this memcpy facility in the case of matching
@ -81,26 +118,30 @@ int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
memcpy(dst, src, src_length * sizeof(T));
return static_cast<int>(src_length);
}
if (num_channels_ == 2) {
const size_t src_length_mono = src_length / num_channels_;
const size_t dst_capacity_mono = dst_capacity / num_channels_;
T* deinterleaved[] = {src_left_.get(), src_right_.get()};
Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
size_t dst_length_mono =
sinc_resampler_->Resample(src_left_.get(), src_length_mono,
dst_left_.get(), dst_capacity_mono);
sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
dst_right_.get(), dst_capacity_mono);
const size_t src_length_mono = src_length / num_channels_;
const size_t dst_capacity_mono = dst_capacity / num_channels_;
deinterleaved[0] = dst_left_.get();
deinterleaved[1] = dst_right_.get();
Interleave(deinterleaved, dst_length_mono, num_channels_, dst);
return static_cast<int>(dst_length_mono * num_channels_);
} else {
return static_cast<int>(
sinc_resampler_->Resample(src, src_length, dst, dst_capacity));
for (size_t ch = 0; ch < num_channels_; ++ch) {
channel_data_array_[ch] = channel_resamplers_[ch].source.data();
}
Deinterleave(src, src_length_mono, num_channels_, channel_data_array_.data());
size_t dst_length_mono = 0;
for (auto& resampler : channel_resamplers_) {
dst_length_mono = resampler.resampler->Resample(
resampler.source.data(), src_length_mono, resampler.destination.data(),
dst_capacity_mono);
}
for (size_t ch = 0; ch < num_channels_; ++ch) {
channel_data_array_[ch] = channel_resamplers_[ch].destination.data();
}
Interleave(channel_data_array_.data(), dst_length_mono, num_channels_, dst);
return static_cast<int>(dst_length_mono * num_channels_);
}
// Explictly generate required instantiations.