webrtc_m130/common_audio/resampler/push_resampler.cc
Tommi cfdbb0d371 Update PushResampler to use a single buffer for source, destination.
PushResampler now uses a single buffer for the deinterleaved
channel sources and another for the deinterleaved destinations.
Before, there was a dedicated buffer per channel (source and dest).
This reduces allocations and allows for using DeinterleavedView for
both which simplifies some checks.

Bug: chromium:335805780
Change-Id: I553a36164109127fa332ab17918d53832d442303
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351542
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42415}
2024-05-30 14:37:50 +00:00

103 lines
3.7 KiB
C++

/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/resampler/include/push_resampler.h"
#include <stdint.h>
#include <string.h>
#include <memory>
#include "api/audio/audio_frame.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "rtc_base/checks.h"
namespace webrtc {
template <typename T>
PushResampler<T>::PushResampler() = default;
template <typename T>
PushResampler<T>::~PushResampler() = default;
template <typename T>
int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
size_t num_channels) {
// These checks used to be factored out of this template function due to
// Windows debug build issues with clang. http://crbug.com/615050
RTC_CHECK_GT(src_sample_rate_hz, 0);
RTC_CHECK_GT(dst_sample_rate_hz, 0);
RTC_CHECK_GT(num_channels, 0);
const size_t src_size_10ms_mono =
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);
if (src_size_10ms_mono == SamplesPerChannel(source_view_) &&
dst_size_10ms_mono == SamplesPerChannel(destination_view_) &&
num_channels == NumChannels(source_view_)) {
// No-op if settings haven't changed.
return 0;
}
// Allocate two buffers for all source and destination channels.
// Then organize source and destination views together with an array of
// resamplers for each channel in the deinterlaved buffers.
source_.reset(new T[src_size_10ms_mono * num_channels]);
destination_.reset(new T[dst_size_10ms_mono * num_channels]);
source_view_ =
DeinterleavedView<T>(source_.get(), src_size_10ms_mono, num_channels);
destination_view_ = DeinterleavedView<T>(destination_.get(),
dst_size_10ms_mono, num_channels);
resamplers_.resize(num_channels);
for (size_t i = 0; i < num_channels; ++i) {
resamplers_[i] = std::make_unique<PushSincResampler>(src_size_10ms_mono,
dst_size_10ms_mono);
}
return 0;
}
template <typename T>
int PushResampler<T>::Resample(InterleavedView<const T> src,
InterleavedView<T> dst) {
RTC_DCHECK_EQ(NumChannels(src), NumChannels(source_view_));
RTC_DCHECK_EQ(NumChannels(dst), NumChannels(destination_view_));
RTC_DCHECK_EQ(SamplesPerChannel(src), SamplesPerChannel(source_view_));
RTC_DCHECK_EQ(SamplesPerChannel(dst), SamplesPerChannel(destination_view_));
if (SamplesPerChannel(src) == SamplesPerChannel(dst)) {
// The old resampler provides this memcpy facility in the case of matching
// sample rates, so reproduce it here for the sinc resampler.
CopySamples(dst, src);
return static_cast<int>(src.data().size());
}
Deinterleave(src, source_view_);
for (size_t i = 0; i < resamplers_.size(); ++i) {
size_t dst_length_mono =
resamplers_[i]->Resample(source_view_[i], destination_view_[i]);
RTC_DCHECK_EQ(dst_length_mono, SamplesPerChannel(dst));
}
Interleave<T>(destination_view_, dst);
return static_cast<int>(dst.size());
}
// Explictly generate required instantiations.
template class PushResampler<int16_t>;
template class PushResampler<float>;
} // namespace webrtc