blob: adcd518ff38fb2c95ae0b0955cada7c826880098 [file] [log] [blame]
/*
* 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()
: src_sample_rate_hz_(0), dst_sample_rate_hz_(0), num_channels_(0) {}
template <typename T>
PushResampler<T>::~PushResampler() {}
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_DCHECK_GT(src_sample_rate_hz, 0);
RTC_DCHECK_GT(dst_sample_rate_hz, 0);
RTC_DCHECK_GT(num_channels, 0);
if (src_sample_rate_hz == src_sample_rate_hz_ &&
dst_sample_rate_hz == dst_sample_rate_hz_ &&
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) {
return -1;
}
src_sample_rate_hz_ = src_sample_rate_hz;
dst_sample_rate_hz_ = dst_sample_rate_hz;
num_channels_ = num_channels;
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);
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(InterleavedView<const T> src,
InterleavedView<T> dst) {
RTC_DCHECK_EQ(NumChannels(src), num_channels_);
RTC_DCHECK_EQ(NumChannels(dst), num_channels_);
RTC_DCHECK_EQ(SamplesPerChannel(src),
SampleRateToDefaultChannelSize(src_sample_rate_hz_));
RTC_DCHECK_EQ(SamplesPerChannel(dst),
SampleRateToDefaultChannelSize(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
// sample rates, so reproduce it here for the sinc resampler.
CopySamples(dst, src);
return static_cast<int>(src.data().size());
}
for (size_t ch = 0; ch < num_channels_; ++ch) {
channel_data_array_[ch] = channel_resamplers_[ch].source.data();
}
// TODO: b/335805780 - Deinterleave should accept InterleavedView<> as input.
Deinterleave(&src.data()[0], src.samples_per_channel(), src.num_channels(),
channel_data_array_.data());
for (auto& resampler : channel_resamplers_) {
size_t dst_length_mono = resampler.resampler->Resample(
resampler.source.data(), src.samples_per_channel(),
resampler.destination.data(), dst.samples_per_channel());
RTC_DCHECK_EQ(dst_length_mono, dst.samples_per_channel());
}
for (size_t ch = 0; ch < num_channels_; ++ch) {
channel_data_array_[ch] = channel_resamplers_[ch].destination.data();
}
// TODO: b/335805780 - Interleave should accept InterleavedView<> as dst.
Interleave(channel_data_array_.data(), dst.samples_per_channel(),
num_channels_, &dst[0]);
return static_cast<int>(dst.size());
}
// Explictly generate required instantiations.
template class PushResampler<int16_t>;
template class PushResampler<float>;
} // namespace webrtc