753eada3aa
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
111 lines
4.0 KiB
C++
111 lines
4.0 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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#include <string.h>
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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namespace webrtc {
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template <typename T>
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PushResampler<T>::PushResampler()
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: src_sample_rate_hz_(0),
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dst_sample_rate_hz_(0),
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num_channels_(0) {
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}
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template <typename T>
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PushResampler<T>::~PushResampler() {
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}
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template <typename T>
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int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
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int dst_sample_rate_hz,
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int num_channels) {
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if (src_sample_rate_hz == src_sample_rate_hz_ &&
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dst_sample_rate_hz == dst_sample_rate_hz_ &&
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num_channels == num_channels_)
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// No-op if settings haven't changed.
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return 0;
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if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
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num_channels <= 0 || num_channels > 2)
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return -1;
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src_sample_rate_hz_ = src_sample_rate_hz;
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dst_sample_rate_hz_ = dst_sample_rate_hz;
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num_channels_ = num_channels;
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const size_t src_size_10ms_mono =
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static_cast<size_t>(src_sample_rate_hz / 100);
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const size_t dst_size_10ms_mono =
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static_cast<size_t>(dst_sample_rate_hz / 100);
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sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
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dst_size_10ms_mono));
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if (num_channels_ == 2) {
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src_left_.reset(new T[src_size_10ms_mono]);
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src_right_.reset(new T[src_size_10ms_mono]);
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dst_left_.reset(new T[dst_size_10ms_mono]);
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dst_right_.reset(new T[dst_size_10ms_mono]);
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sinc_resampler_right_.reset(new PushSincResampler(src_size_10ms_mono,
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dst_size_10ms_mono));
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}
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return 0;
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}
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template <typename T>
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int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
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size_t dst_capacity) {
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const size_t src_size_10ms =
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static_cast<size_t>(src_sample_rate_hz_ * num_channels_ / 100);
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const size_t dst_size_10ms =
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static_cast<size_t>(dst_sample_rate_hz_ * num_channels_ / 100);
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if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
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return -1;
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if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
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// The old resampler provides this memcpy facility in the case of matching
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// sample rates, so reproduce it here for the sinc resampler.
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memcpy(dst, src, src_length * sizeof(T));
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return static_cast<int>(src_length);
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}
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if (num_channels_ == 2) {
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const size_t src_length_mono = src_length / num_channels_;
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const size_t dst_capacity_mono = dst_capacity / num_channels_;
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T* deinterleaved[] = {src_left_.get(), src_right_.get()};
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Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
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size_t dst_length_mono =
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sinc_resampler_->Resample(src_left_.get(), src_length_mono,
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dst_left_.get(), dst_capacity_mono);
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sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
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dst_right_.get(), dst_capacity_mono);
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deinterleaved[0] = dst_left_.get();
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deinterleaved[1] = dst_right_.get();
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Interleave(deinterleaved, dst_length_mono, num_channels_, dst);
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return static_cast<int>(dst_length_mono * num_channels_);
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} else {
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return static_cast<int>(
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sinc_resampler_->Resample(src, src_length, dst, dst_capacity));
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}
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}
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// Explictly generate required instantiations.
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template class PushResampler<int16_t>;
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template class PushResampler<float>;
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} // namespace webrtc
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