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https://github.com/mollyim/webrtc.git
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1f3679884c
Start introducing ArrayView to AudioFrame and code that flows down from there. In this first step: * Add `data_view()` that returns a read-only ArrayView for the audio buffer. When AudioFrame is not initialized however, data_view() will return a nullptr whereas the current data() method never returns nullptr. * Add `mutable_data()` that requires two arguments for properly setting the samples per channel and number of channels that's required for accurately reserving the returned mutable ArrayView. A notable behavior change is that if the requested number of channels is larger than supported or the calculated buffer size is too large, the function will trigger a check. * Add TODOs for following work. Bug: chromium:335805780 Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42202}
123 lines
4.2 KiB
C++
123 lines
4.2 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 "common_audio/resampler/include/push_resampler.h"
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#include <stdint.h>
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#include <string.h>
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#include <memory>
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#include "common_audio/include/audio_util.h"
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#include "common_audio/resampler/push_sinc_resampler.h"
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#include "rtc_base/checks.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), dst_sample_rate_hz_(0), num_channels_(0) {}
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template <typename T>
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PushResampler<T>::~PushResampler() {}
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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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size_t num_channels) {
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// These checks used to be factored out of this template function due to
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// Windows debug build issues with clang. http://crbug.com/615050
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RTC_DCHECK_GT(src_sample_rate_hz, 0);
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RTC_DCHECK_GT(dst_sample_rate_hz, 0);
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RTC_DCHECK_GT(num_channels, 0);
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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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}
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if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 || num_channels <= 0) {
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return -1;
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}
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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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channel_resamplers_.clear();
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for (size_t i = 0; i < num_channels; ++i) {
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channel_resamplers_.push_back(ChannelResampler());
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auto channel_resampler = channel_resamplers_.rbegin();
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channel_resampler->resampler = std::make_unique<PushSincResampler>(
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src_size_10ms_mono, dst_size_10ms_mono);
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channel_resampler->source.resize(src_size_10ms_mono);
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channel_resampler->destination.resize(dst_size_10ms_mono);
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}
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channel_data_array_.resize(num_channels_);
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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(rtc::ArrayView<const T> src,
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rtc::ArrayView<T> dst) {
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// These checks used to be factored out of this template function due to
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// Windows debug build issues with clang. http://crbug.com/615050
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const size_t src_size_10ms = (src_sample_rate_hz_ / 100) * num_channels_;
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const size_t dst_size_10ms = (dst_sample_rate_hz_ / 100) * num_channels_;
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RTC_DCHECK_EQ(src.size(), src_size_10ms);
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RTC_DCHECK_GE(dst.size(), dst_size_10ms);
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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.data(), src.data(), src.size() * sizeof(T));
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return static_cast<int>(src.size());
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}
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const size_t src_length_mono = src.size() / num_channels_;
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const size_t dst_capacity_mono = dst.size() / num_channels_;
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for (size_t ch = 0; ch < num_channels_; ++ch) {
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channel_data_array_[ch] = channel_resamplers_[ch].source.data();
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}
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Deinterleave(src.data(), src_length_mono, num_channels_,
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channel_data_array_.data());
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size_t dst_length_mono = 0;
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for (auto& resampler : channel_resamplers_) {
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dst_length_mono = resampler.resampler->Resample(
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resampler.source.data(), src_length_mono, resampler.destination.data(),
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dst_capacity_mono);
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}
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for (size_t ch = 0; ch < num_channels_; ++ch) {
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channel_data_array_[ch] = channel_resamplers_[ch].destination.data();
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}
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Interleave(channel_data_array_.data(), dst_length_mono, num_channels_,
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dst.data());
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return static_cast<int>(dst_length_mono * num_channels_);
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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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