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2ad75b3956
This CL duplicates a few lines of utility code from //modules/audio_processing:audioproc_test_utils (which contains more testonly things) and allows the possibility to remove testonly from the unpack_aecdump tool. Bug: b/237526033 Change-Id: If2e1dd4cc825429c496091cf8640c67069fb6e6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267701 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37437}
90 lines
3 KiB
C++
90 lines
3 KiB
C++
/*
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* Copyright (c) 2015 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 "modules/audio_processing/test/test_utils.h"
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#include <utility>
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#include "rtc_base/checks.h"
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#include "rtc_base/system/arch.h"
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namespace webrtc {
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ChannelBufferWavReader::ChannelBufferWavReader(std::unique_ptr<WavReader> file)
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: file_(std::move(file)) {}
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ChannelBufferWavReader::~ChannelBufferWavReader() = default;
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bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
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RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
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interleaved_.resize(buffer->size());
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if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
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interleaved_.size()) {
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return false;
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}
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FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
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Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
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buffer->channels());
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return true;
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}
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ChannelBufferWavWriter::ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)
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: file_(std::move(file)) {}
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ChannelBufferWavWriter::~ChannelBufferWavWriter() = default;
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void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
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RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
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interleaved_.resize(buffer.size());
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Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
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&interleaved_[0]);
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FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
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file_->WriteSamples(&interleaved_[0], interleaved_.size());
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}
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ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
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: output_(output) {
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RTC_DCHECK(output_);
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}
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ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;
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void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) {
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// Account for sample rate changes throughout a simulation.
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interleaved_buffer_.resize(buffer.size());
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Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
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interleaved_buffer_.data());
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size_t old_size = output_->size();
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output_->resize(old_size + interleaved_buffer_.size());
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FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
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output_->data() + old_size);
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}
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FILE* OpenFile(const std::string& filename, const char* mode) {
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FILE* file = fopen(filename.c_str(), mode);
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if (!file) {
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printf("Unable to open file %s\n", filename.c_str());
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exit(1);
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}
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return file;
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}
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size_t SamplesFromRate(int rate) {
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return static_cast<size_t>(AudioProcessing::kChunkSizeMs * rate / 1000);
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}
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void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz) {
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frame->sample_rate_hz = sample_rate_hz;
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frame->samples_per_channel =
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AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000;
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}
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} // namespace webrtc
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