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/*
* Copyright (c) 2015 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 "modules/audio_processing/test/test_utils.h"
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/arch.h"
namespace webrtc {
ChannelBufferWavReader::ChannelBufferWavReader(std::unique_ptr<WavReader> file)
: file_(std::move(file)) {}
ChannelBufferWavReader::~ChannelBufferWavReader() = default;
bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
interleaved_.resize(buffer->size());
if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
interleaved_.size()) {
return false;
}
FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
buffer->channels());
return true;
}
ChannelBufferWavWriter::ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)
: file_(std::move(file)) {}
ChannelBufferWavWriter::~ChannelBufferWavWriter() = default;
void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
interleaved_.resize(buffer.size());
InterleavedView<float> view(&interleaved_[0], buffer.num_frames(),
buffer.num_channels());
const float* samples = buffer.channels()[0];
DeinterleavedView<const float> source(samples, buffer.num_frames(),
buffer.num_channels());
Interleave(source, view);
FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
file_->WriteSamples(&interleaved_[0], interleaved_.size());
}
ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
: output_(output) {
RTC_DCHECK(output_);
}
ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;
void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) {
// Account for sample rate changes throughout a simulation.
interleaved_buffer_.resize(buffer.size());
InterleavedView<float> view(&interleaved_buffer_[0], buffer.num_frames(),
buffer.num_channels());
Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
view);
size_t old_size = output_->size();
output_->resize(old_size + interleaved_buffer_.size());
FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
output_->data() + old_size);
}
FILE* OpenFile(absl::string_view filename, absl::string_view mode) {
std::string filename_str(filename);
FILE* file = fopen(filename_str.c_str(), std::string(mode).c_str());
if (!file) {
printf("Unable to open file %s\n", filename_str.c_str());
exit(1);
}
return file;
}
void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz) {
frame->sample_rate_hz = sample_rate_hz;
frame->samples_per_channel =
AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000;
}
} // namespace webrtc
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