blob: 37a20cee1e7da42fa29979f4c5084da12d9d622f [file] [log] [blame]
/*
* 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 <utility>
#include "rtc_base/checks.h"
#include "rtc_base/system/arch.h"
namespace webrtc {
RawFile::RawFile(const std::string& filename)
: file_handle_(fopen(filename.c_str(), "wb")) {}
RawFile::~RawFile() {
fclose(file_handle_);
}
void RawFile::WriteSamples(const int16_t* samples, size_t num_samples) {
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
#error "Need to convert samples to little-endian when writing to PCM file"
#endif
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
void RawFile::WriteSamples(const float* samples, size_t num_samples) {
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
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());
Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
&interleaved_[0]);
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());
Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
interleaved_buffer_.data());
size_t old_size = output_->size();
output_->resize(old_size + interleaved_buffer_.size());
FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
output_->data() + old_size);
}
void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,
RawFile* raw_file) {
if (wav_file) {
wav_file->WriteSamples(data, length);
}
if (raw_file) {
raw_file->WriteSamples(data, length);
}
}
void WriteFloatData(const float* const* data,
size_t samples_per_channel,
size_t num_channels,
WavWriter* wav_file,
RawFile* raw_file) {
size_t length = num_channels * samples_per_channel;
std::unique_ptr<float[]> buffer(new float[length]);
Interleave(data, samples_per_channel, num_channels, buffer.get());
if (raw_file) {
raw_file->WriteSamples(buffer.get(), length);
}
// TODO(aluebs): Use ScaleToInt16Range() from audio_util
for (size_t i = 0; i < length; ++i) {
buffer[i] = buffer[i] > 0
? buffer[i] * std::numeric_limits<int16_t>::max()
: -buffer[i] * std::numeric_limits<int16_t>::min();
}
if (wav_file) {
wav_file->WriteSamples(buffer.get(), length);
}
}
FILE* OpenFile(const std::string& filename, const char* mode) {
FILE* file = fopen(filename.c_str(), mode);
if (!file) {
printf("Unable to open file %s\n", filename.c_str());
exit(1);
}
return file;
}
size_t SamplesFromRate(int rate) {
return static_cast<size_t>(AudioProcessing::kChunkSizeMs * rate / 1000);
}
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;
}
AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
switch (num_channels) {
case 1:
return AudioProcessing::kMono;
case 2:
return AudioProcessing::kStereo;
default:
RTC_CHECK(false);
return AudioProcessing::kMono;
}
}
} // namespace webrtc