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f58ded7cf0
Interleave and Deinterleave now accept two parameters, one for the interleaved buffer and another for the deinterleaved one. The previous versions of the functions still need to exist for test code that uses ChannelBuffer. Bug: chromium:335805780 Change-Id: I20371ab6408766d21e6901e6a04000afa05b3553 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351664 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42412}
95 lines
3.4 KiB
C++
95 lines
3.4 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 <string>
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#include <utility>
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#include "absl/strings/string_view.h"
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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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InterleavedView<float> view(&interleaved_[0], buffer.num_frames(),
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buffer.num_channels());
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const float* samples = buffer.channels()[0];
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DeinterleavedView<const float> source(samples, buffer.num_frames(),
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buffer.num_channels());
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Interleave(source, view);
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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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InterleavedView<float> view(&interleaved_buffer_[0], buffer.num_frames(),
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buffer.num_channels());
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Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
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view);
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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(absl::string_view filename, absl::string_view mode) {
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std::string filename_str(filename);
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FILE* file = fopen(filename_str.c_str(), std::string(mode).c_str());
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if (!file) {
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printf("Unable to open file %s\n", filename_str.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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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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