xref: /aosp_15_r20/external/webrtc/modules/audio_processing/test/test_utils.cc (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #include "modules/audio_processing/test/test_utils.h"
12 
13 #include <string>
14 #include <utility>
15 
16 #include "absl/strings/string_view.h"
17 #include "rtc_base/checks.h"
18 #include "rtc_base/system/arch.h"
19 
20 namespace webrtc {
21 
ChannelBufferWavReader(std::unique_ptr<WavReader> file)22 ChannelBufferWavReader::ChannelBufferWavReader(std::unique_ptr<WavReader> file)
23     : file_(std::move(file)) {}
24 
25 ChannelBufferWavReader::~ChannelBufferWavReader() = default;
26 
Read(ChannelBuffer<float> * buffer)27 bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
28   RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
29   interleaved_.resize(buffer->size());
30   if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
31       interleaved_.size()) {
32     return false;
33   }
34 
35   FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
36   Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
37                buffer->channels());
38   return true;
39 }
40 
ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)41 ChannelBufferWavWriter::ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)
42     : file_(std::move(file)) {}
43 
44 ChannelBufferWavWriter::~ChannelBufferWavWriter() = default;
45 
Write(const ChannelBuffer<float> & buffer)46 void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
47   RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
48   interleaved_.resize(buffer.size());
49   Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
50              &interleaved_[0]);
51   FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
52   file_->WriteSamples(&interleaved_[0], interleaved_.size());
53 }
54 
ChannelBufferVectorWriter(std::vector<float> * output)55 ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
56     : output_(output) {
57   RTC_DCHECK(output_);
58 }
59 
60 ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;
61 
Write(const ChannelBuffer<float> & buffer)62 void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) {
63   // Account for sample rate changes throughout a simulation.
64   interleaved_buffer_.resize(buffer.size());
65   Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
66              interleaved_buffer_.data());
67   size_t old_size = output_->size();
68   output_->resize(old_size + interleaved_buffer_.size());
69   FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
70                   output_->data() + old_size);
71 }
72 
OpenFile(absl::string_view filename,absl::string_view mode)73 FILE* OpenFile(absl::string_view filename, absl::string_view mode) {
74   std::string filename_str(filename);
75   FILE* file = fopen(filename_str.c_str(), std::string(mode).c_str());
76   if (!file) {
77     printf("Unable to open file %s\n", filename_str.c_str());
78     exit(1);
79   }
80   return file;
81 }
82 
SetFrameSampleRate(Int16FrameData * frame,int sample_rate_hz)83 void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz) {
84   frame->sample_rate_hz = sample_rate_hz;
85   frame->samples_per_channel =
86       AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000;
87 }
88 
89 }  // namespace webrtc
90