aboutsummaryrefslogtreecommitdiff
path: root/webrtc/modules/audio_processing/test/test_utils.cc
blob: 74f8b7388210fa741da904129d0c4c3046e1875e (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
/*
 *  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 <utility>

#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_processing/test/test_utils.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(rtc::scoped_ptr<WavReader> file)
    : file_(std::move(file)) {}

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(rtc::scoped_ptr<WavWriter> file)
    : file_(std::move(file)) {}

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());
}

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,
                    int num_channels,
                    WavWriter* wav_file,
                    RawFile* raw_file) {
  size_t length = num_channels * samples_per_channel;
  rtc::scoped_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(AudioFrame* 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(int num_channels) {
  switch (num_channels) {
    case 1:
      return AudioProcessing::kMono;
    case 2:
      return AudioProcessing::kStereo;
    default:
      RTC_CHECK(false);
      return AudioProcessing::kMono;
  }
}

std::vector<Point> ParseArrayGeometry(const std::string& mic_positions) {
  const std::vector<float> values = ParseList<float>(mic_positions);
  const size_t num_mics =
      rtc::CheckedDivExact(values.size(), static_cast<size_t>(3));
  RTC_CHECK_GT(num_mics, 0u) << "mic_positions is not large enough.";

  std::vector<Point> result;
  result.reserve(num_mics);
  for (size_t i = 0; i < values.size(); i += 3) {
    result.push_back(Point(values[i + 0], values[i + 1], values[i + 2]));
  }

  return result;
}

std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
                                      size_t num_mics) {
  std::vector<Point> result = ParseArrayGeometry(mic_positions);
  RTC_CHECK_EQ(result.size(), num_mics)
      << "Could not parse mic_positions or incorrect number of points.";
  return result;
}

}  // namespace webrtc