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 <utility> 12 13 #include "webrtc/base/checks.h" 14 #include "webrtc/modules/audio_processing/test/test_utils.h" 15 16 namespace webrtc { 17 18 RawFile::RawFile(const std::string& filename) 19 : file_handle_(fopen(filename.c_str(), "wb")) {} 20 21 RawFile::~RawFile() { 22 fclose(file_handle_); 23 } 24 25 void RawFile::WriteSamples(const int16_t* samples, size_t num_samples) { 26 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN 27 #error "Need to convert samples to little-endian when writing to PCM file" 28 #endif 29 fwrite(samples, sizeof(*samples), num_samples, file_handle_); 30 } 31 32 void RawFile::WriteSamples(const float* samples, size_t num_samples) { 33 fwrite(samples, sizeof(*samples), num_samples, file_handle_); 34 } 35 36 ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file) 37 : file_(std::move(file)) {} 38 39 bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) { 40 RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels()); 41 interleaved_.resize(buffer->size()); 42 if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) != 43 interleaved_.size()) { 44 return false; 45 } 46 47 FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]); 48 Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(), 49 buffer->channels()); 50 return true; 51 } 52 53 ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file) 54 : file_(std::move(file)) {} 55 56 void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) { 57 RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels()); 58 interleaved_.resize(buffer.size()); 59 Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(), 60 &interleaved_[0]); 61 FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]); 62 file_->WriteSamples(&interleaved_[0], interleaved_.size()); 63 } 64 65 void WriteIntData(const int16_t* data, 66 size_t length, 67 WavWriter* wav_file, 68 RawFile* raw_file) { 69 if (wav_file) { 70 wav_file->WriteSamples(data, length); 71 } 72 if (raw_file) { 73 raw_file->WriteSamples(data, length); 74 } 75 } 76 77 void WriteFloatData(const float* const* data, 78 size_t samples_per_channel, 79 size_t num_channels, 80 WavWriter* wav_file, 81 RawFile* raw_file) { 82 size_t length = num_channels * samples_per_channel; 83 rtc::scoped_ptr<float[]> buffer(new float[length]); 84 Interleave(data, samples_per_channel, num_channels, buffer.get()); 85 if (raw_file) { 86 raw_file->WriteSamples(buffer.get(), length); 87 } 88 // TODO(aluebs): Use ScaleToInt16Range() from audio_util 89 for (size_t i = 0; i < length; ++i) { 90 buffer[i] = buffer[i] > 0 ? 91 buffer[i] * std::numeric_limits<int16_t>::max() : 92 -buffer[i] * std::numeric_limits<int16_t>::min(); 93 } 94 if (wav_file) { 95 wav_file->WriteSamples(buffer.get(), length); 96 } 97 } 98 99 FILE* OpenFile(const std::string& filename, const char* mode) { 100 FILE* file = fopen(filename.c_str(), mode); 101 if (!file) { 102 printf("Unable to open file %s\n", filename.c_str()); 103 exit(1); 104 } 105 return file; 106 } 107 108 size_t SamplesFromRate(int rate) { 109 return static_cast<size_t>(AudioProcessing::kChunkSizeMs * rate / 1000); 110 } 111 112 void SetFrameSampleRate(AudioFrame* frame, 113 int sample_rate_hz) { 114 frame->sample_rate_hz_ = sample_rate_hz; 115 frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs * 116 sample_rate_hz / 1000; 117 } 118 119 AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) { 120 switch (num_channels) { 121 case 1: 122 return AudioProcessing::kMono; 123 case 2: 124 return AudioProcessing::kStereo; 125 default: 126 RTC_CHECK(false); 127 return AudioProcessing::kMono; 128 } 129 } 130 131 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions) { 132 const std::vector<float> values = ParseList<float>(mic_positions); 133 const size_t num_mics = 134 rtc::CheckedDivExact(values.size(), static_cast<size_t>(3)); 135 RTC_CHECK_GT(num_mics, 0u) << "mic_positions is not large enough."; 136 137 std::vector<Point> result; 138 result.reserve(num_mics); 139 for (size_t i = 0; i < values.size(); i += 3) { 140 result.push_back(Point(values[i + 0], values[i + 1], values[i + 2])); 141 } 142 143 return result; 144 } 145 146 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions, 147 size_t num_mics) { 148 std::vector<Point> result = ParseArrayGeometry(mic_positions); 149 RTC_CHECK_EQ(result.size(), num_mics) 150 << "Could not parse mic_positions or incorrect number of points."; 151 return result; 152 } 153 154 } // namespace webrtc 155
