cb05b72eb2
This adds functionality from audioproc_float. The geometry parsing code is now shared from test_utils.h. I removed the "mic_spacing" flag from audioproc_float because it's a redundancy that I suspect isn't very useful. Includes a cleanup of the audio_processing test utils. They're now packaged in targets, with the protobuf-using ones split out to avoid requiring users to depend on protobufs. pcm_utils is no longer needed and removed. The primary motivation for this CL is that AudioProcessing currently doesn't support more than two channels and we'd like a way to pass more channels to the beamformer. R=aluebs@webrtc.org, mgraczyk@chromium.org Review URL: https://webrtc-codereview.appspot.com/50899004 Cr-Commit-Position: refs/heads/master@{#9157}
120 lines
3.5 KiB
C++
120 lines
3.5 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 "webrtc/base/checks.h"
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#include "webrtc/modules/audio_processing/test/test_utils.h"
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namespace webrtc {
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RawFile::RawFile(const std::string& filename)
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: file_handle_(fopen(filename.c_str(), "wb")) {}
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RawFile::~RawFile() {
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fclose(file_handle_);
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}
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void RawFile::WriteSamples(const int16_t* samples, size_t num_samples) {
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#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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#error "Need to convert samples to little-endian when writing to PCM file"
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#endif
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fwrite(samples, sizeof(*samples), num_samples, file_handle_);
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}
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void RawFile::WriteSamples(const float* samples, size_t num_samples) {
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fwrite(samples, sizeof(*samples), num_samples, file_handle_);
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}
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void WriteIntData(const int16_t* data,
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size_t length,
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WavWriter* wav_file,
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RawFile* raw_file) {
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if (wav_file) {
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wav_file->WriteSamples(data, length);
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}
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if (raw_file) {
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raw_file->WriteSamples(data, length);
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}
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}
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void WriteFloatData(const float* const* data,
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int samples_per_channel,
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int num_channels,
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WavWriter* wav_file,
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RawFile* raw_file) {
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size_t length = num_channels * samples_per_channel;
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rtc::scoped_ptr<float[]> buffer(new float[length]);
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Interleave(data, samples_per_channel, num_channels, buffer.get());
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if (raw_file) {
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raw_file->WriteSamples(buffer.get(), length);
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}
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// TODO(aluebs): Use ScaleToInt16Range() from audio_util
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for (size_t i = 0; i < length; ++i) {
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buffer[i] = buffer[i] > 0 ?
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buffer[i] * std::numeric_limits<int16_t>::max() :
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-buffer[i] * std::numeric_limits<int16_t>::min();
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}
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if (wav_file) {
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wav_file->WriteSamples(buffer.get(), length);
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}
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}
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FILE* OpenFile(const std::string& filename, const char* mode) {
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FILE* file = fopen(filename.c_str(), mode);
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if (!file) {
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printf("Unable to open file %s\n", filename.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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int SamplesFromRate(int rate) {
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return AudioProcessing::kChunkSizeMs * rate / 1000;
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}
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void SetFrameSampleRate(AudioFrame* frame,
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int sample_rate_hz) {
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frame->sample_rate_hz_ = sample_rate_hz;
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frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
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sample_rate_hz / 1000;
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}
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AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels) {
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switch (num_channels) {
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case 1:
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return AudioProcessing::kMono;
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case 2:
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return AudioProcessing::kStereo;
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default:
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assert(false);
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return AudioProcessing::kMono;
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}
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}
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std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
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size_t num_mics) {
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const std::vector<float> values = ParseList<float>(mic_positions);
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CHECK_EQ(values.size(), 3 * num_mics) <<
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"Could not parse mic_positions or incorrect number of points.";
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std::vector<Point> result;
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result.reserve(num_mics);
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for (size_t i = 0; i < values.size(); i += 3) {
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double x = values[i + 0];
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double y = values[i + 1];
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double z = values[i + 2];
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result.push_back(Point(x, y, z));
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}
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return result;
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}
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} // namespace webrtc
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