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}
124 lines
3.7 KiB
C++
124 lines
3.7 KiB
C++
/*
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* Copyright (c) 2014 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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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
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#include <math.h>
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#include <iterator>
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#include <limits>
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#include <string>
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#include <vector>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/common_audio/wav_file.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/interface/module_common_types.h"
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namespace webrtc {
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static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
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#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
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class RawFile final {
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public:
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explicit RawFile(const std::string& filename);
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~RawFile();
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void WriteSamples(const int16_t* samples, size_t num_samples);
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void WriteSamples(const float* samples, size_t num_samples);
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private:
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FILE* file_handle_;
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DISALLOW_COPY_AND_ASSIGN(RawFile);
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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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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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// Exits on failure; do not use in unit tests.
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FILE* OpenFile(const std::string& filename, const char* mode);
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int SamplesFromRate(int rate);
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void SetFrameSampleRate(AudioFrame* frame,
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int sample_rate_hz);
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template <typename T>
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void SetContainerFormat(int sample_rate_hz,
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int num_channels,
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AudioFrame* frame,
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rtc::scoped_ptr<ChannelBuffer<T> >* cb) {
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SetFrameSampleRate(frame, sample_rate_hz);
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frame->num_channels_ = num_channels;
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cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
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}
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AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels);
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template <typename T>
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float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
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float mse = 0;
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float mean = 0;
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*variance = 0;
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for (int i = 0; i < length; ++i) {
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T error = ref[i] - test[i];
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mse += error * error;
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*variance += ref[i] * ref[i];
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mean += ref[i];
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}
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mse /= length;
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*variance /= length;
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mean /= length;
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*variance -= mean * mean;
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float snr = 100; // We assign 100 dB to the zero-error case.
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if (mse > 0)
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snr = 10 * log10(*variance / mse);
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return snr;
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}
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// Returns a vector<T> parsed from whitespace delimited values in to_parse,
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// or an empty vector if the string could not be parsed.
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template<typename T>
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std::vector<T> ParseList(const std::string& to_parse) {
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std::vector<T> values;
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std::istringstream str(to_parse);
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std::copy(
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std::istream_iterator<T>(str),
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std::istream_iterator<T>(),
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std::back_inserter(values));
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return values;
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}
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// Parses the array geometry from the command line.
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//
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// If a vector with size != num_mics is returned, an error has occurred and an
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// appropriate error message has been printed to stdout.
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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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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
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