Add WAV and arbitrary geometry support to nlbf test.
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}
This commit is contained in:
Andrew MacDonald
@ -8,13 +8,18 @@
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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 <math.h>
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#include <limits>
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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 "webrtc/audio_processing/debug.pb.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/include/audio_util.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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@ -24,84 +29,38 @@ 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 {
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class RawFile final {
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public:
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RawFile(const std::string& filename)
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: file_handle_(fopen(filename.c_str(), "wb")) {}
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explicit RawFile(const std::string& filename);
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~RawFile();
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~RawFile() {
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fclose(file_handle_);
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}
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void 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 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 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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static inline 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 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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static inline void WriteFloatData(const float* const* data,
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size_t 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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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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static inline 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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FILE* OpenFile(const std::string& filename, const char* mode);
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static inline int SamplesFromRate(int rate) {
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return AudioProcessing::kChunkSizeMs * rate / 1000;
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}
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int SamplesFromRate(int rate);
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static inline 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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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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@ -113,47 +72,7 @@ void SetContainerFormat(int sample_rate_hz,
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cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
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}
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static inline AudioProcessing::ChannelLayout LayoutFromChannels(
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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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// Allocates new memory in the scoped_ptr to fit the raw message and returns the
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// number of bytes read.
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static inline size_t ReadMessageBytesFromFile(
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FILE* file,
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rtc::scoped_ptr<uint8_t[]>* bytes) {
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// The "wire format" for the size is little-endian. Assume we're running on
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// a little-endian machine.
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int32_t size = 0;
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if (fread(&size, sizeof(size), 1, file) != 1)
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return 0;
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if (size <= 0)
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return 0;
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bytes->reset(new uint8_t[size]);
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return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
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}
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// Returns true on success, false on error or end-of-file.
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static inline bool ReadMessageFromFile(FILE* file,
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::google::protobuf::MessageLite* msg) {
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rtc::scoped_ptr<uint8_t[]> bytes;
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size_t size = ReadMessageBytesFromFile(file, &bytes);
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if (!size)
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return false;
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msg->Clear();
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return msg->ParseFromArray(bytes.get(), size);
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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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@ -177,4 +96,28 @@ float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
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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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