blob: 0bf6f27d61b9824714cc0d696e8cbe04a8c600b9 [file] [log] [blame]
/*
* Copyright (c) 2014 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 <math.h>
#include <limits>
#include "webrtc/audio_processing/debug.pb.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
class RawFile {
public:
RawFile(const std::string& filename)
: file_handle_(fopen(filename.c_str(), "wb")) {}
~RawFile() {
fclose(file_handle_);
}
void 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 WriteSamples(const float* samples, size_t num_samples) {
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
private:
FILE* file_handle_;
};
static inline 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);
}
}
static inline 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;
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);
}
}
// Exits on failure; do not use in unit tests.
static inline 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;
}
static inline int SamplesFromRate(int rate) {
return AudioProcessing::kChunkSizeMs * rate / 1000;
}
static inline 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;
}
template <typename T>
void SetContainerFormat(int sample_rate_hz,
int num_channels,
AudioFrame* frame,
scoped_ptr<ChannelBuffer<T> >* cb) {
SetFrameSampleRate(frame, sample_rate_hz);
frame->num_channels_ = num_channels;
cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
}
static inline AudioProcessing::ChannelLayout LayoutFromChannels(
int num_channels) {
switch (num_channels) {
case 1:
return AudioProcessing::kMono;
case 2:
return AudioProcessing::kStereo;
default:
assert(false);
return AudioProcessing::kMono;
}
}
// Allocates new memory in the scoped_ptr to fit the raw message and returns the
// number of bytes read.
static inline size_t ReadMessageBytesFromFile(FILE* file,
scoped_ptr<uint8_t[]>* bytes) {
// The "wire format" for the size is little-endian. Assume we're running on
// a little-endian machine.
int32_t size = 0;
if (fread(&size, sizeof(size), 1, file) != 1)
return 0;
if (size <= 0)
return 0;
bytes->reset(new uint8_t[size]);
return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
}
// Returns true on success, false on error or end-of-file.
static inline bool ReadMessageFromFile(FILE* file,
::google::protobuf::MessageLite* msg) {
scoped_ptr<uint8_t[]> bytes;
size_t size = ReadMessageBytesFromFile(file, &bytes);
if (!size)
return false;
msg->Clear();
return msg->ParseFromArray(bytes.get(), size);
}
template <typename T>
float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
float mse = 0;
float mean = 0;
*variance = 0;
for (int i = 0; i < length; ++i) {
T error = ref[i] - test[i];
mse += error * error;
*variance += ref[i] * ref[i];
mean += ref[i];
}
mse /= length;
*variance /= length;
mean /= length;
*variance -= mean * mean;
float snr = 100; // We assign 100 dB to the zero-error case.
if (mse > 0)
snr = 10 * log10(*variance / mse);
return snr;
}
} // namespace webrtc