ref: fdf38c9b21956b1aab29005f20bd9d1d5af7fe06
dir: /main.c/
////////////////////////////////////////////////////////////////////////////
// **** AUDIO-STRETCH **** //
// Time Domain Harmonic Scaler //
// Copyright (c) 2022 David Bryant //
// All Rights Reserved. //
// Distributed under the BSD Software License (see license.txt) //
////////////////////////////////////////////////////////////////////////////
// main.c
// This module provides a demo for the TDHS library using WAV files.
#include <stdlib.h>
#include <stdint.h>
#include <string.h>
#include <stdio.h>
#include <math.h>
#include "stretch.h"
#define SILENCE_THRESHOLD_DB -40
#define AUDIO_WINDOW_MS 25
static const char *sign_on = "\n"
" AUDIO-STRETCH Time Domain Harmonic Scaling Demo Version 0.4\n"
" Copyright (c) 2022 David Bryant. All Rights Reserved.\n\n";
static const char *usage =
" Usage: AUDIO-STRETCH [-options] infile.wav outfile.wav\n\n"
" Options: -r<n.n> = stretch ratio (0.25 to 4.0, default = 1.0)\n"
" -g<n.n> = gap/silence stretch ratio (if different)\n"
" -u<n> = upper freq period limit (default = 333 Hz)\n"
" -l<n> = lower freq period limit (default = 55 Hz)\n"
" -b<n> = audio buffer/window length (ms, default = 25)\n"
" -t<n> = gap/silence threshold (dB re FS, default = -40)\n"
" -c = cycle through all ratios, starting higher\n"
" -cc = cycle through all ratios, starting lower\n"
" -d = force dual instance even for shallow ratios\n"
" -s = scale rate to preserve duration (not pitch)\n"
" -f = fast pitch detection (default >= 32 kHz)\n"
" -n = normal pitch detection (default < 32 kHz)\n"
" -q = quiet mode (display errors only)\n"
" -v = verbose (display lots of info)\n"
" -y = overwrite outfile if it exists\n\n"
" Web: Visit www.github.com/dbry/audio-stretch for latest version\n\n";
typedef struct {
char ckID [4];
uint32_t ckSize;
char formType [4];
} RiffChunkHeader;
typedef struct {
char ckID [4];
uint32_t ckSize;
} ChunkHeader;
typedef struct {
uint16_t FormatTag, NumChannels;
uint32_t SampleRate, BytesPerSecond;
uint16_t BlockAlign, BitsPerSample;
uint16_t cbSize;
union {
uint16_t ValidBitsPerSample;
uint16_t SamplesPerBlock;
uint16_t Reserved;
} Samples;
int32_t ChannelMask;
uint16_t SubFormat;
char GUID [14];
} WaveHeader;
#define WAVE_FORMAT_PCM 0x1
#define WAVE_FORMAT_EXTENSIBLE 0xfffe
static int write_pcm_wav_header (FILE *outfile, uint32_t num_samples, int num_channels, int bytes_per_sample, uint32_t sample_rate);
double rms_level_dB (int16_t *audio, int samples, int channels);
static int verbose_mode, quiet_mode;
int main (argc, argv) int argc; char **argv;
{
int asked_help = 0, overwrite = 0, scale_rate = 0, force_fast = 0, force_normal = 0, force_dual = 0, cycle_ratio = 0;
float ratio = 1.0, silence_ratio = 0.0, silence_threshold_dB = SILENCE_THRESHOLD_DB;
uint32_t samples_to_process, insamples = 0, outsamples = 0;
int upper_frequency = 333, lower_frequency = 55;
char *infilename = NULL, *outfilename = NULL;
int audio_window_ms = AUDIO_WINDOW_MS;
RiffChunkHeader riff_chunk_header;
WaveHeader WaveHeader = { 0 };
ChunkHeader chunk_header;
StretchHandle stretcher;
FILE *infile, *outfile;
// loop through command-line arguments
while (--argc) {
#ifdef _WIN32
if ((**++argv == '-' || **argv == '/') && (*argv)[1])
#else
if ((**++argv == '-') && (*argv)[1])
#endif
while (*++*argv)
switch (**argv) {
case 'U': case 'u':
upper_frequency = strtol (++*argv, argv, 10);
if (upper_frequency <= 40) {
fprintf (stderr, "\nupper frequency must be at least 40 Hz!\n");
return -1;
}
--*argv;
break;
case 'L': case 'l':
lower_frequency = strtol (++*argv, argv, 10);
if (lower_frequency < 20) {
fprintf (stderr, "\nlower frequency must be at least 20 Hz!\n");
return -1;
}
--*argv;
break;
case 'B': case 'b':
audio_window_ms = strtol (++*argv, argv, 10);
if (audio_window_ms < 1 || audio_window_ms > 100) {
fprintf (stderr, "\naudio window is from 1 to 100 ms!\n");
return -1;
}
--*argv;
break;
case 'R': case 'r':
ratio = strtod (++*argv, argv);
if (ratio < 0.25 || ratio > 4.0) {
fprintf (stderr, "\nratio must be from 0.25 to 4.0!\n");
return -1;
}
--*argv;
break;
case 'G': case 'g':
silence_ratio = strtod (++*argv, argv);
if (silence_ratio < 0.25 || silence_ratio > 4.0) {
fprintf (stderr, "\ngap/silence ratio must be from 0.25 to 4.0!\n");
return -1;
}
--*argv;
break;
case 'T': case 't':
silence_threshold_dB = strtod (++*argv, argv);
if (silence_threshold_dB < -70 || silence_threshold_dB > -10) {
fprintf (stderr, "\nsilence threshold must be from -10 to -70 dB!\n");
return -1;
}
--*argv;
break;
case 'S': case 's':
scale_rate = 1;
break;
case 'C': case 'c':
cycle_ratio++;
break;
case 'D': case 'd':
force_dual = 1;
break;
case 'F': case 'f':
force_fast = 1;
break;
case 'N': case 'n':
force_normal = 1;
break;
case 'H': case 'h':
asked_help = 1;
break;
case 'V': case 'v':
verbose_mode = 1;
break;
case 'Q': case 'q':
quiet_mode = 1;
break;
case 'Y': case 'y':
overwrite = 1;
break;
default:
fprintf (stderr, "\nillegal option: %c !\n", **argv);
return -1;
}
else if (!infilename)
infilename = *argv;
else if (!outfilename)
outfilename = *argv;
else {
fprintf (stderr, "\nextra unknown argument: %s !\n", *argv);
return -1;
}
}
if (!quiet_mode)
fprintf (stderr, "%s", sign_on);
if (!outfilename || asked_help) {
printf ("%s", usage);
return 0;
}
if (!strcmp (infilename, outfilename)) {
fprintf (stderr, "can't overwrite input file (specify different/new output file name)\n");
return -1;
}
if (!overwrite && (outfile = fopen (outfilename, "r"))) {
fclose (outfile);
fprintf (stderr, "output file \"%s\" exists (use -y to overwrite)\n", outfilename);
return -1;
}
if (!(infile = fopen (infilename, "rb"))) {
fprintf (stderr, "can't open file \"%s\" for reading!\n", infilename);
return 1;
}
// read initial RIFF form header
if (!fread (&riff_chunk_header, sizeof (RiffChunkHeader), 1, infile) ||
strncmp (riff_chunk_header.ckID, "RIFF", 4) ||
strncmp (riff_chunk_header.formType, "WAVE", 4)) {
fprintf (stderr, "\"%s\" is not a valid .WAV file!\n", infilename);
return 1;
}
// loop through all elements of the RIFF wav header (until the data chuck)
while (1) {
if (!fread (&chunk_header, sizeof (ChunkHeader), 1, infile)) {
fprintf (stderr, "\"%s\" is not a valid .WAV file!\n", infilename);
return 1;
}
// if it's the format chunk, we want to get some info out of there and
// make sure it's a .wav file we can handle
if (!strncmp (chunk_header.ckID, "fmt ", 4)) {
int format, bits_per_sample;
if (chunk_header.ckSize < 16 || chunk_header.ckSize > sizeof (WaveHeader) ||
!fread (&WaveHeader, chunk_header.ckSize, 1, infile)) {
fprintf (stderr, "\"%s\" is not a valid .WAV file!\n", infilename);
return 1;
}
format = (WaveHeader.FormatTag == WAVE_FORMAT_EXTENSIBLE && chunk_header.ckSize == 40) ?
WaveHeader.SubFormat : WaveHeader.FormatTag;
bits_per_sample = (chunk_header.ckSize == 40 && WaveHeader.Samples.ValidBitsPerSample) ?
WaveHeader.Samples.ValidBitsPerSample : WaveHeader.BitsPerSample;
if (bits_per_sample != 16) {
fprintf (stderr, "\"%s\" is not a 16-bit .WAV file!\n", infilename);
return 1;
}
if (WaveHeader.NumChannels < 1 || WaveHeader.NumChannels > 2) {
fprintf (stderr, "\"%s\" is not a mono or stereo .WAV file!\n", infilename);
return 1;
}
if (WaveHeader.BlockAlign != WaveHeader.NumChannels * 2) {
fprintf (stderr, "\"%s\" is not a valid .WAV file!\n", infilename);
return 1;
}
if (format == WAVE_FORMAT_PCM) {
if (WaveHeader.SampleRate < 8000 || WaveHeader.SampleRate > 48000) {
fprintf (stderr, "\"%s\" sample rate is %lu, must be 8000 to 48000!\n", infilename, (unsigned long) WaveHeader.SampleRate);
return 1;
}
}
else {
fprintf (stderr, "\"%s\" is not a PCM .WAV file!\n", infilename);
return 1;
}
}
else if (!strncmp (chunk_header.ckID, "data", 4)) {
// on the data chunk, get size and exit parsing loop
if (!WaveHeader.SampleRate) { // make sure we saw a "fmt" chunk...
fprintf (stderr, "\"%s\" is not a valid .WAV file!\n", infilename);
return 1;
}
if (!chunk_header.ckSize) {
fprintf (stderr, "this .WAV file has no audio samples, probably is corrupt!\n");
return 1;
}
if (chunk_header.ckSize % WaveHeader.BlockAlign) {
fprintf (stderr, "\"%s\" is not a valid .WAV file!\n", infilename);
return 1;
}
samples_to_process = chunk_header.ckSize / WaveHeader.BlockAlign;
if (!samples_to_process) {
fprintf (stderr, "this .WAV file has no audio samples, probably is corrupt!\n");
return 1;
}
break;
}
else { // just ignore unknown chunks
uint32_t bytes_to_eat = (chunk_header.ckSize + 1) & ~1L;
char dummy;
while (bytes_to_eat--)
if (!fread (&dummy, 1, 1, infile)) {
fprintf (stderr, "\"%s\" is not a valid .WAV file!\n", infilename);
return 1;
}
}
}
if (upper_frequency < lower_frequency * 2 || upper_frequency >= WaveHeader.SampleRate / 2) {
fprintf (stderr, "invalid frequencies specified!\n");
fclose (infile);
return 1;
}
int flags = 0, silence_mode = silence_ratio && !cycle_ratio && silence_ratio != ratio;
int buffer_samples = WaveHeader.SampleRate * (audio_window_ms / 1000.0);
int min_period = WaveHeader.SampleRate / upper_frequency;
int max_period = WaveHeader.SampleRate / lower_frequency;
float max_ratio = ratio;
if (force_dual || ratio < 0.5 || ratio > 2.0 ||
(silence_mode && (silence_ratio < 0.5 || silence_ratio > 2.0)))
flags |= STRETCH_DUAL_FLAG;
if ((force_fast || WaveHeader.SampleRate >= 32000) && !force_normal)
flags |= STRETCH_FAST_FLAG;
if (verbose_mode) {
fprintf (stderr, "file sample rate is %lu Hz (%s), buffer size is %d samples\n",
(unsigned long) WaveHeader.SampleRate, WaveHeader.NumChannels == 2 ? "stereo" : "mono", buffer_samples);
fprintf (stderr, "stretch period range = %d to %d, %d channels, %s, %s\n",
min_period, max_period, WaveHeader.NumChannels, (flags & STRETCH_FAST_FLAG) ? "fast mode" : "normal mode",
(flags & STRETCH_DUAL_FLAG) ? "dual instance" : "single instance");
}
if (!quiet_mode && ratio == 1.0 && !silence_mode && !cycle_ratio)
fprintf (stderr, "warning: a ratio of 1.0 will do nothing but copy the WAV file!\n");
if (!quiet_mode && ratio != 1.0 && cycle_ratio && !scale_rate)
fprintf (stderr, "warning: specifying ratio with cycling doesn't do anything (unless scaling rate)\n");
stretcher = stretch_init (min_period, max_period, WaveHeader.NumChannels, flags);
if (!stretcher) {
fprintf (stderr, "can't initialize stretcher\n");
fclose (infile);
return 1;
}
if (!(outfile = fopen (outfilename, "wb"))) {
fprintf (stderr, "can't open file \"%s\" for writing!\n", outfilename);
fclose (infile);
return 1;
}
uint32_t scaled_rate = scale_rate ? (uint32_t)(WaveHeader.SampleRate * ratio + 0.5) : WaveHeader.SampleRate;
write_pcm_wav_header (outfile, 0, WaveHeader.NumChannels, 2, scaled_rate);
if (cycle_ratio)
max_ratio = (flags & STRETCH_DUAL_FLAG) ? 4.0 : 2.0;
else if (silence_mode && silence_ratio > max_ratio)
max_ratio = silence_ratio;
int max_expected_samples = stretch_output_capacity (stretcher, buffer_samples, max_ratio);
int16_t *inbuffer = malloc (buffer_samples * WaveHeader.BlockAlign), *prebuffer = NULL;
int16_t *outbuffer = malloc (max_expected_samples * WaveHeader.BlockAlign);
int non_silence_frames = 0, silence_frames = 0, used_silence_frames = 0;
int max_generated_stretch = 0, max_generated_flush = 0;
int samples_to_stretch = 0, consecutive_silence_frames = 1;
/* in the gap/silence mode we need an additional buffer to scan the "next" buffer for level */
if (silence_mode)
prebuffer = malloc (buffer_samples * WaveHeader.BlockAlign);
if (!inbuffer || !outbuffer || (silence_mode && !prebuffer)) {
fprintf (stderr, "can't allocate required memory!\n");
fclose (infile);
return 1;
}
/* read the entire file in frames and process with stretch */
while (1) {
int samples_read = fread (silence_mode ? prebuffer : inbuffer, WaveHeader.BlockAlign,
samples_to_process >= buffer_samples ? buffer_samples : samples_to_process, infile);
if (!silence_mode && !samples_read)
break;
insamples += samples_read;
samples_to_process -= samples_read;
/* this is where we scan the frame we just read to see if it's below the silence threshold */
if (silence_mode) {
if (samples_read) {
double level = rms_level_dB (prebuffer, samples_read, WaveHeader.NumChannels);
if (level > silence_threshold_dB) {
consecutive_silence_frames = 0;
non_silence_frames++;
}
else {
consecutive_silence_frames++;
silence_frames++;
}
}
}
else
samples_to_stretch = samples_read;
if (cycle_ratio) {
if (flags & STRETCH_DUAL_FLAG)
ratio = (sin ((double) outsamples / WaveHeader.SampleRate / 2.0) * (cycle_ratio & 1 ? 1.875 : -1.875)) + 2.125;
else
ratio = (sin ((double) outsamples / WaveHeader.SampleRate) * (cycle_ratio & 1 ? 0.75 : -0.75)) + 1.25;
}
if (samples_to_stretch) {
int samples_generated;
/* we use the gap/silence stretch ratio if the current frame, and the ones on either side, measure below the threshold */
if (consecutive_silence_frames >= 3) {
samples_generated = stretch_samples (stretcher, inbuffer, samples_to_stretch, outbuffer, silence_ratio);
used_silence_frames++;
}
else
samples_generated = stretch_samples (stretcher, inbuffer, samples_to_stretch, outbuffer, ratio);
if (samples_generated) {
if (samples_generated > max_generated_stretch)
max_generated_stretch = samples_generated;
fwrite (outbuffer, WaveHeader.BlockAlign, samples_generated, outfile);
outsamples += samples_generated;
if (samples_generated > max_expected_samples) {
fprintf (stderr, "stretch: generated samples (%d) exceeded expected (%d)!\n", samples_generated, max_expected_samples);
fclose (infile);
return 1;
}
}
}
if (silence_mode) {
if (samples_read) {
memcpy (inbuffer, prebuffer, samples_read * WaveHeader.BlockAlign);
samples_to_stretch = samples_read;
}
else
break;
}
}
/* next call the stretch flush function until it returns zero */
while (1) {
int samples_flushed = stretch_flush (stretcher, outbuffer);
if (!samples_flushed)
break;
if (samples_flushed > max_generated_flush)
max_generated_flush = samples_flushed;
fwrite (outbuffer, WaveHeader.BlockAlign, samples_flushed, outfile);
outsamples += samples_flushed;
if (samples_flushed > max_expected_samples) {
fprintf (stderr, "flush: generated samples (%d) exceeded expected (%d)!\n", samples_flushed, max_expected_samples);
fclose (infile);
return 1;
}
}
free (inbuffer);
free (outbuffer);
free (prebuffer);
stretch_deinit (stretcher);
fclose (infile);
rewind (outfile);
write_pcm_wav_header (outfile, outsamples, WaveHeader.NumChannels, 2, scaled_rate);
fclose (outfile);
if (insamples && verbose_mode) {
fprintf (stderr, "done, %lu samples --> %lu samples (ratio = %.3f)\n",
(unsigned long) insamples, (unsigned long) outsamples, (float) outsamples / insamples);
if (scale_rate)
fprintf (stderr, "sample rate changed from %lu Hz to %lu Hz\n",
(unsigned long) WaveHeader.SampleRate, (unsigned long) scaled_rate);
fprintf (stderr, "max expected samples = %d, actually seen = %d stretch, %d flush\n",
max_expected_samples, max_generated_stretch, max_generated_flush);
if (silence_frames || non_silence_frames) {
int total_frames = silence_frames + non_silence_frames;
fprintf (stderr, "%d silence frames detected (%.2f%%), %d actually used (%.2f%%)\n",
silence_frames, silence_frames * 100.0 / total_frames,
used_silence_frames, used_silence_frames * 100.0 / total_frames);
}
}
return 0;
}
static int write_pcm_wav_header (FILE *outfile, uint32_t num_samples, int num_channels, int bytes_per_sample, uint32_t sample_rate)
{
RiffChunkHeader riffhdr;
ChunkHeader datahdr, fmthdr;
WaveHeader wavhdr;
int wavhdrsize = 16;
uint32_t total_data_bytes = num_samples * bytes_per_sample * num_channels;
memset (&wavhdr, 0, sizeof (wavhdr));
wavhdr.FormatTag = WAVE_FORMAT_PCM;
wavhdr.NumChannels = num_channels;
wavhdr.SampleRate = sample_rate;
wavhdr.BytesPerSecond = sample_rate * num_channels * bytes_per_sample;
wavhdr.BlockAlign = bytes_per_sample * num_channels;
wavhdr.BitsPerSample = bytes_per_sample * 8;
memcpy (riffhdr.ckID, "RIFF", sizeof (riffhdr.ckID));
memcpy (riffhdr.formType, "WAVE", sizeof (riffhdr.formType));
riffhdr.ckSize = sizeof (riffhdr) + wavhdrsize + sizeof (datahdr) + total_data_bytes;
memcpy (fmthdr.ckID, "fmt ", sizeof (fmthdr.ckID));
fmthdr.ckSize = wavhdrsize;
memcpy (datahdr.ckID, "data", sizeof (datahdr.ckID));
datahdr.ckSize = total_data_bytes;
return fwrite (&riffhdr, sizeof (riffhdr), 1, outfile) &&
fwrite (&fmthdr, sizeof (fmthdr), 1, outfile) &&
fwrite (&wavhdr, wavhdrsize, 1, outfile) &&
fwrite (&datahdr, sizeof (datahdr), 1, outfile);
}
double rms_level_dB (int16_t *audio, int samples, int channels)
{
double rms_sum = 0.0;
int i;
if (channels == 1)
for (i = 0; i < samples; ++i)
rms_sum += (double) audio [i] * audio [i];
else
for (i = 0; i < samples; ++i) {
double average = (audio [i * 2] + audio [i * 2 + 1]) / 2.0;
rms_sum += average * average;
}
return log10 (rms_sum / samples / (32768.0 * 32767.0 * 0.5)) * 10.0;
}