shithub: audio-stretch

--- a/README

+++ b/README

@@ -40,6 +40,13 @@

 them correctly (by resampling), but it is possible that some applications

 will barf on them.

+For version 0.2 a new option was added to cycle through the full possible

+ratio range in a sinusoidal pattern, starting at 1.0, and either going

+up (-c) or down (-cc) first. In this case any specified ratio is ignored

+(except if the -s option is also specified to scale the sampling rate).

+The total period is fixed at 2π seconds, at which point the output will

+again be exactly aligned with the input.

 To build the demo app:

     $ gcc -O2 *.c -o audio-stretch

@@ -46,7 +53,7 @@

 The "help" display from the demo app:

- AUDIO-STRETCH  Time Domain Harmonic Scaling Demo  Version 0.1

+ AUDIO-STRETCH  Time Domain Harmonic Scaling Demo  Version 0.2

  Copyright (c) 2019 David Bryant. All Rights Reserved.

  Usage:     AUDIO-STRETCH [-options] infile.wav outfile.wav

@@ -54,6 +61,8 @@

  Options:  -r<n.n> = stretch ratio (0.5 to 2.0, default = 1.0)

            -u<n>   = upper freq period limit (default = 333 Hz)

            -l<n>   = lower freq period limit (default = 55 Hz)

+           -c      = cycle through all ratios, starting higher

+           -cc     = cycle through all ratios, starting lower

            -s      = scale rate to preserve duration (not pitch)

            -f      = fast pitch detection (default >= 32 kHz)

            -n      = normal pitch detection (default < 32 kHz)

@@ -68,6 +77,7 @@

 1. The program will handle only mono or stereo files in the WAV format. The

    audio must be 16-bit PCM and the acceptable sampling rates are from 8,000

    to 48,000 Hz. Any additional RIFF info in the WAV file will be discarded.

+   The command-line program is only for little-endian architectures.

 2. For stereo files, the pitch detection is done on a mono conversion of the

    audio, but the scaling transformation is done on the independent channels.

--- a/main.c

+++ b/main.c

@@ -14,11 +14,12 @@

 #include <stdint.h>

 #include <string.h>

 #include <stdio.h>

+#include <math.h>

 #include "stretch.h"

 static const char *sign_on = "\n"

-" AUDIO-STRETCH  Time Domain Harmonic Scaling Demo  Version 0.1\n"

+" AUDIO-STRETCH  Time Domain Harmonic Scaling Demo  Version 0.2\n"

 " Copyright (c) 2019 David Bryant. All Rights Reserved.\n\n";

 static const char *usage =

@@ -26,6 +27,8 @@

 " Options:  -r<n.n> = stretch ratio (0.5 to 2.0, default = 1.0)\n"

 "           -u<n>   = upper freq period limit (default = 333 Hz)\n"

 "           -l<n>   = lower freq period limit (default = 55 Hz)\n"

+"           -c      = cycle through all ratios, starting higher\n"

+"           -cc     = cycle through all ratios, starting lower\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"

@@ -71,7 +74,7 @@

 int main (argc, argv) int argc; char **argv;

-    int asked_help = 0, overwrite = 0, scale_rate = 0, force_fast = 0, force_normal = 0, fast_mode, scaled_rate;

+    int asked_help = 0, overwrite = 0, scale_rate = 0, force_fast = 0, force_normal = 0, cycle_ratio = 0;

     int upper_frequency = 333, lower_frequency = 55, min_period, max_period;

     int samples_to_process, insamples = 0, outsamples = 0;

     char *infilename = NULL, *outfilename = NULL;

@@ -130,6 +133,10 @@

                         scale_rate = 1;

                         break;

+                    case 'C': case 'c':

+                        cycle_ratio++;

+                        break;

                     case 'F': case 'f':

                         force_fast = 1;

                         break;

@@ -301,15 +308,18 @@

     min_period = WaveHeader.SampleRate / upper_frequency;

     max_period = WaveHeader.SampleRate / lower_frequency;

-    fast_mode = (force_fast || WaveHeader.SampleRate >= 32000) && !force_normal;

+    int fast_mode = (force_fast || WaveHeader.SampleRate >= 32000) && !force_normal;

     if (verbose_mode)

         fprintf (stderr, "initializing stretch library with period range = %d to %d, %d channels, %s\n",

             min_period, max_period, WaveHeader.NumChannels, fast_mode ? "fast mode" : "normal mode");

-    if (!quiet_mode && ratio == 1.0)

+    if (!quiet_mode && ratio == 1.0 && !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, fast_mode);

     if (!stretcher) {

@@ -324,11 +334,11 @@

         return 1;

-    scaled_rate = scale_rate ? (int)(WaveHeader.SampleRate * ratio + 0.5) : WaveHeader.SampleRate;

+    int scaled_rate = scale_rate ? (int)(WaveHeader.SampleRate * ratio + 0.5) : WaveHeader.SampleRate;

     write_pcm_wav_header (outfile, 0, WaveHeader.NumChannels, 2, scaled_rate);

     short *inbuffer = malloc (BUFFER_SAMPLES * WaveHeader.BlockAlign);

-    short *outbuffer = malloc ((BUFFER_SAMPLES * 2 + 2048) * WaveHeader.BlockAlign);

+    short *outbuffer = malloc ((BUFFER_SAMPLES * 2 + max_period * 3) * WaveHeader.BlockAlign);

     while (1) {

         int samples_read = fread (inbuffer, WaveHeader.BlockAlign,

@@ -337,6 +347,9 @@

         insamples += samples_read;

         samples_to_process -= samples_read;

+        if (cycle_ratio)

+            ratio = (sin ((double) outsamples / WaveHeader.SampleRate) * (cycle_ratio & 1 ? 0.75 : -0.75)) + 1.25;

         if (samples_read)

             samples_generated = stretch_samples (stretcher, inbuffer, samples_read, outbuffer, ratio);