ref: 07a813e5fbfa132bb7cbf9d2545711a22f33d2f5
parent: 9191cec9babef8179e4345354cbf4eda77e7b183
author: robs <robs>
date: Mon Mar 24 16:41:04 EDT 2008
add some comments
--- a/src/example1.c
+++ b/src/example1.c
@@ -1,7 +1,7 @@
/*
* Simple example of using SoX libraries
*
- * Copyright (c) 2007 robs@users.sourceforge.net
+ * Copyright (c) 2007-8 robs@users.sourceforge.net
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
@@ -25,31 +25,61 @@
#endif
#include <assert.h>
-static sox_format_t * in, * out;
+static sox_format_t * in, * out; /* input and output files */
-static int input_drain(sox_effect_t *effp, sox_sample_t * obuf, sox_size_t * osamp)
+/* The function that will be called to input samples into the effects chain.
+ * In this example, we get samples to process from a SoX-openned audio file.
+ * In a different application, they might be generated or come from a different
+ * part of the application. */
+static int input_drain(
+ sox_effect_t * effp, sox_sample_t * obuf, sox_size_t * osamp)
{+ (void)effp; /* This parameter is not needed in this example */
+
+ /* ensure that *osamp is a multiple of the number of channels. */
*osamp -= *osamp % effp->out_signal.channels;
+
+ /* Read up to *osamp samples into obuf; store the actual number read
+ * back to *osamp */
*osamp = sox_read(in, obuf, *osamp);
- *osamp -= *osamp % effp->out_signal.channels;
+
+ /* sox_read may return a number that is less than was requested; only if
+ * 0 samples is returned does it indicate that end-of-file has been reached
+ * or an error has occurred */
if (!*osamp && in->sox_errno)
fprintf(stderr, "%s: %s\n", in->filename, in->sox_errstr);
return *osamp? SOX_SUCCESS : SOX_EOF;
}
+/* The function that will be called to output samples from the effects chain.
+ * In this example, we store the samples in a SoX-openned audio file.
+ * In a different application, they might perhaps be analysed in some way,
+ * or displayed as a wave-form */
static int output_flow(sox_effect_t *effp UNUSED, sox_sample_t const * ibuf,
sox_sample_t * obuf UNUSED, sox_size_t * isamp, sox_size_t * osamp)
{+ /* Write out *isamp samples */
size_t len = sox_write(out, ibuf, *isamp);
- *osamp = 0;
+ /* len is the number of samples that were actually written out; if this is
+ * different to *isamp, then something has gone wrong--most often, it's
+ * out of disc space */
if (len != *isamp) {fprintf(stderr, "%s: %s\n", out->filename, out->sox_errstr);
return SOX_EOF;
}
- return SOX_SUCCESS;
+
+ /* Outputting is the last `effect' in the effect chain so always passes
+ * 0 samples on to the next effect (as there isn't one!) */
+ *osamp = 0;
+
+ (void)effp; /* This parameter is not needed in this example */
+
+ return SOX_SUCCESS; /* All samples output successfully */
}
+/* A `stub' effect handler to handle inputting samples to the effects
+ * chain; the only function needed for this example is `drain' */
static sox_effect_handler_t const * input_handler(void)
{ static sox_effect_handler_t handler = {@@ -58,6 +88,8 @@
return &handler;
}
+/* A `stub' effect handler to handle outputting samples from the effects
+ * chain; the only function needed for this example is `flow' */
static sox_effect_handler_t const * output_handler(void)
{ static sox_effect_handler_t handler = {@@ -77,29 +109,51 @@
char * vol[] = {"3dB"};assert(argc == 3);
+
+ /* All libSoX applications must start by initialising the SoX library */
assert(sox_format_init() == SOX_SUCCESS);
+ /* Open the input file (with default parameters) */
assert(in = sox_open_read(argv[1], NULL, NULL, NULL));
+
+ /* Open the output file; we must specify the output signal characteristics.
+ * Since we are using only simple effects, they are the same as the input
+ * file characteristics */
assert(out = sox_open_write(NULL, argv[2], &in->signal, NULL, NULL, NULL, 0, NULL, 0));
+ /* Create an effects chain; some effects need to know about the input
+ * or output file encoding so we provide that information here */
chain = sox_create_effects_chain(&in->encoding, &out->encoding);
+ /* The first effect in the effect chain must be something that can source
+ * samples; in this case, we have defined an input handler that inputs
+ * data from an audio file */
sox_create_effect(&e, input_handler());
+ /* This becomes the first `effect' in the chain */
assert(sox_add_effect(chain, &e, &in->signal, &in->signal) == SOX_SUCCESS);
+ /* Create the `vol' effect, and initialise it with the desired parameters: */
sox_create_effect(&e, sox_find_effect("vol"));assert(e.handler.getopts(&e, 1, vol) == SOX_SUCCESS);
+ /* Add the effect to the end of the effects processing chain: */
assert(sox_add_effect(chain, &e, &in->signal, &in->signal) == SOX_SUCCESS);
+ /* Create the `flanger' effect, and initialise it with default parameters: */
sox_create_effect(&e, sox_find_effect("flanger"));assert(e.handler.getopts(&e, 0, NULL) == SOX_SUCCESS);
+ /* Add the effect to the end of the effects processing chain: */
assert(sox_add_effect(chain, &e, &in->signal, &in->signal) == SOX_SUCCESS);
+ /* The last effect in the effect chain must be something that only consumes
+ * samples; in this case, we have defined an output handler that outputs
+ * data to an audio file */
sox_create_effect(&e, output_handler());
assert(sox_add_effect(chain, &e, &in->signal, &in->signal) == SOX_SUCCESS);
+ /* Flow samples through the effects processing chain until EOF is reached */
sox_flow_effects(chain, NULL);
+ /* All done; tidy up: */
sox_delete_effects(chain);
sox_close(out);
sox_close(in);
--- a/src/example2.c
+++ b/src/example2.c
@@ -27,7 +27,6 @@
#endif
#include <assert.h>
-
/*
* Reads input file and displays a few seconds of wave-form, starting from
* a given time through the audio. E.g. example2 song2.au 30.75 1
@@ -37,39 +36,51 @@
sox_format_t * in;
sox_sample_t * buf;
size_t blocks, block_size;
+ /* Period of audio over which we will measure its volume in order to
+ * display the wave-form: */
static const double block_period = 0.025; /* seconds */
double start_secs = 0, period = 2;
char dummy;
sox_size_t seek;
+ /* All libSoX applications must start by initialising the SoX library */
assert(sox_format_init() == SOX_SUCCESS);
assert(argc > 1);
- ++argv, --argc;
+ ++argv, --argc; /* Move to 1st parameter */
+ /* Open the input file (with default parameters) */
assert(in = sox_open_read(*argv, NULL, NULL, NULL));
- ++argv, --argc;
+ ++argv, --argc; /* Move past this parameter */
- if (argc) {+ if (argc) { /* If given, read the start time: */assert(sscanf(*argv, "%lf%c", &start_secs, &dummy) == 1);
- ++argv, --argc;
+ ++argv, --argc; /* Move past this parameter */
}
- if (argc) {+ if (argc) { /* If given, read the period of time to display: */assert(sscanf(*argv, "%lf%c", &period, &dummy) == 1);
- ++argv, --argc;
+ ++argv, --argc; /* Move past this parameter */
}
+ /* Calculate the start position in number of samples: */
seek = start_secs * in->signal.rate * in->signal.channels + .5;
+ /* Make sure that this is at a `wide sample' boundary: */
seek -= seek % in->signal.channels;
+ /* Move the file pointer to the desired starting position */
assert(sox_seek(in, seek, SOX_SEEK_SET) == SOX_SUCCESS);
+ /* Convert block size (in seconds) to a number of samples: */
block_size = block_period * in->signal.rate * in->signal.channels + .5;
+ /* Make sure that this is at a `wide sample' boundary: */
block_size -= block_size % in->signal.channels;
+ /* Allocate a block of memory to store the block of audio samples: */
assert(buf = malloc(sizeof(sox_sample_t) * block_size));
+ /* This application requires that the audio has precisely 2 channels */
assert(in->signal.channels == 2);
+ /* Read in and process blocks of audio for the selected period or until EOF: */
for (blocks = 0; sox_read(in, buf, block_size) == block_size && blocks * block_period < period; ++blocks) {double left = 0, right = 0;
size_t i, clips = 0;
@@ -77,17 +88,28 @@
int l, r;
for (i = 0; i < block_size; ++i) {+ /* convert the sample from SoX's internal format to a `double' for
+ * processing in this application: */
double sample = SOX_SAMPLE_TO_FLOAT_64BIT(buf[i], clips);
+
+ /* The samples for each channel are interleaved; in this example
+ * we allow only stereo audio, so the left channel audio can be found in
+ * even-numbered samples, and the right channel audio in odd-numbered
+ * samples: */
if (i & 1)
- right = max(right, fabs(sample));
+ right = max(right, fabs(sample)); /* Find the peak volume in the block */
else
- left = max(left, fabs(sample));
+ left = max(left, fabs(sample)); /* Find the peak volume in the block */
}
- l = (1 - left) * 35;
- r = (1 - right) * 35;
+
+ /* Build up the wave form by displaying the left & right channel
+ * volume as a line length: */
+ l = (1 - left) * 35 + .5;
+ r = (1 - right) * 35 + .5;
printf("%8.3f%36s|%s\n", start_secs + blocks * block_period, line + l, line + r);}
+ /* All done; tidy up: */
free(buf);
sox_close(in);
sox_format_quit();