/*
* Process options
*/
static int sox_trim_getopts(sox_effect_t * effp, int argc, char **argv)
{
char *end;
priv_t * trim = (priv_t *) effp->priv;
--argc, ++argv;
/* Do not know sample rate yet so hold off on completely parsing
* time related strings.
*/
switch (argc) {
case 2:
end = argv[1];
if (*end == '=') {
trim->end_is_absolute = sox_true;
end++;
} else trim->end_is_absolute = sox_false;
trim->end_str = lsx_malloc(strlen(end)+1);
strcpy(trim->end_str, end);
/* Do a dummy parse to see if it will fail */
if (lsx_parsesamples(0., trim->end_str, &trim->length, 't') == NULL)
return lsx_usage(effp);
case 1:
trim->start_str = lsx_malloc(strlen(argv[0])+1);
strcpy(trim->start_str,argv[0]);
/* Do a dummy parse to see if it will fail */
if (lsx_parsesamples(0., trim->start_str, &trim->start, 't') == NULL)
return lsx_usage(effp);
break;
default:
return lsx_usage(effp);
}
return (SOX_SUCCESS);
}
/*
* Process options
*/
static int sox_trim_getopts(sox_effect_t * effp, int n, char **argv)
{
priv_t * trim = (priv_t *) effp->priv;
/* Do not know sample rate yet so hold off on completely parsing
* time related strings.
*/
switch (n) {
case 2:
trim->length_str = lsx_malloc(strlen(argv[1])+1);
strcpy(trim->length_str,argv[1]);
/* Do a dummy parse to see if it will fail */
if (lsx_parsesamples(0., trim->length_str, &trim->length, 't') == NULL)
return lsx_usage(effp);
case 1:
trim->start_str = lsx_malloc(strlen(argv[0])+1);
strcpy(trim->start_str,argv[0]);
/* Do a dummy parse to see if it will fail */
if (lsx_parsesamples(0., trim->start_str, &trim->start, 't') == NULL)
return lsx_usage(effp);
break;
default:
return lsx_usage(effp);
}
return (SOX_SUCCESS);
}
/*
* Process options
*
* Don't do initialization now.
* The 'info' fields are not yet filled in.
*/
static int sox_mcompand_getopts_1(comp_band_t * l, size_t n, char **argv)
{
char *s;
size_t rates, i, commas;
/* Start by checking the attack and decay rates */
for (s = argv[0], commas = 0; *s; ++s)
if (*s == ',') ++commas;
if (commas % 2 == 0) /* There must be an even number of
attack/decay parameters */
{
lsx_fail("compander: Odd number of attack & decay rate parameters");
return (SOX_EOF);
}
rates = 1 + commas/2;
l->attackRate = lsx_malloc(sizeof(double) * rates);
l->decayRate = lsx_malloc(sizeof(double) * rates);
l->volume = lsx_malloc(sizeof(double) * rates);
l->expectedChannels = rates;
l->delay_buf = NULL;
/* Now tokenise the rates string and set up these arrays. Keep
them in seconds at the moment: we don't know the sample rate yet. */
s = strtok(argv[0], ","); i = 0;
do {
l->attackRate[i] = atof(s); s = strtok(NULL, ",");
l->decayRate[i] = atof(s); s = strtok(NULL, ",");
++i;
} while (s != NULL);
if (!lsx_compandt_parse(&l->transfer_fn, argv[1], n>2 ? argv[2] : 0))
return SOX_EOF;
/* Set the initial "volume" to be attibuted to the input channels.
Unless specified, choose 1.0 (maximum) otherwise clipping will
result if the user has seleced a long attack time */
for (i = 0; i < l->expectedChannels; ++i) {
double v = n>=4 ? pow(10.0, atof(argv[3])/20) : 1.0;
l->volume[i] = v;
/* If there is a delay, store it. */
if (n >= 5) l->delay = atof(argv[4]);
else l->delay = 0.0;
}
return (SOX_SUCCESS);
}
static int sox_silence_start(sox_effect_t * effp)
{
priv_t *silence = (priv_t *)effp->priv;
uint64_t temp;
/* When you want to remove silence, small window sizes are
* better or else RMS will look like non-silence at
* aburpt changes from load to silence.
*/
silence->window_size = (effp->in_signal.rate / 50) *
effp->in_signal.channels;
silence->window = lsx_malloc(silence->window_size * sizeof(double));
clear_rms(effp);
/* Now that we know sample rate, reparse duration. */
if (silence->start)
{
if (lsx_parsesamples(effp->in_signal.rate, silence->start_duration_str,
&temp, 's') == NULL)
return lsx_usage(effp);
silence->start_duration = temp * effp->in_signal.channels;
}
if (silence->stop)
{
if (lsx_parsesamples(effp->in_signal.rate,silence->stop_duration_str,
&temp,'s') == NULL)
return lsx_usage(effp);
silence->stop_duration = temp * effp->in_signal.channels;
}
if (silence->start)
silence->mode = SILENCE_TRIM;
else
silence->mode = SILENCE_COPY;
silence->start_holdoff = lsx_malloc(sizeof(sox_sample_t)*silence->start_duration);
silence->start_holdoff_offset = 0;
silence->start_holdoff_end = 0;
silence->start_found_periods = 0;
silence->stop_holdoff = lsx_malloc(sizeof(sox_sample_t)*silence->stop_duration);
silence->stop_holdoff_offset = 0;
silence->stop_holdoff_end = 0;
silence->stop_found_periods = 0;
effp->out_signal.length = SOX_UNKNOWN_LEN; /* depends on input data */
return(SOX_SUCCESS);
}
static FILE * xfopen(char const * identifier, char const * mode, lsx_io_type * io_type)
{
*io_type = lsx_io_file;
if (*identifier == '|') {
FILE * f = NULL;
#ifdef HAVE_POPEN
#ifndef POPEN_MODE
#define POPEN_MODE "r"
#endif
f = popen(identifier + 1, POPEN_MODE);
*io_type = lsx_io_pipe;
#else
lsx_fail("this build of SoX cannot open pipes");
#endif
return f;
}
else if (is_url(identifier)) {
FILE * f = NULL;
#ifdef HAVE_POPEN
char const * const command_format = "wget --no-check-certificate -q -O- \"%s\"";
char * command = lsx_malloc(strlen(command_format) + strlen(identifier));
sprintf(command, command_format, identifier);
f = popen(command, POPEN_MODE);
free(command);
*io_type = lsx_io_url;
#else
lsx_fail("this build of SoX cannot open URLs");
#endif
return f;
}
return fopen(identifier, mode);
}
static event_t event_new()
{
event_t ev = lsx_malloc(sizeof(struct event_tag));
if (!ev) return 0;
pthread_mutex_init(&ev->mutex, 0);
pthread_cond_init(&ev->cond, 0);
ev->triggered = sox_false;
return ev;
}
static int setup(sox_format_t * ft)
{
priv_t * p = (priv_t *)ft->priv;
snd_pcm_hw_params_t * params = NULL;
snd_pcm_format_mask_t * mask = NULL;
snd_pcm_uframes_t min, max;
unsigned n;
int err;
_(snd_pcm_open, (&p->pcm, ft->filename, ft->mode == 'r'? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 0));
_(snd_pcm_hw_params_malloc, (¶ms));
_(snd_pcm_hw_params_any, (p->pcm, params));
#if SND_LIB_VERSION >= 0x010009 /* Disable alsa-lib resampling: */
_(snd_pcm_hw_params_set_rate_resample, (p->pcm, params, 0));
#endif
_(snd_pcm_hw_params_set_access, (p->pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED));
_(snd_pcm_format_mask_malloc, (&mask)); /* Set format: */
snd_pcm_hw_params_get_format_mask(params, mask);
_(select_format, (&ft->encoding.encoding, &ft->encoding.bits_per_sample, mask, &p->format));
_(snd_pcm_hw_params_set_format, (p->pcm, params, p->format));
snd_pcm_format_mask_free(mask), mask = NULL;
n = ft->signal.rate; /* Set rate: */
_(snd_pcm_hw_params_set_rate_near, (p->pcm, params, &n, 0));
ft->signal.rate = n;
n = ft->signal.channels; /* Set channels: */
_(snd_pcm_hw_params_set_channels_near, (p->pcm, params, &n));
ft->signal.channels = n;
/* Set buf_len > > sox_globals.bufsiz for no underrun: */
p->buf_len = sox_globals.bufsiz * 8 / NBYTES / ft->signal.channels;
_(snd_pcm_hw_params_get_buffer_size_min, (params, &min));
_(snd_pcm_hw_params_get_buffer_size_max, (params, &max));
p->period = range_limit(p->buf_len, min, max) / 8;
p->buf_len = p->period * 8;
_(snd_pcm_hw_params_set_period_size_near, (p->pcm, params, &p->period, 0));
_(snd_pcm_hw_params_set_buffer_size_near, (p->pcm, params, &p->buf_len));
if (p->period * 2 > p->buf_len) {
lsx_fail_errno(ft, SOX_EPERM, "buffer too small");
goto error;
}
_(snd_pcm_hw_params, (p->pcm, params)); /* Configure ALSA */
snd_pcm_hw_params_free(params), params = NULL;
_(snd_pcm_prepare, (p->pcm));
p->buf_len *= ft->signal.channels; /* No longer in `frames' */
p->buf = lsx_malloc(p->buf_len * NBYTES);
return SOX_SUCCESS;
error:
if (mask) snd_pcm_format_mask_free(mask);
if (params) snd_pcm_hw_params_free(params);
return SOX_EOF;
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
static int flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp) {
priv_t * c = (priv_t *) effp->priv;
comp_band_t * l;
size_t len = min(*isamp, *osamp);
size_t band, i;
sox_sample_t *abuf, *bbuf, *cbuf, *oldabuf, *ibuf_copy;
double out;
if (c->band_buf_len < len) {
c->band_buf1 = lsx_realloc(c->band_buf1,len*sizeof(sox_sample_t));
c->band_buf2 = lsx_realloc(c->band_buf2,len*sizeof(sox_sample_t));
c->band_buf3 = lsx_realloc(c->band_buf3,len*sizeof(sox_sample_t));
c->band_buf_len = len;
}
len -= len % effp->out_signal.channels;
ibuf_copy = lsx_malloc(*isamp * sizeof(sox_sample_t));
memcpy(ibuf_copy, ibuf, *isamp * sizeof(sox_sample_t));
/* split ibuf into bands using filters, pipe each band through sox_mcompand_flow_1, then add back together and write to obuf */
memset(obuf,0,len * sizeof *obuf);
for (band=0,abuf=ibuf_copy,bbuf=c->band_buf2,cbuf=c->band_buf1;band<c->nBands;++band) {
l = &c->bands[band];
if (l->topfreq)
crossover_flow(effp, &l->filter, abuf, bbuf, cbuf, len);
else {
bbuf = abuf;
abuf = cbuf;
}
if (abuf == ibuf_copy)
abuf = c->band_buf3;
(void)sox_mcompand_flow_1(effp, c,l,bbuf,abuf,len, (size_t)effp->out_signal.channels);
for (i=0;i<len;++i)
{
out = obuf[i] + abuf[i];
SOX_SAMPLE_CLIP_COUNT(out, effp->clips);
obuf[i] = out;
}
oldabuf = abuf;
abuf = cbuf;
cbuf = oldabuf;
}
*isamp = *osamp = len;
free(ibuf_copy);
return SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
size_t max_delay;
if (!p->max_arg)
return SOX_EFF_NULL;
if (effp->flow < p->argc)
lsx_parsesamples(effp->in_signal.rate, p->argv[effp->flow], &p->buffer_size, 't');
lsx_parsesamples(effp->in_signal.rate, p->max_arg, &max_delay, 't');
p->buffer_index = p->delay = 0;
p->pad = max_delay - p->buffer_size;
p->buffer = lsx_malloc(p->buffer_size * sizeof(*p->buffer));
return SOX_SUCCESS;
}
void sox_append_comments(sox_comments_t * comments, char const * comment)
{
char * end;
if (comment) {
while ((end = strchr(comment, '\n'))) {
size_t len = end - comment;
char * c = lsx_malloc((len + 1) * sizeof(*c));
strncpy(c, comment, len);
c[len] = '\0';
sox_append_comment(comments, c);
comment += len + 1;
free(c);
}
if (*comment)
sox_append_comment(comments, comment);
}
}
int lsx_enum_option(int c, lsx_enum_item const * items)
{
lsx_enum_item const * p = lsx_find_enum_text(lsx_optarg, items, sox_false);
if (p == NULL) {
size_t len = 1;
char * set = lsx_malloc(len);
*set = 0;
for (p = items; p->text; ++p) {
set = lsx_realloc(set, len += 2 + strlen(p->text));
strcat(set, ", "); strcat(set, p->text);
}
lsx_fail("-%c: `%s' is not one of: %s.", c, lsx_optarg, set + 2);
free(set);
return INT_MAX;
}
return p->value;
}
static void start(sox_format_t * ft)
{
priv_t * sf = (priv_t *)ft->priv;
int subtype = ft_enc(ft->encoding.bits_per_sample? ft->encoding.bits_per_sample : ft->signal.precision, ft->encoding.encoding);
sf->log_buffer_ptr = sf->log_buffer = lsx_malloc((size_t)LOG_MAX);
sf->sf_info = lsx_calloc(1, sizeof(SF_INFO));
/* Copy format info */
if (subtype) {
if (strcmp(ft->filetype, "sndfile") == 0)
sf->sf_info->format = name_to_format(ft->filename) | subtype;
else
sf->sf_info->format = name_to_format(ft->filetype) | subtype;
}
sf->sf_info->samplerate = ft->signal.rate;
sf->sf_info->channels = ft->signal.channels;
if (ft->signal.channels)
sf->sf_info->frames = ft->signal.length / ft->signal.channels;
}
static int open_audio(priv_t * ffmpeg, AVStream *st)
{
AVCodecContext *c;
AVCodec *codec;
c = st->codec;
/* find the audio encoder */
codec = avcodec_find_encoder(c->codec_id);
if (!codec) {
lsx_fail("ffmpeg CODEC not found");
return SOX_EOF;
}
/* open it */
if (avcodec_open(c, codec) < 0) {
lsx_fail("ffmpeg could not open CODEC");
return SOX_EOF;
}
ffmpeg->audio_buf_raw = lsx_malloc((size_t)AVCODEC_MAX_AUDIO_FRAME_SIZE + 32);
ffmpeg->audio_buf_aligned = ALIGN16(ffmpeg->audio_buf_raw);
/* ugly hack for PCM codecs (will be removed ASAP with new PCM
support to compute the input frame size in samples */
if (c->frame_size <= 1) {
ffmpeg->audio_input_frame_size = AVCODEC_MAX_AUDIO_FRAME_SIZE / c->channels;
switch(st->codec->codec_id) {
case CODEC_ID_PCM_S16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_U16BE:
ffmpeg->audio_input_frame_size >>= 1;
break;
default:
break;
}
} else
static int startread(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
size_t ReadSize;
sox_bool ignore_length = ft->signal.length == SOX_IGNORE_LENGTH;
int open_library_result;
LSX_DLLIBRARY_OPEN(
p,
mad_dl,
MAD_FUNC_ENTRIES,
"MAD decoder library",
mad_library_names,
open_library_result);
if (open_library_result)
return SOX_EOF;
p->mp3_buffer_size = sox_globals.bufsiz;
p->mp3_buffer = lsx_malloc(p->mp3_buffer_size);
ft->signal.length = SOX_UNSPEC;
if (ft->seekable) {
#ifdef USING_ID3TAG
read_comments(ft);
rewind((FILE*)ft->fp);
if (!ft->signal.length)
#endif
if (!ignore_length)
ft->signal.length = mp3_duration_ms(ft);
}
p->mad_stream_init(&p->Stream);
p->mad_frame_init(&p->Frame);
p->mad_synth_init(&p->Synth);
mad_timer_reset(&p->Timer);
ft->encoding.encoding = SOX_ENCODING_MP3;
/* Decode at least one valid frame to find out the input
* format. The decoded frame will be saved off so that it
* can be processed later.
*/
ReadSize = lsx_readbuf(ft, p->mp3_buffer, p->mp3_buffer_size);
if (ReadSize != p->mp3_buffer_size && ferror((FILE*)ft->fp))
return SOX_EOF;
p->mad_stream_buffer(&p->Stream, p->mp3_buffer, ReadSize);
/* Find a valid frame before starting up. This makes sure
* that we have a valid MP3 and also skips past ID3v2 tags
* at the beginning of the audio file.
*/
p->Stream.error = 0;
while (p->mad_frame_decode(&p->Frame,&p->Stream))
{
/* check whether input buffer needs a refill */
if (p->Stream.error == MAD_ERROR_BUFLEN)
{
if (sox_mp3_input(ft) == SOX_EOF)
return SOX_EOF;
continue;
}
/* Consume any ID3 tags */
sox_mp3_inputtag(ft);
/* FIXME: We should probably detect when we've read
* a bunch of non-ID3 data and still haven't found a
* frame. In that case we can abort early without
* scanning the whole file.
*/
p->Stream.error = 0;
}
if (p->Stream.error)
{
lsx_fail_errno(ft,SOX_EOF,"No valid MP3 frame found");
return SOX_EOF;
}
switch(p->Frame.header.mode)
{
case MAD_MODE_SINGLE_CHANNEL:
case MAD_MODE_DUAL_CHANNEL:
case MAD_MODE_JOINT_STEREO:
case MAD_MODE_STEREO:
ft->signal.channels = MAD_NCHANNELS(&p->Frame.header);
break;
default:
lsx_fail_errno(ft, SOX_EFMT, "Cannot determine number of channels");
return SOX_EOF;
}
p->FrameCount=1;
p->mad_timer_add(&p->Timer,p->Frame.header.duration);
p->mad_synth_frame(&p->Synth,&p->Frame);
ft->signal.precision = MP3_MAD_PRECISION;
ft->signal.rate=p->Synth.pcm.samplerate;
if (ignore_length)
ft->signal.length = SOX_UNSPEC;
else {
ft->signal.length = (uint64_t)(ft->signal.length * .001 * ft->signal.rate + .5);
ft->signal.length *= ft->signal.channels; /* Keep separate from line above! */
}
p->cursamp = 0;
return SOX_SUCCESS;
}
static int sox_fade_getopts(sox_effect_t * effp, int argc, char **argv)
{
priv_t * fade = (priv_t *) effp->priv;
char t_char[2];
int t_argno;
--argc, ++argv;
if (argc < 1 || argc > 4)
return lsx_usage(effp);
/* because sample rate is unavailable at this point we store the
* string off for later computations.
*/
if (sscanf(argv[0], "%1[qhltp]", t_char))
{
fade->in_fadetype = *t_char;
fade->out_fadetype = *t_char;
argv++;
argc--;
}
else
{
/* No type given. */
fade->in_fadetype = 'l';
fade->out_fadetype = 'l';
}
fade->in_stop_str = lsx_malloc(strlen(argv[0])+1);
strcpy(fade->in_stop_str,argv[0]);
/* Do a dummy parse to see if it will fail */
if (lsx_parsesamples(0., fade->in_stop_str, &fade->in_stop, 't') == NULL)
return lsx_usage(effp);
fade->out_start_str = fade->out_stop_str = 0;
for (t_argno = 1; t_argno < argc && t_argno < 3; t_argno++)
{
/* See if there is fade-in/fade-out times/curves specified. */
if(t_argno == 1)
{
fade->out_stop_str = lsx_malloc(strlen(argv[t_argno])+1);
strcpy(fade->out_stop_str,argv[t_argno]);
/* Do a dummy parse to see if it will fail */
if (lsx_parsesamples(0., fade->out_stop_str,
&fade->out_stop, 't') == NULL)
return lsx_usage(effp);
}
else
{
fade->out_start_str = lsx_malloc(strlen(argv[t_argno])+1);
strcpy(fade->out_start_str,argv[t_argno]);
/* Do a dummy parse to see if it will fail */
if (lsx_parsesamples(0., fade->out_start_str,
&fade->out_start, 't') == NULL)
return lsx_usage(effp);
}
} /* End for(t_argno) */
return(SOX_SUCCESS);
}
/*
* Prepare for processing.
*/
static int sox_echo_start(sox_effect_t * effp)
{
priv_t * echo = (priv_t *) effp->priv;
int i;
float sum_in_volume;
long j;
echo->maxsamples = 0;
if ( echo->in_gain < 0.0 )
{
lsx_fail("echo: gain-in must be positive!");
return (SOX_EOF);
}
if ( echo->in_gain > 1.0 )
{
lsx_fail("echo: gain-in must be less than 1.0!");
return (SOX_EOF);
}
if ( echo->out_gain < 0.0 )
{
lsx_fail("echo: gain-in must be positive!");
return (SOX_EOF);
}
for ( i = 0; i < echo->num_delays; i++ ) {
echo->samples[i] = echo->delay[i] * effp->in_signal.rate / 1000.0;
if ( echo->samples[i] < 1 )
{
lsx_fail("echo: delay must be positive!");
return (SOX_EOF);
}
if ( echo->samples[i] > (ptrdiff_t)DELAY_BUFSIZ )
{
lsx_fail("echo: delay must be less than %g seconds!",
DELAY_BUFSIZ / effp->in_signal.rate );
return (SOX_EOF);
}
if ( echo->decay[i] < 0.0 )
{
lsx_fail("echo: decay must be positive!" );
return (SOX_EOF);
}
if ( echo->decay[i] > 1.0 )
{
lsx_fail("echo: decay must be less than 1.0!" );
return (SOX_EOF);
}
if ( echo->samples[i] > echo->maxsamples )
echo->maxsamples = echo->samples[i];
}
echo->delay_buf = lsx_malloc(sizeof (double) * echo->maxsamples);
for ( j = 0; j < echo->maxsamples; ++j )
echo->delay_buf[j] = 0.0;
/* Be nice and check the hint with warning, if... */
sum_in_volume = 1.0;
for ( i = 0; i < echo->num_delays; i++ )
sum_in_volume += echo->decay[i];
if ( sum_in_volume * echo->in_gain > 1.0 / echo->out_gain )
lsx_warn("echo: warning >>> gain-out can cause saturation of output <<<");
echo->counter = 0;
echo->fade_out = echo->maxsamples;
effp->out_signal.length = SOX_UNKNOWN_LEN; /* TODO: calculate actual length */
return (SOX_SUCCESS);
}
/*
* Prepare processing.
* Do all initializations.
*/
static int start(sox_effect_t* effp)
{
priv_t* p = (priv_t*)effp->priv;
int result = SOX_SUCCESS;
spx_int32_t int_val;
float float_val;
if (p->samples_per_frame)
{
p->buffer_end = p->samples_per_frame;
}
else
{
p->buffer_end = effp->in_signal.rate / p->frames_per_second;
if (!p->buffer_end)
{
lsx_fail("frames_per_second too large for the current sample rate.");
return SOX_EOF;
}
}
p->buffer_opos = p->buffer_end;
effp->out_signal.precision = 16;
p->buffer = lsx_malloc(p->buffer_end * sizeof(p->buffer[0]));
if (!p->buffer)
{
result = SOX_ENOMEM;
goto Done;
}
p->sps = speex_preprocess_state_init(p->buffer_end, effp->in_signal.rate);
if (!p->sps)
{
lsx_fail("Failed to initialize preprocessor DSP.");
result = SOX_EOF;
goto Done;
}
int_val = p->agc ? 1 : 2;
speex_preprocess_ctl(p->sps, SPEEX_PREPROCESS_SET_AGC, &int_val);
if (p->agc)
{
float_val = p->agc * 327.68f;
speex_preprocess_ctl(p->sps, SPEEX_PREPROCESS_SET_AGC_LEVEL, &float_val);
}
int_val = p->denoise ? 1 : 2;
speex_preprocess_ctl(p->sps, SPEEX_PREPROCESS_SET_DENOISE, &int_val);
if (p->denoise)
{
int_val = -(spx_int32_t)p->denoise;
speex_preprocess_ctl(p->sps, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &int_val);
}
int_val = p->dereverb ? 1 : 2;
speex_preprocess_ctl(p->sps, SPEEX_PREPROCESS_SET_DEREVERB, &int_val);
Done:
if (result != SOX_SUCCESS)
stop(effp);
return result;
}
static int start_read(sox_format_t * ft)
{
unsigned long header_size_ul = 0, num_samples_ul = 0;
sox_encoding_t encoding = SOX_ENCODING_SIGN2;
size_t header_size, bytes_read;
size_t num_samples = 0;
unsigned bytes_per_sample = 0;
unsigned channels = 1;
unsigned rate = 16000;
char fldname[64], fldtype[16], fldsval[128];
char * buf;
/* Magic header */
if (lsx_reads(ft, fldname, (size_t)8) || strncmp(fldname, "NIST_1A", (size_t)7) != 0) {
lsx_fail_errno(ft, SOX_EHDR, "Sphere header does not begin with magic word `NIST_1A'");
return (SOX_EOF);
}
if (lsx_reads(ft, fldsval, (size_t)8)) {
lsx_fail_errno(ft, SOX_EHDR, "Error reading Sphere header");
return (SOX_EOF);
}
/* Determine header size, and allocate a buffer large enough to hold it. */
sscanf(fldsval, "%lu", &header_size_ul);
buf = lsx_malloc(header_size = header_size_ul);
/* Skip what we have read so far */
header_size -= 16;
if (lsx_reads(ft, buf, header_size) == SOX_EOF) {
lsx_fail_errno(ft, SOX_EHDR, "Error reading Sphere header");
free(buf);
return (SOX_EOF);
}
header_size -= (strlen(buf) + 1);
while (strncmp(buf, "end_head", (size_t)8) != 0) {
if (strncmp(buf, "sample_n_bytes", (size_t)14) == 0)
sscanf(buf, "%63s %15s %u", fldname, fldtype, &bytes_per_sample);
else if (strncmp(buf, "channel_count", (size_t)13) == 0)
sscanf(buf, "%63s %15s %u", fldname, fldtype, &channels);
else if (strncmp(buf, "sample_count ", (size_t)13) == 0)
sscanf(buf, "%53s %15s %lu", fldname, fldtype, &num_samples_ul);
else if (strncmp(buf, "sample_rate ", (size_t)12) == 0)
sscanf(buf, "%53s %15s %u", fldname, fldtype, &rate);
else if (strncmp(buf, "sample_coding", (size_t)13) == 0) {
sscanf(buf, "%63s %15s %127s", fldname, fldtype, fldsval);
if (!strcasecmp(fldsval, "ulaw") || !strcasecmp(fldsval, "mu-law"))
encoding = SOX_ENCODING_ULAW;
else if (!strcasecmp(fldsval, "pcm"))
encoding = SOX_ENCODING_SIGN2;
else {
lsx_fail_errno(ft, SOX_EFMT, "sph: unsupported coding `%s'", fldsval);
free(buf);
return SOX_EOF;
}
}
else if (strncmp(buf, "sample_byte_format", (size_t)18) == 0) {
sscanf(buf, "%53s %15s %127s", fldname, fldtype, fldsval);
if (strcmp(fldsval, "01") == 0) /* Data is little endian. */
ft->encoding.reverse_bytes = MACHINE_IS_BIGENDIAN;
else if (strcmp(fldsval, "10") == 0) /* Data is big endian. */
ft->encoding.reverse_bytes = MACHINE_IS_LITTLEENDIAN;
else if (strcmp(fldsval, "1")) {
lsx_fail_errno(ft, SOX_EFMT, "sph: unsupported coding `%s'", fldsval);
free(buf);
return SOX_EOF;
}
}
if (lsx_reads(ft, buf, header_size) == SOX_EOF) {
lsx_fail_errno(ft, SOX_EHDR, "Error reading Sphere header");
free(buf);
return (SOX_EOF);
}
header_size -= (strlen(buf) + 1);
}
if (!bytes_per_sample)
bytes_per_sample = encoding == SOX_ENCODING_ULAW? 1 : 2;
while (header_size) {
bytes_read = lsx_readbuf(ft, buf, header_size);
if (bytes_read == 0) {
free(buf);
return (SOX_EOF);
}
header_size -= bytes_read;
}
free(buf);
if (ft->seekable) {
/* Check first four bytes of data to see if it's shorten compressed. */
char shorten_check[4];
if (lsx_readchars(ft, shorten_check, sizeof(shorten_check)))
return SOX_EOF;
lsx_seeki(ft, -(off_t)sizeof(shorten_check), SEEK_CUR);
if (!memcmp(shorten_check, "ajkg", sizeof(shorten_check))) {
lsx_fail_errno(ft, SOX_EFMT,
"File uses shorten compression, cannot handle this.");
return (SOX_EOF);
}
}
num_samples = num_samples_ul;
return lsx_check_read_params(ft, channels, (sox_rate_t)rate, encoding,
bytes_per_sample << 3, (uint64_t)num_samples * channels, sox_true);
}
static int start_write(sox_format_t * const ft)
{
priv_t * p = (priv_t *)ft->priv;
FLAC__StreamEncoderState status;
unsigned compression_level = MAX_COMPRESSION; /* Default to "best" */
if (ft->encoding.compression != HUGE_VAL) {
compression_level = ft->encoding.compression;
if (compression_level != ft->encoding.compression ||
compression_level > MAX_COMPRESSION) {
lsx_fail_errno(ft, SOX_EINVAL,
"FLAC compression level must be a whole number from 0 to %i",
MAX_COMPRESSION);
return SOX_EOF;
}
}
p->encoder = FLAC__stream_encoder_new();
if (p->encoder == NULL) {
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder instance");
return SOX_EOF;
}
p->decoded_samples = lsx_malloc(sox_globals.bufsiz * sizeof(FLAC__int32));
p->bits_per_sample = ft->encoding.bits_per_sample;
lsx_report("encoding at %i bits per sample", p->bits_per_sample);
FLAC__stream_encoder_set_channels(p->encoder, ft->signal.channels);
FLAC__stream_encoder_set_bits_per_sample(p->encoder, p->bits_per_sample);
FLAC__stream_encoder_set_sample_rate(p->encoder, (unsigned)(ft->signal.rate + .5));
{ /* Check if rate is streamable: */
static const unsigned streamable_rates[] =
{8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000};
size_t i;
sox_bool streamable = sox_false;
for (i = 0; !streamable && i < array_length(streamable_rates); ++i)
streamable = (streamable_rates[i] == ft->signal.rate);
if (!streamable) {
lsx_report("non-standard rate; output may not be streamable");
FLAC__stream_encoder_set_streamable_subset(p->encoder, sox_false);
}
}
#if FLAC_API_VERSION_CURRENT >= 10
FLAC__stream_encoder_set_compression_level(p->encoder, compression_level);
#else
{
static struct {
unsigned blocksize;
FLAC__bool do_exhaustive_model_search;
FLAC__bool do_mid_side_stereo;
FLAC__bool loose_mid_side_stereo;
unsigned max_lpc_order;
unsigned max_residual_partition_order;
unsigned min_residual_partition_order;
} const options[MAX_COMPRESSION + 1] = {
{1152, sox_false, sox_false, sox_false, 0, 2, 2},
{1152, sox_false, sox_true, sox_true, 0, 2, 2},
{1152, sox_false, sox_true, sox_false, 0, 3, 0},
{4608, sox_false, sox_false, sox_false, 6, 3, 3},
{4608, sox_false, sox_true, sox_true, 8, 3, 3},
{4608, sox_false, sox_true, sox_false, 8, 3, 3},
{4608, sox_false, sox_true, sox_false, 8, 4, 0},
{4608, sox_true, sox_true, sox_false, 8, 6, 0},
{4608, sox_true, sox_true, sox_false, 12, 6, 0},
};
#define SET_OPTION(x) do {\
lsx_report(#x" = %i", options[compression_level].x); \
FLAC__stream_encoder_set_##x(p->encoder, options[compression_level].x);\
} while (0)
SET_OPTION(blocksize);
SET_OPTION(do_exhaustive_model_search);
SET_OPTION(max_lpc_order);
SET_OPTION(max_residual_partition_order);
SET_OPTION(min_residual_partition_order);
if (ft->signal.channels == 2) {
SET_OPTION(do_mid_side_stereo);
SET_OPTION(loose_mid_side_stereo);
}
#undef SET_OPTION
}
#endif
if (ft->signal.length != 0) {
FLAC__stream_encoder_set_total_samples_estimate(p->encoder, (FLAC__uint64)(ft->signal.length / ft->signal.channels));
p->metadata[p->num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_SEEKTABLE);
if (p->metadata[p->num_metadata] == NULL) {
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder seek table template");
return SOX_EOF;
}
{
#if FLAC_API_VERSION_CURRENT >= 8
if (!FLAC__metadata_object_seektable_template_append_spaced_points_by_samples(p->metadata[p->num_metadata], (unsigned)(10 * ft->signal.rate + .5), (FLAC__uint64)(ft->signal.length/ft->signal.channels))) {
#else
size_t samples = 10 * ft->signal.rate;
size_t total_samples = ft->signal.length/ft->signal.channels;
if (!FLAC__metadata_object_seektable_template_append_spaced_points(p->metadata[p->num_metadata], total_samples / samples + (total_samples % samples != 0), (FLAC__uint64)total_samples)) {
#endif
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder seek table points");
return SOX_EOF;
}
}
p->metadata[p->num_metadata]->is_last = sox_false; /* the encoder will set this for us */
++p->num_metadata;
}
if (ft->oob.comments) { /* Make the comment structure */
FLAC__StreamMetadata_VorbisComment_Entry entry;
int i;
p->metadata[p->num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT);
for (i = 0; ft->oob.comments[i]; ++i) {
static const char prepend[] = "Comment=";
char * text = lsx_calloc(strlen(prepend) + strlen(ft->oob.comments[i]) + 1, sizeof(*text));
/* Prepend `Comment=' if no field-name already in the comment */
if (!strchr(ft->oob.comments[i], '='))
strcpy(text, prepend);
entry.entry = (FLAC__byte *) strcat(text, ft->oob.comments[i]);
entry.length = strlen(text);
FLAC__metadata_object_vorbiscomment_append_comment(p->metadata[p->num_metadata], entry, /*copy= */ sox_true);
free(text);
}
++p->num_metadata;
}
if (p->num_metadata)
FLAC__stream_encoder_set_metadata(p->encoder, p->metadata, p->num_metadata);
#if FLAC_API_VERSION_CURRENT <= 7
FLAC__stream_encoder_set_write_callback(p->encoder, flac_stream_encoder_write_callback);
FLAC__stream_encoder_set_metadata_callback(p->encoder, flac_stream_encoder_metadata_callback);
FLAC__stream_encoder_set_client_data(p->encoder, ft);
status = FLAC__stream_encoder_init(p->encoder);
#else
status = FLAC__stream_encoder_init_stream(p->encoder, flac_stream_encoder_write_callback,
flac_stream_encoder_seek_callback, flac_stream_encoder_tell_callback, flac_stream_encoder_metadata_callback, ft);
#endif
if (status != FLAC__STREAM_ENCODER_OK) {
lsx_fail_errno(ft, SOX_EINVAL, "%s", FLAC__StreamEncoderStateString[status]);
return SOX_EOF;
}
return SOX_SUCCESS;
}
static size_t write_samples(sox_format_t * const ft, sox_sample_t const * const sampleBuffer, size_t const len)
{
priv_t * p = (priv_t *)ft->priv;
unsigned i;
for (i = 0; i < len; ++i) {
long pcm = SOX_SAMPLE_TO_SIGNED_32BIT(sampleBuffer[i], ft->clips);
p->decoded_samples[i] = pcm >> (32 - p->bits_per_sample);
switch (p->bits_per_sample) {
case 8: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_8BIT(sampleBuffer[i], ft->clips);
break;
case 16: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_16BIT(sampleBuffer[i], ft->clips);
break;
case 24: p->decoded_samples[i] = /* sign extension: */
SOX_SAMPLE_TO_SIGNED_24BIT(sampleBuffer[i],ft->clips) << 8;
p->decoded_samples[i] >>= 8;
break;
case 32: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_32BIT(sampleBuffer[i],ft->clips);
break;
}
}
FLAC__stream_encoder_process_interleaved(p->encoder, p->decoded_samples, (unsigned) len / ft->signal.channels);
return FLAC__stream_encoder_get_state(p->encoder) == FLAC__STREAM_ENCODER_OK ? len : 0;
}
/*
* Process either isamp or osamp samples, whichever is smaller.
*/
static int sox_pan_flow(sox_effect_t * effp, const sox_sample_t *ibuf, sox_sample_t *obuf,
size_t *isamp, size_t *osamp)
{
priv_t * pan = (priv_t *) effp->priv;
size_t len, done;
sox_sample_t *ibuf_copy;
char ich, och;
double left, right, direction, hdir;
ibuf_copy = lsx_malloc(*isamp * sizeof(sox_sample_t));
memcpy(ibuf_copy, ibuf, *isamp * sizeof(sox_sample_t));
direction = pan->direction; /* -1 <= direction <= 1 */
hdir = 0.5 * direction; /* -0.5 <= hdir <= 0.5 */
left = 0.5 - hdir; /* 0 <= left <= 1 */
right = 0.5 + hdir; /* 0 <= right <= 1 */
ich = effp->in_signal.channels;
och = effp->out_signal.channels;
len = min(*osamp/och,*isamp/ich);
/* report back how much is processed. */
*isamp = len*ich;
*osamp = len*och;
/* 9 different cases to handle: (1,2,4) X (1,2,4) */
switch (och) {
case 1: /* pan on mono channel... not much sense. just avg. */
switch (ich) {
case 1: /* simple copy */
for (done=0; done<len; done++)
*obuf++ = *ibuf_copy++;
break;
case 2: /* average 2 */
for (done=0; done<len; done++)
{
double f;
f = 0.5*ibuf_copy[0] + 0.5*ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
*obuf++ = f;
ibuf_copy += 2;
}
break;
case 4: /* average 4 */
for (done=0; done<len; done++)
{
double f;
f = 0.25*ibuf_copy[0] + 0.25*ibuf_copy[1] +
0.25*ibuf_copy[2] + 0.25*ibuf_copy[3];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
*obuf++ = f;
ibuf_copy += 4;
}
break;
default:
UNEXPECTED_CHANNELS;
break;
} /* end first switch in channel */
break;
case 2:
switch (ich) {
case 1: /* linear */
for (done=0; done<len; done++)
{
double f;
f = left * ibuf_copy[0];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[0] = f;
f = right * ibuf_copy[0];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[1] = f;
obuf += 2;
ibuf_copy++;
}
break;
case 2: /* linear panorama.
* I'm not sure this is the right way to do it.
*/
if (direction <= 0.0) /* to the left */
{
register double volume, cll, clr, cr;
volume = 1.0 - 0.5*direction;
cll = volume*(1.5-left);
clr = volume*(left-0.5);
cr = volume*(1.0+direction);
for (done=0; done<len; done++)
{
double f;
f = cll * ibuf_copy[0] + clr * ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[0] = f;
f = cr * ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[1] = f;
obuf += 2;
ibuf_copy += 2;
}
}
else /* to the right */
{
register double volume, cl, crl, crr;
volume = 1.0 + 0.5*direction;
cl = volume*(1.0-direction);
crl = volume*(right-0.5);
crr = volume*(1.5-right);
for (done=0; done<len; done++)
{
double f;
f = cl * ibuf_copy[0];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[0] = f;
f = crl * ibuf_copy[0] + crr * ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[1] = f;
obuf += 2;
ibuf_copy += 2;
}
}
break;
case 4:
if (direction <= 0.0) /* to the left */
{
register double volume, cll, clr, cr;
volume = 1.0 - 0.5*direction;
cll = volume*(1.5-left);
clr = volume*(left-0.5);
cr = volume*(1.0+direction);
for (done=0; done<len; done++)
{
register double ibuf0, ibuf1, f;
/* build stereo signal */
ibuf0 = 0.5*ibuf_copy[0] + 0.5*ibuf_copy[2];
ibuf1 = 0.5*ibuf_copy[1] + 0.5*ibuf_copy[3];
/* pan it */
f = cll * ibuf0 + clr * ibuf1;
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[0] = f;
f = cr * ibuf1;
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[1] = f;
obuf += 2;
ibuf_copy += 4;
}
}
else /* to the right */
{
register double volume, cl, crl, crr;
volume = 1.0 + 0.5*direction;
cl = volume*(1.0-direction);
crl = volume*(right-0.5);
crr = volume*(1.5-right);
for (done=0; done<len; done++)
{
register double ibuf0, ibuf1, f;
ibuf0 = 0.5*ibuf_copy[0] + 0.5*ibuf_copy[2];
ibuf1 = 0.5*ibuf_copy[1] + 0.5*ibuf_copy[3];
f = cl * ibuf0;
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[0] = f;
f = crl * ibuf0 + crr * ibuf1;
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[1] = f;
obuf += 2;
ibuf_copy += 4;
}
}
break;
default:
UNEXPECTED_CHANNELS;
break;
} /* end second switch in channel */
break;
case 4:
switch (ich) {
case 1: /* linear */
{
register double cr, cl;
cl = 0.5*left;
cr = 0.5*right;
for (done=0; done<len; done++)
{
double f;
f = cl * ibuf_copy[0];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[2] = obuf[0] = f;
f = cr * ibuf_copy[0];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
ibuf_copy[3] = obuf[1] = f;
obuf += 4;
ibuf_copy++;
}
}
break;
case 2: /* simple linear panorama */
if (direction <= 0.0) /* to the left */
{
register double volume, cll, clr, cr;
volume = 0.5 - 0.25*direction;
cll = volume * (1.5-left);
clr = volume * (left-0.5);
cr = volume * (1.0+direction);
for (done=0; done<len; done++)
{
double f;
f = cll * ibuf_copy[0] + clr * ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[2] = obuf[0] = f;
f = cr * ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
ibuf_copy[3] = obuf[1] = f;
obuf += 4;
ibuf_copy += 2;
}
}
else /* to the right */
{
register double volume, cl, crl, crr;
volume = 0.5 + 0.25*direction;
cl = volume * (1.0-direction);
crl = volume * (right-0.5);
crr = volume * (1.5-right);
for (done=0; done<len; done++)
{
double f;
f = cl * ibuf_copy[0];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[2] = obuf[0] =f ;
f = crl * ibuf_copy[0] + crr * ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
ibuf_copy[3] = obuf[1] = f;
obuf += 4;
ibuf_copy += 2;
}
}
break;
case 4:
/* maybe I could improve the formula to reverse...
also, turn only by quarters.
*/
if (direction <= 0.0) /* to the left */
{
register double cown, cright;
cright = -direction;
cown = 1.0 + direction;
for (done=0; done<len; done++)
{
double f;
f = cown*ibuf_copy[0] + cright*ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[0] = f;
f = cown*ibuf_copy[1] + cright*ibuf_copy[3];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[1] = f;
f = cown*ibuf_copy[2] + cright*ibuf_copy[0];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[2] = f;
f = cown*ibuf_copy[3] + cright*ibuf_copy[2];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[3] = f;
obuf += 4;
ibuf_copy += 4;
}
}
else /* to the right */
{
register double cleft, cown;
cleft = direction;
cown = 1.0 - direction;
for (done=0; done<len; done++)
{
double f;
f = cleft*ibuf_copy[2] + cown*ibuf_copy[0];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[0] = f;
f = cleft*ibuf_copy[0] + cown*ibuf_copy[1];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[1] = f;
f = cleft*ibuf_copy[3] + cown*ibuf_copy[2];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[2] = f;
f = cleft*ibuf_copy[1] + cown*ibuf_copy[3];
SOX_SAMPLE_CLIP_COUNT(f, effp->clips);
obuf[3] = f;
obuf += 4;
ibuf_copy += 4;
}
}
break;
default:
UNEXPECTED_CHANNELS;
break;
} /* end third switch in channel */
break;
default:
UNEXPECTED_CHANNELS;
break;
} /* end switch out channel */
free(ibuf_copy - len * ich);
return SOX_SUCCESS;
}
/* common r/w initialization code */
static int ossinit(sox_format_t * ft)
{
int sampletype, samplesize, dsp_stereo;
int tmp, rc;
priv_t *file = (priv_t *)ft->priv;
if (ft->encoding.bits_per_sample == 8) {
sampletype = AFMT_U8;
samplesize = 8;
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN)
ft->encoding.encoding = SOX_ENCODING_UNSIGNED;
if (ft->encoding.encoding != SOX_ENCODING_UNSIGNED) {
lsx_report("OSS driver only supports unsigned with bytes");
lsx_report("Forcing to unsigned");
ft->encoding.encoding = SOX_ENCODING_UNSIGNED;
}
}
else if (ft->encoding.bits_per_sample == 16) {
/* Attempt to use endian that user specified */
if (ft->encoding.reverse_bytes)
sampletype = (MACHINE_IS_BIGENDIAN) ? AFMT_S16_LE : AFMT_S16_BE;
else
sampletype = (MACHINE_IS_BIGENDIAN) ? AFMT_S16_BE : AFMT_S16_LE;
samplesize = 16;
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN)
ft->encoding.encoding = SOX_ENCODING_SIGN2;
if (ft->encoding.encoding != SOX_ENCODING_SIGN2) {
lsx_report("OSS driver only supports signed with words");
lsx_report("Forcing to signed linear");
ft->encoding.encoding = SOX_ENCODING_SIGN2;
}
}
else {
/* Attempt to use endian that user specified */
if (ft->encoding.reverse_bytes)
sampletype = (MACHINE_IS_BIGENDIAN) ? AFMT_S16_LE : AFMT_S16_BE;
else
sampletype = (MACHINE_IS_BIGENDIAN) ? AFMT_S16_BE : AFMT_S16_LE;
samplesize = 16;
ft->encoding.bits_per_sample = 16;
ft->encoding.encoding = SOX_ENCODING_SIGN2;
lsx_report("OSS driver only supports bytes and words");
lsx_report("Forcing to signed linear word");
}
if (ft->signal.channels > 2) ft->signal.channels = 2;
if (ioctl(fileno(ft->fp), (size_t) SNDCTL_DSP_RESET, 0) < 0)
{
lsx_fail_errno(ft,SOX_EOF,"Unable to reset OSS driver. Possibly accessing an invalid file/device");
return(SOX_EOF);
}
/* Query the supported formats and find the best match
*/
rc = ioctl(fileno(ft->fp), SNDCTL_DSP_GETFMTS, &tmp);
if (rc == 0) {
if ((tmp & sampletype) == 0)
{
/* is 16-bit supported? */
if (samplesize == 16 && (tmp & (AFMT_S16_LE|AFMT_S16_BE)) == 0)
{
/* Must not like 16-bits, try 8-bits */
ft->encoding.bits_per_sample = 8;
ft->encoding.encoding = SOX_ENCODING_UNSIGNED;
lsx_report("OSS driver doesn't like signed words");
lsx_report("Forcing to unsigned bytes");
tmp = sampletype = AFMT_U8;
samplesize = 8;
}
/* is 8-bit supported */
else if (samplesize == 8 && (tmp & AFMT_U8) == 0)
{
ft->encoding.bits_per_sample = 16;
ft->encoding.encoding = SOX_ENCODING_SIGN2;
lsx_report("OSS driver doesn't like unsigned bytes");
lsx_report("Forcing to signed words");
sampletype = (MACHINE_IS_BIGENDIAN) ? AFMT_S16_BE : AFMT_S16_LE;
samplesize = 16;
}
/* determine which 16-bit format to use */
if (samplesize == 16 && (tmp & sampletype) == 0)
{
/* Either user requested something not supported
* or hardware doesn't support machine endian.
* Force to opposite as the above test showed
* it supports at least one of the two endians.
*/
sampletype = (sampletype == AFMT_S16_BE) ? AFMT_S16_LE : AFMT_S16_BE;
ft->encoding.reverse_bytes = !ft->encoding.reverse_bytes;
}
}
tmp = sampletype;
rc = ioctl(fileno(ft->fp), SNDCTL_DSP_SETFMT, &tmp);
}
/* Give up and exit */
if (rc < 0 || tmp != sampletype)
{
lsx_fail_errno(ft,SOX_EOF,"Unable to set the sample size to %d", samplesize);
return (SOX_EOF);
}
if (ft->signal.channels == 2) dsp_stereo = 1;
else dsp_stereo = 0;
tmp = dsp_stereo;
if (ioctl(fileno(ft->fp), SNDCTL_DSP_STEREO, &tmp) < 0)
{
lsx_warn("Couldn't set to %s", dsp_stereo? "stereo":"mono");
dsp_stereo = 0;
}
if (tmp != dsp_stereo)
ft->signal.channels = tmp + 1;
tmp = ft->signal.rate;
if (ioctl (fileno(ft->fp), SNDCTL_DSP_SPEED, &tmp) < 0 ||
(int)ft->signal.rate != tmp) {
/* If the rate the sound card is using is not within 1% of what
* the user specified then override the user setting.
* The only reason not to always override this is because of
* clock-rounding problems. Sound cards will sometimes use
* things like 44101 when you ask for 44100. No need overriding
* this and having strange output file rates for something that
* we can't hear anyways.
*/
if ((int)ft->signal.rate - tmp > (tmp * .01) ||
tmp - (int)ft->signal.rate > (tmp * .01))
ft->signal.rate = tmp;
}
/* Find out block size to use last because the driver could compute
* its size based on specific rates/formats.
*/
file->size = 0;
ioctl (fileno(ft->fp), SNDCTL_DSP_GETBLKSIZE, &file->size);
if (file->size < 4 || file->size > 65536) {
lsx_fail_errno(ft,SOX_EOF,"Invalid audio buffer size %lu", (unsigned long)file->size);
return (SOX_EOF);
}
file->count = 0;
file->pos = 0;
file->buf = lsx_malloc(file->size);
if (ioctl(fileno(ft->fp), (size_t) SNDCTL_DSP_SYNC, NULL) < 0) {
lsx_fail_errno(ft,SOX_EOF,"Unable to sync dsp");
return (SOX_EOF);
}
/* Change to non-buffered I/O */
setvbuf(ft->fp, NULL, _IONBF, sizeof(char) * file->size);
return(SOX_SUCCESS);
}
