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Largely simplify format handlers (for file copy etc.)

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collecting common functions in a single place and removing
them from the individual handlers.
The full description is on mantis,
http://bugs.digium.com/view.php?id=6375
and only the ogg_vorbis handler needs to be converted to
the new structure.

As a result of this change, format_au.c and format_pcm_alaw.c
should go away (in a separate commit) as their functionality
(trivial) has been merged in another file.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Luigi Rizzo
2006-04-04 12:59:25 +00:00
parent ec67c650ad
commit 4beb6deeaa
18 changed files with 1748 additions and 2790 deletions
Download Patch File Download Diff File Expand all files Collapse all files

230
formats/format_wav.c
View File

@@ -49,30 +49,16 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
/* Portions of the conversion code are by guido@sienanet.it */
struct ast_filestream {
void *reserved[AST_RESERVED_POINTERS];
/* This is what a filestream means to us */
FILE *f; /* Descriptor */
#define WAV_BUF_SIZE 320
struct wav_desc { /* format-specific parameters */
int bytes;
int needsgain;
struct ast_frame fr; /* Frame information */
char waste[AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */
char empty; /* Empty character */
short buf[160];
int foffset;
int lasttimeout;
int maxlen;
struct timeval last;
};
AST_MUTEX_DEFINE_STATIC(wav_lock);
static int glistcnt = 0;
static char *name = "wav";
static char *desc = "Microsoft WAV format (8000hz Signed Linear)";
static char *exts = "wav";
#define BLOCKSIZE 160
#define GAIN 2 /* 2^GAIN is the multiple to increase the volume by */
@@ -165,7 +151,7 @@ static int check_header(FILE *f)
ast_log(LOG_WARNING, "Read failed (freq)\n");
return -1;
}
if (ltohl(freq) != 8000) {
if (ltohl(freq) != DEFAULT_SAMPLE_RATE) {
ast_log(LOG_WARNING, "Unexpected freqency %d\n", ltohl(freq));
return -1;
}
@@ -233,7 +219,6 @@ static int update_header(FILE *f)
off_t cur,end;
int datalen,filelen,bytes;
cur = ftello(f);
fseek(f, 0, SEEK_END);
end = ftello(f);
@@ -333,135 +318,90 @@ static int write_header(FILE *f)
return 0;
}
static struct ast_filestream *wav_open(FILE *f)
static int wav_open(struct ast_filestream *s)
{
/* We don't have any header to read or anything really, but
if we did, it would go here. We also might want to check
and be sure it's a valid file. */
struct ast_filestream *tmp;
if ((tmp = malloc(sizeof(struct ast_filestream)))) {
memset(tmp, 0, sizeof(struct ast_filestream));
if ((tmp->maxlen = check_header(f)) < 0) {
free(tmp);
return NULL;
}
if (ast_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav list\n");
free(tmp);
return NULL;
}
tmp->f = f;
tmp->needsgain = 1;
tmp->fr.data = tmp->buf;
tmp->fr.frametype = AST_FRAME_VOICE;
tmp->fr.subclass = AST_FORMAT_SLINEAR;
/* datalen will vary for each frame */
tmp->fr.src = name;
tmp->fr.mallocd = 0;
tmp->bytes = 0;
glistcnt++;
ast_mutex_unlock(&wav_lock);
ast_update_use_count();
}
return tmp;
struct wav_desc *tmp = (struct wav_desc *)s->private;
if ((tmp->maxlen = check_header(s->f)) < 0)
return -1;
return 0;
}
static struct ast_filestream *wav_rewrite(FILE *f, const char *comment)
static int wav_rewrite(struct ast_filestream *s, const char *comment)
{
/* We don't have any header to read or anything really, but
if we did, it would go here. We also might want to check
and be sure it's a valid file. */
struct ast_filestream *tmp;
if ((tmp = malloc(sizeof(struct ast_filestream)))) {
memset(tmp, 0, sizeof(struct ast_filestream));
if (write_header(f)) {
free(tmp);
return NULL;
}
if (ast_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav list\n");
free(tmp);
return NULL;
}
tmp->f = f;
glistcnt++;
ast_mutex_unlock(&wav_lock);
ast_update_use_count();
} else
ast_log(LOG_WARNING, "Out of memory\n");
return tmp;
if (write_header(s->f))
return -1;
return 0;
}
static void wav_close(struct ast_filestream *s)
{
char zero = 0;
if (ast_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav list\n");
return;
}
glistcnt--;
ast_mutex_unlock(&wav_lock);
ast_update_use_count();
struct wav_desc *fs = (struct wav_desc *)s->private;
/* Pad to even length */
if (s->bytes & 0x1)
if (fs->bytes & 0x1)
fwrite(&zero, 1, 1, s->f);
fclose(s->f);
free(s);
s = NULL;
}
static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext)
{
int res;
int delay;
int samples; /* actual samples read */
int x;
short tmp[sizeof(s->buf) / 2];
int bytes = sizeof(tmp);
short *tmp;
int bytes = WAV_BUF_SIZE; /* in bytes */
off_t here;
/* Send a frame from the file to the appropriate channel */
struct wav_desc *fs = (struct wav_desc *)s->private;
here = ftello(s->f);
if ((s->maxlen - here) < bytes)
bytes = s->maxlen - here;
if (fs->maxlen - here < bytes) /* truncate if necessary */
bytes = fs->maxlen - here;
if (bytes < 0)
bytes = 0;
/* ast_log(LOG_DEBUG, "here: %d, maxlen: %d, bytes: %d\n", here, s->maxlen, bytes); */
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_SLINEAR;
s->fr.mallocd = 0;
FR_SET_BUF(&s->fr, s->buf, AST_FRIENDLY_OFFSET, bytes);
if ( (res = fread(tmp, 1, bytes, s->f)) <= 0 ) {
if (res) {
if ( (res = fread(s->fr.data, 1, s->fr.datalen, s->f)) <= 0 ) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
}
return NULL;
}
s->fr.datalen = res;
s->fr.samples = samples = res / 2;
tmp = (short *)(s->fr.data);
#if __BYTE_ORDER == __BIG_ENDIAN
for( x = 0; x < sizeof(tmp)/2; x++) tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8);
/* file format is little endian so we need to swap */
for( x = 0; x < samples; x++)
tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8);
#endif
if (s->needsgain) {
for (x=0;x<sizeof(tmp)/2;x++)
if (fs->needsgain) {
for (x=0; x < samples; x++) {
if (tmp[x] & ((1 << GAIN) - 1)) {
/* If it has data down low, then it's not something we've artificially increased gain
on, so we don't need to gain adjust it */
s->needsgain = 0;
fs->needsgain = 0;
break;
}
}
if (s->needsgain) {
for (x=0;x<sizeof(tmp)/2;x++) {
s->buf[x] = tmp[x] >> GAIN;
}
} else {
memcpy(s->buf, tmp, sizeof(s->buf));
if (fs->needsgain) {
for (x=0; x < samples; x++)
tmp[x] = tmp[x] >> GAIN;
}
}
delay = res / 2;
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_SLINEAR;
s->fr.offset = AST_FRIENDLY_OFFSET;
s->fr.datalen = res;
s->fr.data = s->buf;
s->fr.mallocd = 0;
s->fr.samples = delay;
*whennext = delay;
*whennext = samples;
return &s->fr;
}
@@ -470,6 +410,9 @@ static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
int x;
short tmp[8000], *tmpi;
float tmpf;
struct wav_desc *s = (struct wav_desc *)fs->private;
int res;
if (f->frametype != AST_FRAME_VOICE) {
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
@@ -489,33 +432,28 @@ static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
printf("Data Length: %d\n", f->datalen);
#endif
if (fs->buf) {
tmpi = f->data;
/* Volume adjust here to accomodate */
for (x=0;x<f->datalen/2;x++) {
tmpf = ((float)tmpi[x]) * ((float)(1 << GAIN));
if (tmpf > 32767.0)
tmpf = 32767.0;
if (tmpf < -32768.0)
tmpf = -32768.0;
tmp[x] = tmpf;
tmp[x] &= ~((1 << GAIN) - 1);
tmpi = f->data;
/* Volume adjust here to accomodate */
for (x=0;x<f->datalen/2;x++) {
tmpf = ((float)tmpi[x]) * ((float)(1 << GAIN));
if (tmpf > 32767.0)
tmpf = 32767.0;
if (tmpf < -32768.0)
tmpf = -32768.0;
tmp[x] = tmpf;
tmp[x] &= ~((1 << GAIN) - 1);
#if __BYTE_ORDER == __BIG_ENDIAN
tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8);
tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8);
#endif
}
if ((fwrite(tmp, 1, f->datalen, fs->f) != f->datalen) ) {
ast_log(LOG_WARNING, "Bad write (%d): %s\n", errno, strerror(errno));
return -1;
}
} else {
ast_log(LOG_WARNING, "Cannot write data to file.\n");
}
if ((res = fwrite(tmp, 1, f->datalen, fs->f)) != f->datalen ) {
ast_log(LOG_WARNING, "Bad write (%d): %s\n", res, strerror(errno));
return -1;
}
fs->bytes += f->datalen;
s->bytes += f->datalen;
update_header(fs->f);
return 0;
@@ -560,43 +498,45 @@ static off_t wav_tell(struct ast_filestream *fs)
return (offset - 44)/2;
}
static char *wav_getcomment(struct ast_filestream *s)
{
return NULL;
}
static struct ast_format_lock me = { .usecnt = -1 };
static const struct ast_format wav_f = {
.name = "wav",
.exts = "wav",
.format = AST_FORMAT_SLINEAR,
.open = wav_open,
.rewrite = wav_rewrite,
.write = wav_write,
.seek = wav_seek,
.trunc = wav_trunc,
.tell = wav_tell,
.read = wav_read,
.close = wav_close,
.buf_size = WAV_BUF_SIZE + AST_FRIENDLY_OFFSET,
.desc_size = sizeof(struct wav_desc),
.lockp = &me,
};
int load_module()
{
return ast_format_register(name, exts, AST_FORMAT_SLINEAR,
wav_open,
wav_rewrite,
wav_write,
wav_seek,
wav_trunc,
wav_tell,
wav_read,
wav_close,
wav_getcomment);
return ast_format_register(&wav_f);
}
int unload_module()
{
return ast_format_unregister(name);
return ast_format_unregister(wav_f.name);
}
int usecount()
{
return glistcnt;
return me.usecnt;
}
char *description()
{
return desc;
return "Microsoft WAV format (8000hz Signed Linear)";
}
char *key()
{
return ASTERISK_GPL_KEY;