asgardius/alsa-utils
1
0
Fork 0
mirror of https://github.com/alsa-project/alsa-utils synced 2024-12-23 02:26:31 +01:00
Code Issues Projects Releases Packages Wiki Activity

split interleaved files in arecord when reaching 2GB

Browse source 
From: Dirk Jagdmann <doj@cubic.org>

This patch will split files in chunks of aprox. 2GB in the interleaved
recording mode of arecord. This is meant to overcome the 2GB filesize
limit of .wav files, which is easily reached when recording multiple
channels.
This commit is contained in:
Takashi Iwai 2005-12-06 11:25:47 +00:00
parent 504131b582
commit 5f9f0e817c
2 changed files with 116 additions and 49 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

3
aplay/aplay.1
View file

@ -10,7 +10,8 @@ soundcard driver
.SH DESCRIPTION
\fBarecord\fP is a command-line soundfile recorder for the ALSA soundcard
driver. It supports several file formats and multiple soundcards with
multiple devices.
multiple devices. If recording with interleaved mode samples the file is
automatically split before the 2GB filesize.
\fBaplay\fP is much the same, only it plays instead of recording. For
supported soundfile formats, the sampling rate, bit depth, and so

162
aplay/aplay.c
View file

@ -124,11 +124,12 @@ struct fmt_capture {
void (*start) (int fd, size_t count);
void (*end) (int fd);
char *what;
long long max_filesize;
} fmt_rec_table[] = {
{ NULL, end_raw, N_("raw data") },
{ begin_voc, end_voc, N_("VOC") },
{ begin_wave, end_wave, N_("WAVE") },
{ begin_au, end_au, N_("Sparc Audio")}
{ NULL, end_raw, N_("raw data"), LLONG_MAX },
{ begin_voc, end_voc, N_("VOC"), 16000000 },
{ begin_wave, end_wave, N_("WAVE"), 2000000000 },
{ begin_au, end_au, N_("Sparc Audio"), LLONG_MAX }
};
#if __GNUC__ > 2 || (__GNUC__ == 2 && __GNUC_MINOR__ >= 95)
@ -1985,33 +1986,6 @@ void playback_go(int fd, size_t loaded, off64_t count, int rtype, char *name)
snd_pcm_drain(handle);
}
/* capturing raw data, this proc handels WAVE files and .VOCs (as one block) */
void capture_go(int fd, off64_t count, int rtype, char *name)
{
size_t c;
off64_t cur;
ssize_t r, err;
header(rtype, name);
set_params();
do {
for (cur = count; cur > 0; cur -= r) {
c = (cur <= chunk_bytes) ? cur : chunk_bytes;
c = c * 8 / bits_per_frame;
if ((size_t)(r = pcm_read(audiobuf, c)) != c)
break;
r = r * bits_per_frame / 8;
if ((err = write(fd, audiobuf, r)) != r) {
perror(name);
exit(EXIT_FAILURE);
}
if (err > 0)
fdcount += err;
}
} while (rtype == FORMAT_RAW && !timelimit);
}
/*
* let's play or capture it (capture_type says VOC/WAVE/raw)
@ -2072,20 +2046,15 @@ static void playback(char *name)
close(fd);
}
static void capture(char *name)
static void capture(char *name_)
{
pbrec_count = LLONG_MAX;
if (!name || !strcmp(name, "-")) {
fd = fileno(stdout);
name = "stdout";
} else {
remove(name);
if ((fd = open64(name, O_WRONLY | O_CREAT, 0644)) == -1) {
perror(name);
exit(EXIT_FAILURE);
}
}
fdcount = 0;
int tostdout=0; /* boolean which describes output stream */
int filecount=0; /* number of files written */
char *name=name_; /* current filename */
char namebuf[PATH_MAX+1];
off64_t cur; /* number of bytes to capture */
/* get number of bytes to capture */
pbrec_count = calc_count();
/* WAVE-file should be even (I'm not sure), but wasting one byte
isn't a problem (this can only be in 8 bit mono) */
@ -2095,10 +2064,107 @@ static void capture(char *name)
pbrec_count -= pbrec_count % 2;
if (pbrec_count == 0)
pbrec_count -= 2;
if (fmt_rec_table[file_type].start)
fmt_rec_table[file_type].start(fd, pbrec_count);
capture_go(fd, pbrec_count, file_type, name);
fmt_rec_table[file_type].end(fd);
cur = pbrec_count;
/* display verbose output to console */
header(file_type, name);
/* setup sound hardware */
set_params();
/* write to stdout? */
if (!name || !strcmp(name, "-")) {
fd = fileno(stdout);
name = "stdout";
tostdout=1;
fdcount = 0;
}
do {
/* open a file to write */
if(!tostdout) {
/* upon the second file we start the numbering scheme */
if(filecount)
{
/* get a copy of the original filename */
char *s;
char buf[PATH_MAX+1];
strncpy(buf, name_, sizeof(buf));
/* separate extension from filename */
s=buf+strlen(buf);
while(s>buf && *s!='.' && *s!='/')
--s;
if(*s=='.')
*s++=0;
else if(*s=='/')
s=buf+strlen(buf);
/* upon first jump to this if block rename the first file */
if(filecount==1) {
if(*s)
snprintf(namebuf, sizeof(namebuf), "%s-01.%s", buf, s);
else
snprintf(namebuf, sizeof(namebuf), "%s-01", buf);
remove(namebuf);
rename(name, namebuf);
filecount=2;
}
/* name of the current file */
if(*s)
snprintf(namebuf, sizeof(namebuf), "%s-%02i.%s", buf, filecount, s);
else
snprintf(namebuf, sizeof(namebuf), "%s-%02i", buf, filecount);
name=namebuf;
}
/* open a new file */
remove(name);
if ((fd = open64(name, O_WRONLY | O_CREAT, 0644)) == -1) {
perror(name);
exit(EXIT_FAILURE);
}
fdcount = 0;
filecount++;
}
/* setup sample header */
if (fmt_rec_table[file_type].start)
fmt_rec_table[file_type].start(fd, pbrec_count);
/* capture */
do {
ssize_t r=0;
for (; cur > 0 && fdcount<fmt_rec_table[file_type].max_filesize; cur -= r) {
ssize_t err;
size_t c = (cur <= chunk_bytes) ? cur : chunk_bytes;
c = c * 8 / bits_per_frame;
if ((size_t)(r = pcm_read(audiobuf, c)) != c)
break;
r = r * bits_per_frame / 8;
if ((err = write(fd, audiobuf, r)) != r) {
perror(name);
exit(EXIT_FAILURE);
}
if (err > 0)
fdcount += err;
}
/* exit conditions:
1) format_raw and a timelimit
2) all requested samples/bytes have been captured (cur>0)
3) filesize threshold was reached (fdcount<wrthreshold)
*/
} while (file_type == FORMAT_RAW && !timelimit && cur>0 && fdcount<fmt_rec_table[file_type].max_filesize);
/* finish sample container */
fmt_rec_table[file_type].end(fd);
/* repeat the loop when format is raw without timelimit or filesize threshold was reached */
} while((file_type == FORMAT_RAW && !timelimit) || fdcount>=fmt_rec_table[file_type].max_filesize);
}
void playbackv_go(int* fds, unsigned int channels, size_t loaded, off64_t count, int rtype, char **names)