Re: [patch] split interleaved files in arecord when reaching 2GB

[Dirk Jagdmann <doj@cubic.org>]

[-- Attachment #1: Type: text/plain, Size: 559 bytes --]

> It's a nice idea.  But, I prefer this as an option.
> For example, there is no 2GB limit for the raw format.
> We can set this on automatically for WAV format, though...

You're right. it makes no sense to split raw and au files. I have 
modified my patch to work with the fileformat table fmt_rec_table my 
adding a variable max_filesize. File split is set to ~2GB for wav, ~16MB 
for voc and LLONG_MAX for au and raw (which should not be reached in 
practice).

-- 
---> Dirk Jagdmann ^ doj / cubic
----> http://cubic.org/~doj
-----> http://llg.cubic.org

[-- Attachment #2: aplay-splitfiles.patch --]
[-- Type: text/plain, Size: 6920 bytes --]

Index: alsa-utils-1.0.10/aplay/aplay.c
===================================================================
--- alsa-utils-1.0.10.orig/aplay/aplay.c	2005-12-05 23:16:19.000000000 +0100
+++ alsa-utils-1.0.10/aplay/aplay.c	2005-12-05 23:36:44.000000000 +0100
@@ -124,11 +124,12 @@
 	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 @@
 	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 @@
 		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 @@
 		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)
Index: alsa-utils-1.0.10/aplay/aplay.1
===================================================================
--- alsa-utils-1.0.10.orig/aplay/aplay.1	2005-12-05 23:10:36.000000000 +0100
+++ alsa-utils-1.0.10/aplay/aplay.1	2005-12-05 23:23:28.000000000 +0100
@@ -10,7 +10,8 @@
 .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
Index: alsa-utils-1.0.10/aplay/arecord.1
===================================================================
--- alsa-utils-1.0.10.orig/aplay/arecord.1	2005-12-05 23:10:36.000000000 +0100
+++ alsa-utils-1.0.10/aplay/arecord.1	2005-12-05 23:23:28.000000000 +0100
@@ -10,7 +10,8 @@
 .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