[MPlayer-users] audio output is broken on powerpc
past-in at stud.tu-ilmenau.de
past-in at stud.tu-ilmenau.de
Tue Feb 12 06:09:01 CET 2002
Hello.
My powerpc system:
Linux manelan061.rz.tu-ilmenau.de 2.4.18-pre9 #1 Sun Feb 10 03:58:18 UTC 2002 ppc unknown
My test stream (5mb prefix of movie):
http://www.tu-ilmenau.de/~past-in/movie.avi
I've modified libao2/ao_oss.c so that it doesn't touch /dev/dsp,
but instead it writes to /usr/log/sound what it would otherwise write
to /dev/dsp. (Patch included below for completeness.)
On an i386 box, this gives me correct pcm data that I can listen to thus:
$ sox -t raw -r 44100 -c 2 -w -s /usr/log/sound -t ossdsp /dev/dsp
(And because the patch makes `-ao oss' behave essentially like `-ao pcm'
(only very much faster (on my powerpc at least (bug in `-ao pcm'?))),
it should give correct pcm on any machine.)
But on my powerpc, this produces
http://www.tu-ilmenau.de/~past-in/sound.bz2
which i've listened to as
16-bit little-endian signed
and 16-bit big-endian signed
and 16-bit little-endian unsigned
and 16-bit big-endian unsigned,
but it never sounds right.
Hope this helps. Keep up the good work.
-paul
-------------------------------------------------------------------------------
fake-dsp-patch against current CVS tree:
-------------------------------------------------------------------------------
--- ../mplayer-cvs-download/main/libao2/ao_oss.c Tue Jan 15 22:55:28 2002
+++ libao2/ao_oss.c Tue Feb 12 03:09:18 2002
@@ -16,6 +16,18 @@
#include "audio_out.h"
#include "audio_out_internal.h"
+#undef HAVE_AUDIO_SELECT
+static void fill_zz(audio_buf_info *zz)
+{
+ static int once=1;
+ zz->fragments=once?2:1;
+ zz->fragstotal=2;
+ zz->fragsize=32768;
+ zz->bytes=once?65536:32768;
+ if (once)
+ once=0;
+}
+
extern int verbose;
static ao_info_t info =
@@ -106,11 +118,13 @@
if (verbose)
printf("audio_setup: using '%s' dsp device\n", dsp);
+#if 0 //@@@
audio_fd=open(dsp, O_WRONLY);
if(audio_fd<0){
printf("Can't open audio device %s -> nosound\n",dsp);
return 0;
}
+#endif
ao_data.bps=channels*rate;
if(format != AFMT_U8 && format != AFMT_S8)
@@ -120,9 +134,9 @@
ao_data.samplerate=rate;
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
}
-
+
ao_data.format=format;
- ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
+// ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);//@@@
if(format == AFMT_AC3 && ao_data.format != AFMT_AC3) {
printf("Can't set audio device %s to AC3 output\n", dsp);
return 0;
@@ -134,27 +148,34 @@
// We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
ao_data.channels = channels;
if (ao_data.channels > 2) {
+exit(23);
+//freebsd 4.0 doesn't know SNDCTL_DSP_CHANNELS
+#if 0
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
ao_data.channels != channels ) {
printf("audio_setup: Failed to set audio device to %d channels\n", channels);
return 0;
}
+#endif
}
- else {
+ else { //@@@
+#if 0
int c = ao_data.channels-1;
if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
printf("audio_setup: Failed to set audio device to %d channels\n", ao_data.channels);
return 0;
}
+#endif
}
printf("audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
// set rate
ao_data.samplerate=rate;
- ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
+// ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); //@@@
+ao_data.samplerate=44100; //@@@ SAMPLERATE
printf("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
}
- if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
+ if(0&&ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
int r=0;
printf("audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
@@ -164,6 +185,9 @@
printf("audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
}
} else {
+//@@@
+//fprintf(stderr, "init(): fragments=%d fragstotal=%d fragsize=%d bytes=%d\n", zz.fragments,zz.fragstotal,zz.fragsize,zz.bytes);fflush(stderr);
+fill_zz(&zz);
printf("audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
@@ -207,6 +231,7 @@
// stop playing and empty buffers (for seeking/pause)
static void reset(){
+exit(23);
uninit();
audio_fd=open(dsp, O_WRONLY);
if(audio_fd<0){
@@ -217,7 +242,9 @@
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
if(ao_data.format != AFMT_AC3) {
if (ao_data.channels > 2)
- ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
+//freebsd 4.0 doesn't know SNDCTL_DSP_CHANNELS
+exit(23);
+// ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
else {
int c = ao_data.channels-1;
ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
@@ -244,7 +271,10 @@
int playsize=ao_data.outburst;
#ifdef SNDCTL_DSP_GETOSPACE
- if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
+ if(1||ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
+//@@@
+//fprintf(stderr, "get_space(): fragments=%d fragstotal=%d fragsize=%d bytes=%d\n", zz.fragments,zz.fragstotal,zz.fragsize,zz.bytes);fflush(stderr);
+fill_zz(&zz);
// calculate exact buffer space:
playsize = zz.fragments*zz.fragsize;
if (playsize > MAX_OUTBURST)
@@ -273,7 +303,20 @@
// return: number of bytes played
static int play(void* data,int len,int flags){
len/=ao_data.outburst;
- len=write(audio_fd,data,len*ao_data.outburst);
+// len=write(audio_fd,data,len*ao_data.outburst);
+len=len*ao_data.outburst;
+{//@@@ sound logging
+ static int once = 1;
+ static fd = -1;
+ if (once) {
+ once = 0;
+ fd = open("/usr/log/sound", O_CREAT|O_TRUNC|O_WRONLY, 0644);
+ if (fd < 0)
+ exit(23);
+ }
+ if (write(fd, data,len)!=len)
+ exit(23);
+}
return len;
}
@@ -284,15 +327,23 @@
/* Calculate how many bytes/second is sent out */
if(audio_delay_method==2){
int r=0;
- if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
+ if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1) {
+exit(23);
return ((float)r)/(float)ao_data.bps;
+ }
audio_delay_method=1; // fallback if not supported
}
if(audio_delay_method==1){
// SNDCTL_DSP_GETOSPACE
- if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
+ if(1||ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1) {
+//@@@
+//fprintf(stderr, "get_delay(): fragments=%d fragstotal=%d fragsize=%d bytes=%d\n", zz.fragments,zz.fragstotal,zz.fragsize,zz.bytes);fflush(stderr);
+fill_zz(&zz);
return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
+ }
+exit(23);
audio_delay_method=0; // fallback if not supported
}
+exit(23);
return ((float)ao_data.buffersize)/(float)ao_data.bps;
}
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