[Mplayer-dev-eng] Update for libao2 sun audio
Juergen Keil
jk at tools.de
Fri Jun 22 18:23:05 CEST 2001
Hi,
> Sample counts do not work with OSS' /dev/audio emulation. (At least
> not in the evaluation version of the SB LIVE driver.)
Ohh. That's bad.
On the other hand I found an old 3.94 beta version, and that version did
provide a working sample counter (using a SB 16PCI card).
> Did you provide a fallback?
There was a fallback solution, but you had to change the source to enable
it. ``#define USE_BURST_TIMING 1'' near the end of ao_sun.c and you get back
the old behaviour.
Anyway. Appended is an updated patch for ao_sun.c
There's a short timing loop in the ao_sun driver now; it tests the quality of
the sample counter provided by the soundcard/driver and enables the new
sample based timing code only, if the soundcard/driver combination can provide
sufficiently accurate sample counts.
--
Jürgen Keil jk at tools.de
Tools GmbH +49 (228) 9858011
-------------- next part --------------
Index: ao_sun.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_sun.c,v
retrieving revision 1.2
diff -u -B -r1.2 ao_sun.c
--- ao_sun.c 2001/06/08 23:31:06 1.2
+++ ao_sun.c 2001/06/22 16:10:39
@@ -1,12 +1,14 @@
#include <stdio.h>
#include <stdlib.h>
+#include <string.h>
-#include <sys/ioctl.h>
#include <unistd.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <sys/ioctl.h>
#include <sys/time.h>
#include <sys/types.h>
#include <sys/stat.h>
-#include <fcntl.h>
#include <sys/audioio.h>
#ifdef __svr4__
#include <stropts.h>
@@ -20,19 +22,24 @@
static ao_info_t info =
{
- "Sun audio output",
- "sun",
- "jk at tools.de",
- ""
+ "Sun audio output",
+ "sun",
+ "jk at tools.de",
+ ""
};
LIBAO_EXTERN(sun)
+/* These defines are missing on NetBSD */
#ifndef AUDIO_PRECISION_8
-#define AUDIO_PRECISION_8 8
-#define AUDIO_PRECISION_16 16
+#define AUDIO_PRECISION_8 8
+#define AUDIO_PRECISION_16 16
#endif
+#ifndef AUDIO_CHANNELS_MONO
+#define AUDIO_CHANNELS_MONO 1
+#define AUDIO_CHANNELS_STEREO 2
+#endif
// there are some globals:
@@ -43,9 +50,19 @@
// ao_outburst
// ao_buffersize
-static char *dsp="/dev/audio";
+static char *audio_dev = "/dev/audio";
static int queued_bursts = 0;
-static int audio_fd=-1;
+static int queued_samples = 0;
+static int bytes_per_sample = 0;
+static int audio_fd = -1;
+static enum {
+ RTSC_UNKNOWN = 0,
+ RTSC_ENABLED,
+ RTSC_DISABLED
+} enable_sample_timing;
+
+extern int verbose;
+
// convert an OSS audio format specification into a sun audio encoding
static int oss2sunfmt(int oss_format)
@@ -68,16 +85,127 @@
}
}
+// try to figure out, if the soundcard driver provides usable (precise)
+// sample counter information
+static int realtime_samplecounter_available(char *dev)
+{
+ int fd = -1;
+ audio_info_t info;
+ int rtsc_ok = RTSC_DISABLED;
+ int len;
+ void *silence = NULL;
+ struct timeval start, end;
+ struct timespec delay;
+ int usec_delay;
+ unsigned last_samplecnt;
+ unsigned increment;
+ unsigned min_increment;
+
+ len = 44100 * 4 / 4; // amount of data for 0.25sec of 44.1khz, stereo, 16bit
+ silence = calloc(1, len);
+ if (silence == NULL)
+ goto error;
+
+ if ((fd = open(dev, O_WRONLY)) < 0)
+ goto error;
+
+ AUDIO_INITINFO(&info);
+ info.play.sample_rate = 44100;
+ info.play.channels = AUDIO_CHANNELS_STEREO;
+ info.play.precision = AUDIO_PRECISION_16;
+ info.play.encoding = AUDIO_ENCODING_LINEAR;
+ info.play.samples = 0;
+ if (ioctl(fd, AUDIO_SETINFO, &info)) {
+ if (verbose)
+ printf("rtsc: SETINFO failed\n");
+ goto error;
+ }
+
+ if (write(fd, silence, len) != len) {
+ if (verbose)
+ printf("rtsc: write failed");
+ goto error;
+ }
+
+ if (ioctl(fd, AUDIO_GETINFO, &info)) {
+ if (verbose)
+ perror("rtsc: GETINFO1");
+ goto error;
+ }
+
+ last_samplecnt = info.play.samples;
+ min_increment = ~0;
+
+ gettimeofday(&start, NULL);
+ for (;;) {
+ delay.tv_sec = 0;
+ delay.tv_nsec = 10000000;
+ nanosleep(&delay, NULL);
+ gettimeofday(&end, NULL);
+ usec_delay = (end.tv_sec - start.tv_sec) * 1000000
+ + end.tv_usec - start.tv_usec;
+
+ // stop monitoring sample counter after 0.2 seconds
+ if (usec_delay > 200000)
+ break;
+
+ if (ioctl(fd, AUDIO_GETINFO, &info)) {
+ if (verbose)
+ perror("rtsc: GETINFO2 failed");
+ goto error;
+ }
+ if (info.play.samples < last_samplecnt) {
+ if (verbose)
+ printf("rtsc: %d > %d?\n", last_samplecnt, info.play.samples);
+ goto error;
+ }
+
+ if ((increment = info.play.samples - last_samplecnt) > 0) {
+ if (verbose)
+ printf("ao_sun: sample counter increment: %d\n", increment);
+ if (increment < min_increment) {
+ min_increment = increment;
+ if (min_increment < 2000)
+ break; // looks good
+ }
+ }
+ last_samplecnt = info.play.samples;
+ }
+
+ if (min_increment < 2000)
+ rtsc_ok = RTSC_ENABLED;
+
+ if (verbose)
+ printf("ao_sun: minimum sample counter increment per 10msec interval: %d\n"
+ "\t%susing sample counter based timing code\n",
+ min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not ");
+
+
+error:
+ if (silence != NULL) free(silence);
+ if (fd >= 0) {
+#ifdef __svr4__
+ // remove the 0 bytes from the above measurement from the
+ // audio driver's STREAMS queue
+ ioctl(fd, I_FLUSH, FLUSHW);
+#endif
+ //ioctl(fd, AUDIO_DRAIN, 0);
+ close(fd);
+ }
+
+ return rtsc_ok;
+}
+
// to set/get/query special features/parameters
static int control(int cmd,int arg){
- switch(cmd){
- case AOCONTROL_SET_DEVICE:
- dsp=(char*)arg;
- return CONTROL_OK;
- case AOCONTROL_QUERY_FORMAT:
- return CONTROL_TRUE;
- }
- return CONTROL_UNKNOWN;
+ switch(cmd){
+ case AOCONTROL_SET_DEVICE:
+ audio_dev=(char*)arg;
+ return CONTROL_OK;
+ case AOCONTROL_QUERY_FORMAT:
+ return CONTROL_TRUE;
+ }
+ return CONTROL_UNKNOWN;
}
// open & setup audio device
@@ -86,12 +214,18 @@
audio_info_t info;
int byte_per_sec;
+
+ if (enable_sample_timing == RTSC_UNKNOWN
+ && !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) {
+ enable_sample_timing = realtime_samplecounter_available(audio_dev);
+ }
- printf("ao2: %d Hz %d chans 0x%X\n",rate,channels,format);
+ printf("ao2: %d Hz %d chans %s [0x%X]\n",
+ rate,channels,audio_out_format_name(format),format);
- audio_fd=open(dsp, O_WRONLY);
+ audio_fd=open(audio_dev, O_WRONLY);
if(audio_fd<0){
- printf("Can't open audio device %s -> nosound\n",dsp);
+ printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno));
return 0;
}
@@ -101,15 +236,12 @@
info.play.encoding = oss2sunfmt(ao_format = format);
info.play.precision = (format==AFMT_S16_LE? AUDIO_PRECISION_16:AUDIO_PRECISION_8);
info.play.channels = ao_channels = channels;
- --ao_channels;
info.play.sample_rate = ao_samplerate = rate;
- info.play.samples = 0;
- info.play.eof = 0;
if(ioctl (audio_fd, AUDIO_SETINFO, &info)<0)
printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n",channels,audio_out_format_name(format),rate);
- byte_per_sec = (channels * info.play.precision * rate);
- ao_outburst=byte_per_sec > 100000 ? 16384 : 8192;
- queued_bursts = 0;
+ bytes_per_sample = channels * info.play.precision / 8;
+ byte_per_sec = bytes_per_sample * rate;
+ ao_outburst = byte_per_sec > 100000 ? 16384 : 8192;
if(ao_buffersize==-1){
// Measuring buffer size:
@@ -140,12 +272,25 @@
ioctl(audio_fd, AUDIO_DRAIN, 0);
#endif
}
+
+ AUDIO_INITINFO(&info);
+ info.play.samples = 0;
+ info.play.eof = 0;
+ info.play.error = 0;
+ ioctl (audio_fd, AUDIO_SETINFO, &info);
+
+ queued_bursts = 0;
+ queued_samples = 0;
- return 1;
+ return 1;
}
// close audio device
static void uninit(){
+#ifdef __svr4__
+ // throw away buffered data in the audio driver's STREAMS queue
+ ioctl(audio_fd, I_FLUSH, FLUSHW);
+#endif
close(audio_fd);
}
@@ -153,14 +298,10 @@
static void reset(){
audio_info_t info;
-#ifdef __svr4__
- // throw away buffered data in the audio driver's STREAMS queue
- ioctl(audio_fd, I_FLUSH, FLUSHW);
-#endif
uninit();
- audio_fd=open(dsp, O_WRONLY);
+ audio_fd=open(audio_dev, O_WRONLY);
if(audio_fd<0){
- printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE ***\n");
+ printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno));
return;
}
@@ -169,12 +310,14 @@
AUDIO_INITINFO(&info);
info.play.encoding = oss2sunfmt(ao_format);
info.play.precision = (ao_format==AFMT_S16_LE? AUDIO_PRECISION_16:AUDIO_PRECISION_8);
- info.play.channels = ao_channels+1;
+ info.play.channels = ao_channels;
info.play.sample_rate = ao_samplerate;
info.play.samples = 0;
info.play.eof = 0;
+ info.play.error = 0;
ioctl (audio_fd, AUDIO_SETINFO, &info);
queued_bursts = 0;
+ queued_samples = 0;
}
// stop playing, keep buffers (for pause)
@@ -198,49 +341,54 @@
// return: how many bytes can be played without blocking
static int get_space(){
- int playsize=ao_outburst;
+ int playsize = ao_outburst;
+ audio_info_t info;
// check buffer
#ifdef HAVE_AUDIO_SELECT
- { fd_set rfds;
- struct timeval tv;
- FD_ZERO(&rfds);
- FD_SET(audio_fd, &rfds);
- tv.tv_sec = 0;
- tv.tv_usec = 0;
- if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
+ {
+ fd_set rfds;
+ struct timeval tv;
+ FD_ZERO(&rfds);
+ FD_SET(audio_fd, &rfds);
+ tv.tv_sec = 0;
+ tv.tv_usec = 0;
+ if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
}
#endif
- {
- audio_info_t info;
ioctl(audio_fd, AUDIO_GETINFO, &info);
- if(queued_bursts - info.play.eof > 2)
- return 0;
- }
- return ao_outburst;
+ if (queued_bursts - info.play.eof > 2)
+ return 0;
+
+ return ao_outburst;
}
// plays 'len' bytes of 'data'
// it should round it down to outburst*n
// return: number of bytes played
static int play(void* data,int len,int flags){
- len/=ao_outburst;
- len=write(audio_fd,data,len*ao_outburst);
- if(len>0) {
- queued_bursts ++;
- write(audio_fd,data,0);
+ if (len < ao_outburst) return 0;
+ len /= ao_outburst;
+ len = write(audio_fd, data, len*ao_outburst);
+ if(len > 0) {
+ queued_samples += len / bytes_per_sample;
+ if (write(audio_fd,data,0) < 0)
+ perror("ao_sun: send EOF audio record");
+ else
+ queued_bursts ++;
}
return len;
}
-static int audio_delay_method=2;
// return: how many unplayed bytes are in the buffer
static int get_delay(){
- int q;
audio_info_t info;
ioctl(audio_fd, AUDIO_GETINFO, &info);
- return (queued_bursts - info.play.eof) * ao_outburst;
+ if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
+ return (queued_samples - info.play.samples) * bytes_per_sample;
+ else
+ return (queued_bursts - info.play.eof) * ao_outburst;
}
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