mirror of
https://github.com/mpv-player/mpv
synced 2024-11-11 00:15:33 +01:00
ee2ad20548
That is, if the user redirects audio to /dev/sound/1 with the AUDIODEV environment variable, use /dev/sound/1ctl as the mixer device. When AUDIODEV is not set, the default for both devices is /dev/audio & /dev/audioctl. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5894 b3059339-0415-0410-9bf9-f77b7e298cf2
520 lines
13 KiB
C
520 lines
13 KiB
C
#include <stdio.h>
|
|
#include <stdlib.h>
|
|
#include <string.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 <sys/audioio.h>
|
|
#ifdef __svr4__
|
|
#include <stropts.h>
|
|
#endif
|
|
|
|
#include "../config.h"
|
|
#include "../mixer.h"
|
|
|
|
#include "audio_out.h"
|
|
#include "audio_out_internal.h"
|
|
#include "afmt.h"
|
|
|
|
static ao_info_t info =
|
|
{
|
|
"Sun audio output",
|
|
"sun",
|
|
"jk@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
|
|
#endif
|
|
#ifndef AUDIO_CHANNELS_MONO
|
|
#define AUDIO_CHANNELS_MONO 1
|
|
#define AUDIO_CHANNELS_STEREO 2
|
|
#endif
|
|
|
|
|
|
static char *sun_mixer_device = NULL;
|
|
static char *audio_dev = NULL;
|
|
static int queued_bursts = 0;
|
|
static int queued_samples = 0;
|
|
static int bytes_per_sample = 0;
|
|
static int byte_per_sec = 0;
|
|
static int convert_u8_s8;
|
|
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)
|
|
{
|
|
switch (oss_format){
|
|
case AFMT_MU_LAW:
|
|
return AUDIO_ENCODING_ULAW;
|
|
case AFMT_A_LAW:
|
|
return AUDIO_ENCODING_ALAW;
|
|
case AFMT_S16_BE:
|
|
case AFMT_S16_LE:
|
|
return AUDIO_ENCODING_LINEAR;
|
|
#ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1...
|
|
case AFMT_U8:
|
|
return AUDIO_ENCODING_LINEAR8;
|
|
#endif
|
|
#ifdef AUDIO_ENCODING_DVI // Missing on NetBSD...
|
|
case AFMT_IMA_ADPCM:
|
|
return AUDIO_ENCODING_DVI;
|
|
#endif
|
|
default:
|
|
return AUDIO_ENCODING_NONE;
|
|
}
|
|
}
|
|
|
|
// 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. 44kbyte can be sent to all supported
|
|
* sun audio devices without blocking in the
|
|
* "write" below.
|
|
*/
|
|
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;
|
|
}
|
|
|
|
/*
|
|
* For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes
|
|
* chunks (== 4096 samples) to the audio device. If we see a minimum
|
|
* sample counter increment from the soundcard driver of less than
|
|
* 2000 samples, we assume that the driver provides a useable realtime
|
|
* sample counter in the AUDIO_INFO play.samples field. Timing based
|
|
* on sample counts should be much more accurate than counting whole
|
|
* 16kbyte chunks.
|
|
*/
|
|
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:
|
|
audio_dev=(char*)arg;
|
|
return CONTROL_OK;
|
|
case AOCONTROL_QUERY_FORMAT:
|
|
return CONTROL_TRUE;
|
|
case AOCONTROL_GET_VOLUME:
|
|
{
|
|
int fd,v,cmd,devs;
|
|
|
|
fd=open( sun_mixer_device,O_RDONLY );
|
|
if ( fd != -1 )
|
|
{
|
|
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
|
|
struct audio_info info;
|
|
ioctl( fd,AUDIO_GETINFO,&info);
|
|
vol->left=info.play.gain * 100. / AUDIO_MAX_GAIN;
|
|
vol->right=info.play.gain * 100. / AUDIO_MAX_GAIN;
|
|
close( fd );
|
|
return CONTROL_OK;
|
|
}
|
|
return CONTROL_ERROR;
|
|
}
|
|
case AOCONTROL_SET_VOLUME:
|
|
{
|
|
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
|
|
int fd,v,cmd,devs;
|
|
|
|
fd=open( sun_mixer_device,O_RDONLY );
|
|
if ( fd != -1 )
|
|
{
|
|
struct audio_info info;
|
|
AUDIO_INITINFO(&info);
|
|
info.play.gain = (vol->right+vol->left) * AUDIO_MAX_GAIN / 100 / 2;
|
|
ioctl( fd,AUDIO_SETINFO,&info );
|
|
close( fd );
|
|
return CONTROL_OK;
|
|
}
|
|
return CONTROL_ERROR;
|
|
}
|
|
}
|
|
return CONTROL_UNKNOWN;
|
|
}
|
|
|
|
// open & setup audio device
|
|
// return: 1=success 0=fail
|
|
static int init(int rate,int channels,int format,int flags){
|
|
|
|
audio_info_t info;
|
|
int ok;
|
|
|
|
if (audio_dev == NULL) {
|
|
if ((audio_dev = getenv("AUDIODEV")) == NULL)
|
|
audio_dev = "/dev/audio";
|
|
}
|
|
|
|
if (sun_mixer_device == NULL) {
|
|
if ((sun_mixer_device = mixer_device) == NULL) {
|
|
sun_mixer_device = malloc(strlen(audio_dev) + 4);
|
|
strcpy(sun_mixer_device, audio_dev);
|
|
strcat(sun_mixer_device, "ctl");
|
|
}
|
|
}
|
|
|
|
if (ao_subdevice) audio_dev = ao_subdevice;
|
|
|
|
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 %s [0x%X]\n",
|
|
// rate,channels,audio_out_format_name(format),format);
|
|
|
|
audio_fd=open(audio_dev, O_WRONLY);
|
|
if(audio_fd<0){
|
|
printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno));
|
|
return 0;
|
|
}
|
|
|
|
ioctl(audio_fd, AUDIO_DRAIN, 0);
|
|
|
|
AUDIO_INITINFO(&info);
|
|
info.play.encoding = oss2sunfmt(ao_data.format = format);
|
|
info.play.precision =
|
|
(format==AFMT_S16_LE || format==AFMT_S16_BE
|
|
? AUDIO_PRECISION_16
|
|
: AUDIO_PRECISION_8);
|
|
info.play.channels = ao_data.channels = channels;
|
|
info.play.sample_rate = ao_data.samplerate = rate;
|
|
convert_u8_s8 = 0;
|
|
ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
|
|
if (!ok && info.play.encoding == AUDIO_ENCODING_LINEAR8) {
|
|
/* sun audiocs hardware does not support U8 format, try S8... */
|
|
info.play.encoding = AUDIO_ENCODING_LINEAR;
|
|
ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
|
|
if (ok) {
|
|
/* we must perform software U8 -> S8 conversion */
|
|
convert_u8_s8 = 1;
|
|
}
|
|
}
|
|
if (!ok) {
|
|
printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n",
|
|
channels, audio_out_format_name(format), rate);
|
|
return 0;
|
|
}
|
|
|
|
bytes_per_sample = channels * info.play.precision / 8;
|
|
byte_per_sec = bytes_per_sample * rate;
|
|
ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192;
|
|
|
|
#ifdef __not_used__
|
|
/*
|
|
* hmm, ao_data.buffersize is currently not used in this driver, do there's
|
|
* no need to measure it
|
|
*/
|
|
if(ao_data.buffersize==-1){
|
|
// Measuring buffer size:
|
|
void* data;
|
|
ao_data.buffersize=0;
|
|
#ifdef HAVE_AUDIO_SELECT
|
|
data = malloc(ao_data.outburst);
|
|
memset(data, format==AFMT_U8 ? 0x80 : 0, ao_data.outburst);
|
|
while(ao_data.buffersize<0x40000){
|
|
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)) break;
|
|
write(audio_fd,data,ao_data.outburst);
|
|
ao_data.buffersize+=ao_data.outburst;
|
|
}
|
|
free(data);
|
|
if(ao_data.buffersize==0){
|
|
printf("\n *** Your audio driver DOES NOT support select() ***\n");
|
|
printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
|
|
return 0;
|
|
}
|
|
#ifdef __svr4__
|
|
// remove the 0 bytes from the above ao_data.buffersize measurement from the
|
|
// audio driver's STREAMS queue
|
|
ioctl(audio_fd, I_FLUSH, FLUSHW);
|
|
#endif
|
|
ioctl(audio_fd, AUDIO_DRAIN, 0);
|
|
#endif
|
|
}
|
|
#endif /* __not_used__ */
|
|
|
|
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;
|
|
}
|
|
|
|
// 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);
|
|
}
|
|
|
|
// stop playing and empty buffers (for seeking/pause)
|
|
static void reset(){
|
|
audio_info_t info;
|
|
|
|
uninit();
|
|
audio_fd=open(audio_dev, O_WRONLY);
|
|
if(audio_fd<0){
|
|
printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno));
|
|
return;
|
|
}
|
|
|
|
ioctl(audio_fd, AUDIO_DRAIN, 0);
|
|
|
|
AUDIO_INITINFO(&info);
|
|
info.play.encoding = oss2sunfmt(ao_data.format);
|
|
info.play.precision =
|
|
(ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_S16_BE
|
|
? AUDIO_PRECISION_16
|
|
: AUDIO_PRECISION_8);
|
|
info.play.channels = ao_data.channels;
|
|
info.play.sample_rate = ao_data.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)
|
|
static void audio_pause()
|
|
{
|
|
struct audio_info info;
|
|
AUDIO_INITINFO(&info);
|
|
info.play.pause = 1;
|
|
ioctl(audio_fd, AUDIO_SETINFO, &info);
|
|
}
|
|
|
|
// resume playing, after audio_pause()
|
|
static void audio_resume()
|
|
{
|
|
struct audio_info info;
|
|
AUDIO_INITINFO(&info);
|
|
info.play.pause = 0;
|
|
ioctl(audio_fd, AUDIO_SETINFO, &info);
|
|
}
|
|
|
|
|
|
// return: how many bytes can be played without blocking
|
|
static int get_space(){
|
|
int playsize = ao_data.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!
|
|
}
|
|
#endif
|
|
|
|
#ifndef __OpenBSD__
|
|
ioctl(audio_fd, AUDIO_GETINFO, &info);
|
|
if (queued_bursts - info.play.eof > 2)
|
|
return 0;
|
|
#endif
|
|
|
|
return ao_data.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){
|
|
#if WORDS_BIGENDIAN
|
|
int native_endian = AFMT_S16_BE;
|
|
#else
|
|
int native_endian = AFMT_S16_LE;
|
|
#endif
|
|
|
|
if (len < ao_data.outburst) return 0;
|
|
len /= ao_data.outburst;
|
|
len *= ao_data.outburst;
|
|
|
|
/* 16-bit format using the 'wrong' byteorder? swap words */
|
|
if ((ao_data.format == AFMT_S16_LE || ao_data.format == AFMT_S16_BE)
|
|
&& ao_data.format != native_endian) {
|
|
static void *swab_buf;
|
|
static int swab_len;
|
|
if (len > swab_len) {
|
|
if (swab_buf)
|
|
swab_buf = realloc(swab_buf, len);
|
|
else
|
|
swab_buf = malloc(len);
|
|
swab_len = len;
|
|
if (swab_buf == NULL) return 0;
|
|
}
|
|
swab(data, swab_buf, len);
|
|
data = swab_buf;
|
|
} else if (ao_data.format == AFMT_U8 && convert_u8_s8) {
|
|
int i;
|
|
unsigned char *p = data;
|
|
|
|
for (i = 0, p = data; i < len; i++, p++)
|
|
*p ^= 0x80;
|
|
}
|
|
|
|
len = write(audio_fd, data, len);
|
|
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;
|
|
}
|
|
|
|
|
|
// return: delay in seconds between first and last sample in buffer
|
|
static float get_delay(){
|
|
audio_info_t info;
|
|
ioctl(audio_fd, AUDIO_GETINFO, &info);
|
|
#ifdef __OpenBSD__
|
|
return (float) info.play.seek/ (float)byte_per_sec ;
|
|
#else
|
|
if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
|
|
return (float)(queued_samples - info.play.samples) / (float)byte_per_sec;
|
|
else
|
|
return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
|
|
#endif
|
|
}
|
|
|