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mpv/libao2/ao_sun.c

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#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 AUDIO_SWFEATURE_MIXER /* solaris8 or newer? */
# define HAVE_SYS_MIXER_H 1
#endif
#if HAVE_SYS_MIXER_H
# include <sys/mixer.h>
#endif
#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;
}
// match the requested sample rate |sample_rate| against the
// sample rates supported by the audio device |dev|. Return
// a supported sample rate, if that sample rate is close to
// (< 1% difference) the requested rate; return 0 otherwise.
#define MAX_RATE_ERR 1
static unsigned
find_close_samplerate_match(int dev, unsigned sample_rate)
{
#if HAVE_SYS_MIXER_H
am_sample_rates_t *sr;
unsigned i, num, err, best_err, best_rate;
for (num = 16; num < 1024; num *= 2) {
sr = malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(num));
if (!sr)
return 0;
sr->type = AUDIO_PLAY;
sr->flags = 0;
sr->num_samp_rates = num;
if (ioctl(dev, AUDIO_MIXER_GET_SAMPLE_RATES, sr)) {
free(sr);
return 0;
}
if (sr->num_samp_rates <= num)
break;
free(sr);
}
if (sr->flags & MIXER_SR_LIMITS) {
/*
* HW can playback any rate between
* sr->samp_rates[0] .. sr->samp_rates[1]
*/
free(sr);
return 0;
} else {
/* HW supports fixed sample rates only */
best_err = 65535;
best_rate = 0;
for (i = 0; i < sr->num_samp_rates; i++) {
err = abs(sr->samp_rates[i] - sample_rate);
if (err == 0) {
/*
* exact supported sample rate match, no need to
* retry something else
*/
best_rate = 0;
break;
}
if (err < best_err) {
best_err = err;
best_rate = sr->samp_rates[i];
}
}
free(sr);
if (best_rate > 0 && (100/MAX_RATE_ERR)*best_err < sample_rate) {
/* found a supported sample rate with <1% error? */
return best_rate;
}
return 0;
}
#else /* old audioio driver, cannot return list of supported rates */
/* XXX: hardcoded sample rates */
unsigned i, err;
unsigned audiocs_rates[] = {
5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050,
27420, 32000, 33075, 37800, 44100, 48000, 0
};
for (i = 0; audiocs_rates[i]; i++) {
err = abs(audiocs_rates[i] - sample_rate);
if (err == 0) {
/*
* exact supported sample rate match, no need to
* retry something elise
*/
return 0;
}
if ((100/MAX_RATE_ERR)*err < audiocs_rates[i]) {
/* <1% error? */
return audiocs_rates[i];
}
}
return 0;
#endif
}
// return the highest sample rate supported by audio device |dev|.
static unsigned
find_highest_samplerate(int dev)
{
#if HAVE_SYS_MIXER_H
am_sample_rates_t *sr;
unsigned i, num, max_rate;
for (num = 16; num < 1024; num *= 2) {
sr = malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(num));
if (!sr)
return 0;
sr->type = AUDIO_PLAY;
sr->flags = 0;
sr->num_samp_rates = num;
if (ioctl(dev, AUDIO_MIXER_GET_SAMPLE_RATES, sr)) {
free(sr);
return 0;
}
if (sr->num_samp_rates <= num)
break;
free(sr);
}
if (sr->flags & MIXER_SR_LIMITS) {
/*
* HW can playback any rate between
* sr->samp_rates[0] .. sr->samp_rates[1]
*/
max_rate = sr->samp_rates[1];
} else {
/* HW supports fixed sample rates only */
max_rate = 0;
for (i = 0; i < sr->num_samp_rates; i++) {
if (sr->samp_rates[i] > max_rate)
max_rate = sr->samp_rates[i];
}
}
free(sr);
return max_rate;
#else /* old audioio driver, cannot return list of supported rates */
return 44100; /* should be supported even on old ISA SB cards */
#endif
}
static void setup_device_paths()
{
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[0]) {
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;
}
// 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;
if ( !sun_mixer_device ) /* control function is used before init? */
setup_device_paths();
fd=open( sun_mixer_device,O_RDONLY );
if ( fd != -1 )
{
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
float volume;
struct audio_info info;
ioctl( fd,AUDIO_GETINFO,&info);
volume = info.play.gain * 100. / AUDIO_MAX_GAIN;
if ( info.play.balance == AUDIO_MID_BALANCE ) {
vol->right = vol->left = volume;
} else if ( info.play.balance < AUDIO_MID_BALANCE ) {
vol->left = volume;
vol->right = volume * info.play.balance / AUDIO_MID_BALANCE;
} else {
vol->left = volume * (AUDIO_RIGHT_BALANCE-info.play.balance)
/ AUDIO_MID_BALANCE;
vol->right = volume;
}
close( fd );
return CONTROL_OK;
}
return CONTROL_ERROR;
}
case AOCONTROL_SET_VOLUME:
{
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int fd;
if ( !sun_mixer_device ) /* control function is used before init? */
setup_device_paths();
fd=open( sun_mixer_device,O_RDONLY );
if ( fd != -1 )
{
struct audio_info info;
float volume;
AUDIO_INITINFO(&info);
volume = vol->right > vol->left ? vol->right : vol->left;
if ( volume != 0 ) {
info.play.gain = volume * AUDIO_MAX_GAIN / 100;
if ( vol->right == vol->left )
info.play.balance = AUDIO_MID_BALANCE;
else
info.play.balance = (vol->right - vol->left + volume) * AUDIO_RIGHT_BALANCE / (2*volume);
}
#if !defined (__OpenBSD__) && !defined (__NetBSD__)
info.output_muted = (volume == 0);
#endif
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 pass;
int ok;
setup_device_paths();
if (enable_sample_timing == RTSC_UNKNOWN
&& !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) {
enable_sample_timing = realtime_samplecounter_available(audio_dev);
}
#define AF_FILTER_TEST 0
#if AF_FILTER_TEST
/* test code to force use of the audio filter modules */
{
char *s;
if (s = getenv("AF_RATE"))
rate = atoi(s);
if (s = getenv("AF_CHANNELS"))
channels = atoi(s);
if (s = getenv("AF_BITS"))
format = atoi(s) == 16 ? AFMT_S16_NE : AFMT_U8;
}
#endif
// 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);
for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */
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;
if (pass & 1) {
/*
* on some sun audio drivers, 8-bit unsigned LINEAR8 encoding is
* not supported, but 8-bit signed encoding is.
*
* Try S8, and if it works, use our own U8->S8 conversion before
* sending the samples to the sound driver.
*/
if (info.play.encoding != AUDIO_ENCODING_LINEAR8)
continue;
info.play.encoding = AUDIO_ENCODING_LINEAR;
convert_u8_s8 = 1;
}
if (pass & 2) {
/*
* on some sun audio drivers, only certain fixed sample rates are
* supported.
*
* In case the requested sample rate is very close to one of the
* supported rates, use the fixed supported rate instead.
*/
if (!(info.play.sample_rate =
find_close_samplerate_match(audio_fd, rate)))
continue;
/*
* I'm not returning the correct sample rate in
* |ao_data.samplerate|, to avoid software resampling.
*
* ao_data.samplerate = info.play.sample_rate;
*/
}
if (pass & 4) {
/* like "pass & 2", but use the highest supported sample rate */
if (!(info.play.sample_rate
= ao_data.samplerate
= find_highest_samplerate(audio_fd)))
continue;
}
ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
if (ok) {
/* audio format accepted by audio driver */
break;
}
/*
* format not supported?
* retry with different encoding and/or sample rate
*/
}
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;
ao_data.bps = 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(){
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
#if !defined (__OpenBSD__) && !defined(__NetBSD__)
ioctl(audio_fd, AUDIO_GETINFO, &info);
if (queued_bursts - info.play.eof > 2)
return 0;
#endif
#if defined(__NetBSD__) || defined(__OpenBSD__)
ioctl(audio_fd, AUDIO_GETINFO, &info);
return info.hiwat * info.blocksize - info.play.seek;
#else
return ao_data.outburst;
#endif
}
// 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);
#if defined (__OpenBSD__) || defined(__NetBSD__)
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)ao_data.samplerate;
else
return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
#endif
}
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