RepoMirrors
/
mpv
mirror of https://github.com/mpv-player/mpv
1
0
Fork 0
Code Issues Releases Wiki Activity
mpv/libao2/ao_sun.c

506 lines
13 KiB
C
Raw Blame History

#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="/dev/audioctl";
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 )
{
struct audio_info info;
ioctl( fd,AUDIO_GETINFO,&info);
((ao_control_vol_t*)(arg))->left=info.play.gain * 100. / AUDIO_MAX_GAIN;
((ao_control_vol_t*)(arg))->=info.play.gain * 100. / AUDIO_MAX_GAIN;
close( fd );
return CONTROL_OK;
}
return CONTROL_ERROR;
}
case AOCONTROL_SET_VOLUME:
{
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 = (r+l) * 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(mixer_device)
sun_mixer_device=mixer_device;
if (audio_dev == NULL) {
if ((audio_dev = getenv("AUDIODEV")) == NULL)
audio_dev = "/dev/audio";
}
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
ioctl(audio_fd, AUDIO_GETINFO, &info);
if (queued_bursts - info.play.eof > 2)
return 0;
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);
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;
}
Reference in New Issue View Git Blame Copy Permalink