static void
finalize (GObject *object)
{
GeglProperties *o = GEGL_PROPERTIES (object);
if (o->user_data)
{
Priv *p = (Priv*)o->user_data;
flush_audio (o);
flush_video (o);
av_write_trailer (p->oc);
if (p->video_st)
close_video (p, p->oc, p->video_st);
if (p->audio_st)
close_audio (p, p->oc, p->audio_st);
avio_closep (&p->oc->pb);
avformat_free_context (p->oc);
g_free (o->user_data);
o->user_data = NULL;
}
G_OBJECT_CLASS (g_type_class_peek_parent (G_OBJECT_GET_CLASS (object)))->finalize (object);
}
// close audio device
static void uninit(int immed){
// throw away buffered data in the audio driver's STREAMS queue
if (immed)
flush_audio(audio_fd);
else
ioctl(audio_fd, AUDIO_DRAIN, 0);
close(audio_fd);
}
// stop playing and empty buffers (for seeking/pause)
static void reset(void){
audio_info_t info;
flush_audio(audio_fd);
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;
}
// 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 ( mp_msg_test(MSGT_AO,MSGL_V) )
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed);
goto error;
}
if (write(fd, silence, len) != len) {
if ( mp_msg_test(MSGT_AO,MSGL_V) )
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed);
goto error;
}
if (ioctl(fd, AUDIO_GETINFO, &info)) {
if ( mp_msg_test(MSGT_AO,MSGL_V) )
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 ( mp_msg_test(MSGT_AO,MSGL_V) )
perror("rtsc: GETINFO2 failed");
goto error;
}
if (info.play.samples < last_samplecnt) {
if ( mp_msg_test(MSGT_AO,MSGL_V) )
mp_msg(MSGT_AO,MSGL_V,"rtsc: %d > %d?\n", last_samplecnt, info.play.samples);
goto error;
}
if ((increment = info.play.samples - last_samplecnt) > 0) {
if ( mp_msg_test(MSGT_AO,MSGL_V) )
mp_msg(MSGT_AO,MSGL_V,"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 ( mp_msg_test(MSGT_AO,MSGL_V) )
mp_msg(MSGT_AO,MSGL_V,"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) {
// remove the 0 bytes from the above measurement from the
// audio driver's STREAMS queue
flush_audio(fd);
close(fd);
}
return rtsc_ok;
}
