summaryrefslogtreecommitdiffstats
path: root/audio/out/ao_oss.c
diff options
context:
space:
mode:
authorwm4 <wm4@nowhere>2012-11-05 17:02:04 +0100
committerwm4 <wm4@nowhere>2012-11-12 20:06:14 +0100
commitd4bdd0473d6f43132257c9fb3848d829755167a3 (patch)
tree8021c2f7da1841393c8c832105e20cd527826d6c /audio/out/ao_oss.c
parentbd48deba77bd5582c5829d6fe73a7d2571088aba (diff)
downloadmpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.bz2
mpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.xz
Rename directories, move files (step 1 of 2) (does not compile)
Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code.
Diffstat (limited to 'audio/out/ao_oss.c')
-rw-r--r--audio/out/ao_oss.c560
1 files changed, 560 insertions, 0 deletions
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
new file mode 100644
index 0000000000..9d4dde4837
--- /dev/null
+++ b/audio/out/ao_oss.c
@@ -0,0 +1,560 @@
+/*
+ * OSS audio output driver
+ *
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <sys/time.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <string.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "mixer.h"
+
+#ifdef HAVE_SYS_SOUNDCARD_H
+#include <sys/soundcard.h>
+#else
+#ifdef HAVE_SOUNDCARD_H
+#include <soundcard.h>
+#endif
+#endif
+
+#include "libaf/format.h"
+
+#include "audio_out.h"
+#include "audio_out_internal.h"
+
+static const ao_info_t info =
+{
+ "OSS/ioctl audio output",
+ "oss",
+ "A'rpi",
+ ""
+};
+
+/* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */
+
+LIBAO_EXTERN(oss)
+
+static int format2oss(int format)
+{
+ switch(format)
+ {
+ case AF_FORMAT_U8: return AFMT_U8;
+ case AF_FORMAT_S8: return AFMT_S8;
+ case AF_FORMAT_U16_LE: return AFMT_U16_LE;
+ case AF_FORMAT_U16_BE: return AFMT_U16_BE;
+ case AF_FORMAT_S16_LE: return AFMT_S16_LE;
+ case AF_FORMAT_S16_BE: return AFMT_S16_BE;
+#ifdef AFMT_S24_PACKED
+ case AF_FORMAT_S24_LE: return AFMT_S24_PACKED;
+#endif
+#ifdef AFMT_U32_LE
+ case AF_FORMAT_U32_LE: return AFMT_U32_LE;
+#endif
+#ifdef AFMT_U32_BE
+ case AF_FORMAT_U32_BE: return AFMT_U32_BE;
+#endif
+#ifdef AFMT_S32_LE
+ case AF_FORMAT_S32_LE: return AFMT_S32_LE;
+#endif
+#ifdef AFMT_S32_BE
+ case AF_FORMAT_S32_BE: return AFMT_S32_BE;
+#endif
+#ifdef AFMT_FLOAT
+ case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
+#endif
+ // SPECIALS
+ case AF_FORMAT_MU_LAW: return AFMT_MU_LAW;
+ case AF_FORMAT_A_LAW: return AFMT_A_LAW;
+ case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM;
+#ifdef AFMT_MPEG
+ case AF_FORMAT_MPEG2: return AFMT_MPEG;
+#endif
+#ifdef AFMT_AC3
+ case AF_FORMAT_AC3_NE: return AFMT_AC3;
+#endif
+ }
+ mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format));
+ return -1;
+}
+
+static int oss2format(int format)
+{
+ switch(format)
+ {
+ case AFMT_U8: return AF_FORMAT_U8;
+ case AFMT_S8: return AF_FORMAT_S8;
+ case AFMT_U16_LE: return AF_FORMAT_U16_LE;
+ case AFMT_U16_BE: return AF_FORMAT_U16_BE;
+ case AFMT_S16_LE: return AF_FORMAT_S16_LE;
+ case AFMT_S16_BE: return AF_FORMAT_S16_BE;
+#ifdef AFMT_S24_PACKED
+ case AFMT_S24_PACKED: return AF_FORMAT_S24_LE;
+#endif
+#ifdef AFMT_U32_LE
+ case AFMT_U32_LE: return AF_FORMAT_U32_LE;
+#endif
+#ifdef AFMT_U32_BE
+ case AFMT_U32_BE: return AF_FORMAT_U32_BE;
+#endif
+#ifdef AFMT_S32_LE
+ case AFMT_S32_LE: return AF_FORMAT_S32_LE;
+#endif
+#ifdef AFMT_S32_BE
+ case AFMT_S32_BE: return AF_FORMAT_S32_BE;
+#endif
+#ifdef AFMT_FLOAT
+ case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE;
+#endif
+ // SPECIALS
+ case AFMT_MU_LAW: return AF_FORMAT_MU_LAW;
+ case AFMT_A_LAW: return AF_FORMAT_A_LAW;
+ case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM;
+#ifdef AFMT_MPEG
+ case AFMT_MPEG: return AF_FORMAT_MPEG2;
+#endif
+#ifdef AFMT_AC3
+ case AFMT_AC3: return AF_FORMAT_AC3_NE;
+#endif
+ }
+ mp_tmsg(MSGT_GLOBAL,MSGL_ERR,"[AO OSS] Unknown/Unsupported OSS format: %x.\n", format);
+ return -1;
+}
+
+static char *dsp=PATH_DEV_DSP;
+static audio_buf_info zz;
+static int audio_fd=-1;
+static int prepause_space;
+
+static const char *oss_mixer_device = PATH_DEV_MIXER;
+static int oss_mixer_channel = SOUND_MIXER_PCM;
+
+#ifdef SNDCTL_DSP_GETPLAYVOL
+static int volume_oss4(ao_control_vol_t *vol, int cmd) {
+ int v;
+
+ if (audio_fd < 0)
+ return CONTROL_ERROR;
+
+ if (cmd == AOCONTROL_GET_VOLUME) {
+ if (ioctl(audio_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1)
+ return CONTROL_ERROR;
+ vol->right = (v & 0xff00) >> 8;
+ vol->left = v & 0x00ff;
+ return CONTROL_OK;
+ } else if (cmd == AOCONTROL_SET_VOLUME) {
+ v = ((int) vol->right << 8) | (int) vol->left;
+ if (ioctl(audio_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1)
+ return CONTROL_ERROR;
+ return CONTROL_OK;
+ } else
+ return CONTROL_UNKNOWN;
+}
+#endif
+
+// to set/get/query special features/parameters
+static int control(int cmd,void *arg){
+ switch(cmd){
+ case AOCONTROL_GET_VOLUME:
+ case AOCONTROL_SET_VOLUME:
+ {
+ ao_control_vol_t *vol = (ao_control_vol_t *)arg;
+ int fd, v, devs;
+
+#ifdef SNDCTL_DSP_GETPLAYVOL
+ // Try OSS4 first
+ if (volume_oss4(vol, cmd) == CONTROL_OK)
+ return CONTROL_OK;
+#endif
+
+ if(AF_FORMAT_IS_AC3(ao_data.format))
+ return CONTROL_TRUE;
+
+ if ((fd = open(oss_mixer_device, O_RDONLY)) != -1)
+ {
+ ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
+ if (devs & (1 << oss_mixer_channel))
+ {
+ if (cmd == AOCONTROL_GET_VOLUME)
+ {
+ ioctl(fd, MIXER_READ(oss_mixer_channel), &v);
+ vol->right = (v & 0xFF00) >> 8;
+ vol->left = v & 0x00FF;
+ }
+ else
+ {
+ v = ((int)vol->right << 8) | (int)vol->left;
+ ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v);
+ }
+ }
+ else
+ {
+ close(fd);
+ return CONTROL_ERROR;
+ }
+ 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){
+ char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
+ int oss_format;
+ char *mdev = mixer_device, *mchan = mixer_channel;
+
+ mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels,
+ af_fmt2str_short(format));
+
+ if (ao_subdevice) {
+ char *m,*c;
+ m = strchr(ao_subdevice,':');
+ if(m) {
+ c = strchr(m+1,':');
+ if(c) {
+ mchan = c+1;
+ c[0] = '\0';
+ }
+ mdev = m+1;
+ m[0] = '\0';
+ }
+ dsp = ao_subdevice;
+ }
+
+ if(mdev)
+ oss_mixer_device=mdev;
+ else
+ oss_mixer_device=PATH_DEV_MIXER;
+
+ if(mchan){
+ int fd, devs, i;
+
+ if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open mixer device %s: %s\n",
+ oss_mixer_device, strerror(errno));
+ }else{
+ ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
+ close(fd);
+
+ for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
+ if(!strcasecmp(mixer_channels[i], mchan)){
+ if(!(devs & (1 << i))){
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Audio card mixer does not have channel '%s', using default.\n",mchan);
+ i = SOUND_MIXER_NRDEVICES+1;
+ break;
+ }
+ oss_mixer_channel = i;
+ break;
+ }
+ }
+ if(i==SOUND_MIXER_NRDEVICES){
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Audio card mixer does not have channel '%s', using default.\n",mchan);
+ }
+ }
+ } else
+ oss_mixer_channel = SOUND_MIXER_PCM;
+
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp);
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", oss_mixer_device);
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", mixer_channels[oss_mixer_channel]);
+
+#ifdef __linux__
+ audio_fd=open(dsp, O_WRONLY | O_NONBLOCK);
+#else
+ audio_fd=open(dsp, O_WRONLY);
+#endif
+ if(audio_fd<0){
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open audio device %s: %s\n", dsp, strerror(errno));
+ return 0;
+ }
+
+#ifdef __linux__
+ /* Remove the non-blocking flag */
+ if(fcntl(audio_fd, F_SETFL, 0) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't make file descriptor blocking: %s\n", strerror(errno));
+ return 0;
+ }
+#endif
+
+#if defined(FD_CLOEXEC) && defined(F_SETFD)
+ fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
+#endif
+
+ if(AF_FORMAT_IS_AC3(format)) {
+ ao_data.samplerate=rate;
+ ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
+ }
+
+ac3_retry:
+ if (AF_FORMAT_IS_AC3(format))
+ format = AF_FORMAT_AC3_NE;
+ ao_data.format=format;
+ oss_format=format2oss(format);
+ if (oss_format == -1) {
+#if BYTE_ORDER == BIG_ENDIAN
+ oss_format=AFMT_S16_BE;
+#else
+ oss_format=AFMT_S16_LE;
+#endif
+ format=AF_FORMAT_S16_NE;
+ }
+ if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 ||
+ oss_format != format2oss(format)) {
+ mp_tmsg(MSGT_AO,MSGL_WARN, "[AO OSS] Can't set audio device %s to %s output, trying %s...\n", dsp,
+ af_fmt2str_short(format), af_fmt2str_short(AF_FORMAT_S16_NE) );
+ format=AF_FORMAT_S16_NE;
+ goto ac3_retry;
+ }
+#if 0
+ if(oss_format!=format2oss(format))
+ mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %s sample format! Broken audio or bad playback speed are possible! Try with '-af format'\n",audio_out_format_name(format));
+#endif
+
+ ao_data.format = oss2format(oss_format);
+ if (ao_data.format == -1) return 0;
+
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
+ af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
+
+ ao_data.channels = channels;
+ if(!AF_FORMAT_IS_AC3(format)) {
+ // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
+ if (ao_data.channels > 2) {
+ if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
+ ao_data.channels != channels ) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", channels);
+ return 0;
+ }
+ }
+ else {
+ int c = ao_data.channels-1;
+ if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", ao_data.channels);
+ return 0;
+ }
+ ao_data.channels=c+1;
+ }
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
+ // set rate
+ ao_data.samplerate=rate;
+ ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
+ }
+
+ if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
+ int r=0;
+ mp_tmsg(MSGT_AO,MSGL_WARN,"[AO OSS] audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
+ if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
+ } else {
+ ao_data.outburst=r;
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
+ }
+ } else {
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
+ zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
+ if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
+ ao_data.outburst=zz.fragsize;
+ }
+
+ 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,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){
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\n *** Your audio driver DOES NOT support select() ***\n Recompile mpv with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
+ return 0;
+ }
+#endif
+ }
+
+ ao_data.bps=ao_data.channels;
+ switch (ao_data.format & AF_FORMAT_BITS_MASK) {
+ case AF_FORMAT_8BIT:
+ break;
+ case AF_FORMAT_16BIT:
+ ao_data.bps*=2;
+ break;
+ case AF_FORMAT_24BIT:
+ ao_data.bps*=3;
+ break;
+ case AF_FORMAT_32BIT:
+ ao_data.bps*=4;
+ break;
+ }
+
+ ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
+ ao_data.bps*=ao_data.samplerate;
+
+ return 1;
+}
+
+// close audio device
+static void uninit(int immed){
+ if(audio_fd == -1) return;
+#ifdef SNDCTL_DSP_SYNC
+ // to get the buffer played
+ if (!immed)
+ ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL);
+#endif
+#ifdef SNDCTL_DSP_RESET
+ if (immed)
+ ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
+#endif
+ close(audio_fd);
+ audio_fd = -1;
+}
+
+// stop playing and empty buffers (for seeking/pause)
+static void reset(void){
+ int oss_format;
+ uninit(1);
+ audio_fd=open(dsp, O_WRONLY);
+ if(audio_fd < 0){
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno));
+ return;
+ }
+
+#if defined(FD_CLOEXEC) && defined(F_SETFD)
+ fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
+#endif
+
+ oss_format = format2oss(ao_data.format);
+ if(AF_FORMAT_IS_AC3(ao_data.format))
+ ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
+ ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
+ if(!AF_FORMAT_IS_AC3(ao_data.format)) {
+ if (ao_data.channels > 2)
+ ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
+ else {
+ int c = ao_data.channels-1;
+ ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
+ }
+ ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
+ }
+}
+
+// stop playing, keep buffers (for pause)
+static void audio_pause(void)
+{
+ prepause_space = get_space();
+ uninit(1);
+}
+
+// resume playing, after audio_pause()
+static void audio_resume(void)
+{
+ int fillcnt;
+ reset();
+ fillcnt = get_space() - prepause_space;
+ if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) {
+ void *silence = calloc(fillcnt, 1);
+ play(silence, fillcnt, 0);
+ free(silence);
+ }
+}
+
+
+// return: how many bytes can be played without blocking
+static int get_space(void){
+ int playsize=ao_data.outburst;
+
+#ifdef SNDCTL_DSP_GETOSPACE
+ if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
+ // calculate exact buffer space:
+ playsize = zz.fragments*zz.fragsize;
+ return playsize;
+ }
+#endif
+
+ // 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
+
+ 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(len==0)
+ return len;
+ if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
+ len/=ao_data.outburst;
+ len*=ao_data.outburst;
+ }
+ len=write(audio_fd,data,len);
+ return len;
+}
+
+static int audio_delay_method=2;
+
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(void){
+ /* Calculate how many bytes/second is sent out */
+ if(audio_delay_method==2){
+#ifdef SNDCTL_DSP_GETODELAY
+ int r=0;
+ if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
+ return ((float)r)/(float)ao_data.bps;
+#endif
+ audio_delay_method=1; // fallback if not supported
+ }
+ if(audio_delay_method==1){
+ // SNDCTL_DSP_GETOSPACE
+ if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
+ return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
+ audio_delay_method=0; // fallback if not supported
+ }
+ return ((float)ao_data.buffersize)/(float)ao_data.bps;
+}