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-rw-r--r--audio/out/ao_oss.c786
1 files changed, 411 insertions, 375 deletions
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index fa8eccdeea..fa590f7965 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -3,6 +3,9 @@
*
* This file is part of MPlayer.
*
+ * Support for >2 output channels added 2001-11-25
+ * - Steve Davies <steve@daviesfam.org>
+ *
* 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
@@ -31,6 +34,7 @@
#include <string.h>
#include "config.h"
+#include "core/options.h"
#include "core/mp_msg.h"
#include "audio/mixer.h"
@@ -45,124 +49,95 @@
#include "audio/format.h"
#include "ao.h"
-#include "audio_out_internal.h"
-static const ao_info_t info =
-{
- "OSS/ioctl audio output",
- "oss",
- "A'rpi",
- ""
+struct priv {
+ char *dsp;
+ int audio_fd;
+ int prepause_space;
+ const char *oss_mixer_device;
+ int oss_mixer_channel;
+ audio_buf_info zz;
+ int audio_delay_method;
+ int buffersize;
+ int outburst;
};
-/* 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;
+static int format_table[][2] = {
+ {AFMT_U8, AF_FORMAT_U8},
+ {AFMT_S8, AF_FORMAT_S8},
+ {AFMT_U16_LE, AF_FORMAT_U16_LE},
+ {AFMT_U16_BE, AF_FORMAT_U16_BE},
+ {AFMT_S16_LE, AF_FORMAT_S16_LE},
+ {AFMT_S16_BE, AF_FORMAT_S16_BE},
#ifdef AFMT_S24_PACKED
- case AF_FORMAT_S24_LE: return AFMT_S24_PACKED;
+ {AFMT_S24_PACKED, AF_FORMAT_S24_LE},
#endif
#ifdef AFMT_U32_LE
- case AF_FORMAT_U32_LE: return AFMT_U32_LE;
+ {AFMT_U32_LE, AF_FORMAT_U32_LE},
#endif
#ifdef AFMT_U32_BE
- case AF_FORMAT_U32_BE: return AFMT_U32_BE;
+ {AFMT_U32_BE, AF_FORMAT_U32_BE},
#endif
#ifdef AFMT_S32_LE
- case AF_FORMAT_S32_LE: return AFMT_S32_LE;
+ {AFMT_S32_LE, AF_FORMAT_S32_LE},
#endif
#ifdef AFMT_S32_BE
- case AF_FORMAT_S32_BE: return AFMT_S32_BE;
+ {AFMT_S32_BE, AF_FORMAT_S32_BE},
#endif
#ifdef AFMT_FLOAT
- case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
+ {AFMT_FLOAT, AF_FORMAT_FLOAT_NE},
#endif
// SPECIALS
#ifdef AFMT_MPEG
- case AF_FORMAT_MPEG2: return AFMT_MPEG;
+ {AFMT_MPEG, AF_FORMAT_MPEG2},
#endif
#ifdef AFMT_AC3
- case AF_FORMAT_AC3_NE: return AFMT_AC3;
+ {AFMT_AC3, AF_FORMAT_AC3_NE},
#endif
+ {-1, -1}
+};
+
+static int format2oss(int format)
+{
+ for (int n = 0; format_table[n][0] != -1; n++) {
+ if (format_table[n][1] == format)
+ return format_table[n][0];
}
- mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format));
+ 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
-#ifdef AFMT_MPEG
- case AFMT_MPEG: return AF_FORMAT_MPEG2;
-#endif
-#ifdef AFMT_AC3
- case AFMT_AC3: return AF_FORMAT_AC3_NE;
-#endif
+ for (int n = 0; format_table[n][0] != -1; n++) {
+ if (format_table[n][0] == format)
+ return format_table[n][1];
}
- mp_tmsg(MSGT_GLOBAL,MSGL_ERR,"[AO OSS] Unknown/Unsupported OSS format: %x.\n", format);
+ 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) {
+static int volume_oss4(struct ao *ao, ao_control_vol_t *vol, int cmd)
+{
+ struct priv *p = ao->priv;
int v;
- if (audio_fd < 0)
+ if (p->audio_fd < 0)
return CONTROL_ERROR;
if (cmd == AOCONTROL_GET_VOLUME) {
- if (ioctl(audio_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1)
+ if (ioctl(p->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)
+ if (ioctl(p->audio_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1)
return CONTROL_ERROR;
return CONTROL_OK;
} else
@@ -171,384 +146,445 @@ static int volume_oss4(ao_control_vol_t *vol, int cmd) {
#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;
+static int control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+ struct priv *p = ao->priv;
+ 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)
+ if (volume_oss4(ao, 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;
+ if (AF_FORMAT_IS_AC3(ao->format))
+ return CONTROL_TRUE;
+
+ if ((fd = open(p->oss_mixer_device, O_RDONLY)) != -1) {
+ ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
+ if (devs & (1 << p->oss_mixer_channel)) {
+ if (cmd == AOCONTROL_GET_VOLUME) {
+ ioctl(fd, MIXER_READ(p->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(p->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';
+// return: 0=success -1=fail
+static int init(struct ao *ao, char *params)
+{
+ struct MPOpts *opts = ao->opts;
+ char *mixer_channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
+ int oss_format;
+ char *mdev = opts->mixer_device, *mchan = opts->mixer_channel;
+
+ mp_msg(MSGT_AO, MSGL_V, "ao2: %d Hz %d chans %s\n", ao->samplerate,
+ ao->channels.num, af_fmt2str_short(ao->format));
+
+ struct priv *p = talloc(ao, struct priv);
+ *p = (struct priv) {
+ .dsp = PATH_DEV_DSP,
+ .audio_fd = -1,
+ .oss_mixer_device = mdev ? mdev : PATH_DEV_MIXER,
+ .oss_mixer_channel = SOUND_MIXER_PCM,
+ .audio_delay_method = 2,
+ .buffersize = -1,
+ .outburst = 512,
+ };
+ ao->priv = p;
+
+ if (params) {
+ char *m, *c;
+ m = strchr(params, ':');
+ if (m) {
+ c = strchr(m + 1, ':');
+ if (c) {
+ mchan = c + 1;
+ c[0] = '\0';
+ }
+ mdev = m + 1;
+ m[0] = '\0';
+ }
+ p->dsp = talloc_strdup(ao, params);
}
- 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 (mchan) {
+ int fd, devs, i;
+
+ if ((fd = open(p->oss_mixer_device, O_RDONLY)) == -1) {
+ mp_tmsg(MSGT_AO, MSGL_ERR,
+ "[AO OSS] audio_setup: Can't open mixer device %s: %s\n",
+ p->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;
+ }
+ p->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);
+ }
}
- }
- 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 {
+ p->oss_mixer_channel = SOUND_MIXER_PCM;
}
- } 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]);
+ mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' dsp device\n", p->dsp);
+ mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' mixer device\n",
+ p->oss_mixer_device);
+ mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' mixer device\n",
+ mixer_channels[p->oss_mixer_channel]);
#ifdef __linux__
- audio_fd=open(dsp, O_WRONLY | O_NONBLOCK);
+ p->audio_fd = open(p->dsp, O_WRONLY | O_NONBLOCK);
#else
- audio_fd=open(dsp, O_WRONLY);
+ p->audio_fd = open(p->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;
- }
+ if (p->audio_fd < 0) {
+ mp_tmsg(MSGT_AO, MSGL_ERR,
+ "[AO OSS] audio_setup: Can't open audio device %s: %s\n",
+ p->dsp, strerror(errno));
+ return -1;
+ }
#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;
- }
+ /* Remove the non-blocking flag */
+ if (fcntl(p->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 -1;
+ }
#endif
#if defined(FD_CLOEXEC) && defined(F_SETFD)
- fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
+ fcntl(p->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);
- }
+ if (AF_FORMAT_IS_AC3(ao->format)) {
+ ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->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 (AF_FORMAT_IS_AC3(ao->format))
+ ao->format = AF_FORMAT_AC3_NE;
+ oss_format = format2oss(ao->format);
+ if (oss_format == -1) {
#if BYTE_ORDER == BIG_ENDIAN
- oss_format=AFMT_S16_BE;
+ 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));
+ oss_format = AFMT_S16_LE;
#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;
- }
+ ao->format = AF_FORMAT_S16_NE;
}
- 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;
+ if (ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &oss_format) < 0 ||
+ oss_format != format2oss(ao->format))
+ {
+ mp_tmsg(MSGT_AO, MSGL_WARN, "[AO OSS] Can't set audio device %s to %s "
+ "output, trying %s...\n", p->dsp, af_fmt2str_short(ao->format),
+ af_fmt2str_short(AF_FORMAT_S16_NE));
+ ao->format = AF_FORMAT_S16_NE;
+ goto ac3_retry;
}
- 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;
+
+ ao->format = oss2format(oss_format);
+ if (ao->format == -1)
+ return -1;
+
+ mp_msg(MSGT_AO, MSGL_V, "audio_setup: sample format: %s\n",
+ af_fmt2str_short(ao->format));
+
+ if (!AF_FORMAT_IS_AC3(ao->format)) {
+ struct mp_chmap_sel sel = {0};
+ mp_chmap_sel_add_alsa_def(&sel);
+ if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+ return -1;
+ int reqchannels = ao->channels.num;
+ // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
+ if (reqchannels > 2) {
+ int nchannels = reqchannels;
+ if (ioctl(p->audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1 ||
+ nchannels != reqchannels)
+ {
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Failed to "
+ "set audio device to %d channels.\n", reqchannels);
+ return -1;
+ }
+ } else {
+ int c = reqchannels - 1;
+ if (ioctl(p->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", reqchannels);
+ return -1;
+ }
+ if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, c + 1))
+ return -1;
+ }
+ mp_msg(MSGT_AO, MSGL_V,
+ "audio_setup: using %d channels (requested: %d)\n",
+ ao->channels.num, reqchannels);
+ // set rate
+ ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
+ mp_msg(MSGT_AO, MSGL_V, "audio_setup: using %d Hz samplerate\n",
+ ao->samplerate);
}
- 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;
+
+ if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &p->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(p->audio_fd, SNDCTL_DSP_GETBLKSIZE, &r) == -1)
+ mp_msg(MSGT_AO, MSGL_V, "audio_setup: %d bytes/frag (config.h)\n",
+ p->outburst);
+ else {
+ p->outburst = r;
+ mp_msg(MSGT_AO, MSGL_V, "audio_setup: %d bytes/frag (GETBLKSIZE)\n",
+ p->outburst);
+ }
+ } else {
+ mp_msg(MSGT_AO, MSGL_V,
+ "audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
+ p->zz.fragments, p->zz.fragstotal, p->zz.fragsize, p->zz.bytes);
+ p->buffersize = p->zz.bytes;
+ p->outburst = p->zz.fragsize;
}
+
+ if (p->buffersize == -1) {
+ // Measuring buffer size:
+ void *data;
+ p->buffersize = 0;
+#ifdef HAVE_AUDIO_SELECT
+ data = malloc(p->outburst);
+ memset(data, 0, p->outburst);
+ while (p->buffersize < 0x40000) {
+ fd_set rfds;
+ struct timeval tv;
+ FD_ZERO(&rfds);
+ FD_SET(p->audio_fd, &rfds);
+ tv.tv_sec = 0;
+ tv.tv_usec = 0;
+ if (!select(p->audio_fd + 1, NULL, &rfds, NULL, &tv))
+ break;
+ write(p->audio_fd, data, p->outburst);
+ p->buffersize += p->outburst;
+ }
+ free(data);
+ if (p->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 -1;
+ }
#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;
+ }
+
+ ao->bps = ao->channels.num * (af_fmt2bits(ao->format) / 8);
+ p->outburst -= p->outburst % ao->bps; // round down
+ ao->bps *= ao->samplerate;
+
+ return 0;
}
// close audio device
-static void uninit(int immed){
- if(audio_fd == -1) return;
+static void uninit(struct ao *ao, bool immed)
+{
+ struct priv *p = ao->priv;
+ if (p->audio_fd == -1)
+ return;
#ifdef SNDCTL_DSP_SYNC
// to get the buffer played
if (!immed)
- ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL);
+ ioctl(p->audio_fd, SNDCTL_DSP_SYNC, NULL);
#endif
#ifdef SNDCTL_DSP_RESET
if (immed)
- ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
+ ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
#endif
- close(audio_fd);
- audio_fd = -1;
+ close(p->audio_fd);
+ p->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);
+static void close_device(struct ao *ao)
+{
+ struct priv *p = ao->priv;
+#ifdef SNDCTL_DSP_RESET
+ ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
#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);
- }
+ close(p->audio_fd);
+ p->audio_fd = -1;
}
-// stop playing, keep buffers (for pause)
-static void audio_pause(void)
+// stop playing and empty buffers (for seeking/pause)
+static void reset(struct ao *ao)
{
- prepause_space = get_space();
- uninit(1);
-}
+ struct priv *p = ao->priv;
+ int oss_format;
+ close_device(ao);
+ p->audio_fd = open(p->dsp, O_WRONLY);
+ if (p->audio_fd < 0) {
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS]\nFatal error: *** CANNOT "
+ "RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno));
+ return;
+ }
-// 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);
+#if defined(FD_CLOEXEC) && defined(F_SETFD)
+ fcntl(p->audio_fd, F_SETFD, FD_CLOEXEC);
+#endif
+
+ oss_format = format2oss(ao->format);
+ if (AF_FORMAT_IS_AC3(ao->format))
+ ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
+ ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
+ if (!AF_FORMAT_IS_AC3(ao->format)) {
+ if (ao->channels.num > 2)
+ ioctl(p->audio_fd, SNDCTL_DSP_CHANNELS, &ao->channels.num);
+ else {
+ int c = ao->channels.num - 1;
+ ioctl(p->audio_fd, SNDCTL_DSP_STEREO, &c);
+ }
+ ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
}
}
-
// return: how many bytes can be played without blocking
-static int get_space(void){
- int playsize=ao_data.outburst;
+static int get_space(struct ao *ao)
+{
+ struct priv *p = ao->priv;
+ int playsize = p->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;
- }
+ if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &p->zz) != -1) {
+ // calculate exact buffer space:
+ playsize = p->zz.fragments * p->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!
+ {
+ fd_set rfds;
+ struct timeval tv;
+ FD_ZERO(&rfds);
+ FD_SET(p->audio_fd, &rfds);
+ tv.tv_sec = 0;
+ tv.tv_usec = 0;
+ if (!select(p->audio_fd + 1, NULL, &rfds, NULL, &tv))
+ return 0; // not block!
}
#endif
- return ao_data.outburst;
+ return p->outburst;
+}
+
+// stop playing, keep buffers (for pause)
+static void audio_pause(struct ao *ao)
+{
+ struct priv *p = ao->priv;
+ p->prepause_space = get_space(ao);
+ close_device(ao);
}
// 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)
+static int play(struct ao *ao, void *data, int len, int flags)
+{
+ struct priv *p = ao->priv;
+ if (len == 0)
return len;
- if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
- len/=ao_data.outburst;
- len*=ao_data.outburst;
+ if (len > p->outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
+ len /= p->outburst;
+ len *= p->outburst;
}
- len=write(audio_fd,data,len);
+ len = write(p->audio_fd, data, len);
return len;
}
-static int audio_delay_method=2;
+// resume playing, after audio_pause()
+static void audio_resume(struct ao *ao)
+{
+ struct priv *p = ao->priv;
+ int fillcnt;
+ reset(ao);
+ fillcnt = get_space(ao) - p->prepause_space;
+ if (fillcnt > 0 && !(ao->format & AF_FORMAT_SPECIAL_MASK)) {
+ void *silence = calloc(fillcnt, 1);
+ play(ao, silence, fillcnt, 0);
+ free(silence);
+ }
+}
// 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){
+static float get_delay(struct ao *ao)
+{
+ struct priv *p = ao->priv;
+ /* Calculate how many bytes/second is sent out */
+ if (p->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;
+ int r = 0;
+ if (ioctl(p->audio_fd, SNDCTL_DSP_GETODELAY, &r) != -1)
+ return ((float)r) / (float)ao->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;
+ p->audio_delay_method = 1; // fallback if not supported
+ }
+ if (p->audio_delay_method == 1) {
+ // SNDCTL_DSP_GETOSPACE
+ if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &p->zz) != -1) {
+ return ((float)(p->buffersize -
+ p->zz.bytes)) / (float)ao->bps;
+ }
+ p->audio_delay_method = 0; // fallback if not supported
+ }
+ return ((float)p->buffersize) / (float)ao->bps;
}
+
+const struct ao_driver audio_out_oss = {
+ .info = &(const struct ao_info) {
+ "OSS/ioctl audio output",
+ "oss",
+ "A'rpi",
+ ""
+ },
+ .init = init,
+ .uninit = uninit,
+ .control = control,
+ .get_space = get_space,
+ .play = play,
+ .get_delay = get_delay,
+ .pause = audio_pause,
+ .resume = audio_resume,
+ .reset = reset,
+};
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-29 05:13:39 +0000