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mpv/audio/out/ao_oss.c
wm4 4d3a2c7e0d audio/out: remove ao->outburst/buffersize fields 
The core didn't use these fields, and use of them was inconsistent
accross AOs. Some didn't use them at all. Some only set them; the values
were completely unused by the core. Some made full use of them.

Remove these fields. In places where they are still needed, make them
private AO state.

Remove the --abs option. It set the buffer size for ao_oss and ao_dsound
(being ignored by all other AOs), and was already marked as obsolete. If
it turns out that it's still needed for ao_oss or ao_dsound, their
default buffer sizes could be adjusted, and if even that doesn't help,
AO suboptions could be added in these cases.
2013-06-16 19:36:56 +02:00

591 lines
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/*
* OSS audio output driver
*
* 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
* (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 "core/options.h"
#include "core/mp_msg.h"
#include "audio/mixer.h"
#ifdef HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
#else
#ifdef HAVE_SOUNDCARD_H
#include <soundcard.h>
#endif
#endif
#include "audio/format.h"
#include "ao.h"
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;
};
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
{AFMT_S24_PACKED, AF_FORMAT_S24_LE},
#endif
#ifdef AFMT_U32_LE
{AFMT_U32_LE, AF_FORMAT_U32_LE},
#endif
#ifdef AFMT_U32_BE
{AFMT_U32_BE, AF_FORMAT_U32_BE},
#endif
#ifdef AFMT_S32_LE
{AFMT_S32_LE, AF_FORMAT_S32_LE},
#endif
#ifdef AFMT_S32_BE
{AFMT_S32_BE, AF_FORMAT_S32_BE},
#endif
#ifdef AFMT_FLOAT
{AFMT_FLOAT, AF_FORMAT_FLOAT_NE},
#endif
// SPECIALS
#ifdef AFMT_MPEG
{AFMT_MPEG, AF_FORMAT_MPEG2},
#endif
#ifdef 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));
return -1;
}
static int oss2format(int format)
{
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);
return -1;
}
#ifdef SNDCTL_DSP_GETPLAYVOL
static int volume_oss4(struct ao *ao, ao_control_vol_t *vol, int cmd)
{
struct priv *p = ao->priv;
int v;
if (p->audio_fd < 0)
return CONTROL_ERROR;
if (cmd == AOCONTROL_GET_VOLUME) {
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(p->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(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(ao, vol, cmd) == CONTROL_OK)
return CONTROL_OK;
#endif
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: 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);
}
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);
}
}
} else {
p->oss_mixer_channel = SOUND_MIXER_PCM;
}
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__
p->audio_fd = open(p->dsp, O_WRONLY | O_NONBLOCK);
#else
p->audio_fd = open(p->dsp, O_WRONLY);
#endif
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(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(p->audio_fd, F_SETFD, FD_CLOEXEC);
#endif
if (AF_FORMAT_IS_AC3(ao->format)) {
ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
}
ac3_retry:
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;
#else
oss_format = AFMT_S16_LE;
#endif
ao->format = AF_FORMAT_S16_NE;
}
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;
}
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);
}
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->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(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(p->audio_fd, SNDCTL_DSP_SYNC, NULL);
#endif
#ifdef SNDCTL_DSP_RESET
if (immed)
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
#endif
close(p->audio_fd);
p->audio_fd = -1;
}
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
close(p->audio_fd);
p->audio_fd = -1;
}
// stop playing and empty buffers (for seeking/pause)
static void reset(struct ao *ao)
{
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;
}
#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(struct ao *ao)
{
struct priv *p = ao->priv;
int playsize = p->outburst;
#ifdef SNDCTL_DSP_GETOSPACE
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(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 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(struct ao *ao, void *data, int len, int flags)
{
struct priv *p = ao->priv;
if (len == 0)
return len;
if (len > p->outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
len /= p->outburst;
len *= p->outburst;
}
len = write(p->audio_fd, data, len);
return len;
}
// 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(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(p->audio_fd, SNDCTL_DSP_GETODELAY, &r) != -1)
return ((float)r) / (float)ao->bps;
#endif
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,
};
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