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mpv/audio/out/ao_oss.c

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/*
* OSS audio output driver
*
* This file is part of MPlayer.
*
* Original author: A'rpi
* 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 "mpvcore/options.h"
#include "mpvcore/mp_msg.h"
#if HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
#else
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#endif
#endif
#include "audio/format.h"
#include "ao.h"
struct priv {
int audio_fd;
int prepause_space;
int oss_mixer_channel;
audio_buf_info zz;
int audio_delay_method;
int buffersize;
int outburst;
char *dsp;
char *oss_mixer_device;
char *cfg_oss_mixer_channel;
};
static const char *mixer_channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
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},
#endif
// SPECIALS
#ifdef AFMT_MPEG
{AFMT_MPEG, AF_FORMAT_MPEG2},
#endif
#ifdef AFMT_AC3
{AFMT_AC3, AF_FORMAT_AC3},
#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];
}
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];
}
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)
{
struct priv *p = ao->priv;
int oss_format;
#ifdef SNDCTL_DSP_GETPLAYVOL
ao->no_persistent_volume = true;
#endif
const char *mchan = NULL;
if (p->cfg_oss_mixer_channel && p->cfg_oss_mixer_channel[0])
mchan = p->cfg_oss_mixer_channel;
if (mchan) {
int fd, devs, i;
if ((fd = open(p->oss_mixer_device, O_RDONLY)) == -1) {
MP_ERR(ao, "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_ERR(ao, "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_ERR(ao, "Audio card mixer does not have "
"channel '%s', using default.\n", mchan);
}
}
} else {
p->oss_mixer_channel = SOUND_MIXER_PCM;
}
MP_VERBOSE(ao, "using '%s' dsp device\n", p->dsp);
MP_VERBOSE(ao, "using '%s' mixer device\n", p->oss_mixer_device);
MP_VERBOSE(ao, "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_ERR(ao, "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_ERR(ao, "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
ao->format = af_fmt_from_planar(ao->format);
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;
oss_format = format2oss(ao->format);
if (oss_format == -1) {
MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n",
af_fmt_to_str(ao->format));
#if BYTE_ORDER == BIG_ENDIAN
oss_format = AFMT_S16_BE;
#else
oss_format = AFMT_S16_LE;
#endif
ao->format = AF_FORMAT_S16;
}
if (ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &oss_format) < 0 ||
oss_format != format2oss(ao->format))
{
MP_WARN(ao, "Can't set audio device %s to %s output, trying %s...\n",
p->dsp, af_fmt_to_str(ao->format),
af_fmt_to_str(AF_FORMAT_S16));
ao->format = AF_FORMAT_S16;
goto ac3_retry;
}
ao->format = oss2format(oss_format);
if (ao->format == -1) {
MP_ERR(ao, "Unknown/Unsupported OSS format: %x.\n", oss_format);
return -1;
}
MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(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_ERR(ao, "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_ERR(ao, "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_VERBOSE(ao, "using %d channels (requested: %d)\n",
ao->channels.num, reqchannels);
// set rate
ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
MP_VERBOSE(ao, "using %d Hz samplerate\n", ao->samplerate);
}
if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &p->zz) == -1) {
int r = 0;
MP_WARN(ao, "driver doesn't support SNDCTL_DSP_GETOSPACE\n");
if (ioctl(p->audio_fd, SNDCTL_DSP_GETBLKSIZE, &r) == -1)
MP_VERBOSE(ao, "%d bytes/frag (config.h)\n", p->outburst);
else {
p->outburst = r;
MP_VERBOSE(ao, "%d bytes/frag (GETBLKSIZE)\n", p->outburst);
}
} else {
MP_VERBOSE(ao, "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;
#if 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_ERR(ao, "*** Your audio driver DOES NOT support select() ***\n");
MP_ERR(ao, "Recompile mpv with #define HAVE_AUDIO_SELECT 0 in config.h!\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;
}
#ifndef SNDCTL_DSP_RESET
static void close_device(struct ao *ao)
{
struct priv *p = ao->priv;
close(p->audio_fd);
p->audio_fd = -1;
}
#endif
// stop playing and empty buffers (for seeking/pause)
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
int oss_format;
#ifdef SNDCTL_DSP_RESET
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
#else
close_device(ao);
p->audio_fd = open(p->dsp, O_WRONLY);
if (p->audio_fd < 0) {
MP_ERR(ao, "Fatal 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
#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 / ao->sstride;
}
#endif
// check buffer
#if 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 / ao->sstride;
}
// stop playing, keep buffers (for pause)
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
p->prepause_space = get_space(ao) * ao->sstride;
#ifdef SNDCTL_DSP_RESET
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
#else
close_device(ao);
#endif
}
// 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 samples, int flags)
{
struct priv *p = ao->priv;
int len = samples * ao->sstride;
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[0], len);
return len / ao->sstride;
}
// resume playing, after audio_pause()
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
#ifndef SNDCTL_DSP_RESET
reset(ao);
#endif
int fillframes = get_space(ao) - p->prepause_space / ao->sstride;
if (fillframes > 0)
ao_play_silence(ao, fillframes);
}
// 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;
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_oss = {
.description = "OSS/ioctl audio output",
.name = "oss",
.init = init,
.uninit = uninit,
.control = control,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = audio_pause,
.resume = audio_resume,
.reset = reset,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.audio_fd = -1,
.audio_delay_method = 2,
.buffersize = -1,
.outburst = 512,
.oss_mixer_channel = SOUND_MIXER_PCM,
.dsp = PATH_DEV_DSP,
.oss_mixer_device = PATH_DEV_MIXER,
},
.options = (const struct m_option[]) {
OPT_STRING("device", dsp, 0),
OPT_STRING("mixer-device", oss_mixer_device, 0),
OPT_STRING("mixer-channel", cfg_oss_mixer_channel, 0),
{0}
},
};
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