mirror of
https://github.com/mpv-player/mpv
synced 2024-12-20 22:02:59 +00:00
1a2319f3e4
Long planned. Leads to some sanity. There still are some rather gross things. Especially g_groups is ugly, and a hack that can hopefully be removed. (There is a plan for it, but whether it's implemented depends on how much energy is left.)
658 lines
19 KiB
C
658 lines
19 KiB
C
/*
|
|
* OSS audio output driver
|
|
*
|
|
* Original author: A'rpi
|
|
* Support for >2 output channels added 2001-11-25
|
|
* - Steve Davies <steve@daviesfam.org>
|
|
*
|
|
* This file is part of mpv.
|
|
*
|
|
* mpv 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.
|
|
*
|
|
* mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
|
|
*/
|
|
|
|
#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 <poll.h>
|
|
#include <errno.h>
|
|
#include <string.h>
|
|
#include <strings.h>
|
|
|
|
#include "config.h"
|
|
#include "options/options.h"
|
|
#include "common/common.h"
|
|
#include "common/msg.h"
|
|
#include "osdep/timer.h"
|
|
#include "osdep/endian.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"
|
|
#include "internal.h"
|
|
|
|
// Define to 0 if the device must be reopened to reset it (stop all playback,
|
|
// clear the buffer), and the device should be closed when unused.
|
|
// Define to 1 if SNDCTL_DSP_RESET should be used to reset without close.
|
|
#define KEEP_DEVICE (defined(SNDCTL_DSP_RESET) && !defined(__NetBSD__))
|
|
|
|
struct priv {
|
|
int audio_fd;
|
|
int prepause_samples;
|
|
int oss_mixer_channel;
|
|
int audio_delay_method;
|
|
int buffersize;
|
|
int outburst;
|
|
bool device_failed;
|
|
double audio_end;
|
|
|
|
char *dsp;
|
|
char *oss_mixer_device;
|
|
char *cfg_oss_mixer_channel;
|
|
};
|
|
|
|
static const char *const mixer_channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
|
|
|
|
/* like alsa except for 6.1 and 7.1, from pcm/matrix_map.h */
|
|
static const struct mp_chmap oss_layouts[MP_NUM_CHANNELS + 1] = {
|
|
{0}, // empty
|
|
MP_CHMAP_INIT_MONO, // mono
|
|
MP_CHMAP2(FL, FR), // stereo
|
|
MP_CHMAP3(FL, FR, LFE), // 2.1
|
|
MP_CHMAP4(FL, FR, BL, BR), // 4.0
|
|
MP_CHMAP5(FL, FR, BL, BR, FC), // 5.0
|
|
MP_CHMAP6(FL, FR, BL, BR, FC, LFE), // 5.1
|
|
MP_CHMAP7(FL, FR, BL, BR, FC, LFE, BC), // 6.1
|
|
MP_CHMAP8(FL, FR, BL, BR, FC, LFE, SL, SR), // 7.1
|
|
};
|
|
|
|
#if !defined(AFMT_S16_NE) && defined(AFMT_S16_LE) && defined(AFMT_S16_BE)
|
|
#define AFMT_S16_NE MP_SELECT_LE_BE(AFMT_S16_LE, AFMT_S16_BE)
|
|
#endif
|
|
|
|
#if !defined(AFMT_S24_NE) && defined(AFMT_S24_LE) && defined(AFMT_S24_BE)
|
|
#define AFMT_S24_NE MP_SELECT_LE_BE(AFMT_S24_LE, AFMT_S24_BE)
|
|
#endif
|
|
|
|
#if !defined(AFMT_S32_NE) && defined(AFMT_S32_LE) && defined(AFMT_S32_BE)
|
|
#define AFMT_S32_NE AFMT_S32MP_SELECT_LE_BE(AFMT_S32_LE, AFMT_S32_BE)
|
|
#endif
|
|
|
|
static const int format_table[][2] = {
|
|
{AFMT_U8, AF_FORMAT_U8},
|
|
{AFMT_S16_NE, AF_FORMAT_S16},
|
|
#ifdef AFMT_S24_NE
|
|
{AFMT_S24_NE, AF_FORMAT_S24},
|
|
#endif
|
|
#ifdef AFMT_S32_NE
|
|
{AFMT_S32_NE, AF_FORMAT_S32},
|
|
#endif
|
|
#ifdef AFMT_FLOAT
|
|
{AFMT_FLOAT, AF_FORMAT_FLOAT},
|
|
#endif
|
|
#ifdef AFMT_MPEG
|
|
{AFMT_MPEG, AF_FORMAT_S_MP3},
|
|
#endif
|
|
{-1, -1}
|
|
};
|
|
|
|
#ifndef AFMT_AC3
|
|
#define AFMT_AC3 -1
|
|
#endif
|
|
|
|
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 0;
|
|
}
|
|
|
|
|
|
#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_fmt_is_pcm(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;
|
|
}
|
|
#ifdef SNDCTL_DSP_GETPLAYVOL
|
|
case AOCONTROL_HAS_SOFT_VOLUME:
|
|
return CONTROL_TRUE;
|
|
#endif
|
|
}
|
|
return CONTROL_UNKNOWN;
|
|
}
|
|
|
|
// 1: ok, 0: not writable, -1: error
|
|
static int device_writable(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
struct pollfd fd = {.fd = p->audio_fd, .events = POLLOUT};
|
|
return poll(&fd, 1, 0);
|
|
}
|
|
|
|
static void close_device(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
p->device_failed = false;
|
|
if (p->audio_fd == -1)
|
|
return;
|
|
#if defined(SNDCTL_DSP_RESET)
|
|
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
|
|
#endif
|
|
close(p->audio_fd);
|
|
p->audio_fd = -1;
|
|
}
|
|
|
|
// close audio device
|
|
static void uninit(struct ao *ao)
|
|
{
|
|
close_device(ao);
|
|
}
|
|
|
|
static bool try_format(struct ao *ao, int *format)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
|
|
int oss_format = format2oss(*format);
|
|
if (oss_format == -1 && af_fmt_is_spdif(*format))
|
|
oss_format = AFMT_AC3;
|
|
|
|
if (oss_format == -1) {
|
|
MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n",
|
|
af_fmt_to_str(*format));
|
|
*format = 0;
|
|
return false;
|
|
}
|
|
|
|
int actual_format = oss_format;
|
|
if (ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &actual_format) < 0)
|
|
actual_format = -1;
|
|
|
|
if (actual_format == oss_format)
|
|
return true;
|
|
|
|
MP_WARN(ao, "Can't set audio device to %s output.\n", af_fmt_to_str(*format));
|
|
*format = oss2format(actual_format);
|
|
if (actual_format != -1 && !*format)
|
|
MP_ERR(ao, "Unknown/Unsupported OSS format: 0x%x.\n", actual_format);
|
|
return false;
|
|
}
|
|
|
|
static int reopen_device(struct ao *ao, bool allow_format_changes)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
|
|
int samplerate = ao->samplerate;
|
|
int format = ao->format;
|
|
struct mp_chmap channels = ao->channels;
|
|
|
|
#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, mp_strerror(errno));
|
|
goto fail;
|
|
}
|
|
|
|
#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",
|
|
mp_strerror(errno));
|
|
goto fail;
|
|
}
|
|
#endif
|
|
|
|
#if defined(FD_CLOEXEC) && defined(F_SETFD)
|
|
fcntl(p->audio_fd, F_SETFD, FD_CLOEXEC);
|
|
#endif
|
|
|
|
if (af_fmt_is_spdif(format)) {
|
|
if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
|
|
goto fail;
|
|
// Probably could be fixed by setting number of channels; needs testing.
|
|
if (channels.num != 2) {
|
|
MP_ERR(ao, "Format %s not implemented.\n", af_fmt_to_str(format));
|
|
goto fail;
|
|
}
|
|
}
|
|
|
|
int try_formats[AF_FORMAT_COUNT];
|
|
af_get_best_sample_formats(format, try_formats);
|
|
for (int n = 0; try_formats[n]; n++) {
|
|
format = try_formats[n];
|
|
if (try_format(ao, &format))
|
|
break;
|
|
}
|
|
|
|
if (!format) {
|
|
MP_ERR(ao, "Can't set sample format.\n");
|
|
goto fail;
|
|
}
|
|
|
|
MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(format));
|
|
|
|
if (!af_fmt_is_spdif(format)) {
|
|
struct mp_chmap_sel sel = {0};
|
|
for (int n = 0; n < MP_NUM_CHANNELS + 1; n++)
|
|
mp_chmap_sel_add_map(&sel, &oss_layouts[n]);
|
|
if (!ao_chmap_sel_adjust(ao, &sel, &channels))
|
|
goto fail;
|
|
int reqchannels = 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);
|
|
goto fail;
|
|
}
|
|
} 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);
|
|
goto fail;
|
|
}
|
|
if (!ao_chmap_sel_get_def(ao, &sel, &channels, c + 1))
|
|
goto fail;
|
|
}
|
|
MP_VERBOSE(ao, "using %d channels (requested: %d)\n",
|
|
channels.num, reqchannels);
|
|
// set rate
|
|
if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
|
|
goto fail;
|
|
MP_VERBOSE(ao, "using %d Hz samplerate\n", samplerate);
|
|
}
|
|
|
|
audio_buf_info zz = {0};
|
|
if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &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",
|
|
zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
|
|
p->buffersize = zz.bytes;
|
|
p->outburst = zz.fragsize;
|
|
}
|
|
|
|
if (allow_format_changes) {
|
|
ao->format = format;
|
|
ao->samplerate = samplerate;
|
|
ao->channels = channels;
|
|
} else {
|
|
if (format != ao->format || samplerate != ao->samplerate ||
|
|
!mp_chmap_equals(&channels, &ao->channels))
|
|
{
|
|
MP_ERR(ao, "Could not reselect previous audio format.\n");
|
|
goto fail;
|
|
}
|
|
}
|
|
|
|
p->outburst -= p->outburst % (channels.num * af_fmt_to_bytes(format)); // round down
|
|
|
|
return 0;
|
|
|
|
fail:
|
|
close_device(ao);
|
|
return -1;
|
|
}
|
|
|
|
// open & setup audio device
|
|
// return: 0=success -1=fail
|
|
static int init(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
|
|
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, mp_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]);
|
|
|
|
ao->format = af_fmt_from_planar(ao->format);
|
|
|
|
if (reopen_device(ao, true) < 0)
|
|
goto fail;
|
|
|
|
if (p->buffersize == -1) {
|
|
// Measuring buffer size:
|
|
void *data = malloc(p->outburst);
|
|
if (!data) {
|
|
MP_ERR(ao, "Out of memory, or broken outburst size.\n");
|
|
goto fail;
|
|
}
|
|
p->buffersize = 0;
|
|
memset(data, 0, p->outburst);
|
|
while (p->buffersize < 0x40000 && device_writable(ao) > 0) {
|
|
(void)write(p->audio_fd, data, p->outburst);
|
|
p->buffersize += p->outburst;
|
|
}
|
|
free(data);
|
|
if (p->buffersize == 0) {
|
|
MP_ERR(ao, "Your OSS audio driver DOES NOT support poll().\n");
|
|
goto fail;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
|
|
fail:
|
|
uninit(ao);
|
|
return -1;
|
|
}
|
|
|
|
static void drain(struct ao *ao)
|
|
{
|
|
#ifdef SNDCTL_DSP_SYNC
|
|
struct priv *p = ao->priv;
|
|
// to get the buffer played
|
|
if (p->audio_fd != -1)
|
|
ioctl(p->audio_fd, SNDCTL_DSP_SYNC, NULL);
|
|
#endif
|
|
}
|
|
|
|
// stop playing and empty buffers (for seeking/pause)
|
|
static void reset(struct ao *ao)
|
|
{
|
|
#if KEEP_DEVICE
|
|
struct priv *p = ao->priv;
|
|
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
|
|
#else
|
|
close_device(ao);
|
|
#endif
|
|
}
|
|
|
|
// plays 'len' samples of 'data'
|
|
// it should round it down to outburst*n
|
|
// return: number of samples 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 (p->audio_fd < 0 && !p->device_failed && reopen_device(ao, false) < 0)
|
|
MP_ERR(ao, "Fatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE ***\n");
|
|
if (p->audio_fd < 0) {
|
|
// Let playback continue normally, even with a closed device.
|
|
p->device_failed = true;
|
|
double now = mp_time_sec();
|
|
if (p->audio_end < now)
|
|
p->audio_end = now;
|
|
p->audio_end += samples / (double)ao->samplerate;
|
|
return samples;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
// return: delay in seconds between first and last sample in buffer
|
|
static double get_delay(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
if (p->audio_fd < 0) {
|
|
double rest = p->audio_end - mp_time_sec();
|
|
if (rest > 0)
|
|
return rest;
|
|
return 0;
|
|
}
|
|
/* 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 r / (double)ao->bps;
|
|
#endif
|
|
p->audio_delay_method = 1; // fallback if not supported
|
|
}
|
|
if (p->audio_delay_method == 1) {
|
|
audio_buf_info zz = {0};
|
|
if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) {
|
|
return (p->buffersize - zz.bytes) / (double)ao->bps;
|
|
}
|
|
p->audio_delay_method = 0; // fallback if not supported
|
|
}
|
|
return p->buffersize / (double)ao->bps;
|
|
}
|
|
|
|
|
|
// return: how many samples can be played without blocking
|
|
static int get_space(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
|
|
audio_buf_info zz = {0};
|
|
if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) {
|
|
// calculate exact buffer space:
|
|
return zz.fragments * zz.fragsize / ao->sstride;
|
|
}
|
|
|
|
if (p->audio_fd < 0 && p->device_failed && get_delay(ao) > 0.2)
|
|
return 0;
|
|
|
|
if (p->audio_fd < 0 || device_writable(ao) > 0)
|
|
return p->outburst / ao->sstride;
|
|
|
|
return 0;
|
|
}
|
|
|
|
// stop playing, keep buffers (for pause)
|
|
static void audio_pause(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
p->prepause_samples = get_delay(ao) * ao->samplerate;
|
|
#if KEEP_DEVICE
|
|
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
|
|
#else
|
|
close_device(ao);
|
|
#endif
|
|
}
|
|
|
|
// resume playing, after audio_pause()
|
|
static void audio_resume(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
p->audio_end = 0;
|
|
if (p->prepause_samples > 0)
|
|
ao_play_silence(ao, p->prepause_samples);
|
|
}
|
|
|
|
static int audio_wait(struct ao *ao, pthread_mutex_t *lock)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
|
|
struct pollfd fd = {.fd = p->audio_fd, .events = POLLOUT};
|
|
int r = ao_wait_poll(ao, &fd, 1, lock);
|
|
if (fd.revents & (POLLERR | POLLNVAL))
|
|
return -1;
|
|
return r;
|
|
}
|
|
|
|
static void list_devs(struct ao *ao, struct ao_device_list *list)
|
|
{
|
|
if (stat(PATH_DEV_DSP, &(struct stat){0}) == 0)
|
|
ao_device_list_add(list, ao, &(struct ao_device_desc){"", "Default"});
|
|
}
|
|
|
|
#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,
|
|
.drain = drain,
|
|
.wait = audio_wait,
|
|
.wakeup = ao_wakeup_poll,
|
|
.list_devs = list_devs,
|
|
.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, DEVICE_OPT_DEPRECATION),
|
|
OPT_STRING("mixer-device", oss_mixer_device, 0),
|
|
OPT_STRING("mixer-channel", cfg_oss_mixer_channel, 0),
|
|
{0}
|
|
},
|
|
.options_prefix = "oss",
|
|
};
|