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mpv/audio/out/ao_oss.c
wm4 380fc765e4 audio/out: prepare for non-interleaved audio 
This comes with two internal AO API changes:

1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.

2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)

Change all AOs to use the new API.

The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
2013-11-12 23:27:51 +01:00

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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_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];
}
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_NE;
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_NE;
}
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_NE));
ao->format = AF_FORMAT_S16_NE;
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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