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ao_oss: add this audio output again 

Changes:
- code refactored;
- mixer options removed;
- new mpv sound API used;
- add sound devices detect (mpv --audio-device=help will show all available devices);
- only OSSv4 supported now;

Tested on FreeBSD 12.2 amd64.
This commit is contained in:
rim 2020-11-24 04:22:40 +03:00 committed by sfan5
parent e79e455a36
commit 1b2e5137e0
5 changed files with 424 additions and 1 deletions
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5
DOCS/man/ao.rst
View File

@ -14,7 +14,7 @@ in the list.
See ``--ao=help`` for a list of compiled-in audio output drivers. The
driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
where PulseAudio is used.
where PulseAudio is used. On BSD systems, ``--ao=oss`` is preferred.
Available audio output drivers are:
@ -35,6 +35,9 @@ Available audio output drivers are:
with automatic upmixing with shared access, so playing stereo
and multichannel audio at the same time will work as expected.
``oss``
OSS audio output driver
``jack``
JACK (Jack Audio Connection Kit) audio output driver.

4
audio/out/ao.c
View File

@ -35,6 +35,7 @@
#include "common/common.h"
#include "common/global.h"
extern const struct ao_driver audio_out_oss;
extern const struct ao_driver audio_out_audiotrack;
extern const struct ao_driver audio_out_audiounit;
extern const struct ao_driver audio_out_coreaudio;
@ -70,6 +71,9 @@ static const struct ao_driver * const audio_out_drivers[] = {
#endif
#if HAVE_WASAPI
&audio_out_wasapi,
#endif
#if HAVE_OSS_AUDIO
&audio_out_oss,
#endif
// wrappers:
#if HAVE_JACK

410
audio/out/ao_oss.c Normal file
View File

@ -0,0 +1,410 @@
/*
* OSS audio output driver
*
* Original author: A'rpi
* Support for >2 output channels added 2001-11-25
* - Steve Davies <steve@daviesfam.org>
* Rozhuk Ivan <rozhuk.im@gmail.com> 2020
*
* 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 <errno.h>
#include <fcntl.h>
#include <stdio.h>
#include <unistd.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#include <sys/stat.h>
#if defined(__DragonFly__) || defined(__FreeBSD__)
#include <sys/sysctl.h>
#endif
#include <sys/types.h>
#include "config.h"
#include "audio/format.h"
#include "common/msg.h"
#include "options/options.h"
#include "osdep/endian.h"
#include "osdep/io.h"
#include "ao.h"
#include "internal.h"
#ifndef AFMT_AC3
#define AFMT_AC3 -1
#endif
#define PATH_DEV_DSP "/dev/dsp"
struct priv {
int dsp_fd;
bool playing;
double bps; /* Bytes per second. */
};
/* 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_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_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}
};
#define MP_WARN_IOCTL_ERR(__ao) \
MP_WARN((__ao), "%s: ioctl() fail, err = %i: %s\n", \
__FUNCTION__, errno, strerror(errno))
static void uninit(struct ao *ao);
static void device_descr_get(size_t dev_idx, char *buf, size_t buf_size)
{
#if defined(__DragonFly__) || defined(__FreeBSD__)
char dev_path[32];
size_t tmp = (buf_size - 1);
snprintf(dev_path, sizeof(dev_path), "dev.pcm.%zu.%%desc", dev_idx);
if (sysctlbyname(dev_path, buf, &tmp, NULL, 0) != 0) {
tmp = 0;
}
buf[tmp] = 0x00;
#elif defined(SOUND_MIXER_INFO)
size_t tmp = 0;
char dev_path[32];
mixer_info mi;
snprintf(dev_path, sizeof(dev_path), PATH_DEV_MIXER"%zu", dev_idx);
int fd = open(dev_path, O_RDONLY);
if (ioctl(fd, SOUND_MIXER_INFO, &mi) == 0) {
strncpy(buf, mi.name, buf_size);
tmp = (buf_size - 1);
}
close(fd);
buf[tmp] = 0x00;
#else
buf[0] = 0x00;
#endif
}
static int format2oss(int format)
{
for (size_t i = 0; format_table[i][0] != -1; i++) {
if (format_table[i][1] == format)
return format_table[i][0];
}
return -1;
}
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;
}
return (ioctl(p->dsp_fd, SNDCTL_DSP_SETFMT, &oss_format) != -1);
}
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
struct mp_chmap channels = ao->channels;
audio_buf_info info;
size_t i;
int format, samplerate, nchannels, reqchannels, trig = 0;
int best_sample_formats[AF_FORMAT_COUNT + 1];
const char *device = ((ao->device) ? ao->device : PATH_DEV_DSP);
/* Opening device. */
MP_VERBOSE(ao, "Using '%s' audio device.\n", device);
p->dsp_fd = open(device, (O_WRONLY | O_CLOEXEC));
if (p->dsp_fd < 0) {
MP_ERR(ao, "Can't open audio device %s: %s.\n",
device, mp_strerror(errno));
goto err_out;
}
/* Selecting sound format. */
format = af_fmt_from_planar(ao->format);
af_get_best_sample_formats(format, best_sample_formats);
for (i = 0; best_sample_formats[i]; i++) {
format = best_sample_formats[i];
if (try_format(ao, &format))
break;
}
if (!format) {
MP_ERR(ao, "Can't set sample format.\n");
goto err_out;
}
MP_VERBOSE(ao, "Sample format: %s\n", af_fmt_to_str(format));
/* Channels count. */
if (af_fmt_is_spdif(format)) {
/* 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 err_out;
}
} else {
struct mp_chmap_sel sel = {0};
for (i = 0; i < MP_ARRAY_SIZE(oss_layouts); i++) {
mp_chmap_sel_add_map(&sel, &oss_layouts[i]);
}
if (!ao_chmap_sel_adjust(ao, &sel, &channels))
goto err_out;
nchannels = reqchannels = channels.num;
if (ioctl(p->dsp_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1) {
MP_ERR(ao, "Failed to set audio device to %d channels.\n",
reqchannels);
goto err_out_ioctl;
}
if (nchannels != reqchannels) {
/* Update number of channels to OSS suggested value. */
if (!ao_chmap_sel_get_def(ao, &sel, &channels, nchannels))
goto err_out;
}
MP_VERBOSE(ao, "Using %d channels (requested: %d).\n",
channels.num, reqchannels);
}
/* Sample rate. */
samplerate = ao->samplerate;
if (ioctl(p->dsp_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
goto err_out_ioctl;
MP_VERBOSE(ao, "Using %d Hz samplerate.\n", samplerate);
/* Get buffer size. */
if (ioctl(p->dsp_fd, SNDCTL_DSP_GETOSPACE, &info) == -1)
goto err_out_ioctl;
/* See ao.c ao->sstride initializations and get_state(). */
ao->device_buffer = ((info.fragstotal * info.fragsize) /
af_fmt_to_bytes(format));
if (!af_fmt_is_planar(format)) {
ao->device_buffer /= channels.num;
}
/* Do not start playback after data written. */
if (ioctl(p->dsp_fd, SNDCTL_DSP_SETTRIGGER, &trig) == -1)
goto err_out_ioctl;
/* Update sound params. */
ao->format = format;
ao->samplerate = samplerate;
ao->channels = channels;
p->bps = (channels.num * samplerate * af_fmt_to_bytes(format));
p->playing = false;
return 0;
err_out_ioctl:
MP_WARN_IOCTL_ERR(ao);
err_out:
uninit(ao);
return -1;
}
static void uninit(struct ao *ao)
{
struct priv *p = ao->priv;
if (p->dsp_fd == -1)
return;
ioctl(p->dsp_fd, SNDCTL_DSP_HALT, NULL);
close(p->dsp_fd);
p->dsp_fd = -1;
p->playing = false;
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct priv *p = ao->priv;
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int v;
if (p->dsp_fd < 0)
return CONTROL_ERROR;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
if (ioctl(p->dsp_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1) {
MP_WARN_IOCTL_ERR(ao);
return CONTROL_ERROR;
}
vol->right = ((v & 0xff00) >> 8);
vol->left = (v & 0x00ff);
return CONTROL_OK;
case AOCONTROL_SET_VOLUME:
v = ((int)vol->right << 8) | (int)vol->left;
if (ioctl(p->dsp_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1) {
MP_WARN_IOCTL_ERR(ao);
return CONTROL_ERROR;
}
return CONTROL_OK;
}
return CONTROL_UNKNOWN;
}
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
int trig = 0;
/* Clear buf and do not start playback after data written. */
p->playing = false;
if (ioctl(p->dsp_fd, SNDCTL_DSP_HALT, NULL) == -1 ||
ioctl(p->dsp_fd, SNDCTL_DSP_SETTRIGGER, &trig) == -1)
{
MP_WARN_IOCTL_ERR(ao);
MP_WARN(ao, "Force reinitialize audio device.\n");
uninit(ao);
init(ao);
}
}
static void start(struct ao *ao)
{
struct priv *p = ao->priv;
int trig = PCM_ENABLE_OUTPUT;
if (ioctl(p->dsp_fd, SNDCTL_DSP_SETTRIGGER, &trig) == -1) {
MP_WARN_IOCTL_ERR(ao);
return;
}
p->playing = true;
}
static bool audio_write(struct ao *ao, void **data, int samples)
{
struct priv *p = ao->priv;
ssize_t rc;
const size_t size = (samples * ao->sstride);
if (size == 0)
return true;
while ((rc = write(p->dsp_fd, data[0], size)) == -1) {
if (errno == EINTR)
continue;
MP_WARN(ao, "audio_write: write() fail, err = %i: %s.\n",
errno, strerror(errno));
p->playing = false;
return false;
}
if ((size_t)rc != size) {
MP_WARN(ao, "audio_write: unexpected partial write: required: %zu, written: %zu.\n",
size, (size_t)rc);
p->playing = false;
return false;
}
return true;
}
static void get_state(struct ao *ao, struct mp_pcm_state *state)
{
struct priv *p = ao->priv;
audio_buf_info info;
int odelay;
if (ioctl(p->dsp_fd, SNDCTL_DSP_GETOSPACE, &info) == -1 ||
ioctl(p->dsp_fd, SNDCTL_DSP_GETODELAY, &odelay) == -1)
{
MP_WARN_IOCTL_ERR(ao);
p->playing = false;
memset(state, 0x00, sizeof(struct mp_pcm_state));
state->delay = 0.0;
return;
}
state->free_samples = (info.bytes / ao->sstride);
state->queued_samples = (ao->device_buffer - state->free_samples);
state->delay = (odelay / p->bps);
state->playing = p->playing;
}
static void list_devs(struct ao *ao, struct ao_device_list *list)
{
struct stat st;
char dev_path[32] = PATH_DEV_DSP, dev_descr[256] = "Default";
struct ao_device_desc dev = {.name = dev_path, .desc = dev_descr};
if (stat(PATH_DEV_DSP, &st) == 0) {
ao_device_list_add(list, ao, &dev);
}
/* Auto detect. */
for (size_t i = 0, fail_cnt = 0; fail_cnt < 8; i ++, fail_cnt ++) {
snprintf(dev_path, sizeof(dev_path), PATH_DEV_DSP"%zu", i);
if (stat(dev_path, &st) != 0)
continue;
device_descr_get(i, dev_descr, sizeof(dev_descr));
ao_device_list_add(list, ao, &dev);
fail_cnt = 0; /* Reset fail counter. */
}
}
const struct ao_driver audio_out_oss = {
.name = "oss",
.description = "OSS/ioctl audio output",
.init = init,
.uninit = uninit,
.control = control,
.reset = reset,
.start = start,
.write = audio_write,
.get_state = get_state,
.list_devs = list_devs,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.dsp_fd = -1,
.playing = false,
},
};

5
wscript
View File

@ -421,6 +421,11 @@ audio_output_features = [
'desc': 'SDL2 audio output',
'deps': 'sdl2',
'func': check_true,
}, {
'name': '--oss-audio',
'desc': 'OSSv4 audio output',
'func': check_statement(['sys/soundcard.h'], 'int x = SNDCTL_DSP_SETPLAYVOL'),
'deps': 'posix && gpl',
}, {
'name': '--pulse',
'desc': 'PulseAudio audio output',

1
wscript_build.py
View File

@ -244,6 +244,7 @@ def build(ctx):
( "audio/out/ao_null.c" ),
( "audio/out/ao_openal.c", "openal" ),
( "audio/out/ao_opensles.c", "opensles" ),
( "audio/out/ao_oss.c", "oss-audio" ),
( "audio/out/ao_pcm.c" ),
( "audio/out/ao_pulse.c", "pulse" ),
( "audio/out/ao_sdl.c", "sdl2-audio" ),