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authorwm4 <wm4@nowhere>2020-03-28 20:58:03 +0100
committerwm4 <wm4@nowhere>2020-03-28 20:59:31 +0100
commitbca917f6d2c50d9df3a4140b7be7beb6d567aa5b (patch)
tree6bda396bd723d29f0423d720db21fd279fdb2406
parent4583bd8cc7bf538bce424983d49729c934d13a53 (diff)
downloadmpv-bca917f6d2c50d9df3a4140b7be7beb6d567aa5b.tar.bz2
mpv-bca917f6d2c50d9df3a4140b7be7beb6d567aa5b.tar.xz
ao_oss: remove this audio output
Ancient Linux audio output. Apparently it survived until now, because some BSDs (but not all) had use of this. But these should work with ao_sdl or ao_openal too (that's why these AOs exist after all). ao_oss itself has the problem that it's virtually unmaintainable from my point of view due to all the subtle (or non-subtle) difference. Look at the ifdef mess and the multiple code paths (that shouldn't exist) in the removed source code.
-rw-r--r--DOCS/man/ao.rst15
-rw-r--r--audio/out/ao.c4
-rw-r--r--audio/out/ao_oss.c657
-rw-r--r--wscript5
-rw-r--r--wscript_build.py1
5 files changed, 1 insertions, 681 deletions
diff --git a/DOCS/man/ao.rst b/DOCS/man/ao.rst
index 7dace73ec5..5308a7175b 100644
--- a/DOCS/man/ao.rst
+++ b/DOCS/man/ao.rst
@@ -14,8 +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. On BSD systems, ``--ao=oss`` or ``--ao=sndio``
- may work (the latter being experimental).
+ where PulseAudio is used.
Available audio output drivers are:
@@ -36,18 +35,6 @@ 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
-
- The following global options are supported by this audio output:
-
- ``--oss-mixer-device``
- Sets the audio mixer device (default: ``/dev/mixer``).
- ``--oss-mixer-channel``
- Sets the audio mixer channel (default: ``pcm``). Other valid values
- include **vol, pcm, line**. For a complete list of options look for
- ``SOUND_DEVICE_NAMES`` in ``/usr/include/linux/soundcard.h``.
-
``jack``
JACK (Jack Audio Connection Kit) audio output driver.
diff --git a/audio/out/ao.c b/audio/out/ao.c
index 10717d466e..480dad69e0 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -35,7 +35,6 @@
#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;
@@ -72,9 +71,6 @@ static const struct ao_driver * const audio_out_drivers[] = {
#if HAVE_WASAPI
&audio_out_wasapi,
#endif
-#if HAVE_OSS_AUDIO
- &audio_out_oss,
-#endif
// wrappers:
#if HAVE_JACK
&audio_out_jack,
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
deleted file mode 100644
index fe87a3db73..0000000000
--- a/audio/out/ao_oss.c
+++ /dev/null
@@ -1,657 +0,0 @@
-/*
- * 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"
-
-#include <sys/soundcard.h>
-
-#include "audio/format.h"
-
-#include "ao.h"
-#include "internal.h"
-
-#if !HAVE_GPL
-#error GPL only
-#endif
-
-// 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.
-#if defined(SNDCTL_DSP_RESET) && !defined(__NetBSD__)
-#define KEEP_DEVICE 1
-#else
-#define KEEP_DEVICE 0
-#endif
-
-#define PATH_DEV_DSP "/dev/dsp"
-#define PATH_DEV_MIXER "/dev/mixer"
-
-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 *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_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}
-};
-
-#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;
- }
- }
- 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;
-
- const char *device = PATH_DEV_DSP;
- if (ao->device)
- device = ao->device;
-
- MP_VERBOSE(ao, "using '%s' dsp device\n", device);
-#ifdef __linux__
- p->audio_fd = open(device, O_WRONLY | O_NONBLOCK);
-#else
- p->audio_fd = open(device, O_WRONLY);
-#endif
- if (p->audio_fd < 0) {
- MP_ERR(ao, "Can't open audio device %s: %s\n",
- device, 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 + 1];
- 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 c, nchannels, reqchannels;
- nchannels = reqchannels = channels.num;
- // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
- if (reqchannels > 2) {
- if (ioctl(p->audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1) {
- MP_ERR(ao, "Failed to set audio device to %d channels.\n",
- reqchannels);
- goto fail;
- }
- if (nchannels != reqchannels) {
- // Fallback to stereo
- nchannels = 2;
- goto stereo;
- }
- } else {
-stereo:
- c = nchannels - 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;
- }
- }
-
- int sstride = channels.num * af_fmt_to_bytes(format);
- p->outburst -= p->outburst % sstride; // round down
- ao->period_size = p->outburst / sstride;
-
- 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' mixer device\n", p->oss_mixer_device);
- MP_VERBOSE(ao, "using '%s' mixer channel\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,
- .oss_mixer_device = PATH_DEV_MIXER,
- },
- .options = (const struct m_option[]) {
- {"mixer-device", OPT_STRING(oss_mixer_device), .flags = M_OPT_FILE},
- {"mixer-channel", OPT_STRING(cfg_oss_mixer_channel)},
- {0}
- },
- .options_prefix = "oss",
-};
diff --git a/wscript b/wscript
index ccfed67648..47bbf868db 100644
--- a/wscript
+++ b/wscript
@@ -440,11 +440,6 @@ audio_output_features = [
'deps': 'sdl2',
'func': check_true,
}, {
- 'name': '--oss-audio',
- 'desc': 'OSS',
- 'func': check_cc(header_name='sys/soundcard.h'),
- 'deps': 'posix && gpl',
- }, {
'name': '--pulse',
'desc': 'PulseAudio audio output',
'func': check_pkg_config('libpulse', '>= 1.0')
diff --git a/wscript_build.py b/wscript_build.py
index 145b415d75..62f2264efe 100644
--- a/wscript_build.py
+++ b/wscript_build.py
@@ -255,7 +255,6 @@ 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" ),