mirror of https://github.com/mpv-player/mpv.git
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.
This commit is contained in:
wm4
parent
4583bd8cc7
commit
bca917f6d2
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@ -14,8 +14,7 @@ in the list.
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See ``--ao=help`` for a list of compiled-in audio output drivers. The
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driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
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where PulseAudio is used. On BSD systems, ``--ao=oss`` or ``--ao=sndio``
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may work (the latter being experimental).
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where PulseAudio is used.
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Available audio output drivers are:
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@ -36,18 +35,6 @@ Available audio output drivers are:
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with automatic upmixing with shared access, so playing stereo
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and multichannel audio at the same time will work as expected.
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``oss``
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OSS audio output driver
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The following global options are supported by this audio output:
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``--oss-mixer-device``
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Sets the audio mixer device (default: ``/dev/mixer``).
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``--oss-mixer-channel``
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Sets the audio mixer channel (default: ``pcm``). Other valid values
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include **vol, pcm, line**. For a complete list of options look for
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``SOUND_DEVICE_NAMES`` in ``/usr/include/linux/soundcard.h``.
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``jack``
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JACK (Jack Audio Connection Kit) audio output driver.
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@ -35,7 +35,6 @@
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#include "common/common.h"
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#include "common/global.h"
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extern const struct ao_driver audio_out_oss;
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extern const struct ao_driver audio_out_audiotrack;
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extern const struct ao_driver audio_out_audiounit;
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extern const struct ao_driver audio_out_coreaudio;
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@ -71,9 +70,6 @@ static const struct ao_driver * const audio_out_drivers[] = {
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#endif
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#if HAVE_WASAPI
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&audio_out_wasapi,
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#endif
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#if HAVE_OSS_AUDIO
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&audio_out_oss,
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#endif
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// wrappers:
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#if HAVE_JACK
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@ -1,657 +0,0 @@
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/*
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* OSS audio output driver
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*
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* Original author: A'rpi
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* Support for >2 output channels added 2001-11-25
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* - Steve Davies <steve@daviesfam.org>
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*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <sys/ioctl.h>
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#include <unistd.h>
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#include <sys/time.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <fcntl.h>
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#include <poll.h>
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#include <errno.h>
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#include <string.h>
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#include <strings.h>
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#include "config.h"
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#include "options/options.h"
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#include "common/common.h"
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#include "common/msg.h"
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#include "osdep/timer.h"
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#include "osdep/endian.h"
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#include <sys/soundcard.h>
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#include "audio/format.h"
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#include "ao.h"
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#include "internal.h"
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#if !HAVE_GPL
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#error GPL only
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#endif
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// Define to 0 if the device must be reopened to reset it (stop all playback,
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// clear the buffer), and the device should be closed when unused.
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// Define to 1 if SNDCTL_DSP_RESET should be used to reset without close.
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#if defined(SNDCTL_DSP_RESET) && !defined(__NetBSD__)
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#define KEEP_DEVICE 1
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#else
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#define KEEP_DEVICE 0
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#endif
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#define PATH_DEV_DSP "/dev/dsp"
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#define PATH_DEV_MIXER "/dev/mixer"
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struct priv {
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int audio_fd;
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int prepause_samples;
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int oss_mixer_channel;
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int audio_delay_method;
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int buffersize;
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int outburst;
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bool device_failed;
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double audio_end;
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char *oss_mixer_device;
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char *cfg_oss_mixer_channel;
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};
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static const char *const mixer_channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
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/* like alsa except for 6.1 and 7.1, from pcm/matrix_map.h */
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static const struct mp_chmap oss_layouts[MP_NUM_CHANNELS + 1] = {
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{0}, // empty
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MP_CHMAP_INIT_MONO, // mono
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MP_CHMAP2(FL, FR), // stereo
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MP_CHMAP3(FL, FR, LFE), // 2.1
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MP_CHMAP4(FL, FR, BL, BR), // 4.0
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MP_CHMAP5(FL, FR, BL, BR, FC), // 5.0
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MP_CHMAP6(FL, FR, BL, BR, FC, LFE), // 5.1
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MP_CHMAP7(FL, FR, BL, BR, FC, LFE, BC), // 6.1
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MP_CHMAP8(FL, FR, BL, BR, FC, LFE, SL, SR), // 7.1
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};
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#if !defined(AFMT_S16_NE) && defined(AFMT_S16_LE) && defined(AFMT_S16_BE)
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#define AFMT_S16_NE MP_SELECT_LE_BE(AFMT_S16_LE, AFMT_S16_BE)
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#endif
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#if !defined(AFMT_S32_NE) && defined(AFMT_S32_LE) && defined(AFMT_S32_BE)
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#define AFMT_S32_NE AFMT_S32MP_SELECT_LE_BE(AFMT_S32_LE, AFMT_S32_BE)
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#endif
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static const int format_table[][2] = {
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{AFMT_U8, AF_FORMAT_U8},
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{AFMT_S16_NE, AF_FORMAT_S16},
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#ifdef AFMT_S32_NE
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{AFMT_S32_NE, AF_FORMAT_S32},
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#endif
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#ifdef AFMT_FLOAT
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{AFMT_FLOAT, AF_FORMAT_FLOAT},
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#endif
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#ifdef AFMT_MPEG
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{AFMT_MPEG, AF_FORMAT_S_MP3},
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#endif
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{-1, -1}
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};
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#ifndef AFMT_AC3
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#define AFMT_AC3 -1
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#endif
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static int format2oss(int format)
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{
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for (int n = 0; format_table[n][0] != -1; n++) {
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if (format_table[n][1] == format)
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return format_table[n][0];
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}
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return -1;
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}
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static int oss2format(int format)
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{
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for (int n = 0; format_table[n][0] != -1; n++) {
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if (format_table[n][0] == format)
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return format_table[n][1];
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}
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return 0;
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}
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#ifdef SNDCTL_DSP_GETPLAYVOL
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static int volume_oss4(struct ao *ao, ao_control_vol_t *vol, int cmd)
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{
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struct priv *p = ao->priv;
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int v;
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if (p->audio_fd < 0)
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return CONTROL_ERROR;
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if (cmd == AOCONTROL_GET_VOLUME) {
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if (ioctl(p->audio_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1)
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return CONTROL_ERROR;
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vol->right = (v & 0xff00) >> 8;
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vol->left = v & 0x00ff;
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return CONTROL_OK;
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} else if (cmd == AOCONTROL_SET_VOLUME) {
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v = ((int) vol->right << 8) | (int) vol->left;
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if (ioctl(p->audio_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1)
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return CONTROL_ERROR;
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return CONTROL_OK;
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} else
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return CONTROL_UNKNOWN;
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}
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#endif
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// to set/get/query special features/parameters
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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struct priv *p = ao->priv;
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switch (cmd) {
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case AOCONTROL_GET_VOLUME:
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case AOCONTROL_SET_VOLUME: {
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ao_control_vol_t *vol = (ao_control_vol_t *)arg;
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int fd, v, devs;
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#ifdef SNDCTL_DSP_GETPLAYVOL
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// Try OSS4 first
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if (volume_oss4(ao, vol, cmd) == CONTROL_OK)
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return CONTROL_OK;
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#endif
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if (!af_fmt_is_pcm(ao->format))
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return CONTROL_TRUE;
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if ((fd = open(p->oss_mixer_device, O_RDONLY)) != -1) {
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ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
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if (devs & (1 << p->oss_mixer_channel)) {
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if (cmd == AOCONTROL_GET_VOLUME) {
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ioctl(fd, MIXER_READ(p->oss_mixer_channel), &v);
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vol->right = (v & 0xFF00) >> 8;
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vol->left = v & 0x00FF;
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} else {
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v = ((int)vol->right << 8) | (int)vol->left;
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ioctl(fd, MIXER_WRITE(p->oss_mixer_channel), &v);
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}
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} else {
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close(fd);
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return CONTROL_ERROR;
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}
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close(fd);
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return CONTROL_OK;
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}
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return CONTROL_ERROR;
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}
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}
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return CONTROL_UNKNOWN;
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}
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// 1: ok, 0: not writable, -1: error
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static int device_writable(struct ao *ao)
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{
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struct priv *p = ao->priv;
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struct pollfd fd = {.fd = p->audio_fd, .events = POLLOUT};
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return poll(&fd, 1, 0);
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}
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static void close_device(struct ao *ao)
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{
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struct priv *p = ao->priv;
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p->device_failed = false;
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if (p->audio_fd == -1)
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return;
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#if defined(SNDCTL_DSP_RESET)
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ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
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#endif
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close(p->audio_fd);
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p->audio_fd = -1;
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}
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// close audio device
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static void uninit(struct ao *ao)
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{
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close_device(ao);
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}
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static bool try_format(struct ao *ao, int *format)
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{
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struct priv *p = ao->priv;
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int oss_format = format2oss(*format);
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if (oss_format == -1 && af_fmt_is_spdif(*format))
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oss_format = AFMT_AC3;
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if (oss_format == -1) {
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MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n",
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af_fmt_to_str(*format));
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*format = 0;
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return false;
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}
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int actual_format = oss_format;
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if (ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &actual_format) < 0)
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actual_format = -1;
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if (actual_format == oss_format)
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return true;
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MP_WARN(ao, "Can't set audio device to %s output.\n", af_fmt_to_str(*format));
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*format = oss2format(actual_format);
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if (actual_format != -1 && !*format)
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MP_ERR(ao, "Unknown/Unsupported OSS format: 0x%x.\n", actual_format);
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return false;
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}
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static int reopen_device(struct ao *ao, bool allow_format_changes)
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{
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struct priv *p = ao->priv;
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int samplerate = ao->samplerate;
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int format = ao->format;
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struct mp_chmap channels = ao->channels;
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const char *device = PATH_DEV_DSP;
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if (ao->device)
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device = ao->device;
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MP_VERBOSE(ao, "using '%s' dsp device\n", device);
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#ifdef __linux__
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p->audio_fd = open(device, O_WRONLY | O_NONBLOCK);
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#else
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p->audio_fd = open(device, O_WRONLY);
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#endif
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if (p->audio_fd < 0) {
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MP_ERR(ao, "Can't open audio device %s: %s\n",
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device, mp_strerror(errno));
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goto fail;
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}
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#ifdef __linux__
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/* Remove the non-blocking flag */
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if (fcntl(p->audio_fd, F_SETFL, 0) < 0) {
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MP_ERR(ao, "Can't make file descriptor blocking: %s\n",
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mp_strerror(errno));
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goto fail;
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}
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#endif
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#if defined(FD_CLOEXEC) && defined(F_SETFD)
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fcntl(p->audio_fd, F_SETFD, FD_CLOEXEC);
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#endif
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if (af_fmt_is_spdif(format)) {
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if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
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goto fail;
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// Probably could be fixed by setting number of channels; needs testing.
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if (channels.num != 2) {
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MP_ERR(ao, "Format %s not implemented.\n", af_fmt_to_str(format));
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goto fail;
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}
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}
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int try_formats[AF_FORMAT_COUNT + 1];
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af_get_best_sample_formats(format, try_formats);
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for (int n = 0; try_formats[n]; n++) {
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format = try_formats[n];
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if (try_format(ao, &format))
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break;
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}
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if (!format) {
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MP_ERR(ao, "Can't set sample format.\n");
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goto fail;
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}
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MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(format));
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if (!af_fmt_is_spdif(format)) {
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struct mp_chmap_sel sel = {0};
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for (int n = 0; n < MP_NUM_CHANNELS + 1; n++)
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mp_chmap_sel_add_map(&sel, &oss_layouts[n]);
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if (!ao_chmap_sel_adjust(ao, &sel, &channels))
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goto fail;
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int c, nchannels, reqchannels;
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nchannels = reqchannels = channels.num;
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// We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
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if (reqchannels > 2) {
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if (ioctl(p->audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1) {
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MP_ERR(ao, "Failed to set audio device to %d channels.\n",
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reqchannels);
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goto fail;
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}
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if (nchannels != reqchannels) {
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// Fallback to stereo
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nchannels = 2;
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goto stereo;
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}
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} else {
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stereo:
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c = nchannels - 1;
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if (ioctl(p->audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
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MP_ERR(ao, "Failed to set audio device to %d channels.\n",
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reqchannels);
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goto fail;
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}
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if (!ao_chmap_sel_get_def(ao, &sel, &channels, c + 1))
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goto fail;
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}
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MP_VERBOSE(ao, "using %d channels (requested: %d)\n",
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channels.num, reqchannels);
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// set rate
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if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
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goto fail;
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MP_VERBOSE(ao, "using %d Hz samplerate\n", samplerate);
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}
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audio_buf_info zz = {0};
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if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &zz) == -1) {
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int r = 0;
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MP_WARN(ao, "driver doesn't support SNDCTL_DSP_GETOSPACE\n");
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if (ioctl(p->audio_fd, SNDCTL_DSP_GETBLKSIZE, &r) == -1)
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MP_VERBOSE(ao, "%d bytes/frag (config.h)\n", p->outburst);
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else {
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p->outburst = r;
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MP_VERBOSE(ao, "%d bytes/frag (GETBLKSIZE)\n", p->outburst);
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}
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} else {
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MP_VERBOSE(ao, "frags: %3d/%d (%d bytes/frag) free: %6d\n",
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zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
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p->buffersize = zz.bytes;
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p->outburst = zz.fragsize;
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}
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if (allow_format_changes) {
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ao->format = format;
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ao->samplerate = samplerate;
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ao->channels = channels;
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} else {
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if (format != ao->format || samplerate != ao->samplerate ||
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!mp_chmap_equals(&channels, &ao->channels))
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{
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MP_ERR(ao, "Could not reselect previous audio format.\n");
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goto fail;
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}
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}
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int sstride = channels.num * af_fmt_to_bytes(format);
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p->outburst -= p->outburst % sstride; // round down
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ao->period_size = p->outburst / sstride;
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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",
|
||||
};
|
||||
5
wscript
5
wscript
|
|
@ -439,11 +439,6 @@ audio_output_features = [
|
|||
'desc': 'SDL2 audio output',
|
||||
'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',
|
||||
|
|
|
|||
|
|
@ -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" ),
|
||||
|
|
|
|||
Loading…
Reference in New Issue