mirror of
https://github.com/mpv-player/mpv
synced 2024-12-27 17:42:17 +00:00
ecc6e379b2
Make all AOs use what has been introduced in the previous commit. Note that even AOs which can handle all possible layouts (like ao_null) use the new functions. This might be important if in the future ao_select_champ() possibly honors global user options about downmixing and so on.
561 lines
16 KiB
C
561 lines
16 KiB
C
/*
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* OSS audio output driver
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*
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* This file is part of MPlayer.
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*
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* MPlayer 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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* MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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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 <errno.h>
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#include <string.h>
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#include "config.h"
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#include "core/mp_msg.h"
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#include "audio/mixer.h"
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#ifdef HAVE_SYS_SOUNDCARD_H
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#include <sys/soundcard.h>
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#else
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#ifdef HAVE_SOUNDCARD_H
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#include <soundcard.h>
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#endif
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#endif
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#include "audio/format.h"
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#include "ao.h"
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#include "audio_out_internal.h"
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static const ao_info_t info =
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{
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"OSS/ioctl audio output",
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"oss",
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"A'rpi",
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""
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};
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/* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */
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LIBAO_EXTERN(oss)
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static int format2oss(int format)
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{
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switch(format)
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{
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case AF_FORMAT_U8: return AFMT_U8;
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case AF_FORMAT_S8: return AFMT_S8;
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case AF_FORMAT_U16_LE: return AFMT_U16_LE;
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case AF_FORMAT_U16_BE: return AFMT_U16_BE;
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case AF_FORMAT_S16_LE: return AFMT_S16_LE;
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case AF_FORMAT_S16_BE: return AFMT_S16_BE;
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#ifdef AFMT_S24_PACKED
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case AF_FORMAT_S24_LE: return AFMT_S24_PACKED;
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#endif
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#ifdef AFMT_U32_LE
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case AF_FORMAT_U32_LE: return AFMT_U32_LE;
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#endif
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#ifdef AFMT_U32_BE
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case AF_FORMAT_U32_BE: return AFMT_U32_BE;
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#endif
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#ifdef AFMT_S32_LE
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case AF_FORMAT_S32_LE: return AFMT_S32_LE;
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#endif
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#ifdef AFMT_S32_BE
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case AF_FORMAT_S32_BE: return AFMT_S32_BE;
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#endif
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#ifdef AFMT_FLOAT
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case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
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#endif
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// SPECIALS
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#ifdef AFMT_MPEG
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case AF_FORMAT_MPEG2: return AFMT_MPEG;
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#endif
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#ifdef AFMT_AC3
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case AF_FORMAT_AC3_NE: return AFMT_AC3;
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#endif
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}
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mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format));
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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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switch(format)
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{
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case AFMT_U8: return AF_FORMAT_U8;
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case AFMT_S8: return AF_FORMAT_S8;
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case AFMT_U16_LE: return AF_FORMAT_U16_LE;
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case AFMT_U16_BE: return AF_FORMAT_U16_BE;
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case AFMT_S16_LE: return AF_FORMAT_S16_LE;
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case AFMT_S16_BE: return AF_FORMAT_S16_BE;
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#ifdef AFMT_S24_PACKED
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case AFMT_S24_PACKED: return AF_FORMAT_S24_LE;
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#endif
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#ifdef AFMT_U32_LE
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case AFMT_U32_LE: return AF_FORMAT_U32_LE;
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#endif
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#ifdef AFMT_U32_BE
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case AFMT_U32_BE: return AF_FORMAT_U32_BE;
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#endif
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#ifdef AFMT_S32_LE
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case AFMT_S32_LE: return AF_FORMAT_S32_LE;
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#endif
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#ifdef AFMT_S32_BE
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case AFMT_S32_BE: return AF_FORMAT_S32_BE;
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#endif
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#ifdef AFMT_FLOAT
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case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE;
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#endif
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// SPECIALS
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#ifdef AFMT_MPEG
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case AFMT_MPEG: return AF_FORMAT_MPEG2;
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#endif
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#ifdef AFMT_AC3
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case AFMT_AC3: return AF_FORMAT_AC3_NE;
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#endif
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}
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mp_tmsg(MSGT_GLOBAL,MSGL_ERR,"[AO OSS] Unknown/Unsupported OSS format: %x.\n", format);
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return -1;
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}
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static char *dsp=PATH_DEV_DSP;
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static audio_buf_info zz;
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static int audio_fd=-1;
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static int prepause_space;
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static const char *oss_mixer_device = PATH_DEV_MIXER;
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static int oss_mixer_channel = SOUND_MIXER_PCM;
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#ifdef SNDCTL_DSP_GETPLAYVOL
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static int volume_oss4(ao_control_vol_t *vol, int cmd) {
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int v;
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if (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(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(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(int cmd,void *arg){
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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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{
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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(vol, cmd) == CONTROL_OK)
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return CONTROL_OK;
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#endif
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if(AF_FORMAT_IS_AC3(ao_data.format))
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return CONTROL_TRUE;
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if ((fd = open(oss_mixer_device, O_RDONLY)) != -1)
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{
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ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
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if (devs & (1 << oss_mixer_channel))
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{
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if (cmd == AOCONTROL_GET_VOLUME)
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{
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ioctl(fd, MIXER_READ(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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}
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else
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{
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v = ((int)vol->right << 8) | (int)vol->left;
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ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v);
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}
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}
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else
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{
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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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}
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return CONTROL_ERROR;
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}
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return CONTROL_UNKNOWN;
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}
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(int rate,const struct mp_chmap *channels,int format,int flags){
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char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
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int oss_format;
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char *mdev = mixer_device, *mchan = mixer_channel;
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mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,ao_data.channels.num,
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af_fmt2str_short(format));
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if (ao_subdevice) {
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char *m,*c;
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m = strchr(ao_subdevice,':');
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if(m) {
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c = strchr(m+1,':');
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if(c) {
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mchan = c+1;
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c[0] = '\0';
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}
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mdev = m+1;
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m[0] = '\0';
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}
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dsp = ao_subdevice;
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}
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if(mdev)
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oss_mixer_device=mdev;
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else
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oss_mixer_device=PATH_DEV_MIXER;
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if(mchan){
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int fd, devs, i;
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if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open mixer device %s: %s\n",
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oss_mixer_device, strerror(errno));
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}else{
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ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
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close(fd);
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for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
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if(!strcasecmp(mixer_channels[i], mchan)){
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if(!(devs & (1 << i))){
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Audio card mixer does not have channel '%s', using default.\n",mchan);
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i = SOUND_MIXER_NRDEVICES+1;
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break;
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}
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oss_mixer_channel = i;
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break;
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}
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}
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if(i==SOUND_MIXER_NRDEVICES){
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Audio card mixer does not have channel '%s', using default.\n",mchan);
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}
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}
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} else
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oss_mixer_channel = SOUND_MIXER_PCM;
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp);
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", oss_mixer_device);
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", mixer_channels[oss_mixer_channel]);
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#ifdef __linux__
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audio_fd=open(dsp, O_WRONLY | O_NONBLOCK);
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#else
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audio_fd=open(dsp, O_WRONLY);
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#endif
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if(audio_fd<0){
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open audio device %s: %s\n", dsp, strerror(errno));
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return 0;
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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(audio_fd, F_SETFL, 0) < 0) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't make file descriptor blocking: %s\n", strerror(errno));
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return 0;
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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(audio_fd, F_SETFD, FD_CLOEXEC);
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#endif
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if(AF_FORMAT_IS_AC3(format)) {
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ao_data.samplerate=rate;
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ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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}
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ac3_retry:
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if (AF_FORMAT_IS_AC3(format))
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format = AF_FORMAT_AC3_NE;
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ao_data.format=format;
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oss_format=format2oss(format);
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if (oss_format == -1) {
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#if BYTE_ORDER == BIG_ENDIAN
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oss_format=AFMT_S16_BE;
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#else
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oss_format=AFMT_S16_LE;
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#endif
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format=AF_FORMAT_S16_NE;
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}
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if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 ||
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oss_format != format2oss(format)) {
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mp_tmsg(MSGT_AO,MSGL_WARN, "[AO OSS] Can't set audio device %s to %s output, trying %s...\n", dsp,
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af_fmt2str_short(format), af_fmt2str_short(AF_FORMAT_S16_NE) );
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format=AF_FORMAT_S16_NE;
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goto ac3_retry;
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}
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#if 0
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if(oss_format!=format2oss(format))
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mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %s sample format! Broken audio or bad playback speed are possible! Try with '-af format'\n",audio_out_format_name(format));
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#endif
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ao_data.format = oss2format(oss_format);
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if (ao_data.format == -1) return 0;
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
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af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
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if(!AF_FORMAT_IS_AC3(format)) {
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struct mp_chmap_sel sel = {0};
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mp_chmap_sel_add_alsa_def(&sel);
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if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
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return 0;
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int reqchannels = ao_data.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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int nchannels = reqchannels;
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if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1 ||
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nchannels != reqchannels ) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", reqchannels);
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return 0;
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}
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}
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else {
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int c = reqchannels-1;
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if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", reqchannels);
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return 0;
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}
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if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, c + 1))
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return 0;
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}
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels.num, reqchannels);
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// set rate
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ao_data.samplerate=rate;
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ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
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}
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if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
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int r=0;
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mp_tmsg(MSGT_AO,MSGL_WARN,"[AO OSS] audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
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if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
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} else {
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ao_data.outburst=r;
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
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}
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} else {
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: 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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if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
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ao_data.outburst=zz.fragsize;
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}
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if(ao_data.buffersize==-1){
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// Measuring buffer size:
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void* data;
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ao_data.buffersize=0;
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#ifdef HAVE_AUDIO_SELECT
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data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
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while(ao_data.buffersize<0x40000){
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fd_set rfds;
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struct timeval tv;
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FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
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tv.tv_sec=0; tv.tv_usec = 0;
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if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
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write(audio_fd,data,ao_data.outburst);
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ao_data.buffersize+=ao_data.outburst;
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}
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free(data);
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if(ao_data.buffersize==0){
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\n *** Your audio driver DOES NOT support select() ***\n Recompile mpv with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
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return 0;
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}
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#endif
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}
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ao_data.bps=ao_data.channels.num;
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switch (ao_data.format & AF_FORMAT_BITS_MASK) {
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case AF_FORMAT_8BIT:
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break;
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case AF_FORMAT_16BIT:
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ao_data.bps*=2;
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break;
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case AF_FORMAT_24BIT:
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ao_data.bps*=3;
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break;
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case AF_FORMAT_32BIT:
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ao_data.bps*=4;
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break;
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}
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ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
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ao_data.bps*=ao_data.samplerate;
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return 1;
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}
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// close audio device
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static void uninit(int immed){
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if(audio_fd == -1) return;
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#ifdef SNDCTL_DSP_SYNC
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// to get the buffer played
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if (!immed)
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ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL);
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#endif
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#ifdef SNDCTL_DSP_RESET
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if (immed)
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ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
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#endif
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close(audio_fd);
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audio_fd = -1;
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}
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// stop playing and empty buffers (for seeking/pause)
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static void reset(void){
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int oss_format;
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uninit(1);
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audio_fd=open(dsp, O_WRONLY);
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if(audio_fd < 0){
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno));
|
|
return;
|
|
}
|
|
|
|
#if defined(FD_CLOEXEC) && defined(F_SETFD)
|
|
fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
|
|
#endif
|
|
|
|
oss_format = format2oss(ao_data.format);
|
|
if(AF_FORMAT_IS_AC3(ao_data.format))
|
|
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
|
|
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
|
|
if(!AF_FORMAT_IS_AC3(ao_data.format)) {
|
|
if (ao_data.channels.num > 2)
|
|
ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels.num);
|
|
else {
|
|
int c = ao_data.channels.num-1;
|
|
ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
|
|
}
|
|
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
|
|
}
|
|
}
|
|
|
|
// stop playing, keep buffers (for pause)
|
|
static void audio_pause(void)
|
|
{
|
|
prepause_space = get_space();
|
|
uninit(1);
|
|
}
|
|
|
|
// resume playing, after audio_pause()
|
|
static void audio_resume(void)
|
|
{
|
|
int fillcnt;
|
|
reset();
|
|
fillcnt = get_space() - prepause_space;
|
|
if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) {
|
|
void *silence = calloc(fillcnt, 1);
|
|
play(silence, fillcnt, 0);
|
|
free(silence);
|
|
}
|
|
}
|
|
|
|
|
|
// return: how many bytes can be played without blocking
|
|
static int get_space(void){
|
|
int playsize=ao_data.outburst;
|
|
|
|
#ifdef SNDCTL_DSP_GETOSPACE
|
|
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
|
|
// calculate exact buffer space:
|
|
playsize = zz.fragments*zz.fragsize;
|
|
return playsize;
|
|
}
|
|
#endif
|
|
|
|
// check buffer
|
|
#ifdef HAVE_AUDIO_SELECT
|
|
{ fd_set rfds;
|
|
struct timeval tv;
|
|
FD_ZERO(&rfds);
|
|
FD_SET(audio_fd, &rfds);
|
|
tv.tv_sec = 0;
|
|
tv.tv_usec = 0;
|
|
if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
|
|
}
|
|
#endif
|
|
|
|
return ao_data.outburst;
|
|
}
|
|
|
|
// plays 'len' bytes of 'data'
|
|
// it should round it down to outburst*n
|
|
// return: number of bytes played
|
|
static int play(void* data,int len,int flags){
|
|
if(len==0)
|
|
return len;
|
|
if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
|
|
len/=ao_data.outburst;
|
|
len*=ao_data.outburst;
|
|
}
|
|
len=write(audio_fd,data,len);
|
|
return len;
|
|
}
|
|
|
|
static int audio_delay_method=2;
|
|
|
|
// return: delay in seconds between first and last sample in buffer
|
|
static float get_delay(void){
|
|
/* Calculate how many bytes/second is sent out */
|
|
if(audio_delay_method==2){
|
|
#ifdef SNDCTL_DSP_GETODELAY
|
|
int r=0;
|
|
if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
|
|
return ((float)r)/(float)ao_data.bps;
|
|
#endif
|
|
audio_delay_method=1; // fallback if not supported
|
|
}
|
|
if(audio_delay_method==1){
|
|
// SNDCTL_DSP_GETOSPACE
|
|
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
|
|
return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
|
|
audio_delay_method=0; // fallback if not supported
|
|
}
|
|
return ((float)ao_data.buffersize)/(float)ao_data.bps;
|
|
}
|