mirror of
https://github.com/mpv-player/mpv
synced 2024-12-11 09:25:56 +00:00
7be3e8694b
send 0-samples according to the amount of data lost during pause. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@23829 b3059339-0415-0410-9bf9-f77b7e298cf2
534 lines
14 KiB
C
534 lines
14 KiB
C
#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 "mp_msg.h"
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#include "mixer.h"
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#include "help_mp.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 "libaf/af_format.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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static 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_U24_LE
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case AF_FORMAT_U24_LE: return AFMT_U24_LE;
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#endif
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#ifdef AFMT_U24_BE
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case AF_FORMAT_U24_BE: return AFMT_U24_BE;
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#endif
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#ifdef AFMT_S24_LE
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case AF_FORMAT_S24_LE: return AFMT_S24_LE;
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#endif
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#ifdef AFMT_S24_BE
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case AF_FORMAT_S24_BE: return AFMT_S24_BE;
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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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case AF_FORMAT_MU_LAW: return AFMT_MU_LAW;
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case AF_FORMAT_A_LAW: return AFMT_A_LAW;
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case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM;
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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: 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_U24_LE
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case AFMT_U24_LE: return AF_FORMAT_U24_LE;
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#endif
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#ifdef AFMT_U24_BE
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case AFMT_U24_BE: return AF_FORMAT_U24_BE;
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#endif
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#ifdef AFMT_S24_LE
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case AFMT_S24_LE: return AF_FORMAT_S24_LE;
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#endif
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#ifdef AFMT_S24_BE
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case AFMT_S24_BE: return AF_FORMAT_S24_BE;
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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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case AFMT_MU_LAW: return AF_FORMAT_MU_LAW;
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case AFMT_A_LAW: return AF_FORMAT_A_LAW;
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case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM;
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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;
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#endif
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}
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mp_msg(MSGT_GLOBAL,MSGL_ERR,MSGTR_AO_OSS_UnknownUnsupportedFormat, 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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// 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_SET_DEVICE:
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dsp=(char*)arg;
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return CONTROL_OK;
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case AOCONTROL_GET_DEVICE:
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*(char**)arg=dsp;
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return CONTROL_OK;
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#ifdef SNDCTL_DSP_GETFMTS
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case AOCONTROL_QUERY_FORMAT:
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{
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int format;
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if (!ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format))
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if (format & (int)arg)
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return CONTROL_TRUE;
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return CONTROL_FALSE;
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}
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#endif
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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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if(ao_data.format == AF_FORMAT_AC3)
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return CONTROL_TRUE;
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if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
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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,int 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,channels,
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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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenMixer,
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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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenDev, 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantMakeFd, 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(format == AF_FORMAT_AC3) {
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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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ao_data.format=format;
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oss_format=format2oss(format);
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if (oss_format == -1) {
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#ifdef WORDS_BIGENDIAN
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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_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSet, 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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ao_data.channels = channels;
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if(format != AF_FORMAT_AC3) {
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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 (ao_data.channels > 2) {
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if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
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ao_data.channels != channels ) {
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mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, channels);
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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 = ao_data.channels-1;
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if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
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mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, ao_data.channels);
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return 0;
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}
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ao_data.channels=c+1;
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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, channels);
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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_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_OSS_CantUseGetospace);
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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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantUseSelect);
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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;
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if(ao_data.format != AF_FORMAT_U8 && ao_data.format != AF_FORMAT_S8)
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ao_data.bps*=2;
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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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantReopen, strerror(errno));
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return;
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}
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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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oss_format = format2oss(ao_data.format);
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ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
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if(ao_data.format != AF_FORMAT_AC3) {
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if (ao_data.channels > 2)
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ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
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else {
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int c = ao_data.channels-1;
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ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
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}
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ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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}
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause(void)
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{
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prepause_space = get_space();
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uninit(1);
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}
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// resume playing, after audio_pause()
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static void audio_resume(void)
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{
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int fillcnt;
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reset();
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fillcnt = get_space() - prepause_space;
|
|
if (fillcnt > 0) {
|
|
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;
|
|
if (playsize > MAX_OUTBURST)
|
|
playsize = (MAX_OUTBURST / zz.fragsize) * 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;
|
|
}
|