view libmpdemux/ai_oss.c @ 12337:6f1b4c989914

soft skipping for mencoder. rather than skipping decoding/filtering frames that will be skipped, mencoded tells vf_softskip (if present) that it should drop the next frame. this allows filters that need to see every input frame (inverse telecine, denoise3d, ...) to see skipped frames before they get dropped. in principle, a smarter softskip filter could be written that would buffer frames and choose to drop the one with least change, rather than strictly dropping the next one.
author rfelker
date Wed, 28 Apr 2004 04:29:17 +0000
parents 10fb7a24b4fb
children dfbe8cd0e081
line wrap: on
line source

#include <stdio.h>
#include <stdlib.h>

#include "config.h"

#if defined(USE_TV) && (defined(HAVE_TV_V4L) || defined(HAVE_TV_V4L2)) && defined(USE_OSS_AUDIO)

#include <string.h> /* strerror */
#include <fcntl.h>
#include <errno.h>
#include <sys/ioctl.h>

#ifdef HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
#else
#ifdef HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <linux/soundcard.h>
#endif
#endif

#include "audio_in.h"
#include "mp_msg.h"

int ai_oss_set_samplerate(audio_in_t *ai)
{
    int tmp = ai->req_samplerate;
    if (ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &tmp) == -1) return -1;
    ai->samplerate = tmp;
    return 0;
}

int ai_oss_set_channels(audio_in_t *ai)
{
    int err;
    int ioctl_param;

    if (ai->req_channels > 2)
    {
	ioctl_param = ai->req_channels;
	mp_msg(MSGT_TV, MSGL_V, "ioctl dsp channels: %d\n",
	       err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_CHANNELS, &ioctl_param));
	if (err < 0) {
	    mp_msg(MSGT_TV, MSGL_ERR, "Unable to set channel count: %d\n",
		   ai->req_channels);
	    return -1;
	}
	ai->channels = ioctl_param;
    }
    else
    {
	ioctl_param = (ai->req_channels == 2);
	mp_msg(MSGT_TV, MSGL_V, "ioctl dsp stereo: %d (req: %d)\n",
	       err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_STEREO, &ioctl_param),
	       ioctl_param);
	if (err < 0) {
	    mp_msg(MSGT_TV, MSGL_ERR, "Unable to set stereo: %d\n",
		   ai->req_channels == 2);
	    return -1;
	}
	ai->channels = ioctl_param ? 2 : 1;
    }
    return 0;
}

int ai_oss_init(audio_in_t *ai)
{
    int err;
    int ioctl_param;

    ai->oss.audio_fd = open(ai->oss.device, O_RDONLY);
    if (ai->oss.audio_fd < 0)
    {
	mp_msg(MSGT_TV, MSGL_ERR, "unable to open '%s': %s\n",
	       ai->oss.device, strerror(errno));
	return -1;
    }
	
    ioctl_param = 0 ;
    mp_msg(MSGT_TV, MSGL_V, "ioctl dsp getfmt: %d\n",
	   ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETFMTS, &ioctl_param));
	
    mp_msg(MSGT_TV, MSGL_V, "Supported formats: %x\n", ioctl_param);
    if (!(ioctl_param & AFMT_S16_LE))
	mp_msg(MSGT_TV, MSGL_ERR, "notsupported format\n");

    ioctl_param = AFMT_S16_LE;
    mp_msg(MSGT_TV, MSGL_V, "ioctl dsp setfmt: %d\n",
	   err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETFMT, &ioctl_param));
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Unable to set audio format.");
	return -1;
    }

    if (ai_oss_set_channels(ai) < 0) return -1;
	
    ioctl_param = ai->req_samplerate;
    mp_msg(MSGT_TV, MSGL_V, "ioctl dsp speed: %d\n",
	   err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &ioctl_param));
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Unable to set samplerate: %d\n",
	       ai->req_samplerate);
	return -1;
    }
    ai->samplerate = ioctl_param;

    mp_msg(MSGT_TV, MSGL_V, "ioctl dsp trigger: %d\n",
	   ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETTRIGGER, &ioctl_param));
    mp_msg(MSGT_TV, MSGL_V, "trigger: %x\n", ioctl_param);
    ioctl_param = PCM_ENABLE_INPUT;
    mp_msg(MSGT_TV, MSGL_V, "ioctl dsp trigger: %d\n",
	   err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETTRIGGER, &ioctl_param));
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Unable to set trigger: %d\n",
	       PCM_ENABLE_INPUT);
    }

    ai->blocksize = 0;
    mp_msg(MSGT_TV, MSGL_V, "ioctl dsp getblocksize: %d\n",
	   err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETBLKSIZE, &ai->blocksize));
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Unable to get block size!\n");
    }
    mp_msg(MSGT_TV, MSGL_V, "blocksize: %d\n", ai->blocksize);

    // correct the blocksize to a reasonable value
    if (ai->blocksize <= 0) {
	ai->blocksize = 4096*ai->channels*2;
	mp_msg(MSGT_TV, MSGL_ERR, "audio block size is zero, setting to %d!\n", ai->blocksize);
    } else if (ai->blocksize < 4096*ai->channels*2) {
	ai->blocksize *= 4096*ai->channels*2/ai->blocksize;
	mp_msg(MSGT_TV, MSGL_ERR, "audio block size too low, setting to %d!\n", ai->blocksize);
    }

    ai->samplesize = 16;
    ai->bytes_per_sample = 2;

    return 0;
}

#endif /* USE_OSS_AUDIO */