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mirror of https://github.com/mpv-player/mpv synced 2024-10-30 04:46:41 +01:00
mpv/stream/ai_oss.c
wm4 b745c2d005 audio: drop swapped-endian audio formats
Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.

From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.

This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.
2014-09-23 23:09:25 +02:00

159 lines
4.6 KiB
C

/*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include "config.h"
#include <string.h> /* strerror */
#include <fcntl.h>
#include <errno.h>
#include <sys/ioctl.h>
#if HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
#else
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <linux/soundcard.h>
#endif
#endif
#include "osdep/io.h"
#include "audio_in.h"
#include "common/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_VERBOSE(ai, "ioctl dsp channels: %d\n",
err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_CHANNELS, &ioctl_param));
if (err < 0) {
MP_ERR(ai, "Unable to set channel count: %d\n",
ai->req_channels);
return -1;
}
ai->channels = ioctl_param;
}
else
{
ioctl_param = (ai->req_channels == 2);
MP_VERBOSE(ai, "ioctl dsp stereo: %d (req: %d)\n",
err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_STEREO, &ioctl_param),
ioctl_param);
if (err < 0) {
MP_ERR(ai, "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 | O_CLOEXEC);
if (ai->oss.audio_fd < 0)
{
MP_ERR(ai, "Unable to open '%s': %s\n",
ai->oss.device, strerror(errno));
return -1;
}
ioctl_param = 0 ;
MP_VERBOSE(ai, "ioctl dsp getfmt: %d\n",
ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETFMTS, &ioctl_param));
MP_VERBOSE(ai, "Supported formats: %x\n", ioctl_param);
if (!(ioctl_param & AFMT_S16_NE))
MP_ERR(ai, "unsupported format\n");
ioctl_param = AFMT_S16_NE;
MP_VERBOSE(ai, "ioctl dsp setfmt: %d\n",
err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETFMT, &ioctl_param));
if (err < 0) {
MP_ERR(ai, "Unable to set audio format.");
return -1;
}
if (ai_oss_set_channels(ai) < 0) return -1;
ioctl_param = ai->req_samplerate;
MP_VERBOSE(ai, "ioctl dsp speed: %d\n",
err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &ioctl_param));
if (err < 0) {
MP_ERR(ai, "Unable to set samplerate: %d\n",
ai->req_samplerate);
return -1;
}
ai->samplerate = ioctl_param;
MP_VERBOSE(ai, "ioctl dsp trigger: %d\n",
ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETTRIGGER, &ioctl_param));
MP_VERBOSE(ai, "trigger: %x\n", ioctl_param);
ioctl_param = PCM_ENABLE_INPUT;
MP_VERBOSE(ai, "ioctl dsp trigger: %d\n",
err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETTRIGGER, &ioctl_param));
if (err < 0) {
MP_ERR(ai, "Unable to set trigger: %d\n",
PCM_ENABLE_INPUT);
}
ai->blocksize = 0;
MP_VERBOSE(ai, "ioctl dsp getblocksize: %d\n",
err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETBLKSIZE, &ai->blocksize));
if (err < 0) {
MP_ERR(ai, "Unable to get block size!\n");
}
MP_VERBOSE(ai, "blocksize: %d\n", ai->blocksize);
// correct the blocksize to a reasonable value
if (ai->blocksize <= 0) {
ai->blocksize = 4096*ai->channels*2;
MP_ERR(ai, "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_ERR(ai, "Audio block size too low, setting to %d!\n", ai->blocksize);
}
ai->samplesize = 16;
ai->bytes_per_sample = 2;
return 0;
}