Mercurial > mplayer.hg
view stream/ai_oss.c @ 32334:541a774ce8e4
Drop removal of AAC encoder from list of FFmpeg encoders.
The original reason to drop this particular encoder, that it
failed to build on many platforms, should no longer be the case.
author | diego |
---|---|
date | Fri, 01 Oct 2010 16:04:43 +0000 |
parents | ce0122361a39 |
children |
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/* * 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> #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" #include "help_mp.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, MSGTR_MPDEMUX_AIOSS_Unable2SetChanCount, 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, MSGTR_MPDEMUX_AIOSS_Unable2SetStereo, 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, MSGTR_MPDEMUX_AIOSS_Unable2Open, 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, MSGTR_MPDEMUX_AIOSS_UnsupportedFmt); 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, MSGTR_MPDEMUX_AIOSS_Unable2SetAudioFmt); 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, MSGTR_MPDEMUX_AIOSS_Unable2SetSamplerate, 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, MSGTR_MPDEMUX_AIOSS_Unable2SetTrigger, 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, MSGTR_MPDEMUX_AIOSS_Unable2GetBlockSize); } 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, MSGTR_MPDEMUX_AIOSS_AudioBlockSizeZero, ai->blocksize); } else if (ai->blocksize < 4096*ai->channels*2) { ai->blocksize *= 4096*ai->channels*2/ai->blocksize; mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIOSS_AudioBlockSize2Low, ai->blocksize); } ai->samplesize = 16; ai->bytes_per_sample = 2; return 0; }