Mercurial > mplayer.hg
view libao2/ao_oss.c @ 13035:8b823470d1fd
massive sync, rewording, consistency, fixes etc.
author | wight |
---|---|
date | Tue, 17 Aug 2004 17:51:54 +0000 |
parents | 94ba609e53f7 |
children | c1955840883d |
line wrap: on
line source
#include <stdio.h> #include <stdlib.h> #include <sys/ioctl.h> #include <unistd.h> #include <sys/time.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <errno.h> #include <string.h> //#include <sys/soundcard.h> #include "../config.h" #include "../mp_msg.h" #include "../mixer.h" #include "afmt.h" #include "audio_out.h" #include "audio_out_internal.h" static ao_info_t info = { "OSS/ioctl audio output", "oss", "A'rpi", "" }; /* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */ LIBAO_EXTERN(oss) static char *dsp=PATH_DEV_DSP; static audio_buf_info zz; static int audio_fd=-1; char *oss_mixer_device = PATH_DEV_MIXER; int oss_mixer_channel = SOUND_MIXER_PCM; // to set/get/query special features/parameters static int control(int cmd,void *arg){ switch(cmd){ case AOCONTROL_SET_DEVICE: dsp=(char*)arg; return CONTROL_OK; case AOCONTROL_GET_DEVICE: *(char**)arg=dsp; return CONTROL_OK; case AOCONTROL_QUERY_FORMAT: return CONTROL_TRUE; case AOCONTROL_GET_VOLUME: case AOCONTROL_SET_VOLUME: { ao_control_vol_t *vol = (ao_control_vol_t *)arg; int fd, v, devs; if(ao_data.format == AFMT_AC3) return CONTROL_TRUE; if ((fd = open(oss_mixer_device, O_RDONLY)) > 0) { ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); if (devs & (1 << oss_mixer_channel)) { if (cmd == AOCONTROL_GET_VOLUME) { ioctl(fd, MIXER_READ(oss_mixer_channel), &v); vol->right = (v & 0xFF00) >> 8; vol->left = v & 0x00FF; } else { v = ((int)vol->right << 8) | (int)vol->left; ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v); } } else { close(fd); return CONTROL_ERROR; } close(fd); return CONTROL_OK; } } return CONTROL_ERROR; } return CONTROL_UNKNOWN; } // open & setup audio device // return: 1=success 0=fail static int init(int rate,int channels,int format,int flags){ char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES; mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels, audio_out_format_name(format)); if (ao_subdevice) dsp = ao_subdevice; if(mixer_device) oss_mixer_device=mixer_device; if(mixer_channel){ int fd, devs, i; if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){ mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't open mixer device %s: %s\n", oss_mixer_device, strerror(errno)); }else{ ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); close(fd); for (i=0; i<SOUND_MIXER_NRDEVICES; i++){ if(!strcasecmp(mixer_channels[i], mixer_channel)){ if(!(devs & (1 << i))){ mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Audio card mixer does not have channel '%s' using default\n", mixer_channel); i = SOUND_MIXER_NRDEVICES+1; break; } oss_mixer_channel = i; break; } } if(i==SOUND_MIXER_NRDEVICES){ mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't find mixer channel '%s' using default\n", mixer_channel); } } } mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp); mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", oss_mixer_device); mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", mixer_channels[oss_mixer_channel]); #ifdef __linux__ audio_fd=open(dsp, O_WRONLY | O_NONBLOCK); #else audio_fd=open(dsp, O_WRONLY); #endif if(audio_fd<0){ mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't open audio device %s: %s\n", dsp, strerror(errno)); return 0; } #ifdef __linux__ /* Remove the non-blocking flag */ if(fcntl(audio_fd, F_SETFL, 0) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't make filedescriptor blocking: %s\n", strerror(errno)); return 0; } #endif #if defined(FD_CLOEXEC) && defined(F_SETFD) fcntl(audio_fd, F_SETFD, FD_CLOEXEC); #endif if(format == AFMT_AC3) { ao_data.samplerate=rate; ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); } ac3_retry: ao_data.format=format; if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format)<0 || ao_data.format != format) if(format == AFMT_AC3){ mp_msg(MSGT_AO,MSGL_WARN,"Can't set audio device %s to AC3 output, trying S16...\n", dsp); #ifdef WORDS_BIGENDIAN format=AFMT_S16_BE; #else format=AFMT_S16_LE; #endif goto ac3_retry; } mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n", audio_out_format_name(ao_data.format), audio_out_format_name(format)); #if 0 if(ao_data.format!=format) 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 '-aop list=format'\n",audio_out_format_name(format)); #endif ao_data.channels = channels; if(format != AFMT_AC3) { // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it if (ao_data.channels > 2) { if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 || ao_data.channels != channels ) { mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Failed to set audio device to %d channels\n", channels); return 0; } } else { int c = ao_data.channels-1; if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) { mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Failed to set audio device to %d channels\n", ao_data.channels); return 0; } ao_data.channels=c+1; } mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels); // set rate ao_data.samplerate=rate; ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate); #if 0 if(ao_data.samplerate!=rate) mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %d Hz samplerate! A-V sync problems or wrong speed are possible! Try with '-aop list=resample:fout=%d'\n",rate,ao_data.samplerate); #endif } if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){ int r=0; mp_msg(MSGT_AO,MSGL_WARN,"audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n"); if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){ mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst); } else { ao_data.outburst=r; mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst); } } else { mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n", zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes); if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes; ao_data.outburst=zz.fragsize; } if(ao_data.buffersize==-1){ // Measuring buffer size: void* data; ao_data.buffersize=0; #ifdef HAVE_AUDIO_SELECT data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst); while(ao_data.buffersize<0x40000){ 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)) break; write(audio_fd,data,ao_data.outburst); ao_data.buffersize+=ao_data.outburst; } free(data); if(ao_data.buffersize==0){ mp_msg(MSGT_AO,MSGL_ERR,"\n *** Your audio driver DOES NOT support select() ***\n" "Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n"); return 0; } #endif } ao_data.bps=ao_data.channels; if(ao_data.format != AFMT_U8 && ao_data.format != AFMT_S8) ao_data.bps*=2; ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down ao_data.bps*=ao_data.samplerate; return 1; } // close audio device static void uninit(int immed){ if(audio_fd == -1) return; #ifdef SNDCTL_DSP_SYNC // to get the buffer played if (!immed) ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL); #endif #ifdef SNDCTL_DSP_RESET if (immed) ioctl(audio_fd, SNDCTL_DSP_RESET, NULL); #endif close(audio_fd); audio_fd = -1; } // stop playing and empty buffers (for seeking/pause) static void reset(){ uninit(1); audio_fd=open(dsp, O_WRONLY); if(audio_fd < 0){ mp_msg(MSGT_AO,MSGL_ERR,"\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 ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format); if(ao_data.format != AFMT_AC3) { if (ao_data.channels > 2) ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels); else { int c = ao_data.channels-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() { uninit(1); } // resume playing, after audio_pause() static void audio_resume() { reset(); } // return: how many bytes can be played without blocking static int get_space(){ 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){ len/=ao_data.outburst; len=write(audio_fd,data,len*ao_data.outburst); return len; } static int audio_delay_method=2; // return: delay in seconds between first and last sample in buffer static float get_delay(){ /* 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; }