Mercurial > mplayer.hg
view libao2/ao_sgi.c @ 3639:64ee21b3bd09
Modified the sync code once again, commented out hardware pts sync (I'll likely burn in hell before understanding how to get this bastard to sync well)
Added automagic setup of aspect ratio, will remove the "aspect-bug" (I hope). As well as please you rich 16:9 doods ;)
author | mswitch |
---|---|
date | Thu, 20 Dec 2001 20:50:35 +0000 |
parents | 981a9e5118ce |
children | 12b1790038b0 |
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/* ao_sgi - sgi/irix output plugin for MPlayer 22oct2001 oliver.schoenbrunner@jku.at */ #include <stdio.h> #include <stdlib.h> #include <dmedia/audio.h> #include "audio_out.h" #include "audio_out_internal.h" static ao_info_t info = { "sgi audio output", "sgi", "kopflos", "" }; LIBAO_EXTERN(sgi) static ALconfig ao_config; static ALport ao_port; // to set/get/query special features/parameters static int control(int cmd, int arg){ printf("ao_sgi, control\n"); return -1; } // open & setup audio device // return: 1=success 0=fail static int init(int rate, int channels, int format, int flags) { printf("ao_sgi, init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format)); { /* from /usr/share/src/dmedia/audio/setrate.c */ int fd; int rv; double frate; ALpv x[2]; rv = alGetResourceByName(AL_SYSTEM, "out.analog", AL_DEVICE_TYPE); if (!rv) { printf("ao_sgi, play: invalid device\n"); return 0; } frate = rate; x[0].param = AL_RATE; x[0].value.ll = alDoubleToFixed(rate); x[1].param = AL_MASTER_CLOCK; x[1].value.i = AL_CRYSTAL_MCLK_TYPE; if (alSetParams(rv,x, 2)<0) { printf("ao_sgi, init: setparams failed: %s\n", alGetErrorString(oserror())); printf("ao_sgi, init: could not set desired samplerate\n"); } if (x[0].sizeOut < 0) { printf("ao_sgi, init: AL_RATE was not accepted on the given resource\n"); } if (alGetParams(rv,x, 1)<0) { printf("ao_sgi, init: getparams failed: %s\n", alGetErrorString(oserror())); } if (frate != alFixedToDouble(x[0].value.ll)) { printf("ao_sgi, init: samplerate is now %lf (desired rate is %lf)\n", alFixedToDouble(x[0].value.ll), frate); } } ao_data.buffersize=131072; ao_data.outburst = ao_data.buffersize/16; ao_data.channels = channels; ao_config = alNewConfig(); if (!ao_config) { printf("ao_sgi, init: %s\n", alGetErrorString(oserror())); return 0; } if(channels == 2) alSetChannels(ao_config, AL_STEREO); else alSetChannels(ao_config, AL_MONO); alSetWidth(ao_config, AL_SAMPLE_16); alSetSampFmt(ao_config, AL_SAMPFMT_TWOSCOMP); alSetQueueSize(ao_config, 48000); if (alSetDevice(ao_config, AL_DEFAULT_OUTPUT) < 0) { printf("ao_sgi, init: %s\n", alGetErrorString(oserror())); return 0; } ao_port = alOpenPort("mplayer", "w", ao_config); if (!ao_port) { printf("ao_sgi, init: Unable to open audio channel: %s\n", alGetErrorString(oserror())); return 0; } // printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config); return 1; } // close audio device static void uninit() { /* TODO: samplerate should be set back to the value before mplayer was started! */ printf("ao_sgi, uninit: ...\n"); if (ao_port) { while(alGetFilled(ao_port) > 0) sginap(1); alClosePort(ao_port); alFreeConfig(ao_config); } } // stop playing and empty buffers (for seeking/pause) static void reset() { printf("ao_sgi, reset: ...\n"); } // stop playing, keep buffers (for pause) static void audio_pause() { printf("ao_sgi, audio_pause: ...\n"); } // resume playing, after audio_pause() static void audio_resume() { printf("ao_sgi, audio_resume: ...\n"); } // return: how many bytes can be played without blocking static int get_space() { // printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_outburst); // printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port)); return alGetFillable(ao_port)*(2*ao_data.channels); } // 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) { // printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config); // printf("channels %d\n", ao_channels); alWriteFrames(ao_port, data, len/(2*ao_data.channels)); return len; } // return: delay in seconds between first and last sample in buffer static float get_delay(){ // printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize); return 0; }