Mercurial > mplayer.hg
view libao2/ao_sgi.c @ 15533:ddf15d233d58
Do not switch to audio tracks whose codec private data differs from the main audio track's as this will most likely result in messed up audio output. Patch by Michael Behrisch <list () behrisch ! de>
author | mosu |
---|---|
date | Sat, 21 May 2005 06:50:08 +0000 |
parents | d313f591d1a4 |
children | 27a2bc4aad72 |
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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" #include "mp_msg.h" #include "help_mp.h" static ao_info_t info = { "sgi audio output", "sgi", "Oliver Schoenbrunner", "" }; LIBAO_EXTERN(sgi) static ALconfig ao_config; static ALport ao_port; static int sample_rate; static int queue_size; // to set/get/query special features/parameters static int control(int cmd, void *arg){ mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_INFO); return -1; } // open & setup audio device // return: 1=success 0=fail static int init(int rate, int channels, int format, int flags) { mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(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) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice); 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) { mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror())); } if (x[0].sizeOut < 0) { mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate); } if (alGetParams(rv,x, 1)<0) { mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror())); } if (frate != alFixedToDouble(x[0].value.ll)) { mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, alFixedToDouble(x[0].value.ll), frate); } sample_rate = (int)frate; } ao_data.buffersize=131072; ao_data.outburst = ao_data.buffersize/16; ao_data.channels = channels; ao_config = alNewConfig(); if (!ao_config) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, 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) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror())); return 0; } ao_port = alOpenPort("mplayer", "w", ao_config); if (!ao_port) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, alGetErrorString(oserror())); return 0; } // printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config); queue_size = alGetQueueSize(ao_config); return 1; } // close audio device static void uninit(int immed) { /* TODO: samplerate should be set back to the value before mplayer was started! */ mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Uninit); if (ao_port) { if (!immed) while(alGetFilled(ao_port) > 0) sginap(1); alClosePort(ao_port); alFreeConfig(ao_config); } } // stop playing and empty buffers (for seeking/pause) static void reset() { mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Reset); } // stop playing, keep buffers (for pause) static void audio_pause() { mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_PauseInfo); } // resume playing, after audio_pause() static void audio_resume() { mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_ResumeInfo); } // 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; return (float)queue_size/((float)sample_rate); }