Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 13568:1cb0e1833515
Currently vbeGetProtModeInfo call the 0x4f0a function of int 10h the get
a simple 32 bits protected mode interface to some VESA functions. This
protected mode interface is interesting because it's quicker than the
raw int 10h interface.
Unfortunatly, begining with VBE 3.0, the 0x4f0a function is optional,
and some video cards don't implement it (3dfx, intel 845/855/865...).
This protected mode interface is then only used in vbeSetWindow and
vbeSetDisplayStart :
?- vbeSetWindow already implement an alternative methode if protected
mode interface is not available.
?- vbeSetDisplayStart also contain an alternative implementation, but
this one is disabled with a #if 0. I don't exactly know why because
it works well !
So currently, cards which don't have the 0x4f0a function are not
supported. This patch correct this.
?- vbeGetProtModeInfo failure is not fatal.
?- vbeSetDisplayStart has it's alternative implementation reenabled.
? ?it's used only with cards which don't have the 0x4f0a function
? ?so this won't make any difference for cards which were already
? ?working.
This patch also make the failure of vbeGetModeInfo not fatal. The
VBE 3.0 standard state that GetModeInfo can fail with some mode
which are listed as supported if the mode can't be used in the
current situation (not enough video memory for example). So a
failure of vbeGetModeInfo don't mean that other modes won't work
and should really not be fatal.
patch by Aurelien Jacobs <aurel@gnuage.org>
author | faust3 |
---|---|
date | Wed, 06 Oct 2004 08:42:13 +0000 |
parents | c1955840883d |
children | a92101a7eb49 |
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line source
#include "config.h" #include <stdio.h> #include <stdlib.h> #include <string.h> #include "bswap.h" #include "afmt.h" #include "audio_out.h" #include "audio_out_internal.h" #include "../mp_msg.h" #include "../help_mp.h" static ao_info_t info = { "RAW PCM/WAVE file writer audio output", "pcm", "Atmosfear", "" }; LIBAO_EXTERN(pcm) extern int vo_pts; char *ao_outputfilename = NULL; int ao_pcm_waveheader = 1; #define WAV_ID_RIFF 0x46464952 /* "RIFF" */ #define WAV_ID_WAVE 0x45564157 /* "WAVE" */ #define WAV_ID_FMT 0x20746d66 /* "fmt " */ #define WAV_ID_DATA 0x61746164 /* "data" */ #define WAV_ID_PCM 0x0001 struct WaveHeader { uint32_t riff; uint32_t file_length; uint32_t wave; uint32_t fmt; uint32_t fmt_length; uint16_t fmt_tag; uint16_t channels; uint32_t sample_rate; uint32_t bytes_per_second; uint16_t block_align; uint16_t bits; uint32_t data; uint32_t data_length; }; /* init with default values */ static struct WaveHeader wavhdr = { le2me_32(WAV_ID_RIFF), /* same conventions than in sox/wav.c/wavwritehdr() */ 0, //le2me_32(0x7ffff024), le2me_32(WAV_ID_WAVE), le2me_32(WAV_ID_FMT), le2me_32(16), le2me_16(WAV_ID_PCM), le2me_16(2), le2me_32(44100), le2me_32(192000), le2me_16(4), le2me_16(16), le2me_32(WAV_ID_DATA), 0, //le2me_32(0x7ffff000) }; static FILE *fp = NULL; // to set/get/query special features/parameters static int control(int cmd,void *arg){ return -1; } // open & setup audio device // return: 1=success 0=fail static int init(int rate,int channels,int format,int flags){ int bits; if(!ao_outputfilename) { ao_outputfilename = strdup(ao_pcm_waveheader ? "audiodump.wav" : "audiodump.pcm"); } /* bits is only equal to format if (format == 8) or (format == 16); this means that the following "if" is a kludge and should really be a switch to be correct in all cases */ bits=8; switch(format){ case AFMT_S8: format=AFMT_U8; case AFMT_U8: break; default: format=AFMT_S16_LE; bits=16; break; } ao_data.outburst = 65536; ao_data.buffersize= 2*65536; ao_data.channels=channels; ao_data.samplerate=rate; ao_data.format=format; ao_data.bps=channels*rate*(bits/8); wavhdr.channels = le2me_16(ao_data.channels); wavhdr.sample_rate = le2me_32(ao_data.samplerate); wavhdr.bytes_per_second = le2me_32(ao_data.bps); wavhdr.bits = le2me_16(bits); wavhdr.data_length=le2me_32(0x7ffff000); wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format)); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo); fp = fopen(ao_outputfilename, "wb"); if(fp) { if(ao_pcm_waveheader){ /* Reserve space for wave header */ fwrite(&wavhdr,sizeof(wavhdr),1,fp); wavhdr.file_length=wavhdr.data_length=0; } return 1; } mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile, ao_outputfilename); return 0; } // close audio device static void uninit(int immed){ if(ao_pcm_waveheader && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */ wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; wavhdr.file_length = le2me_32(wavhdr.file_length); wavhdr.data_length = le2me_32(wavhdr.data_length); fwrite(&wavhdr,sizeof(wavhdr),1,fp); } fclose(fp); } // stop playing and empty buffers (for seeking/pause) static void reset(){ } // stop playing, keep buffers (for pause) static void audio_pause() { // for now, just call reset(); reset(); } // resume playing, after audio_pause() static void audio_resume() { } // return: how many bytes can be played without blocking static int get_space(){ if(vo_pts) return ao_data.pts < vo_pts ? ao_data.outburst : 0; 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){ // let libaf to do the conversion... #if 0 //#ifdef WORDS_BIGENDIAN if (ao_data.format == AFMT_S16_LE) { unsigned short *buffer = (unsigned short *) data; register int i; for(i = 0; i < len/2; ++i) { buffer[i] = le2me_16(buffer[i]); } } #endif //printf("PCM: Writing chunk!\n"); fwrite(data,len,1,fp); if(ao_pcm_waveheader) wavhdr.data_length += len; return len; } // return: delay in seconds between first and last sample in buffer static float get_delay(){ return 0.0; }