/* * *  output_linux.c *     *	Copyright (C) Aaron Holtzman - May 1999 * *  This file is part of ac3dec, a free Dolby AC-3 stream decoder. *	 *  ac3dec 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, or (at your option) *  any later version. *    *  ac3dec 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 GNU Make; see the file COPYING.  If not, write to *  the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA.  * * */#ifdef HAVE_CONFIG_H#  include <config.h>#endif#include <stdio.h>#include <errno.h>#include <string.h>#include <unistd.h>#include <fcntl.h>#include <math.h>#include <sys/soundcard.h>#include <sys/ioctl.h>#include "ac3.h"#include "decode.h"#include "debug.h"#include "output.h"#include "downmix.h"#include "ring_buffer.h"#define BUFFER_SIZE 1024 static char dev[] = "/dev/dsp";static int fd;//FIXME uncomment all the matlab calls in this module//      and you'll get a handy matlab output file//      of the signal.//#include "matlab.h"//static matlab_file_t *foo;///* * open the audio device for writing to */int output_open(int bits, int rate, int channels){  int tmp;    /*   * Open the device driver   */	fd=open(dev,O_WRONLY | O_NDELAY);  if(fd < 0)   {    dprintf("%s: Opening audio device %s\n",        strerror(errno), dev);    goto ERR;  }	dprintf("Opened audio device \"%s\"\n",dev);	fcntl(fd,F_SETFL,O_NONBLOCK);   tmp = (32 << 16) | (ffs(1024) - 1);   ioctl(fd,SNDCTL_DSP_SETFRAGMENT,&tmp);   tmp = bits;  ioctl(fd,SNDCTL_DSP_SAMPLESIZE,&tmp);  tmp = channels == 2 ? 1 : 0;  ioctl(fd,SNDCTL_DSP_STEREO,&tmp);  tmp = rate;  ioctl(fd,SNDCTL_DSP_SPEED, &tmp);	/* Initialize the ring buffer */	rb_init();	//FIXME remove	//	foo = matlab_open("foo.m");	//		return 1;ERR:  if(fd >= 0) { close(fd); }  return 0;}voidoutput_flush(void){	int i,j = 0;	sint_16 *out_buf = 0;	i = 0;	do	{		out_buf = rb_begin_read();		if(out_buf)			i = write(fd, out_buf,BUFFER_SIZE);		else			break;		if(i == BUFFER_SIZE)		{			rb_end_read();			j++;		}	}	while(i == BUFFER_SIZE);		//FIXME remove	//fprintf(stderr,"(output) Flushed %d blocks, wrote %d bytes last frame\n",j,i);}/* * play the sample to the already opened file descriptor */void output_play(bsi_t *bsi,stream_samples_t *samples){  int i;	float *left,*right;	float norm = 1.0;	float left_tmp = 0.0;	float right_tmp = 0.0;	sint_16 *out_buf;	if(fd < 0)		return;	out_buf = rb_begin_write();	/* Keep trying to dump frames from the ring buffer until we get a 	 * write slot available */	while(!out_buf)	{		usleep(5000);		output_flush();		out_buf = rb_begin_write();	} 	//FIXME remove	//matlab_write(foo,samples->channel[0],512);		//Downmix if necessary 	downmix(bsi,samples);	//Determine a normalization constant if the signal exceeds 	//100% digital [-1.0,1.0]	//	//perhaps use the dynamic range info to do this instead	for(i=0; i< 256;i++)	{    left_tmp = samples->channel[0][i];    right_tmp = samples->channel[1][i];		if(left_tmp > norm)			norm = left_tmp;		if(left_tmp < -norm)			norm = -left_tmp;		if(right_tmp > norm)			norm = right_tmp;		if(right_tmp < -norm)			norm = -right_tmp; 	}	norm = 32000.0/norm;	/* Take the floating point audio data and convert it into	 * 16 bit signed PCM data */	left = samples->channel[0];	right = samples->channel[1];	for(i=0; i < 256; i++)	{	//	if((fabs(*left * norm) > 32768.0) || (fabs(*right * norm) > 32768.0))	//		printf("clipping (%f, %f)\n",*left,*right);		out_buf[i * 2 ]    = (sint_16) (*left++  * norm);		out_buf[i * 2 + 1] = (sint_16) (*right++ * norm);	}	rb_end_write();}voidoutput_close(void){	close(fd);}