view libao2/ao_null.c @ 8514:a1ff87c254ff

I have rewritten the gif89a vo in order to solve some problems I had with it. These are: 1) current code is messy 2) poor comments, if any 3) inaccurate frame dropping and delay code 4) output filename hardcoded 5) output framerate as integer You may specify the output filename and framerate like so: -vo gif89a:4.33 4.33 fps output -vo gif89a:some.gif output to some.gif -vo gif89a:5.02:new.gif output to new.gif at 5.02 fps The filename defaults to out.gif, and the framerate defaults to 5 fps. by Joey Parrish <[e-mail]>

#include <stdio.h>
#include <stdlib.h>
#include <sys/time.h>

#include "afmt.h"
#include "audio_out.h"
#include "audio_out_internal.h"

static ao_info_t info = 
{
	"Null audio output",
	"null",
	"Tobias Diedrich",
	""
};

LIBAO_EXTERN(null)

struct	timeval last_tv;
int	buffer;

static void drain(){
 
    struct timeval now_tv;
    int temp, temp2;

    gettimeofday(&now_tv, 0);
    temp = now_tv.tv_sec - last_tv.tv_sec;
    temp *= ao_data.bps;
    
    temp2 = now_tv.tv_usec - last_tv.tv_usec;
    temp2 /= 1000;
    temp2 *= ao_data.bps;
    temp2 /= 1000;
    temp += temp2;

    buffer-=temp;
    if (buffer<0) buffer=0;

    if(temp>0) last_tv = now_tv;//mplayer is fast
}

// to set/get/query special features/parameters
static int control(int cmd,int arg){
    return -1;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){

    ao_data.buffersize= 65536;
    ao_data.outburst=1024;
    ao_data.channels=channels;
    ao_data.samplerate=rate;
    ao_data.format=format;
    ao_data.bps=channels*rate;
    if (format != AFMT_U8 && format != AFMT_S8)
	ao_data.bps*=2; 
    buffer=0;
    gettimeofday(&last_tv, 0);

    return 1;
}

// close audio device
static void uninit(){

}

// stop playing and empty buffers (for seeking/pause)
static void reset(){
    buffer=0;
}

// 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(){

    drain();
    return ao_data.buffersize - buffer;
}

// 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){

    int maxbursts = (ao_data.buffersize - buffer) / ao_data.outburst;
    int playbursts = len / ao_data.outburst;
    int bursts = playbursts > maxbursts ? maxbursts : playbursts;
    buffer += bursts * ao_data.outburst;
    return bursts * ao_data.outburst;
}

// return: delay in seconds between first and last sample in buffer
static float get_delay(){

    drain();
    return (float) buffer / (float) ao_data.bps;
}