view libao2/ao_null.c @ 24590:2c238fa777ff

ao_alsa: Fix get_space() return values larger than buffersize After a buffer underrun the ALSA get_space() function sometimes returned values larger than the ao had set in ao_data.buffersize. Fix this by replacing the old check against MAX_OUTBURST by one against ao_data.buffersize. There should be no need for the MAX_OUTBURST check; the current MPlayer side should no longer have any constant limit on the amount of data an ao can buffer or request at once. The get_space() values larger than ao_data.buffersize triggered errors in audio decoding causing the current attempt to fill audio buffers to be aborted. I'm not sure how often that caused behavior noticeably worse then an underrun already is.

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

#include "config.h"
#include "libaf/af_format.h"
#include "audio_out.h"
#include "audio_out_internal.h"

static ao_info_t info = 
{
	"Null audio output",
	"null",
	"Tobias Diedrich <[e-mail]>",
	""
};

LIBAO_EXTERN(null)

struct	timeval last_tv;
int	buffer;

static void drain(void){
 
    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,void *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 != AF_FORMAT_U8 && format != AF_FORMAT_S8)
	ao_data.bps*=2; 
    buffer=0;
    gettimeofday(&last_tv, 0);

    return 1;
}

// close audio device
static void uninit(int immed){

}

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

// stop playing, keep buffers (for pause)
static void audio_pause(void)
{
    // for now, just call reset();
    reset();
}

// resume playing, after audio_pause()
static void audio_resume(void)
{
}

// return: how many bytes can be played without blocking
static int get_space(void){

    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(void){

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