view libao2/ao_null.c @ 27645:83d915449a10

Remove IWMMXT optimizations through libavcodec from libmpeg2. According to Siarhei Siamashka libavcodec is faster on ARM so it is better to use it directly instead of creating this hackish mix of two libraries. Plus, these local changes would never be acceptable upstream, so no good reason for keeping it in our local patchset remains.

#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){

    int samplesize = af_fmt2bits(format) / 8;
    ao_data.outburst = 256 * channels * samplesize;
    // A "buffer" for about 0.2 seconds of audio
    ao_data.buffersize = (int)(rate * 0.2 / 256 + 1) * ao_data.outburst;
    ao_data.channels=channels;
    ao_data.samplerate=rate;
    ao_data.format=format;
    ao_data.bps=channels*rate*samplesize;
    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;
}