view libao2/ao_null.c @ 27518:e54c9b7eb0d8

Revert bad changes to SSA/ASS subtitle packet format The following commits are reverted partially or completely: "a valid ASS line contains 9 ',' before actual text" "demux_mkv: output correctly formated ASS packets" "libass: add a new ass_process_data() to process demuxed subtitle packets" These commits converted the internal representation of SSA/ASS subtitle packets from the format used by Matroska to a custom format where each packet has contents exactly matching one line in complete SSA script files. AFAIK no files natively use such a format for muxed subtitles. The stated reason for this change was to use a format that could in principle be muxed into a maximal number of containers. SSA subtitles do not have an implicit duration so both start time and duration or end time need to be specified explicitly; the new format moved timing information inside the codec packet data so it could be muxed without modification into containers that can represent only start time at the container level. However such a change is wrong from the viewpoint of program architecture. Timing information belongs to the demuxer level, but these commits moved not only the duration but also the authoritative value of the start time to inside the codec data. Additionally the new format lost the value of the Matroska ReadOrder field which is used by MPlayer. This commit changes the internal packet format back to that used by Matroska and makes the internal Matroska demuxer output that format again. Libavformat still outputs the "new" format; it could be converted back to the Matroska format in demux_lavf.c, but I'm not adding that code at least yet. The current lavf code has similar problems as the reverted code in MPlayer, and it also currently fails to provide any way to access the value of the ReadOrder field. I hope that the lavf side will be improved; if it isn't conversion can be added later. For now I'll make MPlayer default to the internal Matroska demuxer instead of the lavf one in a separate commit.

#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;
}