view libmpcodecs/ad_ffmpeg.c @ 24900:9079c9745ff9

A/V sync: take audio filter buffers into account Substract the delay caused by filter buffering when calculating currently playing audio position. This matters for af_scaletempo which buffers significant and varying amounts of data. For other current filters the effect is normally insignificant. Instead of the old time-based filter delay field (which was ignored) this version stores the per-filter delay in units of bytes input read without corresponding output. This allows the current scaletempo behavior where other filters before and after it can see the same nominal samplerate even though the real duration of the data varies; in this case the other filters can not know the delay they're causing in terms of real time.

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>

#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"

#include "ad_internal.h"

#include "mpbswap.h"

static ad_info_t info = 
{
	"FFmpeg/libavcodec audio decoders",
	"ffmpeg",
	"Nick Kurshev",
	"ffmpeg.sf.net",
	""
};

LIBAD_EXTERN(ffmpeg)

#define assert(x)

#ifdef USE_LIBAVCODEC_SO
#include <ffmpeg/avcodec.h>
#else
#include "avcodec.h"
#endif

extern int avcodec_inited;

static int preinit(sh_audio_t *sh)
{
  sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
  return 1;
}

static int init(sh_audio_t *sh_audio)
{
    int x;
    AVCodecContext *lavc_context;
    AVCodec *lavc_codec;

    mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
    if(!avcodec_inited){
      avcodec_init();
      avcodec_register_all();
      avcodec_inited=1;
    }
    
    lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
    if(!lavc_codec){
	mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
	return 0;
    }
    
    lavc_context = avcodec_alloc_context();
    sh_audio->context=lavc_context;

    lavc_context->sample_rate = sh_audio->samplerate;
    lavc_context->bit_rate = sh_audio->i_bps * 8;
    if(sh_audio->wf){
	lavc_context->channels = sh_audio->wf->nChannels;
	lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
	lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
	lavc_context->block_align = sh_audio->wf->nBlockAlign;
	lavc_context->bits_per_sample = sh_audio->wf->wBitsPerSample;
    }
    lavc_context->request_channels = audio_output_channels;
    lavc_context->codec_tag = sh_audio->format; //FOURCC
    lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi

    /* alloc extra data */
    if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
        lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
        lavc_context->extradata_size = sh_audio->wf->cbSize;
        memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX), 
               lavc_context->extradata_size);
    }

    // for QDM2
    if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
    {
        lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
        lavc_context->extradata_size = sh_audio->codecdata_len;
        memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, 
               lavc_context->extradata_size);	
    }

    /* open it */
    if (avcodec_open(lavc_context, lavc_codec) < 0) {
        mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
        return 0;
    }
   mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n");
   
//   printf("\nFOURCC: 0x%X\n",sh_audio->format);
   if(sh_audio->format==0x3343414D){
       // MACE 3:1
       sh_audio->ds->ss_div = 2*3; // 1 samples/packet
       sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
   } else
   if(sh_audio->format==0x3643414D){
       // MACE 6:1
       sh_audio->ds->ss_div = 2*6; // 1 samples/packet
       sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
   }

   // Decode at least 1 byte:  (to get header filled)
   x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
   if(x>0) sh_audio->a_buffer_len=x;

  sh_audio->channels=lavc_context->channels;
  sh_audio->samplerate=lavc_context->sample_rate;
  sh_audio->i_bps=lavc_context->bit_rate/8;
  if(sh_audio->wf){
      // If the decoder uses the wrong number of channels all is lost anyway.
      // sh_audio->channels=sh_audio->wf->nChannels;
      if (sh_audio->wf->nSamplesPerSec)
      sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
      if (sh_audio->wf->nAvgBytesPerSec)
      sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
  }
  sh_audio->samplesize=2;
  return 1;
}

static void uninit(sh_audio_t *sh)
{
    AVCodecContext *lavc_context = sh->context;

    if (avcodec_close(lavc_context) < 0)
	mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec);
    av_freep(&lavc_context->extradata);
    av_freep(&lavc_context);
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    AVCodecContext *lavc_context = sh->context;
    switch(cmd){
    case ADCTRL_RESYNC_STREAM:
        avcodec_flush_buffers(lavc_context);
    return CONTROL_TRUE;
    }
    return CONTROL_UNKNOWN;
}

static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
    unsigned char *start=NULL;
    int y,len=-1;
    while(len<minlen){
	int len2=maxlen;
	double pts;
	int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
	if(x<=0) break; // error
	if (pts != MP_NOPTS_VALUE) {
	    sh_audio->pts = pts;
	    sh_audio->pts_bytes = 0;
	}
	y=avcodec_decode_audio2(sh_audio->context,(int16_t*)buf,&len2,start,x);
//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
	if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
	if(y<x) sh_audio->ds->buffer_pos+=y-x;  // put back data (HACK!)
	if(len2>0){
	  //len=len2;break;
	  if(len<0) len=len2; else len+=len2;
	  buf+=len2;
	  maxlen -= len2;
	  sh_audio->pts_bytes += len2;
	}
        mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d  \n",y,len2);
    }
  return len;
}