annotate libmpcodecs/ad_ffmpeg.c @ 16718:044260623695

makes demux_lavf (-demuxer 35) use the framerate specified in the container if it's set and only fall back to the codec framerate if the former is not set. This solves the issue of some avi's playing at 30000/1 fps instead of the correct framerate. Patch by Ivo < ivop AH euronet POIS nl > Original thread: Date: Sep 25, 2005 12:34 AM Subject: [MPlayer-dev-eng] [PATCH] make demux_lavf use container framerate

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

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

#ifdef USE_LIBAVCODEC

#include "ad_internal.h"

#include "bswap.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;

    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->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_malloc(sh_audio->wf->cbSize);
        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;

#if 1
  sh_audio->channels=lavc_context->channels;
  sh_audio->samplerate=lavc_context->sample_rate;
  sh_audio->i_bps=lavc_context->bit_rate/8;
#else
  sh_audio->channels=sh_audio->wf->nChannels;
  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
  sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
#endif
  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=0;
	int x=ds_get_packet(sh_audio->ds,&start);
	if(x<=0) break; // error
	y=avcodec_decode_audio(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;
	}
        mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d  \n",y,len2);
    }
  return len;
}

#endif