annotate libmpcodecs/ad_ffmpeg.c @ 27985:1c77b86d355d

Remove a ColorFill that is not necessary since the surface it is used on has exactly the same size as the video image and the video will be copied into it before it is used the first 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 "libaf/reorder_ch.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)

#include "libavcodec/avcodec.h"

extern int avcodec_initialized;

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_initialized){
      avcodec_init();
      avcodec_register_all();
      avcodec_initialized=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_coded_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){
	  if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
            int src_ch_layout = AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT;
            const char *codec=((AVCodecContext*)sh_audio->context)->codec->name;
            if (!strcasecmp(codec, "ac3"))
              src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_AC3_DEFAULT;
            else if (!strcasecmp(codec, "dca"))
              src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_DCA_DEFAULT;
            else if (!strcasecmp(codec, "libfaad")
                || !strcasecmp(codec, "mpeg4aac"))
              src_ch_layout = AF_CHANNEL_LAYOUT_AAC_DEFAULT;
            else if (!strcasecmp(codec, "liba52"))
              src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_LIBA52_DEFAULT;
            else if (!strcasecmp(codec, "vorbis"))
              src_ch_layout = AF_CHANNEL_LAYOUT_VORBIS_DEFAULT;
            else if (!strcasecmp(codec, "flac"))
              src_ch_layout = AF_CHANNEL_LAYOUT_FLAC_DEFAULT;
            else
              src_ch_layout = AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT;
            reorder_channel_nch(buf, src_ch_layout,
                                AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
                                ((AVCodecContext *)sh_audio->context)->channels,
                                len2 / 2, 2);
	  }
	  //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;
}