view libmpcodecs/ad_ffmpeg.c @ 24129:50e1e79056b6

Make terminal input work more like VO key input The Unix version of getch2() could either return an internally buffered key or do a second-level select() in addition to the input.c one and then read more data. Change getch2() to always add all read keys with mplayer_put_key() (like video output window keyboard input does) and remove the internal select() from the Unix version. Make input.c call mplayer_get_key() directly. The primary motivation for this change is to make combining multiple event sources under one select() easier. Now getch2() only needs to be called when the corresponding fd is readable, and it will be possible to handle events from X-based VOs with the same code.
author uau
date Sat, 25 Aug 2007 04:28:08 +0000
parents 51cd49ee92d6
children 352d7d9422b5
line wrap: on
line source

#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;

    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;
}