view libmpcodecs/ad_ffmpeg.c @ 34102:dd8320c2a2cb

libmpcodec: add vf_lavfi. This filter wraps a complete libavfilter filter graph. As the API of libavfilter is not completely stable yet, the filter is not enabled by default, so as not to let mplayer unbuildable. Some strange behaviours may appear due to the very different model of buffer allocation between mplayer and lavfi.

/*
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#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 "dec_audio.h"
#include "av_helpers.h"
#include "libaf/reorder_ch.h"
#include "fmt-conversion.h"

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

LIBAD_EXTERN(ffmpeg)

#define assert(x)

#include "libavcodec/avcodec.h"


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

static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context)
{
    int broken_srate = 0;
    int samplerate    = lavc_context->sample_rate;
    int sample_format = samplefmt2affmt(lavc_context->sample_fmt);
    if (!sample_format)
        sample_format = sh_audio->sample_format;
    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 (lavc_context->codec_id == CODEC_ID_AAC &&
            samplerate == 2*sh_audio->wf->nSamplesPerSec) {
            broken_srate = 1;
        } else if (sh_audio->wf->nSamplesPerSec)
            samplerate=sh_audio->wf->nSamplesPerSec;
    }
    if (lavc_context->channels != sh_audio->channels ||
        samplerate != sh_audio->samplerate ||
        sample_format != sh_audio->sample_format) {
        sh_audio->channels=lavc_context->channels;
        sh_audio->samplerate=samplerate;
        sh_audio->sample_format = sample_format;
        sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
        if (broken_srate)
            mp_msg(MSGT_DECAUDIO, MSGL_WARN,
                   "Ignoring broken container sample rate for AAC with SBR\n");
        return 1;
    }
    return 0;
}

static int init(sh_audio_t *sh_audio)
{
    int tries = 0;
    int x;
    AVCodecContext *lavc_context;
    AVCodec *lavc_codec;

    mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
    init_avcodec();

    lavc_codec = 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_context3(lavc_codec);
    sh_audio->context=lavc_context;

    lavc_context->drc_scale = drc_level;
    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, sh_audio->wf + 1,
               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_open2(lavc_context, lavc_codec, NULL) < 0) {
        mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
        return 0;
    }
   mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);

//   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)
   do {
       x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
   } while (x <= 0 && tries++ < 5);
   if(x>0) sh_audio->a_buffer_len=x;

  sh_audio->i_bps=lavc_context->bit_rate/8;
  if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
      sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;

  switch (lavc_context->sample_fmt) {
      case AV_SAMPLE_FMT_U8:
      case AV_SAMPLE_FMT_S16:
      case AV_SAMPLE_FMT_S32:
      case AV_SAMPLE_FMT_FLT:
          break;
      default:
          return 0;
  }
  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);
        ds_clear_parser(sh->ds);
    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){
	AVPacket pkt;
	int len2=maxlen;
	double pts;
	int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
	if(x<=0) {
	    start = NULL;
	    x = 0;
	    ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
	    if (x <= 0)
	        break; // error
	} else {
	    int in_size = x;
	    int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
	    sh_audio->ds->buffer_pos -= in_size - consumed;
	}

	av_init_packet(&pkt);
	pkt.data = start;
	pkt.size = x;
	if (pts != MP_NOPTS_VALUE) {
	    sh_audio->pts = pts;
	    sh_audio->pts_bytes = 0;
	}
	y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
	// LATM may need many packets to find mux info
	if (y == AVERROR(EAGAIN))
	    continue;
	if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
	if(!sh_audio->parser && y<x)
	    sh_audio->ds->buffer_pos+=y-x;  // put back data (HACK!)
	if(len2>0){
	  if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
            int samplesize = av_get_bytes_per_sample(((AVCodecContext *)
                                    sh_audio->context)->sample_fmt);
            reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
                                AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
                                ((AVCodecContext *)sh_audio->context)->channels,
                                len2 / samplesize, samplesize);
	  }
	  //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);

        if (setup_format(sh_audio, sh_audio->context))
            break;
    }
  return len;
}