예제 #1
0
static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    switch(cmd)
    {
      case ADCTRL_RESYNC_STREAM:
         aac_sync(sh);
         AACFlushCodec(hAACDecoder);
	 return CONTROL_TRUE;
    }
  return CONTROL_UNKNOWN;
}
예제 #2
0
파일: aac.c 프로젝트: vadmium/deadbeef
// returns -1 on error, 0 on success
int
seek_raw_aac (aac_info_t *info, int sample) {
    uint8_t buf[ADTS_HEADER_SIZE*8];

    int nsamples = 0;
    int stream_sr = 0;
    int stream_ch = 0;

    int eof = 0;
    int bufsize = 0;
    int remaining = 0;

    int frame = 0;

    int frame_samples = 0;
    int curr_sample = 0;

    do {
        curr_sample += frame_samples;
        int size = sizeof (buf) - bufsize;
        if (deadbeef->fread (buf + bufsize, 1, size, info->file) != size) {
            trace ("seek_raw_aac: eof\n");
            break;
        }
        bufsize = sizeof (buf);

        int channels, samplerate, bitrate;
        size = aac_sync (buf, &channels, &samplerate, &bitrate, &frame_samples);
        if (size == 0) {
            memmove (buf, buf+1, sizeof (buf)-1);
            bufsize--;
            continue;
        }
        else {
            //trace ("aac: frame #%d(%d/%d) sync: %d %d %d %d %d\n", frame, curr_sample, sample, channels, samplerate, bitrate, frame_samples, size);
            frame++;
            if (deadbeef->fseek (info->file, size-(int)sizeof(buf), SEEK_CUR) == -1) {
                trace ("seek_raw_aac: invalid seek %d\n", size-sizeof(buf));
                break;
            }
            bufsize = 0;
        }
        if (samplerate <= 24000) {
            frame_samples *= 2;
        }
    } while (curr_sample + frame_samples < sample);

    if (curr_sample + frame_samples < sample) {
        return -1;
    }

    return sample - curr_sample;
}
예제 #3
0
파일: ad_faad.c 프로젝트: BOTCrusher/sagetv
static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    switch(cmd)
    {
      case ADCTRL_RESYNC_STREAM:
         aac_sync(sh);
	 return CONTROL_TRUE;
#if 0      
      case ADCTRL_SKIP_FRAME:
	  return CONTROL_TRUE;
#endif
    }
  return CONTROL_UNKNOWN;
}
예제 #4
0
파일: ad_faad.c 프로젝트: BOTCrusher/sagetv
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
  int j = 0, len = 0, last_dec_len = 1, errors = 0;	      
  void *faac_sample_buffer;

  while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) {

    /* update buffer for raw aac streams: */
  if(!sh->codecdata_len)
    if(sh->a_in_buffer_len < sh->a_in_buffer_size){
      sh->a_in_buffer_len +=
	demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
	sh->a_in_buffer_size - sh->a_in_buffer_len);
    }
	  
#ifdef DUMP_AAC_COMPRESSED
    {int i;
    for (i = 0; i < 16; i++)
      printf ("%02X ", sh->a_in_buffer[i]);
    printf ("\n");}
#endif

  if(!sh->codecdata_len){
   // raw aac stream:
   do {
    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer, sh->a_in_buffer_len);
	
    /* update buffer index after faacDecDecode */
    if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) {
      sh->a_in_buffer_len=0;
    } else {
      sh->a_in_buffer_len-=faac_finfo.bytesconsumed;
      memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len);
    }

    if(faac_finfo.error > 0) {
      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n",
              faacDecGetErrorMessage(faac_finfo.error));
      if (sh->a_in_buffer_len <= 0) {
        errors = MAX_FAAD_ERRORS;
        break;
      }
      sh->a_in_buffer_len--;
      memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len);
      aac_sync(sh);
      errors++;
    } else
      break;
   } while(errors < MAX_FAAD_ERRORS);	  
  } else {
   // packetized (.mp4) aac stream:
    unsigned char* bufptr=NULL;
    double pts;
    int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
    if(buflen<=0) break;
    if (pts != MP_NOPTS_VALUE) {
	sh->pts = pts;
	sh->pts_bytes = 0;
    }
    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen);
  }
  //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]);
  
    if(faac_finfo.error > 0) {
      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
      faacDecGetErrorMessage(faac_finfo.error));
    } else if (faac_finfo.samples == 0) {
      mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
    } else {
      /* XXX: samples already multiplied by channels! */
      mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%ld Bytes)!\n",
      sh->samplesize*faac_finfo.samples);
      memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
      last_dec_len = sh->samplesize*faac_finfo.samples;
      len += last_dec_len;
      sh->pts_bytes += last_dec_len;
    //printf("FAAD: buffer: %d bytes  consumed: %d \n", k, faac_finfo.bytesconsumed);
    }
  }
  return len;
}
예제 #5
0
파일: aac.c 프로젝트: vadmium/deadbeef
static int
parse_aac_stream(DB_FILE *fp, int *psamplerate, int *pchannels, float *pduration, int *ptotalsamples)
{
    size_t framepos = deadbeef->ftell (fp);
    size_t initfpos = framepos;
    int firstframepos = -1;
    int fsize = -1;
    int offs = 0;
    if (!fp->vfs->is_streaming ()) {
        int skip = deadbeef->junk_get_leading_size (fp);
        if (skip >= 0) {
            deadbeef->fseek (fp, skip, SEEK_SET);
        }
        int offs = deadbeef->ftell (fp);
        fsize = deadbeef->fgetlength (fp);
        if (skip > 0) {
            fsize -= skip;
        }
    }

    uint8_t buf[ADTS_HEADER_SIZE*8];

    int nsamples = 0;
    int stream_sr = 0;
    int stream_ch = 0;

    int eof = 0;
    int bufsize = 0;
    int remaining = 0;

    int frame = 0;
    int scanframes = 1000;
    if (fp->vfs->is_streaming ()) {
        scanframes = 1;
    }

    do {
        int size = sizeof (buf) - bufsize;
        if (deadbeef->fread (buf + bufsize, 1, size, fp) != size) {
            trace ("parse_aac_stream: eof\n");
            break;
        }
        bufsize = sizeof (buf);

        int channels, samplerate, bitrate, samples;
        size = aac_sync (buf, &channels, &samplerate, &bitrate, &samples);
        if (size == 0) {
            memmove (buf, buf+1, sizeof (buf)-1);
            bufsize--;
//            trace ("aac_sync fail, framepos: %d\n", framepos);
            if (deadbeef->ftell (fp) - initfpos > 2000) { // how many is enough to make sure?
                break;
            }
            framepos++;
            continue;
        }
        else {
            trace ("aac: frame #%d sync: %dch %d %d %d %d\n", frame, channels, samplerate, bitrate, samples, size);
            frame++;
            nsamples += samples;
            if (!stream_sr) {
                stream_sr = samplerate;
            }
            if (!stream_ch) {
                stream_ch = channels;
            }
            if (firstframepos == -1) {
                firstframepos = framepos;
            }
//            if (fp->vfs->streaming) {
//                *psamplerate = stream_sr;
//                *pchannels = stream_ch;
//            }
            framepos += size;
            if (deadbeef->fseek (fp, size-(int)sizeof(buf), SEEK_CUR) == -1) {
                trace ("parse_aac_stream: invalid seek %d\n", size-sizeof(buf));
                break;
            }
            bufsize = 0;
        }
    } while (ptotalsamples || frame < scanframes);

    if (!frame || !stream_sr || !nsamples) {
        return -1;
    }

    *psamplerate = stream_sr;

    *pchannels = stream_ch;

    if (ptotalsamples) {
        *ptotalsamples = nsamples;
        *pduration = nsamples / (float)stream_sr;
        trace ("aac: duration=%f (%d samples @ %d Hz), fsize=%d, nframes=%d\n", *pduration, *ptotalsamples, stream_sr, fsize, frame);
    }
    else {
        int pos = deadbeef->ftell (fp);
        int totalsamples = (double)fsize / (pos-offs) * nsamples;
        *pduration = totalsamples / (float)stream_sr;
        trace ("aac: duration=%f (%d samples @ %d Hz), fsize=%d\n", *pduration, totalsamples, stream_sr, fsize);
    }

    if (*psamplerate <= 24000) {
        *psamplerate *= 2;
        if (ptotalsamples) {
            *ptotalsamples *= 2;
        }
    }
    return firstframepos;
}
예제 #6
0
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen){
  int len = 0, last_dec_len = 1, err = 0, in_len;
  int outOfData = 0, errors=0;;
  uint8_t *inBuf;

  while(len < minlen && last_dec_len > 0 ) {
    /* update buffer for raw aac streams: */
  if(!sh->codecdata_len){
    if(sh->a_in_buffer_len < sh->a_in_buffer_size){
      sh->a_in_buffer_len +=
        demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
        sh->a_in_buffer_size - sh->a_in_buffer_len);
    }
//#define DUMP_AAC_COMPRESSED
#ifdef DUMP_AAC_COMPRESSED
    {int i;
     inBuf = sh->a_in_buffer;
    for (i = 0; i < 16; i++)
      printf ("%02x ", inBuf[i]);
    printf ("\n");}
#endif
    inBuf  = sh->a_in_buffer;
    in_len = sh->a_in_buffer_len;
    // raw aac stream:
    err = AACDecode(hAACDecoder,  &(inBuf), &(in_len), buf+len);
    if(in_len > 0) {
      memmove(sh->a_in_buffer,inBuf,in_len);
    }
    sh->a_in_buffer_len = in_len;

    if (err  == ERR_AAC_NONE)
    {
        AACGetLastFrameInfo(hAACDecoder, frameInfo);
        reorder_ch_pcm(buf+len, frameInfo->outputSamps, frameInfo->nChans);
        last_dec_len = sh->samplesize*frameInfo->outputSamps*sh->channels/frameInfo->nChans;
        len += last_dec_len;
        sh->pts_bytes += last_dec_len;
        if(errors)
          errors=0;
    }else if(err  == ERR_AAC_INDATA_UNDERFLOW){
      errors++;
    }else if(sh->a_in_buffer_len > 0){
      sh->a_in_buffer_len--;
      memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len);
      aac_sync(sh);
      errors++;
    }
    if(errors > 10){
        break;
    }
   }else{
    /* update buffer for raw aac streams: */
    // packetized (.mp4) aac stream:
    unsigned char* bufptr=NULL;
    double pts;
    int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
    if(buflen<=0) break;
    if (pts != MP_NOPTS_VALUE) {
	sh->pts = pts;
	sh->pts_bytes = 0;
    }
    err = AACDecode(hAACDecoder, &bufptr, &buflen, buf+len);
    if (err  == ERR_AAC_NONE)
    {
        AACGetLastFrameInfo(hAACDecoder, frameInfo);
        reorder_ch_pcm(buf+len, frameInfo->outputSamps, frameInfo->nChans);
        last_dec_len = sh->samplesize*frameInfo->outputSamps*sh->channels/frameInfo->nChans;
        len += last_dec_len;
        sh->pts_bytes += last_dec_len;
    }
   }
  }
  return len;
}
예제 #7
0
파일: aac.c 프로젝트: Koss64/deadbeef
static int
aac_read (DB_fileinfo_t *_info, char *bytes, int size) {
    aac_info_t *info = (aac_info_t *)_info;
    int samplesize = _info->fmt.channels * _info->fmt.bps / 8;
    if (!info->file->vfs->is_streaming ()) {
        if (info->currentsample + size / samplesize > info->endsample) {
            size = (info->endsample - info->currentsample + 1) * samplesize;
            if (size <= 0) {
                trace ("aac_read: eof");
                return 0;
            }
        }
    }

    int initsize = size;
    int eof = 0;

    while (size > 0) {
        if (info->skipsamples > 0 && info->out_remaining > 0) {
            int skip = min (info->out_remaining, info->skipsamples);
            if (skip < info->out_remaining) {
                memmove (info->out_buffer, info->out_buffer + skip * samplesize, (info->out_remaining - skip) * samplesize);
            }
            info->out_remaining -= skip;
            info->skipsamples -= skip;
        }
        if (info->out_remaining > 0) {
            int n = size / samplesize;
            n = min (info->out_remaining, n);

            char *src = info->out_buffer;
            if (info->noremap) {
                memcpy (bytes, src, n * samplesize);
                bytes += n * samplesize;
                src += n * samplesize;
            }
            else {
                int i, j;
                if (info->remap[0] == -1) {
                    // build remap mtx

                    // FIXME: should build channelmask 1st; then remap based on channelmask
                    for (i = 0; i < _info->fmt.channels; i++) {
                        switch (info->frame_info.channel_position[i]) {
                        case FRONT_CHANNEL_CENTER:
                            trace ("FC->%d\n", i);
                            info->remap[2] = i;
                            break;
                        case FRONT_CHANNEL_LEFT:
                            trace ("FL->%d\n", i);
                            info->remap[0] = i;
                            break;
                        case FRONT_CHANNEL_RIGHT:
                            trace ("FR->%d\n", i);
                            info->remap[1] = i;
                            break;
                        case SIDE_CHANNEL_LEFT:
                            trace ("SL->%d\n", i);
                            info->remap[6] = i;
                            break;
                        case SIDE_CHANNEL_RIGHT:
                            trace ("SR->%d\n", i);
                            info->remap[7] = i;
                            break;
                        case BACK_CHANNEL_LEFT:
                            trace ("RL->%d\n", i);
                            info->remap[4] = i;
                            break;
                        case BACK_CHANNEL_RIGHT:
                            trace ("RR->%d\n", i);
                            info->remap[5] = i;
                            break;
                        case BACK_CHANNEL_CENTER:
                            trace ("BC->%d\n", i);
                            info->remap[8] = i;
                            break;
                        case LFE_CHANNEL:
                            trace ("LFE->%d\n", i);
                            info->remap[3] = i;
                            break;
                        default:
                            trace ("aac: unknown ch(%d)->%d\n", info->frame_info.channel_position[i], i);
                            break;
                        }
                    }
                    for (i = 0; i < _info->fmt.channels; i++) {
                        trace ("%d ", info->remap[i]);
                    }
                    trace ("\n");
                    if (info->remap[0] == -1) {
                        info->remap[0] = 0;
                    }
                    if ((_info->fmt.channels == 1 && info->remap[0] == FRONT_CHANNEL_CENTER)
                        || (_info->fmt.channels == 2 && info->remap[0] == FRONT_CHANNEL_LEFT && info->remap[1] == FRONT_CHANNEL_RIGHT)) {
                        info->noremap = 1;
                    }
                }

                for (i = 0; i < n; i++) {
                    for (j = 0; j < _info->fmt.channels; j++) {
                        ((int16_t *)bytes)[j] = ((int16_t *)src)[info->remap[j]];
                    }
                    src += samplesize;
                    bytes += samplesize;
                }
            }
            size -= n * samplesize;

            if (n == info->out_remaining) {
                info->out_remaining = 0;
            }
            else {
                memmove (info->out_buffer, src, (info->out_remaining - n) * samplesize);
                info->out_remaining -= n;
            }
            continue;
        }

        char *samples = NULL;

        if (info->mp4file) {
            unsigned char *buffer = NULL;
            int buffer_size = 0;
#ifdef USE_MP4FF
            int rc = mp4ff_read_sample (info->mp4file, info->mp4track, info->mp4sample, &buffer, &buffer_size);
            if (rc == 0) {
                break;
            }
#else

            buffer = info->samplebuffer;
            buffer_size = info->maxSampleSize;
            MP4Timestamp sampleTime;
            MP4Duration sampleDuration;
            MP4Duration sampleRenderingOffset;
            bool isSyncSample;
            MP4ReadSample (info->mp4file, info->mp4track, info->mp4sample, &buffer, &buffer_size, &sampleTime, &sampleDuration, &sampleRenderingOffset, &isSyncSample);
            // convert timestamp and duration from track time to milliseconds
            u_int64_t myTime = MP4ConvertFromTrackTimestamp (info->mp4file, info->mp4track, 
                    sampleTime, MP4_MSECS_TIME_SCALE);

            u_int64_t myDuration = MP4ConvertFromTrackDuration (info->mp4file, info->mp4track,
                    sampleDuration, MP4_MSECS_TIME_SCALE);
#endif
            if (info->mp4sample >= info->mp4samples) {
                if (buffer) {
                    free (buffer);
                }
                break;
            }
            info->mp4sample++;
            samples = NeAACDecDecode(info->dec, &info->frame_info, buffer, buffer_size);

            if (buffer) {
                free (buffer);
            }
            if (!samples) {
                break;
            }
        }
        else {
            if (info->remaining < AAC_BUFFER_SIZE) {
                trace ("fread from offs %lld\n", deadbeef->ftell (info->file));
                size_t res = deadbeef->fread (info->buffer + info->remaining, 1, AAC_BUFFER_SIZE-info->remaining, info->file);
                if (res == 0) {
                    eof = 1;
                }
                info->remaining += res;
                if (!info->remaining) {
                    break;
                }
            }

            trace ("NeAACDecDecode %d bytes\n", info->remaining)
            samples = NeAACDecDecode (info->dec, &info->frame_info, info->buffer, info->remaining);
            if (!samples) {
                trace ("NeAACDecDecode failed with error %s (%d), consumed=%d\n", NeAACDecGetErrorMessage(info->frame_info.error), (int)info->frame_info.error, info->frame_info.bytesconsumed);

                if (info->num_errors > 10) {
                    trace ("NeAACDecDecode failed %d times, interrupting\n", info->num_errors);
                    break;
                }
                info->num_errors++;
                int s = 0;
                while (!s && info->remaining > 0) {
                    int ch, sr, br, sm;
                    s = aac_sync (info->buffer, &ch, &sr, &br, &sm);
                    if (s == 0) {
                        memmove (info->buffer, info->buffer+1, info->remaining-1);
                        info->remaining--;
                    }
                }
//                info->remaining = 0;
                continue;
            }
            info->num_errors=0;
            int consumed = info->frame_info.bytesconsumed;
            if (consumed > info->remaining) {
                trace ("NeAACDecDecode consumed more than available! wtf?\n");
                break;
            }
            if (consumed == info->remaining) {
                info->remaining = 0;
            }
            else if (consumed > 0) {
                memmove (info->buffer, info->buffer + consumed, info->remaining - consumed);
                info->remaining -= consumed;
            }
        }

        if (info->frame_info.samples > 0) {
            memcpy (info->out_buffer, samples, info->frame_info.samples * 2);
            info->out_remaining = info->frame_info.samples / info->frame_info.channels;
        }
    }

    info->currentsample += (initsize-size) / samplesize;
    trace ("aac_read return: %d\n", initsize-size);
    return initsize-size;
}