Esempio n. 1
0
static int UpdateInput(vorbis* p)
{
	vorbis_block_clear(&p->Block);
	vorbis_dsp_clear(&p->DSP);

	if (p->Codec.In.Format.Type == PACKET_AUDIO)
	{
		vorbis_info* Info = (vorbis_info*)p->Codec.In.Format.Extra;
		if (Info)
		{
			vorbis_synthesis_init(&p->DSP,Info);
			vorbis_block_init(&p->DSP,&p->Block);

			p->Codec.In.Format.ByteRate = Info->bitrate_nominal >> 3;
			p->Codec.In.Format.Format.Audio.SampleRate = Info->rate;
			p->Codec.In.Format.Format.Audio.Channels = Info->channels;
		}

		PacketFormatPCM(&p->Codec.Out.Format,&p->Codec.In.Format,OGG_F_BITS+1);
		p->Codec.Out.Format.Format.Audio.Flags = PCM_PLANES;
	}
Esempio n. 2
0
static GF_Err VORB_AttachStream(GF_BaseDecoder *ifcg, u16 ES_ID, char *decSpecInfo, u32 decSpecInfoSize, u16 DependsOnES_ID, u32 objectTypeIndication, Bool UpStream)
{
    ogg_packet oggpacket;
	GF_BitStream *bs;

	VORBISCTX();
	if (ctx->ES_ID) return GF_BAD_PARAM;
	
	if (!decSpecInfo || !decSpecInfoSize) return GF_NON_COMPLIANT_BITSTREAM;
	if (objectTypeIndication != GPAC_OGG_MEDIA_OTI) return GF_NON_COMPLIANT_BITSTREAM;
	if ((decSpecInfoSize<9) || strncmp(&decSpecInfo[3], "vorbis", 6)) return GF_NON_COMPLIANT_BITSTREAM;

	ctx->ES_ID = ES_ID;

    vorbis_info_init(&ctx->vi);
    vorbis_comment_init(&ctx->vc);

	oggpacket.granulepos = -1;
	oggpacket.b_o_s = 1;
	oggpacket.e_o_s = 0;
	oggpacket.packetno = 0;

	bs = gf_bs_new(decSpecInfo, decSpecInfoSize, GF_BITSTREAM_READ);
	while (gf_bs_available(bs)) {
		oggpacket.bytes = gf_bs_read_u16(bs);
		oggpacket.packet = malloc(sizeof(char) * oggpacket.bytes);
		gf_bs_read_data(bs, oggpacket.packet, oggpacket.bytes);
		if (vorbis_synthesis_headerin(&ctx->vi, &ctx->vc, &oggpacket) < 0 ) {
			free(oggpacket.packet);
			gf_bs_del(bs);
			return GF_NON_COMPLIANT_BITSTREAM;
		}
		free(oggpacket.packet);
	}
	vorbis_synthesis_init(&ctx->vd, &ctx->vi);
	vorbis_block_init(&ctx->vd, &ctx->vb); 
	gf_bs_del(bs);

	return GF_OK;
}
Esempio n. 3
0
encoder_instance encoder_create_vbr(int ch, int bitrate, float quality) {
  encoder_instance state = malloc(sizeof(struct encoder_state));
  state->data = NULL;
  state->data_len = 0;
  
  vorbis_info_init(&state->vi);
  
  if (vorbis_encode_init_vbr(&state->vi, ch, bitrate, quality) != 0) {
    free(state);
    return NULL;
  }
  
  vorbis_comment_init(&state->vc);
  vorbis_comment_add_tag(&state->vc, "ENCODER", "libvorbis.js");
  
  vorbis_analysis_init(&state->vd, &state->vi);
  vorbis_block_init(&state->vd, &state->vb);
  
  srand(time(NULL));
  ogg_stream_init(&state->os, rand());
  
  return state;
}
Esempio n. 4
0
static bool
vorbis_encoder_reinit(struct vorbis_encoder *encoder, GError **error)
{
	vorbis_info_init(&encoder->vi);

	if (encoder->quality >= -1.0) {
		/* a quality was configured (VBR) */

		if (0 != vorbis_encode_init_vbr(&encoder->vi,
						encoder->audio_format.channels,
						encoder->audio_format.sample_rate,
						encoder->quality * 0.1)) {
			g_set_error(error, vorbis_encoder_quark(), 0,
				    "error initializing vorbis vbr");
			vorbis_info_clear(&encoder->vi);
			return false;
		}
	} else {
		/* a bit rate was configured */

		if (0 != vorbis_encode_init(&encoder->vi,
					    encoder->audio_format.channels,
					    encoder->audio_format.sample_rate, -1.0,
					    encoder->bitrate * 1000, -1.0)) {
			g_set_error(error, vorbis_encoder_quark(), 0,
				    "error initializing vorbis encoder");
			vorbis_info_clear(&encoder->vi);
			return false;
		}
	}

	vorbis_analysis_init(&encoder->vd, &encoder->vi);
	vorbis_block_init(&encoder->vd, &encoder->vb);
	ogg_stream_init(&encoder->os, g_random_int());

	return true;
}
Esempio n. 5
0
/* Create an ogg stream and vorbis encoder, with the configuration
 * specified in the encoder_state.
 */
static gboolean
xmms_ices_encoder_create (encoder_state *s, vorbis_comment *vc)
{
	ogg_packet header[3];

	if (s->encoder_inited) {
		XMMS_DBG ("OOPS: xmms_ices_encoder_create called "
		          "with s->encoder_inited == TRUE !");
	}

	XMMS_DBG ("Creating encoder in ABR mode: min/avg/max bitrate %d/%d/%d",
	          s->min_br, s->nom_br, s->max_br);

	/* Create the Vorbis encoder. */
	vorbis_info_init (&s->vi);
	if (vorbis_encode_init (&s->vi, s->channels, s->rate,
	                        s->max_br, s->nom_br, s->min_br) < 0)
		return FALSE;
	vorbis_analysis_init (&s->vd, &s->vi);
	vorbis_block_init (&s->vd, &s->vb);

	/* Initialize the ogg stream and input the vorbis header
	 * packets. */
	ogg_stream_init (&s->os, s->serial++);
	vorbis_analysis_headerout (&s->vd, vc, &header[0], &header[1], &header[2]);
	ogg_stream_packetin (&s->os, &header[0]);
	ogg_stream_packetin (&s->os, &header[1]);
	ogg_stream_packetin (&s->os, &header[2]);

	s->in_header = TRUE;
	s->flushing = FALSE;
	s->samples_in_current_page = 0;
	s->previous_granulepos = 0;
	s->encoder_inited = TRUE;

	return TRUE;
}
Esempio n. 6
0
	StreamEncoder::StreamEncoder()
	{
		//Initialize the info 
		vorbis_info_init(&mVorbisInfo);

		if (ErrorCheck(vorbis_encode_init(&mVorbisInfo, 2, 44100, 100, 80, 60)) == true)
		{
			//Error
			Write("vorbis_encode_init error");
			return;
		}

		if (ErrorCheck(vorbis_analysis_init(&mVorbisDspState, &mVorbisInfo)) == true)
		{
			//Error
			Write("vorbis_analysis_init error");
			return;
		}

		vorbis_comment_init(&mVorbisComment);
		//vorbis_comment_add(&mVorbisComment, "Comments");
		int vahCode = vorbis_analysis_headerout(&mVorbisDspState, &mVorbisComment, &mOggPacketIdentification, &mOggPacketComment, &mOggPacketCodes);
		if (ErrorCheck(vahCode) == true)
		{
			//Error
			Write("vorbis_analysis_init error");
			return;
		}

		if (ErrorCheck(vorbis_block_init(&mVorbisDspState, &mVorbisBlock)) == true)
		{
			//Error
			Write("vorbis_block_init error");
			return;
		}
	}
Esempio n. 7
0
void EncoderVorbis::initStream() {
    // set up analysis state and auxiliary encoding storage
    vorbis_analysis_init(&m_vdsp, &m_vinfo);
    vorbis_block_init(&m_vdsp, &m_vblock);

    // set up packet-to-stream encoder; attach a random serial number
    srand(time(0));
    ogg_stream_init(&m_oggs, getSerial());

    // add comment
    vorbis_comment_init(&m_vcomment);
    vorbis_comment_add_tag(&m_vcomment, "ENCODER", "mixxx/libvorbis");
    if (m_metaDataArtist != NULL) {
        vorbis_comment_add_tag(&m_vcomment, "ARTIST", m_metaDataArtist);
    }
    if (m_metaDataTitle != NULL) {
        vorbis_comment_add_tag(&m_vcomment, "TITLE", m_metaDataTitle);
    }
    if (m_metaDataAlbum != NULL) {
        vorbis_comment_add_tag(&m_vcomment, "ALBUM", m_metaDataAlbum);
    }

    // set up the vorbis headers
    ogg_packet headerInit;
    ogg_packet headerComment;
    ogg_packet headerCode;
    vorbis_analysis_headerout(&m_vdsp, &m_vcomment, &headerInit, &headerComment, &headerCode);
    ogg_stream_packetin(&m_oggs, &headerInit);
    ogg_stream_packetin(&m_oggs, &headerComment);
    ogg_stream_packetin(&m_oggs, &headerCode);

    // The encoder is now inialized. The encode method will start streaming by
    // sending the header first.
    m_header_write = true;
    m_bStreamInitialized = true;
}
Esempio n. 8
0
static GstFlowReturn
vorbis_handle_type_packet (GstVorbisDec * vd)
{
  gint res;

  g_assert (vd->initialized == FALSE);

#ifdef USE_TREMOLO
  if (G_UNLIKELY ((res = vorbis_dsp_init (&vd->vd, &vd->vi))))
    goto synthesis_init_error;
#else
  if (G_UNLIKELY ((res = vorbis_synthesis_init (&vd->vd, &vd->vi))))
    goto synthesis_init_error;

  if (G_UNLIKELY ((res = vorbis_block_init (&vd->vd, &vd->vb))))
    goto block_init_error;
#endif

  vd->initialized = TRUE;

  return GST_FLOW_OK;

  /* ERRORS */
synthesis_init_error:
  {
    GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
        (NULL), ("couldn't initialize synthesis (%d)", res));
    return GST_FLOW_ERROR;
  }
block_init_error:
  {
    GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
        (NULL), ("couldn't initialize block (%d)", res));
    return GST_FLOW_ERROR;
  }
}
Esempio n. 9
0
// Just create the Quicktime objects since this routine is also called
// for reopening.
int FileVorbis::open_file(int rd, int wr)
{
	int result = 0;
	this->rd = rd;
	this->wr = wr;

//printf("FileVorbis::open_file 1\n");
	if(rd)
	{
//printf("FileVorbis::open_file 1\n");
		if(!(fd = fopen(asset->path, "rb")))
		{
			eprintf("Error while opening \"%s\" for reading. \n%m\n", asset->path);
			result = 1;
		}
		else
		{
//printf("FileVorbis::open_file 2 %p %p\n", fd, vf);
			if(ov_open(fd, &vf, NULL, 0) < 0)
			{
				eprintf("Invalid bitstream in %s\n", asset->path);
				result = 1;
			}
			else
			{
//printf("FileVorbis::open_file 1\n");
				vorbis_info *vi = ov_info(&vf, -1);
				asset->channels = vi->channels;
				if(!asset->sample_rate)
					asset->sample_rate = vi->rate;
//printf("FileVorbis::open_file 1\n");
				asset->audio_length = ov_pcm_total(&vf,-1);
//printf("FileVorbis::open_file 1\n");
				asset->audio_data = 1;
// printf("FileVorbis::open_file 1 %d %d %d\n", 
// asset->channels, 
// asset->sample_rate, 
// asset->audio_length);
			}
		}
	}

	if(wr)
	{
		if(!(fd = fopen(asset->path, "wb")))
		{
			eprintf("Error while opening \"%s\" for writing. \n%m\n", asset->path);
			result = 1;
		}
		else
		{
			vorbis_info_init(&vi);
			if(!asset->vorbis_vbr)
				result = vorbis_encode_init(&vi, 
					asset->channels, 
					asset->sample_rate, 
					asset->vorbis_max_bitrate, 
					asset->vorbis_bitrate, 
					asset->vorbis_min_bitrate);
			else
			{
				result = vorbis_encode_setup_managed(&vi,
					asset->channels, 
					asset->sample_rate, 
					asset->vorbis_max_bitrate, 
					asset->vorbis_bitrate, 
					asset->vorbis_min_bitrate);
				result |= vorbis_encode_ctl(&vi, OV_ECTL_RATEMANAGE_AVG, NULL);
				result |= vorbis_encode_setup_init(&vi);
			}

			if(!result)
			{
				vorbis_analysis_init(&vd, &vi);
				vorbis_block_init(&vd, &vb);
				vorbis_comment_init(&vc);
				srand(time(NULL));
				ogg_stream_init(&os, rand());

				ogg_packet header;
				ogg_packet header_comm;
				ogg_packet header_code;
				vorbis_analysis_headerout(&vd, 
					&vc,
					&header,
					&header_comm,
					&header_code);
				ogg_stream_packetin(&os,
					&header);
				ogg_stream_packetin(&os, 
					&header_comm);
				ogg_stream_packetin(&os,
					&header_code);

				while(1)
				{
					int result = ogg_stream_flush(&os, &og);
					if(result == 0) break;
					fwrite(og.header, 1, og.header_len, fd);
					fwrite(og.body, 1, og.body_len, fd);
				}
			}
		}
	}

//printf("FileVorbis::open_file 2\n");
	return result;
}
Esempio n. 10
0
int lame_decode_ogg_initfile( lame_global_flags*  gfp,
                              FILE*               fd,
                              mp3data_struct*     mp3data )
{

  lame_internal_flags *gfc = gfp->internal_flags;
  char *buffer;
  int  bytes;
  int i;


  /********** Decode setup ************/

  ogg_sync_init(&oy); /* Now we can read pages */
  

  /* grab some data at the head of the stream.  We want the first page
     (which is guaranteed to be small and only contain the Vorbis
     stream initial header) We need the first page to get the stream
     serialno. */
  
  /* submit a 4k block to libvorbis' Ogg layer */
  buffer=ogg_sync_buffer(&oy,4096);
  bytes=fread(buffer,1,4096,fd);
  ogg_sync_wrote(&oy,bytes);
  
  /* Get the first page. */
  if(ogg_sync_pageout(&oy,&og)!=1){
    /* error case.  Must not be Vorbis data */
    ERRORF( gfc, "Error initializing Ogg bitstream data.\n" );
    return -1;
  }
  
  /* Get the serial number and set up the rest of decode. */
  /* serialno first; use it to set up a logical stream */
  ogg_stream_init(&os,ogg_page_serialno(&og));
  
  /* extract the initial header from the first page and verify that the
     Ogg bitstream is in fact Vorbis data */
  
  /* I handle the initial header first instead of just having the code
     read all three Vorbis headers at once because reading the initial
     header is an easy way to identify a Vorbis bitstream and it's
     useful to see that functionality seperated out. */
  
  vorbis_info_init(&vi);
  vorbis_comment_init(&vc);
  if(ogg_stream_pagein(&os,&og)<0){ 
    /* error; stream version mismatch perhaps */
    ERRORF( gfc, "Error reading first page of Ogg bitstream data.\n" );
    return -1;
  }
  
  if(ogg_stream_packetout(&os,&op)!=1){ 
    /* no page? must not be vorbis */
    ERRORF( gfc, "Error reading initial header packet.\n" );
    return -1;
  }
  
  if(vorbis_synthesis_headerin(&vi,&vc,&op)<0){ 
    /* error case; not a vorbis header */
    ERRORF( gfc, "This Ogg bitstream does not contain Vorbis "
	    "audio data.\n");
    return -1;
  }
  
  /* At this point, we're sure we're Vorbis.  We've set up the logical
     (Ogg) bitstream decoder.  Get the comment and codebook headers and
     set up the Vorbis decoder */
  
  /* The next two packets in order are the comment and codebook headers.
     They're likely large and may span multiple pages.  Thus we reead
     and submit data until we get our two pacakets, watching that no
     pages are missing.  If a page is missing, error out; losing a
     header page is the only place where missing data is fatal. */
  
  i=0;
  while(i<2){
    while(i<2){
      int result=ogg_sync_pageout(&oy,&og);
      if(result==0)break; /* Need more data */
      /* Don't complain about missing or corrupt data yet.  We'll
	 catch it at the packet output phase */
      if(result==1){
	ogg_stream_pagein(&os,&og); /* we can ignore any errors here
				       as they'll also become apparent
				       at packetout */
	while(i<2){
	  result=ogg_stream_packetout(&os,&op);
	  if(result==0)break;
	  if(result==-1){
	    /* Uh oh; data at some point was corrupted or missing!
	       We can't tolerate that in a header.  Die. */
	    ERRORF( gfc, "Corrupt secondary header.  Exiting.\n" );
	    return -1;
	  }
	  vorbis_synthesis_headerin(&vi,&vc,&op);
	  i++;
	}
      }
    }
    /* no harm in not checking before adding more */
    buffer=ogg_sync_buffer(&oy,4096);
    bytes=fread(buffer,1,4096,fd);
    if(bytes==0 && i<2){
      ERRORF( gfc, "End of file before finding all Vorbis headers!\n" );
      return -1;
    }
    ogg_sync_wrote(&oy,bytes);
  }
  
  /* Throw the comments plus a few lines about the bitstream we're
     decoding */
  {
    /*
    char **ptr=vc.user_comments;
    while(*ptr){
      MSGF( gfc, "%s\n", *ptr );
      ++ptr;
    }
    MSGF( gfc, "\nBitstream is %d channel, %ldHz\n", vi.channels, vi.rate );
    MSGF( gfc, "Encoded by: %s\n\n", vc.vendor );
    */
  }
  
  
  /* OK, got and parsed all three headers. Initialize the Vorbis
     packet->PCM decoder. */
  vorbis_synthesis_init(&vd,&vi); /* central decode state */
  vorbis_block_init(&vd,&vb);     /* local state for most of the decode
				     so multiple block decodes can
				     proceed in parallel.  We could init
				     multiple vorbis_block structures
				     for vd here */
  
  mp3data->stereo = vi.channels;
  mp3data->samplerate = vi.rate;
  mp3data->bitrate = 0; //ov_bitrate_instant(&vf);
  mp3data->nsamp=MAX_U_32_NUM;
  
  return 0;
}
Esempio n. 11
0
static int init(sh_audio_t *sh)
{
  unsigned int offset, i, length, hsizes[3];
  void *headers[3];
  unsigned char* extradata;
  ogg_packet op;
  vorbis_comment vc;
  struct ov_struct_st *ov;
#define ERROR() { \
    vorbis_comment_clear(&vc); \
    vorbis_info_clear(&ov->vi); \
    free(ov); \
    return 0; \
  }

  /// Init the decoder with the 3 header packets
  ov = malloc(sizeof(struct ov_struct_st));
  vorbis_info_init(&ov->vi);
  vorbis_comment_init(&vc);

  if(! sh->wf) {
    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"ad_vorbis, extradata seems to be absent! exit\n");
    ERROR();
  }

  if(! sh->wf->cbSize) {
    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"ad_vorbis, extradata seems to be absent!, exit\n");
    ERROR();
  }

  mp_msg(MSGT_DECAUDIO,MSGL_V,"ad_vorbis, extradata seems is %d bytes long\n", sh->wf->cbSize);
  extradata = (char*) (sh->wf+1);

  if(*extradata != 2) {
    mp_msg (MSGT_DEMUX, MSGL_WARN, "ad_vorbis: Vorbis track does not contain valid headers.\n");
    ERROR();
  }

  offset = 1;
  for (i=0; i < 2; i++) {
    length = 0;
    while ((extradata[offset] == (unsigned char) 0xFF) && length < sh->wf->cbSize) {
      length += 255;
      offset++;
    }
    if(offset >= (sh->wf->cbSize - 1)) {
      mp_msg (MSGT_DEMUX, MSGL_WARN, "ad_vorbis: Vorbis track does not contain valid headers.\n");
      ERROR();
    }
    length += extradata[offset];
    offset++;
    mp_msg (MSGT_DEMUX, MSGL_V, "ad_vorbis, offset: %u, length: %u\n", offset, length);
    hsizes[i] = length;
  }

  headers[0] = &extradata[offset];
  headers[1] = &extradata[offset + hsizes[0]];
  headers[2] = &extradata[offset + hsizes[0] + hsizes[1]];
  hsizes[2] = sh->wf->cbSize - offset - hsizes[0] - hsizes[1];
  mp_msg (MSGT_DEMUX, MSGL_V, "ad_vorbis, header sizes: %d %d %d\n", hsizes[0], hsizes[1], hsizes[2]);

  for(i=0; i<3; i++) {
    op.bytes = hsizes[i];
    op.packet = headers[i];
    op.b_o_s  = (i == 0);
    if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) {
      mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: header n. %d broken! len=%ld\n", i, op.bytes);
      ERROR();
    }
    if(i == 2) {
      float rg_gain=0.f, rg_peak=0.f;
    char **ptr=vc.user_comments;
    while(*ptr){
      mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr);
      /* replaygain */
      read_vorbis_comment( *ptr, "replaygain_album_gain=", "%f", &rg_gain );
      read_vorbis_comment( *ptr, "rg_audiophile=", "%f", &rg_gain );
      if( !rg_gain ) {
	read_vorbis_comment( *ptr, "replaygain_track_gain=", "%f", &rg_gain );
	read_vorbis_comment( *ptr, "rg_radio=", "%f", &rg_gain );
      }
      read_vorbis_comment( *ptr, "replaygain_album_peak=", "%f", &rg_peak );
      if( !rg_peak ) {
	read_vorbis_comment( *ptr, "replaygain_track_peak=", "%f", &rg_peak );
	read_vorbis_comment( *ptr, "rg_peak=", "%f", &rg_peak );
      }
      ++ptr;
    }
    /* replaygain: scale */
    if(!rg_gain)
      ov->rg_scale = 1.f; /* just in case pow() isn't standard-conformant */
    else
      ov->rg_scale = pow(10.f, rg_gain/20);
    /* replaygain: anticlip */
    if(ov->rg_scale * rg_peak > 1.f)
      ov->rg_scale = 1.f / rg_peak;
    /* replaygain: security */
    if(ov->rg_scale > 15.)
      ov->rg_scale = 15.;
#ifdef CONFIG_TREMOR
    ov->rg_scale_int = (int)(ov->rg_scale*64.f);
#endif
    mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel%s, %dHz, %dbit/s %cBR\n",(int)ov->vi.channels,ov->vi.channels>1?"s":"",(int)ov->vi.rate,(int)ov->vi.bitrate_nominal,
	(ov->vi.bitrate_lower!=ov->vi.bitrate_nominal)||(ov->vi.bitrate_upper!=ov->vi.bitrate_nominal)?'V':'C');
    if(rg_gain || rg_peak)
      mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Gain = %+.2f dB, Peak = %.4f, Scale = %.2f\n", rg_gain, rg_peak, ov->rg_scale);
    mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",vc.vendor);
    }
  }

  vorbis_comment_clear(&vc);

//  printf("lower=%d  upper=%d  \n",(int)ov->vi.bitrate_lower,(int)ov->vi.bitrate_upper);

  // Setup the decoder
  sh->channels=ov->vi.channels;
  sh->samplerate=ov->vi.rate;
  sh->samplesize=2;
  // assume 128kbit if bitrate not specified in the header
  sh->i_bps=((ov->vi.bitrate_nominal>0) ? ov->vi.bitrate_nominal : 128000)/8;
  sh->context = ov;

  /// Finish the decoder init
  vorbis_synthesis_init(&ov->vd,&ov->vi);
  vorbis_block_init(&ov->vd,&ov->vb);
  mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");

  return 1;
}
Esempio n. 12
0
RefPtr<MediaDataDecoder::InitPromise>
VorbisDataDecoder::Init()
{
  vorbis_info_init(&mVorbisInfo);
  vorbis_comment_init(&mVorbisComment);
  PodZero(&mVorbisDsp);
  PodZero(&mVorbisBlock);

  AutoTArray<unsigned char*,4> headers;
  AutoTArray<size_t,4> headerLens;
  if (!XiphExtradataToHeaders(headers, headerLens,
                              mInfo.mCodecSpecificConfig->Elements(),
                              mInfo.mCodecSpecificConfig->Length())) {
    return InitPromise::CreateAndReject(
      MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
                  RESULT_DETAIL("Could not get vorbis header.")),
      __func__);
  }
  for (size_t i = 0; i < headers.Length(); i++) {
    if (NS_FAILED(DecodeHeader(headers[i], headerLens[i]))) {
      return InitPromise::CreateAndReject(
        MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
                    RESULT_DETAIL("Could not decode vorbis header.")),
        __func__);
    }
  }

  MOZ_ASSERT(mPacketCount == 3);

  int r = vorbis_synthesis_init(&mVorbisDsp, &mVorbisInfo);
  if (r) {
    return InitPromise::CreateAndReject(
      MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
                  RESULT_DETAIL("Systhesis init fail.")),
      __func__);
  }

  r = vorbis_block_init(&mVorbisDsp, &mVorbisBlock);
  if (r) {
    return InitPromise::CreateAndReject(
      MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
                  RESULT_DETAIL("Block init fail.")),
      __func__);
  }

  if (mInfo.mRate != (uint32_t)mVorbisDsp.vi->rate) {
    LOG(LogLevel::Warning,
        ("Invalid Vorbis header: container and codec rate do not match!"));
  }
  if (mInfo.mChannels != (uint32_t)mVorbisDsp.vi->channels) {
    LOG(LogLevel::Warning,
        ("Invalid Vorbis header: container and codec channels do not match!"));
  }

  AudioConfig::ChannelLayout layout(mVorbisDsp.vi->channels);
  if (!layout.IsValid()) {
    return InitPromise::CreateAndReject(
      MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
                  RESULT_DETAIL("Invalid audio layout.")),
      __func__);
  }

  return InitPromise::CreateAndResolve(TrackInfo::kAudioTrack, __func__);
}
int main(int argc,char *argv[]){

  int i,j;
  ogg_packet op;

  FILE *infile = stdin;

#ifdef _WIN32 /* We need to set stdin/stdout to binary mode. Damn windows. */
  /* Beware the evil ifdef. We avoid these where we can, but this one we
     cannot. Don't add any more, you'll probably go to hell if you do. */
  _setmode( _fileno( stdin ), _O_BINARY );
#endif

  /* open the input file if any */
  if(argc==2){
    infile=fopen(argv[1],"rb");
    if(infile==NULL){
      fprintf(stderr,"Unable to open '%s' for playback.\n", argv[1]);
      exit(1);
    }
  }
  if(argc>2){
      usage();
      exit(1);
  }

  /* start up Ogg stream synchronization layer */
  ogg_sync_init(&oy);

  /* init supporting Vorbis structures needed in header parsing */
  vorbis_info_init(&vi);
  vorbis_comment_init(&vc);

  /* init supporting Theora structures needed in header parsing */
  theora_comment_init(&tc);
  theora_info_init(&ti);

  /* Ogg file open; parse the headers */
  /* Only interested in Vorbis/Theora streams */
  while(!stateflag){
    int ret=buffer_data(infile,&oy);
    if(ret==0)break;
    while(ogg_sync_pageout(&oy,&og)>0){
      ogg_stream_state test;

      /* is this a mandated initial header? If not, stop parsing */
      if(!ogg_page_bos(&og)){
        /* don't leak the page; get it into the appropriate stream */
        queue_page(&og);
        stateflag=1;
        break;
      }

      ogg_stream_init(&test,ogg_page_serialno(&og));
      ogg_stream_pagein(&test,&og);
      ogg_stream_packetout(&test,&op);

      /* identify the codec: try theora */
      if(!theora_p && theora_decode_header(&ti,&tc,&op)>=0){
        /* it is theora */
        memcpy(&to,&test,sizeof(test));
        theora_p=1;
      }else if(!vorbis_p && vorbis_synthesis_headerin(&vi,&vc,&op)>=0){
        /* it is vorbis */
        memcpy(&vo,&test,sizeof(test));
        vorbis_p=1;
      }else{
        /* whatever it is, we don't care about it */
        ogg_stream_clear(&test);
      }
    }
    /* fall through to non-bos page parsing */
  }

  /* we're expecting more header packets. */
  while((theora_p && theora_p<3) || (vorbis_p && vorbis_p<3)){
    int ret;

    /* look for further theora headers */
    while(theora_p && (theora_p<3) && (ret=ogg_stream_packetout(&to,&op))){
      if(ret<0){
        fprintf(stderr,"Error parsing Theora stream headers; corrupt stream?\n");
        exit(1);
      }
      if(theora_decode_header(&ti,&tc,&op)){
        printf("Error parsing Theora stream headers; corrupt stream?\n");
        exit(1);
      }
      theora_p++;
      if(theora_p==3)break;
    }

    /* look for more vorbis header packets */
    while(vorbis_p && (vorbis_p<3) && (ret=ogg_stream_packetout(&vo,&op))){
      if(ret<0){
        fprintf(stderr,"Error parsing Vorbis stream headers; corrupt stream?\n");
        exit(1);
      }
      if(vorbis_synthesis_headerin(&vi,&vc,&op)){
        fprintf(stderr,"Error parsing Vorbis stream headers; corrupt stream?\n");
        exit(1);
      }
      vorbis_p++;
      if(vorbis_p==3)break;
    }

    /* The header pages/packets will arrive before anything else we
       care about, or the stream is not obeying spec */

    if(ogg_sync_pageout(&oy,&og)>0){
      queue_page(&og); /* demux into the appropriate stream */
    }else{
      int ret=buffer_data(infile,&oy); /* someone needs more data */
      if(ret==0){
        fprintf(stderr,"End of file while searching for codec headers.\n");
        exit(1);
      }
    }
  }

  /* and now we have it all.  initialize decoders */
  if(theora_p){
    theora_decode_init(&td,&ti);
    printf("Ogg logical stream %x is Theora %dx%d %.02f fps video\n",
           (unsigned int)to.serialno,ti.width,ti.height, 
           (double)ti.fps_numerator/ti.fps_denominator);
    if(ti.width!=ti.frame_width || ti.height!=ti.frame_height)
      printf("  Frame content is %dx%d with offset (%d,%d).\n",
           ti.frame_width, ti.frame_height, ti.offset_x, ti.offset_y);
    report_colorspace(&ti);
    dump_comments(&tc);
  }else{
    /* tear down the partial theora setup */
    theora_info_clear(&ti);
    theora_comment_clear(&tc);
  }
  if(vorbis_p){
    vorbis_synthesis_init(&vd,&vi);
    vorbis_block_init(&vd,&vb);
    fprintf(stderr,"Ogg logical stream %x is Vorbis %d channel %d Hz audio.\n",
            (unsigned int)vo.serialno,vi.channels,(int)vi.rate);
  }else{
    /* tear down the partial vorbis setup */
    vorbis_info_clear(&vi);
    vorbis_comment_clear(&vc);
  }

  /* open audio */
  if(vorbis_p)open_audio();

  /* open video */
  if(theora_p)open_video();

  /* install signal handler as SDL clobbered the default */
  signal (SIGINT, sigint_handler);

  /* on to the main decode loop.  We assume in this example that audio
     and video start roughly together, and don't begin playback until
     we have a start frame for both.  This is not necessarily a valid
     assumption in Ogg A/V streams! It will always be true of the
     example_encoder (and most streams) though. */

  stateflag=0; /* playback has not begun */
  while(!got_sigint){

    /* we want a video and audio frame ready to go at all times.  If
       we have to buffer incoming, buffer the compressed data (ie, let
       ogg do the buffering) */
    while(vorbis_p && !audiobuf_ready){
      int ret;
      float **pcm;

      /* if there's pending, decoded audio, grab it */
      if((ret=vorbis_synthesis_pcmout(&vd,&pcm))>0){
        int count=audiobuf_fill/2;
        int maxsamples=(audiofd_fragsize-audiobuf_fill)/2/vi.channels;
        for(i=0;i<ret && i<maxsamples;i++)
          for(j=0;j<vi.channels;j++){
            int val=rint(pcm[j][i]*32767.f);
            if(val>32767)val=32767;
            if(val<-32768)val=-32768;
            audiobuf[count++]=val;
          }
        vorbis_synthesis_read(&vd,i);
        audiobuf_fill+=i*vi.channels*2;
        if(audiobuf_fill==audiofd_fragsize)audiobuf_ready=1;
        if(vd.granulepos>=0)
          audiobuf_granulepos=vd.granulepos-ret+i;
        else
          audiobuf_granulepos+=i;
        
      }else{
        
        /* no pending audio; is there a pending packet to decode? */
        if(ogg_stream_packetout(&vo,&op)>0){
          if(vorbis_synthesis(&vb,&op)==0) /* test for success! */
            vorbis_synthesis_blockin(&vd,&vb);
        }else   /* we need more data; break out to suck in another page */
          break;
      }
    }

    while(theora_p && !videobuf_ready){
      /* theora is one in, one out... */
      if(ogg_stream_packetout(&to,&op)>0){

        theora_decode_packetin(&td,&op);
        videobuf_granulepos=td.granulepos;
        
        videobuf_time=theora_granule_time(&td,videobuf_granulepos);

        /* is it already too old to be useful?  This is only actually
           useful cosmetically after a SIGSTOP.  Note that we have to
           decode the frame even if we don't show it (for now) due to
           keyframing.  Soon enough libtheora will be able to deal
           with non-keyframe seeks.  */

        if(videobuf_time>=get_time())
        videobuf_ready=1;
                
      }else
        break;
    }

    if(!videobuf_ready && !audiobuf_ready && feof(infile))break;

    if(!videobuf_ready || !audiobuf_ready){
      /* no data yet for somebody.  Grab another page */
      int bytes=buffer_data(infile,&oy);
      while(ogg_sync_pageout(&oy,&og)>0){
        queue_page(&og);
      }
    }

    /* If playback has begun, top audio buffer off immediately. */
    if(stateflag) audio_write_nonblocking();

    /* are we at or past time for this video frame? */
    if(stateflag && videobuf_ready && videobuf_time<=get_time()){
      video_write();
      videobuf_ready=0;
    }

    if(stateflag &&
       (audiobuf_ready || !vorbis_p) &&
       (videobuf_ready || !theora_p) &&
       !got_sigint){
      /* we have an audio frame ready (which means the audio buffer is
         full), it's not time to play video, so wait until one of the
         audio buffer is ready or it's near time to play video */
        
      /* set up select wait on the audiobuffer and a timeout for video */
      struct timeval timeout;
      fd_set writefs;
      fd_set empty;
      int n=0;

      FD_ZERO(&writefs);
      FD_ZERO(&empty);
      if(audiofd>=0){
        FD_SET(audiofd,&writefs);
        n=audiofd+1;
      }

      if(theora_p){
        long milliseconds=(videobuf_time-get_time())*1000-5;
        if(milliseconds>500)milliseconds=500;
        if(milliseconds>0){
          timeout.tv_sec=milliseconds/1000;
          timeout.tv_usec=(milliseconds%1000)*1000;

          n=select(n,&empty,&writefs,&empty,&timeout);
          if(n)audio_calibrate_timer(0);
        }
      }else{
        select(n,&empty,&writefs,&empty,NULL);
      }
    }

    /* if our buffers either don't exist or are ready to go,
       we can begin playback */
    if((!theora_p || videobuf_ready) &&
       (!vorbis_p || audiobuf_ready))stateflag=1;
    /* same if we've run out of input */
    if(feof(infile))stateflag=1;

  }

  /* tear it all down */

  audio_close();
  SDL_Quit();

  if(vorbis_p){
    ogg_stream_clear(&vo);
    vorbis_block_clear(&vb);
    vorbis_dsp_clear(&vd);
    vorbis_comment_clear(&vc);
    vorbis_info_clear(&vi);
  }
  if(theora_p){
    ogg_stream_clear(&to);
    theora_clear(&td);
    theora_comment_clear(&tc);
    theora_info_clear(&ti);
  }
  ogg_sync_clear(&oy);

  if(infile && infile!=stdin)fclose(infile);

  fprintf(stderr,
          "\r                                                              "
          "\nDone.\n");
  return(0);

}
Esempio n. 14
0
/*!
 * \brief Create a new OGG/Vorbis filestream and set it up for reading.
 * \param s File that points to on disk storage of the OGG/Vorbis data.
 * \return The new filestream.
 */
static int ogg_vorbis_open(struct ast_filestream *s)
{
	int i;
	int bytes;
	int result;
	char **ptr;
	char *buffer;
	struct vorbis_desc *tmp = (struct vorbis_desc *)s->_private;

	tmp->writing = 0;

	ogg_sync_init(&tmp->oy);

	buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
	bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
	ogg_sync_wrote(&tmp->oy, bytes);

	result = ogg_sync_pageout(&tmp->oy, &tmp->og);
	if (result != 1) {
		if(bytes < BLOCK_SIZE) {
			ast_log(LOG_ERROR, "Run out of data...\n");
		} else {
			ast_log(LOG_ERROR, "Input does not appear to be an Ogg bitstream.\n");
		}
		ogg_sync_clear(&tmp->oy);
		return -1;
	}
	
	ogg_stream_init(&tmp->os, ogg_page_serialno(&tmp->og));
	vorbis_info_init(&tmp->vi);
	vorbis_comment_init(&tmp->vc);

	if (ogg_stream_pagein(&tmp->os, &tmp->og) < 0) { 
		ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n");
error:
		ogg_stream_clear(&tmp->os);
		vorbis_comment_clear(&tmp->vc);
		vorbis_info_clear(&tmp->vi);
		ogg_sync_clear(&tmp->oy);
		return -1;
	}
	
	if (ogg_stream_packetout(&tmp->os, &tmp->op) != 1) { 
		ast_log(LOG_ERROR, "Error reading initial header packet.\n");
		goto error;
	}
	
	if (vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op) < 0) { 
		ast_log(LOG_ERROR, "This Ogg bitstream does not contain Vorbis audio data.\n");
		goto error;
	}
	
	for (i = 0; i < 2 ; ) {
		while (i < 2) {
			result = ogg_sync_pageout(&tmp->oy, &tmp->og);
			if (result == 0)
				break;
			if (result == 1) {
				ogg_stream_pagein(&tmp->os, &tmp->og);
				while(i < 2) {
					result = ogg_stream_packetout(&tmp->os,&tmp->op);
					if(result == 0)
						break;
					if(result < 0) {
						ast_log(LOG_ERROR, "Corrupt secondary header.  Exiting.\n");
						goto error;
					}
					vorbis_synthesis_headerin(&tmp->vi, &tmp->vc, &tmp->op);
					i++;
				}
			}
		}

		buffer = ogg_sync_buffer(&tmp->oy, BLOCK_SIZE);
		bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
		if (bytes == 0 && i < 2) {
			ast_log(LOG_ERROR, "End of file before finding all Vorbis headers!\n");
			goto error;
		}
		ogg_sync_wrote(&tmp->oy, bytes);
	}
	
	for (ptr = tmp->vc.user_comments; *ptr; ptr++)
		ast_log(LOG_DEBUG, "OGG/Vorbis comment: %s\n", *ptr);
	ast_log(LOG_DEBUG, "OGG/Vorbis bitstream is %d channel, %ldHz\n", tmp->vi.channels, tmp->vi.rate);
	ast_log(LOG_DEBUG, "OGG/Vorbis file encoded by: %s\n", tmp->vc.vendor);

	if (tmp->vi.channels != 1) {
		ast_log(LOG_ERROR, "Only monophonic OGG/Vorbis files are currently supported!\n");
		goto error;
	}
	
	if (tmp->vi.rate != DEFAULT_SAMPLE_RATE) {
		ast_log(LOG_ERROR, "Only 8000Hz OGG/Vorbis files are currently supported!\n");
		vorbis_block_clear(&tmp->vb);
		vorbis_dsp_clear(&tmp->vd);
		goto error;
	}
	
	vorbis_synthesis_init(&tmp->vd, &tmp->vi);
	vorbis_block_init(&tmp->vd, &tmp->vb);

	return 0;
}
Esempio n. 15
0
static int oggvorbis_decode_init(AVCodecContext *avccontext) {
    OggVorbisDecContext *context = avccontext->priv_data ;
    uint8_t *p= avccontext->extradata;
    int i, hsizes[3];
    unsigned char *headers[3], *extradata = avccontext->extradata;

    vorbis_info_init(&context->vi) ;
    vorbis_comment_init(&context->vc) ;

    if(! avccontext->extradata_size || ! p) {
        av_log(avccontext, AV_LOG_ERROR, "vorbis extradata absent\n");
        return -1;
    }

    if(p[0] == 0 && p[1] == 30) {
        for(i = 0; i < 3; i++){
            hsizes[i] = bytestream_get_be16((const uint8_t **)&p);
            headers[i] = p;
            p += hsizes[i];
        }
    } else if(*p == 2) {
        unsigned int offset = 1;
        p++;
        for(i=0; i<2; i++) {
            hsizes[i] = 0;
            while((*p == 0xFF) && (offset < avccontext->extradata_size)) {
                hsizes[i] += 0xFF;
                offset++;
                p++;
            }
            if(offset >= avccontext->extradata_size - 1) {
                av_log(avccontext, AV_LOG_ERROR,
                       "vorbis header sizes damaged\n");
                return -1;
            }
            hsizes[i] += *p;
            offset++;
            p++;
        }
        hsizes[2] = avccontext->extradata_size - hsizes[0]-hsizes[1]-offset;
#if 0
        av_log(avccontext, AV_LOG_DEBUG,
               "vorbis header sizes: %d, %d, %d, / extradata_len is %d \n",
               hsizes[0], hsizes[1], hsizes[2], avccontext->extradata_size);
#endif
        headers[0] = extradata + offset;
        headers[1] = extradata + offset + hsizes[0];
        headers[2] = extradata + offset + hsizes[0] + hsizes[1];
    } else {
        av_log(avccontext, AV_LOG_ERROR,
               "vorbis initial header len is wrong: %d\n", *p);
        return -1;
    }

    for(i=0; i<3; i++){
        context->op.b_o_s= i==0;
        context->op.bytes = hsizes[i];
        context->op.packet = headers[i];
        if(vorbis_synthesis_headerin(&context->vi, &context->vc, &context->op)<0){
            av_log(avccontext, AV_LOG_ERROR, "%d. vorbis header damaged\n", i+1);
            return -1;
        }
    }

    avccontext->channels = context->vi.channels;
    avccontext->sample_rate = context->vi.rate;
    avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
    avccontext->time_base= (AVRational){1, avccontext->sample_rate};

    vorbis_synthesis_init(&context->vd, &context->vi);
    vorbis_block_init(&context->vd, &context->vb);

    return 0 ;
}
bool Start(void *ctx, int iInChannels, int iInRate, int iInBits,
          const char* title, const char* artist,
          const char* albumartist, const char* album,
          const char* year, const char* track, const char* genre,
          const char* comment, int iTrackLength)
{
  ogg_context *context = (ogg_context *)ctx;
  if (!context || !context->callbacks.write)
    return false;

  // we accept only 2 ch 16bit atm
  if (iInChannels != 2 || iInBits != 16)
    return false;

  if (preset == -1)
    vorbis_encode_init(&context->vorbisInfo, iInChannels, iInRate, -1, bitrate*1000, -1);
  else
    vorbis_encode_init_vbr(&context->vorbisInfo, iInChannels, iInRate, float(preset)/10.0f);

  /* add a comment */
  vorbis_comment comm;
  vorbis_comment_init(&comm);
  vorbis_comment_add_tag(&comm, (char*)"comment", (char*)comment);
  vorbis_comment_add_tag(&comm, (char*)"artist", (char*)artist);
  vorbis_comment_add_tag(&comm, (char*)"title", (char*)title);
  vorbis_comment_add_tag(&comm, (char*)"album", (char*)album);
  vorbis_comment_add_tag(&comm, (char*)"albumartist", (char*)albumartist);
  vorbis_comment_add_tag(&comm, (char*)"genre", (char*)genre);
  vorbis_comment_add_tag(&comm, (char*)"tracknumber", (char*)track);
  vorbis_comment_add_tag(&comm, (char*)"date", (char*)year);

  /* set up the analysis state and auxiliary encoding storage */
  vorbis_analysis_init(&context->vorbisDspState, &context->vorbisInfo);

  vorbis_block_init(&context->vorbisDspState, &context->vorbisBlock);

  /* set up our packet->stream encoder */
  /* pick a random serial number; that way we can more likely build
  chained streams just by concatenation */
  srand((unsigned int)time(NULL));
  ogg_stream_init(&context->oggStreamState, rand());

  /* write out the metadata */
  ogg_packet header;
  ogg_packet header_comm;
  ogg_packet header_code;
  ogg_page   page;

  vorbis_analysis_headerout(&context->vorbisDspState, &comm,
                            &header, &header_comm, &header_code);

  ogg_stream_packetin(&context->oggStreamState, &header);
  ogg_stream_packetin(&context->oggStreamState, &header_comm);
  ogg_stream_packetin(&context->oggStreamState, &header_code);

  while (1)
  {
    /* This ensures the actual
     * audio data will start on a new page, as per spec
     */
    int result = ogg_stream_flush(&context->oggStreamState, &page);
    if (result == 0)
      break;
    context->callbacks.write(context->callbacks.opaque, page.header, page.header_len);
    context->callbacks.write(context->callbacks.opaque, page.body, page.body_len);
  }
  vorbis_comment_clear(&comm);

  context->inited = true;
  return true;
}
Esempio n. 17
0
int ExportOGG::Export(AudacityProject *project,
                       int numChannels,
                       const wxString &fName,
                       bool selectionOnly,
                       double t0,
                       double t1,
                       MixerSpec *mixerSpec,
                       const Tags *metadata,
                       int WXUNUSED(subformat))
{
   double    rate    = project->GetRate();
   const TrackList *tracks = project->GetTracks();
   double    quality = (gPrefs->Read(wxT("/FileFormats/OggExportQuality"), 50)/(float)100.0);

   wxLogNull logNo;            // temporarily disable wxWidgets error messages
   int updateResult = eProgressSuccess;
   int       eos = 0;

   FileIO outFile(fName, FileIO::Output);

   if (!outFile.IsOpened()) {
      wxMessageBox(_("Unable to open target file for writing"));
      return false;
   }

   // All the Ogg and Vorbis encoding data
   ogg_stream_state stream;
   ogg_page         page;
   ogg_packet       packet;

   vorbis_info      info;
   vorbis_comment   comment;
   vorbis_dsp_state dsp;
   vorbis_block     block;

   // Encoding setup
   vorbis_info_init(&info);
   vorbis_encode_init_vbr(&info, numChannels, int(rate + 0.5), quality);

   // Retrieve tags
   if (!FillComment(project, &comment, metadata)) {
      return false;
   }

   // Set up analysis state and auxiliary encoding storage
   vorbis_analysis_init(&dsp, &info);
   vorbis_block_init(&dsp, &block);

   // Set up packet->stream encoder.  According to encoder example,
   // a random serial number makes it more likely that you can make
   // chained streams with concatenation.
   srand(time(NULL));
   ogg_stream_init(&stream, rand());

   // First we need to write the required headers:
   //    1. The Ogg bitstream header, which contains codec setup params
   //    2. The Vorbis comment header
   //    3. The bitstream codebook.
   //
   // After we create those our responsibility is complete, libvorbis will
   // take care of any other ogg bistream constraints (again, according
   // to the example encoder source)
   ogg_packet bitstream_header;
   ogg_packet comment_header;
   ogg_packet codebook_header;

   vorbis_analysis_headerout(&dsp, &comment, &bitstream_header, &comment_header,
         &codebook_header);

   // Place these headers into the stream
   ogg_stream_packetin(&stream, &bitstream_header);
   ogg_stream_packetin(&stream, &comment_header);
   ogg_stream_packetin(&stream, &codebook_header);

   // Flushing these headers now guarentees that audio data will
   // start on a NEW page, which apparently makes streaming easier
   while (ogg_stream_flush(&stream, &page)) {
      outFile.Write(page.header, page.header_len);
      outFile.Write(page.body, page.body_len);
   }

   const WaveTrackConstArray waveTracks =
      tracks->GetWaveTrackConstArray(selectionOnly, false);
   {
      auto mixer = CreateMixer(waveTracks,
         tracks->GetTimeTrack(),
         t0, t1,
         numChannels, SAMPLES_PER_RUN, false,
         rate, floatSample, true, mixerSpec);

      ProgressDialog progress(wxFileName(fName).GetName(),
         selectionOnly ?
         _("Exporting the selected audio as Ogg Vorbis") :
         _("Exporting the entire project as Ogg Vorbis"));

      while (updateResult == eProgressSuccess && !eos) {
         float **vorbis_buffer = vorbis_analysis_buffer(&dsp, SAMPLES_PER_RUN);
         sampleCount samplesThisRun = mixer->Process(SAMPLES_PER_RUN);

         if (samplesThisRun == 0) {
            // Tell the library that we wrote 0 bytes - signalling the end.
            vorbis_analysis_wrote(&dsp, 0);
         }
         else {

            for (int i = 0; i < numChannels; i++) {
               float *temp = (float *)mixer->GetBuffer(i);
               memcpy(vorbis_buffer[i], temp, sizeof(float)*SAMPLES_PER_RUN);
            }

            // tell the encoder how many samples we have
            vorbis_analysis_wrote(&dsp, samplesThisRun);
         }

         // I don't understand what this call does, so here is the comment
         // from the example, verbatim:
         //
         //    vorbis does some data preanalysis, then divvies up blocks
         //    for more involved (potentially parallel) processing. Get
         //    a single block for encoding now
         while (vorbis_analysis_blockout(&dsp, &block) == 1) {

            // analysis, assume we want to use bitrate management
            vorbis_analysis(&block, NULL);
            vorbis_bitrate_addblock(&block);

            while (vorbis_bitrate_flushpacket(&dsp, &packet)) {

               // add the packet to the bitstream
               ogg_stream_packetin(&stream, &packet);

               // From vorbis-tools-1.0/oggenc/encode.c:
               //   If we've gone over a page boundary, we can do actual output,
               //   so do so (for however many pages are available).

               while (!eos) {
                  int result = ogg_stream_pageout(&stream, &page);
                  if (!result) {
                     break;
                  }

                  outFile.Write(page.header, page.header_len);
                  outFile.Write(page.body, page.body_len);

                  if (ogg_page_eos(&page)) {
                     eos = 1;
                  }
               }
            }
         }

         updateResult = progress.Update(mixer->MixGetCurrentTime() - t0, t1 - t0);
      }
   }

   ogg_stream_clear(&stream);

   vorbis_block_clear(&block);
   vorbis_dsp_clear(&dsp);
   vorbis_info_clear(&info);
   vorbis_comment_clear(&comment);

   outFile.Close();

   return updateResult;
}
Esempio n. 18
0
static int
vorbis_read_header (SF_PRIVATE *psf, int log_data)
{	OGG_PRIVATE *odata = (OGG_PRIVATE *) psf->container_data ;
	VORBIS_PRIVATE *vdata = (VORBIS_PRIVATE *) psf->codec_data ;
	char *buffer ;
	int	bytes ;
	int i, nn ;

	odata->eos = 0 ;

	/* Weird stuff happens if these aren't called. */
	ogg_stream_reset (&odata->ostream) ;
	ogg_sync_reset (&odata->osync) ;

	/*
	**	Grab some data at the head of the stream.  We want the first page
	**	(which is guaranteed to be small and only contain the Vorbis
	**	stream initial header) We need the first page to get the stream
	**	serialno.
	*/

	/* Expose the buffer */
	buffer = ogg_sync_buffer (&odata->osync, 4096L) ;

	/* Grab the part of the header that has already been read. */
	memcpy (buffer, psf->header, psf->headindex) ;
	bytes = psf->headindex ;

	/* Submit a 4k block to libvorbis' Ogg layer */
	bytes += psf_fread (buffer + psf->headindex, 1, 4096 - psf->headindex, psf) ;
	ogg_sync_wrote (&odata->osync, bytes) ;

	/* Get the first page. */
	if ((nn = ogg_sync_pageout (&odata->osync, &odata->opage)) != 1)
	{	/* Have we simply run out of data?  If so, we're done. */
		if (bytes < 4096)
			return 0 ;

		/* Error case.  Must not be Vorbis data */
		psf_log_printf (psf, "Input does not appear to be an Ogg bitstream.\n") ;
		return SFE_MALFORMED_FILE ;
	} ;

	/*
	**	Get the serial number and set up the rest of decode.
	**	Serialno first ; use it to set up a logical stream.
	*/
	ogg_stream_clear (&odata->ostream) ;
	ogg_stream_init (&odata->ostream, ogg_page_serialno (&odata->opage)) ;

	if (ogg_stream_pagein (&odata->ostream, &odata->opage) < 0)
	{	/* Error ; stream version mismatch perhaps. */
		psf_log_printf (psf, "Error reading first page of Ogg bitstream data\n") ;
		return SFE_MALFORMED_FILE ;
	} ;

	if (ogg_stream_packetout (&odata->ostream, &odata->opacket) != 1)
	{	/* No page? must not be vorbis. */
		psf_log_printf (psf, "Error reading initial header packet.\n") ;
		return SFE_MALFORMED_FILE ;
	} ;

	/*
	**	This function (vorbis_read_header) gets called multiple times, so the OGG
	**	and vorbis structs have to be cleared every time we pass through to
	**	prevent memory leaks.
	*/
	vorbis_block_clear (&vdata->vblock) ;
	vorbis_dsp_clear (&vdata->vdsp) ;
	vorbis_comment_clear (&vdata->vcomment) ;
	vorbis_info_clear (&vdata->vinfo) ;

	/*
	**	Extract the initial header from the first page and verify that the
	**	Ogg bitstream is in fact Vorbis data.
	**
	**	I handle the initial header first instead of just having the code
	**	read all three Vorbis headers at once because reading the initial
	**	header is an easy way to identify a Vorbis bitstream and it's
	**	useful to see that functionality seperated out.
	*/
	vorbis_info_init (&vdata->vinfo) ;
	vorbis_comment_init (&vdata->vcomment) ;

	if (vorbis_synthesis_headerin (&vdata->vinfo, &vdata->vcomment, &odata->opacket) < 0)
	{	/* Error case ; not a vorbis header. */
		psf_log_printf (psf, "Found Vorbis in stream header, but vorbis_synthesis_headerin failed.\n") ;
		return SFE_MALFORMED_FILE ;
	} ;

	/*
	**	Common Ogg metadata fields?
	**	TITLE, VERSION, ALBUM, TRACKNUMBER, ARTIST, PERFORMER, COPYRIGHT, LICENSE,
	**	ORGANIZATION, DESCRIPTION, GENRE, DATE, LOCATION, CONTACT, ISRC,
	*/

	if (log_data)
	{	int k ;

		for (k = 0 ; k < ARRAY_LEN (vorbis_metatypes) ; k++)
		{	char *dd ;

			dd = vorbis_comment_query (&vdata->vcomment, vorbis_metatypes [k].name, 0) ;
			if (dd == NULL)
				continue ;
			psf_store_string (psf, vorbis_metatypes [k].id, dd) ;
		} ;
	} ;

	/*
	**	At this point, we're sure we're Vorbis.	We've set up the logical (Ogg)
	**	bitstream decoder. Get the comment and codebook headers and set up the
	**	Vorbis decoder.
	**
	**	The next two packets in order are the comment and codebook headers.
	**	They're likely large and may span multiple pages.  Thus we reead
	**	and submit data until we get our two pacakets, watching that no
	**	pages are missing.  If a page is missing, error out ; losing a
	**	header page is the only place where missing data is fatal.
	*/

	i = 0 ;			/* Count of number of packets read */
	while (i < 2)
	{	int result = ogg_sync_pageout (&odata->osync, &odata->opage) ;
		if (result == 0)
		{	/* Need more data */
			buffer = ogg_sync_buffer (&odata->osync, 4096) ;
			bytes = psf_fread (buffer, 1, 4096, psf) ;

			if (bytes == 0 && i < 2)
			{	psf_log_printf (psf, "End of file before finding all Vorbis headers!\n") ;
				return SFE_MALFORMED_FILE ;
			} ;
			nn = ogg_sync_wrote (&odata->osync, bytes) ;
		}
		else if (result == 1)
		{	/*
			**	Don't complain about missing or corrupt data yet. We'll
			**	catch it at the packet output phase.
			**
			**	We can ignore any errors here as they'll also become apparent
			**	at packetout.
			*/
			nn = ogg_stream_pagein (&odata->ostream, &odata->opage) ;
			while (i < 2)
			{	result = ogg_stream_packetout (&odata->ostream, &odata->opacket) ;
				if (result == 0)
					break ;
				if (result < 0)
				{	/*	Uh oh ; data at some point was corrupted or missing!
					**	We can't tolerate that in a header. Die. */
					psf_log_printf (psf, "Corrupt secondary header.	Exiting.\n") ;
					return SFE_MALFORMED_FILE ;
				} ;

				vorbis_synthesis_headerin (&vdata->vinfo, &vdata->vcomment, &odata->opacket) ;
				i++ ;
			} ;
		} ;
	} ;

	if (log_data)
	{	int printed_metadata_msg = 0 ;
		int k ;

		psf_log_printf (psf, "Bitstream is %d channel, %D Hz\n", vdata->vinfo.channels, vdata->vinfo.rate) ;
		psf_log_printf (psf, "Encoded by : %s\n", vdata->vcomment.vendor) ;

		/* Throw the comments plus a few lines about the bitstream we're decoding. */
		for (k = 0 ; k < ARRAY_LEN (vorbis_metatypes) ; k++)
		{	char *dd ;

			dd = vorbis_comment_query (&vdata->vcomment, vorbis_metatypes [k].name, 0) ;
			if (dd == NULL)
				continue ;

			if (printed_metadata_msg == 0)
			{	psf_log_printf (psf, "Metadata :\n") ;
				printed_metadata_msg = 1 ;
			} ;

			psf_store_string (psf, vorbis_metatypes [k].id, dd) ;
			psf_log_printf (psf, "  %-10s : %s\n", vorbis_metatypes [k].name, dd) ;
		} ;

		psf_log_printf (psf, "End\n") ;
	} ;

	psf->sf.samplerate	= vdata->vinfo.rate ;
	psf->sf.channels	= vdata->vinfo.channels ;
	psf->sf.format		= SF_FORMAT_OGG | SF_FORMAT_VORBIS ;

	/*	OK, got and parsed all three headers. Initialize the Vorbis
	**	packet->PCM decoder.
	**	Central decode state. */
	vorbis_synthesis_init (&vdata->vdsp, &vdata->vinfo) ;

	/*	Local state for most of the decode so multiple block decodes can
	**	proceed in parallel. We could init multiple vorbis_block structures
	**	for vd here. */
	vorbis_block_init (&vdata->vdsp, &vdata->vblock) ;

	vdata->loc = 0 ;

	return 0 ;
} /* vorbis_read_header */
Esempio n. 19
0
static int
vorbis_write_header (SF_PRIVATE *psf, int UNUSED (calc_length))
{	OGG_PRIVATE *odata = (OGG_PRIVATE *) psf->container_data ;
	VORBIS_PRIVATE *vdata = (VORBIS_PRIVATE *) psf->codec_data ;
	int k, ret ;

	vorbis_info_init (&vdata->vinfo) ;

	/* The style of encoding should be selectable here, VBR quality mode. */
	ret = vorbis_encode_init_vbr (&vdata->vinfo, psf->sf.channels, psf->sf.samplerate, vdata->quality) ;

#if 0
	ret = vorbis_encode_init (&vdata->vinfo, psf->sf.channels, psf->sf.samplerate, -1, 128000, -1) ; /* average bitrate mode */
	ret = (	vorbis_encode_setup_managed (&vdata->vinfo, psf->sf.channels, psf->sf.samplerate, -1, 128000, -1)
					|| vorbis_encode_ctl (&vdata->vinfo, OV_ECTL_RATEMANAGE_AVG, NULL)
					|| vorbis_encode_setup_init (&vdata->vinfo)
				) ;
#endif
	if (ret)
		return SFE_BAD_OPEN_FORMAT ;

	vdata->loc = 0 ;

	/* add a comment */
	vorbis_comment_init (&vdata->vcomment) ;

	vorbis_comment_add_tag (&vdata->vcomment, "ENCODER", "libsndfile") ;
	for (k = 0 ; k < SF_MAX_STRINGS ; k++)
	{	const char * name ;

		if (psf->strings.data [k].type == 0)
			break ;

		switch (psf->strings.data [k].type)
		{	case SF_STR_TITLE :
				name = "TITLE" ;
				break ;
			case SF_STR_COPYRIGHT :
				name = "COPYRIGHT" ;
				break ;
			case SF_STR_SOFTWARE :
				name = "SOFTWARE" ;
				break ;
			case SF_STR_ARTIST :
				name = "ARTIST" ;
				break ;
			case SF_STR_COMMENT :
				name = "COMMENT" ;
				break ;
			case SF_STR_DATE :
				name = "DATE" ;
				break ;
			case SF_STR_ALBUM :
				name = "ALBUM" ;
				break ;
			case SF_STR_LICENSE :
				name = "LICENSE" ;
				break ;
			case SF_STR_TRACKNUMBER :
				name = "Tracknumber" ;
				break ;
			case SF_STR_GENRE :
				name = "Genre" ;
				break ;

			default :
				continue ;
		} ;

		vorbis_comment_add_tag (&vdata->vcomment, name, psf->strings.storage + psf->strings.data [k].offset) ;
	} ;

	/* set up the analysis state and auxiliary encoding storage */
	vorbis_analysis_init (&vdata->vdsp, &vdata->vinfo) ;
	vorbis_block_init (&vdata->vdsp, &vdata->vblock) ;

	/*
	**	Set up our packet->stream encoder.
	**	Pick a random serial number ; that way we can more likely build
	**	chained streams just by concatenation.
	*/

	ogg_stream_init (&odata->ostream, psf_rand_int32 ()) ;

	/* Vorbis streams begin with three headers ; the initial header (with
	   most of the codec setup parameters) which is mandated by the Ogg
	   bitstream spec.  The second header holds any comment fields.	 The
	   third header holds the bitstream codebook.  We merely need to
	   make the headers, then pass them to libvorbis one at a time ;
	   libvorbis handles the additional Ogg bitstream constraints */

	{	ogg_packet header ;
		ogg_packet header_comm ;
		ogg_packet header_code ;
		int result ;

		vorbis_analysis_headerout (&vdata->vdsp, &vdata->vcomment, &header, &header_comm, &header_code) ;
		ogg_stream_packetin (&odata->ostream, &header) ; /* automatically placed in its own page */
		ogg_stream_packetin (&odata->ostream, &header_comm) ;
		ogg_stream_packetin (&odata->ostream, &header_code) ;

		/* This ensures the actual
		 * audio data will start on a new page, as per spec
		 */
		while ((result = ogg_stream_flush (&odata->ostream, &odata->opage)) != 0)
		{	psf_fwrite (odata->opage.header, 1, odata->opage.header_len, psf) ;
			psf_fwrite (odata->opage.body, 1, odata->opage.body_len, psf) ;
		} ;
	}

	return 0 ;
} /* vorbis_write_header */
int oe_encode ( oe_enc_opt* opt )
{
	ogg_stream_state os;
	ogg_page 		 og;
	ogg_packet 		 op;

	vorbis_dsp_state vd;
	vorbis_block     vb;
	vorbis_info      vi;

	long	samplesdone   = 0;
    int		eos;
	long	bytes_written = 0;
	long	packetsdone   = 0;
	int		ret           = 0;
	
	vorbis_info_init ( &vi );

	if ( opt->quality >= 0.0f )
	{
		if ( vorbis_encode_init_vbr ( &vi, opt->channels, opt->rate, opt->quality ) )
		{
			vorbis_info_clear ( &vi );
			return 1;
		}
	}
	else
	{
		if ( vorbis_encode_init ( 
									&vi,
									opt->channels,
									opt->rate,
									opt->max_bitrate > 0 ? opt->max_bitrate * 1000 : -1,
									opt->bitrate * 1000, 
									opt->min_bitrate > 0 ? opt->min_bitrate * 1000 : -1
								) )
		{
			vorbis_info_clear ( &vi );
			return 1;
		}
	}

	vorbis_analysis_init ( &vd, &vi );
	vorbis_block_init    ( &vd, &vb );

	ogg_stream_init ( &os, opt->serialno );

	ogg_packet header_main;
	ogg_packet header_comments;
	ogg_packet header_codebooks;
	int result;

	vorbis_analysis_headerout ( &vd,opt->comments, &header_main, &header_comments, &header_codebooks );

	ogg_stream_packetin ( &os, &header_main );
	ogg_stream_packetin ( &os, &header_comments );
	ogg_stream_packetin ( &os, &header_codebooks );

	while ( ( result = ogg_stream_flush ( &os, &og ) ) )
	{
		if ( !result )
			break;
		
		ret = oe_write_page ( &og, opt->out );

		if ( ret != og.header_len + og.body_len )
		{
			ret = 1;
			goto cleanup;
		}
		else
			bytes_written += ret;
	}
	
	eos = 0;

	while ( !eos )
	{
		float** buffer       = vorbis_analysis_buffer ( &vd, READSIZE );
		long    samples_read = opt->read_samples ( opt->readdata, buffer, READSIZE );

		if ( samples_read == 0 )
			vorbis_analysis_wrote ( &vd, 0 );
		else
		{
			samplesdone += samples_read;

			vorbis_analysis_wrote ( &vd, samples_read );
		}

		while ( vorbis_analysis_blockout ( &vd, &vb ) == 1 )
		{
			vorbis_analysis         ( &vb, NULL );
			vorbis_bitrate_addblock ( &vb );

			while ( vorbis_bitrate_flushpacket ( &vd, &op ) )
			{
				ogg_stream_packetin ( &os,&op );
				packetsdone++;

				while ( !eos )
				{
					int result = ogg_stream_pageout ( &os, &og );

					if ( !result )
						break;

					ret = oe_write_page ( &og, opt->out );

					if ( ret != og.header_len + og.body_len )
					{
						ret = 1;
						goto cleanup;
					}
					else
						bytes_written += ret; 
	
					if ( ogg_page_eos ( &og ) )
						eos = 1;
				}
			}
		}
	}

	ret = 0;

cleanup:

	ogg_stream_clear ( &os );

	vorbis_block_clear ( &vb );
	vorbis_dsp_clear   ( &vd );
	vorbis_info_clear  ( &vi );

	return ret;
}
Esempio n. 21
0
/*

  return: audio wants more packets
*/
static qboolean OGV_LoadAudio(cinematic_t *cin)
{
	qboolean     anyDataTransferred = qtrue;
	float        **pcm;
	int          frames, frameNeeded;
	int          i, j;
	short        *ptr;
	ogg_packet   op;
	vorbis_block vb;

	memset(&op, 0, sizeof(op));
	memset(&vb, 0, sizeof(vb));
	vorbis_block_init(&g_ogm->vd, &vb);

	while (anyDataTransferred && g_ogm->currentTime + MAX_AUDIO_PRELOAD > (int)(g_ogm->vd.granulepos * 1000 / g_ogm->vi.rate))
	{
		anyDataTransferred = qfalse;

		if ((frames = vorbis_synthesis_pcmout(&g_ogm->vd, &pcm)) > 0)
		{
			// vorbis -> raw
			ptr = (short *)g_ogm->audioBuffer;

			frameNeeded = (SIZEOF_RAWBUFF) / (OGG_SAMPLEWIDTH * g_ogm->vi.channels);

			if (frames < frameNeeded)
			{
				frameNeeded = frames;
			}

			for (i = 0; i < frameNeeded; i++)
			{
				for (j = 0; j < g_ogm->vi.channels; j++)
				{
					*(ptr++) = (short)((pcm[j][i] >= -1.0f && pcm[j][i] <= 1.0f) ? pcm[j][i] * 32767.f : 32767 * ((pcm[j][i] > 0.0f) - (pcm[j][i] < 0.0f)));
				}
			}

			// tell libvorbis how many samples we actually consumed (we ate them all!)
			vorbis_synthesis_read(&g_ogm->vd, frameNeeded);

			if (!(cin->flags & CIN_silent))
			{
				S_RawSamples(0, frameNeeded, g_ogm->vi.rate, OGG_SAMPLEWIDTH, g_ogm->vi.channels, g_ogm->audioBuffer, 1.0f, 1.0f);
			}

			anyDataTransferred = qtrue;
		}

		if (!anyDataTransferred)
		{
			// op -> vorbis
			if (ogg_stream_packetout(&g_ogm->os_audio, &op))
			{
				if (vorbis_synthesis(&vb, &op) == 0)
				{
					vorbis_synthesis_blockin(&g_ogm->vd, &vb);
				}
				anyDataTransferred = qtrue;
			}
		}
	}

	vorbis_block_clear(&vb);

	return (qboolean)(g_ogm->currentTime + MIN_AUDIO_PRELOAD > (int)(g_ogm->vd.granulepos * 1000 / g_ogm->vi.rate));
}
Esempio n. 22
0
S32 encode_vorbis_file(const std::string& in_fname, const std::string& out_fname)
{
#define READ_BUFFER 1024
	unsigned char readbuffer[READ_BUFFER*4+44];   /* out of the data segment, not the stack */	/*Flawfinder: ignore*/

	ogg_stream_state os; /* take physical pages, weld into a logical stream of packets */
	ogg_page         og; /* one Ogg bitstream page.  Vorbis packets are inside */
	ogg_packet       op; /* one raw packet of data for decode */
	
	vorbis_info      vi; /* struct that stores all the static vorbis bitstream settings */
	vorbis_comment   vc; /* struct that stores all the user comments */
	
	vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
	vorbis_block     vb; /* local working space for packet->PCM decode */
	
	int eos=0;
	int result;

	U16 num_channels = 0;
	U32 sample_rate = 0;
	U32 bits_per_sample = 0;

	S32 format_error = 0;
	std::string error_msg;
	if ((format_error = check_for_invalid_wav_formats(in_fname, error_msg)))
	{
		llwarns << error_msg << ": " << in_fname << llendl;
		return(format_error);
	}

#if 1
	unsigned char wav_header[44];	/*Flawfinder: ignore*/

	S32 data_left = 0;

	LLAPRFile infile ;
	infile.open(in_fname,LL_APR_RB);
	if (!infile.getFileHandle())
	{
		llwarns << "Couldn't open temporary ogg file for writing: " << in_fname
			<< llendl;
		return(LLVORBISENC_SOURCE_OPEN_ERR);
	}

	LLAPRFile outfile ;
	outfile.open(out_fname,LL_APR_WPB);
	if (!outfile.getFileHandle())
	{
		llwarns << "Couldn't open upload sound file for reading: " << in_fname
			<< llendl;
		return(LLVORBISENC_DEST_OPEN_ERR);
	}
	
	 // parse the chunks
	 U32 chunk_length = 0;
	 U32 file_pos = 12;  // start at the first chunk (usually fmt but not always)
	 
	 while (infile.eof() != APR_EOF)
	 {
		 infile.seek(APR_SET,file_pos);
		 infile.read(wav_header, 44);
		 
		 chunk_length = ((U32) wav_header[7] << 24) 
			 + ((U32) wav_header[6] << 16) 
			 + ((U32) wav_header[5] << 8) 
			 + wav_header[4];
		 
//		 llinfos << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << llendl;
		 
		 if (!(strncmp((char *)&(wav_header[0]),"fmt ",4)))
		 {
			 num_channels = ((U16) wav_header[11] << 8) + wav_header[10];
			 sample_rate = ((U32) wav_header[15] << 24) 
				 + ((U32) wav_header[14] << 16) 
				 + ((U32) wav_header[13] << 8) 
				 + wav_header[12];
			 bits_per_sample = ((U16) wav_header[23] << 8) + wav_header[22];
		 }
	 	 else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
		 {
			 infile.seek(APR_SET,file_pos+8);
			 // leave the file pointer at the beginning of the data chunk data
			 data_left = chunk_length;			
			 break;
		 }
		 file_pos += (chunk_length + 8);
		 chunk_length = 0;
	 } 
	 

	 /********** Encode setup ************/
	 
	 /* choose an encoding mode */
	 /* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
	 vorbis_info_init(&vi);

	 // always encode to mono

	 // SL-52913 & SL-53779 determined this quality level to be our 'good
	 // enough' general-purpose quality level with a nice low bitrate.
	 // Equivalent to oggenc -q0.5
	 F32 quality = 0.05f;
//	 quality = (bitrate==128000 ? 0.4f : 0.1);

//	 if (vorbis_encode_init(&vi, /* num_channels */ 1 ,sample_rate, -1, bitrate, -1))
	 if (vorbis_encode_init_vbr(&vi, /* num_channels */ 1 ,sample_rate, quality))
//	 if (vorbis_encode_setup_managed(&vi,1,sample_rate,-1,bitrate,-1) ||
//		vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE_AVG,NULL) ||
//		vorbis_encode_setup_init(&vi))
	{
		llwarns << "unable to initialize vorbis codec at quality " << quality << llendl;
		//		llwarns << "unable to initialize vorbis codec at bitrate " << bitrate << llendl;
		return(LLVORBISENC_DEST_OPEN_ERR);
	}
	 
	 /* add a comment */
	 vorbis_comment_init(&vc);
//	 vorbis_comment_add(&vc,"Linden");
	 
	 /* set up the analysis state and auxiliary encoding storage */
	 vorbis_analysis_init(&vd,&vi);
	 vorbis_block_init(&vd,&vb);
	 
	 /* set up our packet->stream encoder */
	 /* pick a random serial number; that way we can more likely build
		chained streams just by concatenation */
	 ogg_stream_init(&os, ll_rand());
	 
	 /* Vorbis streams begin with three headers; the initial header (with
		most of the codec setup parameters) which is mandated by the Ogg
		bitstream spec.  The second header holds any comment fields.  The
		third header holds the bitstream codebook.  We merely need to
		make the headers, then pass them to libvorbis one at a time;
		libvorbis handles the additional Ogg bitstream constraints */
	 
	 {
		 ogg_packet header;
		 ogg_packet header_comm;
		 ogg_packet header_code;
		 
		 vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
		 ogg_stream_packetin(&os,&header); /* automatically placed in its own
											  page */
		 ogg_stream_packetin(&os,&header_comm);
		 ogg_stream_packetin(&os,&header_code);
		 
		 /* We don't have to write out here, but doing so makes streaming 
		  * much easier, so we do, flushing ALL pages. This ensures the actual
		  * audio data will start on a new page
		  */
		 while(!eos){
			 int result=ogg_stream_flush(&os,&og);
			 if(result==0)break;
			 outfile.write(og.header, og.header_len);
			 outfile.write(og.body, og.body_len);
		 }
		 
	 }
	 
	 
	 while(!eos)
	 {
		 long bytes_per_sample = bits_per_sample/8;

		 long bytes=(long)infile.read(readbuffer,llclamp((S32)(READ_BUFFER*num_channels*bytes_per_sample),0,data_left)); /* stereo hardwired here */
		 
		 if (bytes==0)
		 {
			 /* end of file.  this can be done implicitly in the mainline,
				but it's easier to see here in non-clever fashion.
				Tell the library we're at end of stream so that it can handle
				the last frame and mark end of stream in the output properly */

			 vorbis_analysis_wrote(&vd,0);
//			 eos = 1;
			 
		 }
		 else
		 {
			 long i;
			 long samples;
			 int temp;

			 data_left -= bytes;
             /* data to encode */
			 
			 /* expose the buffer to submit data */
			 float **buffer=vorbis_analysis_buffer(&vd,READ_BUFFER);
			
			 i = 0;
			 samples = bytes / (num_channels * bytes_per_sample);

			 if (num_channels == 2)
			 {
				 if (bytes_per_sample == 2)
				 {
					 /* uninterleave samples */
					 for(i=0; i<samples ;i++)
					 {
					 	 temp =  ((signed char *)readbuffer)[i*4+1];	/*Flawfinder: ignore*/
						 temp += ((signed char *)readbuffer)[i*4+3];	/*Flawfinder: ignore*/
						 temp <<= 8;
						 temp += readbuffer[i*4];
						 temp += readbuffer[i*4+2];

						 buffer[0][i] = ((float)temp) / 65536.f;
					 }
				 }
				 else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
				 {
					 /* uninterleave samples */
					 for(i=0; i<samples ;i++)
					 {
					 	 temp  = readbuffer[i*2+0];
						 temp += readbuffer[i*2+1];
						 temp -= 256;
						 buffer[0][i] = ((float)temp) / 256.f;
					 }
				 } 
			 }
			 else if (num_channels == 1)
			 {
				 if (bytes_per_sample == 2)
				 {
					 for(i=0; i < samples ;i++)
					 {
					 	 temp = ((signed char*)readbuffer)[i*2+1];
						 temp <<= 8;
						 temp += readbuffer[i*2];
						 buffer[0][i] = ((float)temp) / 32768.f;
					 }
				 }
				 else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
				 {
					 for(i=0; i < samples ;i++)
					 {
						 temp = readbuffer[i];
						 temp -= 128;
						 buffer[0][i] = ((float)temp) / 128.f;
					 }
				 }
			 }
				
			 /* tell the library how much we actually submitted */
			 vorbis_analysis_wrote(&vd,i);
		 }
			 
		 /* vorbis does some data preanalysis, then divvies up blocks for
			more involved (potentially parallel) processing.  Get a single
			block for encoding now */
		 while(vorbis_analysis_blockout(&vd,&vb)==1)
		 {
			 
			 /* analysis */
			/* Do the main analysis, creating a packet */
			vorbis_analysis(&vb, NULL);
			vorbis_bitrate_addblock(&vb);

			while(vorbis_bitrate_flushpacket(&vd, &op)) 
			{
			 
			 /* weld the packet into the bitstream */
			 ogg_stream_packetin(&os,&op);
			 
			 /* write out pages (if any) */
			 while(!eos)
			 {
				 result = ogg_stream_pageout(&os,&og);

				 if(result==0)
				 	break;

				 outfile.write(og.header, og.header_len);
				 outfile.write(og.body, og.body_len);
				 
				 /* this could be set above, but for illustrative purposes, I do
					it here (to show that vorbis does know where the stream ends) */
				 
				 if(ogg_page_eos(&og))
				 	eos=1;
				 
			 }
			}
		 }
	 }
	 
	 
	 
	 /* clean up and exit.  vorbis_info_clear() must be called last */
	 
	 ogg_stream_clear(&os);
	 vorbis_block_clear(&vb);
	 vorbis_dsp_clear(&vd);
	 vorbis_comment_clear(&vc);
	 vorbis_info_clear(&vi);
	 
	 /* ogg_page and ogg_packet structs always point to storage in
		libvorbis.  They're never freed or manipulated directly */
	 
//	 fprintf(stderr,"Vorbis encoding: Done.\n");
	 llinfos << "Vorbis encoding: Done." << llendl;
	 
#endif
	 return(LLVORBISENC_NOERR);
	 
}
Esempio n. 23
0
 int main(){
  ogg_sync_state   oy; /* sync and verify incoming physical bitstream */
  ogg_stream_state os; /* take physical pages, weld into a logical
                          stream of packets */
  ogg_page         og; /* one Ogg bitstream page. Vorbis packets are inside */
  ogg_packet       op; /* one raw packet of data for decode */

  vorbis_info      vi; /* struct that stores all the static vorbis bitstream
                          settings */
  vorbis_comment   vc; /* struct that stores all the bitstream user comments */
  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
  vorbis_block     vb; /* local working space for packet->PCM decode */

  char *buffer;
  int  bytes;

  FILE *instream;
  FILE *outstream;
  char *inname = "01.ogg";
  char *outname = "esmith2000-09-28d1t15.raw";

//#ifdef _WIN32 /* We need to set stdin/stdout to binary mode. Damn windows. */
//  /* Beware the evil ifdef. We avoid these where we can, but this one we
//     cannot. Don't add any more, you'll probably go to hell if you do. */
//  //_setmode( _fileno( stdin ), _O_BINARY );
//  //_setmode( _fileno( stdout ), _O_BINARY );
//#endif

#if defined(macintosh) && defined(__MWERKS__)
  {
    int argc;
    char **argv;
    argc=ccommand(&argv); /* get a "command line" from the Mac user */
                     /* this also lets the user set stdin and stdout */
  }
#endif

  /********** Decode setup ************/

  //opening the file
  
  
  if( fopen_s( &instream, inname, "rb" ) != 0 )
  {
	  fprintf(stderr,"Can not open file %s\n", inname);
      exit(1);
  };
  
  if( fopen_s( &outstream, outname, "wb" ) != 0 )
  {
	  fprintf(stderr,"Can not open file %s\n", outname);
      exit(1);
  }


  ogg_sync_init(&oy); /* Now we can read pages */
  
  while(1){ /* we repeat if the bitstream is chained */
    int eos=0;
    int i;

    /* grab some data at the head of the stream. We want the first page
       (which is guaranteed to be small and only contain the Vorbis
       stream initial header) We need the first page to get the stream
       serialno. */

    /* submit a 4k block to libvorbis' Ogg layer */
    buffer=ogg_sync_buffer(&oy,4096);
    //bytes=fread(buffer,1,4096,stdin);
	bytes=fread(buffer,1,4096,instream);
    ogg_sync_wrote(&oy,bytes);
    
    /* Get the first page. */
    if(ogg_sync_pageout(&oy,&og)!=1){
      /* have we simply run out of data?  If so, we're done. */
      if(bytes<4096)break;
      
      /* error case.  Must not be Vorbis data */
      fprintf(stderr,"Input does not appear to be an Ogg bitstream.\n");
      exit(1);
    }
  
    /* Get the serial number and set up the rest of decode. */
    /* serialno first; use it to set up a logical stream */
    ogg_stream_init(&os,ogg_page_serialno(&og));
    
    /* extract the initial header from the first page and verify that the
       Ogg bitstream is in fact Vorbis data */
    
    /* I handle the initial header first instead of just having the code
       read all three Vorbis headers at once because reading the initial
       header is an easy way to identify a Vorbis bitstream and it's
       useful to see that functionality seperated out. */
    
    vorbis_info_init(&vi);
    vorbis_comment_init(&vc);
    if(ogg_stream_pagein(&os,&og)<0){ 
      /* error; stream version mismatch perhaps */
      fprintf(stderr,"Error reading first page of Ogg bitstream data.\n");
      exit(1);
    }
    
    if(ogg_stream_packetout(&os,&op)!=1){ 
      /* no page? must not be vorbis */
      fprintf(stderr,"Error reading initial header packet.\n");
      exit(1);
    }
    
    if(vorbis_synthesis_headerin(&vi,&vc,&op)<0){ 
      /* error case; not a vorbis header */
      fprintf(stderr,"This Ogg bitstream does not contain Vorbis "
              "audio data.\n");
      exit(1);
    }
    
    /* At this point, we're sure we're Vorbis. We've set up the logical
       (Ogg) bitstream decoder. Get the comment and codebook headers and
       set up the Vorbis decoder */
    
    /* The next two packets in order are the comment and codebook headers.
       They're likely large and may span multiple pages. Thus we read
       and submit data until we get our two packets, watching that no
       pages are missing. If a page is missing, error out; losing a
       header page is the only place where missing data is fatal. */
    
    i=0;
    while(i<2){
      while(i<2){
        int result=ogg_sync_pageout(&oy,&og);
        if(result==0)break; /* Need more data */
        /* Don't complain about missing or corrupt data yet. We'll
           catch it at the packet output phase */
        if(result==1){
          ogg_stream_pagein(&os,&og); /* we can ignore any errors here
                                         as they'll also become apparent
                                         at packetout */
          while(i<2){
            result=ogg_stream_packetout(&os,&op);
            if(result==0)break;
            if(result<0){
              /* Uh oh; data at some point was corrupted or missing!
                 We can't tolerate that in a header.  Die. */
              fprintf(stderr,"Corrupt secondary header.  Exiting.\n");
              exit(1);
            }
            result=vorbis_synthesis_headerin(&vi,&vc,&op);
            if(result<0){
              fprintf(stderr,"Corrupt secondary header.  Exiting.\n");
              exit(1);
            }
            i++;
          }
        }
      }
      /* no harm in not checking before adding more */
      buffer=ogg_sync_buffer(&oy,4096);
      //bytes=fread(buffer,1,4096,stdin);
	  bytes=fread(buffer,1,4096,instream);
      if(bytes==0 && i<2){
        fprintf(stderr,"End of file before finding all Vorbis headers!\n");
        exit(1);
      }
      ogg_sync_wrote(&oy,bytes);
    }
    
    /* Throw the comments plus a few lines about the bitstream we're
       decoding */
    {
      char **ptr=vc.user_comments;
      while(*ptr){
        fprintf(stderr,"%s\n",*ptr);
        ++ptr;
      }
      fprintf(stderr,"\nBitstream is %d channel, %ldHz\n",vi.channels,vi.rate);
      fprintf(stderr,"Encoded by: %s\n\n",vc.vendor);
    }
    
    convsize=4096/vi.channels;

    /* OK, got and parsed all three headers. Initialize the Vorbis
       packet->PCM decoder. */
    if(vorbis_synthesis_init(&vd,&vi)==0){ /* central decode state */
      vorbis_block_init(&vd,&vb);          /* local state for most of the decode
                                              so multiple block decodes can
                                              proceed in parallel. We could init
                                              multiple vorbis_block structures
                                              for vd here */
      
      /* The rest is just a straight decode loop until end of stream */
      while(!eos){
        while(!eos){
          int result=ogg_sync_pageout(&oy,&og);
          if(result==0)break; /* need more data */
          if(result<0){ /* missing or corrupt data at this page position */
            fprintf(stderr,"Corrupt or missing data in bitstream; "
                    "continuing...\n");
          }else{
            ogg_stream_pagein(&os,&og); /* can safely ignore errors at
                                           this point */
            while(1){
              result=ogg_stream_packetout(&os,&op);
              
              if(result==0)break; /* need more data */
              if(result<0){ /* missing or corrupt data at this page position */
                /* no reason to complain; already complained above */
              }else{
                /* we have a packet.  Decode it */
                float **pcm;
                int samples;
                
                if(vorbis_synthesis(&vb,&op)==0) /* test for success! */
                  vorbis_synthesis_blockin(&vd,&vb);
                /* 
                   
                **pcm is a multichannel float vector.  In stereo, for
                example, pcm[0] is left, and pcm[1] is right.  samples is
                the size of each channel.  Convert the float values
                (-1.<=range<=1.) to whatever PCM format and write it out */
                
                while((samples=vorbis_synthesis_pcmout(&vd,&pcm))>0){
                  int j;
                  int clipflag=0;
                  int bout=(samples<convsize?samples:convsize);
                  
                  /* convert floats to 16 bit signed ints (host order) and
                     interleave */
                  for(i=0;i<vi.channels;i++){
                    ogg_int16_t *ptr=convbuffer+i;
                    float  *mono=pcm[i];
                    for(j=0;j<bout;j++){
#if 1
                      int val=floor(mono[j]*32767.f+.5f);
#else /* optional dither */
                      int val=mono[j]*32767.f+drand48()-0.5f;
#endif
                      /* might as well guard against clipping */
                      if(val>32767){
                        val=32767;
                        clipflag=1;
                      }
                      if(val<-32768){
                        val=-32768;
                        clipflag=1;
                      }
                      *ptr=val;
                      ptr+=vi.channels;
                    }
                  }
                  
                  if(clipflag)
                    fprintf(stderr,"Clipping in frame %ld\n",(long)(vd.sequence));
                  
                  
                  //fwrite(convbuffer,2*vi.channels,bout,stdout);
				  fwrite(convbuffer,2*vi.channels,bout,outstream);
                  
                  vorbis_synthesis_read(&vd,bout); /* tell libvorbis how
                                                      many samples we
                                                      actually consumed */
                }            
              }
            }
            if(ogg_page_eos(&og))eos=1;
          }
        }
        if(!eos){
          buffer=ogg_sync_buffer(&oy,4096);
          //bytes=fread(buffer,1,4096,stdin);
		  bytes=fread(buffer,1,4096,instream);
          ogg_sync_wrote(&oy,bytes);
          if(bytes==0)eos=1;
        }
      }
      
      /* ogg_page and ogg_packet structs always point to storage in
         libvorbis.  They're never freed or manipulated directly */
      
      vorbis_block_clear(&vb);
      vorbis_dsp_clear(&vd);
    }else{
      fprintf(stderr,"Error: Corrupt header during playback initialization.\n");
    }

    /* clean up this logical bitstream; before exit we see if we're
       followed by another [chained] */
    
    ogg_stream_clear(&os);
    vorbis_comment_clear(&vc);
    vorbis_info_clear(&vi);  /* must be called last */
  }

  /* OK, clean up the framer */
  ogg_sync_clear(&oy);
  
  fprintf(stderr,"Done.\n");
  return(0);
}
Esempio n. 24
0
int main(){
  ogg_stream_state os; /* take physical pages, weld into a logical
			  stream of packets */
  ogg_page         og; /* one Ogg bitstream page.  Vorbis packets are inside */
  ogg_packet       op; /* one raw packet of data for decode */
  
  vorbis_info      vi; /* struct that stores all the static vorbis bitstream
			  settings */
  vorbis_comment   vc; /* struct that stores all the user comments */

  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
  vorbis_block     vb; /* local working space for packet->PCM decode */

  int eos=0,ret;
  int i, founddata;

#if defined(macintosh) && defined(__MWERKS__)
  int argc = 0;
  char **argv = NULL;
  argc = ccommand(&argv); /* get a "command line" from the Mac user */
                          /* this also lets the user set stdin and stdout */
#endif

  /* we cheat on the WAV header; we just bypass 44 bytes and never
     verify that it matches 16bit/stereo/44.1kHz.  This is just an
     example, after all. */

#ifdef _WIN32 /* We need to set stdin/stdout to binary mode. Damn windows. */
  /* if we were reading/writing a file, it would also need to in
     binary mode, eg, fopen("file.wav","wb"); */
  /* Beware the evil ifdef. We avoid these where we can, but this one we 
     cannot. Don't add any more, you'll probably go to hell if you do. */
  _setmode( _fileno( stdin ), _O_BINARY );
  _setmode( _fileno( stdout ), _O_BINARY );
#endif


  /* we cheat on the WAV header; we just bypass the header and never
     verify that it matches 16bit/stereo/44.1kHz.  This is just an
     example, after all. */

  readbuffer[0] = '\0';
  for (i=0, founddata=0; i<30 && ! feof(stdin) && ! ferror(stdin); i++)
  {
    fread(readbuffer,1,2,stdin);

    if ( ! strncmp((char*)readbuffer, "da", 2) )
    {
      founddata = 1;
      fread(readbuffer,1,6,stdin);
      break;
    }
  }

  /********** Encode setup ************/

  vorbis_info_init(&vi);

  /* choose an encoding mode.  A few possibilities commented out, one
     actually used: */

  /*********************************************************************
   Encoding using a VBR quality mode.  The usable range is -.1
   (lowest quality, smallest file) to 1. (highest quality, largest file).
   Example quality mode .4: 44kHz stereo coupled, roughly 128kbps VBR 
  
   ret = vorbis_encode_init_vbr(&vi,2,44100,.4);

   ---------------------------------------------------------------------

   Encoding using an average bitrate mode (ABR).
   example: 44kHz stereo coupled, average 128kbps VBR 
  
   ret = vorbis_encode_init(&vi,2,44100,-1,128000,-1);

   ---------------------------------------------------------------------

   Encode using a quality mode, but select that quality mode by asking for
   an approximate bitrate.  This is not ABR, it is true VBR, but selected
   using the bitrate interface, and then turning bitrate management off:

   ret = ( vorbis_encode_setup_managed(&vi,2,44100,-1,128000,-1) ||
           vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE2_SET,NULL) ||
           vorbis_encode_setup_init(&vi));

   *********************************************************************/

  ret=vorbis_encode_init_vbr(&vi,2,44100,0.1);

  /* do not continue if setup failed; this can happen if we ask for a
     mode that libVorbis does not support (eg, too low a bitrate, etc,
     will return 'OV_EIMPL') */

  if(ret)exit(1);

  /* add a comment */
  vorbis_comment_init(&vc);
  vorbis_comment_add_tag(&vc,"ENCODER","encoder_example.c");

  /* set up the analysis state and auxiliary encoding storage */
  vorbis_analysis_init(&vd,&vi);
  vorbis_block_init(&vd,&vb);
  
  /* set up our packet->stream encoder */
  /* pick a random serial number; that way we can more likely build
     chained streams just by concatenation */
  srand(time(NULL));
  ogg_stream_init(&os,rand());

  /* Vorbis streams begin with three headers; the initial header (with
     most of the codec setup parameters) which is mandated by the Ogg
     bitstream spec.  The second header holds any comment fields.  The
     third header holds the bitstream codebook.  We merely need to
     make the headers, then pass them to libvorbis one at a time;
     libvorbis handles the additional Ogg bitstream constraints */

  {
    ogg_packet header;
    ogg_packet header_comm;
    ogg_packet header_code;

    vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
    ogg_stream_packetin(&os,&header); /* automatically placed in its own
					 page */
    ogg_stream_packetin(&os,&header_comm);
    ogg_stream_packetin(&os,&header_code);

	/* This ensures the actual
	 * audio data will start on a new page, as per spec
	 */
	while(!eos){
		int result=ogg_stream_flush(&os,&og);
		if(result==0)break;
		fwrite(og.header,1,og.header_len,stdout);
		fwrite(og.body,1,og.body_len,stdout);
	}

  }
  
  while(!eos){
    long i;
    long bytes=fread(readbuffer,1,READ*4,stdin); /* stereo hardwired here */

    if(bytes==0){
      /* end of file.  this can be done implicitly in the mainline,
         but it's easier to see here in non-clever fashion.
         Tell the library we're at end of stream so that it can handle
         the last frame and mark end of stream in the output properly */
      vorbis_analysis_wrote(&vd,0);

    }else{
      /* data to encode */

      /* expose the buffer to submit data */
      float **buffer=vorbis_analysis_buffer(&vd,READ);
      
      /* uninterleave samples */
      for(i=0;i<bytes/4;i++){
	buffer[0][i]=((readbuffer[i*4+1]<<8)|
		      (0x00ff&(int)readbuffer[i*4]))/32768.f;
	buffer[1][i]=((readbuffer[i*4+3]<<8)|
		      (0x00ff&(int)readbuffer[i*4+2]))/32768.f;
      }
    
      /* tell the library how much we actually submitted */
      vorbis_analysis_wrote(&vd,i);
    }

    /* vorbis does some data preanalysis, then divvies up blocks for
       more involved (potentially parallel) processing.  Get a single
       block for encoding now */
    while(vorbis_analysis_blockout(&vd,&vb)==1){

      /* analysis, assume we want to use bitrate management */
      vorbis_analysis(&vb,NULL);
      vorbis_bitrate_addblock(&vb);

      while(vorbis_bitrate_flushpacket(&vd,&op)){
	
	/* weld the packet into the bitstream */
	ogg_stream_packetin(&os,&op);
	
	/* write out pages (if any) */
	while(!eos){
	  int result=ogg_stream_pageout(&os,&og);
	  if(result==0)break;
	  fwrite(og.header,1,og.header_len,stdout);
	  fwrite(og.body,1,og.body_len,stdout);
	  
	  /* this could be set above, but for illustrative purposes, I do
	     it here (to show that vorbis does know where the stream ends) */
	  
	  if(ogg_page_eos(&og))eos=1;
	}
      }
    }
  }

  /* clean up and exit.  vorbis_info_clear() must be called last */
  
  ogg_stream_clear(&os);
  vorbis_block_clear(&vb);
  vorbis_dsp_clear(&vd);
  vorbis_comment_clear(&vc);
  vorbis_info_clear(&vi);
  
  /* ogg_page and ogg_packet structs always point to storage in
     libvorbis.  They're never freed or manipulated directly */
  
  fprintf(stderr,"Done.\n");
  return(0);
}
Esempio n. 25
0
/*****************************************************************************
 * ProcessHeaders: process Vorbis headers.
 *****************************************************************************/
static int ProcessHeaders( decoder_t *p_dec )
{
    decoder_sys_t *p_sys = p_dec->p_sys;
    ogg_packet oggpacket;

    unsigned pi_size[XIPH_MAX_HEADER_COUNT];
    void *pp_data[XIPH_MAX_HEADER_COUNT];
    unsigned i_count;
    if( xiph_SplitHeaders( pi_size, pp_data, &i_count,
                           p_dec->fmt_in.i_extra, p_dec->fmt_in.p_extra) )
        return VLC_EGENERIC;
    if( i_count < 3 )
        return VLC_EGENERIC;

    oggpacket.granulepos = -1;
    oggpacket.e_o_s = 0;
    oggpacket.packetno = 0;

    /* Take care of the initial Vorbis header */
    oggpacket.b_o_s  = 1; /* yes this actually is a b_o_s packet :) */
    oggpacket.bytes  = pi_size[0];
    oggpacket.packet = pp_data[0];
    if( vorbis_synthesis_headerin( &p_sys->vi, &p_sys->vc, &oggpacket ) < 0 )
    {
        msg_Err( p_dec, "this bitstream does not contain Vorbis audio data");
        return VLC_EGENERIC;
    }

    /* Setup the format */
    p_dec->fmt_out.audio.i_rate     = p_sys->vi.rate;
    p_dec->fmt_out.audio.i_channels = p_sys->vi.channels;

    if( p_dec->fmt_out.audio.i_channels >= ARRAY_SIZE(pi_channels_maps) )
    {
        msg_Err( p_dec, "invalid number of channels (1-%zu): %i",
                 ARRAY_SIZE(pi_channels_maps),
                 p_dec->fmt_out.audio.i_channels );
        return VLC_EGENERIC;
    }

    p_dec->fmt_out.audio.i_physical_channels =
        p_dec->fmt_out.audio.i_original_channels =
            pi_channels_maps[p_sys->vi.channels];
    p_dec->fmt_out.i_bitrate = __MAX( 0, (int32_t) p_sys->vi.bitrate_nominal );

    date_Init( &p_sys->end_date, p_sys->vi.rate, 1 );

    msg_Dbg( p_dec, "channels:%d samplerate:%ld bitrate:%ud",
             p_sys->vi.channels, p_sys->vi.rate, p_dec->fmt_out.i_bitrate );

    /* The next packet in order is the comments header */
    oggpacket.b_o_s  = 0;
    oggpacket.bytes  = pi_size[1];
    oggpacket.packet = pp_data[1];
    if( vorbis_synthesis_headerin( &p_sys->vi, &p_sys->vc, &oggpacket ) < 0 )
    {
        msg_Err( p_dec, "2nd Vorbis header is corrupted" );
        return VLC_EGENERIC;
    }
    ParseVorbisComments( p_dec );

    /* The next packet in order is the codebooks header
     * We need to watch out that this packet is not missing as a
     * missing or corrupted header is fatal. */
    oggpacket.b_o_s  = 0;
    oggpacket.bytes  = pi_size[2];
    oggpacket.packet = pp_data[2];
    if( vorbis_synthesis_headerin( &p_sys->vi, &p_sys->vc, &oggpacket ) < 0 )
    {
        msg_Err( p_dec, "3rd Vorbis header is corrupted" );
        return VLC_EGENERIC;
    }

    if( !p_sys->b_packetizer )
    {
        /* Initialize the Vorbis packet->PCM decoder */
        vorbis_synthesis_init( &p_sys->vd, &p_sys->vi );
        vorbis_block_init( &p_sys->vd, &p_sys->vb );
    }
    else
    {
        p_dec->fmt_out.i_extra = p_dec->fmt_in.i_extra;
        p_dec->fmt_out.p_extra = xrealloc( p_dec->fmt_out.p_extra,
                                           p_dec->fmt_out.i_extra );
        memcpy( p_dec->fmt_out.p_extra,
                p_dec->fmt_in.p_extra, p_dec->fmt_out.i_extra );
    }

    ConfigureChannelOrder(p_sys->pi_chan_table, p_sys->vi.channels,
                          p_dec->fmt_out.audio.i_physical_channels, true);

    return VLC_SUCCESS;
}
Esempio n. 26
0
static gint vorbis_open(void)
{
    ogg_packet header;
    ogg_packet header_comm;
    ogg_packet header_code;

    vorbis_init(NULL);

    vorbis_info_init(&vi);
    vorbis_comment_init(&vc);

    if (tuple)
    {
        gchar tmpstr[32];
        gint scrint;

        add_string_from_tuple (& vc, "title", tuple, FIELD_TITLE);
        add_string_from_tuple (& vc, "artist", tuple, FIELD_ARTIST);
        add_string_from_tuple (& vc, "album", tuple, FIELD_ALBUM);
        add_string_from_tuple (& vc, "genre", tuple, FIELD_GENRE);
        add_string_from_tuple (& vc, "date", tuple, FIELD_DATE);
        add_string_from_tuple (& vc, "comment", tuple, FIELD_COMMENT);

        if ((scrint = tuple_get_int(tuple, FIELD_TRACK_NUMBER, NULL)))
        {
            g_snprintf(tmpstr, sizeof(tmpstr), "%d", scrint);
            vorbis_comment_add_tag(&vc, "tracknumber", tmpstr);
        }

        if ((scrint = tuple_get_int(tuple, FIELD_YEAR, NULL)))
        {
            g_snprintf(tmpstr, sizeof(tmpstr), "%d", scrint);
            vorbis_comment_add_tag(&vc, "year", tmpstr);
        }
    }

    if (vorbis_encode_init_vbr (& vi, input.channels, input.frequency,
     v_base_quality))
    {
        vorbis_info_clear(&vi);
        return 0;
    }

    vorbis_analysis_init(&vd, &vi);
    vorbis_block_init(&vd, &vb);

    srand(time(NULL));
    ogg_stream_init(&os, rand());

    vorbis_analysis_headerout(&vd, &vc, &header, &header_comm, &header_code);

    ogg_stream_packetin(&os, &header);
    ogg_stream_packetin(&os, &header_comm);
    ogg_stream_packetin(&os, &header_code);

    while (ogg_stream_flush (& os, & og))
    {
        write_output(og.header, og.header_len);
        write_output(og.body, og.body_len);
    }

    return 1;
}
Esempio n. 27
0
void VideoStreamPlaybackTheora::set_file(const String &p_file) {

	ERR_FAIL_COND(playing);
	ogg_packet op;
	th_setup_info *ts = NULL;

	file_name = p_file;
	if (file) {
		memdelete(file);
	}
	file = FileAccess::open(p_file, FileAccess::READ);
	ERR_FAIL_COND(!file);

#ifdef THEORA_USE_THREAD_STREAMING
	thread_exit = false;
	thread_eof = false;
	//pre-fill buffer
	int to_read = ring_buffer.space_left();
	int read = file->get_buffer(read_buffer.ptr(), to_read);
	ring_buffer.write(read_buffer.ptr(), read);

	thread = Thread::create(_streaming_thread, this);

#endif

	ogg_sync_init(&oy);

	/* init supporting Vorbis structures needed in header parsing */
	vorbis_info_init(&vi);
	vorbis_comment_init(&vc);

	/* init supporting Theora structures needed in header parsing */
	th_comment_init(&tc);
	th_info_init(&ti);

	theora_eos = false;
	vorbis_eos = false;

	/* Ogg file open; parse the headers */
	/* Only interested in Vorbis/Theora streams */
	int stateflag = 0;

	int audio_track_skip = audio_track;

	while (!stateflag) {
		int ret = buffer_data();
		if (ret == 0) break;
		while (ogg_sync_pageout(&oy, &og) > 0) {
			ogg_stream_state test;

			/* is this a mandated initial header? If not, stop parsing */
			if (!ogg_page_bos(&og)) {
				/* don't leak the page; get it into the appropriate stream */
				queue_page(&og);
				stateflag = 1;
				break;
			}

			ogg_stream_init(&test, ogg_page_serialno(&og));
			ogg_stream_pagein(&test, &og);
			ogg_stream_packetout(&test, &op);

			/* identify the codec: try theora */
			if (!theora_p && th_decode_headerin(&ti, &tc, &ts, &op) >= 0) {
				/* it is theora */
				copymem(&to, &test, sizeof(test));
				theora_p = 1;
			} else if (!vorbis_p && vorbis_synthesis_headerin(&vi, &vc, &op) >= 0) {

				/* it is vorbis */
				if (audio_track_skip) {
					vorbis_info_clear(&vi);
					vorbis_comment_clear(&vc);
					ogg_stream_clear(&test);
					vorbis_info_init(&vi);
					vorbis_comment_init(&vc);

					audio_track_skip--;
				} else {
					copymem(&vo, &test, sizeof(test));
					vorbis_p = 1;
				}
			} else {
				/* whatever it is, we don't care about it */
				ogg_stream_clear(&test);
			}
		}
		/* fall through to non-bos page parsing */
	}

	/* we're expecting more header packets. */
	while ((theora_p && theora_p < 3) || (vorbis_p && vorbis_p < 3)) {
		int ret;

		/* look for further theora headers */
		while (theora_p && (theora_p < 3) && (ret = ogg_stream_packetout(&to, &op))) {
			if (ret < 0) {
				fprintf(stderr, "Error parsing Theora stream headers; "
								"corrupt stream?\n");
				clear();
				return;
			}
			if (!th_decode_headerin(&ti, &tc, &ts, &op)) {
				fprintf(stderr, "Error parsing Theora stream headers; "
								"corrupt stream?\n");
				clear();
				return;
			}
			theora_p++;
		}

		/* look for more vorbis header packets */
		while (vorbis_p && (vorbis_p < 3) && (ret = ogg_stream_packetout(&vo, &op))) {
			if (ret < 0) {
				fprintf(stderr, "Error parsing Vorbis stream headers; corrupt stream?\n");
				clear();
				return;
			}
			ret = vorbis_synthesis_headerin(&vi, &vc, &op);
			if (ret) {
				fprintf(stderr, "Error parsing Vorbis stream headers; corrupt stream?\n");
				clear();
				return;
			}
			vorbis_p++;
			if (vorbis_p == 3) break;
		}

		/* The header pages/packets will arrive before anything else we
		care about, or the stream is not obeying spec */

		if (ogg_sync_pageout(&oy, &og) > 0) {
			queue_page(&og); /* demux into the appropriate stream */
		} else {
			int ret = buffer_data(); /* someone needs more data */
			if (ret == 0) {
				fprintf(stderr, "End of file while searching for codec headers.\n");
				clear();
				return;
			}
		}
	}

	/* and now we have it all.  initialize decoders */
	if (theora_p) {
		td = th_decode_alloc(&ti, ts);
		printf("Ogg logical stream %lx is Theora %dx%d %.02f fps",
				to.serialno, ti.pic_width, ti.pic_height,
				(double)ti.fps_numerator / ti.fps_denominator);
		px_fmt = ti.pixel_fmt;
		switch (ti.pixel_fmt) {
			case TH_PF_420: printf(" 4:2:0 video\n"); break;
			case TH_PF_422: printf(" 4:2:2 video\n"); break;
			case TH_PF_444: printf(" 4:4:4 video\n"); break;
			case TH_PF_RSVD:
			default:
				printf(" video\n  (UNKNOWN Chroma sampling!)\n");
				break;
		}
		if (ti.pic_width != ti.frame_width || ti.pic_height != ti.frame_height)
			printf("  Frame content is %dx%d with offset (%d,%d).\n",
					ti.frame_width, ti.frame_height, ti.pic_x, ti.pic_y);
		th_decode_ctl(td, TH_DECCTL_GET_PPLEVEL_MAX, &pp_level_max,
				sizeof(pp_level_max));
		pp_level = 0;
		th_decode_ctl(td, TH_DECCTL_SET_PPLEVEL, &pp_level, sizeof(pp_level));
		pp_inc = 0;

		int w;
		int h;
		w = (ti.pic_x + ti.frame_width + 1 & ~1) - (ti.pic_x & ~1);
		h = (ti.pic_y + ti.frame_height + 1 & ~1) - (ti.pic_y & ~1);
		size.x = w;
		size.y = h;

		texture->create(w, h, Image::FORMAT_RGBA8, Texture::FLAG_FILTER | Texture::FLAG_VIDEO_SURFACE);

	} else {
		/* tear down the partial theora setup */
		th_info_clear(&ti);
		th_comment_clear(&tc);
	}

	th_setup_free(ts);

	if (vorbis_p) {
		vorbis_synthesis_init(&vd, &vi);
		vorbis_block_init(&vd, &vb);
		fprintf(stderr, "Ogg logical stream %lx is Vorbis %d channel %ld Hz audio.\n",
				vo.serialno, vi.channels, vi.rate);
		//_setup(vi.channels, vi.rate);

	} else {
		/* tear down the partial vorbis setup */
		vorbis_info_clear(&vi);
		vorbis_comment_clear(&vc);
	}

	playing = false;
	buffering = true;
	time = 0;
	audio_frames_wrote = 0;
};
Esempio n. 28
0
/*****************************************************************************
 * OpenEncoder: probe the encoder and return score
 *****************************************************************************/
static int OpenEncoder( vlc_object_t *p_this )
{
    encoder_t *p_enc = (encoder_t *)p_this;
    encoder_sys_t *p_sys;
    int i_quality, i_min_bitrate, i_max_bitrate;
    ogg_packet header[3];

    if( p_enc->fmt_out.i_codec != VLC_CODEC_VORBIS &&
            !p_enc->b_force )
    {
        return VLC_EGENERIC;
    }

    /* Allocate the memory needed to store the decoder's structure */
    if( ( p_sys = (encoder_sys_t *)malloc(sizeof(encoder_sys_t)) ) == NULL )
        return VLC_ENOMEM;
    p_enc->p_sys = p_sys;

    p_enc->pf_encode_audio = Encode;
    p_enc->fmt_in.i_codec  = VLC_CODEC_FL32;
    p_enc->fmt_out.i_codec = VLC_CODEC_VORBIS;

    config_ChainParse( p_enc, ENC_CFG_PREFIX, ppsz_enc_options, p_enc->p_cfg );

    i_quality = var_GetInteger( p_enc, ENC_CFG_PREFIX "quality" );
    if( i_quality > 10 ) i_quality = 10;
    if( i_quality < 0 ) i_quality = 0;

    if( var_GetBool( p_enc, ENC_CFG_PREFIX "cbr" ) ) i_quality = 0;
    i_max_bitrate = var_GetInteger( p_enc, ENC_CFG_PREFIX "max-bitrate" );
    i_min_bitrate = var_GetInteger( p_enc, ENC_CFG_PREFIX "min-bitrate" );

    /* Initialize vorbis encoder */
    vorbis_info_init( &p_sys->vi );

    if( i_quality > 0 )
    {
        /* VBR mode */
        if( vorbis_encode_setup_vbr( &p_sys->vi,
                                     p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.audio.i_rate,
                                     i_quality * 0.1 ) )
        {
            vorbis_info_clear( &p_sys->vi );
            free( p_enc->p_sys );
            msg_Err( p_enc, "VBR mode initialisation failed" );
            return VLC_EGENERIC;
        }

        /* Do we have optional hard quality restrictions? */
        if( i_max_bitrate > 0 || i_min_bitrate > 0 )
        {
            struct ovectl_ratemanage_arg ai;
            vorbis_encode_ctl( &p_sys->vi, OV_ECTL_RATEMANAGE_GET, &ai );

            ai.bitrate_hard_min = i_min_bitrate;
            ai.bitrate_hard_max = i_max_bitrate;
            ai.management_active = 1;

            vorbis_encode_ctl( &p_sys->vi, OV_ECTL_RATEMANAGE_SET, &ai );

        }
        else
        {
            /* Turn off management entirely */
            vorbis_encode_ctl( &p_sys->vi, OV_ECTL_RATEMANAGE_SET, NULL );
        }
    }
    else
    {
        if( vorbis_encode_setup_managed( &p_sys->vi,
                                         p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.audio.i_rate,
                                         i_min_bitrate > 0 ? i_min_bitrate * 1000: -1,
                                         p_enc->fmt_out.i_bitrate,
                                         i_max_bitrate > 0 ? i_max_bitrate * 1000: -1 ) )
        {
            vorbis_info_clear( &p_sys->vi );
            msg_Err( p_enc, "CBR mode initialisation failed" );
            free( p_enc->p_sys );
            return VLC_EGENERIC;
        }
    }

    vorbis_encode_setup_init( &p_sys->vi );

    /* Add a comment */
    vorbis_comment_init( &p_sys->vc);
    vorbis_comment_add_tag( &p_sys->vc, "ENCODER", "VLC media player");

    /* Set up the analysis state and auxiliary encoding storage */
    vorbis_analysis_init( &p_sys->vd, &p_sys->vi );
    vorbis_block_init( &p_sys->vd, &p_sys->vb );

    /* Create and store headers */
    vorbis_analysis_headerout( &p_sys->vd, &p_sys->vc,
                               &header[0], &header[1], &header[2]);
    for( int i = 0; i < 3; i++ )
    {
        if( xiph_AppendHeaders( &p_enc->fmt_out.i_extra, &p_enc->fmt_out.p_extra,
                                header[i].bytes, header[i].packet ) )
        {
            p_enc->fmt_out.i_extra = 0;
            p_enc->fmt_out.p_extra = NULL;
        }
    }

    p_sys->i_channels = p_enc->fmt_in.audio.i_channels;
    p_sys->i_last_block_size = 0;
    p_sys->i_samples_delay = 0;

    ConfigureChannelOrder(p_sys->pi_chan_table, p_sys->vi.channels,
                          p_enc->fmt_in.audio.i_physical_channels, true);

    return VLC_SUCCESS;
}
Esempio n. 29
0
int OggInit(CFile *f, OggData *data)
{
	ogg_packet packet;
	int num_vorbis;
#ifdef USE_THEORA
	int num_theora;
#endif
	int stream_start;
	int ret;

	unsigned magic;
	f->read(&magic, sizeof(magic));
	if (SDL_SwapLE32(magic) != 0x5367674F) { // "OggS" in ASCII
		return -1;
	}
	f->seek(0, SEEK_SET);

	ogg_sync_init(&data->sync);

	vorbis_info_init(&data->vinfo);
	vorbis_comment_init(&data->vcomment);

#ifdef USE_THEORA
	theora_info_init(&data->tinfo);
	theora_comment_init(&data->tcomment);
#endif

#ifdef USE_THEORA
	num_theora = 0;
#endif
	num_vorbis = 0;
	stream_start = 0;
	while (!stream_start) {
		ogg_stream_state test;

		if (OggGetNextPage(&data->page, &data->sync, f)) {
			return -1;
		}

		if (!ogg_page_bos(&data->page)) {
			if (num_vorbis) {
				ogg_stream_pagein(&data->astream, &data->page);
			}
#ifdef USE_THEORA
			if (num_theora) {
				ogg_stream_pagein(&data->vstream, &data->page);
			}
#endif
			stream_start = 1;
			break;
		}

		ogg_stream_init(&test, ogg_page_serialno(&data->page));
		ogg_stream_pagein(&test, &data->page);

		// initial codec headers
		while (ogg_stream_packetout(&test, &packet) == 1) {
#ifdef USE_THEORA
			if (theora_decode_header(&data->tinfo, &data->tcomment, &packet) >= 0) {
				memcpy(&data->vstream, &test, sizeof(test));
				++num_theora;
			} else
#endif
			if (!vorbis_synthesis_headerin(&data->vinfo, &data->vcomment, &packet)) {
				memcpy(&data->astream, &test, sizeof(test));
				++num_vorbis;
			} else {
				ogg_stream_clear(&test);
			}
		}
	}

	data->audio = num_vorbis;
#ifdef USE_THEORA
	data->video = num_theora;
#endif

	// remainint codec headers
	while ((num_vorbis && num_vorbis < 3)
#ifdef USE_THEORA
	  || (num_theora && num_theora < 3) ) {
		// are we in the theora page ?
		while (num_theora && num_theora < 3 &&
		  (ret = ogg_stream_packetout(&data->vstream, &packet))) {
			if (ret < 0) {
				return -1;
			}
			if (theora_decode_header(&data->tinfo, &data->tcomment, &packet)) {
				return -1;
			}
			++num_theora;
		}
#else
	  ) {
#endif

		// are we in the vorbis page ?
		while (num_vorbis && num_vorbis < 3 && 
		  (ret = ogg_stream_packetout(&data->astream, &packet))) {
			if (ret < 0) {
				return -1;
			}
			if (vorbis_synthesis_headerin(&data->vinfo, &data->vcomment, &packet)) {
				return -1;
				
			}
			++num_vorbis;
		}

		if (OggGetNextPage(&data->page, &data->sync, f)) {
				break;
		}

		if (num_vorbis) {
			ogg_stream_pagein(&data->astream, &data->page);
		}
#ifdef USE_THEORA
		if (num_theora) {
			ogg_stream_pagein(&data->vstream, &data->page);
		}
#endif
	}

	if (num_vorbis) {
		vorbis_synthesis_init(&data->vdsp, &data->vinfo);
		vorbis_block_init(&data->vdsp, &data->vblock);
	} else {
    	vorbis_info_clear(&data->vinfo);
    	vorbis_comment_clear(&data->vcomment);
	}

#ifdef USE_THEORA
	if (num_theora) {
		theora_decode_init(&data->tstate, &data->tinfo);
		data->tstate.internal_encode = NULL;  // needed for a bug in libtheora (fixed in next release)
	} else {
    	theora_info_clear(&data->tinfo);
    	theora_comment_clear(&data->tcomment);
	}

	return !(num_vorbis || num_theora);
#else
	return !num_vorbis;
#endif
}
Esempio n. 30
0
int lame_encode_ogg_init(lame_global_flags *gfp)
{
  lame_internal_flags *gfc=gfp->internal_flags;
  char comment[MAX_COMMENT_LENGTH+1];

  
  /********** Encode setup ************/
  
  /* choose an encoding mode */
  /* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
  if (gfp->compression_ratio < 5.01) {
    memcpy(&vi2,&info_E,sizeof(vi2));
    MSGF( gfc, "Encoding with Vorbis mode info_E \n" );
  } else if (gfp->compression_ratio < 6) {
    memcpy(&vi2,&info_D,sizeof(vi2));
    MSGF( gfc, "Encoding with Vorbis mode info_D \n" );
  } else if (gfp->compression_ratio < 8) {
    memcpy(&vi2,&info_C,sizeof(vi2));
    MSGF( gfc, "Encoding with Vorbis mode info_C \n" );
  } else if (gfp->compression_ratio < 10) {
    memcpy(&vi2,&info_B,sizeof(vi2));
    MSGF( gfc, "Encoding with Vorbis mode info_B \n" );
  } else if (gfp->compression_ratio < 12) {
    memcpy(&vi2,&info_A,sizeof(vi2));
    MSGF( gfc, "Encoding with Vorbis mode info_A \n" );
  } else {
    memcpy(&vi2,&info_A,sizeof(vi2));
    MSGF( gfc, "Encoding with Vorbis mode info_A \n" );
  }

  vi2.channels = gfc->channels_out;
  vi2.rate = gfp->out_samplerate;

  
  /* add a comment */
  vorbis_comment_init(&vc2);
  vorbis_comment_add(&vc2,"Track encoded using L.A.M.E. libvorbis interface.");

  /* Add ID3-style comments to the output using (for the time being) the
     "private data members" in the "id3tag_spec" data structure. This was
     from a patch by Ralph Giles <[email protected]> */
     
#ifdef THIS_CODE_IS_NOT_BROKEN_ANYMORE     
  if(gfp->tag_spec.title) {
    strcpy(comment,"TITLE=");
    strncat(comment,gfp->tag_spec.title,MAX_COMMENT_LENGTH-strlen(comment));
    vorbis_comment_add(&vc2,comment);
  }
  if(gfp->tag_spec.artist) {
    strcpy(comment,"ARTIST=");
    strncat(comment,gfp->tag_spec.artist,MAX_COMMENT_LENGTH-strlen(comment));
    vorbis_comment_add(&vc2,comment);
  }
  if(gfp->tag_spec.album) {
    strcpy(comment,"ALBUM=");
    strncat(comment,gfp->tag_spec.album,MAX_COMMENT_LENGTH-strlen(comment));
    vorbis_comment_add(&vc2,comment);
  }
  /* pretend that the ID3 fields are equivalent to the Vorbis fields */
  if(gfp->tag_spec.year) {
    sprintf(comment, "DATE=%d", gfp->tag_spec.year);
    vorbis_comment_add(&vc2,comment);
  }
  if(gfp->tag_spec.comment) {
    strcpy(comment,"DESCRIPTION=");
    strncat(comment,gfp->tag_spec.comment,MAX_COMMENT_LENGTH-strlen(comment));
    vorbis_comment_add(&vc2,comment);
  }
  /* TODO -- support for track and genre */
#endif  

  /* set up the analysis state and auxiliary encoding storage */
  vorbis_analysis_init(&vd2,&vi2);
  vorbis_block_init(&vd2,&vb2);
  
  /* set up our packet->stream encoder */
  /* pick a random serial number; that way we can more likely build
     chained streams just by concatenation */
  srand(time(NULL));
  ogg_stream_init(&os2,rand());
  
  /* Vorbis streams begin with three headers; the initial header (with
     most of the codec setup parameters) which is mandated by the Ogg
     bitstream spec.  The second header holds any comment fields.  The
     third header holds the bitstream codebook.  We merely need to
     make the headers, then pass them to libvorbis one at a time;
     libvorbis handles the additional Ogg bitstream constraints */
  
  {
    ogg_packet header;
    ogg_packet header_comm;
    ogg_packet header_code;
    
    vorbis_analysis_headerout(&vd2,&vc2,&header,&header_comm,&header_code);
    ogg_stream_packetin(&os2,&header); /* automatically placed in its own
					 page */
    ogg_stream_packetin(&os2,&header_comm);
    ogg_stream_packetin(&os2,&header_code);
    
    /* no need to write out here.  We'll get to that in the main loop */
  }
  
  return 0;
}