similar to: encode, decode and encode again

Dear experts, I am trying to use an example I found here: http://svn.xiph.org/trunk/vorbis/examples/encoder_example.c Could you please tell me what float** buffer is needed for (I don't see it used anywhere). Also, could you please tell me what this code is doing: for(i=0;i<bytes/4;i++){ buffer[0][i]=((readbuffer[i*4+1]<<8)|

I'm using seekable_stream_decoder, And., this is my write_callback. I'm not getting the required output. The PCM i get is not the proper music. Am I doing something wrong here? FLAC__StreamDecoderWriteStatus AFLACStreamPlayer::StreamWriteCb ( const FLAC__SeekableStreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data) { int Channels,

I've been trying to piece my way through the encoder_example.c program to better understand how to encode files as ogg/vorbis. I'm stuck on two sections of the code. This is the first /* uninterleave samples */ for(i=0;i<bytes/4;i++){ buffer[0][i]=((readbuffer[i*4+1]<<8)| (0x00ff&(int)readbuffer[i*4]))/32768.f;

Hi there, I've a little question abound programming an ogg-vorbis encoder. The encode_example.c works ok for files with 44100/2/16 - but not for other formats. I don"t really understand what this piece of code is really doing: /* uninterleave samples */ for(i=0;i<bytes/4;i++){ buffer[0][i]=((readbuffer[i*4+1]<<8)|

Hi, I am testing the encoder example and I have a question about it... How can I encode a mono file in 22050 and 44100? I've changed the init values to: ret=vorbis_encode_init_vbr(&vi,1,22050,.5) ; and in the loop: // uninterleave samples for(i=0;i<bytes/2;i++) { buffer[0][i]=((readbuffer[i*2+1]<<8)|(0x00ff&(int)readbuffer[i*2]))/32768.f; } ..

Hi! I'd like to encode from a 22050 Hz source wav file (only in that case), but i get the following error message: "Invalid floating point operation" I use delphi: vorbis_encode_init_vbr(vi,2,22050,0.5) . . . While (i < readCount div 4) Do Begin buffer[0][i] := smallInt((pArray(@readbuffer)[i shl 2 + 1] shl 8) or pArray(@readbuffer)[i shl 2 + 0]) / 32768;

Hi! I have a problem. I hope, you can help me. I use a Delphi conversion (from Aleksandr Shamray), but it doesn't work when I'd like to convert a *.RAW into a mono *.ogg file. vorbis_encode_init_vbr(vi, 1, 44100, 0.5); //because of the mono the program stops at line: //* uninterleave samples */ . . buffer[1][i] := smallInt((pArray(@readbuffer)[i shl 2 + 3] shl 8) or

Hi, While testing ReplayGain (so it could be related to a bug), I noticed the following gain comments for a file: RG_PEAK=1.71580 RG_RADIO=-7.91 dB RG_AUDIOPHILE=-6.72 dB I've never seen such a large peak. Not that I've looked much, nor have I analyzed the file further. I just thought I should mention it. :) The file is the track Board Burner by Mixmaster Mike, available at:

Hi all, I am just coming in the mailing list.... ;-) I'm working on an Ripper/Encoder for GNUstep. At first sight, libvorbisogg seems to be the library that I needed. The problem for me is documentation. So I have a few question about this library : - in order to init an encoding process, we need to provide channels. Well, what does is mean ? - in order to init an encoding process, we need

Hi! I've made a vorbis encoder in delphi, but it's very verey slow! It takes 5 minutes to encode a pcm file. And i don't know why. Could anyone help me??? ---------------------------------------------------------------------------------- result := ogg_stream_flush(os, og); while result <> 0 do begin OutFile.Write(Pointer(og.header)^, og.header_len);

Hi, I made backup-copies of several CDs. First I used eac (exactaudiocopy) for exact ripping to a single wav-file with cue sheet. Then I used flac --cuesheet=<cuefile> <wavfile> to get a flac file with internal cue sheet. I am on a linux system. eac is running in a vmware-windows (actually, eac is the only reason that I have a windows version running). At this point I have two

I'm working on an application where I need to record the current playing position and return to it later. and I need this to be done the most efficient way, so I used ov_raw_tell() and ov_raw_seek() because the documentation says they are the best when speed is a concern. but the problem is that sometimes ov_raw_tell returns the same value before and after calling ov_read; here's an

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

Jan, Here's a hacky patch to add a few commandline options for setting comment header fields in ffmpeg2theora. It's a bit big because I virtualized the global info struct in theorautils.c. In retrospect that probably wasn't necessary, but I think it's cleaner anyway. I didn't test it because I couldn't compile ffmpeg2theora, but modulo bugs it should support

I can't be totally sure what 'short' is on your platform, but if it's (probably) 16-bits, it looks like you're treating the samples in buffer[] as packed 16 bit numbers. but all samples in buffer[] are 32-bit signed integers in host order, regardless of the bits-per-sample of the frame. so to get them down to shorts (assuming they'll fit), do like: FLAC__int32 *

Here's a patch for the current CVS version of libao which makes the IRIX plug-in compile and work properly. Michael --- ./src/plugins/irix/ao_irix.c Fri Aug 3 19:56:17 2001 +++ ../libao-0.8.2/./src/plugins/irix/ao_irix.c Thu Apr 11 20:16:41 2002 @@ -36,12 +36,13 @@ #include <ao/ao.h> +#define AO_IRIX_BUFFER_SIZE 32768 typedef struct ao_irix_internal { - static ALport alport =

Hello, I am trying to implement a realtime encoding then streaming solution using the vorbis codec (would be wrapped in a "7F" type for minimal framing overhead) for sending audio over a low bit rate wireless link. since we need a constant bittrate (i.e. constant packet size) for this solution we want to run the alg in constant bitrate mode; for evaluation of the codec I followed the

... in order to prevent a division by zero (CVE-2014-9638) and integer overflow (CVE-2014-9639). Bug: https://trac.xiph.org/ticket/2136 Bug: https://trac.xiph.org/ticket/2137 --- oggenc/audio.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/oggenc/audio.c b/oggenc/audio.c index 477da8c..1167f1b 100644 --- a/oggenc/audio.c +++ b/oggenc/audio.c @@ -13,6

The Wine development release 1.7.14 is now available. What's new in this release (see below for details): - More Task Scheduler support. - Improvements for AVI encoding support. - More VisualBasic interfaces in MSXML. - Support for deflate content encoding in Wininet. - Some fixes for monochrome printers. - Various bug fixes. The source is available from the following locations:

Hi. I'm trying to implement Ogg/Vorbis support for the MPEG4IP project. The goal is to support Ogg/Vorbis audio for MPEG-4 streaming. So far I have managed to make the encoder save Ogg packets as an Audio object in an .mp4 file. As a side effect, it can also save Ogg pages in an .ogg file playable by xmms, but that's no big deal. So what I'm doing is this. First the init part. a)