search for: uninterleaving

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; } ..

data. On closer inspection (should have done that before, sorry), the procedure uninterleave that you gave is a direct translation of that code, and it also uses little endian (the high byte is at address i*4+1, the low byte at i*4, hence the little byte first, little endian). Procedure uninterleave_little_endian uninterleaves as well, but assumes that the machine it's running on is

data. On closer inspection (should have done that before, sorry), the procedure uninterleave that you gave is a direct translation of that code, and it also uses little endian (the high byte is at address i*4+1, the low byte at i*4, hence the little byte first, little endian). Procedure uninterleave_little_endian uninterleaves as well, but assumes that the machine it's running on is

...shed some light on encoding chunks of PCM data rather than files. My starting point is pretty much that of vorb_enc.dll by Andy Kay and the encoder_example.c from vorbis 1.0 distribution. From what I gather encoder setup goes through fine, but when I get a PCM chunk to encode, I can't get past uninterleaving samples, it seems to die with EXCEPTION_ACCESS_VIOLATION while uninterleaving. Any ideas? I guess what I am really asking is how should I go about doing something like an equivalent of lame_encode_buffer () or lame_encode_buffer_interleaved () in vorbis? Also, a related question. Example encoder...

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)|

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! 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 folks, I'm doing a merge of my current branch onto the mainline (for testing) today. I believe it to be stable. Just a little more vorbisfile testing. After the merge, I have a few more patches to apply, then onto cascading/coupling. New stuff: Floor backend 1 and residue backend 1; both are present, but the mainline modes won't use either yet. Naturally, both are enabled for

Hi, Please brace yourselves for some possibly stupid questions. Why are there only 2 codebooks for floors and many more for residue values? What makes the encoding of floor values fundamentally different from encoding residue values? I have a basic understanding of VQ, and looking at the floor decode function, it all makes sense 1) Read the codebook index from the stream 2) Do a

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);

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

I think I have bumped into the small file error, or I''m doing something stupid. The files are short audio effects for a game (embedded in our own data format). Sample info: Vorbis packets: 1 (4 kb) Samples: 28672 Samplerate: 22 kHz Channels: 2 This is what I''m doing when I want to get a number of bytes from the stream: The problem is that ov_pcm_tell always returns 0. And so

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