search for: uninterleaved

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

Hi everyone, I was wondering if someone could 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

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

...ctral envelope. Advantages: more stable local spectral behavior (no more spreading stereo image at lower bitrates), more control over tuning specific error characteristics. Disadvantages: it takes more bits than LSP, but not by an absurd margin (20%-30% premiun). Residue 1: Same as residue 0, but uninterleaved. The advantages are that this is faster and allows one to more easily exploit adjacent-line structure in the MDCT, assuming that that's what the codebook model wants to do. Vorbisfile optimizations; I took HB's optimization patch, fixed a few infinite loop problems in it, and then decided...

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