similar to: Wav to Ogg Vorbis conversion

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'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 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

On Oct 12, 2008, at 22:32, Paul Davis wrote: > On Sun, 2008-10-12 at 19:26 -0700, Brian Willoughby wrote: >> Is anyone here potentially up to the task of adding support for CAF >> (the CoreAudio Format) into the flac command-line? This would >> present minimal difficulty under OSX, due to the presence of the >> CoreAudio API, but the real challenge would be to support

I have a test program that loads an AIFF or WAV file with libsndfile and plays then using libao. After running around in circles for a while, I found and fixed the problem, but I'm still confused as to why the solution works. Would someone please explain this to me so I don't trip over this again later in my project? I used sf_open_fd() to get a SNDFILE* pointer and an AF_INFO

Hello, I am currently learning the FLAC C API and had the code working with FLAC__stream_decoder_init_file. However, since I'd need the Unicode filename support, I tried _wfopen_s in combination with FLAC__stream_decoder_init_FILE, however I get a runtime crash as sonn as I call FLAC__stream_decoder_process_until_end_of_stream. The same code (partially taken from the examples) is working

I managed to get around it. I used the stream functions and provided my own callbacks for reading and writing. What's strange is that what I've done is just copied the contents of read/write/seek/tell/eof callbacks from the sources to my application and it works just fine, no glitches. When I use the build-in implementation, it just crashes without any reason. It's not a problem to

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've already posted this message month ago, but nobody answered, may be it was not delivered to newsgroup?) Hello, All. I wrote an encoder-decoder based on example from OGG-Vorbis SDK. This encoder can encode a large amount of small WAV-files with equal parameters into one sound archive. Then I decode this sound archive back into large amount of small WAV-files. Theese files are

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

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

2013/2/8 Erik de Castro Lopo <mle+la at mega-nerd.com>: > Collin wrote: > >> It turns out it was an error on my part; sorry for the trouble. I misunderstood >> the format libflac was expecting my samples in. I found the wording in the >> documentation very confusing. >> >> My audio file has 2 channels and 16-bit samples, so I have to pull out each >>

In my experience, with the gcc compiler, cross-compiling is highly reliable. If your code runs on one processor, then it will run on all. Linking and such might be an issue, which you will discover immediately when the first person tries to run what you've built on their system. It still would be great to run these tests on all four. Actually, you should be able to test 32-bit

FYI have anybody seen this demo ? I have use ogg for it. http://techfandu.blogspot.com/2010/03/world-first-projective-html5-video.html Thanks -- ??????????????????????????? ? Narendra Sisodiya ( ???????? ???????? ) ? Society for Knowledge Commons ? Web : http://narendra.techfandu.org ??????????????????????????? -------------- next part -------------- An HTML attachment was scrubbed...

Hello, I have written a libsndfile (http://www.mega-nerd.com/libsndfile/) file output plugin for libao. For the moment I haven't done any changes to the core of libao. However some changes could make this more useful. Before I embark on anything like this I'd like to have your opinion on a few things. --- First, what do you prefer ? 1a. Keeping the AU/WAV/RAW plugins and adding

>I'm currently facing the same problem. >I added the libFLAC++ libraries to my MSVC application. >I implemented the same quality levels (0-8) as used in the FLAC frontend application. >But the resulting files are remarkable different between my application and the FLAC frontend >(although using the same settings). It did turn out to be something in my byte ordering in the end

On 5/2/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Well, I don't quite understand why AC_C_BIGENDIAN and the solution > you're proposing is likely to break other big endian machines (the ones > that don't have __BIG_ENDIAN__). Can you send a patch that addresses > that (i.e. still uses AC_C_BIGENDIAN when it works)? It is not that AC_C_BIGENDIAN