similar to: encoding question

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; } ..

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

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)|

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

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;

(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

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 am having a problem with asterisk voice mail messages that seems to be intermittent. Though the problem occurs most of the time, on rare occasions it will work fine - rare enough that I can't pin down what it is that works. The problem is that voice mail message get played back garbled. Occasionally, I can make out moments of a voice or another sound that may be in the actual message,

Hi Developers, I have a 16-bit PCM data buffer, I want to write that to ogg file. Could you help me to understand how to write pcm data to the ogg? Thanks in advance. Arun balaji -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20150518/6c8c9a63/attachment.htm

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

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

Hello! I've got a little problem with Asterisk (11.14.1), the voicemessages are kinda limited to 40 seconds (average) aproximately; because when a message reach this long I got a cut in the file (*.wav) after I got this message: WARNING[15035][C-000021ef]: format_wav_gsm.c:418 wav_read: Short read (20) (Resource temporarily unavailable)! Does anyone got this problem, any idea of what is

sorry for the mess created. So far I have installed these two library *libogg-1.3.2, libvorbis-1.3.4* successfully. When installing this *vorbis-tools-1.4.0 *: I am getting this error. Making install in man make[2]: Entering directory `/home/arunbalaji/Downloads/vorbis-tools-1.4.0/oggenc/man' make[3]: Entering directory `/home/arunbalaji/Downloads/vorbis-tools-1.4.0/oggenc/man' make[3]:

I finally got around to committing Chris Hanson's MacOS9 patches. Just though you should know. Monty --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

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

>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

With the recent kernel patch viro@www.linux.org.uk[torvalds]|include/linux/ romfs_fs.h|20041006145144|23705 aka "romfs endianness annotations" utils/fstype.c no longer links because cpu_to_be32 is undefined. Options: 1) Change the header to be ok for userspace diff -ur ../linux-2.5/include/linux/romfs_fs.h linux/include/linux/romfs_fs.h --- ../linux-2.5/include/linux/romfs_fs.h