similar to: Crash in mdct_forward

Hi, ??Forgive my poor English, I'm trying to implement a net phone using vorbis codec, which need realtime encode and decode the stream. The problem is, I use vorbis_analysis_buffer() and vorbis_analysis_wrote() push the PCM data in, then use vorbis_analysis_blockout(&vd, &vb) to output a block, it can't output all the data. It seems that some data which length is constant, must

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hi there again.. I'm still writing my particular-case-vorbis-encoder-and-player, and it worked fine until now. I'm having a problem trying to encoding the audio. I need to send the data very quickly, as soon as i capture it from the sound card. Currently i have the following (audio captured 16 bits, mono): void VorbisEncoder::encode(void *data, int r) { char *buf = (char *)

I apologize for bugging you all again, but I've been trying to wrap my head around the encoding process and I'm about to have an aneurysm :P I've gone through oggenc.c, encode.c, and audio.c and I think I have a basic idea of what happens. I got it down in pseudocode as follows: --- 1 get parameters 2 init vorbis_comment -- vorbis_comment_init() 3 init vorbis_info --

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);

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

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 different from source files, but each of them can be normally played by Windows. And now I can't correctly encode

(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

Hey I am playing with a LOT of threads right now, and I want to know if the threads all need their own little vorbis encoders running in them or what exactly is threadsafe in vorbis? So here are the functions that would be called from many threads of with buffer = vorbis_analysis_buffer( &m_vorbisDsp, 4*DATA_CHUNK_SIZE ); vorbis_analysis_wrote( &m_vorbisDsp, dataLength/dataSize );

Hi While implementing OSS Ogg/Vorbis encoding and streaming using RTP/RTCP for both unicast and multicast, I have come across a few issues I need to clear out regarding the numbering of ogg_packets and their granulepos value. Below are the result for two different scenarios. <p>In the first scenario, ogg_packets are generated using a) vorbis_analysis_headerout() b) while

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)

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

It looks like the Mac .exp files were not updated for RC3. The following diff corrects that problem. The CodeWarrior project files also have not been updated. They are in 5.3 format which is quite old. Would anybody object if I submitted updates to them in CodeWarrior Pro 6 format? That's the oldest version I have readily available. If someone else has 5.3 and wants to do it, you need to

Per Monty's suggestions from a while ago, I have [finally] gotten around to playing with different schemes for parallel oggenc. Monty's main suggestion was to have a single thread loop over reading samples and calling vorbis_analysis_blockout(), and then queueing up the resulting blocks to be processed through vorbis_analysis() in other threads (in parallel). To verify that this works, I

Monty -- I'm still running into problems with my threaded encoded, and I think I just figured out why. I have N threads running in parallel calling vorbis_analysis_blockout() and vorbis_analysis() to do the number crunching on the input samples. They all share a single vorbis_dsp_state -- my understanding was that this was ok; they only *read* from the vorbis_dsp_state, therefore not

I noticed that Vorbis has a function called vorbis_analysis_blockout() that breaks up input data into blocks that can be processed in parallel for multi-threaded applications. Ironically, this is much less useful now that processors are so powerful that Vorbis encoding uses up very little processor load. The truly ironic part is that Theora has no analogue to this function yet it is still very

Emscripten is an LLVM-to-JavaScript compiler. It takes LLVM bitcode - which can be generated from C/C++, using llvm-gcc or clang, or any other language that can be converted into LLVM - and compiles that into JavaScript, which can be run on the web (or anywhere else JavaScript can run). I was able to successfully build libogg, libvorbis and libvorbis examples using this tool and generate valid

I got this one this morning, but didn't notice it til now: --- (gdb) bt #0 0x4003be91 in vorbis_block_clear (vb=0xbffff71c) at block.c:156 #1 0x4003ce36 in vorbis_analysis_blockout (v=0xbffff71c, vb=0x8050b80) at block.c:591 #2 0x4002ff23 in _prefetch_all_headers (vf=0xbffff540, dataoffset=-1073749296) at vorbisfile.c:324 #3 0x4003039c in _process_packet (vf=0xbffff540, readp=1)

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;