similar to: Flushing audio

vorbis_analysis(&vblock, &ogg_pkt). Then, your ogg_pkt contains one ogg packet - that's just the raw vorbis data. ogg_pkt.packet, of length ogg_pkt.bytes. You'll need to recreate some of the framing information in ogg - but not all of it. See the vorbis-over-RTP drafts for one way to do this. > > > You might also be able to get away with it (again, this depends: you

vorbis_analysis(&vblock, &ogg_pkt). Then, your ogg_pkt contains one ogg packet - that's just the raw vorbis data. ogg_pkt.packet, of length ogg_pkt.bytes. You'll need to recreate some of the framing information in ogg - but not all of it. See the vorbis-over-RTP drafts for one way to do this. > > > You might also be able to get away with it (again, this depends: you

Hello all, I am playing around, trying to get an OggVorbis encoder implementation working on the Symbian OS. Unfortunately I am getting a crash - here is the stack vorbis_analysis_blockout _ve_envelope_search _ve_amp mdct_forward free User::Free(void *) RHeap::Free(void *) RHeap::GetAddress(const void *)const RHeap::CheckCell(const RHeap::SCell *)const Panic(TCdtPanic)

Hello all, I am playing around, trying to get an OggVorbis encoder implementation working on the Symbian OS. Unfortunately I am getting a crash - here is the stack vorbis_analysis_blockout _ve_envelope_search _ve_amp mdct_forward free User::Free(void *) RHeap::Free(void *) RHeap::GetAddress(const void *)const RHeap::CheckCell(const RHeap::SCell *)const Panic(TCdtPanic)

A variation on questions that I've asked before... In working on the parallel version of oggenc (both threaded and MPI), a profiling run shows that the function vorbis_analysis() takes up the majority of the run time. This seems to be an obvious choice for parallelization -- send each vorbis_block to a different processor, and let them call vorbis_analsis() in parallel with each other.

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

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

1.) after initializing a ogg_stream_state may I just keep calling ogg_stream_packetin with valid ogg_packet's until no more ogg_packets needs to be added? Or do I need to call ogg_stream_pageout after every ogg_stream_packetin? Iaw may I delay calling ogg_stream_pageout until there are no more packets to be added? 2.) I assume I have call ogg_stream_pageout until no more pages can be

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 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

(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, I'm using a Windows development component which uses vorbis.dll, ogg.dll, vorbisenc.dll for encoding a file. When finishing a recording, the component runs these functions... ogg_stream_clear(OggSS); vorbis_block_clear(VBlock); vorbis_dsp_clear(Vdsp); vorbis_comment_clear(VComm); vorbis_info_clear(VInfo); I'm occasionally receiving a hang in the function

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)

Dear theora developer community, currently I'm working on a simple Theora player for Windows. But the code in the player_example.c seems not to have the performance of other implementations like the Direct Show filters by illuminate. In the example player, all important things are done in one thread: decoding the next vorbis or theora packet(s) and reading from the physical stream (+ split

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 Ralph, thanks for your posting. Yes, the standard example player in the theora distribution could also do it when theora would not need so much time. I removed the frame dropping from the example because it's based on some audio stuff under Linux that isn't available on Windows. Currently I've no frame dropping handling build in because the first goal is a good raw performance of

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;

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 --