similar to: Encoding questions

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 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

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 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! 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);

(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

I was wondering. When doing ogg only encoding, how do you mark end of stream? I am setting ogg_packet.e_o_s and then submitting it via ogg_stream_packetin() and then, because its the last potential page ogg_stream_flush()ing it. But when I walk through the file, EOS is not set on that logical bitstream. I have been using oggenc.c/encoder.c and oggdec.c/decoder.c for examples, but because the

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)

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

Michael, > See the examples. If you initialise a managed mode (which this is), you > MUST use vorbis_bitrate_addblock() and vorbis_bitrate_flushpacket(). Thanks for the tip. I added the calls, and it works now. > You SHOULD also give an option to set min/max, since they're now used, > and to use a VBR mode rather than the managed modes here. But when streaming, IMHO it is

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

Hi, It's a bit offtopic for this list, but you might be able to help. For libvorbis rc3, what is the correct bitrate parametrization for the call vorbis_encode_init()? For rc2, it worked calling it the following way: vorbis_encode_init( &vorbisInfo, 2, // two channels 44100, // 44.1kHz

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I download all the tarballs for rc3. I compiled and installed all the libraries successfully, but I wan't able to compile the vorbis tools (1.0rc3) package succesfully. I can run configure successfully, but when I run make it dies with this make[2]: Entering directory `/usr/local/src/vorbis-tools-1.0rc3/oggenc' /bin/sh ../libtool --mode=link gcc -O20 -ffast-math -fsigned-char -o

I'm currently working on a speex + theora webchat. I have both working alone, but to make them work together and to know where the package goes, I'm trying to use ogg packages. With theora there has been no problem, but with speex I can't make it work. The problem is that for a second it works great, but then it begings to gain lag, and in a little time it has 5-10 lag seconds (into

Hi all, I can't compile ices on my Linux Box Here is the output ices.o(.text+0xc0): undefined reference to `log_open_file' encode.o: In function `encode_dataout': encode.o(.text+0x223): undefined reference to `vorbis_bitrate_addblock' encode.o(.text+0x244): undefined reference to `vorbis_bitrate_flushpacket' Thank you -- -- GNU/Linux: il y a moins bien mais plus cher

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

Hello, I have some code that encodes PCM data into a vorbis stream, using the workflow shown in http://www.xiph.org/vorbis/doc/libvorbis/overview.html. By analyzing that process with valgrind/cachegrind I found out that 95% of the time used for encoding is spent in the function "vorbis_analysis(&vb, 0)". My question now is: would it be technically possible to parallelize this? I

Just got around to compiling RC3 under beos and came across an anomaly when using managed bitrates. I haven't changed the code for the beos encoder but I now get double the bitrates so vorbis_encode_init(&vi,mediaFormat.u.raw_audio.channel_count,(long)mediaFormat.u.raw_audio.frame_rate , -1, 128000, -1); now gives me vorbis files that average around 325 - 350. Is this now the correct