similar to: Encoding question

As an optimisation to reading Ogg encapsulated files, I've been thinking about a strategy of reading entire packets out of the stream, and sending them to the decoder. This has the advantages that: * The packet is just a linear array of bits in memory, with length known in advance. * The routine which gets bits out of the packet is then trivial and possibly inlined. * There is no possibility

Hello all, I am providing broadband service through DVB circuit. I have 4KB uplink and 32KB downlink. I want to share 32KB downlink bandwidth among 192.168.0.0/24 (20 users in this subnet i.e 192.168.0.2-192.168.0.22)subnet where each user will get 2KB to 6KB bandwidth. And from 4KB uplink bandwidth each users will get only 1KB to 2KB bandwidth.

Hi, Sorry if this a repost but I want to create the smallest Speex library possible to be put in TI's TMS320 DSP. I'm only interested in one configuration: 5.97 Kbps narrowband. What part of source code can I remove? Currently, when I compiled the version 1.1.12 libspeex.a library with the TI TMS320 and Fixed-Integer options, I get around 522Kb. I would like to reduce it to as small

--- Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > Sorry if this a repost but I want to create the > > smallest Speex library possible to be put in TI's > > TMS320 DSP. I'm only interested in one > configuration: > > 5.97 Kbps narrowband. What part of source code > can I > > remove? Currently, when I compiled the version >

Hello! First of all a question: When you make the encoder tables (ie: mode_e.h) do you use the mapping0_forward function? Because you made the encoder tables, and after this you did a minor correction in the final beta4 ... (this correction was: additional[0]=fabs(additional[0]*scale); in the mapping0_forward function) (I see this was a bug, and the modification was correct, but this correction

Hi, I am a bit confused on how code-words are derived from the codeword lengths. I will appreciate if someone can point me in the correct direction. I will take the example of an actual codebook that i found in a valid vorbis encoded file as shown below. [SK] +------Codebook [0] -------- [SK] Codebook Dimensions = 1 [SK] Codebook Entries = 8 [SK] Unordered [SK] 1, 6, 3, 7, 2, 5, 4, 7, [SK] NO

Hi, Does anybody know how codebooks are generated in OggVorbis encoder? We are porting oggorbis encoder on embedded platform for which VQ codebook memory is hugeeee to imagine. How can we reduce that? Can we do VQ with less codebooks and if yes how? If any help available? Parul Embedded Engineer Einfochips Ltd

I had lunch with an interesting guy who had gotten .com-money to record the whole bible professionally with good actors (in swedish and english, kjv). The idea was to sell custom made compilations of biblical texts on cd over internet. The company is now out of money (surprise!), but all the material is recorded (about 350 hours) and if anyone gets a good idea on what to use the material to, it

Hi, Am I correct in understanding that the codebooks are *not* adaptive during compression? I see that Packet #3 is written to the stream in the beginning of the encode process with no modification. If the codebooks are not adaptive, then why are codebooks included in the stream at all? Why not pass the mode type (A or B or C...) instead of all the mode info and let the decoder load it's

Hi, From the vorbis decoder specification, it is clear that the decoder needs to have all the codebooks before decoding can actually begin. I will appreciate if someone can clear the following questions: 1. I guess the codebooks are derived from the actual input data. Probably the encoder may be making two passes through the input. The first pass finds out the frequency of different symbols

http://www.xiph.org/ogg/vorbis/doc/draft-moffitt-vorbis-rtp-00.txt This is the Internet-Draft I'll be submitting tomorrow and hopefully presenting at the March IETF meeting. If you see anything major, let me know right away, I'll be submitting this in the morning. jack. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

alpha. (that about sums it up) Will encode and play back via an sdp file and multicast on one computer (over the net if you transfer the sdp file over by hand). rc/rtenc3 and src/rtdec3 are in the style of encoder/decoder_example ...so this means you MUST edit it them to suit your system. For example, I specify my rtenc3 to multicast just on eth1 (to avoid pissing off my cable supplier on

Hi all, I'm porting the Tremor code to an ARM platform. Unfortunately, we didn't use the low mem branch code for CPU performance consideration, and now I find the memory use in it is huge for the embedded system. After looking into the code, I found the header parse, especially, the codebook unpack part takes a lot of memory. Tremor decodes the codebooks to a cache to store

I'm in the process of re-encoding lots of albums with rc3. This isn't too hard because my computer lab at school has 24 machines which I can all get ripping/encoding in parallel. Machines are Win95. I'm currently using EAC (www.exactaudiocopy.de) to do clean rips (unfortunately a GUI interface), a cygwin port of normalize (http://www.cs.columbia.edu/~cvaill/normalize/), oggenc,

2009/6/18 Adam Rosi-Kessel <adam at rosi-kessel.org>: > Ivo Emanuel Gon?alves wrote, on 6/18/2009 6:14 AM: >> >> On 6/18/09, Conrad Parker<conrad at metadecks.org> ?wrote: >>> >>> This sounds like it needs a new tool specifically for fixing broken >>> Ogg files. >> >> I see two solutions: >> >> 1) the new vcut which I

Hi, Can someone please tell me how should I go about changing the frame size which is hardcoded to 160 for NB and WB and 320 for UWB. For NB speech(8KHz) the framesize of 160 is 20ms frame but for WB and UWB its 10ms. What are the parameters being affected by simply changing the framesize and sub-frame size in "modes.c" How to change the buffer size and how its affected. can we have a

Hello, I'm trying to write a Vorbis decoder (just as a hobby) but I've got a problem with reading the setup header. According to the Vorbis documentation the first data that should be read after the initial 'vorbis' string is an eight bit integer which when incremented by one, contains the number of codebooks in the header. My problem is that whenever I read this number it seems

I've made a block diagram of the encoder because I tried to find out, how it works http://stoffke.freeshell.65535.net/ogg/block.html Although there are specifiation docs, that give very detailed information about single aspects of the encoding (or decoding) , I'm missing documenations that give a more general overview, about how the encoder works. (Vorbis Illuminated seems a bit

Hi! What happened to this document? Was it just left dying in the IETF system? Tor --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the

Monty Montgomery wrote: > Yes. Without the first three packets (which hold all the codec > settings and all the instruction how to handle the subsequent packets) > the rest of the stream is gibberish. Vorbis can't even unpack the > bits without the codebooks packed into the third header. Curiosity man here. Is there a finite set of predetermined codebooks or is the codebook