similar to: NOP pattern - how to make SPEEX packets bigger?

On Fri, Dec 17, 2004 at 09:29:13AM +0100, Tomasz Pyra wrote: > How can I add some size to SPEEX encoded packets without affecting > decoding results? > > I need it to fit smaller (due to VAD) packets in CBR acm-wav file. I'm not entirely familiar with how speex handles packet lengths, but have you tried just padding the packet out and seeing if the decoder still handles them

Please compare samples: http://www.ivo.pl/ivo2/sound/krwawa_org.wav http://www.ivo.pl/ivo2/sound/krwawa_fixed.wav http://www.ivo.pl/ivo2/sound/krwawa_float.wav krwawa_org.wav is original file. Both output files was created using same program by compressing from wav to speex, then decompressing from speex to wav (using speex_encode_int and speex_decode_int). Only difference was used speex

Hi I have write A function to draw pyramid of age. I have two function draw.pyramide(h,f,l) , pyramide(h,f,l) and a data frame with data. I first use package.skeleton("pyra") I got the package structure Then in my shell I use > RCMD build pyra I get this : check for description ... OK .... removing junk files but "cannot open

How big is difference between encoding speeds of libspeex compiled with arm-asm, and without it? My application is written in EVC++, but there is no inline assembler, so I have to compile speex without arm-asm. But I still trying to find a way to use speex with arm-asm under my EVC++ application. I tried to compile speex with arm-asm under gcc (arm-wince-pe), but generated static link

> If I understand what you said correctly, there is such a call: > speex_bits_read_whole_bytes, which adds a couple byte to the SpeexBits > struct while removing the ones that have already been read. Oh, how could I not have seen that one? That is exactly what I need. Thanks for the pointer. >> It is unusual to require a different sequence of API calls >> depending on

Hallo oggs, i googled it but still can't find any answer: how can i set default audio track? I transcode my dvds to ogg, i use gst-launch for it: gst-launch filesrc location=stream.dump ! mpegpsdemux name=demux \ oggmux name=mux ! filesink location=out.ogg \ demux.audio_80 ! a52dec mode=2 ! audioconvert ! audioresample ! audio/x-raw-float,rate=44100,channels=2 ! audiorate ! \ taginject

Listening to the meeting on granule pos tonight/today it became clear that the issues everyone is concerned with for the most part don't affect my implementations and the issues i have pretty much don't affect anyone else... and in the cases where they overlap, the reasoning seems to be different. And since everyone else has had a lot more time to consider all these issues and i'm

(Cross posted to theora-dev@xiph.org and ffmpeg-devel@mplayerhq.hu) I am working towards adding Theora encoding support to libavcodec in FFmpeg. I am doing this by simply calling libtheora from libavcodec. I am at the point where I can execute: "./ffmpeg -v 100 -i test.wmv -f avi -an -vcodec theora -b 1000000 -y test.avi" I get some whirring and an output file written. My calls to

Just in case my quotation causeed misunderstnaing... Forget about what I quoted before. Probably, by CBR, Monty meant ABR since CBR is stupidly patented and this is the reason why Vorbis has no CBR option. If you know this, it cannot be a problem but I see there is a question asking what is ABR in the mailing list. >On Thu 28 Jun 2001, 23:27:41 PDT, Monty (xiphmont@xiph.org)Wrote: >then

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

I am putting together an ogg mux/demux application which uses two logical streams. One is a vorbis stream, the other is some application specific data. This came together really quickly, following the ogg/vorbis documentation it took only a few hours to get it up and running. Just a small question... My application stream is currently running as a continuous stream but really I want it to be

Hello, I would like to "piggy-back" other data in an Ogg/Vorbis I physical stream (i.e. file) in a manner that does not conflict with specifications and existing decoders/players/editors etc. As far as I can see, I have some possible choices: 1 - Chaining a logical bistream at the end. If this is the way to go, two questions arise to begin with: q1a - How to identify this stream in a

i have been using the asterisk for some three weeks. Previously i was using the softphone iax-phone and now i have to shift to the sip phone xlite. The problem is that there's always unbearable noice in sip to sip calls. Is there any way to get rid of this???? Kindest MM Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Maybe this is documented somewhere, if so, please send me a link to the documentation. My question is: how to set up different (VBR, CBR, ABR) modes when calling the Ogg Vorbis API? Currently I do: CBR: ret = vorbis_encode_setup_managed( &vorbisInfo, getInChannel(), getOutSampleRate(),

Hi, I try to use the video tag to post videos on my blog I'm using ffmeg2theora to transcode videos in theora Resolution of 320x240 pixels and constant bitrate (CBR) of ~400 kbps seems to be ok for publishing on my blog. I've read that CBR is a better choice than VBR for web publishing regarding the way TCP/IP works. ("In TCP it takes a while for a connection to increase its

Hello folks I am encoding music to ogg vorbis format for the first time. From what i have researched, ogg vorbis is natively a vbr format. Also encoding using the -q switch is encouraged instead of specifying a bitrate through -b. Now, while encoding using -b i get this: Encoding "file.wav" to "file.ogg" at approximate bitrate 224 kbps (VBR encoding enabled) Notice

Hi guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

I'm well aware how compression works. But images and document files do not depend on the relative timing of the data to reproduce themselves. They are in essence only two-dimensional in space, whereas the data in a sound file is time-dependent. The question really has more to do with the decoded FLAC stream output, which I presume is a linear PCM file, e.g. WAV. If FLAC is lossless and

> All you say look normal to me. Okay, clarification requires less code to be written than a bug. :) > > Does that makes sense? I would have thought that --dtx alone would have = > doen it, > > but maybe it needs perfect digital silence to work? Or only uses the voi= > ce activation > > to know when there's silence? > > DTX requires the VAD to know when