similar to: Test vectors for encoder

We are working on OggVorbis encoder. In the porting effort we are trying to convert it to fixed point code (both 32 and 24 bit fixed point). Now the issue is how we do the testing. What should be the criteria for our testing. Does anybody has any idea how compliance testing (i.e. some objective tests) is done at encoder side? What is the criteria of testing at the encoder side? If anybody

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

We are implementing oggvorbis decoder on embedded system. In that we want to have support of chaining and grouping. If somebody can throw some light on how it can be implemented, it would be gr8 help. Regards, Parul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20060523/d90e36a0/attachment.htm

Hi, We are working on Oggvorbis encoder porting on embedded platform. Where can we find good test vectors (wav files) for testing OggVorbis encoder? Does anybody have any idea. Parul Embedded Engineer Einfochips Ltd

Hi, Have anybody worked on converting OggVorbis encoder floating point code to fixed point (32 bit or 24 bit). How do you handle _vp_noisemask in that case where precision goes upto 40 bits. Any help available? Parul Embedded Engineer Einfochips Ltd

Hello to All Members of this Group, Can I get the Test Vectors in .wav format for different sampling rates in order to Test the Ogg Vorbis Encoder Code ? If possible Forward the Link where I can get effective testvectors to Test the Encoder. Please Reply As soon as possible . Thanks & Regards , Maulik Desai Embedded Engineer - Embedded Division eInfochips Ltd. Work:

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

hi,All, I want to know whether Opus has AEC features like Speex? Thanks 2012/10/17 <opus-request at xiph.org> > Send opus mailing list submissions to > opus at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.xiph.org/mailman/listinfo/opus > or, via email, send a message with subject or body 'help' to

You asked in the Vorbis list, but your text only mentions OGG. The codec commonly used in OGG containers that is limited to 48 khz is Opus. Maybe you are trying to use the wrong codec (i.e. Opus instead of Vorbis)? Using a 44.1 khz wav file, I was able to encode a 192 khz ogg-vorbis file with the following command: $ oggenc --resample 192000 input.wav Of course, if your original material is

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Hi , I have read that it is posible to encode higher sample rates like 96 khz or 192khz? and the output is 48 khz, the resample is internally.? http://wiki.xiph.org/OpusFAQ But it is possible to encode? 44.1khz. It is resampled to 48khz or I have to make the resample by myself and then encode it with opus. thnx, arctor -------------- next part -------------- An HTML attachment was scrubbed...

Dear Nikos, thanks! But you use Opus only for resampling, not for entirely replacing Speex, don't you? Greetings! Hermie Am 07.05.2013 22:53, schrieb Nikos Chantziaras: > The Opus resampler is actually a bugfixed version of the Speex one. Same > interface/API, but with the bugs removed. It's why I recommended it :-) > Otherwise I would have recommended something entirely

Hello! I would like to know if it is possible to request encoding to a higher samplerate than the original data. For example my uncompressed data was 22 kHz, and I would now like to decode the compressed data to 48 kHz (to feed it to a filter which expects a high "input" format). Is this possible with Speex? Thank you. Hermie

Dear Nikos, thank you very much for your advice! I am not a little bit new to C++, and experiments (like using another third party library) always turn out to be days of work for me. That is why I would first like to try Speex's new internal resampling function. I am currently still using the old 1.0.4 version. I have now downloaded the new 1.2rc1 version. Unfortunately it does not want

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

One thing that has always bothered me about the ogg format is the distortion of high frequency sounds - even at data rates as high 128 and 160 kbps. I find the best way around this is to resample the wav file to 48 kHz (using SoundForge 6.0) before encoding (using CDex) to ogg. It takes a while, and adds a lot of extra wear and tear on my drive, but what a difference! The result is an 80k ogg file

On Aug 07 10:08:43, petrparizek2000 at yahoo.com wrote: > > What sine sweep exactly? > > An exponential sweep. It started slightly below 24 Hz and ended almost at 24 > kHz. And it was 50 seconds long. > > > How did you obtain it, > > I used Angelo Farina's "Aurora" modules. One of them is called "Generate > sine sweep". Can you please

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to

On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > with Opusenc and then decoded the resulting file with Opusdec. What sine sweep exactly? How did you obtain it, and how exactly did you encode and decode it? Jan > The strange > thing was that even though the output wave file was at 48 kHz, it