Le jeu 20/02/2003 à 12:21, X-Fixer a écrit :> just to check that I've got everything right. > the encoder allows as input > - mono onlyWell, there's support for simple stereo (intensity-based) encoding too, but it's probably not necessary to take care of it now.> - 8 or 16 bits as floating point numbers (without scaling to 1.0). > floating-point wavs (IEEE) will also work, but it's better to scale > them to something like 8000.the speexenc/speexdec utility can deal with 8/16 bits, but the library itself takes floating point values with max 32000 scaling (prefered around 8000).> - any sample rate (should be set with speex_encoder_ctl), but prefered > are 8/16/32 kHz. but what modes should I use for a given rate?8 kHz is narrowband (speex_nb_mode), 16 kHz is wideband (speex_wb_mode) and 32 kHz is ultra-wideband (speex_uwb_mode)> also > - what bit rates are allowed?Too many to list here: from around 2 kbps to 44 kbps> - what's the difference between quality and complexity? complexity > does not affect decoder speed while quality does?complexity means: take more (or less) CPU to do a better job at encoding. It doesn't change the bit-rate, but it has a (small) effect on quality. Of course, the quality of the encoding is mainly controlled by the "quality" parameter (which has a direct effect on bit-rate).> also, to have a "legal" ACM codec one should register at microsoft > (get MID, PID and format tag). is this somehow possible? I mean Xiph > is a solid organization.Might want to contact Emmett Plant <emmettfish@mac.com> about that (contact me again if you don't get an answer). Jean-Marc -- Jean-Marc Valin, M.Sc.A. LABORIUS (http://www.gel.usherb.ca/laborius) Université de Sherbrooke, Québec, Canada <p> -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 242 bytes Desc: signature.asc Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20030220/9cb3934b/signature-0001.pgp
>> - any sample rate (should be set with speex_encoder_ctl), but prefered >> are 8/16/32 kHz. but what modes should I use for a given rate?JMV> 8 kHz is narrowband (speex_nb_mode), 16 kHz is wideband (speex_wb_mode) JMV> and 32 kHz is ultra-wideband (speex_uwb_mode) but are other sample rates supported? I ask because ACM does a bad job for resampling and I want to accept any possible sample rate. (and so: what's best mode for an arbitrary sample rate?)>> also >> - what bit rates are allowed?JMV> Too many to list here: from around 2 kbps to 44 kbps JMV> complexity means: take more (or less) CPU to do a better job at JMV> encoding. It doesn't change the bit-rate, but it has a (small) effect on JMV> quality. Of course, the quality of the encoding is mainly controlled by JMV> the "quality" parameter (which has a direct effect on bit-rate). user should be able to select quality for encoding and the best way for it (I think) is to specify it in kbps. so, how should I set up quality/bps (since there are controls for both) to get a proper bps?>> also, to have a "legal" ACM codec one should register at microsoft >> (get MID, PID and format tag). is this somehow possible? I mean Xiph >> is a solid organization.JMV> Might want to contact Emmett Plant <emmettfish@mac.com> about that JMV> (contact me again if you don't get an answer). ok, I'll do that when I get codec working. --- >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 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the list will be ignored/filtered.
just to check that I've got everything right. the encoder allows as input - mono only - 8 or 16 bits as floating point numbers (without scaling to 1.0). floating-point wavs (IEEE) will also work, but it's better to scale them to something like 8000. - any sample rate (should be set with speex_encoder_ctl), but prefered are 8/16/32 kHz. but what modes should I use for a given rate? also - what bit rates are allowed? - what's the difference between quality and complexity? complexity does not affect decoder speed while quality does? also, to have a "legal" ACM codec one should register at microsoft (get MID, PID and format tag). is this somehow possible? I mean Xiph is a solid organization. --- >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 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the list will be ignored/filtered.
Anyone familiar with the Internet Phone (VOIP) app called "Speak Freely"? It has been around since the mid 1990's. I was wondering if anyone would be willing or has thought about adding speex to this program. It is open source and apparently the author is giving up on it, preferring to work on a commercial voip program. Here's the site... and you can find the Visual C 6 code somewhere... http://www.speakfreely.org/ I think this would be one of the best real-world tests of the speex codec. This software doesnt use ACM or directsound api's but uses straight C code. I was thinking the speexenc/speexdec should be easy enough to add. -Dave <p>--- >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 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the list will be ignored/filtered.
> http://www.speakfreely.org/ > > I think this would be one of the best real-world tests of the speex codec. > This software doesnt use ACM or directsound api's but uses straight C code. > I was thinking the speexenc/speexdec should be easy enough to add.The last time I looked at this it was still very much old news - mostly half duplex audio, does not adhere to any signalling standards at all, and not being currently maintained. LinPhone (www.linphone.org) using the OpenSIP stack has used Speex from day one, and it's in wide use. Very real world. Speex is *already* supported in the OpenH323 stack (www.openh323.org) and in use in the ohphone, openphone, Gnomemeeting, etc. VoIP implementations that use that library stack. Very real world. :) Greg <p><p><p><p>--- >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 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the list will be ignored/filtered.