similar to: Opus Quality vs Bit-rate result request

Hi I was just looking at your Opus quality vs bit rate graph.? Latency looks good.? ? Two questions: 1/ Wideband: How did you address channel separation (i.e. What is the interaural time difference (ITD) granularity/resolution in microseconds? ?- assuming two channels. ) ? 2/ Narrowband: I see AMR-NB.? How does Opus compare with modern speach coding and band excitation models? I.e.

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

Hi Jean-Marc, I thought at quality 3 (wideband) - wb_submode1 that the 4-8k band was not using a codebook table. From the code I can see that some sort of "lsp" encoding is performed. What exactly does this encode? (I assume lsp means line-spectral pairs) The reason I am asking is I'm comparing the "effective" spectral bandwidth of Speex against the

Hi Any idea what's causing these errors:- /home/user/build/lib/libopencore-amrnb.a(preemph.o): In function `preemphasis': /home/user/build/opencore-amr-0.1.3/amrnb/../opencore/codecs_v2/audio/gsm_amr/amr_nb/dec/src/preemph.cpp:193: multiple definition of `preemphasis' /home/user/build/lib/libopus.a(celt_encoder.o):/home/user/build/opus/celt/celt_encoder.c:455: first defined here

It appears that both Opus and amrnb have a function called preemphasis(). In the short term, your options are: 1) Not building statically 2) Renaming that function in the Opus code 3) Renaming that function in the amrnb code I can do the rename for the next version of libopus. Cheers, Jean-Marc On 11/13/2013 05:51 PM, batguano999 wrote: > > Hi > Any idea what's causing these

Dear list, I need to convert this data.frame > names(codesM) [1] "key" "AMR.pa1.M" "AMR.pa2.M" "AMR.pa3.M" "AMR.pa4.M" [6] "AMR.pa5.M" "AMR.pa6.M" "AMR.pa7.M" "AMR.pa8.M" "AMR.pa9.M" [11] "AMR.pa10.M" "AMR.ta1.M" "AMR.ta2.M" "AMR.ta3.M"

hello: I have code for upload image file to server file system.This works in rails 1.2.3.But in rails 2.1,it gets an error. The following is the important code: In a view: <%form_for :user,:url=>{:action=>"register"},:html=>{:multipart=>true} do |f|-%> <p> <label>照片</label> <%=f.file_field "file"-%> </p>

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

Hi All, I would like to configure AMR codec in asterisk 1.8.9.1. After lots of research i found " http://sourceforge.net/projects/asterisk-amr/files/" thie link, and follow steps to configure amr. codec_amr.so successfully compiled and load in asterisk. *> core show translation * Translation times between formats (in microseconds) for one second of data Source

Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and "b" A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see

Hi all, supposing to have speex working at 3.95, when VAD/CNG is employed, which is the size of those "silence" samples respect to voice samples? I am wondering if it could be possible to lower even the transmission of VAD/CNG samples by specifying "outbound" in the communication channel only 1 bit instead of sending the whole VAD/CNG packet. I am doing the same for AMR

Hi, I'm not a member of the list, but I think I've run across a (minor) bug. The rsync web page directed me to this list for bug reports. Anyway, I've described it below (including a simple test case with exact commands to reproduce and the output I get from those commands). I hope this helps. Let me know if any more information is needed. Elijah P.S. If it isn't a bug,

Hi All, First of all, I'm sorry if my question is offtopic on this list. In such case please ignore this post and/or contact me directly. I'm asking my questions there because I feel you had similar problem before starting developing Speex. My story: my friend developed 3gpp content creator and he would distribute it in binary form. But there is problem with AMR licensing (the terms

you can use the marker bit on the RTP. - farhan On Tue, Aug 19, 2008 at 12:04 PM, Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote: > Hi all, > > supposing to have speex working at 3.95, when VAD/CNG is employed, which > is the size of those "silence" samples respect to voice samples? > > I am wondering if it could be possible to lower even the

In order to develop a Voip application, today i should make it robust to bit-errors over wireless networks. This is an actually lack of Speex, infact what i've understood is that if a packet arrives corrupted, i must pass NULL to the decoder in order to let it know. My target is to use UDP (with checksum field disabled) and exploit also corrupted packets giving them to the AMR codec. Otherwise

Hello, I'm trying to upgrade from 5-stable to 6-stable; after rebooting to the new kernel, the amr(4) module (amr.ko) refuses to load with this message: link_elf: symbol cam_simq_alloc undefined However, a quick nm(1) on /boot/kernel/cam.ko does show: ... 0000f0ac T cam_sim_set_path 0000efc4 T cam_simq_alloc 0000efd4 T cam_simq_free ... and cam.ko is already loaded before amr.ko.

Fabio Pietrosanti (naif) wrote: > i am not criticizing the technology itself, but the fact that from the > point of view of the implementator that need to make a technological > choice while selecting a codec to be used for a mobile multimedia > application, speex does not provide affordable informations on > "supported platform".

Just noticed that packman has precompiled versions of amr codec. Both wideband and narrowband. Can these be used for asterisk? Heard some nice about AMR (in general) If so, any one around with experience with either?? hw