similar to: Speex and Builder

> 1) May I know how Speex compared with GIPS codec? It seems that Google, > Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking > or fair comparisons? I think these two emails sum up my opinion about Speex vs. iLBC: http://lists.xiph.org/pipermail/speex-dev/2005-June/003410.html http://lists.xiph.org/pipermail/speex-dev/2005-September/003652.html > 2) In

Hi, We are planning to use Speex as the speech codec for a VoIP application. 1) May I know how Speex compared with GIPS codec? It seems that Google, Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking or fair comparisons? 2) In particular, how is the jitter buffer control for Speex in response to intermitent poor connection hiccups? Is it robust enough to smooth out

> Obviously this is Jean-Marc's decision and I'm not telling > him not to support this compiler. I am however pointing > out that this compiler is yet more work for very little > payoff. In the case of my project, the proponent of C++ > Builder sent me a huge, monsterously ugly and totally > unmaintainable patch to add C++ Builder support. Needless > to say, that

On 06/22/2011 09:30 AM, Steve Kann wrote: > Speaking of AEC (thought not quite on topic for this thread), > > Has anyone on this list played with the GIPS code that google just > open-sourced? It looks like their AEC also has code to handle > differential sample rates, though I haven't really evaluated it > thoroughly. > > There is really a lot of code in the drop ?

>> Err, unless I'm totally wrong, there are a few race conditions. >> >> Assume the buffer is full of packets newer than the current pointer, and >> one that is at the current pointer. >> >> get and put start at the same time. >> >> get will find the correct buffer index. Now, just after it finds it's >> index, assume we switch to the

Speaking of AEC (thought not quite on topic for this thread), Has anyone on this list played with the GIPS code that google just open-sourced? It looks like their AEC also has code to handle differential sample rates, though I haven't really evaluated it thoroughly. There is really a lot of code in the drop ? basically all of the GIPS DSP stuff (AGC, VAD, Denoise, echo canceller, etc),

Hi, I'm a developer currently using the speex codec in a VOIP application of ours.. It sounds amazing, especially at wideband.. my question is how do I force it to do a concealment? We have a low latency application, and based on the current API, I'm guessing concealment only kicks in when a packet is lost.. However, our jitter buffer knows when a packet is missing, and I'd like to

(Cross posted to speex-dev from iglance) Enzo -- I haven't tried the fixed point engine, though I've considered it for the WinCE port. For the desktop/laptop edition I'm assuming the slight short->float conversion cost will be made up by the improved performance of the floating point implementation. But I could be wrong: 1) Can anyone recommend whether Speex performs better

Hello Jean-Marc, thank you for your reply. > > I found the cause of the problem. The function shift_timings can > > produce overflows in the timing array if the jitter is huge or the > > time units are very short. After changing the timing values' type from > > spx_int16_t to spx_int32_t it seems to work. > Hmm, I always assumed there wouldn't be any overflows.

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Then why ONE sound card have different capture and playback rate? It must be ONE single physical clock generator which is used by both ADC and DAC in the sound card, isn't it? If you are a hardware engineer. Will you design two different physical clock for ADC and DAC seperately? What on earth causes this problem? Who knows its intrinsic real reason? Isn't there any other solutions? For

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

Hi Does anyone know where one can get hold of a G729 codec for asterisk which effectively can do packet loss concealment using Steve Kann's wonderful new Jitter buffer. The 2 versions that I know of, (digium's and the IPP one) do not perform great at PLC, especially with 4 or 5% loss. Thanks in advance. Clive

Daniel Schmidt a ?crit : > I found the cause of the problem. The function shift_timings can > produce overflows in the timing array if the jitter is huge or the > time units are very short. After changing the timing values' type from > spx_int16_t to spx_int32_t it seems to work. Hmm, I always assumed there wouldn't be any overflows. What parameter range are you using that

> Ok, what you suggest sound logical to me. Currently, I > have done a small trick to prevent this problem. What > I did is that whenever windows request a voice packet > from me and if I do not have the voice packet, I > repeat the previous packet. Hence, all the breaking > portion is filled with previous packet. This trick > seems to work so far. I am not sure what is the

Hi all, I am developing a VoIP application and have used SPEEX for that purpose; I successfully managed to port the codec in fixed-point to an ADSP-21364 processor, and the codec works fine in narrowband, wideband and ultra-wideband mode. Currently, I have chosen, in accordance with the speex manual, a 20 ms frame size in each of these modes: in other words, the input buffer for speex was set to

Steve, The main problem I am having with the system is clipping off the start of someone's speech when they first start talking- the ends of the sentences seem to be handled properly. I am wondering whether this is the fault of the audio playback system or whether this is a speex issue- I also get the musical artifacts problem with the denoiser. This seems to be more of a problem on open