search for: marzullo

...note the -d option), my application does not respond to any request. I''ve tried to use mongrel directly (without requesting apache) and I''ve the same behavior. Do you have any hint why -d does not run correctly ? Thanks for your help Regards. +--------------------+ + Laurent Marzullo +

...---------- Pitch gain4| --------------------------------------------------------------------------- Innovation gain4| --------------------------------------------------------------------------- Innovation VQ4| --------------------------------------------------------------------------- marzullo maio <marzullo63@yahoo.it> ha scritto: Uhm i've to correct my self, now i've a better understanding: ---------------- Frame1 ---------------- Subframe1 ---------------- Subframe2 ---------------- Subframe3 ---------------- Subframe4 ---------------- that is: --------------------...

When i use VBR do i need to set VBR quality or speex quality? Is the speex quality referred to the overall quality and instead VBR quality only if i use VBR? Do these qualities override each other? Thanks. --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail -------------- next part -------------- An HTML attachment was scrubbed...

Do i always need to tell the decoder that some packets has been lost? I noticed that if i discard packets without passing NULL and advising the decoder, sometimes my system hangs and i need to reset my PDA. Since the decoder is mantaining a state, i suppose i always need to tell it a packet has been discarded through a sequence number check. Or can i just pass to speex_decode_int whatever i

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

I noticed my latency is gradually increasing in my audio loopback test application. I just hear what i say, encoding and decoding on the fly. I suppose 8000 Hz for the output device is not 8000 Hz for the input one, they differs since they could refer to different time generators (i suppose...). To solve the problem i rised the output sampling rate from 8000 to 8200 Hz, and now i'm not

Rethinking to BER in UMTS, it seems acceptable to use speex on it do you agree? ----- Messaggio originale ----- Da: marzullo maio <marzullo63@yahoo.it> A: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> Cc: speex-dev@xiph.org Inviato: Sabato 5 maggio 2007, 23:06:40 Oggetto: Re: [Speex-dev] AMR vs Speex on wireless networks. NB for CSD/HSCSD and WB for UMTS/HSDPA. The UMTS/HSDPA services already provides an...

...ing" has to be inserted (zeros, clicks, the content of the kernel memory, ...) instead. As for crashing, it shouldn't happen and I wouldn't be surprised if it had more to do with a soundcard underrun occuring from the fact you're not replacing the packets by anything. Jean-Marc marzullo maio a ?crit : > Do i always need to tell the decoder that some packets has been lost? > I noticed that if i discard packets without passing NULL and advising the decoder, sometimes my system hangs and i need to reset my PDA. > > Since the decoder is mantaining a state, i suppose i alw...

...instance, even though that's a huge problem for VoIP. Ultimately you need to be prepared to resample to one of the de-facto 'standard' rates if timing is important. -- john -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org]On Behalf Of marzullo maio Sent: Tuesday, June 05, 2007 10:06 AM To: speex-dev@xiph.org Subject: [Speex-dev] Output sampling rate slightly increased. Will speexcomplain? I noticed my latency is gradually increasing in my audio loopback test application. I just hear what i say, encoding and decoding on the fly....

My % of packets lost vary usually in function of signal strenght. Since i cannot forecast the % of frames i will lost except with the criteria just said, could be a good approach to set the PLC tuning % dynamically during runtime basing it statistically on the number of ACKed frames? Does the % of frames lost include silence frames (6 bytes frames, i'm using VBR)? So do i need to consider

Does the PLC tuning affect only the quality in case of PL? Does it affect the quality also in good PL conditions? Thanks. --------------------------------- --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've not so much bandwidth to add channel coding but i've few bytes to spare for error detection. With a polynomial CRC-8 i've already more than 99.6 % to detect a burst error in a frame. So let's suppose the critical fields of Speex are: Wideband bit Mode ID LSP OL pitch OL pitch gain OL Exc gain So my plans are to calculate a CRC on these fields for each frame i send, then if i

I would like to know if my vision of things is correct about frame structure. -------------------------------------------------------- Frame header -------------------------------------------------------- Subframe1 header|Subframe1 content| -------------------------------------------------------- Subframe2 header|Subframe2 content| --------------------------------------------------------

Could you please explain what features has been already ported to FIXED_POINT and which others aren't yet? Please add some details on how to understand by looking inside code if a feature is alredy ported or not. For example, where i see float declaration or wherever i see you're using FPU doing float multiplications/divisions etc, this without any alternative code with FIXED_POINT

NB for CSD/HSCSD and WB for UMTS/HSDPA. The UMTS/HSDPA services already provides an excellent turbo code correction. For what i know i can tell you more about corrupted packets. The BER ranges from 10^-5 to 10^-2 for CSD/HSCDS in transparent mode, and from 10^-9 to 10^-6 for UMTS/HSDPA. It depends on signal quality and if you're walking/standing or if you're in real mobility (for example

Can i use a separate instance of the decoder passing NULL to generate white noise or something that is similar to confort noise? I need to feed it to my output audio autput device for each frame i don't receive in time. Or should i just use the same decoder instance of voice and just pass NULL when i discard or loss frames? Is that the best way? How can i accomplish this purpose? Thanks.