similar to: Output sampling rate slightly increased. Will speex complain?

It is a good idea not to use sample rates other than 44100 or 48000 Hz for your final audio I/O. The chipset people just do not give a crap about rates other than that. They don't see a problem with giving you 11100 Hz when you ask for 11025, for 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'

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

Hello, I''ve install ror/mongrel etc, etc on a linux server (rhes). As i''m not root, all package startinf from ruby have been install into ~/gconf/usr ruby-1.8.6 and all of this gem package: -r-xr--r-- 1 dimnce dba 168448 Aug 9 14:53 rails-1.2.3.gem -r-xr--r-- 1 dimnce dba 84480 Aug 9 14:56 rake-0.7.3.gem -r-xr--r-- 1 dimnce dba 217088

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| --------------------------------------------------------

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

Ops i meant: --------------------------------------------------------------------------- Wideband bit|Mode ID|LSP|OL pitch|OL Exc gain| --------------------------------------------------------------------------- Fine pitch1| --------------------------------------------------------------------------- Pitch gain1| ---------------------------------------------------------------------------

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

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

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

I have a video conference like application that I've been working on for a while now, and a recent change is causing some odd problems, and I was wondering if anyone else had seen problems like this. The issue I'm seeing is that when using the sound card for capture, the audio will eventually get about 1-2 seconds out of synch (delayed), from the video. However, if I use USB devices

Hello to everybody. I was wondering... is there a way to force wine to use always highest possible refrash rate - when I ran a game in Fullscreen? I ask because - for excample my good old Incubation works flawless on wine (testet with wine 0.9.27 and 0.9.28) but when I switch to fullscreen(640x480 for this game) the refresh rate is only at 60 HZ - and my eyes hurts after a while... (mty monitor

It appears that both the echo canceller and the noise reducer (NR) introduce rather severe clicking artifacts when presented with audio that has slightly overloaded the A/D converters. I am talking about speech that sounds just slightly distorted, due to clipping, when simply played back. If I pass that speech through the echo canceller or the noise reducer, it acquires really loud clicks and/or

You need to let the know packets are missing so it can generate something meaningful instead. Otherwise, you'll also have less packets to play, which means "something" 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

I have a Gigabyte GeForce 1050 connected by DVI to a monitor (1920x1200 resolution @ 59.9502 Hz) and a TV via HDMI (3840x2160 @30 Hz). The problem is the TV used to work at 59.9685 Hz but then it started showing "No signal" on the TV. I was changing settings trying to get it to work again and I happened to change it to 30Hz and it worked. The specs are here:

Hi All, I am trying to get speex working on the Mac and am running into issues. I got the examples working, but am now trying to make speex, which expects 8000 Hz and 160 samples per buffer (320 bytes per buffer), work with the Mac's built-in audio recording, which uses either 11025, 22050, or 44100 Hz and 1024 samples per buffer (2048 bytes per buffer). I just need to know if

John, Thanks for the reply! You mentioned output sample rates should be 44100 or 48000, should I worry about input (Mic) Sample rates as well? (Currently I was requesting the sample rate on both ends to be 16000 samplesPerSecond, for ease of passing into the codec) Also, do you recommend any particular resampler that I should use, or are any of the ones out there probably okay, or should