search for: 32khz

...AM Subject: Re: [Speex-dev] SPEEX on iPhone ? > 2008/11/13 Vincent Burel <vincent.burel at vb-audio.com>: > > could you explain why 44.1kHz is *not* recommended with SPEEX ? > > my understanding is that the speex modes are tuned for particular > sample rates, so using the 32kHz mode with 44.1kHz data will not yield > as good quality as using the 32kHz mode with 32kHz data. ok, it's strange, by nature, such compression algorithm are able to make down or upsampling very well... Do there is an impact on CPU load ? Vincent

...te: > >> 2008/11/13 Vincent Burel <vincent.burel at vb-audio.com>: > >> > could you explain why 44.1kHz is *not* recommended with SPEEX ? > >> > >> my understanding is that the speex modes are tuned for particular > >> sample rates, so using the 32kHz mode with 44.1kHz data will not yield > >> as good quality as using the 32kHz mode with 32kHz data. > > > > ok, it's strange, by nature, such compression algorithm are able to make > > down or upsampling very well... > > Do there is an impact on CPU load ? >...

hi on my device i can sample only at 32khz and want to use speex at 16khz so i need to down-convert the input signal by factor of 2. does anyone provide me a reference to some code that does that? are there any trick to do that? i tried to add to subsequent sample but the result was very bad. what are the requrment from a decimation filter...

thanks for your reply, could you explain why 44.1kHz is *not* recommended with SPEEX ? Vincent ----- Original Message ----- From: "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Ashhar Farhan" <farhan at phonestack.com>; <speex-dev at xiph.org> Sent: Wednesday, November 12, 2008

...} else otherByte = true } } Basically you need to write every other byte in thePCM sample stream. Clifton Craig Software Engineer http://codeforfun.wordpress.com clifton.craig at gmail.com On Jun 26, 2010, at 9:50 AM, nir elkayam wrote: > hi > > on my device i can sample only at 32khz and want to use speex at 16khz so i need to down-convert the input signal by factor of 2. > does anyone provide me a reference to some code that does that? are there any trick to do that? > > i tried to add to subsequent sample but the result was very bad. > what are the requrment from...

Hi list, I am researcher in VoIP Applications and my challenge now is convert one RTP data frame that is in 32KHz to other RTP data frame in 32KHz. Do someone help me about it? Very thanks, Thiago. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090316/fbbaf566/attachment.htm

My wish for 1.1 is to stream my station with VBR at bitrates averaging 28-32kb/s with a minimum 32khz sample rate. Hopefully this will be achievable with the new low bitrate tuning that is planned. This could put it on a par with Real Audio at 32kb/s. Currently a quality of -1, 32khz sample rate produces an average around 34-38kb/s which is too high for modems. I could currently use a managed b...

I've been playing with speex for use in a VoIP application between PC's. One thing I've found (correlating to the documentation) it that speex runs much faster and produced much better output when it's fed a 32kHz signal instead of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate and feed it 44.1kHz or tell it 32kHz and feed it 44.1kHz. What part of the algorithm is optimized for 32kHz (my initial guess was the codebooks and/or the codebook search) - would it be possible to modify it t...

...;s because they use knowledge > > that speech pitch (and other params) lies in well specified regions. > > Thus if you feed algorithm with wrong freq, you'll fool it and it might > > even not detect speech at all. > > Do you mean that speex is working at 44.1 Khz like at 32Khz without > samplerate conversion ? > > It seems you suggest that working with 44.1 Khz signal instead of 32Khz > (wich is the native samplingrate for the ultra -wide-band mode as far as i > understand) ask for more CPU load. Since YOU have already worked with > SPEEX, > do you k...

...data sampled from microphone device, data sampled from output of soundcard, output pointer, y-buffer) .. which as I understand it should be correct? I've gone to some effort to make sure that the two sets of sampled data are from the exact same time (seems my two soundcards don't sample 32khz at the exact same rate, they differ by about 0.1%), but it doesn't remove any of the echo, it just provides some weird artifacts in addition to the original signal. I'm using a framesize of 640, the same as the speex framesize at 32khz uwb, and tested this with speex 1.1.8 If anyone h...

Le ven 10/01/2003 à 14:39, John Hayes a écrit : > I've been playing with speex for use in a VoIP application between PC's. One > thing I've found (correlating to the documentation) it that speex runs much > faster and produced much better output when it's fed a 32kHz signal instead > of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate > and feed it 44.1kHz or tell it 32kHz and feed it 44.1kHz. First, note that the sampling rate you *tell* it is only used for playing back at the right speed and doesn't influence the algorithm (ye...

Hello Mailing List, I am a Speex supporter and user that would really like to know how much memory Speex uses to decode a 8kHz, 16kHz and 32kHz (primarily the 8kHz) and is it possible to use a 1kBytes of RAM to decode a 8kHz stream? (I was thinking of the possibility of using a ATmega168 to decode Speex) //P?r, Sweden

...l at vb-audio.com> wrote: >> 2008/11/13 Vincent Burel <vincent.burel at vb-audio.com>: >> > could you explain why 44.1kHz is *not* recommended with SPEEX ? >> >> my understanding is that the speex modes are tuned for particular >> sample rates, so using the 32kHz mode with 44.1kHz data will not yield >> as good quality as using the 32kHz mode with 32kHz data. > > ok, it's strange, by nature, such compression algorithm are able to make > down or upsampling very well... > Do there is an impact on CPU load ? Nothing strange if you ever w...

Hi. I am having a strange problem where the source times out during anicecast2 stream -- I was using mp3pro at 32khz and even some other speeds. I have set the source timeout for 1 minute, but it still times out or says connection reset by peer. Now the strange thing is that I also had a windows media stream on the same source machine going to a wm server and it had no problems! So can someone tell me am I miss...

Kirk, Speex was designed for 8kHz, 16kHz, and 32kHz sample rates. If you don't use one of these sample rates, you'll be messing up important assumptions deep within the codec. Why these sample rates? It's telecommunications tradition, rather than PC audio tradition. If you want an efficient and high quality format for voice chat,...

...a 32 or even 24kHz sample rate from speech at 48kHz, much less hear a large quality difference? A number of audiobooks I've listened to have used 22kHz mp3s without being clearly objectionable, and in my personal use I've had decent results using the -voice LAME setting (downsamples to 32kHz and encodes as 56kbps abr). The recent hydrogenaudio tests showed Opus CELT modes trumping the best of breed high-latency codecs at 64kbps despite having only 22.5 ms latency, and the SILK modes do a great job at the opposite of the bitrate spectrum and can make use of larger frame sizes for t...

...odecs format: Speex UltraWideBand and Speex NarrowBand. When I use Speex NarrowBand the Asterisk is able to convert this frame to G711. But when I use Speex UltraWideBand the Asterisk don't convert it. Then I need in my Asterisk Channel Source include the Speex Library in resample this frame in 32KHz to 8KHz. Searching for it in Speex Doc, I found it: SpeexResamplerState *resampler; resampler = speex_resampler_init(nb_channels, input_rate, output_rate, quality, &err); err = speex_resampler_process_int(resampler, channelID, in, &in_length, out, &out_length); But in my source I hav...

...comments from anyone who has got echo cancellation working when recording through WinMM, DirectSound or ASIO. 2. What about the Speex echo cancellor and the different sampling rates. Will it only perform well in narrowband mode (8kHz)? Does it not perform well in wideband (16kHz) or ultra-wideband (32kHz) mode? I read somewhere that you need a longer tail length for higher sample rates. Measured in samples it will of course be longer, but it could be interpreted as longer measured in milliseconds for higher sample rates. Someone please clarify. //Greger -------------- next part -------------- An HT...

This is a really good point, and definitely a recurring theme on this mailing list. :) I wonder, what are some better options for handling this issue, other than to keep saying "just use 8/16/32kHz"? - Extend Speex to support other sample rates (seems unlikely..?) - Integrate a resampling algorithm into libspeex - Maintain a list of recommended resampling libraries that work well with Speex ... maybe even have sample code? Any others? Anecdote - I've noticed one somewhat legiti...

...(1 channel, 22050Hz, 16 bits/sample) in blocks of the frame size. However, the encoder lags significantly - typically upwards of 2000 frames, at complexity <= 1 and quality <= 3. I've a On the decode side, the decoder is not able to do real time at these sampling rates (16kHz, 22kHz and 32kHz). The only sampling rate that seems to work at real time is 8kHz for decode. However, encode at 8kHz is still nowhere near real time. Is there anything I can do to get to the higher sampling rates and real time? Or is this just not possible? FWIW, the basic app uses MFC and the encode and decode...