similar to: amr

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

Has anyone successfully compiled the AMR codec into an Asterisk install, and if so, what steps did you take? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100825/b3db9f37/attachment.htm

Hi, I was wondering if anyone has experimented with Speex's wideband (16kHz) mode at lower quality settings. In particular I have been using quality 3, and with wideband input files the resultant frequency spectrum is limited to about an upper end around 3.5kHz (almost telephony quality bandwidth). Has anyone tried increasing the spectral bandwidth at the expense of lowering the

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.

Because a wideband signal has both a narrowband part and a wideband part the first part of the wideband decode is to call the narrowband decode. In the fixed point version the signal generated from the narrowband decode is downshifted at the end. In this same version the wideband then upshifts the signal by the same amount. I would like to do away with the downshift on the narrowband side

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".

I downloaded speex-1.2beta3-win32 binaries, and used the speexenc.exe and speexdec to process the file male.wav download from http://speex.org/samples I used both the narrowband and ultra-wideband mode: >speexenc -n --quality 3 male.wav n3.spx Encoding 8000 Hz audio using narrowband mode (mono) >speexdec n3.spx n3.wav Decoding 8000 Hz audio using narrowband mode (mono) Encoded with Speex

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

Alfred E. Heggestad wrote: > <...> > > If we don't get any comments in 1 week (by 22. May 2007) we will go ahead > and submit it to the IETF. Of course you can comment on it also after it > has been submitted, but we would like to get the input from the Speex > community first.. > thanks for all the input. please find attached an updated version of the draft. I

Hi Please find an updated version of the Speex I-D attached. The only change is addition of the copyright conditions in Appendix A, as requested by Ivo. Many thanks for your input. I will give you a few more days before submitting to AVT working group /alfred Ivo Emanuel Gon?alves wrote: > Do not forget to add the "Copying conditions" to the RFC. > > Check

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

At 07:55 PM 3/11/03, Jean-Marc Valin wrote: > > - Only narrowband (8 kHz) Speex is currently supported; not > > wideband. (Unfortunately, the assumption that audio sample rate == 8 kHz > > is riddled throughout the Asterisk code.) > >Perhaps it's still possible to send wideband, while telling Asterisk >it's narrowband (the bit-stream is such that you can decode

> I was wondering if anyone has experimented with Speex's wideband (16kHz) > mode at lower quality settings. In particular I have been using quality 3, > and with wideband input files the resultant frequency spectrum is limited to > about an upper end around 3.5kHz (almost telephony quality bandwidth). Has > anyone tried increasing the spectral bandwidth at the expense of

Hello Jean-Marc, Regarding your question - a little background on the various choices, Fair Quality is 15kbps Speex narrowband, Good Quality is 24kbps Speex narrowband, High Quality is 32kbps ADPCM from HawkVoice and Toll Quality is 64kbps PCM. We also have 10 other codecs within the product including GSM, LPC and others that we currently do not expose. To us, quality is a combination of

On Thu, 17 May 2007, Jean-Marc Valin wrote: >>> Consider a device that only has enough ROM to store one set of >>> quantization tables (the limitation could also be about speed, network, >>> ...). If you specify MUST be able to decode, then it means that this >>> device simply *cannot* implement the spec *at all*. This is bad for >>> interoperability.

Jean-Marc Valin wrote: > Hi, > > I just released unstable version 1.1.2 that contains more fixed-point > work. Though it's still not 100% complete, enough have been done to make > it run in real-time on ARM. In order to do that, compile with > --enable-fixed-point --enable-arm-asm. All narrowband modes work in > real-time with complexity 1 (some work with higher

Dear Speex developers, I am currently experimenting with Speex on Symbian smartphones. I have compiled the Speex library, and I am now using it in the following way: 1. Record 320-byte buffers of data in PCM16 format, 8000 Hz sampling rate. 2. Feed the resulting buffer to an instance of a narrowband Speex encoder. 3. Send the encoded data over RTP. 4. Upon receiving on the other side, feed the

In a "normal" frame that doesn't have in-band signalling (which nobody seems to use anyway), the frame starts with the narrowband information. If the frame is in wideband, then this will be followed by a bit of wideband information that starts with a bit set to 1 (narrowband starts with a bit set to 0). Have a look at what the Speex decoder does. You may actually want to have similar

Looks good to me. Jean-Marc Alfred E. Heggestad a ?crit : > Hi > > Please find an updated version of the Speex I-D attached. The only > change is addition of the copyright conditions in Appendix A, > as requested by Ivo. > > Many thanks for your input. > > I will give you a few more days before submitting to AVT working group > > > /alfred > > Ivo

> - Only narrowband (8 kHz) Speex is currently supported; not > wideband. (Unfortunately, the assumption that audio sample rate == 8 kHz > is riddled throughout the Asterisk code.) Perhaps it's still possible to send wideband, while telling Asterisk it's narrowband (the bit-stream is such that you can decode a wideband frame even if you think it's narrowband). > - Some