similar to: Integer Version with low bit rate for Linux Kernel?

Hello Segher Boessenkool I am writing a Linux Kernel Device-Driver for a low bitrate radio link, connected via USB to the PC. Incoming VOIP- IP Pakets go directly to the network interface of the kernel module. A transcoding from e. g. PCM or G723.1 to e. g. speex 2.15 is necessary, and also a header reduction RTP/UDP/IP. Egon -------------- next part -------------- An HTML attachment was

Hello Alexander, I hope, there is no error in speex, if so, of course,I have to fix it. In order to do the transcoding in user space, I have to transfer the voice paket received from VOIP-equipment within kernel to user space, transcode it, send it back to the kernel, send it over the radio link, send it from kernel to user space, transcode it back, send it to the kernel, send it from

Hello Erik, The transfer time from user space to kernel and back is not a problem, it is faster than required. > Have you measured the time taken to transfer a data packet from kernel to > user space? Have you measured the time taken to encode that packet? What > are those measurements? > If you haven't performed those measurements, putting the Speex encoder > in

Hi to everyone. I'm using speex in an application at the minimum bit rate (2.15 kbps), setting the encoder quality at 0, but obviously the quality is terrible. My question is: there are some signal processing that I can do to improve the quality of the signal at very low bit rate? Or can you tell me other codec that works well at that rate? Thank you.

here is the situation i am using htb.init with fwmark to do QoS. i have 2 parent classes with RATE=CEIL which then have some leafs each on his own. the first one works fine (it shapes the packets to the specified rate) class htb 1:21 root rate 1Mbit ceil 1Mbit burst 2909b cburst 2909b Sent 631520262 bytes 651550 pkts (dropped 0, overlimits 0) rate 131573bps 141pps lended: 380595 borrowed: 0

Hi all, It is my understanding that with HTB, the rate and the ceiling are divided over the elements of the class. E.g. using a rate of 100 kb and a ceiling of 2000 kb for a class with 10 elements on a 100Mb NIC, the effect of the ceiling will be that if all elements are generating their maximimum possible trafic they will be effectively limited to ~ ceiling / number of elements, in this case

Hi, I'd like to contribute to Vorbis and I think this may be of some interest for low bitrate coding. I have been experimenting with low bit-rate coding for the high-band (11 kHz to 22 kHz) and, though I haven't yet started quantizing my coefficients (a gain and an LPC filter), I expect to be able to approximate the whole 11-22 kHz band with around 1000 bits/s per channel (maybe even 500

hi how to check the data rate on the nic card in centos i.e, at what speed the data is transmitted and received on ethernet interface in terms of kbps or bps any command is available in centos 5.1 regards, Gopinath M -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm not sure, but I think that Microsoft have the license now.. anyone know if there is the source code to download? Or others codecs.. thank you very much ---------- Original Message ---------------------------------- From: "John Miles" <jmiles@pop.net> Date: Mon, 19 Jun 2006 10:15:59 -0700 >Do a search on "Voxware MetaVoice". I'm not sure who licenses

Hi All, Let me ask this question of the group: When bit rate peeling becomes available, how will the quality of the peeled Vorbis file compare to a file encoded at the target quality directly from the original? So, for: a.wav --> b.ogg (at q6) --> c.ogg (at q2) a.wav --------------------> d.ogg (at q2) how are c.ogg and d.ogg likely to compare in terms of audio

Hello all. This is a question that I should be able to answer, but the old grey cells aren't working. My Google-Fu must not be up to snuff, either. I have a MOXA IP-RS232 converter that I'm going to be using over an RF WAN and I need to confirm the actual data rate that I'll be able to achieve. I have the drivers loaded (/dev/ttyr00) and a loopback plug on the end of the device. I

This is not specifically on-topic for Asterisk, but I have found on many occasions while working with Asterisk that it would have been very handy to be able to measure, with some precision, the bandwidth being used by a particular host, port, or combination of the two. So, I went searching for various tools, none of which were what I wanted. They either were too clever, or too limited in

On Thu, Jun 19, 2014 at 03:30:22PM +0400, lvqcl wrote: > BTW, what can you say about the following place in stream_decoder.c > in read_subframe_lpc_() function: > > /*@@@@@@ technically not pessimistic enough, should be more like > if( (FLAC__uint64)order * ((((FLAC__uint64)1)<<bps)-1) * ((1<<subframe->qlp_coeff_precision)-1) < (((FLAC__uint64)-1)

In the precompute_partition_info_sums_ function, instead of selecting 64-bit accumulator when the signal bps is larger than 16, revert to the original approach based on partition size, but make room for few extra bits to not overflow with unusual signals where the average residual magnitude may be larger than bps. It slightly improves the performance with standard encoding levels and 16-bit files

On Thu, Aug 14, 2014 at 12:34 PM, lvqcl <lvqcl.mail at gmail.com> wrote: > Jose Pablo Carballo <jose.carballo at ridgerun.com> wrote: > >> - channels = 2; >> - bps = 16; >> + channels = ((unsigned)buffer[23] << 8) | buffer[22]; >> + bps = ((unsigned)buffer[35] << 8) | buffer[34]; >> total_samples = (((((((unsigned)buffer[43] << 8)

Hi Guys, I've just faced a wrong warning trying to encode a 24 bits WAV file if(wFormatTag == 1) { if(bps != 8 && bps != 16) { if(bps == 24 || bps == 32) { /* let these slide with a warning since they're unambiguous */ flac__utils_printf(stderr, 1, "%s: WARNING: legacy WAVE file has

Hi, In the last days I've been taking as reference the example found in examples/c/encode/file/main.c. With it I've been able to encode a 2ch, 16 bps, 44100 sample rate input WAV file to a FLAC file. Now I've been trying to modify this example to encode a 2ch, 24 bps, 96000 sample rate WAV file. I have to say I'm a bit lost on how I should read the input file in this case, and

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Hi all, [For the uninitiated: a "peeler" is a program that transforms a Vorbis stream into a smaller, (somewhat) lower quality Vorbis stream, and does so quickly, by just throwing out some data.] After having prototyped several peelers that aim to peel to a certain filesize, or to a certain quality, with mixed success, I've now taken a different route: a peeler that aims for the

Hi all, [For the uninitiated: a "peeler" is a program that transforms a Vorbis stream into a smaller, (somewhat) lower quality Vorbis stream, and does so quickly, by just throwing out some data.] After having prototyped several peelers that aim to peel to a certain filesize, or to a certain quality, with mixed success, I've now taken a different route: a peeler that aims for the