similar to: is it possible to bring speed below 1000 bit/s

Hi all, I'd like to know how well Speex handles multiple input streams. More specifically, if i have 4 streams that are playing at "the same time", how well does the CODEC handle switching from one stream to the next to process the audio. The reason i ask is this....My project up until now uses the MELP CODEC which didn't specifically have the ability to handle multiple

You're asking the wrong question. The question is not "why does it would bad with Speex?", but "why does it sound good with LPC10 and MELP?". And the answer is that both are vocoders. Try dropping frames/subframes with anything else (Vorbis, MP3, G.729, u-law, ...) and it'll sound terrible as well. The only reason it sounds good with vocoders is because the

Hi . I am having a very annoying problem not sure where the root of it lies When running FLdigi it runs fine for about 10 mineuts then i start getting errors leading to a complete lock up that needs a power button to free it up i get the following [ 653.080497] nouveau 0000:01:00.0: fifo: DMA_PUSHER - ch 3 [systemd-logind[462]] get 00000d3a24 [ 653.081461] nouveau 0000:01:00.0: gr:

Hi, Jean-Marc, and thanks for the quick reply. Let me just say I'm a huge fan of speex, and the work you've done. I actually barely understand what I'm reading so far in the source code and documentation, just enough to understand just how cool the algorithms are. LPC10 and MELP allow me to speed up speech with a simple hack on the decoder frame size. Playing fewer samples per

Hi. I'm Bill Cox, and I volunteer a bit for the Vinux project, which is Linux for people with vision impairments. Most blind users use a closed-source speech synthesis tool called voxin, as it's very easy to understand at high speed. I would like to make TTS synthesizers based on large recorded vocabularies of actual speech, but to make it useful for the blind, I need to be able to

I'm trying to run AudioMeter.exe, available for download at http://www.mydarc.de/dg8saq/AudioMeter/index.shtml It is a single executable which measures the sound card input as RMS values and displays them. When I try and run it under Wine, the program's main window (title bar and frame only) and its debug window (title bar and fame only) show up on the screen and I get a message box that

https://bugs.freedesktop.org/show_bug.cgi?id=38630 Summary: 2" of left side of screen appears on the right when using certain apps Product: xorg Version: unspecified Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component:

We have Asterisk1.8.11 and can not move to a newer version right now. But when we run Asterisk as a service, the -r option does not result in giving the CLI prompt? Did the option to get the CLI change? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130722/903d1851/attachment.htm>

Has anyone needed a coded that Asterisk does not natively support, such as MELPe or CVSD? If so, did you find a pure software solution and provided that as an addition to Asterisk? Was that solution successful? Has using an I/F card with a DSP proved to be the better solutions? We are beginners with Asterisk so any help/advice on how to best implement non-native Codecs into Asterisk will be

Dave, I have been interested in some of the same questions that you have been asking. A while back I asked a person at VLSI ( http://www.vlsi.fi/ ) in Finland if they had a product that might work. The exchange of email is included below. In summary, the answer was probably no, but maybe in the future. The VS1005 chip he mentions does not seem to have appeared yet. I currently have a pair

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

Here's one clue about whatever is causing the low quality speech. Speech sounds terrible at 1.01X faster, and it sounds excellent at normal speed (1.0X). So, the main problem is something that breaks with any change in frame size in the decoder. Any idea what that might be? Thanks, Bill On Tue, Oct 19, 2010 at 5:14 PM, Bill Cox <waywardgeek at gmail.com> wrote: > I was able to

Hello Thomas, You are correct - the application is basically for secure remote access to ham CW (morse code) station. I believe the solution will be based on low bit rate MP3 streaming for receiver side (MP3 allows use with existing CODEC in client side, which could include Win CE or Linux palmtop type form factor, and the audio source would just be "voice grade" material on a mono,

Hello Daniele It would be worthwhile to attach an external serial flash or USB thumb drive, if the intent is store data. This allows for far more flexibility in storage Regards Amit On Fri, Aug 26, 2016 at 11:02 AM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 26/08/16 11:40 AM, Daniele Barzotti wrote: > > You're right! I forgot to say that I need only the encoder

Hi, Steve. I tried your the time_scale_tests program, and it works well! Especially for low speed changes, it's the best I've heard so far. For high speed increases, there is what sounds like static added to the sound output. I've attached two sound samples of high speed speech, which is a 4X speed up of a popular TTS voice in the blind community (voxin/Eloquence). I've sped

Hi, I noticed that speex.org has a banner that mentions that Opus is better than Speex in all aspects. The supported bitrate range for Speex seems to be as low as 2kbps though but Opus can only go as low as 6kbps. Is this one aspect where Speex is still preferred? (I understand that it's not a very common scenario though). Thanks, Manpreet. -------------- next part -------------- An HTML

Hi, I would like to add a feature to my email-client (Thunderbird) which stops emails from showing up until a specific time (e.g. having buttons for: show again this evening, tomorrow, next week, next month ...). As far as I've learned this is not possible with IMAP: - storing the 'do-not-show-until-time' within a custom header-field requires down- and uploading of the whole email and

I am totally new to icecast, and would appreciate a pointer in the right direction. My application is to encode audio (at the linux server) at a low bit rate, say ~16 kbps, for transmission to a decoder at another location. The audio source will be bandlimited (basically voice grade circuit such as you'd find in a telephone application.) The decoder end (connected via a TCP/IP session)

Hello Stef, I am working on a Bandwidth management project as part of my college curriculum.I am running TCNG 10-b but I am not able to classify packets based on MAC address using tcng, so I am running tc commands in /tcng/bin and they are working. But is there a way out to classify packets based on MAC address using TCNG. And, I am successful in throttling down the Bandwidth to 8kbit

Hi Keir, this patch changes common code of xen hypervisor, but seems we have to, because current call to do_boot_vcpu is always from domains, however, for SMP VMX guest AP bring up, we need call it from xen hypervisor, so we''d like to split it. What''s your opinion? _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com