similar to: Dropped speex packets

Hi, I have been testing 1.2beta1 fixed point and non fixed point as well as 1.0.5 compiled agains uclibc and run on 150Mhz pentium MMX chip. 1.2 non fixed is fastest 60 seconds encoded in 32 seconds at quality 4 1.0.5 is second with the same file encoding in 39 seconds at quality 4 1.2 fixed is slowest with 60 seconds encoded in 58 seconds at quality 4 Is this as expected? Or am I doing

Speex 1.2beta1 announcement: This new release brings many significant improvements. The quality has been improved, both at the encoder level and the decoder level. These include enhancer improvements (now on by default), input/output high-pass filters, as well as fixing minor regressions in previous 1.1.x releases. A strange and rare instability problem with pure sinusoids has also been fixed. On

Hi Chris, Thanks a lot for your help. I've been able to resolve all the errors except one. I don't have ogg.h header file, so when I try to build Speexdec.exe, I get an error d:\speex\speex-1.1.5\src\speexdec.c(44) : fatal error C1083: Cannot open include file: 'ogg/ogg.h': No such file or directory I don't have this file, but as I wrote in my previous mail, an ogg_static

Hi, I've downloaded the Speex-1.1.5 code and am trying to compile the Speexdec code using Microsoft VC++ 6.0 IDE. When I try to open the speexdec.dsw workspace, I get a dialog asking to locate the ogg_static file. I simply cancel that message, and then the project is opened in the VC++ 6.0 IDE. Now, when I simply try to compile the program, I receive this 5 error messages. error1.

Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase in audio quality and the bitrate is still very low, especially if you take advantage of Speex features such as VBR. 8kHz seems totally inappropriate to me for desktop streaming audio, let alone 8-bit samples. Or perhaps your recording equipment is an original Sound Blaster from 1989? (Even that could record at

I have an audio file compressed by JSpeex that I'm trying to decompress on the iPhone. I've been trying to work from the speexdec example in an attempt to port the core logic to ObjC. I have a class that does the decode and passes the result back to another class. (currently the other class is a unit test.) My other class is writing the converted output to disk. I compare the results of my

Hi, Jean-Marc, I have another quick question, I got following message when compile speexdec.exe: Linking... libspeex.lib(sb_celp.obj) : error LNK2001: unresolved external symbol _lpc_window Release/speexdec.exe : fatal error LNK1120: 1 unresolved externals Where does this lpc_window declare? Thanks a lot, -Allen On 8/8/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: >

I just did: $ speexdec foo.wav foo.wav.wav Segmentation fault There should probably be some sanity check in speexdec. /Ole --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is

On my personal web pages hosted by my ISP I have a collection of speex files with names ending, as usual, in '.spx'. My ISP does not provide for scripting or CGI but thanks to the "Home-Brew Streaming" idea of Dr Tak Auyeng, www.drtak.org/teaches/ARC/speex/, my friends with Windows XP can listen to such a recording without the need for a prior download of the whole file and

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

Hi all, I'm trying to use libspeex 1.1.10 on an ARM926EJ-Sid(wb) rev 3 (v5l). I executed the speexenc and speexdec test files and they can encode and decode. But I'm getting 95% of cpu utilization on the codification and 44% on the decodification. I saw in the post: http://lists.xiph.org/pipermail/speex-dev/2005-June/003485.html that this version of speex works fine on ARM

Thanks a lot for your replies. Where can I get Ogg static libraries header, and Ogg header files. Any place where I can download. Basically I need to create a Speex library for Pocket PC 2003, that can be used from within my .NET Compact Framework application. For this I need to create a DLL file, which essentially is my Speex decoder. For this I simply need to make use of Speexdec, since this

On Fri, Jul 8, 2011 at 6:39 AM, Ashhar Farhan <farhan at phonestack.com> wrote: > what you need to do is this: take the wireshark raw dump, read each > udp packet and write it back to another file. While writing back to > the new file skip the sizeof udp header + rtp header. I can't recall > how many bytes you need to skip, however, I suppose it would be in the > range of

As a note, with the latest versions, I get this "new" failure, when cross compiling for windows using mingw: CCLD speexdec.exe speexdec.o:speexdec.c:(.text+0x4af): undefined reference to `_imp__waveOutGetNumDevs at 0' speexdec.o:speexdec.c:(.text+0x54d): undefined reference to `_imp__waveOutOpen at 24' speexdec.o:speexdec.c:(.text+0x5aa): undefined reference to

Hi Roger, > > On Fri, Jul 8, 2016 at 6:41 PM, Roger Pack <rogerdpack2 at gmail.com> wrote: >> As a note, with the latest versions, I get this "new" failure, when >> cross compiling for windows using mingw: >> >> CCLD speexdec.exe >> speexdec.o:speexdec.c:(.text+0x4af): undefined reference to >> `_imp__waveOutGetNumDevs at 0'

Hello all, I know that SPEEX has a mechanism to handle packet loss. I see there is some handling for packet loss implemented in speexdec. I want to know, if in my own application should i also implement such packet loss handling or is it taken care of within the decoder? I want to test my SPEEX application for GNU Radio for some packet loss,if i just simulate a packet loss mechanism, will it work

I am hoping to use Speex for a embedded project using Philips Arm processor (50 mips) 512kb flash 32kb ram. I found in the manual that decode takes about 0.5 mips so I should have enough processing power and I compiled the lib and it seems to take less then 64K so the only issue is memory usage. I have been testing the speex decode on windows looking at the stack usage and how much is malloc.

OK, that was a stupid error in the le_int prototype. It's fixed in svn now. Jean-Marc Le mercredi 28 septembre 2005 ? 11:53 +1000, Jean-Marc Valin a ?crit : > OK, it seems like I screwed up the wav header code between 1.1.8 and > 1.1.9. Everything seems fine (playing to soundcard also works), except > that the length reported for the wav is too short, which is why it stops >

Hi, I tried the same options suggested in your post and the problem continues. What do you suggest? 2005/9/15, Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca>: > Hi Eduardo, > > All I can say is that the timings you have are a bit odd. What > optimizations options are you using (I suggest -O3)? Also, perhaps you > can try --enable-arm4-asm just in case. I've had the

Hi, What kind of latency is expected using 8,16,32 khz? I am trying to do a realtime stream server, and I am having latency above >500ms ( I capture the sound using the mic, encode it and send it to the client). I am using ALSA system, a SB128 PCI and a 800Mhz P3 . What can I do to lower the latency? I tried a test : arecord -t raw | speexenc - - | speexdec - and I found that I also have a 500