similar to: Well, I'm stumped

Hello, I'm running into problems while trying to build libvorbis. My build platform is Solaris 10 (sparc) (11/06) with GCC 3.4.6. The build fails with the following error : Making all in examples gmake[2]: Entering directory `/afs/cad.njit.edu/u/a/n/andym/ftp/freeware/multimedia/ogg/libvorbis-1.2.0/examples' gcc -DHAVE_CONFIG_H -I. -I.. -I../include

king all in vorbisfile make[2]: Entering directory `/home/ronan/libvorbis-1.0.1/doc/vorbisfile' make[2]: Nothing to be done for `all'. make[2]: Leaving directory `/home/ronan/libvorbis-1.0.1/doc/vorbisfile' Making all in vorbisenc make[2]: Entering directory `/home/ronan/libvorbis-1.0.1/doc/vorbisenc' make[2]: Nothing to be done for `all'. make[2]: Leaving directory

[I just got on to this mailing list to report this problem. If it has already been reported, then I'm sorry for duplicating the report. And I seriously doubt this is not the right topic to talk about on this mailing list, but if it is, sorry for that too.] Okay, here's a new one. I have never seen this error before. This is the latest CVS code I'm trying to compile, on a pretty close

Folks; I am having a hell of a time trying to get IceCast 2.0 installed on an Apple XServe running OS X 10.2.3. The problem seems to stem from one of the libraries required by IceCast, specifically libvorbis. When running MAKE in an attempt to install libvorbis I get the following: <p>Making all in examples /bin/sh ../libtool --mode=link gcc -DDARWIN -fno-common

I hope it isn't too confusing to everyone posting my responses here. Monty requested this thread of discussion be moved to this list. On 14 May 2000 15:54:28 Martin Vogt <mvogt@rhrk.uni-kl.de> wrote: > The anser is "arts". It will become the KDE2.0 soundserver. > Currently I'm integrating the kmpg patch into arts, > then all KDE2.0 apps will be vorbis aware (if

Hi, I fixed the distorted audio bug in the XMMS plugin. A small follows below: --- vorbis.c.orig Mon Oct 11 04:05:14 1999 +++ vorbis.c Mon Oct 11 13:47:42 1999 @@ -161,7 +161,7 @@ /* XXX figure out something to put for the bitrate and how to calculate the length of the stream */ vorbis_ip.output->open_audio(FMT_S16_LE, od.vi.rate, od.vi.channels); - vorbis_ip.set_info(filename,

Hey all...I'm trying to use the decoder_example.c in the examples directory to decode a .ogg file that was encoded by using the binary encoder ogglame.exe and encoded via the encoder_example.c. I'm running into a problem with the decoder though. It plays FINE under the WinAmp plugin but I when I try and use the decoder_example.c to decode the file it says "End of file before finding

I'm trying to adapt decoder_example to to support more of an event driven model. I'm building an RTSP client/server where the client will have a function called everytime a new packet comes in. So, I want to queue up these packets to be played. My current attack is as follows (note this isn't very robust and I realized that): For first and second packet i grab all the header stuff

Okay, here's my first hack attempt at an xmms module. I managed to link (although not test) this with the current libvorbis CVS source (which needed some "help" compiling) and xmms 0.9.5.1. I'm guessing I made some fairly obvious mistakes in here which can be hammered out even without bitstreams to test it on. Tony Arcieri <HR NOSHADE> <UL> <LI>TEXT/x-csrc

I am running the decoder_example file but instead of ouptutting the PCM to sdout I would like to output it to the soundcard so that I can hear the file playing. I am using Visual C++. Could someone please tell me the quickest way to do this. I am quite new to Visual C++ so don't assume that I know about the foundation classes. Thanks, Paul PS I managed to build the GUI using the

I am running the decoder_example file but instead of ouptutting the PCM to stdout I would like to output it to the soundcard so that I can hear the file playing. I am using Visual C++. Could someone please tell me the quickest way to do this. I am quite new to Visual C++ so don't assume that I know about the foundation classes. Thanks, Paul --- >8 ---- List archives:

Well, I updated the player to work with the branch_postbeta2, but it contains some nasty kludges to the project files that probably shouldn't be finalized. I'm going to put it in a separate file (http://oscar.the-rileys.net/programming/vorbis_pb2.tar.gz) for now. Unfortunately, this release doesn't seem to fix the "tearing sound" problem. I think this might be related to

There isn't a lot of guidance on how exactly to run the codec I compiled out of cvs just now, but I expected the following to work: cat test.wav | encoder_example | decoder_example But the decoder fails with: End of file before finding all Vorbis headers! Is this a bug? Or am I missing something fundamental? Should I attach the test.wav? -- Daniel --- >8 ---- List

There isn't a lot of guidance on how exactly to run the codec I compiled out of cvs just now, but I expected the following to work: cat test.wav | encoder_example | decoder_example But the decoder fails with: End of file before finding all Vorbis headers! Is this a bug? Or am I missing something fundamental? Should I attach the test.wav? -- Daniel --- >8 ---- List

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

Hello Hope you don't mind another newbie's question. Further to GIE's email about problems compiling decoder_example.c example (MS VC 7) using debug static ogg/vorbis libs, we'd appreciate any help getting the example files to compile on MS VC 6. The zip files we have downloaded are: libao-0.8.2.zip libogg-1.0rc3.zip libvorbis-1.0rc3.zip vorbis-sdk-1.0rc3.zip

What exactly does it mean when, in the decoder_example.c file, you try to extract the information from the comments and codebook headers and you loose sync? What is actually being check to determine when sync is lost?. I known one part of it is that there is a gaps in the stream. Is the CRC value the determining factor? Please detail! I read a lot of the documents. I still am a little fuzzy.

Hi, I'm speech recognition researcher and would like to do some research on recognition robustness in echo distortion of speech signal. Since VOIP is becoming wide spread, I'd like to simulate (one or more) common echo distortions that mostly appear in voip communications ? Any example, FIR or IIR filter or acoustical system response ? Any other distortion worth researching ? Thanks

Hi, The notch filter in AEC is only used to remove DC signal, and the time of convergence is not important, right? If so, I think preset value of notch_radius is too small, and it causes noticeable distortion(freq < 200hz cut). There is a picture in attachment to show signals under different radius in time-domain. By setting notch_radius to 0.999 for all sampling rates, I found better

Hi, Here are part of my data, > pr16.5    origin dest ldco time.slot distortion 1       1    1    1         1        4.3 2       1    1    1         2        4.7 3       1    1    1         3        5.6 4       1    1    1         4        7.7 5       1    1    2         1        6.8 6       1    1    2         2        7.6 7       1    1    2         3        9.2 8       1    1