similar to: Technical MP3->OGG transcoding question

In case there are any Debian developers around here, I wanted to point out message <20020828154322.GA15114@chulak.naquadah.org> on the debian-devel list this morning. Another Debian developer is proposing to submit an mp3->vorbis transcoding program for inclusion in Debian. I have objected to this on the grounds that the resulting vorbis files will sound like crap, and I have also pointed

I know transcoding is a bad thing. However, i am in the need of being able to lower the quality of an ogg file without the original CD. Whats the best way to do it (least generational loss possible) - "ogg > ogg" or "ogg > wav > ogg" or is there an easier way (as in has someone got a decent working peeler up there sleeve and isn't telling anyone about it)

How do you keep oggenc open long enough to even USE it?!?!?!?!?! Whenever I tell the computer to open the file, it says [Process Completed] and [logout]. What is going on?

Hello, I'd like to develop a codec for the Ogg Vorbis algorithm on a DSP (controller?) hardware platform. We've got 2 (3) main applications: 1) the codec should be used for 2 channel simultaneous message recording/playback on flash EPROM. 2) the codec should be used for streaming audio over IP or Ethernet/intranet networks. 3) for large system, the PC should serve as a storing device We

Asterisk Project Security Advisory - AST-2019-005 Product Asterisk Summary Remote Crash Vulnerability in audio transcoding Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Minor

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

I have a problem that I hope perhaps someone could help me with. I want to relay an mp3 stream (bitrate switching, metadata containing) from an url like http://192.168.0.123:8000/stream.mp3 and serve it to multiple listeners (Winamp clients) using Win32 environment for the relay machine. The original stream is being served by the (free!) SLIMP3 Server software

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

Hi there, up till now I had this two-box setup in mind: * no.1: public IP * no.2: private IP, registers with no.1, serves a small office with clients behind NAT See we'd get something like this: SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA The codec of the SIP client (on the Internet) I don't have full control over, that depends on the

Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice

Forwarding a conversation with a company that provides a management tool for transcoding Vorbis files to MP3 when communicating with the Nomad Jukeboxes from Creative. Good initiative, and probably sounds OK too, but would be even better with FLAC, so I'll ask about it. Linus ---------------------------- Original Message ---------------------------- Subject: Re: Vorbis question From:

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

As far as I can see, transcoding could be usefull for people who do not primarly care about quality but about filesizes. One could assume that such a user would have a collection of mp3's at 128kbps or higher bitrates, and uses an encoder like BladeEnc or Xing. He wants to take uses of ogg's supposed quality and transcode his 128-or-higher files into 96 or 112kbps oggs to save diskspace.

Hello I use asterisk realtime, and I can set the order of codec preference on my realtime allow column. If I could disable transcoding, then I can always ensure a passthrough of the common codec from origin to destination without transcoding (expensive on CPU) - and more or less, force the codec to use by setting the codec preference So, can I disable transcoding?

Before I put myself to the task (next month, maybe) of surveying the CPU costs of transcoding, perhaps someone else has already done this work and would be willing to share it or refer me to a link of previously published data. My reviews of the mailing list with various keywords were unsuccessful in finding adequate references, though I admit I only spent 20 minutes looking. What I seek is

Hello, Asterisk is said to handle call routing and codec translation. I would like to force transcoding function with asterisk but when I try to force transcoding I get the errors: codec not compatible or WARNING[4425]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't make SIP/xxx compatible with SIP/yyy How exactly works asterisk, in order to transcoding? If you have any

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and

Hi, I have a bunch of old radio programs (mystery/drama shows, not music) encoded at 32 kbit (and some 48kbit) mp3 (mono). I want to reencode them in ogg and make them available over gnutella. My question is this. What is the best quality level (-q) for transcoding them. I want to preserve quality, but I want to be sensitive to the many modem based gnutella users. I also want to to

There is a serious bug in Sangoma transcoding cards. The card has an internal, small jitter buffer and it drops samples from the audio stream when there is high jitter in the network. The bandwidth is cheap now so for me the only reason to use transcoding is where I have low-bandwidth-high-jitter links. Sangoma said they will not fix it and we had to go back to software transconding. Do you have

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw