Re: [asterisk-users] drop dead fix
pbx$ man sox allpass frequency[k] width[h|k|o|q] Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width width. An all- pass filter changes the audio's frequency to phase relationship without changing its frequency to amplitude relationship. The filter is described in detail in [1]. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Friday, October 15, 2010 9:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] drop dead fix On 10/15/2010 09:59 AM, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Thanks Danny Nicholas hiring someone to re-record 304 prompts is not simpler and far faster than redeploying an entire system ? sounds like about a 4hr job. or find a better converter. Option 2 is what I have in mind (BTW, with the talent I have, your 4 hrs is closer to 80, after normalizing, trimming and prodding). What I do now is record the file using soundrec, normalize it with Audiograbber, then trim it with Audacity before converting it with Sox. Which of these is letting me down, (or it is the loose nut on the keyboard)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
On Fri, 15 Oct 2010, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Why are you converting them to GSM? Why not convert them to the technology you're using for your phones and trunks? That would be much more efficient. (If you're using g729 for trunks, then that will sound better as GSM to g729 conversion does sound bad) Or maybe it's your conversion software? What are you using? Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
On Fri, 15 Oct 2010, Danny Nicholas wrote: I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Can you post a link to a sample before and after file as well as the command line you are using to convert the file? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
I never had this problem, and this is certainly not asterisk's fault. Probably your conversion is not good. Can you email me a file and I'll do conversion on my end, and if sounds good, let you know how I did it. Then a script can be written to convert them all. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-15 10:25 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 15 Oct 2010, Danny Nicholas wrote: I am about to have to dump Asterisk in f... Can you post a link to a sample before and after file as well as the command line you are using to convert the file? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
You want to pay attention the low-pass and high-pass filter A step conversion will help you see the issues. Go halfway first and look for the change and adjust your filter. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, Oct 15, 2010 at 11:18 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Fri, 15 Oct 2010, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Why are you converting them to GSM? Why not convert them to the technology you're using for your phones and trunks? That would be much more efficient. (If you're using g729 for trunks, then that will sound better as GSM to g729 conversion does sound bad) Or maybe it's your conversion software? What are you using? Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
On Fri, 15 Oct 2010, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Friday, October 15, 2010 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] drop dead fix On Fri, 15 Oct 2010, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Why are you converting them to GSM? Why not convert them to the technology you're using for your phones and trunks? That would be much more efficient. (If you're using g729 for trunks, then that will sound better as GSM to g729 conversion does sound bad) Or maybe it's your conversion software? What are you using? Gordon I did the proof of concept recordings as gsm files. Now that we want to actually do a finished product, the gsm recordings don't sound good enough to make a viable product. Here is a sample Original file http://www.4shared.com/audio/PDGcMDUt/firstmenuwav.html Seems very quiet to me, but I don't have any tools to meansure it where I am right now. The GSM one didn't sounds too bad either, but are you then listening to it after a G729 conversion? Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
On Fri, Oct 15, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote: The original one is super quiet - obviously not Allison in a studio... Listen to the gsm in Asterisk to see my quandary... What is the end use here? Who will be listening to the recordings? Users on PSTN and mobile phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen Sent: Friday, October 15, 2010 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] drop dead fix On Fri, Oct 15, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote: The original one is super quiet - obviously not Allison in a studio... Listen to the gsm in Asterisk to see my quandary... What is the end use here? Who will be listening to the recordings? Users on PSTN and mobile phones? End use is Telephone Banking, so you've nailed the target audience. BTW, the highpass and lowpass filters seem to help, but since I stopped math at pre-calculus, the explanation of the Butterworth filter is beyond my pay grade. Care to offer a better explanation? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
On 10/15/2010 08:59 AM, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? In addition to all the other comments you've received (including the fact that Asterisk does not require GSM format files), keep in mind that *any* product that plays these files over the PSTN is going to have to downsample them to 8KHz and, at a minimum, use G.711 companding. That is what the PSTN uses, so it's not possible to have higher fidelity than that. There were some comments in other replies about your files being 'quiet' (low average volume level)... this won't help your situation at all, because it means that any artifacts caused by resampling and compression/decompression will end up at a relatively high amplitude compared to the original signal (resulting in a low signal-to-noise ratio), and when the listener increases the volume level on their listening device, the noise level will be increased along with it. For these sorts of tasks, you really do want the source material recorded at a fairly high volume level. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
On Fri, Oct 15, 2010 at 11:20 AM, Danny Nicholas da...@debsinc.com wrote: End use is Telephone Banking, so you've nailed the target audience. BTW, the highpass and lowpass filters seem to help, but since I stopped math at pre-calculus, the explanation of the Butterworth filter is beyond my pay grade. Care to offer a better explanation? While, officially, I completed up to calculus 3, the serious lack of use is not helping. You'd be better off taking the highpass number from low to high and listen to the difference, and then do the same for the lowpass number. Your ears will tell you when you have it right (you will definitely hear what each one does), and you can still consider Butterworth inexpensive pancake syrup. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
On Fri, Oct 15, 2010 at 9:35 AM, Danny Nicholas da...@debsinc.com wrote: Don't know if this will make acceptable GSM files, but should help with the WAV ones. Are you using GSM to talk to an ITSP (the idea of banking voice calls going across the internet makes me cringe)? If not, what are you using GSM for? GSM always sounds like garbage (and see below - it's not what you are hearing on your mobile phone! It's not as good as mobile phone codecs). If you are using GSM to save bandwidth, you should really look at a better codec - but I would think a banking system wouldn't use the internet for the voice channel. If you are using a private network and bandwidth is still a concern, I'd look at any of the other codecs (except maybe ilbc, which is even worse than GSM). Any of them would sound better. Somehow, to get to a mobile handset user (who uses GSM), the call will hit the PTSN. The PTSN, as others mentioned, is 8K alaw or ulaw (depending on your country). Get the recordings to sound good on the PTSN (convert to alaw or ulaw 8K, as that's what will happen NO MATTER WHAT when your call hits the PTSN) - don't even try to optimize anything else until then. If you're hitting the PTSN at all (versus a direct connection within an IP-based GSM provider's network - unlikely that you have this), even though the handset user is on GSM, you do NOT want to use GSM as your encoding. Use 8K alaw/ulaw (wav format). I suspect your GSM providers in your area have spent literally millions of dollars on their GSM encoding systems - let them do the work. They'll have to do it even if you played the GSM file, it'll just sound worse if you play GSM, convert it to 8K alaw/ulaw over the PTSN, then have it converted using a different algorithm at the cell site. Finally, not all mobile calls on even GSM networks are gsm format. If they are a different format, converting from one compressed algorithm (gsm) to another (whatever the carrier uses) is going to sound horrible. So don't bother with the GSM format. Few people you are calling/called-by use that codec (not even the mobile phone users). It'll get resampled into something else. You'd be better off using the raw, basically-uncompressed (I know, I know, not quite accurate) alaw/ulaw - which everyone's codecs are designed to handle very well (since every single PTSN call uses it). For reference, Asterisk uses (I believe) the full rate GSM codec. Mobile phones on most GSM networks are using an AMR (not full rate) codec, as it simply sounds better, can deal with bad connections better, and can even use less bandwidth. Of course it is licensed and patented, so Asterisk doesn't implement it. But because of this, Asterisk's gsm doesn't sound as good as a call on a GSM network. Why would you want that? Just don't use it! See http://en.wikipedia.org/wiki/Adaptive_Multi-Rate (What mobile companies use) And http://en.wikipedia.org/wiki/Full_Rate (What Asterisk uses) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming There were some comments in other replies about your files being 'quiet' (low average volume level)... this won't help your situation at all, because it means that any artifacts caused by resampling and compression/decompression will end up at a relatively high amplitude compared to the original signal (resulting in a low signal-to-noise ratio), and when the listener increases the volume level on their listening device, the noise level will be increased along with it. For these sorts of tasks, you really do want the source material recorded at a fairly high volume level. On Fri, 15 Oct 2010, Danny Nicholas wrote: This appears to be the resolution to my problem - #1. Get my recording talent in an isolated environment so I can get clean, loud recordings #2. Dump the Audacity and Audiologic steps and just use SOX with the highpass and lowpass filters. 1) firstmenu.wav.wav is recorded so low, it just looks like line noise in Audacity. Unless you can re-record at a reasonable level, you're always going to be fighting this sow's ear. 2) I use normalize (http://normalize.nongnu.org/) to normalize from the command line, but it does not deal with DC offset like Audacity will. Eliminating your DC offset issue should also be a goal of improving your recording environment. Newer (than provided with CentOS 5.5) versions of sox can do dcshift. 3) Stick with ULAW or PCM (wav). You only have to be concerned with supplying audio encoded appropriately for the first hop in your delivery path. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
