I've for months now been struggling (like Don Quixote, it appears) to find a way to RELIABLY, and with tolerance of any kind of error, concatenate several Ogg Vorbis files-- some with multiple streams in them-- into one.
At first I was trying to do it without decoding. Wrote a bunch of tools in several languages attempting to do that. None of them worked for all cases (some worked for a few, some worked for none).
The existing tools out there-- sox, mplayer, ffmpeg-- all require decoding/re-encoding, but even with that they failed on various edge cases (multiple streams in a file, streams with zero audio data, empty files, etc).
I finally ended up with the only thing that sort of worked and handled ALL the edge cases, even though it requires decoding and recoding audio, which wastes CPU and also might introduce artifacts, which was this magical pipeline:
ogg123 --audio-buffer 0 -q -d wav -f - files-to-concat/*ogg | oggenc -Q - > total-file.ogg
Except for one problem.
Any shows longer than 3:22:53 get concatenated to 3:22:53. What's special about that number? Oh, I bet it is a size_t overflow in probably the total number of decoded bytes at 44.1khz.
If my math is correct, 44.1khz, 2 channels, 16 bits, 3:22:53 is 2,147,317,200 bytes. Ding.
Now what? Go digging through the source of ogg123 or oggenc looking for the problem, and try to patch it?
Ogg Vorbis has been a released standard since, what, 2000? Is it possible in 2014 that I'm the only person who has ever run into this problem?
-ken
... With intel core duo (not core 2 duo).
What a bitch! But I've been able to boot kxstudio and Ubuntu studio (32bit) versions from live USB. I shrunk the HFS+ partition to make room and installed Ubuntu studio to the new ext4 part. Won't install the boot loader (coz of EFI)... rEFIt & rEFInd don't seem to be getting it done to recognize the images on the ext4 partition.
I've read thru many of the Ubuntu and like tutorials. Any simple clues to getting this done? I like Linux on the Mac.. It has optical spdif out on regular sound card for 1 thing. And it's a 2ghz dual core, working a little better than the Vaio 1.6 dual core of the same era.
Thnx in advance.
~ Russell
Hi,
I am wondering, wether there is a way to create click tracks. Audacity
is quite limited, as it only creates quarter note clicks, while the
upper tempo setting is limited, so f.e. you cannot create 16th for
120BPM by means of scaling up the BPM.
Also, there are no triple notes, swing eighth or syncopated options
available, though I am afraid, those will be rather seldom, if availble
at all.
Are there any alternatives, that are at least a little more flexible
than audacity?
I am not talking about recording a plugin, like a softsynth, which is
liable to the internal (midi) clock oscillation, but generating an audio
track with really accurate clicks
Well that was a surprise, to see that ffmpeg has been fully removed from
Ubuntu now. Not sure how much of a better move that is, compared to the
very underhand tactics of having it a renamed package which in reality
had nothing to do with real ffmpeg. Not sure I want to get into the
politics of the situation here though, sure most of you know far more
than I do!
Anyway I have a little script for batch converting flacs to mp3s as I am
rather space conscience while travelling and my mp3 player never liked
ogg. My Bash skills are near zero and what I have working is messy from
searching on the internet for a solution. Hopefully you can help me
modify it to use Sox rather than ffmpeg. (And possibly make it so I can
simply call the command, rather than the more complex command I have to
use currently.)
To start the process I currently use this command in the terminal:
find -type d -exec ~/bin/flac2mp3 "{}" \;
The previously working contents of flac2mp3 was:
#!/bin/bash
if [ -d "${1}" ] ; then
cd "${1}" && for f in *.flac; do ffmpeg -i "$f" -f wav - | lame -b
320 -m j -q 0 - "${f%.flac}.mp3"; done
fi
I tried quickly to edit.
#!/bin/bash
if [ -d "${1}" ] ; then
cd "${1}" && for f in *.flac; do sox "$f" -b 16 "${f%.flac}.wav" - |
lame -b 320 -m j -q 0 - "${f%.flac}.mp3"; done
fi
Seems to think the .wav is an incoming file for concatenation.
#!/bin/bash
if [ -d "${1}" ] ; then
cd "${1}" && for f in *.flac; do sox "$f" -b 16 "${f%.flac}.wav" |
lame -b 320 -m j -q 0 - "${f%.flac}.mp3"; done
fi
Gives me all the wav files converted but doesn't then pass them onto the
lame section of the command.
I'm sure this must be really simple! (As much having it all in one
single script called by just executing the file!)
Regards, Dale.
I'm getting the same crash as recently reported in the bug tracker when trying
to import a MIDI file. Do I have to create an account just to say 'me too'?
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Hi *,
I built a recording machine. Embedded Mainboard with Intel Atom, 4GB Ram, 128GB SSD, 80W PSU and a RME HDSP9652. Running Linux Mint 17 with KX Studio low latency kernel (dont know which version right now).
The first time recording 24Channels at 24/48k went flawless.
The second time I thought my ADCs were broken. Scrambled and distorted I/Os in hdspmixer/ardour3. After restarting, it would not boot anything. Took it from the Studio home. At home it was working fine again.
The third time everything flawless.
The fourth time: see second time.
Ok, calling it an odd behavior is true in two sences. But I refuse to believe, it is only working in odd times. :-D
I experienced something similar, when the PCI connector wasnt mounted properly. But this time it is!
Double checked!
What might cause such behavior?
PSU to small?
Intel CPU stupid?
VooDoo?
I am a little helpless...
Best regards,
Ck
Am 26. Oktober 2014 13:00:01 MEZ, schrieb linux-audio-user-request(a)lists.linuxaudio.org:
>Send Linux-audio-user mailing list submissions to
> linux-audio-user(a)lists.linuxaudio.org
>
>To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>or, via email, send a message with subject or body 'help' to
> linux-audio-user-request(a)lists.linuxaudio.org
>
>You can reach the person managing the list at
> linux-audio-user-owner(a)lists.linuxaudio.org
>
>When replying, please edit your Subject line so it is more specific
>than "Re: Contents of Linux-audio-user digest..."
>
>
>Today's Topics:
>
> 1. Re: Music made with linux: a773/sportin' life (Will Godfrey)
> 2. Re: zita-njbridge > zita-nabridge ? (Carlos sanchiavedraz)
> 3. Re: A survey if you don't mind - results (Will Godfrey)
> 4. Re: A survey if you don't mind - results (Carlos sanchiavedraz)
> 5. Re: A survey if you don't mind - results (Robin Gareus)
> 6. Re: A survey if you don't mind - results (Len Ovens)
> 7. Re: A survey if you don't mind - results (Will J Godfrey)
> 8. Re: A survey if you don't mind - results (Len Ovens)
> 9. Re: A survey if you don't mind - results (Brett McCoy)
> 10. Re: zita-njbridge > zita-nabridge ? (Jeremy Jongepier)
> 11. Re: A survey if you don't mind - results (Ralf Mardorf)
> 12. Re: zita-njbridge > zita-nabridge ? (Rapha?l Mouneyres)
> 13. recordmydesktop stereo (Atte)
> 14. Re: recordmydesktop stereo (Conor Mc Cormack)
>
>
>----------------------------------------------------------------------
>
>Message: 1
>Date: Sat, 25 Oct 2014 16:02:29 +0100
>From: Will Godfrey <willgodfrey(a)musically.me.uk>
>To: linux-audio-user(a)lists.linuxaudio.org
>Subject: Re: [LAU] Music made with linux: a773/sportin' life
>Message-ID: <20141025160229.66407bf0@debian>
>Content-Type: text/plain; charset=US-ASCII
>
>On Fri, 24 Oct 2014 20:24:15 +0200
>Atte <atte(a)youmail.dk> wrote:
>
>> Hi
>>
>> New track out, hope you enjoy!
>>
>> http://soundcloud.com/a773/sportin-life
>>
>> In case you wanna download and/or support:
>> http://a773.bandcamp.com/track/sportin-life
>>
>Really enjoyed this bright lively composition. lots of interesting
>variations,
>but a consistent thread all through.
>
>--
>Will J Godfrey
>http://www.musically.me.uk
>Say you have a poem and I have a tune.
>Exchange them and we can both have a poem, a tune, and a song.
>
>
>------------------------------
>
>Message: 2
>Date: Sat, 25 Oct 2014 17:33:36 +0200
>From: Carlos sanchiavedraz <csanchezgs(a)gmail.com>
>To: raf <rmouneyres(a)gmail.com>
>Cc: linux-audio-user Users <linux-audio-user(a)lists.linuxaudio.org>
>Subject: Re: [LAU] zita-njbridge > zita-nabridge ?
>Message-ID:
> <CA+O6s5ziGUZ43TcLCzXBytGXAutk+FtQVEi0PEhd5NtUgNmXJA(a)mail.gmail.com>
>Content-Type: text/plain; charset="iso-8859-1"
>
>2014-10-24 21:10 GMT+02:00 raf <rmouneyres(a)gmail.com>:
>
>>
>> >> I have a situation in mind : in a LAN a central computer running
>jack
>> server with a lot of audio I/O, and lightweight clients with stereo
>I/O.
>> >> I'd like those clients to be able to send a receive 2 channels of
>audio
>> to the central jack server, lowest latency possible.
>> >>
>> >> With the zita-njbridge and jack server running on the lightweight
>> clients, it works, but considering i don't need the jack server on
>the
>> lightweight clients (no effect plugins, no routing, no synths...)
>could a
>> zita-nabridge exist to capture/override the streams from the client's
>> embedded alsa device ?
>> >
>> > In theory such a thing could exist, it would be an ALSA device
>> > that receives and/or sends audio using the njbridge protocol.
>> > No resampling would be needed as it would be the 'master'
>> > device on the local system.
>> >
>> > In practice that is not very likely to happen, the reason
>> > being that interfacing to Jack is so much more easy than
>> > writing an ALSA driver. Also, passing via Jack does not
>> > add any latency, and in most cases users will want the
>> > flexibility it provides.
>>
>> thanks for the answer, I was expecting this, but hadn't measures the
>> difference between the jack client and alsa driver.
>> So now it looks like I need to learn how to cross compile jack for
>various
>> ARM devices to have it on the lightweight clients :/
>>
>> Rapha?l
>> _______________________________________________
>> Linux-audio-user mailing list
>> Linux-audio-user(a)lists.linuxaudio.org
>> http://lists.linuxaudio.org/listinfo/linux-audio-user
>>
>
>I've thought about a similar idea sometime in the past: A distributed
>audio
>network with thin clients/raspberrys for a home studio or distributed
>via
>some network. I'd be interested in following whatever progress you
>make.
>
>About that of "distributed band" I red a little about programs to jam
>via
>internet: Netjack, Ninjam, Midikit.
>
>--
>
>C. sanchiavedraZ:
>* NEW / NUEVO: www.sanchiavedraZ.com
>* Musix GNU+Linux: www.musix.es
>
Hi,
I like to speed up my boottime of my Mac Mini / Instrument module. I'm
thinking about buying a SSD drive.
How are SSD drives performing these days and how reliable are they? What
about a SSDH drive?
Regards,
Dirk
On Sun, Nov 16, 2014 at 08:56:37AM -0500, Paul Davis wrote:
> you'll find more discussion here; https://community.ardour.org/node/8496
Ah, hmm. It seems maybe this isn't the disaster it seemed. I had a memory
that VST support on Linux was shaky and on dodgy legal footing, but perhaps
it's moved on or I've had the wrong end of the stick.
> fairly sure that he was referring to this post:
> https://community.ardour.org/node/8288
My subscription lapsed a while back and there were some problems getting
it set up again for some reason I forget. At the time I have up but the long
and short of it though is that I *should* be subscribing and so I've tried
again and it seems to be working now.
bjb
Hello,
Very minor thing. Wondering why, in the configuration, interface
option, not all interfaces are directly listed in the drop down box.
Instead, one must press the arrow at the right of the box to see more
interface options. Wouldn't it be simpler to just list them all in one
drop-down box ?
I asked this on the liquidsoap list but nobody there seems too interested in this subject.
I have DJ's streaming from various parts of North America and Europe to a Linux VPS in New York.
Many of the DJ's have periodic dropouts where their sound goes away, and its annoying.
The liquidsoap software logs the buffer status at every packet received.
So I've been logging this stuff. I see a strange pattern.
The DJ's streaming from California who have good internet connectivity (some don't; I'm not including them in my data) show a buffer that steadily decreases, until it drops out and there are 2 seconds of the backing jukebox playing, until the buffer catches up.
The DJ's from New York show perfectly solid buffer status. The buffer fills, and stays filled for hours. One DJ even has a buffer that increases in size as his show goes along!
A separate station I've set up, streams from a Linux PC in California to a Linux VPS in California... rock-solid buffer status.
Now, I've been told that the buffer problem is due to clock synchronization on soundcards. Some soundcards on't stream exactly at 44.1khz, they're like 44.0Khz, and that causes a buffer that gradually declines in size, as it's been explained to me.
But!
Why would everyone in California have "slow" soundcards and everyone in New York have "fast" ones? That seems very unlikely.
I'm assuming everyone is using random soundcards, mostly SoundBlasters or HDAINtels. The mix of OS's is very random: two Linux guys, the rest mostly Winbloze, one or two Macs, and there's no geographic pattern. Almost everyone is using the same GPL streaming software-- The BUTT-- though one is using Nicecast I think.
I'm told that the power grid AC mains sync is not uniform across North America and that the east coast is at 60.1Hz or similar, the west coast at 59.9Hz, but that's irrelevant, it's all rectified/regulated +5V by the time the soundcard (and its clock crystal) sees it.
The obvious answer here is just network latency, since one set of DJ's are streaming locally and the others are traversing a continent. And I might just have to do some kind of time-stretching to avoid dropouts.
But what about this business with the sound cards? Is that a red herring?
-ken
