sound-recorder -c 1 -s 8 -b 8
Typical voice recording command — mono recording with samplerate of 8 kHz and 8 bits per sample:
sound-recorder -c 1 -s 8 -b 8
sound-recorder - Direct-to-disk recording and play-back programs.
Available as a Debian package. Not found on the Internet.
sound-recorder is a direct-to-disk recording program. It uses the recording input from the mixer on your soundcard and records it to a file.
cdsound-recorder is an enhancement to sound-recorder which makes it easy to record tracks or samples from a cdrom to a file.
With the play-sample program you can play the recorded audio or play all other non-compressed and compressed wave-files.
$ sound-recorder
Sound Recorder version 0.06 (Build on Oct 28 2005)
Copyright (C) 1997-2000 by B. Warmerdam under GPL.
This program is free software and comes with ABSOLUTELY NO WARRANTY.
Options:
-c Number of channels [1/2]
-s Samplerate of the recording
-b Bits per sample [8/16]
-k Keep going if destination file already exists (overwrite)
-P Use a higher priority for recording thread
-A Audio device (default /dev/dsp)
-e Execute this statement after recording (eg 'rm $file')
-S The recording time (mm:ss)
-f Output format [wav/pcm/cdr/ima3/ima4/ima5]
-h This information
documented on: 2008-06-22
*Tags*: sound recorder
Newsgroups: comp.os.linux.misc Date: 2002-06-19 09:05:32 PST
> Can you tell me what software I can use to record sound from the > microphone? > I will have to record the English sentencies into files. Lot's of them. > Is there available software to help me to do this and you know about it? > Thank you for your help.
you'll need an audio mixer to set the input levels (something like aumix), and you can use gramofile for recording to files. If there are large breaks between the sentences, then you can probably tune gramofile to break up the sentences in one large recorded file into many small files. If gramofile isn't good enough for splitting the files, then the `snd' sound editor will be of help.
There are many other programs that can help you (see http://sound.condorow.net/ for a comprehensive list); the ones I listed are just the ones I'm used to.
Rob Komar
Newsgroups: alt.os.linux.suse Date: 2002-09-05 21:07:39 PST
|Most users just want to play sound. But I would like to be able to record |sound from the microphone as well.
Works for me, I use arecord. Make sure the microphone is the right type for the sound card input (mine is actually connected to a radio tuner). Some sound card inputs require higher levels than the microphone can supply. Also remember to move up the microphone and input sliders on the mixer panel.
and, additionally, recording can be switched on and off in the mixer! good luck, it works for me, i.e. gramofil for recording. Simon
> So far, I haven't been able to do it. I've tried Broadcast2000 and the > command line tools 'wavrec' and 'rec'. All of them create files filled > with silence.
Run aumix, or some other mixer. Increase all what has to do with Line (Line 1, Line 2, Record Level, Line in, Digital in …), even if you have only one line input.
Go to terminal, type:
wavrec -S -s44100 -t(time) music1.wav
Record for few seconds, and then try to play the file.
If this doesnt work, try to increase Input Gain in mixer. But dont increase it too much, because you will get distorted sound…
If you want to play around with the waveform of the file, record a bunch of stuff and mix it, or divide into separates tracks, I've found audacity to be great:
It DOES require a fair bit of HD space and CPU (or lots of patience ;-) to do this, but it's great fun and a slick way to get old tracks off of cassette or vinyl onto CD.
Rick D.
Gnomemeeting: I have got it up and running under RH7.3 - both gnomemeeting and openh323 tools after fixing "some" problems:
the sound card on the motherboard have not supported full duplex operation (new card installed)
I had to kill sound demon (artsd) to make sound device available to h323 software Use "fuser -uv /dev/dsp" to find out what uses the device
I had to fix firewall rules incoming tcp connections to 1720 port and both way transmission ports 5000:5003
Voice recording: I recorded voice using openam (h323 answering machine) from http://www.openh323.org/
Andrzej
From FAQ: What does OpenAM do? http://www.de.openh323.org/fom-serve/cache/63.html
OPenAM is an answering machine for the H.323 protocol. It will accept incoming H.323 connections and record them to disk. A very simple user interface exists using the DTMF tone indications to get messages played back.
> I recorded voice using openam (h323 answering machine)
or Sound Studio (run studio) that comes with your SuSE.
/Sakis
Newsgroups: alt.os.linux.mandrake Date: 2002-10-28 06:01:26 PST
I have a Creative Esoniq es1370 setup under alsa, which can play music and sound correctly.
I've checked the mixer settings, and tried arecord, grecord, rec, and sox, but simply cannot get sound to record.
ie
sox -r 44100 -t ossdsp -w -s /dev/dsp -t wav /home/test.wav rec test1.wav rec --channels=2 --rate=44100 test.aiff even cat /dev/dsp >myfile.au doesn't work
I'm stumped, and am desperately needing it for GnomeMeeting.
Oh, and yes I did a `killall artsd` to prevent kde from interfering, but get the same results in gnome.
Newsgroups: alt.os.linux.mandrake Date: 2002-10-30 07:14:05 PST
> I downloaded and tried it, without success. The wav file was created, > but no sound.
I've got a 1371, but they're probably similar… Maybe not. Looking through your list, I don't see a "gain" control.
On my ens1371, there's a channel called IGain in kmix…
Let me verify… Yes. No IGain, no sound. I have to have that turned up to record anything on any channel.
(You might check out "Sound Studio." It's a nifty front-end for sox that works pretty well. It has a VU meter, which is what I just used to check this. Instant feedback… If the meter isn't moving, you're not recording.)
Um, you have these channels (filtering out all the crap):
Simple mixer control 'Master',0 Simple mixer control 'Master Mono',0 Simple mixer control 'PCM',0 Simple mixer control 'PCM',1 [...]
> Let's look. You're running the OSS/Free kernel module, which hasn't changed > since 1-31-2001... It says: > > * Module command line parameters: > * joystick if 1 enables the joystick interface on the card; but it still > * needs a driver for joysticks connected to a standard IBM-PC > * joyport. It is tested with the joy-analog driver. This > * module must be loaded before the joystick driver. Kmod will > * not ensure that. > * lineout if 1 the LINE jack is used as an output instead of an input. > * LINE then contains the unmixed dsp output. This can be used > * to make the card a four channel one: use dsp to output two > * channels to LINE and dac to output the other two channels to > * SPKR. Set the mixer to only output synth to SPKR. > * micbias sets the +5V bias to the mic if using an electretmic. > > I don't actually know how you'd make use of this information. One of the > things I've never really had to do. I think you can pass these parameters > to the driver via /etc/modules.conf somehow. BUT none of the parameters > are likely to do you any good anyway, so that's a dead end. > > > I'm at the end of the road. It looks like the card will have to join > > my 'Can't use with linux' stockpile, which now consists of a scanner > > and webcam,...and soundcard. > > I still say you need to turn up the input gain. It's almost definitely the > problem, but I don't know what else to suggest.
Unbelievable. adding this to /etc/modules.conf got it going
options es1370 joystick=1 micbias=1
micbias=1 is supposed to be an option for an 'Electretmic', whatever that is. Talk about fickle ! Anyway, I'm pleased as punch ! Thanks all for helping me work this through
Mark
A final note in case anyone else uses this posting to troubleshoot.
My card is a Creative Esoniq AudioPCI, not a SB which some of the es1370's are based.
Using the es1370 module means using oss. Anything I tried with Alsa snd-ens1370 didn't enable sound recording, though I'm still on the lookout for a fix there.
Mark
> Unbelievable. > adding this to /etc/modules.conf got it going > options es1370 joystick=1 micbias=1
I have tones of 1370's and they work fine, even with microphones.
> micbias=1 is supposed to be an option for an 'Electretmic', whatever that is.
An "electretmic" sounds like they mean a carbon microphone - one that works by changing resistence or some other electric characteristic (dielectric .) in response to sound, and doesn't generate electricity of its own. Yes, you would need to add a bias voltage! Otehrwise you would get sound causing 1% variations in 0V.
Get a proper microphone?
Peter
> An "electretmic" sounds like they mean a carbon microphone - one that
I have an "electret" mic, and it takes a battery. No battery, no sound. It also works about 1000 times better than any "condenser" mic I've ever used, even though it's just some cheap POS from Ripoff Shark (where you can buy yesterday's technology at tomorrow's prices, and if you stop in to buy some diodes they won't let you leave until you buy three cellular phones and satellite TV…)
Michael McIntyre
Many types of microphones require power to operate, as a general rule these types are described as condenser microphones. The power is used for internal pre-amplifiers and polarizing microphone capsules. If internal batteries are to be avoided then the only solution is to supply the power via the microphone signal cable.
Newsgroups: alt.os.linux Date: 2000/08/13
> I wondered yet whether there is a graphical application available > to record sound?
This is the place to look for Linux sound apps:
Erik
Take a look at this one. It's a script application, but it records and plays several formats. Runs on Caldera 2.4 here.
Ken Moffat
Newsgroups: aus.computers.linux, comp.os.linux.misc Date: 2002-11-17 14:32:33 PST
|I have a large collection of old LP's that I want to copy to CD, and |want to use Linux to do it if possible. | |Any suggestions on what is the best way (ie: software wise) to do |this with Linux? | |I dont mind if it is command-line or GUI, as long as I can get some |kind of recoding level meter on the screen, or at least find some |way of stopping clipping of the signal (which sounds terrible in a |digital environment).
I just finished doing my collection. I used arecord from ALSA. As for setting the level, I did it by trial and error, I recorded a few passages and then measured the headroom with sox. It if was too close to 1.0, I lowered the gain. I use something like 1.2. It turns out that pretty much all records use the maximum gain so I only had to do this calibration once and then save the gain settings to an aumix config file. Then I used gramofile to split them into tracks (semi-automatically) and turned them to MP3s with lame.
>pretty much all records use the maximum gain so I only had to do this >calibration once and then save the gain settings to an aumix config >file. Then I used gramofile to split them into tracks >(semi-automatically) and turned them to MP3s with lame.
I found gramofile introduced artifacts on my system - skips and repeats.
I connect my turntable to a pre-amp then into the Soundblaster card.
I record using sox, and split into tracks using audacity. I have to do it manually, using the audacity graph to see the breaks between tracks. I highlight the bits between the breaks, and save the selected bit as WAV, then repeat.
I encode using bladeenc, I could do it in one hit in audacity, but didn't have the stuff it needed to do that and haven't bothered getting it.
I recommend a dedicated sound card like the Soundblaster, I have tried a couple of inbuilt audio chips and they just were not as good.
Zebee
|I found gramofile introduced artifacts on my system - skips and |repeats.
Hmm, a couple of my friends have reported that gramofile can also record at the wrong speed, I haven't experienced this as I don't use its record feature; as mentioned, I use bare arecord. I also don't do any filtering. I live with the pops and clicks.
The most tedious part of track splitting is finding the split points. For pop songs, gramofile does well, and I only need tweak its .tracks a bit. What it doesn't cope well with are: fade in/outs, quiet sections (e.g. classical music), concerts with no breaks, and scratchy records (which make its silence detection fail). When that happens, I listen to the whole side using xmms (or the wav player of your choice) and note the split points. It's not so bad, I use the slider to seek around. But I found that the declared lengths on albums are only vaguely related to the actual length. I generate the metadata file using a quick and dirty script I wrote which takes a list of split points from the command line and generates a .tracks file to feed into gramofile. I split using gramofile, move the tracks to their own directory, and run lame over the lot. After that I enter the album info using a command line tool called mpi and then use xmms to enter the song titles. I run mpi again to rename the files by their title instead of processed01.mp3 and I'm done.
I estimate a real time to play time ratio of about 1.4. But I also get suckered to listening to songs I haven't heard for ages, so it's not an unpleasant task. :-)
>The most tedious part of track splitting is finding the split points.
Yeah. Very easy using audacity as you get a visual guide, as it shows the frequency graph.
No noise, no waves…. you can play that section to be sure.
Zebee
> I record using sox, and split into tracks using audacity. I have to
any reason why you use sox, rather than record straight into audacity?
audacity's a damn good digital editor, i reckon - particularly as it's fairly early into its development. it only seems to have a few very minor bugs. i'm going to have a go at installing it on a couple of machines at the community radio station i work for. they use pro-tools on the main production machine, and quickedit on the news editor, but i doubt much of the production work done there uses any pro-tools/quickedit functions that audacity can't do. if development keeps going, audacity looks set to become the "gimp" of digital editors.
will
>any reason why you use sox, rather than record straight into audacity?
Because I started doing it before I got audacity, and never changed…
>audacity can't do. if development keeps going, audacity looks set to >become the "gimp" of digital editors.
It's got a lot of good and simple features, and is powerful and easy to use.
Zebee
> Any suggestions on what is the best way (ie: software wise) to do > this with Linux?
Use the line-in input to your sound card. They you could use gnoise to copy from the input to files. Unless you copy over one track at a time, it will take a long time (and need much disk space) to edit the sound files.
With gnoise and its interpolate function you can also get rid of the worst of the clicks, but someone really needs to come up with a good automatic program to find and eliminate the clicks. (The RIAA correction on the output to the record player makes this more difficult, as those clicks get averaged out by the equalisation, which makes them harder to find and eliminate.)
Newsgroups: comp.os.linux.misc Date: 2002-09-21 22:28:46 PST
> somebody know?
Erik de Castro Lopo
Newsgroups: aus.computers.linux,comp.os.linux.misc Date: 26 Nov 2002 18:17:09 GMT
> I have a large collection of old LP's that I want to copy to CD, and > want to use Linux to do it if possible.
> Any suggestions on what is the best way (ie: software wise) to do > this with Linux?
I wrote a little utility to do the recording, it doesn't require much:
open /dev/dsp
open output file
set parameters on /dev/dsp for CDROM style audio
read a block
write a block
loop
A few things I noticed:
For soundblaster, the ALSA drivers are better than the free OSS drivers because the ALSA ones seem to be less bothered by interrupts coming from other devices. Fundamentally though, PC DMA is rooted for sound recording because there is no guaranteed way to atomicly read the DMA pointer while DMA is happening — a hardware FIFO approach to sound recording is much better than DMA on PC architecture.
To reduce interrupt loading while recording, use a raw partition as the output device. You lose information about the output file size but you can keep that somewhere else and you are probably going to cut up the recording anyhow. Raw partition has much lower overhead than normal files because it doesn't update directories, inodes and all that.
To prevent bursts of hard-drive IO, open the output with the O_SYNC flag (see man 2 open) so that every block will write as a block and it won't defer writing. When writing to a raw partition, this gives excellent control over the hard drive behaviour by just adjusting the blocksize. Usually 1/2k, 1k, 2k or 4k blocks work best depending on your system.
Scsi discs usually have fewer interrupts and put less load on the CPU. This is important not because the CPU can't handle the load but because you need the CPU to be almost entirely unloaded in order to get real time response on the audio interrupts.
My system:
ad1816: TERRATEC SOUNDSYSTEM BASE 1 detected ad1816: ISAPnP reports 'TERRATEC SOUNDSYSTEM BASE 1' at i/o 0x530, irq 5, dma 1, 3 ad1816: AD1816 sounddriver Copyright (C) 1998 by Thorsten Knabe ad1816: io=0x530, irq=5, dma=1, dma2=3, clockfreq=33000, options=0 isadmabug=0
Running on a 200 MHz system and an Adaptec AIC-7871, it boots off IDE and runs the system off IDE but uses a dedicated SCSI drive just for sound recording. Recording an hour or more of continuous CD quality sound without missed samples works without a problem.
> I dont mind if it is command-line or GUI, as long as I can get some > kind of recoding level meter on the screen, or at least find some > way of stopping clipping of the signal (which sounds terrible in a > digital environment).
Yeah, getting levels right is always a pain. Most sound cards have several volume controls that all affect recording. I generally make sure I mute everything except the input that I want (to reduce noise) and crank the input gain right up then throttle it back with the recording gain. The Teratec Base-1 comes with crappy documentation but Analog Devices provides excellent documentation for the AD1816, basically this is the reason that I'm not using a soundblaster because there is no way to know what is really going on inside the soundblaster.
Tel