Re: audio recording levels
I have worked with audio before, and can confirm internal audio codecs are very good for... trash them. If quality is of any concern for you, just try another adapter, i.e. an inexpensive Behringer UCA 202 USB audio interface. I've tried it with great results on OpenBSD, and you can buy it in Europe for as low as 27eur in Thomann. Regards, and good luck. I've a PMR/O El 16/06/2010 15:05, Paul M escribis: On 16/06/2010, at 6:45 PM, Alexandre Ratchov wrote: On Wed, Jun 16, 2010 at 12:06:40AM +1200, Paul M wrote: On 15/06/2010, at 11:18 PM, Paul M wrote: On 15/06/2010, at 8:25 PM, Alexandre Ratchov wrote: On Mon, Jun 14, 2010 at 08:20:57AM -0600, Ted Roby wrote: Sound cards just get too much noise off the motherboard. well, it depends on the sound card; properly engineered cards don't get noise, including pci ones. It seems the best I can get with the built-in sound on this computer is about 36dB S/N. If the various input gain stages are not set quite Sorry, that's a typo - should be -46 dB that's not very good; 8-bit samples correspond to 48dB; I mean with 48dB S/N, only 8 higher bits are significant. Such a low S/N ratio makes me wonder if your cables, power supply or whatever are ok. BTW, how did you measure the S/N ratio? No, it's not. I measured by writing samples to a file then examining the file. For a source, I used a 440 Hz sine wave. With all the input gains set to minimum, the noise was negligible ( -60 db, which is the smallest I could measure). This is pretty much just the noise of the chip itself. I then tweaked the various input levels (preamp, input stage, adc) till the output of the preamp was just below clipping of the first input stage, and the samples in the file were also just below clipping. This gave me my signal level. Then I disconnected the input and repeated, this gave me my noise - shorting the input may give better results. I repeated this over and over to find the best balance between input stage gain and adc gain. I found the best result was to set the input stage as low as possible, and adjust the adc gain to give full output. (dont forget the input signal is as high as it can be without clipping) Update: Today I reduced the input gain stage to 0 (I didnt try this yesterday) and got about another 6-8 dB, so now I'm 50 dB. Jacob mentioned in and earlier mail that '0' for the input stages on this codec is 0dB gain. FWIW the adc gain is set to 96 (+12dB). Since my moise is measured with no input connected, this rules out any external problems such as cables. The power supply in the computer may be moisy, but I would guess that crappy circuit layout is more likely. (though I will try another power supply some time and see if it helps) paulm
Re: audio recording levels
On Tue, Jun 15, 2010 at 10:46:47AM +0200, Jan Stary wrote: Sound cards just get too much noise off the motherboard. well, it depends on the sound card; properly engineered cards don't get noise, including pci ones. Are some of them known to be better then others in this respect? any card that claims 100dB S/N should be ok, assuming the vendor is not lying. I've a m-audio delta 1010lt, and a esi julia that don't get noise while they are plugged on a machine I use as a wireless access point (which btw is probably the most stupid setup). -- Alexandre
Re: audio recording levels
On Wed, Jun 16, 2010 at 12:06:40AM +1200, Paul M wrote: On 15/06/2010, at 11:18 PM, Paul M wrote: On 15/06/2010, at 8:25 PM, Alexandre Ratchov wrote: On Mon, Jun 14, 2010 at 08:20:57AM -0600, Ted Roby wrote: Sound cards just get too much noise off the motherboard. well, it depends on the sound card; properly engineered cards don't get noise, including pci ones. It seems the best I can get with the built-in sound on this computer is about 36dB S/N. If the various input gain stages are not set quite Sorry, that's a typo - should be -46 dB that's not very good; 8-bit samples correspond to 48dB; I mean with 48dB S/N, only 8 higher bits are significant. Such a low S/N ratio makes me wonder if your cables, power supply or whatever are ok. BTW, how did you measure the S/N ratio? -- Alexandre
Re: audio recording levels
On 16/06/2010, at 6:45 PM, Alexandre Ratchov wrote: On Wed, Jun 16, 2010 at 12:06:40AM +1200, Paul M wrote: On 15/06/2010, at 11:18 PM, Paul M wrote: On 15/06/2010, at 8:25 PM, Alexandre Ratchov wrote: On Mon, Jun 14, 2010 at 08:20:57AM -0600, Ted Roby wrote: Sound cards just get too much noise off the motherboard. well, it depends on the sound card; properly engineered cards don't get noise, including pci ones. It seems the best I can get with the built-in sound on this computer is about 36dB S/N. If the various input gain stages are not set quite Sorry, that's a typo - should be -46 dB that's not very good; 8-bit samples correspond to 48dB; I mean with 48dB S/N, only 8 higher bits are significant. Such a low S/N ratio makes me wonder if your cables, power supply or whatever are ok. BTW, how did you measure the S/N ratio? No, it's not. I measured by writing samples to a file then examining the file. For a source, I used a 440 Hz sine wave. With all the input gains set to minimum, the noise was negligible ( -60 db, which is the smallest I could measure). This is pretty much just the noise of the chip itself. I then tweaked the various input levels (preamp, input stage, adc) till the output of the preamp was just below clipping of the first input stage, and the samples in the file were also just below clipping. This gave me my signal level. Then I disconnected the input and repeated, this gave me my noise - shorting the input may give better results. I repeated this over and over to find the best balance between input stage gain and adc gain. I found the best result was to set the input stage as low as possible, and adjust the adc gain to give full output. (dont forget the input signal is as high as it can be without clipping) Update: Today I reduced the input gain stage to 0 (I didnt try this yesterday) and got about another 6-8 dB, so now I'm 50 dB. Jacob mentioned in and earlier mail that '0' for the input stages on this codec is 0dB gain. FWIW the adc gain is set to 96 (+12dB). Since my moise is measured with no input connected, this rules out any external problems such as cables. The power supply in the computer may be moisy, but I would guess that crappy circuit layout is more likely. (though I will try another power supply some time and see if it helps) paulm
Re: audio recording levels
On Mon, Jun 14, 2010 at 08:20:57AM -0600, Ted Roby wrote: Sound cards just get too much noise off the motherboard. well, it depends on the sound card; properly engineered cards don't get noise, including pci ones. -- Alexandre
Re: audio recording levels
Sound cards just get too much noise off the motherboard. well, it depends on the sound card; properly engineered cards don't get noise, including pci ones. Are some of them known to be better then others in this respect?
Re: audio recording levels
On 15/06/2010, at 8:25 PM, Alexandre Ratchov wrote: On Mon, Jun 14, 2010 at 08:20:57AM -0600, Ted Roby wrote: Sound cards just get too much noise off the motherboard. well, it depends on the sound card; properly engineered cards don't get noise, including pci ones. It seems the best I can get with the built-in sound on this computer is about 36dB S/N. If the various input gain stages are not set quite carefully, it gets much worse. Rf pickup in the input leads is high too. paulm
Re: audio recording levels
On 15/06/2010, at 11:18 PM, Paul M wrote: On 15/06/2010, at 8:25 PM, Alexandre Ratchov wrote: On Mon, Jun 14, 2010 at 08:20:57AM -0600, Ted Roby wrote: Sound cards just get too much noise off the motherboard. well, it depends on the sound card; properly engineered cards don't get noise, including pci ones. It seems the best I can get with the built-in sound on this computer is about 36dB S/N. If the various input gain stages are not set quite Sorry, that's a typo - should be -46 dB carefully, it gets much worse. Rf pickup in the input leads is high too. paulm
Re: audio recording levels
On Jun 14 11:37:52, Paul M wrote: I have a large amount of analog audio I need to digitize and naturaly want to ensure best transfer quality. So I need to set the analog level at the input to the adc as high as possible without clipping. It is good practice to leave a little headroom (say, 6dB) for further processing. You might want to do some noise reduction, compressing, whatever, and the effects will clip if the signal already is saturated. Also, the input to the adc is not all there is to it; there are other mixer settings that affect the signal that will eventually end in your file.wav Ideally, I'll get the workstation hardware set to certin defaults, then adjust the incomming audio as required. There are no defaults. Your analog inputs can (and most probably will) vary greatly. This leads to a couple of questions: Are there (typicaly) any variable gain stages in the analog input path in the computer. Yes. 'mixerctl -a' will shouw you how the azalia 'widgets' are interconnected on your codec. Mixerctl -av (full output below) shows a node called 'record.adc'. It seems reasonable that this might opperate on the analog input to the adc. However there is also 'record.volume', though I would assume this operates on the mixed digital signals at the end of the chain. record.volume Amplifier gain control for widgets listed in record.volume.slaves. Also: a lot of the gain stages have defaults of 120.120. Would it be reasonable to assume that this is the 0 gain setting? What's a 0 gain setting? I believe Jacob Meuser has work going on to make the numbers on the azalia knobs correspond to actual decibels, but I don't know if it's current yet. bios0: ASUSTeK Computer INC. P5QPL-AM azalia0 at pci0 dev 27 function 0 Intel 82801GB HD Audio rev 0x01: apic 2 int 21 (irq 5) azalia0: codecs: Realtek/0x0887 audio0 at azalia0 It would be my guess that this is the audio chip that's integrated with the Asus P5QPL-AM motherboard. If you are really after best transfer quality, you might want to use something else in the first place.
Re: audio recording levels
On 14/06/2010, at 12:49 PM, Jacob Meuser wrote: On Mon, Jun 14, 2010 at 11:37:52AM +1200, Paul M wrote: I have a large amount of analog audio I need to digitize and naturaly want to ensure best transfer quality. So I need to set the analog level at the input to the adc as high as possible without clipping. Ideally, I'll get the workstation hardware set to certin defaults, then adjust the incomming audio as required. This leads to a couple of questions: Are there (typicaly) any variable gain stages in the analog input path in the computer. varies depending on hardware, but often there is a gain at the input and a gain at the ADC. Looking at the mixerctl output again, this does appear to be the case. Mixerctl -av (full output below) shows a node called 'record.adc'. It seems reasonable that this might opperate on the analog input to the adc generically speaking, yes. However there is also 'record.volume', though I would assume this operates on the mixed digital signals at the end of the chain. no. record.volume is essenially an alias. on your hardware with the configuration you've posted, it's a shortcut for setting both record.adc and record.adc2. this is explained in azalia(4) (though maybe that info didn't make it into 4.6, the info in -current azalia(4) is mostly relevant even for 4.6). Also: a lot of the gain stages have defaults of 120.120. Would it be reasonable to assume that this is the 0 gain setting? no. unfortunately, the mixer interface, like a lot of audio(4) related stuff, is designed for consumer usage. so, we just have a range that is essentially 0-100% - it has no relevance to anything except the knob. truly the worst kind of knobs are those that have no outside meaning, but apparently people like this. *shrug* http://marc.info/?l=openbsd-techm=123101323408867w=2 Any thoughts appreciated. if you really want to know how to do this right, your best bet is to find the datasheet for your codec. now, your codec is a Realtek, which is common for azalia(4), and I happen to know them pretty well ... inputs.line=85,85 this is a 0 (0), 10 (85), 20 (170), 30 (255) dB gain on the line-in jack. values in () are the corresponding mixerctl values. record.adc=248,248 this is the ADC input gain. 0 dB should be around '88'. 0..255 here represents the hardware's -16.5 to 30 dB in 1.5 dB steps. these are the only gains on the recording path of your device. Thanks Jacob, this is gold. I'll try and find that datasheet, but what you've given me here is probably enough for me to work out what I need. Thanks again, and thanks to all who worked on giving the audio subsystem the openBSD love. paulm
Re: audio recording levels
On 14/06/2010, at 6:54 PM, Jan Stary wrote: On Jun 14 11:37:52, Paul M wrote: I have a large amount of analog audio I need to digitize and naturaly want to ensure best transfer quality. So I need to set the analog level at the input to the adc as high as possible without clipping. It is good practice to leave a little headroom (say, 6dB) for further processing. You might want to do some noise reduction, compressing, whatever, and the effects will clip if the signal already is saturated. I'll leave enough headroom to allow for the highest peaks, but I'm not planning on doing any additional processing during the initial conversion. I can attenuate the signal later if I add any post processing. Also, the input to the adc is not all there is to it; there are other mixer settings that affect the signal that will eventually end in your file.wav Ideally, I'll get the workstation hardware set to certin defaults, then adjust the incomming audio as required. There are no defaults. Your analog inputs can (and most probably will) vary greatly. I dont really want to futz with the audio hardware once it's set up, so these 'defaults' would be such that a certain input signal level will produce a clean digital file with no clipping and good dynamic range. I'll then use a preamp with decent VU meters to ensure the signal sent to the computer is optimal. This leads to a couple of questions: Are there (typicaly) any variable gain stages in the analog input path in the computer. Yes. 'mixerctl -a' will shouw you how the azalia 'widgets' are interconnected on your codec. Mixerctl -av (full output below) shows a node called 'record.adc'. It seems reasonable that this might opperate on the analog input to the adc. However there is also 'record.volume', though I would assume this operates on the mixed digital signals at the end of the chain. record.volume Amplifier gain control for widgets listed in record.volume.slaves. Thanks, I wondered about the slaves. Also: a lot of the gain stages have defaults of 120.120. Would it be reasonable to assume that this is the 0 gain setting? What's a 0 gain setting? Unity gain, or 0dB gain. Signal level out = signal level in (for that stage). I believe Jacob Meuser has work going on to make the numbers on the azalia knobs correspond to actual decibels, but I don't know if it's current yet. bios0: ASUSTeK Computer INC. P5QPL-AM azalia0 at pci0 dev 27 function 0 Intel 82801GB HD Audio rev 0x01: apic 2 int 21 (irq 5) azalia0: codecs: Realtek/0x0887 audio0 at azalia0 It would be my guess that this is the audio chip that's integrated with the Asus P5QPL-AM motherboard. If you are really after best transfer quality, you might want to use something else in the first place. Good point, thanks for the reality check. Most of what I have to do is not the best quality anyway, but that doesn't mean I'm happy to introduce unnecessary generation loss by being sloppy. There is some though that is very good, so that may well need something better. paulm
Re: audio recording levels
On Mon, Jun 14, 2010 at 2:29 AM, Paul M l...@no-tek.com wrote: On 14/06/2010, at 6:54 PM, Jan Stary wrote: It would be my guess that this is the audio chip that's integrated with the Asus P5QPL-AM motherboard. If you are really after best transfer quality, you might want to use something else in the first place. Good point, thanks for the reality check. Most of what I have to do is not the best quality anyway, but that doesn't mean I'm happy to introduce unnecessary generation loss by being sloppy. There is some though that is very good, so that may well need something better. I heard the Griffin iMic was to be discontinued, but mine is supported under OpenBSD. Your best bet for clean audio is a USB-attached device. Sound cards just get too much noise off the motherboard.
Re: audio recording levels
On 15/06/2010, at 2:20 AM, Ted Roby wrote: On Mon, Jun 14, 2010 at 2:29 AM, Paul M l...@no-tek.com wrote: On 14/06/2010, at 6:54 PM, Jan Stary wrote: It would be my guess that this is the audio chip that's integrated with the Asus P5QPL-AM motherboard. If you are really after best transfer quality, you might want to use something else in the first place. Good point, thanks for the reality check. Most of what I have to do is not the best quality anyway, but that doesn't mean I'm happy to introduce unnecessary generation loss by being sloppy. There is some though that is very good, so that may well need something better. I heard the Griffin iMic was to be discontinued, but mine is supported under OpenBSD. Your best bet for clean audio is a USB-attached device. Sound cards just get too much noise off the motherboard. Interesting. I have one of those kicking around somewhere. I'll have to dig it out. Thanks paulm
audio recording levels
I have a large amount of analog audio I need to digitize and naturaly want to ensure best transfer quality. So I need to set the analog level at the input to the adc as high as possible without clipping. Ideally, I'll get the workstation hardware set to certin defaults, then adjust the incomming audio as required. This leads to a couple of questions: Are there (typicaly) any variable gain stages in the analog input path in the computer. Mixerctl -av (full output below) shows a node called 'record.adc'. It seems reasonable that this might opperate on the analog input to the adc. However there is also 'record.volume', though I would assume this operates on the mixed digital signals at the end of the chain. Also: a lot of the gain stages have defaults of 120.120. Would it be reasonable to assume that this is the 0 gain setting? Any thoughts appreciated. paulm molly:/home/paul mixerctl -a record.adc_mute=off record.adc=248,248 record.adc2_mute=off record.adc2=120,120 inputs.mix_source=mic,mic2,line-in,hp,line inputs.mix_mic=120,120 inputs.mix_mic2=120,120 inputs.mix_line-in=248,248 inputs.mix_hp=120,120 inputs.mix_line=120,120 outputs.mix2=120,120 inputs.mix2_source=dac,mix outputs.mix3=120,120 inputs.mix3_source=dac2,mix outputs.mix4=120,120 inputs.mix4_source=dac3,mix outputs.mix5=120,120 inputs.mix5_source=dac4,mix outputs.line_source=mix2 outputs.line_mute=off inputs.line=85,85 outputs.line_dir=output outputs.line_boost=off outputs.line_eapd=on outputs.mic_source=mix3 outputs.mic_mute=off inputs.mic=85,85 outputs.mic_dir=input-vr80 outputs.mic_boost=off outputs.mic2_source=mix8 outputs.mic2_mute=off inputs.mic2=85,85 outputs.mic2_dir=input-vr80 outputs.mic2_boost=off outputs.line-in_source=mix4 outputs.line-in_mute=off inputs.line-in=85,85 outputs.line-in_dir=input outputs.line-in_boost=off outputs.hp_source=mix5 outputs.hp_mute=off inputs.hp=85,85 outputs.hp_dir=output outputs.hp_boost=off outputs.SPDIF_source=dig-dac record.adc2_source=mic,mic2,line-in,hp,line,mix record.adc_source=mic,mic2,line-in,hp,line,mix outputs.mix8=120,120 inputs.mix8_source=dac5,mix outputs.line_sense=plugged outputs.mic_sense=plugged outputs.mic2_sense=unplugged outputs.line-in_sense=unplugged outputs.hp_sense=unplugged outputs.master=120,120 outputs.master.mute=off outputs.master.slaves=mix2,line record.volume=120,120 record.volume.mute=off record.volume.slaves=adc,adc2 outputs.mode=analog molly:/home/paul dmesg OpenBSD 4.6 (GENERIC.MP.RAID) #0: Mon Mar 29 11:47:19 NZDT 2010 r...@molly.no-tek.local:/molly0/src/sys/arch/i386/compile/ GENERIC.MP.RAID cpu0: Pentium(R) Dual-Core CPU E5400 @ 2.70GHz (GenuineIntel 686-class) 2.73 GHz cpu0: FPU,V86,DE,PSE,TSC,MSR,PAE,MCE,CX8,APIC,SEP,MTRR,PGE,MCA,CMOV,PAT,PSE36, CFLUSH,DS,ACPI,MMX,FXSR,SSE,SSE2,SS,HTT,TM,SBF,SSE3,MWAIT,DS- CPL,VMX,EST,TM2,CX16,xTPR real mem = 1037135872 (989MB) avail mem = 993542144 (947MB) mainbus0 at root bios0 at mainbus0: AT/286+ BIOS, date 09/21/09, BIOS32 rev. 0 @ 0xf0010, SMBIOS rev. 2.5 @ 0xf06c0 (50 entries) bios0: vendor American Megatrends Inc. version 0317 date 09/21/2009 bios0: ASUSTeK Computer INC. P5QPL-AM acpi0 at bios0: rev 0 acpi0: tables DSDT FACP APIC MCFG OEMB HPET GSCI SSDT acpi0: wakeup devices P0P2(S4) P0P3(S4) P0P1(S4) UAR1(S4) PS2K(S4) PS2M(S4) USB0(S4) USB1(S4) USB2(S4) USB3(S4) EUSB(S4) MC97(S4) P0P4(S4) P0P5(S4) P0P6(S4) P0P7(S4) P0P8(S4) P0P9(S4) SLPB(S4) acpitimer0 at acpi0: 3579545 Hz, 24 bits acpimadt0 at acpi0 addr 0xfee0: PC-AT compat cpu0 at mainbus0: apid 0 (boot processor) cpu0: apic clock running at 201MHz cpu1 at mainbus0: apid 1 (application processor) cpu1: Pentium(R) Dual-Core CPU E5400 @ 2.70GHz (GenuineIntel 686-class) 2.73 GHz cpu1: FPU,V86,DE,PSE,TSC,MSR,PAE,MCE,CX8,APIC,SEP,MTRR,PGE,MCA,CMOV,PAT,PSE36, CFLUSH,DS,ACPI,MMX,FXSR,SSE,SSE2,SS,HTT,TM,SBF,SSE3,MWAIT,DS- CPL,VMX,EST,TM2,CX16,xTPR ioapic0 at mainbus0: apid 2 pa 0xfec0, version 20, 24 pins acpihpet0 at acpi0: 14318179 Hz acpiprt0 at acpi0: bus 0 (PCI0) acpiprt1 at acpi0: bus -1 (P0P2) acpiprt2 at acpi0: bus -1 (P0P3) acpiprt3 at acpi0: bus 3 (P0P1) acpiprt4 at acpi0: bus 2 (P0P4) acpiprt5 at acpi0: bus 1 (P0P5) acpicpu0 at acpi0: PSS acpicpu1 at acpi0: PSS acpibtn0 at acpi0: SLPB acpibtn1 at acpi0: PWRB bios0: ROM list: 0xc/0xc600! 0xcc800/0x5800 cpu0: Enhanced SpeedStep 2723 MHz: speeds: 2700, 2003, 1603, 1203 MHz pci0 at mainbus0 bus 0: configuration mode 1 (bios) pchb0 at pci0 dev 0 function 0 Intel G41 Host rev 0x03 vga1 at pci0 dev 2 function 0 Intel G41 Video rev 0x03 wsdisplay0 at vga1 mux 1: console (80x25, vt100 emulation) wsdisplay0: screen 1-5 added (80x25, vt100 emulation) intagp at vga1 not configured azalia0 at pci0 dev 27 function 0 Intel 82801GB HD Audio rev 0x01: apic 2 int 21 (irq 5) azalia0: codecs: Realtek/0x0887 audio0 at azalia0 ppb0 at pci0 dev 28 function 0 Intel 82801GB PCIE rev 0x01: apic 2 int 16 (irq 11) pci1 at ppb0 bus 2 ppb1 at pci0 dev 28 function 1 Intel 82801GB PCIE rev
Re: audio recording levels
Hi, this you might already know, but good rule of thumb is to set the levels manually for each source (according to its dynamics), having peaks around -6dB to -10dB. If you have manual volume/gain control on your recording device/preamp, I'd set all levels in the computer to 80% of the scale and then control everything by hand, on the box. This probably doesn't apply to OpenBSD, but in Linux there have been serious distortion trouble with levels set to 100% on some older cards - the only solution is AFAIK try it out and use what your ears tell you. The good news is you can do other stuff while your LP's spinning and recording :-) -- Martin Pelikan
Re: audio recording levels
On Mon, Jun 14, 2010 at 11:37:52AM +1200, Paul M wrote: I have a large amount of analog audio I need to digitize and naturaly want to ensure best transfer quality. So I need to set the analog level at the input to the adc as high as possible without clipping. Ideally, I'll get the workstation hardware set to certin defaults, then adjust the incomming audio as required. This leads to a couple of questions: Are there (typicaly) any variable gain stages in the analog input path in the computer. varies depending on hardware, but often there is a gain at the input and a gain at the ADC. Mixerctl -av (full output below) shows a node called 'record.adc'. It seems reasonable that this might opperate on the analog input to the adc generically speaking, yes. However there is also 'record.volume', though I would assume this operates on the mixed digital signals at the end of the chain. no. record.volume is essenially an alias. on your hardware with the configuration you've posted, it's a shortcut for setting both record.adc and record.adc2. this is explained in azalia(4) (though maybe that info didn't make it into 4.6, the info in -current azalia(4) is mostly relevant even for 4.6). Also: a lot of the gain stages have defaults of 120.120. Would it be reasonable to assume that this is the 0 gain setting? no. unfortunately, the mixer interface, like a lot of audio(4) related stuff, is designed for consumer usage. so, we just have a range that is essentially 0-100% - it has no relevance to anything except the knob. truly the worst kind of knobs are those that have no outside meaning, but apparently people like this. *shrug* http://marc.info/?l=openbsd-techm=123101323408867w=2 Any thoughts appreciated. if you really want to know how to do this right, your best bet is to find the datasheet for your codec. now, your codec is a Realtek, which is common for azalia(4), and I happen to know them pretty well ... inputs.line=85,85 this is a 0 (0), 10 (85), 20 (170), 30 (255) dB gain on the line-in jack. values in () are the corresponding mixerctl values. record.adc=248,248 this is the ADC input gain. 0 dB should be around '88'. 0..255 here represents the hardware's -16.5 to 30 dB in 1.5 dB steps. these are the only gains on the recording path of your device. -- jake...@sdf.lonestar.org SDF Public Access UNIX System - http://sdf.lonestar.org
Re: audio recording levels
On Mon, Jun 14, 2010 at 02:03:38AM +0200, Martin Pelik?n wrote: Hi, this you might already know, but good rule of thumb is to set the levels manually for each source (according to its dynamics), having peaks around -6dB to -10dB. If you have manual volume/gain control on your recording device/preamp, I'd set all levels in the computer to 80% of the scale and then control everything by hand, on the box. but what does 80% mean? that could be either some attenuation, or a lot of gain. depends on the mixer control. This probably doesn't apply to OpenBSD, but in Linux there have been serious distortion trouble with levels set to 100% on some older cards again, that depends on what 100% means, which varies greatly. - the only solution is AFAIK try it out and use what your ears tell you. ultimately, yes. The good news is you can do other stuff while your LP's spinning and recording :-) well, that depends on *what* you're doing and *how* you're recording. -- jake...@sdf.lonestar.org SDF Public Access UNIX System - http://sdf.lonestar.org
