# Console player and S/PDIF (Toslink)



## ogogon (Nov 22, 2017)

Greetings, colleagues!

Tell me, please, how can I solve my problem.

I use a computer without X.Org. On it, I run the CMus console player and listen to music.
The built-in sound card has both analog outputs and a digital optical output (Toslink).
At the analog output, music is present, but there is no optical output.

For testing, I run Linux Mint on this machine and the optical output worked fine. Therefore, it normally works.

Mother board in my computer - Asus E45M1-I DELUXE (Sound chip - Realtek® ALC892).
OS - FreeBSD 12.
The player is configured to the default sound output system - as to the OSS.
In addition, I ran another player - a flac123. The situation is similar.

How can you use the console programs to play music in digital output?

Ogogon.


----------



## tobik@ (Nov 22, 2017)

ogogon said:


> At the analog output, music is present, but there is no optical output.


Does it appear in /dev/sndstat?


----------



## ogogon (Nov 22, 2017)

tobik@ said:


> Does it appear in /dev/sndstat?




```
root@ot:~ # cat /dev/sndstat
Installed devices:
pcm0: <ATI R6xx (HDMI)> (play)
pcm1: <Realtek ALC892 (Rear Analog)> (play/rec) default
pcm2: <Realtek ALC892 (Front Analog)> (play/rec)
pcm3: <Realtek ALC892 (Rear Digital)> (play)
pcm4: <Realtek ALC892 (Onboard Digital)> (play)
No devices installed from userspace.
root@ot:~ #
```

There are as many as two digital outputs.
I believe that mine is pcm3. What is pcm4 - I do not know. Probably, in addition to the rear, on the motherboard there is also a digital audio connector.


----------



## SirDice (Nov 22, 2017)

ogogon said:


> OS - FreeBSD 12.


Topics about unsupported FreeBSD versions


----------



## tobik@ (Nov 23, 2017)

ogogon said:


> There are as many as two digital outputs.
> I believe that mine is pcm3. What is pcm4 - I do not know. Probably, in addition to the rear, on the motherboard there is also a digital audio connector.


You have already tried setting them as default device via `sysctl hw.snd.default_unit=3` (3 as in pcm3) and it didn't work?


----------



## poorandunlucky (Nov 23, 2017)

ogogon said:


> ```
> root@ot:~ # cat /dev/sndstat
> Installed devices:
> pcm0: <ATI R6xx (HDMI)> (play)
> ...



Did you check the volume?

Check out mixer(8) and sound(4).  You should be able to fix it with that...


----------



## rigoletto@ (Nov 23, 2017)

As far I am concerned the toslink output should be always set at 100% and then controlled by what is working as pre-amplifier in the audio chain (everything to reproduce audio from toslink should have one). The reason is the toslink output is "line-level".

*Just in the case someone concerned with audio quality.*

The volume control should always (as possible) be made in the analog domain.

While in the analog domain the volume control is made by + or - the signal voltage, and so the signal (means audio) will be kept intact (if the pre-amplifer was properly projected and built), in the digital domain the volume is controlled (starting from 100%) actually "removing information": less bits, less resolution, and usually more noise.

"Expensive" digital volume controls, aka dsp controlled ones (or dithered digital attenuation), upscale the audio first to a higher bit-depth (usually 24 or 32 bits) to lower later. Very cleaver, very clean, still degrade it and sound hygienic.


----------



## ogogon (Nov 25, 2017)

tobik@ said:


> You have already tried setting them as default device via `sysctl hw.snd.default_unit=3` (3 as in pcm3) and it didn't work?


Many thanks.
After I said this spell, everything worked as I wanted ...

After the launch of the CMus, the transmitting LED on the Toslink port began to glow and now in my speakers sing the incomparable Tarja Turunen.

Ogogon.


----------



## ogogon (Nov 25, 2017)

By the way, is it possible to look at what mode the player has set up the sound port?
The clock frequency, the depth of sound in bits ...

Ogogon.


----------



## rigoletto@ (Nov 25, 2017)

ogogon said:


> By the way, is it possible to look at what mode the player has set up the sound port?
> The clock frequency, the depth of sound in bits ...
> 
> Ogogon.



It rather depends on how did you set sound(4).

By default it probably is being re-mixed to 48Khz/16Bits wherever the source (audio file). The usual default everywhere.

Assuming you are using a proper external DAC, you could set it to play bitperfect with the downside of playing nothing if the source was encoded by a unsupported (by the DAC) frequency/bit depth.

EDIT:


```
dev.pcm.%d.bitperfect
        Enable or disable bitperfect mode.     When enabled, channels    will
        skip all dsp processing, such as channel matrixing, rate convert-
        ing and equalizing.  The pure sound stream    will be    fed directly
        to    the hardware.  If VCHANs are enabled, the bitperfect mode will
        use the VCHAN format/rate as the definitive format/rate target.
        The recommended way to use    bitperfect mode    is to disable VCHANs
        and enable    this sysctl.  Default is disabled.
```


----------



## ogogon (Nov 25, 2017)

lebarondemerde said:


> It rather depends on how did you set sound(4).
> 
> By default it probably is being re-mixed to 48Khz/16Bits wherever the source (audio file). The usual default everywhere.
> 
> ...


Thank you!
Sound revived, small details appeared, it ceased to be deaf and transistor!

Now in /etc/sysctl.conf it is written:

```
hw.snd.default_unit=3
dev.pcm.3.play.vchans=0
dev.pcm.3.bitperfect=1
```
Did I understand correctly? Or have I missed something?

Ogogon.


----------



## rigoletto@ (Nov 25, 2017)

I would need read sound(4) again to assure at 100%, but the basic thing is when using bitperfect you should disable vchans.


----------



## tobik@ (Nov 25, 2017)

ogogon said:


> By the way, is it possible to look at what mode the player has set up the sound port?


Yes, you can increase verbosity with `sysctl hw.snd.verbose=2` and then /dev/sndstat will contain some information about it.  But it isn't exactly the nicest thing to read... Not sure if we have a tool that makes it more human readable.


----------



## ogogon (Nov 25, 2017)

tobik@ said:


> Yes, you can increase verbosity with `sysctl hw.snd.verbose=2` and then /dev/sndstat will contain some information about it.  But it isn't exactly the nicest thing to read... Not sure if we have a tool that makes it more human readable.


Thank you, he began to tell much more. And where you can see the values of the numbers it leads?

And another question. How to allow maximum digital output modes?

I tried to play a sound file with a sound depth of 24 bits. The driver did not want to let him into Toslink.

`root@ot:/tmp # flac123 1.flac
flac123 version 0.0.12   'flac123 --help' for more info
ao_oss ERROR: Unsupported number of bits: 24.Error opening ao device 0
Error opening 1.flac
root@ot:/tmp #`

However, the hardware of the audio adapter (Realtek® ALC892)) allows this.
http://www.realtek.com.tw/products/productsView.aspx?Langid=1&PFid=28&Level=5&Conn=4&ProdID=284 ("Primary 16/20/24-bit SPDIF-OUT supports 32k/ 44.1k/48k/88.2k/96k/192kHz sample rate")

Does the driver need to allow such modes?


----------



## rigoletto@ (Nov 25, 2017)

Quick reply: I do not know.

EDIT: As far I know you can set it as bitperfect or fixed only.

I guess you are using this ALC892 device just as Toslink output. If so, assuming the bitperfect setup, it is working more like as pass through only.

In this case, what is on the other side of the Toslink connection, because that is the actual DAC?

EDIT_2: is the source you were trying  to play 24/192 or 24/176? Many Toslink transmitters are not that reliable to transmit at those frequencies in 24 bits.


----------



## tobik@ (Nov 25, 2017)

ogogon said:


> I tried to play a sound file with a sound depth of 24 bits. The driver did not want to let him into Toslink.


The OSS backend in audio/libao (which is used by audio/flac123) doesn't support 24 bit samples.

cmus should support it fine.  I submitted a bunch of patches earlier this year to fix it.


----------



## ogogon (Nov 25, 2017)

lebarondemerde said:


> I guess you are using this ALC892 device just as Toslink output. If so, assuming the bitperfect setup, it is working more like as pass through only.


You are absolutely right. In this configuration, the computer is used as digital audio transport. It opens the container FLAC, carries out minimal conversion and sends everything to the input of the DAC.



lebarondemerde said:


> In this case, what is on the other side of the Toslink connection, because that is the actual DAC?


To the output of the computer is connected a inexpensive DAC with a tube output. Xiangsheng DAC-01A.
As a S/PDIF Coax/Toslink receiver, he has Cirrus Logic® CS8416.



lebarondemerde said:


> EDIT_2: is the source you were trying  to play 24/192 or 24/176? Many Toslink transmitters are not that reliable to transmit at those frequencies in 24 bits.


I believe that you need to speak not about the Toslink transmitter. This primitive module, which receives a signal with a TTL level and flashes the transmitting LED.
It is more correct to talk about the S/PDIF driver.
In my case, this is Realtek® ALC892. The documentation says that "Primary 16/20/24-bit SPDIF-OUT supports a sampling frequency of 32k/44.1k/48k/88.2k/96k/192kHz"
Frankly, I have not heard about the frequency of 176k before.

Ogogon.


----------



## rigoletto@ (Nov 25, 2017)

> Frankly, I have not heard about the frequency of 176k before.



Yeah, they are rather uncommon but you still see them around: Example. 

EDIT: The use of 176Khz was reasonable common in the past, like 88Khz too. Probably matching with the advance of the digital recording gear.



> I believe that you need to speak not about the Toslink transmitter. This primitive module, which receives a signal with a TTL level and flashes the transmitting LED.



I was actually talking about the transmitter. Some of them do not work properly on those higher frequencies. While they do support they also fail some times. Why? I do not know.


----------



## ogogon (Nov 25, 2017)

tobik@ said:


> The OSS backend in audio/libao (which is used by audio/flac123) doesn't support 24 bit samples.
> 
> cmus should support it fine.  I submitted a bunch of patches earlier this year to fix it.


I'm sorry, but I'm interested.
As far as I understand, CMus also uses libao. And how do they get around this restriction?

Ogogon.


----------



## rigoletto@ (Nov 25, 2017)

Apparently, the CMus audio/libao support can be disabled in the port - it actually seem to be disabled by default.

I built and installed it in here and indeed there is no audio/libao installed at all.


----------



## ogogon (Nov 26, 2017)

lebarondemerde said:


> Apparently, the CMus audio/libao support can be disabled in the port - it actually seem to be disabled by default.
> 
> I built and installed it in here and indeed there is no audio/libao installed at all.


I'm sorry, I was wrong. There OpenBSD libsndio ...

Ogogon.


----------



## rigoletto@ (Nov 26, 2017)

Just in case you are interested, the 6N3 valve your DAC use is usually replaceable with 5670/2C51 (TungSol are the best) what most people think sounds better, and are inexpensive. The other and better alternative is WE396a but that one is pricey, and probably hard to find these days.


----------



## ogogon (Nov 26, 2017)

lebarondemerde said:


> Just in case you are interested, the 6N3 valve your DAC use is usually replaceable with 5670/2C51 (TungSol are the best) what most people think sounds better, and are inexpensive. The other and better alternative is WE396a but that one is pricey, and probably hard to find these days.


Thank you. In my DAC installed a Soviet 6N3P-E (rus. 6Н3П-Е) produced by "Svetlana" (Leningrad/St. Petersburg). Usually, such an elite firm as "Svetlana" was not distracted by a trifle of 6N3P type, but it was an order from whether the military, or the space agency, and they tried very hard. Tube with the prefix "-E", which should mean increased accuracy of manufacture and durability.


----------



## rigoletto@ (Nov 26, 2017)

Nice.

I never heard a 6N3P but you made me remember of 6N6P what is an amazing valve (they are not interchangeable). Probably one of the best sounding Soviet valves ever.

Well, there is ECC40 what is another very special Soviet valve but with the downside of using a B8A base, what is very hard to find these days.

And yeah, Mil. Specs are always special versions and should last "forever". Western valves usually name the military ones with different names like 10Y became VT-25, etc.

A funny thing about Soviet valves they tends to be more "bassy" than the westerns ones. The GM70 in particular have a very unique "jumbo" sound.

You could talk with THOSE guys if you need more valves, I think they run they business from Turkey and St. Petersburg; however I never did anything with them and so I can't say a thing about they reliability.

EDIT: very expensive but should one of the best DACs in the market: LampizatOr.


----------



## ogogon (Nov 26, 2017)

lebarondemerde said:


> Nice.
> I never heard a 6N3P but you made me remember of 6N6P what is an amazing valve (they are not interchangeable). Probably one of the best sounding Soviet valves ever.


But I - on the contrary. 6N6P even in the hands did not hold, but with 6N3P and 6N2P there was a lot.



lebarondemerde said:


> Well, there is ECC40 what is another very special Soviet valve but with the downside of using a B8A base, what is very hard to find these days.


I believe that you are mistaken. ECC40 is in my opinion "Philips" and does not have a Soviet analog.



lebarondemerde said:


> A funny thing about Soviet valves they tends to be more "bassy" than the westerns ones. The GM70 in particular have a very unique "jumbo" sound.


Lamps of GM series (rus. ГМ-xxx) and GU (rus. ГУ-xxx) were used in amplifying technique, speaking in Russian "not from a good life".
These are powerful generator high-frequency tubes, large, not economical, with a very inconvenient voltage and current of heat.
Simply, there were no others. The Soviet industry did very well what the military needs, but poorly and reluctantly to do what is necessary for normal people.



lebarondemerde said:


> You could talk with THOSE guys if you need more valves, I think they run they business from Turkey and St. Petersburg; however I never did anything with them and so I can't say a thing about they reliability.


Thank you. But now in Russia domestic tubes can be bought much cheaper than these guys sell. At one time they were made in incredible quantities, but the tube boom, alas, ended. Now private individuals with personal hoards can sell almost everything.



lebarondemerde said:


> EDIT: very expensive but should one of the best DACs in the market: LampizatOr.


Yes, thank you, I know this project.

Ogogon.


----------



## rigoletto@ (Nov 26, 2017)

> I believe that you are mistaken. ECC40 is in my opinion "Philips" and does not have a Soviet analog.



Yes, you are right.



> Lamps of GM series (rus. ГМ-xxx) and GU (rus. ГУ-xxx) were used in amplifying technique, speaking in Russian "not from a good life".
> These are powerful generator high-frequency tubes, large, not economical, with a very inconvenient voltage and current of heat.
> Simply, there were no others. The Soviet industry did very well what the military needs, but poorly and reluctantly to do what is necessary for normal people.



Yes, they are transmitting triodes with thoriated tungsten filament, like 211, 10Y, 801A, etc. GM70 is more or less equivalent (in power) to 211, GU48 is replaceable with 833 or 833A (can't remember exactly), and there is the huge GM100.

Those transmitting valves do great SET amplifiers, but they usually are very expensive to make (and run).

One of the characteristics of valves with thoriated tungsten filament is the huge amount of details, but can easily sound thin if not properly designed.

10Y and 801A are my preferred output valves, but they are low power. 10Y should give you less than 1W (usually 0.75W) and 801A about 4W maximum.

The "infamous" Audio Note Ongaku uses 211. ElRog make 300B with thoriated tungsten filament (well, it is not an actual 300B anymore).

At some point I want to manage get GM100 SET amplifiers (would need to be custom made) just to run sub-woofers, just to see how it will sound. The problem would the gigantic size of the transformers (the NAT one is hybrid).

Cheers! 

EDIT: I think it is too off-topic already.


----------



## ogogon (Nov 27, 2017)

tobik@ said:


> The OSS backend in audio/libao (which is used by audio/flac123) doesn't support 24 bit samples.
> 
> cmus should support it fine.  I submitted a bunch of patches earlier this year to fix it.


I believe that you need to make a few more patches. Because with CMus is also not all good.
When I use it to play FLAC files with a clock speed of 44100 - everything plays and sings just wonderful.
When I try to play FLAC files with the frequencies 9600 and 192000 - a short, raucous hiss sounds and CMus freezes.

Here's what sndstat writes at this moment, I give information only for my interface:
`pcm3: <Realtek ALC892 (Rear Digital)> on hdaa1  (1p:0v/0r:0v) default
   snddev flags=0x200003e7<SIMPLEX,AUTOVCHAN,SOFTPCMVOL,BUSY,MPSAFE,REGISTERED,BITPERFECT,VPC,PRIO_WR>
   [pcm3:play:dsp3.p0]: spd 192000, fmt 0x00201000, flags 0x2000510c, 0x00000001, pid 1058 (cmus)
   interrupts 1990, underruns 1986, feed 1990, ready 0 [b:65536/32768/2|bs:131072/32768/4]
   channel flags=0x2000510c<RUNNING,TRIGGERED,BUSY,HAS_SIZE,VCHAN_PASSTHROUGH,BITPERFECT>
   {userland} -> feeder_root(0x00201000) -> {hardware}`

The audio files are not destroyed - Audacity on my MacMini normally play this files.

*What could be wrong?* (Adventure continues!)

Ogogon.


----------



## rigoletto@ (Nov 27, 2017)

ogogon

Have you how to connect using the USB connection? That could narrow the problem due to using the "USB Audio 2.0" driver instead of the Realtek one.


----------



## tobik@ (Nov 27, 2017)

ogogon said:


> When I try to play FLAC files with the frequencies 9600 and 192000 - a short, raucous hiss sounds and CMus freezes.


Send one of the files to me and I'll take a look when I have time.


----------



## ogogon (Nov 27, 2017)

lebarondemerde said:


> ogogon
> 
> Have you how to connect using the USB connection? That could narrow the problem.


I apologize, I did not understand the technical meaning of your question.
I now have a computer, it has a built-in sound card, from its Toslink-output optical cable goes to an external DAC.
With God's help, no USB here!


----------



## ogogon (Nov 27, 2017)

tobik@ said:


> Send one of the files to me and I'll take a look when I have time.


They are very large. How can I give them to you?


----------



## rigoletto@ (Nov 27, 2017)

Your DAC seem to have a USB audio output connection (USB-B). You can use that instead of Toslink, and the DAC will appear as a (external) sound card to the system, it should use the generic USB Audio driver.

EDIT: I see it in HERE, maybe yours could be a slight different version.


----------



## ogogon (Nov 27, 2017)

lebarondemerde said:


> Your DAC seem to have a USB audio output connection (USB-B). You can use that instead of Toslink, and the DAC will appear as a (external) sound card to the system, it should use the generic USB Audio driver.


I've never done this, but tonight I'll try.


----------



## tobik@ (Nov 27, 2017)

ogogon said:


> They are very large. How can I give them to you?


Upload them to Google Drive for example and send me a link to it (tobik@FreeBSD.org).

EDIT: Come to think of it the reason it doesn't work is probably because you enabled bitperfect mode, but we'll see...


----------



## ogogon (Nov 29, 2017)

tobik@ said:


> EDIT: Come to think of it the reason it doesn't work is probably because you enabled bitperfect mode, but we'll see...


Yes you are right. After I made *dev.pcm.3.bitperfect=0* the system started to play.
And why did bitperfect impede it? And why did not impede at speed 44100?


----------



## ogogon (Nov 30, 2017)

lebarondemerde said:


> ogogon
> 
> Have you how to connect using the USB connection? That could narrow the problem due to using the "USB Audio 2.0" driver instead of the Realtek one.


I tried and got slightly different results.

Firstly, I connected a USB module FC-215 on a PCM2704 chip. aliexpress.com/store/product/5V-USB-Powered-HIFI-PCM2704-DAC-to-S-PDIF-Sound-Card-Decoder-Board-3-5mm-Analog/
It was defined as pcm5 and behaved like my built-in audio adapter. In the position dev.pcm.5.bitperfect = 0 played well, and with the value 1 - the hiss and hovered.

`pcm5: <USB audio> at ? kld snd_uaudio (1p:1v/0r:0v) default
   snddev flags=0x200002e3<SIMPLEX,AUTOVCHAN,BUSY,MPSAFE,REGISTERED,VPC,PRIO_WR>
   [pcm5:play:dsp5.p0]: spd 48000, fmt 0x00200010, flags 0x00002108, 0x00000004
   interrupts 11862, underruns 0, feed 11861, ready 0 [b:3072/1536/2|bs:4096/2048/2]
   channel flags=0x2108<TRIGGERED,BUSY,HAS_VCHAN>
   {userland} -> feeder_mixer(0x00200010) -> {hardware}
   pcm5:play:dsp5.p0[pcm5:virtual:dsp5.vp0]: spd 192000/48000, fmt 0x00210000/0x00200010, flags 0x1000110c, 0x0000002b, pid 1031 (cmus)
   interrupts 0, underruns 0, feed 124184, ready 61932 [b:0/0/0|bs:131064/32766/4]
   channel flags=0x1000110c<RUNNING,TRIGGERED,BUSY,HAS_SIZE,VIRTUAL>
   {userland} -> feeder_root(0x00210000) -> feeder_format(0x00210000 -> 0x00200010) -> feeder_rate(0x00200010 q:1 192000 -> 48000) -> feeder_volume(0x00200010) -> {hardware}`

Secondly. I connected the DAC itself.
He behaved differently.
In the dev.pcm.5.bitperfect = 0 state, it played normally.

`pcm5: <USB audio> at ? kld snd_uaudio (1p:0v/1r:1v) default
   snddev flags=0x2e2<AUTOVCHAN,BUSY,MPSAFE,REGISTERED,VPC>
   [pcm5:play:dsp5.p0]: spd 192000/96000, fmt 0x00210000, flags 0x0000110c, 0x0000002b, pid 1033 (cmus)
   interrupts 6272, underruns 47, feed 39957, ready 54696 [b:9216/4608/2|bs:131064/32766/4]
   channel flags=0x110c<RUNNING,TRIGGERED,BUSY,HAS_SIZE>
   {userland} -> feeder_root(0x00210000) -> feeder_format(0x00210000 -> 0x00201000) -> feeder_rate(0x00201000 q:1 192000 -> 96000) -> feeder_volume(0x00201000) -> feeder_format(0x00201000 -> 0x00210000) -> {hardware}
   [pcm5:record:dsp5.r0]: spd 48000, fmt 0x00200010/0x00210000, flags 0x00002100, 0x00000007
   interrupts 0, overruns 0, feed 0, hfree 4608, sfree 4096 [b:4608/2304/2|bs:4096/2048/2]
   channel flags=0x2100<BUSY,HAS_VCHAN>
   {hardware} -> feeder_root(0x00210000) -> feeder_format(0x00210000 -> 0x00200010) -> feeder_mixer(0x00200010) -> {userland}
   pcm5:record:dsp5.r0[pcm5:virtual:dsp5.vr0]: spd 8000, fmt 0x00100008, flags 0x10000000, 0x00000000
   interrupts 0, overruns 0, feed 0, hfree 0, sfree 0 [b:0/0/0|bs:0/0/0]
   channel flags=0x10000000<VIRTUAL>
   {hardware} -> feeder_root(0x00000000) -> {userland}`

But in the position dev.pcm.5.bitperfect = 1 did not hang, but started playing slowly, like a tape recorder turned on at another speed.

`pcm5: <USB audio> at ? kld snd_uaudio (1p:0v/1r:1v) default
   snddev flags=0x3e2<AUTOVCHAN,BUSY,MPSAFE,REGISTERED,BITPERFECT,VPC>
   [pcm5:play:dsp5.p0]: spd 96000, fmt 0x00210000, flags 0x2000110c, 0x00000001, pid 1032 (cmus)
   interrupts 6135, underruns 0, feed 6134, ready 73026 [b:9216/4608/2|bs:131064/32766/4]
   channel flags=0x2000110c<RUNNING,TRIGGERED,BUSY,HAS_SIZE,BITPERFECT>
   {userland} -> feeder_root(0x00210000) -> {hardware}
   [pcm5:record:dsp5.r0]: spd 48000, fmt 0x00200010/0x00210000, flags 0x00002100, 0x00000007
   interrupts 0, overruns 0, feed 0, hfree 4608, sfree 4096 [b:4608/2304/2|bs:4096/2048/2]
   channel flags=0x2100<BUSY,HAS_VCHAN>
   {hardware} -> feeder_root(0x00210000) -> feeder_format(0x00210000 -> 0x00200010) -> feeder_mixer(0x00200010) -> {userland}
   pcm5:record:dsp5.r0[pcm5:virtual:dsp5.vr0]: spd 8000, fmt 0x00100008, flags 0x10000000, 0x00000000
   interrupts 0, overruns 0, feed 0, hfree 0, sfree 0 [b:0/0/0|bs:0/0/0]
   channel flags=0x10000000<VIRTUAL>
   {hardware} -> feeder_root(0x00000000) -> {userland}`

In all cases, was played the same sound file in 24bit/192k format.

Ogogon.


----------



## rigoletto@ (Nov 30, 2017)

tobik@

Can it be related with some buffer not being enough?

ogogon 

Assuming you did all tests using CMus, could you test using something else?

multimedia/ffmpeg maybe?

audio/musicpd + audio/ncmpcpp would be perfect because is know to work, but at cost of taking some time to setup.


----------



## ogogon (Nov 30, 2017)

lebarondemerde said:


> tobik@
> 
> Can it be related with some buffer not being enough?


And how can I look at the sizes of the corresponding buffers and change these sizes?
Are you referring to system buffers or application buffers?


----------



## rigoletto@ (Dec 1, 2017)

You could try to tweak: `hw.snd.latency` and `hw.snd.latency_profile` to see if will help. See sound(4)


----------



## tobik@ (Dec 10, 2017)

I tried the files you uploaded (though I only downloaded 1% of them) and they play fine for me.  Cmus' OSS backend needs to be fixed to actual check the sample rate the system sets and refuse to play files when it's different instead of simply playing them. Normally OSS guarantees that it can be resampled, but in bitperfect mode that assumption breaks.  But besides that I can't see any problems in cmus itself.


----------



## Snurg (Dec 10, 2017)

OT:
1. What does "DAC" mean in this context?
2. ECC 40 is a German tube afaik. I had many of them decades ago in my spares store.
3. If you happen to need German tubes, look if you can get "Valvo" manufactured. (Nowadays hard to find and very expensive)
This is the best German manufacturer in my and my friends' personal experience.
You can almost compare Philips <> Valvo tubes like normal russian tubes <> military version.
I quite often threw away tubes I considered as inferior (Philips, Siemens, Telefunken and lower) just to have only a few really good Valvo tubes of each type in my storage past then...
Many of them I put them into the fireplace - it is fun to watch the tubes shrink when the glass gets soft. Sometimes the results were really funny.


----------



## rigoletto@ (Dec 10, 2017)

> I quite often threw away tubes I considered as inferior (Philips, Siemens, Telefunken and lower) just to have only a few really good Valvo tubes of each type in my storage past then...



Please don't to that. Sell them, or send to me. 

You still can find ECC40 from Valvo around, not that expensive, at least not like old WE valves.

If you manage to find a WE300B, you can easily sell it for at very least US$4K. 

And there is a huge market for Telefuken tubes, specially the Diamond marked 6463.


----------



## Snurg (Dec 10, 2017)

lebarondemerde, of course I won't do that nowadays.
That was in the 1980s when I was schoolboy. Past then many old stuff (radios, TV etc) from the tube era were dumped on the streets and I picked the best stuff of that, repaired and sold much things and kept the very best for me.
Sadly in the late 1990s burglars visited my home and the boxes with my good tubes were gone


----------

