Messages

31

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 26, 2019, 01:59:08 PM »

FYI
I performed a Win10 upgrade installation to 2nd SSD to Win10 1809.
Same behaviour like with Win7.
So I will proceed testing with Win7 as it makes no difference and as Win7 is still my preferred Windows environment.

32

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 26, 2019, 12:55:53 PM »

Sample rate changes work properly.

What irritates me a little is, that the RME driver settings window shows different ASIO buffer sizes for UFX+ and ADI-2 Pro after changing to a song with a different sample rate.

Example: I kept the RME driver setting window open and then playback:
@44.1: UFX+ 32, ADI-2 Pro 32 -> OK
when changing song to 96 kHz:
@96 kHz: UFX 64, ADI-2 Pro 32 (should also be 64) -> unexpected (but works)

After closing RME driver settings dialog and reopening: both are at 64 samples which is correct.
I think this is maybe something where RME might need to take a look.

What not works is still changing the ASIO buffersize while playback, then musicbee hangs and doesn't play a song anymore without a full restart of the application.

P.S.: in previous tests I only replaced the MusicBee.exe, this time I replaced all files from your ZIP archive, to be on the save side that I perform a proper test for you.

33

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 26, 2019, 11:50:16 AM »

Many thanks, sure I can do in the afternoon.

34

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 26, 2019, 11:16:47 AM »

could you run this debug version and send me the info at the end of the error log

https://www.mediafire.com/file/v8yyjayg4k8kd5k/MusicBeeDebugAsio.zip/file

Sure, sorry, had no time the last days.
Now the switching between all sample frequencies works, very nice.

To present you a short informative debug file I tested now exactly this szenario:
- 2 FLAC files (44.1 kHz and 96 kHz)
- ASIO buffersize: 32 samples @44.1 kHz, changes to 64 samples @96 kHz)
11:49:45 - Start of MusicBee
12:00:00 - Start of 44.1 kHz FLAC file (by double clicking to the file on desktop)
12:00:15 - Start of 96 kHz FLAC file (by double clicking to the file on desktop)
12:00:30 - Start of 44.1 kHz FLAC file (by double clicking to the file on desktop)
12:00:45 - Manually stopped MusicBee

26.05.2019 11:59:19 - ASIO reset join
26.05.2019 12:00:01 - ASIO Set rate=44100 (Comment: start of 1st FLAC file)
26.05.2019 12:00:01 - ASIO Set rate2
26.05.2019 12:00:01 - ASIO already started=False
26.05.2019 12:00:01 - ASIO started
26.05.2019 12:00:15 - ASIO Pause (Comment: start of 2nd FLAC file)
26.05.2019 12:00:15 - ASIO Stop2
26.05.2019 12:00:16 - ASIO Set rate=96000
26.05.2019 12:00:16 - ASIO Set rate2
26.05.2019 12:00:16 - ASIO already started=False
26.05.2019 12:00:16 - ASIO started
26.05.2019 12:00:30 - ASIO Pause (Comment: start of 1st FLAC file again)
26.05.2019 12:00:30 - ASIO Stop2
26.05.2019 12:00:31 - ASIO Set rate=44100
26.05.2019 12:00:31 - ASIO Set rate2
26.05.2019 12:00:31 - ASIO already started=False
26.05.2019 12:00:31 - ASIO started
(Comment: stop of playback is not recorded in the sebug log)

What does not work is, if you change the ASIO buffersize during playback, this hangs MusicBee completely.
Nothing helps, different title selection does not restart playback. You can only exit MusicBee and restart.

In this case I started playback of 44.1 kHz FLAC file at around 12:10:00 with ASIO buffersize of 32 samples.
At 12:10:15 I changed ASIO buffersize from 32 to 256 samples and MusicBee immediately stops playing.
At 12:10:30 I changed the buffersize down to 128 samples, still hangs
At 12:10:35 I changed the buffersize down to 64   samples, still hangs
At 12:10:40 I changed the buffersize down to 32   samples, still hangs
At 12:11:45 I tried to stop and restart playing, still hangs
At 12:12:00 I choosed a different FLAC file with 96 kHz, still hangs, start and stop no success.

26.05.2019 12:09:56 - ASIO reset join
26.05.2019 12:10:04 - ASIO Set rate=44100
26.05.2019 12:10:04 - ASIO Set rate2
26.05.2019 12:10:04 - ASIO already started=False
26.05.2019 12:10:04 - ASIO started
26.05.2019 12:10:30 - asioevent=2
26.05.2019 12:10:59 - ASIO Pause
26.05.2019 12:10:59 - ASIO Stop2
26.05.2019 12:11:00 - ASIO Set rate=96000
26.05.2019 12:11:00 - ASIO Set rate2
26.05.2019 12:11:00 - ASIO already started=False
26.05.2019 12:11:00 - ASIO started

Many thanks for your investigations.

35

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 21, 2019, 07:34:09 PM »

After the test requested above, could you restart MB and play the track that played at the wrong sample rate again and also send those results

21.05.2019 19:45:18 - BASS_ASIO_ChannelSetRate(44100,14):True
21.05.2019 19:45:18 - BASS_ASIO_SetRate:True
21.05.2019 19:46:38 - BASS_ASIO_ChannelSetRate(96000,14):True
21.05.2019 19:46:38 - BASS_ASIO_SetRate:True
21.05.2019 19:48:40 - BASS_ASIO_ChannelSetRate(96000,14):True
21.05.2019 19:48:40 - BASS_ASIO_SetRate:True

Maybe to make it clear, there is no "problem sample rate".
The "switching" of one to another sample rate is the problem if the you change from single to double or to quad speed.
Not from 44.1 to 48, but from 44.1 to double (88.2/96) or quad speed (192)

Play something at 44.1 .. select other title with 96 kHz -> issue.
Restart MusicBee, continue playing last title with 96 kHz -> works.

What should work is to be able to select / play songs of any sample rate in one MusicBee session.

When the sample rate is being changed from single to double or quad two things happen:
- ASIO buffer size changes it doubles from single to double and again from double to quad speed
- Amount / Numbering of channels changes, because the amount of ADAT and MADI channels changes
   IF I use AES of UFX+ no change in assignment.
   But if I use the ADI-2 Pro via USB, then the channel assignment becomes wrong as it is not being adjusted.

For ADAT and MADI double speed is being achieved by bonding two, quad speed by bonding 4 channels.

I noticed already that the assignment in MusicBee of Left and Right channel is wrong,
when the amount of channels changes when changing between single / double / quad speed.

Here the list/order of channels and ASIO buffer size as example depening one sample rate:
https://www.dropbox.com/s/6sio1qwf4bw9nec/MADface%20Channel%20Mapping%20UFX%2B%20and%20ADI-2%20Pro.pdf?dl=1

I think the 1st important milestone would be to assume that the channel assignment for Left and Right channel
does not change .. this is the case when I playback via UFX->AES->ADI-2 Pro and get sample rate change stable.

2nd important step would be that MusicBee becomes aware of the exact amount of channels depending on sample rate.
Example: if I playback @44.1 via USB directly to ADI-2 Pro then its channel 95 and 96, but @96 its 55+56
and @192 its 35+36.

36

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 21, 2019, 06:48:12 PM »

Hi, this version wont solve the issue but logs some info into the error log

https://www.mediafire.com/file/tgmk3k1xco0d1ux/MusicBeeDebugAsio.zip/file
unzip and replace the existing musicbee files

stop once you have started playing something with a problem samplerate and then send me the info at the end of the error log (help/ support/ view error log)

21.05.2019 19:45:18 - BASS_ASIO_ChannelSetRate(44100,14):True
21.05.2019 19:45:18 - BASS_ASIO_SetRate:True
21.05.2019 19:46:38 - BASS_ASIO_ChannelSetRate(96000,14):True
21.05.2019 19:46:38 - BASS_ASIO_SetRate:True

37

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 20, 2019, 09:40:22 PM »

Hi,

many thanks, will do that tomorrow and pls read my mail, thanks.

BR
Andreas

38

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 19, 2019, 05:35:34 PM »

Dear Steven,

I retried and even simplified my setup to use only the ADI-2 Pro FS BE via USB.
I turned off the UFX+ and thus didn't use AES, set clock properly to INT...
Also tried to use the USB2 port coming from Chipset (C612, Supermicro X10SRi-F).
Nothing helped on my side so far.

39

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: May 19, 2019, 01:09:08 PM »

Hey Steven, very nice to hear that you are working on this !!

Sorry to say but I see no difference to the older version 3.3.6904, that I have running.
I made a copy of the old folder and compared both versions.

What works is the change from double (88.2/96) or quad speed (192 kHz) to 44.1, but that worked already in 3.3.6904.

What doesn't work is:
- from single speed (44.1) to double or quad speed or
- from double to quad speed

When changing i.e. from 44.1 to 192 kHz (quad speed), the RME driver and device show the correct speed 192 kHz, but the playback appears to be still at the old speed 44.1.

I crosschecked with Foobar 1.4.4b1 there I can change between 44.1 and double speed and between double and quad speed vice versa.

Windows 7 SP1
RME device ADI-2 Pro FS BE
Firmware: v90 (latest, 21.2.19)
RME MADIface driver: 0.9681 (latest, 25.2.19)
Sonnet Allegro Pro USB3 Card
Sonnet Allegro Pro (4xFL1100 chip), Fresco Logic driver version: FLUSB3.0-3.8.34556.0

HW chain: PC -> Sonnet Allegro Pro (USB3) -> RME UFX+ (clock master) -> AES -> ADI-2 Pro (clock slave)

40

Questions / Re: Issues when analysing / writing replay gain tags of my FLAC archieve

« on: April 28, 2019, 11:54:10 AM »

I only know of Ctrl-Shift-V, which is what I use.

41

Questions / Re: Issues when analysing / writing replay gain tags of my FLAC archieve

« on: April 28, 2019, 09:45:41 AM »

Ok thanks, that bringt it to the point.

42

Questions / Re: Issues when analysing / writing replay gain tags of my FLAC archieve

« on: April 28, 2019, 08:30:44 AM »

Thanks for your reply, I use MusicBee.

But sorry, you didn't answer my question, see in bold.

For me is all clear except the point what value shall I dial-in after the warning that for some titles distortion might happen.

43

Questions / Issues when analysing / writing replay gain tags of my FLAC archieve

« on: April 27, 2019, 09:37:02 AM »

Hi,

I am facing some problems when performing volume analysis to write replay gain tags.
I have a big music archive, volume normalization takes quite some time.

After the replay gain analysis I feel like alone with a huge list in a tiny window
where it's nearly impossible to browse through all titles to get an overview which adjustment I have to perform.

Why is it not possible that MusicBee finds a proper value for adjusting the replay gains up or down on it's own ?

I am aware of that we are not peak level optimizing, that we talk about something like subjective perceived loudness.
But still there is the question, when seeing all the different adjustment values, what final adjustment do I need to perform ??

Proposal: additional sortable colums for title and album replay gain tag adjustment would support the review process.

I see that foobar2000 for example 1st analyses and then writes the replay gain tags.

Wouldn't it be better for MusicBee to 1st gather the max level and loudness of all titles and then coming to conclusions to which common loudness all titles could be led ? At the end we write replay gain tags per title.

If the music material is very different I could think of it's tricky to get both .. a common loudness for all titles on one hand and on the other hand not come over 0dB for each of the titles.

But somehow I have the feeling that this should be solveable by algorithms and not letting the user alone to adjust something where he can not really oversee what to do .. because nobody tells him exactly which value to take or what makes sense.

Foobar2000 also claims to be more precise when using oversampling (to increase sample rate by x2, x4, ...).
Would maybe be also a nice idea to make this analysis more thorough.

Any thoughts or recommendations to this to make this process better ?

Many thanks

44

Bug Reports / 3.3 6904: still issues with sample rate changes (ASIO driver)

« on: January 06, 2019, 02:16:27 PM »

Hi,

happy new year.

There is still a major problem if you want to play FLAC files of different sample frequencies using an ASIO driver.
I reported it alreay in another thread 1 approx 6 months ago, but no fix so far.

In contrast with that with Foobar2000 no problem.

Problem description:
When you play a new song with a significatly higher sample frequency, i.e. from 44.1 to 88.2 then the sample rate
becomes changed from 44.1 to 88.2 kHz, but MusicBee plays at the wrong speed most likely as it it would be still
44.1 kHz as it plays veeeeery slow / deep.
The same effect if you change from much higher to much lower speed.
Nothing bad happens if you change from 44.1 to 48 and back or change between 88.2/96 or 176.4/192 kHz.
Only workaround is to close MusicBee, open it and then voila, it plays 88.2.

This is the case now since a few releases and has not been fixed yet :-/
Thats very sad as I have lossless content of different sample rates: 44.1, 48, 88.2, 96 and 192 kHz.
and I need to work with the ASIO driver in my setup.

Hardware and driver in use:
Windows 7 SP1 Professional, 64 bit.
RME ADI-2 Pro FS BE connected via AES to RME UFX+
Clock Master is RME UFX+, ADI-2 Pro FS gets clock from UFX+ via AES.
RME ASIO driver: MADIface 0.9680 (latest)

With foobar2000 v1.4.1 release (using same HW, driver, signal chain) no issues.

Wouldn't it be worth to spend a few brain cycles trying to fix this?
I think it would be worth if MusicBee would work flawlessly with ASIO drivers like foobar2000.

Many thanks
Andreas

45

Bug Reports / Re: 3.1.6512: distortion when playing audio with much different samplerates via ASIO

« on: July 01, 2018, 08:55:22 AM »

Thanks for the pointer, will try the new ASIO DLL.

I am using 3.2.6699 with still the same problem.