Audirvana 3.1

[the_bat]

2 hours ago, bazzvid said:

try -Audirvana -preferences-general-Use large font for playlist text.

 

Thanks.  I didn't think to try that on the grounds it wasn't playlist text I wanted to use large fonts for ?

 

It only increases the font size on the pop up from clicking the album text but that's the place it's most needed. I'd prefer a san-serif font as is used on the track listing, and around the same size but I fully accept that's a personal preference and Damien has more pressing things to look at.

 

Thanks again.

 

Alan

 

 

 

[wdelec]

@damien78

I installed A+ as a control point for streaming DSD files to a upnp dlna machine running MPD. It seems A+ transformed the dsd files into the format that MPD cannot recognized they are dsd and then converted them to PCM (24/382). Here the log of MPD. 

 

(Feature Request) Is that possible to keep the dsd files untouched and just pass the link to a upnp dlna streamer ?  

playlist: play 0:"http://192.168.0.122:49152/audirvana/audio_0_0.dff"

client: [0] command returned 0

decoder_thread: probing plugin ffmpeg

client: [0] process command "status"

client: [0] command returned 0

client: [0] process command "currentsong"

client: [0] command returned 0

client: [0] process command "playlistinfo "1""

client: [0] command returned 0

ffmpeg: detected input format 'iff' (IFF (Interchange File Format))

ffmpeg/iff: DSIFF v1.5.0.0

ffmpeg/iff: Before avformat_find_stream_info() pos: 130 bytes read:4344 seeks:2 nb_streams:1

ffmpeg/iff: All info found

ffmpeg/iff: Estimating duration from bitrate, this may be inaccurate

ffmpeg/iff: stream 0: start_time: -26143344775665.465 duration: 187.320

ffmpeg/iff: format: start_time: -9223372036854.775 duration: 187.320 bitrate=5644 kb/s

ffmpeg/iff: After avformat_find_stream_info() pos: 819330 bytes read:823544 seeks:2 frames:50

ffmpeg: codec 'dsd_msbf'

decoder: audio_format=352800:f:2, seekable=true

alsa_output: opened hw:0,0 type=HW

alsa_output: buffer: size=8..393216 time=22..1114558

alsa_output: period: size=4..196608 time=11..557279

alsa_output: format=S24_LE (Signed 24 bit Little Endian)

alsa_output: buffer_size=176400 period_size=9800

output: opened "botic" (alsa) audio_format=352800:24:2

output: converting in=352800:f:2 -> f=352800:f:2 -> out=352800:24:2

[jimdukey]

A+ doesn't Upsample DSD without PCM Coversion.

[Jud]

1 hour ago, wdelec said:

@damien78

I installed A+ as a control point for streaming DSD files to a upnp dlna machine running MPD. It seems A+ transformed the dsd files into the format that MPD cannot recognized they are dsd and then converted them to PCM (24/382). Here the log of MPD. 

 

(Feature Request) Is that possible to keep the dsd files untouched and just pass the link to a upnp dlna streamer ?  

 

playlist: play 0:"http://192.168.0.122:49152/audirvana/audio_0_0.dff"

client: [0] command returned 0

decoder_thread: probing plugin ffmpeg

client: [0] process command "status"

client: [0] command returned 0

client: [0] process command "currentsong"

client: [0] command returned 0

client: [0] process command "playlistinfo "1""

client: [0] command returned 0

ffmpeg: detected input format 'iff' (IFF (Interchange File Format))

ffmpeg/iff: DSIFF v1.5.0.0

ffmpeg/iff: Before avformat_find_stream_info() pos: 130 bytes read:4344 seeks:2 nb_streams:1

ffmpeg/iff: All info found

ffmpeg/iff: Estimating duration from bitrate, this may be inaccurate

ffmpeg/iff: stream 0: start_time: -26143344775665.465 duration: 187.320

ffmpeg/iff: format: start_time: -9223372036854.775 duration: 187.320 bitrate=5644 kb/s

ffmpeg/iff: After avformat_find_stream_info() pos: 819330 bytes read:823544 seeks:2 frames:50

ffmpeg: codec 'dsd_msbf'

decoder: audio_format=352800:f:2, seekable=true

alsa_output: opened hw:0,0 type=HW

alsa_output: buffer: size=8..393216 time=22..1114558

alsa_output: period: size=4..196608 time=11..557279

alsa_output: format=S24_LE (Signed 24 bit Little Endian)

alsa_output: buffer_size=176400 period_size=9800

output: opened "botic" (alsa) audio_format=352800:24:2

output: converting in=352800:f:2 -> f=352800:f:2 -> out=352800:24:2

 

MPD recognizes and plays DSD files.  So no new feature is needed in A+.  It’s therefore something to do with the MPD configuration and/or the streamer/DAC.  What streamer/DAC do you have, and can you change the MPD configuration or not?

[hykhleif]

@damien78 can i ask please why when i use audirvana with my mac and chord qutest i see no dsd in audio system active audio device tab, the dsd are greyed out

[RunHomeSlow]

7 hours ago, hykhleif said:

@damien78 can i ask please why when i use audirvana with my mac and chord qutest i see no dsd in audio system active audio device tab, the dsd are greyed out

In audio tab of preferences try selecting PCM over DSD 1.1 instead of automatic detection...

[Uni]

I am surprised that PCM upsampling sounds better than DoP in my setup.

 

I have compared two upsampling settings.  Setting 1: DSD over PCM provides stronger bass,  but it provides less details significantly.  In contrast, setting 2 provides much better clarity and details.  I don't know why, so any opinion is welcome.

 

Setting 1: DSD upsampling setting

same as above, convert to DSD128

 

Setting 2: PCM upsampling setting

Engine: iZotope

Steepness : 3db

Filter max length : 2,000,000 samples

Cutoff freq : 1.25

Anti-aliasing : 50

Pre-ringing : 0.36

Forced upsampling : power of 2

EQ: fabfilter Pro Q.  Natural phase mode, High cut 96DB@18k & 19.5k

 

Equipment:

iMac --USB--> NFB11.28 -> AKG K712

digital filter jumper setting: slow roll off, minimum phase (I compared it with fast roll off, linear phase...only this setting won't build up fatigue.  Overall it provides the best sound)

[buonassi]

looks like you read @copy_of_a's posts on best settings for iZotope (with FFPQ2).  I have that EQ software as well.

 

Having played around extensively, I can tell you that the way an impulse looks doesn't mean it will provide the best sound quality.  And that there is more to great sound than just the 'pre ringing'.  

 

I can't know what's causing your 'loss of detail' when upsampling to DSD.  It is possibly your DAC - and the way it handles DSD.  I got plenty of detail when I was doing the DSD software thing in A+.  Didn't miss a thing.  It was an iFi iDAC2 FYI. 

 

The other thing you may want to try is to set your DAC (with jumpers apparently) to use slow slow rolloff LINEAR phase.  Or make sure that you're invoking NOS (non oversampling) when you send the highest 'power of 2' sample rate to the DAC.  If the DAC continues to oversample what you've sent it and is set to Minimum phase, it will introduce more phase shift than you intended with your iZotope filter settings.  Best practice is to control the phase characteristics with your software and leave the DAC linear (if NOS can't be achieved), even fast rolloff linear.  Currently, your software setup is an 'intermediate phase' approach at 0.36.  It introduces a phase shift much closer to what a Min phase filter would do (meaning it's not a subtle shift at all).  If this intermediate phase shifted bitstream is then being oversampled with an additional Minimum phase filter from your DAC, it's just going to confuse/skew the sound (in my opinion).  

[jos]

I got a trial code for Roon for two months, but as a A+ fan I am a little tired ? waiting for the new A+ version... Hopefully I am not go to like Roon in the mean time...

[Uni]

1 hour ago, buonassi said:

looks like you read @copy_of_a's posts on best settings for iZotope (with FFPQ2).  I have that EQ software as well.

 

Having played around extensively, I can tell you that the way an impulse looks doesn't mean it will provide the best sound quality.  And that there is more to great sound than just the 'pre ringing'.  

 

I can't know what's causing your 'loss of detail' when upsampling to DSD.  It is possibly your DAC - and the way it handles DSD.  I got plenty of detail when I was doing the DSD software thing in A+.  Didn't miss a thing.  It was an iFi iDAC2 FYI. 

 

Thanks for your reply a lot  @buonassi .  First of all I should declare that I am a new birdie who didn't know all those digital filtering, upsampling, izotope things two weeks ago.  Fortunately I read a lot of posts and articles on internet these two weeks and able to understand what you were saying.  I share your feeling about it's due to how the DAC handles DSD.  FYI the chip is  Sabre ES9028 Pro 

 

Quote

 

The other thing you may want to try is to set your DAC (with jumpers apparently) to use slow slow rolloff LINEAR phase.  Or make sure that you're invoking NOS (non oversampling) when you send the highest 'power of 2' sample rate to the DAC.  If the DAC continues to oversample what you've sent it and is set to Minimum phase, it will introduce more phase shift than you intended with your iZotope filter settings.  Best practice is to control the phase characteristics with your software and leave the DAC linear (if NOS can't be achieved), even fast rolloff linear.  Currently, your software setup is an 'intermediate phase' approach at 0.36.  It introduces a phase shift much closer to what a Min phase filter would do (meaning it's not a subtle shift at all).  If this intermediate phase shifted bitstream is then being oversampled with an additional Minimum phase filter from your DAC, it's just going to confuse/skew the sound (in my opinion).  

 

I opened the box and rechecked and found that actually I am using slow rolloff linear phase.  Unfortunately there's no way to set NOS.  The combo creator, kingwa, said that since the internal mechanism of the DAC chip was not disclosed, he could never tell if t still does oversampling even audirvana already did.  But since the sound signature still changes when I change the jumper, I assume it's not.

 

My R2R-11 just arrived.  It is a ladder NOS DAC.   But unfortunately I still feel there's lost of details through DoP.  Some other fposts saying that R2R doesn't get well with DSD anyway.  Its a little off topic but I feel this DAC is less irritating and fatigue.

[buonassi]

welcome aboard man!  keep us posted.  

[copy_of_a]

On 12/31/2018 at 9:17 AM, buonassi said:

the way an impulse looks doesn't mean it will provide the best sound quality.  And that there is more to great sound than just the 'pre ringing'. 

I agree! The „extreme“ setting in conjunction with FFProQ2 was aimed at offline upsampling!
As posted somewhere I’ve discarded the quoted workflow and settings myself for a long time.
I’ve also eventually found the strong filtering through FFProQ2 made the sound somewhat dull at the top end and it lost spatiality. iZotopeSRC alone is better suited… and sounds more open. So today I would not consider the respective settings really useful.
However, I still think the 22 (steepness) - 0.95 (cutoff) - 100 (AA) setting - either in minimum or linear phase mode - is pretty good if you prefer (relatively) low ringing. With regard to the transition it’s remotely similar to „poly sinc short“ in HQPlayer but somewhat steeper (and theoretically somewhat more prone to produce aliasing/imaging). With regard to the transition HQPs poly sinc short translated to iZotope would be something like steepness: 16 - cutoff 0.93 - AA: 100.

[davidovada]

Sorry if this is off topic. I just wanted to let you all know how impressed I am with the personal service that Damian provides. I had an odd problem (tracks not advancing) that he hadn't seen before. We exchanged several messages (including extending the trial period) and voila...problem solved. In an age where customer service is absent or inconsistent, this should be acknowledged as being outstanding. Damian, thank you again.

[Uni]

17 hours ago, copy_of_a said:

I agree! The „extreme“ setting in conjunction with FFProQ2 was aimed at offline upsampling!
As posted somewhere I’ve discarded the quoted workflow and settings myself for a long time.
I’ve also eventually found the strong filtering through FFProQ2 made the sound somewhat dull at the top end and it lost spatiality. iZotopeSRC alone is better suited… and sounds more open. So today I would not consider the respective settings really useful.
However, I still think the 22 (steepness) - 0.95 (cutoff) - 100 (AA) setting - either in minimum or linear phase mode - is pretty good if you prefer (relatively) low ringing. With regard to the transition it’s remotely similar to „poly sinc short“ in HQPlayer but somewhat steeper (and theoretically somewhat more prone to produce aliasing/imaging). With regard to the transition HQPs poly sinc short translated to iZotope would be something like steepness: 16 - cutoff 0.93 - AA: 100.

 

@copy_of_a  Hi, great to see you have paid attention to this post.

 

In short, you recommend iZotope setting as steepness: 16 - cutoff 0.93 - AA: 100?  I use "power of 2 oversampling only", which goes to 384k for my DAC.  There's no need to set any EQ cutoff at 18k/19.5k.  Did I get you right for all these?

[Jud]

18 hours ago, copy_of_a said:

With regard to the transition it’s remotely similar to „poly sinc short“ in HQPlayer but somewhat steeper (and theoretically somewhat more prone to produce aliasing/imaging).

 

I'm thinking steeper = less aliasing/imaging, more ringing.

[copy_of_a]

10 hours ago, Jud said:

I'm thinking steeper = less aliasing/imaging, more ringing.

Basically, yes, that’s correct, of course.

However, it also depends on the cutoff shift. If you set an extremely steep filter but leave the cutoff shift at „1“ (=100%) you may produce aliasing/imaging as the filter turns around the nyquist frequency (see the 4th screenshot below with steepness: 150, cutoff: 1). The cutoff shift seems to be the most overlooked setting here on this forum in the iZotope and SRC threads and this is why it took me a while to understand how it works (although it’s actually pretty self-explaining). Generally speaking I’d say a cutoff shift greater than 0.97 is always prone to produce aliasing/imaging … unless you go really, really steep. It’s for a reason that iZotopes „No Aliasing“ presets set the cutoff to 0.96 (in conjunction with appropriate steepness).

 

Regarding my latest post above I’ve confused my settings. Steepness: 22, cutoff: 0.95 and steepness: 16, cutoff: 0.93 hit the nyquist frequency at the same level (only 1db difference)… so regarding aliasing/imaging they are almost the same. For reference some screenshots below showing the transitions of these 2 settings as well as the filter slope of HQPs poly sinc short.

 

 

10 hours ago, Uni said:

In short, you recommend iZotope setting as steepness: 16 - cutoff 0.93 - AA: 100?  I use "power of 2 oversampling only", which goes to 384k for my DAC.  There's no need to set any EQ cutoff at 18k/19.5k.  Did I get you right for all these?

In HQPlayer I like the poly sinc short setting a lot (which - with regard to the roll off - is similar to iZotope steepness: 16, cutoff: 0.93). In Audirvana with iZotope I actually favour my steepness: 22, cutoff: 0.95 setting.

You can chose the one you like better - and, yes, for both the settings there’s no need to further highcut through an EQ.

Basically you can also use the settings for any rates… but personally I always favour „power of 2 oversampling“ in my setup.

 

 

 

[buonassi]

On 1/2/2019 at 1:41 AM, copy_of_a said:

iZotopeSRC alone is better suited… and sounds more open

 

yes, much better than setting a LPF in fab filter, I too agree.  Actually, I only lived with fab filter LPF for about a week, then fell back to iZotope only.

 

my current settings are 16, 1.0 cutoff, 0.6 phase.  I don't care about super HF aliasing.  More concerned about preserving phase up to 16K.  I'm pretty sure these settings don't cause phase shift until the top-most octave.  That's based on my interpretation of graphs over at http://src.infinitewave.ca, but I could be wrong.  

 

So.... I need to change my avatar 'rank' it seems  -  I've re-embraced filters!

 

Hey, on another note, I've also asked miska, but does anyone know what interpolation math iZotope uses?  Is it cubic, sinc function, etc?  

[copy_of_a]

12 hours ago, buonassi said:

my current settings are 16, 1.0 cutoff, 0.6 phase.  I don't care about super HF aliasing.  More concerned about preserving phase up to 16K.  I'm pretty sure these settings don't cause phase shift until the top-most octave.  That's based on my interpretation of graphs over at http://src.infinitewave.ca, but I could be wrong. 

attached the phase of your 16 / 1.0 / 0.6 setting displayed in Room EQ Wizard (orange curve).

You may find 22 / 1.0 / 0.72 (green curve) even better with regard to your goal to preserve linear phase up to 16k (and the somewhat higher amount of ringing at steepness 22 compared to steepness 16 is certainly inaudible).

Then again, if you are happy with your settings, why bother?

 

[macuniverse]

43 minutes ago, copy_of_a said:

attached the phase of your 16 / 1.0 / 0.6 setting displayed in Room EQ Wizard (orange curve).

 

Can you provide some info/link on how we can do this using REW please? Thanks

[copy_of_a]

2 hours ago, macuniverse said:

Can you provide some info/link on how we can do this using REW please? Thanks

I load a 44.1kHz Dirac Impulse into iZotope RX (4), apply the respective SRC settings, upsample to 176.4kHz (without "Anti Aliasing", though, since this is a feature of Audirvana and is not present in the original software), export to 32bit float wav. Then load the processed file into REW.

So... it requires a software that provides iZotope SRC and export function in the first place...

[Musicophile]

4 hours ago, copy_of_a said:

I load a 44.1kHz Dirac Impulse into iZotope RX (4), apply the respective SRC settings, upsample to 176.4kHz (without "Anti Aliasing", though, since this is a feature of Audirvana and is not present in the original software), export to 32bit float wav. Then load the processed file into REW.

So... it requires a software that provides iZotope SRC and export function in the first place...

Could we maybe move the very technical iZotope discussions to a separate thread?

[macuniverse]

7 hours ago, copy_of_a said:

So... it requires a software that provides iZotope SRC and export function in the first place...

 

Thanks

[audiocanyon]

Completely different tangent - has anyone heard when the major Audirvana update might be rolling out for the Mac OS?

[buonassi]

13 hours ago, copy_of_a said:

attached the phase of your 16 / 1.0 / 0.6 setting displayed in Room EQ Wizard (orange curve).

You may find 22 / 1.0 / 0.72 (green curve) even better with regard to your goal to preserve linear phase up to 16k (and the somewhat higher amount of ringing at steepness 22 compared to steepness 16 is certainly inaudible).

Then again, if you are happy with your settings, why bother?

 

 

Absolutely PHENOMENAL!  Thank you so much!  I guess I'm really missing out by not having REW and RX

[hykhleif]

greetings

 

does any one please have a recommended setting for chord qutest, to have give more warm detailed sound using sox or izotope

 

my settings will be chord qutest --- burson conudctor--- audeze lcd mx4

 

thanks in advance, sorry to keep asking as I am new to all the sox and izotope thing

 

also what were the default settings in both sox and izotope as I do not know how to go back to default