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 Post subject: Stereo Tool 9.71 BETA
PostPosted: Tue Jun 08, 2021 5:07 pm 
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Latest version - Beta 046 (August 28 2021)

Downloads
Changes/bugfixes
  • Fix Quick Adjust sliders effect on displayed values for attack/release and Max values.
  • Final Mat/Wess presets check
  • STQ: Fix Input 2 resampling -> Difficult, will take more time. Partially done, nothing noticeable yet.

PortAudio recently released a new version (the first official release in 4 years), and because some recent Windows updates caused issues for some people (certain hardware/driver/Windows combinations) we upgraded to that new release for version 9.70. Since then, some bugs have been found in the new code, which have been fixed. The new version has not officially been released, but when reading through the change log it seemed to be a good idea to update to the latest version.

MicroMPX+ is a new, for now experimental version of MicroMPX for lower bitrates (just like AAC+, MP3Pro etc.). This still has most of the advantages that MicroMPX has at higher bitrates, but (obviously) at a bit lower bitrates. Testers needed.


(*) As mentioned above, due to issues with PortAudio versions and recent Windows updates, it appears that some sound cards only work at very specific I/O block sizes. Sometimes these are different than those that are suggested by the driver!

Because of this, the default block sizes can now be overridden.

Typical values that appear to work are often powers of two (256, 512, 1024, 2048, 4096), but in some cases also multiples of the number of samples per millisecond (320, 441, 480).

We hope to either find values that _always_ work, or a very limited set of values that cover all sound cards.

Please report back to us which values work for your sound card. We need the following info:
- sound card type
- input or output
- sample rate
- block size(s) that work).



TODO's
  • RELEASE
    -
  • STQ: Fix Input 2 resampling -> Difficult, will take more time.
  • Finish BIMP/STAMP interfaces?
  • Multi-instance version: Resampling behavior for input 2 (add it for input 2, and check why enabling it for input 1 causes glitches on input 2)
  • Is a few % phase wobbeling a good idea to improve stereo image and flat frequency response in rooms?
  • Sub-presets (Declipper, Voice detection)
  • MB1 band linking: instead of making it 3 to 2, make it 2 to 3
  • Fake Stereo doesn't work for Quality > 100%
  • AES67 NIC selecting doesn't work on STX (only shows 127.0.0.1 and Automatic) - Fails sometimes, probably due to booting too fast. Needs to be fixed in the AES67 redesign.
  • New web interface in Generic plugin
  • Fix web interface slow animation
  • Allocate big buffers if HD delay is enabled also in Stereo Tool, to avoid glitches
  • PNR message which settings changed
  • Speech detection: Different settings (presets?) for English/Dutch vs Spanish (different S sounds). Different settings for music radio with occasional speech, talk radio with occasional music, and sports radio with background noises. LATER
  • MicroMPX for left/right audio? And/or for AM?
  • Startup screen with logo STXtreme
  • Email/SMS alerts via cloud
  • Gates must work on loudest channel when combining channels? (Wes)
  • Speed linking in dynamic speed mode: Release slowest behavior a bit faster on big drops
  • Try extra clipping stage for highest highs AFTER rest of clipping stages, to protect against shrill sound @ 75 us pre-emphasis
  • Advanced Clipper Bass Killer: Add dynamic behavior (lower strength if there are mids, for example)
  • Improve Highs vs Rest calculation, it's now incorrectly adding up levels. Is it necessary to use real RMS? (Maybe not, verify with random noise ON HIGHS)
  • AGC band linking for extreme behavior? Maybe per band A/B combo a maximum difference? Or, easier, just select a "mid" band and set max offsets for the others.
  • AGC voice detection, reducing thresholds. "IF L/R = Mono, IF Gate = very active, IF Asymetry = detected, IF spectral energy = mostly mids, single base frequency or highs".
  • Add way to override sound card block sizes (at least for Linux, probably also useful for Windows)
  • Login: Per session
  • Login: Update editable status
  • Login: Hide things
  • Login: Show warning when trying to change something without access
  • Put backup file upload in web interface for MicroMPX
  • STX: Edit whitelist on unit
  • Remote jrtplib
  • Create/include BIMP presets
  • Stereo Tool Enterprise: Think of good name
  • Stereo Tool Enterprise: Switching between devices/channels can go wrong
  • Stereo Tool Enterprise: Sample rate mismatch detection is not ok for ASIO inputs
  • Pi PiMP stand alone BUSY
  • Pi AM stand alone
  • ABE GUI BUSY
  • Release
  • Add "Combp phase"-like test tone to uMPX (FB chat Cleiton Araújo)
  • Stereo Tool new I/O framework: Add bypass mode
  • MicroMPX: Silence detector with backup player (timeout - threshold - time ok before going back to stream)
  • Clipper detect where highs are killed and compensate for that
  • Block multiple instances (avoid running multiple Stereo Tool instances with the same settings)
  • Later: MicroMPX: Export version, bitrate and redundancy info in webinterface and library
  • Later: MicroMPX: Export user messages in library
  • Some pages remain visible when you're logged out (password protection)
  • Later: STX: Overview page with meters
  • Monday: Fill new site BUSY
  • Monday: Pens BUSY
  • Tuesday: Make multi-instance version available
  • Tuesday: Add SST multi-instance interface
  • DSP code - on hold
  • MicroMPX decoder 2nd port number - with pull-down menu, and indices.
  • Stereo Tool priorities: Check what prio 15 things are
  • LiquidSoap/Centova setup manual, maybe add some features
  • Website BUSY
  • Support Contract
  • Improve Advanced Limiter shape (Bojcha) -> Problem is not what it seemed to be! There's a memory problem between chunks of audio, solution is to keep track of what happened in the previous block, and keep the previous audio.
  • Improve Advanced Bass Distortion Protection
  • Compressor with curve drawer
  • Disable stereo widening when detecting low bitrate MP3s
  • Add a checkbox under Delossifier that makes Absolute Highs fix the highs.
  • STX: Verify high frequency tilt behavior
  • Put Declipper presets in ST. Selecting one with "APPLY" overrules all existing settings.
  • BS412 bug: For Quality > 100, the compressors are off by a few dB (probably 1.5x or 2/3rds).
  • How to synchronize HD? THU
    -
  • Measure HifiBerry stereo separation -> Improved from 35 to 55 dB. Make RC work with Ignore high frequencies -> Later
  • Loudness project
  • Tom aan de Stegge remarks email 30 september 20:23 - change many defaults
    -
  • Support rewindowing with different Quality/Size settings. Will give better HPF performance and make lower CPU load possible, and PPM patch points.
  • Linux command line version with AES67 input so output can be streamed - seems to be useful... Other options (run a script for the output) might be easier.
  • Finish 9 - final uMPX file name issue -> In next 9 update.
  • PhoneBooster make multi-threaded if needed (it probably is, esp. for Pi)
  • PhoneBooster make Pi version
  • MicroMPX needs exception handling, just in case.
  • WatchCat replace files instead of placing them elsewhere. For playout systems.
  • WatchCat add CMD /C
  • Thimeo School
  • ABDP remove stuff that's no longer needed? BUSY for low CPU
  • Optimize ABDP
  • Input and output gain for command line version (like for sound card)
  • Dynamic adjustment of priorities
  • Sooth effect
  • AES67: Add WheatNet
  • Installer: Make installer aware of programs like MediaMonkey, AIMP, .....
  • HD delay
  • Documentation: "Processing" page contains all kinds of BS412 info - why?
  • Declipper L/R noise
  • Declipper handle resampled audio better (upsample, find oversampling spikes, then detect based on what remains)
  • Console version: Read sample rate and bit size from WAV header
  • # bands selections should allow option to insert a band instead. Or ask what to do when you change it.
  • AES67 thread affinities and priorities
  • Add resampler
Quote:
Latest version - Beta 045 (Aug 26 2021)

Downloads

Changes/bugfixes
  • Create preset test script
  • Replaced more sqrt's by 1/(1/sqrt(x))
  • Bug fix: True Bass levels were calculated incorrectly if certain filters were placed after it. This caused some presets to have more bass than they originally had. Added a "Legacy" setting to choose between old and new behavior.
  • Command line version: Fix "Unexpected end of file".
  • Figure out what's going on with Syrtho
  • Winamp version: Added MB Recaster Free to installer
  • Auto EQ speech sliders didn't work, it always had the maximum effect.
Quote:
Latest version - Beta 040 (Aug 22 2021)

Downloads
Changes/bugfixes
  • Multi-instance version: Fix hangup detection with restarts
  • Multi-instance version: Blocking writes
  • Multi-instance version: Fixed assert on startup (_begin/_end flags)
  • Declipper (as a separate plugin): Fixed some crashes (caused by changes in Stereo Tool code), should be ready for a new release now.
  • Overflows in sqrt(x) = x * 1 / sqrt(x)
  • Output waveform not properly activated on load
  • Reverted 1/(1/sqrt(x)) for Dequantizer and MicroMPX+
Quote:
Latest version - Beta 039 (Aug 19 2021)

Downloads
Changes/bugfixes
  • Update version number
  • Verify that Jack build works
  • Fix Pi ABE builds
  • Rewrote sqrt(x) to x * 1 / sqrt(x), which is faster
  • Verify MicroMPX sound card output, mirrorred signal? - No, must be a sound card issue
  • Check Janne L's preset in command line version - free features only but beeps - does not reproduce.
  • Fixed MPX spectrum drawing, broken since a few beta's
  • Add Speech Detection hooks-per-filter to STX GUI
  • Check sound card input #2 causing buffer underruns - Does not reproduce here, must be a sound card issue.
  • Update presets in SST/STX
  • Matt's preset; add AM "Hot" style DCM preset
  • Matt's preset in Stereo Tool/STXtreme/SST
  • Multi-instance version: smp2
  • Multi-instance version: Quickly closing/opening FM opens in wrong sample rate
Quote:
Latest version - Beta 037 (Aug 13 2021)

Downloads
Changes/bugfixes
  • Scope zoom factors not saved since 9.70 -> Yes they are
  • "For now, please use StereoTool version 9.64. Legacy 1band AGC is broken in later versions!" - viewtopic.php?f=10&t=26400 Fixed, bands 1 and 3 were swapped.
  • Check PortAudio on Mac - Bug fixed, waiting for feedback from others to see if it's fixed there as well.
  • Linux version ignored affinity masks (core pulldown did work, mask did not)
  • Webserver: Fixed bug for massive replies (> 262144 bytes).
  • Update presets
  • Spectrum web interface resizing as in native Tom code
  • STXtreme logo clicking doesn't take you to Quick Adjust (Tom)
  • Scopes in STXtreme sometimes don't work - doesn't reproduce anymore, so apparently fixed by earlier fixes.
  • STXtreme: Added icons
  • Mac input and microphone rights: TEST
  • HD delay missing from STX GUI - no, it's there.
  • Speech detection in STX GUI
  • Scopes disabled mode isn't saved
  • Multi-instance version: Fix clicking caused by resampler
  • Bass EQ ignored Speech Detection in some cases
  • Input meters in STX turn orange for low input levels, and don't show level after Gain
Quote:
Latest version - Beta 034 (Aug 8 2021)

Downloads
Changes/bugfixes
  • STXtreme: HD delay doesn't work. Check fmSampleReadPosOffset_seconds, make it work for STX, and make STX buffer size ALWAYS big enough (14 seconds at least) TEST
  • Test and release ~250 RDS Relay READY
  • WatchCat: Added automatic detection of ffmpeg if installed in C:\, more locations to be added if people request them.
  • WatchCat: Made it impossible to delete built-in presets (that was possible before due to a bug).
  • Web interface: Support for 4K screens in waveforms/spectrums
  • Spectrums: Set maximum width, smaller for 32-bit VST plugins, bigger for STXtreme.
  • Moved overrule block size settings to extreme tweaker mode
  • Spectrums: Scaled drawing in internal interface
  • Spectrums: Fast vertical scaling in internal interface
  • Presets enable Voice
  • Added Wes' new Lunas presets, removed Cerulean and Mousse.
  • MicroMPX+: Decided on final filter settings.
  • Auto-save in ARM version, at the very least when there's no GUI - was already implemented!
  • Check Hot preset - can we do that differently (bass only)? -> No
  • Check that PortAudio settings are correct (high latency as default) - Was ok, no changes
  • Check new PortAudio release - Nothing interesting in the change log, and it seems to be a bit messy right now with some half-finished PR. So, we'll keep what we have for now.
  • FOR TESTING: STANDALONE VERSION NOW EXPLICITLY SETS TIMER INTERVAL TO 1 MS. Was already done by PortAudio for Wasapi/KS so I don't really expect it to help. But who knows... -> Did not help, to be removed again in next build.
Quote:
Latest version - Beta 030 (July 28 2021)

Downloads
Changes/bugfixes
  • AGC display was broken in BETA029.
  • MicroMPX backup player: Added on/off check boxes and hid the "multiple short drops" behavior settings in a deeper menu
  • MicroMPX backup player: Improved "multiple short drops" behavior: It now ignores a single long dropout.
Quote:
Latest version - Beta 029 (July 27 2021)

Downloads
Changes/bugfixes
  • External RDS input in Stereo Tool: Fixed pilot phase errors. To be tested.
  • MicroMPX+: Improved determining of artifact levels.
  • GUI: Fixed display issue in Multiband 2.
  • Web interface: Sliders
  • MicroMPX encoder in Stereo Tool: Fixed pilot frequency issue when using external RDS input.
  • MicroMPX backup player: Make anti-hysterisis clearer (hidden in separate popup page)
Quote:
Latest version - Beta 028 (July 21 2021)

Downloads
Changes/bugfixes
  • First MicroMPX+ experimental version
  • STXtreme: Show RDS data
  • GUI: "AGC" disappears from bottom bar if disabled.
  • Webinterface weirdness latest build: no presets, STXtreme firmware updates broken
  • PNR: Fixed issue: Switching to a different sample rate wasn't detected.
  • Report that PNR doesn't work in WatchCat at 44.1 Does not reproduce, re-test after rebuild
  • RDS relay changed PS to PI
  • Added current PI code to FM info display, so you can see if UECP etc works well.
  • MicroMPX+ VBR: Add target/maximum bitrate. That way we can control talk radio at max quality 95% of the time without ever going too high.
  • Generic plugin 2 instances acts weird, seems to share web interface.
  • Duplicate MpxModulator and OversampledClipper for MicroMPX+ for optimal performance
  • Figure out how to determine whether to use the MicroMPX+ or MicroMPX modulator. Either via user setting (stupid) or automatically (but how do we switch?)
Quote:
Latest version - Beta 025 (July 16 2021)

Downloads
Changes/bugfixes
  • New GUI title uninitialized
  • Webinterface meters upside were down, MicroMPX Encoder destination list looked messy.
  • MicroMPX library: Add dropout reporting function
  • MicroMPX: Add average achieved bitrate display
  • Bug fix: MicroMPX Decoder: Fix reporting of silence times
  • MicroMPX built-in encoder: Check external RDS encoder, something apears to go wrong with that - No, works fine with 2 different RDS encoders, something must be wrong with the used encoder I guess?
  • MicroMPX+: VBR mode. And disable bit reservoir?
  • MicroMPX+: VBR add higher noise thresholds for loud presets, both in encoding and in pulldown menu
  • MicroMPX+: Shrink keyframes
  • MicroMPX+: Add license warning
  • Winamp plugin broken due to Instance update - fix it
  • New "Relay" tool initial version.
Quote:
Latest version - Beta 020 (July 7 2021)

Downloads
Changes/bugfixes
  • PortAudio updated to nightly build from June 28, which has several bug fixes compared to the 19.7.0 release. Some people have reported that this version works better.
  • MicroMPX: Forward Error Correction for V4 encoding was broken, it could in some rare cases cause weird sounds to be played when packets were recovered.
  • GUI: Fixed AGC meters
  • Speech detection: Lowest mono level changed to -25 dB instead of -40, to stay in speech mode with a soft music bed
  • GUI: Speech detection: Bass EQ and Natural Dynamics metering didn't reflect when it's active
  • STXtreme: AES67 settings in STXtreme interface
  • STXtreme: Fix top bar in start screen web interface
  • STXtreme: Improve error message for ASIO timing: Add glitch_too_late_fixed and glitch_too_late_not_fixed debug values DONE, testing needed, maybe split per output?
  • STXtreme: Fix that clicking on Thimeo takes you to the start screen DONE but "Quick Adjust" should not be visible in top bar.
  • STXtreme: Main interface weirdness latest build (lockups, scopes way too small, < > arrows back)
  • MicroMPX+ test DIV2 and other value, make 1 or 2 (?) parameters
  • STXtreme fully zooming out on waveforms in GUI causes audio glitches which in turn trigger the watchdog which causes a reboot
  • SQAP code check ASIO 9.50 vs 9.71 - No issues found, waiting for feedback
  • Web interface: Log DISMISS doesn't work.
  • MicroMPX+ try grouping bins - didn't work (weird!)
  • MicroMPX+ find best start value
  • MicroMPX+ change noise_multiplier value (* 38400/48000)
  • MicroMPX+ reduce reps
  • MicroMPX+ listening tests
  • MicroMPX+ fix peak protection
  • MicroMPX+ add setting in encoder, and transmit it
  • MicroMPX+ fix CPU load issue on ARM for non-+ decoder - probably not interesting, difference is only 3%.
  • MicroMPX+ Add licensing
  • MicroMPX library: umpxGetDecoderStreamVersion missing from library
  • MicroMPX library: send/receive ancillary data
Quote:
Quote:
Latest version - Beta 014 (June 15 2021)

Downloads
Changes/bugfixes
  • Added an extra mode to the block size settings for sound card I/O. No idea if this is good or bad, it appears to work well (and better in some cases, with lower latencies) on my pc though.
  • Increased maximum SFN delay (FM sound card settings; someone needed that to use it for a different goal).
Quote:
Latest version - Beta 013 (June 15 2021)

Downloads
Changes/bugfixes
  • MicroMPX: Improved filtering overlap in ARM builds, which makes filters steeper. Slightly higher CPU usage (3% for Cortex A9 build, about 10% for Pi builds), but it's worth it.
  • VST version: Removed deprecated settings warnings, because (due to how VST's) we can't reliably detect when this message should and shouldn't be shown.
  • Bug fix: After opening a larger number of web interface instances, an outside-of-memory read could occur, which could kill the web interface.
  • Added internal ASIO glitch counters (not yet exported to GUI)
  • AES67 now has priority over ASIO in the settings
  • Bug fix: Simple clipper was not disabled in Bypass mode
  • ShoutCast 2 does not fully work (song info doesn't work), so the GUI now clearly says "ShoutCast 1". Waiting for SC2 support...
  • Added I/O menu to generic-plugin-with-audio version
  • Fixed memory leak in generic plugin version.
  • Bug fix: Generic-plugin-with-audio with audio was accessing low latency thread with does not exist in this mode.
Quote:
Latest version - Beta 011 (June 8 2021)

Downloads
Changes/bugfixes
  • Added Speech detection, part of Advanced Dynamics. When speech is detected, that can be used to disable Natural Dynamics, link the bass band in the AGC and reduce the threshold of the AGC, limit the boosts in Auto EQ, disable Bass EQ, disable True Bass. With this, speech sounds much more natural (not boomy) even if you add a lot of bass to music.
  • PortAudio: Due to issues with PortAudio versions and recent Windows updates, it appears that some sound cards only work at very specific I/O block sizes. The default speeds can now be overridden, please report back to us which values work for your sound card (so: sound card type, input or output, sample rate, block size(s) that work). Based on this we can probably create a pulldown-menu with a limited set of choices. (*)
  • CPU presets: Works different (probably easier) than before. Upgrading from old version should work correctly now (as opposed to in earlier 9.71 beta's).
  • AES67: Added x4 upsampling to make 48 kHz AES67 work in STX together with ASIO at 192 kHz. Tested with LiveWire.
  • AES67 NIC selection didn't work in web interface due to a bug in pulldown menu's.
  • I/O: Wasapi NOT exclusive now has priority over exclusive mode, because some drivers have problems with exclusive mode (hiccups/gliches).
  • Rewrote code that enforces memory alignment (code change only, no effects expected).


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PostPosted: Tue Jun 08, 2021 5:19 pm 
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Regarding buffer configurations: Somebody on the Breakaway forum posted some thoughts about this subject a while back. Some points raised might apply.

https://ceaforum.net/?topic=fine-tuning ... -and-rates
Quote:
3) Buf Size/Buffers
As suggested above, this is the size of each workspace and the number of them that BAE needs to buffer the input from Winamp or other software, process it and pass it to the output device. If you are processing music which was ripped from a CD (sample rate 44.100 kHz), I would focus on a buffer size of multiples of 441 (ten times the sample rate). There are a bunch of technical reasons why, but it stands to reason that samples are going to “fit” better if the buffer is aligned in some way with the number of music samples taken. I have often seen 882 yield a better result, but the theory still holds – it’s a multiple of the sample rate.
Similarly, if you have DVD quality audio sampled at 48kHz, I would experiment with 480, 960 etc

This is probably where most of the tweaking is needed. You might find an input buffer of 441/3 will yield a low input jitter percentage, but the same sized or number of output buffers might produce a high output jitter percentage, so you might end up with 441/3 in, 441/4 out or 441/3 in, 882/2 out.
Would a detected jitter percentage be an appropriate addition to the UI?

I'm excited to mess with the speech detection. :)


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PostPosted: Tue Jun 08, 2021 5:34 pm 
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Quote:
Regarding buffer configurations: Somebody on the Breakaway forum posted some thoughts about this subject a while back. Some points raised might apply.

https://ceaforum.net/?topic=fine-tuning ... -and-rates
Quote:
3) Buf Size/Buffers
As suggested above, this is the size of each workspace and the number of them that BAE needs to buffer the input from Winamp or other software, process it and pass it to the output device. If you are processing music which was ripped from a CD (sample rate 44.100 kHz), I would focus on a buffer size of multiples of 441 (ten times the sample rate). There are a bunch of technical reasons why, but it stands to reason that samples are going to “fit” better if the buffer is aligned in some way with the number of music samples taken. I have often seen 882 yield a better result, but the theory still holds – it’s a multiple of the sample rate.
Similarly, if you have DVD quality audio sampled at 48kHz, I would experiment with 480, 960 etc

This is probably where most of the tweaking is needed. You might find an input buffer of 441/3 will yield a low input jitter percentage, but the same sized or number of output buffers might produce a high output jitter percentage, so you might end up with 441/3 in, 441/4 out or 441/3 in, 882/2 out.
Would a detected jitter percentage be an appropriate addition to the UI?

I'm excited to mess with the speech detection. :)
This kinda looks like what I found. The weird thing that I saw was that if I open my webcam microphone for input (it only supports 32 kHz) and select ANY value that's not a multiple of 320, I get that number of samples but with zeroes in it. So the number of samples is completely wrong. So for example, if I select 340, I get 320 valid samples and 20 0's. The audio continues where it stopped in the next packet, so in fact I would get 34 kHz of data with chunks of silence instead of 32 kHz. What I don't know (yet) is whether this is some weirdness in my sound card, or something that more sound cards do. There is no such thing as jitter in this case (I'm not even sure what "jitter" really means in BAE).

For Stereo Tool itself it would make much more sense to read and write in powers-of-two, since the processing itself happens in powers-of-two. And for many other sound cards, the value suggested by the driver is actually a power-of-two. Just not for my webcam microphone.

I could just follow the driver suggestion, and that usually works. But it appears that that's not always the case. :( Which is why I want more info. Also, I can't think of any logical reason why multiplies of 441, 480 etc would make more sense than anything else, unless the sound card driver itself wants to send data per millisecond instead of per so-many samples. It might indeed work that way for some sound cards though.


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PostPosted: Tue Jun 08, 2021 5:39 pm 
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I've already reported this (I don't know if this was already known in BETA 008. AGC gain reduction is not visible on the AGC meter). 011 is still the same


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PostPosted: Tue Jun 08, 2021 7:02 pm 
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Quote:
I've already reported this (I don't know if this was already known in BETA 008. AGC gain reduction is not visible on the AGC meter). 011 is still the same
Thanks! This is one of those things that are "under development" right now. Some large rewrites of the web interface code for SST, STXtreme and MicroMPX are underway, and one of the changes broke this in the native interface.


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PostPosted: Tue Jun 08, 2021 9:22 pm 
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I already said i first tested with VAC and best options are 441 and 882 for 44.1kHz and 480 and 960 for 48kHz. Same as my USB headphones and my realtek onboard soundcard. However i found and tested that with BA One, where it shows small/acceptable amount of Jitter.
Hans, You say your camera mic is 32kHz, to me it's logical that it works good only with 320 samples and 640 should be good or even better. In BaOne for sample rates you only have selection 44.1 and 48kHz (HD).. so Block Size List is restricted to only those numbers, multiple of 441 and multiple of 480. There is lower or standard numbers like 256 or 384 or half 480 but what i saw till now best/minimal block size for 44.1kHz-441 or double that, for 48kHz-480 or double that, or even 4*.. etc. There is also lower numbers in list but there is no 320 for example since it's not possible to set 32kHz.
So, you can make it standard list (steps) multiply of 441 and 480.. and check box that enables any number. Sometimes for 48kHz 240 block size can work just fine.

I also asked for jitter reading if possible. Easier to find lowest stable block size. What is jitter? Someone think it's myth because nobody can heat jitter error 10% or even more, but it's kinda audio distortion caused by clock deviation. So 0% Jitter is ideal but it's also ok at 4%.. 8% or more (error) since it's still not audible. Atleast I understand it like that.

Gui errors, still that compressor meter in old agc is acting strange.

About Timer Resolution. ST should fix it's usage at some fixed number, 1.000ms probably best number. Exactly what that Axia updated driver done. But to make that to work ST should enforce windows to same 1.0000ms timer resolution when it's started, since it's not permanent settings in windows. ISLC is example software that can do that.

Also ST starts from 3/4 buffer again in this beta (mme used). Buffer bar is also bugged, sometimes settles to middle sometimes not.

_________________
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PostPosted: Wed Jun 09, 2021 2:24 pm 
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1 - I don't know if the behavior in BA One and Stereo Tool is the same, so I don't know if these 441/882 etc numbers make sense. It might, but basically, there's no good reason (except if drivers made things this way) to use the number of samples per millisecond as a basis for anything. In most cases, and also with how modern CPU's work, powers-of-two are much more efficient. For example with ASIO, which is the most optimized I/O layer in Windows, I have never even seen drivers that allow these types of values.

2 - I don't _see_ any jitter when using different values. The only thing that I see is that some sound cards with some sizes insert zeroes (silence) in the audio.

3 - AGC meter: Known issue

4 - Timer resolution for what exactly? We have never assumed anything about timer resolutions, also because older Windows versions already used 16 ms instead of 1 ms. So we have always assumed any timers to be not more accurate than 16 ms. (Something in PortAudio might assume it to be 1 ms though).

5 - Without changing any settings? That's odd... How big is your output buffer?


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PostPosted: Wed Jun 09, 2021 7:44 pm 
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Quote:
5 - Without changing any settings? That's odd... How big is your output buffer?
I keep it 50ms tried 100 and 200.. random bar positions and fuller buffer then normal at start. My input sync need to go for from x1.0004 at start.

Ahh.. also VAC mode "Cable range" is more adequate then "Cable format" now. It was vice-versa few betas ago.

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PostPosted: Thu Jun 10, 2021 3:40 pm 
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Joined: Sun Dec 23, 2018 7:44 pm
Posts: 792
Location: Texas, USA
Seems like the LQLL won't push audio for me to monitor when the dehummer is enabled. Should I send a preset? 64bit windows stand alone, 9.71 b11


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PostPosted: Thu Jun 10, 2021 5:00 pm 
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Posts: 11185
Quote:
Seems like the LQLL won't push audio for me to monitor when the dehummer is enabled. Should I send a preset? 64bit windows stand alone, 9.71 b11
Yes, please do. I just tried it here and there were no issues (there were a few beta's ago, but those issues were solved before BETA011). Please send your .ini file in a PM or to support@thimeo.com, that's usually the best way to reproduce it. You can find it under C:\Users\username, it's normally called something like stereo_tool.ini (depending on which exact version you're using).

Before I look into it, some questions:
- What does the buffer meter of LQLL indicate?
- Does the normal output continue normally?


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