VLC 2.0 with equalizer on 2 pass mode

[corElement]

What exactly happens in the backend with this setting enabled?

At first it seemed like someone put a cushion infront of the speaker and the volume went down by 30% but then I noticed I could hear subtle notes and tones had become more apparant than before. The "openness of the sound" got replaced with strong control over the sound and a darker silent background.

Like.. details in sound like midi notes and piano bass strokes became more prominent.. the reverb that follows on a real piano when you strike a low key I never heard it before on my setup till now.

Is this openness without the 2pass actually what they mean by distortion? Because there was a similar effect when I moved from my Jamo AVR on 2ch stereo to the CA 840a and now with the 2 pass equalizer in VLC it's like another huge improvement in the SQ.

I wish I could make this setting permanent across the system for all sound instead of just VLC. I use the xonar d2x normal driver on 2 channel analog.

 

[alpha1]

I am surprised - its just a software and not supposed to be manipulating anything except what is told.

Need to check.
what about the EQ / pre-amp etc settings?

 

[corElement]

MOTHER OF GOD!!!!

Every single track I'm playing sounds like a knife cutting butter....

SO SMOOOOTH OMG!

I strongly recommend people with bright setups to try out 2 pass equalizer in flat mode on VLC. The tonality it brings out is amazing!!!!

I can even hear gentle drum foot beats in tracks I could never make out before!

 

[alpha1]

I didn't hear anything except the sound reduced by a few decibels.
(on flat setting)

Why don't you ask the creation team behind VLC.
I am sure they will be surprised!

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Perhaps its just that you are trying to (active) listen today instead of (passive) hearing.


And just to be clear - the software player cannot cause distortions unless you overdrive the EQ settings.
As far as official statements from most freeware/open source audio player creation team: sound quality remains same across different players in their basic mode. The more effects you add, the farther away you move from the original sound.

 

[corElement]

The difference is in my face when i turn it on and off back to back.

It's possible your amp/speakers arent revealing the difference because there is a noticeable difference in quality of sound on my setup when I engage 2 pass EQ on flat.

3-6db reduction in gain but about 20% more clarity of subtle details.

I would love to hear some more people comment on this after trying it on their setup.

 

[alpha1]

Did you contact the developers to ask them?
Its a program after all - nothing magical.

 

[corElement]

This is what I got

Let me explain, when you run something with two-pass, you almost always give the software a nominal average bitrate instead of dynamic. Which it tries to achieve by allocating bits appropriately over the span of the material. The programmers may also take the first pass's information to improve general quality by various means.

On the first pass, it won't play the file, Instead, it will record a log file with statistics about the file. Then, on the second pass, when it does output your file, the software has a log file to use, so it doesn't have to guess. This means it can do a better job doling out bitrate, and you get a better quality output.

In other words:

2-pass generally has improved quality due to the ability to view frames collectively, rather than in a linear frame-to-frame manner. This requires more system resources allocated in order to view sections of frames which require more bits, and sections with less need, and allocate bits appropriately. Does this answer your question?

This almost seems like a software version of how Asynchronous USB re-clocks data.

 

[sound1]

Tried it, not much difference. After 2 pass it seems a little 'rounded' but hard to tell for sure.
Probably you are hearing the subtle differences because of your Xonar.

For me foobar2000 with WASAPI is way better than VLC.

 

[corElement]

YES!!!

Thats exactly the word! It sounds round and gentle.

I'm using Xonar with AISO default drivers

What card are you using and how do I try out foobar2000 with wasapi as well?

 

[alpha1]

This is what I got



This almost seems like a software version of how Asynchronous USB re-clocks data.

Are you sure that is the description of 2 pass EQ on VLC?
The prose above sounded more like 2 pass encoding - like when you convert raw video to DIVX or H264.

In which case 2 pass is better way to encode than single pass.

 

[sound1]

YES!!!

Thats exactly the word! It sounds round and gentle.

I'm using Xonar with AISO default drivers

What card are you using and how do I try out foobar2000 with wasapi as well?

I am not using any external card, just the stock one in my laptop with headphone out.
You should really try foobar2000 with WASAPI -- it sounded much better than the ASIO to my ears.
Also try upsampling plugin in foobar (@48kHz). It removes edginess and music sounds more 'rounded', if that's what you are looking for.

 

[alpha1]

I am not using any external card, just the stock one in my laptop with headphone out.
You should really try foobar2000 with WASAPI -- it sounded much better than the ASIO to my ears.
Also try upsampling plugin in foobar (@48kHz). It removes edginess and music sounds more 'rounded', if that's what you are looking for.

I am sure your laptop sound-card is working on 44.1kHz sample rate.
In such cases up-sampling music (44.1kHz) to 48kHz will get down-sampled to 44.1kHz again after coming to the soundcard.

Which will lead to a deteriorated sound quality - not enhanced.

Up-sampler works well only when you have sound card that has native sampling rate of 48/96/192 kHz.
In that case also, software up-sampling is done in case the sound driver up-sampling is of rotten quality (like old Creative cards).

I am surprised that people at HFV prefer the "deteriorated" sound (like 2 pass EQ on VLC, or upsampler on Foobar2000)!!
Perhaps this is why tube amps are preferred on this forum.
LOL.

 

[corElement]

I wouldn't call it deteriorated.... because I can hear mode detail and content... in the tracks I regularly listen to..

 

[sound1]

I am sure your laptop sound-card is working on 44.1kHz sample rate.
In such cases up-sampling music (44.1kHz) to 48kHz will get down-sampled to 44.1kHz again after coming to the soundcard.

Which will lead to a deteriorated sound quality - not enhanced.

Up-sampler works well only when you have sound card that has native sampling rate of 48/96/192 kHz.
In that case also, software up-sampling is done in case the sound driver up-sampling is of rotten quality (like old Creative cards).

I am surprised that people at HFV prefer the "deteriorated" sound (like 2 pass EQ on VLC, or upsampler on Foobar2000)!!
Perhaps this is why tube amps are preferred on this forum.
LOL.

And how exactly you concluded that my laptop supports only 44.1khz without even knowing the model number?

I use HP Elitebook 8440p and according to HP below are audio hardware specs. Complete specs are here.

Frequency Response: 20 Hz - 20 kHz
Signal to Noise Ratio: > 85 dB
THD: 0.01%
Noise Floor: -110 dB
Play/Record Sampling Rates: 8 kHz - 48kHz
DAC: 16, 20 or 24-bit
ADC: 16 or 20-bit
​

Yep, it does support 48kHz.

Even for 44.1kHz cards with ASIO4ALL, it is possible that some sound cards sound better with 48kHz upsampling because the native driver is faulty.

You know, it helps to know the facts before making sweeping generalizations just because you read some DSP theorems in college or internet.

 

[corElement]

Wow chillax man.

 

[alpha1]

And how exactly you concluded that my laptop supports only 44.1khz without even knowing the model number?

I use HP Elitebook 8440p and according to HP below are audio hardware specs. Complete specs are here.

Frequency Response: 20 Hz - 20 kHz
Signal to Noise Ratio: > 85 dB
THD: 0.01%
Noise Floor: -110 dB
Play/Record Sampling Rates: 8 kHz - 48kHz
DAC: 16, 20 or 24-bit
ADC: 16 or 20-bit
​

Yep, it does support 48kHz.

Even for 44.1kHz cards with ASIO4ALL, it is possible that some sound cards sound better with 48kHz upsampling because the native driver is faulty.

You know, it helps to know the facts before making sweeping generalizations just because you read some DSP theorems in college or internet.

Is yours (driver) faulty?
That you have to resort to software program up-sampling (which is normally done by the device driver).

 

[sound1]

Is yours (driver) faulty?

No idea.

That you have to resort to software program up-sampling (which is normally done by the device driver).

I just tried it, sounded better so I am keeping it that way. BTW I couldn't configure the driver s/w to upsample.

 

[alpha1]

^ You don't need to do it manually.
The soundcard is built on a native sampling rate of 48kHz.

But all the audio sources we have - CDs, wave files, mp3 etc are at 44.1kHz.
So they HAVE to be up sampled to reach the soundcard.
This up sampling is already being done by sound driver, OR inside the sound card.
Without this, we will not get output.


Re sampling from 44.1kHz to 48 kHz will anyway introduce distortion (whether via software program or driver or firmware). Whether this distortion is perceivable or not is disputable.
And whether this distortion is pleasant to ears or not is even more disputable.

To understand why I call this exercise futile, refer to foobar's developers FAQ:
foobar2000 FAQ

Q: What resampler settings should I use?
A: First, you shouldnt use resampler at all, unless you can tell the difference between resampling being enabled and disabled; resampler is a resource hog and the differences on normal music are very small and virtually impossible to notice (only certain test signals such as udial.wav may sound obviously different, but nothing like that occurs in real music).
If you really have to use resampler, resample to 48000Hz* and use fastest settings. You dont gain quality by resampling to higher samplerate, its just like stretching a picture to display it on a higher-resolution screen. You will be most likely able to play 96000Hz sample rates on whatever card you have, but samplerates unsupported by hardware will be down sampled by to e.g. 48000Hz before reaching your soundcard.


* Most of currently manufactured consumer soundcards resample internally to 48000Hz (which is absolutely needed for mixing multiple streams), some of them (eg. all SoundBlaster Live! and Audigy series) have issues with resampling (again, mostly noticeable on test signals such as udial.wav, you are very unlikely to be able to tell the difference on real music); those cards are the main reason why resampler DSP became available.


***

I am sure the person who has created an audio player, and works on sound bits, knows far more about what a program does to those bits.

 

[alpha1]

BTW another gem from foobar devs:

Q: Does foobar2000 sound better than other players?
A: No. Most of sound quality differences people hear are placebo effect (at least with real music), as actual differences in produced sound data are below their noise floor (1 or 2 last bits in 16bit samples). Foobar2000 has sound processing features such as software resampling or 24bit output on new high-end soundcards, but most of other mainstream players are capable of doing the same by now.

 

[sound1]

People have noticed & measured difference in sound b/w native rate (48kHz) and PLL derived rate (44.1kHz) even in good quality cards. Example here is Asus Xonar (see attached graphs):
Asus Essence STX Extension Oustide The Case? - Page 5 - diyAudio