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Recent Posts
1
Audio Hardware / Re: is it worth to buy an external dac ?
Last post by francesco -
i'm talking about realtek , they don't release a lot of informations about audio chipsets
As a professional engineer (not audio), I could not build a system around components which had no published specification (for specifications which mattered in the final product).  The reason for that is lack of accountability: for a product to meet its specification, its components have to meet their specifications, or be subject to rigorous testing in-house to determine their specifications (an expensive process).

A typical consumer-grade PC is not aimed at audiophiles, so the makers are not concerned about audiophile specifications.  Therefore an audiophile user cannot rely on guaranteed performance for noise, THD etc.

There are three solutions to this:
  • Find an audiophile supplier of PCs;
  • Use an external DAC with published performance specifications;
  • Listen to (or test) the audio output from your existing PC and decide whether it is good enough for yourself.

Notebooks can get much better simply by unplugging from the mains and running off battery.

I realise you were wondering whether you would notice any improvement by switching to an external DAC and whether the investment would be worthwhile.  Unfortunately only you can decide that, it depends how good your ears are (and how much it matters to you).  Personally, I don't think I would notice any difference.  What you do get with an external DAC is better connectors (the 3.5mm stereo socket is a weak link)!
Hi
I was looking a friend 's computer , pretty good i guess , there is a MSI B350M Gaming Pro , with a Realtek® ALC887 high definition audio card
even looks a pretty desktop with 64Gb of ram , great video card , and fast fast ssd 980 PRO NVMe M.2 SSD 1 TB
i can't find any informations about the Realtek® ALC887 high definition , even should pretty old chipset on a good mothercard
the only informations i can find are these ...
Quote
1. Meets premium audio requirements for Microsoft WLP 3.10
2. Meets stricter performance requirements for future WLP
3. High-performance DACs with 97dB Signal-to-Noise Ratio (SNR), ADCs with 90dB SNR
4. Four stereo DACs (8 channels) support 16/20/24-bit PCM format for 7.1 sound playback.
5. Two stereo ADCs (4 channels) support 16/20/24-bit PCM format recording simultaneously
6. All DACs supports 16/20/24-bit, 44.1k/48k/96k/192kHz sample rate
and
Quote
The ALC887 is a 7.1 Channel High Definition Audio Codec with two independent SPDIF outputs. Featuring eight channels of DAC support 7.1 sound playback, and integrates two stereo ADC that can support a stereo microphone, and feature Acoustic Echo Cancellation (AEC), Beam Forming (BF), and Noise Suppression (NS) for voice applications.  ALC887 is designed not only to meet the premium audio performance requirements in current WLP3.10 (Windows Logo Program), but provides better characteristics for future WLP. That brings user real high fidelity of sound quality.
4
FLAC / Re: FLAC - stored main chunk length differs from written length
Last post by bennetng -
Apart from the observed behaviour of wrong.wav posted above when using the APE/FLAC/WavPack command line tools and various 3rd party decoder/players, I also found that even though there is only one byte of difference between correct.wav and wrong.wav, 7z achieved a much smaller file size when compressing correct.wav and wrong.wav separately (652KB vs 713KB). Also, I cannot reduce file size by putting these two files into a solid 7z. It seems that 7z used different methods to compress correct.wav and wrong.wav so solid compression doesn't work. I tried different settings on the 7z GUI but still cannot find a way to defeat this behaviour, don't know if it can be overridden by using the 7z command line tool or not.

Anyway, good job on flac.exe to reject wrong.wav.
5
General Audio / Looking for a freeware 64 bit Stereo Convolution VST
Last post by Cannonaire -
I hope this is the right place to ask, as it is my first time posting outside the foobar2000 sub forum. I use a couple stereo impulses I found several years ago when I listen to music on headphones, and foobar2000 has a stereo convolution component that works perfectly with them. No problems there. I want to also use these impulses for listening to other things, such as music in a web browser. I have EqualizerAPO installed, and while it doesn't have a built-in stereo convolver (only mono), it does accept 64 bit VSTs. After searching for several hours, I have only found 32 bit VSTs, expensive paid-for VSTs, and one technically-free 64 bit VST that requires me to enter payment information in order to download (hmm...). Would anyone be able to link me to a good one? Again, I need a freeware 64 bit Stereo Convolution VST (usually these are labeled as True Stereo Convolution Reverb VSTs).
6
General Audio / Re: 89 db replaygain too much for classical music
Last post by timcupery -
The dramatically easier way I've found to do this is to encode to +1.5 or +3 or +4.5 (etc.) db above the target volume, and then use fb2k or mp3gain to lower the resulting file by that same amount. I'm doing this with FB2k, and this may engage some anti-clipping algorithm that I'm not aware of and for which I can't find a switch. I haven't tried it with commandline or LameDropXPd to see if baseline LAME encoder does the same thing.

Here's an example of a song that I'd encoded years ago (original was with LAME 3.92) that I just left much quieter than I'd prefer it to be. Recently I scaled it up to volume that sounded right (it's one of these weirdly-mastered indie songs where RG really misdiagnoses the perceptual loudness) and used the method I described above to get the same RG volume, without any clipping. This peak-lowering only decreased the RG value by 0.12 dB.

Note: I'm unable to ABX-differentiate the no-clipping version from the WAV, even though I did a fair comparison with scaling them down to the volume where the WAV isn't clipping, and just turning up my speakers.

Attached are screenshots showing the RG and peak values, as well as wav exports of those two mp3s (the 100% one has clipped peaks that go up to 189%)
8
General Audio / audio editor recommendations (I'm still using EAC's sound processing)
Last post by timcupery -
I got started using the built-in sound processing from Exact Audio Copy, in the fairly early days of HA. I still use it and am comfortable with it, although I'm older and have a family so I don't do much anymore.

EAC's sound processor is limited to normal CD-rip (44.1 16-bit stereo wav) files, and I'm increasingly getting lossless files with higher samplerate or bitdepth (e.g., buying remastered albums through Bancamp).

I've thought for over a decade that I should switch audio editors, but haven't put time into figuring out something that can handle a wider range of files.

Things I like about EAC's sound processing:
* basic wavform display, can zoom in horizontally (time interval) and vertically (wavform height)
* can click on individual samples to see/edit the exact value
* interval bars for CD audio frames (588 samples, 75 per second at 44.1khz)
* options such as "select peak range" and "scale selection"
* doesn't change any sample values unless I tell it to (which I expect to be standard behavior for handling any lossless file)

Audacity is the audio editor I'm most aware of, and I know it handles a wider range of file specifications and types. Will it (or some other app) do the things I'm looking for that EAC's editor does?
9
General Audio / Re: 89 db replaygain too much for classical music
Last post by timcupery -
I don't often encode classical music, but I'm aware of this problem and have encountered it from time to time in rock/pop music (hardly ever with more-recently-produced-or-remastered albums, which tend to be more compressed).

My old solution, when I was in early 20s and single, was to manually select each offending peak (from where that the amplitude line crossed 0 dB, to where it crossed back again, so peak about in the middle of selection) and scale that whole selected area so the peak was compatible with my target scaling on encode. As one might imagine, this was quite time-intensive, and the sort of thing a programmed loop or some other algorithm could have handled much more efficiently.

The dramatically easier way I've found to do this is to encode to +1.5 or +3 or +4.5 (etc.) db above the target volume, and then use fb2k or mp3gain to lower the resulting file by that same amount. I'm doing this with FB2k, and this may engage some anti-clipping algorithm that I'm not aware of and for which I can't find a switch. I haven't tried it with commandline or LameDropXPd to see if baseline LAME encoder does the same thing.
10
General Audio / Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy
Last post by timcupery -
LAME can, by commandline, handle resampling internally while encoding. For my case, starting with a higher-samplerate input file, I would add
--resample 44.1
to the commandline

It doesn't appear to FB2k can tell LAME to do this, nor can LameDropXPd. I've usually tagged my files with RG album gain tags, and with commandline I may have to translate those into --scale X commands. For example, an album gain of -6.0 dB would translate into --scale 0.5