Juha
Rank: Soloist
Posts: 24 
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Post by Juha on Jun 20, 2015 9:26:02 GMT
I did my own audio player about one year ago. Wanted a simple light-weight player that has no settings or menu. My options were Windows Wasapi or ASIO. Decided to do Wasapi, because more sound devices would work. ASIO is available for more semi-pro type recording setups. Wasapi has the Exclusive mode where just one application gets to use of the audio. Here is a picture:  This is a copy-paste player that plays exclusive mode Wasapi on default sound card. Set default sound card from Windows Control Panel / Audio. Here is the download link: StreamerPlus downloadUnzip files to the same directory. Select audio files with Windows Explorer mouse right-click and copy. Start the program StreamerPlus.exe Click the Paste-button and then play. There is .m3u playlist support. Paste a playlist and it is loaded for play. Added history-log playlist. There is a playlist for each day. This contains all files played on that day. Some issues on player programming: 1. There is a temptation to throw in some compression and eq. Microsoft DirectX has these services easily available. Reading and testing many players I am sure this has been done to enhance. I did not do it. 2. Memory play. I don't think so. Microsoft has spent years and years on optimizing everything. All things possible are buffered all over the place. So "memory play" is automatic. Proof for this is HD video. HD video just would not play without heavy buffereing. |
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Juha
Rank: Soloist
Posts: 24
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Post by Juha on Jun 20, 2015 9:29:05 GMT
Did a new version of StreamerPlus player.
Now the player puts itself to highest Windows priority class.
On a fast Windows 8.1 machine it goes to "realtime priority".
On older Windows 7 to "high priority" which is second highest.
Did some testing today:
Upsampling to 96k sounds good. Upsampling done from Control Panel / Audio / Advanced.
Set the Advanced params to a higher rate on default sound device.
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Post by MartinT on Jun 20, 2015 10:16:59 GMT
Nice one, Juha, and thanks for providing this for us to try out.
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Juha
Rank: Soloist
Posts: 24
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Post by Juha on Jun 21, 2015 11:33:01 GMT
Today did some work on the murky waters of
sample rate conversion quality.
The version available for download now up-samples with
32 point sinc interpolation. This is shown on the screen.
I am listening on normal office PC AMD dual core 2.9 ghz.
Hegel usb dac is set to 96k as default.
Is this better than the 8 point sinc I had before?
It takes a couple of days to find out.
One hour later:
Found a couple good settings for the audio library.
Another new version available.
I think this is not bad.
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Post by John on Jun 21, 2015 12:31:24 GMT
I think most players are using sample rate rate conversation to be honest. I like that some players allow you to play with this as the user yourself so you can make your choice to what suits you
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Juha
Rank: Soloist
Posts: 24
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Post by Juha on Jun 21, 2015 12:44:59 GMT
I think most players are using sample rate rate conversation to be honest. I like that some players allow you to play with this as the user yourself so you can make your choice to what suits you I agree. Just went with RaspberryPi / Volumio / IQDac. It took a long time to do the settings. Sample rate conversion seems to be the way to go. What I am doing here is tweaking my own player so that it comes close to the sound of commercial players. This is very compact, no settings or menus, installed in 5 seconds. |
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Post by MartinT on Jun 21, 2015 14:23:38 GMT
I have found upsampling to 24/96 in the Pi 2 to be beneficial. In the original Pi it was too much for the CPU.
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Juha
Rank: Soloist
Posts: 24
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Post by Juha on Jun 24, 2015 7:55:48 GMT
I have changed my mind about memory play.
Memory play version of StreamerPlus will be done soon.
Now that I have up-sampling working well memory play is the next step to try.
Had a look at Bug Head player. We use the same audio player library bass.dll.
Bass.dll from Un4seen Developments runs on the Microsoft Direct Sound api on Windows.
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Post by Clive on Jun 24, 2015 11:08:56 GMT
Juha, have you got the source code for Bug Head? I'm not sure whether it's openly available now but I have a copy in case it would be useful.
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Juha
Rank: Soloist
Posts: 24
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Post by Juha on Aug 9, 2015 19:46:43 GMT
Thank you for the Bug Head code Clive.
I was reading it during my vacation for light entertainment.
They have developed lots of statistical analysis to up-sample music.
When you up-sample, the missing waveform is generated by statistical analysis of existing data.
It is called sinc interpolation.
Does the statistics make better music. Once you know what is going on it loses some magic.
I tried to make my StreamerPlus player totally memory based. After a couple of days no success.
I did change it so that the internal data buffers are much larger. So less hard disk access.
The new version is on the link above.
I am using my own player a lot these days. The up-sampled sound to 96.000 is sort of impressive.
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bud11
Rank: Starter
Posts: 1
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Post by bud11 on Jun 7, 2017 5:39:38 GMT
hi juha, i like the sound of your player. do you still develop this software ?
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Post by MartinT on Jun 7, 2017 6:09:25 GMT
Unfortunately, we haven't heard from Juha in a while. Perhaps he'll see my reference here? |
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Post by yomanze on Jun 8, 2017 19:05:06 GMT
Thank you for the Bug Head code Clive. I was reading it during my vacation for light entertainment. They have developed lots of statistical analysis to up-sample music. When you up-sample, the missing waveform is generated by statistical analysis of existing data. It is called sinc interpolation. Does the statistics make better music. Once you know what is going on it loses some magic. I tried to make my StreamerPlus player totally memory based. After a couple of days no success. I did change it so that the internal data buffers are much larger. So less hard disk access. The new version is on the link above. I am using my own player a lot these days. The up-sampled sound to 96.000 is sort of impressive. Indeed, when you upsample / oversample the original file is effectively 'replaced', it's audible. Having spent quite some time using a non-oversampling DAC & feeding it with 44.1kHz, 88.2kHz & 96kHz I found the most natural was the 44.1kHz, the 88.2kHz better at the extreme top end (due to NOS DAC flaws) and found 96kHz more artificial - these were subtle differences, but enough to impact long-term listening enjoyment. Definitely the even oversampling sounds more natural to me, so is the maths. I also ABX test such files (as well as FLAC & 320k mp3) using Foobar 2000's ABX plugin & the same was confirmed. Funny how people say this stuff doesn't make a difference, but a lot of people seem to come to similar conclusions after doing the technical & theory bits. |
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