FFSoX Player plug-in released

[pbelkner]

Originally Posted by SilverBird775

I'm using my own plugin, "out_maiko.dll".

I've just started to make myself familiar with "out_maiko.dll", and I've made the following observation.

Usually I'm using "in_ffsox.dll" with up-sampling set to 192.0 kHz and "out_asio(dll).dll". This particular setup switches my E-MU 0404 to 192.0 kHz:


If I copy "out_maiko.dll" to "Winamp/Plugins" and the E-MU 0404 is set to 192.0 kHz I get the following error message when starting Winamp:


The only way to resolve this is to manually set the E-MU 0404 to 48.0 kHz:


The same happens if I've played 44.1 kHz audio without re-sampling via ASIO because then the E-MU 0404 is switched to 44.1 kHz.


This lets me suspect that "out_maiko.dll" re-samples to 48.0 kHz (or whatever is set in Window's audio properties) via kmixer regardless on whatever the intention of "in_ffsox.dll" is. Am I wrong?

[SilverBird775]

You have set it this way, not me. My plugin is highly configurable. It's an audio processor atop of WASAPI. It will work the way you have set it not the way you suppose it to work.

The error message above saying that the device driver deny setting WASAPI SHARED mode. I mind you that plugin cannot set it's own format in SHARED mode, it takes whatever Windows Mixer say by default. See the shared mode format in Windows Mixer settings. You have set it to 48000 hz, not me.

For a custom windows mixer independent mode to set the device need to be driven in EXCLUSIVE mode. ASIO is bypassing the windows mixer and have the brand special optimizations. The wrecked drivers (like an early ESI Juli@ beta drivers) could switch the Master Sample Rate for ASIO but could not do so for WASAPI. In that case you have to set any sample rate in manual even for Exclusive mode both in windows software and driver panel. Definately it's not a Microsoft fault.

To understand the plugin settings please read the User Manual. There is really a lot of combinations possible and what it seems to be correct is really depends on what you trying to get:
http://maiko.elementfx.com/forum/index.php?topic=7.0

How to allow Exclusive mode in windows:
http://maiko.elementfx.com/forum/index.php?topic=9.0

How to understand what Maiko gets on input and what it actually output:
http://maiko.elementfx.com/forum/index.php?topic=10.0

For your card it seems there must be a special kick in E-MU panel may be required. Your card is the first one ever reported that unable to work in Shared mode. For a WASAPI mode switch to work the driver should allow automatic sample rate switch. That is the only legit behaviour for Vista and Win7. Your card seems locking the sample rate with exception for it's own E-MU ASIO. How the windows will be able to switch to custom sample rate then?!

[pbelkner]

Originally Posted by SilverBird775

Your card seems locking the sample rate with exception for it's own E-MU ASIO.

That's why I prefer ASIO. Using ASIO the audio is delivered to the DAC without being altered.

Originally Posted by SilverBird775

How the windows will be able to switch to custom sample rate then?!

It doesn't matter, I try to support your plug-in as requested, i.e. support multi-channel. The only thing I have to think of is that the audio is re-sampled to 48.0 kHz before being delivered to the DAC, the same as it is with with Direct Sound, and I have to reset the E-MU to 48.0 kHz before using the respective output plug-in, either your's or DS.

[SilverBird775]

pbelkner
That's right, that's because you had set it this way, to resample. The Maiko plugin just doing exacly what the user had instructed it to do, to resample in your case. Had you wanted it to not resample, you would check the mark on exclusive tab page to slave the output rate to input format.

As i'm already told it's not a dumb copy plugin, but a highly tweakable tool. By default the plugin has set to follow the default windows mixer policy. But it can indeed do such a simple thing as bit-copy, but also conditional bit-copy, forced processing, etc. The plugin follow whatever you instruct it to do but it seems you did not even tried to change the default settings.

I did not asked the in_ffsox to downsample channels either, so lets be reasonable please. If you want my plugin to just copy sound without processing then follow the manual please. It would not work bit-copy way right out of the box and need all the format binding options to be set so the user will be fully responsible for letting the software to reconfigure the hardware.

If you want some help please don't hesitate to ask i'm here to give a hand.

[pbelkner]

Originally Posted by SilverBird775

but a highly tweakable tool.

I think that's the point. The two plug-ins, your's and mine, have some overlap in functionality, and the user has to know how to deal with it.

[SilverBird775]

Way too generalized. The user has to know how to deal with the audio player to play the music, am i right? Yes i am.

The sad story for Maiko plugin did not work right out of the zip on your PC is truly unprecedented and the only reason for this is the below the standard WASAPI driver. It's not a case for any other PC because it just work there. So, some overlapping in functionality thing and knowing how to deal with it thing are not interconnected.

[pbelkner]

Originally Posted by SilverBird775

Way too generalized. The user has to know how to deal with the audio player to play the music, am i right? Yes i am.

Do you know that some people from Turkey really like this combination?

http://forum.donanimhaber.com/m_5372..._1/key_/tm.htm

[kerimcem]

Originally Posted by pbelkner

Do you know that some people from Turkey really like this combination?

http://forum.donanimhaber.com/m_5372..._1/key_/tm.htm

yes
when have this plugin ID3 tag support?
thanks..

[Mark Pollock]

I would like to thank you for sharing your thoughts and time into the stuff you post!!

[pbelkner]

Version 0.4.7 released:

Home: http://in-ffsox.sourceforge.net/
Download: http://sourceforge.net/projects/in-ffsox/files/

What's new?

• According to FFmpeg.org: FFmpeg development has gone into OVERDRIVE. As a consequence the FFSoX Player plugin wasn't compiling any longer using the latest FFmpeg versions. This release ports the FFSoX Player plugin to the latest FFmpeg API.
• Video synchronization has been greatly improved by replacing heuristics with a call to av_opt_ptr(avcodec_get_frame_class(), frame, "best_effort_timestamp");
• This release requires "avformat-54.dll" and "avcodec-54.dll", and is now again in line with the latest FFmpeg builds from http://ffmpeg.zeranoe.com/builds/win32/shared/.

[pbelkner]

Version 0.4.8 released:

Home: http://in-ffsox.sourceforge.net/
Download: http://sourceforge.net/projects/in-ffsox/files/

What's new?

• Upgraded to new SoX 14.4.0.
• Restructured build process for the tool chain.

[Koopa]

pbelkner:

I was adding your FFSoX Player plug-in settings to my Winamp Backup Tool and noticed, that you always write the config file in Program Files\Winamp\Plugins\in_ffsox

That's problematic for users which are no administrator on the system, because they have no permission to save their settings to this folder. Instead you should check the path.ini and write the file in the multi user path (Usually, most installs have their settings in %AppData%\Winamp\Plugins). I'm sure it's documented in the Winamp SDK how to access the multi user path correctly.



-Chris

[MrSinatra]

i am concerned about a much earlier post, i have this sound card:

http://support.creative.com/Products...X-Fi+Elite+Pro

is the contention for that audigy type soundcard that everything is upsampled to 48k for playback? i did not know that if true. that would bother me, i don't like to alter the source specs when ripping or playing back. does this plugin then somehow get around this? how?

[pbelkner]

Originally Posted by Koopa

I'm sure it's documented in the Winamp SDK how to access the multi user path correctly.

Thanks for the hint, despite I'm currently not certain on how to deal with it ...

[DrO]

use IPC_GETINIDIRECTORY(W) from wa_ipc.h and then append \Plugins\in_ffsox to get the correct location - that api has been around since 2.9x

-daz

[pbelkner]

Originally Posted by MrSinatra

i am concerned about a much earlier post, i have this sound card:

http://support.creative.com/Products...X-Fi+Elite+Pro

is the contention for that audigy type soundcard that everything is upsampled to 48k for playback? i did not know that if true. that would bother me, i don't like to alter the source specs when ripping or playing back. does this plugin then somehow get around this? how?

I'm not certain what you are asking for. Elsewhere somebody told me that there are some Audigy sound card sampling up to 48k with bad quality and that FFSoX helps to get around this because if the audio is already delivered to the DAC at 48k there's no need for the DAC to up-sample.

If you don't like FFSoX up-sampling you can switch it off. From my perspective you then miss one of it's most important features ...

[MrSinatra]

i'm sorry, i was unclear. i am not yet using your plugin.

what i was asking, was in response to an earlier post in the thread. someone seemed to suggest that if a person has flacs or mp3s or whatever, ripped from cds, that then have a 44.1khz spec, that the audigy would take those files, and on playback, upsample them to 48khz, and that this was something the card forced on everyone, like it or not.

is that true? b/c i don't like it if so.

and does your plugin get around that? or does it also go to 48khz, but this was desired b/c the plugin did so better than the hardware on the card?

as i said, i have a card like that, (i think), and my "best case scenario" would be for both the card and any software combo to just playback the file native to whatever the file says the specs are... is this possible?

(i am somewhat out of my depth here, so forgive me if this sounds like idiotic rambling, i'm just trying to understand)

[pbelkner]

Originally Posted by Koopa

pbelkner:

I was adding your FFSoX Player plug-in settings to my Winamp Backup Tool and noticed, that you always write the config file in Program Files\Winamp\Plugins\in_ffsox

That's problematic for users which are no administrator on the system, because they have no permission to save their settings to this folder. Instead you should check the path.ini and write the file in the multi user path (Usually, most installs have their settings in %AppData%\Winamp\Plugins). I'm sure it's documented in the Winamp SDK how to access the multi user path correctly.



-Chris

Originally Posted by DrO

use IPC_GETINIDIRECTORY(W) from wa_ipc.h and then append \Plugins\in_ffsox to get the correct location - that api has been around since 2.9x

-daz

Version 0.4.9 released:

Home: http://in-ffsox.sourceforge.net/
Download: http://sourceforge.net/projects/in-ffsox/files/

What's new?

• Changed looking up "in_ffsox.ini" as suggested by DrO.

[Koopa]

Dunno if it's just me, but that version does not remember the settings for me after a Winamp restart. I also cannot find any in_ffsox.ini file in my user profile folder:

C:\Users\Chris\Appdata\Roaming\Winamp\Plugins

Running Winamp on Windows 7.

Once yo read the correct path with the APi do you save the file to the path you got from it?

[pbelkner]

Originally Posted by Koopa

Dunno if it's just me, but that version does not remember the settings for me after a Winamp restart. I also cannot find any in_ffsox.ini file in my user profile folder:

C:\Users\Chris\Appdata\Roaming\Winamp\Plugins

Running Winamp on Windows 7.

Once yo read the correct path with the APi do you save the file to the path you got from it?

Just a minute ago I've uploaded a corrected version. Please tray again. Sorry for any inconvenience.

Regards, Peter

[Koopa]

That's working just fine now.

Will update Backup Tool for your updated settings file.

Cheers,
Chris

[kerimcem]

ffmpeg-latest-win32-shared.7z 26-Apr-2012 02:43 5.7M
library mp3 problem....
http://f1204.hizliresim.com/w/x/51n97.jpg
song play not seek
thanks..

[pbelkner]

Originally Posted by Koopa

That's working just fine now.

Will update Backup Tool for your updated settings file.

Cheers,
Chris

Thanks a lot

[pbelkner]

Originally Posted by kerimcem

ffmpeg-latest-win32-shared.7z 26-Apr-2012 02:43 5.7M
library mp3 problem....
http://f1204.hizliresim.com/w/x/51n97.jpg
song play not seek
thanks..

I can confirm that this particular build (ffmpeg-20120426-git-a4b58fd-win32-shared.7z) from Zeranoe (it's not me) is somehow broken.

On the other hand I've created my own extended build (including MP3) from a fresh GIT checkout this morning (29.04) and it seems to work perfect.

I've send you a respective PM.

[pbelkner]

Originally Posted by MrSinatra

i'm sorry, i was unclear. i am not yet using your plugin.

what i was asking, was in response to an earlier post in the thread. someone seemed to suggest that if a person has flacs or mp3s or whatever, ripped from cds, that then have a 44.1khz spec, that the audigy would take those files, and on playback, upsample them to 48khz, and that this was something the card forced on everyone, like it or not.

is that true? b/c i don't like it if so.

As I've learned from some discussions over there at HA forum (http://www.hydrogenaudio.org/forums/) this is true for EACH descent DAC. It is not evil but it is GOOD because it helps the DAC in reconstructing the original analogue signal out of the digital, i.e. it helps the DAC in smoothing the "digital stair case".

Originally Posted by MrSinatra

and does your plugin get around that? or does it also go to 48khz, but this was desired b/c the plugin did so better than the hardware on the card?

Probably each re-sampling algorithm colors the sound in one or the other way. Some may sound more smooth others a bit harsh. If you don't like the sound of you're DAC's build-in re-sampler you may try SoX.

Originally Posted by MrSinatra

as i said, i have a card like that, (i think), and my "best case scenario" would be for both the card and any software combo to just playback the file native to whatever the file says the specs are... is this possible?

(i am somewhat out of my depth here, so forgive me if this sounds like idiotic rambling, i'm just trying to understand)

As already stated it is up-sampled anyway. The only question is whether up-sampling happens at your PC or at you're DAC.

[pbelkner]

PLEASE NOTE: The Winamp installer destroys this plugin's DLLs! After having (re-)installed Winamp you have to re-install this plugin (including all DLLs from the "in_ffsox" sub-folder) to have working DLLs again!

For details please refer to this message.

[pbelkner]

Originally Posted by pbelkner

PLEASE NOTE: The Winamp installer destroys this plugin's DLLs! After having (re-)installed Winamp you have to re-install this plugin (including all DLLs from the "in_ffsox" sub-folder) to have working DLLs again!

For details please refer to this message.

DJ Egg just announced that the issue is resolved with the next WA release:

Originally Posted by DJ Egg

Done

in_ffsox.dll and the Plugins\in_ffsox folder will not be touched by the BindImageEx Optimizer as from the next release onwards :-)

I just need to know the filenames of any other incompatible plugins now....

[Stone Free]

Originally Posted by pbelkner

I can confirm that this particular build (ffmpeg-20120426-git-a4b58fd-win32-shared.7z) from Zeranoe (it's not me) is somehow broken.

On the other hand I've created my own extended build (including MP3) from a fresh GIT checkout this morning (29.04) and it seems to work perfect.

I've send you a respective PM.

Hi could you do the same for me. It works on a small subset of files but others just don't play, the ones that don't play appear as if the file has been deleted, i.e it shows at the top but there is no information about the file. The same thing happens when the file behind the playlist is deleted.

[Stone Free]

Scrap that it seems to be working all of a sudden. However could you make the next version not ignore Foobars replay gain tagging of m4a files as with your plugin all the replaygain info is not showing.

Also in the plugin section how do you activate the dropdowns for AC3/MP3?
Is there any way to configure out_asio to not cause dropouts when the dialog is dismissed. Also when Winamp starts up there is a thump like plugging in your guitar lead whilst the amp is on.

[Stone Free]

What exactly does the Algorithm choice between ReplayGain and EBU R128, and what is the significance of how the slider is always named the opposite of the choice you select?

[Stone Free]

I've noticed that your input plugin doesn't see any of my AAC ReplayGain tags.

[thinktink]

Would it be possible to add the transcoding/decoder funtions support to this plugin?

The section in the documentation relating to the decoding functions.
winampGetExtendedRead_open
winampGetExtendedRead_open_float
winampGetExtendedRead_openW
winampGetExtendedRead_openW_float
winampGetExtendedRead_getData
winampGetExtendedRead_setTime
winampGetExtendedRead_close

[pbelkner]

Originally Posted by Stone Free

Hi could you do the same for me.

This is not an issue any longer. You should update FFmpeg from Zeranoe's builds: http://ffmpeg.zeranoe.com/builds/win32/shared/.

Originally Posted by Stone Free

It works on a small subset of files but others just don't play, the ones that don't play appear as if the file has been deleted, i.e it shows at the top but there is no information about the file.

You should double check whether you've added the file's extension to the FFSoX configuration (via the option menu).

Please have in mind that FFSoX can only play formats/codecs which are supported by FFmpeg.

[pbelkner]

Originally Posted by Stone Free

However could you make the next version not ignore Foobars replay gain tagging of m4a files as with your plugin all the replaygain info is not showing.

Unfortunately I'm not an FB expert. Could you please let me know which tags FB writes?

Originally Posted by Stone Free

Also in the plugin section how do you activate the dropdowns for AC3/MP3?

These options are not supported any longer. All codecs are taken from FFmpeg. No external library is needed any longer. I should remove the drop down boxes with the next version.

[pbelkner]

Originally Posted by Stone Free

What exactly does the Algorithm choice between ReplayGain and EBU R128, and what is the significance of how the slider is always named the opposite of the choice you select?

RG and EBU R128 are two standards addressing loudness normalization, cf. http://wiki.hydrogenaudio.org/index...._specification and http://tech.ebu.ch/loudness, respectively. The assumed difference between the tow's target loudness is about 5 dB. If you switch between the two the slider is updated accordingly.

[pbelkner]

Originally Posted by Stone Free

I've noticed that your input plugin doesn't see any of my AAC ReplayGain tags.

As already mentioned, you could help improve FFSoX by letting me know which tags are used for RG.

[pbelkner]

Originally Posted by thinktink

Would it be possible to add the transcoding/decoder funtions support to this plugin?

The section in the documentation relating to the decoding functions.
winampGetExtendedRead_open
winampGetExtendedRead_open_float
winampGetExtendedRead_openW
winampGetExtendedRead_openW_float
winampGetExtendedRead_getData
winampGetExtendedRead_setTime
winampGetExtendedRead_close

I have this in mind for a long time. First I have to understand how FFmpeg implements transcoding. This will take some further time.

Many thanks for the link anyway.

[Aminifu]

Originally Posted by pbelkner

RG and EBU R128 are two standards addressing loudness normalization, cf. http://wiki.hydrogenaudio.org/index...._specification and http://tech.ebu.ch/loudness, respectively. The assumed difference between the tow's target loudness is about 5 dB. If you switch between the two the slider is updated accordingly.

Hi,

Which is louder, RG or EBU? I can't tell from the links (different terminology). RB seems to use 89 dB.

[pbelkner]

Originally Posted by Aminifu

Which is louder, RG or EBU?

RG is louder. But you should have in mind that louder means a greater chance of clipping. You should avoid clipping by any means and compensate by using the respective knob of your amplifier.

You should also have in mind that RG is not due to some standards body.

On the other hand EBU R128 is due to the European Broadcasting Union (EBU). In the U.S.A there is the similar standard, ATSC A/85, due to the Advanced Television Systems Committee (ATSC). Both, EBU R128 and ATSC A/85, are flavors of the ITU BS.1770 standard due to the International Telecommunication Union (ITU).

The main difference between EBU R128 and ATSC A/85 is a slight difference of the target loudness, EBU R128 is -23 dBFS and ATSC A/85 is -24 dbFS.

If you are free to choose you should prefer EBU R128 or ATSC A/85, depending on where you are located. You can expect the local TV and radio stations to adapt the respective standard in the near future (if not already done).

You may consider using R128GAIN for loudness analysis and tagging. Please have look at the respective thread at HA forum.

[Aminifu]

Originally Posted by pbelkner

RG is louder. But you should have in mind that louder means a greater chance of clipping. You should avoid clipping by any means and compensate by using the respective knob of your amplifier.

Thanx for the info. I agree with your comments about clipping. So if WA RG's target level is 89 dB, the EBU target level would be about 84 dB.

The vast majority of my files are mp3s and I use "MP3 Gain" (link below) to volume level them between 89.5 and 88 dB to avoid clipping. The 1.5 dB difference is not usually noticeable. This app has a GUI and uses a variation of David Robinson's Replay Gain algorithm. Unlike WA's implementation, the target loudness level is selectable and RG tags are not placed in the files. The 'global gain' field in each mp3 file is directly changed (without having to re-encode the mp3) to achieve the target loudness, up or down. This allows the modified mp3s to play in apps and devices (like my portable media player) that do not recognize RG tags.

http://mp3gain.sourceforge.net/faq.php

Anyway, its good to know that "R128GAIN" is available and provides a selectable target loudness level. Another tool for the toolbox. I assume WA has no problems using the RG tags it makes.

Too bad it does not work with the wavpack format. I prefer wavpack over flac because it uses the ID3 tagging standards. Unless I can find a way to volume level the wavpack format, I may have to switch to flac for a lossless format and learn Vorbis tagging. I do not like to use normal normalization of wav files before converting to wavpack or mp3, it flattens the music's dynamics too much.