I have neither special backroung in linux audio, nor tried another distribution in my system. I know (it is easily audible) that:
1) lowering the latency in both windows and linux improves the sound
2) after setting the same latency Audiophile linux sounds better than Windows XP.
3) Jack sounds better than alsa and pulse. This is possibly due to the ability to adjust latency.
Is there another way besides Jack to lock exclusively the sound device, playback in realtime process priority and -most important- adjust the latency?
Do you have in mind a better audio oriented distribution? I would be happy to try it.
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Unless you did a blind listening test I do not see how you could "know" it. You feel it sounds that way, having read many also subjective reports as well. I have also read a few subjective reports prefering the larger latency. All observations being subjective, there is no point in discussing technical causes.
My blind test on my gear revealed I was not able to tell any difference between low and large latency. How will your test end up? E.g. you can use my rudimentary script Tool for A/B Testing | Blog IVITERA a.s.
Jack adjusts latency via alsa. It is just a layer above it. Most decent playback applications can adjust buffer size in its alsa output module. Locking sound device is just a question of which alsa output device string you use. Jack just shields you away from the lower-level mechanism and hides the actual principles from you. IMO you being a technical guy want to get down to the core and learn the real stuff, not just a subset of features jack developers provided for DAW users/pro's.
It really depends what you are looking for. General-purpose PC, streamlined PC for audio only, embedded headless device - many choices. They all need some linux knowledge to setup reasonably, nothing complicated, just not being afraid of command line (which I assume you as a developer of command-line pureplay are not
).Each one has it's own methods of auditioning a system or component.
After many years, knowing my system very well i tend to listen better with eyes open but i performed some blind tests with friends and all 3 of them preffered audiophile linux with jack and latency as low as possible. The key word here is "prefer", because taste varies. So i like the fast, raw, attacking low latency setting rather than the safe, sweet-ish, fluffy and lazy normal one. Perhaps it's all a mater of taste.
Jack adjusts latency via alsa. It is just a layer above it. Most decent playback applications can adjust buffer size in its alsa output module. Locking sound device is just a question of which alsa output device string you use. Jack just shields you away from the lower-level mechanism and hides the actual principles from you. IMO you being a technical guy want to get down to the core and learn the real stuff, not just a subset of features jack developers provided for DAW users/pro's.
It really depends what you are looking for. General-purpose PC, streamlined PC for audio only, embedded headless device - many choices. They all need some linux knowledge to setup reasonably, nothing complicated, just not being afraid of command line (which I assume you as a developer of command-line pureplay are not
Can you suggest me a player or another direct way to adjust latency? Unfortunately i don't have enough time to "go deep" right now...
ps. your tool looks great!
I have been at this countless times, never ever anyone claiming to hear the low latency effect dared to take the blind test. I did on my gear, with results well below confidence threshold for my ears.
If you want to (which I very much doubt), you can take the test using that script.
Any direct alsa player which lets you setup buffer and/or period size, such as mpd (parameter period_time). For basic players for testing purpose (e.g. the blind test) the easiest tool is aplay with parameters --period-time and --buffer-time. If you want to, we can talk about what these actually do so that you see what this low latency thing actually does in the system. Most of this is hidden from you in windows.
If you want to (which I very much doubt), you can take the test using that script.
Any direct alsa player which lets you setup buffer and/or period size, such as mpd (parameter period_time). For basic players for testing purpose (e.g. the blind test) the easiest tool is aplay with parameters --period-time and --buffer-time. If you want to, we can talk about what these actually do so that you see what this low latency thing actually does in the system. Most of this is hidden from you in windows.
I did the blind test with a help of a friend and it was easy to tell each time which setting is A (10ms latency) and B (2ms latency). I 'm afraid though that the matter is more complicated than concentrating on latency. In both Linux and Windows there are different results with different combinations of buffer size, latency and sample rate. The "key" IMHO is not latency itself but the right combination of the above, although as a general rule things are better as latency is lowered. This may be also dependant on the DAC or USB(firewire) interface used. Moreover, most of the players and plugins sound differently.
In Audiophile Linux IMO Deadbeef is the best, Audacious and gmusicbrowser just good, alsaplayer and musique not so good... Regarding output plugins: Jack is very good, Pulse good, Alsa not so good. So Deadbeef+Jack+low latency is the best for my ears-system but can anyone explain why?
@phofman: Have you tried the above combinations? Is the "audible" playback result similar to you?
In Audiophile Linux IMO Deadbeef is the best, Audacious and gmusicbrowser just good, alsaplayer and musique not so good... Regarding output plugins: Jack is very good, Pulse good, Alsa not so good. So Deadbeef+Jack+low latency is the best for my ears-system but can anyone explain why?
@phofman: Have you tried the above combinations? Is the "audible" playback result similar to you?
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That is very interesting, please yould you provide more details about the test? What command you tested, how the test was administered, what results. Thanks a lot
So basically you say that going through jack into alsa sounds better than going into alsa directly. Do you see where my doubts come from? Jack is just an additional layer with multiple sample conversions.
I will not delve into technical explanation of subjective feelings.
The combination itself says nothing about the actual setup - output plugins, configuration of the output plugins, etc.... There can be countless differences. Plus "sounds better" is extremely individual - everyone prefers something else. The only relatively objectively comparable result is "showing a difference can be identified/heard".
No commands, just changing parameters in jack or pureplayer and playing 15 seconds of the song randomly. It is very easy to guess the settings each time.
Yes. Obviously if players could utilize alsa the way jack does the result would be better without using jack.
So you cannot tell me if you hear any difference with any combination.
I am very sorry, but this is our hobby and if we cancel our hearing there is really nothing to discuss here. Its subjectiveness may be its beauty. Excuse me but i would not like to go any further on this conversation.
I do not belive distinguishing between 2ms and 10ms latencies is "easy" unless something is fundamentally wrong in your system (e.g. audible dropouts in the 2ms setup, interrupts causing audible noise in your system).
I see nothing obvious in this. In addition, the player itself plays the same, whatever latency in jack you setup, only the writing thread (producer) is awoken by jack more often, just like the driver (CPU) is called by the soundcard interrupt more often. The playback application thread and the soundcard DMA transfer run in the same pace, whatever the latency setup in between.
Of course you can setup alsa playback for the same latency like jack, in the end it is always alsa handling the data transfer into the soundcard's DMA region in RAM. Jack itself does not reduce any latency at all (how could it), it is just a plumbing infrastructure.
My ears neither my wife's did not hear any difference in latency on my setup in a properly administered blind test (i.e. we did not know what alternative was being played). Our correct results fell below the threshold of being statistically significant to become more than random guesses.
Discussing technical causes for subjective feelings is definitely not something I want to do. Good luck.
That is one problem that is coming up when asking for blind tests. The informed audiophile just will add to his signature: "All subjective claims are blind tested" and done. No interest in doing this consequent.
I don´t want to smaller the work of npetralias but imagine a double-blind-test shows no result? All those years of work for nothing? The fan crowd, the fun...
I don´t want to smaller the work of npetralias but imagine a double-blind-test shows no result? All those years of work for nothing? The fan crowd, the fun...
Been using Pureplayer on and off for about a month now. I would like to make it my default player but there are a couple of little niggles I need fixed first.
1. More often than not it tries to buffer every song in the list and sits at the end without playing any, really need that working better.
2. A pause button so I don't have to restart a song if disturbed.
3. Not essential but track slide would be nice
Other than those couple of small things its a really great player and the best sound I have heard so far.
1. More often than not it tries to buffer every song in the list and sits at the end without playing any, really need that working better.
2. A pause button so I don't have to restart a song if disturbed.
3. Not essential but track slide would be nice
Other than those couple of small things its a really great player and the best sound I have heard so far.
Let's say you want to load a directory of FLAC files.. you click on "Load Dir" button and select the necessary folder (a quick access link to "Desktop" would be nice) from there you drag the scroll bar to the top (default to top would also be nice) and click on a song and press the play button. Sometimes it will process the song and start playing and other times it will just start processing the next song all the way to the bottom of the list without playing any of them.
Probably ~50% of the time it will just process everything and play nothing.
Probably ~50% of the time it will just process everything and play nothing.
This is not the expected behavior, so i suppose something is wrong with your setup.
Possible causes:
1) The default sound interface (card, dac etc) is not available at the moment.
2) The file (folder path) path is not available ta the moment.
I am guessing (2) especially if you are using a network share or a NAS (network attached storage). These devices sometimes go on idle state and may cause such problems. If the problem occurs again, please try to open the playlist folder using windows explorer and then retry the playback.
Thanks in advance for your feedback.
Possible causes:
1) The default sound interface (card, dac etc) is not available at the moment.
2) The file (folder path) path is not available ta the moment.
I am guessing (2) especially if you are using a network share or a NAS (network attached storage). These devices sometimes go on idle state and may cause such problems. If the problem occurs again, please try to open the playlist folder using windows explorer and then retry the playback.
Thanks in advance for your feedback.
Hello my friend.... I dont think its the players..... Alot of factors goes into it,lets look at them:npetralias said:
1) Digital instead of analogue (We cant hope for much)
2) Format - MP3,WMA,AAC -- I have found 64k WMA to sound the nicest of all formats.
Example 64k WMA stream: http://wms-rly.181.fm/181-awesome80s -- DOESNT THAT SOUND NICE??
3) Speakers your listening with -- Im listening with a tuner which makes it sound BETTER than if i was listening with standard computer speakers......
Im listening to that stream above with WMP9
For some strange reason I can't get it to do it now...
Shouldn't be a file location issue as I moved the files to a "music" folder on c:\ a while ago to make finding them quicker, and it is using the same internal laptop sound card every time.
I have even had the same problem with my home media PC which is connected directly to my active xover via a USB input. Only common thread is the same software and the same songs...
If it happens again I will try to take note of the conditions to help better diagnose. I also notice I am not getting the error message that was coming up now, the one saying the main player file couldn't be located. (fplayer or something like that).
Dude111 player for player this thing is better than any comparable players, yes there might be better ways to get better sound, but as a Windows player in my opinion it is right up there.
Shouldn't be a file location issue as I moved the files to a "music" folder on c:\ a while ago to make finding them quicker, and it is using the same internal laptop sound card every time.
I have even had the same problem with my home media PC which is connected directly to my active xover via a USB input. Only common thread is the same software and the same songs...
If it happens again I will try to take note of the conditions to help better diagnose. I also notice I am not getting the error message that was coming up now, the one saying the main player file couldn't be located. (fplayer or something like that).
Dude111 player for player this thing is better than any comparable players, yes there might be better ways to get better sound, but as a Windows player in my opinion it is right up there.
Beats VLC... I still use VLC when I have playback problems with PurePlayer, but sound quality on PurePlayer is better, more detailed and natural sounding.
Just did a blind test with my partner not telling her what I was doing so as not to skew the results.
I played a Bee Gees song using both PurePlayer and VLC, she liked the "fatter" sound of VLC, but could hear that PP was clearer.
Same song, same equipment and someone who doesn't know the first thing about audio could pick there was a difference. I didn't even tell her what I was testing for so she told me what she heard without any prompting.
Just did a blind test with my partner not telling her what I was doing so as not to skew the results.
I played a Bee Gees song using both PurePlayer and VLC, she liked the "fatter" sound of VLC, but could hear that PP was clearer.
Same song, same equipment and someone who doesn't know the first thing about audio could pick there was a difference. I didn't even tell her what I was testing for so she told me what she heard without any prompting.