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From: Stefan H. <ste...@gm...> - 2011-12-08 19:48:09
|
On Thu, Dec 8, 2011 at 4:44 PM, Petter S. Storvik <pet...@gm...> wrote: > One more thing is; I cant start up computer if device is plugged in. Some > sort of error and it goes to text-mode, and then I have to restart without > the device. But all seems right if I plug it in after startup. This sounds like a kernel panic which can happen if an unexpected error occurs when the driver is running. There is typically a lot of text on the screen and it can contain information that allows developers to fix the bug that caused the panic. If you have a camera on your phone or digital camera, would you be able to capture the error screen? I'm guessing that the kernel panic has nothing to do with recording not working, they are probably different issues but there's a chance they are related. Stefan |
From: Petter S. S. <pet...@gm...> - 2011-12-08 16:44:12
|
Okey, now i have tried everything I could think of and I'm not getting anywhere. When I'm trying to record with audacity and ALSA I cant get any sound from the input. And when I use jack it seems like it instead uses the microphone on the computer. But I can see it with alsamixer and i can change the input. One more thing is; I cant start up computer if device is plugged in. Some sort of error and it goes to text-mode, and then I have to restart without the device. But all seems right if I plug it in after startup. 2011/12/7 Stefan Hajnoczi <ste...@gm...> > On Wed, Dec 7, 2011 at 7:03 AM, "Maciej Łoziński" <loz...@o2...> wrote: > > On Tue, December 6, 2011 10:43 pm, Stefan Hajnoczi wrote: > >> 2011/12/6 Maciej Łoziński <loz...@o2...>: > >>> > >>> I installed latest version from ux2 branch (rev.896) and it indeed has > >>> input source select (my bad, sorry). > >>> > >>> But I didn't manage to record anything from any input (I was using > >>> Audacity 1.3.13 and Arch linux with 3.1.1 kernel). Also JACK was > >>> disconnecting about 30 seconds after connecting, and it wasn't possible > >>> to record with Ardour+JACK. I also encountered a kernel panic once when > >>> plugging in a device. > >> > >> If you have problems with jack please send the debug messages (with > >> qjackctl you can get them by clicking on "Messages"). > >> > >> Stefan > >> > > > > Here is the output from qjackctl. There are some polish messages - > > "Sukces" means 'Success' and "Błędny deskryptor pliku" means 'Invalid > > file descriptor'. > > > > cheers > > Maciek > > > > 07:56:48.270 Patchbay deactivated. > > 07:56:48.292 Statistics reset. > > 07:56:48.346 ALSA connection change. > > 07:56:48.433 ALSA connection graph change. > > 07:57:11.811 JACK is starting... > > 07:57:11.812 /usr/bin/jackd -r -dalsa -dhw:1 -r44100 -p1024 -n2 > > jackd 0.121.3 > > Copyright 2001-2009 Paul Davis, Stephane Letz, Jack O'Quinn, Torben Hohn > > and others. > > jackd comes with ABSOLUTELY NO WARRANTY > > This is free software, and you are welcome to redistribute it > > under certain conditions; see the file COPYING for details > > 07:57:11.857 JACK was started with PID=1927. > > could not open driver .so '/usr/lib/jack/jack_firewire.so': > libffado.so.2: > > cannot open shared object file: No such file or directory > > could not open driver .so '/usr/lib/jack/jack_net.so': libcelt0.so.2: > > cannot open shared object file: No such file or directory > > JACK compiled with System V SHM support. > > loading driver .. > > apparent rate = 44100 > > creating alsa driver ... hw:1|hw:1|1024|2|44100|0|0|nomon|swmeter|-|32bit > > control device hw:1 > > configuring for 44100Hz, period = 1024 frames (23.2 ms), buffer = 2 > periods > > ALSA: final selected sample format for capture: 16bit little-endian > > ALSA: use 2 periods for capture > > ALSA: final selected sample format for playback: 16bit little-endian > > ALSA: use 2 periods for playback > > 07:57:14.088 JACK connection change. > > 07:57:14.121 Server configuration saved to "/home/maciek/.jackdrc". > > 07:57:14.122 Statistics reset. > > 07:57:14.126 Client activated. > > 07:57:14.135 Buffer size change (1024). > > ALSA: poll time out, polled for 34860373 usecs > > Not an error I'm familiar with but it could mean jack was trying to > read/write to the device but gave up after 34 seconds. > > Using arecord(1) or audacity with ALSA is a good first step at > troubleshooting. Adding jack or PulseAudio to the picture can make it > harder to see what is going on. Perhaps you'll get more clues by > running arecord(1) or audacity. For audacity you can check > .xsession-errors to see error messages if you started it from a GUI, > if you start it from a terminal then the error messages get printed to > the terminal. > > Stefan > > > ------------------------------------------------------------------------------ > Cloud Services Checklist: Pricing and Packaging Optimization > This white paper is intended to serve as a reference, checklist and point > of > discussion for anyone considering optimizing the pricing and packaging > model > of a cloud services business. Read Now! > http://www.accelacomm.com/jaw/sfnl/114/51491232/ > _______________________________________________ > Line6linux-devel mailing list > Lin...@li... > https://lists.sourceforge.net/lists/listinfo/line6linux-devel > -- Petter Sakrihei Storvik Ω |
From: Stefan H. <ste...@gm...> - 2011-12-07 08:12:08
|
On Wed, Dec 7, 2011 at 7:03 AM, "Maciej Łoziński" <loz...@o2...> wrote: > On Tue, December 6, 2011 10:43 pm, Stefan Hajnoczi wrote: >> 2011/12/6 Maciej Łoziński <loz...@o2...>: >>> >>> I installed latest version from ux2 branch (rev.896) and it indeed has >>> input source select (my bad, sorry). >>> >>> But I didn't manage to record anything from any input (I was using >>> Audacity 1.3.13 and Arch linux with 3.1.1 kernel). Also JACK was >>> disconnecting about 30 seconds after connecting, and it wasn't possible >>> to record with Ardour+JACK. I also encountered a kernel panic once when >>> plugging in a device. >> >> If you have problems with jack please send the debug messages (with >> qjackctl you can get them by clicking on "Messages"). >> >> Stefan >> > > Here is the output from qjackctl. There are some polish messages - > "Sukces" means 'Success' and "Błędny deskryptor pliku" means 'Invalid > file descriptor'. > > cheers > Maciek > > 07:56:48.270 Patchbay deactivated. > 07:56:48.292 Statistics reset. > 07:56:48.346 ALSA connection change. > 07:56:48.433 ALSA connection graph change. > 07:57:11.811 JACK is starting... > 07:57:11.812 /usr/bin/jackd -r -dalsa -dhw:1 -r44100 -p1024 -n2 > jackd 0.121.3 > Copyright 2001-2009 Paul Davis, Stephane Letz, Jack O'Quinn, Torben Hohn > and others. > jackd comes with ABSOLUTELY NO WARRANTY > This is free software, and you are welcome to redistribute it > under certain conditions; see the file COPYING for details > 07:57:11.857 JACK was started with PID=1927. > could not open driver .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: > cannot open shared object file: No such file or directory > could not open driver .so '/usr/lib/jack/jack_net.so': libcelt0.so.2: > cannot open shared object file: No such file or directory > JACK compiled with System V SHM support. > loading driver .. > apparent rate = 44100 > creating alsa driver ... hw:1|hw:1|1024|2|44100|0|0|nomon|swmeter|-|32bit > control device hw:1 > configuring for 44100Hz, period = 1024 frames (23.2 ms), buffer = 2 periods > ALSA: final selected sample format for capture: 16bit little-endian > ALSA: use 2 periods for capture > ALSA: final selected sample format for playback: 16bit little-endian > ALSA: use 2 periods for playback > 07:57:14.088 JACK connection change. > 07:57:14.121 Server configuration saved to "/home/maciek/.jackdrc". > 07:57:14.122 Statistics reset. > 07:57:14.126 Client activated. > 07:57:14.135 Buffer size change (1024). > ALSA: poll time out, polled for 34860373 usecs Not an error I'm familiar with but it could mean jack was trying to read/write to the device but gave up after 34 seconds. Using arecord(1) or audacity with ALSA is a good first step at troubleshooting. Adding jack or PulseAudio to the picture can make it harder to see what is going on. Perhaps you'll get more clues by running arecord(1) or audacity. For audacity you can check .xsession-errors to see error messages if you started it from a GUI, if you start it from a terminal then the error messages get printed to the terminal. Stefan |
From: Maciej Ł. <loz...@o2...> - 2011-12-07 07:03:19
|
On Tue, December 6, 2011 10:43 pm, Stefan Hajnoczi wrote: > 2011/12/6 Maciej Łoziński <loz...@o2...>: >> >> I installed latest version from ux2 branch (rev.896) and it indeed has >> input source select (my bad, sorry). >> >> But I didn't manage to record anything from any input (I was using >> Audacity 1.3.13 and Arch linux with 3.1.1 kernel). Also JACK was >> disconnecting about 30 seconds after connecting, and it wasn't possible >> to record with Ardour+JACK. I also encountered a kernel panic once when >> plugging in a device. > > If you have problems with jack please send the debug messages (with > qjackctl you can get them by clicking on "Messages"). > > Stefan > Here is the output from qjackctl. There are some polish messages - "Sukces" means 'Success' and "BÅÄdny deskryptor pliku" means 'Invalid file descriptor'. cheers Maciek 07:56:48.270 Patchbay deactivated. 07:56:48.292 Statistics reset. 07:56:48.346 ALSA connection change. 07:56:48.433 ALSA connection graph change. 07:57:11.811 JACK is starting... 07:57:11.812 /usr/bin/jackd -r -dalsa -dhw:1 -r44100 -p1024 -n2 jackd 0.121.3 Copyright 2001-2009 Paul Davis, Stephane Letz, Jack O'Quinn, Torben Hohn and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details 07:57:11.857 JACK was started with PID=1927. could not open driver .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory could not open driver .so '/usr/lib/jack/jack_net.so': libcelt0.so.2: cannot open shared object file: No such file or directory JACK compiled with System V SHM support. loading driver .. apparent rate = 44100 creating alsa driver ... hw:1|hw:1|1024|2|44100|0|0|nomon|swmeter|-|32bit control device hw:1 configuring for 44100Hz, period = 1024 frames (23.2 ms), buffer = 2 periods ALSA: final selected sample format for capture: 16bit little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 16bit little-endian ALSA: use 2 periods for playback 07:57:14.088 JACK connection change. 07:57:14.121 Server configuration saved to "/home/maciek/.jackdrc". 07:57:14.122 Statistics reset. 07:57:14.126 Client activated. 07:57:14.135 Buffer size change (1024). ALSA: poll time out, polled for 34860373 usecs DRIVER NT: could not run driver cycle 07:57:48.413 JACK connection graph change. jack main caught signal 12 cannot read server event (Sukces) cannot continue execution of the processing graph (BÅÄdny deskryptor pliku) zombified - calling shutdown handler 07:57:49.232 Shutdown notification. 07:57:49.236 Client deactivated. 07:57:49.241 JACK is being forced... 07:57:49.442 JACK was stopped successfully. |
From: Stefan H. <ste...@gm...> - 2011-12-06 21:43:44
|
2011/12/6 Maciej Łoziński <loz...@o2...>: > W dniu 06.12.2011 10:25, "Maciej Łoziński" pisze: >>> > >>> > Line6linux driver does not allow to select input source, it's fixed at >>> > default - "instrument input". >> Well, it actually*should* allow selection. Which device are you using (see >> also my previous message)? >> >> Kind regards, >> Markus > > > I installed latest version from ux2 branch (rev.896) and it indeed has > input source select (my bad, sorry). > > But I didn't manage to record anything from any input (I was using > Audacity 1.3.13 and Arch linux with 3.1.1 kernel). Also JACK was > disconnecting about 30 seconds after connecting, and it wasn't possible > to record with Ardour+JACK. I also encountered a kernel panic once when > plugging in a device. If you have problems with jack please send the debug messages (with qjackctl you can get them by clicking on "Messages"). Stefan |
From: Maciej Ł. <loz...@o2...> - 2011-12-06 21:15:50
|
W dniu 06.12.2011 10:25, "Maciej Łoziński" pisze: >> > >> > Line6linux driver does not allow to select input source, it's fixed at >> > default - "instrument input". > Well, it actually*should* allow selection. Which device are you using (see > also my previous message)? > > Kind regards, > Markus I installed latest version from ux2 branch (rev.896) and it indeed has input source select (my bad, sorry). But I didn't manage to record anything from any input (I was using Audacity 1.3.13 and Arch linux with 3.1.1 kernel). Also JACK was disconnecting about 30 seconds after connecting, and it wasn't possible to record with Ardour+JACK. I also encountered a kernel panic once when plugging in a device. Let me know if I could help you by providing some logs or running some debug. cheers Maciek |
From: Petter S. S. <pet...@gm...> - 2011-12-06 19:15:37
|
Now I have downloaded the new version(revision 898) and "make install". QJackCtl shows 2 capture and 4 playback, where capture_1 is the "main" instrument input, as before. But now I got not just PCM and Monitor in alsamixer, but also PCM Capture Source that I can switch to Intstrument, Line, Microphone and Inst & Mic. But it is listed as Playback not Capture if that means something? Will the cupture_1 change to Mic input if I change it in alsamixer(I dont have a microphone here at the moment)? If I use jack and Ardour I get the same low volume and sound. I still cant figure out how to get sound in the headset from the device either. I tested the volume in JAMin with jack and its the same result. But I can use it for recording if I can get monitor in headset and higher amplitude on the in-signal :-) 2011/12/6 "Maciej Łoziński" <loz...@o2...> > On Tue, December 6, 2011 1:07 am, Markus Grabner wrote: > > On Monday 05 December 2011 16:01:14 Maciej Łoziński wrote: > >> > >> Line6linux driver does not allow to select input source, it's fixed at > >> default - "instrument input". > > Well, it actually *should* allow selection. Which device are you using > > (see > > also my previous message)? > > > > I'm using TonePort UX2. But I see you made some updates to the code. I'll > try the latest version from UX2 branch and let you know how it works. > > Btw is POD Studio UX2 the same as TonePort UX2? They look the same, except > color - POD Studio is black, TonePort is red. > > cheers > Maciek > > > > > ------------------------------------------------------------------------------ > Cloud Services Checklist: Pricing and Packaging Optimization > This white paper is intended to serve as a reference, checklist and point > of > discussion for anyone considering optimizing the pricing and packaging > model > of a cloud services business. Read Now! > http://www.accelacomm.com/jaw/sfnl/114/51491232/ > _______________________________________________ > Line6linux-devel mailing list > Lin...@li... > https://lists.sourceforge.net/lists/listinfo/line6linux-devel > -- Petter Sakrihei Storvik Ω |
From: Maciej Ł. <loz...@o2...> - 2011-12-06 09:25:35
|
On Tue, December 6, 2011 1:07 am, Markus Grabner wrote: > On Monday 05 December 2011 16:01:14 Maciej Łoziński wrote: >> >> Line6linux driver does not allow to select input source, it's fixed at >> default - "instrument input". > Well, it actually *should* allow selection. Which device are you using > (see > also my previous message)? > I'm using TonePort UX2. But I see you made some updates to the code. I'll try the latest version from UX2 branch and let you know how it works. Btw is POD Studio UX2 the same as TonePort UX2? They look the same, except color - POD Studio is black, TonePort is red. cheers Maciek |
From: Markus G. <gr...@ic...> - 2011-12-06 00:07:46
|
On Monday 05 December 2011 16:01:14 Maciej Łoziński wrote: > On Mon, December 5, 2011 3:14 pm, Stefan Hajnoczi wrote: > > 2011/12/5 Petter S. Storvik <pet...@gm...>: > >> When I do that I see the POD Studio UX2 card playback and PCM and > >> Monitor is > >> listed. If I press F4, to get to capture devices, it says: > >> This sound device does not have any capture controls. > > > > Ah, right. I get the same with my POD HD300 because there are indeed > > no capture controls (e.g. master volume slider). But recording with > > Audacity, jack applications, or PulseAudio does work :). > > > > Stefan > > Line6linux driver does not allow to select input source, it's fixed at > default - "instrument input". Well, it actually *should* allow selection. Which device are you using (see also my previous message)? Kind regards, Markus -- Markus Grabner Institute for Computer Graphics and Vision Graz University of Technology, Inffeldgasse 16a/II, 8010 Graz, Austria WWW: http://www.icg.tugraz.at/Members/grabner |
From: Markus G. <gr...@ic...> - 2011-12-05 21:03:25
|
On Monday 05 December 2011 16:40:44 Petter S. Storvik wrote: > So there isnt any way to get sound from one of the other three inputs, just > the normal line in? Dont need all of them at the same time, but would like > to have the ability to change it. I now can record my guitar in Ardour when > i use Jack, so thank you for that :-) But the input sound is very, and i > mean very, low. If I have to turn the volume up post-recorded it gets > noisy. Is there some way to get "normal" sound in? I have some problems > with getting sound in the headset connected to the UX2 device too. > > So close now, so close! I just noticed that some features were enabled for the TonePort UX2, but not for the POD Studio UX2. I updated the source code to treat these devices identically when creating controls etc. (revision 896 in the branches/ux2), so please try again with the most recent version! Thanks & kind regards, Markus -- Markus Grabner Institute for Computer Graphics and Vision Graz University of Technology, Inffeldgasse 16a/II, 8010 Graz, Austria WWW: http://www.icg.tugraz.at/Members/grabner |
From: Stefan H. <ste...@gm...> - 2011-12-05 15:55:43
|
2011/12/5 Petter S. Storvik <pet...@gm...>: > So there isnt any way to get sound from one of the other three inputs, just > the normal line in? Dont need all of them at the same time, but would like > to have the ability to change it. I now can record my guitar in Ardour when > i use Jack, so thank you for that :-) But the input sound is very, and i > mean very, low. If I have to turn the volume up post-recorded it gets noisy. > Is there some way to get "normal" sound in? I have some problems with > getting sound in the headset connected to the UX2 device too. > > So close now, so close! At this stage you are entering UX2-specific territory. Sorry I can't help out here because I don't have that device or know much about it. Stefan |
From: Petter S. S. <pet...@gm...> - 2011-12-05 15:45:07
|
So there isnt any way to get sound from one of the other three inputs, just the normal line in? Dont need all of them at the same time, but would like to have the ability to change it. I now can record my guitar in Ardour when i use Jack, so thank you for that :-) But the input sound is very, and i mean very, low. If I have to turn the volume up post-recorded it gets noisy. Is there some way to get "normal" sound in? I have some problems with getting sound in the headset connected to the UX2 device too. So close now, so close! 2011/12/5 "Maciej Łoziński" <loz...@o2...> > On Mon, December 5, 2011 3:14 pm, Stefan Hajnoczi wrote: > > 2011/12/5 Petter S. Storvik <pet...@gm...>: > >> When I do that I see the POD Studio UX2 card playback and PCM and > >> Monitor is > >> listed. If I press F4, to get to capture devices, it says: > >> This sound device does not have any capture controls. > > > > Ah, right. I get the same with my POD HD300 because there are indeed > > no capture controls (e.g. master volume slider). But recording with > > Audacity, jack applications, or PulseAudio does work :). > > > > Stefan > > > > Line6linux driver does not allow to select input source, it's fixed at > default - "instrument input". > > cheers > Maciek > > > -- Petter Sakrihei Storvik Ω |
From: Maciej Ł. <loz...@o2...> - 2011-12-05 15:01:29
|
On Mon, December 5, 2011 3:14 pm, Stefan Hajnoczi wrote: > 2011/12/5 Petter S. Storvik <pet...@gm...>: >> When I do that I see the POD Studio UX2 card playback and PCM and >> Monitor is >> listed. If I press F4, to get to capture devices, it says: >> This sound device does not have any capture controls. > > Ah, right. I get the same with my POD HD300 because there are indeed > no capture controls (e.g. master volume slider). But recording with > Audacity, jack applications, or PulseAudio does work :). > > Stefan > Line6linux driver does not allow to select input source, it's fixed at default - "instrument input". cheers Maciek |
From: Stefan H. <ste...@gm...> - 2011-12-05 14:14:27
|
2011/12/5 Petter S. Storvik <pet...@gm...>: > When I do that I see the POD Studio UX2 card playback and PCM and Monitor is > listed. If I press F4, to get to capture devices, it says: > This sound device does not have any capture controls. Ah, right. I get the same with my POD HD300 because there are indeed no capture controls (e.g. master volume slider). But recording with Audacity, jack applications, or PulseAudio does work :). Stefan |
From: Petter S. S. <pet...@gm...> - 2011-12-05 13:29:56
|
When I do that I see the POD Studio UX2 card playback and PCM and Monitor is listed. If I press F4, to get to capture devices, it says: This sound device does not have any capture controls. 2011/12/5 Stefan Hajnoczi <ste...@gm...> > 2011/12/5 Petter S. Storvik <pet...@gm...>: > > Here are the outputs of dmesg/aplay/arecord and a new sysinfo. The thing > is, > > "alsamixer" dont find any capture devices. > > That's odd, from the output I would expect it to work. > > Did you try "alsamixer -c1" because the UX2 is "Card 1"? > > Stefan > -- Petter Sakrihei Storvik Ω |
From: Stefan H. <ste...@gm...> - 2011-12-05 13:19:38
|
2011/12/5 Petter S. Storvik <pet...@gm...>: > Here are the outputs of dmesg/aplay/arecord and a new sysinfo. The thing is, > "alsamixer" dont find any capture devices. That's odd, from the output I would expect it to work. Did you try "alsamixer -c1" because the UX2 is "Card 1"? Stefan |
From: Petter S. S. <pet...@gm...> - 2011-12-05 13:02:31
|
**** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PODStudioUX2 [POD Studio UX2], device 0: POD Studio UX2 [POD Studio UX2] Subdevices: 1/1 Subdevice #0: subdevice #0 |
From: Stefan H. <ste...@gm...> - 2011-12-05 12:21:02
|
2011/12/5 Petter S. Storvik <pet...@gm...>: > This is my system information. Trying to get the UX2 device work in linux. The lsusb output does not show a Line6 device plugged in. Was your UX2 connected via USB when you collected the system information? What you need in order to try out the line6usb driver: 1. Build and install the line6usb driver for your kernel. 2. Plug in the device. 3. There should be dmesg(1) output showing the driver loading and detecting the device. 4. aplay -l and arecord -l should show ALSA playback/capture devices. 5. You're ready to go! Markus mentioned a UX2 driver branch in a previous thread which you may need to build instead of trunk: https://line6linux.svn.sourceforge.net/svnroot/line6linux/driver/branches/ux2/ Stefan |
From: <ste...@gm...> - 2011-12-03 14:17:22
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From: Stefan Hajnoczi <ste...@gm...> The line6 driver checks struct field addresses for NULL where it does not make sense to do so. The struct has already been checked for NULL and there is no value in checking the first field's address too. Suggested-by: Dan Carpenter <dan...@or...> Signed-off-by: Stefan Hajnoczi <ste...@gm...> --- pod.c | 6 +----- podhd.c | 6 +----- toneport.c | 6 +----- variax.c | 6 +----- 4 files changed, 4 insertions(+), 20 deletions(-) diff --git a/pod.c b/pod.c index d9b3021..4dadc57 100644 --- a/pod.c +++ b/pod.c @@ -1149,14 +1149,10 @@ static struct snd_kcontrol_new pod_control_monitor = { static void pod_destruct(struct usb_interface *interface) { struct usb_line6_pod *pod = usb_get_intfdata(interface); - struct usb_line6 *line6; if (pod == NULL) return; - line6 = &pod->line6; - if (line6 == NULL) - return; - line6_cleanup_audio(line6); + line6_cleanup_audio(&pod->line6); del_timer(&pod->startup_timer); cancel_work_sync(&pod->startup_work); diff --git a/podhd.c b/podhd.c index 6c0f7f2..7ef4543 100644 --- a/podhd.c +++ b/podhd.c @@ -80,14 +80,10 @@ static struct line6_pcm_properties podhd_pcm_properties = { static void podhd_destruct(struct usb_interface *interface) { struct usb_line6_podhd *podhd = usb_get_intfdata(interface); - struct usb_line6 *line6; if (podhd == NULL) return; - line6 = &podhd->line6; - if (line6 == NULL) - return; - line6_cleanup_audio(line6); + line6_cleanup_audio(&podhd->line6); } /* diff --git a/toneport.c b/toneport.c index 879e699..f310578 100644 --- a/toneport.c +++ b/toneport.c @@ -295,14 +295,10 @@ static struct snd_kcontrol_new toneport_control_source = { static void toneport_destruct(struct usb_interface *interface) { struct usb_line6_toneport *toneport = usb_get_intfdata(interface); - struct usb_line6 *line6; if (toneport == NULL) return; - line6 = &toneport->line6; - if (line6 == NULL) - return; - line6_cleanup_audio(line6); + line6_cleanup_audio(&toneport->line6); } /* diff --git a/variax.c b/variax.c index 81241cd..d366222 100644 --- a/variax.c +++ b/variax.c @@ -572,14 +572,10 @@ static DEVICE_ATTR(raw2, S_IWUSR, line6_nop_read, variax_set_raw2); static void variax_destruct(struct usb_interface *interface) { struct usb_line6_variax *variax = usb_get_intfdata(interface); - struct usb_line6 *line6; if (variax == NULL) return; - line6 = &variax->line6; - if (line6 == NULL) - return; - line6_cleanup_audio(line6); + line6_cleanup_audio(&variax->line6); del_timer(&variax->startup_timer1); del_timer(&variax->startup_timer2); -- 1.7.7.3 |
From: <ste...@gm...> - 2011-12-03 14:17:19
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From: Stefan Hajnoczi <ste...@gm...> Playback urbs use the index_out counter to decide which part of the playback buffer to use. Since the urb already has a unique index in range [0, LINE6_ISO_BUFFERS) there is no need to keep a separate counter. Use the urb index instead. This also eliminates the possibility of two urbs using the same playback buffer space if they ever complete out-of-order for some reason. Signed-off-by: Stefan Hajnoczi <ste...@gm...> --- pcm.h | 5 ----- playback.c | 5 +---- 2 files changed, 1 insertions(+), 9 deletions(-) diff --git a/pcm.h b/pcm.h index 77055b3..e41e981 100644 --- a/pcm.h +++ b/pcm.h @@ -149,11 +149,6 @@ struct snd_line6_pcm { unsigned char *buffer_in; /** - Temporary buffer index for playback. - */ - int index_out; - - /** Previously captured frame (for software monitoring). */ unsigned char *prev_fbuf; diff --git a/playback.c b/playback.c index f43f55b..add9537 100644 --- a/playback.c +++ b/playback.c @@ -192,13 +192,10 @@ static int submit_audio_out_urb(struct snd_line6_pcm *line6pcm) urb_frames = urb_size / bytes_per_frame; urb_out->transfer_buffer = line6pcm->buffer_out + - LINE6_ISO_PACKETS * line6pcm->max_packet_size * line6pcm->index_out; + index * LINE6_ISO_PACKETS * line6pcm->max_packet_size; urb_out->transfer_buffer_length = urb_size; urb_out->context = line6pcm; - if (++line6pcm->index_out == LINE6_ISO_BUFFERS) - line6pcm->index_out = 0; - if (test_bit(BIT_PCM_ALSA_PLAYBACK, &line6pcm->flags) && !test_bit(BIT_PAUSE_PLAYBACK, &line6pcm->flags)) { struct snd_pcm_runtime *runtime = -- 1.7.7.3 |
From: <ste...@gm...> - 2011-12-03 14:17:17
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From: Stefan Hajnoczi <ste...@gm...> The .trigger() pcm callbacks are not allowed to block and cannot wait until urbs have completed. We need to ensure that stopping, preparing, and then restarting a stream always works. Currently the driver will sometimes return -EBUSY when restarting the stream because urbs have not completed yet. This can be triggered by jackd from userspace. The solution is to wait on urbs in the .prepare() pcm callback since blocking is allowed in that callback. This guarantees that all urbs are quiesced and ready to be submitted when the start trigger callback is invoked. Signed-off-by: Stefan Hajnoczi <ste...@gm...> --- pcm.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/pcm.c b/pcm.c index 7e3a5f0..7ca6451 100644 --- a/pcm.c +++ b/pcm.c @@ -491,6 +491,20 @@ int snd_line6_prepare(struct snd_pcm_substream *substream) { struct snd_line6_pcm *line6pcm = snd_pcm_substream_chip(substream); + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + line6_unlink_wait_clear_audio_out_urbs(line6pcm); + break; + + case SNDRV_PCM_STREAM_CAPTURE: + line6_unlink_wait_clear_audio_in_urbs(line6pcm); + break; + + default: + MISSING_CASE; + } + + if (!test_and_set_bit(BIT_PREPARED, &line6pcm->flags)) { line6pcm->count_out = 0; line6pcm->pos_out = 0; -- 1.7.7.3 |
From: <ste...@gm...> - 2011-12-03 14:17:17
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From: Stefan Hajnoczi <ste...@gm...> The playback urb transfer buffer calculation does not factor in LINE6_ISO_PACKETS. Buffer memory is organized like this in the driver: Buffer 0 Buffer 1 ... [Packet 0, Packet 1, ...][Packet 0, Packet 1, ...][Packet 0, ...] However, we're lucky that LINE6_ISO_PACKETS is currently defined as 1 so this patch does not change any behavior. It's still worth including this fix in case the LINE6_ISO_PACKETS value is changed in the future. Signed-off-by: Stefan Hajnoczi <ste...@gm...> --- playback.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/playback.c b/playback.c index c39d1f0..f43f55b 100644 --- a/playback.c +++ b/playback.c @@ -192,7 +192,7 @@ static int submit_audio_out_urb(struct snd_line6_pcm *line6pcm) urb_frames = urb_size / bytes_per_frame; urb_out->transfer_buffer = line6pcm->buffer_out + - line6pcm->max_packet_size * line6pcm->index_out; + LINE6_ISO_PACKETS * line6pcm->max_packet_size * line6pcm->index_out; urb_out->transfer_buffer_length = urb_size; urb_out->context = line6pcm; -- 1.7.7.3 |
From: <ste...@gm...> - 2011-12-03 14:17:15
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From: Stefan Hajnoczi <ste...@gm...> The .hw_params() pcm callback can be invoked multiple times in a row. Ensure that the URB data buffer is only allocated once. Signed-off-by: Stefan Hajnoczi <ste...@gm...> --- capture.c | 7 +++++-- playback.c | 7 +++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/capture.c b/capture.c index d9da7ed..c7e3bfa 100644 --- a/capture.c +++ b/capture.c @@ -320,8 +320,11 @@ static int snd_line6_capture_hw_params(struct snd_pcm_substream *substream, } /* -- [FD] end */ - line6pcm->buffer_in = kmalloc(LINE6_ISO_BUFFERS * LINE6_ISO_PACKETS * - line6pcm->max_packet_size, GFP_KERNEL); + /* We may be invoked multiple times in a row so allocate once only */ + if (!line6pcm->buffer_in) + line6pcm->buffer_in = + kmalloc(LINE6_ISO_BUFFERS * LINE6_ISO_PACKETS * + line6pcm->max_packet_size, GFP_KERNEL); if (!line6pcm->buffer_in) { dev_err(line6pcm->line6->ifcdev, diff --git a/playback.c b/playback.c index b344527..c39d1f0 100644 --- a/playback.c +++ b/playback.c @@ -470,8 +470,11 @@ static int snd_line6_playback_hw_params(struct snd_pcm_substream *substream, } /* -- [FD] end */ - line6pcm->buffer_out = kmalloc(LINE6_ISO_BUFFERS * LINE6_ISO_PACKETS * - line6pcm->max_packet_size, GFP_KERNEL); + /* We may be invoked multiple times in a row so allocate once only */ + if (!line6pcm->buffer_out) + line6pcm->buffer_out = + kmalloc(LINE6_ISO_BUFFERS * LINE6_ISO_PACKETS * + line6pcm->max_packet_size, GFP_KERNEL); if (!line6pcm->buffer_out) { dev_err(line6pcm->line6->ifcdev, -- 1.7.7.3 |
From: <ste...@gm...> - 2011-12-03 14:17:13
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From: Stefan Hajnoczi <ste...@gm...> I have been testing PulseAudio, ALSA, and jack applications with a POD HD300 because they tend to produce different call patterns. This is useful for exposing bugs. This series includes the latest fixes as well as some code cleanups. The fixes I'd like to highlight are: Patch 1 resolves a potential memory leak which I introduced when moving buffer allocation into the .hw_params() pcm callback. Patch 3 ensures that the playback buffer is used correctly irrespective of urb completion order. I'm not certain what guarantees we have on iso urb completion order so this may not be a real problem but it at least simplifies the code. Patch 5 makes jackd work by eliminating the -EBUSY error when it tries to start a stream. Stefan Hajnoczi (5): staging: line6: fix memory leak in .hw_params() staging: line6: fix playback urb transfer buffer calculation staging: line6: eliminate useless index_out variable staging: line6: eliminate useless NULL checks staging: line6: wait for urbs in snd_line6_prepare() capture.c | 7 +++++-- pcm.c | 14 ++++++++++++++ pcm.h | 5 ----- playback.c | 12 ++++++------ pod.c | 6 +----- podhd.c | 6 +----- toneport.c | 6 +----- variax.c | 6 +----- 8 files changed, 29 insertions(+), 33 deletions(-) -- 1.7.7.3 |
From: Maciej Ł. <loz...@o2...> - 2011-11-03 19:46:49
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W dniu 28.10.2011 09:32, Maciej Łoziński pisze: > hi! > >> I just added support for source select on the UX2 under the >> assumption that >> it's identical as on the UX1 in the following subversion branch: >> >> https://line6linux.svn.sourceforge.net/svnroot/line6linux/driver >> /branches/ux2 >> >> Please also have a look at >> >> http://sourceforge.net/projects/line6linux/forums/forum/713265/topic >> /3695415 >> >> which is the discussion with an UX1 user finally leading to some of >> UX1's >> features being supported. Can you please provide the list of source >> select >> options of the UX2 (as requested in posting #12 in the above >> mentioned forum >> thread)? >> >> Kind regards, >> Markus > > I've compiled from the UX2 branch, but I cannot see any "PCM Capture > Source" control in alsamixer nor in gnome-alsamixer, Although "PCM" and > "Monitor" sliders are there. > > Btw, device name is not present in gnome-alsamixer, just an empty tab. I > attach a screenshot. > > cheers > Maciek > Hello again! From the mentioned forum post: > *) In which order (from top to bottom) do the entries "Instrument", > "Microphone" and "Inst & Mic" appear in the "source select" field in > the Windows GearBox software? > *) Which of these three options is selected by default when GearBox > is launched? I've installed GearBox and order of input devices is following: Mic 1 Mic 2 Mic Stereo Mic 1 & 2 Line 1 Line 2 Line Stereo Line 1 & 2 Instrument Inst & Mic 1 And "Instrument" was selected when I first ran GearBox. Please let me know if I could give some more help. It would be great to have also all monitoring / VU meter features working in UX2 :-) I'm anxious to help! Cheers Maciek |