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* ALSA: usb-audio: Pass direct struct pointer instead of list_headTakashi Iwai2014-11-041-1/+1
| | | | | | | | | | | | Some functions in mixer.c and endpoint.c receive list_head instead of the object itself. This is not obvious and rather error-prone. Let's pass the proper object directly instead. The functions in midi.c still receive list_head and this can't be changed since the object definition isn't exposed to the outside of midi.c, so left as is. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix races at disconnection and PCM closingTakashi Iwai2014-06-261-0/+1
| | | | | | | | | | | | | | | | | | | | | | | When a USB-audio device is disconnected while PCM is still running, we still see some race: the disconnect callback calls snd_usb_endpoint_free() that calls release_urbs() and then kfree() while a PCM stream would be closed at the same time and calls stop_endpoints() that leads to wait_clear_urbs(). That is, the EP object might be deallocated while a PCM stream is syncing with wait_clear_urbs() with the same EP. Basically calling multiple wait_clear_urbs() would work fine, also calling wait_clear_urbs() and release_urbs() would work, too, as wait_clear_urbs() just reads some fields in ep. The problem is the succeeding kfree() in snd_pcm_endpoint_free(). This patch moves out the EP deallocation into the later point, the destructor callback. At this stage, all PCMs must have been already closed, so it's safe to free the objects. Reported-by: Alan Stern <stern@rowland.harvard.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate()Eldad Zack2013-10-071-1/+1
| | | | | | | | | | The return value of snd_usb_endpoint_deactivate() is not used, make the function have no return value. Update the documentation to reflect what the function is actually doing. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: improve buffer size computations for USB PCM audioAlan Stern2013-09-261-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: spelling correctionEldad Zack2013-04-041-1/+1
| | | | | | | | Correct spelling of snd_usb_endpoint_implict_feedback_sink in all occurances. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: stop both data and sync endpoints asynchronouslyTakashi Iwai2012-11-211-1/+1
| | | | | | | | | | | | | | As we are stopping the endpoints asynchronously now, it's better to trigger the stop of both data and sync endpoints and wait for pending stopping operations, instead of the sequential trigger-and-wait procedure. So the wait argument in snd_usb_endpoint_stop() is dropped, and it's expected that the caller synchronizes explicitly by calling snd_usb_endpoint_sync_pending_stop(). (Actually there is only one place calling this, so it was safe to change.) Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: simplify snd_usb_endpoint_start/stop argumentsTakashi Iwai2012-11-211-3/+2
| | | | | | | | | Reduce the redundant arguments for snd_usb_endpoint_start() and snd_usb_endpoint_stop(). Also replaced from int to bool. No functional changes by this commit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix crash at re-preparing the PCM streamTakashi Iwai2012-11-081-0/+1
| | | | | | | | | | | | | | | | | There are bug reports of a crash with USB-audio devices when PCM prepare is performed immediately after the stream is stopped via trigger callback. It turned out that the problem is that we don't wait until all URBs are killed. This patch adds a new function to synchronize the pending stop operation on an endpoint, and calls in the prepare callback for avoiding the crash above. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181 Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com> Cc: <stable@vger.kernel.org> [v3.6] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Don't require hw_params in endpoint.Dylan Reid2012-09-191-1/+4
| | | | | | | | | Change the interface to configure an endpoint so that it doesn't require a hw_params struct. This will allow it to be called from prepare instead of hw_params, configuring it after system resume. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: fix calls to next_packet_sizeDaniel Mack2012-08-311-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | In order to support devices with implicit feedback streaming models, packet sizes are now stored with each individual urb, and the PCM handling code which fills the buffers purely relies on the size fields now. However, calling snd_usb_audio_next_packet_size() for all possible packets in an URB at once, prior to letting the PCM code do its job does in fact not lead to the same behaviour than what the old code did: The PCM code will break its loop once a period boundary is reached, consequently using up less packets that it really could. As snd_usb_audio_next_packet_size() implements a feedback mechanism to the endpoints phase accumulator, the number of calls to that function matters, and when called too often, the data rate runs out of bounds. Fix this by making the next_packet function public, and call it from the PCM code as before if the packet data sizes are not defined. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: Fix URB cancellation at stream startDaniel Mack2012-08-301-1/+1
| | | | | | | | | | | | | | | | | Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") fixed a scheduling-while-atomic bug that happened when snd_usb_endpoint_start was called from the trigger callback, which is an atmic context. However, the patch breaks the idea of the endpoints reference counting, which is the reason why the driver has been refactored lately. Revert that commit and let snd_usb_endpoint_start() take care of the URB cancellation again. As this function is called from both atomic and non-atomic context, add a flag to denote whether the function may sleep. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: remove old streaming logicDaniel Mack2012-04-131-15/+0
| | | | | Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: switch over to new endpoint streaming logicDaniel Mack2012-04-131-3/+0
| | | | | | | | | With the previous commit that added the new streaming model, all endpoint and streaming related code is now in endpoint.c, and pcm.c only acts as a wrapper for handling the packet's payload. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: implement new endpoint streaming modelDaniel Mack2012-04-131-0/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds a new generic streaming logic for audio over USB. It defines a model (snd_usb_endpoint) that handles everything that is related to an USB endpoint and its streaming. There are functions to activate and deactivate an endpoint (which call usb_set_interface()), and to start and stop its URBs. It also has function pointers to be called when data was received or is about to be sent, and pointer to a sync slave (another snd_usb_endpoint) that is informed when data has been received. A snd_usb_endpoint knows about its state and implements a refcounting, so only the first user will actually start the URBs and only the last one to stop it will tear them down again. With this sort of abstraction, the actual streaming is decoupled from the pcm handling, which makes the "implicit feedback" mechanisms easy to implement. In order to split changes properly, this patch only adds the new implementation but leaves the old one around, so the the driver doesn't change its behaviour. The switch to actually use the new code is submitted separately. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: move code from urb.c to endpoint.cDaniel Mack2011-09-141-0/+17
| | | | | | | | | No code altered at this point, simply preparing for upcoming refactorizations. Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: re-order codeDaniel Mack2011-09-141-7/+0
| | | | | | | | | | | | Move code from endpoint.c into a new file called stream.c and rename functions so that their names actually reflect what they're doing. This way, endpoint.c will be available to functions that hold all the endpoint logic. Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: refactor codeDaniel Mack2010-03-051-0/+11
Clean up the usb audio driver by factoring out a lot of functions to separate files. Code for procfs, quirks, urbs, format parsers etc all got a new home now. Moved almost all special quirk handling to quirks.c and introduced new generic functions to handle them, so the exceptions do not pollute the whole driver. Renamed usbaudio.c to card.c because this is what it actually does now. Renamed usbmidi.c to midi.c for namespace clarity. Removed more things from usbaudio.h. The non-standard drivers were adopted accordingly. Signed-off-by: Daniel Mack <daniel@caiaq.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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