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* ALSA: pcm: Add snd_pcm_stop_xrun() helperTakashi Iwai2014-11-091-3/+1
| | | | | | | | | | | Add a new helper function snd_pcm_stop_xrun() to the standard sequnce lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the existing open codes with this helper. The function checks the PCM running state to prevent setting the wrong state, too, for more safety. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Trigger PCM XRUN at XRUNTakashi Iwai2014-11-061-3/+8
| | | | | | | | | | | The usb-audio driver detects XRUN at its complete callback, but the actual code to trigger PCM XRUN is commented out because it caused deadlock in the past. This patch revives the PCM trigger properly. It resulted in more than just enabling snd_pcm_stop(), but it had to deduce the PCM substream with proper NULL checks and holds the stream lock around the call. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Pass direct struct pointer instead of list_headTakashi Iwai2014-11-041-5/+2
| | | | | | | | | | | | 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 Iwai2014-06-261-3/+14
| | | | | | | | | | | | | | | | | | | | | | | 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: work around corrupted TEAC UD-H01 feedback dataClemens Ladisch2014-05-021-1/+14
| | | | | | | | | | | | | | | | The TEAC UD-H01 firmware sends wrong feedback frequency values, thus causing the PC to send the samples at a wrong rate, which results in clicks and crackles in the output. Add a workaround to detect and fix the corruption. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> [mick37@gmx.de: use sender->udh01_fb_quirk rather than ep->udh01_fb_quirk in snd_usb_handle_sync_urb()] Reported-and-tested-by: Mick <mick37@gmx.de> Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Use standard printk helpersTakashi Iwai2014-02-261-13/+19
| | | | | | | | | | | Convert with dev_err() and co from snd_printk(), etc. As there are too deep indirections (e.g. ep->chip->dev->dev), a few new local macros, usb_audio_err() & co, are introduced. Also, the device numbers in some messages are dropped, as they are shown in the prefix automatically. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: use multiple packets per urb for Wireless USB inbound audioThomas Pugliese2013-11-271-1/+15
| | | | | | | | | | | For Wireless USB audio devices, use multiple isoc packets per URB for inbound endpoints with a datainterval < 5. This allows the WUSB host controller to take advantage of bursting to service endpoints whose logical polling interval is less than the 4ms minimum polling interval limit in WUSB. Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: remove unused endpoint flag EP_FLAG_ACTIVATEDEldad Zack2013-10-071-3/+0
| | | | | | | EP_FLAG_ACTIVATED is never tested for, remove it. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: rename alt_idx to altsettingEldad Zack2013-10-071-3/+3
| | | | | | | | | | | | | | | | As Clemens Ladisch kindly explained: "Please note that there are two methods to identify alternate settings: the number, which is the value in bAlternateSetting, and the index, which is the index in the descriptor array. There might be some wording in the USB spec that these two values must be the same, but in reality, [insert standard rant about firmware writers], bAlternateSetting must be treated as a random ID value." This patch changes the name to express the correct usage semantics. No functional change. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate()Eldad Zack2013-10-071-10/+5
| | | | | | | | | | 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: usb-audio: don't deactivate URBs on in-use EPEldad Zack2013-10-071-3/+3
| | | | | | | | If an endpoint in use, its associated URBs should not be deactivated. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: remove unused parameter from sync_ep_set_paramsEldad Zack2013-10-071-3/+2
| | | | | | | | Since the format is not actually used in sync_ep_set_params(), there is no need to pass it down. 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 Stern2013-09-261-45/+61
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* Merge tag 'asoc-v3.12' of ↵Takashi Iwai2013-08-231-7/+6
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for v3.12 - DAPM is now mandatory for CODEC drivers in order to avoid the repeated regressions in the special cases for non-DAPM CODECs and make it easier to integrate with other components on boards. All existing drivers have had some level of DAPM support added. - A lot of cleanups in DAPM plus support for maintaining controls in a specific state while a DAPM widget all contributed by Lars-Peter Clausen. - Core helpers for bitbanged AC'97 reset from Markus Pargmann. - New drivers and support for Analog Devices ADAU1702 and ADAU1401(a), Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson Microelectronics WM8997. - Support for building drivers that can support it cross-platform for compile test.
| * ALSA: usb-audio: do not trust too-big wMaxPacketSize valuesClemens Ladisch2013-08-081-7/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The driver used to assume that the streaming endpoint's wMaxPacketSize value would be an indication of how much data the endpoint expects or sends, and compute the number of packets per URB using this value. However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes, while only about 88 or 44 bytes are be actually used. This discrepancy would result in URBs with far too few packets, which would not work correctly on the EHCI driver. To get correct URBs, use wMaxPacketSize only as an upper limit on the packet size. Reported-by: James Stone <jamesmstone@gmail.com> Tested-by: James Stone <jamesmstone@gmail.com> Cc: <stable@vger.kernel.org> # 2.6.35+ Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: WARN_ON when alts is passed as NULLEldad Zack2013-08-061-0/+3
|/ | | | | | | | | | Prevent NULL dereference in snd_usb_add_endpoints(), when alts is passed as NULL. In this case, WARN (since this is a non-fatal bug) and return NULL ep. Call sites treat a NULL return value as an error. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB: adjust for changed 3.8 USB APIClemens Ladisch2013-04-291-3/+2
| | | | | | | | | | | | | | The recent changes in the USB API ("implement new semantics for URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the default, and changed this flag to mean that URBs can be delayed. This is not the behaviour wanted by any of the audio drivers because it leads to discontinuous playback with very small period sizes. Therefore, our URBs need to be submitted without this flag. Reported-by: Joe Rayhawk <jrayhawk@fairlystable.org> Cc: <stable@vger.kernel.org> # 3.8 only Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: add support for DSD DOP stream transportDaniel Mack2013-04-181-0/+9
| | | | | | | | | | | | | | | | | | | | | | In order to provide a compatibility way for pushing DSD samples through ordinary PCM channels, the "DoP open Standard" was invented. See http://www.dsd-guide.com for the official document. The host is required to stuff DSD marker bytes (0x05, 0xfa, alternating) in the MSB of 24 bit wide samples on the bus, in addition to the 16 bits of actual DSD sample payload. To support this, the hardware and software stride logic in the driver has to be tweaked a bit, as we make the userspace believe we're operating on 16 bit samples, while we in fact push one more byte per channel down to the hardware. The DOP runtime information is stored in struct snd_usb_substream, so we can keep track of our state across multiple calls to prepare_playback_urb_dsd_dop(). Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: spelling correctionEldad Zack2013-04-041-7/+7
| | | | | | | | 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: convert list_for_each to entry variantEldad Zack2013-04-041-3/+1
| | | | | | | | Change occurances of list_for_each into list_for_each_entry where applicable. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: use sender stride for implicit feedbackEldad Zack2012-11-291-3/+6
| | | | | | | | | | | | | | | For implicit feedback endpoints, the number of bytes for each packet is matched by the corresponding synchronizing endpoint. The size is calculated by taking the actual size and dividing it by the stride - currently by the endpoint's stride, but we should use the synchronization source's stride. This is evident when the number of channels differ between the synchronization source and the implicitly fed-back endpoint, as with M-Audio Fast Track C400 - the synchronization source (capture) has 4 channels, while the implicit feedback mode endpoint has 6. 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 Iwai2012-11-211-6/+5
| | | | | | | | | | | | | | 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 endpoint deactivation codeTakashi Iwai2012-11-211-16/+7
| | | | | | | | For further code simplification, drop the conditional call for usb_kill_urb() with can_wait argument in deactivate_urbs(), and use only usb_unlink_urb() and wait_clear_urbs() pairs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: simplify snd_usb_endpoint_start/stop argumentsTakashi Iwai2012-11-211-9/+8
| | | | | | | | | 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: Deprecate async_unlink optionTakashi Iwai2012-11-211-1/+1
| | | | | | | The async unlink behavior has been working over years. The option was provided only as a workaround for 2.4.x kernel. Let's get rid of it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: use bitmap_weightJoe Perches2012-11-171-6/+1
| | | | | | | Use bitmap_weight to count the total number of bits set in bitmap. Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix crash at re-preparing the PCM streamTakashi Iwai2012-11-081-0/+13
| | | | | | | | | | | | | | | | | 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>
* Merge tag 'sound-3.7' of ↵Linus Torvalds2012-10-091-13/+26
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This contains pretty many small commits covering fairly large range of files in sound/ directory. Partly because of additional API support and partly because of constantly developed ASoC and ARM stuff. Some highlights: - Introduced the helper function and documentation for exposing the channel map via control API, as discussed in Plumbers; most of PCI drivers are covered, will follow more drivers later - Most of drivers have been replaced with the new PM callbacks (if the bus is supported) - HD-audio controller got the support of runtime PM and the support of D3 clock-stop. Also changing the power_save option in sysfs kicks off immediately to enable / disable the power-save mode. - Another significant code change in HD-audio is the rewrite of firmware loading code. Other than that, most of changes in HD-audio are continued cleanups and standardization for the generic auto parser and bug fixes (HBR, device-specific fixups), in addition to the support of channel-map API. - Addition of ASoC bindings for the compressed API, used by the mid-x86 drivers. - Lots of cleanups and API refreshes for ASoC codec drivers and DaVinci. - Conversion of OMAP to dmaengine. - New machine driver for Wolfson Microelectronics Bells. - New CODEC driver for Wolfson Microelectronics WM0010. - Enhancements to the ux500 and wm2000 drivers - A new driver for DA9055 and the support for regulator bypass mode." Fix up various arm soc header file reorg conflicts. * tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits) ALSA: hda - Add new codec ALC283 ALC290 support ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls ALSA: hda - fix indices on boost volume on Conexant ALSA: aloop - add locking to timer access ALSA: hda - Fix hang caused by race during suspend. sound: Remove unnecessary semicolon ALSA: hda/realtek - Fix detection of ALC271X codec ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310 ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event ALSA: hda - make a generic unsol event handler ASoC: codecs: Add DA9055 codec driver ASoC: eukrea-tlv320: Convert it to platform driver ALSA: ASoC: add DT bindings for CS4271 ASoC: wm_hubs: Ensure volume updates are handled during class W startup ASoC: wm5110: Adding missing volume update bits ASoC: wm5110: Add OUT3R support ASoC: wm5110: Add AEC loopback support ASoC: wm5110: Rename EPOUT to HPOUT3 ASoC: arizona: Add more clock rates ASoC: arizona: Add more DSP options for mixer input muxes ...
| * ALSA: usb-audio: Don't require hw_params in endpoint.Dylan Reid2012-09-191-13/+18
| | | | | | | | | | | | | | | | | | 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>
| * Merge branch 'for-linus' into for-nextTakashi Iwai2012-09-111-14/+10
| |\ | | | | | | | | | To merge HD-audio fixes back to 3.7 development line
| * | ALSA: snd-usb: Add quirks for Playback Designs devicesDaniel Mack2012-09-041-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Playback Designs' USB devices have some hardware limitations on their USB interface. In particular: - They need a 20ms delay after each class compliant request as the hardware ACKs the USB packets before the device is actually ready for the next command. Sending data immediately will result in buffer overflows in the hardware. - The devices send bogus feedback data at the start of each stream which confuse the feedback format auto-detection. This patch introduces a new quirks hook that is called after each control packet and which adds a delay for all devices that match Playback Designs' USB VID for now. In addition, it adds a counter to snd_usb_endpoint to drop received packets on the floor. Another new quirks function that is called once an endpoint is started initializes that counter for these devices on their sync endpoint. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com> Supported-by: Demian Martin <demianm_1@yahoo.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: snd-usb: fix next_packet_size calls for pause caseDaniel Mack2012-09-281-1/+7
| |/ |/| | | | | | | | | | | | | | | | | | | | | Also fix the calls to next_packet_size() for the pause case. This was missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size"). Signed-off-by: Daniel Mack <zonque@gmail.com> Reviewed-by: Takashi Iwai <tiwai@suse.de> Reported-and-tested-by: Christian Tefzer <ctrefzer@gmx.de> Cc: stable@kernel.org [ Taking directly because Takashi is on vacation - Linus ] Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* | ALSA: snd-usb: fix calls to next_packet_sizeDaniel Mack2012-08-311-12/+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 Mack2012-08-301-2/+9
|/ | | | | | | | | | | | | | | | | 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: usb-audio: Fix scheduling-while-atomic bug in PCM capture streamTakashi Iwai2012-08-161-4/+0
| | | | | | | | | | | | | | | | A PCM capture stream on usb-audio causes a scheduling-while-atomic BUG, as reported in the bugzilla entry below. It's because snd_usb_endpoint_start() is called at first at trigger START for a capture stream, and this function contains the left-over EP deactivation codes. The problem doesn't happen for a playback stream because the function is called at PCM prepare time, which can sleep. This patch fixes the BUG by moving the EP deactivation code into the PCM prepare callback. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011 Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: move calls to usb_set_interfaceDaniel Mack2012-07-131-67/+6
| | | | | | | | | | | | | | | | | The rework of the snd-usb endpoint logic moved the calls to snd_usb_set_interface() into the snd_usb_endpoint implemenation. This changed the order in which these calls are issued to the device, and thereby caused regressions for some webcams. Fix this by moving the calls back to pcm.c for now to make it work again and use snd_usb_endpoint_activate() to really tear down all remaining URBs in the flight, consequently fixing another regression caused by USB packets on the wire after altsetting 0 has been selected. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Philipp Dreimann <philipp@dreimann.net> Reported-by: Joseph Salisbury <joseph.salisbury@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: fix some typos in endpoint.c documentationDaniel Mack2012-04-241-25/+33
| | | | | | | Also be more specific about some details while at it. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: sound/usb/endpoint.c: suppress warningAndrew Morton2012-04-241-1/+1
| | | | | | | | | sound/usb/endpoint.c: In function 'queue_pending_output_urbs': sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function Cc: Daniel Mack <zonque@gmail.com> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: Fix fill_max flag setTakashi Iwai2012-04-131-1/+1
| | | | | | | | ep->fill_max is a 1 bit flag, thus it has to be boolean. sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params': sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: Remove unused variableTakashi Iwai2012-04-131-1/+0
| | | | | | | sound/usb/endpoint.c: In function ‘deactivate_urbs’: sound/usb/endpoint.c:520:16: warning: unused variable ‘flags’ [-Wunused-variable] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: add some documentationDaniel Mack2012-04-131-11/+171
| | | | | | | | Document the new streaming code and some of the functions so that contributers can catch up easier. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: remove old streaming logicDaniel Mack2012-04-131-844/+6
| | | | | 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 Mack2012-04-131-40/+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 Mack2012-04-131-11/+917
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: usb-audio - Avoid flood of frame-active debug messagesTakashi Iwai2012-01-091-2/+3
| | | | | | | | | | | With some buggy devices, the usb-audio driver may give "frame xxx active" kernel messages too often. Better to keep it as debug-only using snd_printdd(), and also add the rate-limit for avoiding floods. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738681 Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: move code from urb.c to endpoint.cDaniel Mack2011-09-141-0/+948
| | | | | | | | | 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 Mack2011-09-141-433/+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: snd-usb: Accept UAC2 FORMAT_TYPE descriptors with bLength > 6Clemens Ladisch2011-08-041-1/+1
| | | | | | | | | | | The Focusrite Scarlett 18i6 USB has them that way, which is probably a bug. Anyway, the driver should simply ignore this fact. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com> Cc: stable@kernel.org Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio - Add quirks for M-Audio Fast Track Pro and QuattroGuillaume Pellerin2011-07-121-0/+2
| | | | | | | | | | | | | | | | | | | This patch gives M-Audio Fast Track Pro and M-Audio Quattro quirks and endpoints to boot and setup those devices with special options (digital inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are just adapted to match the new global M-Audio parameters. Special configurations can be then loaded through a modprobe conf file. For example, to set the 24 bits mode on the Fast Track Pro add /etc/modprobe.d/fast_track_pro.conf : options snd_usb_audio vid=0x763 pid=0x2012 device_setup=0x08 Here is a list of the possibilities in this example : http://files.parisson.com/debian/fast-track-pro.conf Signed-off-by: Guillaume Pellerin <yomguy@parisson.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'fix/misc' into topic/miscTakashi Iwai2010-09-031-5/+6
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