aboutsummaryrefslogtreecommitdiffstats
path: root/sound/usb/endpoint.c
Commit message (Collapse)AuthorAgeFilesLines
* 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 Mack2012-09-271-1/+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
|\
| * ALSA: usb-audio: fix detection of vendor-specific device protocol settingsClemens Ladisch2010-09-031-5/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Audio Class v2 support code in 2.6.35 added checks for the bInterfaceProtocol field. However, there are devices (usually those detected by vendor-specific quirks) that do not have one of the predefined values in this field, which made the driver reject them. To fix this regression, restore the old behaviour, i.e., assume that a device with an unknown bInterfaceProtocol field (other than UAC_VERSION_2) has more or less UAC-v1-compatible descriptors. [compile warning fixes by tiwai] Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: Daniel Mack <daniel@caiaq.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: fix Fast Track Ultra (8R) 44.1 sample ratesClemens Ladisch2010-09-021-2/+0
|/ | | | | | | | | | The M-Audio Fast Track Ultra series devices did not play sound correctly at 44.1/88.2 kHz. Changing the output endpoint attribute to adaptive fixes this. Signed-off-by: Felix Homann <fexpop@web.de> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: simplify control interface accessDaniel Mack2010-06-231-0/+1
| | | | | | | | | | | As the control interface is now carried in struct snd_usb_audio, we can simplify the API a little and also drop the private ctrlif field from struct usb_mixer_interface. Also remove a left-over function prototype in pcm.h. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: unify UAC macros and struct namesDaniel Mack2010-06-231-2/+2
| | | | | | | | | | Get rid of the last occurances of _v1 suffixes, and move the version number right after the "uac" string. Now things are consitent again. Sorry for the forth and back, but it just looks much nicer this way. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb/endpoint, fix dangling pointer useJiri Slaby2010-06-211-0/+1
| | | | | | | | | | | | | | Stanse found that in snd_usb_parse_audio_endpoints, there is a dangling pointer dereference. When snd_usb_parse_audio_format fails, fp is freed, and continue invoked. On the next loop, there is "fp && fp->altsetting == 1 && fp->channels == 1" test, but fp is set from the last iteration (but is bogus) and thus ilegally dereferenced. Set fp to NULL before "continue". Signed-off-by: Jiri Slaby <jslaby@suse.cz> Acked-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: parse clock topology of UAC2 devicesDaniel Mack2010-05-311-2/+55
| | | | | | | | | | | | | | | | | | | | | | | | | Audio devices which comply to the UAC2 standard can export complex clock topologies in its descriptors and set up links between them. The entities that are defined are - clock sources, which define the end-leafs. - clock selectors, which act as switch to select one out of many possible clocks sources. - clock multipliers, which have an input clock source, and act as clock source again. They can be used to derive one clock from another. All sample rate changes, clock validity queries and the like must go to clock source elements, while clock selectors and multipliers can be used as terminal clock source. The following patch adds a parser for these elements and functions to iterate over the tree and find the leaf nodes (clock sources). The samplerate set functions were moved to the new clock.c file. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: parse UAC2 endpoint descriptors correctlyDaniel Mack2010-05-271-13/+42
| | | | | | | | | | | UAC2 devices have their information about pitch control stored in a different field. Parse it, and emulate the bits for a v1 device. A new struct uac2_iso_endpoint_descriptor is added. Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Greg Kroah-Hartman <gregkh@suse.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: parse more format descriptors with structsDaniel Mack2010-05-271-4/+7
| | | | | Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb - use of kmalloc/kfree requires the include of slab.hStephen Rothwell2010-03-291-0/+1
| | | | | Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usbaudio: Add basic support for M-Audio Fast Track Ultra seriesFelix Homann2010-03-251-0/+2
| | | | | | | | | | | | This adds basic support for M-Audio's Fast Track Ultra series of USB audio interfaces. It is a refactored version of the patch Clemens Ladisch posted some time ago. Neither playback nor capturing work properly at 44100 Hz (don't know why). The other sampling rates work properly. There's no support for the DSP mixer, yet. Signed-off-by: Felix Homann <fexpop@web.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* linux/usb/audio.h: split headerDaniel Mack2010-03-121-0/+1
| | | | | | | | | | | | | - Split the audio.h file in two to clearly denote the differences between the standards. - Add many more defines to audio-v2.h. Most of them are not currently used. - Replaced a magic value with a proper define Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Greg Kroah-Hartman <gregkh@suse.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: add support for samplerate setting on v2 devicesDaniel Mack2010-03-051-2/+2
| | | | | | | | | Sample rate setting is done with a 4-byte long class request that addresses the interface. Signed-off-by: Daniel Mack <daniel@caiaq.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: use a format bitmask per alternate settingClemens Ladisch2010-03-051-2/+2
| | | | | | | | In preparation for USB audio 2.0 support, change the audioformat structure so that it uses a bitmask to specify possible formats. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
generated by cgit v1.1 at 2024-12-23 18:16:30 +0000