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* [media] Revert "[media] sound/usb: Use Media Controller API to share media ↵Mauro Carvalho Chehab2016-03-311-23/+5
| | | | | | | | | | | | | | | | | | | resources" Unfortunately, this patch caused several regressions at au0828 and snd-usb-audio, like this one: https://bugzilla.kernel.org/show_bug.cgi?id=115561 It also showed several troubles at the MC core that handles pretty poorly the memory protections and data lifetime management. So, better to revert it and fix the core before reapplying this change. This reverts commit aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources")' Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
* Merge tag 'sound-4.6-rc1' of ↵Linus Torvalds2016-03-181-0/+2
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "After a heavy storm by syzkaller in 4.5 cycle, we have relatively few changes in the core at this time while a lot of changes are found in the driver side, unsurprisingly. Below are some highlights: ALSA core: - A few more hardening in ALSA timer codes - An extension of sequencer API for advertising the card / pid - Small fixes in compress-offload and jack layers HD-audio: - Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming DP-MST support - Lots of code refactoring for sharing with ASoC SKL driver - Regression fixes for Intel HDMI/DP - Fixups for CX20724 codec, Lenovo AiO USB-audio: - Add quirk_alias option to make quirk debugging easier - Fixes for possible Oops by malformed firmware Firewire: - Add support for FW-1804 in tascam driver - Improvements / changes in card registration, multi stream handling, etc for DICE - Lots of code refactoring ASoC: - Enhancements of still ongoing topology API - Lots of commits for Intel Skylake support including HDMI support - A few Intel Atom driver updates for recent devices - Lots of improvements to the Renesas drivers - Capture support for Qualcomm drivers - Support for TI DaVinci DRA7xxx devices - New machine drivers for Freescale systems with Cirrus CODECs, Mediatek systems with RT5650 CODECs - New CPU drivers for Allwinner S/PDIF controllers - New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514" * tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (291 commits) ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation ALSA: mixart: silence an uninitialized variable warning ALSA: usb-audio: Add sanity checks for endpoint accesses ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk() ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk() ALSA: hda - Limit i915 HDMI binding only for HSW and later ALSA: hda - Fix unconditional GPIO toggle via automute ALSA: mixart: silence unitialized variable warnings ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41. ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda ASoC: rsnd: add simplified module explanation ASoC: hdac_hdmi: Add broxton device ID ASoC: Intel: Bxtn: Add Broxton PCI ID ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops ASoC: Intel: add dmabuffer to common sst_dsp ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core ASoC: Intel: Skylake: Fix whitepsace issues ASoC: Intel: Skylake: Move module id defines ...
| * ALSA: usb-audio: Add sanity checks for endpoint accessesTakashi Iwai2016-03-161-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | Add some sanity check codes before actually accessing the endpoint via get_endpoint() in order to avoid the invalid access through a malformed USB descriptor. Mostly just checking bNumEndpoints, but in one place (snd_microii_spdif_default_get()), the validity of iface and altsetting index is checked as well. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | [media] sound/usb: Use Media Controller API to share media resourcesShuah Khan2016-03-031-5/+23
|/ | | | | | | | | | | | | | | | | | | Change ALSA driver to use Media Controller API to share media resources with DVB and V4L2 drivers on a AU0828 media device. Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Signed-off-by: Shuah Khan <shuahkh@osg.samsung.com> Acked-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
* ALSA: USB-audio: Add quirk for Zoom R16/24 playbackRicard Wanderlof2015-10-191-1/+31
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Zoom R16/24 have a nonstandard playback format where each isochronous packet contains a length descriptor in the first four bytes. (Curiously, capture data does not contain this and requires no quirk.) The quirk involves adding the extra length descriptor whenever outgoing isochronous packets are generated, both in pcm.c (outgoing audio) and endpoint.c (silent data). In order to make the quirk as unintrusive as possible, for pcm.c:prepare_playback_urb(), the isochronous packet descriptors are initially set up in the same way no matter if the quirk is enabled or not. Once it is time to actually copy the data into the outgoing packet buffer (together with the added length descriptors) the isochronous descriptors are adjusted in order take the increased payload length into account. For endpoint.c:prepare_silent_urb() it makes more sense to modify the actual function, partly because the function is less complex to start with and partly because it is not as time-critical as prepare_playback_urb() (whose bulk is run with interrupts disabled), so the (minute) additional time spent in the non-quirk case is motivated by the simplicity of having a single function for all cases. The quirk is controlled by the new tx_length_quirk member in struct snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c and endpoint.c from quirks.c in a similar manner to the txfr_quirk member in the same structs. In contrast to txfr_quirk however, the quirk is enabled directly in quirks.c:create_standard_audio_quirk() by checking the USB ID in that function. Another option would be to introduce a new QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk very plain to see in the quirk table, but it was felt that the additional code needed to implement it this way would just make the implementation more complex with no real gain. Tested with a Zoom R16, both by doing capture and playback separately using arecord and aplay (8 channel capture and 2 channel playback, respectively), as well as capture and playback together using Ardour, as well as Audacity and Qtractor together with jackd. The R24 is reportedly compatible with the R16 when used as an audio interface. Both devices share the same USB ID and have the same number of inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the patch. Regression tested using an Edirol UA-5 in both class compliant (16-bit) and "advanced" (24 bit, forces the use of quirks) modes. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Tested-by: Panu Matilainen <pmatilai@laiskiainen.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Add offset parameter to copy_to_urb()Ricard Wanderlof2015-10-191-6/+6
| | | | | | | | Preparation for adding Zoom R16/24 quirk. No functional change. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Also move out hwptr_done wrap from prepare_playback_urb()Ricard Wanderlof2015-10-191-3/+6
| | | | | | | | Refactoring in preparation for adding Zoom R16/24 quirk. No functional change. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: USB-audio: Break out copying to urb from prepare_playback_urb()Ricard Wanderlof2015-10-191-14/+21
| | | | | | | | Refactoring in preparation for adding Zoom R16/24 quirk. No functional change. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Avoid nested autoresume callsTakashi Iwai2015-08-261-17/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After the recent fix of runtime PM for USB-audio driver, we got a lockdep warning like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc8+ #61 Not tainted --------------------------------------------- pulseaudio/980 is trying to acquire lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] but task is already holding lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] This comes from snd_usb_autoresume() invoking down_read() and it's used in a nested way. Although it's basically safe, per se (as these are read locks), it's better to reduce such spurious warnings. The read lock is needed to guarantee the execution of "shutdown" (cleanup at disconnection) task after all concurrent tasks are finished. This can be implemented in another better way. Also, the current check of chip->in_pm isn't good enough for protecting the racy execution of multiple auto-resumes. This patch rewrites the logic of snd_usb_autoresume() & co; namely, - The recursive call of autopm is avoided by the new refcount, chip->active. The chip->in_pm flag is removed accordingly. - Instead of rwsem, another refcount, chip->usage_count, is introduced for tracking the period to delay the shutdown procedure. At the last clear of this refcount, wake_up() to the shutdown waiter is called. - The shutdown flag is replaced with shutdown atomic count; this is for reducing the lock. - Two new helpers are introduced to simplify the management of these refcounts; snd_usb_lock_shutdown() increases the usage_count, checks the shutdown state, and does autoresume. snd_usb_unlock_shutdown() does the opposite. Most of mixer and other codes just need this, and simply returns an error if it receives an error from lock. Fixes: 9003ebb13f61 ('ALSA: usb-audio: Fix runtime PM unbalance') Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: handle descriptor with SYNC_NONE illegal valuePierre-Louis Bossart2015-08-161-2/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The M-Audio Transit exposes an interface with a SYNC_NONE attribute. This is not a valid value according to the USB audio classspec. However there is a sync endpoint associated to this record. Changing the logic to try to use this sync endpoint allows for seamless transitions between altset 2 and altset 3. If any errors happen, the behavior remains the same. $ more /proc/asound/card1/stream0 M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (NONE) Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: fix corrupted pointers due to interface setting changePierre-Louis Bossart2015-08-161-0/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When a transition occurs between alternate settings that do not use the same synchronization method, the substream pointers were not reset. This prevents audio from being played during the second transition. Identified and tested with M-Audio Transit device (0763:2006 Midiman M-Audio Transit) Details of the issue: First playback to adaptive endpoint: $ aplay -Dhw:1,0 ~/24_96.wav Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes, Rate 96000 Hz, Stereo [ 3169.297556] usb 1-2: setting usb interface 1:1 [ 3169.297568] usb 1-2: Creating new playback data endpoint #3 [ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0 [ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000 first playback to asynchronous endpoint: $ aplay -Dhw:1,0 ~/16_48.wav Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo [ 3204.520251] usb 1-2: setting usb interface 1:3 [ 3204.520264] usb 1-2: Creating new playback data endpoint #3 [ 3204.520272] usb 1-2: Creating new capture sync endpoint #83 [ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0 [ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0 [ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000 [ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000 second playback to adaptive endpoint: no audio and error on terminal: $ aplay -Dhw:1,0 ~/24_96.wav Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes, Rate 96000 Hz, Stereo aplay: pcm_write:1939: write error: Input/output error [ 3239.483589] usb 1-2: setting usb interface 1:1 [ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000 [ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0 [ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0 This last line shows that a sync endpoint is used when it shouldn't. The sync endpoint is no longer valid and the pointers are corrupted Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: update trigger timestamp on first non-zero URB submittedPierre-Louis Bossart2015-02-091-0/+9
| | | | | | | | | | | | | | | | The first URBs are submitted during the prepare stage. When .trigger is called, the ALSA core saves a trigger tstamp that doesn't correspond to the actual time when the samples are submitted. The trigger_tstamp is now updated when the first data are submitted to avoid any time offsets. A usb-specific trigger_tstamp_pending_update flag is used for now, at some point the flag would need to move to the ALSA core, USB is not the only interface where silent block transfers are programmed as part of the prepare stage, with actual data enabled when .trigger is called. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Add mode select quirk for Denon/Marantz DACsJurgen Kramer2014-11-281-0/+5
| | | | | | | | | | | | | Denon/Marantz USB DACs need a specific vendor command to switch between PCM and DSD mode. This patch adds a new quirk function to switch between the two modes using the specific USB vendor command. This patch applies to the following devices: - Marantz SA-14S1 - Marantz HD-DAC1 Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while ↵Sander Eikelenboom2014-05-021-3/+2
| | | | | | | | | | | | | | | | | | | | | | | | DEBUG not defined This (widely used) construction: if(printk_ratelimit()) dev_dbg() Causes the ratelimiting to spam the kernel log with the "callbacks suppressed" message below, even while the dev_dbg it is supposed to rate limit wouldn't print anything because DEBUG is not defined for this device. [ 533.803964] retire_playback_urb: 852 callbacks suppressed [ 538.807930] retire_playback_urb: 852 callbacks suppressed [ 543.811897] retire_playback_urb: 852 callbacks suppressed [ 548.815745] retire_playback_urb: 852 callbacks suppressed [ 553.819826] retire_playback_urb: 852 callbacks suppressed So use dev_dbg_ratelimited() instead of this construction. Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()Tim Gardner2014-04-091-1/+2
| | | | | | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1305133 Malfunctioning or slow devices can cause a flood of dmesg SPAM. I've ignored checkpatch.pl complaints about the use of printk_ratelimit() in favour of prior art in sound/usb/pcm.c. WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit + if (printk_ratelimit() && Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.de> Cc: Eldad Zack <eldad@fogrefinery.com> Cc: Daniel Mack <zonque@gmail.com> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Tim Gardner <tim.gardner@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Use standard printk helpersTakashi Iwai2014-02-261-34/+49
| | | | | | | | | | | 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-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: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on errorEldad Zack2013-10-071-0/+1
| | | | | | | | If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED should be cleared. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: remove deactivate_endpoints()Eldad Zack2013-10-071-17/+2
| | | | | | | | | | | The only call site for deactivate_endpoints() at snd_usb_hw_free(). The return value is not checked there, as it is irrelevant if it fails on hw_free. This patch moves the deactivation of the endpoints directly into snd_usb_hw_free(). 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 Stern2013-09-261-2/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: remove implicit_fb from quirkEldad Zack2013-08-061-7/+1
| | | | | | | | | Since the quirks all apply to implicit feedback (the source endpoint is always a data endpoint), there's no need to set and check a flag for it. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: remove is_playback from implicit feedback quirksEldad Zack2013-08-061-21/+19
| | | | | | | | | | | An implicit feedback endpoint can only be a capture source. The consumer (sink) of the implicit feedback endpoint is therefore limited to playback EPs. Check if the target endpoint is a playback first and remove redundant checks. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: do not initialize and check implicit_fbEldad Zack2013-08-061-5/+3
| | | | | | | | | | | Since implicit_fb is not changed, !implicit_fb will always be true - it is set only after these checks. Similarly, there's also no need to set it at the top of the function. Change the type of implicit_fb to bool (more appropriate). Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: reverse condition logic in set_sync_endpointEldad Zack2013-08-061-39/+42
| | | | | | | | Reverse logic on the conditions required to qualify for a sync endpoint and remove one level of indendation. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: move implicit fb quirks to separate functionEldad Zack2013-08-061-15/+45
| | | | | | | | Separate setting implicit feedback quirks from setting a sync endpoint (which may also be explicit feedback or async). Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: separate sync endpoint setting from set_formatEldad Zack2013-08-061-55/+73
| | | | | | | | | Setting the sync endpoint currently takes up about half of set_format(). Move it to a dedicated function. No functional change. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: remove assignment from if conditionEldad Zack2013-08-061-1/+2
| | | | | | | Following general kernel style. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: remove disabled debug code in set_formatEldad Zack2013-08-061-9/+0
| | | | | | | Code block does not compile when enabled. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: detect implicit feedback on Roland devicesClemens Ladisch2013-06-271-0/+41
| | | | | | | | | | | | | | | All the Roland/Edirol/BOSS USB audio devices that need implicit feedback show this unambiguously in their descriptors, so it might be a good idea to let the driver detect this. This should make playback work correctly (at least with Jack) with the following devices: - BOSS GT-100 - BOSS JS-8 Jam Station - Edirol M-16DX - Roland GAIA SH-01 Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
* ALSA: usb-audio: store protocol version in struct audioformatClemens Ladisch2013-06-271-3/+1
| | | | | | | | | Instead of reading bInterfaceProtocol from the descriptor whenever it's needed, store this value in the audioformat structure. Besides simplifying some code, this will allow us to correctly handle vendor- specific devices where the descriptors are marked with other values. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
* ALSA: pcm_format_to_bits strong-typed conversionEldad Zack2013-04-291-2/+2
| | | | | | | | | | | Add a function to handle conversion from snd_pcm_format_t to bitwise with proper typing. Change such conversions to use this function and silence sparse warnings. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: add support for bit-reversed byte formatsDaniel Mack2013-04-181-1/+18
| | | | | | | | | | | | | | | | | There is quite some confusion around the bit-ordering in DSD samples, and no general agreement that defines whether hardware is supposed to expect the oldest sample in the MSB or the LSB of a byte. ALSA will hence set the rule that on the software API layer, bytes always carry the oldest bit in the most significant bit of a byte, and the driver has to translate that at runtime in order to match the hardware layout. This patch adds support for this by adding a boolean flag to the audio format struct. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: add support for DSD DOP stream transportDaniel Mack2013-04-181-11/+76
| | | | | | | | | | | | | | | | | | | | | | 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: snd-usb: use ep->stride from urb callbacksDaniel Mack2013-04-181-7/+7
| | | | | | | | | | For normal PCM transfer, this change has no effect, as the endpoint's stride is always frame_bits/8. For DSD DOP streams, however, which is added later, the hardware stride differs from the software stride, and the endpoint has the correct information in these cases. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: Add quirk for 192KHz recording on E-Mu devicesCalvin Owens2013-04-131-1/+1
| | | | | | | | | | | | | | | | | | | | | When recording at 176.2KHz or 192Khz, the device adds a 32-bit length header to the capture packets, which obviously needs to be ignored for recording to work properly. Userspace expected: L0 L1 L2 R0 R1 R2 ...but actually got: R2 L0 L1 L2 R0 R1 Also, the last byte of the length header being interpreted as L0 of the first sample caused spikes every 0.5ms, resulting in a loud 16KHz tone (about the highest 'B' on a piano) being present throughout captures. Tested at all sample rates on an E-Mu 0404USB, and tested for regressions on a generic USB headset. Signed-off-by: Calvin Owens <jcalvinowens@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locationsDaniel Mack2013-04-101-6/+1
| | | | | | | | | It turns out the devices from Playback Design need the delay quirk after usb_set_interface from clocks.c as well. Make it a proper quirks function and factor out the code to quirks.c. Signed-off-by: Daniel Mack <zonque@gmail.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: convert list_for_each to entry variantEldad Zack2013-04-041-20/+10
| | | | | | | | 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: snd-usb: add delay quirk for "Playback Design" productsDaniel Mack2013-03-181-0/+7
| | | | | | | | | "Playback Design" products need a 50ms delay after setting the USB interface. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: add support for M-Audio FT C600Matt Gruskin2013-02-111-0/+1
| | | | | | | | | | Adds quirks and mixer support for the M-Audio Fast Track C600 USB audio interface. This device is very similar to the C400 - the C600 simply has some more inputs and outputs, so the existing C400 support is extended to support this device as well. Signed-off-by: Matt Gruskin <matthew.gruskin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Make snd_printd() and snd_printdd() inlineTakashi Iwai2013-01-251-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | Because currently snd_printd() and snd_printdd() macros are expanded to empty when CONFIG_SND_DEBUG=n, a compile warning like below appears sometimes, and we had to covert it by ugly ifdefs: sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’: sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable] For "fixing" these issues better, this patch replaces snd_printd() and snd_printdd() definitions with empty inline functions instead of macros. This should have the same effect but shut up warnings like above. But since we had already put ifdefs, changing to inline functions would trigger compile errors. So, such ifdefs is removed in this patch. In addition, snd_pci_quirk name field is defined only when CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in snd_printdd() argument triggers the build errors, too. For avoiding these errors, introduce a new macro snd_pci_quirk_name() that is defined no matter how the debug option is set. Reported-by: Stratos Karafotis <stratosk@semaphore.gr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2013-01-231-0/+10
|\ | | | | | | | | This is a preliminary merge before the upcoming merge of generic parser branch.
| * ALSA: usb-audio: Fix NULL dereference by access to non-existing substreamTakashi Iwai2013-01-111-0/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for audioformat mismatch] introduced the correction of parameters to be set for sync EP. But since the new code assumes that the sync EP is always paired with the data EP of another direction, it triggers Oops when a device only with a single direction is used. This patch adds a proper check of sync EP type and the presence of the paired substream for avoiding the crash. Reported-and-tested-by: Jens Axboe <axboe@kernel.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: support delay calculation on capture streamsPierre-Louis Bossart2012-12-241-3/+20
|/ | | | | | | | | | | | Enable delay report on capture path. The delay is reset when an URB is retired and increment at each call to .pointer based on frame counter changes. The precision of the delay information is limited to 1ms as in the playback case. This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: sync ep init fix for audioformat mismatchEldad Zack2012-12-041-7/+99
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 947d299686aa9cc8aecf749d54e8475c6e498956 , "ALSA: snd-usb: properly initialize the sync endpoint", while correcting the initialization of the sync endpoint when opening just the data endpoint, prevents devices that has a sync endpoint, with a channel number different than that of the data endpoint, from functioning. Due to a different channel and period bytes count, attempting to initialize the sync endpoint will fail at the usb host driver. For example, when using xhci: cannot submit urb 0, error -90: internal error With this patch, if a sync endpoint has multiple audioformats, a matching audioformat is preferred. An audioformat must be found with at least one channel and support the requested sample rate and PCM format, otherwise the stream will not be opened. If the number of channels differ between the selected audioformat and the requested format, adjust the period bytes count accordingly. It is safe to perform the calculation on the basis of the channel count, since the requested PCM audio format and the rate must be supported by the selected audioformat. Cc: Jeffrey Barish <jeff_barish@earthlink.net> Cc: Daniel Mack <zonque@gmail.com> Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: FT C400 sync playback EP to capture EPEldad Zack2012-11-291-0/+13
| | | | | | | | | The playback endpoint uses implicit feedback mode, similar to the M-Audio FTU. Like with the FTU, we need to associate the sync pipe ourselves. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: replace hardcoded value with constEldad Zack2012-11-291-1/+1
| | | | | | | In this context, 0x01 is USB_ENDPOINT_XFER_ISOC. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: fix delay account during pauseTakashi Iwai2012-11-231-1/+17
| | | | | | | | | | | | | | When a playback stream is paused, the stream isn't actually stopped, thus we still need to take care of the in-flight data amount for the delay calculation. Otherwise the value of subs->last_delay is no longer reliable and can give a bogus value after resuming from pause. This will result in "delay: estimated XX, actual YY" error messages. Also, during pause after all in flight data are processed (i.e. last_delay = 0), we don't have to calculate the actual delay from the current frame. Give a short path in such a case. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: ignore delay calculation for capture streamTakashi Iwai2012-11-231-1/+2
| | | | | | | | It doesn't make sense to calculate the delay for capture streams in the current implementation. It's always zero, so we should skip the computation in snd_usb_pcm_pointer() in the case of capture. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2012-11-221-1/+1
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