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2018-06-12treewide: kmalloc() -> kmalloc_array()Kees Cook1-1/+1
The kmalloc() function has a 2-factor argument form, kmalloc_array(). This patch replaces cases of: kmalloc(a * b, gfp) with: kmalloc_array(a * b, gfp) as well as handling cases of: kmalloc(a * b * c, gfp) with: kmalloc(array3_size(a, b, c), gfp) as it's slightly less ugly than: kmalloc_array(array_size(a, b), c, gfp) This does, however, attempt to ignore constant size factors like: kmalloc(4 * 1024, gfp) though any constants defined via macros get caught up in the conversion. Any factors with a sizeof() of "unsigned char", "char", and "u8" were dropped, since they're redundant. The tools/ directory was manually excluded, since it has its own implementation of kmalloc(). The Coccinelle script used for this was: // Fix redundant parens around sizeof(). @@ type TYPE; expression THING, E; @@ ( kmalloc( - (sizeof(TYPE)) * E + sizeof(TYPE) * E , ...) | kmalloc( - (sizeof(THING)) * E + sizeof(THING) * E , ...) ) // Drop single-byte sizes and redundant parens. @@ expression COUNT; typedef u8; typedef __u8; @@ ( kmalloc( - sizeof(u8) * (COUNT) + COUNT , ...) | kmalloc( - sizeof(__u8) * (COUNT) + COUNT , ...) | kmalloc( - sizeof(char) * (COUNT) + COUNT , ...) | kmalloc( - sizeof(unsigned char) * (COUNT) + COUNT , ...) | kmalloc( - sizeof(u8) * COUNT + COUNT , ...) | kmalloc( - sizeof(__u8) * COUNT + COUNT , ...) | kmalloc( - sizeof(char) * COUNT + COUNT , ...) | kmalloc( - sizeof(unsigned char) * COUNT + COUNT , ...) ) // 2-factor product with sizeof(type/expression) and identifier or constant. @@ type TYPE; expression THING; identifier COUNT_ID; constant COUNT_CONST; @@ ( - kmalloc + kmalloc_array ( - sizeof(TYPE) * (COUNT_ID) + COUNT_ID, sizeof(TYPE) , ...) | - kmalloc + kmalloc_array ( - sizeof(TYPE) * COUNT_ID + COUNT_ID, sizeof(TYPE) , ...) | - kmalloc + kmalloc_array ( - sizeof(TYPE) * (COUNT_CONST) + COUNT_CONST, sizeof(TYPE) , ...) | - kmalloc + kmalloc_array ( - sizeof(TYPE) * COUNT_CONST + COUNT_CONST, sizeof(TYPE) , ...) | - kmalloc + kmalloc_array ( - sizeof(THING) * (COUNT_ID) + COUNT_ID, sizeof(THING) , ...) | - kmalloc + kmalloc_array ( - sizeof(THING) * COUNT_ID + COUNT_ID, sizeof(THING) , ...) | - kmalloc + kmalloc_array ( - sizeof(THING) * (COUNT_CONST) + COUNT_CONST, sizeof(THING) , ...) | - kmalloc + kmalloc_array ( - sizeof(THING) * COUNT_CONST + COUNT_CONST, sizeof(THING) , ...) ) // 2-factor product, only identifiers. @@ identifier SIZE, COUNT; @@ - kmalloc + kmalloc_array ( - SIZE * COUNT + COUNT, SIZE , ...) // 3-factor product with 1 sizeof(type) or sizeof(expression), with // redundant parens removed. @@ expression THING; identifier STRIDE, COUNT; type TYPE; @@ ( kmalloc( - sizeof(TYPE) * (COUNT) * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | kmalloc( - sizeof(TYPE) * (COUNT) * STRIDE + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | kmalloc( - sizeof(TYPE) * COUNT * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | kmalloc( - sizeof(TYPE) * COUNT * STRIDE + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | kmalloc( - sizeof(THING) * (COUNT) * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | kmalloc( - sizeof(THING) * (COUNT) * STRIDE + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | kmalloc( - sizeof(THING) * COUNT * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | kmalloc( - sizeof(THING) * COUNT * STRIDE + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) ) // 3-factor product with 2 sizeof(variable), with redundant parens removed. @@ expression THING1, THING2; identifier COUNT; type TYPE1, TYPE2; @@ ( kmalloc( - sizeof(TYPE1) * sizeof(TYPE2) * COUNT + array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2)) , ...) | kmalloc( - sizeof(TYPE1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2)) , ...) | kmalloc( - sizeof(THING1) * sizeof(THING2) * COUNT + array3_size(COUNT, sizeof(THING1), sizeof(THING2)) , ...) | kmalloc( - sizeof(THING1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(THING1), sizeof(THING2)) , ...) | kmalloc( - sizeof(TYPE1) * sizeof(THING2) * COUNT + array3_size(COUNT, sizeof(TYPE1), sizeof(THING2)) , ...) | kmalloc( - sizeof(TYPE1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(TYPE1), sizeof(THING2)) , ...) ) // 3-factor product, only identifiers, with redundant parens removed. @@ identifier STRIDE, SIZE, COUNT; @@ ( kmalloc( - (COUNT) * STRIDE * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | kmalloc( - COUNT * (STRIDE) * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | kmalloc( - COUNT * STRIDE * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | kmalloc( - (COUNT) * (STRIDE) * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | kmalloc( - COUNT * (STRIDE) * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | kmalloc( - (COUNT) * STRIDE * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | kmalloc( - (COUNT) * (STRIDE) * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | kmalloc( - COUNT * STRIDE * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) ) // Any remaining multi-factor products, first at least 3-factor products, // when they're not all constants... @@ expression E1, E2, E3; constant C1, C2, C3; @@ ( kmalloc(C1 * C2 * C3, ...) | kmalloc( - (E1) * E2 * E3 + array3_size(E1, E2, E3) , ...) | kmalloc( - (E1) * (E2) * E3 + array3_size(E1, E2, E3) , ...) | kmalloc( - (E1) * (E2) * (E3) + array3_size(E1, E2, E3) , ...) | kmalloc( - E1 * E2 * E3 + array3_size(E1, E2, E3) , ...) ) // And then all remaining 2 factors products when they're not all constants, // keeping sizeof() as the second factor argument. @@ expression THING, E1, E2; type TYPE; constant C1, C2, C3; @@ ( kmalloc(sizeof(THING) * C2, ...) | kmalloc(sizeof(TYPE) * C2, ...) | kmalloc(C1 * C2 * C3, ...) | kmalloc(C1 * C2, ...) | - kmalloc + kmalloc_array ( - sizeof(TYPE) * (E2) + E2, sizeof(TYPE) , ...) | - kmalloc + kmalloc_array ( - sizeof(TYPE) * E2 + E2, sizeof(TYPE) , ...) | - kmalloc + kmalloc_array ( - sizeof(THING) * (E2) + E2, sizeof(THING) , ...) | - kmalloc + kmalloc_array ( - sizeof(THING) * E2 + E2, sizeof(THING) , ...) | - kmalloc + kmalloc_array ( - (E1) * E2 + E1, E2 , ...) | - kmalloc + kmalloc_array ( - (E1) * (E2) + E1, E2 , ...) | - kmalloc + kmalloc_array ( - E1 * E2 + E1, E2 , ...) ) Signed-off-by: Kees Cook <keescook@chromium.org>
2018-05-29ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffersTakashi Iwai1-5/+54
Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as this serves merely as an intermediate buffer that is copied to each URB transfer buffer. This works well in general on x86, but on some archs this may result in cache coherency issues when mmap is used. OTOH, it works also on such arch unless mmap is used. This patch is a step for mitigating the inconvenience; a new module option "use_vmalloc" is provided so that user can choose to allocate the DMA coherent buffer instead of the existing vmalloc buffer. The drawback is that it'd be the standard dma_alloc_coherent() calls and the system would require contiguous pages on non-x86 archs. Note that it's a global option and not dynamically switchable since the buffer is pre-allocated at the probe time. In theory, it's possible to be switchable, but it'd be trickier and racier. As default use_vmalloc option is set to true, so that the old behavior is kept. For allowing the coherent mmap on ARM or MIPS, pass use_vmalloc=0 option explicitly. Reported-and-tested-by: Daniel Danzberger <daniel@dd-wrt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-28ALSA: usb-audio: Follow standard coding styleTakashi Iwai1-8/+10
Avoid if ((err = ...) style and expand to multiple lines instead. No change in the end result, but just the beautification. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-28ALSA: usb-audio: Move autoresume call at the end of openTakashi Iwai1-36/+37
... so that we can avoid the extra goto lines. Also beautify the code to follow the standard codex. No functional changes. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-28ALSA: usb-audio: Simplify PCM open/close callbacksTakashi Iwai1-26/+8
The stream direction in open and close callbacks can be retrieved from substream->direction, hence we don't have to stick with the unique PCM ops hard-coded for each direction. Rewrite the common open/close callback functions. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-16ALSA: usb-audio: Apply rate limit to warning messages in URB complete callbackTakashi Iwai1-1/+1
retire_capture_urb() may print warning messages when the given URB doesn't align, and this may flood the system log easily. Put the rate limit to the message for avoiding it. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1093485 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-02ALSA: usb-audio: Add keep_iface flagTakashi Iwai1-7/+10
Introduce a new flag to struct snd_usb_audio for allowing the device to skip usb_set_interface() calls at changing or closing the stream. As of this patch, the flag is nowhere set, so it's just a place holder. The dynamic switching will be added in the following patch. A background information for this change: Dell WD15 dock with Realtek chip gives a very long pause at each time the driver changes the altset, which eventually happens at every PCM stream open/close and parameter change. As the long pause happens in each usb_set_interface() call, there is nothing we can do as long as it's called. The workaround is to reduce calling it as much as possible, and this flag indicates that behavior. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-02ALSA: usb-audio: Avoid superfluous usb_set_interface() callsTakashi Iwai1-7/+4
This is a preliminary change for the upcoming quirk implementation. Currently USB-audio driver tries to call usb_set_interface() whenever the format change with interface/altset modification happens. In this patch, the check is replaced with the comparison of cur_altsetting and the targeted altsetting pointer, so that the driver may skip the unnecessary function calls. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-04-19ALSA: usb-audio: add implicit fb quirk for Axe-Fx IIIAlberto Aguirre1-0/+4
The Axe-Fx III implicit feedback end point and the data sink endpoint are in different interface descriptors. Add quirk to ensure a sync endpoint is properly configured. Signed-off-by: Alberto Aguirre <albaguirre@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-04-19ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirkAlberto Aguirre1-32/+20
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-02-12ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204Lassi Ylikojola1-0/+9
Add quirk to ensure a sync endpoint is properly configured. This patch is a fix for same symptoms on Behringer UFX1204 as patch from Albertto Aquirre on Dec 8 2016 for Axe-Fx II. Signed-off-by: Lassi Ylikojola <lassi.ylikojola@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-08-19ALSA: usb: constify snd_pcm_ops structuresArvind Yadav1-2/+2
snd_pcm_ops are not supposed to change at runtime. All functions working with snd_pcm_ops provided by <sound/pcm.h> work with const snd_pcm_ops. So mark the non-const structs as const. Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-08-17ALSA: usb: make snd_pcm_hardware constBhumika Goyal1-1/+1
Make this const as it is only used in a copy operation. Done using Coccinelle. Signed-off-by: Bhumika Goyal <bhumirks@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-01-05ALSA: usb-audio: Fix irq/process data synchronizationIoan-Adrian Ratiu1-5/+5
Commit 16200948d83 ("ALSA: usb-audio: Fix race at stopping the stream") was incomplete causing another more severe kernel panic, so it got reverted. This fixes both the original problem and its fallout kernel race/crash. The original fix is to move the endpoint member NULL clearing logic inside wait_clear_urbs() so the irq triggering the urb completion doesn't call retire_capture/playback_urb() after the NULL clearing and generate a panic. However this creates a new race between snd_usb_endpoint_start()'s call to wait_clear_urbs() and the irq urb completion handler which again calls retire_capture/playback_urb() leading to a new NULL dereference. We keep the EP deactivation code in snd_usb_endpoint_start() because removing it will break the EP reference counting (see [1] [2] for info), however we don't need the "can_sleep" mechanism anymore because a new function was introduced (snd_usb_endpoint_sync_pending_stop()) which synchronizes pending stops and gets called inside the pcm prepare callback. It also makes sense to remove can_sleep because it was also removed from deactivate_urbs() signature in [3] so we benefit from more simplification. [1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start") [2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") [3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code") Fixes: f8114f8583bb ("Revert "ALSA: usb-audio: Fix race at stopping the stream"") Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-09ALSA: usb-audio: add implicit fb quirk for Axe-Fx IIAlberto Aguirre1-0/+10
The Axe-Fx II implicit feedback end point and the data sync endpoint are in different interface descriptors. Add quirk to ensure a sync endpoint is properly configured. Signed-off-by: Alberto Aguirre <albaguirre@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-06ALSA: usb-audio: avoid setting of sample rate multiple times on busDaniel Girnus1-10/+11
Some of userland applications call 'snd_pcm_hw_params()' and 'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()' is called twice and the second 'snd_pcm_hw_prepare()' is called in 'SNDRV_PCM_STATE_PREPARED' state. Some devices are not able to manage this and they will stop playback if the sample rate will be configured several times over USB protocol. V2: updated Changelog Signed-off-by: Daniel Girnus <dgirnus@de.adit-jv.com> Signed-off-by: Jens Lorenz <jlorenz@de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-31[media] Revert "[media] sound/usb: Use Media Controller API to share media ↵Mauro Carvalho Chehab1-23/+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>
2016-03-18Merge tag 'sound-4.6-rc1' of ↵Linus Torvalds1-0/+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 ...
2016-03-16ALSA: usb-audio: Add sanity checks for endpoint accessesTakashi Iwai1-0/+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>
2016-03-03[media] sound/usb: Use Media Controller API to share media resourcesShuah Khan1-5/+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>
2015-10-19ALSA: USB-audio: Add quirk for Zoom R16/24 playbackRicard Wanderlof1-1/+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>
2015-10-19ALSA: USB-audio: Add offset parameter to copy_to_urb()Ricard Wanderlof1-6/+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>
2015-10-19ALSA: USB-audio: Also move out hwptr_done wrap from prepare_playback_urb()Ricard Wanderlof1-3/+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>
2015-10-19ALSA: USB-audio: Break out copying to urb from prepare_playback_urb()Ricard Wanderlof1-14/+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>
2015-08-26ALSA: usb-audio: Avoid nested autoresume callsTakashi Iwai1-17/+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>
2015-08-16ALSA: usb: handle descriptor with SYNC_NONE illegal valuePierre-Louis Bossart1-2/+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>
2015-08-16ALSA: usb: fix corrupted pointers due to interface setting changePierre-Louis Bossart1-0/+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>
2015-02-09ALSA: usb: update trigger timestamp on first non-zero URB submittedPierre-Louis Bossart1-0/+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>
2014-11-28ALSA: usb-audio: Add mode select quirk for Denon/Marantz DACsJurgen Kramer1-0/+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>
2014-05-02ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while ↵Sander Eikelenboom1-3/+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>
2014-04-09ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()sound-fix-3.15-rc1Tim Gardner1-1/+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>
2014-02-26ALSA: usb-audio: Use standard printk helpersTakashi Iwai1-34/+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>
2013-10-07ALSA: usb-audio: rename alt_idx to altsettingEldad Zack1-3/+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>
2013-10-07ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on errorEldad Zack1-0/+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>
2013-10-07ALSA: usb-audio: remove deactivate_endpoints()Eldad Zack1-17/+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>
2013-09-26ALSA: improve buffer size computations for USB PCM audioAlan Stern1-2/+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>
2013-08-06ALSA: usb-audio: remove implicit_fb from quirkEldad Zack1-7/+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>
2013-08-06ALSA: usb-audio: remove is_playback from implicit feedback quirksEldad Zack1-21/+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>
2013-08-06ALSA: usb-audio: do not initialize and check implicit_fbEldad Zack1-5/+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>
2013-08-06ALSA: usb-audio: reverse condition logic in set_sync_endpointEldad Zack1-39/+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>
2013-08-06ALSA: usb-audio: move implicit fb quirks to separate functionEldad Zack1-15/+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>
2013-08-06ALSA: usb-audio: separate sync endpoint setting from set_formatEldad Zack1-55/+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>
2013-08-06ALSA: usb-audio: remove assignment from if conditionEldad Zack1-1/+2
Following general kernel style. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06ALSA: usb-audio: remove disabled debug code in set_formatEldad Zack1-9/+0
Code block does not compile when enabled. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-27ALSA: usb-audio: detect implicit feedback on Roland devicesClemens Ladisch1-0/+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>
2013-06-27ALSA: usb-audio: store protocol version in struct audioformatClemens Ladisch1-3/+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>
2013-04-29ALSA: pcm_format_to_bits strong-typed conversionEldad Zack1-2/+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>
2013-04-18ALSA: snd-usb: add support for bit-reversed byte formatsDaniel Mack1-1/+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>
2013-04-18ALSA: snd-usb: add support for DSD DOP stream transportDaniel Mack1-11/+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>
2013-04-18ALSA: snd-usb: use ep->stride from urb callbacksDaniel Mack1-7/+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>
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