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2016-08-22ALSA: usb: use TEAC UD-H01 quirk for more devicesDaniel Mack
The quirk seems to be necessary not only for TEAC UD-H01 devices, but to more that are based on the Tenor 8802TL chipset. Devices built by T+A are affected too, and they apparently all use the same USB PID:PID. Extend the quirky handling for that device as well, and rename the quirks flag. Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de> Signed-off-by: Daniel Mack <daniel@zonque.org> 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 Chehab
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>
2016-03-03[media] sound/usb: Use Media Controller API to share media resourcesShuah Khan
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 Wanderlof
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>
2015-02-09ALSA: usb: update trigger timestamp on first non-zero URB submittedPierre-Louis Bossart
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>
2014-05-02ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback dataClemens Ladisch
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus causing the PC to send the samples at a wrong rate, which results in clicks and crackles in the output. Add a workaround to detect and fix the corruption. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> [mick37@gmx.de: use sender->udh01_fb_quirk rather than ep->udh01_fb_quirk in snd_usb_handle_sync_urb()] Reported-and-tested-by: Mick <mick37@gmx.de> Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07ALSA: usb-audio: rename alt_idx to altsettingEldad Zack
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>
2013-09-26ALSA: improve buffer size computations for USB PCM audioAlan Stern
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>
2013-06-27ALSA: usb-audio: store protocol version in struct audioformatClemens Ladisch
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>
2013-04-18ALSA: snd-usb: add support for bit-reversed byte formatsDaniel Mack
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 Mack
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>
2013-04-13ALSA: usb: Add quirk for 192KHz recording on E-Mu devicesCalvin Owens
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>
2012-11-26ALSA: usb-audio: add channel map supportTakashi Iwai
Add the support for channel maps of the PCM streams on USB audio devices. The channel map information is already found in ChannelConfig descriptor entries, which haven't been referred until now. Each chmap entry is added to audioformat list entry and copied to TLV dynamically instead of creating a whole chmap array. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30ALSA: usb-audio: Fix races at disconnectionTakashi Iwai
Close some races at disconnection of a USB audio device by adding the chip->shutdown_mutex and chip->shutdown check at appropriate places. The spots to put bandaids are: - PCM prepare, hw_params and hw_free - where the usb device is accessed for communication or get speed, in mixer.c and others; the device speed is now cached in subs->speed instead of accessing to chip->dev The accesses in PCM open and close don't need the mutex protection because these are already handled in the core PCM disconnection code. The autosuspend/autoresume codes are still uncovered by this patch because of possible mutex deadlocks. They'll be covered by the upcoming change to rwsem. Also the mixer codes are untouched, too. These will be fixed in another patch, too. Reported-by: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19ALSA: usb-audio: Avoid unnecessary EP setups in prepareTakashi Iwai
The recent fix for USB suspend breakage moved the code to set up EP from hw_params to prepare, but it means also the EP setup might be called multiple times unnecessarily because the prepare callback can be called multiple times without starting the stream (e.g. OSS emulation). This patch adds a new flag to struct snd_usb_substream indicating whether the setup of EP is required, and do it only when necessary, i.e. right after hw_params or suspend. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19ALSA: usb-audio: Move configuration to prepare.Dylan Reid
Move interface and endpoint configuration from hw_params to prepare callback. During system suspend/resume when the USB device power isn't cycled the interface and endpoint configuration need to be set before audio playback can continue. Resume involves another call to prepare but not to hw_params, moving it here allows a playing stream to continue after resume. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-04ALSA: snd-usb: Add quirks for Playback Designs devicesDaniel Mack
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>
2012-08-28ALSA: usb-audio: Remove obsoleted fields in struct snd_usb_substreamTakashi Iwai
The two entries are duplicated in struct snd_usb_endpoint. Seems forgotten in the last clean-up. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-08ALSA: usb-audio: Fix substream assignmentsTakashi Iwai
In 3.5 kernel, the endpoint is assigned dynamically for the substreams, but the PCM assignment still checks the presence of the endpoint pointer. This ended up in duplicated PCM substream creations at probing time, resulting in kernel warnings like: WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6() Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2 Call Trace: [<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c [<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48 [<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b [<ffffffff811292f0>] proc_register+0x169/0x1a6 [<ffffffff8112962e>] create_proc_entry+0x74/0x8c [<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd] [<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm] [<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio] [<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio] [<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb [<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio] [<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio] [<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio] ..... This patch fixes the regression by checking the fixed endpoint number for each substream instead of the endpoint pointer. Reported-and-tested-by: Jamie Heilman <jamie@audible.transient.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-21ALSA: snd-usb: remove refactorization left-oversDaniel Mack
Drop some struct members and definitions that became obsolete during the refactorization of the driver. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13ALSA: usb: Remove obsoleted fields from struct snd_usb_substreamTakashi Iwai
Many fields have been moved to struct snd_usb_endpoint. Also fix the proc output to correspond to the new structure. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13ALSA: snd-usb: switch over to new endpoint streaming logicDaniel Mack
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>
2012-04-13ALSA: snd-usb: implement new endpoint streaming modelDaniel Mack
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>
2012-02-15ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk()Xi Wang
A malicious USB device could feed in a large nr_rates value. This would cause the subsequent call to kmemdup() to allocate a smaller buffer than expected, leading to out-of-bounds access. This patch validates the nr_rates value and reuses the limit introduced in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()"). Signed-off-by: Xi Wang <xi.wang@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12ALSA: usb: refine delay information with USB frame counterPierre-Louis Bossart
Existing code only updates the audio delay when URBs were submitted/retired. This can introduce an uncertainty of 8ms on the number of samples played out with the default settings, and a lot more when URBs convey more packets to reduce the interrupt rate and power consumption. This patch relies on the USB frame counter to reduce the uncertainty to less than 2ms worst-case. The delay information essentially becomes independent of the URB size and number of packets. This should help applications like PulseAudio which require accurate audio timing. Clemens Ladisch reported a decrease of mplayer's A-V difference from nrpacks down to at most 1ms. Thanks to Clemens for also pointing out that the implementation of frame counters varies between different HCDs. Only the 8 lowest-bits are used to estimate the delay. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> [clemens: changed debug code] Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-10-27ALSA: usb-audio: automatically detect feedback formatClemens Ladisch
There are two USB Audio Class specifications (v1 and v2), but neither of them clearly defines the feedback format for high-speed UAC v1 devices. Add to this whatever the Creative and M-Audio firmware writers have been smoking, and it becomes impossible to predict the exact feedback format used by a particular device. Therefore, automatically detect the feedback format by looking at the magnitude of the first received feedback value. Also, this allows us to get rid of some special cases for E-Mu devices. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31ALSA: usb-audio: parse clock topology of UAC2 devicesDaniel Mack
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>
2010-03-05ALSA: usb-audio: use a format bitmask per alternate settingClemens Ladisch
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>
2010-03-05ALSA: usb-audio: rename substream format field to altset_idxClemens Ladisch
The snd_usb_substream::format field actually contains the index of the current alternate setting, so rename it to altset_idx to avoid confusion. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-03-05ALSA: usb-audio: refactor codeDaniel Mack
Clean up the usb audio driver by factoring out a lot of functions to separate files. Code for procfs, quirks, urbs, format parsers etc all got a new home now. Moved almost all special quirk handling to quirks.c and introduced new generic functions to handle them, so the exceptions do not pollute the whole driver. Renamed usbaudio.c to card.c because this is what it actually does now. Renamed usbmidi.c to midi.c for namespace clarity. Removed more things from usbaudio.h. The non-standard drivers were adopted accordingly. Signed-off-by: Daniel Mack <daniel@caiaq.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>