- 10 4月, 2014 1 次提交
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由 Tim Gardner 提交于
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: NTim Gardner <tim.gardner@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 2月, 2014 1 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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- 07 10月, 2013 3 次提交
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由 Eldad Zack 提交于
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: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED should be cleared. Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
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: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 9月, 2013 1 次提交
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由 Alan Stern 提交于
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: NAlan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: NDaniel Mack <zonque@gmail.com> Tested-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 06 8月, 2013 8 次提交
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由 Eldad Zack 提交于
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: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
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: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
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: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
Reverse logic on the conditions required to qualify for a sync endpoint and remove one level of indendation. Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
Separate setting implicit feedback quirks from setting a sync endpoint (which may also be explicit feedback or async). Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
Setting the sync endpoint currently takes up about half of set_format(). Move it to a dedicated function. No functional change. Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
Following general kernel style. Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
Code block does not compile when enabled. Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 28 6月, 2013 2 次提交
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由 Clemens Ladisch 提交于
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: NClemens Ladisch <clemens@ladisch.de>
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由 Clemens Ladisch 提交于
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: NClemens Ladisch <clemens@ladisch.de>
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- 29 4月, 2013 1 次提交
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由 Eldad Zack 提交于
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: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 4月, 2013 3 次提交
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由 Daniel Mack 提交于
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: NDaniel Mack <zonque@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
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: NDaniel Mack <zonque@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
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: NDaniel Mack <zonque@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 4月, 2013 1 次提交
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由 Calvin Owens 提交于
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: NCalvin Owens <jcalvinowens@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 4月, 2013 1 次提交
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由 Daniel Mack 提交于
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: NDaniel Mack <zonque@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 04 4月, 2013 2 次提交
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由 Eldad Zack 提交于
Correct spelling of snd_usb_endpoint_implict_feedback_sink in all occurances. Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
Change occurances of list_for_each into list_for_each_entry where applicable. Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 3月, 2013 1 次提交
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由 Daniel Mack 提交于
"Playback Design" products need a 50ms delay after setting the USB interface. Signed-off-by: NDaniel Mack <zonque@gmail.com> Reported-by: NAndreas Koch <andreas@akdesigninc.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 11 2月, 2013 1 次提交
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由 Matt Gruskin 提交于
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: NMatt Gruskin <matthew.gruskin@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 1月, 2013 1 次提交
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由 Takashi Iwai 提交于
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: NStratos Karafotis <stratosk@semaphore.gr> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 11 1月, 2013 1 次提交
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由 Takashi Iwai 提交于
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: NJens Axboe <axboe@kernel.dk> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 24 12月, 2012 1 次提交
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由 Pierre-Louis Bossart 提交于
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 3f94fad0. Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 04 12月, 2012 1 次提交
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由 Eldad Zack 提交于
Commit 947d2996 , "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: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 29 11月, 2012 2 次提交
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由 Eldad Zack 提交于
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: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Eldad Zack 提交于
In this context, 0x01 is USB_ENDPOINT_XFER_ISOC. Signed-off-by: NEldad Zack <eldad@fogrefinery.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 23 11月, 2012 3 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
Jeffrey Barish reported an obvious bug in the pcm part of the usb-audio driver which causes the code to not initialize the sync endpoint from configure_endpoint(). Reported-by: NJeffrey Barish <jeff_barish@earthlink.net> Signed-off-by: NDaniel Mack <zonque@gmail.com> Cc: stable@kernel.org [3.5+] Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 21 11月, 2012 3 次提交
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由 Takashi Iwai 提交于
PCM hw_free and close should wait until all the pending stop operations have been finished. Basically only PCM trigger callback should use non-wait calls. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
As we are stopping the endpoints asynchronously now, it's better to trigger the stop of both data and sync endpoints and wait for pending stopping operations, instead of the sequential trigger-and-wait procedure. So the wait argument in snd_usb_endpoint_stop() is dropped, and it's expected that the caller synchronizes explicitly by calling snd_usb_endpoint_sync_pending_stop(). (Actually there is only one place calling this, so it was safe to change.) Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Reduce the redundant arguments for snd_usb_endpoint_start() and snd_usb_endpoint_stop(). Also replaced from int to bool. No functional changes by this commit. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 08 11月, 2012 1 次提交
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由 Takashi Iwai 提交于
There are bug reports of a crash with USB-audio devices when PCM prepare is performed immediately after the stream is stopped via trigger callback. It turned out that the problem is that we don't wait until all URBs are killed. This patch adds a new function to synchronize the pending stop operation on an endpoint, and calls in the prepare callback for avoiding the crash above. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181Reported-and-tested-by: NArtem S. Tashkinov <t.artem@lycos.com> Cc: <stable@vger.kernel.org> [v3.6] Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 30 10月, 2012 1 次提交
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由 Takashi Iwai 提交于
Replace mutex with rwsem for codec->shutdown protection so that concurrent accesses are allowed. Also add the protection to snd_usb_autosuspend() and snd_usb_autoresume(), too. Reported-by: NMatthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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