10 Apr, 2014
1 commit
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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__ratelimited to printk_ratelimit
+ if (printk_ratelimit() &&Cc: Jaroslav Kysela
Cc: Takashi Iwai
Cc: Eldad Zack
Cc: Daniel Mack
Cc: Clemens Ladisch
Signed-off-by: Tim Gardner
Signed-off-by: Takashi Iwai
26 Feb, 2014
1 commit
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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
07 Oct, 2013
3 commits
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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
Signed-off-by: Takashi Iwai -
If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED
should be cleared.Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai -
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
Signed-off-by: Takashi Iwai
26 Sep, 2013
1 commit
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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
CC: Clemens Ladisch
Tested-by: Daniel Mack
Tested-by: Eldad Zack
Signed-off-by: Takashi Iwai
06 Aug, 2013
8 commits
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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
Signed-off-by: Takashi Iwai -
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
Signed-off-by: Takashi Iwai -
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
Signed-off-by: Takashi Iwai -
Reverse logic on the conditions required to qualify for a sync endpoint
and remove one level of indendation.Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai -
Separate setting implicit feedback quirks from setting
a sync endpoint (which may also be explicit feedback or async).Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai -
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
Signed-off-by: Takashi Iwai -
Following general kernel style.
Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai -
Code block does not compile when enabled.
Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai
28 Jun, 2013
2 commits
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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-01Signed-off-by: Clemens Ladisch
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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
29 Apr, 2013
1 commit
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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
Signed-off-by: Takashi Iwai
18 Apr, 2013
3 commits
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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
Signed-off-by: Takashi Iwai -
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
Signed-off-by: Takashi Iwai -
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
Signed-off-by: Takashi Iwai
13 Apr, 2013
1 commit
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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 R1Also, 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
Cc:
Signed-off-by: Takashi Iwai
10 Apr, 2013
1 commit
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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
Signed-off-by: Takashi Iwai
04 Apr, 2013
2 commits
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Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai -
Change occurances of list_for_each into list_for_each_entry where
applicable.Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai
18 Mar, 2013
1 commit
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"Playback Design" products need a 50ms delay after setting the USB
interface.Signed-off-by: Daniel Mack
Reported-by: Andreas Koch
Signed-off-by: Takashi Iwai
11 Feb, 2013
1 commit
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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
Signed-off-by: Takashi Iwai
26 Jan, 2013
1 commit
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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
Signed-off-by: Takashi Iwai
23 Jan, 2013
1 commit
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This is a preliminary merge before the upcoming merge of generic parser
branch.
11 Jan, 2013
1 commit
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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
Signed-off-by: Takashi Iwai
24 Dec, 2012
1 commit
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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
Signed-off-by: Takashi Iwai
04 Dec, 2012
1 commit
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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
Cc: Daniel Mack
Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai
29 Nov, 2012
2 commits
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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
Signed-off-by: Takashi Iwai -
In this context, 0x01 is USB_ENDPOINT_XFER_ISOC.
Signed-off-by: Eldad Zack
Signed-off-by: Takashi Iwai
23 Nov, 2012
4 commits
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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
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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
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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: Jeffrey Barish
Signed-off-by: Daniel Mack
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai
21 Nov, 2012
3 commits
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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: Takashi Iwai
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As we are stopping the endpoints asynchronously now, it's better to
trigger the stop of both data and sync endpoints and wait for pending
stopping operations, instead of the sequential trigger-and-wait
procedure.So the wait argument in snd_usb_endpoint_stop() is dropped, and it's
expected that the caller synchronizes explicitly by calling
snd_usb_endpoint_sync_pending_stop(). (Actually there is only one
place calling this, so it was safe to change.)Signed-off-by: Takashi Iwai
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Reduce the redundant arguments for snd_usb_endpoint_start() and
snd_usb_endpoint_stop(). Also replaced from int to bool.No functional changes by this commit.
Signed-off-by: Takashi Iwai