So far, the hda_codec object kept the hda_pcm list in an array, and
the codec driver was expected to assign the array. However, this
makes the object life cycle management harder, because the assigned
array is freed at the codec driver detach while it might be still
accessed by the opened streams.
In this patch, we allocate each hda_pcm object dynamically and manage
it as a linked list. Each object has a kref refcount, and both the
codec driver binder and the PCM open/close touches it, so that the
object won't be freed while in use.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The memory allocators should have already given the kernel warning
messages, thus we don't have to annoy again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the codec object to have an individual card pointer. Not only
this simplifies the redirections in many places, also this will allow
us to make each codec assigned to a different card object.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_build_pcms() does actually three things: let the codec driver
build up hda_pcm list, set the PCM default values, and call the
attach_pcm bus ops for each hda_pcm instance. The former two are
basically independent from the bus implementation, so it'd make the
code a bit more readable.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We leave the pcm field of struct hda_pcm at removal of each device, so
far. This hasn't been a problem since unbinding the codec driver
isn't supposed to happen and another route via snd_hda_codec_reset()
clears all the once. However, for a proper unbind implementation, we
need to care about it.
This patch does the thing above properly:
- Include struct hda_pcm pointer instead of struct hda_pcm_stream
pointers in struct azx_dev. This allows us to point the hda_pcm
object at dev_free callback.
- Introduce to_hda_pcm_stream() macro for better readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, we let the controller driver power down the all codecs at the
end of probe. But this can be done better in the codec's dev_register
callback. This results in the reduction of duplicated codes in each
control driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now the final bit of runtime PM cleanup: instead of manual
notification of the power up/down of the codec via hda_bus pm_notify
ops, use the standard runtime PM feature.
The child codec device will kick off the runtime PM of the parent
(PCI) device upon suspend/resume automatically. For managing whether
the link can be really turned off, we use the bit flags
bus->codec_powered instead of the earlier bus->power_keep_link_on.
flag. Each codec driver is responsible to set/clear the bit flag, and
the controller device can be turned off only when all these bits are
cleared.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We used to pass the power_save option value to hda_bus via a given
pointer. This was needed to refer to the value from the HD-audio core
side. However, after the transition to the runtime PM, this is no
longer needed.
This patch drops the power_save value indirection in hda_bus above,
and let the controller driver reprograms the autosuspend value
explicitly by a new helper, snd_hda_set_power_save(). Without this
call, the HD-audio core doesn't set up the autosuspend and flip the
runtime PM. (User may still be able to set up via sysfs, though.)
Along with this change, the pointer argument of azx_bus_create() is
dropped as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous transition of suspend/resume, now move the
power-save code to the standard runtime PM. As usual for runtime PM,
it's a bit tricky, but this simplified codes a lot in the end.
For keeping the usage compatibility, power_save module option still
controls the whole power-saving behavior on all codecs. The value is
translated to pm_runtime_*_autosuspend() and pm_runtime_allow() /
pm_runtime_forbid() calls.
snd_hda_power_up() and snd_hda_power_down() are translated to
pm_runtime_get_sync() and pm_runtime_put_autosuspend(), respectively.
Since we can do call pm_runtime_get_sync() more reliably, the sync
version is used always and snd_hda_power_up_d3wait() is dropped.
Another slight difference is that snd_hda_power_up()/down() don't call
runtime_pm code during the suspend/resume transition phase. Calling
them there isn't safe unlike our own code, resulted in unexpected
behavior (endless wakeups).
The hda_power_count tracepoint was removed, as it doesn't match well
with the new code.
Last but not least, we need to set ignore_children flag in the parent
dev.power field so that the runtime PM of the controller chip won't
get confused. The notification is still done in the bus pm_notify
callback. We'll get rid of this hack in the later patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch moves the suspend/resume mechanisms down to each codec
driver level, as we have a proper codec driver bound on the bus now.
Then we get the asynchronous PM gratis without fiddling much in the
driver level.
As a soft-landing transition, implement the common suspend/resume pm
ops for hda_codec_driver and keep the each codec driver intact. Only
the callers of suspend/resume in the controller side (azx_suspend()
and azx_resume()) are removed.
Another involved place is azx_bus_reset() calling the temporary
suspend and resume as a hackish method of bus reset. The HD-audio
core provide a helper function snd_hda_bus_reset() instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we create the standard HD-audio bus (/sys/bus/hdaudio), and bind
the codec driver with the codec device over there. This is the first
step of the whole transition so that the changes to each codec driver
are kept as minimal as possible.
Each codec driver needs to register hda_codec_driver struct containing
the currently existing preset via the new helper macro
module_hda_codec_driver(). The old hda_codec_preset_list is replaced
with this infrastructure. The generic parsers (for HDMI and other)
are also included in the preset with the special IDs to bind
uniquely.
In HD-audio core side, the device binding code is split to
hda_bind.c. It provides the snd_hda_bus_type implementation to match
the codec driver with the given codec vendor ID. It also manages the
module auto-loading by itself like before: when the matching isn't
found, it tries to probe the corresponding codec modules, and finally
falls back to the generic drivers. (The special ID mentioned above is
set at this stage.)
The only visible change to outside is that the hdaudio sysfs entry now
appears in /sys/bus/devices, not as a sound class device.
More works to move the suspend/resume and remove ops will be
(hopefully) done in later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for the hda_bus implementation, removing
the parent device setup to codec device. Since the bus and the class
devices can't be crossed over, leave the sound devices to the default
parent device as is.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A part of these drivers, especially BeBoB driver, are programmed to wait
some events. Thus the drivers should not destroy any data in .remove()
context.
This commit moves some destructors from 'struct fw_driver.remove()' to
'struct snd_card.private_free()' to shutdown safely.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently stream destructor in each driver has a problem to be called in
a context in which sound card object is released, because the destructors
call amdtp_stream_pcm_abort() and touch PCM runtime data.
The PCM runtime data is destroyed in application's context with
snd_pcm_close(), on the other hand PCM substream data is destroyed after
sound card object is released, in most case after all of ALSA character
devices are released. When PCM runtime is destroyed and PCM substream is
remained, amdtp_stream_pcm_abort() touches PCM runtime data and causes
Null-pointer-dereference.
This commit changes stream destructors and allows each driver to call
it after releasing runtime.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AMDTP helper functions increment/decrement reference counter for an
instance of FireWire unit, while it's complicated for each driver to
process error state.
In previous commit, each driver has the role of reference counting. This
commit removes this role from the helper function.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks and Dice drivers try to touch instances of FireWire unit after
sound card object is released, while references to the unit is decremented
in .remove(). When unplugging during streaming, sound card object is
released after .remove(), thus Fireworks and Dice drivers causes GPF or
Null-pointer-dereferencing to application processes because an instance of
FireWire unit was already released.
This commit adds reference-counting for FireWire unit in drivers to allow
them to touch an instance of FireWire unit after .remove(). In most case,
any operations after .remove() may be failed safely.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS doesn't seem to set up pins for 5.1 and the SPDIF out, so we need
to give explicitly here.
Reported-and-tested-by: Misan Thropos <misanthropos@gmx.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sign for microsecond (U+0085, MICRO SIGN) was encoded to '0x c2 b5'
by UTF-8 character encoding scheme. But the byte sequence was converted
to '0x c3 82 c2 b5' in a previous commit. As a result, the byte
sequence cannot represent microsecond sign in UTF-8 or ASCII. This
may confuse developers.
This commit replaces the sign to string expression with 'microseconds'
to purge superfluous troubles.
Fixes: 5c697e5b46ef("ALSA: firewire-lib: remove rx_blocks_for_midi quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here are a few small fix patches for 3.20-rc1:
- Quirks for Denon and Lifecam USB-audio devices and HD-audio
on HP laptops
- A long-time regression fix for HDSP eMADI
- Add missing DRAIN_TRIGGER flag set for ASoC intel-sst
- Trivial fixes for sequencer core and HD-audio Tegra, a LINE6
cleanup
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Merge tag 'sound-fix-3.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a few small fix patches for 3.20-rc1:
- Quirks for Denon and Lifecam USB-audio devices and HD-audio on HP
laptops
- A long-time regression fix for HDSP eMADI
- Add missing DRAIN_TRIGGER flag set for ASoC intel-sst
- Trivial fixes for sequencer core and HD-audio Tegra, a LINE6
cleanup"
* tag 'sound-fix-3.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb: Fix support for Denon DA-300USB DAC (ID 154e:1003)
ASoC: Intel: add SNDRV_PCM_INFO_DRAIN_TRIGGER flag
ALSA: usb-audio: Don't attempt to get Lifecam HD-5000 sample rate
ALSA: hda/tegra check correct return value from ioremap_resource
ALSA: hdspm - Constrain periods to 2 on older cards
ALSA: hda - enable mute led quirk for one more hp machine.
ALSA: seq: potential out of bounds in do_control()
ALSA: line6: Improve line6_read/write_data() interfaces
Some codes in snd_pcm_dev_disconnect() are still valid even for
internal PCMs, but they are skipped because of the check of
list_empty(&pcm->list) at the beginning. Remove this check and put
pcm->internal checks appropriately for internal PCM object to process
through this function.
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An internal PCM object shouldn't be added to the PCM device list, as
it's never accessed directly from the user-space, and it has no proc
or any similar accesses. Currently, it's excluded in snd_pcm_get()
and snd_pcm_next(), but it's easier not to add such an object to the
list.
Actually, the whole snd_pcm_dev_register() can be skipped for an
internal PCM. So this patch changes the code there, but also
addresses the uninitialized list_head access.
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No real functional change, only take wall clock and system time
in same routine and add accuracy report.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce more generic .get_time_info to retrieve
system timestamp and audio timestamp in single routine.
Backwards compatibility is preserved with same functionality
as with .wall_clock method (to be removed in following commits
to avoid breaking git bisect)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let userspace select audio timestamp config, ignore and zero all
other fields
Use audio_tstamp_data to retrieve config and pass report back to
user space
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let userspace select audio timestamp config when the
STATUS_EXT ioctl is used, ignore and zero all
other fields
No change for the existing STATUS ioctl, parameters
are treated as read-only.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the mixer amp is touched by control elements, we don't have to
power up always; if the codec was suspended at the time, we can just
update the amp cache and it's reflected to the hardware upon resume.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
azx_create_codec() function does actually two things: create a bus and
probe codecs. For the future work, split this to two logical
functions, azx_bus_create() and azx_probe_codecs().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For making the debugging of runtime PM easier, introduce
azx_has_pm_runtime() and use it in all places checking the runtime pm
driver capability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull dmaengine updates from Vinod Koul:
"This update brings:
- the big cleanup up by Maxime for device control and slave
capabilities. This makes the API much cleaner.
- new IMG MDC driver by Andrew
- new Renesas R-Car Gen2 DMA Controller driver by Laurent along with
bunch of fixes on rcar drivers
- odd fixes and updates spread over driver"
* 'for-linus' of git://git.infradead.org/users/vkoul/slave-dma: (130 commits)
dmaengine: pl330: add DMA_PAUSE feature
dmaengine: pl330: improve pl330_tx_status() function
dmaengine: rcar-dmac: Disable channel 0 when using IOMMU
dmaengine: rcar-dmac: Work around descriptor mode IOMMU errata
dmaengine: rcar-dmac: Allocate hardware descriptors with DMAC device
dmaengine: rcar-dmac: Fix oops due to unintialized list in error ISR
dmaengine: rcar-dmac: Fix spinlock issues in interrupt
dmaenegine: edma: fix sparse warnings
dmaengine: rcar-dmac: Fix uninitialized variable usage
dmaengine: shdmac: extend PM methods
dmaengine: shdmac: use SET_RUNTIME_PM_OPS()
dmaengine: pl330: fix bug that cause start the same descs in cyclic
dmaengine: at_xdmac: allow muliple dwidths when doing slave transfers
dmaengine: at_xdmac: simplify channel configuration stuff
dmaengine: at_xdmac: introduce save_cc field
dmaengine: at_xdmac: wait for in-progress transaction to complete after pausing a channel
ioat: fail self-test if wait_for_completion times out
dmaengine: dw: define DW_DMA_MAX_NR_MASTERS
dmaengine: dw: amend description of dma_dev field
dmatest: move src_off, dst_off, len inside loop
...
Fix problem where playback of Denon DA-300USB DAC sometimes does not
start and leads to error messages like "clock source 41 is not valid,
cannot use".
Solution: Treat this device the same as other Denon/Marantz devices in
sound/usb/quirks.c.
Tested with both PCM and DSD formats.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=93261
Signed-off-by: Frank C Guenther <bugzilla.frnkcg@spamgourmet.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no big merit to handle hda_bus_unsolicited object
individually, as it's tightly coupled with the hda_bus object itself.
Embedding it makes the code simpler in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of copying from the given template, let the caller fills the
fields after creation. This simplifies the code after all.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add SNDRV_PCM_INFO_DRAIN_TRIGGER in snd_pcm_hardware.info to
call sst_hsw_stream_set_silence_start() when PCM draining.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit removes code duplication between
snd_hdspm_{capture,playback}_ops. No semantic changes intended, this is
purely cosmetic.
Signed-off-by: Adrian Knoth <aknoth@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit removes code duplication between
snd_hdspm_{capture,playback}_release. No semantic changes intended, this
is purely cosmetic.
Signed-off-by: Adrian Knoth <aknoth@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit removes code duplication between
snd_hdspm_{capture,playback}_open. No semantic changes intended, this is
purely cosmetic.
Signed-off-by: Adrian Knoth <aknoth@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A really small cleanup to consolidate snd_find_free_minor() and
snd_kernel_minor() so that we can get rid of one more ifdef.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds a quirk to disable the check that the sample rate has been set correctly, as the Lifecam does not support getting the sample rate.
This means that we don't need to wait for the USB timeout when attempting to get the sample rate. Waiting for the timeout causes problems in some applications, which give up on the device acquisition process before it has had time to complete, resulting in no sound.
[minor tidy up by tiwai]
Signed-off-by: Joe Turner <joe@oampo.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
