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updated solution to the problem reported with randconfig:
CONFIG_SND_SOC_SOF_IMX depends on CONFIG_SND_SOC_SOF, but is in
turn referenced by the sof-of-dev driver. This creates a reverse
dependency that manifests in a link error when CONFIG_SND_SOC_SOF_OF
is built-in but CONFIG_SND_SOC_SOF_IMX=m:
sound/soc/sof/sof-of-dev.o:(.data+0x118): undefined reference to `sof_imx8_ops'
use def_trisate to propagate the right settings without select.
Fixes: f4df4e4042 ("ASoC: SOF: imx8: Fix COMPILE_TEST error")
Fixes: 202acc565a ("ASoC: SOF: imx: Add i.MX8 HW support")
Suggested-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191101173045.27099-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When ASoC card instance is removed containing a HDA codec,
hdac_hda_codec_remove() may run in parallel with codec resume.
This will cause problems if the HDA link is freed with
snd_hdac_ext_bus_link_put() while the codec is still in
middle of its resume process.
To fix this, change the order such that pm_runtime_disable()
is called before the link is freed. This will ensure any
pending runtime PM action is completed before proceeding
to free the link.
This issue can be easily hit with e.g. SOF driver by loading and
unloading the drivers.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191101170635.26389-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The check for the mmap support via hw_support_mmap() function misses
the case where the device is with SNDRV_DMA_TYPE_DEV_UC, which should
have been treated equally as SNDRV_DMA_TYPE_DEV. Let's fix it.
Note that this bug doesn't hit any practical problem, because
SNDRV_DMA_TYPE_DEV_UC is used only for x86-specific drivers
(snd-hda-intel and snd-intel8x0) for the specific platforms that need
the non-cached buffers. And, on such platforms, hw_support_mmap()
already returns true in anyway. That's the reason I didn't put
Cc-to-stable mark here. This is only for any theoretical future
extension.
Fixes: 425da15970 ("ALSA: pcm: use dma_can_mmap() to check if a device supports dma_mmap_*")
Fixes: 42e748a0b3 ("ALSA: memalloc: Add non-cached buffer type")
Link: https://lore.kernel.org/r/20191104101115.27311-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Focusrite Saffire Pro i/o, the lowest 8 bits of register represents
configured source of sampling clock. The next lowest 8 bits represents
whether the configured source is actually detected or not just after
the register is changed for the source.
Current implementation evaluates whole the register to detect configured
source. This results in failure due to the next lowest 8 bits when the
source is connected in advance.
This commit fixes the bug.
Fixes: 25784ec2d0 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series")
Cc: <stable@vger.kernel.org> # v3.16+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191102150920.20367-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally BeBeB ASICs and firmware supports clock mode to synchronizing
to syt field of received isoc packet. This mode is known as 'SYT Match'
slightly described in IEC 61883-6 (but no detail mechanisms). In this
mode, drivers can control sampling clock in device. Driver for Windows
and macOS uses this feature to perform synchronization for devices
on the same bus.
In this mode, a plug of Music subunit for synchronization is connected
to a plug of isoc unit for incoming packet streaming, then the order to
establish connections is INPUT_PLUG first, OUTPUT_PLUG second.
This commit implements the above.
Actually each device works with its own clock for sampling, therefore
the original design is hardly implemented to vendor's products.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191101131323.17300-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as I investigated, there's some cases about the delay for device
between establishing OUTPUT_PLUG and transmitting first isoc packet. For
devices which support BeBoB protocol version 1 can transmit the packet
within several hundred milliseconds, while for devices which support
BeBoB protocol version 3 can transmit the packet within 2 seconds.
Devices with protocol version 1:
* Edirol FA-66
* Yamaha GO46
* Terratec Phase x24 FW
* M-Audio FireWire AudioPhile
* M-Audio FireWire Solo
* M-Audio FireWire 1814
* M-Audio FireWire 410
* Focusrite Saffire Pro 26 I/O
Devices with protocol version 3:
* M-Audio Profire Lightbridge
* Behringer FCA610
* Phonic Firefly 202
At present ALSA bebob driver postpones starting IR context during
1.5 sec for all of supported devices. The delay is too long for
devices with protocol version 1, while it's not enough for devices with
protocol version 3.
This commit improves the delay for these protocols.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191101131323.17300-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as I investigated, some devices with BeBoB protocol version 1
can be freezed during several hundreds milliseconds after breaking
connections. When accessing during the freezed time, any transaction
is corrupted. In the worst case, the device is going to reboot.
I can see this issue in:
* Roland FA-66
* M-Audio FireWire Solo
This commit expands sleep just after breaking connections to avoid
the freezed time as much as possible. I note that the freeze/reboot
behaviour is similar to below models:
* Focusrite Saffire Pro 10 I/O
* Focusrite Saffire Pro 26 I/O
The above models certainly reboot after breaking connections.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191101131323.17300-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a card is disconnected while in use, the system waits until all
opened files are closed then releases the card. This is done via
put_device() of the card device in each device release code.
The recently reported mutex deadlock bug happens in this code path;
snd_timer_close() for the timer device deals with the global
register_mutex and it calls put_device() there. When this timer
device is the last one, the card gets freed and it eventually calls
snd_timer_free(), which has again the protection with the global
register_mutex -- boom.
Basically put_device() call itself is race-free, so a relative simple
workaround is to move this put_device() call out of the mutex. For
achieving that, in this patch, snd_timer_close_locked() got a new
argument to store the card device pointer in return, and each caller
invokes put_device() with the returned object after the mutex unlock.
Reported-and-tested-by: Kirill A. Shutemov <kirill.shutemov@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit ade49db337 ("ALSA: hda/hdmi - Allow audio component for
AMD/ATI and Nvidia HDMI") introduced the spec->pcm_lock mutex lock to
the whole generic_hdmi_init() function for avoiding the race with the
audio component registration. However, this caused a dead lock when
the unsolicited event is handled without the audio component, as the
codec gets runtime-resumed in hdmi_present_sense() which is already
inside the spec->pcm_lock in its caller.
For avoiding this deadlock, add a new mutex only for the audio
component binding that is used in both generic_hdmi_init() and the
audio notifier registration where the jack callbacks are handled /
re-registered.
Fixes: ade49db337 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI")
Reported-and-tested-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
Link: https://lore.kernel.org/r/s5himo7i89i.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_tplg_component_remove() is pair of snd_soc_tplg_component_load(),
and it is topology related cleanup function.
The driver which called _load() needs to call _remove() by its responsibility.
Today, skl-pcm and topology are the user, and these are calling both
_load() and _remove().
soc-core doesn't need to call it.
This patch remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736fbdnwt.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixup this warning
LINUX/sound/soc/codecs/rt5677-spi.c: In function ‘rt5677_spi_pcm_close’:
LINUX/sound/soc/codecs/rt5677-spi.c:114:30: warning: unused variable ‘rtd’ [-Wunused-variable]
struct snd_soc_pcm_runtime *rtd = substream->private_data;
^~~
Fixes: a0e0d13542 ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a79idajh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In MOTU FireWire series, devices which support protocol version 2 have
several types of hardware design to process audio data frames for isoc
packet. Roughly devices are categorized into three groups:
- 828mkII
- Traveler/896HD
- UltraLite/8pre FireWire
Some bit flags in register addressed by 0x'ffff'f000'0b14
includes device-specific effects.
This commit cleanups implementation of protocol v2 in this point.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191030080644.1704-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In MOTU FireWire series, devices have a mode to generate sampling clock
from a sequence of source packet header (SPH) included in each data block
of received packet. This mode is used for several purposes such as mode
for SMPTE time code, sync to the other sound cards and so on.
This commit adds support for the SPH mode.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191030080644.1704-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To support the DP-MST multiple streams via single connector feature,
the HDMI driver was extended with the concept of backup PCMs. See
commit 9152085def ("ALSA: hda - add DP MST audio support").
This implementation works fine with snd_hda_intel.c as PCM topology
is fully managed within the single driver.
When the HDA codec driver is used from ASoC components, the concept
of backup PCMs no longer fits. For ASoC topologies, the physical
HDMI converters are presented as backend DAIs and these should match
with hardware capabilities. The ASoC topology may define arbitrary
PCMs (i.e. frontend DAIs) and have processing elements before eventual
routing to the HDMI BE DAIs. With backup PCMs, the link between
FE and BE DAIs would become dynamic and change when monitors are
(un)plugged. This would lead to modifying the topology every time
hotplug events happen, which is not currently possible in ASoC and
there does not seem to be any obvious benefits from this design.
To overcome above problems and enable the HDMI driver to be used
from ASoC, this patch adds a new mode (mst_no_extra_pcms flags) to
patch_hdmi.c. In this mode, the codec driver does not assume
the backup PCMs to be created.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191029134017.18901-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Support three different use cases with rockchip_max98090 driver.
The three use cases:
- max98090 only: Current usage, where HDMI link is missing.
- HDMI only: Needed use case for veyron_mickey board.
- max98090 + HDMI: Ideal use case for veyron_jerry, veyron_minnie..etc.
The presence of max98090 is determined by the presence of
rockchip,audio-codec device property.
The presence of HDMI is determined by the presence of
rockchip,hdmi-codec device property.
Create different sound card based on the configuration.
Note that we keep max98090 only usage for backward compatibility.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20191028071930.145899-4-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add one optional property "rockchip,hdmi-codec" to let user specify HDMI
device node in DTS so machine driver can find hdmi-codec device node for
HDMI codec DAI.
Use the presence of rockchip,audio-codec and rockchip,hdmi-codec to
specify the use case.
Use max98090 only : specify rockchip,audio-codec.
Use HDMI only: specify rockchip,hdmi-codec.
Use both max98090 and HDMI: specify rockchip,audio-codec and
rockchip,hdmi-codec.
Move these properties to optional because they are not needed for
HDMI-only use case.
"rockchip,audio-codec": The phandle of the MAX98090 audio codec
"rockchip,headset-codec": The phandle of Ext chip for jack detection
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20191028071930.145899-3-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow codec driver register callback function for plug event.
The callback registration flow:
dw-hdmi <--- hw-hdmi-i2s-audio <--- hdmi-codec
dw-hdmi-i2s-audio implements hook_plugged_cb op
so codec driver can register the callback.
dw-hdmi exports a function dw_hdmi_set_plugged_cb so platform device
can register the callback.
When connector plug/unplug event happens, report this event using the
callback.
Make sure that audio and drm are using the single source of truth for
connector status.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20191028071930.145899-2-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
