ASoC: Updates for v6.5
A fairly quiet release from a core and framework point of view, but a
very big one from the point of view of new drivers:
- More refectoring from Morimoto-san, this time mainly around DAI
links and how we control the ordering of trigger() callbacks.
- Convert a lot of drivers to use maple tree based caches.
- Lots of work on the x86 driver stack.
- Compressed audio support for Qualcomm.
- Support for AMD SoundWire, Analog Devices SSM3515, Google Chameleon,
Ingenic X1000, Intel systems with various CODECs, Longsoon platforms,
Maxim MAX98388, Mediatek MT8188, Nuvoton NAU8825C, NXP platforms with
NAU8822, Qualcomm WSA884x, StarFive JH7110, Texas Instruments TAS2781.
According to CTA 861 the channel/speaker allocation info in the
audio infoframe only applies to uncompressed (PCM) audio streams.
The channel count info should indicate the number of channels
in the transmitted audio, which usually won't match the number of
channels used to transmit the compressed bitstream.
Some devices (eg some Sony TVs) will refuse to decode compressed
audio if these values are not set correctly.
To fix this we can simply set the channel count to 0 (which means
"refer to stream header") and set the channel/speaker allocation to 0
as well (which would mean stereo FL/FR for PCM, a safe value all sinks
will support) when transmitting compressed audio.
Signed-off-by: Matthias Reichl <hias@horus.com>
Link: https://lore.kernel.org/r/20230624165232.5751-1-hias@horus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SADs of compressed formats contain the channel and sample rate
info of the audio data inside the compressed stream, but when
building constraints we must use the rates and channels used to
transport the compressed streams.
eg 48kHz 6ch EAC3 needs to be transmitted as a 2ch 192kHz stream.
This patch fixes the constraints for the common AC3 and DTS formats,
the constraints for the less common MPEG, DSD etc formats are copied
directly from the info in the SADs as before as I don't have the specs
and equipment to test those.
Signed-off-by: Matthias Reichl <hias@horus.com>
Link: https://lore.kernel.org/r/20230624165216.5719-1-hias@horus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The generic snd_soc_dai_get_dlc() contains a default translation function
for DAI names which has factored out common code in a number of card
drivers, resolving the dai_name and of_node either using a driver provided
translation function or with a generic implementation. Unfortunately the
of_node can't be set by the translation function since it currently doesn't
have an interface to do that but snd_soc_dai_get_dlc() only initialises the
of_node in the case where there is no translation function.
This breaks the Meson support after conversion to use the generic helpers
since the DPCM cards for it check which component of the SoC is connected
to each link by checking the compatible for the component and the Meson
components provide a custom operation so don't use the generic code.
Fix this and potentially other cards by unconditionally storing the node in
the dai_link_component, there shouldn't be a binding specific of_node
selected since that's how we determine the translation function.
Fixes: 2e1dbea1f8 ("ASoC: meson: use snd_soc_{of_}get_dlc()")
Fixes: 3c8b586185 ("ASoC: soc-core.c: add index on snd_soc_of_get_dai_name()")
Link: https://lore.kernel.org/r/Message-Id: <20230623-asoc-fix-meson-probe-v1-1-82b2c2ec5ca4@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Yet more preliminary work for the upcoming USB gadget support.
Now export the helpers to convert between legacy MIDI1 and UMP data
for handling the MIDI 1.0 USB interface. The header file is moved to
include/sound.
The API functions are slightly changed, so that they can be used
without the direct access to snd_ump object. The allocation is done
in ump.c itself as it's a simple kcalloc().
Link: https://lore.kernel.org/r/20230623075530.10976-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new driver has two modules that both get enabled for build testing
when all codecs are selected. The comlib part has an i2c dependency,
so this remains disabled on builds without i2c, but then the other
one fails to link:
ERROR: modpost: "tasdevice_dev_bulk_write" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined!
ERROR: modpost: "tasdevice_dev_update_bits" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined!
ERROR: modpost: "tasdevice_dev_bulk_read" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined!
ERROR: modpost: "tasdevice_dev_read" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined!
ERROR: modpost: "tasdevice_dev_write" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined!
There are many ways to address this, adding an explicit dependency seems
to be the clearest method that keeps the structure of the driver otherwise
unchanged.
Fixes: ef3bcde75d ("ASoC: tas2781: Add tas2781 driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/Message-Id: <20230622101205.3180938-1-arnd@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
TDM mode for max98090 is dsp_a compatible with such limitations:
1) Up to four timeslots supported.
2) Only 16 bits timeslots supported.
3) Only 2 active timeslots (L/R) supported.
We want to setup TDM mode only when dsp_a mode is selected. So move
M98090_REG_TDM_FORMAT/M98090_REG_TDM_CONTROL registers setup from
max98090_set_tdm_slot() to the max98090_dai_set_fmt(). Also extend
max98090_set_tdm_slot() with all TDM limitations check.
Signed-off-by: Maxim Kochetkov <fido_max@inbox.ru>
Link: https://lore.kernel.org/r/Message-Id: <20230622142038.63388-1-fido_max@inbox.ru>
Signed-off-by: Mark Brown <broonie@kernel.org>
The i2s driver uses the mapped __iomem address of the FIFO as the DMA
address for the device. This apparently works on loongarch because of
the way it handles __iomem pointers as aliases of physical addresses,
but this is not portable to other architectures and causes a compiler
warning when dma addresses are not the same size as pointers:
sound/soc/loongson/loongson_i2s_pci.c: In function 'loongson_i2s_pci_probe':
sound/soc/loongson/loongson_i2s_pci.c:110:29: error: cast from pointer to integer of different size [-Werror=pointer-to-int-cast]
110 | tx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_TX_DATA;
| ^
sound/soc/loongson/loongson_i2s_pci.c:113:29: error: cast from pointer to integer of different size [-Werror=pointer-to-int-cast]
113 | rx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_RX_DATA;
| ^
Change the driver to instead use the physical address as stored in the
PCI BAR resource directly. Since 'dev_addr' is a 32-bit address, I think
this results in the same truncated address on loongarch but is otherwise
closer to portable code and avoids the warning.
Fixes: d84881e068 ("ASoC: Add support for Loongson I2S controller")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/Message-Id: <20230622101235.3230941-1-arnd@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds the required quirk to enable the Cirrus amp and correct pins
on the ASUS ROG GV601V series.
While this works if the related _DSD properties are made available, these
aren't included in the ACPI of these laptops (yet).
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Link: https://lore.kernel.org/r/20230621085715.5382-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a small patch set to change the UMP core for the upcoming
gadget driver support. Basically exporting a couple of helper
functions and adding a flag to suppress the internal UMP handling.
No functional changes by those alone.
Link: https://lore.kernel.org/r/20230621110241.4751-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is another preliminary patch for USB MIDI 2.0 gadget driver.
Export the currently local snd_ump_receive_ump_val(). It can be used
by the gadget driver for processing the UMP data.
Link: https://lore.kernel.org/r/20230621110241.4751-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For gapless playback it is possible that each track can have different
codec profile with same decoder, for example we have WMA album,
we may have different tracks as WMA v9, WMA v10 and so on
Or if DSP's like QDSP have abililty to switch decoders on single stream
for each track, then this call could be used to set new codec parameters.
Existing code does not allow to change this profile while doing gapless
playback.
Reuse existing SNDRV_COMPRESS_SET_PARAMS to set this new track params along
some additional checks to enforce proper state machine.
With this new changes now the user can call SNDRV_COMPRESS_SET_PARAMS
anytime after setting next track and additional check in write should
also ensure that params are set before writing new data.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20230619092805.21649-1-srinivas.kandagatla@linaro.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Vijendar Mukunda <Vijendar.Mukunda@amd.com>:
This patch series add support for
- Platform device creation for SoundWire Manager instances and
PDM controller.
- SoundWire DMA driver.
- Interrupt handling for SoundWire manager related interrupts,
SoundWire DMA interrupts and ACP error interrupts.
- ACP PCI driver PM ops modification with respect to SoundWire
Power modes.
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name
for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need
.of_node too. Therefor user need to arrange.
It will be more useful if it gets both .dai_name and .of_node.
This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it.
AMD SoundWire manager supports different power modes.
acp_reset flag will be set to false only when SoundWire manager power
mode is selected as ClockStop Mode. For rest of the combinations
(ACP PDM + SDW), acp_reset flag will be set to true.
When acp_reset flag is set, acp de-init and acp init sequence should
be invoked during suspend/resume callbacks.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230612095903.2113464-10-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Based on ACP pin configuration and scanning child devices
under ACP pci device ACPI scope, platform device configuration
(pdev_config) and platform device count(pdev_count) will be
calculated.
Using pdev_config and pdev_count values, ACP PCI driver will
create platform devices for Pink Sardine platform.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230612095903.2113464-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:
This patchset adds compressed offload support to Qualcomm audioreach drivers.
Currently it supports AAC, MP3 and FALC along with gapless.
Tested this on SM8450 and sc7280.
Merge series from Mark Brown <broonie@kernel.org>:
Several of the Everest Semi CODECs only support single register read and
write operations and therefore do not benefit from using the rbtree
cache over the maple tree cache, convert them to the more modern maple
tree cache.
