Cezary Rojewski [Mon, 26 Oct 2020 10:01:29 +0000 (11:01 +0100)]
ASoC: pcm: DRAIN support reactivation
soc-pcm's dpcm_fe_dai_do_trigger() supported DRAIN commnad up to kernel
v5.4 where explicit switch(cmd) has been introduced which takes into
account all SNDRV_PCM_TRIGGER_xxx but SNDRV_PCM_TRIGGER_DRAIN. Update
switch statement to reactive support for it.
As DRAIN is somewhat unique by lacking negative/stop counterpart, bring
behaviour of dpcm_fe_dai_do_trigger() for said command back to its
pre-v5.4 state by adding it to START/RESUME/PAUSE_RELEASE group.
Fixes: 24463e83c5e3 ("ASoC: pcm: update FE/BE trigger order based on the command") Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20201026100129.8216-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Zou Wei [Thu, 5 Nov 2020 12:28:07 +0000 (20:28 +0800)]
ASoC: mediatek: mt8192: Make some symbols static
Fix the following sparse warnings:
./mt8192-dai-i2s.c:2040:5: warning: symbol 'mt8192_dai_i2s_get_share' was not declared. Should it be static?
./mt8192-dai-i2s.c:2060:5: warning: symbol 'mt8192_dai_i2s_set_priv' was not declared. Should it be static?
./mt8192-afe-gpio.c:15:16: warning: symbol 'aud_pinctrl' was not declared. Should it be static?
./mt8192-afe-pcm.c:70:5: warning: symbol 'mt8192_get_memif_pbuf_size' was not declared. Should it be static?
./mt8192-afe-pcm.c:2137:39: warning: symbol 'mt8192_afe_component' was not declared. Should it be static?
Sudip Mukherjee [Thu, 5 Nov 2020 12:47:47 +0000 (12:47 +0000)]
ASoC: mediatek: mt8192: Fix build failure
A build of arm64 allmodconfig with next-20201105 fails with the error:
ERROR: modpost: "mt8192_afe_gpio_request" undefined!
ERROR: modpost: "mt8192_afe_gpio_init" undefined!
Export the symbols so that mt8192-mt6359-rt1015-rt5682.ko finds it.
Mark Brown [Wed, 4 Nov 2020 20:40:39 +0000 (20:40 +0000)]
Merge series "ASoC: Mediatek: Add support for MT8192 SoC" from Jiaxin Yu <jiaxin.yu@mediatek.com>:
This series of patches adds support for Mediatek AFE for MT8192 SoC. At the same
time, the calibration function of MT6359 is completed with real machine driver.
The patch is based on broonie tree "for-next" branch.
Change since v3:
- use normal conditional statements to improve legiblity in [v3,3/9]
- remove mtk_i2s_hd_en_event as there's trace in the core
- impove mt8192_i2s_enum and mt8192_adda_enum
Change since v2:
- split the dai driver files as a separate patch
- fix dt-bindings to GPL-2.0-only License
- remove unnecessary preperty descriptions such as 'maxItems'
Change since v1:
- fixed some typos related to dt-bindings in [v1,3/5] and [v1,5/5]
- add vendor prefix to the properties, such as "mediatek,apmixedsys"
- add a dependency description to indicate the required header files
Jiaxin Yu (9):
ASoC: mediatek: mt6359: add the calibration functions
ASoC: mediatek: mt8192: add platform driver
ASoC: mediatek: mt8192: support i2s in platform driver
ASoC: mediatek: mt8192: support adda in platform driver
ASoC: mediatek: mt8192: support pcm in platform driver
ASoC: mediatek: mt8192: support tdm in platform driver
dt-bindings: mediatek: mt8192: add audio afe document
ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and
rt5682
dt-bindings: mediatek: mt8192: add mt8192-mt6358-rt1015-rt5682
document
Now that enum and mixer kcontrols are freed by resource management
framework, removing kcontrol becomes one function call, so simplify code
in remove_widget.
ASoC: topology: Change allocations to resource managed
In order for topology to be resource managed, change all allocations to
be resource managed:
k*alloc -> devm_k*alloc
kstrdup -> devm_kstrdup
Exceptions where non resource managed allocation is left is
soc_tplg_dapm_widget_create(), as it uses pointer to memory locally and
frees it up after use, as well as soc_tplg_dapm_graph_elems_load(),
which has temporary pointer to table of routes.
After conversion all redundant calls in error and clean up paths were
removed.
Also removed some variables which become unneeded when there is no calls
using them.
In few places tplg->comp->dev is used, while everywhere else tplg->dev
is being used. Unify those uses towards tplg->dev, as it is being set to
comp->dev during initialization anyway.
In theory topology can be loaded in multiple steps by providing index to
snd_soc_tplg_component_load, however, from usability point of view it
doesn't make sense, as can be seen from all current users loading
topology in one go. Remove the unnecessary parameter.
ASoC: topology: Remove unused functions from topology API
Topology API exposes snd_soc_tplg_widget_remove and
snd_soc_tplg_widget_remove_all, but both are nowhere used. All current
users load and unload topology as a whole. As following commits
introduce resource managed memory, remove them to simplify code and
reduce maintenance burden.
Zhang Qilong [Mon, 2 Nov 2020 10:34:28 +0000 (18:34 +0800)]
ASoC: ti: davinci-mcasp: remove always zero of davinci_mcasp_get_dt_params
davinci_mcasp_get_dt_params alway return zero, and its return value
could be ignored by the caller. So make it 'void' type to avoid the
check its return value.
Fixes: a32d8aaa04ff7 ("ASoC: ti: davinci-mcasp: Support for auxclk-fs-ratio") Signed-off-by: Zhang Qilong <zhangqilong3@huawei.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20201102103428.32678-1-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Besides the fact that muxclk is optional, muxclk can be set using
assigned-clocks, removing the need to set it in driver. The warning is
thus unneeded, so we can transform it in a debug print, eventually to just
reflect that muxclk was not set by the driver.
Brent Lu [Fri, 30 Oct 2020 17:05:59 +0000 (01:05 +0800)]
ASoC: intel: sof_rt5682: Add quirk for Dooly
This DMI product family string of this board is "Google_Hatch" so the
DMI quirk will take place. However, this board is using rt1015 speaker
amp instead of max98357a specified in the quirk. Therefore, we need an
new DMI quirk for this board.
Brent Lu [Fri, 30 Oct 2020 17:05:58 +0000 (01:05 +0800)]
ASoC: intel: sof_rt5682: Add support for cml_rt1015_rt5682
This patch adds the driver data and updates quirk info for cml with
rt1015 speaker amp and rt5682 headset codec. Due to different mclk
frequency on JSL and CML, we need to use 4 slot TDM 100fs to avoid
the SSP m/n counter.
Jernej Skrabec [Fri, 30 Oct 2020 14:46:43 +0000 (15:46 +0100)]
ASoC: sun4i-i2s: Add H6 compatible
H6 I2S is very similar to H3, except that it supports up to 16 channels
and thus few registers have fields on different position.
Signed-off-by: Jernej Skrabec <jernej.skrabec@siol.net> Signed-off-by: Marcus Cooper <codekipper@gmail.com> Acked-by: Maxime Ripard <mripard@kernel.org> Acked-by: Rob Herring <robh@kernel.org> Acked-by: Chen-Yu Tsai <wens@csie.org> Signed-off-by: Clément Péron <peron.clem@gmail.com> Link: https://lore.kernel.org/r/20201030144648.397824-11-peron.clem@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Samuel Holland [Fri, 30 Oct 2020 14:46:41 +0000 (15:46 +0100)]
ASoC: sun4i-i2s: Fix setting of FIFO modes
Because SUN4I_I2S_FIFO_CTRL_REG is volatile, writes done while the
regmap is cache-only are ignored. To work around this, move the
configuration to a callback that runs while the ASoC core has a
runtime PM reference to the device.
Signed-off-by: Samuel Holland <samuel@sholland.org> Reviewed-by: Chen-Yu Tsai <wens@csie.org> Acked-by: Maxime Ripard <mripard@kernel.org> Signed-off-by: Clément Péron <peron.clem@gmail.com> Link: https://lore.kernel.org/r/20201030144648.397824-9-peron.clem@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Marcus Cooper [Fri, 30 Oct 2020 14:46:39 +0000 (15:46 +0100)]
ASoC: sun4i-i2s: Add 20 and 24 bit support
Extend the functionality of the driver to include support of 20 and
24 bits per sample.
Signed-off-by: Marcus Cooper <codekipper@gmail.com> Acked-by: Maxime Ripard <mripard@kernel.org> Reviewed-by: Chen-Yu Tsai <wens@csie.org> Signed-off-by: Clément Péron <peron.clem@gmail.com> Link: https://lore.kernel.org/r/20201030144648.397824-7-peron.clem@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Marcus Cooper [Fri, 30 Oct 2020 14:46:38 +0000 (15:46 +0100)]
ASoC: sun4i-i2s: Set sign extend sample
On the newer SoCs such as the H3 and A64 this is set by default
to transfer a 0 after each sample in each slot. However the A10
and A20 SoCs that this driver was developed on had a default
setting where it padded the audio gain with zeros.
This isn't a problem while we have only support for 16bit audio
but with larger sample resolution rates in the pipeline then SEXT
bits should be cleared so that they also pad at the LSB. Without
this the audio gets distorted.
Set sign extend sample for all the sunxi generations even if they
are not affected. This will keep consistency and avoid relying on
default.
Signed-off-by: Marcus Cooper <codekipper@gmail.com> Reviewed-by: Chen-Yu Tsai <wens@csie.org> Acked-by: Maxime Ripard <mripard@kernel.org> Signed-off-by: Clément Péron <peron.clem@gmail.com> Link: https://lore.kernel.org/r/20201030144648.397824-6-peron.clem@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Clément Péron [Fri, 30 Oct 2020 14:46:37 +0000 (15:46 +0100)]
ASoC: sun4i-i2s: Change get_sr() and get_wss() to be more explicit
We are actually using a complex formula to just return a bunch of
simple values. Also this formula is wrong for sun4i when calling
get_wss() the function return 4 instead of 3.
Replace this with a simpler switch case.
Also drop the i2s params which is unused and return a simple int as
returning an error code could be out of range for an s8 and there is
no optim to return a s8 here.
Fixes: f34584b78dcb ("ASoC: sun4i-i2s: Change SR and WSS computation") Reviewed-by: Chen-Yu Tsai <wens@csie.org> Acked-by: Maxime Ripard <mripard@kernel.org> Signed-off-by: Clément Péron <peron.clem@gmail.com> Link: https://lore.kernel.org/r/20201030144648.397824-5-peron.clem@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Clément Péron [Fri, 30 Oct 2020 14:46:35 +0000 (15:46 +0100)]
ASoC: sun4i-i2s: Change set_chan_cfg() params
As slots and slot_width can be set manually using set_tdm().
These values are then kept in sun4i_i2s struct.
So we need to check if these values are set or not.
This is not done actually and will trigger a bug.
For example, if we set to the simple soundcard in the device-tree
dai-tdm-slot-width = <32> and then start a stream using S16_LE,
currently we would calculate BCLK for 32-bit slots, but program
lrck_period for 16-bit slots, making the sample rate double what we
expected.
To fix this, we need to check if these values are set or not but as
this logic is already done by the caller. Avoid duplicating this
logic and just pass the required values as params to set_chan_cfg().
Clément Péron [Fri, 30 Oct 2020 14:46:34 +0000 (15:46 +0100)]
ASoC: sun4i-i2s: Fix lrck_period computation for I2S justified mode
Left and Right justified mode are computed using the same formula
as DSP_A and DSP_B mode.
Which is wrong and the user manual explicitly says:
LRCK_PERDIOD:
PCM Mode: Number of BCLKs within (Left + Right) channel width.
I2S/Left-Justified/Right-Justified Mode: Number of BCLKs within each
individual channel width(Left or Right)
Fix this by using the same formula as the I2S mode.
Fixes: 3cf5b0f45a36 ("ASoC: sun4i-i2s: Add support for DSP formats") Signed-off-by: Clément Péron <peron.clem@gmail.com> Acked-by: Maxime Ripard <mripard@kernel.org> Link: https://lore.kernel.org/r/20201030144648.397824-2-peron.clem@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Update compatible string as board compatible and device compatible
should not be same!. New compatible is now suffixed with -sndcard
to be inline with other Qualcomm Sound cards.
This also fixes the warnings/error reported by dt_binding_check.
Fixes: 11e7d05c4c6b ("ASoC: qcom: dt-bindings: Add SM8250 sound card bindings") Reported-by: Rob Herring <robh@kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20201029101550.31695-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
Colin Ian King [Wed, 28 Oct 2020 11:51:12 +0000 (11:51 +0000)]
ASoC: qcom: fix unsigned int bitwidth compared to less than zero
The check for an error return from the call to snd_pcm_format_width
is never true as the unsigned int bitwidth can never be less than
zero. Fix this by making bitwidth an int.
ASoC: qcom: sm8250: Fix array out of bounds access
Static analysis Coverity had detected a potential array out-of-bounds
write issue due to the fact that MAX AFE port Id was set to 16 instead
of using AFE_PORT_MAX macro.
Fix this by properly using AFE_PORT_MAX macro.
Fixes: bff1b86219ea ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Reported-by: Colin Ian King <colin.king@canonical.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20201028142001.22431-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Tue, 27 Oct 2020 20:36:10 +0000 (20:36 +0000)]
Merge series "Add documentation and machine driver for SC7180 sound card" from Cheng-Yi Chiang <cychiang@chromium.org>:
Note:
- The machine driver patch is made by the collaboration of
Cheng-Yi Chiang <cychiang@chromium.org>
Rohit kumar <rohitkr@codeaurora.org>
Ajit Pandey <ajitp@codeaurora.org>
But Ajit has left codeaurora.
Changes from v1 to v2:
- Ducumentation: Addressed all suggestions from Doug.
- Machine driver:
- Fix comment style for license.
- Sort includes.
- Remove sc7180_snd_hw_params.
- Remove sc7180_dai_init and use aux device instead for headset jack registration.
- Statically define format for Primary MI2S.
- Atomic is not a concern because there is mutex in card to make sure
startup and shutdown happen sequentially.
- Fix missing return -EINVAL in startup.
- Use static sound card.
- Use devm_kzalloc to avoid kfree.
Changes from v2 to v3:
- Ducumentation: Addressed suggestions from Srini.
- Machine driver:
- Reuse qcom_snd_parse_of to parse properties.
- Remove playback-only and capture-only.
- Misc fixes to address comments.
Changes from v3 to v4:
- Ducumentation: Addressed suggestions from Rob.
- Remove definition of dai.
- Use 'sound-dai: true' for sound-dai schema.
- Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver.
- Machine driver:
- Add Reviewed-by: Tzung-Bi Shih <tzungbi@google.com>
Changes from v4 to v5:
- Documentation: Addressed suggestions from Rob.
- Add definition for "#address-cells" and "#size-cells".
- Add additionalProperties: false
- Add required properties.
Changes from v5 to v6:
- Documentation: Addressed suggestions from Rob.
- Drop contains in compatible strings.
- Only allow dai-link@[0-9]
- Remove reg ref since it has a type definition already.
Changes from v6 to v7
- Documentation:
- Add headset-jack and hdmi-jack to specify the codec
responsible for jack detection.
- HDMI codec driver:
- Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect.
- Machine driver:
- Removed aux device following Stephan's suggestion.
- Use headset-jack and hdmi-jack to specify the codec
responsible for jack detection.
- Add support for HDMI(actually DP) playback.
Changes from v7 to v8
- Documentation:
- Remove headset-jack and hdmi-jack.
- Machine driver:
- Let machine driver decide whether there is a jack on the DAI.
Changes from v8 to v9
- hdmi-codec driver:
- Fixed the naming.
- Machine driver:
- Fixed unused fields.
- Moved snd_soc_card_set_drvdata
- Keep the naming of HDMI as dai name until v5 of lpass-hdmi patches.
Changes from v9 to v10
- Documentation:
- Let compatible string be more specific for board configuration to allow
for future changes.
- Machine driver:
- Fixed unused include and macro.
- Add temporary macro SC7180_LPASS_DP for future change in sc7180-lpass.h.
- Let sound card be dynamically allocated.
- Change compatible string accordingly.
Changes from v10 to v11
- Machine driver:
- Use temporary macro LPASS_DP_RX for future change in sc7180-lpass.h.
Changes from v11 to v12
- Documentation:
- Change the file and title name for new compatible string google,sc7180-trogdor.
- Change the example of model name.
- Machine driver:
- Use the definitaion of index LPASS_DP_RX in sc7180-lpass.h.
- Fix for compatible string.
- Replace a comma with semicolon.
Ajit Pandey [Tue, 27 Oct 2020 03:22:34 +0000 (11:22 +0800)]
ASoC: qcom: sc7180: Add machine driver for sound card registration
Add new driver to register sound card on sc7180 trogdor board and
do the required configuration for lpass cpu dai and external codecs
connected over MI2S interfaces.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20201027032234.1705835-3-cychiang@chromium.org Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Mon, 26 Oct 2020 18:37:16 +0000 (18:37 +0000)]
Merge series "DAI driver for new XCVR IP" from "Viorel Suman (OSS)" <viorel.suman@oss.nxp.com>
Viorel Suman <viorel.suman@nxp.com>:
From: Viorel Suman <viorel.suman@nxp.com>
DAI driver for new XCVR IP found in i.MX8MP.
Viorel Suman (2):
ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver
ASoC: dt-bindings: fsl_xcvr: Add document for XCVR
Changes since v1:
- improved 6- and 12-ch layout comment
- used regmap polling function, improved
clocks handling in runtime_resume
- added FW size check in FW load function,
improved IRQ handler, removed dummy IRQ handlers
- fixed yaml file
Changes since v2:
- used devm_reset_control_get_exclusive instead of of_reset_control_get
- moved reset_control_assert into runtime_suspend
Changes since v3:
- removed "firmware-name" DTS property from both documentation and
source code by porting it into SoC specific 'compatible' data structure.
This kind of duplicated code can be a hotbed of bugs,
thus, this patch-set share soc_pcm_hw_free() and rollback.
Kuninori Morimoto (6):
ASoC: soc.h: remove for_each_rtd_dais_rollback()
ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params()
ASoC: soc-link: add mark for snd_soc_link_hw_params/free()
ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free()
ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free()
ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free()
Mark Brown [Mon, 26 Oct 2020 18:37:13 +0000 (18:37 +0000)]
Merge series "ASoC: qcom: add support for QRB5165 RB5 machine" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:
This patchset adds support to Qualcomm Robotics RB5 Development Kit based on
QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard
DMIC connected to internal LPASS codec via WSA and VA macros respectively.
// special cases where newlines are needed (hope for no more than 5)
@@
expression e1,e2;
statement S;
position p != bad.p;
position p1;
position p2 :
script:ocaml(p1) { infunction p1 && combined p1 p2 };
@@
- e1@p1,@S@p e2@p2;
+ e1; e2;
@@
expression e1,e2;
statement S;
position p != bad.p;
position p1;
position p2 :
script:ocaml(p1) { infunction p1 && combined p1 p2 };
@@
- e1@p1,@S@p e2@p2;
+ e1; e2;
@@
expression e1,e2;
statement S;
position p != bad.p;
position p1;
position p2 :
script:ocaml(p1) { infunction p1 && combined p1 p2 };
@@
- e1@p1,@S@p e2@p2;
+ e1; e2;
@@
expression e1,e2;
statement S;
position p != bad.p;
position p1;
position p2 :
script:ocaml(p1) { infunction p1 && combined p1 p2 };
@@
- e1@p1,@S@p e2@p2;
+ e1; e2;
@@
expression e1,e2;
statement S;
position p != bad.p;
position p1;
position p2 :
script:ocaml(p1) { infunction p1 && combined p1 p2 };
@@
- e1@p1,@S@p e2@p2;
+ e1; e2;
@r@
expression e1,e2;
statement S;
position p != bad.p;
@@
e1 ,@S@p e2;
@@
expression e1,e2;
position p1;
position p2 :
script:ocaml(p1) { infunction p1 && not(combined p1 p2) };
statement S;
position r.p;
@@
Mark Brown [Mon, 26 Oct 2020 18:37:11 +0000 (18:37 +0000)]
Merge series "ASoC: sun8i-codec: support for AIF2 and AIF3" from Samuel Holland <samuel@sholland.org>:
This series adds support the other two AIFs present in the sun8i codec,
which can be used for codec2codec DAI links.
This series first fills out the DAI driver, removing assumptions that
were made for AIF1 (16 bits, 2 channels, certain clock inversions). Some
new logic is required to handle 3 DAIs and the ADC/DAC sharing the same
clock. Finally, it adds the new DAIs, and hooks them up with DAPM
widgets and routes per the hardware topology.
To minimize the number of patches in this series, related device tree
patches (increasing #sound-dai-cells, adding new DAI links) will be sent
separately.
Changes from v1:
- Patches 1-8 from v1 (DAPM changes) were merged
- Prefixed AIF constants with "SUN8I_CODEC_" [1, 7, 10, 16, 17]
- Renamed variables in sun8i_codec_set_fmt for clarity [3]
- Update sysclk->sysclk_rate if later calls to hw_params change the
sample rate (thanks Chen-Yu for reminding me of this) [11]
- Select COMMON_CLK for clk_set_rate_exclusive [12]
- Add comments and hopefully clarify the clock protection logic [12]
- Make the error message more concise and put it on one line [12]
- Drop the "reg" variable holding SUN8I_AIF_CLK_CTRL(dai->id) [15]
- Rename "div_reg" to "clk_reg" and adjust comments for clarity [17]
- Improve the AIF2/AIF3 rate mismatch error message [17]
- Minor grammar/wording fixes in commit messages [2, 4, 7, 8, 16, 17]
- Added Maxime's Acked-by: [1-2, 4-9, 11, 13-14, 16]
Samuel Holland (17):
ASoC: sun8i-codec: Prepare to extend the DAI driver
ASoC: sun8i-codec: Program DAI format before clock inversion
ASoC: sun8i-codec: Enable all supported clock inversions
ASoC: sun8i-codec: Use the provided word size
ASoC: sun8i-codec: Round up the LRCK divisor
ASoC: sun8i-codec: Correct the BCLK divisor calculation
ASoC: sun8i-codec: Support the TDM slot binding
ASoC: sun8i-codec: Enforce symmetric DAI parameters
ASoC: sun8i-codec: Enable all supported sample rates
ASoC: sun8i-codec: Automatically set the system sample rate
ASoC: sun8i-codec: Constrain to compatible sample rates
ASoC: sun8i-codec: Protect the clock rate while streams are open
ASoC: sun8i-codec: Require an exact BCLK divisor match
ASoC: sun8i-codec: Enable all supported PCM formats
ASoC: sun8i-codec: Generalize AIF clock control
ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes
ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes
Mark Brown [Mon, 26 Oct 2020 18:37:10 +0000 (18:37 +0000)]
Merge series "dt-bindings: stm32: convert audio dfsdm to json-schema" from Olivier Moysan <olivier.moysan@st.com>:
Some audio properties documented in st,stm32-adfsdm.txt are already documented
in Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml bindings.
Move remaining properties from st,stm32-adfsdm.txt to st,stm32-dfsdm-adc.yaml,
and remove st,stm32-adfsdm.txt.
Changes in v2:
- Complete st,stm32-dfsdm-adc.yaml rather than converting st,stm32-adfsdm.txt
Shengjiu Wang [Thu, 15 Oct 2020 05:28:48 +0000 (13:28 +0800)]
ASoC: fsl_spdif: Add support for i.MX8QM platform
On i.MX8QM, there are separate interrupts for TX and RX.
As the EDMA can't be configured to swing back to first FIFO
after writing the second FIFO, so we need to force the burst
size to be 2 on i.MX8QM. And EDMA don't support to shift
the data from S24_LE to S16_LE, so the supported TX format
is also different on i.MX8QM.
ASoC: wm5102: Use get_unaligned_be16() for dac_comp_coeff
Replace the two-step copy-and-convert in
wm5102_out_comp_coeff_put() with get_unaligned_be16(). Apart from
looking nicer, it avoids this sparse warning:
wm5102.c:687:35: sparse: sparse: cast to restricted __be16
ASoC: adau1977: remove platform data and move micbias bindings include
The change removes the platform_data include/definition. It only contains
some values for the MICBIAS.
These are moved into 'dt-bindings/sound/adi,adau1977.h' so that they can be
used inside device-trees. When moving then, they need to be converted to
pre-compiler defines, so that the DT compiler can understand them.
The driver then, also needs to include the new
'dt-bindings/sound/adi,adau1977.h' file.
The TI PCM5102A codec driver can be used with the generic sound card
drivers, so it should be selectable. For example, with the addition
of #sound-dai-cells = <0> property in DT, it can be used with simple/graph
card drivers.
ASoC: jz4740-i2s: Remove manual DMA peripheral ID assignment
All platforms that use the jz4740-i2s driver have been switched to
devicetree for a while now and the assignment of the DMA peripheral ID
is done in the devicetree.
It is no longer necessary to manually assign the peripheral ID in the
driver, so remove that. The DMA driver does not even look at the value
assigned in the driver anymore and always uses the value provided by the
devicetree.
ASoC: qcom: sm8250: add sound card qrb5165-rb5 support
Add support to Qualcomm Robotics RB5 Development Kit based on
QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers
with onboard DMIC connected to internal LPASS codec via WSA
and VA macros respectively.
This patch adds bindings required for SM8250 based soundcards
for example Qualcomm Robotics RB5 Development Kit which makes
use of ADSP and Internal LPASS codec.
Device tree audio configuration for STM32 DFSDM is already
covered in the following binding:
Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml
Remove stm32-adfsdm.txt obsolete binding.
The difference is
soc_pcm_hw_free() is for all dai/component/substream,
rollback is for succeeded part only.
This kind of duplicated code can be a hotbed of bugs,
thus, we want to share soc_pcm_hw_free() and rollback.
Now, soc_pcm_hw_params/free() are handling
1) snd_soc_link_hw_params/free()
2) snd_soc_pcm_component_hw_params/free()
3) snd_soc_dai_hw_params/free()
Now, 1) to 3) are handled.
This patch adds new soc_pcm_hw_clean() and call it from
soc_pcm_hw_params() as rollback, and from soc_pcm_hw_free() as
normal close handler.
Other difference is that soc_pcm_hw_free() handles digital mute
if it was last user. Rollback also handles it by this patch.
This patch is for 3) snd_soc_dai_hw_params/free().
The idea of having bit-flag or counter is not enough for this purpose.
For example if one DAI is used for 2xPlaybacks for some reasons,
and if 1st Playback was succeeded but 2nd Playback was failed,
2nd Playback rollback doesn't need to call shutdown.
But it has succeeded bit-flag or counter via 1st Playback,
thus, 2nd Playback rollback will call unneeded shutdown.
And 1st Playback's necessary shutdown will not be called,
because bit-flag or counter was cleared by wrong 2nd Playback rollback.
To avoid such case, this patch marks substream pointer when hw_params() was
succeeded. If rollback needed, it will check rollback flag and marked
substream pointer.
One note here is that it cares *previous* hw_params() only now,
but we might want to check *whole* marked substream in the future.
This patch is using macro named "push/pop", so that it can be easily
update.
This patch is for 2) snd_soc_pcm_component_hw_params/free().
The idea of having bit-flag or counter is not enough for this purpose.
For example if one DAI is used for 2xPlaybacks for some reasons,
and if 1st Playback was succeeded but 2nd Playback was failed,
2nd Playback rollback doesn't need to call shutdown.
But it has succeeded bit-flag or counter via 1st Playback,
thus, 2nd Playback rollback will call unneeded shutdown.
And 1st Playback's necessary shutdown will not be called,
because bit-flag or counter was cleared by wrong 2nd Playback rollback.
To avoid such case, this patch marks substream pointer when hw_params() was
succeeded. If rollback needed, it will check rollback flag and marked
substream pointer.
One note here is that it cares *previous* hw_params() only now,
but we might want to check *whole* marked substream in the future.
This patch is using macro named "push/pop", so that it can be easily
update.
This patch is for 1) snd_soc_link_hw_params/free().
The idea of having bit-flag or counter is not enough for this purpose.
For example if one DAI is used for 2xPlaybacks for some reasons,
and if 1st Playback was succeeded but 2nd Playback was failed,
2nd Playback rollback doesn't need to call shutdown.
But it has succeeded bit-flag or counter via 1st Playback,
thus, 2nd Playback rollback will call unneeded shutdown.
And 1st Playback's necessary shutdown will not be called,
because bit-flag or counter was cleared by wrong 2nd Playback rollback.
To avoid such case, this patch marks substream pointer when hw_params() was
succeeded. If rollback needed, it will check rollback flag and marked
substream pointer.
One note here ist that it cares *previous* hw_params() only now,
but we might want to check *whole* marked substream in the future.
This patch is using macro named "push/pop", so that it can be easily
update.
commit bda7c1eef87d ("ASoC: soc-pcm: add soc_pcm_clean() and call it
from soc_pcm_open/close()") uses soc_pcm_clean() and then
for_each_rtd_dais_rollback() is no longer used.
This patch removes it.
Samuel Holland [Wed, 14 Oct 2020 06:19:41 +0000 (01:19 -0500)]
ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes
AIF3 has some differences from AIF1 and AIF2:
- It supports one channel only
- It supports master mode only
- It is not directly connected to any of the mixers; instead all audio
goes through a mux with AIF2.
- It does not have its own clock dividers; instead it reuses AIF2 BCLK
and LRCK. This means that when both AIF2 and AIF3 are active, they
must use the same sample rate and total frame width. Since AIF2 and
AIF3 are only used for codec2codec DAI links, constraints are not
applicable here; the only thing we can do when the rates don't match
is report an error.
Make the necessary adjustments to support this AIF.
Samuel Holland [Wed, 14 Oct 2020 06:19:39 +0000 (01:19 -0500)]
ASoC: sun8i-codec: Generalize AIF clock control
The AIF clock control register has the same layout for all three AIFs.
The only difference between them is that AIF3 is missing some fields. We
can reuse the same register field definitions for all three registers,
and use the DAI ID to select the correct register address.
Samuel Holland [Wed, 14 Oct 2020 06:19:38 +0000 (01:19 -0500)]
ASoC: sun8i-codec: Enable all supported PCM formats
Now that the DAI clock setup is correct for all hardware-supported PCM
formats, we can enable them in the driver. With the appropriate support
in the CPU DAI driver, this allows userspace to access the additional
formats.
Since this codec is connected to the CPU via a DAI, not directly, we do
not care if the CPU DAI is using 3-byte or 4-byte formats, so we can
support them both.
Samuel Holland [Wed, 14 Oct 2020 06:19:37 +0000 (01:19 -0500)]
ASoC: sun8i-codec: Require an exact BCLK divisor match
Now that we guarantee that SYSCLK is running at the optimal rate when
hw_params succeeds, and that it will continue running at that rate,
SYSCLK will always be an integer multiple of BCLK. So we can always
pick the exact divider, not just the closest divider.
Samuel Holland [Wed, 14 Oct 2020 06:19:36 +0000 (01:19 -0500)]
ASoC: sun8i-codec: Protect the clock rate while streams are open
The codec's clock input is shared among all AIFs, and shared with other
audio-related hardware in the SoC, including I2S and SPDIF controllers.
To ensure sample rates selected by userspace or by codec2codec DAI links
are maintained, the clock rate must be protected while it is in use.
Samuel Holland [Wed, 14 Oct 2020 06:19:35 +0000 (01:19 -0500)]
ASoC: sun8i-codec: Constrain to compatible sample rates
While another stream is active, only allow userspace to use sample rates
that are compatible with the current SYSCLK frequency. This ensures the
actual sample rate will always match what is given in hw_params.
Samuel Holland [Wed, 14 Oct 2020 06:19:34 +0000 (01:19 -0500)]
ASoC: sun8i-codec: Automatically set the system sample rate
The sun8i codec has three clock/sample rate domains:
- The AIF1 domain, with a sample rate equal to AIF1 LRCK
- The AIF2 domain, with a sample rate equal to AIF2 LRCK
- The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC),
with a sample rate given by a divisor from SYSCLK. The divisor is
controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending
on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact
sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz.
When an AIF (currently only AIF1) is active, the ADC and DAC should run
at that sample rate to avoid artifacting. Sample rate conversion is only
available when multiple AIFs are active and are routed to each other;
this means the sample rate conversion hardware usually cannot be used.
Only attach the event hook to the channel 0 AIF widgets, since we only
need one event when a DAI stream starts or stops. Channel 0 is always
brought up with a DAI stream, regardless of the number of channels in
the stream.
The ADC and DAC (along with their effects blocks) can be used even if
no AIFs are in use. In that case, we should select an appropriate sample
rate divisor, instead of keeping the last-used AIF sample rate.
44.1/48 kHz was chosen to balance audio quality and power consumption.
Since the sample rate is tied to active AIF paths, disabling pmdown_time
allows switching to the optimal sample rate immediately, instead of
after a 5 second delay.
Samuel Holland [Wed, 14 Oct 2020 06:19:33 +0000 (01:19 -0500)]
ASoC: sun8i-codec: Enable all supported sample rates
The system sample rate programmed into the hardware is really a clock
divider from SYSCLK to the ADC and DAC. Since we support two SYSCLK
frequencies, we can use all sample rates corresponding to one of those
frequencies divided by any available divisor.
This commit enables support for those sample rates. It also stops
advertising support for a 64 kHz sample rate, which is not supported.
projects / kernel.git / log