The TGL RVP can be configured in many ways. We initially supported a
mixed configuration with RT711 in SoundWire mode and RT1308 in
TDM mode.
However Intel teams no longer have any hardware with this
configuration and there are no commercially-available devices using it
either, so let's remove this entry. The corresponding topology will
also be removed from the SOF tree.
This patch partially reverts Commit d985d208bf8f ("ASoC: Intel: common: add match tables for TGL w/ SoundWire")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Bard Liao <bard.liao@intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Link: https://lore.kernel.org/r/20210301235637.1177525-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Intel: soc-acpi: remove unused TGL table with rt5682 only
This patch partially reverts Commit 095ee71907ea ("ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver").
This commit was added as an enabling patch before the Maxim98373 codec
driver was available. This codec is now fully functional and the
topology with only RT5682 no longer maintained or used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Bard Liao <bard.liao@intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Link: https://lore.kernel.org/r/20210301235637.1177525-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Mon, 1 Mar 2021 23:31:55 +0000 (23:31 +0000)]
Merge series "ASoC: rsnd: cleanup ppcheck warning for Renesas sound driver" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Hi Mark, Pierre-Louis
These patches are based on below patch-set which from Pierre-Louis,
and cleanup cppcheck warnings for Rensas sound driver.
[3/5] has Reported-by Pierre-Louis tag.
Mark Brown [Mon, 1 Mar 2021 23:31:48 +0000 (23:31 +0000)]
Merge series "ASoC: fsl: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
Nothing critical and no functional changes.
The only change that needs attention if the 'fsl_ssi: remove
unnecessary tests' patch, where variables are to zero, then tested to
set register fields. Either the tests are indeed redundant or the
entire programming sequence is incorrect.
Mark Brown [Mon, 1 Mar 2021 23:31:47 +0000 (23:31 +0000)]
Merge series "ASoC: rt*: Constify static structs" from Rikard Falkeborn <rikard.falkeborn@gmail.com>:
Constify a number of static structs that are never modified in RealTek
codecs. The most important patches are the first two, which constifies
snd_soc_dai_ops and sdw_slave_ops, both which contain function pointers.
The other two patches are for good measure, since I was already touching
the code there.
When doing this, I discovered sound/soc/codecs/rt1016.c is not in a
Makefile, so there is not really any way to build it (I added locally to
the Makefile to compile-test my changes). Is this expected or an oversight?
Cezary Rojewski [Mon, 25 Jan 2021 11:54:41 +0000 (12:54 +0100)]
ASoC: Intel: Skylake: Compile when any configuration is selected
Skylake is dependent on SND_SOC_INTEL_SKYLAKE (aka "all SST platforms")
whereas selecting specific configuration such as KBL-only will not
cause driver code to compile. Switch to SND_SOC_INTEL_SKYLAKE_COMMON
dependency so selecting any configuration causes the driver to be built.
Reported-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Suggested-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Fixes: 35bc99aaa1a3 ("ASoC: Intel: Skylake: Add more platform granularity") Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20210125115441.10383-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Kai Vehmanen [Wed, 24 Feb 2021 14:15:41 +0000 (16:15 +0200)]
ASoC: SOF: Intel: hda: turn off display power in resume
Turn off display power at the end of controller resume flow. This is now
possible with the changes done in commit 87fc20e4a0cb ("ASoC: SOF:
Intel: hda: use hdac_ext fine-grained link management"). As codec driver
is able to request the HDA link to be brought back up, the controller
no longer needs to blindly keep display power enabled.
Co-developed-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Keyon Jie <yang.jie@intel.com> Link: https://lore.kernel.org/r/20210224141541.3331254-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Colin Ian King [Fri, 26 Feb 2021 18:56:53 +0000 (18:56 +0000)]
ASoC: Intel: boards: sof-wm8804: add check for PLL setting
Currently the return from snd_soc_dai_set_pll is not checking for
failure, this is the only driver in the kernel that ignores this,
so it probably should be added for sake of completeness. Fix this
by adding an error return check.
Addresses-Coverity: ("Unchecked return value") Fixes: f139546fb7d4 ("ASoC: Intel: boards: sof-wm8804: support for Hifiberry Digiplus boards") Signed-off-by: Colin Ian King <colin.king@canonical.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210226185653.1071321-1-colin.king@canonical.com Signed-off-by: Mark Brown <broonie@kernel.org>
Colin Ian King [Mon, 15 Feb 2021 20:05:01 +0000 (20:05 +0000)]
ASoC: codecs: lpass-rx-macro: remove redundant initialization of variable hph_pwr_mode
The variable hph_pwr_mode is being initialized with a value that is
never read and it is being updated later with a new value. The
initialization is redundant and can be removed.
Shengjiu Wang [Mon, 22 Feb 2021 08:40:20 +0000 (16:40 +0800)]
ASoC: fsl_sai: Add pm qos cpu latency support
On SoCs such as i.MX7ULP, cpuidle has some levels which
may disable system/bus clocks, so need to add pm_qos to
prevent cpuidle from entering low level idles and make sure
system/bus clocks are enabled when sai is active.
ASoC: constify of_phandle_args in snd_soc_get_dai_name()
The pointer to of_phandle_args passed to snd_soc_get_dai_name() and
of_xlate_dai_name() implementations is not modified. Since it is being
used only to translate passed OF node to a DAI name, it should not be
modified, so mark it as const for correctness and safer code.
sound/soc/sh/rcar/adg.c:67:9: style: The scope of the variable 'ratio' can be reduced. [variableScope]
int i, ratio;
^
sound/soc/sh/rcar/adg.c:114:6: style: The scope of the variable 'idx' can be reduced. [variableScope]
int idx, sel, div, step;
^
sound/soc/sh/rcar/adg.c:114:21: style: The scope of the variable 'step' can be reduced. [variableScope]
int idx, sel, div, step;
^
sound/soc/sh/rcar/adg.c:397:14: style: The scope of the variable 'clk' can be reduced. [variableScope]
struct clk *clk;
^
sound/soc/sh/rcar/cmd.c:46:20: style: The scope of the variable 'src' can be reduced. [variableScope]
struct rsnd_mod *src;
^
sound/soc/sh/rcar/cmd.c:47:27: style: The scope of the variable 'tio' can be reduced. [variableScope]
struct rsnd_dai_stream *tio;
^
sound/soc/sh/rcar/cmd.c:145:13: style: The scope of the variable 'ret' can be reduced. [variableScope]
int i, nr, ret;
^
sound/soc/sh/rcar/core.c:233:26: style: The scope of the variable 'io' can be reduced. [variableScope]
struct rsnd_dai_stream *io;
^
sound/soc/sh/rcar/core.c:489:19: style: The scope of the variable 'mod' can be reduced. [variableScope]
struct rsnd_mod *mod;
^
sound/soc/sh/rcar/core.c:1064:9: style: The scope of the variable 'j' can be reduced. [variableScope]
int i, j;
^
sound/soc/sh/rcar/core.c:1143:19: style: The scope of the variable 'mod' can be reduced. [variableScope]
struct rsnd_mod *mod;
^
sound/soc/sh/rcar/core.c:1261:22: style: The scope of the variable 'playback' can be reduced. [variableScope]
struct device_node *playback, *capture;
^
sound/soc/sh/rcar/core.c:1261:33: style: The scope of the variable 'capture' can be reduced. [variableScope]
struct device_node *playback, *capture;
^
sound/soc/sh/rcar/core.c:1419:29: style: The scope of the variable 'be_params' can be reduced. [variableScope]
struct snd_pcm_hw_params *be_params;
^
sound/soc/sh/rcar/core.c:1369:22: style: Local variable 'rdai' shadows outer variable [shadowVariable]
struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i);
^
sound/soc/sh/rcar/core.c:1338:19: note: Shadowed declaration
struct rsnd_dai *rdai;
^
sound/soc/sh/rcar/core.c:1369:22: note: Shadow variable
struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i);
^
sound/soc/sh/rcar/core.c:1380:22: style: Local variable 'rdai' shadows outer variable [shadowVariable]
struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i);
^
sound/soc/sh/rcar/ssi.c:170:19: style: The scope of the variable 'mod' can be reduced. [variableScope]
struct rsnd_mod *mod;
^
sound/soc/sh/rcar/ssi.c:535:6: style: The scope of the variable 'i' can be reduced. [variableScope]
int i;
^
sound/soc/sh/rcar/ssi.c:1212:19: style: The scope of the variable 'mod' can be reduced. [variableScope]
struct rsnd_mod *mod;
^
sound/soc/sh/rcar/ssi.c:328:16: portability: Shifting signed 32-bit value by 31 bits is implementation-defined behaviour [shiftTooManyBitsSigned]
ssi->cr_clk = FORCE | rsnd_rdai_width_to_swl(rdai) |
^
sound/soc/sh/rcar/ssi.c:387:12: portability: Shifting signed 32-bit value by 31 bits is implementation-defined behaviour [shiftTooManyBitsSigned]
cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai);
^
sound/soc/sh/rcar/ssiu.c:212:10: style: The scope of the variable 'shift' can be reduced. [variableScope]
int i, shift;
^
sound/soc/sh/rcar/ssiu.c:337:19: style: The scope of the variable 'mod' can be reduced. [variableScope]
struct rsnd_mod *mod;
^
sound/soc/sh/rcar/ssiu.c:362:22: style: The scope of the variable 'np' can be reduced. [variableScope]
struct device_node *np;
^
sound/soc/sh/rcar/ssiu.c:363:19: style: The scope of the variable 'mod' can be reduced. [variableScope]
struct rsnd_mod *mod;
^
sound/soc/sh/rcar/ssiu.c:366:6: style: The scope of the variable 'i' can be reduced. [variableScope]
int i;
^
sound/soc/sh/rcar/ssiu.c:397:13: style: The scope of the variable 'ret' can be reduced. [variableScope]
int i, nr, ret;
^
Julia Lawall [Sat, 13 Feb 2021 10:19:07 +0000 (11:19 +0100)]
ASoC: fsl: drop unneeded snd_soc_dai_set_drvdata
snd_soc_dai_set_drvdata is not needed when the set data comes from
snd_soc_dai_get_drvdata or dev_get_drvdata. The problem was fixed
usingthe following semantic patch: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,y,e;
@@
x = dev_get_drvdata(y->dev)
... when != x = e
- snd_soc_dai_set_drvdata(y,x);
@@
expression x,y,e;
@@
x = snd_soc_dai_get_drvdata(y)
... when != x = e
- snd_soc_dai_set_drvdata(y,x);
// </smpl>
Julia Lawall [Sat, 13 Feb 2021 10:19:06 +0000 (11:19 +0100)]
ASoC: sun4i-i2s: drop unneeded snd_soc_dai_set_drvdata
snd_soc_dai_set_drvdata is not needed when the set data comes from
snd_soc_dai_get_drvdata or dev_get_drvdata. The problem was fixed
usingthe following semantic patch: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,y,e;
@@
x = dev_get_drvdata(y->dev)
... when != x = e
- snd_soc_dai_set_drvdata(y,x);
@@
expression x,y,e;
@@
x = snd_soc_dai_get_drvdata(y)
... when != x = e
- snd_soc_dai_set_drvdata(y,x);
// </smpl>
Julia Lawall [Sat, 13 Feb 2021 10:19:05 +0000 (11:19 +0100)]
ASoC: mxs-saif: drop unneeded snd_soc_dai_set_drvdata
snd_soc_dai_set_drvdata is not needed when the set data comes from
snd_soc_dai_get_drvdata or dev_get_drvdata. The problem was fixed
usingthe following semantic patch: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,y,e;
@@
x = dev_get_drvdata(y->dev)
... when != x = e
- snd_soc_dai_set_drvdata(y,x);
@@
expression x,y,e;
@@
x = snd_soc_dai_get_drvdata(y)
... when != x = e
- snd_soc_dai_set_drvdata(y,x);
// </smpl>
In this case, the whole probe function then does nothing, so drop it.
Julia Lawall [Sat, 13 Feb 2021 10:19:04 +0000 (11:19 +0100)]
ASoC: mmp-sspa: drop unneeded snd_soc_dai_set_drvdata
snd_soc_dai_set_drvdata is not needed when the set data comes from
snd_soc_dai_get_drvdata or dev_get_drvdata. The problem was fixed
usingthe following semantic patch: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,y,e;
@@
x = dev_get_drvdata(y->dev)
... when != x = e
- snd_soc_dai_set_drvdata(y,x);
@@
expression x,y,e;
@@
x = snd_soc_dai_get_drvdata(y)
... when != x = e
- snd_soc_dai_set_drvdata(y,x);
// </smpl>
sound/soc/sh/siu_pcm.c:375:5: style: Redundant initialization for
'rt'. The initialized value is overwritten before it is
read. [redundantInitialization]
sound/soc/sh/rcar/ssi.c:403:6: style: Redundant initialization for
'wsr'. The initialized value is overwritten before it is
read. [redundantInitialization]
wsr = ssi->wsr;
^
sound/soc/sh/rcar/ssi.c:372:11: note: wsr is initialized
u32 wsr = ssi->wsr;
^
sound/soc/sh/rcar/ssi.c:403:6: note: wsr is overwritten
wsr = ssi->wsr;
^
sound/soc/sh/rcar/ctu.c:212:6: style: Variable 'ret' is reassigned a
value before the old one has been used. [redundantAssignment]
ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0",
^
sound/soc/sh/rcar/ctu.c:205:6: note: ret is assigned
ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass",
^
sound/soc/sh/rcar/ctu.c:212:6: note: ret is overwritten
ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0",
^
All the kcontrol creations are checked for errors, except for one. Add
the missing error check.
sound/soc/sh/rcar/core.c:219:9: warning: Identical condition and
return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/sh/rcar/core.c:210:6: note: If condition 'ret' is true, the
function will return/exit
if (ret)
^
sound/soc/sh/rcar/core.c:219:9: note: Returning identical expression
'ret'
return ret;
^
sound/soc/soc-pcm.c:2398:7: style: Variable 'ret' is reassigned a
value before the old one has been used. [redundantAssignment]
ret = -EINVAL;
^
sound/soc/soc-pcm.c:2395:7: note: ret is assigned
ret = -EINVAL;
^
sound/soc/soc-pcm.c:2398:7: note: ret is overwritten
ret = -EINVAL;
^
sound/soc/samsung/snow.c:112:2: style:inconclusive: Found duplicate
branches for 'if' and 'else'. [duplicateBranch]
if (rtd->num_codecs > 1)
^
sound/soc/samsung/snow.c:114:2: note: Found duplicate branches for
'if' and 'else'.
else
^
sound/soc/samsung/snow.c:112:2: note: Found duplicate branches for
'if' and 'else'.
if (rtd->num_codecs > 1)
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Fixes: 7de6b6bc1a58 ("ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer") Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org> Link: https://lore.kernel.org/r/20210219230918.5058-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/samsung/smdk_wm8994.c:179:6: style: Variable 'ret' is
reassigned a value before the old one has been
used. [redundantAssignment]
ret = devm_snd_soc_register_card(&pdev->dev, card);
^
sound/soc/samsung/smdk_wm8994.c:166:8: note: ret is assigned
ret = -EINVAL;
^
sound/soc/samsung/smdk_wm8994.c:179:6: note: ret is overwritten
ret = devm_snd_soc_register_card(&pdev->dev, card);
^
The initial authors bothered to set ret to -EINVAL and throw a
dev_err() message, so it looks like there is a missing return to avoid
continuing if the property is missing.
sound/soc/samsung/s3c24xx_simtec.c:191:7: style: Variable 'ret' is
assigned a value that is never used. [unreadVariable]
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
^
Looking at the code, it's not clear why the return value is checked in
the two other cases but not here, so mirror the behavior and add a
check.
dpcm_fe_dai_startup() (= A) calls dpcm_set_fe_runtime() (= B) to setup
DPCM runtime. From *naming point of view*, it sounds like setup function
for FE.
(A) static int dpcm_fe_dai_startup(...)
{
...
(B) dpcm_set_fe_runtime(...);
...
}
But in dpcm_set_fe_runtime() (= B),
It setups FE by dpcm_runtime_setup_fe() (= X),
and setups BE by dpcm_runtime_merge_xxx() (= Y).
dpcm_fe_dai_startup() (= A) calls dpcm_set_fe_runtime() (= B) to setup
DPCM runtime. From *naming point of view*, it sounds like setup function
for FE.
(A) static int dpcm_fe_dai_startup(...)
{
...
(B) dpcm_set_fe_runtime(...);
...
}
But in dpcm_set_fe_runtime() (= B),
It setups FE by dpcm_runtime_setup_fe() (= X),
and setups BE by dpcm_runtime_merge_xxx() (= Y).
dpcm_fe_dai_startup() (= A) calls dpcm_set_fe_runtime() (= B) to setup
DPCM runtime. From *naming point of view*, it sounds like setup function
for FE.
(A) static int dpcm_fe_dai_startup(...)
{
...
(B) dpcm_set_fe_runtime(...);
...
}
But in dpcm_set_fe_runtime() (= B),
It setups FE by for_each loop (= X),
and setups BE by dpcm_runtime_merge_xxx() (= Y).
If this for_each_rtd_cpu_dais() loop (= C) calls
dpcm_init_runtime_hw() (= A) multiple times, this means it is Multi-CPU.
If we focus to format operation at (D), using mask (= D1) is understandable
because it restricts unsupported format.
But, enabling format when zero format case (= D2) is very strange,
because it might enables unsupported format.
This runtime->hw.formats is initialized by ULLONG_MAX at soc_pcm_hw_init(),
thus becoming zero format means it can't use such format.
And doing this strange format operation is only here.
This patch removes strange format operation (= D2), and use standard
soc_pcm_hw_update_format() for it.
sound/soc/fsl/p1022_ds.c:344:6: style: Redundant initialization for
'ret'. The initialized value is overwritten before it is
read. [redundantInitialization]
ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
^
sound/soc/fsl/p1022_ds.c:203:10: note: ret is initialized
int ret = -ENODEV;
^
sound/soc/fsl/p1022_ds.c:344:6: note: ret is overwritten
ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
^
sound/soc/fsl/mpc8610_hpcd.c:333:6: style: Redundant initialization
for 'ret'. The initialized value is overwritten before it is
read. [redundantInitialization]
ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma",
^
sound/soc/fsl/mpc8610_hpcd.c:193:10: note: ret is initialized
int ret = -ENODEV;
^
sound/soc/fsl/mpc8610_hpcd.c:333:6: note: ret is overwritten
ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma",
^
ASoC: fsl: mpc5200: signed parameter in snprintf format
cppcheck warning:
sound/soc/fsl/mpc5200_dma.c:414:2: warning: %u in format
string (no. 1) requires 'unsigned int' but the argument type is
'signed int'. [invalidPrintfArgType_uint]
snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id);
^
Also fix sizeof use, missing parentheses reported by checkpatch.pl
sound/soc/fsl/fsl_easrc.c:751:53: style: Variable 'st2_mem_alloc' is
assigned a value that is never used. [unreadVariable]
int st1_chanxexp, st1_mem_alloc = 0, st2_mem_alloc = 0;
^
sound/soc/fsl/fsl_easrc.c:1331:11: style: Variable 'size' is assigned
a value that is never used. [unreadVariable]
int size = 0;
^
Rikard Falkeborn [Wed, 24 Feb 2021 21:19:16 +0000 (22:19 +0100)]
ASoC: rt*: Constify static struct snd_soc_dai_ops
The only usage of them is to assign their address to the ops field in
the snd_soc_dai_driver struct, which is a pointer to const. Make them
const to allow the compiler to put them in read-only memory.
Rikard Falkeborn [Wed, 24 Feb 2021 21:19:15 +0000 (22:19 +0100)]
ASoC: rt*: Constify static struct sdw_slave_ops
The only usage of these is to assign their address to the ops field in
the sdw_driver struct, which is a pointer to const. Make them const to
allow the compiler to put them in read-only memory.
sound/soc/codecs/lpass-wsa-macro.c:958:6: style: Variable 'ret' is
reassigned a value before the old one has been
used. [redundantAssignment]
ret = wsa_macro_set_prim_interpolator_rate(dai, (u8) rate_val, sample_rate);
^
sound/soc/codecs/lpass-wsa-macro.c:946:6: note: ret is assigned
ret = wsa_macro_set_mix_interpolator_rate(dai, (u8) rate_val, sample_rate);
^
sound/soc/codecs/lpass-wsa-macro.c:958:6: note: ret is overwritten
ret = wsa_macro_set_prim_interpolator_rate(dai, (u8) rate_val, sample_rate);
^
set_mix_interpolator_rate can return -EINVAL, add a test and bail on error.
sound/soc/codecs/wcd934x.c:1571:9: warning: Identical condition and
return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/codecs/wcd934x.c:1568:6: note: If condition 'ret' is true,
the function will return/exit
if (ret)
^
sound/soc/codecs/wcd934x.c:1571:9: note: Returning identical
expression 'ret'
return ret;
^
sound/soc/codecs/wcd9335.c:5216:9: warning: Identical condition and
return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/codecs/wcd9335.c:5211:6: note: If condition 'ret' is true,
the function will return/exit
if (ret)
^
sound/soc/codecs/wcd9335.c:5216:9: note: Returning identical
expression 'ret'
return ret;
^
sound/soc/qcom/qdsp6/q6afe.c:848:25: note: Assignment 'p=NULL',
assigned value is 0
struct q6afe_port *p = NULL;
^
sound/soc/qcom/qdsp6/q6afe.c:854:7: note: Null pointer dereference
if (p->token == token) {
^
sound/soc/qcom/qdsp6/q6afe.c:939:8: style: Redundant initialization
for 'wait'. The initialized value is overwritten before it is
read. [redundantInitialization]
wait = &port->wait;
^
sound/soc/qcom/qdsp6/q6afe.c:933:26: note: wait is initialized
wait_queue_head_t *wait = &port->wait;
^
sound/soc/qcom/qdsp6/q6afe.c:939:8: note: wait is overwritten
wait = &port->wait;
^
sound/soc/qcom/qdsp6/q6afe.c:1191:10: style: Redundant initialization
for 'port_id'. The initialized value is overwritten before it is
read. [redundantInitialization]
port_id = port->id;
^
sound/soc/qcom/qdsp6/q6afe.c:1186:14: note: port_id is initialized
int port_id = port->id;
^
sound/soc/qcom/qdsp6/q6afe.c:1191:10: note: port_id is overwritten
port_id = port->id;
^
sound/soc/qcom/lpass-hdmi.c:189:9: warning: Identical condition and
return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/qcom/lpass-hdmi.c:186:6: note: If condition 'ret' is true,
the function will return/exit
if (ret)
^
sound/soc/qcom/lpass-hdmi.c:189:9: note: Returning identical
expression 'ret'
return ret;
^
sound/soc/qcom/lpass-hdmi.c:206:9: warning: Identical condition and
return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/qcom/lpass-hdmi.c:203:6: note: If condition 'ret' is true,
the function will return/exit
if (ret)
^
sound/soc/qcom/lpass-hdmi.c:206:9: note: Returning identical
expression 'ret'
return ret;
^
Mark Brown [Tue, 9 Mar 2021 19:04:28 +0000 (19:04 +0000)]
Merge series "ASoC: sdm845: array out of bound issues" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:
During testing John Stultz and Amit reported few array our bound issues
after enabling bound sanitizer
This patch series attempts to fix those!
changes since v1:
- make sure the wcd is not de-referenced without intialization
Srinivas Kandagatla (3):
ASoC: qcom: sdm845: Fix array out of bounds access
ASoC: qcom: sdm845: Fix array out of range on rx slim channels
ASoC: codecs: wcd934x: add a sanity check in set channel map
ASoC: codecs: wcd934x: add a sanity check in set channel map
set channel map can be passed with a channel maps, however if
the number of channels that are passed are more than the actual
supported channels then we would be accessing array out of bounds.
So add a sanity check to validate these numbers!
Fixes: a61f3b4f476e ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec") Reported-by: John Stultz <john.stultz@linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20210309142129.14182-4-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: qcom: sdm845: Fix array out of range on rx slim channels
WCD934x has only 13 RX SLIM ports however we are setting it as 16
in set_channel_map, this will lead to array out of bounds error!
Orignally caught by enabling USBAN array out of bounds check:
Fixes: 5caf64c633a3 ("ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga") Reported-by: John Stultz <john.stultz@linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20210309142129.14182-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: qcom: sdm845: 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: 1b93a8843147 ("ASoC: qcom: sdm845: handle soundwire stream") Reported-by: John Stultz <john.stultz@linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20210309142129.14182-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Mon, 8 Mar 2021 16:03:32 +0000 (16:03 +0000)]
Merge series "Report jack and button detection + Capture Support" from Lucas Tanure <tanureal@opensource.cirrus.com>:
Hi All,
Here is a patch series for reporting to user space jack and button events and
add the support for Capture. With some cleanups and fixes along the way.
Regards,
Lucas Tanure
Lucas Tanure (12):
ASoC: cs42l42: Fix Bitclock polarity inversion
ASoC: cs42l42: Fix channel width support
ASoC: cs42l42: Fix mixer volume control
ASoC: cs42l42: Don't enable/disable regulator at Bias Level
ASoC: cs42l42: Always wait at least 3ms after reset
ASoC: cs42l42: Remove power if the driver is being removed
ASoC: cs42l42: Disable regulators if probe fails
ASoC: cs42l42: Provide finer control on playback path
ASoC: cs42l42: Set clock source for both ways of stream
ASoC: cs42l42: Add Capture Support
ASoC: cs42l42: Report jack and button detection
ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called
Richard Fitzgerald (3):
ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT
ASoC: cs42l42: Only start PLL if it is needed
ASoC: cs42l42: Wait for PLL to lock before switching to it
Peter Robinson [Sun, 7 Mar 2021 16:23:37 +0000 (16:23 +0000)]
ASoC: remove remnants of sirf prima/atlas audio codec
In 61fbeb5 the sirf prima/atlas drivers were removed. This cleans
up a stray header and some Kconfig entries for the codec that
were missed in the process.
Fixes: 61fbeb5dcb3d (ASoC: remove sirf prima/atlas drivers) Signed-off-by: Peter Robinson <pbrobinson@gmail.com> Cc: Arnd Bergmann <arnd@arndb.de> Cc: Mark Brown <broonie@kernel.org> Acked-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20210307162338.1160604-1-pbrobinson@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these
controls are incorrectly toggling the first bit of the register, which
is part of the FS_RATE field.
Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX"
control, which is to use SND_SOC_NOPM as the register and use an enum in
the shift field instead.
Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route") Signed-off-by: Jonathan Marek <jonathan@marek.ca> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20210305005049.24726-1-jonathan@marek.ca Signed-off-by: Mark Brown <broonie@kernel.org>
Jonathan Marek [Thu, 4 Mar 2021 21:56:46 +0000 (16:56 -0500)]
ASoC: codecs: lpass-va-macro: mute/unmute all active decimators
An interface can have multiple decimators enabled, so loop over all active
decimators. Otherwise only one channel will be unmuted, and other channels
will be zero. This fixes recording from dual DMIC as a single two channel
stream.
Also remove the now unused "active_decimator" field.
Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Jonathan Marek <jonathan@marek.ca> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20210304215646.17956-1-jonathan@marek.ca Signed-off-by: Mark Brown <broonie@kernel.org>
Lucas Tanure [Fri, 5 Mar 2021 17:34:28 +0000 (17:34 +0000)]
ASoC: cs42l42: Fix Bitclock polarity inversion
The driver was setting bit clock polarity opposite to intended polarity.
Also simplify the code by grouping ADC and DAC clock configurations into
a single field.
Jon Hunter [Wed, 3 Mar 2021 11:55:26 +0000 (11:55 +0000)]
ASoC: soc-core: Prevent warning if no DMI table is present
Many systems do not use ACPI and hence do not provide a DMI table. On
non-ACPI systems a warning, such as the following, is printed on boot.
WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name!
The variable 'dmi_available' is not exported and so currently cannot be
used by kernel modules without adding an accessor. However, it is
possible to use the function is_acpi_device_node() to determine if the
sound card is an ACPI device and hence indicate if we expect a DMI table
to be present. Therefore, call is_acpi_device_node() to see if we are
using ACPI and only parse the DMI table if we are booting with ACPI.
Mark Brown [Mon, 1 Mar 2021 23:31:40 +0000 (23:31 +0000)]
Merge series "AsoC: rt5640/rt5651: Volume control fixes" from Hans de Goede <hdegoede@redhat.com>:
Hi All,
Here is a series of rt5640/rt5651 volume-control fixes which I wrote
while working on a bytcr-rt5640 UCM profile patch-series adding
hardware-volume control to devices using this UCM profile.
The UCM series will also work on older kernels, but it works best on
kernels with this series applied, giving e.g. finer grained volume
control and support for hardware muting the outputs.
Regards,
Hans
Hans de Goede (5):
ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor
of 10
ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor
of 10
ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control
ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback
Volume'
ASoC: Intel: bytcr_rt5640: Add used AIF to the components string
Hans de Goede [Sun, 28 Feb 2021 16:04:41 +0000 (17:04 +0100)]
ASoC: es8316: Simplify adc_pga_gain_tlv table
Most steps in this table are steps of 3dB (300 centi-dB), so we can
simplify the table.
This not only reduces the amount of space it takes inside the kernel,
this also makes alsa-lib's mixer code actually accept the table, where
as before this change alsa-lib saw the "ADC PGA Gain" control as a
control without a dB scale.
The original default value written to the DAP_AVC_CTRL register during
sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to
bits 4 and 10, which are defined as RESERVED. It would also not set
bits 12 and 14 to their correct RESET values of 0x1, and instead set
them to 0x0. While the DAP_AVC module is effectively disabled because
the EN bit is 0, this default value is still writing invalid values to
registers that are marked as read-only and RESERVED as well as not
setting bits 12 and 14 to their correct default values as defined by the
datasheet.
The correct value that should be written to the DAP_AVC_CTRL register is
0x5100, which configures the register bits to the default values defined
by the datasheet, and prevents any writes to bits defined as
'read-only'. Generally speaking, it is best practice to NOT attempt to
write values to registers/bits defined as RESERVED, as it generally
produces unwanted/undefined behavior, or errors.
Also, all credit for this patch should go to my colleague Dan MacDonald
<dmacdonald@curbellmedical.com> for finding this error in the first
place.
Hans de Goede [Fri, 26 Feb 2021 14:38:14 +0000 (15:38 +0100)]
ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Hans de Goede [Fri, 26 Feb 2021 14:38:13 +0000 (15:38 +0100)]
ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc-
vol-tlv values being off by a factor of 10") which made the exact same
change to the rt5670 codec driver.
Hans de Goede [Wed, 24 Feb 2021 10:50:52 +0000 (11:50 +0100)]
ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 10-p0XX OVCD current threshold
When I added the quirk for the "HP Pavilion x2 10-p0XX" I copied the
byt_rt5640_quirk_table[] entry for the HP Pavilion x2 10-k0XX / 10-n0XX
models since these use almost the same settings.
While doing this I accidentally also copied and kept the non-standard
OVCD_TH_1500UA setting used on those models. This too low threshold is
causing headsets to often be seen as headphones (without a headset-mic)
and when correctly identified it is causing ghost play/pause
button-presses to get detected.
Correct the HP Pavilion x2 10-p0XX quirk to use the default OVCD_TH_2000UA
setting, fixing these problems.
Fixes: fbdae7d6d04d ("ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks") Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210224105052.42116-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Wed, 24 Feb 2021 16:28:50 +0000 (16:28 +0000)]
Merge series "ASoC: rt5670: Various kcontrol fixes" from Hans de Goede <hdegoede@redhat.com>:
Hi All,
While working on adding hardware-volume control support to the UCM
profile for the rt5672 and on adding LED trigger support to the
rt5670 codec driver. I hit / noticed a couple of issues this series
fixes these issues.
Regards,
Hans
Hans de Goede (4):
ASoC: rt5670: Remove 'OUT Channel Switch' control
ASoC: rt5670: Remove 'HP Playback Switch' control
ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC mixer
settings
ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control
Hans de Goede [Mon, 15 Feb 2021 14:21:18 +0000 (15:21 +0100)]
ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control
For reliable output-mute LED control we need a "DAC1 Playback Switch"
control. The "DAC Playback volume" control is the only control in the
path from the DAC1 data input to the speaker output, so the UCM profile
for the speaker output will have its PlaybackMixerElem set to "DAC1".
But userspace (pulseaudio) will set the "DAC1 Playback Volume" control to
its softest setting (which is not fully muted) while still showing the
speaker as being enabled at a low volume in the UI.
If we were to set the SNDRV_CTL_ELEM_ACCESS_SPK_LED on the "DAC1 Playback
Volume" control, this would mean then what pressing KEY_VOLUMEDOWN the
speaker-mute LED (embedded in the volume-mute toggle key) would light
while the UI is still showing the speaker as being enabled at a low
volume, meaning that the UI and the LED are out of sync.
Only after an _extra_ KEY_VOLUMEDOWN press would the UI show the
speaker as being muted.
The path from DAC1 data input to the speaker output does have
a digital mixer with DAC1's data as one of its inputs direclty after
the "DAC1 Playback Volume" control.
This commit adds an emulated "DAC1 Playback Switch" control by:
1. Declaring the enable flag for that mixers DAC1 input as well as the
"DAC1 Playback Switch" control both as SND_SOC_NOPM controls.
2. Storing the settings of both controls as driver-private data
3. Only clearing the mute flag for the DAC1 input of that mixer if the
stored values indicate both controls are enabled.
This is a preparation patch for adding "audio-mute" LED trigger support.
projects / thirdparty / kernel / linux.git / log
