| Age | Commit message (Collapse) | Author | Files | Lines |
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Kernel crashes when an ASoC component rebinding.
The dai_link->platforms has been reset to NULL by soc_cleanup_platform()
in soc_cleanup_card_resources() when un-registering component. However,
it has no chance to re-allocate the dai_link->platforms when registering
the component again.
Move the DAI pre-links initiation from snd_soc_register_card() to
snd_soc_instantiate_card() to make sure all DAI pre-links get initiated
when component rebinding.
As an example, by using the following commands:
- echo -n max98357a > /sys/bus/platform/drivers/max98357a/unbind
- echo -n max98357a > /sys/bus/platform/drivers/max98357a/bind
Got the error message:
"Unable to handle kernel NULL pointer dereference at virtual address".
The call trace:
snd_soc_is_matching_component+0x30/0x6c
soc_bind_dai_link+0x16c/0x240
snd_soc_bind_card+0x1e4/0xb10
snd_soc_add_component+0x270/0x300
snd_soc_register_component+0x54/0x6c
Signed-off-by: Tzung-Bi Shih <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The platform override code uses devm_ functions to allocate memory for
the new name but the card device is not initialized. Fix by moving the
init earlier.
Fixes: f403906da05cd ("ASoC: Intel: cht_bsw_rt5672: platform name fixup support")
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The platform override code uses devm_ functions to allocate memory for
the new name but the card device is not initialized. Fix by moving the
init earlier.
Fixes: 4506db8043341 ("ASoC: Intel: cht_bsw_nau8824: platform name fixup support")
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The platform override code uses devm_ functions to allocate memory for
the new name but the card device is not initialized. Fix by moving the
init earlier.
Fixes: e4bc6b1195f64 ("ASoC: Intel: bytcht_es8316: platform name fixup support")
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The platform override code uses devm_ functions to allocate memory for
the new name but the card device is not initialized. Fix by moving the
init earlier.
Fixes: 7e7e24d7c7ff0 ("ASoC: Intel: cht_bsw_max98090_ti: platform name fixup support")
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Whilst testing the capture functionality of the i2s on the newer
SoCs it was noticed that the recording was somewhat distorted.
This was due to the offset not being set correctly on the receiver
side.
Signed-off-by: Marcus Cooper <[email protected]>
Acked-by: Maxime Ripard <[email protected]>
Acked-by: Chen-Yu Tsai <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Although not causing any noticeable issues, the mask for the
channel offset is covering too many bits.
Signed-off-by: Marcus Cooper <[email protected]>
Acked-by: Maxime Ripard <[email protected]>
Acked-by: Chen-Yu Tsai <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The supported formats are S16_LE and S24_LE now. However, by datasheet
of max98090, S24_LE is only supported when it is in the right justified
mode. We should remove 24-bit format if it is not in that mode to avoid
triggering error.
Signed-off-by: Yu-Hsuan Hsu <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Fix gcc build error while CONFIG_I2C is not set
sound/soc/codecs/da7219.c:2640:1: warning: data definition has no type or storage class
module_i2c_driver(da7219_i2c_driver);
^~~~~~~~~~~~~~~~~
sound/soc/codecs/da7219.c:2640:1: error: type defaults to int in declaration of module_i2c_driver [-Werror=implicit-int]
sound/soc/codecs/da7219.c:2640:1: warning: parameter names (without types) in function declaration
sound/soc/codecs/da7219.c:2629:26: warning: da7219_i2c_driver defined but not used [-Wunused-variable]
Reported-by: Hulk Robot <[email protected]>
Fixes: 6d817c0e9fd7 ("ASoC: codecs: Add da7219 codec driver")
Signed-off-by: YueHaibing <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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while building without PCI:
sound/soc/sof/intel/hda.o: In function `hda_dsp_probe':
hda.c:(.text+0x79c): undefined reference to `pci_ioremap_bar'
hda.c:(.text+0x79c): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `pci_ioremap_bar'
hda.c:(.text+0x7c4): undefined reference to `pci_ioremap_bar'
hda.c:(.text+0x7c4): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `pci_ioremap_bar'
Reported-by: Hulk Robot <[email protected]>
Fixes: e13ef82a9ab8 ("ASoC: SOF: add COMPILE_TEST for PCI options")
Signed-off-by: YueHaibing <[email protected]>
Acked-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The definitions for DSP oops structures were not aligned
correctly to current FW ABI version 3.6.0, leading to
invalid data being printed out to debug logs. Fix the structs
and update related platform code accordingly.
Signed-off-by: Kai Vehmanen <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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HDA_DEV_ASOC type codec device refcounts are managed differently
from HDA_DEV_LEGACY devices. The refcount is released explicitly
in snd_hdac_ext_bus_device_remove() for ASOC type devices.
So, remove the put_device() call in snd_hda_codec_dev_free()
for such devices to make the refcount balanced. This will prevent
the NULL pointer exception when the codec driver is released
after the card is freed.
Signed-off-by: Ranjani Sridharan <[email protected]>
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Previously the structure used bitfields, which do not guarantee bit
ordering.
This change makes sure the order is clearly defined. It also renames
and repurposes the field for general use.
Signed-off-by: Slawomir Blauciak <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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We had a couple of misses with ABI changes, e.g. for Xtensa oops
information and the integration of sound trigger, before we set-up a
formal process to track evolutions.
With this patch, the SOF kernel patches are officially aligned with
the firmware 3.6 level. Changing this level has no impact on existing
users and is fully backwards-compatible.
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Add soundwire dai type and update ABI version.
Signed-off-by: Pan Xiuli <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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We missed these two definitions for GDB support and component
notifications, they are defined for the SOF firmware. Since they are
not used by the kernel so far, we can still add them without any ABI
change.
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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This tablet has an incorrect acpi identifier just like
Thinkpad10 tablet, which is why it is trying to load the RT5640 driver
instead of the RT5762 driver. The RT5640 driver, on the other hand, checks
the hardware ID, so no driver are loaded during boot. This fix resolves to
load the RT5672 driver on this tablet during boot. It also provides the
correct IO configuration, like the jack detect mode 3, for 1.8V pullup. I
would like to thank Pierre-Louis Bossart for helping with this patch.
Signed-off-by: Kovács Tamás <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Allwinner DAC seems to have a delay in the Speaker audio routing. When
playing a sound for the first time, the sound gets chopped. On a second
play the sound is played correctly. After some time (~5s) the issue gets
back.
This commit seems to be fixing the same issue as bf14da7 but
for another codepath.
This is the DTS that was used to debug the problem.
&codec {
allwinner,pa-gpios = <&r_pio 0 11 GPIO_ACTIVE_HIGH>; /* PL11 */
allwinner,audio-routing =
"Speaker", "LINEOUT";
status = "okay";
}
Signed-off-by: Georgii Staroselskii <[email protected]>
Reviewed-by: Chen-Yu Tsai <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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re-write hda_init_caps and remove the HDA reset, clean HDA
streams and clear interrupt steps in hda_dsp_probe so the
HDA init steps will not be called twice if the
CONFIG_SND_SOC_SOF_HDA is true.
Fixes: 8a300c8fb17 ("ASoC: SOF: Intel: Add HDA controller for Intel DSP")
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Zhu Yingjiang <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Currently on all supported platforms the IPC IRQ thread first signals
the sender when an IPC response is received from the DSP, then unmasks
the IPC interrupt. Those actions are performed without holding any
locks, so the thread can be interrupted between them. IPC timeouts
have been observed in such scenarios: if the sender is woken up and it
proceeds with sending the next message without unmasking the IPC
interrupt, it can miss the next response. This patch takes a spin-lock
to prevent the IRQ thread from being preempted at that point. It also
makes sure, that the next IPC transmission by the host cannot take
place before the IRQ thread has finished updating all the required IPC
registers.
Fixes: 53e0c72d98b ("ASoC: SOF: Add support for IPC IO between DSP and Host")
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Guennadi Liakhovetski <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The size for the bytes kcontrol should include the abi header, that is,
data->size + sizeof(*data), it is also aligned with get method after
this change.
Fixes: c3078f53970 ("ASoC: SOF: Add Sound Open Firmware KControl support")
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Keyon Jie <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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If the SOF hw_params() fail, typically with an IPC error thrown by the
firmware, the period_elapsed workqueue is not initialized, but we
still cancel it in hw_free(), which results in a kernel warning.
Move the initialization to the .open callback. Tested on Broadwell
(Samus) and IceLake.
Fixes: e2803e610ae ("ASoC: SOF: PCM: add period_elapsed work to fix
race condition in interrupt context")
GitHub issue: https://github.com/thesofproject/linux/issues/932
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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sof_pcm_hw_params() can only be called once to setup the FW hw_params.
So after calling sof_pcm_hw_params(), hw_params_upon_resume flag must
be cleared to avoid multiple invoking sof_pcm_hw_params() by prepare.
For example, after resume, there is an xrun happened, prepare() will
be called. As the hw_params_upon_resume flag is not cleared,
sof_pcm_hw_params() will be called and this will cause IPC timeout.
This patch fixes such issues.
Fixes: 868bd00f495 ("ASoC: SOF: Add PCM operations support")
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Libin Yang <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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In some configurations, it's a requirement to split the probe in two,
with a second part handled in a workqueue (e.g. for HDMI support
which depends on the DRM modules).
SOF already handles these configurations but the error flow is
incorrect. When an error occurs in the workqueue, the probe has
technically already completed. If we release the resources on errors,
this generates kernel oops/use-after-free when the resources are
released a second time on module removal.
GitHub issue: https://github.com/thesofproject/linux/issues/945
Fixes: c16211d6226 ("ASoC: SOF: Add Sound Open Firmware driver core")
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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No need to call snd_soc_unregister_component in case of error
because the component device is resource-managed.
Fixes: c16211d6226 ("ASoC: SOF: Add Sound Open Firmware driver core")
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Ranjani Sridharan <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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snd_sof_remove() disables the DSP and unmaps the DSP BAR.
Removing topology after disabling the DSP results in a
kernel panic while unloading the pipeline widget. This is
because pipeline widget unload attempts to power down
the core it is scheduled on by accessing the DSP registers.
So, the suggested fix here is to unregister the machine driver
first to remove the topology and then disable the DSP
to avoid the situation described above.
Note that the kernel panic only happens in cases where the
HDaudio link is not managed by the hdac library,
e.g. no codec or when HDMI is not supported.
When the hdac library is used, snd_sof_remove() calls
snd_hdac_ext_bus_device_remove() to remove the codec which
unregisters the component driver thereby also removing the
topology before the DSP is disabled.
Fixes: c16211d6226 ("ASoC: SOF: Add Sound Open Firmware driver core")
Reviewed-by: Takashi Iwai <[email protected]>
Signed-off-by: Ranjani Sridharan <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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commit 53e947a0e1f7 ("ASoC: soc-core: merge card resources cleanup
method") merged cleanup method of snd_soc_instantiate_card() and
soc_cleanup_card_resources().
But, after this commit, if user uses unbind/bind to Component factor
drivers, Kernel might indicates refcount error at
soc_cleanup_card_resources().
The 1st reason is card->snd_card is still exist even though
snd_card_free() was called, but it is already cleaned.
We need to set NULL to it.
2nd is card->dapm and card create debugfs, but its dentry is still
exist even though it was removed. We need to set NULL to it.
Fixes: 53e947a0e1f7 ("ASoC: soc-core: merge card resources cleanup method")
Cc: [email protected] # for v5.1
Signed-off-by: Kuninori Morimoto <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Removing link components results in topology unloading. So,
acquire the client_mutex before removing components in
soc_remove_link_components. This will prevent the lockdep warning
seen when dai links are removed during topology removal.
Signed-off-by: Ranjani Sridharan <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Revert commit 069d037aea98 ("ASoC: simple-card: Fix configuration of
DAI format"). During further review, it turns out that the actual issue
was caused by an incorrectly formatted device-tree node describing the
soundcard.
The following is incorrect because the simple-audio-card
'bitclock-master' and 'frame-master' properties should not reference the
actual codec phandle ...
sound {
compatible = "simple-audio-card";
...
=> simple-audio-card,bitclock-master = <&codec>;
=> simple-audio-card,frame-master = <&codec>;
...
simple-audio-card,cpu {
sound-dai = <&xxx>;
};
simple-audio-card,codec {
=> sound-dai = <&codec>;
};
};
Rather, these properties should reference the phandle to the
'simple-audio-card,codec' property as shown below ...
sound {
compatible = "simple-audio-card";
...
=> simple-audio-card,bitclock-master = <&codec>;
=> simple-audio-card,frame-master = <&codec>;
...
simple-audio-card,cpu {
sound-dai = <&xxx>;
};
=> codec: simple-audio-card,codec { /* simple-card wants here */
sound-dai = <&xxx>; /* not here */
};
};
Signed-off-by: Jon Hunter <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The use of BIT/GENMASK was incorrect, fix.
Signed-off-by: Sathya Prakash M R <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The RT5682 codec button mapping, initially copied from the DA7219 one,
needs to be corrected.
Signed-off-by: Sathya Prakash M R <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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When configuring a codec to be both bit-clock and frame-master, it was
found that the codec was always configured as bit-clock and frame-slave.
Looking at the simple_dai_link_of() function there appears to be two
problems with the configuration of the DAI format, which are ...
1. The function asoc_simple_parse_daifmt() is called before the function
asoc_simple_parse_codec() and this means that the device-tree node
for the codec has not been parsed yet, which is needed by the
function asoc_simple_parse_daifmt() to determine who is the codec.
2. The phandle passed to asoc_simple_parse_daifmt() is the phandle to
the 'codec' node and not the phandle of the actual codec defined by
the 'sound-dai' property under the 'codec' node.
Fix the above by moving the call to asoc_simple_parse_daifmt() after the
the call to asoc_simple_parse_codec() and pass the phandle for the codec
to asoc_simple_parse_daifmt().
Signed-off-by: Jon Hunter <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The selection order of m/c in match table is corrected
to use common codec as last in the list.
Signed-off-by: Sathya Prakash M R <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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There is a case when a we want to read a large number of bytes that
require a burst but is not a multiple of the word size (8). When this
happens rt5677_spi_reverse will run off the end of the buffer. The
solution is to tell spi_reverse the actual size of the destination and
stop if we reach it even if we have data left that we read.
Cc: Ben Zhang <[email protected]>
Signed-off-by: Curtis Malainey <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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snd_soc_dai_link_event() is updating snd_soc_dai :: active,
but it is unbalance.
It counts up if it has startup callback.
case SND_SOC_DAPM_PRE_PMU:
...
snd_soc_dapm_widget_for_each_source_path(w, path) {
...
if (source->driver->ops->startup) {
...
=> source->active++;
}
...
}
...
But, always counts down
case SND_SOC_DAPM_PRE_PMD:
...
snd_soc_dapm_widget_for_each_source_path(w, path) {
...
=> source->active--;
...
}
This patch always counts up when SND_SOC_DAPM_PRE_PMD.
Signed-off-by: Kuninori Morimoto <[email protected]>
Reviewed-by: Vinod Koul <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Add regcache_mark_dirty before regcache_sync for power
of codec may be lost at suspend, then all the register
need to be reconfigured.
Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver
support for CS42448/CS42888")
Cc: <[email protected]>
Signed-off-by: Shengjiu Wang <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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When the output sample rate is [8kHz, 30kHz], the limitation
of the supported ratio range is [1/24, 8]. In the driver
we use (8kHz, 30kHz) instead of [8kHz, 30kHz].
So this patch is to fix this issue and the potential rounding
issue with divider.
Fixes: fff6e03c7b65 ("ASoC: fsl_asrc: add support for 8-30kHz
output sample rate")
Cc: <[email protected]>
Signed-off-by: Shengjiu Wang <[email protected]>
Acked-by: Nicolin Chen <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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snd_soc_component_update_bits() may return 1 if operation
was successful and the value of the register changed.
Return a non-zero in ak4458_rstn_control for an error only.
Signed-off-by: Shengjiu Wang <[email protected]>
Signed-off-by: Viorel Suman <[email protected]>
Reviewed-by: Daniel Baluta <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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If playback/capture is paused and system enters S3, after system returns
from suspend, BE dai needs to call prepare() callback when playback/capture
is released from pause if RESUME_INFO flag is not set.
Currently, the dpcm_be_dai_prepare() function will block calling prepare()
if the pcm is in SND_SOC_DPCM_STATE_PAUSED state. This will cause the
following test case fail if the pcm uses BE:
playback -> pause -> S3 suspend -> S3 resume -> pause release
The playback may exit abnormally when pause is released because the BE dai
prepare() is not called.
This patch allows dpcm_be_dai_prepare() to call dai prepare() callback in
SND_SOC_DPCM_STATE_PAUSED state.
Signed-off-by: Libin Yang <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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AK4458 is probed successfully even if AK4458 is not present - this
is caused by probe function returning no error on i2c access failure.
Return an error on probe if i2c access has failed.
Signed-off-by: Shengjiu Wang <[email protected]>
Signed-off-by: Viorel Suman <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The cs4265_readable_register function stopped short of the maximum
register.
An example bug is taken from :
https://github.com/Audio-Injector/Ultra/issues/25
Where alsactl store fails with :
Cannot read control '2,0,0,C Data Buffer,0': Input/output error
This patch fixes the bug by setting the cs4265 to have readable
registers up to the maximum hardware register CS4265_MAX_REGISTER.
Signed-off-by: Matt Flax <[email protected]>
Reviewed-by: Charles Keepax <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Remove the erroneous addition of "SET_VALUE" to the GLB
IPC command string.
Signed-off-by: Ranjani Sridharan <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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A race condition exists in handling firmware boot timeout.
If FW sends FW_READY just after boot timeout has expired in
driver, a kernel exception will result as FW_READY handler
will be run while the state is still being cleaned up in
snd_sof_run_firmware(). Avoid the race by setting
boot_complete also in the error case.
Signed-off-by: Kai Vehmanen <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The nocodec option can be selected individually, leading to the following
issue:
sound/soc/sof/core.o: In function `snd_sof_device_probe':
core.c:(.text+0x4af): undefined reference to `sof_nocodec_setup'
Fix by selecting the SND_SOF_NOCODEC option as needed.
Reported-by: Hulk Robot <[email protected]>
Reported-by: YueHaibing <[email protected]>
Signed-off-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The recent fix for the build fix caused a couple of unused variable
compiler warnings when CONFIG_SND_SOC_SOF_NOCODEC isn't set:
sound/soc/sof/core.c:263:6: warning: unused variable ‘ret’ [-Wunused-variable]
sound/soc/sof/core.c:262:28: warning: unused variable ‘machine’ [-Wunused-variable]
Fix them by adding another ifdef.
Fixes: ce38a75089f7 ("ASoC: SOF: core: fix undefined nocodec reference")
Signed-off-by: Takashi Iwai <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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- Set period minimum size. Ensure at least 5ms period
up to 48kHz/16 bits to prevent underrun/overrun.
- Remove MDMA constraints on period maximum size and
set period maximum to half the buffer maximum size.
Signed-off-by: Olivier Moysan <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Some userspace apps, like pulseaudio, may call open, hw_params,
prepare to judge whether the pcm is ready or not. Current hdac_hdmi
will return -ENODEV if monitor is not connected, which will cause
the apps believe the pcm is not ready. Actually PCM for hdmi is ready,
even the monitor is not connected.
This patch removes the check of monitor presence in hw_params, just like
what the legacy HD-Audio driver does.
Signed-off-by: Libin Yang <[email protected]>
Acked-by: Pierre-Louis Bossart <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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mt6358_amic_disable() resets PGA to 0.
Save the gain settings from mixer control and restore them when using
the microphone.
Signed-off-by: Tzung-Bi Shih <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Output volume settings from mixer controls would be lost.
Imagine that "Headphone Volume" has set to -10dB via amixer:
- in mtk_hp_enable()
- hp_store_gain() saves the volume setting -10dB from regmap_read()
to ana_gain[AUDIO_ANALOG_VOLUME_HPOUTL]
- headset_volume_ramp() ramps up from -10dB to -10dB
- in mtk_hp_disable()
- headset_volume_ramp() ramps down from -10dB to -40dB
Next time in mtk_hp_enable(), hp_store_gain() would save -40dB but
not -10dB. As a result, headset_volume_ramp() would ramp from -10dB to
-40dB (which is mute).
Signed-off-by: Tzung-Bi Shih <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Mt6358 ramps up from the smallest volume (i.e. -10dB) to target dB when
opening and ramps down from target dB to mute (i.e. -40dB) when closing.
If target is equal to -10dB when opening, headset_volume_ramp() simply
leaves current setting (which may not be -10dB) unchanged.
Execute the loop at least once to initialize the setting to the
starting point (i.e. from).
Signed-off-by: Tzung-Bi Shih <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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