- 03 Jun, 2019 1 commit
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Kovács Tamás authored
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 <kepszlok@zohomail.eu> Signed-off-by:
Mark Brown <broonie@kernel.org>
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- 29 May, 2019 1 commit
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Georgii Staroselskii authored
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 bf14da7e 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 <georgii.staroselskii@emlid.com> Reviewed-by:
Chen-Yu Tsai <wens@csie.org> Signed-off-by:
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- 28 May, 2019 9 commits
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Zhu Yingjiang authored
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: 8a300c8f ("ASoC: SOF: Intel: Add HDA controller for Intel DSP") Reviewed-by:
Takashi Iwai <tiwai@suse.de> Signed-off-by:
Zhu Yingjiang <yingjiang.zhu@linux.intel.com> Signed-off-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by:
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Guennadi Liakhovetski authored
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: 53e0c72d ("ASoC: SOF: Add support for IPC IO between DSP and Host") Reviewed-by:
Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Signed-off-by:
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Keyon Jie authored
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: c3078f53 ("ASoC: SOF: Add Sound Open Firmware KControl support") Reviewed-by:
Keyon Jie <yang.jie@linux.intel.com> Signed-off-by:
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Pierre-Louis Bossart authored
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: e2803e61 ("ASoC: SOF: PCM: add period_elapsed work to fix race condition in interrupt context") GitHub issue: https://github.com/thesofproject/linux/issues/932Reviewed-by:
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Libin Yang authored
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: 868bd00f ("ASoC: SOF: Add PCM operations support") Reviewed-by:
Libin Yang <libin.yang@intel.com> Signed-off-by:
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Pierre-Louis Bossart authored
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: c16211d6 ("ASoC: SOF: Add Sound Open Firmware driver core") Reviewed-by:
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Ranjani Sridharan authored
No need to call snd_soc_unregister_component in case of error because the component device is resource-managed. Fixes: c16211d6 ("ASoC: SOF: Add Sound Open Firmware driver core") Reviewed-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by:
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Ranjani Sridharan authored
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: c16211d6 ("ASoC: SOF: Add Sound Open Firmware driver core") Reviewed-by:
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Kuninori Morimoto authored
commit 53e947a0 ("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: 53e947a0 ("ASoC: soc-core: merge card resources cleanup method") Cc: stable@vger.kernel.org # for v5.1 Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by:
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- 24 May, 2019 2 commits
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Ranjani Sridharan authored
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:
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Jon Hunter authored
Revert commit 069d037a ("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 <jonathanh@nvidia.com> Signed-off-by:
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- 21 May, 2019 2 commits
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Sathya Prakash M R authored
The use of BIT/GENMASK was incorrect, fix. Signed-off-by:
Sathya Prakash M R <sathya.prakash.m.r@intel.com> Signed-off-by:
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Sathya Prakash M R authored
The RT5682 codec button mapping, initially copied from the DA7219 one, needs to be corrected. Signed-off-by:
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- 20 May, 2019 4 commits
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Jon Hunter authored
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:
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Sathya Prakash M R authored
The selection order of m/c in match table is corrected to use common codec as last in the list. Signed-off-by:
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Curtis Malainey authored
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 <benzh@chromium.org> Signed-off-by:
Curtis Malainey <cujomalainey@chromium.org> Signed-off-by:
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Kuninori Morimoto authored
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:Vinod Koul <vkoul@kernel.org> Signed-off-by:
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- 17 May, 2019 1 commit
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S.j. Wang authored
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: 0c516b4f ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Cc: <stable@vger.kernel.org> Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Signed-off-by:
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- 16 May, 2019 1 commit
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S.j. Wang authored
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: fff6e03c ("ASoC: fsl_asrc: add support for 8-30kHz output sample rate") Cc: <stable@vger.kernel.org> Signed-off-by:
Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by:
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- 13 May, 2019 7 commits
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Viorel Suman authored
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:
Viorel Suman <viorel.suman@nxp.com> Reviewed-by:
Daniel Baluta <daniel.baluta@nxp.com> Signed-off-by:
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Libin Yang authored
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:
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Viorel Suman authored
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:
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Matt Flax authored
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 <flatmax@flatmax.org> Reviewed-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by:
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Ranjani Sridharan authored
Remove the erroneous addition of "SET_VALUE" to the GLB IPC command string. Signed-off-by:
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Kai Vehmanen authored
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 <kai.vehmanen@linux.intel.com> Signed-off-by:
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Pierre-Louis Bossart authored
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 <hulkci@huawei.com> Reported-by:
YueHaibing <yuehaibing@huawei.com> Signed-off-by:
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- 09 May, 2019 1 commit
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Takashi Iwai authored
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: ce38a750 ("ASoC: SOF: core: fix undefined nocodec reference") Signed-off-by:
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- 08 May, 2019 8 commits
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Olivier Moysan authored
- 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 <olivier.moysan@st.com> Signed-off-by:
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Libin Yang authored
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:
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Tzung-Bi Shih authored
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 <tzungbi@google.com> Signed-off-by:
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Tzung-Bi Shih authored
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 authored
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:
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Pierre-Louis Bossart authored
The existing code mistakenly uses IS_ENABLED in C code instead of as in conditional compilation, leading to the following error: ld: sound/soc/sof/core.o: in function `sof_machine_check': sound/soc/sof/core.c:279: undefined reference to `sof_nocodec_setup' Fix by using #if !IS_ENABLED() Reported-by:
kbuild test robot <lkp@intel.com> Signed-off-by:
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Pierre-Louis Bossart authored
The SND_SOC_INTEL_COMMON Kconfig was removed months ago from SOF Kconfig files but is still selected instead of the correct SND_SOC_SOF_INTEL_COMMON kconfig which does select xtensa stuff, leading to the following errors. ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x120): undefined reference to `sof_xtensa_arch_ops' ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x180): undefined reference to `sof_xtensa_arch_ops' ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x1e0): undefined reference to `sof_xtensa_arch_ops' Reported-by:
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Takashi Iwai authored
This fixes a compile warning below by properly handling the error code from sof_get_ctrl_copy_params(): include/linux/kernel.h:843:43: warning: 'sparams.pl_size' may be used uninitialized in this function [-Wmaybe-uninitialized] sound/soc/sof/ipc.c:639:34: note: 'sparams.pl_size' was declared here The function returns an error before setting sparams.pl_size, so it'd assign an uninitialized value at a later point. Fixes: 53e0c72d ("ASoC: SOF: Add support for IPC IO between DSP and Host") Signed-off-by:
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- 07 May, 2019 3 commits
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Takashi Iwai authored
Paper over a trivial case leading to an uninitialized variable compile warning: sound/soc/rockchip/rockchip_pdm.c:179:3: warning: ‘clk_out’ may be used uninitialized in this function [-Wmaybe-uninitialized] Fixes: 624e8e00 ("ASoC: rockchip: pdm: fixup pdm fractional div") Signed-off-by:
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Takashi Iwai authored
A trivial fix for the randconfig build error: sound/soc/sof/ops.c:20:6: warning: ‘ret’ is used uninitialized in this function [-Wuninitialized] Fixes: d1d95fcb ("ASoC: SOF: Add DSP HW abstraction operations") Signed-off-by:
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Takashi Iwai authored
A trivial fix for the randconfig build error: sound/soc/codecs/da7219.c:2366:6: warning: unused variable ‘i’ [-Wunused-variable] Fixes: d90ba6c8 ("ASoC: da7219: Expose BCLK and WCLK control through CCF") Signed-off-by:
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