- 19 Aug, 2020 1 commit
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Mateusz Gorski authored
Different modules for HDMI codec are used depending on the "hda_codec_use_common_hdmi" option being enabled or not. Driver private context for both of them is different. This leads to null-pointer dereference error when driver tries to set autosuspend delay for HDMI codec while the option is off (hdac_hdmi module is used for HDMI). Change the string in conditional statement to "ehdaudio0D0" to ensure that only the HDAudio codec is handled by this function. Fixes: 5bf73b1b ("ASoC: intel/skl/hda - fix oops on systems without i915 audio codec") Signed-off-by:
Mateusz Gorski <mateusz.gorski@linux.intel.com> Reviewed-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200722173524.30161-1-mateusz.gorski@linux.intel.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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- 17 Aug, 2020 4 commits
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Dinghao Liu authored
When power_up_sst() fails, stream needs to be freed just like when try_module_get() fails. However, current code is returning directly and ends up leaking memory. Fixes: 0121327c ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp") Signed-off-by:
Dinghao Liu <dinghao.liu@zju.edu.cn> Acked-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200813084112.26205-1-dinghao.liu@zju.edu.cnSigned-off-by:
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Sylwester Nawrocki authored
The driver supports WM1811, WM8994, WM8958 devices but according to documentation and the regmap definitions the WM8958_DSP2_* registers are only available on WM8958. In current code these registers are being accessed as if they were available on all the three chips. When starting playback on WM1811 CODEC multiple errors like: "wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5" can be seen, which is caused by attempts to read an unavailable WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent commit "e2329eeb ASoC: soc-component: add soc_component_err()". This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling is only done for WM8958. Signed-off-by:
Sylwester Nawrocki <s.nawrocki@samsung.com> Acked-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.comSigned-off-by:
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Srinivas Kandagatla authored
For some reason interrupt set and clear register offsets are not set correctly. This patch corrects them! Fixes: 585e881e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Tested-by:
Stephan Gerhold <stephan@gerhold.net> Reviewed-by:
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Sylwester Nawrocki authored
The ADC2 and DAC2 are not available on WM1811 device. This patch moves the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing them and avoid unreadable register access on WM1811. This allows to get rid of warnings during boot like: wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5 Signed-off-by:
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- 11 Aug, 2020 3 commits
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Srinivas Kandagatla authored
Most of the DAPM widgets for DSP ASoC components reuse reg field of the widgets for its internal calculations, however these are not real registers. So read/writes to these numbers are not really valid. However ASoC core will read these registers to get default state during startup. With recent changes to ASoC core, every register read/write failures are reported very verbosely. Prior to this fails to reads are totally ignored, so we never saw any error messages. To fix this add dummy read/write function to return default value. Fixes: e3a33673 ("ASoC: qdsp6: q6routing: Add q6routing driver") Reported-by:
John Stultz <john.stultz@linaro.org> Signed-off-by:
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Srinivas Kandagatla authored
Looks like the q6afe-dai dapm widget registers are set as "0", which is a not correct. As this registers will be read by ASoC core during startup which will throw up errors, Fix this by making the registers as SND_SOC_NOPM as these should be never used. With recent changes to ASoC core, every register read/write failures are reported very verbosely. Prior to this fails to reads are totally ignored, so we never saw any error messages. Fixes: 24c4cbcf ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Reported-by:
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Takashi Iwai authored
Along with the recent unification of snd_soc_component_read*() functions, the behavior of snd_soc_component_read() was changed slightly; namely it returns the register read value directly, and even if an error happens, it returns zero (but it prints an error message). That said, the caller side can't know whether it's an error or not any longer. Ideally this shouldn't matter much, but in practice this seems causing a regression, as John reported. And, grepping the tree revealed that there are still plenty of callers that do check the error code, so we'll need to deal with them in anyway. As a quick band-aid over the regression, this patch changes the return value of snd_soc_component_read() again to the negative error code. It can't work, obviously, for 32bit register values, but it should be enough for the known regressions, so far. Fixes: cf6e26c7 ("ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32()") Reported-by:
Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200810134631.19742-1-tiwai@suse.deSigned-off-by:
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- 10 Aug, 2020 1 commit
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Ravulapati Vishnu vardhan rao authored
Replacing string compare with "codec_dai->name" instead of comparing with "codec_dai->component->name" in hw_params because, Here the component name for codec RT1015 is "i2c-10EC5682:00" and will never be "rt1015-aif1" as it is codec-dai->name. So, strcmp() always compares and fails to set the sysclk,pll,bratio for expected codec-dai="rt1015-aif1". Signed-off-by:
Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com> Link: https://lore.kernel.org/r/20200807161046.17932-1-Vishnuvardhanrao.Ravulapati@amd.comSigned-off-by:
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- 03 Aug, 2020 10 commits
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Mark Brown authored
Hi, this is a trivial patch set to add the missing __maybe_unused for covering the compile warnings with CONFIG_PM=n. Takashi === Takashi Iwai (5): ASoC: tegra: tegra186_dspk: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_admaif: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n sound/soc/tegra/tegra186_dspk.c | 4 ++-- sound/soc/tegra/tegra210_admaif.c | 4 ++-- sound/soc/tegra/tegra210_ahub.c | 4 ++-- sound/soc/tegra/tegra210_dmic.c | 4 ++-- sound/soc/tegra/tegra210_i2s.c | 4 ++-- 5 files changed, 10 insertions(+), 10 deletions(-) -- 2.16.4
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Takashi Iwai authored
The variable rtd was left unused in psc_dma_free(), even unnoticed during conversion to a new style: sound/soc/fsl/mpc5200_dma.c:342:30: warning: unused variable 'rtd' [-Wunused-variable] Drop the superfluous one. Fixes: 6d1048bc ("ASoC: fsl: mpc5200_dma: remove snd_pcm_ops") Signed-off-by:
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Takashi Iwai authored
Fix trivial compile warnings wrt unused functions by adding __maybe_unused prefix: sound/soc/tegra/tegra210_i2s.c:167:12: warning: 'tegra210_i2s_runtime_suspend' defined but not used [-Wunused-function] sound/soc/tegra/tegra210_i2s.c:179:12: warning: 'tegra210_i2s_runtime_resume' defined but not used [-Wunused-function] Fixes: c0bfa983 ("ASoC: tegra: Add Tegra210 based I2S driver") Signed-off-by:
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Takashi Iwai authored
Fix trivial compile warnings wrt unused functions by adding __maybe_unused prefix: sound/soc/tegra/tegra210_dmic.c:43:12: warning: 'tegra210_dmic_runtime_suspend' defined but not used [-Wunused-function] sound/soc/tegra/tegra210_dmic.c:55:12: warning: 'tegra210_dmic_runtime_resume' defined but not used [-Wunused-function] Fixes: 8c8ff982 ("ASoC: tegra: Add Tegra210 based DMIC driver") Signed-off-by:
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Takashi Iwai authored
Fix trivial compile warnings wrt unused functions by adding __maybe_unused prefix: sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function] sound/soc/tegra/tegra210_ahub.c:579:12: warning: 'tegra_ahub_runtime_resume' defined but not used [-Wunused-function] Fixes: 16e1bcc2 ("ASoC: tegra: Add Tegra210 based AHUB driver") Signed-off-by:
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Takashi Iwai authored
Fix trivial compile warnings wrt unused functions by adding __maybe_unused prefix: sound/soc/tegra/tegra210_admaif.c:232:12: warning: 'tegra_admaif_runtime_resume' defined but not used [-Wunused-function] sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function] Fixes: f74028e1 ("ASoC: tegra: Add Tegra210 based ADMAIF driver") Signed-off-by:
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Takashi Iwai authored
Fix trivial compile warnings wrt unused functions by adding __maybe_unused prefix: sound/soc/tegra/tegra186_dspk.c:74:12: warning: 'tegra186_dspk_runtime_suspend' defined but not used [-Wunused-function] sound/soc/tegra/tegra186_dspk.c:86:12: warning: 'tegra186_dspk_runtime_resume' defined but not used [-Wunused-function] Fixes: 327ef647 ("ASoC: tegra: Add Tegra186 based DSPK driver") Signed-off-by:
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Hui Wang authored
Recently we found an issue about the suspend and resume. If dmic is recording the sound, and we run suspend and resume, after the resume, the dmic can't work well anymore. we need to close the app and reopen the app, then the dmic could record the sound again. For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2 -r 48000 test.wav", then suspend and resume, after the system resume back, we speak to the dmic. then stop the arecord, use aplay to play the test.wav, we could hear the sound recorded after resume is weird, it is not what we speak to the dmic. I found two registers are set in the dai_hw_params(), if the two registers are set during the resume, this issue could be fixed. Move the code of the dai_hw_params() into the pdm_dai_trigger(), then these two registers will be set during resume since pdm_dai_trigger() will be called during resume. And delete the empty function dai_hw_params(). Signed-off-by:
Hui Wang <hui.wang@canonical.com> Reviewed-by:
Vijendar Mukunda <Vijendar.Mukunda@amd.com> Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20200730123138.5659-1-hui.wang@canonical.comSigned-off-by:
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Fabio Estevam authored
Removing ADDITIONAL_CONTROL_4 from the list of readable registers cause audio distortion. This change was sent as a comment below the --- line when submitting commit 658bb297 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE"), so it was not supposed to get merged. Keep WM8962_ADDITIONAL_CONTROL_4 inside wm8962_readable_register() to fix the regression. Fixes: 658bb297 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE") Reported-by:
Shengjiu Wang <shengjiu.wang@gmail.com> Signed-off-by:
Fabio Estevam <festevam@gmail.com> Link: https://lore.kernel.org/r/20200803115233.19034-1-festevam@gmail.comSigned-off-by:
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Shengjiu Wang authored
With this case: aplay -Dhw:x 16khz.wav 24khz.wav There is sound distortion for 24khz.wav. The reason is that setting PLL of WM8962 with set_bias_level function, the bias level is not changed when 24khz.wav is played, then the PLL won't be reset, the clock is not correct, so distortion happens. The resolution of this issue is to remove fsl_asoc_card_set_bias_level. Move PLL configuration to hw_params and hw_free. After removing fsl_asoc_card_set_bias_level, also test WM8960 case, it can work. Fixes: 708b4351 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by:
Nicolin Chen <nicoleotsuka@gmail.com> Link: https://lore.kernel.org/r/1596420811-16690-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by:
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- 31 Jul, 2020 5 commits
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Mark Brown authored
Merge series "ASoC: core: Two step component registration" from Cezary Rojewski <cezary.rojewski@intel.com>: Provide a mechanism for true two-step component registration. This mimics device registration flow where initialization is the first step while addition goes as second in line. Drivers may choose to modify component's fields before registering component to ASoC subsystem via snd_soc_add_component. Patchset achieves status quo - behavior of snd_soc_register_component remains unchanged. Cezary Rojewski (3): ASoC: core: Relocate and expose snd_soc_component_initialize ASoC: core: Simplify snd_soc_component_initialize declaration ASoC: core: Two step component registration include/sound/soc-component.h | 3 -- include/sound/soc.h | 11 +++--- sound/soc/soc-component.c | 16 --------- sound/soc/soc-core.c | 52 +++++++++++++++++---------- sound/soc/soc-generic-dmaengine-pcm.c | 14 +++++--- sound/soc/stm/stm32_adfsdm.c | 9 +++-- 6 files changed, 55 insertions(+), 50 deletions(-) -- 2.17.1
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Cezary Rojewski authored
Modify snd_soc_add_component so it calls snd_soc_component_initialize no longer and thus providing true two-step registration. Drivers may choose to change component's fields before actually adding it to ASoC subsystem. Signed-off-by:
Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20200731144146.6678-4-cezary.rojewski@intel.comSigned-off-by:
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Cezary Rojewski authored
Move 'name' field initialization responsibility back to snd_soc_component_initialize to prepare snd_soc_add_component function for being called separatelly as a second registration step. Signed-off-by:
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Cezary Rojewski authored
To allow for two-step component registration, expose snd_soc_component_initialize function and move it back to soc-core.c. Signed-off-by:
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Laurent Pinchart authored
Enabling a whole subsystem from a single driver 'select' is frowned upon and won't be accepted in new drivers, that need to use 'depends on' instead. Existing selection of DMADEVICES will then cause circular dependencies. Replace them with a dependency. Signed-off-by:
Laurent Pinchart <laurent.pinchart@ideasonboard.com> Acked-by:
Randy Dunlap <rdunlap@infradead.org> Link: https://lore.kernel.org/r/20200731152433.1297-3-laurent.pinchart@ideasonboard.comSigned-off-by:
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- 30 Jul, 2020 16 commits
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Mark Brown authored
The various list iterators are able to handle an empty list. The only effect of avoiding the loop is not initializing some index variables. Drop list_empty tests in cases where these variables are not used. The semantic patch that makes these changes is as follows: (http://coccinelle.lip6.fr/) <smpl> @@ expression x,e; iterator name list_for_each_entry; statement S; identifier i; @@ -if (!(list_empty(x))) { list_for_each_entry(i,x,...) S - } ... when != i ? i = e @@ expression x,e; iterator name list_for_each_entry_safe; statement S; identifier i,j; @@ -if (!(list_empty(x))) { list_for_each_entry_safe(i,j,x,...) S - } ... when != i when != j ( i = e; | ? j = e; ) @@ expression x,e; iterator name list_for_each; statement S; identifier i; @@ -if (!(list_empty(x))) { list_for_each(i,x) S - } ... when != i ? i = e @@ expression x,e; iterator name list_for_each_safe; statement S; identifier i,j; @@ -if (!(list_empty(x))) { list_for_each_safe(i,j,x) S - } ... when != i when != j ( i = e; | ? j = e; ) // ------------------- @@ expression x,e; statement S; identifier i; @@ -if (!(list_empty(x))) list_for_each_entry(i,x,...) S ... when != i ? i = e @@ expression x,e; statement S; identifier i,j; @@ -if (!(list_empty(x))) list_for_each_entry_safe(i,j,x,...) S ... when != i when != j ( i = e; | ? j = e; ) @@ expression x,e; statement S; identifier i; @@ -if (!(list_empty(x))) list_for_each(i,x) S ... when != i ? i = e @@ expression x,e; statement S; identifier i,j; @@ -if (!(list_empty(x))) list_for_each_safe(i,j,x) S ... when != i when != j ( i = e; | ? j = e; ) </smpl> --- drivers/media/pci/saa7134/saa7134-core.c | 14 ++--- drivers/media/usb/cx231xx/cx231xx-core.c | 16 ++---- drivers/media/usb/tm6000/tm6000-core.c | 24 +++------- drivers/net/ethernet/mellanox/mlx5/core/steering/dr_matcher.c | 13 ++--- drivers/net/ethernet/mellanox/mlx5/core/steering/dr_rule.c | 5 -- drivers/net/ethernet/sfc/ptp.c | 20 +++----- drivers/net/wireless/ath/dfs_pattern_detector.c | 15 ++---- sound/soc/intel/atom/sst/sst_loader.c | 10 +--- sound/soc/intel/skylake/skl-pcm.c | 8 +-- sound/soc/intel/skylake/skl-topology.c | 5 -- 10 files changed, 53 insertions(+), 77 deletions(-)
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Alper Nebi Yasak authored
PulseAudio (and perhaps other userspace utilities) can not detect any jack for rk3399_gru_sound as the driver doesn't expose related Jack kcontrols. This patch adds two DAPM pins to the headset jack, where the snd_soc_card_jack_new() call automatically creates "Headphones Jack" and "Headset Mic Jack" kcontrols from them. With an appropriate ALSA UCM config specifying JackControl fields for the "Headphones" and "Headset" (mic) devices, PulseAudio can detect plug/unplug events for both of them after this patch. Signed-off-by:
Alper Nebi Yasak <alpernebiyasak@gmail.com> Link: https://lore.kernel.org/r/20200721182709.6895-1-alpernebiyasak@gmail.comSigned-off-by:
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Fabio Estevam authored
According to the WM8962 datasheet, there is no register at address 0x200. WM8962_GPIO_BASE is just a base address for the GPIO registers and not a real register, so remove it from wm8962_readable_register(). Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip its access. This fixes the following errors: wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 Signed-off-by:
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Julia Lawall authored
Use resource_size rather than a verbose computation on the end and start fields. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) <smpl> @@ struct resource ptr; @@ - (ptr.end - ptr.start + 1) + resource_size(&ptr) </smpl> Signed-off-by:
Julia Lawall <Julia.Lawall@inria.fr> Link: https://lore.kernel.org/r/1595751933-4952-1-git-send-email-Julia.Lawall@inria.frSigned-off-by:
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Julia Lawall authored
list_for_each_entry_safe is able to handle an empty list. The only effect of avoiding the loop is not initializing the index variable. Drop list_empty tests in cases where these variables are not used. Note that list_for_each_entry_safe is defined in terms of list_first_entry, which indicates that it should not be used on an empty list. But in list_for_each_entry_safe, the element obtained by list_first_entry is not really accessed, only the address of its list_head field is compared to the address of the list head, so the list_first_entry is safe. The semantic patch that makes this change is as follows (with another variant for the no brace case): (http://coccinelle.lip6.fr/) <smpl> @@ expression x,e; iterator name list_for_each_entry_safe; statement S; identifier i,j; @@ -if (!(list_empty(x))) { list_for_each_entry_safe(i,j,x,...) S - } ... when != i when != j ( i = e; | ? j = e; ) </smpl> Signed-off-by:
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Mark Brown authored
Merge series "ASoC: Intel: KMB: TDM Enablement patches" from Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>: This patch series is to enable multiple features on the Keembay Platform Michael Sit Wei Hong (4): ASoC: Intel: KMB: Add 8kHz audio support ASoC: Intel: KMB: Rework disable channel function ASoC: Intel: KMB: Enable TDM audio capture dt-bindings: sound: intel,keembay-i2s: Add channel-max property .../bindings/sound/intel,keembay-i2s.yaml | 8 + sound/soc/intel/keembay/kmb_platform.c | 137 +++++++++++++----- sound/soc/intel/keembay/kmb_platform.h | 1 + 3 files changed, 112 insertions(+), 34 deletions(-) -- 2.17.1
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Mark Brown authored
This patcheset is collection of fixes for the TDM input and output the axg audio architecture. Its fixes: - slave mode format setting - g12 and sm1 skew offset - tdm clock inversion - standard daifmt props names which don't require a specific prefix Jerome Brunet (4): ASoC: meson: axg-tdm-interface: fix link fmt setup ASoC: meson: axg-tdmin: fix g12a skew ASoC: meson: axg-tdm-formatters: fix sclk inversion ASoC: meson: cards: remove DT_PREFIX for standard daifmt properties sound/soc/meson/axg-tdm-formatter.c | 11 ++++++----- sound/soc/meson/axg-tdm-formatter.h | 1 - sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++--------- sound/soc/meson/axg-tdmin.c | 16 +++++++++++++++- sound/soc/meson/axg-tdmout.c | 3 --- sound/soc/meson/meson-card-utils.c | 2 +- 6 files changed, 39 insertions(+), 20 deletions(-) -- 2.25.4
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Kuninori Morimoto authored
This patch switches from .txt base to .yaml base Document. Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by:
Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/87mu4cxlo2.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87o8pf3923.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/873659bpbk.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Ravulapati Vishnu vardhan rao authored
Adding rt1015 hw_params which set Bit-clock ratio, PLL and appropriate sys clk specific with RTK1015. Signed-off-by:
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Ravulapati Vishnu vardhan rao authored
DAI link support for RTK 1015 and providing the codec details depending on the snd_soc_card selected by ACPI ID. Signed-off-by:
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Ravulapati Vishnu vardhan rao authored
Adding support for ALC1015 RTK codec in machine driver. Passing specific card structure based on its ACPI ID. Signed-off-by:
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Ravulapati Vishnu vardhan rao authored
Passing specific snd_soc_card structure depending on the ACPI ID. In future we can add other IDs in the ACPI table and pass the structure. Signed-off-by:
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Ravulapati Vishnu vardhan rao authored
As in future our machine driver supports multiple codecs So changing naming convention of snd_soc_card struct and its fields. Signed-off-by:
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Simon Shields authored
This patch adds support for voice and BT calls, along with standard audio output via the speaker, earpiece, headphone jack, HDMI, and any accessories compatible with Midas boards. This patch also supports headphone/headset detection and headsets with inline buttons. [m.szyprowski: adaptation to v5.1+ kernels (DAI links initialization)] [s.nawrocki: removal of the clk API calls for CODEC MCLK, the jack data structure moved to struct midas_priv, coding style and typo fixes, conversion to new cpu/codec/dai-node binding] Signed-off-by:
Simon Shields <simon@lineageos.org> Signed-off-by:
Marek Szyprowski <m.szyprowski@samsung.com> Signed-off-by:
Krzysztof Kozlowski <krzk@kernel.org> Link: https://lore.kernel.org/r/20200728131111.14334-2-s.nawrocki@samsung.comSigned-off-by:
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Sylwester Nawrocki authored
This patch adds documentation of DT biding for the Midas sound complex. Partially based on the *txt version by Simon Shields <simon@lineageos.org>. Signed-off-by:
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Dan Murphy authored
Reset the device before programming the registers or all programming will be lost as the device resets registers to default settings. Signed-off-by:
Dan Murphy <dmurphy@ti.com> Link: https://lore.kernel.org/r/20200730142419.28205-2-dmurphy@ti.comSigned-off-by:
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