- 02 Dec, 2021 2 commits
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Lucas Tanure authored
Set SPI_MASTER as dependency as is using CS35L41 SPI driver Fixes: 96792fdd ("ASoC: amd: enable vangogh platform machine driver build") Signed-off-by:
Lucas Tanure <tanureal@opensource.cirrus.com> Reported-by:
kernel test robot <lkp@intel.com> Link: https://lore.kernel.org/r/20211201180004.1402156-1-tanureal@opensource.cirrus.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Oder Chiou authored
The patch fixes the wrong state of the JD with 1M pull up resistor in the HDA header. Signed-off-by:
Oder Chiou <oder_chiou@realtek.com> Link: https://lore.kernel.org/r/20211201095629.21818-1-oder_chiou@realtek.comSigned-off-by:
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- 01 Dec, 2021 3 commits
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Srinivas Kandagatla authored
return value form snd_soc_dapm_put_enum_double() directly instead of taking this in another redundant variable. Signed-off-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20211130160507.22180-3-srinivas.kandagatla@linaro.orgSigned-off-by:
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Sameer Pujar authored
The MVC module has a per channel control bit, based on which it decides to apply channel specific volume/mute settings. When per channel control bit is enabled (which is the default HW configuration), all MVC channel volume/mute can be independently controlled. If the control is disabled, channel-0 volume/mute setting is applied by HW to all remaining channels. Thus add support to leverage this HW feature by exposing master controls for volume/mute. With this, now there are per channel and master volume/mute controls. Users need to just use controls which are suitable for their applications. The per channel control enable/disable is mananged in driver and hidden from users, so that they need to just worry about respective volume/mute controls. Signed-off-by:
Sameer Pujar <spujar@nvidia.com> Link: https://lore.kernel.org/r/1638278605-28225-1-git-send-email-spujar@nvidia.comSigned-off-by:
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Mark Brown authored
Merge branch 'for-5.16' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-5.17 so we can apply new Tegra work
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- 30 Nov, 2021 9 commits
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Mac Chiang authored
support 2 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2/4 speakers. 2 or 4 speakers playback add echo reference capture add bt offload support add DMI_OEM_STRING for board variants add ALC5682I-VS support Signed-off-by:
Mark Hsieh <mark_hsieh@wistron.corp-partner.google.com> Signed-off-by:
Mac Chiang <mac.chiang@intel.com> Signed-off-by:
Kieth Tzeng <keith.tzeng@quantatw.com> Signed-off-by:
Brent Lu <brent.lu@intel.com> Acked-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20211125030453.4382-1-mac.chiang@intel.comSigned-off-by:
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Colin Ian King authored
Don't populate various arrays on the stack but instead make them static const. Also makes the object code smaller by a few hundred bytes. Signed-off-by:
Colin Ian King <colin.i.king@gmail.com> Link: https://lore.kernel.org/r/20211129224236.506883-1-colin.i.king@gmail.comSigned-off-by:
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Charles Keepax authored
Move the lockdep asserts until after the ctl pointer has been checked for NULL, to avoid potentially NULL pointer dereferences. Fixes: fb2f364fb5b9 ("firmware: cs_dsp: Add lockdep asserts to interface functions") Reported-by:Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20211130102842.26410-1-ckeepax@opensource.cirrus.comSigned-off-by:
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Rikard Falkeborn authored
These are only assigned to the ops fields in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by:
Rikard Falkeborn <rikard.falkeborn@gmail.com> Acked-by:
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Chris Down authored
On my T14s Gen2 I saw the following: [ 16.057258] skl_hda_dsp_generic skl_hda_dsp_generic: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 3 [ 16.057261] skl_hda_dsp_generic skl_hda_dsp_generic: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 4 [ 16.057263] skl_hda_dsp_generic skl_hda_dsp_generic: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 5 [...and so on.] It looks like the double newline is a mistake, so remove one. Signed-off-by:Chris Down <chris@chrisdown.name> Acked-by:
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Rikard Falkeborn authored
These are only assigned to the ops field in the snd_soc_dai_link which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by:
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Hui Wang authored
On the latest Lenovo Thinkstation laptops, we often experience the speaker failure after rebooting, check the dmesg, we could see: sof-audio-pci-intel-tgl 0000:00:1f.3: codec #0 probe error, ret: -5 The analogue codec on the machine is ALC287, then we designed a testcase to reboot and check the codec probing result repeatedly, we found the analogue codec probing always failed at least once within several minutes to several hours (roughly 1 reboot per min). This issue happens on all laptops of this Thinkstation model, but with legacy HDA driver, we couldn't reproduce this issue on those laptops. And so far, this issue is not reproduced on machines which don't belong to this model. We tried to make the hda_dsp_ctrl_init_chip() same as hda_intel_init_chip() which is the controller init routine in the legacy HDA driver, but it didn't help. We found when issue happens, the resp is -1, and if we let driver re-run send_cmd() and get_response(), it will get the correct response 10ec0287, then driver continues the rest work, finally boot to the desktop and all audio function work well. Here adding codec probing retries to 3 times, it could fix the issue on this Thinkstation model, and it doesn't bring impact to other machines. Reviewed-by:
Bard Liao <bard.liao@intel.com> Reviewed-by:
Hui Wang <hui.wang@canonical.com> Signed-off-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20211130090606.529348-1-kai.vehmanen@linux.intel.comSigned-off-by:
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Dan Carpenter authored
The "index" is potentially used without being initialized on the error path. Fixes: fc329c1d ("ASoC: amd: add platform devices for acp6x pdm driver and dmic driver") Signed-off-by:
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Nicolas Frattaroli authored
Previously, the DAI template was used directly, which lead to fun bugs such as "why is my channels_max changing?" when one instantiated more than one i2s_tdm IP block in a device tree. This change makes it so that we instead duplicate the template struct, and then use that. Fixes: 081068fd ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by:
Nicolas Frattaroli <frattaroli.nicolas@gmail.com> Link: https://lore.kernel.org/r/20211125084900.417102-1-frattaroli.nicolas@gmail.comSigned-off-by:
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- 29 Nov, 2021 9 commits
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Mark Brown authored
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>: Current soc_pcm_pointer() is checking runtime->delay, but it might be updated silently by component's .point callback. It is strange and difficult to find/know the issue. This patch adds .delay callback for component, and solve the issue.
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Jernej Skrabec authored
SPDIF core is capable of sending custom status. Implement IEC958 control handling. Signed-off-by:
Jernej Skrabec <jernej.skrabec@gmail.com> Link: https://lore.kernel.org/r/20211117194458.2249643-1-jernej.skrabec@gmail.comSigned-off-by:
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lvzhaoxiong authored
Qcom machine driver adds rt5682s support in this patch. Card name can be specified from dts by model property, and driver makes use of the name to distinguish which headset codec is on the board. Signed-off-by:
lvzhaoxiong <lvzhaoxiong@huaqin.corp-partner.google.com> Link: https://lore.kernel.org/r/20211123024329.21998-1-lvzhaoxiong@huaqin.corp-partner.google.comSigned-off-by:
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Samuel Holland authored
This allows changing the volume of each digital input/output independently, and provides the only "master volume" for the DAC. (The ADC also has a gain control on the analog side.) While the hardware supports digital gain up to +72dB, the controls here are limited to +24dB maximum, as any gain above that level makes volume sliders difficult to use, and is extremely likely to cause clipping. Signed-off-by:
Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20211118033645.43524-1-samuel@sholland.orgSigned-off-by:
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Kuninori Morimoto authored
No driver directly updates runtime->delay in .pointer. This patch cleanups its method. Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87zgq4wnkx.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Kuninori Morimoto authored
Now ALSA SoC supports .delay for component. This patch uses it, and not update runtime->delay on .pointer directly / secretly. Signed-off-by:
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Kuninori Morimoto authored
Now ALSA SoC supports .delay for component. This patch uses it, and not update runtime->delay on .pointer directly / secretly. Signed-off-by:
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Kuninori Morimoto authored
Current soc-pcm.c :: soc_pcm_pointer() is assuming that component driver might update runtime->delay silently in snd_soc_pcm_component_pointer() (= A). static snd_pcm_uframes_t soc_pcm_pointer(...) { ... /* clearing the previous total delay */ => runtime->delay = 0; (A) offset = snd_soc_pcm_component_pointer(substream); /* base delay if assigned in pointer callback */ => delay = runtime->delay; ... } 1) The behavior that ".pointer callback secretly updates runtime->delay" is strange and confusable. 2) Current snd_soc_pcm_component_pointer() uses 1st found component's .pointer callback only, thus it is no problem for now. But runtime->delay might be overwrote if it adjusted to multiple components in the future. 3) Component delay is updated at .pointer callback timing (secretly). But some components which doesn't have .pointer callback might want to increase runtime->delay for some reasons. We already have .delay function for DAI, but not have for Component. This patch adds new snd_soc_pcm_component_delay() for it. Signed-off-by: -
Kuninori Morimoto authored
Current soc_pcm_pointer() is manually calculating both CPU-DAI's max delay (= A) and Codec-DAI's max delay (= B). static snd_pcm_uframes_t soc_pcm_pointer(...) { ... ^ for_each_rtd_cpu_dais(rtd, i, cpu_dai) (A) cpu_delay = max(cpu_delay, ...); v delay += cpu_delay; ^ for_each_rtd_codec_dais(rtd, i, codec_dai) (B) codec_delay = max(codec_delay, ...); v delay += codec_delay; runtime->delay = delay; ... } Current soc_pcm_pointer() and the total delay calculating is not readable / difficult to understand. This patch update snd_soc_dai_delay() to snd_soc_pcm_dai_delay(), and calcule both CPU/Codec delay in one function. Link: https://lore.kernel.org/r/87fszl4yrq.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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- 27 Nov, 2021 2 commits
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Mark Brown authored
Merge series from Kai Vehmanen <kai.vehmanen@linux.intel.com>: Implement an updated programming sequence to handle DMA stop for Intel HD-Audio DMA. The new flow is only used if the firmware is sufficiently new to support the feature. SOF1.9.2 is the first release with the updated flow. The kernel changes are backwards compatible with old firmware releases. Likewise new firmware releases will work with old kernel. Series reviewed originally at: https://github.com/thesofproject/linux/pull/3167
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Mark Brown authored
Merge series from Sameer Pujar <spujar@nvidia.com>: This series addresses following problems: * The runtime PM is not balanced in MVC driver, whenever mute or volume mixer controls are set. * Some of the AHUB devices (SFC, MVC, Mixer, AMX and ADX) use late system sleep. Suspend failure is seen on Jetson TX2 platform.
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- 26 Nov, 2021 15 commits
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Geert Uytterhoeven authored
On 32-bit with CONFIG_ARCH_DMA_ADDR_T_64BIT=n: sound/soc/sof/mediatek/mt8195/mt8195.c: In function ‘platform_parse_resource’: sound/soc/sof/mediatek/mt8195/mt8195.c:51:15: error: format ‘%llx’ expects argument of type ‘long long unsigned int’, but argument 4 has type ‘unsigned int’ [-Werror=format=] 51 | dev_dbg(dev, "DMA pbase=0x%llx, size=0x%llx\n", | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/sof/mediatek/mt8195/mt8195.c: In function ‘adsp_memory_remap_init’: sound/soc/sof/mediatek/mt8195/mt8195.c:167:15: error: format ‘%llx’ expects argument of type ‘long long unsigned int’, but argument 4 has type ‘phys_addr_t’ {aka ‘unsigned int’} [-Werror=format=] 167 | dev_dbg(dev, "adsp->pa_dram %llx, offset %#x\n", adsp->pa_dram, offset); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/sof/mediatek/mt8195/mt8195.c: In function ‘adsp_shared_base_ioremap’: sound/soc/sof/mediatek/mt8195/mt8195.c:196:15: error: format ‘%llx’ expects argument of type ‘long long unsigned int’, but argument 5 has type ‘phys_addr_t’ {aka ‘unsigned int’} [-Werror=format=] 196 | dev_dbg(dev, "shared-dram vbase=%p, phy addr :%llx, size=%#x\n", | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Fix the first cases by printing the full resource using %pR. Fix the other cases by printing the physical addresses using %pa. Reported-by: noreply@ellerman.id.au Fixes: 32d7e03d ("ASoC: SOF: mediatek: Add mt8195 hardware support") Signed-off-by:Geert Uytterhoeven <geert@linux-m68k.org> Link: https://lore.kernel.org/r/20211123103013.73645-1-geert@linux-m68k.orgSigned-off-by:
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Lucas Tanure authored
Can't link I2C and SPI to the same binary, better to move CS35L41 to 3 modules approach. And instead of exposing cs35l41_reg, volatile_reg, readable_reg and precious_reg arrays, move cs35l41_regmap_i2c/spi to new module and expose it. Signed-off-by:
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Paul Cercueil authored
The capture and playback paths both have a configurable gain after their respective mixer, which can be set from -31 dB to 0 dB in 32 steps. Signed-off-by:
Paul Cercueil <paul@crapouillou.net> Link: https://lore.kernel.org/r/20211125232543.117074-1-paul@crapouillou.netSigned-off-by:
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Ye Guojin authored
This was found by coccicheck: ./sound/soc/fsl/imx-hdmi.c,209,1-7,ERROR missing put_device; call of_find_device_by_node on line 119, but without a corresponding object release within this function. Reported-by:
Zeal Robot <zealci@zte.com.cn> Signed-off-by:
Ye Guojin <ye.guojin@zte.com.cn> Link: https://lore.kernel.org/r/20211110002910.134915-1-ye.guojin@zte.com.cnSigned-off-by:
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Ranjani Sridharan authored
For HDA DAI's the DMA must be paused after the RUN bit is cleared by the host. So, send the DAI_CONFIG IPC with just the SOF_DAI_CONFIG_FLAGS_PAUSE flag set to indicate this to the firmware. Signed-off-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by:
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Ranjani Sridharan authored
Some DAI components, such as HDaudio, need to be stopped in two steps a) stop the DAI component b) stop the DAI DMA This patch enables this two-step stop by expanding the DAI_CONFIG IPC flags and split them into 2 parts. The 4 LSB bits indicate when the DAI_CONFIG IPC is sent, ex: hw_params, hw_free or pause. The 4 MSB bits are used as the quirk flags to be used along with the command flags. The quirk flag called SOF_DAI_CONFIG_FLAGS_2_STEP_STOP shall be set along with the HW_PARAMS command flag, i.e. before the pipeline is started so that the stop/pause trigger op in the FW can take the appropriate action to either perform/skip the DMA stop. If set, the DMA stop will be executed when the DAI_CONFIG IPC is sent during hw_free. In the case of pause, DMA pause will be handled when the DAI_CONFIG IPC is sent with the PAUSE command flag. Along with this, modify the signature for the hda_ctrl_dai_widget_setup/ hda_ctrl_dai_widget_free() functions to take additional flags as an argument and modify all users to pass the appropriate quirk flags. Only the HDA DAI's need to pass the SOF_DAI_CONFIG_FLAGS_2_STEP_STOP quirk flag during hw_params to indicate that it supports two-step stop and pause. Signed-off-by:
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Ranjani Sridharan authored
Even though the order of stopping the DMA and freeing the widget list is not important, align the sequence to match with the stop trigger to avoid confusion. Signed-off-by:
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Ranjani Sridharan authored
Move the check for the prepared flag inside snd_pcm_dsp_pcm_free() to avoid having to check it before every invocation of the function. Signed-off-by:
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Ranjani Sridharan authored
Add a helper function to free PCM in the FW, stop the DMA and free the widget list. These actions are performed both during PCM trigger STOP and when a paused stream is freed during system suspend. Signed-off-by:
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Ranjani Sridharan authored
Paused streams must be stopped and platform hw_free should be invoked during system suspend so they can be restarted properly after system resume. Signed-off-by:
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Ranjani Sridharan authored
snd_sof_pcm_platform_hw_params() will be called when the stream is restarted with a prepare ioctl. This happens in two cases i.e. when a suspended stream is resumed or when a stream is restarted without intermediate call to sof_pcm_hw_free(). Make sure to call snd_sof_pcm_platform_hw_free() in both these cases to keep it balanced. Signed-off-by:
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Ranjani Sridharan authored
The recommended programming sequence for HD-Audio DMA is to reset the stream before coupling the link and host DMA's. Signed-off-by:
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Ranjani Sridharan authored
Add a helper function to perform stream reset. Signed-off-by:
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Ranjani Sridharan authored
The DAI_CONFIG IPC that is sent during the STOP trigger is used for stopping the DMA in the FW. This must be done after the DMA RUN bit is cleared by the host. So move the call to snd_hdac_ext_link_stream_clear() before hda_link_dai_widget_update() to follow the correct programming sequence for DMA stop for HDA DAIs. Signed-off-by:
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Rob Clark authored
Move the declaration of temporary arrays to somewhere that won't go out of scope before the devm_clk_hw_register() call, lest we be at the whim of the compiler for whether those stack variables get overwritten. Fixes a crash seen with gcc version 11.2.1 20210728 (Red Hat 11.2.1-1) Fixes: bdd229ab ("ASoC: rt5682s: Add driver for ALC5682I-VS codec") Signed-off-by:
Rob Clark <robdclark@chromium.org> Reviewed-by:
Stephen Boyd <swboyd@chromium.org> Link: https://lore.kernel.org/r/20211118010453.843286-2-robdclark@gmail.comSigned-off-by:
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