December 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v10 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 02 Dec '21

02 Dec '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v9: - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops changes since v8: - remove struct adsp_mbox_ch_info - add get_mtk_adsp_mbox_priv - use mtk_adsp_mbox_priv va_mboxreg address in adsp mbox driver - add struct mtk_adsp_mbox_cfg for DSP mbox register offset - remove adsp mbox register offset hard code define - remove mtk-adsp-ipc.h reference in adsp mbox driver changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 135 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 176 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 10 files changed, 450 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

1 3

[PATCH v6 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 02 Dec '21

02 Dec '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 ++++++++++ sound/soc/qcom/lpass-cpu.c | 244 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 +++- sound/soc/qcom/lpass-platform.c | 578 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 434 ++++++++++++++++ sound/soc/qcom/lpass.h | 151 ++++++ 9 files changed, 1784 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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alsautil: init procedure is executed during a store and restore process when the final snd_ctl_elem is not readable
by 易先生 02 Dec '21

02 Dec '21

hi, team When i execute the following commands sequencitally, alsa store -f 1.txt alsa restore -f 1.txt my playing volume was initialized. After a debugging process, I found the problem: During the "store" process, a control element is saved to the configuration file only when this element is readable.While during the "restore" process, after all the control element from the configuration file is processed, an "additional control element check" which causes a "init procedure" if the result is true is processed. Thus, if total control element count is 5 with the 5th element is not readable, then there will be only 4 control element stored in the configuration file, ranges from "control.1" to "control.4". Then the "additional control checking" process produces "true" and the variable "err" is evaluated as "-EAGAIN".

1 0

[PATCH] ALSA: usb-audio: Drop CONFIG_PM
by Takashi Iwai 02 Dec '21

02 Dec '21

Practically seen, CONFIG_PM is almost mandatory. Let's drop the ugly ifdef lines and simplify the code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 7 ------- sound/usb/mixer.c | 4 ---- sound/usb/mixer.h | 2 -- sound/usb/mixer_quirks.c | 2 -- sound/usb/mixer_quirks.h | 2 -- sound/usb/power.h | 10 ---------- 6 files changed, 27 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 1764b9302d46..376962291c4d 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -987,8 +987,6 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) wake_up(&chip->shutdown_wait); } -#ifdef CONFIG_PM - int snd_usb_autoresume(struct snd_usb_audio *chip) { int i, err; @@ -1100,11 +1098,6 @@ static int usb_audio_resume(struct usb_interface *intf) atomic_dec(&chip->active); /* allow autopm after this point */ return err; } -#else -#define usb_audio_suspend NULL -#define usb_audio_resume NULL -#define usb_audio_resume NULL -#endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { #include "quirks-table.h" diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 5b9fd07ce2a2..e8f3f8d622ec 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3629,7 +3629,6 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) mixer->disconnected = true; } -#ifdef CONFIG_PM /* stop any bus activity of a mixer */ static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) { @@ -3711,7 +3710,6 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) return snd_usb_mixer_activate(mixer); } -#endif void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, struct usb_mixer_interface *mixer, @@ -3720,7 +3718,5 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->mixer = mixer; list->id = unitid; list->dump = snd_usb_mixer_dump_cval; -#ifdef CONFIG_PM list->resume = restore_mixer_value; -#endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 98ea24d91d80..d43895c1ae5c 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -118,10 +118,8 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv); -#ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); -#endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index db194ad168d0..1f9863725c7c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3280,7 +3280,6 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) return err; } -#ifdef CONFIG_PM void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer) { switch (mixer->chip->usb_id) { @@ -3289,7 +3288,6 @@ void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer) break; } } -#endif void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid) diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index 52be26db558f..4ba01ba3fe8b 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -14,9 +14,7 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *cval, int unitid, struct snd_kcontrol *kctl); -#ifdef CONFIG_PM void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer); -#endif #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/sound/usb/power.h b/sound/usb/power.h index 6004231a7c75..396e3e51440a 100644 --- a/sound/usb/power.h +++ b/sound/usb/power.h @@ -21,17 +21,7 @@ struct snd_usb_power_domain * snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, unsigned char id); -#ifdef CONFIG_PM int snd_usb_autoresume(struct snd_usb_audio *chip); void snd_usb_autosuspend(struct snd_usb_audio *chip); -#else -static inline int snd_usb_autoresume(struct snd_usb_audio *chip) -{ - return 0; -} -static inline void snd_usb_autosuspend(struct snd_usb_audio *chip) -{ -} -#endif #endif /* __USBAUDIO_POWER_H */ -- 2.31.1

1 0

[PATCH] ALSA: usb-audio: Drop superfluous '0' in Presonus Studio 1810c's ID
by Takashi Iwai 02 Dec '21

02 Dec '21

The vendor ID of Presonus Studio 1810c had a superfluous '0' in its USB ID. Drop it. Fixes: 8dc5efe3d17c ("ALSA: usb-audio: Add support for Presonus Studio 1810c") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/format.c | 2 +- sound/usb/mixer_quirks.c | 2 +- sound/usb/quirks.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index f5e676a51b30..405dc0bf6678 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -375,7 +375,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, for (rate = min; rate <= max; rate += res) { /* Filter out invalid rates on Presonus Studio 1810c */ - if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + if (chip->usb_id == USB_ID(0x194f, 0x010c) && !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d489c1de3bae..db194ad168d0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3254,7 +3254,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_rme_controls_create(mixer); break; - case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + case USB_ID(0x194f, 0x010c): /* Presonus Studio 1810c */ err = snd_sc1810_init_mixer(mixer); break; case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 64e1c20311ed..ab9f3da49941 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1290,7 +1290,7 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); /* presonus studio 1810c: skip altsets incompatible with device_setup */ - if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + if (chip->usb_id == USB_ID(0x194f, 0x010c)) return s1810c_skip_setting_quirk(chip, iface, altno); -- 2.31.1

1 0

[PATCH] ALSA: hda/hdmi: Consider ELD is invalid when no SAD is present
by Kai-Heng Feng 02 Dec '21

02 Dec '21

There's a system that reports a bogus HDMI audio interface: $ cat eld#2.0 monitor_present 1 eld_valid 1 monitor_name connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xe430 product_id 0x690 port_id 0x0 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers [0xffff] FL/FR LFE FC RL/RR RC FLC/FRC RLC/RRC FLW/FRW FLH/FRH TC FCH sad_count 0 Since playing audio is not possible without SAD, also consider ELD is invalid for this case. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 415701bd10ac8..e7c2f3167f311 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1535,7 +1535,7 @@ static void update_eld(struct hda_codec *codec, } } - if (!eld->eld_valid || eld->eld_size <= 0) { + if (!eld->eld_valid || eld->eld_size <= 0 || eld->info.sad_count <= 0) { eld->eld_valid = false; eld->eld_size = 0; } -- 2.32.0

2 1

[PATCH v9 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 02 Dec '21

02 Dec '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v8: - remove struct adsp_mbox_ch_info - add get_mtk_adsp_mbox_priv - use mtk_adsp_mbox_priv va_mboxreg address in adsp mbox driver - add struct mtk_adsp_mbox_cfg for DSP mbox register offset - remove adsp mbox register offset hard code define - remove mtk-adsp-ipc.h reference in adsp mbox driver changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 135 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 176 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 10 files changed, 450 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 5

[PATCH v2 00/20] Support HDMI audio on NVIDIA Tegra20
by Dmitry Osipenko 02 Dec '21

02 Dec '21

This series revives Tegra20 S/PDIF driver which was upstreamed long time ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI audio CODEC provider. Finally, HDMI audio is enabled in device-trees. For now the audio is enable only for Acer A500 tablet and Toshiba AC100 netbook because they're already supported by upstream, later on ASUS TF101 tablet will join them. I based S/PDIF patches on Arnd's Bergmann patch from a separate series [1] that removes obsolete slave_id. This eases merging of the patches by removing the merge conflict. [1] https://patchwork.ozlabs.org/project/linux-tegra/list/?submitter=80402 Changelog: v1: - Corrected I2S yaml problem that was reported by the DT bot for v1 by removing the non-existent required clock-names property. - Removed assigned-clocks property from S/PDIF yaml since this property is now inherited from the clocks property. - Reordered the "tegra20: spdif: Set FIFO trigger level" patch, making it the first sound/soc patch in the series, like it was suggested by Mark Brown in the comment to v1. Also reworded commit message of this patch to *not* make it looks like it should be backported to stable kernels. Arnd Bergmann (1): ASoC: tegra20-spdif: stop setting slave_id Dmitry Osipenko (19): ASoC: dt-bindings: Add binding for Tegra20 S/PDIF ASoC: dt-bindings: tegra20-i2s: Convert to schema ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property dt-bindings: host1x: Document optional HDMI sound-dai-cells ASoC: tegra20: spdif: Set FIFO trigger level ASoC: tegra20: spdif: Support device-tree ASoC: tegra20: spdif: Improve driver's code ASoC: tegra20: spdif: Use more resource-managed helpers ASoC: tegra20: spdif: Reset hardware ASoC: tegra20: spdif: Support system suspend ASoC: tegra20: spdif: Filter out unsupported rates ASoC: tegra20: i2s: Filter out unsupported rates drm/tegra: hdmi: Unwind tegra_hdmi_init() errors drm/tegra: hdmi: Register audio CODEC on Tegra20 ARM: tegra_defconfig: Enable S/PDIF driver ARM: tegra: Add S/PDIF node to Tegra20 device-tree ARM: tegra: Add HDMI audio graph to Tegra20 device-tree ARM: tegra: acer-a500: Enable S/PDIF and HDMI audio ARM: tegra: paz00: Enable S/PDIF and HDMI audio .../display/tegra/nvidia,tegra20-host1x.txt | 1 + .../bindings/sound/nvidia,tegra20-i2s.txt | 30 --- .../bindings/sound/nvidia,tegra20-i2s.yaml | 77 +++++++ .../bindings/sound/nvidia,tegra20-spdif.yaml | 85 ++++++++ .../boot/dts/tegra20-acer-a500-picasso.dts | 8 + arch/arm/boot/dts/tegra20-paz00.dts | 8 + arch/arm/boot/dts/tegra20.dtsi | 40 +++- arch/arm/configs/tegra_defconfig | 1 + drivers/gpu/drm/tegra/Kconfig | 3 + drivers/gpu/drm/tegra/hdmi.c | 168 +++++++++++++-- sound/soc/tegra/tegra20_i2s.c | 49 +++++ sound/soc/tegra/tegra20_spdif.c | 195 +++++++++++++----- sound/soc/tegra/tegra20_spdif.h | 1 + sound/soc/tegra/tegra_pcm.c | 6 + sound/soc/tegra/tegra_pcm.h | 1 + 15 files changed, 571 insertions(+), 102 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml -- 2.33.1

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[PATCH v2 0/4] ARM: dts: stm32: merge spi and i2s nodes
by Olivier Moysan 02 Dec '21

02 Dec '21

When a STM32 SPI instance offers I2S feature, two nodes are defined in STM32 SoC device tree to support both SPI and I2S. Merge SPI node and I2S nodes into a single node, to avoid hardware description duplication and compilation warnings. Changes in v2: - Squash some patches in the serie to avoid dtbs check warnings on intermediate compilation steps. Olivier Moysan (4): ASoC: dt-bindings: stm32: i2s: add audio-graph-card port ASoC: dt-bindings: stm32: i2s: allow additional properties. ASoC: dt-bindings: stm32: i2s: update example ARM: dts: stm32: merge spi and i2s nodes .../bindings/sound/st,stm32-i2s.yaml | 9 ++++- arch/arm/boot/dts/stm32mp151.dtsi | 39 ++----------------- arch/arm/boot/dts/stm32mp157c-ev1.dts | 2 +- .../arm/boot/dts/stm32mp15xx-dhcom-drc02.dtsi | 2 +- .../boot/dts/stm32mp15xx-dhcor-avenger96.dtsi | 2 +- arch/arm/boot/dts/stm32mp15xx-dkx.dtsi | 7 +++- 6 files changed, 18 insertions(+), 43 deletions(-) -- 2.17.1

2 8

[PATCH v2] ASoC: tegra: Add master volume/mute control support
by Sameer Pujar 01 Dec '21

01 Dec '21

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(a)nvidia.com> --- changes in v2: * Kcontrol put() related comments, received during v1, are addressed in another series [0]. * Thus v2 is rebased on top of recent changes. * In doing so, this patch also addresses optimization comment received on [0] which suggested usage of regmap_update_bits_check() in the MVC driver. [0] https://lkml.org/lkml/2021/11/18/930 sound/soc/tegra/tegra210_mvc.c | 209 ++++++++++++++++++++++++++++++++--------- sound/soc/tegra/tegra210_mvc.h | 5 + 2 files changed, 169 insertions(+), 45 deletions(-) diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index 93cbeb6..2d79138 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -108,67 +108,152 @@ static void tegra210_mvc_conv_vol(struct tegra210_mvc *mvc, u8 chan, s32 val) } } -static int tegra210_mvc_get_mute(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static u32 tegra210_mvc_get_ctrl_reg(struct snd_kcontrol *kcontrol) { struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - u8 mute_mask; u32 val; pm_runtime_get_sync(cmpnt->dev); regmap_read(mvc->regmap, TEGRA210_MVC_CTRL, &val); pm_runtime_put(cmpnt->dev); - mute_mask = (val >> TEGRA210_MVC_MUTE_SHIFT) & - TEGRA210_MUTE_MASK_EN; + return val; +} + +static int tegra210_mvc_get_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 val = tegra210_mvc_get_ctrl_reg(kcontrol); + u8 mute_mask = TEGRA210_GET_MUTE_VAL(val); - ucontrol->value.integer.value[0] = mute_mask; + /* + * If per channel control is enabled, then return + * exact mute/unmute setting of all channels. + * + * Else report setting based on CH0 bit to reflect + * the correct HW state. + */ + if (val & TEGRA210_MVC_PER_CHAN_CTRL_EN) { + ucontrol->value.integer.value[0] = mute_mask; + } else { + if (mute_mask & TEGRA210_MVC_CH0_MUTE_EN) + ucontrol->value.integer.value[0] = + TEGRA210_MUTE_MASK_EN; + else + ucontrol->value.integer.value[0] = 0; + } return 0; } -static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_mvc_get_master_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 val = tegra210_mvc_get_ctrl_reg(kcontrol); + u8 mute_mask = TEGRA210_GET_MUTE_VAL(val); + + /* + * If per channel control is disabled, then return + * master mute/unmute setting based on CH0 bit. + * + * Else report settings based on state of all + * channels. + */ + if (!(val & TEGRA210_MVC_PER_CHAN_CTRL_EN)) { + ucontrol->value.integer.value[0] = + mute_mask & TEGRA210_MVC_CH0_MUTE_EN; + } else { + if (mute_mask == TEGRA210_MUTE_MASK_EN) + ucontrol->value.integer.value[0] = + TEGRA210_MVC_CH0_MUTE_EN; + else + ucontrol->value.integer.value[0] = 0; + } + + return 0; +} + +static int tegra210_mvc_volume_switch_timeout(struct snd_soc_component *cmpnt) { - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - unsigned int value; - u8 new_mask, old_mask; + u32 value; int err; - pm_runtime_get_sync(cmpnt->dev); - - /* Check if VOLUME_SWITCH is triggered */ err = regmap_read_poll_timeout(mvc->regmap, TEGRA210_MVC_SWITCH, value, !(value & TEGRA210_MVC_VOLUME_SWITCH_MASK), 10, 10000); if (err < 0) - goto end; + dev_err(cmpnt->dev, + "Volume switch trigger is still active, err = %d\n", + err); - regmap_read(mvc->regmap, TEGRA210_MVC_CTRL, &value); + return err; +} - old_mask = (value >> TEGRA210_MVC_MUTE_SHIFT) & TEGRA210_MUTE_MASK_EN; - new_mask = ucontrol->value.integer.value[0]; +static int tegra210_mvc_update_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + bool per_chan_ctrl) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); + u32 mute_val = ucontrol->value.integer.value[0]; + u32 per_ch_ctrl_val; + bool change = false; + int err; - if (new_mask == old_mask) { - err = 0; + pm_runtime_get_sync(cmpnt->dev); + + err = tegra210_mvc_volume_switch_timeout(cmpnt); + if (err < 0) goto end; + + if (per_chan_ctrl) { + per_ch_ctrl_val = TEGRA210_MVC_PER_CHAN_CTRL_EN; + } else { + per_ch_ctrl_val = 0; + + if (mute_val) + mute_val = TEGRA210_MUTE_MASK_EN; } - err = regmap_update_bits(mvc->regmap, mc->reg, + regmap_update_bits_check(mvc->regmap, TEGRA210_MVC_CTRL, TEGRA210_MVC_MUTE_MASK, - new_mask << TEGRA210_MVC_MUTE_SHIFT); - if (err < 0) - goto end; + mute_val << TEGRA210_MVC_MUTE_SHIFT, + &change); - err = 1; + if (change) { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, + per_ch_ctrl_val); + + regmap_update_bits(mvc->regmap, TEGRA210_MVC_SWITCH, + TEGRA210_MVC_VOLUME_SWITCH_MASK, + TEGRA210_MVC_VOLUME_SWITCH_TRIGGER); + } end: pm_runtime_put(cmpnt->dev); - return err; + + if (err < 0) + return err; + + if (change) + return 1; + + return 0; +} + +static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_update_mute(kcontrol, ucontrol, true); +} + +static int tegra210_mvc_put_master_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_update_mute(kcontrol, ucontrol, false); } static int tegra210_mvc_get_vol(struct snd_kcontrol *kcontrol, @@ -178,7 +263,7 @@ static int tegra210_mvc_get_vol(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - u8 chan = (mc->reg - TEGRA210_MVC_TARGET_VOL) / REG_SIZE; + u8 chan = TEGRA210_MVC_GET_CHAN(mc->reg, TEGRA210_MVC_TARGET_VOL); s32 val = mvc->volume[chan]; if (mvc->curve_type == CURVE_POLY) { @@ -193,44 +278,55 @@ static int tegra210_mvc_get_vol(struct snd_kcontrol *kcontrol, return 0; } -static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_mvc_get_master_vol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_get_vol(kcontrol, ucontrol); +} + +static int tegra210_mvc_update_vol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + bool per_ch_enable) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - unsigned int reg = mc->reg; - unsigned int value; - u8 chan; - int err, old_volume; + u8 chan = TEGRA210_MVC_GET_CHAN(mc->reg, TEGRA210_MVC_TARGET_VOL); + int old_volume = mvc->volume[chan]; + int err, i; pm_runtime_get_sync(cmpnt->dev); - /* Check if VOLUME_SWITCH is triggered */ - err = regmap_read_poll_timeout(mvc->regmap, TEGRA210_MVC_SWITCH, - value, !(value & TEGRA210_MVC_VOLUME_SWITCH_MASK), - 10, 10000); + err = tegra210_mvc_volume_switch_timeout(cmpnt); if (err < 0) goto end; - chan = (reg - TEGRA210_MVC_TARGET_VOL) / REG_SIZE; - old_volume = mvc->volume[chan]; - - tegra210_mvc_conv_vol(mvc, chan, - ucontrol->value.integer.value[0]); + tegra210_mvc_conv_vol(mvc, chan, ucontrol->value.integer.value[0]); if (mvc->volume[chan] == old_volume) { err = 0; goto end; } + if (per_ch_enable) { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, + TEGRA210_MVC_PER_CHAN_CTRL_EN); + } else { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, 0); + + for (i = 1; i < TEGRA210_MVC_MAX_CHAN_COUNT; i++) + mvc->volume[i] = mvc->volume[chan]; + } + /* Configure init volume same as target volume */ regmap_write(mvc->regmap, TEGRA210_MVC_REG_OFFSET(TEGRA210_MVC_INIT_VOL, chan), mvc->volume[chan]); - regmap_write(mvc->regmap, reg, mvc->volume[chan]); + regmap_write(mvc->regmap, mc->reg, mvc->volume[chan]); regmap_update_bits(mvc->regmap, TEGRA210_MVC_SWITCH, TEGRA210_MVC_VOLUME_SWITCH_MASK, @@ -240,9 +336,22 @@ static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, end: pm_runtime_put(cmpnt->dev); + return err; } +static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_update_vol(kcontrol, ucontrol, true); +} + +static int tegra210_mvc_put_master_vol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_update_vol(kcontrol, ucontrol, false); +} + static void tegra210_mvc_reset_vol_settings(struct tegra210_mvc *mvc, struct device *dev) { @@ -438,6 +547,16 @@ static const struct snd_kcontrol_new tegra210_mvc_vol_ctrl[] = { TEGRA210_MVC_CTRL, 0, TEGRA210_MUTE_MASK_EN, 0, tegra210_mvc_get_mute, tegra210_mvc_put_mute), + /* Master volume */ + SOC_SINGLE_EXT("Volume", TEGRA210_MVC_TARGET_VOL, 0, 16000, 0, + tegra210_mvc_get_master_vol, + tegra210_mvc_put_master_vol), + + /* Master mute */ + SOC_SINGLE_EXT("Mute", TEGRA210_MVC_CTRL, 0, 1, 0, + tegra210_mvc_get_master_mute, + tegra210_mvc_put_master_mute), + SOC_ENUM_EXT("Curve Type", tegra210_mvc_curve_type_ctrl, tegra210_mvc_get_curve_type, tegra210_mvc_put_curve_type), }; diff --git a/sound/soc/tegra/tegra210_mvc.h b/sound/soc/tegra/tegra210_mvc.h index def29c4..d775335 100644 --- a/sound/soc/tegra/tegra210_mvc.h +++ b/sound/soc/tegra/tegra210_mvc.h @@ -59,6 +59,7 @@ #define TEGRA210_MUTE_MASK_EN 0xff #define TEGRA210_MVC_MUTE_MASK (TEGRA210_MUTE_MASK_EN << TEGRA210_MVC_MUTE_SHIFT) #define TEGRA210_MVC_MUTE_EN (TEGRA210_MUTE_MASK_EN << TEGRA210_MVC_MUTE_SHIFT) +#define TEGRA210_MVC_CH0_MUTE_EN 1 #define TEGRA210_MVC_PER_CHAN_CTRL_EN_SHIFT 30 #define TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK (1 << TEGRA210_MVC_PER_CHAN_CTRL_EN_SHIFT) @@ -92,6 +93,10 @@ #define TEGRA210_MVC_MAX_CHAN_COUNT 8 #define TEGRA210_MVC_REG_OFFSET(reg, i) (reg + (REG_SIZE * i)) +#define TEGRA210_MVC_GET_CHAN(reg, base) (((reg) - (base)) / REG_SIZE) + +#define TEGRA210_GET_MUTE_VAL(val) (((val) >> TEGRA210_MVC_MUTE_SHIFT) & TEGRA210_MUTE_MASK_EN) + #define NUM_GAIN_POLY_COEFFS 9 enum { -- 2.7.4

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