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[PATCH v2] ALSA: usb-audio: Re-apply implicit feedback mode to Pioneer devices
by Takashi Iwai 20 Apr '21

20 Apr '21

Pioneer devices are supposed to be working with the implicit feedback mode, but so far the attempt to apply the implicit feedback caused issues, hence we explicitly skipped the implicit feedback mode for them. Recently, Geraldo discovered that the device actually works if you skip the generic matching of the sync EPs for the capture stream. That is, we should apply the implicit feedback setup for the playback like other similar devices, while we need to return 1 from audioformat_capture_quirk() so that no further matching will be done. And, later on, Olivia reported later that the fiddling with the capture quirk alone doesn't suffice for the test with speaker-test program. This seems to be a similar case like the recently fixed BOSS devices. Indeed, the problem could be addressed by setting playback_first flag, which indicates that the playback URBs have to be sent out at first even in the implicit feedback mode. This patch implements the application of the implicit feedback to Pioneer devices as described in the above. The former skip_pioneer_sync_ep() was dropped, and instead we provide is_pioneer_implicit_fb() to check the Pioneer devices that need the implicit feedback. In the audioformat_implicit_fb_quirk(), simply apply the implicit fb for playback and set chip->playback_first flag if matching, and in audioformat_capture_quirk()(), it returns 1 for skipping the generic EP sync handling. Reported-by: Geraldo <geraldogabriel(a)gmail.com> Tested-by: Olivia Mackintosh <livvy(a)base.nu> Link: https://lore.kernel.org/r/s5ha6pygqfz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/implicit.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 428e3a449f5d..77ffcea294dd 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -230,18 +230,27 @@ static int add_roland_implicit_fb(struct snd_usb_audio *chip, ifnum, alts); } -/* Playback and capture EPs on Pioneer devices share the same iface/altset, - * but they don't seem working with the implicit fb mode well, hence we - * just return as if the sync were already set up. +/* Playback and capture EPs on Pioneer devices share the same iface/altset + * for the implicit feedback operation */ -static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, - struct audioformat *fmt, - struct usb_host_interface *alts) +static bool is_pioneer_implicit_fb(struct snd_usb_audio *chip, + struct usb_host_interface *alts) + { struct usb_endpoint_descriptor *epd; + if (USB_ID_VENDOR(chip->usb_id) != 0x2b73 && + USB_ID_VENDOR(chip->usb_id) != 0x08e4) + return false; + if (alts->desc.bInterfaceClass != USB_CLASS_VENDOR_SPEC) + return false; if (alts->desc.bNumEndpoints != 2) - return 0; + return false; + + epd = get_endpoint(alts, 0); + if (!usb_endpoint_is_isoc_out(epd) || + (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC) + return false; epd = get_endpoint(alts, 1); if (!usb_endpoint_is_isoc_in(epd) || @@ -250,8 +259,9 @@ static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, USB_ENDPOINT_USAGE_DATA && (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != USB_ENDPOINT_USAGE_IMPLICIT_FB)) - return 0; - return 1; /* don't handle with the implicit fb, just skip sync EP */ + return false; + + return true; } static int __add_generic_implicit_fb(struct snd_usb_audio *chip, @@ -367,12 +377,12 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Pioneer devices with vendor spec class */ - if (attr == USB_ENDPOINT_SYNC_ASYNC && - alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && - (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ - USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) { - if (skip_pioneer_sync_ep(chip, fmt, alts)) - return 1; + if (is_pioneer_implicit_fb(chip, alts)) { + chip->playback_first = 1; + return add_implicit_fb_sync_ep(chip, fmt, + get_endpoint(alts, 1)->bEndpointAddress, + 1, alts->desc.bInterfaceNumber, + alts); } /* Try the generic implicit fb if available */ @@ -394,6 +404,8 @@ static int audioformat_capture_quirk(struct snd_usb_audio *chip, if (p && (p->type == IMPLICIT_FB_FIXED || p->type == IMPLICIT_FB_BOTH)) return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0, p->iface, NULL); + if (is_pioneer_implicit_fb(chip, alts)) + return 1; /* skip the quirk, also don't handle generic sync EP */ return 0; } -- 2.26.2

2 1

[PATCH 0/5] ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack
by David Ward 19 Apr '21

19 Apr '21

The last two patches in this series fix a longstanding issue that prevented the ALC3263 codec from using a headset mic. This codec can be found on Dell systems including the Latitude 13 7350, Venue 11 Pro 7140, and XPS 13 9343. In fact, there is an ACPI quirk for the XPS 13 9343, which forces it to use legacy HD Audio just to avoid this issue: https://lore.kernel.org/alsa-devel/CAPeXnHv07HkvcHrYFmZMr8OTp7U7F=k_k=LPYnU… This may allow that ACPI quirk to be removed. Either way, the other systems mentioned above do not support this quirk and already use the ASoC driver, so this fix is necessary for headset mic support on those systems. Note: there is likely other handling for this codec that only exists in the HDA driver, but which also belongs in the ASoC driver. Commit 394c97f824fa ("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that does not seem to be resolved in the ASoC driver, to give an example. Other patches in this series are not specific to the ALC3263. These patches set the correct combo jack configuration when headphones are inserted, and fix a misaligned value set in the DMIC2 Configuration Default register. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=114171 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> David Ward (5): ASoC: rt286: Fix upper byte in DMIC2 configuration ASoC: rt286: Configure combo jack for headphones ASoC: rt298: Configure combo jack for headphones ASoC: rt286: Make RT286_SET_GPIO_* readable and writable ASoC: rt286: Generalize support for ALC3263 codec sound/soc/codecs/rt286.c | 34 +++++++++++++++++++++------------- sound/soc/codecs/rt298.c | 9 +++++++-- 2 files changed, 28 insertions(+), 15 deletions(-) -- 2.31.1

3 7

[PATCH 0/5] ASoC: remove more cppcheck warnings
by Pierre-Louis Bossart 19 Apr '21

19 Apr '21

I missed those warnings in the previous rounds, they are mostly trivial and shouldn't change the behavior. Pierre-Louis Bossart (5): ASoC: soc-acpi: remove useless initialization ASoC: soc-core: fix signed/unsigned issue ASoC: soc-core: fix always-false condition ASoC: codecs: lpass-rx-macro: remove useless return ASoC: codecs: rt5682: clarify expression sound/soc/codecs/lpass-rx-macro.c | 2 -- sound/soc/codecs/rt5682.c | 2 +- sound/soc/soc-acpi.c | 2 +- sound/soc/soc-core.c | 7 +------ 4 files changed, 3 insertions(+), 10 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: sigmadsp: Disable cache mechanism for readbacks
by Niklas Carlsson 19 Apr '21

19 Apr '21

From: Niklas Carlsson <niklasc(a)axis.com> The ALSA control readback functionality only works for non-volatile controls, i.e. control values that does not change on their own without driver interaction. This doesn't work for readbacks since the DSP firmware updates the control value. Disable the cache mechanism in the driver if the control name matches the prefix used for readbacks to ensure that the control value is valid. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/sigmadsp.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 76c77dc8ecf7..b992216aee55 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -24,6 +24,8 @@ #define SIGMA_FW_CHUNK_TYPE_CONTROL 1 #define SIGMA_FW_CHUNK_TYPE_SAMPLERATES 2 +#define READBACK_CTRL_NAME "ReadBack" + struct sigmadsp_control { struct list_head head; uint32_t samplerates; @@ -31,6 +33,7 @@ struct sigmadsp_control { unsigned int num_bytes; const char *name; struct snd_kcontrol *kcontrol; + bool is_readback; bool cached; uint8_t cache[]; }; @@ -141,7 +144,8 @@ static int sigmadsp_ctrl_put(struct snd_kcontrol *kcontrol, if (ret == 0) { memcpy(ctrl->cache, data, ctrl->num_bytes); - ctrl->cached = true; + if (!ctrl->is_readback) + ctrl->cached = true; } mutex_unlock(&sigmadsp->lock); @@ -164,7 +168,8 @@ static int sigmadsp_ctrl_get(struct snd_kcontrol *kcontrol, } if (ret == 0) { - ctrl->cached = true; + if (!ctrl->is_readback) + ctrl->cached = true; memcpy(ucontrol->value.bytes.data, ctrl->cache, ctrl->num_bytes); } @@ -231,6 +236,15 @@ static int sigma_fw_load_control(struct sigmadsp *sigmadsp, name[name_len] = '\0'; ctrl->name = name; + /* + * Readbacks doesn't work with non-volatile controls, since the + * firmware updates the control value without driver interaction. Mark + * the readbacks to ensure that the values are not cached. + */ + if (ctrl->name && strncmp(ctrl->name, READBACK_CTRL_NAME, + (sizeof(READBACK_CTRL_NAME) - 1)) == 0) + ctrl->is_readback = true; + ctrl->addr = le16_to_cpu(ctrl_chunk->addr); ctrl->num_bytes = num_bytes; ctrl->samplerates = le32_to_cpu(chunk->samplerates); -- 2.20.1

2 1

[PATCH v2] ASoC: ak4458: check reset control status
by Shengjiu Wang 19 Apr '21

19 Apr '21

From: Viorel Suman <viorel.suman(a)nxp.com> check the return value of ak4458_rstn_control. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - fix "use of undeclared identifier 'ret'" reported by kernel test robot sound/soc/codecs/ak4458.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 85a1d00894a9..3088128816bb 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -334,7 +334,7 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int pcm_width = max(params_physical_width(params), ak4458->slot_width); u8 format, dsdsel0, dsdsel1; - int nfs1, dsd_bclk; + int nfs1, dsd_bclk, ret; nfs1 = params_rate(params); ak4458->fs = nfs1; @@ -419,8 +419,13 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AK4458_00_CONTROL1, AK4458_DIF_MASK, format); - ak4458_rstn_control(component, 0); - ak4458_rstn_control(component, 1); + ret = ak4458_rstn_control(component, 0); + if (ret) + return ret; + + ret = ak4458_rstn_control(component, 1); + if (ret) + return ret; return 0; } @@ -429,6 +434,7 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); + int ret; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* Slave Mode */ @@ -461,8 +467,13 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) ak4458->fmt == SND_SOC_DAIFMT_PDM ? AK4458_DP_MASK : 0); - ak4458_rstn_control(component, 0); - ak4458_rstn_control(component, 1); + ret = ak4458_rstn_control(component, 0); + if (ret) + return ret; + + ret = ak4458_rstn_control(component, 1); + if (ret) + return ret; return 0; } -- 2.27.0

2 1

[v6 2/3] ASoC: Add Rockchip rk817 audio CODEC support
by Chris Morgan 19 Apr '21

19 Apr '21

Add support for the Rockchip rk817 audio codec integrated into the rk817 PMIC. This is based on the sources provided by Rockchip from their BSP kernel. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in seperate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 560 +++++++++++++++++++++++++++++++++ 3 files changed, 568 insertions(+) create mode 100644 sound/soc/codecs/rk817_codec.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c87b42606c9..7624475efa08 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -155,6 +155,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_RK3328 + imply SND_SOC_RK817 imply SND_SOC_RT274 imply SND_SOC_RT286 imply SND_SOC_RT298 @@ -1058,6 +1059,11 @@ config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO +config SND_SOC_RK817 + tristate "Rockchip RK817 audio CODEC" + depends on MFD_RK808 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..4e5e9b28775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -165,6 +165,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rk3328-objs := rk3328_codec.o +snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o +obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c new file mode 100644 index 000000000000..6938a7a1f02b --- /dev/null +++ b/sound/soc/codecs/rk817_codec.c @@ -0,0 +1,560 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk817 ALSA SoC Audio driver +// +// Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mfd/rk808.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +struct rk817_codec_priv { + struct snd_soc_component *component; + struct rk808 *rk808; + struct clk *mclk; + unsigned int stereo_sysclk; + bool mic_in_differential; +}; + +/* + * This sets the codec up with the values defined in the default implementation including the APLL + * from the Rockchip vendor kernel. I do not know if these values are universal despite differing + * from the default values defined above and taken from the datasheet, or implementation specific. + * I don't have another implementation to compare from the Rockchip sources. Hard-coding for now. + * Additionally, I do not know according to the documentation the units accepted for the clock + * values, so for the moment those are left unvalidated. + */ + +static int rk817_init(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_write(component, RK817_CODEC_DDAC_POPD_DACST, 0x02); + snd_soc_component_write(component, RK817_CODEC_DDAC_SR_LMT0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DADC_SR_ACL0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DTOP_VUCTIME, 0xf4); + if (rk817->mic_in_differential) { + snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, + MIC_DIFF_EN); + }; + return 0; +} + +static int rk817_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + /* Set resistor value and charge pump current for PLL. */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG1, 0x58); + /* Set the PLL feedback clock divide value (values not documented). */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG2, 0x2d); + /* Set the PLL pre-divide value (values not documented). */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG3, 0x0c); + /* Set the PLL VCO output clock divide and PLL divided ratio of PLL High Clk (values not + * documented). + */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG4, 0xa5); + + return 0; +} + +/* + * DDAC/DADC L/R volume setting + * 0db~-95db, 0.375db/step, for example: + * 0x00: 0dB + * 0xff: -95dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_vol_tlv, -9500, 0); + +/* + * PGA GAIN L/R volume setting + * 27db~-18db, 3db/step, for example: + * 0x0: -18dB + * 0xf: 27dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_gain_tlv, -1800, 2700); + +static const struct snd_kcontrol_new rk817_volume_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume", RK817_CODEC_DDAC_VOLL, + RK817_CODEC_DDAC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume", RK817_CODEC_DADC_VOLL, + RK817_CODEC_DADC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_TLV("Mic Capture Gain", RK817_CODEC_DMIC_PGA_GAIN, 4, 0, 0xf, 0, + rk817_gain_tlv), +}; + +/* Since the speaker output and L headphone pin are internally the same, make audio path mutually + * exclusive with a mux. + */ + +static const char *dac_mux_text[] = { + "HP", + "SPK", +}; + +static SOC_ENUM_SINGLE_VIRT_DECL(dac_enum, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = + SOC_DAPM_ENUM("Playback Mux", dac_enum); + +static const struct snd_soc_dapm_widget rk817_dapm_widgets[] = { + + /* capture/playback common */ + SND_SOC_DAPM_SUPPLY("LDO Regulator", RK817_CODEC_AREF_RTCFG1, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBIAS Block", RK817_CODEC_AREF_RTCFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VAvg Buffer", RK817_CODEC_AREF_RTCFG1, 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL Power", RK817_CODEC_APLL_CFG5, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX1 Transfer Start", RK817_CODEC_DI2S_RXCMD_TSD, 5, 0, NULL, 0), + + /* capture path common */ + SND_SOC_DAPM_SUPPLY("ADC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC Power On", RK817_CODEC_AMIC_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Transfer Start", RK817_CODEC_DI2S_TXCR3_TXCMD, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Right Justified", RK817_CODEC_DI2S_TXCR3_TXCMD, 3, 0, NULL, 0), + + /* capture path L */ + SND_SOC_DAPM_ADC("ADC L", "Capture", RK817_CODEC_AADC_CFG0, 7, 1), + SND_SOC_DAPM_SUPPLY("PGA L Power On", RK817_CODEC_AMIC_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L2", RK817_CODEC_AMIC_CFG0, 2, 0, NULL, 0), + + /* capture path R */ + SND_SOC_DAPM_ADC("ADC R", "Capture", RK817_CODEC_AADC_CFG0, 6, 1), + SND_SOC_DAPM_SUPPLY("PGA R Power On", RK817_CODEC_AMIC_CFG0, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R2", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + + /* playback path common */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Bias", RK817_CODEC_ADAC_CFG1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mute Off", RK817_CODEC_DDAC_MUTE_MIXCTL, 0, 1, NULL, 0), + + /* playback path speaker */ + SND_SOC_DAPM_SUPPLY("Class D Mode", RK817_CODEC_DDAC_MUTE_MIXCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("High Pass Filter", RK817_CODEC_DDAC_MUTE_MIXCTL, 7, 0, NULL, 0), + SND_SOC_DAPM_DAC("SPK DAC", "Playback", RK817_CODEC_ADAC_CFG1, 2, 1), + SND_SOC_DAPM_SUPPLY("Enable Class D", RK817_CODEC_ACLASSD_CFG1, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Disable Class D Mute Ramp", RK817_CODEC_ACLASSD_CFG1, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 1", RK817_CODEC_ACLASSD_CFG1, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 2", RK817_CODEC_ACLASSD_CFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 2", RK817_CODEC_ACLASSD_CFG2, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 3", RK817_CODEC_ACLASSD_CFG2, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 2", RK817_CODEC_ACLASSD_CFG2, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 3", RK817_CODEC_ACLASSD_CFG2, 0, 0, NULL, 0), + + /* playback path headphones */ + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", RK817_CODEC_AHP_CP, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone CP Discharge LDO", RK817_CODEC_AHP_CP, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone OStage", RK817_CODEC_AHP_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Pre Amp", RK817_CODEC_AHP_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", "Playback", RK817_CODEC_ADAC_CFG1, 1, 1), + SND_SOC_DAPM_DAC("DAC R", "Playback", RK817_CODEC_ADAC_CFG1, 0, 1), + + /* Mux for input/output path selection */ + SND_SOC_DAPM_MUX("Playback Mux", SND_SOC_NOPM, 1, 0, &dac_mux), + + /* Pins for Simple Card Bindings */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("SPKO"), +}; + +static const struct snd_soc_dapm_route rk817_dapm_routes[] = { + + /* capture path */ + /* left mic */ + {"ADC L", NULL, "LDO Regulator"}, + {"ADC L", NULL, "IBIAS Block"}, + {"ADC L", NULL, "VAvg Buffer"}, + {"ADC L", NULL, "PLL Power"}, + {"ADC L", NULL, "ADC Clock"}, + {"ADC L", NULL, "I2S TX Clock"}, + {"ADC L", NULL, "ADC Channel Enable"}, + {"ADC L", NULL, "I2S TX Channel Enable"}, + {"ADC L", NULL, "I2S TX1 Transfer Start"}, + {"MICL", NULL, "MIC Power On"}, + {"MICL", NULL, "PGA L Power On"}, + {"MICL", NULL, "Mic Boost L1"}, + {"MICL", NULL, "Mic Boost L2"}, + {"MICL", NULL, "I2S TX3 Transfer Start"}, + {"MICL", NULL, "I2S TX3 Right Justified"}, + {"ADC L", NULL, "MICL"}, + + /* right mic */ + {"ADC R", NULL, "LDO Regulator"}, + {"ADC R", NULL, "IBIAS Block"}, + {"ADC R", NULL, "VAvg Buffer"}, + {"ADC R", NULL, "PLL Power"}, + {"ADC R", NULL, "ADC Clock"}, + {"ADC R", NULL, "I2S TX Clock"}, + {"ADC R", NULL, "ADC Channel Enable"}, + {"ADC R", NULL, "I2S TX Channel Enable"}, + {"ADC R", NULL, "I2S TX1 Transfer Start"}, + {"MICR", NULL, "MIC Power On"}, + {"MICR", NULL, "PGA R Power On"}, + {"MICR", NULL, "Mic Boost R1"}, + {"MICR", NULL, "Mic Boost R2"}, + {"MICR", NULL, "I2S TX3 Transfer Start"}, + {"MICR", NULL, "I2S TX3 Right Justified"}, + {"ADC R", NULL, "MICR"}, + + /* playback path */ + /* speaker path */ + {"SPK DAC", NULL, "LDO Regulator"}, + {"SPK DAC", NULL, "IBIAS Block"}, + {"SPK DAC", NULL, "VAvg Buffer"}, + {"SPK DAC", NULL, "PLL Power"}, + {"SPK DAC", NULL, "I2S TX1 Transfer Start"}, + {"SPK DAC", NULL, "DAC Clock"}, + {"SPK DAC", NULL, "I2S RX Clock"}, + {"SPK DAC", NULL, "DAC Channel Enable"}, + {"SPK DAC", NULL, "I2S RX Channel Enable"}, + {"SPK DAC", NULL, "Class D Mode"}, + {"SPK DAC", NULL, "DAC Bias"}, + {"SPK DAC", NULL, "DAC Mute Off"}, + {"SPK DAC", NULL, "Enable Class D"}, + {"SPK DAC", NULL, "Disable Class D Mute Ramp"}, + {"SPK DAC", NULL, "Class D Mute Rate 1"}, + {"SPK DAC", NULL, "Class D Mute Rate 2"}, + {"SPK DAC", NULL, "Class D OCPP 2"}, + {"SPK DAC", NULL, "Class D OCPP 3"}, + {"SPK DAC", NULL, "Class D OCPN 2"}, + {"SPK DAC", NULL, "Class D OCPN 3"}, + {"SPK DAC", NULL, "High Pass Filter"}, + + /* headphone path L */ + {"DAC L", NULL, "LDO Regulator"}, + {"DAC L", NULL, "IBIAS Block"}, + {"DAC L", NULL, "VAvg Buffer"}, + {"DAC L", NULL, "PLL Power"}, + {"DAC L", NULL, "I2S TX1 Transfer Start"}, + {"DAC L", NULL, "DAC Clock"}, + {"DAC L", NULL, "I2S RX Clock"}, + {"DAC L", NULL, "DAC Channel Enable"}, + {"DAC L", NULL, "I2S RX Channel Enable"}, + {"DAC L", NULL, "DAC Bias"}, + {"DAC L", NULL, "DAC Mute Off"}, + {"DAC L", NULL, "Headphone Charge Pump"}, + {"DAC L", NULL, "Headphone CP Discharge LDO"}, + {"DAC L", NULL, "Headphone OStage"}, + {"DAC L", NULL, "Headphone Pre Amp"}, + + /* headphone path R */ + {"DAC R", NULL, "LDO Regulator"}, + {"DAC R", NULL, "IBIAS Block"}, + {"DAC R", NULL, "VAvg Buffer"}, + {"DAC R", NULL, "PLL Power"}, + {"DAC R", NULL, "I2S TX1 Transfer Start"}, + {"DAC R", NULL, "DAC Clock"}, + {"DAC R", NULL, "I2S RX Clock"}, + {"DAC R", NULL, "DAC Channel Enable"}, + {"DAC R", NULL, "I2S RX Channel Enable"}, + {"DAC R", NULL, "DAC Bias"}, + {"DAC R", NULL, "DAC Mute Off"}, + {"DAC R", NULL, "Headphone Charge Pump"}, + {"DAC R", NULL, "Headphone CP Discharge LDO"}, + {"DAC R", NULL, "Headphone OStage"}, + {"DAC R", NULL, "Headphone Pre Amp"}, + + /* mux path for output selection */ + {"Playback Mux", "HP", "DAC L"}, + {"Playback Mux", "HP", "DAC R"}, + {"Playback Mux", "SPK", "SPK DAC"}, + {"SPKO", NULL, "Playback Mux"}, + {"HPOL", NULL, "Playback Mux"}, + {"HPOR", NULL, "Playback Mux"}, +}; + +static int rk817_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817->stereo_sysclk = freq; + + return 0; +} + +static int rk817_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + unsigned int i2s_mst = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s_mst |= RK817_I2S_MODE_SLV; + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2s_mst |= RK817_I2S_MODE_MST; + break; + default: + dev_err(component->dev, "%s : set master mask failed!\n", __func__); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RK817_CODEC_DI2S_CKM, + RK817_I2S_MODE_MASK, i2s_mst); + + return 0; +} + +static int rk817_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_16BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_16BITS); + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_24BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_24BITS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_ENABLE); + else + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_DISABLE); + + return 0; +} + +#define RK817_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops rk817_dai_ops = { + .hw_params = rk817_hw_params, + .set_fmt = rk817_set_dai_fmt, + .set_sysclk = rk817_set_dai_sysclk, + .mute_stream = rk817_digital_mute, + .no_capture_mute = 1, +}; + +static struct snd_soc_dai_driver rk817_dai[] = { + { + .name = "rk817-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = RK817_PLAYBACK_RATES, + .formats = RK817_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RK817_CAPTURE_RATES, + .formats = RK817_FORMATS, + }, + .ops = &rk817_dai_ops, + }, +}; + +static int rk817_probe(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + struct rk808 *rk808 = dev_get_drvdata(component->dev->parent); + int ret; + + snd_soc_component_init_regmap(component, rk808->regmap); + rk817->component = component; + + ret = snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + + rk817_init(component); + + /* setting initial pll values so that we can continue to leverage simple-audio-card. + * The values aren't important since no parameters are used. + */ + + snd_soc_component_set_pll(component, 0, 0, 0, 0); + + return 0; +} + +static void rk817_remove(struct snd_soc_component *component) +{ + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver soc_codec_dev_rk817 = { + .probe = rk817_probe, + .remove = rk817_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .controls = rk817_volume_controls, + .num_controls = ARRAY_SIZE(rk817_volume_controls), + .dapm_routes = rk817_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rk817_dapm_routes), + .dapm_widgets = rk817_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk817_dapm_widgets), + .set_pll = rk817_set_component_pll, +}; + +static int rk817_codec_parse_dt_property(struct device *dev, + struct rk817_codec_priv *rk817) +{ + struct device_node *node = dev->parent->of_node; + + if (!node) { + dev_err(dev, "%s() dev->parent->of_node is NULL\n", + __func__); + return -ENODEV; + } + + node = of_get_child_by_name(dev->parent->of_node, "codec"); + if (!node) { + dev_err(dev, "%s() Can not get child: codec\n", + __func__); + return -ENODEV; + } + + rk817->mic_in_differential = + of_property_read_bool(node, "mic-in-differential"); + + return 0; +} + +static int rk817_platform_probe(struct platform_device *pdev) +{ + struct rk808 *rk808 = dev_get_drvdata(pdev->dev.parent); + struct rk817_codec_priv *rk817_codec_data; + int ret; + + if (!rk808) { + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); + return -EINVAL; + } + + rk817_codec_data = devm_kzalloc(&pdev->dev, + sizeof(struct rk817_codec_priv), + GFP_KERNEL); + if (!rk817_codec_data) + return -ENOMEM; + + platform_set_drvdata(pdev, rk817_codec_data); + + rk817_codec_data->rk808 = rk808; + + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); + if (ret < 0) { + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", + __func__, ret); + goto err_; + } + + rk817_codec_data->mclk = clk_get(pdev->dev.parent, "mclk"); + if (IS_ERR(rk817_codec_data->mclk)) { + dev_err(&pdev->dev, "Unable to get mclk\n"); + ret = -ENXIO; + goto err_; + } + + ret = clk_prepare_enable(rk817_codec_data->mclk); + if (ret < 0) { + dev_err(&pdev->dev, "%s() clock prepare error %d\n", + __func__, ret); + goto err_; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, + rk817_dai, ARRAY_SIZE(rk817_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "%s() register codec error %d\n", + __func__, ret); + goto err_; + } + + return 0; +err_: + + return ret; +} + +static int rk817_platform_remove(struct platform_device *pdev) +{ + struct rk817_codec_priv *rk817 = platform_get_drvdata(pdev); + + clk_disable_unprepare(rk817->mclk); + + return 0; +} + +MODULE_DEVICE_TABLE(of, rk817_codec_dt_ids); + +static struct platform_driver rk817_codec_driver = { + .driver = { + .name = "rk817-codec", + }, + .probe = rk817_platform_probe, + .remove = rk817_platform_remove, +}; + +module_platform_driver(rk817_codec_driver); + +MODULE_DESCRIPTION("ASoC RK817 codec driver"); +MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); -- 2.25.1

2 1

[RFC PATCH 0/3] Separate BE DAI HW constraints from FE ones
by Codrin Ciubotariu 19 Apr '21

19 Apr '21

HW constraints are needed to set limitations for HW parameters used to configure the DAIs. All DAIs on the same link must agree upon the HW parameters, so the parameters are affected by the DAIs' features and their limitations. In case of DPCM, the FE DAIs can be used to perform different kind of conversions, such as format or rate changing, bringing the audio stream to a configuration supported by all the DAIs of the BE's link. For this reason, the limitations of the BE DAIs are not always important for the HW parameters between user-space and FE, only for the paratemers between FE and BE DAI links. This brings us to this patch-set, which aims to separate the FE HW constraints from the BE HW constraints. This way, the FE can be used to perform more efficient HW conversions, on the initial audio stream parameters, to parameters supported by the BE DAIs. To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs. This change affects many sound drivers (and one gpu drm driver). All these changes are included in the first patch, to have a better overview of the implications created by this change. The second patch adds a new struct snd_pcm_hw_constraints under struct snd_soc_dpcm_runtime, which is used to store the HW constraint rules added by the BE DAIs. This structure is initialized with a subset of the runtime constraint rules which does not include the rules that affect the buffer or period size. snd_pcm_hw_rule_add() will add the BE rules to the new struct snd_pcm_hw_constraints. The third and last patch will apply the BE rule constraints, after the fixup callback. If the fixup HW parameters do not respect the BE constraint rules, the rules will exit with an error. The FE mask and interval constraints are copied to the BE ones, to satisfy the dai_link->dpcm_merged_* flags. The dai_link->dpcm_merged_* flags are used to know if the FE does format or sampling rate conversion. I tested with ad1934 and wm8731 codecs as BEs, with a not-yet-mainlined ASRC as FE, that can also do format conversion. I realize that the change to snd_pcm_hw_rule_add() has a big impact, even though all the drivers use snd_pcm_hw_rule_add() with substream->runtime, so passing substream instead of runtime is not that risky. Codrin Ciubotariu (3): pcm: use substream instead of runtime in snd_pcm_hw_rule_add() ASoC: soc-pcm: add hw_constraints for BE DAI links ASoC: soc-pcm: apply BE HW constraint rules drivers/gpu/drm/vc4/vc4_hdmi.c | 2 +- include/sound/ac97_codec.h | 2 +- include/sound/pcm.h | 29 +++++-- include/sound/pcm_drm_eld.h | 2 +- include/sound/soc-dpcm.h | 1 + sound/arm/aaci.c | 4 +- sound/arm/pxa2xx-pcm-lib.c | 4 +- sound/core/pcm_drm_eld.c | 6 +- sound/core/pcm_lib.c | 66 ++++++++------- sound/core/pcm_native.c | 83 +++++++++++-------- sound/drivers/aloop.c | 8 +- sound/drivers/vx/vx_pcm.c | 8 +- sound/firewire/amdtp-am824.c | 8 +- sound/firewire/amdtp-am824.h | 2 +- sound/firewire/amdtp-stream.c | 12 +-- sound/firewire/amdtp-stream.h | 2 +- sound/firewire/bebob/bebob_pcm.c | 6 +- sound/firewire/dice/dice-pcm.c | 6 +- sound/firewire/digi00x/amdtp-dot.c | 6 +- sound/firewire/digi00x/digi00x-pcm.c | 6 +- sound/firewire/digi00x/digi00x.h | 2 +- sound/firewire/fireface/amdtp-ff.c | 6 +- sound/firewire/fireface/ff-pcm.c | 6 +- sound/firewire/fireface/ff.h | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 6 +- sound/firewire/motu/amdtp-motu.c | 6 +- sound/firewire/motu/motu-pcm.c | 6 +- sound/firewire/motu/motu.h | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 6 +- sound/firewire/tascam/amdtp-tascam.c | 6 +- sound/firewire/tascam/tascam-pcm.c | 2 +- sound/firewire/tascam/tascam.h | 2 +- sound/pci/ac97/ac97_pcm.c | 8 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 6 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 8 +- sound/pci/ca0106/ca0106_main.c | 4 +- sound/pci/cmipci.c | 14 ++-- sound/pci/cs4281.c | 4 +- sound/pci/cs46xx/cs46xx_lib.c | 4 +- sound/pci/echoaudio/echoaudio.c | 24 +++--- sound/pci/emu10k1/emu10k1x.c | 4 +- sound/pci/emu10k1/emupcm.c | 10 +-- sound/pci/ens1370.c | 12 +-- sound/pci/es1938.c | 4 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 6 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.c | 4 +- sound/pci/hda/patch_hdmi.c | 8 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/ice1712/ice1712.c | 8 +- sound/pci/ice1712/ice1724.c | 30 +++---- sound/pci/intel8x0.c | 10 +-- sound/pci/intel8x0m.c | 2 +- sound/pci/lola/lola_pcm.c | 4 +- sound/pci/mixart/mixart.c | 8 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen_pcm.c | 8 +- sound/pci/pcxhr/pcxhr.c | 4 +- sound/pci/rme32.c | 14 ++-- sound/pci/rme96.c | 17 ++-- sound/pci/rme9652/hdsp.c | 24 +++--- sound/pci/rme9652/hdspm.c | 12 +-- sound/pci/rme9652/rme9652.c | 20 ++--- sound/pci/sonicvibes.c | 4 +- sound/pci/via82xx.c | 4 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 4 +- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-da7219-max98357a.c | 40 ++++----- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 4 +- sound/soc/bcm/cygnus-pcm.c | 5 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak4613.c | 6 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 4 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4234.c | 4 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs43130.c | 4 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/pcm512x.c | 6 +- sound/soc/codecs/sigmadsp.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/uda1334.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/fsl/fsl_asrc.c | 4 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 9 +- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 12 +-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 +-- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/aiu-fifo.c | 4 +- sound/soc/meson/axg-fifo.c | 4 +- sound/soc/sh/rcar/core.c | 6 +- sound/soc/soc-pcm.c | 76 ++++++++++++++++- sound/soc/sti/uniperif_player.c | 4 +- sound/soc/sti/uniperif_reader.c | 4 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- sound/soc/ti/davinci-mcasp.c | 12 +-- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/uniphier/aio-dma.c | 4 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/usb/hiface/pcm.c | 2 +- sound/usb/line6/capture.c | 2 +- sound/usb/line6/playback.c | 2 +- sound/usb/misc/ua101.c | 2 +- sound/usb/pcm.c | 24 +++--- 135 files changed, 535 insertions(+), 429 deletions(-) -- 2.27.0

5 20

[PATCH] ALSA: seq: oss: Fix inconsistent indenting
by Yang Li 19 Apr '21

19 Apr '21

Kernel test robot throws below warning -> smatch warnings: sound/core/seq/oss/seq_oss_event.c:297 note_on_event() warn: inconsistent indenting Fixed the inconsistent indenting. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/core/seq/oss/seq_oss_event.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 7b7c925..9a42713 100644 --- a/sound/core/seq/oss/seq_oss_event.c +++ b/sound/core/seq/oss/seq_oss_event.c @@ -294,7 +294,7 @@ /* set volume to zero -- note off */ // type = SNDRV_SEQ_EVENT_NOTEOFF; //else - if (info->ch[ch].vel) + if (info->ch[ch].vel) /* sample already started -- volume change */ type = SNDRV_SEQ_EVENT_KEYPRESS; else -- 1.8.3.1

3 2

[RESEND PATCH] pinctrl: mediatek: use spin lock in mtk_rmw
by Tzung-Bi Shih 19 Apr '21

19 Apr '21

Commit 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") uses mutex lock in mtk_rmw. However the function is possible called from atomic context. For example call trace: mutex_lock+0x28/0x64 mtk_rmw+0x38/0x80 mtk_hw_set_value+0x100/0x138 mtk_gpio_set+0x48/0x58 gpiod_set_raw_value_commit+0xf4/0x110 gpiod_set_value_nocheck+0x4c/0x80 gpiod_set_value+0x4c/0x6c max98357a_daiops_trigger+0x8c/0x9c soc_pcm_trigger+0x5c/0x10c The max98357a_daiops_trigger() could run in either atomic or non-atomic context. As a result, dmesg shows some similar messages: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:254". Uses spin lock in mtk_rmw instead. Fixes: 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- drivers/pinctrl/mediatek/pinctrl-moore.c | 2 +- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c | 4 ++-- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h | 2 +- drivers/pinctrl/mediatek/pinctrl-paris.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/pinctrl/mediatek/pinctrl-moore.c b/drivers/pinctrl/mediatek/pinctrl-moore.c index f77921957f15..3a4a23c40a71 100644 --- a/drivers/pinctrl/mediatek/pinctrl-moore.c +++ b/drivers/pinctrl/mediatek/pinctrl-moore.c @@ -619,7 +619,7 @@ int mtk_moore_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); /* Copy from internal struct mtk_pin_desc to register to the core */ pins = devm_kmalloc_array(&pdev->dev, hw->soc->npins, sizeof(*pins), diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c index fcf7c3eeee4a..9ffe01a8ceca 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c @@ -58,14 +58,14 @@ void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set) { u32 val; - mutex_lock(&pctl->lock); + spin_lock(&pctl->lock); val = mtk_r32(pctl, i, reg); val &= ~mask; val |= set; mtk_w32(pctl, i, reg, val); - mutex_unlock(&pctl->lock); + spin_unlock(&pctl->lock); } static int mtk_hw_pin_field_lookup(struct mtk_pinctrl *hw, diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h index 65eac708a3b3..13836bb188b7 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h @@ -252,7 +252,7 @@ struct mtk_pinctrl { struct mtk_pinctrl_group *groups; const char **grp_names; /* lock pin's register resource to avoid multiple threads issue*/ - struct mutex lock; + spinlock_t lock; }; void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set); diff --git a/drivers/pinctrl/mediatek/pinctrl-paris.c b/drivers/pinctrl/mediatek/pinctrl-paris.c index 48e823f6d293..85db2e4377f0 100644 --- a/drivers/pinctrl/mediatek/pinctrl-paris.c +++ b/drivers/pinctrl/mediatek/pinctrl-paris.c @@ -970,7 +970,7 @@ int mtk_paris_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); err = mtk_pctrl_build_state(pdev); if (err) { -- 2.31.1.368.gbe11c130af-goog

2 2

[PATCH v5] sound: rawmidi: Add framing mode
by David Henningsson 19 Apr '21

19 Apr '21

This commit adds a new framing mode that frames all MIDI data into 32-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Testing on a Celeron N3150 with this mode has a max jitter of 2.8 ms, compared to the in-kernel seq implementation which has a max jitter of 5 ms during idle and much worse when running scheduler stress tests in parallel. Signed-off-by: David Henningsson <coding(a)diwic.se> --- This version of the patch has been compile tested only, and compared to v4, has only the changes with no controversy. include/sound/rawmidi.h | 2 + include/uapi/sound/asound.h | 26 ++++++++++++- sound/core/rawmidi.c | 73 ++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 6 ++- 4 files changed, 103 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..b0057a193c31 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + u8 framing; /* whether to frame input data */ + clockid_t clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..124ac74a13e9 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,34 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +enum { + SNDRV_RAWMIDI_FRAMING_NONE = 0, + SNDRV_RAWMIDI_FRAMING_TSTAMP, + SNDRV_RAWMIDI_FRAMING_LAST = SNDRV_RAWMIDI_FRAMING_TSTAMP, +}; + +#define SND_RAWMIDI_FRAMING_DATA_LENGTH 16 + +struct snd_rawmidi_framing_tstamp { + /* For now, frame_type is always 0. Midi 2.0 is expected to add new + * types here. Applications are expected to skip unknown frame types. + */ + u8 frame_type; + u8 length; /* number of valid bytes in data field */ + u8 reserved[2]; + u32 tv_nsec; /* nanoseconds */ + u64 tv_sec; /* seconds */ + u8 data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; +}; + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned char framing; /* For input data only, frame incoming data */ + unsigned char clock_type; /* Type of clock to use for framing, same as clockid_t */ + unsigned char reserved[14]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..c3a4940a919d 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -683,6 +683,8 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (params->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP && (params->buffer_size & 0x1f) != 0) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +722,16 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + if (params->framing) { + if (params->framing > SNDRV_RAWMIDI_FRAMING_LAST) + return -EINVAL; + /* framing requires a valid clock type */ + if (params->clock_type != CLOCK_MONOTONIC_RAW && params->clock_type != CLOCK_MONOTONIC) + return -EINVAL; + } snd_rawmidi_drain_input(substream); + substream->framing = params->framing; + substream->clock_type = params->clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +974,56 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, const struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp *dest_ptr; + struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec }; + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + BUILD_BUG_ON(frame_size != 0x20); + if (snd_BUG_ON((runtime->hw_ptr & 0x1f) != 0)) + return -EINVAL; + + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + break; + } + if (src_count >= SND_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SND_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SND_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + +struct timespec64 get_framing_tstamp(struct snd_rawmidi_substream *substream) +{ + struct timespec64 ts64 = {0, 0}; + + if (substream->framing != SNDRV_RAWMIDI_FRAMING_TSTAMP) + return ts64; + if (substream->clock_type == CLOCK_MONOTONIC_RAW) + ktime_get_raw_ts64(&ts64); + else + ktime_get_ts64(&ts64); + return ts64; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1038,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1049,11 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } + spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -1596,6 +1661,12 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Avail : %lu\n" " Overruns : %lu\n", buffer_size, avail, xruns); + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + snd_iprintf(buffer, + " Framing : tstamp\n" + " Clock type : %s\n", + substream->clock_type == CLOCK_MONOTONIC_RAW ? "monotonic raw" : "monotonic"); + } } } } diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 7397130976d0..2603d2dd8abb 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -13,7 +13,9 @@ struct snd_rawmidi_params32 { u32 buffer_size; u32 avail_min; unsigned int no_active_sensing; /* avoid bit-field */ - unsigned char reserved[16]; + unsigned char framing; + unsigned char clock_type; + unsigned char reserved[14]; } __attribute__((packed)); static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, @@ -25,6 +27,8 @@ static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, if (get_user(params.stream, &src->stream) || get_user(params.buffer_size, &src->buffer_size) || get_user(params.avail_min, &src->avail_min) || + get_user(params.framing, &src->framing) || + get_user(params.clock_type, &src->clock_type) || get_user(val, &src->no_active_sensing)) return -EFAULT; params.no_active_sensing = val; -- 2.25.1

4 7