| Message ID | 20240314232201.2102178-1-jan.dakinevich@salutedevices.com |
|---|---|
| Headers | show |
| Series | Introduce support of audio for Amlogic A1 SoC family | expand |
On Fri 15 Mar 2024 at 02:21, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > This controller provides clocks and reset functionality for audio > peripherals on Amlogic A1 SoC family. > > The driver is almost identical to 'axg-audio', however it would be better > to keep it separate due to following reasons: > > - significant amount of bits has another definition. I will bring there > a mess of new defines with A1_ suffixes. > > - registers of this controller are located in two separate regions. It > will give a lot of complications for 'axg-audio' to support this. > > Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> > --- > drivers/clk/meson/Kconfig | 13 + > drivers/clk/meson/Makefile | 1 + > drivers/clk/meson/a1-audio.c | 556 +++++++++++++++++++++++++++++++++++ > drivers/clk/meson/a1-audio.h | 58 ++++ > 4 files changed, 628 insertions(+) > create mode 100644 drivers/clk/meson/a1-audio.c > create mode 100644 drivers/clk/meson/a1-audio.h > > diff --git a/drivers/clk/meson/Kconfig b/drivers/clk/meson/Kconfig > index d6a2fa5f7e88..80c4a18c83d2 100644 > --- a/drivers/clk/meson/Kconfig > +++ b/drivers/clk/meson/Kconfig > @@ -133,6 +133,19 @@ config COMMON_CLK_A1_PERIPHERALS > device, A1 SoC Family. Say Y if you want A1 Peripherals clock > controller to work. > > +config COMMON_CLK_A1_AUDIO > + tristate "Amlogic A1 SoC Audio clock controller support" > + depends on ARM64 > + select COMMON_CLK_MESON_REGMAP > + select COMMON_CLK_MESON_CLKC_UTILS > + select COMMON_CLK_MESON_PHASE > + select COMMON_CLK_MESON_SCLK_DIV > + select COMMON_CLK_MESON_AUDIO_RSTC > + help > + Support for the Audio clock controller on Amlogic A113L based > + device, A1 SoC Family. Say Y if you want A1 Audio clock controller > + to work. > + > config COMMON_CLK_G12A > tristate "G12 and SM1 SoC clock controllers support" > depends on ARM64 > diff --git a/drivers/clk/meson/Makefile b/drivers/clk/meson/Makefile > index 88d94921a4dc..4968fc7ad555 100644 > --- a/drivers/clk/meson/Makefile > +++ b/drivers/clk/meson/Makefile > @@ -20,6 +20,7 @@ obj-$(CONFIG_COMMON_CLK_AXG) += axg.o axg-aoclk.o > obj-$(CONFIG_COMMON_CLK_AXG_AUDIO) += axg-audio.o > obj-$(CONFIG_COMMON_CLK_A1_PLL) += a1-pll.o > obj-$(CONFIG_COMMON_CLK_A1_PERIPHERALS) += a1-peripherals.o > +obj-$(CONFIG_COMMON_CLK_A1_AUDIO) += a1-audio.o > obj-$(CONFIG_COMMON_CLK_GXBB) += gxbb.o gxbb-aoclk.o > obj-$(CONFIG_COMMON_CLK_G12A) += g12a.o g12a-aoclk.o > obj-$(CONFIG_COMMON_CLK_MESON8B) += meson8b.o meson8-ddr.o > diff --git a/drivers/clk/meson/a1-audio.c b/drivers/clk/meson/a1-audio.c > new file mode 100644 > index 000000000000..6039116c93ba > --- /dev/null > +++ b/drivers/clk/meson/a1-audio.c > @@ -0,0 +1,556 @@ > +// SPDX-License-Identifier: (GPL-2.0 OR MIT) > +/* > + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. > + * > + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> > + */ > + > +#include <linux/clk.h> > +#include <linux/clk-provider.h> > +#include <linux/init.h> > +#include <linux/of_device.h> > +#include <linux/module.h> > +#include <linux/platform_device.h> > +#include <linux/regmap.h> > +#include <linux/reset.h> > +#include <linux/reset-controller.h> > +#include <linux/slab.h> > + > +#include "meson-clkc-utils.h" > +#include "meson-audio-rstc.h" > +#include "clk-regmap.h" > +#include "clk-phase.h" > +#include "sclk-div.h" > +#include "a1-audio.h" > + > +#define AUDIO_PDATA(_name) \ > + ((const struct clk_parent_data[]) { { .hw = &(_name).hw } }) Not a fan - yet another level of macro. > + > +#define AUDIO_MUX(_name, _reg, _mask, _shift, _pdata) \ > +static struct clk_regmap _name = { \ > + .map = AUDIO_REG_MAP(_reg), \ > + .data = &(struct clk_regmap_mux_data){ \ > + .offset = AUDIO_REG_OFFSET(_reg), \ > + .mask = (_mask), \ > + .shift = (_shift), \ > + }, \ > + .hw.init = &(struct clk_init_data) { \ > + .name = #_name, \ > + .ops = &clk_regmap_mux_ops, \ > + .parent_data = (_pdata), \ > + .num_parents = ARRAY_SIZE(_pdata), \ > + .flags = CLK_SET_RATE_PARENT, \ > + }, \ > +} > + > +#define AUDIO_DIV(_name, _reg, _shift, _width, _pdata) \ > +static struct clk_regmap _name = { \ > + .map = AUDIO_REG_MAP(_reg), \ > + .data = &(struct clk_regmap_div_data){ \ > + .offset = AUDIO_REG_OFFSET(_reg), \ > + .shift = (_shift), \ > + .width = (_width), \ > + }, \ > + .hw.init = &(struct clk_init_data) { \ > + .name = #_name, \ > + .ops = &clk_regmap_divider_ops, \ > + .parent_data = (_pdata), \ > + .num_parents = 1, \ > + .flags = CLK_SET_RATE_PARENT, \ > + }, \ > +} > + > +#define AUDIO_GATE(_name, _reg, _bit, _pdata) \ > +static struct clk_regmap _name = { \ > + .map = AUDIO_REG_MAP(_reg), \ > + .data = &(struct clk_regmap_gate_data){ \ > + .offset = AUDIO_REG_OFFSET(_reg), \ > + .bit_idx = (_bit), \ > + }, \ > + .hw.init = &(struct clk_init_data) { \ > + .name = #_name, \ > + .ops = &clk_regmap_gate_ops, \ > + .parent_data = (_pdata), \ > + .num_parents = 1, \ > + .flags = CLK_SET_RATE_PARENT, \ > + }, \ > +} > + > +#define AUDIO_SCLK_DIV(_name, _reg, _div_shift, _div_width, \ > + _hi_shift, _hi_width, _pdata, _set_rate_parent) \ > +static struct clk_regmap _name = { \ > + .map = AUDIO_REG_MAP(_reg), \ > + .data = &(struct meson_sclk_div_data) { \ > + .div = { \ > + .reg_off = AUDIO_REG_OFFSET(_reg), \ > + .shift = (_div_shift), \ > + .width = (_div_width), \ > + }, \ > + .hi = { \ > + .reg_off = AUDIO_REG_OFFSET(_reg), \ > + .shift = (_hi_shift), \ > + .width = (_hi_width), \ > + }, \ > + }, \ > + .hw.init = &(struct clk_init_data) { \ > + .name = #_name, \ > + .ops = &meson_sclk_div_ops, \ > + .parent_data = (_pdata), \ > + .num_parents = 1, \ > + .flags = (_set_rate_parent) ? CLK_SET_RATE_PARENT : 0, \ Does not help readeability. Just pass the flag as axg-audio does. > + }, \ > +} > + > +#define AUDIO_TRIPHASE(_name, _reg, _width, _shift0, _shift1, _shift2, \ > + _pdata) \ > +static struct clk_regmap _name = { \ > + .map = AUDIO_REG_MAP(_reg), \ > + .data = &(struct meson_clk_triphase_data) { \ > + .ph0 = { \ > + .reg_off = AUDIO_REG_OFFSET(_reg), \ > + .shift = (_shift0), \ > + .width = (_width), \ > + }, \ > + .ph1 = { \ > + .reg_off = AUDIO_REG_OFFSET(_reg), \ > + .shift = (_shift1), \ > + .width = (_width), \ > + }, \ > + .ph2 = { \ > + .reg_off = AUDIO_REG_OFFSET(_reg), \ > + .shift = (_shift2), \ > + .width = (_width), \ > + }, \ > + }, \ > + .hw.init = &(struct clk_init_data) { \ > + .name = #_name, \ > + .ops = &meson_clk_triphase_ops, \ > + .parent_data = (_pdata), \ > + .num_parents = 1, \ > + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ > + }, \ > +} > + > +#define AUDIO_SCLK_WS(_name, _reg, _width, _shift_ph, _shift_ws, \ > + _pdata) \ > +static struct clk_regmap _name = { \ > + .map = AUDIO_REG_MAP(_reg), \ > + .data = &(struct meson_sclk_ws_inv_data) { \ > + .ph = { \ > + .reg_off = AUDIO_REG_OFFSET(_reg), \ > + .shift = (_shift_ph), \ > + .width = (_width), \ > + }, \ > + .ws = { \ > + .reg_off = AUDIO_REG_OFFSET(_reg), \ > + .shift = (_shift_ws), \ > + .width = (_width), \ > + }, \ > + }, \ > + .hw.init = &(struct clk_init_data) { \ > + .name = #_name, \ > + .ops = &meson_sclk_ws_inv_ops, \ > + .parent_data = (_pdata), \ > + .num_parents = 1, \ > + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ > + }, \ > +} All the above does essentially the same things as the macro of axg-audio, to some minor differences. Yet it is another set to maintain. I'd much prefer if you put the axg-audio macro in a header a re-used those. There would a single set to maintain. You may then specialize the included in the driver C file, to avoid redundant parameters Rework axg-audio to use clk_parent_data if you must, but not in the same series please. > + > +static const struct clk_parent_data a1_pclk_pdata[] = { > + { .fw_name = "pclk", }, > +}; > + > +AUDIO_GATE(audio_ddr_arb, AUDIO_CLK_GATE_EN0, 0, a1_pclk_pdata); > +AUDIO_GATE(audio_tdmin_a, AUDIO_CLK_GATE_EN0, 1, a1_pclk_pdata); > +AUDIO_GATE(audio_tdmin_b, AUDIO_CLK_GATE_EN0, 2, a1_pclk_pdata); > +AUDIO_GATE(audio_tdmin_lb, AUDIO_CLK_GATE_EN0, 3, a1_pclk_pdata); > +AUDIO_GATE(audio_loopback, AUDIO_CLK_GATE_EN0, 4, a1_pclk_pdata); > +AUDIO_GATE(audio_tdmout_a, AUDIO_CLK_GATE_EN0, 5, a1_pclk_pdata); > +AUDIO_GATE(audio_tdmout_b, AUDIO_CLK_GATE_EN0, 6, a1_pclk_pdata); > +AUDIO_GATE(audio_frddr_a, AUDIO_CLK_GATE_EN0, 7, a1_pclk_pdata); > +AUDIO_GATE(audio_frddr_b, AUDIO_CLK_GATE_EN0, 8, a1_pclk_pdata); > +AUDIO_GATE(audio_toddr_a, AUDIO_CLK_GATE_EN0, 9, a1_pclk_pdata); > +AUDIO_GATE(audio_toddr_b, AUDIO_CLK_GATE_EN0, 10, a1_pclk_pdata); > +AUDIO_GATE(audio_spdifin, AUDIO_CLK_GATE_EN0, 11, a1_pclk_pdata); > +AUDIO_GATE(audio_resample, AUDIO_CLK_GATE_EN0, 12, a1_pclk_pdata); > +AUDIO_GATE(audio_eqdrc, AUDIO_CLK_GATE_EN0, 13, a1_pclk_pdata); > +AUDIO_GATE(audio_audiolocker, AUDIO_CLK_GATE_EN0, 14, a1_pclk_pdata); This is what I mean by redundant parameter ^ > + > +AUDIO_GATE(audio2_ddr_arb, AUDIO2_CLK_GATE_EN0, 0, a1_pclk_pdata); > +AUDIO_GATE(audio2_pdm, AUDIO2_CLK_GATE_EN0, 1, a1_pclk_pdata); > +AUDIO_GATE(audio2_tdmin_vad, AUDIO2_CLK_GATE_EN0, 2, a1_pclk_pdata); > +AUDIO_GATE(audio2_toddr_vad, AUDIO2_CLK_GATE_EN0, 3, a1_pclk_pdata); > +AUDIO_GATE(audio2_vad, AUDIO2_CLK_GATE_EN0, 4, a1_pclk_pdata); > +AUDIO_GATE(audio2_audiotop, AUDIO2_CLK_GATE_EN0, 7, a1_pclk_pdata); > + > +static const struct clk_parent_data a1_mst_pdata[] = { > + { .fw_name = "dds_in" }, > + { .fw_name = "fclk_div2" }, > + { .fw_name = "fclk_div3" }, > + { .fw_name = "hifi_pll" }, > + { .fw_name = "xtal" }, > +}; > + > +#define AUDIO_MST_MCLK(_name, _reg) \ > + AUDIO_MUX(_name##_mux, (_reg), 0x7, 24, a1_mst_pdata); \ > + AUDIO_DIV(_name##_div, (_reg), 0, 16, \ > + AUDIO_PDATA(_name##_mux)); \ > + AUDIO_GATE(_name, (_reg), 31, AUDIO_PDATA(_name##_div)) > + > +AUDIO_MST_MCLK(audio_mst_a_mclk, AUDIO_MCLK_A_CTRL); > +AUDIO_MST_MCLK(audio_mst_b_mclk, AUDIO_MCLK_B_CTRL); > +AUDIO_MST_MCLK(audio_mst_c_mclk, AUDIO_MCLK_C_CTRL); > +AUDIO_MST_MCLK(audio_mst_d_mclk, AUDIO_MCLK_D_CTRL); > +AUDIO_MST_MCLK(audio_spdifin_clk, AUDIO_CLK_SPDIFIN_CTRL); > +AUDIO_MST_MCLK(audio_eqdrc_clk, AUDIO_CLK_EQDRC_CTRL); > + > +AUDIO_MUX(audio_resample_clk_mux, AUDIO_CLK_RESAMPLE_CTRL, 0xf, 24, > + a1_mst_pdata); > +AUDIO_DIV(audio_resample_clk_div, AUDIO_CLK_RESAMPLE_CTRL, 0, 8, > + AUDIO_PDATA(audio_resample_clk_mux)); > +AUDIO_GATE(audio_resample_clk, AUDIO_CLK_RESAMPLE_CTRL, 31, > + AUDIO_PDATA(audio_resample_clk_div)); > + > +AUDIO_MUX(audio_locker_in_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 8, > + a1_mst_pdata); > +AUDIO_DIV(audio_locker_in_clk_div, AUDIO_CLK_LOCKER_CTRL, 0, 8, > + AUDIO_PDATA(audio_locker_in_clk_mux)); > +AUDIO_GATE(audio_locker_in_clk, AUDIO_CLK_LOCKER_CTRL, 15, > + AUDIO_PDATA(audio_locker_in_clk_div)); > + > +AUDIO_MUX(audio_locker_out_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 24, > + a1_mst_pdata); > +AUDIO_DIV(audio_locker_out_clk_div, AUDIO_CLK_LOCKER_CTRL, 16, 8, > + AUDIO_PDATA(audio_locker_out_clk_mux)); > +AUDIO_GATE(audio_locker_out_clk, AUDIO_CLK_LOCKER_CTRL, 31, > + AUDIO_PDATA(audio_locker_out_clk_div)); > + > +AUDIO_MST_MCLK(audio2_vad_mclk, AUDIO2_MCLK_VAD_CTRL); > +AUDIO_MST_MCLK(audio2_vad_clk, AUDIO2_CLK_VAD_CTRL); > +AUDIO_MST_MCLK(audio2_pdm_dclk, AUDIO2_CLK_PDMIN_CTRL0); > +AUDIO_MST_MCLK(audio2_pdm_sysclk, AUDIO2_CLK_PDMIN_CTRL1); > + > +#define AUDIO_MST_SCLK(_name, _reg0, _reg1, _pdata) \ > + AUDIO_GATE(_name##_pre_en, (_reg0), 31, (_pdata)); \ > + AUDIO_SCLK_DIV(_name##_div, (_reg0), 20, 10, 0, 0, \ > + AUDIO_PDATA(_name##_pre_en), true); \ > + AUDIO_GATE(_name##_post_en, (_reg0), 30, \ > + AUDIO_PDATA(_name##_div)); \ > + AUDIO_TRIPHASE(_name, (_reg1), 1, 0, 2, 4, \ > + AUDIO_PDATA(_name##_post_en)) > + Again, I'm not a fan of this many levels of macro. I can live with it but certainly don't want the burden of reviewing and maintaining for clock driver. AXG / G12 and A1 are obviously closely related, so make it common. > +#define AUDIO_MST_LRCLK(_name, _reg0, _reg1, _pdata) \ > + AUDIO_SCLK_DIV(_name##_div, (_reg0), 0, 10, 10, 10, \ > + (_pdata), false); \ > + AUDIO_TRIPHASE(_name, (_reg1), 1, 1, 3, 5, \ > + AUDIO_PDATA(_name##_div)) > + > +AUDIO_MST_SCLK(audio_mst_a_sclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, > + AUDIO_PDATA(audio_mst_a_mclk)); > +AUDIO_MST_SCLK(audio_mst_b_sclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, > + AUDIO_PDATA(audio_mst_b_mclk)); > +AUDIO_MST_SCLK(audio_mst_c_sclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, > + AUDIO_PDATA(audio_mst_c_mclk)); > +AUDIO_MST_SCLK(audio_mst_d_sclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, > + AUDIO_PDATA(audio_mst_d_mclk)); > + > +AUDIO_MST_LRCLK(audio_mst_a_lrclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, > + AUDIO_PDATA(audio_mst_a_sclk_post_en)); > +AUDIO_MST_LRCLK(audio_mst_b_lrclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, > + AUDIO_PDATA(audio_mst_b_sclk_post_en)); > +AUDIO_MST_LRCLK(audio_mst_c_lrclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, > + AUDIO_PDATA(audio_mst_c_sclk_post_en)); > +AUDIO_MST_LRCLK(audio_mst_d_lrclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, > + AUDIO_PDATA(audio_mst_d_sclk_post_en)); > + > +static const struct clk_parent_data a1_mst_sclk_pdata[] = { > + { .hw = &audio_mst_a_sclk.hw }, > + { .hw = &audio_mst_b_sclk.hw }, > + { .hw = &audio_mst_c_sclk.hw }, > + { .hw = &audio_mst_d_sclk.hw }, > + { .fw_name = "slv_sclk0" }, > + { .fw_name = "slv_sclk1" }, > + { .fw_name = "slv_sclk2" }, > + { .fw_name = "slv_sclk3" }, > + { .fw_name = "slv_sclk4" }, > + { .fw_name = "slv_sclk5" }, > + { .fw_name = "slv_sclk6" }, > + { .fw_name = "slv_sclk7" }, > + { .fw_name = "slv_sclk8" }, > + { .fw_name = "slv_sclk9" }, > +}; > + > +static const struct clk_parent_data a1_mst_lrclk_pdata[] = { > + { .hw = &audio_mst_a_lrclk.hw }, > + { .hw = &audio_mst_b_lrclk.hw }, > + { .hw = &audio_mst_c_lrclk.hw }, > + { .hw = &audio_mst_d_lrclk.hw }, > + { .fw_name = "slv_lrclk0" }, > + { .fw_name = "slv_lrclk1" }, > + { .fw_name = "slv_lrclk2" }, > + { .fw_name = "slv_lrclk3" }, > + { .fw_name = "slv_lrclk4" }, > + { .fw_name = "slv_lrclk5" }, > + { .fw_name = "slv_lrclk6" }, > + { .fw_name = "slv_lrclk7" }, > + { .fw_name = "slv_lrclk8" }, > + { .fw_name = "slv_lrclk9" }, > +}; > + > +#define AUDIO_TDM_SCLK(_name, _reg) \ > + AUDIO_MUX(_name##_mux, (_reg), 0xf, 24, a1_mst_sclk_pdata); \ > + AUDIO_GATE(_name##_pre_en, (_reg), 31, \ > + AUDIO_PDATA(_name##_mux)); \ > + AUDIO_GATE(_name##_post_en, (_reg), 30, \ > + AUDIO_PDATA(_name##_pre_en)); \ > + AUDIO_SCLK_WS(_name, (_reg), 1, 29, 28, \ > + AUDIO_PDATA(_name##_post_en)) > + > +#define AUDIO_TDM_LRCLK(_name, _reg) \ > + AUDIO_MUX(_name, (_reg), 0xf, 20, a1_mst_lrclk_pdata) > + > +AUDIO_TDM_SCLK(audio_tdmin_a_sclk, AUDIO_CLK_TDMIN_A_CTRL); > +AUDIO_TDM_SCLK(audio_tdmin_b_sclk, AUDIO_CLK_TDMIN_B_CTRL); > +AUDIO_TDM_SCLK(audio_tdmin_lb_sclk, AUDIO_CLK_TDMIN_LB_CTRL); > +AUDIO_TDM_SCLK(audio_tdmout_a_sclk, AUDIO_CLK_TDMOUT_A_CTRL); > +AUDIO_TDM_SCLK(audio_tdmout_b_sclk, AUDIO_CLK_TDMOUT_B_CTRL); > + > +AUDIO_TDM_LRCLK(audio_tdmin_a_lrclk, AUDIO_CLK_TDMIN_A_CTRL); > +AUDIO_TDM_LRCLK(audio_tdmin_b_lrclk, AUDIO_CLK_TDMIN_B_CTRL); > +AUDIO_TDM_LRCLK(audio_tdmin_lb_lrclk, AUDIO_CLK_TDMIN_LB_CTRL); > +AUDIO_TDM_LRCLK(audio_tdmout_a_lrclk, AUDIO_CLK_TDMOUT_A_CTRL); > +AUDIO_TDM_LRCLK(audio_tdmout_b_lrclk, AUDIO_CLK_TDMOUT_B_CTRL); > + > +static struct clk_hw *a1_audio_hw_clks[] = { > + [AUD_CLKID_DDR_ARB] = &audio_ddr_arb.hw, > + [AUD_CLKID_TDMIN_A] = &audio_tdmin_a.hw, > + [AUD_CLKID_TDMIN_B] = &audio_tdmin_b.hw, > + [AUD_CLKID_TDMIN_LB] = &audio_tdmin_lb.hw, > + [AUD_CLKID_LOOPBACK] = &audio_loopback.hw, > + [AUD_CLKID_TDMOUT_A] = &audio_tdmout_a.hw, > + [AUD_CLKID_TDMOUT_B] = &audio_tdmout_b.hw, > + [AUD_CLKID_FRDDR_A] = &audio_frddr_a.hw, > + [AUD_CLKID_FRDDR_B] = &audio_frddr_b.hw, > + [AUD_CLKID_TODDR_A] = &audio_toddr_a.hw, > + [AUD_CLKID_TODDR_B] = &audio_toddr_b.hw, > + [AUD_CLKID_SPDIFIN] = &audio_spdifin.hw, > + [AUD_CLKID_RESAMPLE] = &audio_resample.hw, > + [AUD_CLKID_EQDRC] = &audio_eqdrc.hw, > + [AUD_CLKID_LOCKER] = &audio_audiolocker.hw, > + [AUD_CLKID_MST_A_MCLK_SEL] = &audio_mst_a_mclk_mux.hw, > + [AUD_CLKID_MST_A_MCLK_DIV] = &audio_mst_a_mclk_div.hw, > + [AUD_CLKID_MST_A_MCLK] = &audio_mst_a_mclk.hw, > + [AUD_CLKID_MST_B_MCLK_SEL] = &audio_mst_b_mclk_mux.hw, > + [AUD_CLKID_MST_B_MCLK_DIV] = &audio_mst_b_mclk_div.hw, > + [AUD_CLKID_MST_B_MCLK] = &audio_mst_b_mclk.hw, > + [AUD_CLKID_MST_C_MCLK_SEL] = &audio_mst_c_mclk_mux.hw, > + [AUD_CLKID_MST_C_MCLK_DIV] = &audio_mst_c_mclk_div.hw, > + [AUD_CLKID_MST_C_MCLK] = &audio_mst_c_mclk.hw, > + [AUD_CLKID_MST_D_MCLK_SEL] = &audio_mst_d_mclk_mux.hw, > + [AUD_CLKID_MST_D_MCLK_DIV] = &audio_mst_d_mclk_div.hw, > + [AUD_CLKID_MST_D_MCLK] = &audio_mst_d_mclk.hw, > + [AUD_CLKID_RESAMPLE_CLK_SEL] = &audio_resample_clk_mux.hw, > + [AUD_CLKID_RESAMPLE_CLK_DIV] = &audio_resample_clk_div.hw, > + [AUD_CLKID_RESAMPLE_CLK] = &audio_resample_clk.hw, > + [AUD_CLKID_LOCKER_IN_CLK_SEL] = &audio_locker_in_clk_mux.hw, > + [AUD_CLKID_LOCKER_IN_CLK_DIV] = &audio_locker_in_clk_div.hw, > + [AUD_CLKID_LOCKER_IN_CLK] = &audio_locker_in_clk.hw, > + [AUD_CLKID_LOCKER_OUT_CLK_SEL] = &audio_locker_out_clk_mux.hw, > + [AUD_CLKID_LOCKER_OUT_CLK_DIV] = &audio_locker_out_clk_div.hw, > + [AUD_CLKID_LOCKER_OUT_CLK] = &audio_locker_out_clk.hw, > + [AUD_CLKID_SPDIFIN_CLK_SEL] = &audio_spdifin_clk_mux.hw, > + [AUD_CLKID_SPDIFIN_CLK_DIV] = &audio_spdifin_clk_div.hw, > + [AUD_CLKID_SPDIFIN_CLK] = &audio_spdifin_clk.hw, > + [AUD_CLKID_EQDRC_CLK_SEL] = &audio_eqdrc_clk_mux.hw, > + [AUD_CLKID_EQDRC_CLK_DIV] = &audio_eqdrc_clk_div.hw, > + [AUD_CLKID_EQDRC_CLK] = &audio_eqdrc_clk.hw, > + [AUD_CLKID_MST_A_SCLK_PRE_EN] = &audio_mst_a_sclk_pre_en.hw, > + [AUD_CLKID_MST_A_SCLK_DIV] = &audio_mst_a_sclk_div.hw, > + [AUD_CLKID_MST_A_SCLK_POST_EN] = &audio_mst_a_sclk_post_en.hw, > + [AUD_CLKID_MST_A_SCLK] = &audio_mst_a_sclk.hw, > + [AUD_CLKID_MST_B_SCLK_PRE_EN] = &audio_mst_b_sclk_pre_en.hw, > + [AUD_CLKID_MST_B_SCLK_DIV] = &audio_mst_b_sclk_div.hw, > + [AUD_CLKID_MST_B_SCLK_POST_EN] = &audio_mst_b_sclk_post_en.hw, > + [AUD_CLKID_MST_B_SCLK] = &audio_mst_b_sclk.hw, > + [AUD_CLKID_MST_C_SCLK_PRE_EN] = &audio_mst_c_sclk_pre_en.hw, > + [AUD_CLKID_MST_C_SCLK_DIV] = &audio_mst_c_sclk_div.hw, > + [AUD_CLKID_MST_C_SCLK_POST_EN] = &audio_mst_c_sclk_post_en.hw, > + [AUD_CLKID_MST_C_SCLK] = &audio_mst_c_sclk.hw, > + [AUD_CLKID_MST_D_SCLK_PRE_EN] = &audio_mst_d_sclk_pre_en.hw, > + [AUD_CLKID_MST_D_SCLK_DIV] = &audio_mst_d_sclk_div.hw, > + [AUD_CLKID_MST_D_SCLK_POST_EN] = &audio_mst_d_sclk_post_en.hw, > + [AUD_CLKID_MST_D_SCLK] = &audio_mst_d_sclk.hw, > + [AUD_CLKID_MST_A_LRCLK_DIV] = &audio_mst_a_lrclk_div.hw, > + [AUD_CLKID_MST_A_LRCLK] = &audio_mst_a_lrclk.hw, > + [AUD_CLKID_MST_B_LRCLK_DIV] = &audio_mst_b_lrclk_div.hw, > + [AUD_CLKID_MST_B_LRCLK] = &audio_mst_b_lrclk.hw, > + [AUD_CLKID_MST_C_LRCLK_DIV] = &audio_mst_c_lrclk_div.hw, > + [AUD_CLKID_MST_C_LRCLK] = &audio_mst_c_lrclk.hw, > + [AUD_CLKID_MST_D_LRCLK_DIV] = &audio_mst_d_lrclk_div.hw, > + [AUD_CLKID_MST_D_LRCLK] = &audio_mst_d_lrclk.hw, > + [AUD_CLKID_TDMIN_A_SCLK_SEL] = &audio_tdmin_a_sclk_mux.hw, > + [AUD_CLKID_TDMIN_A_SCLK_PRE_EN] = &audio_tdmin_a_sclk_pre_en.hw, > + [AUD_CLKID_TDMIN_A_SCLK_POST_EN] = &audio_tdmin_a_sclk_post_en.hw, > + [AUD_CLKID_TDMIN_A_SCLK] = &audio_tdmin_a_sclk.hw, > + [AUD_CLKID_TDMIN_A_LRCLK] = &audio_tdmin_a_lrclk.hw, > + [AUD_CLKID_TDMIN_B_SCLK_SEL] = &audio_tdmin_b_sclk_mux.hw, > + [AUD_CLKID_TDMIN_B_SCLK_PRE_EN] = &audio_tdmin_b_sclk_pre_en.hw, > + [AUD_CLKID_TDMIN_B_SCLK_POST_EN] = &audio_tdmin_b_sclk_post_en.hw, > + [AUD_CLKID_TDMIN_B_SCLK] = &audio_tdmin_b_sclk.hw, > + [AUD_CLKID_TDMIN_B_LRCLK] = &audio_tdmin_b_lrclk.hw, > + [AUD_CLKID_TDMIN_LB_SCLK_SEL] = &audio_tdmin_lb_sclk_mux.hw, > + [AUD_CLKID_TDMIN_LB_SCLK_PRE_EN] = &audio_tdmin_lb_sclk_pre_en.hw, > + [AUD_CLKID_TDMIN_LB_SCLK_POST_EN] = &audio_tdmin_lb_sclk_post_en.hw, > + [AUD_CLKID_TDMIN_LB_SCLK] = &audio_tdmin_lb_sclk.hw, > + [AUD_CLKID_TDMIN_LB_LRCLK] = &audio_tdmin_lb_lrclk.hw, > + [AUD_CLKID_TDMOUT_A_SCLK_SEL] = &audio_tdmout_a_sclk_mux.hw, > + [AUD_CLKID_TDMOUT_A_SCLK_PRE_EN] = &audio_tdmout_a_sclk_pre_en.hw, > + [AUD_CLKID_TDMOUT_A_SCLK_POST_EN] = &audio_tdmout_a_sclk_post_en.hw, > + [AUD_CLKID_TDMOUT_A_SCLK] = &audio_tdmout_a_sclk.hw, > + [AUD_CLKID_TDMOUT_A_LRCLK] = &audio_tdmout_a_lrclk.hw, > + [AUD_CLKID_TDMOUT_B_SCLK_SEL] = &audio_tdmout_b_sclk_mux.hw, > + [AUD_CLKID_TDMOUT_B_SCLK_PRE_EN] = &audio_tdmout_b_sclk_pre_en.hw, > + [AUD_CLKID_TDMOUT_B_SCLK_POST_EN] = &audio_tdmout_b_sclk_post_en.hw, > + [AUD_CLKID_TDMOUT_B_SCLK] = &audio_tdmout_b_sclk.hw, > + [AUD_CLKID_TDMOUT_B_LRCLK] = &audio_tdmout_b_lrclk.hw, > + > + [AUD2_CLKID_DDR_ARB] = &audio2_ddr_arb.hw, > + [AUD2_CLKID_PDM] = &audio2_pdm.hw, > + [AUD2_CLKID_TDMIN_VAD] = &audio2_tdmin_vad.hw, > + [AUD2_CLKID_TODDR_VAD] = &audio2_toddr_vad.hw, > + [AUD2_CLKID_VAD] = &audio2_vad.hw, > + [AUD2_CLKID_AUDIOTOP] = &audio2_audiotop.hw, > + [AUD2_CLKID_VAD_MCLK_SEL] = &audio2_vad_mclk_mux.hw, > + [AUD2_CLKID_VAD_MCLK_DIV] = &audio2_vad_mclk_div.hw, > + [AUD2_CLKID_VAD_MCLK] = &audio2_vad_mclk.hw, > + [AUD2_CLKID_VAD_CLK_SEL] = &audio2_vad_clk_mux.hw, > + [AUD2_CLKID_VAD_CLK_DIV] = &audio2_vad_clk_div.hw, > + [AUD2_CLKID_VAD_CLK] = &audio2_vad_clk.hw, > + [AUD2_CLKID_PDM_DCLK_SEL] = &audio2_pdm_dclk_mux.hw, > + [AUD2_CLKID_PDM_DCLK_DIV] = &audio2_pdm_dclk_div.hw, > + [AUD2_CLKID_PDM_DCLK] = &audio2_pdm_dclk.hw, > + [AUD2_CLKID_PDM_SYSCLK_SEL] = &audio2_pdm_sysclk_mux.hw, > + [AUD2_CLKID_PDM_SYSCLK_DIV] = &audio2_pdm_sysclk_div.hw, > + [AUD2_CLKID_PDM_SYSCLK] = &audio2_pdm_sysclk.hw, > +}; > + > +static struct meson_clk_hw_data a1_audio_clks = { > + .hws = a1_audio_hw_clks, > + .num = ARRAY_SIZE(a1_audio_hw_clks), > +}; > + > +static struct regmap *a1_audio_map(struct platform_device *pdev, > + unsigned int index) > +{ > + char name[32]; > + const struct regmap_config cfg = { > + .reg_bits = 32, > + .val_bits = 32, > + .reg_stride = 4, > + .name = name, Not necessary > + }; > + void __iomem *base; > + > + base = devm_platform_ioremap_resource(pdev, index); > + if (IS_ERR(base)) > + return base; > + > + scnprintf(name, sizeof(name), "%d", index); > + return devm_regmap_init_mmio(&pdev->dev, base, &cfg); > +} That is overengineered. Please keep it simple. Declare the regmap_config as static const global, and do it like axg-audio please. > + > +static int a1_register_clk(struct platform_device *pdev, > + struct regmap *map0, struct regmap *map1, > + struct clk_hw *hw) > +{ > + struct clk_regmap *clk = container_of(hw, struct clk_regmap, hw); > + > + if (!hw) > + return 0; > + > + switch ((unsigned long)clk->map) { > + case AUDIO_RANGE_0: > + clk->map = map0; > + break; > + case AUDIO_RANGE_1: > + clk->map = map1; > + break; ... fishy > + default: > + WARN_ON(1); > + return -EINVAL; > + } > + > + return devm_clk_hw_register(&pdev->dev, hw); > +} > + > +static int a1_audio_clkc_probe(struct platform_device *pdev) > +{ > + struct regmap *map0, *map1; > + struct clk *clk; > + unsigned int i; > + int ret; > + > + clk = devm_clk_get_enabled(&pdev->dev, "pclk"); > + if (WARN_ON(IS_ERR(clk))) > + return PTR_ERR(clk); > + > + map0 = a1_audio_map(pdev, 0); > + if (IS_ERR(map0)) > + return PTR_ERR(map0); > + > + map1 = a1_audio_map(pdev, 1); > + if (IS_ERR(map1)) > + return PTR_ERR(map1); No - Looks to me you just have two clock controllers you are trying force into one. > + > + /* > + * Register and enable AUD2_CLKID_AUDIOTOP clock first. Unless > + * it is enabled any read/write to 'map0' hangs the CPU. > + */ > + > + ret = a1_register_clk(pdev, map0, map1, > + a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]); > + if (ret) > + return ret; > + > + ret = clk_prepare_enable(a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]->clk); > + if (ret) > + return ret; Again, this shows 2 devices. The one related to your 'map0' should request AUD2_CLKID_AUDIOTOP as input and enable it right away. > + > + for (i = 0; i < a1_audio_clks.num; i++) { > + if (i == AUD2_CLKID_AUDIOTOP) > + continue; > + > + ret = a1_register_clk(pdev, map0, map1, a1_audio_clks.hws[i]); > + if (ret) > + return ret; > + } > + > + ret = devm_of_clk_add_hw_provider(&pdev->dev, meson_clk_hw_get, > + &a1_audio_clks); > + if (ret) > + return ret; > + > + BUILD_BUG_ON((unsigned long)AUDIO_REG_MAP(AUDIO_SW_RESET0) != > + AUDIO_RANGE_0); Why is that necessary ? > + return meson_audio_rstc_register(&pdev->dev, map0, > + AUDIO_REG_OFFSET(AUDIO_SW_RESET0), 32); > +} > + > +static const struct of_device_id a1_audio_clkc_match_table[] = { > + { .compatible = "amlogic,a1-audio-clkc", }, > + {} > +}; > +MODULE_DEVICE_TABLE(of, a1_audio_clkc_match_table); > + > +static struct platform_driver a1_audio_clkc_driver = { > + .probe = a1_audio_clkc_probe, > + .driver = { > + .name = "a1-audio-clkc", > + .of_match_table = a1_audio_clkc_match_table, > + }, > +}; > +module_platform_driver(a1_audio_clkc_driver); > + > +MODULE_DESCRIPTION("Amlogic A1 Audio Clock driver"); > +MODULE_AUTHOR("Jan Dakinevich <jan.dakinevich@salutedevices.com>"); > +MODULE_LICENSE("GPL"); > diff --git a/drivers/clk/meson/a1-audio.h b/drivers/clk/meson/a1-audio.h > new file mode 100644 > index 000000000000..f994e87276cd > --- /dev/null > +++ b/drivers/clk/meson/a1-audio.h > @@ -0,0 +1,58 @@ > +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ > +/* > + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. > + * > + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> > + */ > + > +#ifndef __A1_AUDIO_H > +#define __A1_AUDIO_H > + > +#define AUDIO_RANGE_0 0xa > +#define AUDIO_RANGE_1 0xb > +#define AUDIO_RANGE_SHIFT 16 > + > +#define AUDIO_REG(_range, _offset) \ > + (((_range) << AUDIO_RANGE_SHIFT) + (_offset)) > + > +#define AUDIO_REG_OFFSET(_reg) \ > + ((_reg) & ((1 << AUDIO_RANGE_SHIFT) - 1)) > + > +#define AUDIO_REG_MAP(_reg) \ > + ((void *)((_reg) >> AUDIO_RANGE_SHIFT)) That is seriouly overengineered. The following are offset. Just write what they are. There is not reason to put that into a header. It is only going to be used by a single driver. > + > +#define AUDIO_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_0, 0x000) > +#define AUDIO_MCLK_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x008) > +#define AUDIO_MCLK_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x00c) > +#define AUDIO_MCLK_C_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x010) > +#define AUDIO_MCLK_D_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x014) > +#define AUDIO_MCLK_E_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x018) > +#define AUDIO_MCLK_F_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x01c) > +#define AUDIO_SW_RESET0 AUDIO_REG(AUDIO_RANGE_0, 0x028) > +#define AUDIO_MST_A_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x040) > +#define AUDIO_MST_A_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x044) > +#define AUDIO_MST_B_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x048) > +#define AUDIO_MST_B_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x04c) > +#define AUDIO_MST_C_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x050) > +#define AUDIO_MST_C_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x054) > +#define AUDIO_MST_D_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x058) > +#define AUDIO_MST_D_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x05c) > +#define AUDIO_CLK_TDMIN_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x080) > +#define AUDIO_CLK_TDMIN_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x084) > +#define AUDIO_CLK_TDMIN_LB_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x08c) > +#define AUDIO_CLK_TDMOUT_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x090) > +#define AUDIO_CLK_TDMOUT_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x094) > +#define AUDIO_CLK_SPDIFIN_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x09c) > +#define AUDIO_CLK_RESAMPLE_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a4) > +#define AUDIO_CLK_LOCKER_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a8) > +#define AUDIO_CLK_EQDRC_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0c0) > + > +#define AUDIO2_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_1, 0x00c) > +#define AUDIO2_MCLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x040) > +#define AUDIO2_CLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x044) > +#define AUDIO2_CLK_PDMIN_CTRL0 AUDIO_REG(AUDIO_RANGE_1, 0x058) > +#define AUDIO2_CLK_PDMIN_CTRL1 AUDIO_REG(AUDIO_RANGE_1, 0x05c) > + > +#include <dt-bindings/clock/amlogic,a1-audio-clkc.h> > + > +#endif /* __A1_AUDIO_H */
On Fri 15 Mar 2024 at 02:21, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > This series includes the following: > > - new audio clock and reset controller data and adaptation for it of existing > code (patches 0001..0004); > > - adaptation of existing audio components for A1 Soc (patches 0005..0021); > > - handy cosmetics for dai-link naming (patches 0022..0023); > > - integration of audio devices into common trees (patch 0024); > > - audio support bring up on Amlogic ad402 reference board (patch 0025). This > patch is not actually checked on real hardware (because all ad402 that we had > were burned out). This patch is based on ad402's schematics and on experience > with our own hardware (which is very close to reference board); > > Dmitry Rokosov (2): > ASoC: dt-bindings: meson: introduce link-name optional property > ASoC: meson: implement link-name optional property in meson card utils > > Jan Dakinevich (23): > clk: meson: a1: restrict an amount of 'hifi_pll' params > clk: meson: axg: move reset controller's code to separate module > dt-bindings: clock: meson: add A1 audio clock and reset controller > bindings > clk: meson: a1: add the audio clock controller driver > ASoC: meson: codec-glue: add support for capture stream > ASoC: meson: g12a-toacodec: fix "Lane Select" width > ASoC: meson: g12a-toacodec: rework the definition of bits > ASoC: dt-bindings: meson: g12a-toacodec: add support for A1 SoC family > ASoC: meson: g12a-toacodec: add support for A1 SoC family > ASoC: meson: t9015: prepare to adding new platforms > ASoC: dt-bindings: meson: t9015: add support for A1 SoC family > ASoC: meson: t9015: add support for A1 SoC family > ASoC: dt-bindings: meson: axg-pdm: document 'sysrate' property > ASoC: meson: axg-pdm: introduce 'sysrate' property > pinctrl/meson: fix typo in PDM's pin name > ASoC: dt-bindings: meson: meson-axg-audio-arb: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-fifo: claim support of A1 SoC family > ASoC: dt-bindings: meson: axg-pdm: claim support of A1 SoC family > ASoC: dt-bindings: meson: axg-sound-card: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-tdm-formatters: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-tdm-iface: claim support of A1 SoC > family > arm64: dts: meson: a1: add audio devices > arm64: dts: ad402: enable audio I'm sorry but a 25 patches series is just way too big, especially when spamming multiple sub systems. Please try to make one series per sub systems and general topic, I see at least * A1 audio clocks * G12 acodec fix * Acodec rework * PDM * pinctrl * arm64 I did not review all but I think I've made enough comments to keep you busy for a while > > .../bindings/clock/amlogic,a1-audio-clkc.yaml | 83 +++ > .../reset/amlogic,meson-axg-audio-arb.yaml | 10 +- > .../bindings/sound/amlogic,axg-fifo.yaml | 8 + > .../bindings/sound/amlogic,axg-pdm.yaml | 5 + > .../sound/amlogic,axg-sound-card.yaml | 12 +- > .../sound/amlogic,axg-tdm-formatters.yaml | 22 +- > .../bindings/sound/amlogic,axg-tdm-iface.yaml | 6 +- > .../bindings/sound/amlogic,g12a-toacodec.yaml | 1 + > .../bindings/sound/amlogic,gx-sound-card.yaml | 6 + > .../bindings/sound/amlogic,t9015.yaml | 4 +- > .../arm64/boot/dts/amlogic/meson-a1-ad402.dts | 126 ++++ > arch/arm64/boot/dts/amlogic/meson-a1.dtsi | 471 +++++++++++++++ > drivers/clk/meson/Kconfig | 18 + > drivers/clk/meson/Makefile | 2 + > drivers/clk/meson/a1-audio.c | 556 ++++++++++++++++++ > drivers/clk/meson/a1-audio.h | 58 ++ > drivers/clk/meson/a1-pll.c | 8 +- > drivers/clk/meson/axg-audio.c | 95 +-- > drivers/clk/meson/meson-audio-rstc.c | 109 ++++ > drivers/clk/meson/meson-audio-rstc.h | 12 + > drivers/pinctrl/meson/pinctrl-meson-a1.c | 6 +- > .../dt-bindings/clock/amlogic,a1-audio-clkc.h | 122 ++++ > .../reset/amlogic,meson-a1-audio-reset.h | 29 + > .../dt-bindings/sound/meson-g12a-toacodec.h | 5 + > sound/soc/meson/axg-pdm.c | 10 +- > sound/soc/meson/g12a-toacodec.c | 298 ++++++++-- > sound/soc/meson/meson-card-utils.c | 12 +- > sound/soc/meson/meson-codec-glue.c | 174 ++++-- > sound/soc/meson/meson-codec-glue.h | 23 + > sound/soc/meson/t9015.c | 326 +++++++++- > 30 files changed, 2394 insertions(+), 223 deletions(-) > create mode 100644 Documentation/devicetree/bindings/clock/amlogic,a1-audio-clkc.yaml > create mode 100644 drivers/clk/meson/a1-audio.c > create mode 100644 drivers/clk/meson/a1-audio.h > create mode 100644 drivers/clk/meson/meson-audio-rstc.c > create mode 100644 drivers/clk/meson/meson-audio-rstc.h > create mode 100644 include/dt-bindings/clock/amlogic,a1-audio-clkc.h > create mode 100644 include/dt-bindings/reset/amlogic,meson-a1-audio-reset.h
On Fri 15 Mar 2024 at 11:00, Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> wrote: > On 15/03/2024 00:21, Jan Dakinevich wrote: >> This option allow to redefine the rate of DSP system clock. > > And why is it suitable for bindings? Describe the hardware, not what you > want to do in the driver. > >> >> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >> --- >> Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml | 4 ++++ >> 1 file changed, 4 insertions(+) >> >> diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >> index df21dd72fc65..d2f23a59a6b6 100644 >> --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >> +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >> @@ -40,6 +40,10 @@ properties: >> resets: >> maxItems: 1 >> >> + sysrate: >> + $ref: /schemas/types.yaml#/definitions/uint32 >> + description: redefine rate of DSP system clock > > No vendor prefix, so is it a generic property? Also, missing unit > suffix, but more importantly I don't understand why this is a property > of hardware. +1. The appropriate way to set rate of the clock before the driver take over is 'assigned-rate', if you need to customize this for different platform. Then you don't have to deal with it in the device driver. > > Best regards, > Krzysztof
On Fri, 15 Mar 2024 02:21:36 +0300, Jan Dakinevich wrote: > This series includes the following: > > - new audio clock and reset controller data and adaptation for it of existing > code (patches 0001..0004); > > - adaptation of existing audio components for A1 Soc (patches 0005..0021); > > - handy cosmetics for dai-link naming (patches 0022..0023); > > - integration of audio devices into common trees (patch 0024); > > - audio support bring up on Amlogic ad402 reference board (patch 0025). This > patch is not actually checked on real hardware (because all ad402 that we had > were burned out). This patch is based on ad402's schematics and on experience > with our own hardware (which is very close to reference board); > > Dmitry Rokosov (2): > ASoC: dt-bindings: meson: introduce link-name optional property > ASoC: meson: implement link-name optional property in meson card utils > > Jan Dakinevich (23): > clk: meson: a1: restrict an amount of 'hifi_pll' params > clk: meson: axg: move reset controller's code to separate module > dt-bindings: clock: meson: add A1 audio clock and reset controller > bindings > clk: meson: a1: add the audio clock controller driver > ASoC: meson: codec-glue: add support for capture stream > ASoC: meson: g12a-toacodec: fix "Lane Select" width > ASoC: meson: g12a-toacodec: rework the definition of bits > ASoC: dt-bindings: meson: g12a-toacodec: add support for A1 SoC family > ASoC: meson: g12a-toacodec: add support for A1 SoC family > ASoC: meson: t9015: prepare to adding new platforms > ASoC: dt-bindings: meson: t9015: add support for A1 SoC family > ASoC: meson: t9015: add support for A1 SoC family > ASoC: dt-bindings: meson: axg-pdm: document 'sysrate' property > ASoC: meson: axg-pdm: introduce 'sysrate' property > pinctrl/meson: fix typo in PDM's pin name > ASoC: dt-bindings: meson: meson-axg-audio-arb: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-fifo: claim support of A1 SoC family > ASoC: dt-bindings: meson: axg-pdm: claim support of A1 SoC family > ASoC: dt-bindings: meson: axg-sound-card: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-tdm-formatters: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-tdm-iface: claim support of A1 SoC > family > arm64: dts: meson: a1: add audio devices > arm64: dts: ad402: enable audio > > .../bindings/clock/amlogic,a1-audio-clkc.yaml | 83 +++ > .../reset/amlogic,meson-axg-audio-arb.yaml | 10 +- > .../bindings/sound/amlogic,axg-fifo.yaml | 8 + > .../bindings/sound/amlogic,axg-pdm.yaml | 5 + > .../sound/amlogic,axg-sound-card.yaml | 12 +- > .../sound/amlogic,axg-tdm-formatters.yaml | 22 +- > .../bindings/sound/amlogic,axg-tdm-iface.yaml | 6 +- > .../bindings/sound/amlogic,g12a-toacodec.yaml | 1 + > .../bindings/sound/amlogic,gx-sound-card.yaml | 6 + > .../bindings/sound/amlogic,t9015.yaml | 4 +- > .../arm64/boot/dts/amlogic/meson-a1-ad402.dts | 126 ++++ > arch/arm64/boot/dts/amlogic/meson-a1.dtsi | 471 +++++++++++++++ > drivers/clk/meson/Kconfig | 18 + > drivers/clk/meson/Makefile | 2 + > drivers/clk/meson/a1-audio.c | 556 ++++++++++++++++++ > drivers/clk/meson/a1-audio.h | 58 ++ > drivers/clk/meson/a1-pll.c | 8 +- > drivers/clk/meson/axg-audio.c | 95 +-- > drivers/clk/meson/meson-audio-rstc.c | 109 ++++ > drivers/clk/meson/meson-audio-rstc.h | 12 + > drivers/pinctrl/meson/pinctrl-meson-a1.c | 6 +- > .../dt-bindings/clock/amlogic,a1-audio-clkc.h | 122 ++++ > .../reset/amlogic,meson-a1-audio-reset.h | 29 + > .../dt-bindings/sound/meson-g12a-toacodec.h | 5 + > sound/soc/meson/axg-pdm.c | 10 +- > sound/soc/meson/g12a-toacodec.c | 298 ++++++++-- > sound/soc/meson/meson-card-utils.c | 12 +- > sound/soc/meson/meson-codec-glue.c | 174 ++++-- > sound/soc/meson/meson-codec-glue.h | 23 + > sound/soc/meson/t9015.c | 326 +++++++++- > 30 files changed, 2394 insertions(+), 223 deletions(-) > create mode 100644 Documentation/devicetree/bindings/clock/amlogic,a1-audio-clkc.yaml > create mode 100644 drivers/clk/meson/a1-audio.c > create mode 100644 drivers/clk/meson/a1-audio.h > create mode 100644 drivers/clk/meson/meson-audio-rstc.c > create mode 100644 drivers/clk/meson/meson-audio-rstc.h > create mode 100644 include/dt-bindings/clock/amlogic,a1-audio-clkc.h > create mode 100644 include/dt-bindings/reset/amlogic,meson-a1-audio-reset.h > > -- > 2.34.1 > > > My bot found new DTB warnings on the .dts files added or changed in this series. Some warnings may be from an existing SoC .dtsi. Or perhaps the warnings are fixed by another series. Ultimately, it is up to the platform maintainer whether these warnings are acceptable or not. No need to reply unless the platform maintainer has comments. If you already ran DT checks and didn't see these error(s), then make sure dt-schema is up to date: pip3 install dtschema --upgrade New warnings running 'make CHECK_DTBS=y amlogic/meson-a1-ad402.dtb' for 20240314232201.2102178-1-jan.dakinevich@salutedevices.com: arch/arm64/boot/dts/amlogic/meson-a1-ad402.dtb: audio-controller@4800: 'power-domains' does not match any of the regexes: 'pinctrl-[0-9]+' from schema $id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml# arch/arm64/boot/dts/amlogic/meson-a1-ad402.dtb: audio-controller@1000: Unevaluated properties are not allowed ('power-domains' was unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,axg-pdm.yaml#
Hi Jan! On 15/03/2024 00:21, Jan Dakinevich wrote: > This series includes the following: > > - new audio clock and reset controller data and adaptation for it of existing > code (patches 0001..0004); > > - adaptation of existing audio components for A1 Soc (patches 0005..0021); > > - handy cosmetics for dai-link naming (patches 0022..0023); > > - integration of audio devices into common trees (patch 0024); > > - audio support bring up on Amlogic ad402 reference board (patch 0025). This > patch is not actually checked on real hardware (because all ad402 that we had > were burned out). This patch is based on ad402's schematics and on experience > with our own hardware (which is very close to reference board); Thanks for your serie, it's nice you're working on upstreaming this feature. In my opinion it's fine to have a "big" initial serie if you're not sure if your changes are ok, but next time add the RFC tag so we know it's not a final changeset and you seek advices. Overall the code is clean and your patch order makes sense if they were meant to be applied by a single maintainer, but in this case it will be split into multiple subsystems so it's better to split them as Jerome explained to ease review and the maintainers process. Don't hesitate discussing with us in the #linux-amlogic IRC channel on Libera.Chat, the goal is to reduce the number of patch version and ease the review and maintainance process. Concerning the link-name property, I think it should be done afterwards since it's not necessary to support audio on A1, and I think it could be extended to other SoC boards (which would be a great feature). Neil > > Dmitry Rokosov (2): > ASoC: dt-bindings: meson: introduce link-name optional property > ASoC: meson: implement link-name optional property in meson card utils > > Jan Dakinevich (23): > clk: meson: a1: restrict an amount of 'hifi_pll' params > clk: meson: axg: move reset controller's code to separate module > dt-bindings: clock: meson: add A1 audio clock and reset controller > bindings > clk: meson: a1: add the audio clock controller driver > ASoC: meson: codec-glue: add support for capture stream > ASoC: meson: g12a-toacodec: fix "Lane Select" width > ASoC: meson: g12a-toacodec: rework the definition of bits > ASoC: dt-bindings: meson: g12a-toacodec: add support for A1 SoC family > ASoC: meson: g12a-toacodec: add support for A1 SoC family > ASoC: meson: t9015: prepare to adding new platforms > ASoC: dt-bindings: meson: t9015: add support for A1 SoC family > ASoC: meson: t9015: add support for A1 SoC family > ASoC: dt-bindings: meson: axg-pdm: document 'sysrate' property > ASoC: meson: axg-pdm: introduce 'sysrate' property > pinctrl/meson: fix typo in PDM's pin name > ASoC: dt-bindings: meson: meson-axg-audio-arb: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-fifo: claim support of A1 SoC family > ASoC: dt-bindings: meson: axg-pdm: claim support of A1 SoC family > ASoC: dt-bindings: meson: axg-sound-card: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-tdm-formatters: claim support of A1 SoC > family > ASoC: dt-bindings: meson: axg-tdm-iface: claim support of A1 SoC > family > arm64: dts: meson: a1: add audio devices > arm64: dts: ad402: enable audio > > .../bindings/clock/amlogic,a1-audio-clkc.yaml | 83 +++ > .../reset/amlogic,meson-axg-audio-arb.yaml | 10 +- > .../bindings/sound/amlogic,axg-fifo.yaml | 8 + > .../bindings/sound/amlogic,axg-pdm.yaml | 5 + > .../sound/amlogic,axg-sound-card.yaml | 12 +- > .../sound/amlogic,axg-tdm-formatters.yaml | 22 +- > .../bindings/sound/amlogic,axg-tdm-iface.yaml | 6 +- > .../bindings/sound/amlogic,g12a-toacodec.yaml | 1 + > .../bindings/sound/amlogic,gx-sound-card.yaml | 6 + > .../bindings/sound/amlogic,t9015.yaml | 4 +- > .../arm64/boot/dts/amlogic/meson-a1-ad402.dts | 126 ++++ > arch/arm64/boot/dts/amlogic/meson-a1.dtsi | 471 +++++++++++++++ > drivers/clk/meson/Kconfig | 18 + > drivers/clk/meson/Makefile | 2 + > drivers/clk/meson/a1-audio.c | 556 ++++++++++++++++++ > drivers/clk/meson/a1-audio.h | 58 ++ > drivers/clk/meson/a1-pll.c | 8 +- > drivers/clk/meson/axg-audio.c | 95 +-- > drivers/clk/meson/meson-audio-rstc.c | 109 ++++ > drivers/clk/meson/meson-audio-rstc.h | 12 + > drivers/pinctrl/meson/pinctrl-meson-a1.c | 6 +- > .../dt-bindings/clock/amlogic,a1-audio-clkc.h | 122 ++++ > .../reset/amlogic,meson-a1-audio-reset.h | 29 + > .../dt-bindings/sound/meson-g12a-toacodec.h | 5 + > sound/soc/meson/axg-pdm.c | 10 +- > sound/soc/meson/g12a-toacodec.c | 298 ++++++++-- > sound/soc/meson/meson-card-utils.c | 12 +- > sound/soc/meson/meson-codec-glue.c | 174 ++++-- > sound/soc/meson/meson-codec-glue.h | 23 + > sound/soc/meson/t9015.c | 326 +++++++++- > 30 files changed, 2394 insertions(+), 223 deletions(-) > create mode 100644 Documentation/devicetree/bindings/clock/amlogic,a1-audio-clkc.yaml > create mode 100644 drivers/clk/meson/a1-audio.c > create mode 100644 drivers/clk/meson/a1-audio.h > create mode 100644 drivers/clk/meson/meson-audio-rstc.c > create mode 100644 drivers/clk/meson/meson-audio-rstc.h > create mode 100644 include/dt-bindings/clock/amlogic,a1-audio-clkc.h > create mode 100644 include/dt-bindings/reset/amlogic,meson-a1-audio-reset.h >
On 3/15/24 13:22, Jerome Brunet wrote: > > On Fri 15 Mar 2024 at 11:00, Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> wrote: > >> On 15/03/2024 00:21, Jan Dakinevich wrote: >>> This option allow to redefine the rate of DSP system clock. >> >> And why is it suitable for bindings? Describe the hardware, not what you >> want to do in the driver. >> >>> >>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>> --- >>> Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml | 4 ++++ >>> 1 file changed, 4 insertions(+) >>> >>> diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>> index df21dd72fc65..d2f23a59a6b6 100644 >>> --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>> +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>> @@ -40,6 +40,10 @@ properties: >>> resets: >>> maxItems: 1 >>> >>> + sysrate: >>> + $ref: /schemas/types.yaml#/definitions/uint32 >>> + description: redefine rate of DSP system clock >> >> No vendor prefix, so is it a generic property? Also, missing unit >> suffix, but more importantly I don't understand why this is a property >> of hardware. > > +1. > > The appropriate way to set rate of the clock before the driver take over > is 'assigned-rate', if you need to customize this for different > platform. > It would be great, but it doesn't work. Below, is what I want to see: assigned-clocks = <&clkc_audio AUD2_CLKID_PDM_SYSCLK_SEL>, <&clkc_audio AUD2_CLKID_PDM_SYSCLK_DIV>; assigned-clock-parents = <&clkc_pll CLKID_FCLK_DIV3>, <0>; assigned-clock-rates = <0>, <256000000>; But regardles of this declaration, PDM's driver unconditionally sets sysclk'rate to 250MHz and throws away everything that was configured before, reparents audio2_pdm_sysclk_mux to hifi_pll and changes hifi_pll's rate. This value 250MHz is declared here: static const struct axg_pdm_cfg axg_pdm_config = { .filters = &axg_default_filters, .sys_rate = 250000000, }; The property 'sysrate' is intended to redefine hardcoded 'sys_rate' value in 'axg_pdm_config'. > Then you don't have to deal with it in the device driver. > >> >> Best regards, >> Krzysztof > >
On Sun 17 Mar 2024 at 18:52, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > On 3/15/24 13:22, Jerome Brunet wrote: >> >> On Fri 15 Mar 2024 at 11:00, Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> wrote: >> >>> On 15/03/2024 00:21, Jan Dakinevich wrote: >>>> This option allow to redefine the rate of DSP system clock. >>> >>> And why is it suitable for bindings? Describe the hardware, not what you >>> want to do in the driver. >>> >>>> >>>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>> --- >>>> Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml | 4 ++++ >>>> 1 file changed, 4 insertions(+) >>>> >>>> diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>> index df21dd72fc65..d2f23a59a6b6 100644 >>>> --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>> +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>> @@ -40,6 +40,10 @@ properties: >>>> resets: >>>> maxItems: 1 >>>> >>>> + sysrate: >>>> + $ref: /schemas/types.yaml#/definitions/uint32 >>>> + description: redefine rate of DSP system clock >>> >>> No vendor prefix, so is it a generic property? Also, missing unit >>> suffix, but more importantly I don't understand why this is a property >>> of hardware. >> >> +1. >> >> The appropriate way to set rate of the clock before the driver take over >> is 'assigned-rate', if you need to customize this for different >> platform. >> > > It would be great, but it doesn't work. Below, is what I want to see: > > assigned-clocks = > <&clkc_audio AUD2_CLKID_PDM_SYSCLK_SEL>, > <&clkc_audio AUD2_CLKID_PDM_SYSCLK_DIV>; > assigned-clock-parents = > <&clkc_pll CLKID_FCLK_DIV3>, > <0>; > assigned-clock-rates = > <0>, > <256000000>; > > But regardles of this declaration, PDM's driver unconditionally sets > sysclk'rate to 250MHz and throws away everything that was configured > before, reparents audio2_pdm_sysclk_mux to hifi_pll and changes > hifi_pll's rate. > > This value 250MHz is declared here: > > static const struct axg_pdm_cfg axg_pdm_config = { > .filters = &axg_default_filters, > .sys_rate = 250000000, > }; > > The property 'sysrate' is intended to redefine hardcoded 'sys_rate' > value in 'axg_pdm_config'. What is stopping you from removing that from the driver and adding assigned-rate to 250M is the existing platform ? > >> Then you don't have to deal with it in the device driver. >> >>> >>> Best regards, >>> Krzysztof >> >>
On Mon 18 Mar 2024 at 11:55, Jerome Brunet <jbrunet@baylibre.com> wrote: > On Sun 17 Mar 2024 at 18:52, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > >> On 3/15/24 13:22, Jerome Brunet wrote: >>> >>> On Fri 15 Mar 2024 at 11:00, Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> wrote: >>> >>>> On 15/03/2024 00:21, Jan Dakinevich wrote: >>>>> This option allow to redefine the rate of DSP system clock. >>>> >>>> And why is it suitable for bindings? Describe the hardware, not what you >>>> want to do in the driver. >>>> >>>>> >>>>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>> --- >>>>> Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml | 4 ++++ >>>>> 1 file changed, 4 insertions(+) >>>>> >>>>> diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>>> index df21dd72fc65..d2f23a59a6b6 100644 >>>>> --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>>> +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>>> @@ -40,6 +40,10 @@ properties: >>>>> resets: >>>>> maxItems: 1 >>>>> >>>>> + sysrate: >>>>> + $ref: /schemas/types.yaml#/definitions/uint32 >>>>> + description: redefine rate of DSP system clock >>>> >>>> No vendor prefix, so is it a generic property? Also, missing unit >>>> suffix, but more importantly I don't understand why this is a property >>>> of hardware. >>> >>> +1. >>> >>> The appropriate way to set rate of the clock before the driver take over >>> is 'assigned-rate', if you need to customize this for different >>> platform. >>> >> >> It would be great, but it doesn't work. Below, is what I want to see: >> >> assigned-clocks = >> <&clkc_audio AUD2_CLKID_PDM_SYSCLK_SEL>, >> <&clkc_audio AUD2_CLKID_PDM_SYSCLK_DIV>; >> assigned-clock-parents = >> <&clkc_pll CLKID_FCLK_DIV3>, >> <0>; >> assigned-clock-rates = >> <0>, >> <256000000>; >> >> But regardles of this declaration, PDM's driver unconditionally sets >> sysclk'rate to 250MHz and throws away everything that was configured >> before, reparents audio2_pdm_sysclk_mux to hifi_pll and changes >> hifi_pll's rate. >> >> This value 250MHz is declared here: >> >> static const struct axg_pdm_cfg axg_pdm_config = { >> .filters = &axg_default_filters, >> .sys_rate = 250000000, >> }; >> >> The property 'sysrate' is intended to redefine hardcoded 'sys_rate' >> value in 'axg_pdm_config'. > > What is stopping you from removing that from the driver and adding > assigned-rate to 250M is the existing platform ? ... Also, considering how PDM does work, I'm not sure I get the point of the doing all this to go from 250MHz to 256Mhz. PDM value is sampled at ~75% of the half period. That clock basically feeds a counter and the threshold is adjusted based on the clock rate. So there is no need to change the rate. Changing it is only necessary when the captured audio rate is extremely slow (<8kHz) and the counter may overflow. The driver already adjust this automatically. So changing the input rate from 250MHz to 256MHz should not make any difference. > >> >>> Then you don't have to deal with it in the device driver. >>> >>>> >>>> Best regards, >>>> Krzysztof >>> >>>
On 3/18/24 15:19, Jerome Brunet wrote: > > On Mon 18 Mar 2024 at 11:55, Jerome Brunet <jbrunet@baylibre.com> wrote: > >> On Sun 17 Mar 2024 at 18:52, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >> >>> On 3/15/24 13:22, Jerome Brunet wrote: >>>> >>>> On Fri 15 Mar 2024 at 11:00, Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> wrote: >>>> >>>>> On 15/03/2024 00:21, Jan Dakinevich wrote: >>>>>> This option allow to redefine the rate of DSP system clock. >>>>> >>>>> And why is it suitable for bindings? Describe the hardware, not what you >>>>> want to do in the driver. >>>>> >>>>>> >>>>>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>>> --- >>>>>> Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml | 4 ++++ >>>>>> 1 file changed, 4 insertions(+) >>>>>> >>>>>> diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>>>> index df21dd72fc65..d2f23a59a6b6 100644 >>>>>> --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>>>> +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>>>> @@ -40,6 +40,10 @@ properties: >>>>>> resets: >>>>>> maxItems: 1 >>>>>> >>>>>> + sysrate: >>>>>> + $ref: /schemas/types.yaml#/definitions/uint32 >>>>>> + description: redefine rate of DSP system clock >>>>> >>>>> No vendor prefix, so is it a generic property? Also, missing unit >>>>> suffix, but more importantly I don't understand why this is a property >>>>> of hardware. >>>> >>>> +1. >>>> >>>> The appropriate way to set rate of the clock before the driver take over >>>> is 'assigned-rate', if you need to customize this for different >>>> platform. >>>> >>> >>> It would be great, but it doesn't work. Below, is what I want to see: >>> >>> assigned-clocks = >>> <&clkc_audio AUD2_CLKID_PDM_SYSCLK_SEL>, >>> <&clkc_audio AUD2_CLKID_PDM_SYSCLK_DIV>; >>> assigned-clock-parents = >>> <&clkc_pll CLKID_FCLK_DIV3>, >>> <0>; >>> assigned-clock-rates = >>> <0>, >>> <256000000>; >>> >>> But regardles of this declaration, PDM's driver unconditionally sets >>> sysclk'rate to 250MHz and throws away everything that was configured >>> before, reparents audio2_pdm_sysclk_mux to hifi_pll and changes >>> hifi_pll's rate. >>> >>> This value 250MHz is declared here: >>> >>> static const struct axg_pdm_cfg axg_pdm_config = { >>> .filters = &axg_default_filters, >>> .sys_rate = 250000000, >>> }; >>> >>> The property 'sysrate' is intended to redefine hardcoded 'sys_rate' >>> value in 'axg_pdm_config'. >> >> What is stopping you from removing that from the driver and adding >> assigned-rate to 250M is the existing platform ? > > ... Also, considering how PDM does work, I'm not sure I get the point of > the doing all this to go from 250MHz to 256Mhz. > The point is to use fclk_div3 clock as source for PDM's sysclock and keep hiff_pll clock free for TDM. Because, I can get 256MHz from any hifi_pll and fclk_div3, but only hifi_pll is able to provide accurate 48kHz (after several divider). > PDM value is sampled at ~75% of the half period. That clock basically > feeds a counter and the threshold is adjusted based on the clock rate. > > So there is no need to change the rate. Changing it is only necessary > when the captured audio rate is extremely slow (<8kHz) and the counter > may overflow. The driver already adjust this automatically. > > So changing the input rate from 250MHz to 256MHz should not make any > difference. > Thank you for the explanation. >> >>> >>>> Then you don't have to deal with it in the device driver. >>>> >>>>> >>>>> Best regards, >>>>> Krzysztof >>>> >>>> > >
Let's start from the end: > No - Looks to me you just have two clock controllers you are trying force into one. > Again, this shows 2 devices. The one related to your 'map0' should request AUD2_CLKID_AUDIOTOP as input and enable it right away. Most of fishy workarounds that you commented is caused the fact the mmio of this clock controller is divided into two parts. Compare it with axg-audio driver, things that was part of contigous memory region (like pdm) here are moved to second region. Is this enough to make a guess that these are two devices? Concerning AUD2_CLKID_AUDIOTOP clock, as it turned out, it must be enabled before enabling of clocks from second region too. That is AUD2_CLKID_AUDIOTOP clock feeds both parts of this clock controller. On 3/15/24 12:20, Jerome Brunet wrote: > > On Fri 15 Mar 2024 at 02:21, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > >> This controller provides clocks and reset functionality for audio >> peripherals on Amlogic A1 SoC family. >> >> The driver is almost identical to 'axg-audio', however it would be better >> to keep it separate due to following reasons: >> >> - significant amount of bits has another definition. I will bring there >> a mess of new defines with A1_ suffixes. >> >> - registers of this controller are located in two separate regions. It >> will give a lot of complications for 'axg-audio' to support this. >> >> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >> --- >> drivers/clk/meson/Kconfig | 13 + >> drivers/clk/meson/Makefile | 1 + >> drivers/clk/meson/a1-audio.c | 556 +++++++++++++++++++++++++++++++++++ >> drivers/clk/meson/a1-audio.h | 58 ++++ >> 4 files changed, 628 insertions(+) >> create mode 100644 drivers/clk/meson/a1-audio.c >> create mode 100644 drivers/clk/meson/a1-audio.h >> >> diff --git a/drivers/clk/meson/Kconfig b/drivers/clk/meson/Kconfig >> index d6a2fa5f7e88..80c4a18c83d2 100644 >> --- a/drivers/clk/meson/Kconfig >> +++ b/drivers/clk/meson/Kconfig >> @@ -133,6 +133,19 @@ config COMMON_CLK_A1_PERIPHERALS >> device, A1 SoC Family. Say Y if you want A1 Peripherals clock >> controller to work. >> >> +config COMMON_CLK_A1_AUDIO >> + tristate "Amlogic A1 SoC Audio clock controller support" >> + depends on ARM64 >> + select COMMON_CLK_MESON_REGMAP >> + select COMMON_CLK_MESON_CLKC_UTILS >> + select COMMON_CLK_MESON_PHASE >> + select COMMON_CLK_MESON_SCLK_DIV >> + select COMMON_CLK_MESON_AUDIO_RSTC >> + help >> + Support for the Audio clock controller on Amlogic A113L based >> + device, A1 SoC Family. Say Y if you want A1 Audio clock controller >> + to work. >> + >> config COMMON_CLK_G12A >> tristate "G12 and SM1 SoC clock controllers support" >> depends on ARM64 >> diff --git a/drivers/clk/meson/Makefile b/drivers/clk/meson/Makefile >> index 88d94921a4dc..4968fc7ad555 100644 >> --- a/drivers/clk/meson/Makefile >> +++ b/drivers/clk/meson/Makefile >> @@ -20,6 +20,7 @@ obj-$(CONFIG_COMMON_CLK_AXG) += axg.o axg-aoclk.o >> obj-$(CONFIG_COMMON_CLK_AXG_AUDIO) += axg-audio.o >> obj-$(CONFIG_COMMON_CLK_A1_PLL) += a1-pll.o >> obj-$(CONFIG_COMMON_CLK_A1_PERIPHERALS) += a1-peripherals.o >> +obj-$(CONFIG_COMMON_CLK_A1_AUDIO) += a1-audio.o >> obj-$(CONFIG_COMMON_CLK_GXBB) += gxbb.o gxbb-aoclk.o >> obj-$(CONFIG_COMMON_CLK_G12A) += g12a.o g12a-aoclk.o >> obj-$(CONFIG_COMMON_CLK_MESON8B) += meson8b.o meson8-ddr.o >> diff --git a/drivers/clk/meson/a1-audio.c b/drivers/clk/meson/a1-audio.c >> new file mode 100644 >> index 000000000000..6039116c93ba >> --- /dev/null >> +++ b/drivers/clk/meson/a1-audio.c >> @@ -0,0 +1,556 @@ >> +// SPDX-License-Identifier: (GPL-2.0 OR MIT) >> +/* >> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >> + * >> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >> + */ >> + >> +#include <linux/clk.h> >> +#include <linux/clk-provider.h> >> +#include <linux/init.h> >> +#include <linux/of_device.h> >> +#include <linux/module.h> >> +#include <linux/platform_device.h> >> +#include <linux/regmap.h> >> +#include <linux/reset.h> >> +#include <linux/reset-controller.h> >> +#include <linux/slab.h> >> + >> +#include "meson-clkc-utils.h" >> +#include "meson-audio-rstc.h" >> +#include "clk-regmap.h" >> +#include "clk-phase.h" >> +#include "sclk-div.h" >> +#include "a1-audio.h" >> + >> +#define AUDIO_PDATA(_name) \ >> + ((const struct clk_parent_data[]) { { .hw = &(_name).hw } }) > > Not a fan - yet another level of macro. > >> + >> +#define AUDIO_MUX(_name, _reg, _mask, _shift, _pdata) \ >> +static struct clk_regmap _name = { \ >> + .map = AUDIO_REG_MAP(_reg), \ >> + .data = &(struct clk_regmap_mux_data){ \ >> + .offset = AUDIO_REG_OFFSET(_reg), \ >> + .mask = (_mask), \ >> + .shift = (_shift), \ >> + }, \ >> + .hw.init = &(struct clk_init_data) { \ >> + .name = #_name, \ >> + .ops = &clk_regmap_mux_ops, \ >> + .parent_data = (_pdata), \ >> + .num_parents = ARRAY_SIZE(_pdata), \ >> + .flags = CLK_SET_RATE_PARENT, \ >> + }, \ >> +} >> + >> +#define AUDIO_DIV(_name, _reg, _shift, _width, _pdata) \ >> +static struct clk_regmap _name = { \ >> + .map = AUDIO_REG_MAP(_reg), \ >> + .data = &(struct clk_regmap_div_data){ \ >> + .offset = AUDIO_REG_OFFSET(_reg), \ >> + .shift = (_shift), \ >> + .width = (_width), \ >> + }, \ >> + .hw.init = &(struct clk_init_data) { \ >> + .name = #_name, \ >> + .ops = &clk_regmap_divider_ops, \ >> + .parent_data = (_pdata), \ >> + .num_parents = 1, \ >> + .flags = CLK_SET_RATE_PARENT, \ >> + }, \ >> +} >> + >> +#define AUDIO_GATE(_name, _reg, _bit, _pdata) \ >> +static struct clk_regmap _name = { \ >> + .map = AUDIO_REG_MAP(_reg), \ >> + .data = &(struct clk_regmap_gate_data){ \ >> + .offset = AUDIO_REG_OFFSET(_reg), \ >> + .bit_idx = (_bit), \ >> + }, \ >> + .hw.init = &(struct clk_init_data) { \ >> + .name = #_name, \ >> + .ops = &clk_regmap_gate_ops, \ >> + .parent_data = (_pdata), \ >> + .num_parents = 1, \ >> + .flags = CLK_SET_RATE_PARENT, \ >> + }, \ >> +} >> + >> +#define AUDIO_SCLK_DIV(_name, _reg, _div_shift, _div_width, \ >> + _hi_shift, _hi_width, _pdata, _set_rate_parent) \ >> +static struct clk_regmap _name = { \ >> + .map = AUDIO_REG_MAP(_reg), \ >> + .data = &(struct meson_sclk_div_data) { \ >> + .div = { \ >> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >> + .shift = (_div_shift), \ >> + .width = (_div_width), \ >> + }, \ >> + .hi = { \ >> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >> + .shift = (_hi_shift), \ >> + .width = (_hi_width), \ >> + }, \ >> + }, \ >> + .hw.init = &(struct clk_init_data) { \ >> + .name = #_name, \ >> + .ops = &meson_sclk_div_ops, \ >> + .parent_data = (_pdata), \ >> + .num_parents = 1, \ >> + .flags = (_set_rate_parent) ? CLK_SET_RATE_PARENT : 0, \ > > Does not help readeability. Just pass the flag as axg-audio does. > >> + }, \ >> +} >> + >> +#define AUDIO_TRIPHASE(_name, _reg, _width, _shift0, _shift1, _shift2, \ >> + _pdata) \ >> +static struct clk_regmap _name = { \ >> + .map = AUDIO_REG_MAP(_reg), \ >> + .data = &(struct meson_clk_triphase_data) { \ >> + .ph0 = { \ >> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >> + .shift = (_shift0), \ >> + .width = (_width), \ >> + }, \ >> + .ph1 = { \ >> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >> + .shift = (_shift1), \ >> + .width = (_width), \ >> + }, \ >> + .ph2 = { \ >> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >> + .shift = (_shift2), \ >> + .width = (_width), \ >> + }, \ >> + }, \ >> + .hw.init = &(struct clk_init_data) { \ >> + .name = #_name, \ >> + .ops = &meson_clk_triphase_ops, \ >> + .parent_data = (_pdata), \ >> + .num_parents = 1, \ >> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >> + }, \ >> +} >> + >> +#define AUDIO_SCLK_WS(_name, _reg, _width, _shift_ph, _shift_ws, \ >> + _pdata) \ >> +static struct clk_regmap _name = { \ >> + .map = AUDIO_REG_MAP(_reg), \ >> + .data = &(struct meson_sclk_ws_inv_data) { \ >> + .ph = { \ >> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >> + .shift = (_shift_ph), \ >> + .width = (_width), \ >> + }, \ >> + .ws = { \ >> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >> + .shift = (_shift_ws), \ >> + .width = (_width), \ >> + }, \ >> + }, \ >> + .hw.init = &(struct clk_init_data) { \ >> + .name = #_name, \ >> + .ops = &meson_sclk_ws_inv_ops, \ >> + .parent_data = (_pdata), \ >> + .num_parents = 1, \ >> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >> + }, \ >> +} > > All the above does essentially the same things as the macro of > axg-audio, to some minor differences. Yet it is another set to maintain. > Except one thing... Here I keep memory identifier to which this clock belongs: .map = AUDIO_REG_MAP(_reg), It is workaround, but ->map the only common field in clk_regmap that could be used for this purpose. > I'd much prefer if you put the axg-audio macro in a header a re-used > those. There would a single set to maintain. You may then specialize the > included in the driver C file, to avoid redundant parameters > > Rework axg-audio to use clk_parent_data if you must, but not in the same > series please. > >> + >> +static const struct clk_parent_data a1_pclk_pdata[] = { >> + { .fw_name = "pclk", }, >> +}; >> + >> +AUDIO_GATE(audio_ddr_arb, AUDIO_CLK_GATE_EN0, 0, a1_pclk_pdata); >> +AUDIO_GATE(audio_tdmin_a, AUDIO_CLK_GATE_EN0, 1, a1_pclk_pdata); >> +AUDIO_GATE(audio_tdmin_b, AUDIO_CLK_GATE_EN0, 2, a1_pclk_pdata); >> +AUDIO_GATE(audio_tdmin_lb, AUDIO_CLK_GATE_EN0, 3, a1_pclk_pdata); >> +AUDIO_GATE(audio_loopback, AUDIO_CLK_GATE_EN0, 4, a1_pclk_pdata); >> +AUDIO_GATE(audio_tdmout_a, AUDIO_CLK_GATE_EN0, 5, a1_pclk_pdata); >> +AUDIO_GATE(audio_tdmout_b, AUDIO_CLK_GATE_EN0, 6, a1_pclk_pdata); >> +AUDIO_GATE(audio_frddr_a, AUDIO_CLK_GATE_EN0, 7, a1_pclk_pdata); >> +AUDIO_GATE(audio_frddr_b, AUDIO_CLK_GATE_EN0, 8, a1_pclk_pdata); >> +AUDIO_GATE(audio_toddr_a, AUDIO_CLK_GATE_EN0, 9, a1_pclk_pdata); >> +AUDIO_GATE(audio_toddr_b, AUDIO_CLK_GATE_EN0, 10, a1_pclk_pdata); >> +AUDIO_GATE(audio_spdifin, AUDIO_CLK_GATE_EN0, 11, a1_pclk_pdata); >> +AUDIO_GATE(audio_resample, AUDIO_CLK_GATE_EN0, 12, a1_pclk_pdata); >> +AUDIO_GATE(audio_eqdrc, AUDIO_CLK_GATE_EN0, 13, a1_pclk_pdata); >> +AUDIO_GATE(audio_audiolocker, AUDIO_CLK_GATE_EN0, 14, a1_pclk_pdata); > This is what I mean by redundant parameter ^ > Yep. I could define something like AUDIO_PCLK_GATE(). >> + >> +AUDIO_GATE(audio2_ddr_arb, AUDIO2_CLK_GATE_EN0, 0, a1_pclk_pdata); >> +AUDIO_GATE(audio2_pdm, AUDIO2_CLK_GATE_EN0, 1, a1_pclk_pdata); >> +AUDIO_GATE(audio2_tdmin_vad, AUDIO2_CLK_GATE_EN0, 2, a1_pclk_pdata); >> +AUDIO_GATE(audio2_toddr_vad, AUDIO2_CLK_GATE_EN0, 3, a1_pclk_pdata); >> +AUDIO_GATE(audio2_vad, AUDIO2_CLK_GATE_EN0, 4, a1_pclk_pdata); >> +AUDIO_GATE(audio2_audiotop, AUDIO2_CLK_GATE_EN0, 7, a1_pclk_pdata); >> + >> +static const struct clk_parent_data a1_mst_pdata[] = { >> + { .fw_name = "dds_in" }, >> + { .fw_name = "fclk_div2" }, >> + { .fw_name = "fclk_div3" }, >> + { .fw_name = "hifi_pll" }, >> + { .fw_name = "xtal" }, >> +}; >> + >> +#define AUDIO_MST_MCLK(_name, _reg) \ >> + AUDIO_MUX(_name##_mux, (_reg), 0x7, 24, a1_mst_pdata); \ >> + AUDIO_DIV(_name##_div, (_reg), 0, 16, \ >> + AUDIO_PDATA(_name##_mux)); \ >> + AUDIO_GATE(_name, (_reg), 31, AUDIO_PDATA(_name##_div)) >> + >> +AUDIO_MST_MCLK(audio_mst_a_mclk, AUDIO_MCLK_A_CTRL); >> +AUDIO_MST_MCLK(audio_mst_b_mclk, AUDIO_MCLK_B_CTRL); >> +AUDIO_MST_MCLK(audio_mst_c_mclk, AUDIO_MCLK_C_CTRL); >> +AUDIO_MST_MCLK(audio_mst_d_mclk, AUDIO_MCLK_D_CTRL); >> +AUDIO_MST_MCLK(audio_spdifin_clk, AUDIO_CLK_SPDIFIN_CTRL); >> +AUDIO_MST_MCLK(audio_eqdrc_clk, AUDIO_CLK_EQDRC_CTRL); >> + >> +AUDIO_MUX(audio_resample_clk_mux, AUDIO_CLK_RESAMPLE_CTRL, 0xf, 24, >> + a1_mst_pdata); >> +AUDIO_DIV(audio_resample_clk_div, AUDIO_CLK_RESAMPLE_CTRL, 0, 8, >> + AUDIO_PDATA(audio_resample_clk_mux)); >> +AUDIO_GATE(audio_resample_clk, AUDIO_CLK_RESAMPLE_CTRL, 31, >> + AUDIO_PDATA(audio_resample_clk_div)); >> + >> +AUDIO_MUX(audio_locker_in_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 8, >> + a1_mst_pdata); >> +AUDIO_DIV(audio_locker_in_clk_div, AUDIO_CLK_LOCKER_CTRL, 0, 8, >> + AUDIO_PDATA(audio_locker_in_clk_mux)); >> +AUDIO_GATE(audio_locker_in_clk, AUDIO_CLK_LOCKER_CTRL, 15, >> + AUDIO_PDATA(audio_locker_in_clk_div)); >> + >> +AUDIO_MUX(audio_locker_out_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 24, >> + a1_mst_pdata); >> +AUDIO_DIV(audio_locker_out_clk_div, AUDIO_CLK_LOCKER_CTRL, 16, 8, >> + AUDIO_PDATA(audio_locker_out_clk_mux)); >> +AUDIO_GATE(audio_locker_out_clk, AUDIO_CLK_LOCKER_CTRL, 31, >> + AUDIO_PDATA(audio_locker_out_clk_div)); >> + >> +AUDIO_MST_MCLK(audio2_vad_mclk, AUDIO2_MCLK_VAD_CTRL); >> +AUDIO_MST_MCLK(audio2_vad_clk, AUDIO2_CLK_VAD_CTRL); >> +AUDIO_MST_MCLK(audio2_pdm_dclk, AUDIO2_CLK_PDMIN_CTRL0); >> +AUDIO_MST_MCLK(audio2_pdm_sysclk, AUDIO2_CLK_PDMIN_CTRL1); >> + >> +#define AUDIO_MST_SCLK(_name, _reg0, _reg1, _pdata) \ >> + AUDIO_GATE(_name##_pre_en, (_reg0), 31, (_pdata)); \ >> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 20, 10, 0, 0, \ >> + AUDIO_PDATA(_name##_pre_en), true); \ >> + AUDIO_GATE(_name##_post_en, (_reg0), 30, \ >> + AUDIO_PDATA(_name##_div)); \ >> + AUDIO_TRIPHASE(_name, (_reg1), 1, 0, 2, 4, \ >> + AUDIO_PDATA(_name##_post_en)) >> + > > Again, I'm not a fan of this many levels of macro. I can live with it > but certainly don't want the burden of reviewing and maintaining for > clock driver. AXG / G12 and A1 are obviously closely related, so make it common. > >> +#define AUDIO_MST_LRCLK(_name, _reg0, _reg1, _pdata) \ >> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 0, 10, 10, 10, \ >> + (_pdata), false); \ >> + AUDIO_TRIPHASE(_name, (_reg1), 1, 1, 3, 5, \ >> + AUDIO_PDATA(_name##_div)) >> + >> +AUDIO_MST_SCLK(audio_mst_a_sclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >> + AUDIO_PDATA(audio_mst_a_mclk)); >> +AUDIO_MST_SCLK(audio_mst_b_sclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >> + AUDIO_PDATA(audio_mst_b_mclk)); >> +AUDIO_MST_SCLK(audio_mst_c_sclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >> + AUDIO_PDATA(audio_mst_c_mclk)); >> +AUDIO_MST_SCLK(audio_mst_d_sclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >> + AUDIO_PDATA(audio_mst_d_mclk)); >> + >> +AUDIO_MST_LRCLK(audio_mst_a_lrclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >> + AUDIO_PDATA(audio_mst_a_sclk_post_en)); >> +AUDIO_MST_LRCLK(audio_mst_b_lrclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >> + AUDIO_PDATA(audio_mst_b_sclk_post_en)); >> +AUDIO_MST_LRCLK(audio_mst_c_lrclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >> + AUDIO_PDATA(audio_mst_c_sclk_post_en)); >> +AUDIO_MST_LRCLK(audio_mst_d_lrclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >> + AUDIO_PDATA(audio_mst_d_sclk_post_en)); >> + >> +static const struct clk_parent_data a1_mst_sclk_pdata[] = { >> + { .hw = &audio_mst_a_sclk.hw }, >> + { .hw = &audio_mst_b_sclk.hw }, >> + { .hw = &audio_mst_c_sclk.hw }, >> + { .hw = &audio_mst_d_sclk.hw }, >> + { .fw_name = "slv_sclk0" }, >> + { .fw_name = "slv_sclk1" }, >> + { .fw_name = "slv_sclk2" }, >> + { .fw_name = "slv_sclk3" }, >> + { .fw_name = "slv_sclk4" }, >> + { .fw_name = "slv_sclk5" }, >> + { .fw_name = "slv_sclk6" }, >> + { .fw_name = "slv_sclk7" }, >> + { .fw_name = "slv_sclk8" }, >> + { .fw_name = "slv_sclk9" }, >> +}; >> + >> +static const struct clk_parent_data a1_mst_lrclk_pdata[] = { >> + { .hw = &audio_mst_a_lrclk.hw }, >> + { .hw = &audio_mst_b_lrclk.hw }, >> + { .hw = &audio_mst_c_lrclk.hw }, >> + { .hw = &audio_mst_d_lrclk.hw }, >> + { .fw_name = "slv_lrclk0" }, >> + { .fw_name = "slv_lrclk1" }, >> + { .fw_name = "slv_lrclk2" }, >> + { .fw_name = "slv_lrclk3" }, >> + { .fw_name = "slv_lrclk4" }, >> + { .fw_name = "slv_lrclk5" }, >> + { .fw_name = "slv_lrclk6" }, >> + { .fw_name = "slv_lrclk7" }, >> + { .fw_name = "slv_lrclk8" }, >> + { .fw_name = "slv_lrclk9" }, >> +}; >> + >> +#define AUDIO_TDM_SCLK(_name, _reg) \ >> + AUDIO_MUX(_name##_mux, (_reg), 0xf, 24, a1_mst_sclk_pdata); \ >> + AUDIO_GATE(_name##_pre_en, (_reg), 31, \ >> + AUDIO_PDATA(_name##_mux)); \ >> + AUDIO_GATE(_name##_post_en, (_reg), 30, \ >> + AUDIO_PDATA(_name##_pre_en)); \ >> + AUDIO_SCLK_WS(_name, (_reg), 1, 29, 28, \ >> + AUDIO_PDATA(_name##_post_en)) >> + >> +#define AUDIO_TDM_LRCLK(_name, _reg) \ >> + AUDIO_MUX(_name, (_reg), 0xf, 20, a1_mst_lrclk_pdata) >> + >> +AUDIO_TDM_SCLK(audio_tdmin_a_sclk, AUDIO_CLK_TDMIN_A_CTRL); >> +AUDIO_TDM_SCLK(audio_tdmin_b_sclk, AUDIO_CLK_TDMIN_B_CTRL); >> +AUDIO_TDM_SCLK(audio_tdmin_lb_sclk, AUDIO_CLK_TDMIN_LB_CTRL); >> +AUDIO_TDM_SCLK(audio_tdmout_a_sclk, AUDIO_CLK_TDMOUT_A_CTRL); >> +AUDIO_TDM_SCLK(audio_tdmout_b_sclk, AUDIO_CLK_TDMOUT_B_CTRL); >> + >> +AUDIO_TDM_LRCLK(audio_tdmin_a_lrclk, AUDIO_CLK_TDMIN_A_CTRL); >> +AUDIO_TDM_LRCLK(audio_tdmin_b_lrclk, AUDIO_CLK_TDMIN_B_CTRL); >> +AUDIO_TDM_LRCLK(audio_tdmin_lb_lrclk, AUDIO_CLK_TDMIN_LB_CTRL); >> +AUDIO_TDM_LRCLK(audio_tdmout_a_lrclk, AUDIO_CLK_TDMOUT_A_CTRL); >> +AUDIO_TDM_LRCLK(audio_tdmout_b_lrclk, AUDIO_CLK_TDMOUT_B_CTRL); >> + >> +static struct clk_hw *a1_audio_hw_clks[] = { >> + [AUD_CLKID_DDR_ARB] = &audio_ddr_arb.hw, >> + [AUD_CLKID_TDMIN_A] = &audio_tdmin_a.hw, >> + [AUD_CLKID_TDMIN_B] = &audio_tdmin_b.hw, >> + [AUD_CLKID_TDMIN_LB] = &audio_tdmin_lb.hw, >> + [AUD_CLKID_LOOPBACK] = &audio_loopback.hw, >> + [AUD_CLKID_TDMOUT_A] = &audio_tdmout_a.hw, >> + [AUD_CLKID_TDMOUT_B] = &audio_tdmout_b.hw, >> + [AUD_CLKID_FRDDR_A] = &audio_frddr_a.hw, >> + [AUD_CLKID_FRDDR_B] = &audio_frddr_b.hw, >> + [AUD_CLKID_TODDR_A] = &audio_toddr_a.hw, >> + [AUD_CLKID_TODDR_B] = &audio_toddr_b.hw, >> + [AUD_CLKID_SPDIFIN] = &audio_spdifin.hw, >> + [AUD_CLKID_RESAMPLE] = &audio_resample.hw, >> + [AUD_CLKID_EQDRC] = &audio_eqdrc.hw, >> + [AUD_CLKID_LOCKER] = &audio_audiolocker.hw, >> + [AUD_CLKID_MST_A_MCLK_SEL] = &audio_mst_a_mclk_mux.hw, >> + [AUD_CLKID_MST_A_MCLK_DIV] = &audio_mst_a_mclk_div.hw, >> + [AUD_CLKID_MST_A_MCLK] = &audio_mst_a_mclk.hw, >> + [AUD_CLKID_MST_B_MCLK_SEL] = &audio_mst_b_mclk_mux.hw, >> + [AUD_CLKID_MST_B_MCLK_DIV] = &audio_mst_b_mclk_div.hw, >> + [AUD_CLKID_MST_B_MCLK] = &audio_mst_b_mclk.hw, >> + [AUD_CLKID_MST_C_MCLK_SEL] = &audio_mst_c_mclk_mux.hw, >> + [AUD_CLKID_MST_C_MCLK_DIV] = &audio_mst_c_mclk_div.hw, >> + [AUD_CLKID_MST_C_MCLK] = &audio_mst_c_mclk.hw, >> + [AUD_CLKID_MST_D_MCLK_SEL] = &audio_mst_d_mclk_mux.hw, >> + [AUD_CLKID_MST_D_MCLK_DIV] = &audio_mst_d_mclk_div.hw, >> + [AUD_CLKID_MST_D_MCLK] = &audio_mst_d_mclk.hw, >> + [AUD_CLKID_RESAMPLE_CLK_SEL] = &audio_resample_clk_mux.hw, >> + [AUD_CLKID_RESAMPLE_CLK_DIV] = &audio_resample_clk_div.hw, >> + [AUD_CLKID_RESAMPLE_CLK] = &audio_resample_clk.hw, >> + [AUD_CLKID_LOCKER_IN_CLK_SEL] = &audio_locker_in_clk_mux.hw, >> + [AUD_CLKID_LOCKER_IN_CLK_DIV] = &audio_locker_in_clk_div.hw, >> + [AUD_CLKID_LOCKER_IN_CLK] = &audio_locker_in_clk.hw, >> + [AUD_CLKID_LOCKER_OUT_CLK_SEL] = &audio_locker_out_clk_mux.hw, >> + [AUD_CLKID_LOCKER_OUT_CLK_DIV] = &audio_locker_out_clk_div.hw, >> + [AUD_CLKID_LOCKER_OUT_CLK] = &audio_locker_out_clk.hw, >> + [AUD_CLKID_SPDIFIN_CLK_SEL] = &audio_spdifin_clk_mux.hw, >> + [AUD_CLKID_SPDIFIN_CLK_DIV] = &audio_spdifin_clk_div.hw, >> + [AUD_CLKID_SPDIFIN_CLK] = &audio_spdifin_clk.hw, >> + [AUD_CLKID_EQDRC_CLK_SEL] = &audio_eqdrc_clk_mux.hw, >> + [AUD_CLKID_EQDRC_CLK_DIV] = &audio_eqdrc_clk_div.hw, >> + [AUD_CLKID_EQDRC_CLK] = &audio_eqdrc_clk.hw, >> + [AUD_CLKID_MST_A_SCLK_PRE_EN] = &audio_mst_a_sclk_pre_en.hw, >> + [AUD_CLKID_MST_A_SCLK_DIV] = &audio_mst_a_sclk_div.hw, >> + [AUD_CLKID_MST_A_SCLK_POST_EN] = &audio_mst_a_sclk_post_en.hw, >> + [AUD_CLKID_MST_A_SCLK] = &audio_mst_a_sclk.hw, >> + [AUD_CLKID_MST_B_SCLK_PRE_EN] = &audio_mst_b_sclk_pre_en.hw, >> + [AUD_CLKID_MST_B_SCLK_DIV] = &audio_mst_b_sclk_div.hw, >> + [AUD_CLKID_MST_B_SCLK_POST_EN] = &audio_mst_b_sclk_post_en.hw, >> + [AUD_CLKID_MST_B_SCLK] = &audio_mst_b_sclk.hw, >> + [AUD_CLKID_MST_C_SCLK_PRE_EN] = &audio_mst_c_sclk_pre_en.hw, >> + [AUD_CLKID_MST_C_SCLK_DIV] = &audio_mst_c_sclk_div.hw, >> + [AUD_CLKID_MST_C_SCLK_POST_EN] = &audio_mst_c_sclk_post_en.hw, >> + [AUD_CLKID_MST_C_SCLK] = &audio_mst_c_sclk.hw, >> + [AUD_CLKID_MST_D_SCLK_PRE_EN] = &audio_mst_d_sclk_pre_en.hw, >> + [AUD_CLKID_MST_D_SCLK_DIV] = &audio_mst_d_sclk_div.hw, >> + [AUD_CLKID_MST_D_SCLK_POST_EN] = &audio_mst_d_sclk_post_en.hw, >> + [AUD_CLKID_MST_D_SCLK] = &audio_mst_d_sclk.hw, >> + [AUD_CLKID_MST_A_LRCLK_DIV] = &audio_mst_a_lrclk_div.hw, >> + [AUD_CLKID_MST_A_LRCLK] = &audio_mst_a_lrclk.hw, >> + [AUD_CLKID_MST_B_LRCLK_DIV] = &audio_mst_b_lrclk_div.hw, >> + [AUD_CLKID_MST_B_LRCLK] = &audio_mst_b_lrclk.hw, >> + [AUD_CLKID_MST_C_LRCLK_DIV] = &audio_mst_c_lrclk_div.hw, >> + [AUD_CLKID_MST_C_LRCLK] = &audio_mst_c_lrclk.hw, >> + [AUD_CLKID_MST_D_LRCLK_DIV] = &audio_mst_d_lrclk_div.hw, >> + [AUD_CLKID_MST_D_LRCLK] = &audio_mst_d_lrclk.hw, >> + [AUD_CLKID_TDMIN_A_SCLK_SEL] = &audio_tdmin_a_sclk_mux.hw, >> + [AUD_CLKID_TDMIN_A_SCLK_PRE_EN] = &audio_tdmin_a_sclk_pre_en.hw, >> + [AUD_CLKID_TDMIN_A_SCLK_POST_EN] = &audio_tdmin_a_sclk_post_en.hw, >> + [AUD_CLKID_TDMIN_A_SCLK] = &audio_tdmin_a_sclk.hw, >> + [AUD_CLKID_TDMIN_A_LRCLK] = &audio_tdmin_a_lrclk.hw, >> + [AUD_CLKID_TDMIN_B_SCLK_SEL] = &audio_tdmin_b_sclk_mux.hw, >> + [AUD_CLKID_TDMIN_B_SCLK_PRE_EN] = &audio_tdmin_b_sclk_pre_en.hw, >> + [AUD_CLKID_TDMIN_B_SCLK_POST_EN] = &audio_tdmin_b_sclk_post_en.hw, >> + [AUD_CLKID_TDMIN_B_SCLK] = &audio_tdmin_b_sclk.hw, >> + [AUD_CLKID_TDMIN_B_LRCLK] = &audio_tdmin_b_lrclk.hw, >> + [AUD_CLKID_TDMIN_LB_SCLK_SEL] = &audio_tdmin_lb_sclk_mux.hw, >> + [AUD_CLKID_TDMIN_LB_SCLK_PRE_EN] = &audio_tdmin_lb_sclk_pre_en.hw, >> + [AUD_CLKID_TDMIN_LB_SCLK_POST_EN] = &audio_tdmin_lb_sclk_post_en.hw, >> + [AUD_CLKID_TDMIN_LB_SCLK] = &audio_tdmin_lb_sclk.hw, >> + [AUD_CLKID_TDMIN_LB_LRCLK] = &audio_tdmin_lb_lrclk.hw, >> + [AUD_CLKID_TDMOUT_A_SCLK_SEL] = &audio_tdmout_a_sclk_mux.hw, >> + [AUD_CLKID_TDMOUT_A_SCLK_PRE_EN] = &audio_tdmout_a_sclk_pre_en.hw, >> + [AUD_CLKID_TDMOUT_A_SCLK_POST_EN] = &audio_tdmout_a_sclk_post_en.hw, >> + [AUD_CLKID_TDMOUT_A_SCLK] = &audio_tdmout_a_sclk.hw, >> + [AUD_CLKID_TDMOUT_A_LRCLK] = &audio_tdmout_a_lrclk.hw, >> + [AUD_CLKID_TDMOUT_B_SCLK_SEL] = &audio_tdmout_b_sclk_mux.hw, >> + [AUD_CLKID_TDMOUT_B_SCLK_PRE_EN] = &audio_tdmout_b_sclk_pre_en.hw, >> + [AUD_CLKID_TDMOUT_B_SCLK_POST_EN] = &audio_tdmout_b_sclk_post_en.hw, >> + [AUD_CLKID_TDMOUT_B_SCLK] = &audio_tdmout_b_sclk.hw, >> + [AUD_CLKID_TDMOUT_B_LRCLK] = &audio_tdmout_b_lrclk.hw, >> + >> + [AUD2_CLKID_DDR_ARB] = &audio2_ddr_arb.hw, >> + [AUD2_CLKID_PDM] = &audio2_pdm.hw, >> + [AUD2_CLKID_TDMIN_VAD] = &audio2_tdmin_vad.hw, >> + [AUD2_CLKID_TODDR_VAD] = &audio2_toddr_vad.hw, >> + [AUD2_CLKID_VAD] = &audio2_vad.hw, >> + [AUD2_CLKID_AUDIOTOP] = &audio2_audiotop.hw, >> + [AUD2_CLKID_VAD_MCLK_SEL] = &audio2_vad_mclk_mux.hw, >> + [AUD2_CLKID_VAD_MCLK_DIV] = &audio2_vad_mclk_div.hw, >> + [AUD2_CLKID_VAD_MCLK] = &audio2_vad_mclk.hw, >> + [AUD2_CLKID_VAD_CLK_SEL] = &audio2_vad_clk_mux.hw, >> + [AUD2_CLKID_VAD_CLK_DIV] = &audio2_vad_clk_div.hw, >> + [AUD2_CLKID_VAD_CLK] = &audio2_vad_clk.hw, >> + [AUD2_CLKID_PDM_DCLK_SEL] = &audio2_pdm_dclk_mux.hw, >> + [AUD2_CLKID_PDM_DCLK_DIV] = &audio2_pdm_dclk_div.hw, >> + [AUD2_CLKID_PDM_DCLK] = &audio2_pdm_dclk.hw, >> + [AUD2_CLKID_PDM_SYSCLK_SEL] = &audio2_pdm_sysclk_mux.hw, >> + [AUD2_CLKID_PDM_SYSCLK_DIV] = &audio2_pdm_sysclk_div.hw, >> + [AUD2_CLKID_PDM_SYSCLK] = &audio2_pdm_sysclk.hw, >> +}; >> + >> +static struct meson_clk_hw_data a1_audio_clks = { >> + .hws = a1_audio_hw_clks, >> + .num = ARRAY_SIZE(a1_audio_hw_clks), >> +}; >> + >> +static struct regmap *a1_audio_map(struct platform_device *pdev, >> + unsigned int index) >> +{ >> + char name[32]; >> + const struct regmap_config cfg = { >> + .reg_bits = 32, >> + .val_bits = 32, >> + .reg_stride = 4, >> + .name = name, > > Not necessary > This implementation uses two regmaps, and this field allow to avoid errors like this: [ 0.145530] debugfs: Directory 'fe050000.audio-clock-controller' with parent 'regmap' already present! >> + }; >> + void __iomem *base; >> + >> + base = devm_platform_ioremap_resource(pdev, index); >> + if (IS_ERR(base)) >> + return base; >> + >> + scnprintf(name, sizeof(name), "%d", index); >> + return devm_regmap_init_mmio(&pdev->dev, base, &cfg); >> +} > > That is overengineered. Please keep it simple. Declare the regmap_config > as static const global, and do it like axg-audio please. > This only reason why it is not "static const" because I need to set unique name for each regmap. >> + >> +static int a1_register_clk(struct platform_device *pdev, >> + struct regmap *map0, struct regmap *map1, >> + struct clk_hw *hw) >> +{ >> + struct clk_regmap *clk = container_of(hw, struct clk_regmap, hw); >> + >> + if (!hw) >> + return 0; >> + >> + switch ((unsigned long)clk->map) { >> + case AUDIO_RANGE_0: >> + clk->map = map0; >> + break; >> + case AUDIO_RANGE_1: >> + clk->map = map1; >> + break; > > ... fishy > >> + default: >> + WARN_ON(1); >> + return -EINVAL; >> + } >> + >> + return devm_clk_hw_register(&pdev->dev, hw); >> +} >> + >> +static int a1_audio_clkc_probe(struct platform_device *pdev) >> +{ >> + struct regmap *map0, *map1; >> + struct clk *clk; >> + unsigned int i; >> + int ret; >> + >> + clk = devm_clk_get_enabled(&pdev->dev, "pclk"); >> + if (WARN_ON(IS_ERR(clk))) >> + return PTR_ERR(clk); >> + >> + map0 = a1_audio_map(pdev, 0); >> + if (IS_ERR(map0)) >> + return PTR_ERR(map0); >> + >> + map1 = a1_audio_map(pdev, 1); >> + if (IS_ERR(map1)) >> + return PTR_ERR(map1); > > No - Looks to me you just have two clock controllers you are trying > force into one. > See the begining. >> + >> + /* >> + * Register and enable AUD2_CLKID_AUDIOTOP clock first. Unless >> + * it is enabled any read/write to 'map0' hangs the CPU. >> + */ >> + >> + ret = a1_register_clk(pdev, map0, map1, >> + a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]); >> + if (ret) >> + return ret; >> + >> + ret = clk_prepare_enable(a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]->clk); >> + if (ret) >> + return ret; > > Again, this shows 2 devices. The one related to your 'map0' should > request AUD2_CLKID_AUDIOTOP as input and enable it right away. > See the begining. >> + >> + for (i = 0; i < a1_audio_clks.num; i++) { >> + if (i == AUD2_CLKID_AUDIOTOP) >> + continue; >> + >> + ret = a1_register_clk(pdev, map0, map1, a1_audio_clks.hws[i]); >> + if (ret) >> + return ret; >> + } >> + >> + ret = devm_of_clk_add_hw_provider(&pdev->dev, meson_clk_hw_get, >> + &a1_audio_clks); >> + if (ret) >> + return ret; >> + >> + BUILD_BUG_ON((unsigned long)AUDIO_REG_MAP(AUDIO_SW_RESET0) != >> + AUDIO_RANGE_0); > > Why is that necessary ? > A little paranoia. Here AUDIO_SW_RESET0 is handled as map0's register, and I want to assert it. >> + return meson_audio_rstc_register(&pdev->dev, map0, >> + AUDIO_REG_OFFSET(AUDIO_SW_RESET0), 32); >> +} >> + >> +static const struct of_device_id a1_audio_clkc_match_table[] = { >> + { .compatible = "amlogic,a1-audio-clkc", }, >> + {} >> +}; >> +MODULE_DEVICE_TABLE(of, a1_audio_clkc_match_table); >> + >> +static struct platform_driver a1_audio_clkc_driver = { >> + .probe = a1_audio_clkc_probe, >> + .driver = { >> + .name = "a1-audio-clkc", >> + .of_match_table = a1_audio_clkc_match_table, >> + }, >> +}; >> +module_platform_driver(a1_audio_clkc_driver); >> + >> +MODULE_DESCRIPTION("Amlogic A1 Audio Clock driver"); >> +MODULE_AUTHOR("Jan Dakinevich <jan.dakinevich@salutedevices.com>"); >> +MODULE_LICENSE("GPL"); >> diff --git a/drivers/clk/meson/a1-audio.h b/drivers/clk/meson/a1-audio.h >> new file mode 100644 >> index 000000000000..f994e87276cd >> --- /dev/null >> +++ b/drivers/clk/meson/a1-audio.h >> @@ -0,0 +1,58 @@ >> +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ >> +/* >> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >> + * >> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >> + */ >> + >> +#ifndef __A1_AUDIO_H >> +#define __A1_AUDIO_H >> + >> +#define AUDIO_RANGE_0 0xa >> +#define AUDIO_RANGE_1 0xb >> +#define AUDIO_RANGE_SHIFT 16 >> + >> +#define AUDIO_REG(_range, _offset) \ >> + (((_range) << AUDIO_RANGE_SHIFT) + (_offset)) >> + >> +#define AUDIO_REG_OFFSET(_reg) \ >> + ((_reg) & ((1 << AUDIO_RANGE_SHIFT) - 1)) >> + >> +#define AUDIO_REG_MAP(_reg) \ >> + ((void *)((_reg) >> AUDIO_RANGE_SHIFT)) > > That is seriouly overengineered. > The following are offset. Just write what they are. > This is all in order to keep range's identifier together with offset and then use it to store the identifier in clk_regmaps. > There is not reason to put that into a header. It is only going to be > used by a single driver. > >> + >> +#define AUDIO_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_0, 0x000) >> +#define AUDIO_MCLK_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x008) >> +#define AUDIO_MCLK_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x00c) >> +#define AUDIO_MCLK_C_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x010) >> +#define AUDIO_MCLK_D_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x014) >> +#define AUDIO_MCLK_E_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x018) >> +#define AUDIO_MCLK_F_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x01c) >> +#define AUDIO_SW_RESET0 AUDIO_REG(AUDIO_RANGE_0, 0x028) >> +#define AUDIO_MST_A_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x040) >> +#define AUDIO_MST_A_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x044) >> +#define AUDIO_MST_B_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x048) >> +#define AUDIO_MST_B_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x04c) >> +#define AUDIO_MST_C_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x050) >> +#define AUDIO_MST_C_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x054) >> +#define AUDIO_MST_D_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x058) >> +#define AUDIO_MST_D_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x05c) >> +#define AUDIO_CLK_TDMIN_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x080) >> +#define AUDIO_CLK_TDMIN_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x084) >> +#define AUDIO_CLK_TDMIN_LB_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x08c) >> +#define AUDIO_CLK_TDMOUT_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x090) >> +#define AUDIO_CLK_TDMOUT_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x094) >> +#define AUDIO_CLK_SPDIFIN_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x09c) >> +#define AUDIO_CLK_RESAMPLE_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a4) >> +#define AUDIO_CLK_LOCKER_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a8) >> +#define AUDIO_CLK_EQDRC_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0c0) >> + >> +#define AUDIO2_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_1, 0x00c) >> +#define AUDIO2_MCLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x040) >> +#define AUDIO2_CLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x044) >> +#define AUDIO2_CLK_PDMIN_CTRL0 AUDIO_REG(AUDIO_RANGE_1, 0x058) >> +#define AUDIO2_CLK_PDMIN_CTRL1 AUDIO_REG(AUDIO_RANGE_1, 0x05c) >> + >> +#include <dt-bindings/clock/amlogic,a1-audio-clkc.h> >> + >> +#endif /* __A1_AUDIO_H */ > >
On 17/03/2024 16:52, Jan Dakinevich wrote: > > > On 3/15/24 13:22, Jerome Brunet wrote: >> >> On Fri 15 Mar 2024 at 11:00, Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> wrote: >> >>> On 15/03/2024 00:21, Jan Dakinevich wrote: >>>> This option allow to redefine the rate of DSP system clock. >>> >>> And why is it suitable for bindings? Describe the hardware, not what you >>> want to do in the driver. >>> >>>> >>>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>> --- >>>> Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml | 4 ++++ >>>> 1 file changed, 4 insertions(+) >>>> >>>> diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>> index df21dd72fc65..d2f23a59a6b6 100644 >>>> --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>> +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml >>>> @@ -40,6 +40,10 @@ properties: >>>> resets: >>>> maxItems: 1 >>>> >>>> + sysrate: >>>> + $ref: /schemas/types.yaml#/definitions/uint32 >>>> + description: redefine rate of DSP system clock >>> >>> No vendor prefix, so is it a generic property? Also, missing unit >>> suffix, but more importantly I don't understand why this is a property >>> of hardware. >> >> +1. >> >> The appropriate way to set rate of the clock before the driver take over >> is 'assigned-rate', if you need to customize this for different >> platform. >> > > It would be great, but it doesn't work. Below, is what I want to see: > > assigned-clocks = > <&clkc_audio AUD2_CLKID_PDM_SYSCLK_SEL>, > <&clkc_audio AUD2_CLKID_PDM_SYSCLK_DIV>; > assigned-clock-parents = > <&clkc_pll CLKID_FCLK_DIV3>, > <0>; > assigned-clock-rates = > <0>, > <256000000>; > > But regardles of this declaration, PDM's driver unconditionally sets That's driver's problem. You do not change bindings, just because your driver behaves differently. Just fix driver. > sysclk'rate to 250MHz and throws away everything that was configured > before, reparents audio2_pdm_sysclk_mux to hifi_pll and changes > hifi_pll's rate. > > This value 250MHz is declared here: > > static const struct axg_pdm_cfg axg_pdm_config = { > .filters = &axg_default_filters, > .sys_rate = 250000000, > }; > > The property 'sysrate' is intended to redefine hardcoded 'sys_rate' > value in 'axg_pdm_config'. What does it have to do with bindings? Change driver if you are not happy how it operates. Best regards, Krzysztof
On Tue 19 Mar 2024 at 04:47, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > Let's start from the end: > >> No - Looks to me you just have two clock controllers you are trying > force into one. > >> Again, this shows 2 devices. The one related to your 'map0' should > request AUD2_CLKID_AUDIOTOP as input and enable it right away. > > Most of fishy workarounds that you commented is caused the fact the mmio > of this clock controller is divided into two parts. Compare it with > axg-audio driver, things that was part of contigous memory region (like > pdm) here are moved to second region. Is this enough to make a guess > that these are two devices? I see obsolutely no reason to think it is a single device nor to add all the quirks you have the way you did. So yes, in that case, 2 zones, 2 devices. > > Concerning AUD2_CLKID_AUDIOTOP clock, as it turned out, it must be > enabled before enabling of clocks from second region too. That is > AUD2_CLKID_AUDIOTOP clock feeds both parts of this clock controller. > Yes. I understood the first time around and already commented on that. > > On 3/15/24 12:20, Jerome Brunet wrote: >> >> On Fri 15 Mar 2024 at 02:21, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >> >>> This controller provides clocks and reset functionality for audio >>> peripherals on Amlogic A1 SoC family. >>> >>> The driver is almost identical to 'axg-audio', however it would be better >>> to keep it separate due to following reasons: >>> >>> - significant amount of bits has another definition. I will bring there >>> a mess of new defines with A1_ suffixes. >>> >>> - registers of this controller are located in two separate regions. It >>> will give a lot of complications for 'axg-audio' to support this. >>> >>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>> --- >>> drivers/clk/meson/Kconfig | 13 + >>> drivers/clk/meson/Makefile | 1 + >>> drivers/clk/meson/a1-audio.c | 556 +++++++++++++++++++++++++++++++++++ >>> drivers/clk/meson/a1-audio.h | 58 ++++ >>> 4 files changed, 628 insertions(+) >>> create mode 100644 drivers/clk/meson/a1-audio.c >>> create mode 100644 drivers/clk/meson/a1-audio.h >>> >>> diff --git a/drivers/clk/meson/Kconfig b/drivers/clk/meson/Kconfig >>> index d6a2fa5f7e88..80c4a18c83d2 100644 >>> --- a/drivers/clk/meson/Kconfig >>> +++ b/drivers/clk/meson/Kconfig >>> @@ -133,6 +133,19 @@ config COMMON_CLK_A1_PERIPHERALS >>> device, A1 SoC Family. Say Y if you want A1 Peripherals clock >>> controller to work. >>> >>> +config COMMON_CLK_A1_AUDIO >>> + tristate "Amlogic A1 SoC Audio clock controller support" >>> + depends on ARM64 >>> + select COMMON_CLK_MESON_REGMAP >>> + select COMMON_CLK_MESON_CLKC_UTILS >>> + select COMMON_CLK_MESON_PHASE >>> + select COMMON_CLK_MESON_SCLK_DIV >>> + select COMMON_CLK_MESON_AUDIO_RSTC >>> + help >>> + Support for the Audio clock controller on Amlogic A113L based >>> + device, A1 SoC Family. Say Y if you want A1 Audio clock controller >>> + to work. >>> + >>> config COMMON_CLK_G12A >>> tristate "G12 and SM1 SoC clock controllers support" >>> depends on ARM64 >>> diff --git a/drivers/clk/meson/Makefile b/drivers/clk/meson/Makefile >>> index 88d94921a4dc..4968fc7ad555 100644 >>> --- a/drivers/clk/meson/Makefile >>> +++ b/drivers/clk/meson/Makefile >>> @@ -20,6 +20,7 @@ obj-$(CONFIG_COMMON_CLK_AXG) += axg.o axg-aoclk.o >>> obj-$(CONFIG_COMMON_CLK_AXG_AUDIO) += axg-audio.o >>> obj-$(CONFIG_COMMON_CLK_A1_PLL) += a1-pll.o >>> obj-$(CONFIG_COMMON_CLK_A1_PERIPHERALS) += a1-peripherals.o >>> +obj-$(CONFIG_COMMON_CLK_A1_AUDIO) += a1-audio.o >>> obj-$(CONFIG_COMMON_CLK_GXBB) += gxbb.o gxbb-aoclk.o >>> obj-$(CONFIG_COMMON_CLK_G12A) += g12a.o g12a-aoclk.o >>> obj-$(CONFIG_COMMON_CLK_MESON8B) += meson8b.o meson8-ddr.o >>> diff --git a/drivers/clk/meson/a1-audio.c b/drivers/clk/meson/a1-audio.c >>> new file mode 100644 >>> index 000000000000..6039116c93ba >>> --- /dev/null >>> +++ b/drivers/clk/meson/a1-audio.c >>> @@ -0,0 +1,556 @@ >>> +// SPDX-License-Identifier: (GPL-2.0 OR MIT) >>> +/* >>> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >>> + * >>> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>> + */ >>> + >>> +#include <linux/clk.h> >>> +#include <linux/clk-provider.h> >>> +#include <linux/init.h> >>> +#include <linux/of_device.h> >>> +#include <linux/module.h> >>> +#include <linux/platform_device.h> >>> +#include <linux/regmap.h> >>> +#include <linux/reset.h> >>> +#include <linux/reset-controller.h> >>> +#include <linux/slab.h> >>> + >>> +#include "meson-clkc-utils.h" >>> +#include "meson-audio-rstc.h" >>> +#include "clk-regmap.h" >>> +#include "clk-phase.h" >>> +#include "sclk-div.h" >>> +#include "a1-audio.h" >>> + >>> +#define AUDIO_PDATA(_name) \ >>> + ((const struct clk_parent_data[]) { { .hw = &(_name).hw } }) >> >> Not a fan - yet another level of macro. >> >>> + >>> +#define AUDIO_MUX(_name, _reg, _mask, _shift, _pdata) \ >>> +static struct clk_regmap _name = { \ >>> + .map = AUDIO_REG_MAP(_reg), \ >>> + .data = &(struct clk_regmap_mux_data){ \ >>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>> + .mask = (_mask), \ >>> + .shift = (_shift), \ >>> + }, \ >>> + .hw.init = &(struct clk_init_data) { \ >>> + .name = #_name, \ >>> + .ops = &clk_regmap_mux_ops, \ >>> + .parent_data = (_pdata), \ >>> + .num_parents = ARRAY_SIZE(_pdata), \ >>> + .flags = CLK_SET_RATE_PARENT, \ >>> + }, \ >>> +} >>> + >>> +#define AUDIO_DIV(_name, _reg, _shift, _width, _pdata) \ >>> +static struct clk_regmap _name = { \ >>> + .map = AUDIO_REG_MAP(_reg), \ >>> + .data = &(struct clk_regmap_div_data){ \ >>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>> + .shift = (_shift), \ >>> + .width = (_width), \ >>> + }, \ >>> + .hw.init = &(struct clk_init_data) { \ >>> + .name = #_name, \ >>> + .ops = &clk_regmap_divider_ops, \ >>> + .parent_data = (_pdata), \ >>> + .num_parents = 1, \ >>> + .flags = CLK_SET_RATE_PARENT, \ >>> + }, \ >>> +} >>> + >>> +#define AUDIO_GATE(_name, _reg, _bit, _pdata) \ >>> +static struct clk_regmap _name = { \ >>> + .map = AUDIO_REG_MAP(_reg), \ >>> + .data = &(struct clk_regmap_gate_data){ \ >>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>> + .bit_idx = (_bit), \ >>> + }, \ >>> + .hw.init = &(struct clk_init_data) { \ >>> + .name = #_name, \ >>> + .ops = &clk_regmap_gate_ops, \ >>> + .parent_data = (_pdata), \ >>> + .num_parents = 1, \ >>> + .flags = CLK_SET_RATE_PARENT, \ >>> + }, \ >>> +} >>> + >>> +#define AUDIO_SCLK_DIV(_name, _reg, _div_shift, _div_width, \ >>> + _hi_shift, _hi_width, _pdata, _set_rate_parent) \ >>> +static struct clk_regmap _name = { \ >>> + .map = AUDIO_REG_MAP(_reg), \ >>> + .data = &(struct meson_sclk_div_data) { \ >>> + .div = { \ >>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>> + .shift = (_div_shift), \ >>> + .width = (_div_width), \ >>> + }, \ >>> + .hi = { \ >>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>> + .shift = (_hi_shift), \ >>> + .width = (_hi_width), \ >>> + }, \ >>> + }, \ >>> + .hw.init = &(struct clk_init_data) { \ >>> + .name = #_name, \ >>> + .ops = &meson_sclk_div_ops, \ >>> + .parent_data = (_pdata), \ >>> + .num_parents = 1, \ >>> + .flags = (_set_rate_parent) ? CLK_SET_RATE_PARENT : 0, \ >> >> Does not help readeability. Just pass the flag as axg-audio does. >> >>> + }, \ >>> +} >>> + >>> +#define AUDIO_TRIPHASE(_name, _reg, _width, _shift0, _shift1, _shift2, \ >>> + _pdata) \ >>> +static struct clk_regmap _name = { \ >>> + .map = AUDIO_REG_MAP(_reg), \ >>> + .data = &(struct meson_clk_triphase_data) { \ >>> + .ph0 = { \ >>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>> + .shift = (_shift0), \ >>> + .width = (_width), \ >>> + }, \ >>> + .ph1 = { \ >>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>> + .shift = (_shift1), \ >>> + .width = (_width), \ >>> + }, \ >>> + .ph2 = { \ >>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>> + .shift = (_shift2), \ >>> + .width = (_width), \ >>> + }, \ >>> + }, \ >>> + .hw.init = &(struct clk_init_data) { \ >>> + .name = #_name, \ >>> + .ops = &meson_clk_triphase_ops, \ >>> + .parent_data = (_pdata), \ >>> + .num_parents = 1, \ >>> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >>> + }, \ >>> +} >>> + >>> +#define AUDIO_SCLK_WS(_name, _reg, _width, _shift_ph, _shift_ws, \ >>> + _pdata) \ >>> +static struct clk_regmap _name = { \ >>> + .map = AUDIO_REG_MAP(_reg), \ >>> + .data = &(struct meson_sclk_ws_inv_data) { \ >>> + .ph = { \ >>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>> + .shift = (_shift_ph), \ >>> + .width = (_width), \ >>> + }, \ >>> + .ws = { \ >>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>> + .shift = (_shift_ws), \ >>> + .width = (_width), \ >>> + }, \ >>> + }, \ >>> + .hw.init = &(struct clk_init_data) { \ >>> + .name = #_name, \ >>> + .ops = &meson_sclk_ws_inv_ops, \ >>> + .parent_data = (_pdata), \ >>> + .num_parents = 1, \ >>> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >>> + }, \ >>> +} >> >> All the above does essentially the same things as the macro of >> axg-audio, to some minor differences. Yet it is another set to maintain. >> > > Except one thing... Here I keep memory identifier to which this clock > belongs: > > .map = AUDIO_REG_MAP(_reg), > > It is workaround, but ->map the only common field in clk_regmap that > could be used for this purpose. > > >> I'd much prefer if you put the axg-audio macro in a header a re-used >> those. There would a single set to maintain. You may then specialize the >> included in the driver C file, to avoid redundant parameters >> >> Rework axg-audio to use clk_parent_data if you must, but not in the same >> series please. >> >>> + >>> +static const struct clk_parent_data a1_pclk_pdata[] = { >>> + { .fw_name = "pclk", }, >>> +}; >>> + >>> +AUDIO_GATE(audio_ddr_arb, AUDIO_CLK_GATE_EN0, 0, a1_pclk_pdata); >>> +AUDIO_GATE(audio_tdmin_a, AUDIO_CLK_GATE_EN0, 1, a1_pclk_pdata); >>> +AUDIO_GATE(audio_tdmin_b, AUDIO_CLK_GATE_EN0, 2, a1_pclk_pdata); >>> +AUDIO_GATE(audio_tdmin_lb, AUDIO_CLK_GATE_EN0, 3, a1_pclk_pdata); >>> +AUDIO_GATE(audio_loopback, AUDIO_CLK_GATE_EN0, 4, a1_pclk_pdata); >>> +AUDIO_GATE(audio_tdmout_a, AUDIO_CLK_GATE_EN0, 5, a1_pclk_pdata); >>> +AUDIO_GATE(audio_tdmout_b, AUDIO_CLK_GATE_EN0, 6, a1_pclk_pdata); >>> +AUDIO_GATE(audio_frddr_a, AUDIO_CLK_GATE_EN0, 7, a1_pclk_pdata); >>> +AUDIO_GATE(audio_frddr_b, AUDIO_CLK_GATE_EN0, 8, a1_pclk_pdata); >>> +AUDIO_GATE(audio_toddr_a, AUDIO_CLK_GATE_EN0, 9, a1_pclk_pdata); >>> +AUDIO_GATE(audio_toddr_b, AUDIO_CLK_GATE_EN0, 10, a1_pclk_pdata); >>> +AUDIO_GATE(audio_spdifin, AUDIO_CLK_GATE_EN0, 11, a1_pclk_pdata); >>> +AUDIO_GATE(audio_resample, AUDIO_CLK_GATE_EN0, 12, a1_pclk_pdata); >>> +AUDIO_GATE(audio_eqdrc, AUDIO_CLK_GATE_EN0, 13, a1_pclk_pdata); >>> +AUDIO_GATE(audio_audiolocker, AUDIO_CLK_GATE_EN0, 14, a1_pclk_pdata); >> This is what I mean by redundant parameter ^ >> > > Yep. I could define something like AUDIO_PCLK_GATE(). > >>> + >>> +AUDIO_GATE(audio2_ddr_arb, AUDIO2_CLK_GATE_EN0, 0, a1_pclk_pdata); >>> +AUDIO_GATE(audio2_pdm, AUDIO2_CLK_GATE_EN0, 1, a1_pclk_pdata); >>> +AUDIO_GATE(audio2_tdmin_vad, AUDIO2_CLK_GATE_EN0, 2, a1_pclk_pdata); >>> +AUDIO_GATE(audio2_toddr_vad, AUDIO2_CLK_GATE_EN0, 3, a1_pclk_pdata); >>> +AUDIO_GATE(audio2_vad, AUDIO2_CLK_GATE_EN0, 4, a1_pclk_pdata); >>> +AUDIO_GATE(audio2_audiotop, AUDIO2_CLK_GATE_EN0, 7, a1_pclk_pdata); >>> + >>> +static const struct clk_parent_data a1_mst_pdata[] = { >>> + { .fw_name = "dds_in" }, >>> + { .fw_name = "fclk_div2" }, >>> + { .fw_name = "fclk_div3" }, >>> + { .fw_name = "hifi_pll" }, >>> + { .fw_name = "xtal" }, >>> +}; >>> + >>> +#define AUDIO_MST_MCLK(_name, _reg) \ >>> + AUDIO_MUX(_name##_mux, (_reg), 0x7, 24, a1_mst_pdata); \ >>> + AUDIO_DIV(_name##_div, (_reg), 0, 16, \ >>> + AUDIO_PDATA(_name##_mux)); \ >>> + AUDIO_GATE(_name, (_reg), 31, AUDIO_PDATA(_name##_div)) >>> + >>> +AUDIO_MST_MCLK(audio_mst_a_mclk, AUDIO_MCLK_A_CTRL); >>> +AUDIO_MST_MCLK(audio_mst_b_mclk, AUDIO_MCLK_B_CTRL); >>> +AUDIO_MST_MCLK(audio_mst_c_mclk, AUDIO_MCLK_C_CTRL); >>> +AUDIO_MST_MCLK(audio_mst_d_mclk, AUDIO_MCLK_D_CTRL); >>> +AUDIO_MST_MCLK(audio_spdifin_clk, AUDIO_CLK_SPDIFIN_CTRL); >>> +AUDIO_MST_MCLK(audio_eqdrc_clk, AUDIO_CLK_EQDRC_CTRL); >>> + >>> +AUDIO_MUX(audio_resample_clk_mux, AUDIO_CLK_RESAMPLE_CTRL, 0xf, 24, >>> + a1_mst_pdata); >>> +AUDIO_DIV(audio_resample_clk_div, AUDIO_CLK_RESAMPLE_CTRL, 0, 8, >>> + AUDIO_PDATA(audio_resample_clk_mux)); >>> +AUDIO_GATE(audio_resample_clk, AUDIO_CLK_RESAMPLE_CTRL, 31, >>> + AUDIO_PDATA(audio_resample_clk_div)); >>> + >>> +AUDIO_MUX(audio_locker_in_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 8, >>> + a1_mst_pdata); >>> +AUDIO_DIV(audio_locker_in_clk_div, AUDIO_CLK_LOCKER_CTRL, 0, 8, >>> + AUDIO_PDATA(audio_locker_in_clk_mux)); >>> +AUDIO_GATE(audio_locker_in_clk, AUDIO_CLK_LOCKER_CTRL, 15, >>> + AUDIO_PDATA(audio_locker_in_clk_div)); >>> + >>> +AUDIO_MUX(audio_locker_out_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 24, >>> + a1_mst_pdata); >>> +AUDIO_DIV(audio_locker_out_clk_div, AUDIO_CLK_LOCKER_CTRL, 16, 8, >>> + AUDIO_PDATA(audio_locker_out_clk_mux)); >>> +AUDIO_GATE(audio_locker_out_clk, AUDIO_CLK_LOCKER_CTRL, 31, >>> + AUDIO_PDATA(audio_locker_out_clk_div)); >>> + >>> +AUDIO_MST_MCLK(audio2_vad_mclk, AUDIO2_MCLK_VAD_CTRL); >>> +AUDIO_MST_MCLK(audio2_vad_clk, AUDIO2_CLK_VAD_CTRL); >>> +AUDIO_MST_MCLK(audio2_pdm_dclk, AUDIO2_CLK_PDMIN_CTRL0); >>> +AUDIO_MST_MCLK(audio2_pdm_sysclk, AUDIO2_CLK_PDMIN_CTRL1); >>> + >>> +#define AUDIO_MST_SCLK(_name, _reg0, _reg1, _pdata) \ >>> + AUDIO_GATE(_name##_pre_en, (_reg0), 31, (_pdata)); \ >>> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 20, 10, 0, 0, \ >>> + AUDIO_PDATA(_name##_pre_en), true); \ >>> + AUDIO_GATE(_name##_post_en, (_reg0), 30, \ >>> + AUDIO_PDATA(_name##_div)); \ >>> + AUDIO_TRIPHASE(_name, (_reg1), 1, 0, 2, 4, \ >>> + AUDIO_PDATA(_name##_post_en)) >>> + >> >> Again, I'm not a fan of this many levels of macro. I can live with it >> but certainly don't want the burden of reviewing and maintaining for >> clock driver. AXG / G12 and A1 are obviously closely related, so make it common. >> >>> +#define AUDIO_MST_LRCLK(_name, _reg0, _reg1, _pdata) \ >>> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 0, 10, 10, 10, \ >>> + (_pdata), false); \ >>> + AUDIO_TRIPHASE(_name, (_reg1), 1, 1, 3, 5, \ >>> + AUDIO_PDATA(_name##_div)) >>> + >>> +AUDIO_MST_SCLK(audio_mst_a_sclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >>> + AUDIO_PDATA(audio_mst_a_mclk)); >>> +AUDIO_MST_SCLK(audio_mst_b_sclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >>> + AUDIO_PDATA(audio_mst_b_mclk)); >>> +AUDIO_MST_SCLK(audio_mst_c_sclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >>> + AUDIO_PDATA(audio_mst_c_mclk)); >>> +AUDIO_MST_SCLK(audio_mst_d_sclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >>> + AUDIO_PDATA(audio_mst_d_mclk)); >>> + >>> +AUDIO_MST_LRCLK(audio_mst_a_lrclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >>> + AUDIO_PDATA(audio_mst_a_sclk_post_en)); >>> +AUDIO_MST_LRCLK(audio_mst_b_lrclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >>> + AUDIO_PDATA(audio_mst_b_sclk_post_en)); >>> +AUDIO_MST_LRCLK(audio_mst_c_lrclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >>> + AUDIO_PDATA(audio_mst_c_sclk_post_en)); >>> +AUDIO_MST_LRCLK(audio_mst_d_lrclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >>> + AUDIO_PDATA(audio_mst_d_sclk_post_en)); >>> + >>> +static const struct clk_parent_data a1_mst_sclk_pdata[] = { >>> + { .hw = &audio_mst_a_sclk.hw }, >>> + { .hw = &audio_mst_b_sclk.hw }, >>> + { .hw = &audio_mst_c_sclk.hw }, >>> + { .hw = &audio_mst_d_sclk.hw }, >>> + { .fw_name = "slv_sclk0" }, >>> + { .fw_name = "slv_sclk1" }, >>> + { .fw_name = "slv_sclk2" }, >>> + { .fw_name = "slv_sclk3" }, >>> + { .fw_name = "slv_sclk4" }, >>> + { .fw_name = "slv_sclk5" }, >>> + { .fw_name = "slv_sclk6" }, >>> + { .fw_name = "slv_sclk7" }, >>> + { .fw_name = "slv_sclk8" }, >>> + { .fw_name = "slv_sclk9" }, >>> +}; >>> + >>> +static const struct clk_parent_data a1_mst_lrclk_pdata[] = { >>> + { .hw = &audio_mst_a_lrclk.hw }, >>> + { .hw = &audio_mst_b_lrclk.hw }, >>> + { .hw = &audio_mst_c_lrclk.hw }, >>> + { .hw = &audio_mst_d_lrclk.hw }, >>> + { .fw_name = "slv_lrclk0" }, >>> + { .fw_name = "slv_lrclk1" }, >>> + { .fw_name = "slv_lrclk2" }, >>> + { .fw_name = "slv_lrclk3" }, >>> + { .fw_name = "slv_lrclk4" }, >>> + { .fw_name = "slv_lrclk5" }, >>> + { .fw_name = "slv_lrclk6" }, >>> + { .fw_name = "slv_lrclk7" }, >>> + { .fw_name = "slv_lrclk8" }, >>> + { .fw_name = "slv_lrclk9" }, >>> +}; >>> + >>> +#define AUDIO_TDM_SCLK(_name, _reg) \ >>> + AUDIO_MUX(_name##_mux, (_reg), 0xf, 24, a1_mst_sclk_pdata); \ >>> + AUDIO_GATE(_name##_pre_en, (_reg), 31, \ >>> + AUDIO_PDATA(_name##_mux)); \ >>> + AUDIO_GATE(_name##_post_en, (_reg), 30, \ >>> + AUDIO_PDATA(_name##_pre_en)); \ >>> + AUDIO_SCLK_WS(_name, (_reg), 1, 29, 28, \ >>> + AUDIO_PDATA(_name##_post_en)) >>> + >>> +#define AUDIO_TDM_LRCLK(_name, _reg) \ >>> + AUDIO_MUX(_name, (_reg), 0xf, 20, a1_mst_lrclk_pdata) >>> + >>> +AUDIO_TDM_SCLK(audio_tdmin_a_sclk, AUDIO_CLK_TDMIN_A_CTRL); >>> +AUDIO_TDM_SCLK(audio_tdmin_b_sclk, AUDIO_CLK_TDMIN_B_CTRL); >>> +AUDIO_TDM_SCLK(audio_tdmin_lb_sclk, AUDIO_CLK_TDMIN_LB_CTRL); >>> +AUDIO_TDM_SCLK(audio_tdmout_a_sclk, AUDIO_CLK_TDMOUT_A_CTRL); >>> +AUDIO_TDM_SCLK(audio_tdmout_b_sclk, AUDIO_CLK_TDMOUT_B_CTRL); >>> + >>> +AUDIO_TDM_LRCLK(audio_tdmin_a_lrclk, AUDIO_CLK_TDMIN_A_CTRL); >>> +AUDIO_TDM_LRCLK(audio_tdmin_b_lrclk, AUDIO_CLK_TDMIN_B_CTRL); >>> +AUDIO_TDM_LRCLK(audio_tdmin_lb_lrclk, AUDIO_CLK_TDMIN_LB_CTRL); >>> +AUDIO_TDM_LRCLK(audio_tdmout_a_lrclk, AUDIO_CLK_TDMOUT_A_CTRL); >>> +AUDIO_TDM_LRCLK(audio_tdmout_b_lrclk, AUDIO_CLK_TDMOUT_B_CTRL); >>> + >>> +static struct clk_hw *a1_audio_hw_clks[] = { >>> + [AUD_CLKID_DDR_ARB] = &audio_ddr_arb.hw, >>> + [AUD_CLKID_TDMIN_A] = &audio_tdmin_a.hw, >>> + [AUD_CLKID_TDMIN_B] = &audio_tdmin_b.hw, >>> + [AUD_CLKID_TDMIN_LB] = &audio_tdmin_lb.hw, >>> + [AUD_CLKID_LOOPBACK] = &audio_loopback.hw, >>> + [AUD_CLKID_TDMOUT_A] = &audio_tdmout_a.hw, >>> + [AUD_CLKID_TDMOUT_B] = &audio_tdmout_b.hw, >>> + [AUD_CLKID_FRDDR_A] = &audio_frddr_a.hw, >>> + [AUD_CLKID_FRDDR_B] = &audio_frddr_b.hw, >>> + [AUD_CLKID_TODDR_A] = &audio_toddr_a.hw, >>> + [AUD_CLKID_TODDR_B] = &audio_toddr_b.hw, >>> + [AUD_CLKID_SPDIFIN] = &audio_spdifin.hw, >>> + [AUD_CLKID_RESAMPLE] = &audio_resample.hw, >>> + [AUD_CLKID_EQDRC] = &audio_eqdrc.hw, >>> + [AUD_CLKID_LOCKER] = &audio_audiolocker.hw, >>> + [AUD_CLKID_MST_A_MCLK_SEL] = &audio_mst_a_mclk_mux.hw, >>> + [AUD_CLKID_MST_A_MCLK_DIV] = &audio_mst_a_mclk_div.hw, >>> + [AUD_CLKID_MST_A_MCLK] = &audio_mst_a_mclk.hw, >>> + [AUD_CLKID_MST_B_MCLK_SEL] = &audio_mst_b_mclk_mux.hw, >>> + [AUD_CLKID_MST_B_MCLK_DIV] = &audio_mst_b_mclk_div.hw, >>> + [AUD_CLKID_MST_B_MCLK] = &audio_mst_b_mclk.hw, >>> + [AUD_CLKID_MST_C_MCLK_SEL] = &audio_mst_c_mclk_mux.hw, >>> + [AUD_CLKID_MST_C_MCLK_DIV] = &audio_mst_c_mclk_div.hw, >>> + [AUD_CLKID_MST_C_MCLK] = &audio_mst_c_mclk.hw, >>> + [AUD_CLKID_MST_D_MCLK_SEL] = &audio_mst_d_mclk_mux.hw, >>> + [AUD_CLKID_MST_D_MCLK_DIV] = &audio_mst_d_mclk_div.hw, >>> + [AUD_CLKID_MST_D_MCLK] = &audio_mst_d_mclk.hw, >>> + [AUD_CLKID_RESAMPLE_CLK_SEL] = &audio_resample_clk_mux.hw, >>> + [AUD_CLKID_RESAMPLE_CLK_DIV] = &audio_resample_clk_div.hw, >>> + [AUD_CLKID_RESAMPLE_CLK] = &audio_resample_clk.hw, >>> + [AUD_CLKID_LOCKER_IN_CLK_SEL] = &audio_locker_in_clk_mux.hw, >>> + [AUD_CLKID_LOCKER_IN_CLK_DIV] = &audio_locker_in_clk_div.hw, >>> + [AUD_CLKID_LOCKER_IN_CLK] = &audio_locker_in_clk.hw, >>> + [AUD_CLKID_LOCKER_OUT_CLK_SEL] = &audio_locker_out_clk_mux.hw, >>> + [AUD_CLKID_LOCKER_OUT_CLK_DIV] = &audio_locker_out_clk_div.hw, >>> + [AUD_CLKID_LOCKER_OUT_CLK] = &audio_locker_out_clk.hw, >>> + [AUD_CLKID_SPDIFIN_CLK_SEL] = &audio_spdifin_clk_mux.hw, >>> + [AUD_CLKID_SPDIFIN_CLK_DIV] = &audio_spdifin_clk_div.hw, >>> + [AUD_CLKID_SPDIFIN_CLK] = &audio_spdifin_clk.hw, >>> + [AUD_CLKID_EQDRC_CLK_SEL] = &audio_eqdrc_clk_mux.hw, >>> + [AUD_CLKID_EQDRC_CLK_DIV] = &audio_eqdrc_clk_div.hw, >>> + [AUD_CLKID_EQDRC_CLK] = &audio_eqdrc_clk.hw, >>> + [AUD_CLKID_MST_A_SCLK_PRE_EN] = &audio_mst_a_sclk_pre_en.hw, >>> + [AUD_CLKID_MST_A_SCLK_DIV] = &audio_mst_a_sclk_div.hw, >>> + [AUD_CLKID_MST_A_SCLK_POST_EN] = &audio_mst_a_sclk_post_en.hw, >>> + [AUD_CLKID_MST_A_SCLK] = &audio_mst_a_sclk.hw, >>> + [AUD_CLKID_MST_B_SCLK_PRE_EN] = &audio_mst_b_sclk_pre_en.hw, >>> + [AUD_CLKID_MST_B_SCLK_DIV] = &audio_mst_b_sclk_div.hw, >>> + [AUD_CLKID_MST_B_SCLK_POST_EN] = &audio_mst_b_sclk_post_en.hw, >>> + [AUD_CLKID_MST_B_SCLK] = &audio_mst_b_sclk.hw, >>> + [AUD_CLKID_MST_C_SCLK_PRE_EN] = &audio_mst_c_sclk_pre_en.hw, >>> + [AUD_CLKID_MST_C_SCLK_DIV] = &audio_mst_c_sclk_div.hw, >>> + [AUD_CLKID_MST_C_SCLK_POST_EN] = &audio_mst_c_sclk_post_en.hw, >>> + [AUD_CLKID_MST_C_SCLK] = &audio_mst_c_sclk.hw, >>> + [AUD_CLKID_MST_D_SCLK_PRE_EN] = &audio_mst_d_sclk_pre_en.hw, >>> + [AUD_CLKID_MST_D_SCLK_DIV] = &audio_mst_d_sclk_div.hw, >>> + [AUD_CLKID_MST_D_SCLK_POST_EN] = &audio_mst_d_sclk_post_en.hw, >>> + [AUD_CLKID_MST_D_SCLK] = &audio_mst_d_sclk.hw, >>> + [AUD_CLKID_MST_A_LRCLK_DIV] = &audio_mst_a_lrclk_div.hw, >>> + [AUD_CLKID_MST_A_LRCLK] = &audio_mst_a_lrclk.hw, >>> + [AUD_CLKID_MST_B_LRCLK_DIV] = &audio_mst_b_lrclk_div.hw, >>> + [AUD_CLKID_MST_B_LRCLK] = &audio_mst_b_lrclk.hw, >>> + [AUD_CLKID_MST_C_LRCLK_DIV] = &audio_mst_c_lrclk_div.hw, >>> + [AUD_CLKID_MST_C_LRCLK] = &audio_mst_c_lrclk.hw, >>> + [AUD_CLKID_MST_D_LRCLK_DIV] = &audio_mst_d_lrclk_div.hw, >>> + [AUD_CLKID_MST_D_LRCLK] = &audio_mst_d_lrclk.hw, >>> + [AUD_CLKID_TDMIN_A_SCLK_SEL] = &audio_tdmin_a_sclk_mux.hw, >>> + [AUD_CLKID_TDMIN_A_SCLK_PRE_EN] = &audio_tdmin_a_sclk_pre_en.hw, >>> + [AUD_CLKID_TDMIN_A_SCLK_POST_EN] = &audio_tdmin_a_sclk_post_en.hw, >>> + [AUD_CLKID_TDMIN_A_SCLK] = &audio_tdmin_a_sclk.hw, >>> + [AUD_CLKID_TDMIN_A_LRCLK] = &audio_tdmin_a_lrclk.hw, >>> + [AUD_CLKID_TDMIN_B_SCLK_SEL] = &audio_tdmin_b_sclk_mux.hw, >>> + [AUD_CLKID_TDMIN_B_SCLK_PRE_EN] = &audio_tdmin_b_sclk_pre_en.hw, >>> + [AUD_CLKID_TDMIN_B_SCLK_POST_EN] = &audio_tdmin_b_sclk_post_en.hw, >>> + [AUD_CLKID_TDMIN_B_SCLK] = &audio_tdmin_b_sclk.hw, >>> + [AUD_CLKID_TDMIN_B_LRCLK] = &audio_tdmin_b_lrclk.hw, >>> + [AUD_CLKID_TDMIN_LB_SCLK_SEL] = &audio_tdmin_lb_sclk_mux.hw, >>> + [AUD_CLKID_TDMIN_LB_SCLK_PRE_EN] = &audio_tdmin_lb_sclk_pre_en.hw, >>> + [AUD_CLKID_TDMIN_LB_SCLK_POST_EN] = &audio_tdmin_lb_sclk_post_en.hw, >>> + [AUD_CLKID_TDMIN_LB_SCLK] = &audio_tdmin_lb_sclk.hw, >>> + [AUD_CLKID_TDMIN_LB_LRCLK] = &audio_tdmin_lb_lrclk.hw, >>> + [AUD_CLKID_TDMOUT_A_SCLK_SEL] = &audio_tdmout_a_sclk_mux.hw, >>> + [AUD_CLKID_TDMOUT_A_SCLK_PRE_EN] = &audio_tdmout_a_sclk_pre_en.hw, >>> + [AUD_CLKID_TDMOUT_A_SCLK_POST_EN] = &audio_tdmout_a_sclk_post_en.hw, >>> + [AUD_CLKID_TDMOUT_A_SCLK] = &audio_tdmout_a_sclk.hw, >>> + [AUD_CLKID_TDMOUT_A_LRCLK] = &audio_tdmout_a_lrclk.hw, >>> + [AUD_CLKID_TDMOUT_B_SCLK_SEL] = &audio_tdmout_b_sclk_mux.hw, >>> + [AUD_CLKID_TDMOUT_B_SCLK_PRE_EN] = &audio_tdmout_b_sclk_pre_en.hw, >>> + [AUD_CLKID_TDMOUT_B_SCLK_POST_EN] = &audio_tdmout_b_sclk_post_en.hw, >>> + [AUD_CLKID_TDMOUT_B_SCLK] = &audio_tdmout_b_sclk.hw, >>> + [AUD_CLKID_TDMOUT_B_LRCLK] = &audio_tdmout_b_lrclk.hw, >>> + >>> + [AUD2_CLKID_DDR_ARB] = &audio2_ddr_arb.hw, >>> + [AUD2_CLKID_PDM] = &audio2_pdm.hw, >>> + [AUD2_CLKID_TDMIN_VAD] = &audio2_tdmin_vad.hw, >>> + [AUD2_CLKID_TODDR_VAD] = &audio2_toddr_vad.hw, >>> + [AUD2_CLKID_VAD] = &audio2_vad.hw, >>> + [AUD2_CLKID_AUDIOTOP] = &audio2_audiotop.hw, >>> + [AUD2_CLKID_VAD_MCLK_SEL] = &audio2_vad_mclk_mux.hw, >>> + [AUD2_CLKID_VAD_MCLK_DIV] = &audio2_vad_mclk_div.hw, >>> + [AUD2_CLKID_VAD_MCLK] = &audio2_vad_mclk.hw, >>> + [AUD2_CLKID_VAD_CLK_SEL] = &audio2_vad_clk_mux.hw, >>> + [AUD2_CLKID_VAD_CLK_DIV] = &audio2_vad_clk_div.hw, >>> + [AUD2_CLKID_VAD_CLK] = &audio2_vad_clk.hw, >>> + [AUD2_CLKID_PDM_DCLK_SEL] = &audio2_pdm_dclk_mux.hw, >>> + [AUD2_CLKID_PDM_DCLK_DIV] = &audio2_pdm_dclk_div.hw, >>> + [AUD2_CLKID_PDM_DCLK] = &audio2_pdm_dclk.hw, >>> + [AUD2_CLKID_PDM_SYSCLK_SEL] = &audio2_pdm_sysclk_mux.hw, >>> + [AUD2_CLKID_PDM_SYSCLK_DIV] = &audio2_pdm_sysclk_div.hw, >>> + [AUD2_CLKID_PDM_SYSCLK] = &audio2_pdm_sysclk.hw, >>> +}; >>> + >>> +static struct meson_clk_hw_data a1_audio_clks = { >>> + .hws = a1_audio_hw_clks, >>> + .num = ARRAY_SIZE(a1_audio_hw_clks), >>> +}; >>> + >>> +static struct regmap *a1_audio_map(struct platform_device *pdev, >>> + unsigned int index) >>> +{ >>> + char name[32]; >>> + const struct regmap_config cfg = { >>> + .reg_bits = 32, >>> + .val_bits = 32, >>> + .reg_stride = 4, >>> + .name = name, >> >> Not necessary >> > > This implementation uses two regmaps, and this field allow to avoid > errors like this: > > [ 0.145530] debugfs: Directory 'fe050000.audio-clock-controller' with > parent 'regmap' already present! > >>> + }; >>> + void __iomem *base; >>> + >>> + base = devm_platform_ioremap_resource(pdev, index); >>> + if (IS_ERR(base)) >>> + return base; >>> + >>> + scnprintf(name, sizeof(name), "%d", index); >>> + return devm_regmap_init_mmio(&pdev->dev, base, &cfg); >>> +} >> >> That is overengineered. Please keep it simple. Declare the regmap_config >> as static const global, and do it like axg-audio please. >> > > This only reason why it is not "static const" because I need to set > unique name for each regmap. > >>> + >>> +static int a1_register_clk(struct platform_device *pdev, >>> + struct regmap *map0, struct regmap *map1, >>> + struct clk_hw *hw) >>> +{ >>> + struct clk_regmap *clk = container_of(hw, struct clk_regmap, hw); >>> + >>> + if (!hw) >>> + return 0; >>> + >>> + switch ((unsigned long)clk->map) { >>> + case AUDIO_RANGE_0: >>> + clk->map = map0; >>> + break; >>> + case AUDIO_RANGE_1: >>> + clk->map = map1; >>> + break; >> >> ... fishy >> >>> + default: >>> + WARN_ON(1); >>> + return -EINVAL; >>> + } >>> + >>> + return devm_clk_hw_register(&pdev->dev, hw); >>> +} >>> + >>> +static int a1_audio_clkc_probe(struct platform_device *pdev) >>> +{ >>> + struct regmap *map0, *map1; >>> + struct clk *clk; >>> + unsigned int i; >>> + int ret; >>> + >>> + clk = devm_clk_get_enabled(&pdev->dev, "pclk"); >>> + if (WARN_ON(IS_ERR(clk))) >>> + return PTR_ERR(clk); >>> + >>> + map0 = a1_audio_map(pdev, 0); >>> + if (IS_ERR(map0)) >>> + return PTR_ERR(map0); >>> + >>> + map1 = a1_audio_map(pdev, 1); >>> + if (IS_ERR(map1)) >>> + return PTR_ERR(map1); >> >> No - Looks to me you just have two clock controllers you are trying >> force into one. >> > > See the begining. > >>> + >>> + /* >>> + * Register and enable AUD2_CLKID_AUDIOTOP clock first. Unless >>> + * it is enabled any read/write to 'map0' hangs the CPU. >>> + */ >>> + >>> + ret = a1_register_clk(pdev, map0, map1, >>> + a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]); >>> + if (ret) >>> + return ret; >>> + >>> + ret = clk_prepare_enable(a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]->clk); >>> + if (ret) >>> + return ret; >> >> Again, this shows 2 devices. The one related to your 'map0' should >> request AUD2_CLKID_AUDIOTOP as input and enable it right away. >> > > See the begining. > >>> + >>> + for (i = 0; i < a1_audio_clks.num; i++) { >>> + if (i == AUD2_CLKID_AUDIOTOP) >>> + continue; >>> + >>> + ret = a1_register_clk(pdev, map0, map1, a1_audio_clks.hws[i]); >>> + if (ret) >>> + return ret; >>> + } >>> + >>> + ret = devm_of_clk_add_hw_provider(&pdev->dev, meson_clk_hw_get, >>> + &a1_audio_clks); >>> + if (ret) >>> + return ret; >>> + >>> + BUILD_BUG_ON((unsigned long)AUDIO_REG_MAP(AUDIO_SW_RESET0) != >>> + AUDIO_RANGE_0); >> >> Why is that necessary ? >> > > A little paranoia. Here AUDIO_SW_RESET0 is handled as map0's register, > and I want to assert it. > >>> + return meson_audio_rstc_register(&pdev->dev, map0, >>> + AUDIO_REG_OFFSET(AUDIO_SW_RESET0), 32); >>> +} >>> + >>> +static const struct of_device_id a1_audio_clkc_match_table[] = { >>> + { .compatible = "amlogic,a1-audio-clkc", }, >>> + {} >>> +}; >>> +MODULE_DEVICE_TABLE(of, a1_audio_clkc_match_table); >>> + >>> +static struct platform_driver a1_audio_clkc_driver = { >>> + .probe = a1_audio_clkc_probe, >>> + .driver = { >>> + .name = "a1-audio-clkc", >>> + .of_match_table = a1_audio_clkc_match_table, >>> + }, >>> +}; >>> +module_platform_driver(a1_audio_clkc_driver); >>> + >>> +MODULE_DESCRIPTION("Amlogic A1 Audio Clock driver"); >>> +MODULE_AUTHOR("Jan Dakinevich <jan.dakinevich@salutedevices.com>"); >>> +MODULE_LICENSE("GPL"); >>> diff --git a/drivers/clk/meson/a1-audio.h b/drivers/clk/meson/a1-audio.h >>> new file mode 100644 >>> index 000000000000..f994e87276cd >>> --- /dev/null >>> +++ b/drivers/clk/meson/a1-audio.h >>> @@ -0,0 +1,58 @@ >>> +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ >>> +/* >>> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >>> + * >>> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>> + */ >>> + >>> +#ifndef __A1_AUDIO_H >>> +#define __A1_AUDIO_H >>> + >>> +#define AUDIO_RANGE_0 0xa >>> +#define AUDIO_RANGE_1 0xb >>> +#define AUDIO_RANGE_SHIFT 16 >>> + >>> +#define AUDIO_REG(_range, _offset) \ >>> + (((_range) << AUDIO_RANGE_SHIFT) + (_offset)) >>> + >>> +#define AUDIO_REG_OFFSET(_reg) \ >>> + ((_reg) & ((1 << AUDIO_RANGE_SHIFT) - 1)) >>> + >>> +#define AUDIO_REG_MAP(_reg) \ >>> + ((void *)((_reg) >> AUDIO_RANGE_SHIFT)) >> >> That is seriouly overengineered. >> The following are offset. Just write what they are. >> > > This is all in order to keep range's identifier together with offset and > then use it to store the identifier in clk_regmaps. > >> There is not reason to put that into a header. It is only going to be >> used by a single driver. >> >> + >>> +#define AUDIO_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_0, 0x000) >>> +#define AUDIO_MCLK_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x008) >>> +#define AUDIO_MCLK_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x00c) >>> +#define AUDIO_MCLK_C_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x010) >>> +#define AUDIO_MCLK_D_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x014) >>> +#define AUDIO_MCLK_E_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x018) >>> +#define AUDIO_MCLK_F_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x01c) >>> +#define AUDIO_SW_RESET0 AUDIO_REG(AUDIO_RANGE_0, 0x028) >>> +#define AUDIO_MST_A_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x040) >>> +#define AUDIO_MST_A_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x044) >>> +#define AUDIO_MST_B_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x048) >>> +#define AUDIO_MST_B_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x04c) >>> +#define AUDIO_MST_C_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x050) >>> +#define AUDIO_MST_C_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x054) >>> +#define AUDIO_MST_D_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x058) >>> +#define AUDIO_MST_D_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x05c) >>> +#define AUDIO_CLK_TDMIN_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x080) >>> +#define AUDIO_CLK_TDMIN_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x084) >>> +#define AUDIO_CLK_TDMIN_LB_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x08c) >>> +#define AUDIO_CLK_TDMOUT_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x090) >>> +#define AUDIO_CLK_TDMOUT_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x094) >>> +#define AUDIO_CLK_SPDIFIN_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x09c) >>> +#define AUDIO_CLK_RESAMPLE_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a4) >>> +#define AUDIO_CLK_LOCKER_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a8) >>> +#define AUDIO_CLK_EQDRC_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0c0) >>> + >>> +#define AUDIO2_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_1, 0x00c) >>> +#define AUDIO2_MCLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x040) >>> +#define AUDIO2_CLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x044) >>> +#define AUDIO2_CLK_PDMIN_CTRL0 AUDIO_REG(AUDIO_RANGE_1, 0x058) >>> +#define AUDIO2_CLK_PDMIN_CTRL1 AUDIO_REG(AUDIO_RANGE_1, 0x05c) >>> + >>> +#include <dt-bindings/clock/amlogic,a1-audio-clkc.h> >>> + >>> +#endif /* __A1_AUDIO_H */ >> >>
On Sat 23 Mar 2024 at 21:02, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > Jerome, I have reworked my driver reusing axg-audio code as most as I > could and now I have one more question. Lets see on this definition from > axg-audio: > > #define AUD_MST_MUX(_name, _reg, _flag) \ > AUD_MUX(_name##_sel, _reg, 0x7, 24, _flag, \ > mst_mux_parent_data, 0) > > #define AUD_MST_MCLK_MUX(_name, _reg) \ > AUD_MST_MUX(_name, _reg, CLK_MUX_ROUND_CLOSEST) > > CLK_SET_RATE_PARENT is not set here. But why? It means, that topmost pll > clock will not be reconfigured at runtime to satisfy the rate that was > requested from axg-tdm. > Yes, that is by design. It is another area where mainline audio differs greatly from AML vendor code. The PLLs are expected be to fixed and the audio master clock will reparent to the most adequate PLL source depending on the use case. This is how we manage to satisfy all audio interfaces with a very limited number of PLLs On AXG/G12 there is at most 6 concurrent interfaces (3 FRDDR/TODDR) - 8 on sm1 - and we can satisfy on that with 3 PLLs. That would not be possible if interfaces were having their way with the PLLs, reseting it everytime a stream is started. The PLL rate should be carefully chosen so it can be derived easily. On AXG/G12/SM1 that is: * one PLL per rate family, to maximize clock precision * x24 x32: to handle different sample sizes * x2 until we reach the PLL limits to allow higher rates such as 384kHz or even higher If you have less PLLs on A1, you'll have to make compromises, like a less precise clock to support multiple family with one PLL. This is why the PLLs are set for each platform in DT because that choice may depend on the platform use case. > > On 3/19/24 11:30, Jerome Brunet wrote: >> >> On Tue 19 Mar 2024 at 04:47, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >> >>> Let's start from the end: >>> >>>> No - Looks to me you just have two clock controllers you are trying >>> force into one. >>> >>>> Again, this shows 2 devices. The one related to your 'map0' should >>> request AUD2_CLKID_AUDIOTOP as input and enable it right away. >>> >>> Most of fishy workarounds that you commented is caused the fact the mmio >>> of this clock controller is divided into two parts. Compare it with >>> axg-audio driver, things that was part of contigous memory region (like >>> pdm) here are moved to second region. Is this enough to make a guess >>> that these are two devices? >> >> I see obsolutely no reason to think it is a single device nor to add all the quirks >> you have the way you did. So yes, in that case, 2 zones, 2 devices. >> >>> >>> Concerning AUD2_CLKID_AUDIOTOP clock, as it turned out, it must be >>> enabled before enabling of clocks from second region too. That is >>> AUD2_CLKID_AUDIOTOP clock feeds both parts of this clock controller. >>> >> >> Yes. I understood the first time around and already commented on that. >> >>> >>> On 3/15/24 12:20, Jerome Brunet wrote: >>>> >>>> On Fri 15 Mar 2024 at 02:21, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >>>> >>>>> This controller provides clocks and reset functionality for audio >>>>> peripherals on Amlogic A1 SoC family. >>>>> >>>>> The driver is almost identical to 'axg-audio', however it would be better >>>>> to keep it separate due to following reasons: >>>>> >>>>> - significant amount of bits has another definition. I will bring there >>>>> a mess of new defines with A1_ suffixes. >>>>> >>>>> - registers of this controller are located in two separate regions. It >>>>> will give a lot of complications for 'axg-audio' to support this. >>>>> >>>>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>> --- >>>>> drivers/clk/meson/Kconfig | 13 + >>>>> drivers/clk/meson/Makefile | 1 + >>>>> drivers/clk/meson/a1-audio.c | 556 +++++++++++++++++++++++++++++++++++ >>>>> drivers/clk/meson/a1-audio.h | 58 ++++ >>>>> 4 files changed, 628 insertions(+) >>>>> create mode 100644 drivers/clk/meson/a1-audio.c >>>>> create mode 100644 drivers/clk/meson/a1-audio.h >>>>> >>>>> diff --git a/drivers/clk/meson/Kconfig b/drivers/clk/meson/Kconfig >>>>> index d6a2fa5f7e88..80c4a18c83d2 100644 >>>>> --- a/drivers/clk/meson/Kconfig >>>>> +++ b/drivers/clk/meson/Kconfig >>>>> @@ -133,6 +133,19 @@ config COMMON_CLK_A1_PERIPHERALS >>>>> device, A1 SoC Family. Say Y if you want A1 Peripherals clock >>>>> controller to work. >>>>> >>>>> +config COMMON_CLK_A1_AUDIO >>>>> + tristate "Amlogic A1 SoC Audio clock controller support" >>>>> + depends on ARM64 >>>>> + select COMMON_CLK_MESON_REGMAP >>>>> + select COMMON_CLK_MESON_CLKC_UTILS >>>>> + select COMMON_CLK_MESON_PHASE >>>>> + select COMMON_CLK_MESON_SCLK_DIV >>>>> + select COMMON_CLK_MESON_AUDIO_RSTC >>>>> + help >>>>> + Support for the Audio clock controller on Amlogic A113L based >>>>> + device, A1 SoC Family. Say Y if you want A1 Audio clock controller >>>>> + to work. >>>>> + >>>>> config COMMON_CLK_G12A >>>>> tristate "G12 and SM1 SoC clock controllers support" >>>>> depends on ARM64 >>>>> diff --git a/drivers/clk/meson/Makefile b/drivers/clk/meson/Makefile >>>>> index 88d94921a4dc..4968fc7ad555 100644 >>>>> --- a/drivers/clk/meson/Makefile >>>>> +++ b/drivers/clk/meson/Makefile >>>>> @@ -20,6 +20,7 @@ obj-$(CONFIG_COMMON_CLK_AXG) += axg.o axg-aoclk.o >>>>> obj-$(CONFIG_COMMON_CLK_AXG_AUDIO) += axg-audio.o >>>>> obj-$(CONFIG_COMMON_CLK_A1_PLL) += a1-pll.o >>>>> obj-$(CONFIG_COMMON_CLK_A1_PERIPHERALS) += a1-peripherals.o >>>>> +obj-$(CONFIG_COMMON_CLK_A1_AUDIO) += a1-audio.o >>>>> obj-$(CONFIG_COMMON_CLK_GXBB) += gxbb.o gxbb-aoclk.o >>>>> obj-$(CONFIG_COMMON_CLK_G12A) += g12a.o g12a-aoclk.o >>>>> obj-$(CONFIG_COMMON_CLK_MESON8B) += meson8b.o meson8-ddr.o >>>>> diff --git a/drivers/clk/meson/a1-audio.c b/drivers/clk/meson/a1-audio.c >>>>> new file mode 100644 >>>>> index 000000000000..6039116c93ba >>>>> --- /dev/null >>>>> +++ b/drivers/clk/meson/a1-audio.c >>>>> @@ -0,0 +1,556 @@ >>>>> +// SPDX-License-Identifier: (GPL-2.0 OR MIT) >>>>> +/* >>>>> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >>>>> + * >>>>> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>> + */ >>>>> + >>>>> +#include <linux/clk.h> >>>>> +#include <linux/clk-provider.h> >>>>> +#include <linux/init.h> >>>>> +#include <linux/of_device.h> >>>>> +#include <linux/module.h> >>>>> +#include <linux/platform_device.h> >>>>> +#include <linux/regmap.h> >>>>> +#include <linux/reset.h> >>>>> +#include <linux/reset-controller.h> >>>>> +#include <linux/slab.h> >>>>> + >>>>> +#include "meson-clkc-utils.h" >>>>> +#include "meson-audio-rstc.h" >>>>> +#include "clk-regmap.h" >>>>> +#include "clk-phase.h" >>>>> +#include "sclk-div.h" >>>>> +#include "a1-audio.h" >>>>> + >>>>> +#define AUDIO_PDATA(_name) \ >>>>> + ((const struct clk_parent_data[]) { { .hw = &(_name).hw } }) >>>> >>>> Not a fan - yet another level of macro. >>>> >>>>> + >>>>> +#define AUDIO_MUX(_name, _reg, _mask, _shift, _pdata) \ >>>>> +static struct clk_regmap _name = { \ >>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>> + .data = &(struct clk_regmap_mux_data){ \ >>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>> + .mask = (_mask), \ >>>>> + .shift = (_shift), \ >>>>> + }, \ >>>>> + .hw.init = &(struct clk_init_data) { \ >>>>> + .name = #_name, \ >>>>> + .ops = &clk_regmap_mux_ops, \ >>>>> + .parent_data = (_pdata), \ >>>>> + .num_parents = ARRAY_SIZE(_pdata), \ >>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>> + }, \ >>>>> +} >>>>> + >>>>> +#define AUDIO_DIV(_name, _reg, _shift, _width, _pdata) \ >>>>> +static struct clk_regmap _name = { \ >>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>> + .data = &(struct clk_regmap_div_data){ \ >>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>> + .shift = (_shift), \ >>>>> + .width = (_width), \ >>>>> + }, \ >>>>> + .hw.init = &(struct clk_init_data) { \ >>>>> + .name = #_name, \ >>>>> + .ops = &clk_regmap_divider_ops, \ >>>>> + .parent_data = (_pdata), \ >>>>> + .num_parents = 1, \ >>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>> + }, \ >>>>> +} >>>>> + >>>>> +#define AUDIO_GATE(_name, _reg, _bit, _pdata) \ >>>>> +static struct clk_regmap _name = { \ >>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>> + .data = &(struct clk_regmap_gate_data){ \ >>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>> + .bit_idx = (_bit), \ >>>>> + }, \ >>>>> + .hw.init = &(struct clk_init_data) { \ >>>>> + .name = #_name, \ >>>>> + .ops = &clk_regmap_gate_ops, \ >>>>> + .parent_data = (_pdata), \ >>>>> + .num_parents = 1, \ >>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>> + }, \ >>>>> +} >>>>> + >>>>> +#define AUDIO_SCLK_DIV(_name, _reg, _div_shift, _div_width, \ >>>>> + _hi_shift, _hi_width, _pdata, _set_rate_parent) \ >>>>> +static struct clk_regmap _name = { \ >>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>> + .data = &(struct meson_sclk_div_data) { \ >>>>> + .div = { \ >>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>> + .shift = (_div_shift), \ >>>>> + .width = (_div_width), \ >>>>> + }, \ >>>>> + .hi = { \ >>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>> + .shift = (_hi_shift), \ >>>>> + .width = (_hi_width), \ >>>>> + }, \ >>>>> + }, \ >>>>> + .hw.init = &(struct clk_init_data) { \ >>>>> + .name = #_name, \ >>>>> + .ops = &meson_sclk_div_ops, \ >>>>> + .parent_data = (_pdata), \ >>>>> + .num_parents = 1, \ >>>>> + .flags = (_set_rate_parent) ? CLK_SET_RATE_PARENT : 0, \ >>>> >>>> Does not help readeability. Just pass the flag as axg-audio does. >>>> >>>>> + }, \ >>>>> +} >>>>> + >>>>> +#define AUDIO_TRIPHASE(_name, _reg, _width, _shift0, _shift1, _shift2, \ >>>>> + _pdata) \ >>>>> +static struct clk_regmap _name = { \ >>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>> + .data = &(struct meson_clk_triphase_data) { \ >>>>> + .ph0 = { \ >>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>> + .shift = (_shift0), \ >>>>> + .width = (_width), \ >>>>> + }, \ >>>>> + .ph1 = { \ >>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>> + .shift = (_shift1), \ >>>>> + .width = (_width), \ >>>>> + }, \ >>>>> + .ph2 = { \ >>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>> + .shift = (_shift2), \ >>>>> + .width = (_width), \ >>>>> + }, \ >>>>> + }, \ >>>>> + .hw.init = &(struct clk_init_data) { \ >>>>> + .name = #_name, \ >>>>> + .ops = &meson_clk_triphase_ops, \ >>>>> + .parent_data = (_pdata), \ >>>>> + .num_parents = 1, \ >>>>> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >>>>> + }, \ >>>>> +} >>>>> + >>>>> +#define AUDIO_SCLK_WS(_name, _reg, _width, _shift_ph, _shift_ws, \ >>>>> + _pdata) \ >>>>> +static struct clk_regmap _name = { \ >>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>> + .data = &(struct meson_sclk_ws_inv_data) { \ >>>>> + .ph = { \ >>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>> + .shift = (_shift_ph), \ >>>>> + .width = (_width), \ >>>>> + }, \ >>>>> + .ws = { \ >>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>> + .shift = (_shift_ws), \ >>>>> + .width = (_width), \ >>>>> + }, \ >>>>> + }, \ >>>>> + .hw.init = &(struct clk_init_data) { \ >>>>> + .name = #_name, \ >>>>> + .ops = &meson_sclk_ws_inv_ops, \ >>>>> + .parent_data = (_pdata), \ >>>>> + .num_parents = 1, \ >>>>> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >>>>> + }, \ >>>>> +} >>>> >>>> All the above does essentially the same things as the macro of >>>> axg-audio, to some minor differences. Yet it is another set to maintain. >>>> >>> >>> Except one thing... Here I keep memory identifier to which this clock >>> belongs: >>> >>> .map = AUDIO_REG_MAP(_reg), >>> >>> It is workaround, but ->map the only common field in clk_regmap that >>> could be used for this purpose. >>> >>> >>>> I'd much prefer if you put the axg-audio macro in a header a re-used >>>> those. There would a single set to maintain. You may then specialize the >>>> included in the driver C file, to avoid redundant parameters >>>> >>>> Rework axg-audio to use clk_parent_data if you must, but not in the same >>>> series please. >>>> >>>>> + >>>>> +static const struct clk_parent_data a1_pclk_pdata[] = { >>>>> + { .fw_name = "pclk", }, >>>>> +}; >>>>> + >>>>> +AUDIO_GATE(audio_ddr_arb, AUDIO_CLK_GATE_EN0, 0, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_tdmin_a, AUDIO_CLK_GATE_EN0, 1, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_tdmin_b, AUDIO_CLK_GATE_EN0, 2, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_tdmin_lb, AUDIO_CLK_GATE_EN0, 3, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_loopback, AUDIO_CLK_GATE_EN0, 4, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_tdmout_a, AUDIO_CLK_GATE_EN0, 5, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_tdmout_b, AUDIO_CLK_GATE_EN0, 6, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_frddr_a, AUDIO_CLK_GATE_EN0, 7, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_frddr_b, AUDIO_CLK_GATE_EN0, 8, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_toddr_a, AUDIO_CLK_GATE_EN0, 9, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_toddr_b, AUDIO_CLK_GATE_EN0, 10, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_spdifin, AUDIO_CLK_GATE_EN0, 11, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_resample, AUDIO_CLK_GATE_EN0, 12, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_eqdrc, AUDIO_CLK_GATE_EN0, 13, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio_audiolocker, AUDIO_CLK_GATE_EN0, 14, a1_pclk_pdata); >>>> This is what I mean by redundant parameter ^ >>>> >>> >>> Yep. I could define something like AUDIO_PCLK_GATE(). >>> >>>>> + >>>>> +AUDIO_GATE(audio2_ddr_arb, AUDIO2_CLK_GATE_EN0, 0, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio2_pdm, AUDIO2_CLK_GATE_EN0, 1, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio2_tdmin_vad, AUDIO2_CLK_GATE_EN0, 2, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio2_toddr_vad, AUDIO2_CLK_GATE_EN0, 3, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio2_vad, AUDIO2_CLK_GATE_EN0, 4, a1_pclk_pdata); >>>>> +AUDIO_GATE(audio2_audiotop, AUDIO2_CLK_GATE_EN0, 7, a1_pclk_pdata); >>>>> + >>>>> +static const struct clk_parent_data a1_mst_pdata[] = { >>>>> + { .fw_name = "dds_in" }, >>>>> + { .fw_name = "fclk_div2" }, >>>>> + { .fw_name = "fclk_div3" }, >>>>> + { .fw_name = "hifi_pll" }, >>>>> + { .fw_name = "xtal" }, >>>>> +}; >>>>> + >>>>> +#define AUDIO_MST_MCLK(_name, _reg) \ >>>>> + AUDIO_MUX(_name##_mux, (_reg), 0x7, 24, a1_mst_pdata); \ >>>>> + AUDIO_DIV(_name##_div, (_reg), 0, 16, \ >>>>> + AUDIO_PDATA(_name##_mux)); \ >>>>> + AUDIO_GATE(_name, (_reg), 31, AUDIO_PDATA(_name##_div)) >>>>> + >>>>> +AUDIO_MST_MCLK(audio_mst_a_mclk, AUDIO_MCLK_A_CTRL); >>>>> +AUDIO_MST_MCLK(audio_mst_b_mclk, AUDIO_MCLK_B_CTRL); >>>>> +AUDIO_MST_MCLK(audio_mst_c_mclk, AUDIO_MCLK_C_CTRL); >>>>> +AUDIO_MST_MCLK(audio_mst_d_mclk, AUDIO_MCLK_D_CTRL); >>>>> +AUDIO_MST_MCLK(audio_spdifin_clk, AUDIO_CLK_SPDIFIN_CTRL); >>>>> +AUDIO_MST_MCLK(audio_eqdrc_clk, AUDIO_CLK_EQDRC_CTRL); >>>>> + >>>>> +AUDIO_MUX(audio_resample_clk_mux, AUDIO_CLK_RESAMPLE_CTRL, 0xf, 24, >>>>> + a1_mst_pdata); >>>>> +AUDIO_DIV(audio_resample_clk_div, AUDIO_CLK_RESAMPLE_CTRL, 0, 8, >>>>> + AUDIO_PDATA(audio_resample_clk_mux)); >>>>> +AUDIO_GATE(audio_resample_clk, AUDIO_CLK_RESAMPLE_CTRL, 31, >>>>> + AUDIO_PDATA(audio_resample_clk_div)); >>>>> + >>>>> +AUDIO_MUX(audio_locker_in_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 8, >>>>> + a1_mst_pdata); >>>>> +AUDIO_DIV(audio_locker_in_clk_div, AUDIO_CLK_LOCKER_CTRL, 0, 8, >>>>> + AUDIO_PDATA(audio_locker_in_clk_mux)); >>>>> +AUDIO_GATE(audio_locker_in_clk, AUDIO_CLK_LOCKER_CTRL, 15, >>>>> + AUDIO_PDATA(audio_locker_in_clk_div)); >>>>> + >>>>> +AUDIO_MUX(audio_locker_out_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 24, >>>>> + a1_mst_pdata); >>>>> +AUDIO_DIV(audio_locker_out_clk_div, AUDIO_CLK_LOCKER_CTRL, 16, 8, >>>>> + AUDIO_PDATA(audio_locker_out_clk_mux)); >>>>> +AUDIO_GATE(audio_locker_out_clk, AUDIO_CLK_LOCKER_CTRL, 31, >>>>> + AUDIO_PDATA(audio_locker_out_clk_div)); >>>>> + >>>>> +AUDIO_MST_MCLK(audio2_vad_mclk, AUDIO2_MCLK_VAD_CTRL); >>>>> +AUDIO_MST_MCLK(audio2_vad_clk, AUDIO2_CLK_VAD_CTRL); >>>>> +AUDIO_MST_MCLK(audio2_pdm_dclk, AUDIO2_CLK_PDMIN_CTRL0); >>>>> +AUDIO_MST_MCLK(audio2_pdm_sysclk, AUDIO2_CLK_PDMIN_CTRL1); >>>>> + >>>>> +#define AUDIO_MST_SCLK(_name, _reg0, _reg1, _pdata) \ >>>>> + AUDIO_GATE(_name##_pre_en, (_reg0), 31, (_pdata)); \ >>>>> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 20, 10, 0, 0, \ >>>>> + AUDIO_PDATA(_name##_pre_en), true); \ >>>>> + AUDIO_GATE(_name##_post_en, (_reg0), 30, \ >>>>> + AUDIO_PDATA(_name##_div)); \ >>>>> + AUDIO_TRIPHASE(_name, (_reg1), 1, 0, 2, 4, \ >>>>> + AUDIO_PDATA(_name##_post_en)) >>>>> + >>>> >>>> Again, I'm not a fan of this many levels of macro. I can live with it >>>> but certainly don't want the burden of reviewing and maintaining for >>>> clock driver. AXG / G12 and A1 are obviously closely related, so make it common. >>>> >>>>> +#define AUDIO_MST_LRCLK(_name, _reg0, _reg1, _pdata) \ >>>>> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 0, 10, 10, 10, \ >>>>> + (_pdata), false); \ >>>>> + AUDIO_TRIPHASE(_name, (_reg1), 1, 1, 3, 5, \ >>>>> + AUDIO_PDATA(_name##_div)) >>>>> + >>>>> +AUDIO_MST_SCLK(audio_mst_a_sclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >>>>> + AUDIO_PDATA(audio_mst_a_mclk)); >>>>> +AUDIO_MST_SCLK(audio_mst_b_sclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >>>>> + AUDIO_PDATA(audio_mst_b_mclk)); >>>>> +AUDIO_MST_SCLK(audio_mst_c_sclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >>>>> + AUDIO_PDATA(audio_mst_c_mclk)); >>>>> +AUDIO_MST_SCLK(audio_mst_d_sclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >>>>> + AUDIO_PDATA(audio_mst_d_mclk)); >>>>> + >>>>> +AUDIO_MST_LRCLK(audio_mst_a_lrclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >>>>> + AUDIO_PDATA(audio_mst_a_sclk_post_en)); >>>>> +AUDIO_MST_LRCLK(audio_mst_b_lrclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >>>>> + AUDIO_PDATA(audio_mst_b_sclk_post_en)); >>>>> +AUDIO_MST_LRCLK(audio_mst_c_lrclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >>>>> + AUDIO_PDATA(audio_mst_c_sclk_post_en)); >>>>> +AUDIO_MST_LRCLK(audio_mst_d_lrclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >>>>> + AUDIO_PDATA(audio_mst_d_sclk_post_en)); >>>>> + >>>>> +static const struct clk_parent_data a1_mst_sclk_pdata[] = { >>>>> + { .hw = &audio_mst_a_sclk.hw }, >>>>> + { .hw = &audio_mst_b_sclk.hw }, >>>>> + { .hw = &audio_mst_c_sclk.hw }, >>>>> + { .hw = &audio_mst_d_sclk.hw }, >>>>> + { .fw_name = "slv_sclk0" }, >>>>> + { .fw_name = "slv_sclk1" }, >>>>> + { .fw_name = "slv_sclk2" }, >>>>> + { .fw_name = "slv_sclk3" }, >>>>> + { .fw_name = "slv_sclk4" }, >>>>> + { .fw_name = "slv_sclk5" }, >>>>> + { .fw_name = "slv_sclk6" }, >>>>> + { .fw_name = "slv_sclk7" }, >>>>> + { .fw_name = "slv_sclk8" }, >>>>> + { .fw_name = "slv_sclk9" }, >>>>> +}; >>>>> + >>>>> +static const struct clk_parent_data a1_mst_lrclk_pdata[] = { >>>>> + { .hw = &audio_mst_a_lrclk.hw }, >>>>> + { .hw = &audio_mst_b_lrclk.hw }, >>>>> + { .hw = &audio_mst_c_lrclk.hw }, >>>>> + { .hw = &audio_mst_d_lrclk.hw }, >>>>> + { .fw_name = "slv_lrclk0" }, >>>>> + { .fw_name = "slv_lrclk1" }, >>>>> + { .fw_name = "slv_lrclk2" }, >>>>> + { .fw_name = "slv_lrclk3" }, >>>>> + { .fw_name = "slv_lrclk4" }, >>>>> + { .fw_name = "slv_lrclk5" }, >>>>> + { .fw_name = "slv_lrclk6" }, >>>>> + { .fw_name = "slv_lrclk7" }, >>>>> + { .fw_name = "slv_lrclk8" }, >>>>> + { .fw_name = "slv_lrclk9" }, >>>>> +}; >>>>> + >>>>> +#define AUDIO_TDM_SCLK(_name, _reg) \ >>>>> + AUDIO_MUX(_name##_mux, (_reg), 0xf, 24, a1_mst_sclk_pdata); \ >>>>> + AUDIO_GATE(_name##_pre_en, (_reg), 31, \ >>>>> + AUDIO_PDATA(_name##_mux)); \ >>>>> + AUDIO_GATE(_name##_post_en, (_reg), 30, \ >>>>> + AUDIO_PDATA(_name##_pre_en)); \ >>>>> + AUDIO_SCLK_WS(_name, (_reg), 1, 29, 28, \ >>>>> + AUDIO_PDATA(_name##_post_en)) >>>>> + >>>>> +#define AUDIO_TDM_LRCLK(_name, _reg) \ >>>>> + AUDIO_MUX(_name, (_reg), 0xf, 20, a1_mst_lrclk_pdata) >>>>> + >>>>> +AUDIO_TDM_SCLK(audio_tdmin_a_sclk, AUDIO_CLK_TDMIN_A_CTRL); >>>>> +AUDIO_TDM_SCLK(audio_tdmin_b_sclk, AUDIO_CLK_TDMIN_B_CTRL); >>>>> +AUDIO_TDM_SCLK(audio_tdmin_lb_sclk, AUDIO_CLK_TDMIN_LB_CTRL); >>>>> +AUDIO_TDM_SCLK(audio_tdmout_a_sclk, AUDIO_CLK_TDMOUT_A_CTRL); >>>>> +AUDIO_TDM_SCLK(audio_tdmout_b_sclk, AUDIO_CLK_TDMOUT_B_CTRL); >>>>> + >>>>> +AUDIO_TDM_LRCLK(audio_tdmin_a_lrclk, AUDIO_CLK_TDMIN_A_CTRL); >>>>> +AUDIO_TDM_LRCLK(audio_tdmin_b_lrclk, AUDIO_CLK_TDMIN_B_CTRL); >>>>> +AUDIO_TDM_LRCLK(audio_tdmin_lb_lrclk, AUDIO_CLK_TDMIN_LB_CTRL); >>>>> +AUDIO_TDM_LRCLK(audio_tdmout_a_lrclk, AUDIO_CLK_TDMOUT_A_CTRL); >>>>> +AUDIO_TDM_LRCLK(audio_tdmout_b_lrclk, AUDIO_CLK_TDMOUT_B_CTRL); >>>>> + >>>>> +static struct clk_hw *a1_audio_hw_clks[] = { >>>>> + [AUD_CLKID_DDR_ARB] = &audio_ddr_arb.hw, >>>>> + [AUD_CLKID_TDMIN_A] = &audio_tdmin_a.hw, >>>>> + [AUD_CLKID_TDMIN_B] = &audio_tdmin_b.hw, >>>>> + [AUD_CLKID_TDMIN_LB] = &audio_tdmin_lb.hw, >>>>> + [AUD_CLKID_LOOPBACK] = &audio_loopback.hw, >>>>> + [AUD_CLKID_TDMOUT_A] = &audio_tdmout_a.hw, >>>>> + [AUD_CLKID_TDMOUT_B] = &audio_tdmout_b.hw, >>>>> + [AUD_CLKID_FRDDR_A] = &audio_frddr_a.hw, >>>>> + [AUD_CLKID_FRDDR_B] = &audio_frddr_b.hw, >>>>> + [AUD_CLKID_TODDR_A] = &audio_toddr_a.hw, >>>>> + [AUD_CLKID_TODDR_B] = &audio_toddr_b.hw, >>>>> + [AUD_CLKID_SPDIFIN] = &audio_spdifin.hw, >>>>> + [AUD_CLKID_RESAMPLE] = &audio_resample.hw, >>>>> + [AUD_CLKID_EQDRC] = &audio_eqdrc.hw, >>>>> + [AUD_CLKID_LOCKER] = &audio_audiolocker.hw, >>>>> + [AUD_CLKID_MST_A_MCLK_SEL] = &audio_mst_a_mclk_mux.hw, >>>>> + [AUD_CLKID_MST_A_MCLK_DIV] = &audio_mst_a_mclk_div.hw, >>>>> + [AUD_CLKID_MST_A_MCLK] = &audio_mst_a_mclk.hw, >>>>> + [AUD_CLKID_MST_B_MCLK_SEL] = &audio_mst_b_mclk_mux.hw, >>>>> + [AUD_CLKID_MST_B_MCLK_DIV] = &audio_mst_b_mclk_div.hw, >>>>> + [AUD_CLKID_MST_B_MCLK] = &audio_mst_b_mclk.hw, >>>>> + [AUD_CLKID_MST_C_MCLK_SEL] = &audio_mst_c_mclk_mux.hw, >>>>> + [AUD_CLKID_MST_C_MCLK_DIV] = &audio_mst_c_mclk_div.hw, >>>>> + [AUD_CLKID_MST_C_MCLK] = &audio_mst_c_mclk.hw, >>>>> + [AUD_CLKID_MST_D_MCLK_SEL] = &audio_mst_d_mclk_mux.hw, >>>>> + [AUD_CLKID_MST_D_MCLK_DIV] = &audio_mst_d_mclk_div.hw, >>>>> + [AUD_CLKID_MST_D_MCLK] = &audio_mst_d_mclk.hw, >>>>> + [AUD_CLKID_RESAMPLE_CLK_SEL] = &audio_resample_clk_mux.hw, >>>>> + [AUD_CLKID_RESAMPLE_CLK_DIV] = &audio_resample_clk_div.hw, >>>>> + [AUD_CLKID_RESAMPLE_CLK] = &audio_resample_clk.hw, >>>>> + [AUD_CLKID_LOCKER_IN_CLK_SEL] = &audio_locker_in_clk_mux.hw, >>>>> + [AUD_CLKID_LOCKER_IN_CLK_DIV] = &audio_locker_in_clk_div.hw, >>>>> + [AUD_CLKID_LOCKER_IN_CLK] = &audio_locker_in_clk.hw, >>>>> + [AUD_CLKID_LOCKER_OUT_CLK_SEL] = &audio_locker_out_clk_mux.hw, >>>>> + [AUD_CLKID_LOCKER_OUT_CLK_DIV] = &audio_locker_out_clk_div.hw, >>>>> + [AUD_CLKID_LOCKER_OUT_CLK] = &audio_locker_out_clk.hw, >>>>> + [AUD_CLKID_SPDIFIN_CLK_SEL] = &audio_spdifin_clk_mux.hw, >>>>> + [AUD_CLKID_SPDIFIN_CLK_DIV] = &audio_spdifin_clk_div.hw, >>>>> + [AUD_CLKID_SPDIFIN_CLK] = &audio_spdifin_clk.hw, >>>>> + [AUD_CLKID_EQDRC_CLK_SEL] = &audio_eqdrc_clk_mux.hw, >>>>> + [AUD_CLKID_EQDRC_CLK_DIV] = &audio_eqdrc_clk_div.hw, >>>>> + [AUD_CLKID_EQDRC_CLK] = &audio_eqdrc_clk.hw, >>>>> + [AUD_CLKID_MST_A_SCLK_PRE_EN] = &audio_mst_a_sclk_pre_en.hw, >>>>> + [AUD_CLKID_MST_A_SCLK_DIV] = &audio_mst_a_sclk_div.hw, >>>>> + [AUD_CLKID_MST_A_SCLK_POST_EN] = &audio_mst_a_sclk_post_en.hw, >>>>> + [AUD_CLKID_MST_A_SCLK] = &audio_mst_a_sclk.hw, >>>>> + [AUD_CLKID_MST_B_SCLK_PRE_EN] = &audio_mst_b_sclk_pre_en.hw, >>>>> + [AUD_CLKID_MST_B_SCLK_DIV] = &audio_mst_b_sclk_div.hw, >>>>> + [AUD_CLKID_MST_B_SCLK_POST_EN] = &audio_mst_b_sclk_post_en.hw, >>>>> + [AUD_CLKID_MST_B_SCLK] = &audio_mst_b_sclk.hw, >>>>> + [AUD_CLKID_MST_C_SCLK_PRE_EN] = &audio_mst_c_sclk_pre_en.hw, >>>>> + [AUD_CLKID_MST_C_SCLK_DIV] = &audio_mst_c_sclk_div.hw, >>>>> + [AUD_CLKID_MST_C_SCLK_POST_EN] = &audio_mst_c_sclk_post_en.hw, >>>>> + [AUD_CLKID_MST_C_SCLK] = &audio_mst_c_sclk.hw, >>>>> + [AUD_CLKID_MST_D_SCLK_PRE_EN] = &audio_mst_d_sclk_pre_en.hw, >>>>> + [AUD_CLKID_MST_D_SCLK_DIV] = &audio_mst_d_sclk_div.hw, >>>>> + [AUD_CLKID_MST_D_SCLK_POST_EN] = &audio_mst_d_sclk_post_en.hw, >>>>> + [AUD_CLKID_MST_D_SCLK] = &audio_mst_d_sclk.hw, >>>>> + [AUD_CLKID_MST_A_LRCLK_DIV] = &audio_mst_a_lrclk_div.hw, >>>>> + [AUD_CLKID_MST_A_LRCLK] = &audio_mst_a_lrclk.hw, >>>>> + [AUD_CLKID_MST_B_LRCLK_DIV] = &audio_mst_b_lrclk_div.hw, >>>>> + [AUD_CLKID_MST_B_LRCLK] = &audio_mst_b_lrclk.hw, >>>>> + [AUD_CLKID_MST_C_LRCLK_DIV] = &audio_mst_c_lrclk_div.hw, >>>>> + [AUD_CLKID_MST_C_LRCLK] = &audio_mst_c_lrclk.hw, >>>>> + [AUD_CLKID_MST_D_LRCLK_DIV] = &audio_mst_d_lrclk_div.hw, >>>>> + [AUD_CLKID_MST_D_LRCLK] = &audio_mst_d_lrclk.hw, >>>>> + [AUD_CLKID_TDMIN_A_SCLK_SEL] = &audio_tdmin_a_sclk_mux.hw, >>>>> + [AUD_CLKID_TDMIN_A_SCLK_PRE_EN] = &audio_tdmin_a_sclk_pre_en.hw, >>>>> + [AUD_CLKID_TDMIN_A_SCLK_POST_EN] = &audio_tdmin_a_sclk_post_en.hw, >>>>> + [AUD_CLKID_TDMIN_A_SCLK] = &audio_tdmin_a_sclk.hw, >>>>> + [AUD_CLKID_TDMIN_A_LRCLK] = &audio_tdmin_a_lrclk.hw, >>>>> + [AUD_CLKID_TDMIN_B_SCLK_SEL] = &audio_tdmin_b_sclk_mux.hw, >>>>> + [AUD_CLKID_TDMIN_B_SCLK_PRE_EN] = &audio_tdmin_b_sclk_pre_en.hw, >>>>> + [AUD_CLKID_TDMIN_B_SCLK_POST_EN] = &audio_tdmin_b_sclk_post_en.hw, >>>>> + [AUD_CLKID_TDMIN_B_SCLK] = &audio_tdmin_b_sclk.hw, >>>>> + [AUD_CLKID_TDMIN_B_LRCLK] = &audio_tdmin_b_lrclk.hw, >>>>> + [AUD_CLKID_TDMIN_LB_SCLK_SEL] = &audio_tdmin_lb_sclk_mux.hw, >>>>> + [AUD_CLKID_TDMIN_LB_SCLK_PRE_EN] = &audio_tdmin_lb_sclk_pre_en.hw, >>>>> + [AUD_CLKID_TDMIN_LB_SCLK_POST_EN] = &audio_tdmin_lb_sclk_post_en.hw, >>>>> + [AUD_CLKID_TDMIN_LB_SCLK] = &audio_tdmin_lb_sclk.hw, >>>>> + [AUD_CLKID_TDMIN_LB_LRCLK] = &audio_tdmin_lb_lrclk.hw, >>>>> + [AUD_CLKID_TDMOUT_A_SCLK_SEL] = &audio_tdmout_a_sclk_mux.hw, >>>>> + [AUD_CLKID_TDMOUT_A_SCLK_PRE_EN] = &audio_tdmout_a_sclk_pre_en.hw, >>>>> + [AUD_CLKID_TDMOUT_A_SCLK_POST_EN] = &audio_tdmout_a_sclk_post_en.hw, >>>>> + [AUD_CLKID_TDMOUT_A_SCLK] = &audio_tdmout_a_sclk.hw, >>>>> + [AUD_CLKID_TDMOUT_A_LRCLK] = &audio_tdmout_a_lrclk.hw, >>>>> + [AUD_CLKID_TDMOUT_B_SCLK_SEL] = &audio_tdmout_b_sclk_mux.hw, >>>>> + [AUD_CLKID_TDMOUT_B_SCLK_PRE_EN] = &audio_tdmout_b_sclk_pre_en.hw, >>>>> + [AUD_CLKID_TDMOUT_B_SCLK_POST_EN] = &audio_tdmout_b_sclk_post_en.hw, >>>>> + [AUD_CLKID_TDMOUT_B_SCLK] = &audio_tdmout_b_sclk.hw, >>>>> + [AUD_CLKID_TDMOUT_B_LRCLK] = &audio_tdmout_b_lrclk.hw, >>>>> + >>>>> + [AUD2_CLKID_DDR_ARB] = &audio2_ddr_arb.hw, >>>>> + [AUD2_CLKID_PDM] = &audio2_pdm.hw, >>>>> + [AUD2_CLKID_TDMIN_VAD] = &audio2_tdmin_vad.hw, >>>>> + [AUD2_CLKID_TODDR_VAD] = &audio2_toddr_vad.hw, >>>>> + [AUD2_CLKID_VAD] = &audio2_vad.hw, >>>>> + [AUD2_CLKID_AUDIOTOP] = &audio2_audiotop.hw, >>>>> + [AUD2_CLKID_VAD_MCLK_SEL] = &audio2_vad_mclk_mux.hw, >>>>> + [AUD2_CLKID_VAD_MCLK_DIV] = &audio2_vad_mclk_div.hw, >>>>> + [AUD2_CLKID_VAD_MCLK] = &audio2_vad_mclk.hw, >>>>> + [AUD2_CLKID_VAD_CLK_SEL] = &audio2_vad_clk_mux.hw, >>>>> + [AUD2_CLKID_VAD_CLK_DIV] = &audio2_vad_clk_div.hw, >>>>> + [AUD2_CLKID_VAD_CLK] = &audio2_vad_clk.hw, >>>>> + [AUD2_CLKID_PDM_DCLK_SEL] = &audio2_pdm_dclk_mux.hw, >>>>> + [AUD2_CLKID_PDM_DCLK_DIV] = &audio2_pdm_dclk_div.hw, >>>>> + [AUD2_CLKID_PDM_DCLK] = &audio2_pdm_dclk.hw, >>>>> + [AUD2_CLKID_PDM_SYSCLK_SEL] = &audio2_pdm_sysclk_mux.hw, >>>>> + [AUD2_CLKID_PDM_SYSCLK_DIV] = &audio2_pdm_sysclk_div.hw, >>>>> + [AUD2_CLKID_PDM_SYSCLK] = &audio2_pdm_sysclk.hw, >>>>> +}; >>>>> + >>>>> +static struct meson_clk_hw_data a1_audio_clks = { >>>>> + .hws = a1_audio_hw_clks, >>>>> + .num = ARRAY_SIZE(a1_audio_hw_clks), >>>>> +}; >>>>> + >>>>> +static struct regmap *a1_audio_map(struct platform_device *pdev, >>>>> + unsigned int index) >>>>> +{ >>>>> + char name[32]; >>>>> + const struct regmap_config cfg = { >>>>> + .reg_bits = 32, >>>>> + .val_bits = 32, >>>>> + .reg_stride = 4, >>>>> + .name = name, >>>> >>>> Not necessary >>>> >>> >>> This implementation uses two regmaps, and this field allow to avoid >>> errors like this: >>> >>> [ 0.145530] debugfs: Directory 'fe050000.audio-clock-controller' with >>> parent 'regmap' already present! >>> >>>>> + }; >>>>> + void __iomem *base; >>>>> + >>>>> + base = devm_platform_ioremap_resource(pdev, index); >>>>> + if (IS_ERR(base)) >>>>> + return base; >>>>> + >>>>> + scnprintf(name, sizeof(name), "%d", index); >>>>> + return devm_regmap_init_mmio(&pdev->dev, base, &cfg); >>>>> +} >>>> >>>> That is overengineered. Please keep it simple. Declare the regmap_config >>>> as static const global, and do it like axg-audio please. >>>> >>> >>> This only reason why it is not "static const" because I need to set >>> unique name for each regmap. >>> >>>>> + >>>>> +static int a1_register_clk(struct platform_device *pdev, >>>>> + struct regmap *map0, struct regmap *map1, >>>>> + struct clk_hw *hw) >>>>> +{ >>>>> + struct clk_regmap *clk = container_of(hw, struct clk_regmap, hw); >>>>> + >>>>> + if (!hw) >>>>> + return 0; >>>>> + >>>>> + switch ((unsigned long)clk->map) { >>>>> + case AUDIO_RANGE_0: >>>>> + clk->map = map0; >>>>> + break; >>>>> + case AUDIO_RANGE_1: >>>>> + clk->map = map1; >>>>> + break; >>>> >>>> ... fishy >>>> >>>>> + default: >>>>> + WARN_ON(1); >>>>> + return -EINVAL; >>>>> + } >>>>> + >>>>> + return devm_clk_hw_register(&pdev->dev, hw); >>>>> +} >>>>> + >>>>> +static int a1_audio_clkc_probe(struct platform_device *pdev) >>>>> +{ >>>>> + struct regmap *map0, *map1; >>>>> + struct clk *clk; >>>>> + unsigned int i; >>>>> + int ret; >>>>> + >>>>> + clk = devm_clk_get_enabled(&pdev->dev, "pclk"); >>>>> + if (WARN_ON(IS_ERR(clk))) >>>>> + return PTR_ERR(clk); >>>>> + >>>>> + map0 = a1_audio_map(pdev, 0); >>>>> + if (IS_ERR(map0)) >>>>> + return PTR_ERR(map0); >>>>> + >>>>> + map1 = a1_audio_map(pdev, 1); >>>>> + if (IS_ERR(map1)) >>>>> + return PTR_ERR(map1); >>>> >>>> No - Looks to me you just have two clock controllers you are trying >>>> force into one. >>>> >>> >>> See the begining. >>> >>>>> + >>>>> + /* >>>>> + * Register and enable AUD2_CLKID_AUDIOTOP clock first. Unless >>>>> + * it is enabled any read/write to 'map0' hangs the CPU. >>>>> + */ >>>>> + >>>>> + ret = a1_register_clk(pdev, map0, map1, >>>>> + a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]); >>>>> + if (ret) >>>>> + return ret; >>>>> + >>>>> + ret = clk_prepare_enable(a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]->clk); >>>>> + if (ret) >>>>> + return ret; >>>> >>>> Again, this shows 2 devices. The one related to your 'map0' should >>>> request AUD2_CLKID_AUDIOTOP as input and enable it right away. >>>> >>> >>> See the begining. >>> >>>>> + >>>>> + for (i = 0; i < a1_audio_clks.num; i++) { >>>>> + if (i == AUD2_CLKID_AUDIOTOP) >>>>> + continue; >>>>> + >>>>> + ret = a1_register_clk(pdev, map0, map1, a1_audio_clks.hws[i]); >>>>> + if (ret) >>>>> + return ret; >>>>> + } >>>>> + >>>>> + ret = devm_of_clk_add_hw_provider(&pdev->dev, meson_clk_hw_get, >>>>> + &a1_audio_clks); >>>>> + if (ret) >>>>> + return ret; >>>>> + >>>>> + BUILD_BUG_ON((unsigned long)AUDIO_REG_MAP(AUDIO_SW_RESET0) != >>>>> + AUDIO_RANGE_0); >>>> >>>> Why is that necessary ? >>>> >>> >>> A little paranoia. Here AUDIO_SW_RESET0 is handled as map0's register, >>> and I want to assert it. >>> >>>>> + return meson_audio_rstc_register(&pdev->dev, map0, >>>>> + AUDIO_REG_OFFSET(AUDIO_SW_RESET0), 32); >>>>> +} >>>>> + >>>>> +static const struct of_device_id a1_audio_clkc_match_table[] = { >>>>> + { .compatible = "amlogic,a1-audio-clkc", }, >>>>> + {} >>>>> +}; >>>>> +MODULE_DEVICE_TABLE(of, a1_audio_clkc_match_table); >>>>> + >>>>> +static struct platform_driver a1_audio_clkc_driver = { >>>>> + .probe = a1_audio_clkc_probe, >>>>> + .driver = { >>>>> + .name = "a1-audio-clkc", >>>>> + .of_match_table = a1_audio_clkc_match_table, >>>>> + }, >>>>> +}; >>>>> +module_platform_driver(a1_audio_clkc_driver); >>>>> + >>>>> +MODULE_DESCRIPTION("Amlogic A1 Audio Clock driver"); >>>>> +MODULE_AUTHOR("Jan Dakinevich <jan.dakinevich@salutedevices.com>"); >>>>> +MODULE_LICENSE("GPL"); >>>>> diff --git a/drivers/clk/meson/a1-audio.h b/drivers/clk/meson/a1-audio.h >>>>> new file mode 100644 >>>>> index 000000000000..f994e87276cd >>>>> --- /dev/null >>>>> +++ b/drivers/clk/meson/a1-audio.h >>>>> @@ -0,0 +1,58 @@ >>>>> +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ >>>>> +/* >>>>> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >>>>> + * >>>>> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>> + */ >>>>> + >>>>> +#ifndef __A1_AUDIO_H >>>>> +#define __A1_AUDIO_H >>>>> + >>>>> +#define AUDIO_RANGE_0 0xa >>>>> +#define AUDIO_RANGE_1 0xb >>>>> +#define AUDIO_RANGE_SHIFT 16 >>>>> + >>>>> +#define AUDIO_REG(_range, _offset) \ >>>>> + (((_range) << AUDIO_RANGE_SHIFT) + (_offset)) >>>>> + >>>>> +#define AUDIO_REG_OFFSET(_reg) \ >>>>> + ((_reg) & ((1 << AUDIO_RANGE_SHIFT) - 1)) >>>>> + >>>>> +#define AUDIO_REG_MAP(_reg) \ >>>>> + ((void *)((_reg) >> AUDIO_RANGE_SHIFT)) >>>> >>>> That is seriouly overengineered. >>>> The following are offset. Just write what they are. >>>> >>> >>> This is all in order to keep range's identifier together with offset and >>> then use it to store the identifier in clk_regmaps. >>> >>>> There is not reason to put that into a header. It is only going to be >>>> used by a single driver. >>>>>> + >>>>> +#define AUDIO_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_0, 0x000) >>>>> +#define AUDIO_MCLK_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x008) >>>>> +#define AUDIO_MCLK_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x00c) >>>>> +#define AUDIO_MCLK_C_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x010) >>>>> +#define AUDIO_MCLK_D_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x014) >>>>> +#define AUDIO_MCLK_E_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x018) >>>>> +#define AUDIO_MCLK_F_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x01c) >>>>> +#define AUDIO_SW_RESET0 AUDIO_REG(AUDIO_RANGE_0, 0x028) >>>>> +#define AUDIO_MST_A_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x040) >>>>> +#define AUDIO_MST_A_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x044) >>>>> +#define AUDIO_MST_B_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x048) >>>>> +#define AUDIO_MST_B_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x04c) >>>>> +#define AUDIO_MST_C_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x050) >>>>> +#define AUDIO_MST_C_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x054) >>>>> +#define AUDIO_MST_D_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x058) >>>>> +#define AUDIO_MST_D_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x05c) >>>>> +#define AUDIO_CLK_TDMIN_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x080) >>>>> +#define AUDIO_CLK_TDMIN_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x084) >>>>> +#define AUDIO_CLK_TDMIN_LB_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x08c) >>>>> +#define AUDIO_CLK_TDMOUT_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x090) >>>>> +#define AUDIO_CLK_TDMOUT_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x094) >>>>> +#define AUDIO_CLK_SPDIFIN_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x09c) >>>>> +#define AUDIO_CLK_RESAMPLE_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a4) >>>>> +#define AUDIO_CLK_LOCKER_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a8) >>>>> +#define AUDIO_CLK_EQDRC_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0c0) >>>>> + >>>>> +#define AUDIO2_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_1, 0x00c) >>>>> +#define AUDIO2_MCLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x040) >>>>> +#define AUDIO2_CLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x044) >>>>> +#define AUDIO2_CLK_PDMIN_CTRL0 AUDIO_REG(AUDIO_RANGE_1, 0x058) >>>>> +#define AUDIO2_CLK_PDMIN_CTRL1 AUDIO_REG(AUDIO_RANGE_1, 0x05c) >>>>> + >>>>> +#include <dt-bindings/clock/amlogic,a1-audio-clkc.h> >>>>> + >>>>> +#endif /* __A1_AUDIO_H */ >>>> >>>> >> >>
On 3/26/24 18:26, Jerome Brunet wrote: > > On Sat 23 Mar 2024 at 21:02, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > >> Jerome, I have reworked my driver reusing axg-audio code as most as I >> could and now I have one more question. Lets see on this definition from >> axg-audio: >> >> #define AUD_MST_MUX(_name, _reg, _flag) \ >> AUD_MUX(_name##_sel, _reg, 0x7, 24, _flag, \ >> mst_mux_parent_data, 0) >> >> #define AUD_MST_MCLK_MUX(_name, _reg) \ >> AUD_MST_MUX(_name, _reg, CLK_MUX_ROUND_CLOSEST) >> >> CLK_SET_RATE_PARENT is not set here. But why? It means, that topmost pll >> clock will not be reconfigured at runtime to satisfy the rate that was >> requested from axg-tdm. >> > > Yes, that is by design. It is another area where mainline audio differs > greatly from AML vendor code. The PLLs are expected be to fixed and the > audio master clock will reparent to the most adequate PLL source > depending on the use case. > > This is how we manage to satisfy all audio interfaces with a very > limited number of PLLs > > On AXG/G12 there is at most 6 concurrent interfaces (3 FRDDR/TODDR) - 8 > on sm1 - and we can satisfy on that with 3 PLLs. That would not be > possible if interfaces were having their way with the PLLs, reseting it > everytime a stream is started. > > The PLL rate should be carefully chosen so it can be derived easily. On > AXG/G12/SM1 that is: > * one PLL per rate family, to maximize clock precision > * x24 x32: to handle different sample sizes > * x2 until we reach the PLL limits to allow higher rates such as 384kHz > or even higher > Thank you. Now it has become much clearer. > If you have less PLLs on A1, you'll have to make compromises, like a less > precise clock to support multiple family with one PLL. > This is why the PLLs are set for each platform in DT because that choice > may depend on the platform use case. > Unfortunately, on A1 we have only one PLL. Yes, for us it would be better to have hifi_pll with predefined rate. For instance it will allow to avoid that ugly workaround in PDM (sysrate property, etc). But what whould be preferred for upstream? I can imagine a scenario where samples with different rate should be played, PDM attached to fclk_divN and there are no conflicts with TDM. In this case reconfiguration of hifi_pll on demand could better satisfy somebody's requirements. >> >> On 3/19/24 11:30, Jerome Brunet wrote: >>> >>> On Tue 19 Mar 2024 at 04:47, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >>> >>>> Let's start from the end: >>>> >>>>> No - Looks to me you just have two clock controllers you are trying >>>> force into one. >>>> >>>>> Again, this shows 2 devices. The one related to your 'map0' should >>>> request AUD2_CLKID_AUDIOTOP as input and enable it right away. >>>> >>>> Most of fishy workarounds that you commented is caused the fact the mmio >>>> of this clock controller is divided into two parts. Compare it with >>>> axg-audio driver, things that was part of contigous memory region (like >>>> pdm) here are moved to second region. Is this enough to make a guess >>>> that these are two devices? >>> >>> I see obsolutely no reason to think it is a single device nor to add all the quirks >>> you have the way you did. So yes, in that case, 2 zones, 2 devices. >>> >>>> >>>> Concerning AUD2_CLKID_AUDIOTOP clock, as it turned out, it must be >>>> enabled before enabling of clocks from second region too. That is >>>> AUD2_CLKID_AUDIOTOP clock feeds both parts of this clock controller. >>>> >>> >>> Yes. I understood the first time around and already commented on that. >>> >>>> >>>> On 3/15/24 12:20, Jerome Brunet wrote: >>>>> >>>>> On Fri 15 Mar 2024 at 02:21, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >>>>> >>>>>> This controller provides clocks and reset functionality for audio >>>>>> peripherals on Amlogic A1 SoC family. >>>>>> >>>>>> The driver is almost identical to 'axg-audio', however it would be better >>>>>> to keep it separate due to following reasons: >>>>>> >>>>>> - significant amount of bits has another definition. I will bring there >>>>>> a mess of new defines with A1_ suffixes. >>>>>> >>>>>> - registers of this controller are located in two separate regions. It >>>>>> will give a lot of complications for 'axg-audio' to support this. >>>>>> >>>>>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>>> --- >>>>>> drivers/clk/meson/Kconfig | 13 + >>>>>> drivers/clk/meson/Makefile | 1 + >>>>>> drivers/clk/meson/a1-audio.c | 556 +++++++++++++++++++++++++++++++++++ >>>>>> drivers/clk/meson/a1-audio.h | 58 ++++ >>>>>> 4 files changed, 628 insertions(+) >>>>>> create mode 100644 drivers/clk/meson/a1-audio.c >>>>>> create mode 100644 drivers/clk/meson/a1-audio.h >>>>>> >>>>>> diff --git a/drivers/clk/meson/Kconfig b/drivers/clk/meson/Kconfig >>>>>> index d6a2fa5f7e88..80c4a18c83d2 100644 >>>>>> --- a/drivers/clk/meson/Kconfig >>>>>> +++ b/drivers/clk/meson/Kconfig >>>>>> @@ -133,6 +133,19 @@ config COMMON_CLK_A1_PERIPHERALS >>>>>> device, A1 SoC Family. Say Y if you want A1 Peripherals clock >>>>>> controller to work. >>>>>> >>>>>> +config COMMON_CLK_A1_AUDIO >>>>>> + tristate "Amlogic A1 SoC Audio clock controller support" >>>>>> + depends on ARM64 >>>>>> + select COMMON_CLK_MESON_REGMAP >>>>>> + select COMMON_CLK_MESON_CLKC_UTILS >>>>>> + select COMMON_CLK_MESON_PHASE >>>>>> + select COMMON_CLK_MESON_SCLK_DIV >>>>>> + select COMMON_CLK_MESON_AUDIO_RSTC >>>>>> + help >>>>>> + Support for the Audio clock controller on Amlogic A113L based >>>>>> + device, A1 SoC Family. Say Y if you want A1 Audio clock controller >>>>>> + to work. >>>>>> + >>>>>> config COMMON_CLK_G12A >>>>>> tristate "G12 and SM1 SoC clock controllers support" >>>>>> depends on ARM64 >>>>>> diff --git a/drivers/clk/meson/Makefile b/drivers/clk/meson/Makefile >>>>>> index 88d94921a4dc..4968fc7ad555 100644 >>>>>> --- a/drivers/clk/meson/Makefile >>>>>> +++ b/drivers/clk/meson/Makefile >>>>>> @@ -20,6 +20,7 @@ obj-$(CONFIG_COMMON_CLK_AXG) += axg.o axg-aoclk.o >>>>>> obj-$(CONFIG_COMMON_CLK_AXG_AUDIO) += axg-audio.o >>>>>> obj-$(CONFIG_COMMON_CLK_A1_PLL) += a1-pll.o >>>>>> obj-$(CONFIG_COMMON_CLK_A1_PERIPHERALS) += a1-peripherals.o >>>>>> +obj-$(CONFIG_COMMON_CLK_A1_AUDIO) += a1-audio.o >>>>>> obj-$(CONFIG_COMMON_CLK_GXBB) += gxbb.o gxbb-aoclk.o >>>>>> obj-$(CONFIG_COMMON_CLK_G12A) += g12a.o g12a-aoclk.o >>>>>> obj-$(CONFIG_COMMON_CLK_MESON8B) += meson8b.o meson8-ddr.o >>>>>> diff --git a/drivers/clk/meson/a1-audio.c b/drivers/clk/meson/a1-audio.c >>>>>> new file mode 100644 >>>>>> index 000000000000..6039116c93ba >>>>>> --- /dev/null >>>>>> +++ b/drivers/clk/meson/a1-audio.c >>>>>> @@ -0,0 +1,556 @@ >>>>>> +// SPDX-License-Identifier: (GPL-2.0 OR MIT) >>>>>> +/* >>>>>> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >>>>>> + * >>>>>> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>>> + */ >>>>>> + >>>>>> +#include <linux/clk.h> >>>>>> +#include <linux/clk-provider.h> >>>>>> +#include <linux/init.h> >>>>>> +#include <linux/of_device.h> >>>>>> +#include <linux/module.h> >>>>>> +#include <linux/platform_device.h> >>>>>> +#include <linux/regmap.h> >>>>>> +#include <linux/reset.h> >>>>>> +#include <linux/reset-controller.h> >>>>>> +#include <linux/slab.h> >>>>>> + >>>>>> +#include "meson-clkc-utils.h" >>>>>> +#include "meson-audio-rstc.h" >>>>>> +#include "clk-regmap.h" >>>>>> +#include "clk-phase.h" >>>>>> +#include "sclk-div.h" >>>>>> +#include "a1-audio.h" >>>>>> + >>>>>> +#define AUDIO_PDATA(_name) \ >>>>>> + ((const struct clk_parent_data[]) { { .hw = &(_name).hw } }) >>>>> >>>>> Not a fan - yet another level of macro. >>>>> >>>>>> + >>>>>> +#define AUDIO_MUX(_name, _reg, _mask, _shift, _pdata) \ >>>>>> +static struct clk_regmap _name = { \ >>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>> + .data = &(struct clk_regmap_mux_data){ \ >>>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .mask = (_mask), \ >>>>>> + .shift = (_shift), \ >>>>>> + }, \ >>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>> + .name = #_name, \ >>>>>> + .ops = &clk_regmap_mux_ops, \ >>>>>> + .parent_data = (_pdata), \ >>>>>> + .num_parents = ARRAY_SIZE(_pdata), \ >>>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>>> + }, \ >>>>>> +} >>>>>> + >>>>>> +#define AUDIO_DIV(_name, _reg, _shift, _width, _pdata) \ >>>>>> +static struct clk_regmap _name = { \ >>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>> + .data = &(struct clk_regmap_div_data){ \ >>>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .shift = (_shift), \ >>>>>> + .width = (_width), \ >>>>>> + }, \ >>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>> + .name = #_name, \ >>>>>> + .ops = &clk_regmap_divider_ops, \ >>>>>> + .parent_data = (_pdata), \ >>>>>> + .num_parents = 1, \ >>>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>>> + }, \ >>>>>> +} >>>>>> + >>>>>> +#define AUDIO_GATE(_name, _reg, _bit, _pdata) \ >>>>>> +static struct clk_regmap _name = { \ >>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>> + .data = &(struct clk_regmap_gate_data){ \ >>>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .bit_idx = (_bit), \ >>>>>> + }, \ >>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>> + .name = #_name, \ >>>>>> + .ops = &clk_regmap_gate_ops, \ >>>>>> + .parent_data = (_pdata), \ >>>>>> + .num_parents = 1, \ >>>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>>> + }, \ >>>>>> +} >>>>>> + >>>>>> +#define AUDIO_SCLK_DIV(_name, _reg, _div_shift, _div_width, \ >>>>>> + _hi_shift, _hi_width, _pdata, _set_rate_parent) \ >>>>>> +static struct clk_regmap _name = { \ >>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>> + .data = &(struct meson_sclk_div_data) { \ >>>>>> + .div = { \ >>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .shift = (_div_shift), \ >>>>>> + .width = (_div_width), \ >>>>>> + }, \ >>>>>> + .hi = { \ >>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .shift = (_hi_shift), \ >>>>>> + .width = (_hi_width), \ >>>>>> + }, \ >>>>>> + }, \ >>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>> + .name = #_name, \ >>>>>> + .ops = &meson_sclk_div_ops, \ >>>>>> + .parent_data = (_pdata), \ >>>>>> + .num_parents = 1, \ >>>>>> + .flags = (_set_rate_parent) ? CLK_SET_RATE_PARENT : 0, \ >>>>> >>>>> Does not help readeability. Just pass the flag as axg-audio does. >>>>> >>>>>> + }, \ >>>>>> +} >>>>>> + >>>>>> +#define AUDIO_TRIPHASE(_name, _reg, _width, _shift0, _shift1, _shift2, \ >>>>>> + _pdata) \ >>>>>> +static struct clk_regmap _name = { \ >>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>> + .data = &(struct meson_clk_triphase_data) { \ >>>>>> + .ph0 = { \ >>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .shift = (_shift0), \ >>>>>> + .width = (_width), \ >>>>>> + }, \ >>>>>> + .ph1 = { \ >>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .shift = (_shift1), \ >>>>>> + .width = (_width), \ >>>>>> + }, \ >>>>>> + .ph2 = { \ >>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .shift = (_shift2), \ >>>>>> + .width = (_width), \ >>>>>> + }, \ >>>>>> + }, \ >>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>> + .name = #_name, \ >>>>>> + .ops = &meson_clk_triphase_ops, \ >>>>>> + .parent_data = (_pdata), \ >>>>>> + .num_parents = 1, \ >>>>>> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >>>>>> + }, \ >>>>>> +} >>>>>> + >>>>>> +#define AUDIO_SCLK_WS(_name, _reg, _width, _shift_ph, _shift_ws, \ >>>>>> + _pdata) \ >>>>>> +static struct clk_regmap _name = { \ >>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>> + .data = &(struct meson_sclk_ws_inv_data) { \ >>>>>> + .ph = { \ >>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .shift = (_shift_ph), \ >>>>>> + .width = (_width), \ >>>>>> + }, \ >>>>>> + .ws = { \ >>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>> + .shift = (_shift_ws), \ >>>>>> + .width = (_width), \ >>>>>> + }, \ >>>>>> + }, \ >>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>> + .name = #_name, \ >>>>>> + .ops = &meson_sclk_ws_inv_ops, \ >>>>>> + .parent_data = (_pdata), \ >>>>>> + .num_parents = 1, \ >>>>>> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >>>>>> + }, \ >>>>>> +} >>>>> >>>>> All the above does essentially the same things as the macro of >>>>> axg-audio, to some minor differences. Yet it is another set to maintain. >>>>> >>>> >>>> Except one thing... Here I keep memory identifier to which this clock >>>> belongs: >>>> >>>> .map = AUDIO_REG_MAP(_reg), >>>> >>>> It is workaround, but ->map the only common field in clk_regmap that >>>> could be used for this purpose. >>>> >>>> >>>>> I'd much prefer if you put the axg-audio macro in a header a re-used >>>>> those. There would a single set to maintain. You may then specialize the >>>>> included in the driver C file, to avoid redundant parameters >>>>> >>>>> Rework axg-audio to use clk_parent_data if you must, but not in the same >>>>> series please. >>>>> >>>>>> + >>>>>> +static const struct clk_parent_data a1_pclk_pdata[] = { >>>>>> + { .fw_name = "pclk", }, >>>>>> +}; >>>>>> + >>>>>> +AUDIO_GATE(audio_ddr_arb, AUDIO_CLK_GATE_EN0, 0, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_tdmin_a, AUDIO_CLK_GATE_EN0, 1, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_tdmin_b, AUDIO_CLK_GATE_EN0, 2, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_tdmin_lb, AUDIO_CLK_GATE_EN0, 3, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_loopback, AUDIO_CLK_GATE_EN0, 4, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_tdmout_a, AUDIO_CLK_GATE_EN0, 5, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_tdmout_b, AUDIO_CLK_GATE_EN0, 6, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_frddr_a, AUDIO_CLK_GATE_EN0, 7, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_frddr_b, AUDIO_CLK_GATE_EN0, 8, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_toddr_a, AUDIO_CLK_GATE_EN0, 9, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_toddr_b, AUDIO_CLK_GATE_EN0, 10, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_spdifin, AUDIO_CLK_GATE_EN0, 11, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_resample, AUDIO_CLK_GATE_EN0, 12, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_eqdrc, AUDIO_CLK_GATE_EN0, 13, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio_audiolocker, AUDIO_CLK_GATE_EN0, 14, a1_pclk_pdata); >>>>> This is what I mean by redundant parameter ^ >>>>> >>>> >>>> Yep. I could define something like AUDIO_PCLK_GATE(). >>>> >>>>>> + >>>>>> +AUDIO_GATE(audio2_ddr_arb, AUDIO2_CLK_GATE_EN0, 0, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio2_pdm, AUDIO2_CLK_GATE_EN0, 1, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio2_tdmin_vad, AUDIO2_CLK_GATE_EN0, 2, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio2_toddr_vad, AUDIO2_CLK_GATE_EN0, 3, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio2_vad, AUDIO2_CLK_GATE_EN0, 4, a1_pclk_pdata); >>>>>> +AUDIO_GATE(audio2_audiotop, AUDIO2_CLK_GATE_EN0, 7, a1_pclk_pdata); >>>>>> + >>>>>> +static const struct clk_parent_data a1_mst_pdata[] = { >>>>>> + { .fw_name = "dds_in" }, >>>>>> + { .fw_name = "fclk_div2" }, >>>>>> + { .fw_name = "fclk_div3" }, >>>>>> + { .fw_name = "hifi_pll" }, >>>>>> + { .fw_name = "xtal" }, >>>>>> +}; >>>>>> + >>>>>> +#define AUDIO_MST_MCLK(_name, _reg) \ >>>>>> + AUDIO_MUX(_name##_mux, (_reg), 0x7, 24, a1_mst_pdata); \ >>>>>> + AUDIO_DIV(_name##_div, (_reg), 0, 16, \ >>>>>> + AUDIO_PDATA(_name##_mux)); \ >>>>>> + AUDIO_GATE(_name, (_reg), 31, AUDIO_PDATA(_name##_div)) >>>>>> + >>>>>> +AUDIO_MST_MCLK(audio_mst_a_mclk, AUDIO_MCLK_A_CTRL); >>>>>> +AUDIO_MST_MCLK(audio_mst_b_mclk, AUDIO_MCLK_B_CTRL); >>>>>> +AUDIO_MST_MCLK(audio_mst_c_mclk, AUDIO_MCLK_C_CTRL); >>>>>> +AUDIO_MST_MCLK(audio_mst_d_mclk, AUDIO_MCLK_D_CTRL); >>>>>> +AUDIO_MST_MCLK(audio_spdifin_clk, AUDIO_CLK_SPDIFIN_CTRL); >>>>>> +AUDIO_MST_MCLK(audio_eqdrc_clk, AUDIO_CLK_EQDRC_CTRL); >>>>>> + >>>>>> +AUDIO_MUX(audio_resample_clk_mux, AUDIO_CLK_RESAMPLE_CTRL, 0xf, 24, >>>>>> + a1_mst_pdata); >>>>>> +AUDIO_DIV(audio_resample_clk_div, AUDIO_CLK_RESAMPLE_CTRL, 0, 8, >>>>>> + AUDIO_PDATA(audio_resample_clk_mux)); >>>>>> +AUDIO_GATE(audio_resample_clk, AUDIO_CLK_RESAMPLE_CTRL, 31, >>>>>> + AUDIO_PDATA(audio_resample_clk_div)); >>>>>> + >>>>>> +AUDIO_MUX(audio_locker_in_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 8, >>>>>> + a1_mst_pdata); >>>>>> +AUDIO_DIV(audio_locker_in_clk_div, AUDIO_CLK_LOCKER_CTRL, 0, 8, >>>>>> + AUDIO_PDATA(audio_locker_in_clk_mux)); >>>>>> +AUDIO_GATE(audio_locker_in_clk, AUDIO_CLK_LOCKER_CTRL, 15, >>>>>> + AUDIO_PDATA(audio_locker_in_clk_div)); >>>>>> + >>>>>> +AUDIO_MUX(audio_locker_out_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 24, >>>>>> + a1_mst_pdata); >>>>>> +AUDIO_DIV(audio_locker_out_clk_div, AUDIO_CLK_LOCKER_CTRL, 16, 8, >>>>>> + AUDIO_PDATA(audio_locker_out_clk_mux)); >>>>>> +AUDIO_GATE(audio_locker_out_clk, AUDIO_CLK_LOCKER_CTRL, 31, >>>>>> + AUDIO_PDATA(audio_locker_out_clk_div)); >>>>>> + >>>>>> +AUDIO_MST_MCLK(audio2_vad_mclk, AUDIO2_MCLK_VAD_CTRL); >>>>>> +AUDIO_MST_MCLK(audio2_vad_clk, AUDIO2_CLK_VAD_CTRL); >>>>>> +AUDIO_MST_MCLK(audio2_pdm_dclk, AUDIO2_CLK_PDMIN_CTRL0); >>>>>> +AUDIO_MST_MCLK(audio2_pdm_sysclk, AUDIO2_CLK_PDMIN_CTRL1); >>>>>> + >>>>>> +#define AUDIO_MST_SCLK(_name, _reg0, _reg1, _pdata) \ >>>>>> + AUDIO_GATE(_name##_pre_en, (_reg0), 31, (_pdata)); \ >>>>>> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 20, 10, 0, 0, \ >>>>>> + AUDIO_PDATA(_name##_pre_en), true); \ >>>>>> + AUDIO_GATE(_name##_post_en, (_reg0), 30, \ >>>>>> + AUDIO_PDATA(_name##_div)); \ >>>>>> + AUDIO_TRIPHASE(_name, (_reg1), 1, 0, 2, 4, \ >>>>>> + AUDIO_PDATA(_name##_post_en)) >>>>>> + >>>>> >>>>> Again, I'm not a fan of this many levels of macro. I can live with it >>>>> but certainly don't want the burden of reviewing and maintaining for >>>>> clock driver. AXG / G12 and A1 are obviously closely related, so make it common. >>>>> >>>>>> +#define AUDIO_MST_LRCLK(_name, _reg0, _reg1, _pdata) \ >>>>>> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 0, 10, 10, 10, \ >>>>>> + (_pdata), false); \ >>>>>> + AUDIO_TRIPHASE(_name, (_reg1), 1, 1, 3, 5, \ >>>>>> + AUDIO_PDATA(_name##_div)) >>>>>> + >>>>>> +AUDIO_MST_SCLK(audio_mst_a_sclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >>>>>> + AUDIO_PDATA(audio_mst_a_mclk)); >>>>>> +AUDIO_MST_SCLK(audio_mst_b_sclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >>>>>> + AUDIO_PDATA(audio_mst_b_mclk)); >>>>>> +AUDIO_MST_SCLK(audio_mst_c_sclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >>>>>> + AUDIO_PDATA(audio_mst_c_mclk)); >>>>>> +AUDIO_MST_SCLK(audio_mst_d_sclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >>>>>> + AUDIO_PDATA(audio_mst_d_mclk)); >>>>>> + >>>>>> +AUDIO_MST_LRCLK(audio_mst_a_lrclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >>>>>> + AUDIO_PDATA(audio_mst_a_sclk_post_en)); >>>>>> +AUDIO_MST_LRCLK(audio_mst_b_lrclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >>>>>> + AUDIO_PDATA(audio_mst_b_sclk_post_en)); >>>>>> +AUDIO_MST_LRCLK(audio_mst_c_lrclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >>>>>> + AUDIO_PDATA(audio_mst_c_sclk_post_en)); >>>>>> +AUDIO_MST_LRCLK(audio_mst_d_lrclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >>>>>> + AUDIO_PDATA(audio_mst_d_sclk_post_en)); >>>>>> + >>>>>> +static const struct clk_parent_data a1_mst_sclk_pdata[] = { >>>>>> + { .hw = &audio_mst_a_sclk.hw }, >>>>>> + { .hw = &audio_mst_b_sclk.hw }, >>>>>> + { .hw = &audio_mst_c_sclk.hw }, >>>>>> + { .hw = &audio_mst_d_sclk.hw }, >>>>>> + { .fw_name = "slv_sclk0" }, >>>>>> + { .fw_name = "slv_sclk1" }, >>>>>> + { .fw_name = "slv_sclk2" }, >>>>>> + { .fw_name = "slv_sclk3" }, >>>>>> + { .fw_name = "slv_sclk4" }, >>>>>> + { .fw_name = "slv_sclk5" }, >>>>>> + { .fw_name = "slv_sclk6" }, >>>>>> + { .fw_name = "slv_sclk7" }, >>>>>> + { .fw_name = "slv_sclk8" }, >>>>>> + { .fw_name = "slv_sclk9" }, >>>>>> +}; >>>>>> + >>>>>> +static const struct clk_parent_data a1_mst_lrclk_pdata[] = { >>>>>> + { .hw = &audio_mst_a_lrclk.hw }, >>>>>> + { .hw = &audio_mst_b_lrclk.hw }, >>>>>> + { .hw = &audio_mst_c_lrclk.hw }, >>>>>> + { .hw = &audio_mst_d_lrclk.hw }, >>>>>> + { .fw_name = "slv_lrclk0" }, >>>>>> + { .fw_name = "slv_lrclk1" }, >>>>>> + { .fw_name = "slv_lrclk2" }, >>>>>> + { .fw_name = "slv_lrclk3" }, >>>>>> + { .fw_name = "slv_lrclk4" }, >>>>>> + { .fw_name = "slv_lrclk5" }, >>>>>> + { .fw_name = "slv_lrclk6" }, >>>>>> + { .fw_name = "slv_lrclk7" }, >>>>>> + { .fw_name = "slv_lrclk8" }, >>>>>> + { .fw_name = "slv_lrclk9" }, >>>>>> +}; >>>>>> + >>>>>> +#define AUDIO_TDM_SCLK(_name, _reg) \ >>>>>> + AUDIO_MUX(_name##_mux, (_reg), 0xf, 24, a1_mst_sclk_pdata); \ >>>>>> + AUDIO_GATE(_name##_pre_en, (_reg), 31, \ >>>>>> + AUDIO_PDATA(_name##_mux)); \ >>>>>> + AUDIO_GATE(_name##_post_en, (_reg), 30, \ >>>>>> + AUDIO_PDATA(_name##_pre_en)); \ >>>>>> + AUDIO_SCLK_WS(_name, (_reg), 1, 29, 28, \ >>>>>> + AUDIO_PDATA(_name##_post_en)) >>>>>> + >>>>>> +#define AUDIO_TDM_LRCLK(_name, _reg) \ >>>>>> + AUDIO_MUX(_name, (_reg), 0xf, 20, a1_mst_lrclk_pdata) >>>>>> + >>>>>> +AUDIO_TDM_SCLK(audio_tdmin_a_sclk, AUDIO_CLK_TDMIN_A_CTRL); >>>>>> +AUDIO_TDM_SCLK(audio_tdmin_b_sclk, AUDIO_CLK_TDMIN_B_CTRL); >>>>>> +AUDIO_TDM_SCLK(audio_tdmin_lb_sclk, AUDIO_CLK_TDMIN_LB_CTRL); >>>>>> +AUDIO_TDM_SCLK(audio_tdmout_a_sclk, AUDIO_CLK_TDMOUT_A_CTRL); >>>>>> +AUDIO_TDM_SCLK(audio_tdmout_b_sclk, AUDIO_CLK_TDMOUT_B_CTRL); >>>>>> + >>>>>> +AUDIO_TDM_LRCLK(audio_tdmin_a_lrclk, AUDIO_CLK_TDMIN_A_CTRL); >>>>>> +AUDIO_TDM_LRCLK(audio_tdmin_b_lrclk, AUDIO_CLK_TDMIN_B_CTRL); >>>>>> +AUDIO_TDM_LRCLK(audio_tdmin_lb_lrclk, AUDIO_CLK_TDMIN_LB_CTRL); >>>>>> +AUDIO_TDM_LRCLK(audio_tdmout_a_lrclk, AUDIO_CLK_TDMOUT_A_CTRL); >>>>>> +AUDIO_TDM_LRCLK(audio_tdmout_b_lrclk, AUDIO_CLK_TDMOUT_B_CTRL); >>>>>> + >>>>>> +static struct clk_hw *a1_audio_hw_clks[] = { >>>>>> + [AUD_CLKID_DDR_ARB] = &audio_ddr_arb.hw, >>>>>> + [AUD_CLKID_TDMIN_A] = &audio_tdmin_a.hw, >>>>>> + [AUD_CLKID_TDMIN_B] = &audio_tdmin_b.hw, >>>>>> + [AUD_CLKID_TDMIN_LB] = &audio_tdmin_lb.hw, >>>>>> + [AUD_CLKID_LOOPBACK] = &audio_loopback.hw, >>>>>> + [AUD_CLKID_TDMOUT_A] = &audio_tdmout_a.hw, >>>>>> + [AUD_CLKID_TDMOUT_B] = &audio_tdmout_b.hw, >>>>>> + [AUD_CLKID_FRDDR_A] = &audio_frddr_a.hw, >>>>>> + [AUD_CLKID_FRDDR_B] = &audio_frddr_b.hw, >>>>>> + [AUD_CLKID_TODDR_A] = &audio_toddr_a.hw, >>>>>> + [AUD_CLKID_TODDR_B] = &audio_toddr_b.hw, >>>>>> + [AUD_CLKID_SPDIFIN] = &audio_spdifin.hw, >>>>>> + [AUD_CLKID_RESAMPLE] = &audio_resample.hw, >>>>>> + [AUD_CLKID_EQDRC] = &audio_eqdrc.hw, >>>>>> + [AUD_CLKID_LOCKER] = &audio_audiolocker.hw, >>>>>> + [AUD_CLKID_MST_A_MCLK_SEL] = &audio_mst_a_mclk_mux.hw, >>>>>> + [AUD_CLKID_MST_A_MCLK_DIV] = &audio_mst_a_mclk_div.hw, >>>>>> + [AUD_CLKID_MST_A_MCLK] = &audio_mst_a_mclk.hw, >>>>>> + [AUD_CLKID_MST_B_MCLK_SEL] = &audio_mst_b_mclk_mux.hw, >>>>>> + [AUD_CLKID_MST_B_MCLK_DIV] = &audio_mst_b_mclk_div.hw, >>>>>> + [AUD_CLKID_MST_B_MCLK] = &audio_mst_b_mclk.hw, >>>>>> + [AUD_CLKID_MST_C_MCLK_SEL] = &audio_mst_c_mclk_mux.hw, >>>>>> + [AUD_CLKID_MST_C_MCLK_DIV] = &audio_mst_c_mclk_div.hw, >>>>>> + [AUD_CLKID_MST_C_MCLK] = &audio_mst_c_mclk.hw, >>>>>> + [AUD_CLKID_MST_D_MCLK_SEL] = &audio_mst_d_mclk_mux.hw, >>>>>> + [AUD_CLKID_MST_D_MCLK_DIV] = &audio_mst_d_mclk_div.hw, >>>>>> + [AUD_CLKID_MST_D_MCLK] = &audio_mst_d_mclk.hw, >>>>>> + [AUD_CLKID_RESAMPLE_CLK_SEL] = &audio_resample_clk_mux.hw, >>>>>> + [AUD_CLKID_RESAMPLE_CLK_DIV] = &audio_resample_clk_div.hw, >>>>>> + [AUD_CLKID_RESAMPLE_CLK] = &audio_resample_clk.hw, >>>>>> + [AUD_CLKID_LOCKER_IN_CLK_SEL] = &audio_locker_in_clk_mux.hw, >>>>>> + [AUD_CLKID_LOCKER_IN_CLK_DIV] = &audio_locker_in_clk_div.hw, >>>>>> + [AUD_CLKID_LOCKER_IN_CLK] = &audio_locker_in_clk.hw, >>>>>> + [AUD_CLKID_LOCKER_OUT_CLK_SEL] = &audio_locker_out_clk_mux.hw, >>>>>> + [AUD_CLKID_LOCKER_OUT_CLK_DIV] = &audio_locker_out_clk_div.hw, >>>>>> + [AUD_CLKID_LOCKER_OUT_CLK] = &audio_locker_out_clk.hw, >>>>>> + [AUD_CLKID_SPDIFIN_CLK_SEL] = &audio_spdifin_clk_mux.hw, >>>>>> + [AUD_CLKID_SPDIFIN_CLK_DIV] = &audio_spdifin_clk_div.hw, >>>>>> + [AUD_CLKID_SPDIFIN_CLK] = &audio_spdifin_clk.hw, >>>>>> + [AUD_CLKID_EQDRC_CLK_SEL] = &audio_eqdrc_clk_mux.hw, >>>>>> + [AUD_CLKID_EQDRC_CLK_DIV] = &audio_eqdrc_clk_div.hw, >>>>>> + [AUD_CLKID_EQDRC_CLK] = &audio_eqdrc_clk.hw, >>>>>> + [AUD_CLKID_MST_A_SCLK_PRE_EN] = &audio_mst_a_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_MST_A_SCLK_DIV] = &audio_mst_a_sclk_div.hw, >>>>>> + [AUD_CLKID_MST_A_SCLK_POST_EN] = &audio_mst_a_sclk_post_en.hw, >>>>>> + [AUD_CLKID_MST_A_SCLK] = &audio_mst_a_sclk.hw, >>>>>> + [AUD_CLKID_MST_B_SCLK_PRE_EN] = &audio_mst_b_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_MST_B_SCLK_DIV] = &audio_mst_b_sclk_div.hw, >>>>>> + [AUD_CLKID_MST_B_SCLK_POST_EN] = &audio_mst_b_sclk_post_en.hw, >>>>>> + [AUD_CLKID_MST_B_SCLK] = &audio_mst_b_sclk.hw, >>>>>> + [AUD_CLKID_MST_C_SCLK_PRE_EN] = &audio_mst_c_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_MST_C_SCLK_DIV] = &audio_mst_c_sclk_div.hw, >>>>>> + [AUD_CLKID_MST_C_SCLK_POST_EN] = &audio_mst_c_sclk_post_en.hw, >>>>>> + [AUD_CLKID_MST_C_SCLK] = &audio_mst_c_sclk.hw, >>>>>> + [AUD_CLKID_MST_D_SCLK_PRE_EN] = &audio_mst_d_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_MST_D_SCLK_DIV] = &audio_mst_d_sclk_div.hw, >>>>>> + [AUD_CLKID_MST_D_SCLK_POST_EN] = &audio_mst_d_sclk_post_en.hw, >>>>>> + [AUD_CLKID_MST_D_SCLK] = &audio_mst_d_sclk.hw, >>>>>> + [AUD_CLKID_MST_A_LRCLK_DIV] = &audio_mst_a_lrclk_div.hw, >>>>>> + [AUD_CLKID_MST_A_LRCLK] = &audio_mst_a_lrclk.hw, >>>>>> + [AUD_CLKID_MST_B_LRCLK_DIV] = &audio_mst_b_lrclk_div.hw, >>>>>> + [AUD_CLKID_MST_B_LRCLK] = &audio_mst_b_lrclk.hw, >>>>>> + [AUD_CLKID_MST_C_LRCLK_DIV] = &audio_mst_c_lrclk_div.hw, >>>>>> + [AUD_CLKID_MST_C_LRCLK] = &audio_mst_c_lrclk.hw, >>>>>> + [AUD_CLKID_MST_D_LRCLK_DIV] = &audio_mst_d_lrclk_div.hw, >>>>>> + [AUD_CLKID_MST_D_LRCLK] = &audio_mst_d_lrclk.hw, >>>>>> + [AUD_CLKID_TDMIN_A_SCLK_SEL] = &audio_tdmin_a_sclk_mux.hw, >>>>>> + [AUD_CLKID_TDMIN_A_SCLK_PRE_EN] = &audio_tdmin_a_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_TDMIN_A_SCLK_POST_EN] = &audio_tdmin_a_sclk_post_en.hw, >>>>>> + [AUD_CLKID_TDMIN_A_SCLK] = &audio_tdmin_a_sclk.hw, >>>>>> + [AUD_CLKID_TDMIN_A_LRCLK] = &audio_tdmin_a_lrclk.hw, >>>>>> + [AUD_CLKID_TDMIN_B_SCLK_SEL] = &audio_tdmin_b_sclk_mux.hw, >>>>>> + [AUD_CLKID_TDMIN_B_SCLK_PRE_EN] = &audio_tdmin_b_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_TDMIN_B_SCLK_POST_EN] = &audio_tdmin_b_sclk_post_en.hw, >>>>>> + [AUD_CLKID_TDMIN_B_SCLK] = &audio_tdmin_b_sclk.hw, >>>>>> + [AUD_CLKID_TDMIN_B_LRCLK] = &audio_tdmin_b_lrclk.hw, >>>>>> + [AUD_CLKID_TDMIN_LB_SCLK_SEL] = &audio_tdmin_lb_sclk_mux.hw, >>>>>> + [AUD_CLKID_TDMIN_LB_SCLK_PRE_EN] = &audio_tdmin_lb_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_TDMIN_LB_SCLK_POST_EN] = &audio_tdmin_lb_sclk_post_en.hw, >>>>>> + [AUD_CLKID_TDMIN_LB_SCLK] = &audio_tdmin_lb_sclk.hw, >>>>>> + [AUD_CLKID_TDMIN_LB_LRCLK] = &audio_tdmin_lb_lrclk.hw, >>>>>> + [AUD_CLKID_TDMOUT_A_SCLK_SEL] = &audio_tdmout_a_sclk_mux.hw, >>>>>> + [AUD_CLKID_TDMOUT_A_SCLK_PRE_EN] = &audio_tdmout_a_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_TDMOUT_A_SCLK_POST_EN] = &audio_tdmout_a_sclk_post_en.hw, >>>>>> + [AUD_CLKID_TDMOUT_A_SCLK] = &audio_tdmout_a_sclk.hw, >>>>>> + [AUD_CLKID_TDMOUT_A_LRCLK] = &audio_tdmout_a_lrclk.hw, >>>>>> + [AUD_CLKID_TDMOUT_B_SCLK_SEL] = &audio_tdmout_b_sclk_mux.hw, >>>>>> + [AUD_CLKID_TDMOUT_B_SCLK_PRE_EN] = &audio_tdmout_b_sclk_pre_en.hw, >>>>>> + [AUD_CLKID_TDMOUT_B_SCLK_POST_EN] = &audio_tdmout_b_sclk_post_en.hw, >>>>>> + [AUD_CLKID_TDMOUT_B_SCLK] = &audio_tdmout_b_sclk.hw, >>>>>> + [AUD_CLKID_TDMOUT_B_LRCLK] = &audio_tdmout_b_lrclk.hw, >>>>>> + >>>>>> + [AUD2_CLKID_DDR_ARB] = &audio2_ddr_arb.hw, >>>>>> + [AUD2_CLKID_PDM] = &audio2_pdm.hw, >>>>>> + [AUD2_CLKID_TDMIN_VAD] = &audio2_tdmin_vad.hw, >>>>>> + [AUD2_CLKID_TODDR_VAD] = &audio2_toddr_vad.hw, >>>>>> + [AUD2_CLKID_VAD] = &audio2_vad.hw, >>>>>> + [AUD2_CLKID_AUDIOTOP] = &audio2_audiotop.hw, >>>>>> + [AUD2_CLKID_VAD_MCLK_SEL] = &audio2_vad_mclk_mux.hw, >>>>>> + [AUD2_CLKID_VAD_MCLK_DIV] = &audio2_vad_mclk_div.hw, >>>>>> + [AUD2_CLKID_VAD_MCLK] = &audio2_vad_mclk.hw, >>>>>> + [AUD2_CLKID_VAD_CLK_SEL] = &audio2_vad_clk_mux.hw, >>>>>> + [AUD2_CLKID_VAD_CLK_DIV] = &audio2_vad_clk_div.hw, >>>>>> + [AUD2_CLKID_VAD_CLK] = &audio2_vad_clk.hw, >>>>>> + [AUD2_CLKID_PDM_DCLK_SEL] = &audio2_pdm_dclk_mux.hw, >>>>>> + [AUD2_CLKID_PDM_DCLK_DIV] = &audio2_pdm_dclk_div.hw, >>>>>> + [AUD2_CLKID_PDM_DCLK] = &audio2_pdm_dclk.hw, >>>>>> + [AUD2_CLKID_PDM_SYSCLK_SEL] = &audio2_pdm_sysclk_mux.hw, >>>>>> + [AUD2_CLKID_PDM_SYSCLK_DIV] = &audio2_pdm_sysclk_div.hw, >>>>>> + [AUD2_CLKID_PDM_SYSCLK] = &audio2_pdm_sysclk.hw, >>>>>> +}; >>>>>> + >>>>>> +static struct meson_clk_hw_data a1_audio_clks = { >>>>>> + .hws = a1_audio_hw_clks, >>>>>> + .num = ARRAY_SIZE(a1_audio_hw_clks), >>>>>> +}; >>>>>> + >>>>>> +static struct regmap *a1_audio_map(struct platform_device *pdev, >>>>>> + unsigned int index) >>>>>> +{ >>>>>> + char name[32]; >>>>>> + const struct regmap_config cfg = { >>>>>> + .reg_bits = 32, >>>>>> + .val_bits = 32, >>>>>> + .reg_stride = 4, >>>>>> + .name = name, >>>>> >>>>> Not necessary >>>>> >>>> >>>> This implementation uses two regmaps, and this field allow to avoid >>>> errors like this: >>>> >>>> [ 0.145530] debugfs: Directory 'fe050000.audio-clock-controller' with >>>> parent 'regmap' already present! >>>> >>>>>> + }; >>>>>> + void __iomem *base; >>>>>> + >>>>>> + base = devm_platform_ioremap_resource(pdev, index); >>>>>> + if (IS_ERR(base)) >>>>>> + return base; >>>>>> + >>>>>> + scnprintf(name, sizeof(name), "%d", index); >>>>>> + return devm_regmap_init_mmio(&pdev->dev, base, &cfg); >>>>>> +} >>>>> >>>>> That is overengineered. Please keep it simple. Declare the regmap_config >>>>> as static const global, and do it like axg-audio please. >>>>> >>>> >>>> This only reason why it is not "static const" because I need to set >>>> unique name for each regmap. >>>> >>>>>> + >>>>>> +static int a1_register_clk(struct platform_device *pdev, >>>>>> + struct regmap *map0, struct regmap *map1, >>>>>> + struct clk_hw *hw) >>>>>> +{ >>>>>> + struct clk_regmap *clk = container_of(hw, struct clk_regmap, hw); >>>>>> + >>>>>> + if (!hw) >>>>>> + return 0; >>>>>> + >>>>>> + switch ((unsigned long)clk->map) { >>>>>> + case AUDIO_RANGE_0: >>>>>> + clk->map = map0; >>>>>> + break; >>>>>> + case AUDIO_RANGE_1: >>>>>> + clk->map = map1; >>>>>> + break; >>>>> >>>>> ... fishy >>>>> >>>>>> + default: >>>>>> + WARN_ON(1); >>>>>> + return -EINVAL; >>>>>> + } >>>>>> + >>>>>> + return devm_clk_hw_register(&pdev->dev, hw); >>>>>> +} >>>>>> + >>>>>> +static int a1_audio_clkc_probe(struct platform_device *pdev) >>>>>> +{ >>>>>> + struct regmap *map0, *map1; >>>>>> + struct clk *clk; >>>>>> + unsigned int i; >>>>>> + int ret; >>>>>> + >>>>>> + clk = devm_clk_get_enabled(&pdev->dev, "pclk"); >>>>>> + if (WARN_ON(IS_ERR(clk))) >>>>>> + return PTR_ERR(clk); >>>>>> + >>>>>> + map0 = a1_audio_map(pdev, 0); >>>>>> + if (IS_ERR(map0)) >>>>>> + return PTR_ERR(map0); >>>>>> + >>>>>> + map1 = a1_audio_map(pdev, 1); >>>>>> + if (IS_ERR(map1)) >>>>>> + return PTR_ERR(map1); >>>>> >>>>> No - Looks to me you just have two clock controllers you are trying >>>>> force into one. >>>>> >>>> >>>> See the begining. >>>> >>>>>> + >>>>>> + /* >>>>>> + * Register and enable AUD2_CLKID_AUDIOTOP clock first. Unless >>>>>> + * it is enabled any read/write to 'map0' hangs the CPU. >>>>>> + */ >>>>>> + >>>>>> + ret = a1_register_clk(pdev, map0, map1, >>>>>> + a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]); >>>>>> + if (ret) >>>>>> + return ret; >>>>>> + >>>>>> + ret = clk_prepare_enable(a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]->clk); >>>>>> + if (ret) >>>>>> + return ret; >>>>> >>>>> Again, this shows 2 devices. The one related to your 'map0' should >>>>> request AUD2_CLKID_AUDIOTOP as input and enable it right away. >>>>> >>>> >>>> See the begining. >>>> >>>>>> + >>>>>> + for (i = 0; i < a1_audio_clks.num; i++) { >>>>>> + if (i == AUD2_CLKID_AUDIOTOP) >>>>>> + continue; >>>>>> + >>>>>> + ret = a1_register_clk(pdev, map0, map1, a1_audio_clks.hws[i]); >>>>>> + if (ret) >>>>>> + return ret; >>>>>> + } >>>>>> + >>>>>> + ret = devm_of_clk_add_hw_provider(&pdev->dev, meson_clk_hw_get, >>>>>> + &a1_audio_clks); >>>>>> + if (ret) >>>>>> + return ret; >>>>>> + >>>>>> + BUILD_BUG_ON((unsigned long)AUDIO_REG_MAP(AUDIO_SW_RESET0) != >>>>>> + AUDIO_RANGE_0); >>>>> >>>>> Why is that necessary ? >>>>> >>>> >>>> A little paranoia. Here AUDIO_SW_RESET0 is handled as map0's register, >>>> and I want to assert it. >>>> >>>>>> + return meson_audio_rstc_register(&pdev->dev, map0, >>>>>> + AUDIO_REG_OFFSET(AUDIO_SW_RESET0), 32); >>>>>> +} >>>>>> + >>>>>> +static const struct of_device_id a1_audio_clkc_match_table[] = { >>>>>> + { .compatible = "amlogic,a1-audio-clkc", }, >>>>>> + {} >>>>>> +}; >>>>>> +MODULE_DEVICE_TABLE(of, a1_audio_clkc_match_table); >>>>>> + >>>>>> +static struct platform_driver a1_audio_clkc_driver = { >>>>>> + .probe = a1_audio_clkc_probe, >>>>>> + .driver = { >>>>>> + .name = "a1-audio-clkc", >>>>>> + .of_match_table = a1_audio_clkc_match_table, >>>>>> + }, >>>>>> +}; >>>>>> +module_platform_driver(a1_audio_clkc_driver); >>>>>> + >>>>>> +MODULE_DESCRIPTION("Amlogic A1 Audio Clock driver"); >>>>>> +MODULE_AUTHOR("Jan Dakinevich <jan.dakinevich@salutedevices.com>"); >>>>>> +MODULE_LICENSE("GPL"); >>>>>> diff --git a/drivers/clk/meson/a1-audio.h b/drivers/clk/meson/a1-audio.h >>>>>> new file mode 100644 >>>>>> index 000000000000..f994e87276cd >>>>>> --- /dev/null >>>>>> +++ b/drivers/clk/meson/a1-audio.h >>>>>> @@ -0,0 +1,58 @@ >>>>>> +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ >>>>>> +/* >>>>>> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >>>>>> + * >>>>>> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>>> + */ >>>>>> + >>>>>> +#ifndef __A1_AUDIO_H >>>>>> +#define __A1_AUDIO_H >>>>>> + >>>>>> +#define AUDIO_RANGE_0 0xa >>>>>> +#define AUDIO_RANGE_1 0xb >>>>>> +#define AUDIO_RANGE_SHIFT 16 >>>>>> + >>>>>> +#define AUDIO_REG(_range, _offset) \ >>>>>> + (((_range) << AUDIO_RANGE_SHIFT) + (_offset)) >>>>>> + >>>>>> +#define AUDIO_REG_OFFSET(_reg) \ >>>>>> + ((_reg) & ((1 << AUDIO_RANGE_SHIFT) - 1)) >>>>>> + >>>>>> +#define AUDIO_REG_MAP(_reg) \ >>>>>> + ((void *)((_reg) >> AUDIO_RANGE_SHIFT)) >>>>> >>>>> That is seriouly overengineered. >>>>> The following are offset. Just write what they are. >>>>> >>>> >>>> This is all in order to keep range's identifier together with offset and >>>> then use it to store the identifier in clk_regmaps. >>>> >>>>> There is not reason to put that into a header. It is only going to be >>>>> used by a single driver. >>>>>>> + >>>>>> +#define AUDIO_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_0, 0x000) >>>>>> +#define AUDIO_MCLK_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x008) >>>>>> +#define AUDIO_MCLK_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x00c) >>>>>> +#define AUDIO_MCLK_C_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x010) >>>>>> +#define AUDIO_MCLK_D_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x014) >>>>>> +#define AUDIO_MCLK_E_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x018) >>>>>> +#define AUDIO_MCLK_F_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x01c) >>>>>> +#define AUDIO_SW_RESET0 AUDIO_REG(AUDIO_RANGE_0, 0x028) >>>>>> +#define AUDIO_MST_A_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x040) >>>>>> +#define AUDIO_MST_A_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x044) >>>>>> +#define AUDIO_MST_B_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x048) >>>>>> +#define AUDIO_MST_B_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x04c) >>>>>> +#define AUDIO_MST_C_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x050) >>>>>> +#define AUDIO_MST_C_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x054) >>>>>> +#define AUDIO_MST_D_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x058) >>>>>> +#define AUDIO_MST_D_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x05c) >>>>>> +#define AUDIO_CLK_TDMIN_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x080) >>>>>> +#define AUDIO_CLK_TDMIN_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x084) >>>>>> +#define AUDIO_CLK_TDMIN_LB_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x08c) >>>>>> +#define AUDIO_CLK_TDMOUT_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x090) >>>>>> +#define AUDIO_CLK_TDMOUT_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x094) >>>>>> +#define AUDIO_CLK_SPDIFIN_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x09c) >>>>>> +#define AUDIO_CLK_RESAMPLE_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a4) >>>>>> +#define AUDIO_CLK_LOCKER_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a8) >>>>>> +#define AUDIO_CLK_EQDRC_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0c0) >>>>>> + >>>>>> +#define AUDIO2_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_1, 0x00c) >>>>>> +#define AUDIO2_MCLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x040) >>>>>> +#define AUDIO2_CLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x044) >>>>>> +#define AUDIO2_CLK_PDMIN_CTRL0 AUDIO_REG(AUDIO_RANGE_1, 0x058) >>>>>> +#define AUDIO2_CLK_PDMIN_CTRL1 AUDIO_REG(AUDIO_RANGE_1, 0x05c) >>>>>> + >>>>>> +#include <dt-bindings/clock/amlogic,a1-audio-clkc.h> >>>>>> + >>>>>> +#endif /* __A1_AUDIO_H */ >>>>> >>>>> >>> >>> > >
On Tue 26 Mar 2024 at 21:44, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: > On 3/26/24 18:26, Jerome Brunet wrote: >> >> On Sat 23 Mar 2024 at 21:02, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >> >>> Jerome, I have reworked my driver reusing axg-audio code as most as I >>> could and now I have one more question. Lets see on this definition from >>> axg-audio: >>> >>> #define AUD_MST_MUX(_name, _reg, _flag) \ >>> AUD_MUX(_name##_sel, _reg, 0x7, 24, _flag, \ >>> mst_mux_parent_data, 0) >>> >>> #define AUD_MST_MCLK_MUX(_name, _reg) \ >>> AUD_MST_MUX(_name, _reg, CLK_MUX_ROUND_CLOSEST) >>> >>> CLK_SET_RATE_PARENT is not set here. But why? It means, that topmost pll >>> clock will not be reconfigured at runtime to satisfy the rate that was >>> requested from axg-tdm. >>> >> >> Yes, that is by design. It is another area where mainline audio differs >> greatly from AML vendor code. The PLLs are expected be to fixed and the >> audio master clock will reparent to the most adequate PLL source >> depending on the use case. >> >> This is how we manage to satisfy all audio interfaces with a very >> limited number of PLLs >> >> On AXG/G12 there is at most 6 concurrent interfaces (3 FRDDR/TODDR) - 8 >> on sm1 - and we can satisfy on that with 3 PLLs. That would not be >> possible if interfaces were having their way with the PLLs, reseting it >> everytime a stream is started. >> > The PLL rate should be carefully chosen so it can be derived easily. On >> AXG/G12/SM1 that is: >> * one PLL per rate family, to maximize clock precision >> * x24 x32: to handle different sample sizes >> * x2 until we reach the PLL limits to allow higher rates such as 384kHz >> or even higher >> > > Thank you. Now it has become much clearer. > >> If you have less PLLs on A1, you'll have to make compromises, like a less >> precise clock to support multiple family with one PLL. >> This is why the PLLs are set for each platform in DT because that choice >> may depend on the platform use case. >> > > Unfortunately, on A1 we have only one PLL. > That where compromises comes in. Pick a rate known as 'audio friendly' which match some rates and appromixate others, or use codec clock master. > Yes, for us it would be better to have hifi_pll with predefined rate. > For instance it will allow to avoid that ugly workaround in PDM (sysrate > property, etc). That is another problem entirely. Krzysztof and I already covered what you should do for this. The exact rate the PDM system clock does not matter at all because the driver queries the actual rate of the clocks and adapts. You have problems here only because you added CLK_SET_RATE_PARENT and you are triggering the concurrent usage problem I explain below. > > But what whould be preferred for upstream? I can imagine a scenario > where samples with different rate should be played, PDM attached to > fclk_divN and there are no conflicts with TDM. You are considering only PDM and 1 TDM. The SoC has 2 TDMs which could be active concurrently a different rates. > In this case > reconfiguration of hifi_pll on demand could better satisfy somebody's > requirements. No this is not possible. Doing so does not allow all the interfaces to be used concurrently. * If the PLL is not protected, existing streams get broken by new starting stream when the PLL is reconfigured * If the PLL is protected, new stream effectively starve because no clock may provide a 'good enough' rate for them This is why the reference rate must carefully chosen, something CCF cannot do for you. For example, a source of 12.288MHz (and its powers of 2) allows to match 48 and 32kHz sample rates and approximate 44.1kHz rates with an acceptable drift. > >>> >>> On 3/19/24 11:30, Jerome Brunet wrote: >>>> >>>> On Tue 19 Mar 2024 at 04:47, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >>>> >>>>> Let's start from the end: >>>>> >>>>>> No - Looks to me you just have two clock controllers you are trying >>>>> force into one. >>>>> >>>>>> Again, this shows 2 devices. The one related to your 'map0' should >>>>> request AUD2_CLKID_AUDIOTOP as input and enable it right away. >>>>> >>>>> Most of fishy workarounds that you commented is caused the fact the mmio >>>>> of this clock controller is divided into two parts. Compare it with >>>>> axg-audio driver, things that was part of contigous memory region (like >>>>> pdm) here are moved to second region. Is this enough to make a guess >>>>> that these are two devices? >>>> >>>> I see obsolutely no reason to think it is a single device nor to add all the quirks >>>> you have the way you did. So yes, in that case, 2 zones, 2 devices. >>>> >>>>> >>>>> Concerning AUD2_CLKID_AUDIOTOP clock, as it turned out, it must be >>>>> enabled before enabling of clocks from second region too. That is >>>>> AUD2_CLKID_AUDIOTOP clock feeds both parts of this clock controller. >>>>> >>>> >>>> Yes. I understood the first time around and already commented on that. >>>> >>>>> >>>>> On 3/15/24 12:20, Jerome Brunet wrote: >>>>>> >>>>>> On Fri 15 Mar 2024 at 02:21, Jan Dakinevich <jan.dakinevich@salutedevices.com> wrote: >>>>>> >>>>>>> This controller provides clocks and reset functionality for audio >>>>>>> peripherals on Amlogic A1 SoC family. >>>>>>> >>>>>>> The driver is almost identical to 'axg-audio', however it would be better >>>>>>> to keep it separate due to following reasons: >>>>>>> >>>>>>> - significant amount of bits has another definition. I will bring there >>>>>>> a mess of new defines with A1_ suffixes. >>>>>>> >>>>>>> - registers of this controller are located in two separate regions. It >>>>>>> will give a lot of complications for 'axg-audio' to support this. >>>>>>> >>>>>>> Signed-off-by: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>>>> --- >>>>>>> drivers/clk/meson/Kconfig | 13 + >>>>>>> drivers/clk/meson/Makefile | 1 + >>>>>>> drivers/clk/meson/a1-audio.c | 556 +++++++++++++++++++++++++++++++++++ >>>>>>> drivers/clk/meson/a1-audio.h | 58 ++++ >>>>>>> 4 files changed, 628 insertions(+) >>>>>>> create mode 100644 drivers/clk/meson/a1-audio.c >>>>>>> create mode 100644 drivers/clk/meson/a1-audio.h >>>>>>> >>>>>>> diff --git a/drivers/clk/meson/Kconfig b/drivers/clk/meson/Kconfig >>>>>>> index d6a2fa5f7e88..80c4a18c83d2 100644 >>>>>>> --- a/drivers/clk/meson/Kconfig >>>>>>> +++ b/drivers/clk/meson/Kconfig >>>>>>> @@ -133,6 +133,19 @@ config COMMON_CLK_A1_PERIPHERALS >>>>>>> device, A1 SoC Family. Say Y if you want A1 Peripherals clock >>>>>>> controller to work. >>>>>>> >>>>>>> +config COMMON_CLK_A1_AUDIO >>>>>>> + tristate "Amlogic A1 SoC Audio clock controller support" >>>>>>> + depends on ARM64 >>>>>>> + select COMMON_CLK_MESON_REGMAP >>>>>>> + select COMMON_CLK_MESON_CLKC_UTILS >>>>>>> + select COMMON_CLK_MESON_PHASE >>>>>>> + select COMMON_CLK_MESON_SCLK_DIV >>>>>>> + select COMMON_CLK_MESON_AUDIO_RSTC >>>>>>> + help >>>>>>> + Support for the Audio clock controller on Amlogic A113L based >>>>>>> + device, A1 SoC Family. Say Y if you want A1 Audio clock controller >>>>>>> + to work. >>>>>>> + >>>>>>> config COMMON_CLK_G12A >>>>>>> tristate "G12 and SM1 SoC clock controllers support" >>>>>>> depends on ARM64 >>>>>>> diff --git a/drivers/clk/meson/Makefile b/drivers/clk/meson/Makefile >>>>>>> index 88d94921a4dc..4968fc7ad555 100644 >>>>>>> --- a/drivers/clk/meson/Makefile >>>>>>> +++ b/drivers/clk/meson/Makefile >>>>>>> @@ -20,6 +20,7 @@ obj-$(CONFIG_COMMON_CLK_AXG) += axg.o axg-aoclk.o >>>>>>> obj-$(CONFIG_COMMON_CLK_AXG_AUDIO) += axg-audio.o >>>>>>> obj-$(CONFIG_COMMON_CLK_A1_PLL) += a1-pll.o >>>>>>> obj-$(CONFIG_COMMON_CLK_A1_PERIPHERALS) += a1-peripherals.o >>>>>>> +obj-$(CONFIG_COMMON_CLK_A1_AUDIO) += a1-audio.o >>>>>>> obj-$(CONFIG_COMMON_CLK_GXBB) += gxbb.o gxbb-aoclk.o >>>>>>> obj-$(CONFIG_COMMON_CLK_G12A) += g12a.o g12a-aoclk.o >>>>>>> obj-$(CONFIG_COMMON_CLK_MESON8B) += meson8b.o meson8-ddr.o >>>>>>> diff --git a/drivers/clk/meson/a1-audio.c b/drivers/clk/meson/a1-audio.c >>>>>>> new file mode 100644 >>>>>>> index 000000000000..6039116c93ba >>>>>>> --- /dev/null >>>>>>> +++ b/drivers/clk/meson/a1-audio.c >>>>>>> @@ -0,0 +1,556 @@ >>>>>>> +// SPDX-License-Identifier: (GPL-2.0 OR MIT) >>>>>>> +/* >>>>>>> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >>>>>>> + * >>>>>>> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>>>> + */ >>>>>>> + >>>>>>> +#include <linux/clk.h> >>>>>>> +#include <linux/clk-provider.h> >>>>>>> +#include <linux/init.h> >>>>>>> +#include <linux/of_device.h> >>>>>>> +#include <linux/module.h> >>>>>>> +#include <linux/platform_device.h> >>>>>>> +#include <linux/regmap.h> >>>>>>> +#include <linux/reset.h> >>>>>>> +#include <linux/reset-controller.h> >>>>>>> +#include <linux/slab.h> >>>>>>> + >>>>>>> +#include "meson-clkc-utils.h" >>>>>>> +#include "meson-audio-rstc.h" >>>>>>> +#include "clk-regmap.h" >>>>>>> +#include "clk-phase.h" >>>>>>> +#include "sclk-div.h" >>>>>>> +#include "a1-audio.h" >>>>>>> + >>>>>>> +#define AUDIO_PDATA(_name) \ >>>>>>> + ((const struct clk_parent_data[]) { { .hw = &(_name).hw } }) >>>>>> >>>>>> Not a fan - yet another level of macro. >>>>>> >>>>>>> + >>>>>>> +#define AUDIO_MUX(_name, _reg, _mask, _shift, _pdata) \ >>>>>>> +static struct clk_regmap _name = { \ >>>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>>> + .data = &(struct clk_regmap_mux_data){ \ >>>>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .mask = (_mask), \ >>>>>>> + .shift = (_shift), \ >>>>>>> + }, \ >>>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>>> + .name = #_name, \ >>>>>>> + .ops = &clk_regmap_mux_ops, \ >>>>>>> + .parent_data = (_pdata), \ >>>>>>> + .num_parents = ARRAY_SIZE(_pdata), \ >>>>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>>>> + }, \ >>>>>>> +} >>>>>>> + >>>>>>> +#define AUDIO_DIV(_name, _reg, _shift, _width, _pdata) \ >>>>>>> +static struct clk_regmap _name = { \ >>>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>>> + .data = &(struct clk_regmap_div_data){ \ >>>>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .shift = (_shift), \ >>>>>>> + .width = (_width), \ >>>>>>> + }, \ >>>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>>> + .name = #_name, \ >>>>>>> + .ops = &clk_regmap_divider_ops, \ >>>>>>> + .parent_data = (_pdata), \ >>>>>>> + .num_parents = 1, \ >>>>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>>>> + }, \ >>>>>>> +} >>>>>>> + >>>>>>> +#define AUDIO_GATE(_name, _reg, _bit, _pdata) \ >>>>>>> +static struct clk_regmap _name = { \ >>>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>>> + .data = &(struct clk_regmap_gate_data){ \ >>>>>>> + .offset = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .bit_idx = (_bit), \ >>>>>>> + }, \ >>>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>>> + .name = #_name, \ >>>>>>> + .ops = &clk_regmap_gate_ops, \ >>>>>>> + .parent_data = (_pdata), \ >>>>>>> + .num_parents = 1, \ >>>>>>> + .flags = CLK_SET_RATE_PARENT, \ >>>>>>> + }, \ >>>>>>> +} >>>>>>> + >>>>>>> +#define AUDIO_SCLK_DIV(_name, _reg, _div_shift, _div_width, \ >>>>>>> + _hi_shift, _hi_width, _pdata, _set_rate_parent) \ >>>>>>> +static struct clk_regmap _name = { \ >>>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>>> + .data = &(struct meson_sclk_div_data) { \ >>>>>>> + .div = { \ >>>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .shift = (_div_shift), \ >>>>>>> + .width = (_div_width), \ >>>>>>> + }, \ >>>>>>> + .hi = { \ >>>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .shift = (_hi_shift), \ >>>>>>> + .width = (_hi_width), \ >>>>>>> + }, \ >>>>>>> + }, \ >>>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>>> + .name = #_name, \ >>>>>>> + .ops = &meson_sclk_div_ops, \ >>>>>>> + .parent_data = (_pdata), \ >>>>>>> + .num_parents = 1, \ >>>>>>> + .flags = (_set_rate_parent) ? CLK_SET_RATE_PARENT : 0, \ >>>>>> >>>>>> Does not help readeability. Just pass the flag as axg-audio does. >>>>>> >>>>>>> + }, \ >>>>>>> +} >>>>>>> + >>>>>>> +#define AUDIO_TRIPHASE(_name, _reg, _width, _shift0, _shift1, _shift2, \ >>>>>>> + _pdata) \ >>>>>>> +static struct clk_regmap _name = { \ >>>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>>> + .data = &(struct meson_clk_triphase_data) { \ >>>>>>> + .ph0 = { \ >>>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .shift = (_shift0), \ >>>>>>> + .width = (_width), \ >>>>>>> + }, \ >>>>>>> + .ph1 = { \ >>>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .shift = (_shift1), \ >>>>>>> + .width = (_width), \ >>>>>>> + }, \ >>>>>>> + .ph2 = { \ >>>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .shift = (_shift2), \ >>>>>>> + .width = (_width), \ >>>>>>> + }, \ >>>>>>> + }, \ >>>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>>> + .name = #_name, \ >>>>>>> + .ops = &meson_clk_triphase_ops, \ >>>>>>> + .parent_data = (_pdata), \ >>>>>>> + .num_parents = 1, \ >>>>>>> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >>>>>>> + }, \ >>>>>>> +} >>>>>>> + >>>>>>> +#define AUDIO_SCLK_WS(_name, _reg, _width, _shift_ph, _shift_ws, \ >>>>>>> + _pdata) \ >>>>>>> +static struct clk_regmap _name = { \ >>>>>>> + .map = AUDIO_REG_MAP(_reg), \ >>>>>>> + .data = &(struct meson_sclk_ws_inv_data) { \ >>>>>>> + .ph = { \ >>>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .shift = (_shift_ph), \ >>>>>>> + .width = (_width), \ >>>>>>> + }, \ >>>>>>> + .ws = { \ >>>>>>> + .reg_off = AUDIO_REG_OFFSET(_reg), \ >>>>>>> + .shift = (_shift_ws), \ >>>>>>> + .width = (_width), \ >>>>>>> + }, \ >>>>>>> + }, \ >>>>>>> + .hw.init = &(struct clk_init_data) { \ >>>>>>> + .name = #_name, \ >>>>>>> + .ops = &meson_sclk_ws_inv_ops, \ >>>>>>> + .parent_data = (_pdata), \ >>>>>>> + .num_parents = 1, \ >>>>>>> + .flags = CLK_SET_RATE_PARENT | CLK_DUTY_CYCLE_PARENT, \ >>>>>>> + }, \ >>>>>>> +} >>>>>> >>>>>> All the above does essentially the same things as the macro of >>>>>> axg-audio, to some minor differences. Yet it is another set to maintain. >>>>>> >>>>> >>>>> Except one thing... Here I keep memory identifier to which this clock >>>>> belongs: >>>>> >>>>> .map = AUDIO_REG_MAP(_reg), >>>>> >>>>> It is workaround, but ->map the only common field in clk_regmap that >>>>> could be used for this purpose. >>>>> >>>>> >>>>>> I'd much prefer if you put the axg-audio macro in a header a re-used >>>>>> those. There would a single set to maintain. You may then specialize the >>>>>> included in the driver C file, to avoid redundant parameters >>>>>> >>>>>> Rework axg-audio to use clk_parent_data if you must, but not in the same >>>>>> series please. >>>>>> >>>>>>> + >>>>>>> +static const struct clk_parent_data a1_pclk_pdata[] = { >>>>>>> + { .fw_name = "pclk", }, >>>>>>> +}; >>>>>>> + >>>>>>> +AUDIO_GATE(audio_ddr_arb, AUDIO_CLK_GATE_EN0, 0, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_tdmin_a, AUDIO_CLK_GATE_EN0, 1, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_tdmin_b, AUDIO_CLK_GATE_EN0, 2, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_tdmin_lb, AUDIO_CLK_GATE_EN0, 3, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_loopback, AUDIO_CLK_GATE_EN0, 4, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_tdmout_a, AUDIO_CLK_GATE_EN0, 5, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_tdmout_b, AUDIO_CLK_GATE_EN0, 6, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_frddr_a, AUDIO_CLK_GATE_EN0, 7, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_frddr_b, AUDIO_CLK_GATE_EN0, 8, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_toddr_a, AUDIO_CLK_GATE_EN0, 9, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_toddr_b, AUDIO_CLK_GATE_EN0, 10, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_spdifin, AUDIO_CLK_GATE_EN0, 11, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_resample, AUDIO_CLK_GATE_EN0, 12, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_eqdrc, AUDIO_CLK_GATE_EN0, 13, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio_audiolocker, AUDIO_CLK_GATE_EN0, 14, a1_pclk_pdata); >>>>>> This is what I mean by redundant parameter ^ >>>>>> >>>>> >>>>> Yep. I could define something like AUDIO_PCLK_GATE(). >>>>> >>>>>>> + >>>>>>> +AUDIO_GATE(audio2_ddr_arb, AUDIO2_CLK_GATE_EN0, 0, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio2_pdm, AUDIO2_CLK_GATE_EN0, 1, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio2_tdmin_vad, AUDIO2_CLK_GATE_EN0, 2, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio2_toddr_vad, AUDIO2_CLK_GATE_EN0, 3, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio2_vad, AUDIO2_CLK_GATE_EN0, 4, a1_pclk_pdata); >>>>>>> +AUDIO_GATE(audio2_audiotop, AUDIO2_CLK_GATE_EN0, 7, a1_pclk_pdata); >>>>>>> + >>>>>>> +static const struct clk_parent_data a1_mst_pdata[] = { >>>>>>> + { .fw_name = "dds_in" }, >>>>>>> + { .fw_name = "fclk_div2" }, >>>>>>> + { .fw_name = "fclk_div3" }, >>>>>>> + { .fw_name = "hifi_pll" }, >>>>>>> + { .fw_name = "xtal" }, >>>>>>> +}; >>>>>>> + >>>>>>> +#define AUDIO_MST_MCLK(_name, _reg) \ >>>>>>> + AUDIO_MUX(_name##_mux, (_reg), 0x7, 24, a1_mst_pdata); \ >>>>>>> + AUDIO_DIV(_name##_div, (_reg), 0, 16, \ >>>>>>> + AUDIO_PDATA(_name##_mux)); \ >>>>>>> + AUDIO_GATE(_name, (_reg), 31, AUDIO_PDATA(_name##_div)) >>>>>>> + >>>>>>> +AUDIO_MST_MCLK(audio_mst_a_mclk, AUDIO_MCLK_A_CTRL); >>>>>>> +AUDIO_MST_MCLK(audio_mst_b_mclk, AUDIO_MCLK_B_CTRL); >>>>>>> +AUDIO_MST_MCLK(audio_mst_c_mclk, AUDIO_MCLK_C_CTRL); >>>>>>> +AUDIO_MST_MCLK(audio_mst_d_mclk, AUDIO_MCLK_D_CTRL); >>>>>>> +AUDIO_MST_MCLK(audio_spdifin_clk, AUDIO_CLK_SPDIFIN_CTRL); >>>>>>> +AUDIO_MST_MCLK(audio_eqdrc_clk, AUDIO_CLK_EQDRC_CTRL); >>>>>>> + >>>>>>> +AUDIO_MUX(audio_resample_clk_mux, AUDIO_CLK_RESAMPLE_CTRL, 0xf, 24, >>>>>>> + a1_mst_pdata); >>>>>>> +AUDIO_DIV(audio_resample_clk_div, AUDIO_CLK_RESAMPLE_CTRL, 0, 8, >>>>>>> + AUDIO_PDATA(audio_resample_clk_mux)); >>>>>>> +AUDIO_GATE(audio_resample_clk, AUDIO_CLK_RESAMPLE_CTRL, 31, >>>>>>> + AUDIO_PDATA(audio_resample_clk_div)); >>>>>>> + >>>>>>> +AUDIO_MUX(audio_locker_in_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 8, >>>>>>> + a1_mst_pdata); >>>>>>> +AUDIO_DIV(audio_locker_in_clk_div, AUDIO_CLK_LOCKER_CTRL, 0, 8, >>>>>>> + AUDIO_PDATA(audio_locker_in_clk_mux)); >>>>>>> +AUDIO_GATE(audio_locker_in_clk, AUDIO_CLK_LOCKER_CTRL, 15, >>>>>>> + AUDIO_PDATA(audio_locker_in_clk_div)); >>>>>>> + >>>>>>> +AUDIO_MUX(audio_locker_out_clk_mux, AUDIO_CLK_LOCKER_CTRL, 0xf, 24, >>>>>>> + a1_mst_pdata); >>>>>>> +AUDIO_DIV(audio_locker_out_clk_div, AUDIO_CLK_LOCKER_CTRL, 16, 8, >>>>>>> + AUDIO_PDATA(audio_locker_out_clk_mux)); >>>>>>> +AUDIO_GATE(audio_locker_out_clk, AUDIO_CLK_LOCKER_CTRL, 31, >>>>>>> + AUDIO_PDATA(audio_locker_out_clk_div)); >>>>>>> + >>>>>>> +AUDIO_MST_MCLK(audio2_vad_mclk, AUDIO2_MCLK_VAD_CTRL); >>>>>>> +AUDIO_MST_MCLK(audio2_vad_clk, AUDIO2_CLK_VAD_CTRL); >>>>>>> +AUDIO_MST_MCLK(audio2_pdm_dclk, AUDIO2_CLK_PDMIN_CTRL0); >>>>>>> +AUDIO_MST_MCLK(audio2_pdm_sysclk, AUDIO2_CLK_PDMIN_CTRL1); >>>>>>> + >>>>>>> +#define AUDIO_MST_SCLK(_name, _reg0, _reg1, _pdata) \ >>>>>>> + AUDIO_GATE(_name##_pre_en, (_reg0), 31, (_pdata)); \ >>>>>>> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 20, 10, 0, 0, \ >>>>>>> + AUDIO_PDATA(_name##_pre_en), true); \ >>>>>>> + AUDIO_GATE(_name##_post_en, (_reg0), 30, \ >>>>>>> + AUDIO_PDATA(_name##_div)); \ >>>>>>> + AUDIO_TRIPHASE(_name, (_reg1), 1, 0, 2, 4, \ >>>>>>> + AUDIO_PDATA(_name##_post_en)) >>>>>>> + >>>>>> >>>>>> Again, I'm not a fan of this many levels of macro. I can live with it >>>>>> but certainly don't want the burden of reviewing and maintaining for >>>>>> clock driver. AXG / G12 and A1 are obviously closely related, so make it common. >>>>>> >>>>>>> +#define AUDIO_MST_LRCLK(_name, _reg0, _reg1, _pdata) \ >>>>>>> + AUDIO_SCLK_DIV(_name##_div, (_reg0), 0, 10, 10, 10, \ >>>>>>> + (_pdata), false); \ >>>>>>> + AUDIO_TRIPHASE(_name, (_reg1), 1, 1, 3, 5, \ >>>>>>> + AUDIO_PDATA(_name##_div)) >>>>>>> + >>>>>>> +AUDIO_MST_SCLK(audio_mst_a_sclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >>>>>>> + AUDIO_PDATA(audio_mst_a_mclk)); >>>>>>> +AUDIO_MST_SCLK(audio_mst_b_sclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >>>>>>> + AUDIO_PDATA(audio_mst_b_mclk)); >>>>>>> +AUDIO_MST_SCLK(audio_mst_c_sclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >>>>>>> + AUDIO_PDATA(audio_mst_c_mclk)); >>>>>>> +AUDIO_MST_SCLK(audio_mst_d_sclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >>>>>>> + AUDIO_PDATA(audio_mst_d_mclk)); >>>>>>> + >>>>>>> +AUDIO_MST_LRCLK(audio_mst_a_lrclk, AUDIO_MST_A_SCLK_CTRL0, AUDIO_MST_A_SCLK_CTRL1, >>>>>>> + AUDIO_PDATA(audio_mst_a_sclk_post_en)); >>>>>>> +AUDIO_MST_LRCLK(audio_mst_b_lrclk, AUDIO_MST_B_SCLK_CTRL0, AUDIO_MST_B_SCLK_CTRL1, >>>>>>> + AUDIO_PDATA(audio_mst_b_sclk_post_en)); >>>>>>> +AUDIO_MST_LRCLK(audio_mst_c_lrclk, AUDIO_MST_C_SCLK_CTRL0, AUDIO_MST_C_SCLK_CTRL1, >>>>>>> + AUDIO_PDATA(audio_mst_c_sclk_post_en)); >>>>>>> +AUDIO_MST_LRCLK(audio_mst_d_lrclk, AUDIO_MST_D_SCLK_CTRL0, AUDIO_MST_D_SCLK_CTRL1, >>>>>>> + AUDIO_PDATA(audio_mst_d_sclk_post_en)); >>>>>>> + >>>>>>> +static const struct clk_parent_data a1_mst_sclk_pdata[] = { >>>>>>> + { .hw = &audio_mst_a_sclk.hw }, >>>>>>> + { .hw = &audio_mst_b_sclk.hw }, >>>>>>> + { .hw = &audio_mst_c_sclk.hw }, >>>>>>> + { .hw = &audio_mst_d_sclk.hw }, >>>>>>> + { .fw_name = "slv_sclk0" }, >>>>>>> + { .fw_name = "slv_sclk1" }, >>>>>>> + { .fw_name = "slv_sclk2" }, >>>>>>> + { .fw_name = "slv_sclk3" }, >>>>>>> + { .fw_name = "slv_sclk4" }, >>>>>>> + { .fw_name = "slv_sclk5" }, >>>>>>> + { .fw_name = "slv_sclk6" }, >>>>>>> + { .fw_name = "slv_sclk7" }, >>>>>>> + { .fw_name = "slv_sclk8" }, >>>>>>> + { .fw_name = "slv_sclk9" }, >>>>>>> +}; >>>>>>> + >>>>>>> +static const struct clk_parent_data a1_mst_lrclk_pdata[] = { >>>>>>> + { .hw = &audio_mst_a_lrclk.hw }, >>>>>>> + { .hw = &audio_mst_b_lrclk.hw }, >>>>>>> + { .hw = &audio_mst_c_lrclk.hw }, >>>>>>> + { .hw = &audio_mst_d_lrclk.hw }, >>>>>>> + { .fw_name = "slv_lrclk0" }, >>>>>>> + { .fw_name = "slv_lrclk1" }, >>>>>>> + { .fw_name = "slv_lrclk2" }, >>>>>>> + { .fw_name = "slv_lrclk3" }, >>>>>>> + { .fw_name = "slv_lrclk4" }, >>>>>>> + { .fw_name = "slv_lrclk5" }, >>>>>>> + { .fw_name = "slv_lrclk6" }, >>>>>>> + { .fw_name = "slv_lrclk7" }, >>>>>>> + { .fw_name = "slv_lrclk8" }, >>>>>>> + { .fw_name = "slv_lrclk9" }, >>>>>>> +}; >>>>>>> + >>>>>>> +#define AUDIO_TDM_SCLK(_name, _reg) \ >>>>>>> + AUDIO_MUX(_name##_mux, (_reg), 0xf, 24, a1_mst_sclk_pdata); \ >>>>>>> + AUDIO_GATE(_name##_pre_en, (_reg), 31, \ >>>>>>> + AUDIO_PDATA(_name##_mux)); \ >>>>>>> + AUDIO_GATE(_name##_post_en, (_reg), 30, \ >>>>>>> + AUDIO_PDATA(_name##_pre_en)); \ >>>>>>> + AUDIO_SCLK_WS(_name, (_reg), 1, 29, 28, \ >>>>>>> + AUDIO_PDATA(_name##_post_en)) >>>>>>> + >>>>>>> +#define AUDIO_TDM_LRCLK(_name, _reg) \ >>>>>>> + AUDIO_MUX(_name, (_reg), 0xf, 20, a1_mst_lrclk_pdata) >>>>>>> + >>>>>>> +AUDIO_TDM_SCLK(audio_tdmin_a_sclk, AUDIO_CLK_TDMIN_A_CTRL); >>>>>>> +AUDIO_TDM_SCLK(audio_tdmin_b_sclk, AUDIO_CLK_TDMIN_B_CTRL); >>>>>>> +AUDIO_TDM_SCLK(audio_tdmin_lb_sclk, AUDIO_CLK_TDMIN_LB_CTRL); >>>>>>> +AUDIO_TDM_SCLK(audio_tdmout_a_sclk, AUDIO_CLK_TDMOUT_A_CTRL); >>>>>>> +AUDIO_TDM_SCLK(audio_tdmout_b_sclk, AUDIO_CLK_TDMOUT_B_CTRL); >>>>>>> + >>>>>>> +AUDIO_TDM_LRCLK(audio_tdmin_a_lrclk, AUDIO_CLK_TDMIN_A_CTRL); >>>>>>> +AUDIO_TDM_LRCLK(audio_tdmin_b_lrclk, AUDIO_CLK_TDMIN_B_CTRL); >>>>>>> +AUDIO_TDM_LRCLK(audio_tdmin_lb_lrclk, AUDIO_CLK_TDMIN_LB_CTRL); >>>>>>> +AUDIO_TDM_LRCLK(audio_tdmout_a_lrclk, AUDIO_CLK_TDMOUT_A_CTRL); >>>>>>> +AUDIO_TDM_LRCLK(audio_tdmout_b_lrclk, AUDIO_CLK_TDMOUT_B_CTRL); >>>>>>> + >>>>>>> +static struct clk_hw *a1_audio_hw_clks[] = { >>>>>>> + [AUD_CLKID_DDR_ARB] = &audio_ddr_arb.hw, >>>>>>> + [AUD_CLKID_TDMIN_A] = &audio_tdmin_a.hw, >>>>>>> + [AUD_CLKID_TDMIN_B] = &audio_tdmin_b.hw, >>>>>>> + [AUD_CLKID_TDMIN_LB] = &audio_tdmin_lb.hw, >>>>>>> + [AUD_CLKID_LOOPBACK] = &audio_loopback.hw, >>>>>>> + [AUD_CLKID_TDMOUT_A] = &audio_tdmout_a.hw, >>>>>>> + [AUD_CLKID_TDMOUT_B] = &audio_tdmout_b.hw, >>>>>>> + [AUD_CLKID_FRDDR_A] = &audio_frddr_a.hw, >>>>>>> + [AUD_CLKID_FRDDR_B] = &audio_frddr_b.hw, >>>>>>> + [AUD_CLKID_TODDR_A] = &audio_toddr_a.hw, >>>>>>> + [AUD_CLKID_TODDR_B] = &audio_toddr_b.hw, >>>>>>> + [AUD_CLKID_SPDIFIN] = &audio_spdifin.hw, >>>>>>> + [AUD_CLKID_RESAMPLE] = &audio_resample.hw, >>>>>>> + [AUD_CLKID_EQDRC] = &audio_eqdrc.hw, >>>>>>> + [AUD_CLKID_LOCKER] = &audio_audiolocker.hw, >>>>>>> + [AUD_CLKID_MST_A_MCLK_SEL] = &audio_mst_a_mclk_mux.hw, >>>>>>> + [AUD_CLKID_MST_A_MCLK_DIV] = &audio_mst_a_mclk_div.hw, >>>>>>> + [AUD_CLKID_MST_A_MCLK] = &audio_mst_a_mclk.hw, >>>>>>> + [AUD_CLKID_MST_B_MCLK_SEL] = &audio_mst_b_mclk_mux.hw, >>>>>>> + [AUD_CLKID_MST_B_MCLK_DIV] = &audio_mst_b_mclk_div.hw, >>>>>>> + [AUD_CLKID_MST_B_MCLK] = &audio_mst_b_mclk.hw, >>>>>>> + [AUD_CLKID_MST_C_MCLK_SEL] = &audio_mst_c_mclk_mux.hw, >>>>>>> + [AUD_CLKID_MST_C_MCLK_DIV] = &audio_mst_c_mclk_div.hw, >>>>>>> + [AUD_CLKID_MST_C_MCLK] = &audio_mst_c_mclk.hw, >>>>>>> + [AUD_CLKID_MST_D_MCLK_SEL] = &audio_mst_d_mclk_mux.hw, >>>>>>> + [AUD_CLKID_MST_D_MCLK_DIV] = &audio_mst_d_mclk_div.hw, >>>>>>> + [AUD_CLKID_MST_D_MCLK] = &audio_mst_d_mclk.hw, >>>>>>> + [AUD_CLKID_RESAMPLE_CLK_SEL] = &audio_resample_clk_mux.hw, >>>>>>> + [AUD_CLKID_RESAMPLE_CLK_DIV] = &audio_resample_clk_div.hw, >>>>>>> + [AUD_CLKID_RESAMPLE_CLK] = &audio_resample_clk.hw, >>>>>>> + [AUD_CLKID_LOCKER_IN_CLK_SEL] = &audio_locker_in_clk_mux.hw, >>>>>>> + [AUD_CLKID_LOCKER_IN_CLK_DIV] = &audio_locker_in_clk_div.hw, >>>>>>> + [AUD_CLKID_LOCKER_IN_CLK] = &audio_locker_in_clk.hw, >>>>>>> + [AUD_CLKID_LOCKER_OUT_CLK_SEL] = &audio_locker_out_clk_mux.hw, >>>>>>> + [AUD_CLKID_LOCKER_OUT_CLK_DIV] = &audio_locker_out_clk_div.hw, >>>>>>> + [AUD_CLKID_LOCKER_OUT_CLK] = &audio_locker_out_clk.hw, >>>>>>> + [AUD_CLKID_SPDIFIN_CLK_SEL] = &audio_spdifin_clk_mux.hw, >>>>>>> + [AUD_CLKID_SPDIFIN_CLK_DIV] = &audio_spdifin_clk_div.hw, >>>>>>> + [AUD_CLKID_SPDIFIN_CLK] = &audio_spdifin_clk.hw, >>>>>>> + [AUD_CLKID_EQDRC_CLK_SEL] = &audio_eqdrc_clk_mux.hw, >>>>>>> + [AUD_CLKID_EQDRC_CLK_DIV] = &audio_eqdrc_clk_div.hw, >>>>>>> + [AUD_CLKID_EQDRC_CLK] = &audio_eqdrc_clk.hw, >>>>>>> + [AUD_CLKID_MST_A_SCLK_PRE_EN] = &audio_mst_a_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_MST_A_SCLK_DIV] = &audio_mst_a_sclk_div.hw, >>>>>>> + [AUD_CLKID_MST_A_SCLK_POST_EN] = &audio_mst_a_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_MST_A_SCLK] = &audio_mst_a_sclk.hw, >>>>>>> + [AUD_CLKID_MST_B_SCLK_PRE_EN] = &audio_mst_b_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_MST_B_SCLK_DIV] = &audio_mst_b_sclk_div.hw, >>>>>>> + [AUD_CLKID_MST_B_SCLK_POST_EN] = &audio_mst_b_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_MST_B_SCLK] = &audio_mst_b_sclk.hw, >>>>>>> + [AUD_CLKID_MST_C_SCLK_PRE_EN] = &audio_mst_c_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_MST_C_SCLK_DIV] = &audio_mst_c_sclk_div.hw, >>>>>>> + [AUD_CLKID_MST_C_SCLK_POST_EN] = &audio_mst_c_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_MST_C_SCLK] = &audio_mst_c_sclk.hw, >>>>>>> + [AUD_CLKID_MST_D_SCLK_PRE_EN] = &audio_mst_d_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_MST_D_SCLK_DIV] = &audio_mst_d_sclk_div.hw, >>>>>>> + [AUD_CLKID_MST_D_SCLK_POST_EN] = &audio_mst_d_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_MST_D_SCLK] = &audio_mst_d_sclk.hw, >>>>>>> + [AUD_CLKID_MST_A_LRCLK_DIV] = &audio_mst_a_lrclk_div.hw, >>>>>>> + [AUD_CLKID_MST_A_LRCLK] = &audio_mst_a_lrclk.hw, >>>>>>> + [AUD_CLKID_MST_B_LRCLK_DIV] = &audio_mst_b_lrclk_div.hw, >>>>>>> + [AUD_CLKID_MST_B_LRCLK] = &audio_mst_b_lrclk.hw, >>>>>>> + [AUD_CLKID_MST_C_LRCLK_DIV] = &audio_mst_c_lrclk_div.hw, >>>>>>> + [AUD_CLKID_MST_C_LRCLK] = &audio_mst_c_lrclk.hw, >>>>>>> + [AUD_CLKID_MST_D_LRCLK_DIV] = &audio_mst_d_lrclk_div.hw, >>>>>>> + [AUD_CLKID_MST_D_LRCLK] = &audio_mst_d_lrclk.hw, >>>>>>> + [AUD_CLKID_TDMIN_A_SCLK_SEL] = &audio_tdmin_a_sclk_mux.hw, >>>>>>> + [AUD_CLKID_TDMIN_A_SCLK_PRE_EN] = &audio_tdmin_a_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_TDMIN_A_SCLK_POST_EN] = &audio_tdmin_a_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_TDMIN_A_SCLK] = &audio_tdmin_a_sclk.hw, >>>>>>> + [AUD_CLKID_TDMIN_A_LRCLK] = &audio_tdmin_a_lrclk.hw, >>>>>>> + [AUD_CLKID_TDMIN_B_SCLK_SEL] = &audio_tdmin_b_sclk_mux.hw, >>>>>>> + [AUD_CLKID_TDMIN_B_SCLK_PRE_EN] = &audio_tdmin_b_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_TDMIN_B_SCLK_POST_EN] = &audio_tdmin_b_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_TDMIN_B_SCLK] = &audio_tdmin_b_sclk.hw, >>>>>>> + [AUD_CLKID_TDMIN_B_LRCLK] = &audio_tdmin_b_lrclk.hw, >>>>>>> + [AUD_CLKID_TDMIN_LB_SCLK_SEL] = &audio_tdmin_lb_sclk_mux.hw, >>>>>>> + [AUD_CLKID_TDMIN_LB_SCLK_PRE_EN] = &audio_tdmin_lb_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_TDMIN_LB_SCLK_POST_EN] = &audio_tdmin_lb_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_TDMIN_LB_SCLK] = &audio_tdmin_lb_sclk.hw, >>>>>>> + [AUD_CLKID_TDMIN_LB_LRCLK] = &audio_tdmin_lb_lrclk.hw, >>>>>>> + [AUD_CLKID_TDMOUT_A_SCLK_SEL] = &audio_tdmout_a_sclk_mux.hw, >>>>>>> + [AUD_CLKID_TDMOUT_A_SCLK_PRE_EN] = &audio_tdmout_a_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_TDMOUT_A_SCLK_POST_EN] = &audio_tdmout_a_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_TDMOUT_A_SCLK] = &audio_tdmout_a_sclk.hw, >>>>>>> + [AUD_CLKID_TDMOUT_A_LRCLK] = &audio_tdmout_a_lrclk.hw, >>>>>>> + [AUD_CLKID_TDMOUT_B_SCLK_SEL] = &audio_tdmout_b_sclk_mux.hw, >>>>>>> + [AUD_CLKID_TDMOUT_B_SCLK_PRE_EN] = &audio_tdmout_b_sclk_pre_en.hw, >>>>>>> + [AUD_CLKID_TDMOUT_B_SCLK_POST_EN] = &audio_tdmout_b_sclk_post_en.hw, >>>>>>> + [AUD_CLKID_TDMOUT_B_SCLK] = &audio_tdmout_b_sclk.hw, >>>>>>> + [AUD_CLKID_TDMOUT_B_LRCLK] = &audio_tdmout_b_lrclk.hw, >>>>>>> + >>>>>>> + [AUD2_CLKID_DDR_ARB] = &audio2_ddr_arb.hw, >>>>>>> + [AUD2_CLKID_PDM] = &audio2_pdm.hw, >>>>>>> + [AUD2_CLKID_TDMIN_VAD] = &audio2_tdmin_vad.hw, >>>>>>> + [AUD2_CLKID_TODDR_VAD] = &audio2_toddr_vad.hw, >>>>>>> + [AUD2_CLKID_VAD] = &audio2_vad.hw, >>>>>>> + [AUD2_CLKID_AUDIOTOP] = &audio2_audiotop.hw, >>>>>>> + [AUD2_CLKID_VAD_MCLK_SEL] = &audio2_vad_mclk_mux.hw, >>>>>>> + [AUD2_CLKID_VAD_MCLK_DIV] = &audio2_vad_mclk_div.hw, >>>>>>> + [AUD2_CLKID_VAD_MCLK] = &audio2_vad_mclk.hw, >>>>>>> + [AUD2_CLKID_VAD_CLK_SEL] = &audio2_vad_clk_mux.hw, >>>>>>> + [AUD2_CLKID_VAD_CLK_DIV] = &audio2_vad_clk_div.hw, >>>>>>> + [AUD2_CLKID_VAD_CLK] = &audio2_vad_clk.hw, >>>>>>> + [AUD2_CLKID_PDM_DCLK_SEL] = &audio2_pdm_dclk_mux.hw, >>>>>>> + [AUD2_CLKID_PDM_DCLK_DIV] = &audio2_pdm_dclk_div.hw, >>>>>>> + [AUD2_CLKID_PDM_DCLK] = &audio2_pdm_dclk.hw, >>>>>>> + [AUD2_CLKID_PDM_SYSCLK_SEL] = &audio2_pdm_sysclk_mux.hw, >>>>>>> + [AUD2_CLKID_PDM_SYSCLK_DIV] = &audio2_pdm_sysclk_div.hw, >>>>>>> + [AUD2_CLKID_PDM_SYSCLK] = &audio2_pdm_sysclk.hw, >>>>>>> +}; >>>>>>> + >>>>>>> +static struct meson_clk_hw_data a1_audio_clks = { >>>>>>> + .hws = a1_audio_hw_clks, >>>>>>> + .num = ARRAY_SIZE(a1_audio_hw_clks), >>>>>>> +}; >>>>>>> + >>>>>>> +static struct regmap *a1_audio_map(struct platform_device *pdev, >>>>>>> + unsigned int index) >>>>>>> +{ >>>>>>> + char name[32]; >>>>>>> + const struct regmap_config cfg = { >>>>>>> + .reg_bits = 32, >>>>>>> + .val_bits = 32, >>>>>>> + .reg_stride = 4, >>>>>>> + .name = name, >>>>>> >>>>>> Not necessary >>>>>> >>>>> >>>>> This implementation uses two regmaps, and this field allow to avoid >>>>> errors like this: >>>>> >>>>> [ 0.145530] debugfs: Directory 'fe050000.audio-clock-controller' with >>>>> parent 'regmap' already present! >>>>> >>>>>>> + }; >>>>>>> + void __iomem *base; >>>>>>> + >>>>>>> + base = devm_platform_ioremap_resource(pdev, index); >>>>>>> + if (IS_ERR(base)) >>>>>>> + return base; >>>>>>> + >>>>>>> + scnprintf(name, sizeof(name), "%d", index); >>>>>>> + return devm_regmap_init_mmio(&pdev->dev, base, &cfg); >>>>>>> +} >>>>>> >>>>>> That is overengineered. Please keep it simple. Declare the regmap_config >>>>>> as static const global, and do it like axg-audio please. >>>>>> >>>>> >>>>> This only reason why it is not "static const" because I need to set >>>>> unique name for each regmap. >>>>> >>>>>>> + >>>>>>> +static int a1_register_clk(struct platform_device *pdev, >>>>>>> + struct regmap *map0, struct regmap *map1, >>>>>>> + struct clk_hw *hw) >>>>>>> +{ >>>>>>> + struct clk_regmap *clk = container_of(hw, struct clk_regmap, hw); >>>>>>> + >>>>>>> + if (!hw) >>>>>>> + return 0; >>>>>>> + >>>>>>> + switch ((unsigned long)clk->map) { >>>>>>> + case AUDIO_RANGE_0: >>>>>>> + clk->map = map0; >>>>>>> + break; >>>>>>> + case AUDIO_RANGE_1: >>>>>>> + clk->map = map1; >>>>>>> + break; >>>>>> >>>>>> ... fishy >>>>>> >>>>>>> + default: >>>>>>> + WARN_ON(1); >>>>>>> + return -EINVAL; >>>>>>> + } >>>>>>> + >>>>>>> + return devm_clk_hw_register(&pdev->dev, hw); >>>>>>> +} >>>>>>> + >>>>>>> +static int a1_audio_clkc_probe(struct platform_device *pdev) >>>>>>> +{ >>>>>>> + struct regmap *map0, *map1; >>>>>>> + struct clk *clk; >>>>>>> + unsigned int i; >>>>>>> + int ret; >>>>>>> + >>>>>>> + clk = devm_clk_get_enabled(&pdev->dev, "pclk"); >>>>>>> + if (WARN_ON(IS_ERR(clk))) >>>>>>> + return PTR_ERR(clk); >>>>>>> + >>>>>>> + map0 = a1_audio_map(pdev, 0); >>>>>>> + if (IS_ERR(map0)) >>>>>>> + return PTR_ERR(map0); >>>>>>> + >>>>>>> + map1 = a1_audio_map(pdev, 1); >>>>>>> + if (IS_ERR(map1)) >>>>>>> + return PTR_ERR(map1); >>>>>> >>>>>> No - Looks to me you just have two clock controllers you are trying >>>>>> force into one. >>>>>> >>>>> >>>>> See the begining. >>>>> >>>>>>> + >>>>>>> + /* >>>>>>> + * Register and enable AUD2_CLKID_AUDIOTOP clock first. Unless >>>>>>> + * it is enabled any read/write to 'map0' hangs the CPU. >>>>>>> + */ >>>>>>> + >>>>>>> + ret = a1_register_clk(pdev, map0, map1, >>>>>>> + a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]); >>>>>>> + if (ret) >>>>>>> + return ret; >>>>>>> + >>>>>>> + ret = clk_prepare_enable(a1_audio_clks.hws[AUD2_CLKID_AUDIOTOP]->clk); >>>>>>> + if (ret) >>>>>>> + return ret; >>>>>> >>>>>> Again, this shows 2 devices. The one related to your 'map0' should >>>>>> request AUD2_CLKID_AUDIOTOP as input and enable it right away. >>>>>> >>>>> >>>>> See the begining. >>>>> >>>>>>> + >>>>>>> + for (i = 0; i < a1_audio_clks.num; i++) { >>>>>>> + if (i == AUD2_CLKID_AUDIOTOP) >>>>>>> + continue; >>>>>>> + >>>>>>> + ret = a1_register_clk(pdev, map0, map1, a1_audio_clks.hws[i]); >>>>>>> + if (ret) >>>>>>> + return ret; >>>>>>> + } >>>>>>> + >>>>>>> + ret = devm_of_clk_add_hw_provider(&pdev->dev, meson_clk_hw_get, >>>>>>> + &a1_audio_clks); >>>>>>> + if (ret) >>>>>>> + return ret; >>>>>>> + >>>>>>> + BUILD_BUG_ON((unsigned long)AUDIO_REG_MAP(AUDIO_SW_RESET0) != >>>>>>> + AUDIO_RANGE_0); >>>>>> >>>>>> Why is that necessary ? >>>>>> >>>>> >>>>> A little paranoia. Here AUDIO_SW_RESET0 is handled as map0's register, >>>>> and I want to assert it. >>>>> >>>>>>> + return meson_audio_rstc_register(&pdev->dev, map0, >>>>>>> + AUDIO_REG_OFFSET(AUDIO_SW_RESET0), 32); >>>>>>> +} >>>>>>> + >>>>>>> +static const struct of_device_id a1_audio_clkc_match_table[] = { >>>>>>> + { .compatible = "amlogic,a1-audio-clkc", }, >>>>>>> + {} >>>>>>> +}; >>>>>>> +MODULE_DEVICE_TABLE(of, a1_audio_clkc_match_table); >>>>>>> + >>>>>>> +static struct platform_driver a1_audio_clkc_driver = { >>>>>>> + .probe = a1_audio_clkc_probe, >>>>>>> + .driver = { >>>>>>> + .name = "a1-audio-clkc", >>>>>>> + .of_match_table = a1_audio_clkc_match_table, >>>>>>> + }, >>>>>>> +}; >>>>>>> +module_platform_driver(a1_audio_clkc_driver); >>>>>>> + >>>>>>> +MODULE_DESCRIPTION("Amlogic A1 Audio Clock driver"); >>>>>>> +MODULE_AUTHOR("Jan Dakinevich <jan.dakinevich@salutedevices.com>"); >>>>>>> +MODULE_LICENSE("GPL"); >>>>>>> diff --git a/drivers/clk/meson/a1-audio.h b/drivers/clk/meson/a1-audio.h >>>>>>> new file mode 100644 >>>>>>> index 000000000000..f994e87276cd >>>>>>> --- /dev/null >>>>>>> +++ b/drivers/clk/meson/a1-audio.h >>>>>>> @@ -0,0 +1,58 @@ >>>>>>> +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ >>>>>>> +/* >>>>>>> + * Copyright (c) 2024, SaluteDevices. All Rights Reserved. >>>>>>> + * >>>>>>> + * Author: Jan Dakinevich <jan.dakinevich@salutedevices.com> >>>>>>> + */ >>>>>>> + >>>>>>> +#ifndef __A1_AUDIO_H >>>>>>> +#define __A1_AUDIO_H >>>>>>> + >>>>>>> +#define AUDIO_RANGE_0 0xa >>>>>>> +#define AUDIO_RANGE_1 0xb >>>>>>> +#define AUDIO_RANGE_SHIFT 16 >>>>>>> + >>>>>>> +#define AUDIO_REG(_range, _offset) \ >>>>>>> + (((_range) << AUDIO_RANGE_SHIFT) + (_offset)) >>>>>>> + >>>>>>> +#define AUDIO_REG_OFFSET(_reg) \ >>>>>>> + ((_reg) & ((1 << AUDIO_RANGE_SHIFT) - 1)) >>>>>>> + >>>>>>> +#define AUDIO_REG_MAP(_reg) \ >>>>>>> + ((void *)((_reg) >> AUDIO_RANGE_SHIFT)) >>>>>> >>>>>> That is seriouly overengineered. >>>>>> The following are offset. Just write what they are. >>>>>> >>>>> >>>>> This is all in order to keep range's identifier together with offset and >>>>> then use it to store the identifier in clk_regmaps. >>>>> >>>>>> There is not reason to put that into a header. It is only going to be >>>>>> used by a single driver. >>>>>>>> + >>>>>>> +#define AUDIO_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_0, 0x000) >>>>>>> +#define AUDIO_MCLK_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x008) >>>>>>> +#define AUDIO_MCLK_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x00c) >>>>>>> +#define AUDIO_MCLK_C_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x010) >>>>>>> +#define AUDIO_MCLK_D_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x014) >>>>>>> +#define AUDIO_MCLK_E_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x018) >>>>>>> +#define AUDIO_MCLK_F_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x01c) >>>>>>> +#define AUDIO_SW_RESET0 AUDIO_REG(AUDIO_RANGE_0, 0x028) >>>>>>> +#define AUDIO_MST_A_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x040) >>>>>>> +#define AUDIO_MST_A_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x044) >>>>>>> +#define AUDIO_MST_B_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x048) >>>>>>> +#define AUDIO_MST_B_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x04c) >>>>>>> +#define AUDIO_MST_C_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x050) >>>>>>> +#define AUDIO_MST_C_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x054) >>>>>>> +#define AUDIO_MST_D_SCLK_CTRL0 AUDIO_REG(AUDIO_RANGE_0, 0x058) >>>>>>> +#define AUDIO_MST_D_SCLK_CTRL1 AUDIO_REG(AUDIO_RANGE_0, 0x05c) >>>>>>> +#define AUDIO_CLK_TDMIN_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x080) >>>>>>> +#define AUDIO_CLK_TDMIN_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x084) >>>>>>> +#define AUDIO_CLK_TDMIN_LB_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x08c) >>>>>>> +#define AUDIO_CLK_TDMOUT_A_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x090) >>>>>>> +#define AUDIO_CLK_TDMOUT_B_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x094) >>>>>>> +#define AUDIO_CLK_SPDIFIN_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x09c) >>>>>>> +#define AUDIO_CLK_RESAMPLE_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a4) >>>>>>> +#define AUDIO_CLK_LOCKER_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0a8) >>>>>>> +#define AUDIO_CLK_EQDRC_CTRL AUDIO_REG(AUDIO_RANGE_0, 0x0c0) >>>>>>> + >>>>>>> +#define AUDIO2_CLK_GATE_EN0 AUDIO_REG(AUDIO_RANGE_1, 0x00c) >>>>>>> +#define AUDIO2_MCLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x040) >>>>>>> +#define AUDIO2_CLK_VAD_CTRL AUDIO_REG(AUDIO_RANGE_1, 0x044) >>>>>>> +#define AUDIO2_CLK_PDMIN_CTRL0 AUDIO_REG(AUDIO_RANGE_1, 0x058) >>>>>>> +#define AUDIO2_CLK_PDMIN_CTRL1 AUDIO_REG(AUDIO_RANGE_1, 0x05c) >>>>>>> + >>>>>>> +#include <dt-bindings/clock/amlogic,a1-audio-clkc.h> >>>>>>> + >>>>>>> +#endif /* __A1_AUDIO_H */ >>>>>> >>>>>> >>>> >>>> >> >>
This series includes the following: - new audio clock and reset controller data and adaptation for it of existing code (patches 0001..0004); - adaptation of existing audio components for A1 Soc (patches 0005..0021); - handy cosmetics for dai-link naming (patches 0022..0023); - integration of audio devices into common trees (patch 0024); - audio support bring up on Amlogic ad402 reference board (patch 0025). This patch is not actually checked on real hardware (because all ad402 that we had were burned out). This patch is based on ad402's schematics and on experience with our own hardware (which is very close to reference board); Dmitry Rokosov (2): ASoC: dt-bindings: meson: introduce link-name optional property ASoC: meson: implement link-name optional property in meson card utils Jan Dakinevich (23): clk: meson: a1: restrict an amount of 'hifi_pll' params clk: meson: axg: move reset controller's code to separate module dt-bindings: clock: meson: add A1 audio clock and reset controller bindings clk: meson: a1: add the audio clock controller driver ASoC: meson: codec-glue: add support for capture stream ASoC: meson: g12a-toacodec: fix "Lane Select" width ASoC: meson: g12a-toacodec: rework the definition of bits ASoC: dt-bindings: meson: g12a-toacodec: add support for A1 SoC family ASoC: meson: g12a-toacodec: add support for A1 SoC family ASoC: meson: t9015: prepare to adding new platforms ASoC: dt-bindings: meson: t9015: add support for A1 SoC family ASoC: meson: t9015: add support for A1 SoC family ASoC: dt-bindings: meson: axg-pdm: document 'sysrate' property ASoC: meson: axg-pdm: introduce 'sysrate' property pinctrl/meson: fix typo in PDM's pin name ASoC: dt-bindings: meson: meson-axg-audio-arb: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-fifo: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-pdm: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-sound-card: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-tdm-formatters: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-tdm-iface: claim support of A1 SoC family arm64: dts: meson: a1: add audio devices arm64: dts: ad402: enable audio .../bindings/clock/amlogic,a1-audio-clkc.yaml | 83 +++ .../reset/amlogic,meson-axg-audio-arb.yaml | 10 +- .../bindings/sound/amlogic,axg-fifo.yaml | 8 + .../bindings/sound/amlogic,axg-pdm.yaml | 5 + .../sound/amlogic,axg-sound-card.yaml | 12 +- .../sound/amlogic,axg-tdm-formatters.yaml | 22 +- .../bindings/sound/amlogic,axg-tdm-iface.yaml | 6 +- .../bindings/sound/amlogic,g12a-toacodec.yaml | 1 + .../bindings/sound/amlogic,gx-sound-card.yaml | 6 + .../bindings/sound/amlogic,t9015.yaml | 4 +- .../arm64/boot/dts/amlogic/meson-a1-ad402.dts | 126 ++++ arch/arm64/boot/dts/amlogic/meson-a1.dtsi | 471 +++++++++++++++ drivers/clk/meson/Kconfig | 18 + drivers/clk/meson/Makefile | 2 + drivers/clk/meson/a1-audio.c | 556 ++++++++++++++++++ drivers/clk/meson/a1-audio.h | 58 ++ drivers/clk/meson/a1-pll.c | 8 +- drivers/clk/meson/axg-audio.c | 95 +-- drivers/clk/meson/meson-audio-rstc.c | 109 ++++ drivers/clk/meson/meson-audio-rstc.h | 12 + drivers/pinctrl/meson/pinctrl-meson-a1.c | 6 +- .../dt-bindings/clock/amlogic,a1-audio-clkc.h | 122 ++++ .../reset/amlogic,meson-a1-audio-reset.h | 29 + .../dt-bindings/sound/meson-g12a-toacodec.h | 5 + sound/soc/meson/axg-pdm.c | 10 +- sound/soc/meson/g12a-toacodec.c | 298 ++++++++-- sound/soc/meson/meson-card-utils.c | 12 +- sound/soc/meson/meson-codec-glue.c | 174 ++++-- sound/soc/meson/meson-codec-glue.h | 23 + sound/soc/meson/t9015.c | 326 +++++++++- 30 files changed, 2394 insertions(+), 223 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/amlogic,a1-audio-clkc.yaml create mode 100644 drivers/clk/meson/a1-audio.c create mode 100644 drivers/clk/meson/a1-audio.h create mode 100644 drivers/clk/meson/meson-audio-rstc.c create mode 100644 drivers/clk/meson/meson-audio-rstc.h create mode 100644 include/dt-bindings/clock/amlogic,a1-audio-clkc.h create mode 100644 include/dt-bindings/reset/amlogic,meson-a1-audio-reset.h