[alsa-devel] [PATCH resend v5 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
Hi,
This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface.
This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html
This patchset has been tested on Cygnus wireless audio bcm958305K board. It is rebased on top of v4.6-rc1 and is available from github:
repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v5
Changes from v4: - Fix power suspend function and add power resume function - Remove clock initialization code from audio driver to clock framework Changes from v3: - Fix the subject lines to match the style for the subsystem Changes from v2: - Split patchset 2/2 from v2 into patchsets 2/3 and 3/3. - Remove SND_SOC_CYGNUS_DIAG. Diagnostics can be performed using standard kernel trace infrastructure. - Fix interrupt handler. Acknowledge only those interrupts that are handledby ISR. - Modify configure_vco() and the pll_macro_entry() struct to make it better readable. The functionality did not change. - Remove casts on macros - Removed surround sound channel grouping from the driver. Changes from v1: - Address code review comments. Fixed print format of type size_t and pointer.
Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver
.../bindings/sound/brcm,cygnus-audio.txt | 67 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1513 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++ 6 files changed, 2594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h
From: Simran Rai ssimran@broadcom.com
Add bindings for audio driver in Broadcom Cygnus.
Signed-off-by: Lori Hikichi lhikichi@broadcom.com Signed-off-by: Simran Rai ssimran@broadcom.com Reviewed-by: Ray Jui rjui@broadcom.com Reviewed-by: Scott Branden sbranden@broadcom.com Signed-off-by: Simran Rai simran.rai@broadcom.com --- .../bindings/sound/brcm,cygnus-audio.txt | 67 ++++++++++++++++++++ 1 file changed, 67 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
diff --git a/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt new file mode 100644 index 0000000..b139e66 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt @@ -0,0 +1,67 @@ +BROADCOM Cygnus Audio I2S/TDM/SPDIF controller + +Required properties: + - compatible : "brcm,cygnus-audio" + - #address-cells: 32bit valued, 1 cell. + - #size-cells: 32bit valued, 0 cell. + - reg : Should contain audio registers location and length + - reg-names: names of the registers listed in "reg" property + Valid names are "aud" and "i2s_in". "aud" contains a + set of DMA, I2S_OUT and SPDIF registers. "i2s_in" contains + a set of I2S_IN registers. + - clocks: PLL and leaf clocks used by audio ports + - assigned-clocks: PLL and leaf clocks + - assigned-clock-parents: parent clocks of the assigned clocks + (usually the PLL) + - assigned-clock-rates: List of clock frequencies of the + assigned clocks + - clock-names: names of 3 leaf clocks used by audio ports + Valid names are "ch0_audio", "ch1_audio", "ch2_audio" + - interrupts: audio DMA interrupt number + +SSP Subnode properties: +- reg: The index of ssp port interface to use + Valid value are 0, 1, 2, or 3 (for spdif) + +Example: + cygnus_audio: audio@180ae000 { + compatible = "brcm,cygnus-audio"; + #address-cells = <1>; + #size-cells = <0>; + reg = <0x180ae000 0xafd>, <0x180aec00 0x1f8>; + reg-names = "aud", "i2s_in"; + clocks = <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clocks = <&audiopll BCM_CYGNUS_AUDIOPLL>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clock-parents = <&audiopll BCM_CYGNUS_AUDIOPLL>; + assigned-clock-rates = <1769470191>, + <0>, + <0>, + <0>; + clock-names = "ch0_audio", "ch1_audio", "ch2_audio"; + interrupts = <GIC_SPI 143 IRQ_TYPE_LEVEL_HIGH>; + + ssp0: ssp_port@0 { + reg = <0>; + status = "okay"; + }; + + ssp1: ssp_port@1 { + reg = <1>; + status = "disabled"; + }; + + ssp2: ssp_port@2 { + reg = <2>; + status = "disabled"; + }; + + spdif: spdif_port@3 { + reg = <3>; + status = "disabled"; + }; + };
On Tue, Mar 29, 2016 at 11:46:30AM -0700, Simran Rai wrote:
From: Simran Rai ssimran@broadcom.com
Add bindings for audio driver in Broadcom Cygnus.
Signed-off-by: Lori Hikichi lhikichi@broadcom.com Signed-off-by: Simran Rai ssimran@broadcom.com Reviewed-by: Ray Jui rjui@broadcom.com Reviewed-by: Scott Branden sbranden@broadcom.com Signed-off-by: Simran Rai simran.rai@broadcom.com
.../bindings/sound/brcm,cygnus-audio.txt | 67 ++++++++++++++++++++ 1 file changed, 67 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
I acked v3 of this already. Did something change? If not, please add acks when sending new versions. If things did change, explain that here.
And this v5 is not a resend if you didn't already send out v5 before.
Hi Rob,
On Thu, Mar 31, 2016 at 7:13 AM, Rob Herring robh@kernel.org wrote:
On Tue, Mar 29, 2016 at 11:46:30AM -0700, Simran Rai wrote:
From: Simran Rai ssimran@broadcom.com
Add bindings for audio driver in Broadcom Cygnus.
Signed-off-by: Lori Hikichi lhikichi@broadcom.com Signed-off-by: Simran Rai ssimran@broadcom.com Reviewed-by: Ray Jui rjui@broadcom.com Reviewed-by: Scott Branden sbranden@broadcom.com Signed-off-by: Simran Rai simran.rai@broadcom.com
.../bindings/sound/brcm,cygnus-audio.txt | 67 ++++++++++++++++++++ 1 file changed, 67 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
I acked v3 of this already. Did something change? If not, please add acks when sending new versions. If things did change, explain that here.
There was a change made in v4 that introduced assigned-clocks, assigned-clock-parents, and assigned-clock-rates, to allow initial configuration of default parent clock and its frequency.
And this v5 is not a resend if you didn't already send out v5 before.
I had sent out v5 on March 01 but looks like it did not make it to through. Yesterday I sent out another patch (v6) to address Mark's comments. Can you please ack PATCH v6.
Thanks, Simran
On Thu, Mar 31, 2016 at 11:14:53AM -0700, Simran Rai wrote:
There was a change made in v4 that introduced assigned-clocks, assigned-clock-parents, and assigned-clock-rates, to allow initial configuration of default parent clock and its frequency.
...all of which are standard clock binding properties.
The patch
ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio
has been applied to the asoc tree at
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying to this mail.
Thanks, Mark
From 36e5ecc2986f4712d8fdfc05ed1e5d39dda7096d Mon Sep 17 00:00:00 2001
From: Simran Rai ssimran@broadcom.com Date: Tue, 17 May 2016 17:01:07 -0700 Subject: [PATCH] ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio
Add bindings for audio driver in Broadcom Cygnus.
Signed-off-by: Lori Hikichi lhikichi@broadcom.com Signed-off-by: Simran Rai ssimran@broadcom.com Reviewed-by: Ray Jui rjui@broadcom.com Reviewed-by: Scott Branden sbranden@broadcom.com Acked-by: Rob Herring robh@kernel.org Signed-off-by: Mark Brown broonie@kernel.org --- .../bindings/sound/brcm,cygnus-audio.txt | 67 ++++++++++++++++++++++ 1 file changed, 67 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
diff --git a/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt new file mode 100644 index 000000000000..b139e66d2a11 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt @@ -0,0 +1,67 @@ +BROADCOM Cygnus Audio I2S/TDM/SPDIF controller + +Required properties: + - compatible : "brcm,cygnus-audio" + - #address-cells: 32bit valued, 1 cell. + - #size-cells: 32bit valued, 0 cell. + - reg : Should contain audio registers location and length + - reg-names: names of the registers listed in "reg" property + Valid names are "aud" and "i2s_in". "aud" contains a + set of DMA, I2S_OUT and SPDIF registers. "i2s_in" contains + a set of I2S_IN registers. + - clocks: PLL and leaf clocks used by audio ports + - assigned-clocks: PLL and leaf clocks + - assigned-clock-parents: parent clocks of the assigned clocks + (usually the PLL) + - assigned-clock-rates: List of clock frequencies of the + assigned clocks + - clock-names: names of 3 leaf clocks used by audio ports + Valid names are "ch0_audio", "ch1_audio", "ch2_audio" + - interrupts: audio DMA interrupt number + +SSP Subnode properties: +- reg: The index of ssp port interface to use + Valid value are 0, 1, 2, or 3 (for spdif) + +Example: + cygnus_audio: audio@180ae000 { + compatible = "brcm,cygnus-audio"; + #address-cells = <1>; + #size-cells = <0>; + reg = <0x180ae000 0xafd>, <0x180aec00 0x1f8>; + reg-names = "aud", "i2s_in"; + clocks = <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clocks = <&audiopll BCM_CYGNUS_AUDIOPLL>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clock-parents = <&audiopll BCM_CYGNUS_AUDIOPLL>; + assigned-clock-rates = <1769470191>, + <0>, + <0>, + <0>; + clock-names = "ch0_audio", "ch1_audio", "ch2_audio"; + interrupts = <GIC_SPI 143 IRQ_TYPE_LEVEL_HIGH>; + + ssp0: ssp_port@0 { + reg = <0>; + status = "okay"; + }; + + ssp1: ssp_port@1 { + reg = <1>; + status = "disabled"; + }; + + ssp2: ssp_port@2 { + reg = <2>; + status = "disabled"; + }; + + spdif: spdif_port@3 { + reg = <3>; + status = "disabled"; + }; + };
From: Simran Rai ssimran@broadcom.com
This patch adds Cygnus audio DAI driver. It supports I2S, TDM and SPDIF modes and uses three clocks derived from PLL.
This patchset has been tested on Cygnus wireless audio bcm958305K board.
Signed-off-by: Lori Hikichi lhikichi@broadcom.com Signed-off-by: Simran Rai ssimran@broadcom.com Reviewed-by: Ray Jui rjui@broadcom.com Reviewed-by: Arun Parameswaran arunp@broadcom.com Reviewed-by: Scott Branden sbranden@broadcom.com Signed-off-by: Simran Rai simran.rai@broadcom.com --- sound/soc/bcm/cygnus-ssp.c | 1513 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++++ 2 files changed, 1652 insertions(+) create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c new file mode 100644 index 0000000..42d2543 --- /dev/null +++ b/sound/soc/bcm/cygnus-ssp.c @@ -0,0 +1,1513 @@ +/* + * Copyright (C) 2014-2015 Broadcom Corporation + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; without even the implied warranty + * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "cygnus-ssp.h" + +#define DEFAULT_VCO 1354750204 + +#define CYGNUS_TDM_RATE \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define CAPTURE_FCI_ID_BASE 0x180 +#define CYGNUS_SSP_TRISTATE_MASK 0x001fff +#define CYGNUS_PLLCLKSEL_MASK 0xf + +/* Used with stream_on field to indicate which streams are active */ +#define PLAYBACK_STREAM_MASK BIT(0) +#define CAPTURE_STREAM_MASK BIT(1) + +#define I2S_STREAM_CFG_MASK 0xff003ff +#define I2S_CAP_STREAM_CFG_MASK 0xf0 +#define SPDIF_STREAM_CFG_MASK 0x3ff +#define CH_GRP_STEREO 0x1 + +/* Begin register offset defines */ +#define AUD_MISC_SEROUT_OE_REG_BASE 0x01c +#define AUD_MISC_SEROUT_SPDIF_OE 13 +#define AUD_MISC_SEROUT_MCLK_OE 3 +#define AUD_MISC_SEROUT_LRCK_OE 2 +#define AUD_MISC_SEROUT_SCLK_OE 1 +#define AUD_MISC_SEROUT_SDAT_OE 0 + +/* AUD_FMM_BF_CTRL_xxx regs */ +#define BF_DST_CFG0_OFFSET 0x100 +#define BF_DST_CFG1_OFFSET 0x104 +#define BF_DST_CFG2_OFFSET 0x108 + +#define BF_DST_CTRL0_OFFSET 0x130 +#define BF_DST_CTRL1_OFFSET 0x134 +#define BF_DST_CTRL2_OFFSET 0x138 + +#define BF_SRC_CFG0_OFFSET 0x148 +#define BF_SRC_CFG1_OFFSET 0x14c +#define BF_SRC_CFG2_OFFSET 0x150 +#define BF_SRC_CFG3_OFFSET 0x154 + +#define BF_SRC_CTRL0_OFFSET 0x1c0 +#define BF_SRC_CTRL1_OFFSET 0x1c4 +#define BF_SRC_CTRL2_OFFSET 0x1c8 +#define BF_SRC_CTRL3_OFFSET 0x1cc + +#define BF_SRC_GRP0_OFFSET 0x1fc +#define BF_SRC_GRP1_OFFSET 0x200 +#define BF_SRC_GRP2_OFFSET 0x204 +#define BF_SRC_GRP3_OFFSET 0x208 + +#define BF_SRC_GRP_EN_OFFSET 0x320 +#define BF_SRC_GRP_FLOWON_OFFSET 0x324 +#define BF_SRC_GRP_SYNC_DIS_OFFSET 0x328 + +/* AUD_FMM_IOP_OUT_I2S_xxx regs */ +#define OUT_I2S_0_STREAM_CFG_OFFSET 0xa00 +#define OUT_I2S_0_CFG_OFFSET 0xa04 +#define OUT_I2S_0_MCLK_CFG_OFFSET 0xa0c + +#define OUT_I2S_1_STREAM_CFG_OFFSET 0xa40 +#define OUT_I2S_1_CFG_OFFSET 0xa44 +#define OUT_I2S_1_MCLK_CFG_OFFSET 0xa4c + +#define OUT_I2S_2_STREAM_CFG_OFFSET 0xa80 +#define OUT_I2S_2_CFG_OFFSET 0xa84 +#define OUT_I2S_2_MCLK_CFG_OFFSET 0xa8c + +/* AUD_FMM_IOP_OUT_SPDIF_xxx regs */ +#define SPDIF_STREAM_CFG_OFFSET 0xac0 +#define SPDIF_CTRL_OFFSET 0xac4 +#define SPDIF_FORMAT_CFG_OFFSET 0xad8 +#define SPDIF_MCLK_CFG_OFFSET 0xadc + +/* AUD_FMM_IOP_PLL_0_xxx regs */ +#define IOP_PLL_0_MACRO_OFFSET 0xb00 +#define IOP_PLL_0_MDIV_Ch0_OFFSET 0xb14 +#define IOP_PLL_0_MDIV_Ch1_OFFSET 0xb18 +#define IOP_PLL_0_MDIV_Ch2_OFFSET 0xb1c + +#define IOP_PLL_0_ACTIVE_MDIV_Ch0_OFFSET 0xb30 +#define IOP_PLL_0_ACTIVE_MDIV_Ch1_OFFSET 0xb34 +#define IOP_PLL_0_ACTIVE_MDIV_Ch2_OFFSET 0xb38 + +/* AUD_FMM_IOP_xxx regs */ +#define IOP_PLL_0_CONTROL_OFFSET 0xb04 +#define IOP_PLL_0_USER_NDIV_OFFSET 0xb08 +#define IOP_PLL_0_ACTIVE_NDIV_OFFSET 0xb20 +#define IOP_PLL_0_RESET_OFFSET 0xb5c + +/* AUD_FMM_IOP_IN_I2S_xxx regs */ +#define IN_I2S_0_STREAM_CFG_OFFSET 0x00 +#define IN_I2S_0_CFG_OFFSET 0x04 +#define IN_I2S_1_STREAM_CFG_OFFSET 0x40 +#define IN_I2S_1_CFG_OFFSET 0x44 +#define IN_I2S_2_STREAM_CFG_OFFSET 0x80 +#define IN_I2S_2_CFG_OFFSET 0x84 + +/* AUD_FMM_IOP_MISC_xxx regs */ +#define IOP_SW_INIT_LOGIC 0x1c0 + +/* End register offset defines */ + + +/* AUD_FMM_IOP_OUT_I2S_x_MCLK_CFG_0_REG */ +#define I2S_OUT_MCLKRATE_SHIFT 16 + +/* AUD_FMM_IOP_OUT_I2S_x_MCLK_CFG_REG */ +#define I2S_OUT_PLLCLKSEL_SHIFT 0 + +/* AUD_FMM_IOP_OUT_I2S_x_STREAM_CFG */ +#define I2S_OUT_STREAM_ENA 31 +#define I2S_OUT_STREAM_CFG_GROUP_ID 20 +#define I2S_OUT_STREAM_CFG_CHANNEL_GROUPING 24 + +/* AUD_FMM_IOP_IN_I2S_x_CAP */ +#define I2S_IN_STREAM_CFG_CAP_ENA 31 +#define I2S_IN_STREAM_CFG_0_GROUP_ID 4 + +/* AUD_FMM_IOP_OUT_I2S_x_I2S_CFG_REG */ +#define I2S_OUT_CFGX_CLK_ENA 0 +#define I2S_OUT_CFGX_DATA_ENABLE 1 +#define I2S_OUT_CFGX_DATA_ALIGNMENT 6 +#define I2S_OUT_CFGX_BITS_PER_SLOT 13 +#define I2S_OUT_CFGX_VALID_SLOT 14 +#define I2S_OUT_CFGX_FSYNC_WIDTH 18 +#define I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32 26 +#define I2S_OUT_CFGX_SLAVE_MODE 30 +#define I2S_OUT_CFGX_TDM_MODE 31 + +/* AUD_FMM_BF_CTRL_SOURCECH_CFGx_REG */ +#define BF_SRC_CFGX_SFIFO_ENA 0 +#define BF_SRC_CFGX_BUFFER_PAIR_ENABLE 1 +#define BF_SRC_CFGX_SAMPLE_CH_MODE 2 +#define BF_SRC_CFGX_SFIFO_SZ_DOUBLE 5 +#define BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY 10 +#define BF_SRC_CFGX_BIT_RES 20 +#define BF_SRC_CFGX_PROCESS_SEQ_ID_VALID 31 + +/* AUD_FMM_BF_CTRL_DESTCH_CFGx_REG */ +#define BF_DST_CFGX_CAP_ENA 0 +#define BF_DST_CFGX_BUFFER_PAIR_ENABLE 1 +#define BF_DST_CFGX_DFIFO_SZ_DOUBLE 2 +#define BF_DST_CFGX_NOT_PAUSE_WHEN_FULL 11 +#define BF_DST_CFGX_FCI_ID 12 +#define BF_DST_CFGX_CAP_MODE 24 +#define BF_DST_CFGX_PROC_SEQ_ID_VALID 31 + +/* AUD_FMM_IOP_OUT_SPDIF_xxx */ +#define SPDIF_0_OUT_DITHER_ENA 3 +#define SPDIF_0_OUT_STREAM_ENA 31 + +/* AUD_FMM_IOP_PLL_0_USER */ +#define IOP_PLL_0_USER_NDIV_FRAC 10 + +/* AUD_FMM_IOP_PLL_0_ACTIVE */ +#define IOP_PLL_0_ACTIVE_NDIV_FRAC 10 + + +#define INIT_SSP_REGS(num) (struct cygnus_ssp_regs){ \ + .i2s_stream_cfg = OUT_I2S_ ##num## _STREAM_CFG_OFFSET, \ + .i2s_cap_stream_cfg = IN_I2S_ ##num## _STREAM_CFG_OFFSET, \ + .i2s_cfg = OUT_I2S_ ##num## _CFG_OFFSET, \ + .i2s_cap_cfg = IN_I2S_ ##num## _CFG_OFFSET, \ + .i2s_mclk_cfg = OUT_I2S_ ##num## _MCLK_CFG_OFFSET, \ + .bf_destch_ctrl = BF_DST_CTRL ##num## _OFFSET, \ + .bf_destch_cfg = BF_DST_CFG ##num## _OFFSET, \ + .bf_sourcech_ctrl = BF_SRC_CTRL ##num## _OFFSET, \ + .bf_sourcech_cfg = BF_SRC_CFG ##num## _OFFSET, \ + .bf_sourcech_grp = BF_SRC_GRP ##num## _OFFSET \ +} + +struct pll_macro_entry { + u32 mclk; + u32 pll_ch_num; +}; + +/* + * PLL has 3 output channels (1x, 2x, and 4x). Below are + * the common MCLK frequencies used by audio driver + */ +static const struct pll_macro_entry pll_predef_mclk[] = { + { 4096000, 0}, + { 8192000, 1}, + {16384000, 2}, + + { 5644800, 0}, + {11289600, 1}, + {22579200, 2}, + + { 6144000, 0}, + {12288000, 1}, + {24576000, 2}, + + {12288000, 0}, + {24576000, 1}, + {49152000, 2}, + + {22579200, 0}, + {45158400, 1}, + {90316800, 2}, + + {24576000, 0}, + {49152000, 1}, + {98304000, 2}, +}; + +/* List of valid frame sizes for tdm mode */ +static const int ssp_valid_tdm_framesize[] = {32, 64, 128, 256, 512}; + +/* + * Use this relationship to derive the sampling rate (lrclk) + * lrclk = (mclk) / ((2*mclk_to_sclk_ratio) * (32 * SCLK))). + * + * Use mclk and pll_ch from the table above + * + * Valid SCLK = 0/1/2/4/8/12 + * + * mclk_to_sclk_ratio = number of MCLK per SCLK. Division is twice the + * value programmed in this field. + * Valid mclk_to_sclk_ratio = 1 through to 15 + * + * eg: To set lrclk = 48khz, set mclk = 12288000, mclk_to_sclk_ratio = 2, + * SCLK = 64 + */ +struct _ssp_clk_coeff { + u32 mclk; + u32 sclk_rate; + u32 rate; + u32 mclk_rate; +}; + +static const struct _ssp_clk_coeff ssp_clk_coeff[] = { + { 4096000, 32, 16000, 4}, + { 4096000, 32, 32000, 2}, + { 4096000, 64, 8000, 4}, + { 4096000, 64, 16000, 2}, + { 4096000, 64, 32000, 1}, + { 4096000, 128, 8000, 2}, + { 4096000, 128, 16000, 1}, + { 4096000, 256, 8000, 1}, + + { 6144000, 32, 16000, 6}, + { 6144000, 32, 32000, 3}, + { 6144000, 32, 48000, 2}, + { 6144000, 32, 96000, 1}, + { 6144000, 64, 8000, 6}, + { 6144000, 64, 16000, 3}, + { 6144000, 64, 48000, 1}, + { 6144000, 128, 8000, 3}, + + { 8192000, 32, 32000, 4}, + { 8192000, 64, 16000, 4}, + { 8192000, 64, 32000, 2}, + { 8192000, 128, 8000, 4}, + { 8192000, 128, 16000, 2}, + { 8192000, 128, 32000, 1}, + { 8192000, 256, 8000, 2}, + { 8192000, 256, 16000, 1}, + { 8192000, 512, 8000, 1}, + + {12288000, 32, 32000, 6}, + {12288000, 32, 48000, 4}, + {12288000, 32, 96000, 2}, + {12288000, 32, 192000, 1}, + {12288000, 64, 16000, 6}, + {12288000, 64, 32000, 3}, + {12288000, 64, 48000, 2}, + {12288000, 64, 96000, 1}, + {12288000, 128, 8000, 6}, + {12288000, 128, 16000, 3}, + {12288000, 128, 48000, 1}, + {12288000, 256, 8000, 3}, + + {16384000, 64, 32000, 4}, + {16384000, 128, 16000, 4}, + {16384000, 128, 32000, 2}, + {16384000, 256, 8000, 4}, + {16384000, 256, 16000, 2}, + {16384000, 256, 32000, 1}, + {16384000, 512, 8000, 2}, + {16384000, 512, 16000, 1}, + + {24576000, 32, 96000, 4}, + {24576000, 32, 192000, 2}, + {24576000, 64, 32000, 6}, + {24576000, 64, 48000, 4}, + {24576000, 64, 96000, 2}, + {24576000, 64, 192000, 1}, + {24576000, 128, 16000, 6}, + {24576000, 128, 32000, 3}, + {24576000, 128, 48000, 2}, + {24576000, 256, 8000, 6}, + {24576000, 256, 16000, 3}, + {24576000, 256, 48000, 1}, + {24576000, 512, 8000, 3}, + + {49152000, 32, 192000, 4}, + {49152000, 64, 96000, 4}, + {49152000, 64, 192000, 2}, + {49152000, 128, 32000, 6}, + {49152000, 128, 48000, 4}, + {49152000, 128, 96000, 2}, + {49152000, 128, 192000, 1}, + {49152000, 256, 16000, 6}, + {49152000, 256, 32000, 3}, + {49152000, 256, 48000, 2}, + {49152000, 256, 96000, 1}, + {49152000, 512, 8000, 6}, + {49152000, 512, 16000, 3}, + {49152000, 512, 48000, 1}, + + { 5644800, 32, 22050, 4}, + { 5644800, 32, 44100, 2}, + { 5644800, 32, 88200, 1}, + { 5644800, 64, 11025, 4}, + { 5644800, 64, 22050, 2}, + { 5644800, 64, 44100, 1}, + + {11289600, 32, 44100, 4}, + {11289600, 32, 88200, 2}, + {11289600, 32, 176400, 1}, + {11289600, 64, 22050, 4}, + {11289600, 64, 44100, 2}, + {11289600, 64, 88200, 1}, + {11289600, 128, 11025, 4}, + {11289600, 128, 22050, 2}, + {11289600, 128, 44100, 1}, + + {22579200, 32, 88200, 4}, + {22579200, 32, 176400, 2}, + {22579200, 64, 44100, 4}, + {22579200, 64, 88200, 2}, + {22579200, 64, 176400, 1}, + {22579200, 128, 22050, 4}, + {22579200, 128, 44100, 2}, + {22579200, 128, 88200, 1}, + {22579200, 256, 11025, 4}, + {22579200, 256, 22050, 2}, + {22579200, 256, 44100, 1}, + + {45158400, 32, 176400, 4}, + {45158400, 64, 88200, 4}, + {45158400, 64, 176400, 2}, + {45158400, 128, 44100, 4}, + {45158400, 128, 88200, 2}, + {45158400, 128, 176400, 1}, + {45158400, 256, 22050, 4}, + {45158400, 256, 44100, 2}, + {45158400, 256, 88200, 1}, + {45158400, 512, 11025, 4}, + {45158400, 512, 22050, 2}, + {45158400, 512, 44100, 1}, +}; + +static struct cygnus_aio_port *cygnus_dai_get_portinfo(struct snd_soc_dai *dai) +{ + struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai); + + return &cygaud->portinfo[dai->id]; +} + +static int audio_ssp_init_portregs(struct cygnus_aio_port *aio) +{ + u32 value, fci_id; + int status = 0; + + switch (aio->port_type) { + case PORT_TDM: + value = readl(aio->cygaud->audio + aio->regs.i2s_stream_cfg); + value &= ~I2S_STREAM_CFG_MASK; + + /* Set Group ID */ + writel(aio->portnum, + aio->cygaud->audio + aio->regs.bf_sourcech_grp); + + /* Configure the AUD_FMM_IOP_OUT_I2S_x_STREAM_CFG reg */ + value |= aio->portnum << I2S_OUT_STREAM_CFG_GROUP_ID; + value |= aio->portnum; /* FCI ID is the port num */ + value |= CH_GRP_STEREO << I2S_OUT_STREAM_CFG_CHANNEL_GROUPING; + writel(value, aio->cygaud->audio + aio->regs.i2s_stream_cfg); + + /* Configure the AUD_FMM_BF_CTRL_SOURCECH_CFGX reg */ + value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + value &= ~BIT(BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY); + value |= BIT(BF_SRC_CFGX_SFIFO_SZ_DOUBLE); + value |= BIT(BF_SRC_CFGX_PROCESS_SEQ_ID_VALID); + writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + + /* Configure the AUD_FMM_IOP_IN_I2S_x_CAP_STREAM_CFG_0 reg */ + value = readl(aio->cygaud->i2s_in + + aio->regs.i2s_cap_stream_cfg); + value &= ~I2S_CAP_STREAM_CFG_MASK; + value |= aio->portnum << I2S_IN_STREAM_CFG_0_GROUP_ID; + writel(value, aio->cygaud->i2s_in + + aio->regs.i2s_cap_stream_cfg); + + /* Configure the AUD_FMM_BF_CTRL_DESTCH_CFGX_REG_BASE reg */ + fci_id = CAPTURE_FCI_ID_BASE + aio->portnum; + + value = readl(aio->cygaud->audio + aio->regs.bf_destch_cfg); + value |= BIT(BF_DST_CFGX_DFIFO_SZ_DOUBLE); + value &= ~BIT(BF_DST_CFGX_NOT_PAUSE_WHEN_FULL); + value |= (fci_id << BF_DST_CFGX_FCI_ID); + value |= BIT(BF_DST_CFGX_PROC_SEQ_ID_VALID); + writel(value, aio->cygaud->audio + aio->regs.bf_destch_cfg); + + /* Enable the transmit pin for this port */ + value = readl(aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE); + value &= ~BIT((aio->portnum * 4) + AUD_MISC_SEROUT_SDAT_OE); + writel(value, aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE); + break; + case PORT_SPDIF: + writel(aio->portnum, aio->cygaud->audio + BF_SRC_GRP3_OFFSET); + + value = readl(aio->cygaud->audio + SPDIF_CTRL_OFFSET); + value |= BIT(SPDIF_0_OUT_DITHER_ENA); + writel(value, aio->cygaud->audio + SPDIF_CTRL_OFFSET); + + /* Enable and set the FCI ID for the SPDIF channel */ + value = readl(aio->cygaud->audio + SPDIF_STREAM_CFG_OFFSET); + value &= ~SPDIF_STREAM_CFG_MASK; + value |= aio->portnum; /* FCI ID is the port num */ + value |= BIT(SPDIF_0_OUT_STREAM_ENA); + writel(value, aio->cygaud->audio + SPDIF_STREAM_CFG_OFFSET); + + value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + value &= ~BIT(BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY); + value |= BIT(BF_SRC_CFGX_SFIFO_SZ_DOUBLE); + value |= BIT(BF_SRC_CFGX_PROCESS_SEQ_ID_VALID); + writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + + /* Enable the spdif output pin */ + value = readl(aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE); + value &= ~BIT(AUD_MISC_SEROUT_SPDIF_OE); + writel(value, aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE); + break; + default: + dev_err(aio->cygaud->dev, "Port not supported\n"); + status = -EINVAL; + } + + return status; +} + +static void audio_ssp_in_enable(struct cygnus_aio_port *aio) +{ + u32 value; + + value = readl(aio->cygaud->audio + aio->regs.bf_destch_cfg); + value |= BIT(BF_DST_CFGX_CAP_ENA); + writel(value, aio->cygaud->audio + aio->regs.bf_destch_cfg); + + writel(0x1, aio->cygaud->audio + aio->regs.bf_destch_ctrl); + + value = readl(aio->cygaud->audio + aio->regs.i2s_cfg); + value |= BIT(I2S_OUT_CFGX_CLK_ENA); + value |= BIT(I2S_OUT_CFGX_DATA_ENABLE); + writel(value, aio->cygaud->audio + aio->regs.i2s_cfg); + + value = readl(aio->cygaud->i2s_in + aio->regs.i2s_cap_stream_cfg); + value |= BIT(I2S_IN_STREAM_CFG_CAP_ENA); + writel(value, aio->cygaud->i2s_in + aio->regs.i2s_cap_stream_cfg); + + aio->streams_on |= CAPTURE_STREAM_MASK; +} + +static void audio_ssp_in_disable(struct cygnus_aio_port *aio) +{ + u32 value; + + value = readl(aio->cygaud->i2s_in + aio->regs.i2s_cap_stream_cfg); + value &= ~BIT(I2S_IN_STREAM_CFG_CAP_ENA); + writel(value, aio->cygaud->i2s_in + aio->regs.i2s_cap_stream_cfg); + + aio->streams_on &= ~CAPTURE_STREAM_MASK; + + /* If both playback and capture are off */ + if (!aio->streams_on) { + value = readl(aio->cygaud->audio + aio->regs.i2s_cfg); + value &= ~BIT(I2S_OUT_CFGX_CLK_ENA); + value &= ~BIT(I2S_OUT_CFGX_DATA_ENABLE); + writel(value, aio->cygaud->audio + aio->regs.i2s_cfg); + } + + writel(0x0, aio->cygaud->audio + aio->regs.bf_destch_ctrl); + + value = readl(aio->cygaud->audio + aio->regs.bf_destch_cfg); + value &= ~BIT(BF_DST_CFGX_CAP_ENA); + writel(value, aio->cygaud->audio + aio->regs.bf_destch_cfg); +} + +static int audio_ssp_out_enable(struct cygnus_aio_port *aio) +{ + u32 value; + int status = 0; + + switch (aio->port_type) { + case PORT_TDM: + value = readl(aio->cygaud->audio + aio->regs.i2s_stream_cfg); + value |= BIT(I2S_OUT_STREAM_ENA); + writel(value, aio->cygaud->audio + aio->regs.i2s_stream_cfg); + + writel(1, aio->cygaud->audio + aio->regs.bf_sourcech_ctrl); + + value = readl(aio->cygaud->audio + aio->regs.i2s_cfg); + value |= BIT(I2S_OUT_CFGX_CLK_ENA); + value |= BIT(I2S_OUT_CFGX_DATA_ENABLE); + writel(value, aio->cygaud->audio + aio->regs.i2s_cfg); + + value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + value |= BIT(BF_SRC_CFGX_SFIFO_ENA); + writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + + aio->streams_on |= PLAYBACK_STREAM_MASK; + break; + case PORT_SPDIF: + value = readl(aio->cygaud->audio + SPDIF_FORMAT_CFG_OFFSET); + value |= 0x3; + writel(value, aio->cygaud->audio + SPDIF_FORMAT_CFG_OFFSET); + + writel(1, aio->cygaud->audio + aio->regs.bf_sourcech_ctrl); + + value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + value |= BIT(BF_SRC_CFGX_SFIFO_ENA); + writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + break; + default: + dev_err(aio->cygaud->dev, + "Port not supported %d\n", aio->portnum); + status = -EINVAL; + } + + return status; +} + +static int audio_ssp_out_disable(struct cygnus_aio_port *aio) +{ + u32 value; + int status = 0; + + switch (aio->port_type) { + case PORT_TDM: + aio->streams_on &= ~PLAYBACK_STREAM_MASK; + + /* If both playback and capture are off */ + if (!aio->streams_on) { + value = readl(aio->cygaud->audio + aio->regs.i2s_cfg); + value &= ~BIT(I2S_OUT_CFGX_CLK_ENA); + value &= ~BIT(I2S_OUT_CFGX_DATA_ENABLE); + writel(value, aio->cygaud->audio + aio->regs.i2s_cfg); + } + + /* set group_sync_dis = 1 */ + value = readl(aio->cygaud->audio + BF_SRC_GRP_SYNC_DIS_OFFSET); + value |= BIT(aio->portnum); + writel(value, aio->cygaud->audio + BF_SRC_GRP_SYNC_DIS_OFFSET); + + writel(0, aio->cygaud->audio + aio->regs.bf_sourcech_ctrl); + + value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + value &= ~BIT(BF_SRC_CFGX_SFIFO_ENA); + writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + + /* set group_sync_dis = 0 */ + value = readl(aio->cygaud->audio + BF_SRC_GRP_SYNC_DIS_OFFSET); + value &= ~BIT(aio->portnum); + writel(value, aio->cygaud->audio + BF_SRC_GRP_SYNC_DIS_OFFSET); + + value = readl(aio->cygaud->audio + aio->regs.i2s_stream_cfg); + value &= ~BIT(I2S_OUT_STREAM_ENA); + writel(value, aio->cygaud->audio + aio->regs.i2s_stream_cfg); + + /* IOP SW INIT on OUT_I2S_x */ + value = readl(aio->cygaud->i2s_in + IOP_SW_INIT_LOGIC); + value |= BIT(aio->portnum); + writel(value, aio->cygaud->i2s_in + IOP_SW_INIT_LOGIC); + value &= ~BIT(aio->portnum); + writel(value, aio->cygaud->i2s_in + IOP_SW_INIT_LOGIC); + break; + case PORT_SPDIF: + value = readl(aio->cygaud->audio + SPDIF_FORMAT_CFG_OFFSET); + value &= ~0x3; + writel(value, aio->cygaud->audio + SPDIF_FORMAT_CFG_OFFSET); + writel(0, aio->cygaud->audio + aio->regs.bf_sourcech_ctrl); + + value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + value &= ~BIT(BF_SRC_CFGX_SFIFO_ENA); + writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + break; + default: + dev_err(aio->cygaud->dev, + "Port not supported %d\n", aio->portnum); + status = -EINVAL; + } + + return status; +} + +static int pll_configure_mclk(struct cygnus_audio *cygaud, u32 mclk, + struct cygnus_aio_port *aio) +{ + int i = 0, error; + bool found = false; + const struct pll_macro_entry *p_entry; + struct clk *ch_clk; + + for (i = 0; i < ARRAY_SIZE(pll_predef_mclk); i++) { + p_entry = &pll_predef_mclk[i]; + if (p_entry->mclk == mclk) { + found = true; + break; + } + } + if (!found) { + dev_err(cygaud->dev, + "%s No valid mclk freq (%u) found!\n", __func__, mclk); + return -EINVAL; + } + + ch_clk = cygaud->audio_clk[p_entry->pll_ch_num]; + + if ((aio->clk_trace.cap_en) && (!aio->clk_trace.cap_clk_en)) { + error = clk_prepare_enable(ch_clk); + if (error) { + dev_err(cygaud->dev, "%s clk_prepare_enable failed %d\n", + __func__, error); + return error; + } + aio->clk_trace.cap_clk_en = true; + } + + if ((aio->clk_trace.play_en) && (!aio->clk_trace.play_clk_en)) { + error = clk_prepare_enable(ch_clk); + if (error) { + dev_err(cygaud->dev, "%s clk_prepare_enable failed %d\n", + __func__, error); + return error; + } + aio->clk_trace.play_clk_en = true; + } + + error = clk_set_rate(ch_clk, mclk); + if (error) { + dev_err(cygaud->dev, "%s Set MCLK rate failed: %d\n", + __func__, error); + return error; + } + + return p_entry->pll_ch_num; +} + +static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio, + struct cygnus_audio *cygaud) +{ + u32 value, i = 0; + u32 mask = 0xf; + u32 sclk; + bool found = false; + const struct _ssp_clk_coeff *p_entry = NULL; + + if ((!aio->lrclk) || (!aio->bit_per_frame)) { + dev_err(aio->cygaud->dev, "First set up port through hw_params()\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(ssp_clk_coeff); i++) { + p_entry = &ssp_clk_coeff[i]; + if ((p_entry->rate == aio->lrclk) && + (p_entry->sclk_rate == aio->bit_per_frame) && + (p_entry->mclk == aio->mclk)) { + found = true; + break; + } + } + if (!found) { + dev_err(aio->cygaud->dev, + "No valid match found in ssp_clk_coeff array\n"); + dev_err(aio->cygaud->dev, "lrclk = %u, bits/frame = %u, mclk = %u\n", + aio->lrclk, aio->bit_per_frame, aio->mclk); + return -EINVAL; + } + + sclk = aio->bit_per_frame; + if (sclk == 512) + sclk = 0; + /* sclks_per_1fs_div = sclk cycles/32 */ + sclk /= 32; + /* Set sclk rate */ + if (aio->port_type == PORT_TDM) { + /* Set number of bitclks per frame */ + value = readl(aio->cygaud->audio + aio->regs.i2s_cfg); + value &= ~(mask << I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32); + value |= sclk << I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32; + writel(value, aio->cygaud->audio + aio->regs.i2s_cfg); + dev_dbg(aio->cygaud->dev, + "SCLKS_PER_1FS_DIV32 = 0x%x\n", value); + } + + /* Set MCLK_RATE ssp port (spdif and ssp are the same) */ + value = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg); + value &= ~(0xf << I2S_OUT_MCLKRATE_SHIFT); + value |= (p_entry->mclk_rate << I2S_OUT_MCLKRATE_SHIFT); + writel(value, aio->cygaud->audio + aio->regs.i2s_mclk_cfg); + + dev_dbg(aio->cygaud->dev, "mclk cfg reg = 0x%x\n", value); + dev_dbg(aio->cygaud->dev, "bits per frame = %u, mclk = %u Hz, lrclk = %u Hz\n", + aio->bit_per_frame, aio->mclk, aio->lrclk); + return 0; +} + +static int cygnus_ssp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai); + struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai); + int rate, bitres; + u32 value; + u32 mask = 0x1f; + int ret = 0; + + dev_dbg(aio->cygaud->dev, "%s port = %d\n", __func__, aio->portnum); + dev_dbg(aio->cygaud->dev, "params_channels %d\n", + params_channels(params)); + dev_dbg(aio->cygaud->dev, "rate %d\n", params_rate(params)); + dev_dbg(aio->cygaud->dev, "format %d\n", params_format(params)); + + rate = params_rate(params); + + switch (aio->mode) { + case CYGNUS_SSPMODE_TDM: + if ((rate == 192000) && (params_channels(params) > 4)) { + dev_err(aio->cygaud->dev, "Cannot run %d channels at %dHz\n", + params_channels(params), rate); + return -EINVAL; + } + break; + case CYGNUS_SSPMODE_I2S: + aio->bit_per_frame = 64; /* I2S must be 64 bit per frame */ + break; + default: + dev_err(aio->cygaud->dev, + "%s port running in unknown mode\n", __func__); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Configure channels as mono or stereo */ + if (params_channels(params) == 1) { + value = readl(aio->cygaud->audio + + aio->regs.bf_sourcech_cfg); + value |= BIT(BF_SRC_CFGX_SAMPLE_CH_MODE); + value &= ~BIT(BF_SRC_CFGX_BUFFER_PAIR_ENABLE); + writel(value, aio->cygaud->audio + + aio->regs.bf_sourcech_cfg); + } else { + value = readl(aio->cygaud->audio + + aio->regs.bf_sourcech_cfg); + value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE); + writel(value, aio->cygaud->audio + + aio->regs.bf_sourcech_cfg); + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bitres = 8; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + bitres = 16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S24_LE: + /* 32 bit mode is coded as 0 */ + bitres = 0; + break; + + default: + return -EINVAL; + } + + value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + value &= ~(mask << BF_SRC_CFGX_BIT_RES); + value |= (bitres << BF_SRC_CFGX_BIT_RES); + writel(value, aio->cygaud->audio + aio->regs.bf_sourcech_cfg); + + } else { + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + value = readl(aio->cygaud->audio + + aio->regs.bf_destch_cfg); + value |= BIT(BF_DST_CFGX_CAP_MODE); + writel(value, aio->cygaud->audio + + aio->regs.bf_destch_cfg); + break; + + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S24_LE: + value = readl(aio->cygaud->audio + + aio->regs.bf_destch_cfg); + value &= ~BIT(BF_DST_CFGX_CAP_MODE); + writel(value, aio->cygaud->audio + + aio->regs.bf_destch_cfg); + break; + + default: + return -EINVAL; + } + } + + aio->lrclk = rate; + + if (!aio->is_slave) + ret = cygnus_ssp_set_clocks(aio, cygaud); + + return ret; +} + +/* + * This function sets the mclk frequency for pll clock + */ +static int cygnus_ssp_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + int sel; + u32 value; + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai); + struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai); + + dev_dbg(aio->cygaud->dev, + "%s Enter port = %d\n", __func__, aio->portnum); + sel = pll_configure_mclk(cygaud, freq, aio); + if (sel < 0) { + dev_err(aio->cygaud->dev, + "%s Setting mclk failed.\n", __func__); + return -EINVAL; + } + + aio->mclk = freq; + + dev_dbg(aio->cygaud->dev, "%s Setting MCLKSEL to %d\n", __func__, sel); + value = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg); + value &= ~(0xf << I2S_OUT_PLLCLKSEL_SHIFT); + value |= (sel << I2S_OUT_PLLCLKSEL_SHIFT); + writel(value, aio->cygaud->audio + aio->regs.i2s_mclk_cfg); + + return 0; +} + +static int cygnus_ssp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai); + + snd_soc_dai_set_dma_data(dai, substream, aio); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aio->clk_trace.play_en = true; + else + aio->clk_trace.cap_en = true; + + return 0; +} + +static void cygnus_ssp_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aio->clk_trace.play_en = false; + else + aio->clk_trace.cap_en = false; + + if (!aio->is_slave) { + u32 val; + + val = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg); + val &= CYGNUS_PLLCLKSEL_MASK; + if (val >= ARRAY_SIZE(aio->cygaud->audio_clk)) { + dev_err(aio->cygaud->dev, "Clk index %u is out of bounds\n", + val); + return; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (aio->clk_trace.play_clk_en) { + clk_disable_unprepare(aio->cygaud-> + audio_clk[val]); + aio->clk_trace.play_clk_en = false; + } + } else { + if (aio->clk_trace.cap_clk_en) { + clk_disable_unprepare(aio->cygaud-> + audio_clk[val]); + aio->clk_trace.cap_clk_en = false; + } + } + } +} + +/* + * Bit Update Notes + * 31 Yes TDM Mode (1 = TDM, 0 = i2s) + * 30 Yes Slave Mode (1 = Slave, 0 = Master) + * 29:26 No Sclks per frame + * 25:18 Yes FS Width + * 17:14 No Valid Slots + * 13 No Bits (1 = 16 bits, 0 = 32 bits) + * 12:08 No Bits per samp + * 07 Yes Justifcation (1 = LSB, 0 = MSB) + * 06 Yes Alignment (1 = Delay 1 clk, 0 = no delay + * 05 Yes SCLK polarity (1 = Rising, 0 = Falling) + * 04 Yes LRCLK Polarity (1 = High for left, 0 = Low for left) + * 03:02 Yes Reserved - write as zero + * 01 No Data Enable + * 00 No CLK Enable + */ +#define I2S_OUT_CFG_REG_UPDATE_MASK 0x3C03FF03 + +/* Input cfg is same as output, but the FS width is not a valid field */ +#define I2S_IN_CFG_REG_UPDATE_MASK (I2S_OUT_CFG_REG_UPDATE_MASK | 0x03FC0000) + +int cygnus_ssp_set_custom_fsync_width(struct snd_soc_dai *cpu_dai, int len) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai); + + if ((len > 0) && (len < 256)) { + aio->fsync_width = len; + return 0; + } else { + return -EINVAL; + } +} + +static int cygnus_ssp_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai); + u32 ssp_curcfg; + u32 ssp_newcfg; + u32 ssp_outcfg; + u32 ssp_incfg; + u32 val; + u32 mask; + + dev_dbg(aio->cygaud->dev, "%s Enter fmt: %x\n", __func__, fmt); + + if (aio->port_type == PORT_SPDIF) + return -EINVAL; + + ssp_newcfg = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ssp_newcfg |= BIT(I2S_OUT_CFGX_SLAVE_MODE); + aio->is_slave = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ssp_newcfg &= ~BIT(I2S_OUT_CFGX_SLAVE_MODE); + aio->is_slave = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT); + ssp_newcfg |= BIT(I2S_OUT_CFGX_FSYNC_WIDTH); + aio->mode = CYGNUS_SSPMODE_I2S; + break; + + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + ssp_newcfg |= BIT(I2S_OUT_CFGX_TDM_MODE); + + /* DSP_A = data after FS, DSP_B = data during FS */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_DSP_A) + ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT); + + if ((aio->fsync_width > 0) && (aio->fsync_width < 256)) + ssp_newcfg |= + (aio->fsync_width << I2S_OUT_CFGX_FSYNC_WIDTH); + else + ssp_newcfg |= BIT(I2S_OUT_CFGX_FSYNC_WIDTH); + + aio->mode = CYGNUS_SSPMODE_TDM; + break; + + default: + return -EINVAL; + } + + /* + * SSP out cfg. + * Retain bits we do not want to update, then OR in new bits + */ + ssp_curcfg = readl(aio->cygaud->audio + aio->regs.i2s_cfg); + ssp_outcfg = (ssp_curcfg & I2S_OUT_CFG_REG_UPDATE_MASK) | ssp_newcfg; + writel(ssp_outcfg, aio->cygaud->audio + aio->regs.i2s_cfg); + + /* + * SSP in cfg. + * Retain bits we do not want to update, then OR in new bits + */ + ssp_curcfg = readl(aio->cygaud->i2s_in + aio->regs.i2s_cap_cfg); + ssp_incfg = (ssp_curcfg & I2S_IN_CFG_REG_UPDATE_MASK) | ssp_newcfg; + writel(ssp_incfg, aio->cygaud->i2s_in + aio->regs.i2s_cap_cfg); + + val = readl(aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE); + + /* + * Configure the word clk and bit clk as output or tristate + * Each port has 4 bits for controlling its pins. + * Shift the mask based upon port number. + */ + mask = BIT(AUD_MISC_SEROUT_LRCK_OE) + | BIT(AUD_MISC_SEROUT_SCLK_OE) + | BIT(AUD_MISC_SEROUT_MCLK_OE); + mask = mask << (aio->portnum * 4); + if (aio->is_slave) + /* Set bit for tri-state */ + val |= mask; + else + /* Clear bit for drive */ + val &= ~mask; + + dev_dbg(aio->cygaud->dev, "%s Set OE bits 0x%x\n", __func__, val); + writel(val, aio->cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE); + + return 0; +} + +static int cygnus_ssp_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai); + struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai); + + dev_dbg(aio->cygaud->dev, + "%s cmd %d at port = %d\n", __func__, cmd, aio->portnum); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + audio_ssp_out_enable(aio); + else + audio_ssp_in_enable(aio); + cygaud->active_ports++; + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + audio_ssp_out_disable(aio); + else + audio_ssp_in_disable(aio); + cygaud->active_ports--; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai); + u32 value; + int bits_per_slot = 0; /* default to 32-bits per slot */ + int frame_bits; + unsigned int active_slots; + bool found = false; + int i; + + if (tx_mask != rx_mask) { + dev_err(aio->cygaud->dev, + "%s tx_mask must equal rx_mask\n", __func__); + return -EINVAL; + } + + active_slots = hweight32(tx_mask); + + if ((active_slots < 0) || (active_slots > 16)) + return -EINVAL; + + /* Slot value must be even */ + if (active_slots % 2) + return -EINVAL; + + /* We encode 16 slots as 0 in the reg */ + if (active_slots == 16) + active_slots = 0; + + /* Slot Width is either 16 or 32 */ + switch (slot_width) { + case 16: + bits_per_slot = 1; + break; + case 32: + bits_per_slot = 0; + break; + default: + bits_per_slot = 0; + dev_warn(aio->cygaud->dev, + "%s Defaulting Slot Width to 32\n", __func__); + } + + frame_bits = slots * slot_width; + + for (i = 0; i < ARRAY_SIZE(ssp_valid_tdm_framesize); i++) { + if (ssp_valid_tdm_framesize[i] == frame_bits) { + found = true; + break; + } + } + + if (!found) { + dev_err(aio->cygaud->dev, + "%s In TDM mode, frame bits INVALID (%d)\n", + __func__, frame_bits); + return -EINVAL; + } + + aio->bit_per_frame = frame_bits; + + dev_dbg(aio->cygaud->dev, "%s active_slots %u, bits per frame %d\n", + __func__, active_slots, frame_bits); + + /* Set capture side of ssp port */ + value = readl(aio->cygaud->i2s_in + aio->regs.i2s_cap_cfg); + value &= ~(0xf << I2S_OUT_CFGX_VALID_SLOT); + value |= (active_slots << I2S_OUT_CFGX_VALID_SLOT); + value &= ~BIT(I2S_OUT_CFGX_BITS_PER_SLOT); + value |= (bits_per_slot << I2S_OUT_CFGX_BITS_PER_SLOT); + writel(value, aio->cygaud->i2s_in + aio->regs.i2s_cap_cfg); + + /* Set playback side of ssp port */ + value = readl(aio->cygaud->audio + aio->regs.i2s_cfg); + value &= ~(0xf << I2S_OUT_CFGX_VALID_SLOT); + value |= (active_slots << I2S_OUT_CFGX_VALID_SLOT); + value &= ~BIT(I2S_OUT_CFGX_BITS_PER_SLOT); + value |= (bits_per_slot << I2S_OUT_CFGX_BITS_PER_SLOT); + writel(value, aio->cygaud->audio + aio->regs.i2s_cfg); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int cygnus_ssp_suspend(struct snd_soc_dai *cpu_dai) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai); + + if (!aio->is_slave) { + u32 val; + + val = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg); + val &= CYGNUS_PLLCLKSEL_MASK; + if (val >= ARRAY_SIZE(aio->cygaud->audio_clk)) { + dev_err(aio->cygaud->dev, "Clk index %u is out of bounds\n", + val); + return -EINVAL; + } + + if (aio->clk_trace.cap_clk_en) + clk_disable_unprepare(aio->cygaud->audio_clk[val]); + if (aio->clk_trace.play_clk_en) + clk_disable_unprepare(aio->cygaud->audio_clk[val]); + + aio->pll_clk_num = val; + } + + return 0; +} + +static int cygnus_ssp_resume(struct snd_soc_dai *cpu_dai) +{ + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai); + + if (!aio->is_slave) { + if (aio->clk_trace.cap_clk_en) + clk_prepare_enable(aio->cygaud-> + audio_clk[aio->pll_clk_num]); + if (aio->clk_trace.play_clk_en) + clk_prepare_enable(aio->cygaud-> + audio_clk[aio->pll_clk_num]); + } + + return 0; +} +#else +#define cygnus_ssp_suspend NULL +#define cygnus_ssp_resume NULL +#endif + +static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = { + .startup = cygnus_ssp_startup, + .shutdown = cygnus_ssp_shutdown, + .trigger = cygnus_ssp_trigger, + .hw_params = cygnus_ssp_hw_params, + .set_fmt = cygnus_ssp_set_fmt, + .set_sysclk = cygnus_ssp_set_sysclk, + .set_tdm_slot = cygnus_set_dai_tdm_slot, +}; + + +#define INIT_CPU_DAI(num) { \ + .name = "cygnus-ssp" #num, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 16, \ + .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000, \ + .formats = SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 16, \ + .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ + }, \ + .ops = &cygnus_ssp_dai_ops, \ + .suspend = cygnus_ssp_suspend, \ + .resume = cygnus_ssp_resume, \ +} + +static const struct snd_soc_dai_driver cygnus_ssp_dai_info[] = { + INIT_CPU_DAI(0), + INIT_CPU_DAI(1), + INIT_CPU_DAI(2), +}; + +static struct snd_soc_dai_driver cygnus_spdif_dai_info = { + .name = "cygnus-spdif", + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &cygnus_ssp_dai_ops, + .suspend = cygnus_ssp_suspend, + .resume = cygnus_ssp_resume, +}; + +static struct snd_soc_dai_driver cygnus_ssp_dai[CYGNUS_MAX_PORTS]; + +static const struct snd_soc_component_driver cygnus_ssp_component = { + .name = "cygnus-audio", +}; + +/* + * Return < 0 if error + * Return 0 if disabled + * Return 1 if enabled and node is parsed successfully + */ +static int parse_ssp_child_node(struct platform_device *pdev, + struct device_node *dn, + struct cygnus_audio *cygaud, + struct snd_soc_dai_driver *p_dai) +{ + struct cygnus_aio_port *aio; + struct cygnus_ssp_regs ssp_regs[3]; + u32 rawval; + int portnum = -1; + enum cygnus_audio_port_type port_type; + + if (of_property_read_u32(dn, "reg", &rawval)) { + dev_err(&pdev->dev, "Missing reg property\n"); + return -EINVAL; + } + + portnum = rawval; + switch (rawval) { + case 0: + ssp_regs[0] = INIT_SSP_REGS(0); + port_type = PORT_TDM; + break; + case 1: + ssp_regs[1] = INIT_SSP_REGS(1); + port_type = PORT_TDM; + break; + case 2: + ssp_regs[2] = INIT_SSP_REGS(2); + port_type = PORT_TDM; + break; + case 3: + port_type = PORT_SPDIF; + break; + default: + dev_err(&pdev->dev, "Bad value for reg %u\n", rawval); + return -EINVAL; + } + + aio = &cygaud->portinfo[portnum]; + aio->cygaud = cygaud; + aio->portnum = portnum; + aio->port_type = port_type; + aio->fsync_width = -1; + + if (port_type == PORT_TDM) { + aio->regs = ssp_regs[portnum]; + + *p_dai = cygnus_ssp_dai_info[portnum]; + aio->mode = CYGNUS_SSPMODE_UNKNOWN; + + } else { /* SPDIF case */ + aio->regs.bf_sourcech_cfg = BF_SRC_CFG3_OFFSET; + aio->regs.bf_sourcech_ctrl = BF_SRC_CTRL3_OFFSET; + aio->regs.i2s_mclk_cfg = SPDIF_MCLK_CFG_OFFSET; + aio->regs.i2s_stream_cfg = SPDIF_STREAM_CFG_OFFSET; + + *p_dai = cygnus_spdif_dai_info; + + /* For the purposes of this code SPDIF can be I2S mode */ + aio->mode = CYGNUS_SSPMODE_I2S; + } + + dev_dbg(&pdev->dev, "%s portnum = %d\n", __func__, aio->portnum); + aio->streams_on = 0; + aio->cygaud->dev = &pdev->dev; + aio->clk_trace.play_en = false; + aio->clk_trace.cap_en = false; + + audio_ssp_init_portregs(aio); + return 0; +} + +static int audio_clk_init(struct platform_device *pdev, + struct cygnus_audio *cygaud) +{ + int i; + char clk_name[PROP_LEN_MAX]; + + for (i = 0; i < ARRAY_SIZE(cygaud->audio_clk); i++) { + snprintf(clk_name, PROP_LEN_MAX, "ch%d_audio", i); + + cygaud->audio_clk[i] = devm_clk_get(&pdev->dev, clk_name); + if (IS_ERR(cygaud->audio_clk[i])) + return PTR_ERR(cygaud->audio_clk[i]); + } + + return 0; +} + +static int cygnus_ssp_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct device_node *child_node; + struct resource *res = pdev->resource; + struct cygnus_audio *cygaud; + int err = -EINVAL; + int node_count; + int active_port_count; + + cygaud = devm_kzalloc(dev, sizeof(struct cygnus_audio), GFP_KERNEL); + if (!cygaud) + return -ENOMEM; + + dev_set_drvdata(dev, cygaud); + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "aud"); + cygaud->audio = devm_ioremap_resource(dev, res); + if (IS_ERR(cygaud->audio)) + return PTR_ERR(cygaud->audio); + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "i2s_in"); + cygaud->i2s_in = devm_ioremap_resource(dev, res); + if (IS_ERR(cygaud->i2s_in)) + return PTR_ERR(cygaud->i2s_in); + + /* Tri-state all controlable pins until we know that we need them */ + writel(CYGNUS_SSP_TRISTATE_MASK, + cygaud->audio + AUD_MISC_SEROUT_OE_REG_BASE); + + node_count = of_get_child_count(pdev->dev.of_node); + if ((node_count < 1) || (node_count > CYGNUS_MAX_PORTS)) { + dev_err(dev, "child nodes is %d. Must be between 1 and %d\n", + node_count, CYGNUS_MAX_PORTS); + return -EINVAL; + } + + active_port_count = 0; + + for_each_available_child_of_node(pdev->dev.of_node, child_node) { + err = parse_ssp_child_node(pdev, child_node, cygaud, + &cygnus_ssp_dai[active_port_count]); + + /* negative is err, 0 is active and good, 1 is disabled */ + if (err < 0) + return err; + else if (!err) { + dev_dbg(dev, "Activating DAI: %s\n", + cygnus_ssp_dai[active_port_count].name); + active_port_count++; + } + } + + cygaud->dev = dev; + cygaud->active_ports = 0; + + dev_dbg(dev, "Registering %d DAIs\n", active_port_count); + err = snd_soc_register_component(dev, &cygnus_ssp_component, + cygnus_ssp_dai, active_port_count); + if (err) { + dev_err(dev, "snd_soc_register_dai failed\n"); + return err; + } + + cygaud->irq_num = platform_get_irq(pdev, 0); + if (cygaud->irq_num <= 0) { + dev_err(dev, "platform_get_irq failed\n"); + err = cygaud->irq_num; + goto err_irq; + } + + err = audio_clk_init(pdev, cygaud); + if (err) { + dev_err(dev, "audio clock initialization failed\n"); + goto err_irq; + } + + err = cygnus_soc_platform_register(dev, cygaud); + if (err) { + dev_err(dev, "platform reg error %d\n", err); + goto err_irq; + } + + return 0; + +err_irq: + snd_soc_unregister_component(dev); + return err; +} + +static int cygnus_ssp_remove(struct platform_device *pdev) +{ + cygnus_soc_platform_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static const struct of_device_id cygnus_ssp_of_match[] = { + { .compatible = "brcm,cygnus-audio" }, + {}, +}; +MODULE_DEVICE_TABLE(of, cygnus_ssp_of_match); + +static struct platform_driver cygnus_ssp_driver = { + .probe = cygnus_ssp_probe, + .remove = cygnus_ssp_remove, + .driver = { + .name = "cygnus-ssp", + .of_match_table = cygnus_ssp_of_match, + }, +}; + +module_platform_driver(cygnus_ssp_driver); + +MODULE_ALIAS("platform:cygnus-ssp"); +MODULE_LICENSE("GPL v2"); +MODULE_AUTHOR("Broadcom"); +MODULE_DESCRIPTION("Cygnus ASoC SSP Interface"); diff --git a/sound/soc/bcm/cygnus-ssp.h b/sound/soc/bcm/cygnus-ssp.h new file mode 100644 index 0000000..33dd343 --- /dev/null +++ b/sound/soc/bcm/cygnus-ssp.h @@ -0,0 +1,139 @@ +/* + * Copyright (C) 2014-2015 Broadcom Corporation + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; without even the implied warranty + * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#ifndef __CYGNUS_SSP_H__ +#define __CYGNUS_SSP_H__ + +#define CYGNUS_TDM_DAI_MAX_SLOTS 16 + +#define CYGNUS_MAX_PLAYBACK_PORTS 4 +#define CYGNUS_MAX_CAPTURE_PORTS 3 +#define CYGNUS_MAX_I2S_PORTS 3 +#define CYGNUS_MAX_PORTS CYGNUS_MAX_PLAYBACK_PORTS +#define CYGNUS_AUIDO_MAX_NUM_CLKS 3 + +#define CYGNUS_SSP_FRAMEBITS_DIV 1 + +#define CYGNUS_SSPMODE_I2S 0 +#define CYGNUS_SSPMODE_TDM 1 +#define CYGNUS_SSPMODE_UNKNOWN -1 + +#define CYGNUS_SSP_CLKSRC_PLL 0 + +/* Max string length of our dt property names */ +#define PROP_LEN_MAX 40 + +struct ringbuf_regs { + unsigned rdaddr; + unsigned wraddr; + unsigned baseaddr; + unsigned endaddr; + unsigned fmark; /* freemark for play, fullmark for caputure */ + unsigned period_bytes; + unsigned buf_size; +}; + +#define RINGBUF_REG_PLAYBACK(num) ((struct ringbuf_regs) { \ + .rdaddr = SRC_RBUF_ ##num## _RDADDR_OFFSET, \ + .wraddr = SRC_RBUF_ ##num## _WRADDR_OFFSET, \ + .baseaddr = SRC_RBUF_ ##num## _BASEADDR_OFFSET, \ + .endaddr = SRC_RBUF_ ##num## _ENDADDR_OFFSET, \ + .fmark = SRC_RBUF_ ##num## _FREE_MARK_OFFSET, \ + .period_bytes = 0, \ + .buf_size = 0, \ +}) + +#define RINGBUF_REG_CAPTURE(num) ((struct ringbuf_regs) { \ + .rdaddr = DST_RBUF_ ##num## _RDADDR_OFFSET, \ + .wraddr = DST_RBUF_ ##num## _WRADDR_OFFSET, \ + .baseaddr = DST_RBUF_ ##num## _BASEADDR_OFFSET, \ + .endaddr = DST_RBUF_ ##num## _ENDADDR_OFFSET, \ + .fmark = DST_RBUF_ ##num## _FULL_MARK_OFFSET, \ + .period_bytes = 0, \ + .buf_size = 0, \ +}) + +enum cygnus_audio_port_type { + PORT_TDM, + PORT_SPDIF, +}; + +struct cygnus_ssp_regs { + u32 i2s_stream_cfg; + u32 i2s_cfg; + u32 i2s_cap_stream_cfg; + u32 i2s_cap_cfg; + u32 i2s_mclk_cfg; + + u32 bf_destch_ctrl; + u32 bf_destch_cfg; + u32 bf_sourcech_ctrl; + u32 bf_sourcech_cfg; + u32 bf_sourcech_grp; +}; + +struct cygnus_track_clk { + bool cap_en; + bool play_en; + bool cap_clk_en; + bool play_clk_en; +}; + +struct cygnus_aio_port { + int portnum; + int mode; + bool is_slave; + int streams_on; /* will be 0 if both capture and play are off */ + int fsync_width; + int port_type; + + u32 mclk; + u32 lrclk; + u32 bit_per_frame; + u32 pll_clk_num; + + struct cygnus_audio *cygaud; + struct cygnus_ssp_regs regs; + + struct ringbuf_regs play_rb_regs; + struct ringbuf_regs capture_rb_regs; + + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + + struct cygnus_track_clk clk_trace; +}; + + +struct cygnus_audio { + struct cygnus_aio_port portinfo[CYGNUS_MAX_PORTS]; + + int irq_num; + void __iomem *audio; + struct device *dev; + void __iomem *i2s_in; + + struct clk *audio_clk[CYGNUS_AUIDO_MAX_NUM_CLKS]; + int active_ports; + unsigned long vco_rate; +}; + +extern int cygnus_ssp_get_mode(struct snd_soc_dai *cpu_dai); +extern int cygnus_ssp_add_pll_tweak_controls(struct snd_soc_pcm_runtime *rtd); +extern int cygnus_ssp_set_custom_fsync_width(struct snd_soc_dai *cpu_dai, + int len); +extern int cygnus_soc_platform_register(struct device *dev, + struct cygnus_audio *cygaud); +extern int cygnus_soc_platform_unregister(struct device *dev); +extern int cygnus_ssp_set_custom_fsync_width(struct snd_soc_dai *cpu_dai, + int len); +#endif
On Tue, Mar 29, 2016 at 11:46:31AM -0700, Simran Rai wrote:
A few issues here, a lot of them are stylistic though there's what look like a couple of small bugs here too.
+static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio,
struct cygnus_audio *cygaud)
+{
- u32 value, i = 0;
- u32 mask = 0xf;
- u32 sclk;
- bool found = false;
- const struct _ssp_clk_coeff *p_entry = NULL;
- if ((!aio->lrclk) || (!aio->bit_per_frame)) {
dev_err(aio->cygaud->dev, "First set up port through hw_params()\n");
return -EINVAL;
- }
This function is only ever called from one site in hw_prams(). What is this defending against? A check like this seems very worrying, if it ever goes off that seems to indicate either something is seriously wrong or we should be recording something then coming back and trying again later.
- for (i = 0; i < ARRAY_SIZE(ssp_clk_coeff); i++) {
p_entry = &ssp_clk_coeff[i];
if ((p_entry->rate == aio->lrclk) &&
(p_entry->sclk_rate == aio->bit_per_frame) &&
(p_entry->mclk == aio->mclk)) {
Why the strange indentation here?
- /* Set sclk rate */
- if (aio->port_type == PORT_TDM) {
switch statment here for extensibility.
/* Configure channels as mono or stereo */
if (params_channels(params) == 1) {
value = readl(aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
value |= BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
value &= ~BIT(BF_SRC_CFGX_BUFFER_PAIR_ENABLE);
writel(value, aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
} else {
value = readl(aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
writel(value, aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
}
Either this should be a switch statement or the comment should say we support more than stereo. It's also not clear to me how BUFFER_PAIR_ENABLE gets set again if we go from mono to stereo.
- if (!aio->is_slave) {
if (aio->clk_trace.cap_clk_en)
clk_prepare_enable(aio->cygaud->
audio_clk[aio->pll_clk_num]);
Should check the return value of clk_prepare_enable().
- .playback = {
.channels_min = 2,
.channels_max = 2,
.rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
SNDRV_PCM_RATE_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S32_LE,
According to hw_params() the driver also supports S8 and S24.
- if (port_type == PORT_TDM) {
- } else { /* SPDIF case */
switch statement...
Hi Mark,
On Tue, Mar 29, 2016 at 3:16 PM, Mark Brown broonie@kernel.org wrote:
On Tue, Mar 29, 2016 at 11:46:31AM -0700, Simran Rai wrote:
A few issues here, a lot of them are stylistic though there's what look like a couple of small bugs here too.
+static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio,
struct cygnus_audio *cygaud)
+{
u32 value, i = 0;
u32 mask = 0xf;
u32 sclk;
bool found = false;
const struct _ssp_clk_coeff *p_entry = NULL;
if ((!aio->lrclk) || (!aio->bit_per_frame)) {
dev_err(aio->cygaud->dev, "First set up port through
hw_params()\n");
return -EINVAL;
}
This function is only ever called from one site in hw_prams(). What is this defending against? A check like this seems very worrying, if it ever goes off that seems to indicate either something is seriously wrong or we should be recording something then coming back and trying again later.
Will remove it.
for (i = 0; i < ARRAY_SIZE(ssp_clk_coeff); i++) {
p_entry = &ssp_clk_coeff[i];
if ((p_entry->rate == aio->lrclk) &&
(p_entry->sclk_rate == aio->bit_per_frame)
&&
(p_entry->mclk == aio->mclk)) {
Why the strange indentation here?
Will fix this.
/* Set sclk rate */
if (aio->port_type == PORT_TDM) {
switch statment here for extensibility.
Will replace if statement with switch statement
/* Configure channels as mono or stereo */
if (params_channels(params) == 1) {
value = readl(aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
value |= BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
value &= ~BIT(BF_SRC_CFGX_BUFFER_PAIR_ENABLE);
writel(value, aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
} else {
value = readl(aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
writel(value, aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
}
Either this should be a switch statement or the comment should say we support more than stereo. It's also not clear to me how BUFFER_PAIR_ENABLE gets set again if we go from mono to stereo.
Will change to switch statement. Thanks for exposing the bug related to BUFFER_PAIR_ENABLE. Will fix it.
if (!aio->is_slave) {
if (aio->clk_trace.cap_clk_en)
clk_prepare_enable(aio->cygaud->
audio_clk[aio->pll_clk_num]);
Should check the return value of clk_prepare_enable().
Will do.
.playback = {
.channels_min = 2,
.channels_max = 2,
.rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
SNDRV_PCM_RATE_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S32_LE,
According to hw_params() the driver also supports S8 and S24.
SSP and TDM playback supports S8, S16 and S32. SSP and TDM capture supports S16 and S32. SPDIF playback supports S16 and S32. Will fix the code to reflect this.
if (port_type == PORT_TDM) {
} else { /* SPDIF case */
switch statement...
Will do.
Thanks, Simran
Hi Mark, Resending my replies as plain text.Apologies for the duplicate.
On Tue, Mar 29, 2016 at 3:16 PM, Mark Brown broonie@kernel.org wrote:
On Tue, Mar 29, 2016 at 11:46:31AM -0700, Simran Rai wrote:
A few issues here, a lot of them are stylistic though there's what look like a couple of small bugs here too.
+static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio,
struct cygnus_audio *cygaud)
+{
u32 value, i = 0;
u32 mask = 0xf;
u32 sclk;
bool found = false;
const struct _ssp_clk_coeff *p_entry = NULL;
if ((!aio->lrclk) || (!aio->bit_per_frame)) {
dev_err(aio->cygaud->dev, "First set up port through hw_params()\n");
return -EINVAL;
}
This function is only ever called from one site in hw_prams(). What is this defending against? A check like this seems very worrying, if it ever goes off that seems to indicate either something is seriously wrong or we should be recording something then coming back and trying again later.
Will remove it.
for (i = 0; i < ARRAY_SIZE(ssp_clk_coeff); i++) {
p_entry = &ssp_clk_coeff[i];
if ((p_entry->rate == aio->lrclk) &&
(p_entry->sclk_rate == aio->bit_per_frame) &&
(p_entry->mclk == aio->mclk)) {
Why the strange indentation here?
Will fix this.
/* Set sclk rate */
if (aio->port_type == PORT_TDM) {
switch statment here for extensibility.
Will replace.
/* Configure channels as mono or stereo */
if (params_channels(params) == 1) {
value = readl(aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
value |= BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
value &= ~BIT(BF_SRC_CFGX_BUFFER_PAIR_ENABLE);
writel(value, aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
} else {
value = readl(aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
writel(value, aio->cygaud->audio +
aio->regs.bf_sourcech_cfg);
}
Either this should be a switch statement or the comment should say we support more than stereo. It's also not clear to me how BUFFER_PAIR_ENABLE gets set again if we go from mono to stereo.
Will change to switch statement. Thanks for exposing the bug related to BUFFER_PAIR_ENABLE. Will fix it.
if (!aio->is_slave) {
if (aio->clk_trace.cap_clk_en)
clk_prepare_enable(aio->cygaud->
audio_clk[aio->pll_clk_num]);
Should check the return value of clk_prepare_enable().
Will do.
.playback = {
.channels_min = 2,
.channels_max = 2,
.rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
SNDRV_PCM_RATE_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S32_LE,
According to hw_params() the driver also supports S8 and S24.
SSP and TDM playback supports S8, S16 and S32. SSP and TDM capture supports S16 and S32. SPDIF playback supports S16 and S32. Will fix the code to reflect this.
if (port_type == PORT_TDM) {
} else { /* SPDIF case */
switch statement...
Will do.
Thanks, Simran
From: Simran Rai ssimran@broadcom.com
This patch adds Cygnus audio DMA driver. It supports playback and capture modes and uses ringbuffers for data transfer.
Signed-off-by: Lori Hikichi lhikichi@broadcom.com Signed-off-by: Simran Rai ssimran@broadcom.com Reviewed-by: Ray Jui rjui@broadcom.com Reviewed-by: Arun Parameswaran arunp@broadcom.com Reviewed-by: Scott Branden sbranden@broadcom.com Signed-off-by: Simran Rai simran.rai@broadcom.com --- sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 875 insertions(+) create mode 100644 sound/soc/bcm/cygnus-pcm.c
diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 6a834e1..d528aac 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -7,3 +7,12 @@ config SND_BCM2835_SOC_I2S Say Y or M if you want to add support for codecs attached to the BCM2835 I2S interface. You will also need to select the audio interfaces to support below. + +config SND_SOC_CYGNUS + tristate "SoC platform audio for Broadcom Cygnus chips" + depends on ARCH_BCM_CYGNUS || COMPILE_TEST + help + Say Y if you want to add support for ASoC audio on Broadcom + Cygnus chips (bcm958300, bcm958305, bcm911360) + + If you don't know what to do here, say N. \ No newline at end of file diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile index bc816b7..fc739d0 100644 --- a/sound/soc/bcm/Makefile +++ b/sound/soc/bcm/Makefile @@ -3,3 +3,8 @@ snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o
obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o
+# CYGNUS Platform Support +snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o + +obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o + diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c new file mode 100644 index 0000000..d616e096 --- /dev/null +++ b/sound/soc/bcm/cygnus-pcm.c @@ -0,0 +1,861 @@ +/* + * Copyright (C) 2014-2015 Broadcom Corporation + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; without even the implied warranty + * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#include <linux/debugfs.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/timer.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "cygnus-ssp.h" + +/* Register offset needed for ASoC PCM module */ + +#define INTH_R5F_STATUS_OFFSET 0x040 +#define INTH_R5F_CLEAR_OFFSET 0x048 +#define INTH_R5F_MASK_SET_OFFSET 0x050 +#define INTH_R5F_MASK_CLEAR_OFFSET 0x054 + +#define BF_REARM_FREE_MARK_OFFSET 0x344 +#define BF_REARM_FULL_MARK_OFFSET 0x348 + +/* Ring Buffer Ctrl Regs --- Start */ +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_RDADDR_REG_BASE */ +#define SRC_RBUF_0_RDADDR_OFFSET 0x500 +#define SRC_RBUF_1_RDADDR_OFFSET 0x518 +#define SRC_RBUF_2_RDADDR_OFFSET 0x530 +#define SRC_RBUF_3_RDADDR_OFFSET 0x548 +#define SRC_RBUF_4_RDADDR_OFFSET 0x560 +#define SRC_RBUF_5_RDADDR_OFFSET 0x578 +#define SRC_RBUF_6_RDADDR_OFFSET 0x590 + +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_WRADDR_REG_BASE */ +#define SRC_RBUF_0_WRADDR_OFFSET 0x504 +#define SRC_RBUF_1_WRADDR_OFFSET 0x51c +#define SRC_RBUF_2_WRADDR_OFFSET 0x534 +#define SRC_RBUF_3_WRADDR_OFFSET 0x54c +#define SRC_RBUF_4_WRADDR_OFFSET 0x564 +#define SRC_RBUF_5_WRADDR_OFFSET 0x57c +#define SRC_RBUF_6_WRADDR_OFFSET 0x594 + +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_BASEADDR_REG_BASE */ +#define SRC_RBUF_0_BASEADDR_OFFSET 0x508 +#define SRC_RBUF_1_BASEADDR_OFFSET 0x520 +#define SRC_RBUF_2_BASEADDR_OFFSET 0x538 +#define SRC_RBUF_3_BASEADDR_OFFSET 0x550 +#define SRC_RBUF_4_BASEADDR_OFFSET 0x568 +#define SRC_RBUF_5_BASEADDR_OFFSET 0x580 +#define SRC_RBUF_6_BASEADDR_OFFSET 0x598 + +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_ENDADDR_REG_BASE */ +#define SRC_RBUF_0_ENDADDR_OFFSET 0x50c +#define SRC_RBUF_1_ENDADDR_OFFSET 0x524 +#define SRC_RBUF_2_ENDADDR_OFFSET 0x53c +#define SRC_RBUF_3_ENDADDR_OFFSET 0x554 +#define SRC_RBUF_4_ENDADDR_OFFSET 0x56c +#define SRC_RBUF_5_ENDADDR_OFFSET 0x584 +#define SRC_RBUF_6_ENDADDR_OFFSET 0x59c + +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_FREE_MARK_REG_BASE */ +#define SRC_RBUF_0_FREE_MARK_OFFSET 0x510 +#define SRC_RBUF_1_FREE_MARK_OFFSET 0x528 +#define SRC_RBUF_2_FREE_MARK_OFFSET 0x540 +#define SRC_RBUF_3_FREE_MARK_OFFSET 0x558 +#define SRC_RBUF_4_FREE_MARK_OFFSET 0x570 +#define SRC_RBUF_5_FREE_MARK_OFFSET 0x588 +#define SRC_RBUF_6_FREE_MARK_OFFSET 0x5a0 + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_RDADDR_REG_BASE */ +#define DST_RBUF_0_RDADDR_OFFSET 0x5c0 +#define DST_RBUF_1_RDADDR_OFFSET 0x5d8 +#define DST_RBUF_2_RDADDR_OFFSET 0x5f0 +#define DST_RBUF_3_RDADDR_OFFSET 0x608 +#define DST_RBUF_4_RDADDR_OFFSET 0x620 +#define DST_RBUF_5_RDADDR_OFFSET 0x638 + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_WRADDR_REG_BASE */ +#define DST_RBUF_0_WRADDR_OFFSET 0x5c4 +#define DST_RBUF_1_WRADDR_OFFSET 0x5dc +#define DST_RBUF_2_WRADDR_OFFSET 0x5f4 +#define DST_RBUF_3_WRADDR_OFFSET 0x60c +#define DST_RBUF_4_WRADDR_OFFSET 0x624 +#define DST_RBUF_5_WRADDR_OFFSET 0x63c + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_BASEADDR_REG_BASE */ +#define DST_RBUF_0_BASEADDR_OFFSET 0x5c8 +#define DST_RBUF_1_BASEADDR_OFFSET 0x5e0 +#define DST_RBUF_2_BASEADDR_OFFSET 0x5f8 +#define DST_RBUF_3_BASEADDR_OFFSET 0x610 +#define DST_RBUF_4_BASEADDR_OFFSET 0x628 +#define DST_RBUF_5_BASEADDR_OFFSET 0x640 + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_ENDADDR_REG_BASE */ +#define DST_RBUF_0_ENDADDR_OFFSET 0x5cc +#define DST_RBUF_1_ENDADDR_OFFSET 0x5e4 +#define DST_RBUF_2_ENDADDR_OFFSET 0x5fc +#define DST_RBUF_3_ENDADDR_OFFSET 0x614 +#define DST_RBUF_4_ENDADDR_OFFSET 0x62c +#define DST_RBUF_5_ENDADDR_OFFSET 0x644 + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_FULL_MARK_REG_BASE */ +#define DST_RBUF_0_FULL_MARK_OFFSET 0x5d0 +#define DST_RBUF_1_FULL_MARK_OFFSET 0x5e8 +#define DST_RBUF_2_FULL_MARK_OFFSET 0x600 +#define DST_RBUF_3_FULL_MARK_OFFSET 0x618 +#define DST_RBUF_4_FULL_MARK_OFFSET 0x630 +#define DST_RBUF_5_FULL_MARK_OFFSET 0x648 +/* Ring Buffer Ctrl Regs --- End */ + +/* Error Status Regs --- Start */ +/* AUD_FMM_BF_ESR_ESRX_STATUS_REG_BASE */ +#define ESR0_STATUS_OFFSET 0x900 +#define ESR1_STATUS_OFFSET 0x918 +#define ESR2_STATUS_OFFSET 0x930 +#define ESR3_STATUS_OFFSET 0x948 +#define ESR4_STATUS_OFFSET 0x960 + +/* AUD_FMM_BF_ESR_ESRX_STATUS_CLEAR_REG_BASE */ +#define ESR0_STATUS_CLR_OFFSET 0x908 +#define ESR1_STATUS_CLR_OFFSET 0x920 +#define ESR2_STATUS_CLR_OFFSET 0x938 +#define ESR3_STATUS_CLR_OFFSET 0x950 +#define ESR4_STATUS_CLR_OFFSET 0x968 + +/* AUD_FMM_BF_ESR_ESRX_MASK_REG_BASE */ +#define ESR0_MASK_STATUS_OFFSET 0x90c +#define ESR1_MASK_STATUS_OFFSET 0x924 +#define ESR2_MASK_STATUS_OFFSET 0x93c +#define ESR3_MASK_STATUS_OFFSET 0x954 +#define ESR4_MASK_STATUS_OFFSET 0x96c + +/* AUD_FMM_BF_ESR_ESRX_MASK_SET_REG_BASE */ +#define ESR0_MASK_SET_OFFSET 0x910 +#define ESR1_MASK_SET_OFFSET 0x928 +#define ESR2_MASK_SET_OFFSET 0x940 +#define ESR3_MASK_SET_OFFSET 0x958 +#define ESR4_MASK_SET_OFFSET 0x970 + +/* AUD_FMM_BF_ESR_ESRX_MASK_CLEAR_REG_BASE */ +#define ESR0_MASK_CLR_OFFSET 0x914 +#define ESR1_MASK_CLR_OFFSET 0x92c +#define ESR2_MASK_CLR_OFFSET 0x944 +#define ESR3_MASK_CLR_OFFSET 0x95c +#define ESR4_MASK_CLR_OFFSET 0x974 +/* Error Status Regs --- End */ + +#define R5F_ESR0_SHIFT 0 /* esr0 = fifo underflow */ +#define R5F_ESR1_SHIFT 1 /* esr1 = ringbuf underflow */ +#define R5F_ESR2_SHIFT 2 /* esr2 = ringbuf overflow */ +#define R5F_ESR3_SHIFT 3 /* esr3 = freemark */ +#define R5F_ESR4_SHIFT 4 /* esr4 = fullmark */ + + +/* Mask for R5F register. Set all relevant interrupt for playback handler */ +#define ANY_PLAYBACK_IRQ (BIT(R5F_ESR0_SHIFT) | \ + BIT(R5F_ESR1_SHIFT) | \ + BIT(R5F_ESR3_SHIFT)) + +/* Mask for R5F register. Set all relevant interrupt for capture handler */ +#define ANY_CAPTURE_IRQ (BIT(R5F_ESR2_SHIFT) | BIT(R5F_ESR4_SHIFT)) + +/* + * PERIOD_BYTES_MIN is the number of bytes to at which the interrupt will tick. + * This number should be a multiple of 256. Minimum value is 256 + */ +#define PERIOD_BYTES_MIN 0x100 + +static const struct snd_pcm_hardware cygnus_pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + + /* A period is basically an interrupt */ + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = 0x10000, + + /* period_min/max gives range of approx interrupts per buffer */ + .periods_min = 2, + .periods_max = 8, + + /* + * maximum buffer size in bytes = period_bytes_max * periods_max + * We allocate this amount of data for each enabled channel + */ + .buffer_bytes_max = 4 * 0x8000, +}; + +static u64 cygnus_dma_dmamask = DMA_BIT_MASK(32); + +static struct cygnus_aio_port *cygnus_dai_get_dma_data( + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + + return snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); +} + +static void ringbuf_set_initial(void __iomem *audio_io, + struct ringbuf_regs *p_rbuf, + bool is_playback, + u32 start, + u32 periodsize, + u32 bufsize) +{ + u32 initial_rd; + u32 initial_wr; + u32 end; + u32 fmark_val; /* free or full mark */ + + p_rbuf->period_bytes = periodsize; + p_rbuf->buf_size = bufsize; + + if (is_playback) { + /* Set the pointers to indicate full (flip uppermost bit) */ + initial_rd = start; + initial_wr = initial_rd ^ BIT(31); + } else { + /* Set the pointers to indicate empty */ + initial_wr = start; + initial_rd = initial_wr; + } + + end = start + bufsize - 1; + + /* + * The interrupt will fire when free/full mark is *exceeded* + * The fmark value must be multiple of PERIOD_BYTES_MIN so set fmark + * to be PERIOD_BYTES_MIN less than the period size. + */ + fmark_val = periodsize - PERIOD_BYTES_MIN; + + writel(start, audio_io + p_rbuf->baseaddr); + writel(end, audio_io + p_rbuf->endaddr); + writel(fmark_val, audio_io + p_rbuf->fmark); + writel(initial_rd, audio_io + p_rbuf->rdaddr); + writel(initial_wr, audio_io + p_rbuf->wraddr); +} + +static int configure_ringbuf_regs(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + struct ringbuf_regs *p_rbuf; + int status = 0; + + aio = cygnus_dai_get_dma_data(substream); + + /* Map the ssp portnum to a set of ring buffers. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + p_rbuf = &aio->play_rb_regs; + + switch (aio->portnum) { + case 0: + *p_rbuf = RINGBUF_REG_PLAYBACK(0); + break; + case 1: + *p_rbuf = RINGBUF_REG_PLAYBACK(2); + break; + case 2: + *p_rbuf = RINGBUF_REG_PLAYBACK(4); + break; + case 3: /* SPDIF */ + *p_rbuf = RINGBUF_REG_PLAYBACK(6); + break; + default: + status = -EINVAL; + } + } else { + p_rbuf = &aio->capture_rb_regs; + + switch (aio->portnum) { + case 0: + *p_rbuf = RINGBUF_REG_CAPTURE(0); + break; + case 1: + *p_rbuf = RINGBUF_REG_CAPTURE(2); + break; + case 2: + *p_rbuf = RINGBUF_REG_CAPTURE(4); + break; + default: + status = -EINVAL; + } + } + + return status; +} + +static struct ringbuf_regs *get_ringbuf(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + struct ringbuf_regs *p_rbuf = NULL; + + aio = cygnus_dai_get_dma_data(substream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + p_rbuf = &aio->play_rb_regs; + else + p_rbuf = &aio->capture_rb_regs; + + return p_rbuf; +} + +static void enable_intr(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + u32 clear_mask; + + aio = cygnus_dai_get_dma_data(substream); + + /* The port number maps to the bit position to be cleared */ + clear_mask = BIT(aio->portnum); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Clear interrupt status before enabling them */ + writel(clear_mask, aio->cygaud->audio + ESR0_STATUS_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR1_STATUS_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR3_STATUS_CLR_OFFSET); + /* Unmask the interrupts of the given port*/ + writel(clear_mask, aio->cygaud->audio + ESR0_MASK_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR1_MASK_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR3_MASK_CLR_OFFSET); + + writel(ANY_PLAYBACK_IRQ, + aio->cygaud->audio + INTH_R5F_MASK_CLEAR_OFFSET); + } else { + writel(clear_mask, aio->cygaud->audio + ESR2_STATUS_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR4_STATUS_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR2_MASK_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR4_MASK_CLR_OFFSET); + + writel(ANY_CAPTURE_IRQ, + aio->cygaud->audio + INTH_R5F_MASK_CLEAR_OFFSET); + } + +} + +static void disable_intr(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct cygnus_aio_port *aio; + u32 set_mask; + + aio = cygnus_dai_get_dma_data(substream); + + dev_dbg(rtd->cpu_dai->dev, "%s on port %d\n", __func__, aio->portnum); + + /* The port number maps to the bit position to be set */ + set_mask = BIT(aio->portnum); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Mask the interrupts of the given port*/ + writel(set_mask, aio->cygaud->audio + ESR0_MASK_SET_OFFSET); + writel(set_mask, aio->cygaud->audio + ESR1_MASK_SET_OFFSET); + writel(set_mask, aio->cygaud->audio + ESR3_MASK_SET_OFFSET); + } else { + writel(set_mask, aio->cygaud->audio + ESR2_MASK_SET_OFFSET); + writel(set_mask, aio->cygaud->audio + ESR4_MASK_SET_OFFSET); + } + +} + +static int cygnus_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + enable_intr(substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + disable_intr(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static void cygnus_pcm_period_elapsed(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + struct ringbuf_regs *p_rbuf = NULL; + u32 regval; + + aio = cygnus_dai_get_dma_data(substream); + + p_rbuf = get_ringbuf(substream); + + /* + * If free/full mark interrupt occurs, provide timestamp + * to ALSA and update appropriate idx by period_bytes + */ + snd_pcm_period_elapsed(substream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Set the ring buffer to full */ + regval = readl(aio->cygaud->audio + p_rbuf->rdaddr); + regval = regval ^ BIT(31); + writel(regval, aio->cygaud->audio + p_rbuf->wraddr); + } else { + /* Set the ring buffer to empty */ + regval = readl(aio->cygaud->audio + p_rbuf->wraddr); + writel(regval, aio->cygaud->audio + p_rbuf->rdaddr); + } +} + +/* + * ESR0/1/3 status Description + * 0x1 I2S0_out port caused interrupt + * 0x2 I2S1_out port caused interrupt + * 0x4 I2S2_out port caused interrupt + * 0x8 SPDIF_out port caused interrupt + */ +static void handle_playback_irq(struct cygnus_audio *cygaud) +{ + void __iomem *audio_io; + u32 port; + u32 esr_status0, esr_status1, esr_status3; + + audio_io = cygaud->audio; + + /* + * ESR status gets updates with/without interrupts enabled. + * So, check the ESR mask, which provides interrupt enable/ + * disable status and use it to determine which ESR status + * should be serviced. + */ + esr_status0 = readl(audio_io + ESR0_STATUS_OFFSET); + esr_status0 &= ~readl(audio_io + ESR0_MASK_STATUS_OFFSET); + esr_status1 = readl(audio_io + ESR1_STATUS_OFFSET); + esr_status1 &= ~readl(audio_io + ESR1_MASK_STATUS_OFFSET); + esr_status3 = readl(audio_io + ESR3_STATUS_OFFSET); + esr_status3 &= ~readl(audio_io + ESR3_MASK_STATUS_OFFSET); + + for (port = 0; port < CYGNUS_MAX_PLAYBACK_PORTS; port++) { + u32 esrmask = BIT(port); + + /* + * Ringbuffer or FIFO underflow + * If we get this interrupt then, it is also true that we have + * not yet responded to the freemark interrupt. + * Log a debug message. The freemark handler below will + * handle getting everything going again. + */ + if ((esrmask & esr_status1) || (esrmask & esr_status0)) { + dev_dbg(cygaud->dev, + "Underrun: esr0=0x%x, esr1=0x%x esr3=0x%x\n", + esr_status0, esr_status1, esr_status3); + } + + /* + * Freemark is hit. This is the normal interrupt. + * In typical operation the read and write regs will be equal + */ + if (esrmask & esr_status3) { + struct snd_pcm_substream *playstr; + + playstr = cygaud->portinfo[port].play_stream; + cygnus_pcm_period_elapsed(playstr); + } + } + + /* Clear ESR interrupt */ + writel(esr_status0, audio_io + ESR0_STATUS_CLR_OFFSET); + writel(esr_status1, audio_io + ESR1_STATUS_CLR_OFFSET); + writel(esr_status3, audio_io + ESR3_STATUS_CLR_OFFSET); + /* Rearm freemark logic by writing 1 to the correct bit */ + writel(esr_status3, audio_io + BF_REARM_FREE_MARK_OFFSET); +} + +/* + * ESR2/4 status Description + * 0x1 I2S0_in port caused interrupt + * 0x2 I2S1_in port caused interrupt + * 0x4 I2S2_in port caused interrupt + */ +static void handle_capture_irq(struct cygnus_audio *cygaud) +{ + void __iomem *audio_io; + u32 port; + u32 esr_status2, esr_status4; + + audio_io = cygaud->audio; + + /* + * ESR status gets updates with/without interrupts enabled. + * So, check the ESR mask, which provides interrupt enable/ + * disable status and use it to determine which ESR status + * should be serviced. + */ + esr_status2 = readl(audio_io + ESR2_STATUS_OFFSET); + esr_status2 &= ~readl(audio_io + ESR2_MASK_STATUS_OFFSET); + esr_status4 = readl(audio_io + ESR4_STATUS_OFFSET); + esr_status4 &= ~readl(audio_io + ESR4_MASK_STATUS_OFFSET); + + for (port = 0; port < CYGNUS_MAX_CAPTURE_PORTS; port++) { + u32 esrmask = BIT(port); + + /* + * Ringbuffer or FIFO overflow + * If we get this interrupt then, it is also true that we have + * not yet responded to the fullmark interrupt. + * Log a debug message. The fullmark handler below will + * handle getting everything going again. + */ + if (esrmask & esr_status2) + dev_dbg(cygaud->dev, + "Overflow: esr2=0x%x\n", esr_status2); + + if (esrmask & esr_status4) { + struct snd_pcm_substream *capstr; + + capstr = cygaud->portinfo[port].capture_stream; + cygnus_pcm_period_elapsed(capstr); + } + } + + writel(esr_status2, audio_io + ESR2_STATUS_CLR_OFFSET); + writel(esr_status4, audio_io + ESR4_STATUS_CLR_OFFSET); + /* Rearm fullmark logic by writing 1 to the correct bit */ + writel(esr_status4, audio_io + BF_REARM_FULL_MARK_OFFSET); +} + +static irqreturn_t cygnus_dma_irq(int irq, void *data) +{ + u32 r5_status; + struct cygnus_audio *cygaud = data; + + /* + * R5 status bits Description + * 0 ESR0 (playback FIFO interrupt) + * 1 ESR1 (playback rbuf interrupt) + * 2 ESR2 (capture rbuf interrupt) + * 3 ESR3 (Freemark play. interrupt) + * 4 ESR4 (Fullmark capt. interrupt) + */ + r5_status = readl(cygaud->audio + INTH_R5F_STATUS_OFFSET); + + if (!(r5_status & (ANY_PLAYBACK_IRQ | ANY_CAPTURE_IRQ))) + return IRQ_NONE; + + /* If playback interrupt happened */ + if (ANY_PLAYBACK_IRQ & r5_status) { + handle_playback_irq(cygaud); + writel(ANY_PLAYBACK_IRQ & r5_status, + cygaud->audio + INTH_R5F_CLEAR_OFFSET); + } + + /* If capture interrupt happened */ + if (ANY_CAPTURE_IRQ & r5_status) { + handle_capture_irq(cygaud); + writel(ANY_CAPTURE_IRQ & r5_status, + cygaud->audio + INTH_R5F_CLEAR_OFFSET); + } + + return IRQ_HANDLED; +} + +static int cygnus_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct cygnus_aio_port *aio; + int ret; + + aio = cygnus_dai_get_dma_data(substream); + if (!aio) + return -ENODEV; + + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw); + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, PERIOD_BYTES_MIN); + if (ret < 0) + return ret; + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, PERIOD_BYTES_MIN); + if (ret < 0) + return ret; + /* + * Keep track of which substream belongs to which port. + * This info is needed by snd_pcm_period_elapsed() in irq_handler + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aio->play_stream = substream; + else + aio->capture_stream = substream; + + return 0; +} + +static int cygnus_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct cygnus_aio_port *aio; + + aio = cygnus_dai_get_dma_data(substream); + + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aio->play_stream = NULL; + else + aio->capture_stream = NULL; + + if (!aio->play_stream && !aio->capture_stream) + dev_dbg(rtd->cpu_dai->dev, "freed port %d\n", aio->portnum); + + return 0; +} + +static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct cygnus_aio_port *aio; + int ret = 0; + + aio = cygnus_dai_get_dma_data(substream); + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + return ret; +} + +static int cygnus_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct cygnus_aio_port *aio; + + aio = cygnus_dai_get_dma_data(substream); + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int cygnus_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct cygnus_aio_port *aio; + unsigned long bufsize, periodsize; + int ret = 0; + bool is_play; + u32 start; + struct ringbuf_regs *p_rbuf = NULL; + + aio = cygnus_dai_get_dma_data(substream); + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + bufsize = snd_pcm_lib_buffer_bytes(substream); + periodsize = snd_pcm_lib_period_bytes(substream); + + dev_dbg(rtd->cpu_dai->dev, "%s (buf_size %lu) (period_size %lu)\n", + __func__, bufsize, periodsize); + + configure_ringbuf_regs(substream); + + p_rbuf = get_ringbuf(substream); + + start = runtime->dma_addr; + + is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0; + + ringbuf_set_initial(aio->cygaud->audio, p_rbuf, is_play, start, + periodsize, bufsize); + + return ret; +} + +static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + unsigned int res = 0, cur = 0, base = 0; + struct ringbuf_regs *p_rbuf = NULL; + + aio = cygnus_dai_get_dma_data(substream); + + /* + * Get the offset of the current read (for playack) or write + * index (for capture). Report this value back to the asoc framework. + */ + p_rbuf = get_ringbuf(substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cur = readl(aio->cygaud->audio + p_rbuf->rdaddr); + else + cur = readl(aio->cygaud->audio + p_rbuf->wraddr); + + base = readl(aio->cygaud->audio + p_rbuf->baseaddr); + + /* + * Mask off the MSB of the rdaddr,wraddr and baseaddr + * since MSB is not part of the address + */ + res = (cur & 0x7fffffff) - (base & 0x7fffffff); + + return bytes_to_frames(substream->runtime, res); +} + +static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size; + + size = cygnus_pcm_hw.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%zx @ %pK\n", + __func__, size, buf->area); + + if (!buf->area) { + dev_err(rtd->cpu_dai->dev, "%s: dma_alloc failed\n", __func__); + return -ENOMEM; + } + buf->bytes = size; + + return 0; +} + + +static const struct snd_pcm_ops cygnus_pcm_ops = { + .open = cygnus_pcm_open, + .close = cygnus_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = cygnus_pcm_hw_params, + .hw_free = cygnus_pcm_hw_free, + .prepare = cygnus_pcm_prepare, + .trigger = cygnus_pcm_trigger, + .pointer = cygnus_pcm_pointer, +}; + +static void cygnus_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (substream) { + buf = &substream->dma_buffer; + if (buf->area) { + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } + } + + substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + if (substream) { + buf = &substream->dma_buffer; + if (buf->area) { + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } + } +} + +static int cygnus_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &cygnus_dma_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = cygnus_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = cygnus_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) { + cygnus_dma_free_dma_buffers(pcm); + return ret; + } + } + + return 0; +} + +static struct snd_soc_platform_driver cygnus_soc_platform = { + .ops = &cygnus_pcm_ops, + .pcm_new = cygnus_dma_new, + .pcm_free = cygnus_dma_free_dma_buffers, +}; + +int cygnus_soc_platform_register(struct device *dev, + struct cygnus_audio *cygaud) +{ + int rc = 0; + + dev_dbg(dev, "%s Enter\n", __func__); + + rc = devm_request_irq(dev, cygaud->irq_num, cygnus_dma_irq, + IRQF_SHARED, "cygnus-audio", cygaud); + if (rc) { + dev_err(dev, "%s request_irq error %d\n", __func__, rc); + return rc; + } + + rc = snd_soc_register_platform(dev, &cygnus_soc_platform); + if (rc) { + dev_err(dev, "%s failed\n", __func__); + return rc; + } + + return 0; +} + +int cygnus_soc_platform_unregister(struct device *dev) +{ + snd_soc_unregister_platform(dev); + + return 0; +} + +MODULE_LICENSE("GPL v2"); +MODULE_AUTHOR("Broadcom"); +MODULE_DESCRIPTION("Cygnus ASoC PCM module");
On Tue, Mar 29, 2016 at 11:46:32AM -0700, Simran Rai wrote:
From: Simran Rai ssimran@broadcom.com
This patch adds Cygnus audio DMA driver. It supports playback and capture modes and uses ringbuffers for data transfer.
This looks good.
The patch
ASoC: cygnus: Add Cygnus audio DMA driver
has been applied to the asoc tree at
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying to this mail.
Thanks, Mark
From 1200a7d9b2c65ffb2dd673add65cd5dc95671489 Mon Sep 17 00:00:00 2001
From: Simran Rai ssimran@broadcom.com Date: Tue, 17 May 2016 17:01:09 -0700 Subject: [PATCH] ASoC: cygnus: Add Cygnus audio DMA driver
This patch adds Cygnus audio DMA driver. It supports playback and capture modes and uses ringbuffers for data transfer.
Signed-off-by: Lori Hikichi lhikichi@broadcom.com Signed-off-by: Simran Rai ssimran@broadcom.com Reviewed-by: Ray Jui rjui@broadcom.com Reviewed-by: Arun Parameswaran arunp@broadcom.com Reviewed-by: Scott Branden sbranden@broadcom.com Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 875 insertions(+) create mode 100644 sound/soc/bcm/cygnus-pcm.c
diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 6a834e109f1d..d528aaceaad9 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -7,3 +7,12 @@ config SND_BCM2835_SOC_I2S Say Y or M if you want to add support for codecs attached to the BCM2835 I2S interface. You will also need to select the audio interfaces to support below. + +config SND_SOC_CYGNUS + tristate "SoC platform audio for Broadcom Cygnus chips" + depends on ARCH_BCM_CYGNUS || COMPILE_TEST + help + Say Y if you want to add support for ASoC audio on Broadcom + Cygnus chips (bcm958300, bcm958305, bcm911360) + + If you don't know what to do here, say N. \ No newline at end of file diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile index bc816b71e5a4..fc739d007884 100644 --- a/sound/soc/bcm/Makefile +++ b/sound/soc/bcm/Makefile @@ -3,3 +3,8 @@ snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o
obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o
+# CYGNUS Platform Support +snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o + +obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o + diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c new file mode 100644 index 000000000000..d616e096462e --- /dev/null +++ b/sound/soc/bcm/cygnus-pcm.c @@ -0,0 +1,861 @@ +/* + * Copyright (C) 2014-2015 Broadcom Corporation + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; without even the implied warranty + * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#include <linux/debugfs.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/timer.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "cygnus-ssp.h" + +/* Register offset needed for ASoC PCM module */ + +#define INTH_R5F_STATUS_OFFSET 0x040 +#define INTH_R5F_CLEAR_OFFSET 0x048 +#define INTH_R5F_MASK_SET_OFFSET 0x050 +#define INTH_R5F_MASK_CLEAR_OFFSET 0x054 + +#define BF_REARM_FREE_MARK_OFFSET 0x344 +#define BF_REARM_FULL_MARK_OFFSET 0x348 + +/* Ring Buffer Ctrl Regs --- Start */ +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_RDADDR_REG_BASE */ +#define SRC_RBUF_0_RDADDR_OFFSET 0x500 +#define SRC_RBUF_1_RDADDR_OFFSET 0x518 +#define SRC_RBUF_2_RDADDR_OFFSET 0x530 +#define SRC_RBUF_3_RDADDR_OFFSET 0x548 +#define SRC_RBUF_4_RDADDR_OFFSET 0x560 +#define SRC_RBUF_5_RDADDR_OFFSET 0x578 +#define SRC_RBUF_6_RDADDR_OFFSET 0x590 + +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_WRADDR_REG_BASE */ +#define SRC_RBUF_0_WRADDR_OFFSET 0x504 +#define SRC_RBUF_1_WRADDR_OFFSET 0x51c +#define SRC_RBUF_2_WRADDR_OFFSET 0x534 +#define SRC_RBUF_3_WRADDR_OFFSET 0x54c +#define SRC_RBUF_4_WRADDR_OFFSET 0x564 +#define SRC_RBUF_5_WRADDR_OFFSET 0x57c +#define SRC_RBUF_6_WRADDR_OFFSET 0x594 + +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_BASEADDR_REG_BASE */ +#define SRC_RBUF_0_BASEADDR_OFFSET 0x508 +#define SRC_RBUF_1_BASEADDR_OFFSET 0x520 +#define SRC_RBUF_2_BASEADDR_OFFSET 0x538 +#define SRC_RBUF_3_BASEADDR_OFFSET 0x550 +#define SRC_RBUF_4_BASEADDR_OFFSET 0x568 +#define SRC_RBUF_5_BASEADDR_OFFSET 0x580 +#define SRC_RBUF_6_BASEADDR_OFFSET 0x598 + +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_ENDADDR_REG_BASE */ +#define SRC_RBUF_0_ENDADDR_OFFSET 0x50c +#define SRC_RBUF_1_ENDADDR_OFFSET 0x524 +#define SRC_RBUF_2_ENDADDR_OFFSET 0x53c +#define SRC_RBUF_3_ENDADDR_OFFSET 0x554 +#define SRC_RBUF_4_ENDADDR_OFFSET 0x56c +#define SRC_RBUF_5_ENDADDR_OFFSET 0x584 +#define SRC_RBUF_6_ENDADDR_OFFSET 0x59c + +/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_FREE_MARK_REG_BASE */ +#define SRC_RBUF_0_FREE_MARK_OFFSET 0x510 +#define SRC_RBUF_1_FREE_MARK_OFFSET 0x528 +#define SRC_RBUF_2_FREE_MARK_OFFSET 0x540 +#define SRC_RBUF_3_FREE_MARK_OFFSET 0x558 +#define SRC_RBUF_4_FREE_MARK_OFFSET 0x570 +#define SRC_RBUF_5_FREE_MARK_OFFSET 0x588 +#define SRC_RBUF_6_FREE_MARK_OFFSET 0x5a0 + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_RDADDR_REG_BASE */ +#define DST_RBUF_0_RDADDR_OFFSET 0x5c0 +#define DST_RBUF_1_RDADDR_OFFSET 0x5d8 +#define DST_RBUF_2_RDADDR_OFFSET 0x5f0 +#define DST_RBUF_3_RDADDR_OFFSET 0x608 +#define DST_RBUF_4_RDADDR_OFFSET 0x620 +#define DST_RBUF_5_RDADDR_OFFSET 0x638 + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_WRADDR_REG_BASE */ +#define DST_RBUF_0_WRADDR_OFFSET 0x5c4 +#define DST_RBUF_1_WRADDR_OFFSET 0x5dc +#define DST_RBUF_2_WRADDR_OFFSET 0x5f4 +#define DST_RBUF_3_WRADDR_OFFSET 0x60c +#define DST_RBUF_4_WRADDR_OFFSET 0x624 +#define DST_RBUF_5_WRADDR_OFFSET 0x63c + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_BASEADDR_REG_BASE */ +#define DST_RBUF_0_BASEADDR_OFFSET 0x5c8 +#define DST_RBUF_1_BASEADDR_OFFSET 0x5e0 +#define DST_RBUF_2_BASEADDR_OFFSET 0x5f8 +#define DST_RBUF_3_BASEADDR_OFFSET 0x610 +#define DST_RBUF_4_BASEADDR_OFFSET 0x628 +#define DST_RBUF_5_BASEADDR_OFFSET 0x640 + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_ENDADDR_REG_BASE */ +#define DST_RBUF_0_ENDADDR_OFFSET 0x5cc +#define DST_RBUF_1_ENDADDR_OFFSET 0x5e4 +#define DST_RBUF_2_ENDADDR_OFFSET 0x5fc +#define DST_RBUF_3_ENDADDR_OFFSET 0x614 +#define DST_RBUF_4_ENDADDR_OFFSET 0x62c +#define DST_RBUF_5_ENDADDR_OFFSET 0x644 + +/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_FULL_MARK_REG_BASE */ +#define DST_RBUF_0_FULL_MARK_OFFSET 0x5d0 +#define DST_RBUF_1_FULL_MARK_OFFSET 0x5e8 +#define DST_RBUF_2_FULL_MARK_OFFSET 0x600 +#define DST_RBUF_3_FULL_MARK_OFFSET 0x618 +#define DST_RBUF_4_FULL_MARK_OFFSET 0x630 +#define DST_RBUF_5_FULL_MARK_OFFSET 0x648 +/* Ring Buffer Ctrl Regs --- End */ + +/* Error Status Regs --- Start */ +/* AUD_FMM_BF_ESR_ESRX_STATUS_REG_BASE */ +#define ESR0_STATUS_OFFSET 0x900 +#define ESR1_STATUS_OFFSET 0x918 +#define ESR2_STATUS_OFFSET 0x930 +#define ESR3_STATUS_OFFSET 0x948 +#define ESR4_STATUS_OFFSET 0x960 + +/* AUD_FMM_BF_ESR_ESRX_STATUS_CLEAR_REG_BASE */ +#define ESR0_STATUS_CLR_OFFSET 0x908 +#define ESR1_STATUS_CLR_OFFSET 0x920 +#define ESR2_STATUS_CLR_OFFSET 0x938 +#define ESR3_STATUS_CLR_OFFSET 0x950 +#define ESR4_STATUS_CLR_OFFSET 0x968 + +/* AUD_FMM_BF_ESR_ESRX_MASK_REG_BASE */ +#define ESR0_MASK_STATUS_OFFSET 0x90c +#define ESR1_MASK_STATUS_OFFSET 0x924 +#define ESR2_MASK_STATUS_OFFSET 0x93c +#define ESR3_MASK_STATUS_OFFSET 0x954 +#define ESR4_MASK_STATUS_OFFSET 0x96c + +/* AUD_FMM_BF_ESR_ESRX_MASK_SET_REG_BASE */ +#define ESR0_MASK_SET_OFFSET 0x910 +#define ESR1_MASK_SET_OFFSET 0x928 +#define ESR2_MASK_SET_OFFSET 0x940 +#define ESR3_MASK_SET_OFFSET 0x958 +#define ESR4_MASK_SET_OFFSET 0x970 + +/* AUD_FMM_BF_ESR_ESRX_MASK_CLEAR_REG_BASE */ +#define ESR0_MASK_CLR_OFFSET 0x914 +#define ESR1_MASK_CLR_OFFSET 0x92c +#define ESR2_MASK_CLR_OFFSET 0x944 +#define ESR3_MASK_CLR_OFFSET 0x95c +#define ESR4_MASK_CLR_OFFSET 0x974 +/* Error Status Regs --- End */ + +#define R5F_ESR0_SHIFT 0 /* esr0 = fifo underflow */ +#define R5F_ESR1_SHIFT 1 /* esr1 = ringbuf underflow */ +#define R5F_ESR2_SHIFT 2 /* esr2 = ringbuf overflow */ +#define R5F_ESR3_SHIFT 3 /* esr3 = freemark */ +#define R5F_ESR4_SHIFT 4 /* esr4 = fullmark */ + + +/* Mask for R5F register. Set all relevant interrupt for playback handler */ +#define ANY_PLAYBACK_IRQ (BIT(R5F_ESR0_SHIFT) | \ + BIT(R5F_ESR1_SHIFT) | \ + BIT(R5F_ESR3_SHIFT)) + +/* Mask for R5F register. Set all relevant interrupt for capture handler */ +#define ANY_CAPTURE_IRQ (BIT(R5F_ESR2_SHIFT) | BIT(R5F_ESR4_SHIFT)) + +/* + * PERIOD_BYTES_MIN is the number of bytes to at which the interrupt will tick. + * This number should be a multiple of 256. Minimum value is 256 + */ +#define PERIOD_BYTES_MIN 0x100 + +static const struct snd_pcm_hardware cygnus_pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + + /* A period is basically an interrupt */ + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = 0x10000, + + /* period_min/max gives range of approx interrupts per buffer */ + .periods_min = 2, + .periods_max = 8, + + /* + * maximum buffer size in bytes = period_bytes_max * periods_max + * We allocate this amount of data for each enabled channel + */ + .buffer_bytes_max = 4 * 0x8000, +}; + +static u64 cygnus_dma_dmamask = DMA_BIT_MASK(32); + +static struct cygnus_aio_port *cygnus_dai_get_dma_data( + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + + return snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); +} + +static void ringbuf_set_initial(void __iomem *audio_io, + struct ringbuf_regs *p_rbuf, + bool is_playback, + u32 start, + u32 periodsize, + u32 bufsize) +{ + u32 initial_rd; + u32 initial_wr; + u32 end; + u32 fmark_val; /* free or full mark */ + + p_rbuf->period_bytes = periodsize; + p_rbuf->buf_size = bufsize; + + if (is_playback) { + /* Set the pointers to indicate full (flip uppermost bit) */ + initial_rd = start; + initial_wr = initial_rd ^ BIT(31); + } else { + /* Set the pointers to indicate empty */ + initial_wr = start; + initial_rd = initial_wr; + } + + end = start + bufsize - 1; + + /* + * The interrupt will fire when free/full mark is *exceeded* + * The fmark value must be multiple of PERIOD_BYTES_MIN so set fmark + * to be PERIOD_BYTES_MIN less than the period size. + */ + fmark_val = periodsize - PERIOD_BYTES_MIN; + + writel(start, audio_io + p_rbuf->baseaddr); + writel(end, audio_io + p_rbuf->endaddr); + writel(fmark_val, audio_io + p_rbuf->fmark); + writel(initial_rd, audio_io + p_rbuf->rdaddr); + writel(initial_wr, audio_io + p_rbuf->wraddr); +} + +static int configure_ringbuf_regs(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + struct ringbuf_regs *p_rbuf; + int status = 0; + + aio = cygnus_dai_get_dma_data(substream); + + /* Map the ssp portnum to a set of ring buffers. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + p_rbuf = &aio->play_rb_regs; + + switch (aio->portnum) { + case 0: + *p_rbuf = RINGBUF_REG_PLAYBACK(0); + break; + case 1: + *p_rbuf = RINGBUF_REG_PLAYBACK(2); + break; + case 2: + *p_rbuf = RINGBUF_REG_PLAYBACK(4); + break; + case 3: /* SPDIF */ + *p_rbuf = RINGBUF_REG_PLAYBACK(6); + break; + default: + status = -EINVAL; + } + } else { + p_rbuf = &aio->capture_rb_regs; + + switch (aio->portnum) { + case 0: + *p_rbuf = RINGBUF_REG_CAPTURE(0); + break; + case 1: + *p_rbuf = RINGBUF_REG_CAPTURE(2); + break; + case 2: + *p_rbuf = RINGBUF_REG_CAPTURE(4); + break; + default: + status = -EINVAL; + } + } + + return status; +} + +static struct ringbuf_regs *get_ringbuf(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + struct ringbuf_regs *p_rbuf = NULL; + + aio = cygnus_dai_get_dma_data(substream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + p_rbuf = &aio->play_rb_regs; + else + p_rbuf = &aio->capture_rb_regs; + + return p_rbuf; +} + +static void enable_intr(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + u32 clear_mask; + + aio = cygnus_dai_get_dma_data(substream); + + /* The port number maps to the bit position to be cleared */ + clear_mask = BIT(aio->portnum); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Clear interrupt status before enabling them */ + writel(clear_mask, aio->cygaud->audio + ESR0_STATUS_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR1_STATUS_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR3_STATUS_CLR_OFFSET); + /* Unmask the interrupts of the given port*/ + writel(clear_mask, aio->cygaud->audio + ESR0_MASK_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR1_MASK_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR3_MASK_CLR_OFFSET); + + writel(ANY_PLAYBACK_IRQ, + aio->cygaud->audio + INTH_R5F_MASK_CLEAR_OFFSET); + } else { + writel(clear_mask, aio->cygaud->audio + ESR2_STATUS_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR4_STATUS_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR2_MASK_CLR_OFFSET); + writel(clear_mask, aio->cygaud->audio + ESR4_MASK_CLR_OFFSET); + + writel(ANY_CAPTURE_IRQ, + aio->cygaud->audio + INTH_R5F_MASK_CLEAR_OFFSET); + } + +} + +static void disable_intr(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct cygnus_aio_port *aio; + u32 set_mask; + + aio = cygnus_dai_get_dma_data(substream); + + dev_dbg(rtd->cpu_dai->dev, "%s on port %d\n", __func__, aio->portnum); + + /* The port number maps to the bit position to be set */ + set_mask = BIT(aio->portnum); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Mask the interrupts of the given port*/ + writel(set_mask, aio->cygaud->audio + ESR0_MASK_SET_OFFSET); + writel(set_mask, aio->cygaud->audio + ESR1_MASK_SET_OFFSET); + writel(set_mask, aio->cygaud->audio + ESR3_MASK_SET_OFFSET); + } else { + writel(set_mask, aio->cygaud->audio + ESR2_MASK_SET_OFFSET); + writel(set_mask, aio->cygaud->audio + ESR4_MASK_SET_OFFSET); + } + +} + +static int cygnus_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + enable_intr(substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + disable_intr(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static void cygnus_pcm_period_elapsed(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + struct ringbuf_regs *p_rbuf = NULL; + u32 regval; + + aio = cygnus_dai_get_dma_data(substream); + + p_rbuf = get_ringbuf(substream); + + /* + * If free/full mark interrupt occurs, provide timestamp + * to ALSA and update appropriate idx by period_bytes + */ + snd_pcm_period_elapsed(substream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Set the ring buffer to full */ + regval = readl(aio->cygaud->audio + p_rbuf->rdaddr); + regval = regval ^ BIT(31); + writel(regval, aio->cygaud->audio + p_rbuf->wraddr); + } else { + /* Set the ring buffer to empty */ + regval = readl(aio->cygaud->audio + p_rbuf->wraddr); + writel(regval, aio->cygaud->audio + p_rbuf->rdaddr); + } +} + +/* + * ESR0/1/3 status Description + * 0x1 I2S0_out port caused interrupt + * 0x2 I2S1_out port caused interrupt + * 0x4 I2S2_out port caused interrupt + * 0x8 SPDIF_out port caused interrupt + */ +static void handle_playback_irq(struct cygnus_audio *cygaud) +{ + void __iomem *audio_io; + u32 port; + u32 esr_status0, esr_status1, esr_status3; + + audio_io = cygaud->audio; + + /* + * ESR status gets updates with/without interrupts enabled. + * So, check the ESR mask, which provides interrupt enable/ + * disable status and use it to determine which ESR status + * should be serviced. + */ + esr_status0 = readl(audio_io + ESR0_STATUS_OFFSET); + esr_status0 &= ~readl(audio_io + ESR0_MASK_STATUS_OFFSET); + esr_status1 = readl(audio_io + ESR1_STATUS_OFFSET); + esr_status1 &= ~readl(audio_io + ESR1_MASK_STATUS_OFFSET); + esr_status3 = readl(audio_io + ESR3_STATUS_OFFSET); + esr_status3 &= ~readl(audio_io + ESR3_MASK_STATUS_OFFSET); + + for (port = 0; port < CYGNUS_MAX_PLAYBACK_PORTS; port++) { + u32 esrmask = BIT(port); + + /* + * Ringbuffer or FIFO underflow + * If we get this interrupt then, it is also true that we have + * not yet responded to the freemark interrupt. + * Log a debug message. The freemark handler below will + * handle getting everything going again. + */ + if ((esrmask & esr_status1) || (esrmask & esr_status0)) { + dev_dbg(cygaud->dev, + "Underrun: esr0=0x%x, esr1=0x%x esr3=0x%x\n", + esr_status0, esr_status1, esr_status3); + } + + /* + * Freemark is hit. This is the normal interrupt. + * In typical operation the read and write regs will be equal + */ + if (esrmask & esr_status3) { + struct snd_pcm_substream *playstr; + + playstr = cygaud->portinfo[port].play_stream; + cygnus_pcm_period_elapsed(playstr); + } + } + + /* Clear ESR interrupt */ + writel(esr_status0, audio_io + ESR0_STATUS_CLR_OFFSET); + writel(esr_status1, audio_io + ESR1_STATUS_CLR_OFFSET); + writel(esr_status3, audio_io + ESR3_STATUS_CLR_OFFSET); + /* Rearm freemark logic by writing 1 to the correct bit */ + writel(esr_status3, audio_io + BF_REARM_FREE_MARK_OFFSET); +} + +/* + * ESR2/4 status Description + * 0x1 I2S0_in port caused interrupt + * 0x2 I2S1_in port caused interrupt + * 0x4 I2S2_in port caused interrupt + */ +static void handle_capture_irq(struct cygnus_audio *cygaud) +{ + void __iomem *audio_io; + u32 port; + u32 esr_status2, esr_status4; + + audio_io = cygaud->audio; + + /* + * ESR status gets updates with/without interrupts enabled. + * So, check the ESR mask, which provides interrupt enable/ + * disable status and use it to determine which ESR status + * should be serviced. + */ + esr_status2 = readl(audio_io + ESR2_STATUS_OFFSET); + esr_status2 &= ~readl(audio_io + ESR2_MASK_STATUS_OFFSET); + esr_status4 = readl(audio_io + ESR4_STATUS_OFFSET); + esr_status4 &= ~readl(audio_io + ESR4_MASK_STATUS_OFFSET); + + for (port = 0; port < CYGNUS_MAX_CAPTURE_PORTS; port++) { + u32 esrmask = BIT(port); + + /* + * Ringbuffer or FIFO overflow + * If we get this interrupt then, it is also true that we have + * not yet responded to the fullmark interrupt. + * Log a debug message. The fullmark handler below will + * handle getting everything going again. + */ + if (esrmask & esr_status2) + dev_dbg(cygaud->dev, + "Overflow: esr2=0x%x\n", esr_status2); + + if (esrmask & esr_status4) { + struct snd_pcm_substream *capstr; + + capstr = cygaud->portinfo[port].capture_stream; + cygnus_pcm_period_elapsed(capstr); + } + } + + writel(esr_status2, audio_io + ESR2_STATUS_CLR_OFFSET); + writel(esr_status4, audio_io + ESR4_STATUS_CLR_OFFSET); + /* Rearm fullmark logic by writing 1 to the correct bit */ + writel(esr_status4, audio_io + BF_REARM_FULL_MARK_OFFSET); +} + +static irqreturn_t cygnus_dma_irq(int irq, void *data) +{ + u32 r5_status; + struct cygnus_audio *cygaud = data; + + /* + * R5 status bits Description + * 0 ESR0 (playback FIFO interrupt) + * 1 ESR1 (playback rbuf interrupt) + * 2 ESR2 (capture rbuf interrupt) + * 3 ESR3 (Freemark play. interrupt) + * 4 ESR4 (Fullmark capt. interrupt) + */ + r5_status = readl(cygaud->audio + INTH_R5F_STATUS_OFFSET); + + if (!(r5_status & (ANY_PLAYBACK_IRQ | ANY_CAPTURE_IRQ))) + return IRQ_NONE; + + /* If playback interrupt happened */ + if (ANY_PLAYBACK_IRQ & r5_status) { + handle_playback_irq(cygaud); + writel(ANY_PLAYBACK_IRQ & r5_status, + cygaud->audio + INTH_R5F_CLEAR_OFFSET); + } + + /* If capture interrupt happened */ + if (ANY_CAPTURE_IRQ & r5_status) { + handle_capture_irq(cygaud); + writel(ANY_CAPTURE_IRQ & r5_status, + cygaud->audio + INTH_R5F_CLEAR_OFFSET); + } + + return IRQ_HANDLED; +} + +static int cygnus_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct cygnus_aio_port *aio; + int ret; + + aio = cygnus_dai_get_dma_data(substream); + if (!aio) + return -ENODEV; + + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw); + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, PERIOD_BYTES_MIN); + if (ret < 0) + return ret; + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, PERIOD_BYTES_MIN); + if (ret < 0) + return ret; + /* + * Keep track of which substream belongs to which port. + * This info is needed by snd_pcm_period_elapsed() in irq_handler + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aio->play_stream = substream; + else + aio->capture_stream = substream; + + return 0; +} + +static int cygnus_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct cygnus_aio_port *aio; + + aio = cygnus_dai_get_dma_data(substream); + + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aio->play_stream = NULL; + else + aio->capture_stream = NULL; + + if (!aio->play_stream && !aio->capture_stream) + dev_dbg(rtd->cpu_dai->dev, "freed port %d\n", aio->portnum); + + return 0; +} + +static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct cygnus_aio_port *aio; + int ret = 0; + + aio = cygnus_dai_get_dma_data(substream); + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + return ret; +} + +static int cygnus_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct cygnus_aio_port *aio; + + aio = cygnus_dai_get_dma_data(substream); + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int cygnus_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct cygnus_aio_port *aio; + unsigned long bufsize, periodsize; + int ret = 0; + bool is_play; + u32 start; + struct ringbuf_regs *p_rbuf = NULL; + + aio = cygnus_dai_get_dma_data(substream); + dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + + bufsize = snd_pcm_lib_buffer_bytes(substream); + periodsize = snd_pcm_lib_period_bytes(substream); + + dev_dbg(rtd->cpu_dai->dev, "%s (buf_size %lu) (period_size %lu)\n", + __func__, bufsize, periodsize); + + configure_ringbuf_regs(substream); + + p_rbuf = get_ringbuf(substream); + + start = runtime->dma_addr; + + is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0; + + ringbuf_set_initial(aio->cygaud->audio, p_rbuf, is_play, start, + periodsize, bufsize); + + return ret; +} + +static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct cygnus_aio_port *aio; + unsigned int res = 0, cur = 0, base = 0; + struct ringbuf_regs *p_rbuf = NULL; + + aio = cygnus_dai_get_dma_data(substream); + + /* + * Get the offset of the current read (for playack) or write + * index (for capture). Report this value back to the asoc framework. + */ + p_rbuf = get_ringbuf(substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cur = readl(aio->cygaud->audio + p_rbuf->rdaddr); + else + cur = readl(aio->cygaud->audio + p_rbuf->wraddr); + + base = readl(aio->cygaud->audio + p_rbuf->baseaddr); + + /* + * Mask off the MSB of the rdaddr,wraddr and baseaddr + * since MSB is not part of the address + */ + res = (cur & 0x7fffffff) - (base & 0x7fffffff); + + return bytes_to_frames(substream->runtime, res); +} + +static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size; + + size = cygnus_pcm_hw.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%zx @ %pK\n", + __func__, size, buf->area); + + if (!buf->area) { + dev_err(rtd->cpu_dai->dev, "%s: dma_alloc failed\n", __func__); + return -ENOMEM; + } + buf->bytes = size; + + return 0; +} + + +static const struct snd_pcm_ops cygnus_pcm_ops = { + .open = cygnus_pcm_open, + .close = cygnus_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = cygnus_pcm_hw_params, + .hw_free = cygnus_pcm_hw_free, + .prepare = cygnus_pcm_prepare, + .trigger = cygnus_pcm_trigger, + .pointer = cygnus_pcm_pointer, +}; + +static void cygnus_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (substream) { + buf = &substream->dma_buffer; + if (buf->area) { + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } + } + + substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + if (substream) { + buf = &substream->dma_buffer; + if (buf->area) { + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } + } +} + +static int cygnus_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &cygnus_dma_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = cygnus_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = cygnus_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) { + cygnus_dma_free_dma_buffers(pcm); + return ret; + } + } + + return 0; +} + +static struct snd_soc_platform_driver cygnus_soc_platform = { + .ops = &cygnus_pcm_ops, + .pcm_new = cygnus_dma_new, + .pcm_free = cygnus_dma_free_dma_buffers, +}; + +int cygnus_soc_platform_register(struct device *dev, + struct cygnus_audio *cygaud) +{ + int rc = 0; + + dev_dbg(dev, "%s Enter\n", __func__); + + rc = devm_request_irq(dev, cygaud->irq_num, cygnus_dma_irq, + IRQF_SHARED, "cygnus-audio", cygaud); + if (rc) { + dev_err(dev, "%s request_irq error %d\n", __func__, rc); + return rc; + } + + rc = snd_soc_register_platform(dev, &cygnus_soc_platform); + if (rc) { + dev_err(dev, "%s failed\n", __func__); + return rc; + } + + return 0; +} + +int cygnus_soc_platform_unregister(struct device *dev) +{ + snd_soc_unregister_platform(dev); + + return 0; +} + +MODULE_LICENSE("GPL v2"); +MODULE_AUTHOR("Broadcom"); +MODULE_DESCRIPTION("Cygnus ASoC PCM module");
On Tue, Mar 29, 2016 at 11:46:29AM -0700, Simran Rai wrote:
Changes from v4:
- Fix power suspend function and add power resume function
- Remove clock initialization code from audio driver to clock framework
I can't find any sign of anything since your v4 having been submitted previously so this isn't a resend (and including noise like that in the subject line just means less space for content anyway).
On 29/03/16 12:09, Mark Brown wrote:
On Tue, Mar 29, 2016 at 11:46:29AM -0700, Simran Rai wrote:
Changes from v4:
- Fix power suspend function and add power resume function
- Remove clock initialization code from audio driver to clock framework
I can't find any sign of anything since your v4 having been submitted previously so this isn't a resend (and including noise like that in the subject line just means less space for content anyway).
Subject apart, and thanks for the education, is there something wrong with this patch series? -- Florian
On Tue, Mar 29, 2016 at 12:20:25PM -0700, Florian Fainelli wrote:
Subject apart, and thanks for the education, is there something wrong with this patch series?
I've not yet reviewed it, I was just doing first pass triage and was about to complain that my review comments had been ignored since it was flagged as a resend but there were issues with the last version. It's fortunate that I actually went to look for that version to paste my comments back in...
On Wed, 30 Mar 2016 17:51:40 +0200, Mark Brown wrote:
On Wed, Mar 30, 2016 at 05:35:14PM +0200, Takashi Iwai wrote:
Empty mail?
Sorry, discard it. Just finger slipped.
Takashi
participants (5)
-
Florian Fainelli
-
Mark Brown
-
Rob Herring
-
Simran Rai
-
Takashi Iwai