linux/include/sound/sof/dai-intel.h
Masahiro Yamada dc3bf49ea3 treewide: remove SPDX "WITH Linux-syscall-note" from kernel-space headers again
The "WITH Linux-syscall-note" exception exists for headers exported to
user space. It is strange to add it to non-exported headers.

Commit 687a3e4d8e ("treewide: remove SPDX "WITH Linux-syscall-note"
from kernel-space headers") did cleanups some months ago, but it looks
like we need to do this periodically.

This patch was generated by the following script:

  git grep -l -e Linux-syscall-note \
    -- :*.h :^arch/*/include/uapi/asm/*.h :^include/uapi/ :^tools |
  while read file
  do
          sed -i -e 's/(\(GPL-[^[:space:]]*\) WITH Linux-syscall-note)/\1/g' \
          -e 's/ WITH Linux-syscall-note//g' $file
  done

I did not commit drivers/staging/android/uapi/ion.h . This header is
not currently exported, but somebody may plan to move it to include/uapi/
when the time comes. I am not sure. Anyway, it will be better to check
the license inconsistency in drivers/staging/android/uapi/.

Signed-off-by: Masahiro Yamada <yamada.masahiro@socionext.com>
Reviewed-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-07-25 11:05:10 +02:00

180 lines
6.4 KiB
C

/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
/*
* This file is provided under a dual BSD/GPLv2 license. When using or
* redistributing this file, you may do so under either license.
*
* Copyright(c) 2018 Intel Corporation. All rights reserved.
*/
#ifndef __INCLUDE_SOUND_SOF_DAI_INTEL_H__
#define __INCLUDE_SOUND_SOF_DAI_INTEL_H__
#include <sound/sof/header.h>
/* ssc1: TINTE */
#define SOF_DAI_INTEL_SSP_QUIRK_TINTE (1 << 0)
/* ssc1: PINTE */
#define SOF_DAI_INTEL_SSP_QUIRK_PINTE (1 << 1)
/* ssc2: SMTATF */
#define SOF_DAI_INTEL_SSP_QUIRK_SMTATF (1 << 2)
/* ssc2: MMRATF */
#define SOF_DAI_INTEL_SSP_QUIRK_MMRATF (1 << 3)
/* ssc2: PSPSTWFDFD */
#define SOF_DAI_INTEL_SSP_QUIRK_PSPSTWFDFD (1 << 4)
/* ssc2: PSPSRWFDFD */
#define SOF_DAI_INTEL_SSP_QUIRK_PSPSRWFDFD (1 << 5)
/* ssc1: LBM */
#define SOF_DAI_INTEL_SSP_QUIRK_LBM (1 << 6)
/* here is the possibility to define others aux macros */
#define SOF_DAI_INTEL_SSP_FRAME_PULSE_WIDTH_MAX 38
#define SOF_DAI_INTEL_SSP_SLOT_PADDING_MAX 31
/* SSP clocks control settings
*
* Macros for clks_control field in sof_ipc_dai_ssp_params struct.
*/
/* mclk 0 disable */
#define SOF_DAI_INTEL_SSP_MCLK_0_DISABLE BIT(0)
/* mclk 1 disable */
#define SOF_DAI_INTEL_SSP_MCLK_1_DISABLE BIT(1)
/* mclk keep active */
#define SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_KA BIT(2)
/* bclk keep active */
#define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_KA BIT(3)
/* fs keep active */
#define SOF_DAI_INTEL_SSP_CLKCTRL_FS_KA BIT(4)
/* bclk idle */
#define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_IDLE_HIGH BIT(5)
/* SSP Configuration Request - SOF_IPC_DAI_SSP_CONFIG */
struct sof_ipc_dai_ssp_params {
struct sof_ipc_hdr hdr;
uint16_t reserved1;
uint16_t mclk_id;
uint32_t mclk_rate; /* mclk frequency in Hz */
uint32_t fsync_rate; /* fsync frequency in Hz */
uint32_t bclk_rate; /* bclk frequency in Hz */
/* TDM */
uint32_t tdm_slots;
uint32_t rx_slots;
uint32_t tx_slots;
/* data */
uint32_t sample_valid_bits;
uint16_t tdm_slot_width;
uint16_t reserved2; /* alignment */
/* MCLK */
uint32_t mclk_direction;
uint16_t frame_pulse_width;
uint16_t tdm_per_slot_padding_flag;
uint32_t clks_control;
uint32_t quirks;
} __packed;
/* HDA Configuration Request - SOF_IPC_DAI_HDA_CONFIG */
struct sof_ipc_dai_hda_params {
struct sof_ipc_hdr hdr;
uint32_t link_dma_ch;
} __packed;
/* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */
/* This struct is defined per 2ch PDM controller available in the platform.
* Normally it is sufficient to set the used microphone specific enables to 1
* and keep other parameters as zero. The customizations are:
*
* 1. If a device mixes different microphones types with different polarity
* and/or the absolute polarity matters the PCM signal from a microphone
* can be inverted with the controls.
*
* 2. If the microphones in a stereo pair do not appear in captured stream
* in desired order due to board schematics choises they can be swapped with
* the clk_edge parameter.
*
* 3. If PDM bit errors are seen in capture (poor quality) the skew parameter
* that delays the sampling time of data by half cycles of DMIC source clock
* can be tried for improvement. However there is no guarantee for this to fix
* data integrity problems.
*/
struct sof_ipc_dai_dmic_pdm_ctrl {
struct sof_ipc_hdr hdr;
uint16_t id; /**< PDM controller ID */
uint16_t enable_mic_a; /**< Use A (left) channel mic (0 or 1)*/
uint16_t enable_mic_b; /**< Use B (right) channel mic (0 or 1)*/
uint16_t polarity_mic_a; /**< Optionally invert mic A signal (0 or 1) */
uint16_t polarity_mic_b; /**< Optionally invert mic B signal (0 or 1) */
uint16_t clk_edge; /**< Optionally swap data clock edge (0 or 1) */
uint16_t skew; /**< Adjust PDM data sampling vs. clock (0..15) */
uint16_t reserved[3]; /**< Make sure the total size is 4 bytes aligned */
} __packed;
/* This struct contains the global settings for all 2ch PDM controllers. The
* version number used in configuration data is checked vs. version used by
* device driver src/drivers/dmic.c need to match. It is incremented from
* initial value 1 if updates done for the to driver would alter the operation
* of the microhone.
*
* Note: The microphone clock (pdmclk_min, pdmclk_max, duty_min, duty_max)
* parameters need to be set as defined in microphone data sheet. E.g. clock
* range 1.0 - 3.2 MHz is usually supported microphones. Some microphones are
* multi-mode capable and there may be denied mic clock frequencies between
* the modes. In such case set the clock range limits of the desired mode to
* avoid the driver to set clock to an illegal rate.
*
* The duty cycle could be set to 48-52% if not known. Generally these
* parameters can be altered within data sheet specified limits to match
* required audio application performance power.
*
* The microphone clock needs to be usually about 50-80 times the used audio
* sample rate. With highest sample rates above 48 kHz this can relaxed
* somewhat.
*
* The parameter wake_up_time describes how long time the microphone needs
* for the data line to produce valid output from mic clock start. The driver
* will mute the captured audio for the given time. The min_clock_on_time
* parameter is used to prevent too short clock bursts to happen. The driver
* will keep the clock active after capture stop if this time is not yet
* met. The unit for both is microseconds (us). Exceed of 100 ms will be
* treated as an error.
*/
struct sof_ipc_dai_dmic_params {
struct sof_ipc_hdr hdr;
uint32_t driver_ipc_version; /**< Version (1..N) */
uint32_t pdmclk_min; /**< Minimum microphone clock in Hz (100000..N) */
uint32_t pdmclk_max; /**< Maximum microphone clock in Hz (min...N) */
uint32_t fifo_fs; /**< FIFO sample rate in Hz (8000..96000) */
uint32_t reserved_1; /**< Reserved */
uint16_t fifo_bits; /**< FIFO word length (16 or 32) */
uint16_t reserved_2; /**< Reserved */
uint16_t duty_min; /**< Min. mic clock duty cycle in % (20..80) */
uint16_t duty_max; /**< Max. mic clock duty cycle in % (min..80) */
uint32_t num_pdm_active; /**< Number of active pdm controllers */
uint32_t wake_up_time; /**< Time from clock start to data (us) */
uint32_t min_clock_on_time; /**< Min. time that clk is kept on (us) */
uint32_t unmute_ramp_time; /**< Length of logarithmic gain ramp (ms) */
/* reserved for future use */
uint32_t reserved[5];
/**< variable number of pdm controller config */
struct sof_ipc_dai_dmic_pdm_ctrl pdm[0];
} __packed;
#endif