mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-12-27 00:25:21 +07:00
dc3bf49ea3
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>
180 lines
6.4 KiB
C
180 lines
6.4 KiB
C
/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
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/*
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* This file is provided under a dual BSD/GPLv2 license. When using or
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* redistributing this file, you may do so under either license.
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*
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* Copyright(c) 2018 Intel Corporation. All rights reserved.
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*/
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#ifndef __INCLUDE_SOUND_SOF_DAI_INTEL_H__
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#define __INCLUDE_SOUND_SOF_DAI_INTEL_H__
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#include <sound/sof/header.h>
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/* ssc1: TINTE */
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#define SOF_DAI_INTEL_SSP_QUIRK_TINTE (1 << 0)
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/* ssc1: PINTE */
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#define SOF_DAI_INTEL_SSP_QUIRK_PINTE (1 << 1)
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/* ssc2: SMTATF */
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#define SOF_DAI_INTEL_SSP_QUIRK_SMTATF (1 << 2)
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/* ssc2: MMRATF */
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#define SOF_DAI_INTEL_SSP_QUIRK_MMRATF (1 << 3)
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/* ssc2: PSPSTWFDFD */
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#define SOF_DAI_INTEL_SSP_QUIRK_PSPSTWFDFD (1 << 4)
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/* ssc2: PSPSRWFDFD */
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#define SOF_DAI_INTEL_SSP_QUIRK_PSPSRWFDFD (1 << 5)
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/* ssc1: LBM */
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#define SOF_DAI_INTEL_SSP_QUIRK_LBM (1 << 6)
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/* here is the possibility to define others aux macros */
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#define SOF_DAI_INTEL_SSP_FRAME_PULSE_WIDTH_MAX 38
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#define SOF_DAI_INTEL_SSP_SLOT_PADDING_MAX 31
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/* SSP clocks control settings
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*
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* Macros for clks_control field in sof_ipc_dai_ssp_params struct.
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*/
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/* mclk 0 disable */
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#define SOF_DAI_INTEL_SSP_MCLK_0_DISABLE BIT(0)
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/* mclk 1 disable */
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#define SOF_DAI_INTEL_SSP_MCLK_1_DISABLE BIT(1)
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/* mclk keep active */
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#define SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_KA BIT(2)
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/* bclk keep active */
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#define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_KA BIT(3)
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/* fs keep active */
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#define SOF_DAI_INTEL_SSP_CLKCTRL_FS_KA BIT(4)
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/* bclk idle */
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#define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_IDLE_HIGH BIT(5)
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/* SSP Configuration Request - SOF_IPC_DAI_SSP_CONFIG */
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struct sof_ipc_dai_ssp_params {
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struct sof_ipc_hdr hdr;
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uint16_t reserved1;
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uint16_t mclk_id;
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uint32_t mclk_rate; /* mclk frequency in Hz */
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uint32_t fsync_rate; /* fsync frequency in Hz */
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uint32_t bclk_rate; /* bclk frequency in Hz */
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/* TDM */
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uint32_t tdm_slots;
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uint32_t rx_slots;
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uint32_t tx_slots;
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/* data */
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uint32_t sample_valid_bits;
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uint16_t tdm_slot_width;
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uint16_t reserved2; /* alignment */
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/* MCLK */
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uint32_t mclk_direction;
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uint16_t frame_pulse_width;
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uint16_t tdm_per_slot_padding_flag;
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uint32_t clks_control;
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uint32_t quirks;
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} __packed;
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/* HDA Configuration Request - SOF_IPC_DAI_HDA_CONFIG */
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struct sof_ipc_dai_hda_params {
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struct sof_ipc_hdr hdr;
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uint32_t link_dma_ch;
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} __packed;
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/* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */
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/* This struct is defined per 2ch PDM controller available in the platform.
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* Normally it is sufficient to set the used microphone specific enables to 1
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* and keep other parameters as zero. The customizations are:
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*
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* 1. If a device mixes different microphones types with different polarity
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* and/or the absolute polarity matters the PCM signal from a microphone
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* can be inverted with the controls.
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*
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* 2. If the microphones in a stereo pair do not appear in captured stream
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* in desired order due to board schematics choises they can be swapped with
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* the clk_edge parameter.
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*
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* 3. If PDM bit errors are seen in capture (poor quality) the skew parameter
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* that delays the sampling time of data by half cycles of DMIC source clock
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* can be tried for improvement. However there is no guarantee for this to fix
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* data integrity problems.
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*/
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struct sof_ipc_dai_dmic_pdm_ctrl {
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struct sof_ipc_hdr hdr;
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uint16_t id; /**< PDM controller ID */
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uint16_t enable_mic_a; /**< Use A (left) channel mic (0 or 1)*/
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uint16_t enable_mic_b; /**< Use B (right) channel mic (0 or 1)*/
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uint16_t polarity_mic_a; /**< Optionally invert mic A signal (0 or 1) */
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uint16_t polarity_mic_b; /**< Optionally invert mic B signal (0 or 1) */
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uint16_t clk_edge; /**< Optionally swap data clock edge (0 or 1) */
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uint16_t skew; /**< Adjust PDM data sampling vs. clock (0..15) */
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uint16_t reserved[3]; /**< Make sure the total size is 4 bytes aligned */
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} __packed;
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/* This struct contains the global settings for all 2ch PDM controllers. The
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* version number used in configuration data is checked vs. version used by
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* device driver src/drivers/dmic.c need to match. It is incremented from
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* initial value 1 if updates done for the to driver would alter the operation
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* of the microhone.
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*
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* Note: The microphone clock (pdmclk_min, pdmclk_max, duty_min, duty_max)
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* parameters need to be set as defined in microphone data sheet. E.g. clock
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* range 1.0 - 3.2 MHz is usually supported microphones. Some microphones are
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* multi-mode capable and there may be denied mic clock frequencies between
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* the modes. In such case set the clock range limits of the desired mode to
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* avoid the driver to set clock to an illegal rate.
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*
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* The duty cycle could be set to 48-52% if not known. Generally these
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* parameters can be altered within data sheet specified limits to match
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* required audio application performance power.
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*
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* The microphone clock needs to be usually about 50-80 times the used audio
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* sample rate. With highest sample rates above 48 kHz this can relaxed
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* somewhat.
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*
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* The parameter wake_up_time describes how long time the microphone needs
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* for the data line to produce valid output from mic clock start. The driver
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* will mute the captured audio for the given time. The min_clock_on_time
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* parameter is used to prevent too short clock bursts to happen. The driver
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* will keep the clock active after capture stop if this time is not yet
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* met. The unit for both is microseconds (us). Exceed of 100 ms will be
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* treated as an error.
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*/
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struct sof_ipc_dai_dmic_params {
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struct sof_ipc_hdr hdr;
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uint32_t driver_ipc_version; /**< Version (1..N) */
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uint32_t pdmclk_min; /**< Minimum microphone clock in Hz (100000..N) */
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uint32_t pdmclk_max; /**< Maximum microphone clock in Hz (min...N) */
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uint32_t fifo_fs; /**< FIFO sample rate in Hz (8000..96000) */
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uint32_t reserved_1; /**< Reserved */
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uint16_t fifo_bits; /**< FIFO word length (16 or 32) */
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uint16_t reserved_2; /**< Reserved */
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uint16_t duty_min; /**< Min. mic clock duty cycle in % (20..80) */
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uint16_t duty_max; /**< Max. mic clock duty cycle in % (min..80) */
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uint32_t num_pdm_active; /**< Number of active pdm controllers */
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uint32_t wake_up_time; /**< Time from clock start to data (us) */
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uint32_t min_clock_on_time; /**< Min. time that clk is kept on (us) */
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uint32_t unmute_ramp_time; /**< Length of logarithmic gain ramp (ms) */
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/* reserved for future use */
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uint32_t reserved[5];
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/**< variable number of pdm controller config */
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struct sof_ipc_dai_dmic_pdm_ctrl pdm[0];
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} __packed;
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#endif
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