From 168eed447129899611098219b70ef97b605bc6e1 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 2 Nov 2021 12:10:17 +0200 Subject: ASoC: SOF: IPC: Add new IPC command to free trace DMA Add a new SOF_IPC_TRACE_DMA_FREE IPC command to stop and free trace DMA in the FW. Signed-off-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20211102101019.14037-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/header.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 4c747c52e01b..b97a76bcb655 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -119,6 +119,7 @@ #define SOF_IPC_TRACE_DMA_POSITION SOF_CMD_TYPE(0x002) #define SOF_IPC_TRACE_DMA_PARAMS_EXT SOF_CMD_TYPE(0x003) #define SOF_IPC_TRACE_FILTER_UPDATE SOF_CMD_TYPE(0x004) /**< ABI3.17 */ +#define SOF_IPC_TRACE_DMA_FREE SOF_CMD_TYPE(0x005) /**< ABI3.20 */ /* debug */ #define SOF_IPC_DEBUG_MEM_USAGE SOF_CMD_TYPE(0x001) -- cgit v1.2.3 From 749303055b78bc38ec0790ccc596cae235446367 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 15 Nov 2021 12:02:15 +0000 Subject: firmware: cs_dsp: tidy includes in cs_dsp.c and cs_dsp.h This patch removes unused included header files and moves others into cs_dsp.h to ensure that types referenced in the header file are properly described to prevent compiler warnings. Signed-off-by: Simon Trimmer Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20211115120215.56824-1-simont@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/cs_dsp.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include') diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 9ad9eaaaa552..3a54b1afc48f 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -11,6 +11,11 @@ #ifndef __CS_DSP_H #define __CS_DSP_H +#include +#include +#include +#include + #define CS_ADSP2_REGION_0 BIT(0) #define CS_ADSP2_REGION_1 BIT(1) #define CS_ADSP2_REGION_2 BIT(2) -- cgit v1.2.3 From 7206998f578d5553989bc01ea2e544b622e79539 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 08:24:59 +0100 Subject: ALSA: hda: Fix potential deadlock at codec unbinding When a codec is unbound dynamically via sysfs while its stream is in use, we may face a potential deadlock at the proc remove or a UAF. This happens since the hda_pcm is managed by a linked list, as it handles the hda_pcm object release via kref. When a PCM is opened at the unbinding time, the release of hda_pcm gets delayed and it ends up with the close of the PCM stream releasing the associated hda_pcm object of its own. The hda_pcm destructor contains the PCM device release that includes the removal of procfs entries. And, this removal has the sync of the close of all in-use files -- which would never finish because it's called from the PCM file descriptor itself, i.e. it's trying to shoot its foot. For addressing the deadlock above, this patch changes the way to manage and release the hda_pcm object. The kref of hda_pcm is dropped, and instead a simple refcount is introduced in hda_codec for keeping the track of the active PCM streams, and at each PCM open and close, this refcount is adjusted accordingly. At unbinding, the driver calls snd_device_disconnect() for each PCM stream, then synchronizes with the refcount finish, and finally releases the object resources. Fixes: bbbc7e8502c9 ("ALSA: hda - Allocate hda_pcm objects dynamically") Link: https://lore.kernel.org/r/20211116072459.18930-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/hda_codec.h | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 0e45963bb767..82d9daa17851 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -8,7 +8,7 @@ #ifndef __SOUND_HDA_CODEC_H #define __SOUND_HDA_CODEC_H -#include +#include #include #include #include @@ -166,8 +166,8 @@ struct hda_pcm { bool own_chmap; /* codec driver provides own channel maps */ /* private: */ struct hda_codec *codec; - struct kref kref; struct list_head list; + unsigned int disconnected:1; }; /* codec information */ @@ -187,6 +187,8 @@ struct hda_codec { /* PCM to create, set by patch_ops.build_pcms callback */ struct list_head pcm_list_head; + refcount_t pcm_ref; + wait_queue_head_t remove_sleep; /* codec specific info */ void *spec; @@ -420,7 +422,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm) { - kref_get(&pcm->kref); + refcount_inc(&pcm->codec->pcm_ref); } void snd_hda_codec_pcm_put(struct hda_pcm *pcm); -- cgit v1.2.3 From 2c95b92ecd92e784785b1db8cccc4f0f2bfa850c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 08:33:58 +0100 Subject: ALSA: memalloc: Unify x86 SG-buffer handling (take#3) This is a second attempt to unify the x86-specific SG-buffer handling code with the new standard non-contiguous page handler. The first try (in commit 2d9ea39917a4) failed due to the wrong page and address calculations, hence reverted. (And the second try failed due to a copy&paste error.) Now it's corrected with the previous fix for noncontig pages, and the proper sg page iteration by this patch. After the migration, SNDRV_DMA_TYPE_DMA_SG becomes identical with SNDRV_DMA_TYPE_NONCONTIG on x86, while others still fall back to SNDRV_DMA_TYPE_DEV. Tested-by: Alex Xu (Hello71) Tested-by: Harald Arnesen Link: https://lore.kernel.org/r/20211017074859.24112-4-tiwai@suse.de Link: https://lore.kernel.org/r/20211109062235.22310-1-tiwai@suse.de Link: https://lore.kernel.org/r/20211116073358.19741-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'include') diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 1051b84e8579..653dfffb3ac8 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -36,13 +36,6 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ #define SNDRV_DMA_TYPE_DEV_WC 5 /* continuous write-combined */ -#ifdef CONFIG_SND_DMA_SGBUF -#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ -#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ -#else -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ -#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC -#endif #ifdef CONFIG_GENERIC_ALLOCATOR #define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ #else @@ -51,6 +44,13 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_VMALLOC 7 /* vmalloc'ed buffer */ #define SNDRV_DMA_TYPE_NONCONTIG 8 /* non-coherent SG buffer */ #define SNDRV_DMA_TYPE_NONCOHERENT 9 /* non-coherent buffer */ +#ifdef CONFIG_SND_DMA_SGBUF +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_NONCONTIG +#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC +#endif /* * info for buffer allocation -- cgit v1.2.3 From efb931cdc4b94a0f7ed17a76844f08cef1bdffe5 Mon Sep 17 00:00:00 2001 From: Ajit Kumar Pandey Date: Wed, 17 Nov 2021 11:37:24 +0200 Subject: ASoC: SOF: topology: Add support for AMD ACP DAIs Add new sof dais and config to pass topology file configuration to SOF firmware running on ACP's DSP core. ACP firmware support I2S_BT, I2S_SP and DMIC controller hence add three new dais to the list of supported sof_dais Signed-off-by: Ajit Kumar Pandey Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Signed-off-by: Daniel Baluta Link: https://lore.kernel.org/r/20211117093734.17407-12-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- include/sound/sof/dai-amd.h | 21 +++++++++++++++++++++ include/sound/sof/dai.h | 7 +++++++ 2 files changed, 28 insertions(+) create mode 100644 include/sound/sof/dai-amd.h (limited to 'include') diff --git a/include/sound/sof/dai-amd.h b/include/sound/sof/dai-amd.h new file mode 100644 index 000000000000..90d09dbdd709 --- /dev/null +++ b/include/sound/sof/dai-amd.h @@ -0,0 +1,21 @@ +/* SPDX-License-Identifier: (GPL-2.0-only 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) 2021 Advanced Micro Devices, Inc.. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_DAI_AMD_H__ +#define __INCLUDE_SOUND_SOF_DAI_AMD_H__ + +#include + +/* ACP Configuration Request - SOF_IPC_DAI_AMD_CONFIG */ +struct sof_ipc_dai_acp_params { + struct sof_ipc_hdr hdr; + + uint32_t fsync_rate; /* FSYNC frequency in Hz */ + uint32_t tdm_slots; +} __packed; +#endif diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 9625f47557b8..3782127a7095 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -12,6 +12,7 @@ #include #include #include +#include /* * DAI Configuration. @@ -66,6 +67,9 @@ enum sof_ipc_dai_type { SOF_DAI_INTEL_ALH, /**< Intel ALH */ SOF_DAI_IMX_SAI, /**< i.MX SAI */ SOF_DAI_IMX_ESAI, /**< i.MX ESAI */ + SOF_DAI_AMD_BT, /**< AMD ACP BT*/ + SOF_DAI_AMD_SP, /**< AMD ACP SP */ + SOF_DAI_AMD_DMIC, /**< AMD ACP DMIC */ }; /* general purpose DAI configuration */ @@ -90,6 +94,9 @@ struct sof_ipc_dai_config { struct sof_ipc_dai_alh_params alh; struct sof_ipc_dai_esai_params esai; struct sof_ipc_dai_sai_params sai; + struct sof_ipc_dai_acp_params acpbt; + struct sof_ipc_dai_acp_params acpsp; + struct sof_ipc_dai_acp_params acpdmic; }; } __packed; -- cgit v1.2.3 From 2925748eadc33cba3bded7b69475a1b002b124ac Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 17 Nov 2021 13:22:53 +0000 Subject: firmware: cs_dsp: Add version checks on coefficient loading The firmware coefficient files contain version information that is currently ignored by the cs_dsp code. This information specifies which version of the firmware the coefficient were generated for. Add a check into the code which prints a warning in the case the coefficient and firmware differ in version, in many cases this will be ok but it is not always, so best to let the user know there is a potential issue. Co-authored-by: Simon Trimmer Signed-off-by: Simon Trimmer Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20211117132300.1290-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/cs_dsp.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 3a54b1afc48f..ce54705e2bec 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -54,12 +54,14 @@ struct cs_dsp_region { * struct cs_dsp_alg_region - Describes a logical algorithm region in DSP address space * @list: List node for internal use * @alg: Algorithm id + * @ver: Expected algorithm version * @type: Memory region type * @base: Address of region */ struct cs_dsp_alg_region { struct list_head list; unsigned int alg; + unsigned int ver; int type; unsigned int base; }; -- cgit v1.2.3 From 14055b5a3a23204c4702ae5d3f2a819ee081ce33 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 17 Nov 2021 13:22:54 +0000 Subject: firmware: cs_dsp: Add pre_run callback The code already has a post_run callback, add a matching pre_run callback to the client_ops that is called before execution is started. This callback provides a convenient place for the client code to set DSP controls or hardware that requires configuration before the DSP core actually starts execution. Note that placing this callback before cs_dsp_coeff_sync_controls is important to ensure that any control values are then correctly synced out to the chip. Co-authored-by: Simon Trimmer Signed-off-by: Simon Trimmer Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20211117132300.1290-4-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/cs_dsp.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index ce54705e2bec..0bf849baeaa5 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -187,7 +187,8 @@ struct cs_dsp { * struct cs_dsp_client_ops - client callbacks * @control_add: Called under the pwr_lock when a control is created * @control_remove: Called under the pwr_lock when a control is destroyed - * @post_run: Called under the pwr_lock by cs_dsp_run() + * @pre_run: Called under the pwr_lock by cs_dsp_run() before the core is started + * @post_run: Called under the pwr_lock by cs_dsp_run() after the core is started * @post_stop: Called under the pwr_lock by cs_dsp_stop() * @watchdog_expired: Called when a watchdog expiry is detected * @@ -197,6 +198,7 @@ struct cs_dsp { struct cs_dsp_client_ops { int (*control_add)(struct cs_dsp_coeff_ctl *ctl); void (*control_remove)(struct cs_dsp_coeff_ctl *ctl); + int (*pre_run)(struct cs_dsp *dsp); int (*post_run)(struct cs_dsp *dsp); void (*post_stop)(struct cs_dsp *dsp); void (*watchdog_expired)(struct cs_dsp *dsp); -- cgit v1.2.3 From b329b3d39497a9fdb175d7e4fd77ae7170d5d26c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 17 Nov 2021 13:22:58 +0000 Subject: firmware: cs_dsp: Clarify some kernel doc comments Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20211117132300.1290-8-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/cs_dsp.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 0bf849baeaa5..1ad1b173417a 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -76,8 +76,8 @@ struct cs_dsp_alg_region { * @enabled: Flag indicating whether control is enabled * @list: List node for internal use * @cache: Cached value of the control - * @offset: Offset of control within alg_region - * @len: Length of the cached value + * @offset: Offset of control within alg_region in words + * @len: Length of the cached value in bytes * @set: Flag indicating the value has been written by the user * @flags: Bitfield of WMFW_CTL_FLAG_ control flags defined in wmfw.h * @type: One of the WMFW_CTL_TYPE_ control types defined in wmfw.h -- cgit v1.2.3 From f444da38ac924748de696c393327a44c4b8d727e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 17 Nov 2021 13:22:59 +0000 Subject: firmware: cs_dsp: Add offset to cs_dsp read/write Provide a mechanism to access only part of a control through the cs_dsp interface. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20211117132300.1290-9-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/cs_dsp.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 1ad1b173417a..38b4da3ddfe4 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -232,8 +232,10 @@ void cs_dsp_init_debugfs(struct cs_dsp *dsp, struct dentry *debugfs_root); void cs_dsp_cleanup_debugfs(struct cs_dsp *dsp); int cs_dsp_coeff_write_acked_control(struct cs_dsp_coeff_ctl *ctl, unsigned int event_id); -int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, const void *buf, size_t len); -int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl, void *buf, size_t len); +int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off, + const void *buf, size_t len); +int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off, + void *buf, size_t len); struct cs_dsp_coeff_ctl *cs_dsp_get_ctl(struct cs_dsp *dsp, const char *name, int type, unsigned int alg); -- cgit v1.2.3 From 5c903f64ce97172d63f7591cfa9e37cba58867b2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 17 Nov 2021 13:23:00 +0000 Subject: firmware: cs_dsp: Allow creation of event controls Some firmwares contain controls intended to convey firmware state back to the host. Whilst more infrastructure will probably be needed for these in time, as a first step allow creation of the controls, so said firmwares arn't completely rejected. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20211117132300.1290-10-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/wmfw.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/linux/firmware/cirrus/wmfw.h b/include/linux/firmware/cirrus/wmfw.h index a19bf7c6fc8b..74e5a4f6c13a 100644 --- a/include/linux/firmware/cirrus/wmfw.h +++ b/include/linux/firmware/cirrus/wmfw.h @@ -29,6 +29,7 @@ #define WMFW_CTL_TYPE_ACKED 0x1000 /* acked control */ #define WMFW_CTL_TYPE_HOSTEVENT 0x1001 /* event control */ #define WMFW_CTL_TYPE_HOST_BUFFER 0x1002 /* host buffer pointer */ +#define WMFW_CTL_TYPE_FWEVENT 0x1004 /* firmware event control */ struct wmfw_header { char magic[4]; -- cgit v1.2.3 From e6feefa541f309afed8aa54431681261bc57bcde Mon Sep 17 00:00:00 2001 From: YC Hung Date: Thu, 18 Nov 2021 12:07:43 +0200 Subject: ASoC: SOF: tokens: add token for Mediatek AFE MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add the definition for Mediatek audio front end(AFE) tokens,include AFE sampling rate, channels, and format. Signed-off-by: YC Hung Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Reviewed-by: Guennadi Liakhovetski Reviewed-by: Daniel Baluta Signed-off-by: Daniel Baluta Link: https://lore.kernel.org/r/20211118100749.54628-3-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- include/uapi/sound/sof/tokens.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index 02b71a8deea4..b72fa385bebf 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -140,4 +140,9 @@ #define SOF_TKN_INTEL_HDA_RATE 1500 #define SOF_TKN_INTEL_HDA_CH 1501 +/* AFE */ +#define SOF_TKN_MEDIATEK_AFE_RATE 1600 +#define SOF_TKN_MEDIATEK_AFE_CH 1601 +#define SOF_TKN_MEDIATEK_AFE_FORMAT 1602 + #endif -- cgit v1.2.3 From b72bfcffcfc11858a8fc92998733372606db485e Mon Sep 17 00:00:00 2001 From: YC Hung Date: Thu, 18 Nov 2021 12:07:44 +0200 Subject: ASoC: SOF: topology: Add support for Mediatek AFE DAI MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add new sof dai and config to pass topology file configuration to SOF firmware running on Mediatek platform DSP core. Add mediatek audio front end(AFE) to the list of supported sof_dais Signed-off-by: YC Hung Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Reviewed-by: Guennadi Liakhovetski Reviewed-by: Daniel Baluta Signed-off-by: Daniel Baluta Link: https://lore.kernel.org/r/20211118100749.54628-4-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- include/sound/sof/dai-mediatek.h | 23 +++++++++++++++++++++++ include/sound/sof/dai.h | 3 +++ 2 files changed, 26 insertions(+) create mode 100644 include/sound/sof/dai-mediatek.h (limited to 'include') diff --git a/include/sound/sof/dai-mediatek.h b/include/sound/sof/dai-mediatek.h new file mode 100644 index 000000000000..62dd4720558d --- /dev/null +++ b/include/sound/sof/dai-mediatek.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ +/* + * Copyright(c) 2021 Mediatek Corporation. All rights reserved. + * + * Author: Bo Pan + */ + +#ifndef __INCLUDE_SOUND_SOF_DAI_MEDIATEK_H__ +#define __INCLUDE_SOUND_SOF_DAI_MEDIATEK_H__ + +#include + +struct sof_ipc_dai_mtk_afe_params { + struct sof_ipc_hdr hdr; + u32 channels; + u32 rate; + u32 format; + u32 stream_id; + u32 reserved[4]; /* reserve for future */ +} __packed; + +#endif + diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 3782127a7095..5132bc60f54b 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -13,6 +13,7 @@ #include #include #include +#include /* * DAI Configuration. @@ -70,6 +71,7 @@ enum sof_ipc_dai_type { SOF_DAI_AMD_BT, /**< AMD ACP BT*/ SOF_DAI_AMD_SP, /**< AMD ACP SP */ SOF_DAI_AMD_DMIC, /**< AMD ACP DMIC */ + SOF_DAI_MEDIATEK_AFE, /**< Mediatek AFE */ }; /* general purpose DAI configuration */ @@ -97,6 +99,7 @@ struct sof_ipc_dai_config { struct sof_ipc_dai_acp_params acpbt; struct sof_ipc_dai_acp_params acpsp; struct sof_ipc_dai_acp_params acpdmic; + struct sof_ipc_dai_mtk_afe_params afe; }; } __packed; -- cgit v1.2.3 From b456abe63f60ad93c83a526d33b71574bc32656c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 19 Nov 2021 17:08:50 -0600 Subject: ALSA: pcm: introduce INFO_NO_REWINDS flag When the hardware can only deal with a monotonically increasing appl_ptr, this flag can be set. In case the application requests a rewind, be it with a snd_pcm_rewind() or with a direct change of a mmap'ed pointer followed by a SNDRV_PCM_IOCTL_SYNC_PTR, this patch checks if a rewind occurred and returns an error. Credits to Takashi Iwai for identifying the path with SYNC_PTR and suggesting the pointer checks. Suggested-by: Takashi Iwai Reviewed-by: Takashi Iwai Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211119230852.206310-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 5fbb79e30819..ff7e638221c5 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -300,7 +300,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ #define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ #define SNDRV_PCM_INFO_EXPLICIT_SYNC 0x10000000 /* needs explicit sync of pointers and data */ - +#define SNDRV_PCM_INFO_NO_REWINDS 0x20000000 /* hardware can only support monotonic changes of appl_ptr */ #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ -- cgit v1.2.3 From 083a7fba38885a8ffa03a2857e383421cefd36e6 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Thu, 25 Nov 2021 13:58:11 +0800 Subject: ASoC: rt5640: Add the binding include file for the HDA header support The patch adds the binding include file for the HDA header support. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20211125055812.8911-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- include/dt-bindings/sound/rt5640.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/dt-bindings/sound/rt5640.h b/include/dt-bindings/sound/rt5640.h index 154c9b4414f2..655f6946388a 100644 --- a/include/dt-bindings/sound/rt5640.h +++ b/include/dt-bindings/sound/rt5640.h @@ -16,6 +16,7 @@ #define RT5640_JD_SRC_GPIO2 4 #define RT5640_JD_SRC_GPIO3 5 #define RT5640_JD_SRC_GPIO4 6 +#define RT5640_JD_SRC_HDA_HEADER 7 #define RT5640_OVCD_SF_0P5 0 #define RT5640_OVCD_SF_0P75 1 -- cgit v1.2.3 From a0f84dfb3f6d9f78f862cbe885036d3e4449fc6f Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Thu, 25 Nov 2021 12:15:19 +0200 Subject: ASoC: SOF: IPC: dai: Expand DAI_CONFIG IPC flags Some DAI components, such as HDaudio, need to be stopped in two steps a) stop the DAI component b) stop the DAI DMA This patch enables this two-step stop by expanding the DAI_CONFIG IPC flags and split them into 2 parts. The 4 LSB bits indicate when the DAI_CONFIG IPC is sent, ex: hw_params, hw_free or pause. The 4 MSB bits are used as the quirk flags to be used along with the command flags. The quirk flag called SOF_DAI_CONFIG_FLAGS_2_STEP_STOP shall be set along with the HW_PARAMS command flag, i.e. before the pipeline is started so that the stop/pause trigger op in the FW can take the appropriate action to either perform/skip the DMA stop. If set, the DMA stop will be executed when the DAI_CONFIG IPC is sent during hw_free. In the case of pause, DMA pause will be handled when the DAI_CONFIG IPC is sent with the PAUSE command flag. Along with this, modify the signature for the hda_ctrl_dai_widget_setup/ hda_ctrl_dai_widget_free() functions to take additional flags as an argument and modify all users to pass the appropriate quirk flags. Only the HDA DAI's need to pass the SOF_DAI_CONFIG_FLAGS_2_STEP_STOP quirk flag during hw_params to indicate that it supports two-step stop and pause. Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20211125101520.291581-10-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/dai.h | 25 +++++++++++++++++++------ 1 file changed, 19 insertions(+), 6 deletions(-) (limited to 'include') diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 5132bc60f54b..59ee50ac7705 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -52,12 +52,25 @@ #define SOF_DAI_FMT_INV_MASK 0x0f00 #define SOF_DAI_FMT_CLOCK_PROVIDER_MASK 0xf000 -/* DAI_CONFIG flags */ -#define SOF_DAI_CONFIG_FLAGS_MASK 0x3 -#define SOF_DAI_CONFIG_FLAGS_NONE (0 << 0) /**< DAI_CONFIG sent without stage information */ -#define SOF_DAI_CONFIG_FLAGS_HW_PARAMS (1 << 0) /**< DAI_CONFIG sent during hw_params stage */ -#define SOF_DAI_CONFIG_FLAGS_HW_FREE (2 << 0) /**< DAI_CONFIG sent during hw_free stage */ -#define SOF_DAI_CONFIG_FLAGS_RFU (3 << 0) /**< not used, reserved for future use */ +/* + * DAI_CONFIG flags. The 4 LSB bits are used for the commands, HW_PARAMS, HW_FREE and PAUSE + * representing when the IPC is sent. The 4 MSB bits are used to add quirks along with the above + * commands. + */ +#define SOF_DAI_CONFIG_FLAGS_CMD_MASK 0xF +#define SOF_DAI_CONFIG_FLAGS_NONE 0 /**< DAI_CONFIG sent without stage information */ +#define SOF_DAI_CONFIG_FLAGS_HW_PARAMS BIT(0) /**< DAI_CONFIG sent during hw_params stage */ +#define SOF_DAI_CONFIG_FLAGS_HW_FREE BIT(1) /**< DAI_CONFIG sent during hw_free stage */ +/**< DAI_CONFIG sent during pause trigger. Only available ABI 3.20 onwards */ +#define SOF_DAI_CONFIG_FLAGS_PAUSE BIT(2) +#define SOF_DAI_CONFIG_FLAGS_QUIRK_SHIFT 4 +#define SOF_DAI_CONFIG_FLAGS_QUIRK_MASK (0xF << SOF_DAI_CONFIG_FLAGS_QUIRK_SHIFT) +/* + * This should be used along with the SOF_DAI_CONFIG_FLAGS_HW_PARAMS to indicate that pipeline + * stop/pause and DAI DMA stop/pause should happen in two steps. This change is only available + * ABI 3.20 onwards. + */ +#define SOF_DAI_CONFIG_FLAGS_2_STEP_STOP BIT(0) /** \brief Types of DAI */ enum sof_ipc_dai_type { -- cgit v1.2.3 From 8544f08c816292c2219f28c6eaa69236b978bfb9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 16 Nov 2021 16:45:12 +0900 Subject: ASoC: soc-dai: update snd_soc_dai_delay() to snd_soc_pcm_dai_delay() Current soc_pcm_pointer() is manually calculating both CPU-DAI's max delay (= A) and Codec-DAI's max delay (= B). static snd_pcm_uframes_t soc_pcm_pointer(...) { ... ^ for_each_rtd_cpu_dais(rtd, i, cpu_dai) (A) cpu_delay = max(cpu_delay, ...); v delay += cpu_delay; ^ for_each_rtd_codec_dais(rtd, i, codec_dai) (B) codec_delay = max(codec_delay, ...); v delay += codec_delay; runtime->delay = delay; ... } Current soc_pcm_pointer() and the total delay calculating is not readable / difficult to understand. This patch update snd_soc_dai_delay() to snd_soc_pcm_dai_delay(), and calcule both CPU/Codec delay in one function. Link: https://lore.kernel.org/r/87fszl4yrq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/875yssy25z.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0dcb361a98bb..5d4dd7c5450b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -208,8 +208,6 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int rollback); -snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, - struct snd_pcm_substream *substream); void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, @@ -238,6 +236,8 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, int cmd); +void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, + snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); int snd_soc_dai_compr_startup(struct snd_soc_dai *dai, struct snd_compr_stream *cstream); -- cgit v1.2.3 From 403f830e7a0be5a9e33c7a9d208574f79887ec57 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 16 Nov 2021 16:45:18 +0900 Subject: ASoC: soc-component: add snd_soc_pcm_component_delay() Current soc-pcm.c :: soc_pcm_pointer() is assuming that component driver might update runtime->delay silently in snd_soc_pcm_component_pointer() (= A). static snd_pcm_uframes_t soc_pcm_pointer(...) { ... /* clearing the previous total delay */ => runtime->delay = 0; (A) offset = snd_soc_pcm_component_pointer(substream); /* base delay if assigned in pointer callback */ => delay = runtime->delay; ... } 1) The behavior that ".pointer callback secretly updates runtime->delay" is strange and confusable. 2) Current snd_soc_pcm_component_pointer() uses 1st found component's .pointer callback only, thus it is no problem for now. But runtime->delay might be overwrote if it adjusted to multiple components in the future. 3) Component delay is updated at .pointer callback timing (secretly). But some components which doesn't have .pointer callback might want to increase runtime->delay for some reasons. We already have .delay function for DAI, but not have for Component. This patch adds new snd_soc_pcm_component_delay() for it. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/874k8cy25t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index a4317144ab62..a52080407b98 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -148,6 +148,8 @@ struct snd_soc_component_driver { struct vm_area_struct *vma); int (*ack)(struct snd_soc_component *component, struct snd_pcm_substream *substream); + snd_pcm_sframes_t (*delay)(struct snd_soc_component *component, + struct snd_pcm_substream *substream); const struct snd_compress_ops *compress_ops; @@ -505,5 +507,7 @@ int snd_soc_pcm_component_pm_runtime_get(struct snd_soc_pcm_runtime *rtd, void snd_soc_pcm_component_pm_runtime_put(struct snd_soc_pcm_runtime *rtd, void *stream, int rollback); int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream); +void snd_soc_pcm_component_delay(struct snd_pcm_substream *substream, + snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); #endif /* __SOC_COMPONENT_H */ -- cgit v1.2.3 From 15fa179f3f45415696d376abc84e0098a9586b33 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 26 Nov 2021 15:03:53 +0100 Subject: ALSA: hda: Fill gaps in NHLT endpoint-interface MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Two key operations missings are: endpoint presence-check and retrieval of matching endpoint hardware configuration (blob). Add operations for both use cases. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20211126140355.1042684-2-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- include/sound/intel-nhlt.h | 37 +++++++++++++++++++++++++++++-------- 1 file changed, 29 insertions(+), 8 deletions(-) (limited to 'include') diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index d0574805865f..089a760d36eb 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -10,6 +10,14 @@ #include +enum nhlt_link_type { + NHLT_LINK_HDA = 0, + NHLT_LINK_DSP = 1, + NHLT_LINK_DMIC = 2, + NHLT_LINK_SSP = 3, + NHLT_LINK_INVALID +}; + #if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_INTEL_NHLT) struct wav_fmt { @@ -33,14 +41,6 @@ struct wav_fmt_ext { u8 sub_fmt[16]; } __packed; -enum nhlt_link_type { - NHLT_LINK_HDA = 0, - NHLT_LINK_DSP = 1, - NHLT_LINK_DMIC = 2, - NHLT_LINK_SSP = 3, - NHLT_LINK_INVALID -}; - enum nhlt_device_type { NHLT_DEVICE_BT = 0, NHLT_DEVICE_DMIC = 1, @@ -132,6 +132,12 @@ void intel_nhlt_free(struct nhlt_acpi_table *addr); int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt); +bool intel_nhlt_has_endpoint_type(struct nhlt_acpi_table *nhlt, u8 link_type); +struct nhlt_specific_cfg * +intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, + u32 bus_id, u8 link_type, u8 vbps, u8 bps, + u8 num_ch, u32 rate, u8 dir, u8 dev_type); + #else struct nhlt_acpi_table; @@ -150,6 +156,21 @@ static inline int intel_nhlt_get_dmic_geo(struct device *dev, { return 0; } + +static inline bool intel_nhlt_has_endpoint_type(struct nhlt_acpi_table *nhlt, + u8 link_type) +{ + return false; +} + +static inline struct nhlt_specific_cfg * +intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, + u32 bus_id, u8 link_type, u8 vbps, u8 bps, + u8 num_ch, u32 rate, u8 dir, u8 dev_type) +{ + return NULL; +} + #endif #endif -- cgit v1.2.3 From 7cfa3d00730a4c0694b55fb1974823baeab8815b Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Wed, 8 Dec 2021 18:17:18 +0800 Subject: ASoC: rt5682s: add delay time to fix pop sound issue There is a pop noise at the beginning of the capture data. This patch adds the delay time before stereo1 ADC unmute to fix the pop sound issue. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20211208101718.28945-1-shumingf@realtek.com Signed-off-by: Mark Brown --- include/sound/rt5682s.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/rt5682s.h b/include/sound/rt5682s.h index accfbc2dcdd2..f18d91308b9a 100644 --- a/include/sound/rt5682s.h +++ b/include/sound/rt5682s.h @@ -40,6 +40,7 @@ struct rt5682s_platform_data { enum rt5682s_jd_src jd_src; unsigned int dmic_clk_rate; unsigned int dmic_delay; + unsigned int amic_delay; bool dmic_clk_driving_high; const char *dai_clk_names[RT5682S_DAI_NUM_CLKS]; -- cgit v1.2.3 From fb6723daf89083a0d2290f3a0abc777e40766c84 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 May 2021 12:33:53 +0900 Subject: ALSA: pcm: comment about relation between msbits hw parameter and [S|U]32 formats Regarding to handling [U|S][32|24] PCM formats, many userspace application developers and driver developers have confusion, since they require them to understand justification or padding. It easily loses consistency and soundness to operate with many type of devices. In this commit, I attempt to solve the situation by adding comment about relation between [S|U]32 formats and 'msbits' hardware parameter. The formats are used for 'left-justified' sample format, and the available bit count in most significant bit is delivered to userspace in msbits hardware parameter (struct snd_pcm_hw_params.msbits), which is decided by msbits constraint added by pcm drivers (snd_pcm_hw_constraint_msbits()). In driver side, the msbits constraint includes two elements; the physical width of format and the available width of the format in most significant bit. The former is used to match SAMPLE_BITS of format. (For my convenience, I ignore wildcard in the usage of the constraint.) As a result of interaction between ALSA pcm core and ALSA pcm application, when the format in which SAMPLE_BITS equals to physical width of the msbits constaint, the msbits parameter is set by referring to the available width of the constraint. When the msbits parameter is not changed in the above process, ALSA pcm core set it alternatively with SAMPLE_BIT of chosen format. In userspace application side, the msbits is only available after calling ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS request. Even if the hardware parameter structure includes somewhat value of SAMPLE_BITS interval parameter as width of format, all of the width is not always available since msbits can be less than the width. I note that [S|U]24 formats are used for 'right-justified' 24 bit sample formats within 32 bit frame. The first byte in most significant bit should be invalidated. Although the msbits exposed to userspace should be zero as invalid value, actually it is 32 from physical width of format. [ corrected typos -- tiwai ] Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20210529033353.21641-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 3 +++ include/uapi/sound/asound.h | 3 +++ 2 files changed, 6 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 33451f8ff755..9b187d86e1bd 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -147,6 +147,9 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_S24_BE _SNDRV_PCM_FMTBIT(S24_BE) #define SNDRV_PCM_FMTBIT_U24_LE _SNDRV_PCM_FMTBIT(U24_LE) #define SNDRV_PCM_FMTBIT_U24_BE _SNDRV_PCM_FMTBIT(U24_BE) +// For S32/U32 formats, 'msbits' hardware parameter is often used to deliver information about the +// available bit count in most significant bit. It's for the case of so-called 'left-justified' or +// `right-padding` sample which has less width than 32 bit. #define SNDRV_PCM_FMTBIT_S32_LE _SNDRV_PCM_FMTBIT(S32_LE) #define SNDRV_PCM_FMTBIT_S32_BE _SNDRV_PCM_FMTBIT(S32_BE) #define SNDRV_PCM_FMTBIT_U32_LE _SNDRV_PCM_FMTBIT(U32_LE) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 5fbb79e30819..cf1d20e34167 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -202,6 +202,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S24_BE ((__force snd_pcm_format_t) 7) /* low three bytes */ #define SNDRV_PCM_FORMAT_U24_LE ((__force snd_pcm_format_t) 8) /* low three bytes */ #define SNDRV_PCM_FORMAT_U24_BE ((__force snd_pcm_format_t) 9) /* low three bytes */ +// For S32/U32 formats, 'msbits' hardware parameter is often used to deliver information about the +// available bit count in most significant bit. It's for the case of so-called 'left-justified' or +// `right-padding` sample which has less width than 32 bit. #define SNDRV_PCM_FORMAT_S32_LE ((__force snd_pcm_format_t) 10) #define SNDRV_PCM_FORMAT_S32_BE ((__force snd_pcm_format_t) 11) #define SNDRV_PCM_FORMAT_U32_LE ((__force snd_pcm_format_t) 12) -- cgit v1.2.3 From 55b71f6c29f2a78af42dd453dfed895eba516cb4 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 13 Dec 2021 17:12:57 +0900 Subject: ALSA: uapi: use C90 comment style instead of C99 style UAPI headers are built with compiler option for C90, thus double-slashes comment introduced in C99 is not preferable. Fixes: fb6723daf890 ("ALSA: pcm: comment about relation between msbits hw parameter and [S|U]32 formats") Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20211213081257.36097-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index cf1d20e34167..1834f58b8ede 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -202,9 +202,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S24_BE ((__force snd_pcm_format_t) 7) /* low three bytes */ #define SNDRV_PCM_FORMAT_U24_LE ((__force snd_pcm_format_t) 8) /* low three bytes */ #define SNDRV_PCM_FORMAT_U24_BE ((__force snd_pcm_format_t) 9) /* low three bytes */ -// For S32/U32 formats, 'msbits' hardware parameter is often used to deliver information about the -// available bit count in most significant bit. It's for the case of so-called 'left-justified' or -// `right-padding` sample which has less width than 32 bit. +/* + * For S32/U32 formats, 'msbits' hardware parameter is often used to deliver information about the + * available bit count in most significant bit. It's for the case of so-called 'left-justified' or + * `right-padding` sample which has less width than 32 bit. + */ #define SNDRV_PCM_FORMAT_S32_LE ((__force snd_pcm_format_t) 10) #define SNDRV_PCM_FORMAT_S32_BE ((__force snd_pcm_format_t) 11) #define SNDRV_PCM_FORMAT_U32_LE ((__force snd_pcm_format_t) 12) -- cgit v1.2.3 From b7898396f4bbe160f546d0c5e9fa17cca9a7d153 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Dec 2021 11:37:42 -0600 Subject: ASoC: soc-pcm: Fix and cleanup DPCM locking The existing locking for DPCM has several issues a) a confusing mix of card->mutex and card->pcm_mutex. b) a dpcm_lock spinlock added inconsistently and on paths that could be recursively taken. The use of irqsave/irqrestore was also overkill. The suggested model is: 1) The pcm_mutex is the top-most protection of BE links in the FE. The pcm_mutex is applied always on either the top PCM callbacks or the external call from DAPM, not taken in the internal functions. 2) the FE stream lock is taken in higher levels before invoking dpcm_be_dai_trigger() 3) when adding and deleting a BE, both the pcm_mutex and FE stream lock are taken. Signed-off-by: Takashi Iwai [clarification of commit message by plbossart] Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20211207173745.15850-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e6dd8a257c5..5872a8864f3b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -893,8 +893,6 @@ struct snd_soc_card { struct mutex pcm_mutex; enum snd_soc_pcm_subclass pcm_subclass; - spinlock_t dpcm_lock; - int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); int (*remove)(struct snd_soc_card *card); -- cgit v1.2.3 From 848aedfdc6ba25ad5652797db9266007773e44dd Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Dec 2021 11:37:44 -0600 Subject: ASoC: soc-pcm: test refcount before triggering On start/pause_release/resume, when more than one FE is connected to the same BE, it's possible that the trigger is sent more than once. This is not desirable, we only want to trigger a BE once, which is straightforward to implement with a refcount. For stop/pause/suspend, the problem is more complicated: the check implemented in snd_soc_dpcm_can_be_free_stop() may fail due to a conceptual deadlock when we trigger the BE before the FE. In this case, the FE states have not yet changed, so there are corner cases where the TRIGGER_STOP is never sent - the dual case of start where multiple triggers might be sent. This patch suggests an unconditional trigger in all cases, without checking the FE states, using a refcount protected by the BE PCM stream lock. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20211207173745.15850-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index bc7af90099a8..75b92d883976 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -101,6 +101,8 @@ struct snd_soc_dpcm_runtime { enum snd_soc_dpcm_state state; int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ + + int be_start; /* refcount protected by BE stream pcm lock */ }; #define for_each_dpcm_fe(be, stream, _dpcm) \ -- cgit v1.2.3 From bdecfceffeeb9000e78b0f613069f5c06974b347 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 22 Nov 2021 23:21:54 +0100 Subject: ASoC: dai_dma: remove slave_id field This field is no longer set from any driver now, so remove the last references as well. Signed-off-by: Arnd Bergmann Acked-by: Mark Brown Link: https://lore.kernel.org/r/20211122222203.4103644-3-arnd@kernel.org Signed-off-by: Vinod Koul --- include/sound/dmaengine_pcm.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 96666efddb39..38ea046e653c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -60,7 +60,6 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @maxburst: Maximum number of words(note: words, as in units of the * src_addr_width member, not bytes) that can be send to or received from the * DAI in one burst. - * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. * @chan_name: Custom channel name to use when requesting DMA channel. @@ -74,7 +73,6 @@ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; enum dma_slave_buswidth addr_width; u32 maxburst; - unsigned int slave_id; void *filter_data; const char *chan_name; unsigned int fifo_size; -- cgit v1.2.3 From 03de6b273805b3c552ff158f8688555937375926 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 22 Nov 2021 23:22:00 +0100 Subject: dmaengine: qcom-adm: stop abusing slave_id config The slave_id was previously used to pick one DMA slave instead of another, but this is now done through the DMA descriptors in device tree. For the qcom_adm driver, the configuration is documented in the DT binding to contain a tuple of device identifier and a "crci" field, but the implementation ends up using only a single cell for identifying the slave, with the crci getting passed in nonstandard properties of the device, and passed through the dma driver using the old slave_id field. Part of the problem apparently is that the nand driver ends up using only a single DMA request ID, but requires distinct values for "crci" depending on the type of transfer. Change both the dmaengine driver and the two slave drivers to allow the documented binding to work in addition to the ad-hoc passing of crci values. In order to no longer abuse the slave_id field, pass the data using the "peripheral_config" mechanism instead. Signed-off-by: Arnd Bergmann Acked-by: Mark Brown Link: https://lore.kernel.org/r/20211122222203.4103644-9-arnd@kernel.org Signed-off-by: Vinod Koul --- include/linux/dma/qcom_adm.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) create mode 100644 include/linux/dma/qcom_adm.h (limited to 'include') diff --git a/include/linux/dma/qcom_adm.h b/include/linux/dma/qcom_adm.h new file mode 100644 index 000000000000..af20df674f0c --- /dev/null +++ b/include/linux/dma/qcom_adm.h @@ -0,0 +1,12 @@ +// SPDX-License-Identifier: GPL-2.0-only +#ifndef __LINUX_DMA_QCOM_ADM_H +#define __LINUX_DMA_QCOM_ADM_H + +#include + +struct qcom_adm_peripheral_config { + u32 crci; + u32 mux; +}; + +#endif /* __LINUX_DMA_QCOM_ADM_H */ -- cgit v1.2.3 From 93cdb5b0dc56cc7a8b87a61146495f3bdc93d7ba Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 22 Nov 2021 23:22:01 +0100 Subject: dmaengine: xilinx_dpdma: stop using slave_id field The display driver wants to pass a custom flag to the DMA engine driver, which it started doing by using the slave_id field that was traditionally used for a different purpose. As there is no longer a correct use for the slave_id field, it should really be removed, and the remaining users changed over to something different. The new mechanism for passing nonstandard settings is using the .peripheral_config field, so use that to pass a newly defined structure here, making it clear that this will not work in portable drivers. Reviewed-by: Laurent Pinchart Signed-off-by: Arnd Bergmann Acked-by: Mark Brown Link: https://lore.kernel.org/r/20211122222203.4103644-10-arnd@kernel.org Signed-off-by: Vinod Koul --- include/linux/dma/xilinx_dpdma.h | 11 +++++++++++ 1 file changed, 11 insertions(+) create mode 100644 include/linux/dma/xilinx_dpdma.h (limited to 'include') diff --git a/include/linux/dma/xilinx_dpdma.h b/include/linux/dma/xilinx_dpdma.h new file mode 100644 index 000000000000..83a1377f03f8 --- /dev/null +++ b/include/linux/dma/xilinx_dpdma.h @@ -0,0 +1,11 @@ +// SPDX-License-Identifier: GPL-2.0 +#ifndef __LINUX_DMA_XILINX_DPDMA_H +#define __LINUX_DMA_XILINX_DPDMA_H + +#include + +struct xilinx_dpdma_peripheral_config { + bool video_group; +}; + +#endif /* __LINUX_DMA_XILINX_DPDMA_H */ -- cgit v1.2.3 From 3c219644075795a99271d345efdfa8b256e55161 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 22 Nov 2021 23:22:03 +0100 Subject: dmaengine: remove slave_id config field All references to the slave_id field have been removed, so remove the field as well to prevent new references from creeping in again. Originally this allowed slave DMA drivers to configure which device is accessed with the dmaengine_slave_config() call, but this was inconsistent, as the same information is also passed while requesting a channel, and never changes in practice. In modern kernels, the device is always selected when requesting the channel, so the .slave_id field is no longer useful. Reviewed-by: Laurent Pinchart Signed-off-by: Arnd Bergmann Acked-by: Mark Brown Link: https://lore.kernel.org/r/20211122222203.4103644-12-arnd@kernel.org Signed-off-by: Vinod Koul --- include/linux/dmaengine.h | 4 ---- 1 file changed, 4 deletions(-) (limited to 'include') diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 9000f3ffce8b..0349b35235e6 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -418,9 +418,6 @@ enum dma_slave_buswidth { * @device_fc: Flow Controller Settings. Only valid for slave channels. Fill * with 'true' if peripheral should be flow controller. Direction will be * selected at Runtime. - * @slave_id: Slave requester id. Only valid for slave channels. The dma - * slave peripheral will have unique id as dma requester which need to be - * pass as slave config. * @peripheral_config: peripheral configuration for programming peripheral * for dmaengine transfer * @peripheral_size: peripheral configuration buffer size @@ -448,7 +445,6 @@ struct dma_slave_config { u32 src_port_window_size; u32 dst_port_window_size; bool device_fc; - unsigned int slave_id; void *peripheral_config; size_t peripheral_size; }; -- cgit v1.2.3 From 60ded273e4c047aec364f70195aced71a4082f90 Mon Sep 17 00:00:00 2001 From: Karol Trzcinski Date: Thu, 16 Dec 2021 17:24:22 -0600 Subject: ipc: debug: Add shared memory heap to memory scan Newly added shared heap zones should be taken into account during memory usage scanning. Reviewed-by: Kai Vehmanen Reviewed-by: Liam Girdwood Signed-off-by: Karol Trzcinski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211216232422.345164-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/debug.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/sof/debug.h b/include/sound/sof/debug.h index 3ecb5793789d..38693e3fb514 100644 --- a/include/sound/sof/debug.h +++ b/include/sound/sof/debug.h @@ -19,6 +19,8 @@ enum sof_ipc_dbg_mem_zone { SOF_IPC_MEM_ZONE_SYS_RUNTIME = 1, /**< System-runtime zone */ SOF_IPC_MEM_ZONE_RUNTIME = 2, /**< Runtime zone */ SOF_IPC_MEM_ZONE_BUFFER = 3, /**< Buffer zone */ + SOF_IPC_MEM_ZONE_RUNTIME_SHARED = 4, /**< System runtime zone */ + SOF_IPC_MEM_ZONE_SYS_SHARED = 5, /**< System shared zone */ }; /** ABI3.18 */ -- cgit v1.2.3 From e047d0372689f5d4231eefb731b60ac64720bbf0 Mon Sep 17 00:00:00 2001 From: Ricard Wanderlof Date: Wed, 15 Dec 2021 18:01:24 +0100 Subject: ASoC: tlv320adc3xxx: New codec bindings DT bindings for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof Link: https://lore.kernel.org/r/alpine.DEB.2.21.2112151759170.27889@lap5cg0092dnk.se.axis.com Signed-off-by: Mark Brown --- include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h (limited to 'include') diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..ec988439da20 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- cgit v1.2.3 From fc179420fde3821c4d191e81b4f7b05c1dab87e2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 23 Dec 2021 13:36:17 +0200 Subject: ASoC: SOF: Move the definition of enum snd_sof_fw_state to global header Move the enum snd_sof_fw_state to include/sound/sof.h to be accessible outside of the core SOF stack. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Reviewed-by: Paul Olaru Link: https://lore.kernel.org/r/20211223113628.18582-10-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) (limited to 'include') diff --git a/include/sound/sof.h b/include/sound/sof.h index 23b374311d16..b9131c01eefd 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -17,6 +17,28 @@ struct snd_sof_dsp_ops; +/** + * enum snd_sof_fw_state - DSP firmware state definitions + * @SOF_FW_BOOT_NOT_STARTED: firmware boot is not yet started + * @SOF_FW_BOOT_PREPARE: preparing for boot (firmware loading for exaqmple) + * @SOF_FW_BOOT_IN_PROGRESS: firmware boot is in progress + * @SOF_FW_BOOT_FAILED: firmware boot failed + * @SOF_FW_BOOT_READY_FAILED: firmware booted but fw_ready op failed + * @SOF_FW_BOOT_READY_OK: firmware booted and fw_ready op passed + * @SOF_FW_BOOT_COMPLETE: firmware is booted up and functional + * @SOF_FW_CRASHED: firmware crashed after successful boot + */ +enum snd_sof_fw_state { + SOF_FW_BOOT_NOT_STARTED = 0, + SOF_FW_BOOT_PREPARE, + SOF_FW_BOOT_IN_PROGRESS, + SOF_FW_BOOT_FAILED, + SOF_FW_BOOT_READY_FAILED, + SOF_FW_BOOT_READY_OK, + SOF_FW_BOOT_COMPLETE, + SOF_FW_CRASHED, +}; + /* * SOF Platform data. */ -- cgit v1.2.3 From d41607d37c1385da799f9a2ddb10c460e573687e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 23 Dec 2021 13:36:18 +0200 Subject: ASoC: SOF: Rename 'enum snd_sof_fw_state' to 'enum sof_fw_state' Since there is nothing SND about the firmware state, rename the enum from `snd_sof_fw_state` to simply `sof_fw_state` Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Reviewed-by: Paul Olaru Link: https://lore.kernel.org/r/20211223113628.18582-11-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/sof.h b/include/sound/sof.h index b9131c01eefd..813680ab9aad 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -18,7 +18,7 @@ struct snd_sof_dsp_ops; /** - * enum snd_sof_fw_state - DSP firmware state definitions + * enum sof_fw_state - DSP firmware state definitions * @SOF_FW_BOOT_NOT_STARTED: firmware boot is not yet started * @SOF_FW_BOOT_PREPARE: preparing for boot (firmware loading for exaqmple) * @SOF_FW_BOOT_IN_PROGRESS: firmware boot is in progress @@ -28,7 +28,7 @@ struct snd_sof_dsp_ops; * @SOF_FW_BOOT_COMPLETE: firmware is booted up and functional * @SOF_FW_CRASHED: firmware crashed after successful boot */ -enum snd_sof_fw_state { +enum sof_fw_state { SOF_FW_BOOT_NOT_STARTED = 0, SOF_FW_BOOT_PREPARE, SOF_FW_BOOT_IN_PROGRESS, -- cgit v1.2.3 From 3d4641a42ccf1593b3f3a474ee7541727acbb8e0 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Tue, 14 Dec 2021 15:20:46 +0100 Subject: ASoC: core: Add snd_soc_of_parse_pin_switches() from simple-card-utils The ASoC core already has several helpers to parse card properties from the device tree. Move the parsing code for "pin-switches" from simple-card-utils to a shared snd_soc_of_parse_pin_switches() function so other drivers can also use it to set up pin switches configured in the device tree. Cc: Paul Cercueil Signed-off-by: Stephan Gerhold Link: https://lore.kernel.org/r/20211214142049.20422-2-stephan@gerhold.net Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 5872a8864f3b..7a1650b303f1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1211,6 +1211,7 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, const char *propname); +int snd_soc_of_parse_pin_switches(struct snd_soc_card *card, const char *prop); int snd_soc_of_get_slot_mask(struct device_node *np, const char *prop_name, unsigned int *mask); -- cgit v1.2.3 From b86947b52f0d0e5b6e6f0510933ca13aad266e47 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart