From 03eec9b4eb897dde5985579508c978e7a29052bd Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 4 Dec 2019 15:15:50 -0600 Subject: ASoC: SOF: Introduce default_fw_filename member in sof_dev_desc Currently the FW filename is obtained from the ACPI matching table when determining which machine driver to use. In preparation for making the machine driver ACPI match optional for Device Tree platforms and moving the machine driver selection out of the SOF core, this patch introduces the default_fw_filename member in struct sof_dev_desc. Once the machine driver selection is moved out of SOF core, the nocodec_fw_filename will become obsolete and will be removed. Signed-off-by: Ranjani Sridharan Signed-off-by: Daniel Baluta Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191204211556.12671-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/sof.h b/include/sound/sof.h index 479101736ee0..1723478db4a2 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -91,6 +91,9 @@ struct sof_dev_desc { const char *default_fw_path; const char *default_tplg_path; + /* default firmware name */ + const char *default_fw_filename; + const struct snd_sof_dsp_ops *ops; const struct sof_arch_ops *arch_ops; }; -- cgit v1.2.3 From f4e4113b2aec2d344276ac07f78e80460eb8ebf8 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 4 Dec 2019 15:15:54 -0600 Subject: ASoC: SOF: remove nocodec_fw_filename Remove nocodec_fw_filename from struct sof_dev_desc as it is not longer needed. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191204211556.12671-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/sof.h b/include/sound/sof.h index 1723478db4a2..98a757d3a67d 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -84,7 +84,6 @@ struct sof_dev_desc { const void *chip_info; /* defaults for no codec mode */ - const char *nocodec_fw_filename; const char *nocodec_tplg_filename; /* defaults paths for firmware and topology files */ -- cgit v1.2.3 From 5ad1cece81db9b389526499ae2e784013c85f136 Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Wed, 4 Dec 2019 15:15:55 -0600 Subject: ASoC: SOF: Remove unused drv_name in sof_pdata This field is only set but never used. Let's remove it to make code cleaner. Signed-off-by: Daniel Baluta Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191204211556.12671-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/sof.h b/include/sound/sof.h index 98a757d3a67d..96625355aa94 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -22,7 +22,6 @@ struct snd_sof_dsp_ops; */ struct snd_sof_pdata { const struct firmware *fw; - const char *drv_name; const char *name; const char *platform; -- cgit v1.2.3 From d612b455f120d05a42c95ccd7469fa13efb8d307 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 4 Dec 2019 15:15:56 -0600 Subject: ASoC: SOF: nocodec: Amend arguments for sof_nocodec_setup() Set the drv_name and tplg_filename for nocodec machine driver in sof_machine_check(). This means the sof_nocodec_setup() does not need the mach, plat_data or desc arguments any longer. Signed-off-by: Daniel Baluta Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191204211556.12671-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'include') diff --git a/include/sound/sof.h b/include/sound/sof.h index 96625355aa94..6ea74f1a9ec2 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -97,8 +97,5 @@ struct sof_dev_desc { }; int sof_nocodec_setup(struct device *dev, - struct snd_sof_pdata *sof_pdata, - struct snd_soc_acpi_mach *mach, - const struct sof_dev_desc *desc, const struct snd_sof_dsp_ops *ops); #endif -- cgit v1.2.3 From 433363e779ecb772c2e9ffea5c9f266115c24441 Mon Sep 17 00:00:00 2001 From: Seppo Ingalsuo Date: Mon, 9 Dec 2019 18:48:47 -0600 Subject: ASoC: SOF: Add asynchronous sample rate converter topology support This patch adds into SOF topology the handling of ASRC DAPM type, adds the tokens to configure the ASRC, and implement component IPC into the driver. Signed-off-by: Seppo Ingalsuo Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191210004854.16845-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/topology.h | 27 +++++++++++++++++++++++++++ include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 6 ++++++ 3 files changed, 34 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h index c47b36240920..8e76178fedf0 100644 --- a/include/sound/sof/topology.h +++ b/include/sound/sof/topology.h @@ -36,6 +36,7 @@ enum sof_comp_type { SOF_COMP_KPB, /* A key phrase buffer component */ SOF_COMP_SELECTOR, /**< channel selector component */ SOF_COMP_DEMUX, + SOF_COMP_ASRC, /**< Asynchronous sample rate converter */ /* keep FILEREAD/FILEWRITE as the last ones */ SOF_COMP_FILEREAD = 10000, /**< host test based file IO */ SOF_COMP_FILEWRITE = 10001, /**< host test based file IO */ @@ -147,6 +148,32 @@ struct sof_ipc_comp_src { uint32_t rate_mask; /**< SOF_RATE_ supported rates */ } __packed; +/* generic ASRC component */ +struct sof_ipc_comp_asrc { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; + /* either source or sink rate must be non zero */ + uint32_t source_rate; /**< Define fixed source rate or */ + /**< use 0 to indicate need to get */ + /**< the rate from stream */ + uint32_t sink_rate; /**< Define fixed sink rate or */ + /**< use 0 to indicate need to get */ + /**< the rate from stream */ + uint32_t asynchronous_mode; /**< synchronous 0, asynchronous 1 */ + /**< When 1 the ASRC tracks and */ + /**< compensates for drift. */ + uint32_t operation_mode; /**< push 0, pull 1, In push mode the */ + /**< ASRC consumes a defined number */ + /**< of frames at input, with varying */ + /**< number of frames at output. */ + /**< In pull mode the ASRC outputs */ + /**< a defined number of frames while */ + /**< number of input frames varies. */ + + /* reserved for future use */ + uint32_t reserved[4]; +} __attribute__((packed)); + /* generic MUX component */ struct sof_ipc_comp_mux { struct sof_ipc_comp comp; diff --git a/include/uapi/sound/sof/abi.h b/include/uapi/sound/sof/abi.h index ebfdc20ca081..c0ef1643c753 100644 --- a/include/uapi/sound/sof/abi.h +++ b/include/uapi/sound/sof/abi.h @@ -26,7 +26,7 @@ /* SOF ABI version major, minor and patch numbers */ #define SOF_ABI_MAJOR 3 -#define SOF_ABI_MINOR 11 +#define SOF_ABI_MINOR 12 #define SOF_ABI_PATCH 0 /* SOF ABI version number. Format within 32bit word is MMmmmppp */ diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index 76883e6fb750..a9a5c4d0a892 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -57,6 +57,12 @@ #define SOF_TKN_SRC_RATE_IN 300 #define SOF_TKN_SRC_RATE_OUT 301 +/* ASRC */ +#define SOF_TKN_ASRC_RATE_IN 320 +#define SOF_TKN_ASRC_RATE_OUT 321 +#define SOF_TKN_ASRC_ASYNCHRONOUS_MODE 322 +#define SOF_TKN_ASRC_OPERATION_MODE 323 + /* PCM */ #define SOF_TKN_PCM_DMAC_CONFIG 353 -- cgit v1.2.3 From b7c5986489b5c55ebebc430037dd4691a6bbc99c Mon Sep 17 00:00:00 2001 From: Slawomir Blauciak Date: Mon, 9 Dec 2019 18:48:50 -0600 Subject: ASoC: SOF: ipc: channel map structures This change adds stream map and channel map structures used for channel re-routing and stream aggregation. Signed-off-by: Slawomir Blauciak Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191210004854.16845-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/channel_map.h | 61 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 61 insertions(+) create mode 100644 include/sound/sof/channel_map.h (limited to 'include') diff --git a/include/sound/sof/channel_map.h b/include/sound/sof/channel_map.h new file mode 100644 index 000000000000..21044eb5f377 --- /dev/null +++ b/include/sound/sof/channel_map.h @@ -0,0 +1,61 @@ +/* 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) 2019 Intel Corporation. All rights reserved. + */ + +#ifndef __IPC_CHANNEL_MAP_H__ +#define __IPC_CHANNEL_MAP_H__ + +#include +#include + +/** + * \brief Channel map, specifies transformation of one-to-many or many-to-one. + * + * In case of one-to-many specifies how the output channels are computed out of + * a single source channel, + * in case of many-to-one specifies how a single target channel is computed + * from a multichannel input stream. + * + * Channel index specifies position of the channel in the stream on the 'one' + * side. + * + * Ext ID is the identifier of external part of the transformation. Depending + * on the context, it may be pipeline ID, dai ID, ... + * + * Channel mask describes which channels are taken into account on the "many" + * side. Bit[i] set to 1 means that i-th channel is used for computation + * (either as source or as a target). + * + * Channel mask is followed by array of coefficients in Q2.30 format, + * one per each channel set in the mask (left to right, LS bit set in the + * mask corresponds to ch_coeffs[0]). + */ +struct sof_ipc_channel_map { + uint32_t ch_index; + uint32_t ext_id; + uint32_t ch_mask; + uint32_t reserved; + int32_t ch_coeffs[0]; +} __packed; + +/** + * \brief Complete map for each channel of a multichannel stream. + * + * num_ch_map Specifies number of items in the ch_map. + * More than one transformation per a single channel is allowed (in case + * multiple external entities are transformed). + * A channel may be skipped in the transformation list, then it is filled + * with 0's by the transformation function. + */ +struct sof_ipc_stream_map { + struct sof_ipc_cmd_hdr hdr; + uint32_t num_ch_map; + uint32_t reserved[3]; + struct sof_ipc_channel_map ch_map[0]; +} __packed; + +#endif /* __IPC_CHANNEL_MAP_H__ */ -- cgit v1.2.3 From cc73390008c9a47c49ad73c459b5590fd4c4c890 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Dec 2019 09:33:40 +0900 Subject: ASoC: soc-core: remove dai_link_list ASoC is using many lists. Now, used dai_link is listed to card as dai_link_list. [card]->[dai_link]->[dai_link]->... BTW, this "dai_link" is used to create "rtd". And this rtd is listed to card as rtd_list. [card]->[rtd]->[rtd]->... Here, each rtd has dai_link. This means, we can track all dai_link via rtd list. This patch removes card dai_link_list, and uses rtd_list instead of it. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87fthtyq6z.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 7 ------- 1 file changed, 7 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index c28a1ed5e8df..b7ba3b91d080 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -852,7 +852,6 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; - struct list_head list; /* DAI link list of the soc card */ #ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; /* For topology */ #endif @@ -1037,7 +1036,6 @@ struct snd_soc_card { /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; /* predefined links only */ int num_links; /* predefined links only */ - struct list_head dai_link_list; /* all links */ struct list_head rtd_list; int num_rtd; @@ -1107,11 +1105,6 @@ struct snd_soc_card { ((i) < (card)->num_aux_devs) && ((aux) = &(card)->aux_dev[i]); \ (i)++) -#define for_each_card_links(card, link) \ - list_for_each_entry(link, &(card)->dai_link_list, list) -#define for_each_card_links_safe(card, link, _link) \ - list_for_each_entry_safe(link, _link, &(card)->dai_link_list, list) - #define for_each_card_rtds(card, rtd) \ list_for_each_entry(rtd, &(card)->rtd_list, list) #define for_each_card_rtds_safe(card, rtd, _rtd) \ -- cgit v1.2.3 From 8babfb7030573f7338b141d038c2094b7bb95034 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Dec 2019 09:33:55 +0900 Subject: ASoC: soc-core: remove snd_soc_get_dai_substream() No driver is using snd_soc_get_dai_substream(), and snd_soc_get_pcm_runtime() is enough for such purpose. We can revival it if it was needed in the future. Let's remove unused function. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87d0cxyq6k.wl-kuninori.morimoto.gx@renesas.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 b7ba3b91d080..68ec5a051afe 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -464,8 +464,6 @@ static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) void snd_soc_disconnect_sync(struct device *dev); -struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, - const char *dai_link, int stream); struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, const char *dai_link); -- cgit v1.2.3 From 4468189ff307f294491628a49702a04de22bffb8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Dec 2019 09:34:08 +0900 Subject: ASoC: soc-core: find rtd via dai_link pointer at snd_soc_get_pcm_runtime() Current snd_soc_get_pcm_runtime() is finding rtd by checking dai_link name. But, it is strange and waste of CPU power, because its user want to get from rtd from dai_link, not from dai_link name. This patch find rtd via dai_link pointer instead of its name. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87a781yq67.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 68ec5a051afe..40c2a677f531 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -465,7 +465,7 @@ static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) void snd_soc_disconnect_sync(struct device *dev); struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, - const char *dai_link); + struct snd_soc_dai_link *dai_link); bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd); void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream); -- cgit v1.2.3 From d6f31e0e6d09594717ed21c7c9238d9fbdb30ccb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Dec 2019 09:34:14 +0900 Subject: ASoC: soc-core: move snd_soc_find_dai_link() snd_soc_find_dai_link() is soc-topology specific function. We don't need to have it at soc-core. This patch moves it to soc-topology.c Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/878snlyq61.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 40c2a677f531..09d3d9b615c2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1327,9 +1327,6 @@ int snd_soc_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); void snd_soc_remove_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); -struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card, - int id, const char *name, - const char *stream_name); struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv, -- cgit v1.2.3 From 0c04800424c42ec3fbe87422d3e04b5c978fc177 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Dec 2019 09:34:19 +0900 Subject: ASoC: soc-core: rename snd_soc_add_dai_link() to snd_soc_add_pcm_runtime() Now soc-core and soc-topology is using snd_soc_add_dai_link(). The abstract of this function is "create pcm_runtime from dai_link information and connect it to card". Thus, "add dai_link" is wrong/confusable naming. This patch renames function name. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/877e35yq5w.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 09d3d9b615c2..b223b2d39950 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1323,8 +1323,8 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, struct snd_soc_dai_link *dai_link); void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link); -int snd_soc_add_dai_link(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link); +int snd_soc_add_pcm_runtime(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link); void snd_soc_remove_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); -- cgit v1.2.3 From 50cd9b5317d5593d0a33f4227f56ddcc1bf66604 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Dec 2019 09:34:23 +0900 Subject: ASoC: soc-core: rename snd_soc_remove_dai_link() to snd_soc_remove_pcm_runtime() Now soc-core and soc-topology is using snd_soc_remove_dai_link(). It removes pcm_runtime (= rtd) and disconnect it from card. The purpose is removing pcm_runtime, not dai_link. This patch renames function name. Signed-off-by: Kuninori Morimoto Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/875zipyq5s.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index b223b2d39950..3923178ad050 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1325,8 +1325,8 @@ void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link); int snd_soc_add_pcm_runtime(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); -void snd_soc_remove_dai_link(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link); +void snd_soc_remove_pcm_runtime(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd); struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv, -- cgit v1.2.3 From 88452da92ba2b264a3922218c2cec13aac51c502 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Dec 2019 15:57:27 +0100 Subject: ALSA: hda: Use standard waitqueue for RIRB wakeup The HD-audio CORB/RIRB communication was programmed in a way that was documented in the reference in decades ago, which is essentially a polling in the waiter side. It's working fine but costs CPU cycles on some platforms that support only slow communications. Also, for some platforms that had unreliable communications, we put longer wait time (2 ms), which accumulate quite long time if you execute many verbs in a shot (e.g. at the initialization or resume phase). This patch attempts to improve the situation by introducing the standard waitqueue in the RIRB waiter side instead of polling. The test results on my machine show significant improvements. The time spent for "cat /proc/asound/card*/codec#*" were changed like: * Intel SKL + Realtek codec before the patch: 0.00user 0.04system 0:00.10elapsed 40.0%CPU after the patch: 0.00user 0.01system 0:00.10elapsed 10.0%CPU * Nvidia GP107GL + Nvidia HDMI codec before the patch: 0.00user 0.00system 0:02.76elapsed 0.0%CPU after the patch: 0.00user 0.00system 0:00.01elapsed 17.0%CPU So, for Intel chips, the total time is same, while the total time is greatly reduced (from 2.76 to 0.01s) for Nvidia chips. The only negative data here is the increase of CPU time for Nvidia, but this is the unavoidable cost for faster wakeups, supposedly. Link: https://lore.kernel.org/r/20191210145727.22054-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index e05b95e83d5a..81373a2efd96 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -317,6 +317,7 @@ struct hdac_bus { struct hdac_rb corb; struct hdac_rb rirb; unsigned int last_cmd[HDA_MAX_CODECS]; /* last sent command */ + wait_queue_head_t rirb_wq; /* CORB/RIRB and position buffers */ struct snd_dma_buffer rb; -- cgit v1.2.3 From 8a6a6a38f86885982891197840e7b8eccad5ef50 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 11 Dec 2019 13:32:05 +0900 Subject: ASoC: soc-core: tidyup for CONFIG_DMI soc-core.c has 2 #ifdef CONFIG_DMI, but we can merge these. OTOH, soc.h has dmi_longname, but it is needed if CONFIG_DMI was defined. In other words, It is not needed if CONFIG_DMI was not defined. This patch tidyup these. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87eexbbhyy.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index c920f17a5647..a6a3a7d54c70 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -986,7 +986,9 @@ struct snd_soc_card { const char *long_name; const char *driver_name; const char *components; +#ifdef CONFIG_DMI char dmi_longname[80]; +#endif /* CONFIG_DMI */ char topology_shortname[32]; struct device *dev; -- cgit v1.2.3 From 01fec8cce7cc68bb88b1ff55bc03da5f3e6625e3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Dec 2019 15:54:06 +0100 Subject: ASoC: Drop snd_soc_pcm_lib_ioctl() Now all snd_soc_pcm_lib_ioctl() calls were dropped, and it became superfluous. Let's kill it. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20191210145406.21419-24-tiwai@suse.de Signed-off-by: Mark Brown --- include/sound/soc.h | 5 ----- 1 file changed, 5 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a3a7d54c70..82e65235c60d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1399,11 +1399,6 @@ static inline void snd_soc_dapm_mutex_unlock(struct snd_soc_dapm_context *dapm) mutex_unlock(&dapm->card->dapm_mutex); } -/* bypass */ -int snd_soc_pcm_lib_ioctl(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - unsigned int cmd, void *arg); - #include #endif -- cgit v1.2.3 From fcae40c99fb3d09f4407f549a7f17761abe5e1bc Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Tue, 24 Apr 2018 20:06:08 +0800 Subject: ALSA: Replace timespec with timespec64 Since timespec is not year 2038 safe on 32bit system, and we need to convert all timespec variables to timespec64 type for sound subsystem. This patch is used to do preparation for following patches, that will convert all structures defined in uapi/sound/asound.h to use 64-bit time_t. Signed-off-by: Baolin Wang Signed-off-by: Arnd Bergmann --- include/sound/pcm.h | 18 +++++++++--------- include/sound/soc-component.h | 4 ++-- include/sound/timer.h | 4 ++-- 3 files changed, 13 insertions(+), 13 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8a89fa6fdd5e..cb407fade933 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -62,7 +62,7 @@ struct snd_pcm_ops { int (*sync_stop)(struct snd_pcm_substream *substream); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); int (*get_time_info)(struct snd_pcm_substream *substream, - struct timespec *system_ts, struct timespec *audio_ts, + struct timespec64 *system_ts, struct timespec64 *audio_ts, struct snd_pcm_audio_tstamp_config *audio_tstamp_config, struct snd_pcm_audio_tstamp_report *audio_tstamp_report); int (*fill_silence)(struct snd_pcm_substream *substream, int channel, @@ -343,7 +343,7 @@ static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; - struct timespec trigger_tstamp; /* trigger timestamp */ + struct timespec64 trigger_tstamp; /* trigger timestamp */ bool trigger_tstamp_latched; /* trigger timestamp latched in low-level driver/hardware */ int overrange; snd_pcm_uframes_t avail_max; @@ -421,7 +421,7 @@ struct snd_pcm_runtime { /* -- audio timestamp config -- */ struct snd_pcm_audio_tstamp_config audio_tstamp_config; struct snd_pcm_audio_tstamp_report audio_tstamp_report; - struct timespec driver_tstamp; + struct timespec64 driver_tstamp; #if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ @@ -1155,22 +1155,22 @@ static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substrea } /** - * snd_pcm_gettime - Fill the timespec depending on the timestamp mode + * snd_pcm_gettime - Fill the timespec64 depending on the timestamp mode * @runtime: PCM runtime instance - * @tv: timespec to fill + * @tv: timespec64 to fill */ static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime, - struct timespec *tv) + struct timespec64 *tv) { switch (runtime->tstamp_type) { case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC: - ktime_get_ts(tv); + ktime_get_ts64(tv); break; case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: - getrawmonotonic(tv); + ktime_get_raw_ts64(tv); break; default: - getnstimeofday(tv); + ktime_get_real_ts64(tv); break; } } diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 506f72a6b2c2..154d02fbbfed 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -93,8 +93,8 @@ struct snd_soc_component_driver { snd_pcm_uframes_t (*pointer)(struct snd_soc_component *component, struct snd_pcm_substream *substream); int (*get_time_info)(struct snd_soc_component *component, - struct snd_pcm_substream *substream, struct timespec *system_ts, - struct timespec *audio_ts, + struct snd_pcm_substream *substream, struct timespec64 *system_ts, + struct timespec64 *audio_ts, struct snd_pcm_audio_tstamp_config *audio_tstamp_config, struct snd_pcm_audio_tstamp_report *audio_tstamp_report); int (*copy_user)(struct snd_soc_component *component, diff --git a/include/sound/timer.h b/include/sound/timer.h index a53e37bcd746..23e885d31525 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -89,7 +89,7 @@ struct snd_timer_instance { unsigned long ticks, unsigned long resolution); void (*ccallback) (struct snd_timer_instance * timeri, int event, - struct timespec * tstamp, + struct timespec64 * tstamp, unsigned long resolution); void (*disconnect)(struct snd_timer_instance *timeri); void *callback_data; @@ -113,7 +113,7 @@ struct snd_timer_instance { */ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, struct snd_timer **rtimer); -void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstamp); +void snd_timer_notify(struct snd_timer *timer, int event, struct timespec64 *tstamp); int snd_timer_global_new(char *id, int device, struct snd_timer **rtimer); int snd_timer_global_free(struct snd_timer *timer); int snd_timer_global_register(struct snd_timer *timer); -- cgit v1.2.3 From a07804cc7472d8aa5db03ea5d75f3d8d80abb687 Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Tue, 24 Apr 2018 20:06:09 +0800 Subject: ALSA: Avoid using timespec for struct snd_timer_status struct snd_timer_status uses 'timespec' type variables to record timestamp, which will be changed to an incompatible layout with updated user space using 64-bit time_t. To handle both the old and the new layout on 32-bit architectures, this patch introduces 'struct snd_timer_status32' and 'struct snd_timer_status64' to handle 32bit time_t and 64bit time_t in native mode and compat mode, which replaces timespec with s64 type. When glibc changes time_t to 64-bit, any recompiled program will issue ioctl commands that the kernel does not understand without this patch. In the public uapi header, snd_timer_status is now guarded by an #ifndef __KERNEL__ to avoid referencing 'struct timespec'. The timespec definition will be removed from the kernel to prevent new y2038 bugs and to avoid the conflict with an incompatible libc type of the same name. Signed-off-by: Baolin Wang Signed-off-by: Arnd Bergmann --- include/uapi/sound/asound.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index df1153cea0b7..930854f67fd3 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -761,6 +761,7 @@ struct snd_timer_params { unsigned char reserved[60]; /* reserved */ }; +#ifndef __KERNEL__ struct snd_timer_status { struct timespec tstamp; /* Timestamp - last update */ unsigned int resolution; /* current period resolution in ns */ @@ -769,6 +770,7 @@ struct snd_timer_status { unsigned int queue; /* used queue size */ unsigned char reserved[64]; /* reserved */ }; +#endif #define SNDRV_TIMER_IOCTL_PVERSION _IOR('T', 0x00, int) #define SNDRV_TIMER_IOCTL_NEXT_DEVICE _IOWR('T', 0x01, struct snd_timer_id) -- cgit v1.2.3 From a4e7dd35b9dac21fa7b33e8788b51c7fbc7a49f1 Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Tue, 24 Apr 2018 20:06:10 +0800 Subject: ALSA: Avoid using timespec for struct snd_ctl_elem_value The struct snd_ctl_elem_value will use 'timespec' type variables to record timestamp, which is not year 2038 safe on 32bits system. Since there are no drivers will implemented the tstamp member of the struct snd_ctl_elem_value, and also the stucture size will not be changed if we change timespec to s64 for tstamp member of struct snd_ctl_elem_value. From Takashi's comments, "In the library, applications are not expected to access to this structure directly. The applications get opaque pointer to the structure and must use any control APIs to operate it. Actually the library produce no API to handle 'struct snd_ctl_elem_value.tstamp'. This means that we can drop this member from alsa-lib without decline of functionality." Thus we can simply remove the tstamp member to avoid using the type which is not year 2038 safe on 32bits system. Signed-off-by: Baolin Wang Signed-off-by: Arnd Bergmann --- include/uapi/sound/asound.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 930854f67fd3..40a23d8418fe 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -957,8 +957,7 @@ struct snd_ctl_elem_value { } bytes; struct snd_aes_iec958 iec958; } value; /* RO */ - struct timespec tstamp; - unsigned char reserved[128-sizeof(struct timespec)]; + unsigned char reserved[128]; }; struct snd_ctl_tlv { -- cgit v1.2.3 From 3ddee7f88aaf2dee38f7016ac8fd48dd9fdb43e3 Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Tue, 24 Apr 2018 20:06:11 +0800 Subject: ALSA: Avoid using timespec for struct snd_pcm_status The struct snd_pcm_status will use 'timespec' type variables to record timestamp, which is not year 2038 safe on 32bits system. Userspace will use SNDRV_PCM_IOCTL_STATUS and SNDRV_PCM_IOCTL_STATUS_EXT as commands to issue ioctl() to fill the 'snd_pcm_status' structure in userspace. The command number is always defined through _IOR/_IOW/IORW, so when userspace changes the definition of 'struct timespec' to use 64-bit types, the command number also changes. Thus in the kernel, we now need to define two versions of each such ioctl and corresponding ioctl commands to handle 32bit time_t and 64bit time_t in native mode: struct snd_pcm_status32 { ...... s32 trigger_tstamp_sec; s32 trigger_tstamp_nsec; ...... s32 audio_tstamp_sec; s32 audio_tstamp_nsec; ...... }; struct snd_pcm_status64 { ...... s32 trigger_tstamp_sec; s32 trigger_tstamp_nsec; ...... s32 audio_tstamp_sec; s32 audio_tstamp_nsec; ...... }; Moreover in compat file, we renamed or introduced new structures to handle 32bit/64bit time_t in compatible mode. The 'struct snd_pcm_status32' and snd_pcm_status_user32() are used to handle 32bit time_t in compat mode. 'struct compat_snd_pcm_status64' and snd_pcm_status_user_compat64() are used to handle 64bit time_t. The implicit padding before timespec is made explicit to avoid incompatible structure layout between 32-bit and 64-bit x86 due to the different alignment requirements, and the snd_pcm_status structure is now hidden from the kernel to avoid relying on the timespec definitio definitionn Finally we can replace SNDRV_PCM_IOCTL_STATUS and SNDRV_PCM_IOCTL_STATUS_EXT with new commands and introduce new functions to fill new 'struct snd_pcm_status64' instead of using unsafe 'struct snd_pcm_status'. Then in future, the new commands can be matched when userspace changes 'timespec' to 64bit type to make a size change of 'struct snd_pcm_status'. When glibc changes time_t to 64-bit, any recompiled program will issue ioctl commands that the kernel does not understand without this patch. Signed-off-by: Baolin Wang Signed-off-by: Arnd Bergmann --- include/sound/pcm.h | 56 +++++++++++++++++++++++++++++++++++++++++++-- include/uapi/sound/asound.h | 6 +++++ 2 files changed, 60 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cb407fade933..5a31525e2df6 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -44,6 +44,7 @@ struct snd_pcm_hardware { size_t fifo_size; /* fifo size in bytes */ }; +struct snd_pcm_status64; struct snd_pcm_substream; struct snd_pcm_audio_tstamp_config; /* definitions further down */ @@ -558,8 +559,8 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree); int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info); int snd_pcm_info_user(struct snd_pcm_substream *substream, struct snd_pcm_info __user *info); -int snd_pcm_status(struct snd_pcm_substream *substream, - struct snd_pcm_status *status); +int snd_pcm_status64(struct snd_pcm_substream *substream, + struct snd_pcm_status64 *status); int snd_pcm_start(struct snd_pcm_substream *substream); int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status); int snd_pcm_drain_done(struct snd_pcm_substream *substream); @@ -1422,4 +1423,55 @@ static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format) #define pcm_dbg(pcm, fmt, args...) \ dev_dbg((pcm)->card->dev, fmt, ##args) +struct snd_pcm_status64 { + snd_pcm_state_t state; /* stream state */ + u8 rsvd[4]; + s64 trigger_tstamp_sec; /* time when stream was started/stopped/paused */ + s64 trigger_tstamp_nsec; + s64 tstamp_sec; /* reference timestamp */ + s64 tstamp_nsec; + snd_pcm_uframes_t appl_ptr; /* appl ptr */ + snd_pcm_uframes_t hw_ptr; /* hw ptr */ + snd_pcm_sframes_t delay; /* current delay in frames */ + snd_pcm_uframes_t avail; /* number of frames available */ + snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ + snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ + snd_pcm_state_t suspended_state; /* suspended stream state */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + s64 audio_tstamp_sec; /* sample counter, wall clock, PHC or on-demand sync'ed */ + s64 audio_tstamp_nsec; + s64 driver_tstamp_sec; /* useful in case reference system tstamp is reported with delay */ + s64 driver_tstamp_nsec; + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-4*sizeof(s64)]; /* must be filled with zero */ +}; + +#define SNDRV_PCM_IOCTL_STATUS64 _IOR('A', 0x20, struct snd_pcm_status64) +#define SNDRV_PCM_IOCTL_STATUS_EXT64 _IOWR('A', 0x24, struct snd_pcm_status64) + +struct snd_pcm_status32 { + s32 state; /* stream state */ + s32 trigger_tstamp_sec; /* time when stream was started/stopped/paused */ + s32 trigger_tstamp_nsec; + s32 tstamp_sec; /* reference timestamp */ + s32 tstamp_nsec; + u32 appl_ptr; /* appl ptr */ + u32 hw_ptr; /* hw ptr */ + s32 delay; /* current delay in frames */ + u32 avail; /* number of frames available */ + u32 avail_max; /* max frames available on hw since last status */ + u32 overrange; /* count of ADC (capture) overrange detections from last status */ + s32 suspended_state; /* suspended stream state */ + u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + s32 audio_tstamp_sec; /* sample counter, wall clock, PHC or on-demand sync'ed */ + s32 audio_tstamp_nsec; + s32 driver_tstamp_sec; /* useful in case reference system tstamp is reported with delay */ + s32 driver_tstamp_nsec; + u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-4*sizeof(s32)]; /* must be filled with zero */ +}; + +#define SNDRV_PCM_IOCTL_STATUS32 _IOR('A', 0x20, struct snd_pcm_status32) +#define SNDRV_PCM_IOCTL_STATUS_EXT32 _IOWR('A', 0x24, struct snd_pcm_status32) + #endif /* __SOUND_PCM_H */ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 40a23d8418fe..d2c88c098b20 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -456,8 +456,13 @@ enum { SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED }; +#ifndef __KERNEL__ +/* explicit padding avoids incompatibility between i386 and x86-64 */ +typedef struct { unsigned char pad[sizeof(time_t) - sizeof(int)] __time_pad; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ + __time_pad pad1; /* align to timespec */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ struct timespec tstamp; /* reference timestamp */ snd_pcm_uframes_t appl_ptr; /* appl ptr */ @@ -473,6 +478,7 @@ struct snd_pcm_status { __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; +#endif struct snd_pcm_mmap_status { snd_pcm_state_t state; /* RO: state - SNDRV_PCM_STATE_XXXX */ -- cgit v1.2.3 From d9e5582c4bb219f3459e39f65410f0e5128fbe91 Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Tue, 24 Apr 2018 20:06:12 +0800 Subject: ALSA: Avoid using timespec for struct snd_rawmidi_status The struct snd_rawmidi_status will use 'timespec' type variables to record timestamp, which is not year 2038 safe on 32bits system. Thus we introduced 'struct snd_rawmidi_status32' and 'struct snd_rawmidi_status64' to handle 32bit time_t and 64bit time_t in native mode, which replace timespec with s64 type. In compat mode, we renamed or introduced new structures to handle 32bit/64bit time_t in compatible mode. The 'struct snd_rawmidi_status32' and snd_rawmidi_ioctl_status32() are used to handle 32bit time_t in compat mode. 'struct compat_snd_rawmidi_status64' is used to handle 64bit time_t. When glibc changes time_t to 64-bit, any recompiled program will issue ioctl commands that the kernel does not understand without this patch. Signed-off-by: Baolin Wang Signed-off-by: Arnd Bergmann --- include/uapi/sound/asound.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index d2c88c098b20..e0ada33afa1e 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -654,13 +654,16 @@ struct snd_rawmidi_params { unsigned char reserved[16]; /* reserved for future use */ }; +#ifndef __KERNEL__ struct snd_rawmidi_status { int stream; + __time_pad pad1; struct timespec tstamp; /* Timestamp */ size_t avail; /* available bytes */ size_t xruns; /* count of overruns since last status (in bytes) */ unsigned char reserved[16]; /* reserved for future use */ }; +#endif #define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int) #define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info) -- cgit v1.2.3 From 528be501b7d4a64e04672a38ebfc9e19c555e770 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 11 Dec 2019 19:44:57 -0600 Subject: soundwire: sdw_slave: add probe_complete structure and new fields When a Slave device becomes synchronized with the bus, it may report its presence in PING frames, as well as optionally asserting an in-band PREQ signal. The bus driver will detect a new Device0, start the enumeration process and assign it a non-zero device number. The SoundWire enumeration provides an arbitration to deal with multiple Slaves reporting ATTACHED at the same time. The bus driver will also invoke the driver .probe() callback associated with this device. The probe() depends on the Linux device core, which handles the match operations and may result in modules being loaded. Once the non-zero device number is programmed, the Slave will report its new status in PING frames and the Master hardware will typically report this status change with an interrupt. At this point, the .update_status() callback of the codec driver will be invoked (usually from an interrupt thread or workqueue scheduled from the interrupt thread). The first race condition which can happen is between the .probe(), which allocates the resources, and .update_status() where initializations are typically handled. The .probe() is only called once during the initial boot, while .update_status() will be called for every bus hardware reset and if the Slave device loses synchronization (an unlikely event but with non-zero probability). The time difference between the end of the enumeration process and a change of status reported by the hardware may be as small as one SoundWire PING frame. The scheduling of the interrupt thread, which invokes .update_status() is not deterministic, but can be small enough to create a race condition. With a 48 kHz frame rate and ideal scheduling cases, the .probe() may be pre-empted within double-digit microseconds. Since there is no guarantee that the .probe() completes by the time .update_status() is invoked as a result of an interrupt, it's not unusual for the .update_status() to rely on data structures that have not been allocated yet, leading to kernel oopses. This patch adds a probe_complete utility, which is used in the sdw_update_slave_status() routine. The codec driver does not need to do anything and can safely assume all resources are allocated in its update_status() callback. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 28745b9ba279..cb1db4a7475d 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -547,6 +547,10 @@ struct sdw_slave_ops { * @node: node for bus list * @port_ready: Port ready completion flag for each Slave port * @dev_num: Device Number assigned by Bus + * @probed: boolean tracking driver state + * @probe_complete: completion utility to control potential races + * on startup between driver probe/initialization and SoundWire + * Slave state changes/implementation-defined interrupts */ struct sdw_slave { struct sdw_slave_id id; @@ -561,6 +565,8 @@ struct sdw_slave { struct list_head node; struct completion *port_ready; u16 dev_num; + bool probed; + struct completion probe_complete; }; #define dev_to_sdw_dev(_dev) container_of(_dev, struct sdw_slave, dev) -- cgit v1.2.3 From fbbff36325079fd9d2fcd30063c84f4b38a0ad9b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 11 Dec 2019 19:44:58 -0600 Subject: soundwire: sdw_slave: add enumeration_complete structure When the Master starts the bus (be it during the initial boot or system resume), it usually performs a HardReset to make sure electrical levels are correct, then enables the control channel. While the PM framework guarantees that the Slave devices will only become 'active' once the Master completes the bus initialization, there is still a risk of a race condition: the Slave enumeration is handled in a separate interrupt thread triggered by hardware status changes, so the Slave device may not be ready to accept commands when the Slave driver tries to access the registers and restore settings in its resume or pm_runtime_resume callbacks. In those cases, any read/write commands from/to the Slave device will result in a timeout. This patch adds an enumeration_complete structure. When the bus is goes through a HardReset sequence and restarted, the Slave will be marked as UNATTACHED, which will result in a call to init_completion(). When the Slave reports its presence during PING frames as a non-zero Device, the Master hardware will issue an interrupt and the bus driver will invoke complete(). The order between init_completion()/complete() is predictable since this is a Master-initiated transition. The Slave driver may use wait_for_completion() in its resume callback. When regmap is used, the Slave driver will typically set its regmap in cache-only mode on suspend, then on resume block on wait_for_completion(&enumeration_complete) to guarantee it is safe to start read/write transactions. It may then exit the cache-only mode and use a regmap_sync to restore settings. All these steps are optional, their use completely depends on the Slave device capabilities and how the Slave driver is implemented. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index cb1db4a7475d..3fa8d875b16b 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -551,6 +551,9 @@ struct sdw_slave_ops { * @probe_complete: completion utility to control potential races * on startup between driver probe/initialization and SoundWire * Slave state changes/implementation-defined interrupts + * @enumeration_complete: completion utility to control potential races + * on startup between device enumeration and read/write access to the + * Slave device */ struct sdw_slave { struct sdw_slave_id id; @@ -567,6 +570,7 @@ struct sdw_slave { u16 dev_num; bool probed; struct completion probe_complete; + struct completion enumeration_complete; }; #define dev_to_sdw_dev(_dev) container_of(_dev, struct sdw_slave, dev) -- cgit v1.2.3 From 7afc50e441af0afc8055920a64cff70b648e4b44 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 11 Dec 2019 19:44:59 -0600 Subject: soundwire: sdw_slave: add initialization_complete definition Slave drivers may have different ways of handling their settings, with or without regmap. During the integration of codec drivers, done in partnership between Intel and Realtek, it became desirable to implement a predictable order between low-level initializations performed in .update_status() (invoked by an interrupt thread) and the settings restored in the resume steps (invoked by the PM core). This patch builds on the previous solution to wait for the Slave device to be fully enumerated. The complete() in this case is signaled not before the .update_status() is called, but after .update_status() returns. Without this patch, the settings were not properly restored, leading to timing-dependent 'no sound after resume' or 'no headset detected after resume' bug reports. Depending on how initialization is handled, a Slave device driver may wait for enumeration_complete, or for initialization_complete, both are valid synchronization points. They are initialized at the same time, they only differ on when complete() is invoked. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 3fa8d875b16b..ed42cd79eab7 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -554,6 +554,8 @@ struct sdw_slave_ops { * @enumeration_complete: completion utility to control potential races * on startup between device enumeration and read/write access to the * Slave device + * @initialization_complete: completion utility to control potential races + * on startup between device enumeration and settings being restored */ struct sdw_slave { struct sdw_slave_id id; @@ -571,6 +573,7 @@ struct sdw_slave { bool probed; struct completion probe_complete; struct completion enumeration_complete; + struct completion initialization_complete; }; #define dev_to_sdw_dev(_dev) container_of(_dev, struct sdw_slave, dev) -- cgit v1.2.3 From b2bd75f806c49929d7ab5a860c0a69b0a17c59d2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 11 Dec 2019 19:45:00 -0600 Subject: soundwire: sdw_slave: track unattach_request to handle all init sequences The Slave device initialization can be split in 4 different cases: 1. Master-initiated hardware reset, system suspend-resume and pm_runtime based on clock-stop mode1. To avoid timeouts and a bad audio experience, the Slave device resume operations need to wait for the Slave device to be re-enumerated and its settings restored. 2. Exit from clock-stop mode0. In this case, the Slave device is required to remain enumerated and its context preserved while the clock is stopped, so no re-initialization or wait_for_completion() is necessary. 3. Slave-initiated pm_runtime D3 transition. With the parent child relationship, it is possible that a Slave device becomes 'suspended' while its parent is still 'active' with the bus clock still toggling. In this case, during the pm_runtime resume operation, there is no need to wait for any settings to be restored. 4. Slave reset (sync loss or implementation-defined). In that case the bus remains operational and the Slave device will be re-initialized when it becomes ATTACHED again. In previous patches, we suggested the use of wait_for_completion() to deal with the case #1, but case #2 and #3 do not need any wait. To account for those differences, this patch adds an unattach_request field. The field is explicitly set by the Master for the case #1, and if non-zero the Slave device shall wait on resume. In all other cases, the Slave resume operations can proceed without wait. The only request tracked so far is Master HardReset, but the request is declared as a bit mask for future extensions (if needed). The definition for this value is added in bus.h and does not need to be exposed in sdw.h Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ed42cd79eab7..b7c9eca4332a 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -556,6 +556,11 @@ struct sdw_slave_ops { * Slave device * @initialization_complete: completion utility to control potential races * on startup between device enumeration and settings being restored + * @unattach_request: mask field to keep track why the Slave re-attached and + * was re-initialized. This is useful to deal with potential race conditions + * between the Master suspending and the codec resuming, and make sure that + * when the Master triggered a reset the Slave is properly enumerated and + * initialized */ struct sdw_slave { struct sdw_slave_id id; @@ -574,6 +579,7 @@ struct sdw_slave { struct completion probe_complete; struct completion enumeration_complete; struct completion initialization_complete; + u32 unattach_request; }; #define dev_to_sdw_dev(_dev) container_of(_dev, struct sdw_slave, dev) -- cgit v1.2.3 From f98f690fb03c2a8d21dfa31aa1042480cf6f7f9b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 11 Dec 2019 19:45:01 -0600 Subject: soundwire: intel: update interfaces between ASoC and SoundWire The current interfaces between ASoC and SoundWire are limited by the platform_device infrastructure to an init() and exit() (mapped to the platform driver.probe and .remove) To help with the platform detection, machine driver selection and management of power dependencies between DSP and SoundWire IP, the ASoC side requires: a) an ACPI scan helper, to report if any devices are exposed in the DSDT tables, and if any links are disabled by the BIOS. b) a probe helper that allocates the resources without actually starting the bus. c) a startup helper which does start the bus when all power dependencies are settled. d) an exit helper to free all resources e) an interrupt_enable/disable helper, typically invoked after the startup helper but also used in suspend routines. This patch moves all required interfaces to sdw_intel.h, mainly to allow SoundWire and ASoC parts to be merged separately once the header files are shared between trees. To avoid compilation issues, the conflicts in intel_init.c are blindly removed. This would in theory prevent the code from working, but since there are no users of the Intel Soundwire driver this has no impact. Functionality will be restored when the removal of platform devices is complete. Support for SoundWire + SOF builds will only be provided once all the required pieces are upstream. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw_intel.h | 77 ++++++++++++++++++++++++++++++++++--- 1 file changed, 72 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index c9427cb6020b..034eca8df748 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -16,24 +16,91 @@ struct sdw_intel_ops { }; /** - * struct sdw_intel_res - Soundwire Intel resource structure + * struct sdw_intel_acpi_info - Soundwire Intel information found in ACPI tables + * @handle: ACPI controller handle + * @count: link count found with "sdw-master-count" property + * @link_mask: bit-wise mask listing links enabled by BIOS menu + * + * this structure could be expanded to e.g. provide all the _ADR + * information in case the link_mask is not sufficient to identify + * platform capabilities. + */ +struct sdw_intel_acpi_info { + acpi_handle handle; + int count; + u32 link_mask; +}; + +struct sdw_intel_link_res; + +/** + * struct sdw_intel_ctx - context allocated by the controller + * driver probe + * @count: link count + * @mmio_base: mmio base of SoundWire registers, only used to check + * hardware capabilities after all power dependencies are settled. + * @link_mask: bit-wise mask listing SoundWire links reported by the + * Controller + * @handle: ACPI parent handle + * @links: information for each link (controller-specific and kept + * opaque here) + */ +struct sdw_intel_ctx { + int count; + void __iomem *mmio_base; + u32 link_mask; + acpi_handle handle; + struct sdw_intel_link_res *links; +}; + +/** + * struct sdw_intel_res - Soundwire Intel global resource structure, + * typically populated by the DSP driver + * + * @count: link count * @mmio_base: mmio base of SoundWire registers * @irq: interrupt number * @handle: ACPI parent handle * @parent: parent device * @ops: callback ops - * @arg: callback arg + * @dev: device implementing hwparams and free callbacks + * @link_mask: bit-wise mask listing links selected by the DSP driver + * This mask may be a subset of the one reported by the controller since + * machine-specific quirks are handled in the DSP driver. */ struct sdw_intel_res { + int count; void __iomem *mmio_base; int irq; acpi_handle handle; struct device *parent; const struct sdw_intel_ops *ops; - void *arg; + struct device *dev; + u32 link_mask; }; -void *sdw_intel_init(acpi_handle *parent_handle, struct sdw_intel_res *res); -void sdw_intel_exit(void *arg); +/* + * On Intel platforms, the SoundWire IP has dependencies on power + * rails shared with the DSP, and the initialization steps are split + * in three. First an ACPI scan to check what the firmware describes + * in DSDT tables, then an allocation step (with no hardware + * configuration but with all the relevant devices created) and last + * the actual hardware configuration. The final stage is a global + * interrupt enable which is controlled by the DSP driver. Splitting + * these phases helps simplify the boot flow and make early decisions + * on e.g. which machine driver to select (I2S mode, HDaudio or + * SoundWire). + */ +int sdw_intel_acpi_scan(acpi_handle *parent_handle, + struct sdw_intel_acpi_info *info); + +struct sdw_intel_ctx * +sdw_intel_probe(struct sdw_intel_res *res); + +int sdw_intel_startup(struct sdw_intel_ctx *ctx); + +void sdw_intel_exit(struct sdw_intel_ctx *ctx); + +void sdw_intel_enable_irq(void __iomem *mmio_base, bool enable); #endif -- cgit v1.2.3 From 4b206d34b92224496c42226c4b6d92719056c8b6 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Wed, 11 Dec 2019 19:45:02 -0600 Subject: soundwire: intel: update stream callbacks for hwparams/free stream operations The SoundWire DAIs for Intel platform are created in drivers/soundwire/intel.c, while the communication with the Intel DSP is all controlled in soc/sof/intel When the DAI status changes, a callback is used to bridge the gap between the two subsystems. The naming of the existing 'config_stream' callback does not map well with any of ALSA/ASoC concepts. This patch renames it as 'params_stream' to be more self-explanatory. A new 'free_stream' callback is added in case any resources allocated in the 'params_stream' stage need to be released. In the SOF implementation, this is used in the hw_free case to release the DMA channels over IPC. These two callbacks now rely on structures which expose the link_id and alh_stream_id (required by the firmware IPC), instead of a list of parameters. The 'void *' definitions are changed to use explicit types, as suggested on alsa-devel during earlier reviews. Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw_intel.h | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 034eca8df748..3ccb38d48eef 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -4,15 +4,39 @@ #ifndef __SDW_INTEL_H #define __SDW_INTEL_H +/** + * struct sdw_intel_stream_params_data: configuration passed during + * the @params_stream callback, e.g. for interaction with DSP + * firmware. + */ +struct sdw_intel_stream_params_data { + struct snd_pcm_substream *substream; + struct snd_soc_dai *dai; + struct snd_pcm_hw_params *hw_params; + int link_id; + int alh_stream_id; +}; + +/** + * struct sdw_intel_stream_free_data: configuration passed during + * the @free_stream callback, e.g. for interaction with DSP + * firmware. + */ +struct sdw_intel_stream_free_data { + struct snd_pcm_substream *substream; + struct snd_soc_dai *dai; + int link_id; +}; + /** * struct sdw_intel_ops: Intel audio driver callback ops * - * @config_stream: configure the stream with the hw_params - * the first argument containing the context is mandatory */ struct sdw_intel_ops { - int (*config_stream)(void *arg, void *substream, - void *dai, void *hw_params, int stream_num); + int (*params_stream)(struct device *dev, + struct sdw_intel_stream_params_data *params_data); + int (*free_stream)(struct device *dev, + struct sdw_intel_stream_free_data *free_data); }; /** -- cgit v1.2.3 From 6cd1d670bee641d5d10b11d58c7c99ac1ddf8068 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 11 Dec 2019 19:45:03 -0600 Subject: soundwire: intel: update headers for interrupts The existing use of 6 handlers is problematic in MSI mode. Update headers so that all shared interrupts can be handled with a single handler. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw_intel.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 3ccb38d48eef..2ce3e9ecc4b6 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -4,6 +4,8 @@ #ifndef __SDW_INTEL_H #define __SDW_INTEL_H +#include + /** * struct sdw_intel_stream_params_data: configuration passed during * the @params_stream callback, e.g. for interaction with DSP @@ -127,4 +129,6 @@ void sdw_intel_exit(struct sdw_intel_ctx *ctx); void sdw_intel_enable_irq(void __iomem *mmio_base, bool enable); +irqreturn_t sdw_intel_thread(int irq, void *dev_id); + #endif -- cgit v1.2.3 From eae0b60d64834c75a460d96b1d1e0e187381e341 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 11 Dec 2019 19:45:04 -0600 Subject: soundwire: intel: add link_list to handle interrupts with a single thread In MSI mode, the use of separate handlers and threads for the Intel IPC, stream and SoundWire shared interrupt leads to timeouts and lost interrupts. The solution is to merge all interrupt handling across all links with a single thread function. The use of a linked list enables this thread function to walk through all contexts and figure out which link needs attention. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw_intel.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 2ce3e9ecc4b6..2a56180bc9dc 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -70,6 +70,7 @@ struct sdw_intel_link_res; * @handle: ACPI parent handle * @links: information for each link (controller-specific and kept * opaque here) + * @link_list: list to handle interrupts across all links */ struct sdw_intel_ctx { int count; @@ -77,6 +78,7 @@ struct sdw_intel_ctx { u32 link_mask; acpi_handle handle; struct sdw_intel_link_res *links; + struct list_head link_list; }; /** -- cgit v1.2.3 From 905b5a81afe15e8252e5892b8ca1ff1c1adfb79d Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Wed, 11 Dec 2019 19:45:05 -0600 Subject: soundwire: intel: add prototype for WAKEEN interrupt processing In ClockStop mode, the PCI device will be notified of a wake, which will be handled from an interrupt thread. Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw_intel.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 2a56180bc9dc..073121c49695 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -122,6 +122,8 @@ struct sdw_intel_res { int sdw_intel_acpi_scan(acpi_handle *parent_handle, struct sdw_intel_acpi_info *info); +void sdw_intel_process_wakeen_event(struct sdw_intel_ctx *ctx); + struct sdw_intel_ctx * sdw_intel_probe(struct sdw_intel_res *res); -- cgit v1.2.3 From 4da0680f24c9af2de8406ded68c4ef967f448de3 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 11 Dec 2019 19:45:06 -0600 Subject: soundwire: intel: add mutex for shared SHIM register access Some of the Intel SoundWire SHIM registers contain fields for different links. Without protection, the master drivers for the different links will access these shared registers, leading to invalid configurations and timeouts (specifically when changing CPA/SPA power-related registers and polling for the changes to be applied). A mutex is added to make sure all rmw access to those registers are serialized. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw_intel.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 073121c49695..45fa6d93197f 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -71,6 +71,7 @@ struct sdw_intel_link_res; * @links: information for each link (controller-specific and kept * opaque here) * @link_list: list to handle interrupts across all links + * @shim_lock: mutex to handle concurrent rmw access to shared SHIM registers. */ struct sdw_intel_ctx { int count; @@ -79,6 +80,7 @@ struct sdw_intel_ctx { acpi_handle handle; struct sdw_intel_link_res *links; struct list_head link_list; + struct mutex shim_lock; /* lock for access to shared SHIM registers */ }; /** -- cgit v1.2.3 From 09f6a72d014386939d21899921dd379006471a4b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 11 Dec 2019 19:45:07 -0600 Subject: soundwire: intel: add clock stop quirks Due to power rail dependencies, the SoundWire Master driver cannot make decisions on its own when entering pm runtime suspend. Add quirk mask for each link, so that the SOF parent driver can inform the SoundWire master driver of the desired behavior: a) leave clock on b) power-off instead of clock stop c) power-off if all devices cannot generate wakes d) force bus reset on clock restart Note that for now the interface with the SOF driver relies on a single mask for all links. If needed, the interface might be modified at a later point to provide more freedom. The code at the lower level does not assume any commonality between links. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191212014507.28050-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul --- include/linux/soundwire/sdw_intel.h | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) (limited to 'include') diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 45fa6d93197f..93b83bdf8035 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -59,6 +59,40 @@ struct sdw_intel_acpi_info { struct sdw_intel_link_res; +/* Intel clock-stop/pm_runtime quirk definitions */ + +/* + * Force the clock to remain on during pm_runtime suspend. This might + * be needed if Slave devices do not have an alternate clock source or + * if the latency requirements are very strict. + */ +#define SDW_INTEL_CLK_STOP_NOT_ALLOWED BIT(0) + +/* + * Stop the bus during pm_runtime suspend. If set, a complete bus + * reset and re-enumeration will be performed when the bus + * restarts. This mode shall not be used if Slave devices can generate + * in-band wakes. + */ +#define SDW_INTEL_CLK_STOP_TEARDOWN BIT(1) + +/* + * Stop the bus during pm_suspend if Slaves are not wake capable + * (e.g. speaker amplifiers). The clock-stop mode is typically + * slightly higher power than when the IP is completely powered-off. + */ +#define SDW_INTEL_CLK_STOP_WAKE_CAPABLE_ONLY BIT(2) + +/* + * Require a bus reset (and complete re-enumeration) when exiting + * clock stop modes. This may be needed if the controller power was + * turned off and all context lost. This quirk shall not be used if a + * Slave device needs to remain enumerated and keep its context, + * e.g. to provide the reasons for the wake, report acoustic events or + * pass a history buffer. + */ +#define SDW_INTEL_CLK_STOP_BUS_RESET BIT(3) + /** * struct sdw_intel_ctx - context allocated by the controller * driver probe @@ -97,6 +131,8 @@ struct sdw_intel_ctx { * @link_mask: bit-wise mask listing links selected by the DSP driver * This mask may be a subset of the one reported by the controller since * machine-specific quirks are handled in the DSP driver. + * @clock_stop_quirks: mask array of possible behaviors requested by the + * DSP driver. The quirks are common for all links for now. */ struct sdw_intel_res { int count; @@ -107,6 +143,7 @@ struct sdw_intel_res { const struct sdw_intel_ops *ops; struct device *dev; u32 link_mask; + u32 clock_stop_quirks; }; /* -- cgit v1.2.3 From 07094ae6f9527279de6fd0c59e88f6d0423585b1 Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Tue, 24 Apr 2018 20:06:13 +0800 Subject: ALSA: Avoid using timespec for struct snd_timer_tread The struct snd_timer_tread will use 'timespec' type variables to record timestamp, which is not year 2038 safe on 32bits system. Since the struct snd_timer_tread is passed through read() rather than ioctl(), and the read syscall has no command number that lets us pick between the 32-bit or 64-bit version of this structure. Thus we introduced one new command SNDRV_TIMER_IOCTL_TREAD64 and new struct snd_timer_tread64 replacing timespec with s64 type to handle 64bit time_t. That means we will set tu->tread = TREAD_FORMAT_64BIT when user space has a 64bit time_t, then we will copy to user with struct snd_timer_tread64. Otherwise we will use 32bit time_t variables when copying to user. Moreover this patch replaces timespec type with timespec64 type and related y2038 safe APIs. Signed-off-by: Baolin Wang Signed-off-by: Arnd Bergmann --- include/uapi/sound/asound.h | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index e0ada33afa1e..ad86c5a7a1e2 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -783,7 +783,7 @@ struct snd_timer_status { #define SNDRV_TIMER_IOCTL_PVERSION _IOR('T', 0x00, int) #define SNDRV_TIMER_IOCTL_NEXT_DEVICE _IOWR('T', 0x01, struct snd_timer_id) -#define SNDRV_TIMER_IOCTL_TREAD _IOW('T', 0x02, int) +#define SNDRV_TIMER_IOCTL_TREAD_OLD _IOW('T', 0x02, int) #define SNDRV_TIMER_IOCTL_GINFO _IOWR('T', 0x03, struct snd_timer_ginfo) #define SNDRV_TIMER_IOCTL_GPARAMS _IOW('T', 0x04, struct snd_timer_gparams) #define SNDRV_TIMER_IOCTL_GSTATUS _IOWR('T', 0x05, struct snd_timer_gstatus) @@ -796,6 +796,15 @@ struct snd_timer_status { #define SNDRV_TIMER_IOCTL_STOP _IO('T', 0xa1) #define SNDRV_TIMER_IOCTL_CONTINUE _IO('T', 0xa2) #define SNDRV_TIMER_IOCTL_PAUSE _IO('T', 0xa3) +#define SNDRV_TIMER_IOCTL_TREAD64 _IOW('T', 0xa4, int) + +#if __BITS_PER_LONG == 64 +#define SNDRV_TIMER_IOCTL_TREAD SNDRV_TIMER_IOCTL_TREAD_OLD +#else +#define SNDRV_TIMER_IOCTL_TREAD ((sizeof(__kernel_long_t) >= sizeof(time_t)) ? \ + SNDRV_TIMER_IOCTL_TREAD_OLD : \ + SNDRV_TIMER_IOCTL_TREAD64) +#endif struct snd_timer_read { unsigned int resolution; @@ -821,11 +830,15 @@ enum { SNDRV_TIMER_EVENT_MRESUME = SNDRV_TIMER_EVENT_RESUME + 10, }; +#ifndef __KERNEL__ struct snd_timer_tread { int event; + __time_pad pad1; struct timespec tstamp; unsigned int val; + __time_pad pad2; }; +#endif /**************************************************************************** * * -- cgit v1.2.3 From 80fe7430c7085951d1246d83f638cc17e6c0be36 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 24 Apr 2018 20:06:15 +0800 Subject: ALSA: add new 32-bit layout for snd_pcm_mmap_status/control The snd_pcm_mmap_status and snd_pcm_mmap_control interfaces are one of the trickiest areas to get right when moving to 64-bit time_t in user space. The snd_pcm_mmap_status structure layout is incompatible with user space that uses a 64-bit time_t, so we need a new layout for it. Since the SNDRV_PCM_IOCTL_SYNC_PTR ioctl combines it with snd_pcm_mmap_control into snd_pcm_sync_ptr, we need to change those two as well. Both structures are also exported via an mmap() operation on certain architectures, and this suffers from incompatibility between 32-bit and 64-bit user space. As we have to change both structures anyway, this is a good opportunity to fix the mmap() problem as well, so let's standardize on the existing 64-bit layout of the structure where possible. The downside is that we lose mmap() support for existing 32-bit x86 and powerpc applications, adding that would introduce very noticeable runtime overhead and complexity. My assumption here is that not too many people will miss the removed feature, given that: - Almost all x86 and powerpc users these days are on 64-bit kernels, the majority of today's 32-bit users are on architectures that never supported mmap (ARM, MIPS, ...). - It never worked in compat mode (it was intentionally disabled there) - The application already needs to work with a fallback to SNDRV_PCM_IOCTL_SYNC_PTR, which will keep working with both the old and new structure layout. Both the ioctl() and mmap() based interfaces are changed at the same time, as they are based on the same structures. Unlike other interfaces, we change the uapi header to export both the traditional structure and a version that is portable between 32-bit and 64-bit user space code and that corresponds to the existing 64-bit layout. We further check the __USE_TIME_BITS64 macro that will be defined by future C library versions whenever we use the new time_t definition, so any existing user space source code will not see any changes until it gets rebuilt against a new C library. However, the new structures are all visible in addition to the old ones, allowing applications to explicitly request the new structures. In order to detect the difference between the old snd_pcm_mmap_status and the new __snd_pcm_mmap_status64 structure from the ioctl command number, we rely on one quirk in the structure definition: snd_pcm_mmap_status must be aligned to alignof(time_t), which leads the compiler to insert four bytes of padding in struct snd_pcm_sync_ptr after 'flags' and a corresponding change in the size of snd_pcm_sync_ptr itself. On x86-32 (and only there), the compiler doesn't use 64-bit alignment in structure, so I'm adding an explicit pad in the structure that has no effect on the existing 64-bit architectures but ensures that the layout matches for x86. The snd_pcm_uframes_t type compatibility requires another hack: we can't easily make that 64 bit wide, so I leave the type as 'unsigned long', but add padding before and after it, to ensure that the data is properly aligned to the respective 64-bit field in the in-kernel structure. For the SNDRV_PCM_MMAP_OFFSET_STATUS/CONTROL constants that are used as the virtual file offset in the mmap() function, we also have to introduce new constants that depend on hte __USE_TIME_BITS64 macro: The existing macros are renamed to SNDRV_PCM_MMAP_OFFSET_STATUS_OLD and SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD, they continue to work fine on 64-bit architectures, but stop working on native 32-bit user space. The replacement _NEW constants are now used by default for user space built with __USE_TIME_BITS64, those now work on all new kernels for x86, ppc and alpha (32 and 64 bit, native and compat). It might be a good idea for a future alsa-lib to support both the _OLD and _NEW macros and use the corresponding structures directly. Unmodified alsa-lib source code will retain the current behavior, so it will no longer be able to use mmap() for the status/control structures on 32-bit systems, until either the C library gets updated to 64-bit time_t or alsa-lib gets updated to support both mmap() layouts. Co-developed-with: Baolin Wang Signed-off-by: Baolin Wang Signed-off-by: Arnd Bergmann --- include/uapi/sound/asound.h | 110 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 100 insertions(+), 10 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index ad86c5a7a1e2..df9983e7ead5 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -35,6 +35,8 @@ #include #endif +#include + /* * protocol version */ @@ -301,7 +303,9 @@ typedef int __bitwise snd_pcm_subformat_t; #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 */ - +#if (__BITS_PER_LONG == 32 && defined(__USE_TIME_BITS64)) || defined __KERNEL__ +#define __SND_STRUCT_TIME64 +#endif typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ @@ -317,8 +321,17 @@ typedef int __bitwise snd_pcm_state_t; enum { SNDRV_PCM_MMAP_OFFSET_DATA = 0x00000000, - SNDRV_PCM_MMAP_OFFSET_STATUS = 0x80000000, - SNDRV_PCM_MMAP_OFFSET_CONTROL = 0x81000000, + SNDRV_PCM_MMAP_OFFSET_STATUS_OLD = 0x80000000, + SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD = 0x81000000, + SNDRV_PCM_MMAP_OFFSET_STATUS_NEW = 0x82000000, + SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW = 0x83000000, +#ifdef __SND_STRUCT_TIME64 + SNDRV_PCM_MMAP_OFFSET_STATUS = SNDRV_PCM_MMAP_OFFSET_STATUS_NEW, + SNDRV_PCM_MMAP_OFFSET_CONTROL = SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW, +#else + SNDRV_PCM_MMAP_OFFSET_STATUS = SNDRV_PCM_MMAP_OFFSET_STATUS_OLD, + SNDRV_PCM_MMAP_OFFSET_CONTROL = SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD, +#endif }; union snd_pcm_sync_id { @@ -480,16 +493,46 @@ struct snd_pcm_status { }; #endif -struct snd_pcm_mmap_status { +/* + * For mmap operations, we need the 64-bit layout, both for compat mode, + * and for y2038 compatibility. For 64-bit applications, the two definitions + * are identical, so we keep the traditional version. + */ +#ifdef __SND_STRUCT_TIME64 +#define __snd_pcm_mmap_status64 snd_pcm_mmap_status +#define __snd_pcm_mmap_control64 snd_pcm_mmap_control +#define __snd_pcm_sync_ptr64 snd_pcm_sync_ptr +#ifdef __KERNEL__ +#define __snd_timespec64 __kernel_timespec +#else +#define __snd_timespec64 timespec +#endif +struct __snd_timespec { + __s32 tv_sec; + __s32 tv_nsec; +}; +#else +#define __snd_pcm_mmap_status snd_pcm_mmap_status +#define __snd_pcm_mmap_control snd_pcm_mmap_control +#define __snd_pcm_sync_ptr snd_pcm_sync_ptr +#define __snd_timespec timespec +struct __snd_timespec64 { + __s64 tv_sec; + __s64 tv_nsec; +}; + +#endif + +struct __snd_pcm_mmap_status { snd_pcm_state_t state; /* RO: state - SNDRV_PCM_STATE_XXXX */ int pad1; /* Needed for 64 bit alignment */ snd_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ - struct timespec tstamp; /* Timestamp */ + struct __snd_timespec tstamp; /* Timestamp */ snd_pcm_state_t suspended_state; /* RO: suspended stream state */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ + struct __snd_timespec audio_tstamp; /* from sample counter or wall clock */ }; -struct snd_pcm_mmap_control { +struct __snd_pcm_mmap_control { snd_pcm_uframes_t appl_ptr; /* RW: appl ptr (0...boundary-1) */ snd_pcm_uframes_t avail_min; /* RW: min available frames for wakeup */ }; @@ -498,14 +541,59 @@ struct snd_pcm_mmap_control { #define SNDRV_PCM_SYNC_PTR_APPL (1<<1) /* get appl_ptr from driver (r/w op) */ #define SNDRV_PCM_SYNC_PTR_AVAIL_MIN (1<<2) /* get avail_min from driver */ -struct snd_pcm_sync_ptr { +struct __snd_pcm_sync_ptr { unsigned int flags; union { - struct snd_pcm_mmap_status status; + struct __snd_pcm_mmap_status status; + unsigned char reserved[64]; + } s; + union { + struct __snd_pcm_mmap_control control; + unsigned char reserved[64]; + } c; +}; + +#if defined(__BYTE_ORDER) ? __BYTE_ORDER == __BIG_ENDIAN : defined(__BIG_ENDIAN) +typedef char __pad_before_uframe[sizeof(__u64) - sizeof(snd_pcm_uframes_t)]; +typedef char __pad_after_uframe[0]; +#endif + +#if defined(__BYTE_ORDER) ? __BYTE_ORDER == __LITTLE_ENDIAN : defined(__LITTLE_ENDIAN) +typedef char __pad_before_uframe[0]; +typedef char __pad_after_uframe[sizeof(__u64) - sizeof(snd_pcm_uframes_t)]; +#endif + +struct __snd_pcm_mmap_status64 { + __s32 state; /* RO: state - SNDRV_PCM_STATE_XXXX */ + __u32 pad1; /* Needed for 64 bit alignment */ + __pad_before_uframe __pad1; + snd_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ + __pad_after_uframe __pad2; + struct __snd_timespec64 tstamp; /* Timestamp */ + __s32 suspended_state; /* RO: suspended stream state */ + __u32 pad3; /* Needed for 64 bit alignment */ + struct __snd_timespec64 audio_tstamp; /* sample counter or wall clock */ +}; + +struct __snd_pcm_mmap_control64 { + __pad_before_uframe __pad1; + snd_pcm_uframes_t appl_ptr; /* RW: appl ptr (0...boundary-1) */ + __pad_before_uframe __pad2; + + __pad_before_uframe __pad3; + snd_pcm_uframes_t avail_min; /* RW: min available frames for wakeup */ + __pad_after_uframe __pad4; +}; + +struct __snd_pcm_sync_ptr64 { + __u32 flags; + __u32 pad1; + union { + struct __snd_pcm_mmap_status64 status; unsigned char reserved[64]; } s; union { - struct snd_pcm_mmap_control control; + struct __snd_pcm_mmap_control64 control; unsigned char reserved[64]; } c; }; @@ -590,6 +678,8 @@ enum { #define SNDRV_PCM_IOCTL_STATUS _IOR('A', 0x20, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) +#define __SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct __snd_pcm_sync_ptr) +#define __SNDRV_PCM_IOCTL_SYNC_PTR64 _IOWR('A', 0x23, struct __snd_pcm_sync_ptr64) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) #define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) -- cgit v1.2.3 From 1cfaef9617033f38eba9cc725809ed32bcdb3dc5 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 13 Nov 2019 17:49:14 +0100 Subject: ALSA: bump uapi version numbers Change SNDRV_PCM_VERSION, SNDRV_RAWMIDI_VERSION and SNDRV_TIMER_VERSION to indicate the addition of the time64 version of the mmap interface and these ioctl commands: SNDRV_PCM_IOCTL_SYNC SNDRV_RAWMIDI_IOCTL_STATUS SNDRV_PCM_IOCTL_STATUS SNDRV_PCM_IOCTL_STATUS_EXT SNDRV_TIMER_IOCTL_TREAD SNDRV_TIMER_IOCTL_STATUS 32-bit applications built with 64-bit time_t require both the headers and the running kernel to support at least the new API version. When built with earlier kernel headers, some of these may not work correctly, so applications are encouraged to fail compilation like #if SNDRV_PCM_VERSION < SNDRV_PROTOCOL_VERSION(2, 0, 15) extern int __fail_build_for_time_64[sizeof(long) - sizeof(time_t)]; #endif or provide their own updated copy of the header file. At runtime, the interface is unchanged for 32-bit time_t, but new kernels are required to work with user compiled with 64-bit time_t. A runtime check can be used to detect old kernel versions and warn about those. Signed-off-by: Arnd Bergmann --- include/uapi/sound/asound.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index df9983e7ead5..e6a958b8aff1 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -156,7 +156,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 14) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 15) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 0) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -766,7 +766,7 @@ struct snd_rawmidi_status { * Timer section - /dev/snd/timer */ -#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6) +#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 7) enum { SNDRV_TIMER_CLASS_NONE = -1, -- cgit v1.2.3 From 5f2cb361d798fb39adb79fab4e5235e307c70e9a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Dec 2019 20:11:01 +0100 Subject: ALSA: hda: Unify get_response handling Now most of the get_response handling became quite similar between HDA-core and legacy drivers, and the only differences are: - the handling of extra-long polling delay for some codecs - the debug message for the stalled communication and both are worth to share in the common code. This patch unifies the code into snd_hdac_bus_get_response(), and use this from the legacy get_response callback. It results in a good amount of code reduction in the end. Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20191212191101.19517-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/hda_codec.h | 1 - include/sound/hdaudio.h | 1 + 2 files changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index ac18f428eda6..3ee8036f5436 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -51,7 +51,6 @@ struct hda_bus { DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); /* misc op flags */ - unsigned int needs_damn_long_delay :1; unsigned int allow_bus_reset:1; /* allow bus reset at fatal error */ /* status for codec/controller */ unsigned int shutdown :1; /* being unloaded */ diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 81373a2efd96..bc2f77a6f17b 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -338,6 +338,7 @@ struct hdac_bus { bool reverse_assign:1; /* assign devices in reverse order */ bool corbrp_self_clear:1; /* CORBRP clears itself after reset */ bool polling_mode:1; + bool needs_damn_long_delay:1; int poll_count; -- cgit v1.2.3 From c13493a2460b7ba8f6e75fe6e1a3b732cc294f8f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 13 Dec 2019 09:54:36 +0900 Subject: ASoC: soc-core: support snd_soc_dai_link_component for codec_conf To find codec_conf component, it is using dev_name, of_node. But, we already has this kind of finding component method by snd_soc_dai_link_component, and snd_soc_is_matching_component(). We shouldn't have duplicate implementation to do same things. This patch adds snd_soc_dai_link_component support to find codec_conf component. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87lfrh59kj.wl-kuninori.morimoto.gx@renesas.com Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 82e65235c60d..a94e5d2fc2b2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -949,6 +949,7 @@ struct snd_soc_dai_link { #define COMP_CODEC(_name, _dai) { .name = _name, .dai_name = _dai, } #define COMP_PLATFORM(_name) { .name = _name } #define COMP_AUX(_name) { .name = _name } +#define COMP_CODEC_CONF(_name) { .name = _name } #define COMP_DUMMY() { .name = "snd-soc-dummy", .dai_name = "snd-soc-dummy-dai", } extern struct snd_soc_dai_link_component null_dailink_component[0]; @@ -962,6 +963,8 @@ struct snd_soc_codec_conf { const char *dev_name; struct device_node *of_node; + struct snd_soc_dai_link_component dlc; + /* * optional map of kcontrol, widget and path name prefixes that are * associated per device -- cgit v1.2.3 From ee8f537fd8b71c555d01a89e0834413bbf5373d4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 13 Dec 2019 09:55:55 +0900 Subject: ASoC: soc-core: remove legacy style of codec_conf Now all driver is using snd_soc_dai_link_component for codec_conf. Let's remove legacy style Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/871rt959ic.wl-kuninori.morimoto.gx@renesas.com Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Signed-off-by: Mark Brown --- include/sound/soc.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index a94e5d2fc2b2..9787c80e548b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -960,9 +960,6 @@ struct snd_soc_codec_conf { * specify device either by device name, or by * DT/OF node, but not both. */ - const char *dev_name; - struct device_node *of_node; - struct snd_soc_dai_link_component dlc; /* -- cgit v1.2.3 From e8b7cab8cec3386598de29bbca70d8d31aacc709 Mon Sep 17 00:00:00 2001 From: Karol Trzcinski Date: Tue, 17 Dec 2019 18:26:10 -0600 Subject: ASoC: SOF: define struct with compiler name and version Add compiler information structure sof_ipc_cc_version. Add new enum value in sof_ipc_ext_data for new structure. This struct will be used to show more information about firmware in host system. It will be helpful during debugging. Signed-off-by: Karol Trzcinski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191218002616.7652-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/info.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'include') diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h index a9156b4a062c..1c560144996c 100644 --- a/include/sound/sof/info.h +++ b/include/sound/sof/info.h @@ -30,6 +30,7 @@ enum sof_ipc_ext_data { SOF_IPC_EXT_DMA_BUFFER = 0, SOF_IPC_EXT_WINDOW, + SOF_IPC_EXT_CC_INFO, }; /* FW version - SOF_IPC_GLB_VERSION */ @@ -115,4 +116,18 @@ struct sof_ipc_window { struct sof_ipc_window_elem window[]; } __packed; +struct sof_ipc_cc_version { + struct sof_ipc_ext_data_hdr ext_hdr; + uint32_t major; + uint32_t minor; + uint32_t micro; + + /* reserved for future use */ + uint32_t reserved[4]; + + char name[16]; /* null terminated compiler name */ + char optim[4]; /* null terminated compiler -O flag value */ + char desc[]; /* null terminated compiler description */ +} __packed; + #endif -- cgit v1.2.3 From 9c1d4cf6ac26f890d82278326f6c7552c53ffb65 Mon Sep 17 00:00:00 2001 From: Guido Roncarolo Date: Tue, 17 Dec 2019 18:26:15 -0600 Subject: ASoC: SOF: imx: Describe SAI parameters to be sent to DSP Introduce sof_ipc_dai_sai_params to keep information that we get from topology and we send to DSP FW. For the moment it is identical to ESAI one but it will evolve shortly independently Signed-off-by: Guido Roncarolo Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191218002616.7652-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/dai-imx.h | 20 ++++++++++++++++++++ include/sound/sof/dai.h | 1 + include/uapi/sound/sof/tokens.h | 3 +-- 3 files changed, 22 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/sof/dai-imx.h b/include/sound/sof/dai-imx.h index e02fb0b0fae1..ff9088dcc6f2 100644 --- a/include/sound/sof/dai-imx.h +++ b/include/sound/sof/dai-imx.h @@ -31,4 +31,24 @@ struct sof_ipc_dai_esai_params { uint16_t reserved2; /* alignment */ } __packed; +/* SAI Configuration Request - SOF_IPC_DAI_SAI_CONFIG */ +struct sof_ipc_dai_sai_params { + struct sof_ipc_hdr hdr; + + /* MCLK */ + uint16_t reserved1; + uint16_t mclk_id; + uint32_t mclk_direction; + + 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; + uint16_t tdm_slot_width; + uint16_t reserved2; /* alignment */ +} __packed; #endif diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index c229565767e5..2565edd336f1 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -75,6 +75,7 @@ struct sof_ipc_dai_config { struct sof_ipc_dai_hda_params hda; struct sof_ipc_dai_alh_params alh; struct sof_ipc_dai_esai_params esai; + struct sof_ipc_dai_sai_params sai; }; } __packed; diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index a9a5c4d0a892..2a25cd8da503 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -113,8 +113,7 @@ #define SOF_TKN_EFFECT_TYPE SOF_TKN_PROCESS_TYPE /* SAI */ -#define SOF_TKN_IMX_SAI_FIRST_TOKEN 1000 -/* TODO: Add SAI tokens */ +#define SOF_TKN_IMX_SAI_MCLK_ID 1000 /* ESAI */ #define SOF_TKN_IMX_ESAI_MCLK_ID 1100 -- cgit v1.2.3 From 0f501c7cde4086d15c396a95c59631b05dbc0351 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 17 Dec 2019 14:22:30 -0600 Subject: ASoC: SOF: move arch_ops under ops The current structures are not well designed. We include Xtensa information from the ACPI and PCI levels, but at the Kconfig/module level everything Xtensa related is included at the sof/intel level. Move the arch_ops under ops so that Xtensa is hidden in the DSP ops, with a structure that follows the Kconfig/module partition. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191217202231.18259-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/sof.h b/include/sound/sof.h index 6ea74f1a9ec2..a0cbca021230 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -93,7 +93,6 @@ struct sof_dev_desc { const char *default_fw_filename; const struct snd_sof_dsp_ops *ops; - const struct sof_arch_ops *arch_ops; }; int sof_nocodec_setup(struct device *dev, -- cgit v1.2.3 From 2e4688676392767e16c1adeca4cc2c083e2db13f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Dec 2019 16:34:11 +0100 Subject: ALSA: emu10k1: Make uapi/emu10k1.h compilable again Recently we updated the content in alsa-lib uapi header files by just copying from the latest Linus kernel uapi/*.h, and noticed that it broke the build of some alsa-tools programs. The reason is that we used to have a modified version in the past, so that the program can be built without referring to the unexported stuff like snd_ctl_elem_id or __user prefix. This patch attempts to restore that, i.e. dropping the stuff that can't be referred in the user-space. For adapting the changes in uapi/emu10k1.h, the emu10k1 driver code is also slightly modified. Most of changes are pointer cast. Link: https://lore.kernel.org/r/20191220153415.2740-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 38 ++++++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 12 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index 042c5a6f16ee..c1150e4d0231 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,9 +23,6 @@ #ifndef _UAPI__SOUND_EMU10K1_H #define _UAPI__SOUND_EMU10K1_H -#include -#include - /* * ---- FX8010 ---- */ @@ -282,8 +279,22 @@ struct snd_emu10k1_fx8010_info { #define EMU10K1_GPR_TRANSLATION_TREBLE 3 #define EMU10K1_GPR_TRANSLATION_ONOFF 4 +enum emu10k1_ctl_elem_iface { + EMU10K1_CTL_ELEM_IFACE_MIXER = 2, /* virtual mixer device */ + EMU10K1_CTL_ELEM_IFACE_PCM = 3, /* PCM device */ +}; + +struct emu10k1_ctl_elem_id { + unsigned int pad; /* don't use */ + int iface; /* interface identifier */ + unsigned int device; /* device/client number */ + unsigned int subdevice; /* subdevice (substream) number */ + unsigned char name[44]; /* ASCII name of item */ + unsigned int index; /* index of item */ +}; + struct snd_emu10k1_fx8010_control_gpr { - struct snd_ctl_elem_id id; /* full control ID definition */ + struct emu10k1_ctl_elem_id id; /* full control ID definition */ unsigned int vcount; /* visible count */ unsigned int count; /* count of GPR (1..16) */ unsigned short gpr[32]; /* GPR number(s) */ @@ -296,7 +307,7 @@ struct snd_emu10k1_fx8010_control_gpr { /* old ABI without TLV support */ struct snd_emu10k1_fx8010_control_old_gpr { - struct snd_ctl_elem_id id; + struct emu10k1_ctl_elem_id id; unsigned int vcount; unsigned int count; unsigned short gpr[32]; @@ -310,24 +321,24 @@ struct snd_emu10k1_fx8010_code { char name[128]; __EMU10K1_DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ - __u32 __user *gpr_map; /* initializers */ + __u32 *gpr_map; /* initializers */ unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */ - struct snd_emu10k1_fx8010_control_gpr __user *gpr_add_controls; /* GPR controls to add/replace */ + struct snd_emu10k1_fx8010_control_gpr *gpr_add_controls; /* GPR controls to add/replace */ unsigned int gpr_del_control_count; /* count of GPR controls to remove */ - struct snd_ctl_elem_id __user *gpr_del_controls; /* IDs of GPR controls to remove */ + struct emu10k1_ctl_elem_id *gpr_del_controls; /* IDs of GPR controls to remove */ unsigned int gpr_list_control_count; /* count of GPR controls to list */ unsigned int gpr_list_control_total; /* total count of GPR controls */ - struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */ + struct snd_emu10k1_fx8010_control_gpr *gpr_list_controls; /* listed GPR controls */ __EMU10K1_DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ - __u32 __user *tram_data_map; /* data initializers */ - __u32 __user *tram_addr_map; /* map initializers */ + __u32 *tram_data_map; /* data initializers */ + __u32 *tram_addr_map; /* map initializers */ __EMU10K1_DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ - __u32 __user *code; /* one instruction - 64 bits */ + __u32 *code; /* one instruction - 64 bits */ }; struct snd_emu10k1_fx8010_tram { @@ -371,11 +382,14 @@ struct snd_emu10k1_fx8010_pcm_rec { #define SNDRV_EMU10K1_IOCTL_SINGLE_STEP _IOW ('H', 0x83, int) #define SNDRV_EMU10K1_IOCTL_DBG_READ _IOR ('H', 0x84, int) +#ifndef __KERNEL__ /* typedefs for compatibility to user-space */ typedef struct snd_emu10k1_fx8010_info emu10k1_fx8010_info_t; typedef struct snd_emu10k1_fx8010_control_gpr emu10k1_fx8010_control_gpr_t; typedef struct snd_emu10k1_fx8010_code emu10k1_fx8010_code_t; typedef struct snd_emu10k1_fx8010_tram emu10k1_fx8010_tram_t; typedef struct snd_emu10k1_fx8010_pcm_rec emu10k1_fx8010_pcm_t; +typedef struct emu10k1_ctl_elem_id emu10k1_ctl_elem_id_t; +#endif #endif /* _UAPI__SOUND_EMU10K1_H */ -- cgit v1.2.3 From d63e63d4210713ddc7eb1b2d2afc4a8d42ccf91e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Dec 2019 16:34:12 +0100 Subject: ALSA: hdsp: Make uapi/hdsp.h compilable again Recently alsa-lib updated its content of sound/hdsp.h just by copying the latest Linus kernel uapi/*.h, and this broke the build of alsa-tools programs. We used to modify the headers so that they can be built without asoundlib.h and linux kernel headers, and the verbatim copy doesn't work as is. This patch removes again the linux/types.h inclusion and drop __user prefix that broke the build and adjusts the corresponding code. Link: https://lore.kernel.org/r/20191220153415.2740-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/hdsp.h | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/hdsp.h b/include/uapi/sound/hdsp.h index 5dc0c3db0a4c..88c92a3fb477 100644 --- a/include/uapi/sound/hdsp.h +++ b/include/uapi/sound/hdsp.h @@ -20,8 +20,6 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ -#include - #define HDSP_MATRIX_MIXER_SIZE 2048 enum HDSP_IO_Type { @@ -74,7 +72,7 @@ struct hdsp_config_info { #define SNDRV_HDSP_IOCTL_GET_CONFIG_INFO _IOR('H', 0x41, struct hdsp_config_info) struct hdsp_firmware { - void __user *firmware_data; /* 24413 x 4 bytes */ + void *firmware_data; /* 24413 x 4 bytes */ }; #define SNDRV_HDSP_IOCTL_UPLOAD_FIRMWARE _IOW('H', 0x42, struct hdsp_firmware) -- cgit v1.2.3 From 4fa406caf950fd46ae06bccf9a4c72171401d203 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Dec 2019 16:34:13 +0100 Subject: ALSA: hdspm: Drop linux/types.h inclusion in uapi header The hdspm.h uapi header has been used also from non-Linux or platforms that don't have linux/*.h. It was OK in the past because alsa-lib contained the modified version of this header file, but now it tries to the verbatim copy, so it broke the build. This fixes it again. Link: https://lore.kernel.org/r/20191220153415.2740-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/hdspm.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index a38f3f79beb7..2d91f90eb5e1 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -21,8 +21,6 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ -#include - /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 -- cgit v1.2.3 From 7fd7d6c5045113350fcf78e865ced8a80dbde9fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Dec 2019 16:34:14 +0100 Subject: ALSA: uapi: Fix typos and header inclusion in asound.h The recent changes in uapi/asoundlib.h caused some build errors in alsa-lib side because of a typo and the new included files. Basically asound.h is supposed to be usable also on non-Linux systems, so we've tried to avoid the Linux-specific include files. This patch is an attempt to recover from those changes. Fixes: 3ddee7f88aaf ("ALSA: Avoid using timespec for struct snd_pcm_status") Fixes: 80fe7430c708 ("ALSA: add new 32-bit layout for snd_pcm_mmap_status/control") Link: https://lore.kernel.org/r/20191220153415.2740-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index e6a958b8aff1..e7943302359e 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -26,7 +26,9 @@ #if defined(__KERNEL__) || defined(__linux__) #include +#include #else +#include #include #endif @@ -35,8 +37,6 @@ #include #endif -#include - /* * protocol version */ @@ -471,7 +471,7 @@ enum { #ifndef __KERNEL__ /* explicit padding avoids incompatibility between i386 and x86-64 */ -typedef struct { unsigned char pad[sizeof(time_t) - sizeof(int)] __time_pad; +typedef struct { unsigned char pad[sizeof(time_t) - sizeof(int)]; } __time_pad; struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ -- cgit v1.2.3 From 645c08f17f477915f6d900b767e789852f150054 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Dec 2019 16:34:15 +0100 Subject: ALSA: uapi: Drop asound.h inclusion from asoc.h The asound.h isn't always available while asoc.h itself is distributed in alsa-lib package. So we need to avoid the unnecessary inclusion of asound.h from there. Link: https://lore.kernel.org/r/20191220153415.2740-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/asoc.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index a74ca232f1fc..6048553c119d 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -17,7 +17,6 @@ #define __LINUX_UAPI_SND_ASOC_H #include -#include /* * Maximum number of channels topology kcontrol can represent. -- cgit v1.2.3 From bfea224d9250e56ed5475bf72190783a7e1d8e43 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Dec 2019 17:15:55 +0100 Subject: ALSA: uapi: Drop unneeded typedefs We kept some typedefs in uapi/sound/*.h so that the programs in alsa-tools can be built. Now that alsa-lib takes these and applies the workarounds in its own, we don't need these typedefs any longer in the kernel uapi side. Let's drop them. Link: https://lore.kernel.org/r/20191220161555.20232-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 10 ---------- include/uapi/sound/hdsp.h | 9 --------- include/uapi/sound/hdspm.h | 8 -------- 3 files changed, 27 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index c1150e4d0231..6bcd76f64c1c 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -382,14 +382,4 @@ struct snd_emu10k1_fx8010_pcm_rec { #define SNDRV_EMU10K1_IOCTL_SINGLE_STEP _IOW ('H', 0x83, int) #define SNDRV_EMU10K1_IOCTL_DBG_READ _IOR ('H', 0x84, int) -#ifndef __KERNEL__ -/* typedefs for compatibility to user-space */ -typedef struct snd_emu10k1_fx8010_info emu10k1_fx8010_info_t; -typedef struct snd_emu10k1_fx8010_control_gpr emu10k1_fx8010_control_gpr_t; -typedef struct snd_emu10k1_fx8010_code emu10k1_fx8010_code_t; -typedef struct snd_emu10k1_fx8010_tram emu10k1_fx8010_tram_t; -typedef struct snd_emu10k1_fx8010_pcm_rec emu10k1_fx8010_pcm_t; -typedef struct emu10k1_ctl_elem_id emu10k1_ctl_elem_id_t; -#endif - #endif /* _UAPI__SOUND_EMU10K1_H */ diff --git a/include/uapi/sound/hdsp.h b/include/uapi/sound/hdsp.h index 88c92a3fb477..7ac2d3f2a9b3 100644 --- a/include/uapi/sound/hdsp.h +++ b/include/uapi/sound/hdsp.h @@ -97,13 +97,4 @@ struct hdsp_9632_aeb { #define SNDRV_HDSP_IOCTL_GET_9632_AEB _IOR('H', 0x45, struct hdsp_9632_aeb) -/* typedefs for compatibility to user-space */ -typedef enum HDSP_IO_Type HDSP_IO_Type; -typedef struct hdsp_peak_rms hdsp_peak_rms_t; -typedef struct hdsp_config_info hdsp_config_info_t; -typedef struct hdsp_firmware hdsp_firmware_t; -typedef struct hdsp_version hdsp_version_t; -typedef struct hdsp_mixer hdsp_mixer_t; -typedef struct hdsp_9632_aeb hdsp_9632_aeb_t; - #endif /* __SOUND_HDSP_H */ diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 2d91f90eb5e1..3fbfd9dc5f51 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -219,12 +219,4 @@ struct hdspm_mixer_ioctl { /* use indirect access due to the limit of ioctl bit size */ #define SNDRV_HDSPM_IOCTL_GET_MIXER _IOR('H', 0x44, struct hdspm_mixer_ioctl) -/* typedefs for compatibility to user-space */ -typedef struct hdspm_peak_rms hdspm_peak_rms_t; -typedef struct hdspm_config_info hdspm_config_info_t; -typedef struct hdspm_version hdspm_version_t; -typedef struct hdspm_channelfader snd_hdspm_channelfader_t; -typedef struct hdspm_mixer hdspm_mixer_t; - - #endif -- cgit v1.2.3 From 54228356667920658cd33a63e12b7bff3e13c880 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 23 Dec 2019 11:39:18 +0900 Subject: ALSA: ctl: remove unused macro for timestamping of elem_value In a former commit, 'tstamp' member was removed from 'struct snd_ctl_elem_value' in a middle way toward solution of Y2038 issue. In a protocol of ALSA control interface, this member is designed to deliver timestamp information in the value structure when the target element supports SNDRV_CTL_ELEM_ACCESS_TIMESTAMP flag. Actually, the feature is neither used by kernel space nor user space, especiall alsa-lib has no API for the feature. Therefore it's reasonable to remove both of them. Practically, the timestamp information corresponds to no information about type of clock ID. It can bring confusions to applications. Reference: a4e7dd35b9da ("ALSA: Avoid using timespec for struct snd_ctl_elem_value") Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191223023921.8151-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- 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 e7943302359e..efd9e1398e07 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -974,7 +974,7 @@ typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_ACCESS_WRITE (1<<1) #define SNDRV_CTL_ELEM_ACCESS_READWRITE (SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE) #define SNDRV_CTL_ELEM_ACCESS_VOLATILE (1<<2) /* control value may be changed without a notification */ -#define SNDRV_CTL_ELEM_ACCESS_TIMESTAMP (1<<3) /* when was control changed */ +// (1 << 3) is unused. #define SNDRV_CTL_ELEM_ACCESS_TLV_READ (1<<4) /* TLV read is possible */ #define SNDRV_CTL_ELEM_ACCESS_TLV_WRITE (1<<5) /* TLV write is possible */ #define SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE (SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) -- cgit v1.2.3 From ff16351e3f302a2913bd17da6ed8f195ab2139fd Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 23 Dec 2019 11:39:20 +0900 Subject: ALSA: ctl: remove dimen member from elem_info structure The 'dimen' member of 'struct snd_ctl_elem_info' is designed to deliver information to use an array of value as multi-dimensional values. This feature is used just by echoaudio PCI driver, and fortunately it's not used by the other applications than 'echomixer' in alsa-tools. In a previous commit, usage of 'dimen' member is removed from echoaudio PCI driver. Nowadays no driver/application use the feature. This commit removes the member from structure. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191223023921.8151-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index efd9e1398e07..06033fc78ee4 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -1040,11 +1040,7 @@ struct snd_ctl_elem_info { } enumerated; unsigned char reserved[128]; } value; - union { - unsigned short d[4]; /* dimensions */ - unsigned short *d_ptr; /* indirect - obsoleted */ - } dimen; - unsigned char reserved[64-4*sizeof(unsigned short)]; + unsigned char reserved[64]; }; struct snd_ctl_elem_value { -- cgit v1.2.3 From bd3eb4e87eb399a9fe15ef1b59db78faf9c20359 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 23 Dec 2019 11:39:21 +0900 Subject: ALSA: ctl: bump protocol version up to v2.1.0 In a development period for v5.6 kernel, some changes are introduced to structures in ALSA control interface: - 'tstamp' member is removed from 'struct snd_ctl_elem_value - 'TSTAMP' flag is removed from a set of access flags for 'struct snd_ctl_elem_info' - 'dimen' member is removed from 'struct snd_ctl_elem_info Although these changes were introduced with enough consideration for backward compatibility, they include slightly lose of it. This commit bumps protocol version of ALSA control interface up to v2.1.0. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191223023921.8151-5-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- 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 06033fc78ee4..e36dadaf84ba 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -936,7 +936,7 @@ struct snd_timer_tread { * * ****************************************************************************/ -#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 7) +#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 1, 0) struct snd_ctl_card_info { int card; /* card number */ -- cgit v1.2.3 From a103a3989993859091a26936b9337a8e09330fc3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Dec 2019 11:03:53 +0100 Subject: ALSA: control: Fix incompatible protocol error The recent change to bump the ALSA control API protocol version from 2.0.7 to 2.1.0 caused a regression on user-space; while the user-space expects both the major and the minor versions to be identical with the supported numbers, we changed the minor number from 0 to 1. For recovering from the incompatibility, this patch changes the protocol version again to 2.0.8, which is compatible, but yet higher than the original number 2.0.7, indicating that the protocol change. Fixes: bd3eb4e87eb3 ("ALSA: ctl: bump protocol version up to v2.1.0") Reported-by: Paul Menzel Tested-by: Paul Menzel Link: https://lore.kernel.org/r/s5h1rsr769i.wl-tiwai@suse.de Signed-off-by: Takashi Iwai --- 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 e36dadaf84ba..30ebb2a42983 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -936,7 +936,7 @@ struct snd_timer_tread { * * ****************************************************************************/ -#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 1, 0) +#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 8) struct snd_ctl_card_info { int card; /* card number */ -- cgit v1.2.3 From d06ed0c20960f124642ce77cd80056a35ffebe55 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Dec 2019 22:27:42 +0100 Subject: ALSA: uapi: Add linux/types.h include back (but carefully) A few uapi/sound/*.h headers have been corrected for recovering from the compile errors with the existing user-space code (alsa-lib) by the recent commits. OTOH, these introduced another regression, as now linux/types.h inclusion became mandatory for the uapi header checks. As a compromise, this patch re-adds linux/types.h inclusions again, but conditionally not to break other non-standard user-space stuff again. Fixes: 2e4688676392 ("ALSA: emu10k1: Make uapi/emu10k1.h compilable again") Fixes: d63e63d42107 ("ALSA: hdsp: Make uapi/hdsp.h compilable again") Fixes: 4fa406caf950 ("ALSA: hdspm: Drop linux/types.h inclusion in uapi header") Reported-by: kbuild test robot Link: https://lore.kernel.org/r/20191230212742.28925-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 4 ++++ include/uapi/sound/hdsp.h | 4 ++++ include/uapi/sound/hdspm.h | 4 ++++ 3 files changed, 12 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index 6bcd76f64c1c..88609cc0524c 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,6 +23,10 @@ #ifndef _UAPI__SOUND_EMU10K1_H #define _UAPI__SOUND_EMU10K1_H +#ifdef __linux__ +#include +#endif + /* * ---- FX8010 ---- */ diff --git a/include/uapi/sound/hdsp.h b/include/uapi/sound/hdsp.h index 7ac2d3f2a9b3..b8df62b60f4d 100644 --- a/include/uapi/sound/hdsp.h +++ b/include/uapi/sound/hdsp.h @@ -20,6 +20,10 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __linux__ +#include +#endif + #define HDSP_MATRIX_MIXER_SIZE 2048 enum HDSP_IO_Type { diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 3fbfd9dc5f51..14af3d00ea3f 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -21,6 +21,10 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __linux__ +#include +#endif + /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 -- cgit v1.2.3 From 8b575824304dd18a90616cbcf0f81ca1ed1fb5d2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jan 2020 09:16:19 +0100 Subject: ALSA: core: Treat snd_device_ops as const This is a preliminary patch to allow const for snd_device_ops definitions in each driver's code. The ops reference is read-only, hence it can be declared as const for further optimization. There should be no functional changes by this patch. Link: https://lore.kernel.org/r/20200103081714.9560-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/core.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index af3dce956c17..0e14b7a3e67b 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -69,7 +69,7 @@ struct snd_device { enum snd_device_state state; /* state of the device */ enum snd_device_type type; /* device type */ void *device_data; /* device structure */ - struct snd_device_ops *ops; /* operations */ + const struct snd_device_ops *ops; /* operations */ }; #define snd_device(n) list_entry(n, struct snd_device, list) @@ -256,7 +256,7 @@ static inline void snd_card_unref(struct snd_card *card) /* device.c */ int snd_device_new(struct snd_card *card, enum snd_device_type type, - void *device_data, struct snd_device_ops *ops); + void *device_data, const struct snd_device_ops *ops); int snd_device_register(struct snd_card *card, void *device_data); int snd_device_register_all(struct snd_card *card); void snd_device_disconnect(struct snd_card *card, void *device_data); -- cgit v1.2.3 From 19260818a9747768b09c1c1f10b4125b4d00595e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jan 2020 09:16:38 +0100 Subject: ALSA: ac97: Treat snd_ac97_bus_ops as const This is a preliminary patch to allow const for snd_ac97_bus_ops definitions in each driver's code. The ops reference is read-only, hence it can be declared as const for further optimization. There should be no functional changes by this patch. Link: https://lore.kernel.org/r/20200103081714.9560-23-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index cc383991c0fe..49200ec26dc4 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -197,7 +197,7 @@ struct snd_ac97_bus_ops { struct snd_ac97_bus { /* -- lowlevel (hardware) driver specific -- */ - struct snd_ac97_bus_ops *ops; + const struct snd_ac97_bus_ops *ops; void *private_data; void (*private_free) (struct snd_ac97_bus *bus); /* --- */ @@ -310,7 +310,8 @@ static inline int ac97_can_spdif(struct snd_ac97 * ac97) /* functions */ /* create new AC97 bus */ -int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops, +int snd_ac97_bus(struct snd_card *card, int num, + const struct snd_ac97_bus_ops *ops, void *private_data, struct snd_ac97_bus **rbus); /* create mixer controls */ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, -- cgit v1.2.3 From d25ff26840bd0af3283d8e478669abc104bb873a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jan 2020 09:16:44 +0100 Subject: ALSA: info: Make snd_info_entry_ops as const The reference to snd_info_entry_ops is rather read-only, so declare it as a const pointer. This allows a bit more optimization. There should be no functional changes by this patch. Link: https://lore.kernel.org/r/20200103081714.9560-29-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/info.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/info.h b/include/sound/info.h index b01a22913400..7c13bf52cc81 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -64,7 +64,7 @@ struct snd_info_entry { unsigned short content; union { struct snd_info_entry_text text; - struct snd_info_entry_ops *ops; + const struct snd_info_entry_ops *ops; } c; struct snd_info_entry *parent; struct module *module; -- cgit v1.2.3 From aad7ebb544072bcb9335fa4eb0fbd1b85a6c495b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jan 2020 09:16:45 +0100 Subject: ALSA: seq: Constify struct snd_midi_op Change the argument of snd_midi_process_event() to receive a const snd_midi_op pointer and its callers respectively. This allows further optimizations. There should be no functional changes by this patch. Link: https://lore.kernel.org/r/20200103081714.9560-30-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/seq_midi_emul.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/seq_midi_emul.h b/include/sound/seq_midi_emul.h index d6b74059b4b1..88799d1e1f53 100644 --- a/include/sound/seq_midi_emul.h +++ b/include/sound/seq_midi_emul.h @@ -174,7 +174,8 @@ enum { }; /* Prototypes for midi_process.c */ -void snd_midi_process_event(struct snd_midi_op *ops, struct snd_seq_event *ev, +void snd_midi_process_event(const struct snd_midi_op *ops, + struct snd_seq_event *ev, struct snd_midi_channel_set *chanset); void snd_midi_channel_set_clear(struct snd_midi_channel_set *chset); struct snd_midi_channel_set *snd_midi_channel_alloc_set(int n); -- cgit v1.2.3 From f8ae2d2919481817d2e942617c203fc792687c66 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jan 2020 09:16:46 +0100 Subject: ALSA: vx: Constify snd_vx_hardware and snd_vx_ops definitions Both snd_vx_hardware and snd_vx_ops are only referred without modification, hence they can be constified gracefully for further optimizations. There should be no functional changes by this patch. Link: https://lore.kernel.org/r/20200103081714.9560-31-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/vx_core.h | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h index 84569ddf85e1..1ddd3036bdfc 100644 --- a/include/sound/vx_core.h +++ b/include/sound/vx_core.h @@ -147,8 +147,8 @@ struct vx_core { /* ports are defined externally */ /* low-level functions */ - struct snd_vx_hardware *hw; - struct snd_vx_ops *ops; + const struct snd_vx_hardware *hw; + const struct snd_vx_ops *ops; struct mutex lock; @@ -193,8 +193,9 @@ struct vx_core { /* * constructor */ -struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw, - struct snd_vx_ops *ops, int extra_size); +struct vx_core *snd_vx_create(struct snd_card *card, + const struct snd_vx_hardware *hw, + const struct snd_vx_ops *ops, int extra_size); int snd_vx_setup_firmware(struct vx_core *chip); int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *dsp); int snd_vx_dsp_boot(struct vx_core *chip, const struct firmware *dsp); -- cgit v1.2.3 From fbd3eb7f66c5b4f37a959bc2deaeb1d7b5ddf0d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 4 Jan 2020 09:35:56 +0100 Subject: ALSA: control: Add verification for kctl accesses The current implementation of ALSA control API fully relies on the callbacks of each driver, and there is no verification of the values passed via API. This patch is an attempt to improve the situation slightly by adding the validation code for the values stored via info and get callbacks. The patch adds a new kconfig, CONFIG_SND_CTL_VALIDATION. It depends on CONFIG_SND_DEBUG and off as default since the validation would require a slight overhead including the additional call of info callback at each get callback invocation. When this config is enabled, the values stored by each info callback invocation are verified, namely: - Whether the info type is valid - Whether the number of enum items is non-zero - Whether the given info count is within the allowed boundary Similarly, the values stored at each get callback are verified as well: - Whether the values are within the given range - Whether the values are aligned with the given step - Whether any further changes are seen in the data array over the given info count The last point helps identifying a possibly invalid data type access, typically a case where the info callback declares the type being SNDRV_CTL_ELEM_TYPE_ENUMERATED while the get/put callbacks store the values in value.integer.value[] array. When a validation fails, the ALSA core logs an error message including the device and the control ID, and the API call also returns an error. So, with the new validation turned on, the driver behavior difference may be visible on user-space, too -- it's intentional, though, so that we can catch an error more clearly. The patch also introduces a new ctl access type, SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK. A driver may pass this flag with other access bits to indicate that the ctl element won't be verified. It's useful when a driver code is specially written to access the data greater than info->count size by some reason. For example, this flag is actually set now in HD-audio HDMI codec driver which needs to clear the data array in the case of the disconnected monitor. Also, the PCM channel-map helper code is slightly modified to avoid the false-positive hit by this validation code, too. Link: https://lore.kernel.org/r/20200104083556.27789-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/control.h | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'include') diff --git a/include/sound/control.h b/include/sound/control.h index 5d7c99475684..11feeee31e35 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -22,6 +22,16 @@ typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol, unsigned int size, unsigned int __user *tlv); +/* internal flag for skipping validations */ +#ifdef CONFIG_SND_CTL_VALIDATION +#define SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK (1 << 27) +#define snd_ctl_skip_validation(info) \ + ((info)->access & SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK) +#else +#define SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK 0 +#define snd_ctl_skip_validation(info) true +#endif + enum { SNDRV_CTL_TLV_OP_READ = 0, SNDRV_CTL_TLV_OP_WRITE = 1, -- cgit v1.2.3 From f5f87abfb745aa6eb5c5f9bdf8b5a64600692506 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 5 Jan 2020 15:47:16 +0100 Subject: ALSA: Allow const arrays for legacy resource management helpers Declare the arrays passed to the helper functions for legacy resources (mostly for ISA drivers) as const, so that each caller can make its static data as const for minor optimizations, too. Link: https://lore.kernel.org/r/20200105144823.29547-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/initval.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/initval.h b/include/sound/initval.h index 3ddae2b4177e..a1ff3b4865b4 100644 --- a/include/sound/initval.h +++ b/include/sound/initval.h @@ -37,7 +37,7 @@ #define SNDRV_DEFAULT_PTR SNDRV_DEFAULT_STR #ifdef SNDRV_LEGACY_FIND_FREE_IOPORT -static long snd_legacy_find_free_ioport(long *port_table, long size) +static long snd_legacy_find_free_ioport(const long *port_table, long size) { while (*port_table != -1) { if (request_region(*port_table, size, "ALSA test")) { @@ -58,7 +58,7 @@ static irqreturn_t snd_legacy_empty_irq_handler(int irq, void *dev_id) return IRQ_HANDLED; } -static int snd_legacy_find_free_irq(int *irq_table) +static int snd_legacy_find_free_irq(const int *irq_table) { while (*irq_table != -1) { if (!request_irq(*irq_table, snd_legacy_empty_irq_handler, @@ -74,7 +74,7 @@ static int snd_legacy_find_free_irq(int *irq_table) #endif #ifdef SNDRV_LEGACY_FIND_FREE_DMA -static int snd_legacy_find_free_dma(int *dma_table) +static int snd_legacy_find_free_dma(const int *dma_table) { while (*dma_table != -1) { if (!request_dma(*dma_table, "ALSA Test DMA")) { -- cgit v1.2.3 From 613fb50059cf19aa6acbc503a00265d9151c0b09 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 10 Jan 2020 11:35:21 +0900 Subject: ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto Reviewed-By: Ranjani Sridharan Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 9787c80e548b..0513f30a0209 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -736,19 +736,9 @@ struct snd_soc_compr_ops { int (*trigger)(struct snd_compr_stream *); }; -struct snd_soc_rtdcom_list { - struct snd_soc_component *component; - struct list_head list; /* rtd::component_list */ -}; struct snd_soc_component* snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, const char *driver_name); -#define for_each_rtd_components(rtd, rtdcom, _component) \ - for (rtdcom = list_first_entry(&(rtd)->component_list, \ - typeof(*rtdcom), list); \ - (&rtdcom->list != &(rtd)->component_list) && \ - (_component = rtdcom->component); \ - rtdcom = list_next_entry(rtdcom, list)) struct snd_soc_dai_link_component { const char *name; @@ -1150,12 +1140,18 @@ struct snd_soc_pcm_runtime { unsigned int num; /* 0-based and monotonic increasing */ struct list_head list; /* rtd list of the soc card */ - struct list_head component_list; /* list of connected components */ /* bit field */ unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ + + int num_components; + struct snd_soc_component *components[0]; /* CPU/Codec/Platform */ }; +#define for_each_rtd_components(rtd, i, component) \ + for ((i) = 0; \ + ((i) < rtd->num_components) && ((component) = rtd->components[i]);\ + (i)++) #define for_each_rtd_codec_dai(rtd, i, dai)\ for ((i) = 0; \ ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ -- cgit v1.2.3 From 83f94a2e293d617a98e077680ea00b2830a9ca22 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 10 Jan 2020 11:36:17 +0900 Subject: ASoC: soc-core: add snd_soc_close_delayed_work() We need to setup rtd->close_delayed_work_func. It will be set at snd_soc_dai_compress_new() or soc_new_pcm(). But these setups close_delayed_work() which is same name / same implemantaion, but different local code. To reduce duplicate code, this patch moves it as snd_soc_close_delayed_work() and share same code. Signed-off-by: Kuninori Morimoto Reviewed-By: Ranjani Sridharan Link: https://lore.kernel.org/r/8736cot4v2.wl-kuninori.morimoto.gx@renesas.com 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 0513f30a0209..f0e4f36f83bf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1159,6 +1159,7 @@ struct snd_soc_pcm_runtime { #define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);) +void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); /* mixer control */ struct soc_mixer_control { -- cgit v1.2.3 From 3f4cf797939cb3ccdb6f989da53f1899d30432dc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 10 Jan 2020 11:36:23 +0900 Subject: ASoC: soc-dapm: add snd_soc_dapm_stream_stop() When we stop stream, if it was Playback, we might need to care about power down time. In such case, we need to use delayed work. We have same implementation for it at soc-pcm.c and soc-compress.c, but we don't want to have duplicate code. This patch adds snd_soc_dapm_stream_stop(), and share same code. Signed-off-by: Kuninori Morimoto Reviewed-By: Ranjani Sridharan Link: https://lore.kernel.org/r/871rs8t4uw.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6e8a31225383..1b6afbc1a4ed 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -434,6 +434,7 @@ void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm); /* dapm events */ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event); +void snd_soc_dapm_stream_stop(struct snd_soc_pcm_runtime *rtd, int stream); void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ -- cgit v1.2.3 From 1a462be52f4505a2719631fb5aa7bfdbd37bfd8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Jan 2020 10:01:04 +0100 Subject: ALSA: hda: Manage concurrent reg access more properly In the commit 8e85def5723e ("ALSA: hda: enable regmap internal locking"), we re-enabled the regmap lock due to the reported regression that showed the possible concurrent accesses. It was a temporary workaround, and there are still a few opened races even after the revert. In this patch, we cover those still opened windows with a proper mutex lock and disable the regmap internal lock again. First off, the patch introduces a new snd_hdac_device.regmap_lock mutex that is applied for each snd_hdac_regmap_*() call, including read, write and update helpers. The mutex is applied carefully so that it won't block the self-power-up procedure in the helper function. Also, this assures the protection for the accesses without regmap, too. The snd_hdac_regmap_update_raw() is refactored to use the standard regmap_update_bits_check() function instead of the open-code. The non-regmap case is still open-coded but it's an easy part. The all read and write operations are in the single mutex protection, so it's now race-free. In addition, a couple of new helper functions are added: snd_hdac_regmap_update_raw_once() and snd_hdac_regmap_sync(). Both are called from HD-audio legacy driver. The former is to initialize the given verb bits but only once when it's not initialized yet. Due to this condition, the function invokes regcache_cache_only(), and it's now performed inside the regmap_lock (formerly it was racy) too. The latter function is for simply invoking regcache_sync() inside the regmap_lock, which is called from the codec resume call path. Along with that, the HD-audio codec driver code is slightly modified / simplified to adapt those new functions. And finally, snd_hdac_regmap_read_raw(), *_write_raw(), etc are rewritten with the helper macro. It's just for simplification because the code logic is identical among all those functions. Tested-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200109090104.26073-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/hda_regmap.h | 3 +++ include/sound/hdaudio.h | 1 + 2 files changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index 5141f8ffbb12..4c1b9bebbd60 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -24,6 +24,9 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, unsigned int val); int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, unsigned int mask, unsigned int val); +int snd_hdac_regmap_update_raw_once(struct hdac_device *codec, unsigned int reg, + unsigned int mask, unsigned int val); +void snd_hdac_regmap_sync(struct hdac_device *codec); /** * snd_hdac_regmap_encode_verb - encode the verb to a pseudo register diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index bc2f77a6f17b..541ca99b154b 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -86,6 +86,7 @@ struct hdac_device { /* regmap */ struct regmap *regmap; + struct mutex regmap_lock; struct snd_array vendor_verbs; bool lazy_cache:1; /* don't wake up for writes */ bool caps_overwriting:1; /* caps overwrite being in process */ -- cgit v1.2.3 From efb6f3159e874f09992b7318cea12e8e27e8389b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 10 Jan 2020 16:25:25 -0600 Subject: ASoC: soc-acpi: add _ADR-based link descriptors For SoundWire support, we added a 'link_mask' to describe the PCB hardware layout. This helped form a signature that can be used as a first-order way of detecting the hardware and selecting the machine driver. The concept of link_mask is however not enough. Some BIOS enable all links, even when there are no devices physically connected. We can also see variations with multiple devices attached on one link, or different types of devices connected on the same link. To accurately represent the hardware, we need to build static tables where each link exposes a list of expected devices represented by the 64-bit _ADR field (which uniquely identifies each device). The new 'links' field is optional when the link_mask is sufficient to represent a platform in a unique way. The existing mechanism to support I2C devices is left as is, it'd be too invasive to change the existing support for _HID and the notion of link is not relevant either. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard liao Link: https://lore.kernel.org/r/20200110222530.30303-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'include') diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index c4c997bd0379..a217a87cae86 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -61,6 +61,8 @@ static inline struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) * @platform: string used for HDaudio codec support * @codec_mask: used for HDAudio support * @common_hdmi_codec_drv: use commom HDAudio HDMI codec driver + * @link_mask: links enabled on the board + * @links: array of link _ADR descriptors, null terminated */ struct snd_soc_acpi_mach_params { u32 acpi_ipc_irq_index; @@ -68,6 +70,23 @@ struct snd_soc_acpi_mach_params { u32 codec_mask; u32 dmic_num; bool common_hdmi_codec_drv; + u32 link_mask; + const struct snd_soc_acpi_link_adr *links; +}; + +/** + * snd_soc_acpi_link_adr: ACPI-based list of _ADR, with a variable + * number of devices per link + * + * @mask: one bit set indicates the link this list applies to + * @num_adr: ARRAY_SIZE of adr + * @adr: array of _ADR (represented as u64). + */ + +struct snd_soc_acpi_link_adr { + const u32 mask; + const u32 num_adr; + const u64 *adr; }; /** @@ -78,6 +97,7 @@ struct snd_soc_acpi_mach_params { * * @id: ACPI ID (usually the codec's) used to find a matching machine driver. * @link_mask: describes required board layout, e.g. for SoundWire. + * @links: array of link _ADR descriptors, null terminated. * @drv_name: machine driver name * @fw_filename: firmware file name. Used when SOF is not enabled. * @board: board name @@ -94,6 +114,7 @@ struct snd_soc_acpi_mach_params { struct snd_soc_acpi_mach { const u8 id[ACPI_ID_LEN]; const u32 link_mask; + const struct snd_soc_acpi_link_adr *links; const char *drv_name; const char *fw_filename; const char *board; -- cgit v1.2.3 From cf61a42066eae88b4bb0e5528ec9f5f8e50c5343 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 10 Jan 2020 16:25:26 -0600 Subject: ASoC: Intel: common: soc-acpi: declare new tables for SoundWire We cannot really lump SoundWire-based configurations into the same tables since the mechanisms to identify boards is based on link configurations and _ADR instead of _HID for I2S, so define new tables Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200110222530.30303-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-match.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 20c0bee3b959..ab6f75a86611 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -31,6 +31,12 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cfl_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[]; + /* * generic table used for HDA codec-based platforms, possibly with * additional ACPI-enumerated codecs -- cgit v1.2.3 From 4d024fe8f806e20e577cc934204c5784c7063293 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Jan 2020 11:41:27 +0100 Subject: ALSA: hda: Apply aligned MMIO access only conditionally It turned out that the recent simplification of HD-audio bus access helpers caused a regression on the virtual HD-audio device on QEMU with ARM platforms. The driver got a CORB/RIRB timeout and couldn't probe any codecs. The essential difference that caused a problem was the enforced aligned MMIO accesses by simplification. Since snd-hda-tegra driver is enabled on ARM, it enables CONFIG_SND_HDA_ALIGNED_MMIO, which makes the all HD-audio drivers using the aligned MMIO accesses. While this is mandatory for snd-hda-tegra, it seems that snd-hda-intel on ARM gets broken by this access pattern. For addressing the regression, this patch introduces a new flag, aligned_mmio, to hdac_bus object, and applies the aligned MMIO only when this flag is set. This change affects only platforms with CONFIG_SND_HDA_ALIGNED_MMIO set, i.e. mostly only for ARM platforms. Unfortunately the patch became a big bigger than it should be, just because the former calls didn't take hdac_bus object in the argument, hence we had to extend the call patterns. Fixes: 19abfefd4c76 ("ALSA: hda: Direct MMIO accesses") BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1161152 Cc: Link: https://lore.kernel.org/r/20200120104127.28985-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 77 ++++++++++++++++++++++++++++++++++--------------- 1 file changed, 53 insertions(+), 24 deletions(-) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index e05b95e83d5a..fb9dce4c6928 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -8,6 +8,7 @@ #include #include +#include #include #include #include @@ -330,6 +331,7 @@ struct hdac_bus { bool chip_init:1; /* h/w initialized */ /* behavior flags */ + bool aligned_mmio:1; /* aligned MMIO access */ bool sync_write:1; /* sync after verb write */ bool use_posbuf:1; /* use position buffer */ bool snoop:1; /* enable snooping */ @@ -405,34 +407,61 @@ void snd_hdac_bus_free_stream_pages(struct hdac_bus *bus); unsigned int snd_hdac_aligned_read(void __iomem *addr, unsigned int mask); void snd_hdac_aligned_write(unsigned int val, void __iomem *addr, unsigned int mask); -#define snd_hdac_reg_writeb(v, addr) snd_hdac_aligned_write(v, addr, 0xff) -#define snd_hdac_reg_writew(v, addr) snd_hdac_aligned_write(v, addr, 0xffff) -#define snd_hdac_reg_readb(addr) snd_hdac_aligned_read(addr, 0xff) -#define snd_hdac_reg_readw(addr) snd_hdac_aligned_read(addr, 0xffff) -#else /* CONFIG_SND_HDA_ALIGNED_MMIO */ -#define snd_hdac_reg_writeb(val, addr) writeb(val, addr) -#define snd_hdac_reg_writew(val, addr) writew(val, addr) -#define snd_hdac_reg_readb(addr) readb(addr) -#define snd_hdac_reg_readw(addr) readw(addr) -#endif /* CONFIG_SND_HDA_ALIGNED_MMIO */ -#define snd_hdac_reg_writel(val, addr) writel(val, addr) -#define snd_hdac_reg_readl(addr) readl(addr) +#define snd_hdac_aligned_mmio(bus) (bus)->aligned_mmio +#else +#define snd_hdac_aligned_mmio(bus) false +#define snd_hdac_aligned_read(addr, mask) 0 +#define snd_hdac_aligned_write(val, addr, mask) do {} while (0) +#endif + +static inline void snd_hdac_reg_writeb(struct hdac_bus *bus, void __iomem *addr, + u8 val) +{ + if (snd_hdac_aligned_mmio(bus)) + snd_hdac_aligned_write(val, addr, 0xff); + else + writeb(val, addr); +} + +static inline void snd_hdac_reg_writew(struct hdac_bus *bus, void __iomem *addr, + u16 val) +{ + if (snd_hdac_aligned_mmio(bus)) + snd_hdac_aligned_write(val, addr, 0xffff); + else + writew(val, addr); +} + +static inline u8 snd_hdac_reg_readb(struct hdac_bus *bus, void __iomem *addr) +{ + return snd_hdac_aligned_mmio(bus) ? + snd_hdac_aligned_read(addr, 0xff) : readb(addr); +} + +static inline u16 snd_hdac_reg_readw(struct hdac_bus *bus, void __iomem *addr) +{ + return snd_hdac_aligned_mmio(bus) ? + snd_hdac_aligned_read(addr, 0xffff) : readw(addr); +} + +#define snd_hdac_reg_writel(bus, addr, val) writel(val, addr) +#define snd_hdac_reg_readl(bus, addr) readl(addr) /* * macros for easy use */ #define _snd_hdac_chip_writeb(chip, reg, value) \ - snd_hdac_reg_writeb(value, (chip)->remap_addr + (reg)) + snd_hdac_reg_writeb(chip, (chip)->remap_addr + (reg), value) #define _snd_hdac_chip_readb(chip, reg) \ - snd_hdac_reg_readb((chip)->remap_addr + (reg)) + snd_hdac_reg_readb(chip, (chip)->remap_addr + (reg)) #define _snd_hdac_chip_writew(chip, reg, value) \ - snd_hdac_reg_writew(value, (chip)->remap_addr + (reg)) + snd_hdac_reg_writew(chip, (chip)->remap_addr + (reg), value) #define _snd_hdac_chip_readw(chip, reg) \ - snd_hdac_reg_readw((chip)->remap_addr + (reg)) + snd_hdac_reg_readw(chip, (chip)->remap_addr + (reg)) #define _snd_hdac_chip_writel(chip, reg, value) \ - snd_hdac_reg_writel(value, (chip)->remap_addr + (reg)) + snd_hdac_reg_writel(chip, (chip)->remap_addr + (reg), value) #define _snd_hdac_chip_readl(chip, reg) \ - snd_hdac_reg_readl((chip)->remap_addr + (reg)) + snd_hdac_reg_readl(chip, (chip)->remap_addr + (reg)) /* read/write a register, pass without AZX_REG_ prefix */ #define snd_hdac_chip_writel(chip, reg, value) \ @@ -540,17 +569,17 @@ int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus, */ /* read/write a register, pass without AZX_REG_ prefix */ #define snd_hdac_stream_writel(dev, reg, value) \ - snd_hdac_reg_writel(value, (dev)->sd_addr + AZX_REG_ ## reg) + snd_hdac_reg_writel((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg, value) #define snd_hdac_stream_writew(dev, reg, value) \ - snd_hdac_reg_writew(value, (dev)->sd_addr + AZX_REG_ ## reg) + snd_hdac_reg_writew((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg, value) #define snd_hdac_stream_writeb(dev, reg, value) \ - snd_hdac_reg_writeb(value, (dev)->sd_addr + AZX_REG_ ## reg) + snd_hdac_reg_writeb((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg, value) #define snd_hdac_stream_readl(dev, reg) \ - snd_hdac_reg_readl((dev)->sd_addr + AZX_REG_ ## reg) + snd_hdac_reg_readl((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg) #define snd_hdac_stream_readw(dev, reg) \ - snd_hdac_reg_readw((dev)->sd_addr + AZX_REG_ ## reg) + snd_hdac_reg_readw((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg) #define snd_hdac_stream_readb(dev, reg) \ - snd_hdac_reg_readb((dev)->sd_addr + AZX_REG_ ## reg) + snd_hdac_reg_readb((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg) /* update a register, pass without AZX_REG_ prefix */ #define snd_hdac_stream_updatel(dev, reg, mask, val) \ -- cgit v1.2.3 From d4cfb30fce03093ad944e0b44bd8f40bdad5330e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Jan 2020 13:44:22 +0100 Subject: ALSA: pcm: Set per-card upper limit of PCM buffer allocations Currently, the available buffer allocation size for a PCM stream depends on the preallocated size; when a buffer has been preallocated, the max buffer size is set to that size, so that application won't re-allocate too much memory. OTOH, when no preallocation is done, each substream may allocate arbitrary size of buffers as long as snd_pcm_hardware.buffer_bytes_max allows -- which can be quite high, HD-audio sets 1GB there. It means that the system may consume a high amount of pages for PCM buffers, and they are pinned and never swapped out. This can lead to OOM easily. For avoiding such a situation, this patch adds the upper limit per card. Each snd_pcm_lib_malloc_pages() and _free_pages() calls are tracked and it will return an error if the total amount of buffers goes over the defined upper limit. The default value is set to 32MB, which should be really large enough for usual operations. If larger buffers are needed for any specific usage, it can be adjusted (also dynamically) via snd_pcm.max_alloc_per_card option. Setting zero there means no chceck is performed, and again, unlimited amount of buffers are allowed. Link: https://lore.kernel.org/r/20200120124423.11862-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/core.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index 0e14b7a3e67b..ac8b692b69b4 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -120,6 +120,9 @@ struct snd_card { int sync_irq; /* assigned irq, used for PCM sync */ wait_queue_head_t remove_sleep; + size_t total_pcm_alloc_bytes; /* total amount of allocated buffers */ + struct mutex memory_mutex; /* protection for the above */ + #ifdef CONFIG_PM unsigned int power_state; /* power state */ wait_queue_head_t power_sleep; -- cgit v1.2.3 From 450312b640f9c6caeeaf4bd8d4130f86ea27f456 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 20 Jan 2020 10:05:03 +0900 Subject: ASoC: soc-core: remove DAI suspend/resume Historically, CPU and Codec were implemented different, but now it is merged as Component. ALSA SoC is supporting suspend/resume at DAI and Component level. The method is like below. 1) Suspend/Resume all CPU DAI if bus-control was 0 2) Suspend/Resume all Component 3) Suspend/Resume all CPU DAI if bus-control was 1 Historically 2) was Codec special operation. Because CPU and Codec were merged into Component, CPU suspend/resume has 3 chance to suspend(= 1/2/3), but Codec suspend/resume has 1 chance (= 2). Here, DAI side suspend/resume is caring bus-control, but no driver which is supporting suspend/resume is setting bus-control. This means 3) was never used. Here, used parameter for suspend/resume component->dev and dai->dev are same pointer. For that reason, we can merge DAI and Component suspend/resume. One note is that we should use 2), because it is caring BIAS level. This patch removes 1) and 3). Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87r1zvx7i8.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 939c73db6a03..2ccecf3e03d5 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -286,8 +286,6 @@ struct snd_soc_dai_driver { /* DAI driver callbacks */ int (*probe)(struct snd_soc_dai *dai); int (*remove)(struct snd_soc_dai *dai); - int (*suspend)(struct snd_soc_dai *dai); - int (*resume)(struct snd_soc_dai *dai); /* compress dai */ int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); /* Optional Callback used at pcm creation*/ -- cgit v1.2.3 From 250a15cf575a655097151ad887cea02dd0977136 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 20 Jan 2020 10:05:07 +0900 Subject: ASoC: soc-core: remove bus_control Now, snd_soc_dai_driver::bus_control is used for how to resume. But, no driver which has bus_control has DAI driver suspend/resume support. This patch removes pointless bus_control from ALSA SoC. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87pnffx7i4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2ccecf3e03d5..eaaeb00e9e84 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -302,7 +302,6 @@ struct snd_soc_dai_driver { unsigned int symmetric_rates:1; unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; - unsigned int bus_control:1; /* DAI is also used for the control bus */ /* probe ordering - for components with runtime dependencies */ int probe_order; -- cgit v1.2.3 From 839284e794822ebc8b66f32006a5d4cad863e1fb Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Fri, 17 Jan 2020 15:38:13 +0800 Subject: ASoC: dapm: add snd_soc_dapm_put_enum_double_locked Adds snd_soc_dapm_put_enum_double_locked() for those use cases if dapm_mutex has already locked. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20200117073814.82441-3-tzungbi@google.com Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1b6afbc1a4ed..2a306c6f3fbc 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -392,6 +392,8 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_enum_double_locked(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, -- cgit v1.2.3