January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: simple-card: rename to asoc_simple_card_canonicalize_platform()
by Kuninori Morimoto 21 Jan '19

21 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current simple-card is using asoc_simple_card_canonicalize_dailink(). Its naming is "dailink", but is for "platform". We already have asoc_simple_card_canonicalize_cpu() for "cpu", let's follow same naming rule. It never return error, so, void function is better idea. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 11 +++-------- sound/soc/generic/simple-card-utils.c | 7 ++----- sound/soc/generic/simple-card.c | 11 +++-------- 4 files changed, 9 insertions(+), 22 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index ab5a2ba..7afe453 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -108,7 +108,7 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, int asoc_simple_card_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai); -int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link); +void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link); void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 42b077c..bb12351 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -307,14 +307,12 @@ static int graph_dai_link_of_dpcm(struct graph_priv *priv, "prefix"); } + asoc_simple_card_canonicalize_platform(dai_link); + ret = asoc_simple_card_of_parse_tdm(ep, dai); if (ret) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) @@ -405,10 +403,6 @@ static int graph_dai_link_of(struct graph_priv *priv, if (ret < 0) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, @@ -419,6 +413,7 @@ static int graph_dai_link_of(struct graph_priv *priv, dai_link->ops = &graph_ops; dai_link->init = graph_dai_init; + asoc_simple_card_canonicalize_platform(dai_link); asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 3c0901d..5c1424f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -394,16 +394,13 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); -int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) +void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ if (!dai_link->platforms->of_node) dai_link->platforms->of_node = dai_link->cpu_of_node; - - return 0; - } -EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink); +EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_platform); void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d8a0d1e..08df261 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -297,14 +297,12 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, simple_get_conversion(dev, np, &dai_props->adata); + asoc_simple_card_canonicalize_platform(dai_link); + ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - snprintf(prop, sizeof(prop), "%smclk-fs", prefix); of_property_read_u32(top, PREFIX "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(node, prop, &dai_props->mclk_fs); @@ -409,10 +407,6 @@ static int simple_dai_link_of(struct simple_priv *priv, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - goto dai_link_of_err; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, @@ -424,6 +418,7 @@ static int simple_dai_link_of(struct simple_priv *priv, dai_link->init = simple_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_card_canonicalize_platform(dai_link); dai_link_of_err: of_node_put(node); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: dapm: Skip suspending widgets with ignore flag
by Zhiwei Jiang 21 Jan '19

21 Jan '19

For endpoint wigdets which have set the suspend ignore flag, asoc shouldn't mark them as dirty when ASoC suspend function is called. Signed-off-by: Zhiwei Jiang <qq282012236(a)gmail.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..68c49090f561 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -278,7 +278,7 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &card->widgets, list) { - if (w->is_ep) { + if (w->is_ep && !w->ignore_suspend) { dapm_mark_dirty(w, "Rechecking endpoints"); if (w->is_ep & SND_SOC_DAPM_EP_SINK) dapm_widget_invalidate_output_paths(w); -- 2.19.0

2 1

[alsa-devel] testing hardware with alsaloop.
by tom＠hitechsys.com 21 Jan '19

21 Jan '19

Hello Im running alsaloop test to test the audio on our hardware. Im specifically running this command - alsaloop -C hw:0,0 -P hw:1,0 -S 1 -l 480 -c 6 -v When I measure the delay through my hardware I get 11mSec - which is fine, and the audio quality is good as well. However after several hours of running this loop test the delay has increased to 90mSec. Can someone advice what is going wrong here please. Best regards Tom Favell | Hi Tech Systems Ltd http://www.hitechsys.com For support go to https://support.hitechsys.com/ T: +44 (0) 1256 780880 M: +44 (0) 79797 55359 http://www.hitechsys.com © 2019 Hi Tech Systems Ltd. Holbrook House, Oakley Lane, Basingstoke, RG23 7JY, UK Registered in England No. 2130346 | VAT No. GB 450390370 The information contained in this e-mail is confidential and is intended solely for the use of the named addressee. Access, copying or re-use of the information in it by any other person is not authorized. The Internet is not a secure transmission medium and therefore the content of this e-mail may have been changed without the consent of the originator. If you are not the intended recipient, please destroy this email and contact the sender.http://www.hitechsys.com

1 0

[alsa-devel] [PATCH] ALSA: hda - Add mute LED support for HP ProBook 470 G5
by Anthony Wong 21 Jan '19

21 Jan '19

From: Anthony Wong <anthony.wong(a)canonical.com> Support speaker and mic mute LEDs on HP ProBook 470 G5. BugLink: https://bugs.launchpad.net/bugs/1811254 Signed-off-by: Anthony Wong <anthony.wong(a)canonical.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 88ce2f1..8b5071b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -966,6 +966,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.7.4

2 1

[alsa-devel] [PATCH 0/9] ALSA: fireface: add support for Fireface UCX
by Takashi Sakamoto 21 Jan '19

21 Jan '19

Hi, In 2012, RME GmbH shipped Fireface UFX. This model is one of latter models of Fireface series and support both of IEEE 1394 bus and USB. Furthermore, it supports two types of remote control unit (Basic Remote and Advanced Remote Control) with by 9pin mini-din connector. This patchset adds support for this model, as a part of devices by ALSA firewire stack (therefore it's not a device of USB). At present, maintain of isochronous communication and synchronization status are cleared but asynchronous communication for MIDI messages are not yet, therefore this commit adds support for isochronous communication for PCM frames. This work is based on my packet analysis from rent device for a short term (2 weeks). I'm happy to get your responses till next weekend. Driver code in a remote branch is available for backport till Linux kernel v4.17 or later for your test. https://github.com/takaswie/snd-firewire-improve/tree/topic/ff800-midi Regards Takashi Sakamoto (9): ALSA: fireface: rename protocol layer for former models ALSA: fireface: unify protocol layer for FF400/FF800 ALSA: fireface: obsolete proc node to leave one node ALSA: fireface: add protocol-dependent operation to dump status ALSA: fireface: add protocol-dependent operation to switch mode to fetch PCM frame ALSA: fireface: add protocol-dependent operation to get clock status ALSA: fireface: code refactoring for dump of sync status ALSA: fireface: code refactoring to parse of clock configuration ALSA: fireface: add support for Fireface UCX sound/firewire/Kconfig | 1 + sound/firewire/fireface/Makefile | 4 +- sound/firewire/fireface/ff-pcm.c | 2 +- sound/firewire/fireface/ff-proc.c | 217 +------- sound/firewire/fireface/ff-protocol-ff400.c | 161 ------ sound/firewire/fireface/ff-protocol-ff800.c | 143 ----- sound/firewire/fireface/ff-protocol-former.c | 549 +++++++++++++++++++ sound/firewire/fireface/ff-protocol-latter.c | 273 +++++++++ sound/firewire/fireface/ff-stream.c | 40 +- sound/firewire/fireface/ff-transaction.c | 63 --- sound/firewire/fireface/ff.c | 41 +- sound/firewire/fireface/ff.h | 13 +- 12 files changed, 886 insertions(+), 621 deletions(-) delete mode 100644 sound/firewire/fireface/ff-protocol-ff400.c delete mode 100644 sound/firewire/fireface/ff-protocol-ff800.c create mode 100644 sound/firewire/fireface/ff-protocol-former.c create mode 100644 sound/firewire/fireface/ff-protocol-latter.c -- 2.19.1

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[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Switch to modern UUID API
by Andy Shevchenko 21 Jan '19

21 Jan '19

Switch the driver to use modern UUID API, i.e. guid_t type and accompanying functions, such as guid_equal(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 12 ++++++------ sound/soc/intel/skylake/skl-sst-dsp.h | 6 +++--- sound/soc/intel/skylake/skl-sst-utils.c | 23 +++++++---------------- sound/soc/intel/skylake/skl-sst.c | 4 ++-- sound/soc/intel/skylake/skl-topology.c | 24 ++++++++++++------------ sound/soc/intel/skylake/skl-topology.h | 6 +++--- 6 files changed, 33 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..86f8ebc77a32 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1268,12 +1268,12 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; struct skl_module_inst_id *pin_id; - uuid_le *uuid_mod, *uuid_tplg; + guid_t *uuid_mod, *uuid_tplg; struct skl_module *skl_module; struct uuid_module *module; int i, ret = -EIO; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (list_empty(&ctx->uuid_list)) { dev_err(ctx->dev, "Module list is empty\n"); @@ -1281,7 +1281,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) } list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { mconfig->id.module_id = module->id; if (mconfig->module) mconfig->module->loadable = module->is_loadable; @@ -1298,7 +1298,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; - if (!uuid_le_cmp(*uuid_mod, *uuid_tplg)) { + if (guid_equal(uuid_mod, uuid_tplg)) { mconfig->module = skl_module; ret = 0; break; @@ -1310,13 +1310,13 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) list_for_each_entry(module, &ctx->uuid_list, list) { for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } for (i = 0; i < MAX_OUT_QUEUE; i++) { pin_id = &mconfig->m_out_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } } diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index e1d6f6719f7e..cbc7a93d56c2 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -177,7 +177,7 @@ struct skl_dsp_loader_ops { #define MAX_INSTANCE_BUFF 2 struct uuid_module { - uuid_le uuid; + guid_t uuid; int id; int is_loadable; int max_instance; @@ -241,8 +241,8 @@ void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, unsigned int offset, int index); -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id); -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id); +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id); +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id); int skl_get_pvt_instance_id_map(struct skl_sst *ctx, int module_id, int instance_id); void skl_freeup_uuid_list(struct skl_sst *ctx); diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 2ae405617876..85551321c35b 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -21,17 +21,11 @@ #include "../common/sst-dsp-priv.h" #include "skl-sst-ipc.h" - -#define UUID_STR_SIZE 37 #define DEFAULT_HASH_SHA256_LEN 32 /* FW Extended Manifest Header id = $AE1 */ #define SKL_EXT_MANIFEST_HEADER_MAGIC 0x31454124 -struct UUID { - u8 id[16]; -}; - union seg_flags { u32 ul; struct { @@ -65,7 +59,7 @@ struct module_type { struct adsp_module_entry { u32 struct_id; u8 name[8]; - struct UUID uuid; + u8 uuid[16]; struct module_type type; u8 hash1[DEFAULT_HASH_SHA256_LEN]; u32 entry_point; @@ -184,13 +178,13 @@ static inline int skl_pvtid_128(struct uuid_module *module) * This generates a 128 bit private unique id for a module TYPE so that * module instance is unique */ -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id) +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id) { struct uuid_module *module; int pvt_id; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { pvt_id = skl_pvtid_128(module); if (pvt_id >= 0) { @@ -214,13 +208,13 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id); * * This frees a 128 bit private unique id previously generated */ -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id) +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id) { int i; struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { if (*pvt_id != 0) i = (*pvt_id) / 64; @@ -247,7 +241,6 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, struct adsp_fw_hdr *adsp_hdr; struct adsp_module_entry *mod_entry; int i, num_entry, size; - uuid_le *uuid_bin; const char *buf; struct skl_sst *skl = ctx->thread_context; struct uuid_module *module; @@ -279,8 +272,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, return -EINVAL; } - mod_entry = (struct adsp_module_entry *) - (buf + offset + adsp_hdr->len); + mod_entry = (struct adsp_module_entry *)(buf + offset + adsp_hdr->len); num_entry = adsp_hdr->num_modules; @@ -307,8 +299,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, goto free_uuid_list; } - uuid_bin = (uuid_le *)mod_entry->uuid.id; - memcpy(&module->uuid, uuid_bin, sizeof(module->uuid)); + guid_copy(&module->uuid, (guid_t *)&mod_entry->uuid); module->id = (i | (index << 12)); module->is_loadable = mod_entry->type.load_type; diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 5951bbdf1f1a..92806119a755 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -420,9 +420,9 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - uuid_le *uuid_mod; + guid_t *uuid_mod; - uuid_mod = (uuid_le *)guid; + uuid_mod = (guid_t *)guid; snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", "intel/dsp_fw_", uuid_mod, ".bin"); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 76dde12cc2bb..160a5c864b3f 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -577,7 +577,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) int ret = 0; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; w = w_module->w; mconfig = w->priv; @@ -585,7 +585,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) if (mconfig->id.module_id < 0) { dev_err(skl->skl_sst->dev, "module %pUL id not populated\n", - (uuid_le *)mconfig->guid); + (guid_t *)mconfig->guid); return -EIO; } @@ -619,7 +619,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) * FE/BE params */ skl_tplg_update_module_params(w, ctx); - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; mconfig->id.pvt_id = skl_get_pvt_id(ctx, uuid_mod, mconfig->id.instance_id); if (mconfig->id.pvt_id < 0) @@ -658,9 +658,9 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, struct skl_module_cfg *mconfig = NULL; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; mconfig = w_module->w->priv; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod && mconfig->m_state > SKL_MODULE_UNINIT) { @@ -916,12 +916,12 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, return 0; } -static int skl_get_module_id(struct skl_sst *ctx, uuid_le *uuid) +static int skl_get_module_id(struct skl_sst *ctx, guid_t *uuid) { struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid, module->uuid) == 0) + if (guid_equal(uuid, &module->uuid)) return module->id; } @@ -2121,11 +2121,11 @@ static int skl_tplg_add_pipe(struct device *dev, return 0; } -static int skl_tplg_get_uuid(struct device *dev, u8 *guid, +static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) { if (uuid_tkn->token == SKL_TKN_UUID) { - memcpy(guid, &uuid_tkn->uuid, 16); + guid_copy(guid, (guid_t *)&uuid_tkn->uuid); return 0; } @@ -2151,7 +2151,7 @@ static int skl_tplg_fill_pin(struct device *dev, break; case SKL_TKN_UUID: - ret = skl_tplg_get_uuid(dev, m_pin[pin_index].id.mod_uuid.b, + ret = skl_tplg_get_uuid(dev, &m_pin[pin_index].id.mod_uuid, (struct snd_soc_tplg_vendor_uuid_elem *)tkn_elem); if (ret < 0) return ret; @@ -2666,7 +2666,7 @@ static int skl_tplg_get_tokens(struct device *dev, case SND_SOC_TPLG_TUPLE_TYPE_UUID: if (is_module_guid) { - ret = skl_tplg_get_uuid(dev, mconfig->guid, + ret = skl_tplg_get_uuid(dev, (guid_t *)mconfig->guid, array->uuid); is_module_guid = false; } else { @@ -3484,7 +3484,7 @@ static int skl_tplg_get_manifest_uuid(struct device *dev, if (uuid_tkn->token == SKL_TKN_UUID) { mod = skl->modules[ref_count]; - memcpy(&mod->uuid, &uuid_tkn->uuid, sizeof(uuid_tkn->uuid)); + guid_copy(&mod->uuid, (guid_t *)&uuid_tkn->uuid); ref_count++; } else { dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 82282cac9751..9311e248b6f9 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -224,7 +224,7 @@ struct skl_mod_inst_map { struct skl_uuid_inst_map { u16 inst_id; u16 reserved; - uuid_le mod_uuid; + guid_t mod_uuid; } __packed; struct skl_kpb_params { @@ -236,7 +236,7 @@ struct skl_kpb_params { }; struct skl_module_inst_id { - uuid_le mod_uuid; + guid_t mod_uuid; int module_id; u32 instance_id; int pvt_id; @@ -369,7 +369,7 @@ struct skl_module_res { }; struct skl_module { - uuid_le uuid; + guid_t uuid; u8 loadable; u8 input_pin_type; u8 output_pin_type; -- 2.18.0

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[alsa-devel] [PATCH 0/2] ASoC: soc-core: use snd_soc_dai_link_component for cpu
by Kuninori Morimoto 21 Jan '19

21 Jan '19

Hi Mark These add snd_soc_dai_link_component support for CPU. I know multi-CPU support patch was posted to ML, but I think it is not yet accepted (or rejected ?). If these patches were accepted, we can finally switch to modern style dai_link, and remove legacy style dai_link. Kuninori Morimoto (2): ASoC: soc-core: add .num_platform for dai_link ASoC: soc-core: use snd_soc_dai_link_component for cpu For switching to modern, I added "use modern style dai_link" and "remove legacy style dai_link" patches as sample. If these have no objection, I will post full patch to ML. Kuninori Morimoto (3): ASoC: audio-graph/simple-card: use modern style dai_link for CPU ASoC: atmel: atmel-classd: use modern style dai_link ASoC: soc-core: remove regacy dai_link binding method --- Kuninori Morimoto

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[alsa-devel] [PATCH v2] ASoC: soc-core: add .num_platforms for dai_link
by Kuninori Morimoto 21 Jan '19

21 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current snd_soc_dai_link is starting to use snd_soc_dai_link_component style for Platform, but it is assuming single Platform so far. But We will need to have multi Platform support in the not far future. Currently only simple card is using it as sound card driver, and other drivers are converted to it from legacy style by snd_soc_init_platform(). To avoid future problem of multi Platform support, let's add num_platforms before it is too late. In the same time, to make it same naming mothed, "platform" should be "platforms". This patch fixup it too. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - pickup this patch from posted patch-set include/sound/simple_card_utils.h | 2 +- include/sound/soc.h | 3 ++- sound/soc/generic/audio-graph-card.c | 5 +++-- sound/soc/generic/simple-card-utils.c | 4 ++-- sound/soc/generic/simple-card.c | 7 ++++--- sound/soc/soc-core.c | 21 +++++++++++++++------ 6 files changed, 27 insertions(+), 15 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 6d69ed2..ab5a2ba 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -75,7 +75,7 @@ void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); &dai_link->codec_dai_name, \ list_name, cells_name, NULL) #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, dai_link->platform, \ + asoc_simple_card_parse_dai(node, dai_link->platforms, \ &dai_link->platform_of_node, \ NULL, list_name, cells_name, NULL) int asoc_simple_card_parse_dai(struct device_node *node, diff --git a/include/sound/soc.h b/include/sound/soc.h index c31b6d1..3089257 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -961,7 +961,8 @@ struct snd_soc_dai_link { */ const char *platform_name; struct device_node *platform_of_node; - struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link_component *platforms; + unsigned int num_platforms; int id; /* optional ID for machine driver link identification */ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 3ec96cd..42b077c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -26,7 +26,7 @@ struct graph_priv { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ - struct snd_soc_dai_link_component platform; + struct snd_soc_dai_link_component platforms; struct asoc_simple_card_data adata; struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; @@ -687,7 +687,8 @@ static int graph_probe(struct platform_device *pdev) for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; + dai_link[i].platforms = &dai_props[i].platforms; + dai_link[i].num_platforms = 1; } priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 336895f..3c0901d 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -397,8 +397,8 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ - if (!dai_link->platform->of_node) - dai_link->platform->of_node = dai_link->cpu_of_node; + if (!dai_link->platforms->of_node) + dai_link->platforms->of_node = dai_link->cpu_of_node; return 0; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 479de23..d8a0d1e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -21,7 +21,7 @@ struct simple_priv { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ - struct snd_soc_dai_link_component platform; + struct snd_soc_dai_link_component platforms; struct asoc_simple_card_data adata; struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; @@ -732,7 +732,8 @@ static int simple_probe(struct platform_device *pdev) for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; + dai_link[i].platforms = &dai_props[i].platforms; + dai_link[i].num_platforms = 1; } priv->dai_props = dai_props; @@ -782,7 +783,7 @@ static int simple_probe(struct platform_device *pdev) codecs->name = cinfo->codec; codecs->dai_name = cinfo->codec_dai.name; - platform = dai_link->platform; + platform = dai_link->platforms; platform->name = cinfo->platform; card->name = (cinfo->card) ? cinfo->card : cinfo->name; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bf405d6..58218d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -921,7 +921,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* find one from the set of registered platforms */ for_each_component(component) { - if (!snd_soc_is_matching_component(dai_link->platform, + if (!snd_soc_is_matching_component(dai_link->platforms, component)) continue; @@ -1035,7 +1035,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card) static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { - struct snd_soc_dai_link_component *platform = dai_link->platform; + struct snd_soc_dai_link_component *platform = dai_link->platforms; /* * REMOVE ME @@ -1055,7 +1055,8 @@ static int snd_soc_init_platform(struct snd_soc_card *card, if (!platform) return -ENOMEM; - dai_link->platform = platform; + dai_link->platforms = platform; + dai_link->num_platforms = 1; dai_link->legacy_platform = 1; platform->name = dai_link->platform_name; platform->of_node = dai_link->platform_of_node; @@ -1145,11 +1146,19 @@ static int soc_init_dai_link(struct snd_soc_card *card, } } + /* FIXME */ + if (link->num_platforms > 1) { + dev_err(card->dev, + "ASoC: multi platform is not yet supported %s\n", + link->name); + return -EINVAL; + } + /* * Platform may be specified by either name or OF node, but * can be left unspecified, and a dummy platform will be used. */ - if (link->platform->name && link->platform->of_node) { + if (link->platforms->name && link->platforms->of_node) { dev_err(card->dev, "ASoC: Both platform name/of_node are set for %s\n", link->name); @@ -1160,7 +1169,7 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Defer card registartion if platform dai component is not added to * component list. */ - if (!soc_find_component(link->platform->of_node, link->platform->name)) + if (!soc_find_component(link->platforms->of_node, link->platforms->name)) return -EPROBE_DEFER; /* @@ -1959,7 +1968,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dev_err(card->dev, "init platform error"); continue; } - dai_link->platform->name = component->name; + dai_link->platforms->name = component->name; /* convert non BE into BE */ dai_link->no_pcm = 1; -- 2.7.4

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[alsa-devel] simple-audio-card and headphone jack detect
by Paul Cercueil 19 Jan '19

19 Jan '19

Hello, I am using the simple-card and simple-amplifier drivers with the following devicetree: amp: audio-amp { compatible = "simple-audio-amplifier"; enable-gpios = <&gpc 15 GPIO_ACTIVE_HIGH>; }; sound-card { compatible = "simple-audio-card"; simple-audio-card,name = "rs90-audio"; simple-audio-card,format = "i2s"; simple-audio-card,widgets = "Speaker", "Speaker", "Headphone", "Headphones"; simple-audio-card,routing = "INL", "LHPOUT", "INR", "RHPOUT", "Headphones", "LHPOUT", "Headphones", "RHPOUT", "Speaker", "OUTL", "Speaker", "OUTR"; simple-audio-card,hp-det-gpio = <&gpd 16 GPIO_ACTIVE_LOW>; simple-audio-card,aux-devs = <&amp>; simple-audio-card,cpu { sound-dai = <&aic>; }; simple-audio-card,codec { sound-dai = <&codec>; bitclock-master; frame-master; }; }; My codec outputs on the LHPOUT/RHPOUT lines. INL/INR and OUTL/OUTR are the inputs/outputs of the simple-amplifier. So far so good, when no audio is playing the audio amp is disabled (GPIO is low), and it gets enabled as soon as I start to play audio. What is not working, and what I am trying to achieve, is to have the external audio amplifier disabled as soon as headphones are plugged. Is that something the core handles for me? Any suggestions to make it work? Thanks, -Paul

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[alsa-devel] [PATCH] ASoC: fsl_sai: Remove expensive print in irq handler
by Daniel Baluta 18 Jan '19

18 Jan '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> When stopping audio, ASoC will first stop DMA then CPU DAI. Sometimes there is a delay between DMA stop and CPU DAI stop, which triggers an underrun error. Now, because of the delay introduced by dev_err another underrun error will occur causing a vicious circle making impossible to stop CPU DAI. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4163f2cfc06f..db9e0872f73d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -62,10 +62,10 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) dev_dbg(dev, "isr: Start of Tx word detected\n"); if (flags & FSL_SAI_CSR_SEF) - dev_warn(dev, "isr: Tx Frame sync error detected\n"); + dev_dbg(dev, "isr: Tx Frame sync error detected\n"); if (flags & FSL_SAI_CSR_FEF) { - dev_warn(dev, "isr: Transmit underrun detected\n"); + dev_dbg(dev, "isr: Transmit underrun detected\n"); /* FIFO reset for safety */ xcsr |= FSL_SAI_CSR_FR; } @@ -96,10 +96,10 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) dev_dbg(dev, "isr: Start of Rx word detected\n"); if (flags & FSL_SAI_CSR_SEF) - dev_warn(dev, "isr: Rx Frame sync error detected\n"); + dev_dbg(dev, "isr: Rx Frame sync error detected\n"); if (flags & FSL_SAI_CSR_FEF) { - dev_warn(dev, "isr: Receive overflow detected\n"); + dev_dbg(dev, "isr: Receive overflow detected\n"); /* FIFO reset for safety */ xcsr |= FSL_SAI_CSR_FR; } -- 2.17.1

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