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[alsa-devel] [PATCH][next] ALSA: xen-front: remove redundant error check on ret
by Colin King 28 May '18

28 May '18

From: Colin Ian King <colin.king(a)canonical.com> The error for a -ve value in ret is redundant as all previous assignments to ret have an associated -ve check and hence it is impossible for ret to be less that zero at the point of the check. Remove this redundant error check. Detected by CoveritScan, CID#1469407 ("Logically Dead code") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/xen/xen_snd_front_evtchnl.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c index d70a62e7f910..102d6e096cc8 100644 --- a/sound/xen/xen_snd_front_evtchnl.c +++ b/sound/xen/xen_snd_front_evtchnl.c @@ -351,8 +351,6 @@ int xen_snd_front_evtchnl_create_all(struct xen_snd_front_info *front_info, } } } - if (ret < 0) - goto fail; front_info->num_evt_pairs = num_streams; return 0; -- 2.17.0

3 2

[alsa-devel] [PATCH] sound: Use octal not symbolic permissions
by Joe Perches 28 May '18

28 May '18

Convert the S_<FOO> symbolic permissions to their octal equivalents as using octal and not symbolic permissions is preferred by many as more readable. see: https://lkml.org/lkml/2016/8/2/1945 Done with automated conversion via: $ ./scripts/checkpatch.pl -f --types=SYMBOLIC_PERMS --fix-inplace <files...> Miscellanea: o Wrapped one multi-line call to a single line Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/core/compress_offload.c | 2 +- sound/core/info.c | 6 +++--- sound/core/init.c | 4 ++-- sound/core/oss/mixer_oss.c | 2 +- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 10 +++++----- sound/core/pcm_memory.c | 2 +- sound/drivers/dummy.c | 2 +- sound/drivers/mts64.c | 6 +++--- sound/drivers/opl4/opl4_proc.c | 2 +- sound/drivers/portman2x4.c | 6 +++--- sound/firewire/bebob/bebob_proc.c | 2 +- sound/firewire/dice/dice-proc.c | 2 +- sound/firewire/digi00x/digi00x-proc.c | 2 +- sound/firewire/fireface/ff-proc.c | 2 +- sound/firewire/fireworks/fireworks_proc.c | 2 +- sound/firewire/motu/motu-proc.c | 2 +- sound/firewire/oxfw/oxfw-proc.c | 2 +- sound/firewire/tascam/tascam-proc.c | 2 +- sound/isa/msnd/msnd_pinnacle.c | 32 +++++++++++++++---------------- sound/pci/ac97/ac97_proc.c | 4 ++-- sound/pci/asihpi/asihpi.c | 12 ++++++------ sound/pci/asihpi/hpioctl.c | 4 ++-- sound/pci/ca0106/ca0106_proc.c | 6 +++--- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos.c | 14 +++++++------- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/ctxfi/cttimer.c | 2 +- sound/pci/ctxfi/xfi.c | 4 ++-- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/emu10k1/emuproc.c | 22 ++++++++++----------- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/ice1712/pontis.c | 2 +- sound/pci/ice1712/prodigy_hifi.c | 2 +- sound/pci/lola/lola_proc.c | 2 +- sound/pci/pcxhr/pcxhr.c | 2 +- sound/soc/codecs/cs43130.c | 8 ++++---- sound/soc/codecs/wm_adsp.c | 11 +++++------ sound/soc/fsl/fsl_ssi_dbg.c | 2 +- sound/sound_core.c | 6 +++--- sound/sparc/dbri.c | 2 +- 41 files changed, 102 insertions(+), 103 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 4563432badba..4b01a37c836e 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -1001,7 +1001,7 @@ static int snd_compress_proc_init(struct snd_compr *compr) compr->card->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; diff --git a/sound/core/info.c b/sound/core/info.c index 4b36767af9e1..fe502bc5e6d2 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -454,7 +454,7 @@ static struct snd_info_entry *create_subdir(struct module *mod, entry = snd_info_create_module_entry(mod, name, NULL); if (!entry) return NULL; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return NULL; @@ -470,7 +470,7 @@ int __init snd_info_init(void) snd_proc_root = snd_info_create_entry("asound", NULL); if (!snd_proc_root) return -ENOMEM; - snd_proc_root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + snd_proc_root->mode = S_IFDIR | 0555; snd_proc_root->p = proc_mkdir("asound", NULL); if (!snd_proc_root->p) goto error; @@ -716,7 +716,7 @@ snd_info_create_entry(const char *name, struct snd_info_entry *parent) kfree(entry); return NULL; } - entry->mode = S_IFREG | S_IRUGO; + entry->mode = S_IFREG | 0444; entry->content = SNDRV_INFO_CONTENT_TEXT; mutex_init(&entry->access); INIT_LIST_HEAD(&entry->children); diff --git a/sound/core/init.c b/sound/core/init.c index 79b4df1c1dc7..4849c611c0fe 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -703,7 +703,7 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, return count; } -static DEVICE_ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr); +static DEVICE_ATTR(id, 0644, card_id_show_attr, card_id_store_attr); static ssize_t card_number_show_attr(struct device *dev, @@ -713,7 +713,7 @@ card_number_show_attr(struct device *dev, return scnprintf(buf, PAGE_SIZE, "%i\n", card->number); } -static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); +static DEVICE_ATTR(number, 0444, card_number_show_attr, NULL); static struct attribute *card_dev_attrs[] = { &dev_attr_id.attr, diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 379bf486ccc7..64d904bee8bb 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1247,7 +1247,7 @@ static void snd_mixer_oss_proc_init(struct snd_mixer_oss *mixer) if (! entry) return; entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = snd_mixer_oss_proc_read; entry->c.text.write = snd_mixer_oss_proc_write; entry->private_data = mixer; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 1980f68246cb..905a53c1cde5 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -3045,7 +3045,7 @@ static void snd_pcm_oss_proc_init(struct snd_pcm *pcm) continue; if ((entry = snd_info_create_card_entry(pcm->card, "oss", pstr->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = snd_pcm_oss_proc_read; entry->c.text.write = snd_pcm_oss_proc_write; entry->private_data = pstr; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 66ac89aad681..c352bfb973cc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -530,7 +530,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) pcm->card->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; @@ -552,7 +552,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) if (entry) { entry->c.text.read = snd_pcm_xrun_debug_read; entry->c.text.write = snd_pcm_xrun_debug_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = pstr; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -590,7 +590,7 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) substream->pstr->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; @@ -647,7 +647,7 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) entry->private_data = substream; entry->c.text.read = NULL; entry->c.text.write = snd_pcm_xrun_injection_write; - entry->mode = S_IFREG | S_IWUSR; + entry->mode = S_IFREG | 0200; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -1087,7 +1087,7 @@ static ssize_t show_pcm_class(struct device *dev, return snprintf(buf, PAGE_SIZE, "%s\n", str); } -static DEVICE_ATTR(pcm_class, S_IRUGO, show_pcm_class, NULL); +static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); static struct attribute *pcm_dev_attrs[] = { &dev_attr_pcm_class.attr, NULL diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index ae33e456708c..4b5356a10315 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -201,7 +201,7 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc", substream->proc_root)) != NULL) { entry->c.text.read = snd_pcm_lib_preallocate_proc_read; entry->c.text.write = snd_pcm_lib_preallocate_proc_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = substream; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 8fb9a54fe8ba..9af154db530a 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1042,7 +1042,7 @@ static void dummy_proc_init(struct snd_dummy *chip) if (!snd_card_proc_new(chip->card, "dummy_pcm", &entry)) { snd_info_set_text_ops(entry, chip, dummy_proc_read); entry->c.text.write = dummy_proc_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = chip; } } diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index f32e81342247..b68e71ca7abd 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -41,11 +41,11 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); MODULE_AUTHOR("Matthias Koenig <mk(a)phasorlab.de>"); diff --git a/sound/drivers/opl4/opl4_proc.c b/sound/drivers/opl4/opl4_proc.c index cd2c07fa2ef4..16b24091d799 100644 --- a/sound/drivers/opl4/opl4_proc.c +++ b/sound/drivers/opl4/opl4_proc.c @@ -104,7 +104,7 @@ int snd_opl4_create_proc(struct snd_opl4 *opl4) if (entry) { if (opl4->hardware < OPL3_HW_OPL4_ML) { /* OPL4 can access 4 MB external ROM/SRAM */ - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->size = 4 * 1024 * 1024; } else { /* OPL4-ML has 1 MB internal ROM */ diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index ec8a94325ef6..3cdf0a88d71b 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -60,11 +60,11 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); MODULE_AUTHOR("Levent Guendogdu, Tobias Gehrig, Matthias Koenig"); diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c index ec24f96794f5..8096891af913 100644 --- a/sound/firewire/bebob/bebob_proc.c +++ b/sound/firewire/bebob/bebob_proc.c @@ -183,7 +183,7 @@ void snd_bebob_proc_init(struct snd_bebob *bebob) bebob->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index faf8c854d256..bb870fc73f99 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -305,7 +305,7 @@ void snd_dice_create_proc(struct snd_dice *dice) dice->card->proc_root); if (!root) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/digi00x/digi00x-proc.c b/sound/firewire/digi00x/digi00x-proc.c index a1d601f31165..6996d5a6ff5f 100644 --- a/sound/firewire/digi00x/digi00x-proc.c +++ b/sound/firewire/digi00x/digi00x-proc.c @@ -79,7 +79,7 @@ void snd_dg00x_proc_init(struct snd_dg00x *dg00x) if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/fireface/ff-proc.c b/sound/firewire/fireface/ff-proc.c index 69441d121f71..40ccbfd8ef89 100644 --- a/sound/firewire/fireface/ff-proc.c +++ b/sound/firewire/fireface/ff-proc.c @@ -52,7 +52,7 @@ void snd_ff_proc_init(struct snd_ff *ff) ff->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/fireworks/fireworks_proc.c b/sound/firewire/fireworks/fireworks_proc.c index 9c21f31b8b21..779ecec5af62 100644 --- a/sound/firewire/fireworks/fireworks_proc.c +++ b/sound/firewire/fireworks/fireworks_proc.c @@ -219,7 +219,7 @@ void snd_efw_proc_init(struct snd_efw *efw) efw->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c index 4edc064999ed..ab6830a6d242 100644 --- a/sound/firewire/motu/motu-proc.c +++ b/sound/firewire/motu/motu-proc.c @@ -107,7 +107,7 @@ void snd_motu_proc_init(struct snd_motu *motu) motu->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c index 8ba4f9f262b8..27dac071bc73 100644 --- a/sound/firewire/oxfw/oxfw-proc.c +++ b/sound/firewire/oxfw/oxfw-proc.c @@ -103,7 +103,7 @@ void snd_oxfw_proc_init(struct snd_oxfw *oxfw) oxfw->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/tascam/tascam-proc.c b/sound/firewire/tascam/tascam-proc.c index bfd4a4c06914..fee3bf32a0da 100644 --- a/sound/firewire/tascam/tascam-proc.c +++ b/sound/firewire/tascam/tascam-proc.c @@ -78,7 +78,7 @@ void snd_tscm_proc_init(struct snd_tscm *tscm) tscm->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 45e561c425bf..6c584d9b6c42 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -757,9 +757,9 @@ static int snd_msnd_pinnacle_cfg_reset(int cfg) static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard."); static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; @@ -801,22 +801,22 @@ MODULE_LICENSE("GPL"); MODULE_FIRMWARE(INITCODEFILE); MODULE_FIRMWARE(PERMCODEFILE); -module_param_hw_array(io, long, ioport, NULL, S_IRUGO); +module_param_hw_array(io, long, ioport, NULL, 0444); MODULE_PARM_DESC(io, "IO port #"); -module_param_hw_array(irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(mem, long, iomem, NULL, S_IRUGO); -module_param_array(write_ndelay, int, NULL, S_IRUGO); -module_param(calibrate_signal, int, S_IRUGO); +module_param_hw_array(irq, int, irq, NULL, 0444); +module_param_hw_array(mem, long, iomem, NULL, 0444); +module_param_array(write_ndelay, int, NULL, 0444); +module_param(calibrate_signal, int, 0444); #ifndef MSND_CLASSIC -module_param_array(digital, int, NULL, S_IRUGO); -module_param_hw_array(cfg, long, ioport, NULL, S_IRUGO); -module_param_array(reset, int, 0, S_IRUGO); -module_param_hw_array(mpu_io, long, ioport, NULL, S_IRUGO); -module_param_hw_array(mpu_irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(ide_io0, long, ioport, NULL, S_IRUGO); -module_param_hw_array(ide_io1, long, ioport, NULL, S_IRUGO); -module_param_hw_array(ide_irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(joystick_io, long, ioport, NULL, S_IRUGO); +module_param_array(digital, int, NULL, 0444); +module_param_hw_array(cfg, long, ioport, NULL, 0444); +module_param_array(reset, int, 0, 0444); +module_param_hw_array(mpu_io, long, ioport, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); +module_param_hw_array(ide_io0, long, ioport, NULL, 0444); +module_param_hw_array(ide_io1, long, ioport, NULL, 0444); +module_param_hw_array(ide_irq, int, irq, NULL, 0444); +module_param_hw_array(joystick_io, long, ioport, NULL, 0444); #endif diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 6320bf084e47..e120a11c69e8 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -448,7 +448,7 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) { snd_info_set_text_ops(entry, ac97, snd_ac97_proc_regs_read); #ifdef CONFIG_SND_DEBUG - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = snd_ac97_proc_regs_write; #endif if (snd_info_register(entry) < 0) { @@ -474,7 +474,7 @@ void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus) sprintf(name, "codec97#%d", bus->num); if ((entry = snd_info_create_card_entry(bus->card, name, bus->card->proc_root)) != NULL) { - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 720361455c60..64e0961f93ba 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -69,27 +69,27 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static bool enable_hpi_hwdep = 1; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "ALSA index value for AudioScience soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ALSA ID string for AudioScience soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "ALSA enable AudioScience soundcard."); -module_param(enable_hpi_hwdep, bool, S_IRUGO|S_IWUSR); +module_param(enable_hpi_hwdep, bool, 0644); MODULE_PARM_DESC(enable_hpi_hwdep, "ALSA enable HPI hwdep for AudioScience soundcard "); /* identify driver */ #ifdef KERNEL_ALSA_BUILD static char *build_info = "Built using headers from kernel source"; -module_param(build_info, charp, S_IRUGO); +module_param(build_info, charp, 0444); MODULE_PARM_DESC(build_info, "Built using headers from kernel source"); #else static char *build_info = "Built within ALSA source"; -module_param(build_info, charp, S_IRUGO); +module_param(build_info, charp, 0444); MODULE_PARM_DESC(build_info, "Built within ALSA source"); #endif diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index b1a2a7ea4172..7d049569012c 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -46,14 +46,14 @@ MODULE_FIRMWARE("asihpi/dsp8900.bin"); #endif static int prealloc_stream_buf; -module_param(prealloc_stream_buf, int, S_IRUGO); +module_param(prealloc_stream_buf, int, 0444); MODULE_PARM_DESC(prealloc_stream_buf, "Preallocate size for per-adapter stream buffer"); /* Allow the debug level to be changed after module load. E.g. echo 2 > /sys/module/asihpi/parameters/hpiDebugLevel */ -module_param(hpi_debug_level, int, S_IRUGO | S_IWUSR); +module_param(hpi_debug_level, int, 0644); MODULE_PARM_DESC(hpi_debug_level, "debug verbosity 0..5"); /* List of adapters found */ diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 9b2b8b38122f..a2c85cc37972 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -431,7 +431,7 @@ int snd_ca0106_proc_init(struct snd_ca0106 *emu) if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32); entry->c.text.write = snd_ca0106_proc_reg_write32; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16); @@ -440,12 +440,12 @@ int snd_ca0106_proc_init(struct snd_ca0106 *emu) if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1); entry->c.text.write = snd_ca0106_proc_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { entry->c.text.write = snd_ca0106_proc_i2c_write; entry->private_data = emu; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2); diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 0020fd0efc46..ed1251c5f449 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2849,7 +2849,7 @@ static int snd_cs46xx_proc_init(struct snd_card *card, struct snd_cs46xx *chip) entry->private_data = chip; entry->c.ops = &snd_cs46xx_proc_io_ops; entry->size = region->size; - entry->mode = S_IFREG | S_IRUSR; + entry->mode = S_IFREG | 0400; } } #ifdef CONFIG_SND_CS46XX_NEW_DSP diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index aa61615288ff..c44eadef64ae 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -798,7 +798,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -814,7 +814,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "spos_symbols", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_symbol_table_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -826,7 +826,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "spos_modules", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_modules_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -838,7 +838,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "parameter", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_parameter_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -850,7 +850,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "sample", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_sample_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -862,7 +862,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "task_tree", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_task_tree_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -874,7 +874,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "scb_info", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_scb_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a770..abb01ce66983 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -271,7 +271,7 @@ void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip, entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = scb_info; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_scb_info_read; diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 08e874e9a7f6..2099e9ce441a 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -17,7 +17,7 @@ static bool use_system_timer; MODULE_PARM_DESC(use_system_timer, "Force to use system-timer"); -module_param(use_system_timer, bool, S_IRUGO); +module_param(use_system_timer, bool, 0444); struct ct_timer_ops { void (*init)(struct ct_timer_instance *); diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index f2f32779de98..b2874220be1b 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -26,9 +26,9 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}"); static unsigned int reference_rate = 48000; static unsigned int multiple = 2; MODULE_PARM_DESC(reference_rate, "Reference rate (default=48000)"); -module_param(reference_rate, uint, S_IRUGO); +module_param(reference_rate, uint, 0444); MODULE_PARM_DESC(multiple, "Rate multiplier (default=2)"); -module_param(multiple, uint, S_IRUGO); +module_param(multiple, uint, 0444); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 2c2b12a06177..611589cbdad6 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1070,7 +1070,7 @@ static int snd_emu10k1x_proc_init(struct emu10k1x *emu) if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read); entry->c.text.write = snd_emu10k1x_proc_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = emu; } diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index bde0d1954f56..dc3d3512154f 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -574,32 +574,32 @@ int snd_emu10k1_proc_init(struct snd_emu10k1 *emu) if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); entry->c.text.write = snd_emu_proc_io_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a); entry->c.text.write = snd_emu_proc_ptr_reg_write00; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b); entry->c.text.write = snd_emu_proc_ptr_reg_write00; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } #endif @@ -621,35 +621,35 @@ int snd_emu10k1_proc_init(struct snd_emu10k1 *emu) if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_GPR : TOTAL_SIZE_GPR; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_tram_data", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_DATA : TOTAL_SIZE_TANKMEM_DATA ; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_tram_addr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_ADDR : TOTAL_SIZE_TANKMEM_ADDR ; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_code", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_CODE : TOTAL_SIZE_CODE; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_acode", &entry)) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->c.text.read = snd_emu10k1_proc_acode_read; } return 0; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7d7eb1354eee..8840daf9c6a3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -510,7 +510,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) snd_info_set_text_ops(entry, per_pin, print_eld_info); entry->c.text.write = write_eld_info; - entry->mode |= S_IWUSR; + entry->mode |= 0200; per_pin->proc_entry = entry; return 0; diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 5101f40f6fbd..93b8cfc6636f 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -662,7 +662,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = wm_proc_regs_write; } } diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 8dabd4d0211d..d7366ade5a25 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -926,7 +926,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) struct snd_info_entry *entry; if (!snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = wm_proc_regs_write; } } diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c index c241dc06dd92..904e3c4f4dfe 100644 --- a/sound/pci/lola/lola_proc.c +++ b/sound/pci/lola/lola_proc.c @@ -214,7 +214,7 @@ void lola_proc_debug_new(struct lola *chip) snd_info_set_text_ops(entry, chip, lola_proc_codec_read); if (!snd_card_proc_new(chip->card, "codec_rw", &entry)) { snd_info_set_text_ops(entry, chip, lola_proc_codec_rw_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = lola_proc_codec_rw_write; } if (!snd_card_proc_new(chip->card, "regs", &entry)) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index f9ae72f28ddc..e57da4036231 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1465,7 +1465,7 @@ static void pcxhr_proc_init(struct snd_pcxhr *chip) !snd_card_proc_new(chip->card, "gpio", &entry)) { snd_info_set_text_ops(entry, chip, pcxhr_proc_gpio_read); entry->c.text.write = pcxhr_proc_gpo_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (!snd_card_proc_new(chip->card, "ltc", &entry)) snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc); diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index feca0a672976..80dc42197154 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1733,10 +1733,10 @@ static ssize_t cs43130_show_ac_r(struct device *dev, return cs43130_show_ac(dev, buf, HP_RIGHT); } -static DEVICE_ATTR(hpload_dc_l, S_IRUGO, cs43130_show_dc_l, NULL); -static DEVICE_ATTR(hpload_dc_r, S_IRUGO, cs43130_show_dc_r, NULL); -static DEVICE_ATTR(hpload_ac_l, S_IRUGO, cs43130_show_ac_l, NULL); -static DEVICE_ATTR(hpload_ac_r, S_IRUGO, cs43130_show_ac_r, NULL); +static DEVICE_ATTR(hpload_dc_l, 0444, cs43130_show_dc_l, NULL); +static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); +static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); +static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); static struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b7b914963c62..2175dccdf388 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -627,22 +627,21 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, if (!root) goto err; - if (!debugfs_create_bool("booted", S_IRUGO, root, &dsp->booted)) + if (!debugfs_create_bool("booted", 0444, root, &dsp->booted)) goto err; - if (!debugfs_create_bool("running", S_IRUGO, root, &dsp->running)) + if (!debugfs_create_bool("running", 0444, root, &dsp->running)) goto err; - if (!debugfs_create_x32("fw_id", S_IRUGO, root, &dsp->fw_id)) + if (!debugfs_create_x32("fw_id", 0444, root, &dsp->fw_id)) goto err; - if (!debugfs_create_x32("fw_version", S_IRUGO, root, - &dsp->fw_id_version)) + if (!debugfs_create_x32("fw_version", 0444, root, &dsp->fw_id_version)) goto err; for (i = 0; i < ARRAY_SIZE(wm_adsp_debugfs_fops); ++i) { if (!debugfs_create_file(wm_adsp_debugfs_fops[i].name, - S_IRUGO, root, dsp, + 0444, root, dsp, &wm_adsp_debugfs_fops[i].fops)) goto err; } diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 1bacfa24ba7f..1255dfe19eef 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -142,7 +142,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) if (!ssi_dbg->dbg_dir) return -ENOMEM; - ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, + ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops); if (!ssi_dbg->dbg_stats) { diff --git a/sound/sound_core.c b/sound/sound_core.c index b4efb22db561..40ad000c2e3c 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -413,7 +413,7 @@ int register_sound_special_device(const struct file_operations *fops, int unit, break; } return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, - name, S_IRUSR | S_IWUSR, dev); + name, 0600, dev); } EXPORT_SYMBOL(register_sound_special_device); @@ -440,7 +440,7 @@ EXPORT_SYMBOL(register_sound_special); int register_sound_mixer(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[0], fops, dev, 0, 128, - "mixer", S_IRUSR | S_IWUSR, NULL); + "mixer", 0600, NULL); } EXPORT_SYMBOL(register_sound_mixer); @@ -468,7 +468,7 @@ EXPORT_SYMBOL(register_sound_mixer); int register_sound_dsp(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[3], fops, dev, 3, 131, - "dsp", S_IWUSR | S_IRUSR, NULL); + "dsp", 0600, NULL); } EXPORT_SYMBOL(register_sound_dsp); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index f0e713527e91..7609eceba1a2 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2518,7 +2518,7 @@ static void snd_dbri_proc(struct snd_card *card) #ifdef DBRI_DEBUG if (!snd_card_proc_new(card, "debug", &entry)) { snd_info_set_text_ops(entry, dbri, dbri_debug_read); - entry->mode = S_IFREG | S_IRUGO; /* Readable only. */ + entry->mode = S_IFREG | 0444; /* Readable only. */ } #endif } -- 2.15.0

3 2

[alsa-devel] [PATCH 1/2] ASoC: omap: sdma-pcm: add module license
by Arnd Bergmann 28 May '18

28 May '18

Kbuild warns that we should have a MODULE_LICENSE() tag in the new module: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o This adds both the license and author field corresponding to the information at the top of the file. Fixes: dde637f2daf1 ("ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/omap/sdma-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/omap/sdma-pcm.c b/sound/soc/omap/sdma-pcm.c index f7b2b5b69d27..e9981bff8bd2 100644 --- a/sound/soc/omap/sdma-pcm.c +++ b/sound/soc/omap/sdma-pcm.c @@ -4,6 +4,7 @@ * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> */ +#include <linux/module.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -66,3 +67,6 @@ int sdma_pcm_platform_register(struct device *dev, return devm_snd_dmaengine_pcm_register(dev, config, flags); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); + +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi(a)ti.com>"); +MODULE_LICENSE("GPL v2"); -- 2.9.0

3 4

[alsa-devel] [PATCH] ALSA: usb-audio: Simplify PCM open/close callbacks
by Takashi Iwai 27 May '18

27 May '18

The stream direction in open and close callbacks can be retrieved from substream->direction, hence we don't have to stick with the unique PCM ops hard-coded for each direction. Rewrite the common open/close callback functions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 34 ++++++++-------------------------- 1 file changed, 8 insertions(+), 26 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 20bed1c7a312..d5b9c30d3bb1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1227,8 +1227,9 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; } -static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) +static int snd_usb_pcm_open(struct snd_pcm_substream *substream) { + int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; @@ -1248,8 +1249,9 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) return setup_hw_info(runtime, subs); } -static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) +static int snd_usb_pcm_close(struct snd_pcm_substream *substream) { + int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; @@ -1611,26 +1613,6 @@ static void retire_playback_urb(struct snd_usb_substream *subs, spin_unlock_irqrestore(&subs->lock, flags); } -static int snd_usb_playback_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_playback_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_capture_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_CAPTURE); -} - -static int snd_usb_capture_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); -} - static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -1692,8 +1674,8 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream } static const struct snd_pcm_ops snd_usb_playback_ops = { - .open = snd_usb_playback_open, - .close = snd_usb_playback_close, + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_usb_hw_params, .hw_free = snd_usb_hw_free, @@ -1705,8 +1687,8 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { }; static const struct snd_pcm_ops snd_usb_capture_ops = { - .open = snd_usb_capture_open, - .close = snd_usb_capture_close, + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_usb_hw_params, .hw_free = snd_usb_hw_free, -- 2.16.3

1 0

[alsa-devel] [PATCH] ALSA: aloop: Reduced duplicated PCM ops definition
by Takashi Iwai 27 May '18

27 May '18

The PCM ops defined for playback and capture are identical. Just use the single one for both. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/aloop.c | 19 +++---------------- 1 file changed, 3 insertions(+), 16 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index eab7f594ebe7..78a2fdc38531 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -768,20 +768,7 @@ static int loopback_close(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops loopback_playback_ops = { - .open = loopback_open, - .close = loopback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = loopback_hw_params, - .hw_free = loopback_hw_free, - .prepare = loopback_prepare, - .trigger = loopback_trigger, - .pointer = loopback_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, -}; - -static const struct snd_pcm_ops loopback_capture_ops = { +static const struct snd_pcm_ops loopback_pcm_ops = { .open = loopback_open, .close = loopback_close, .ioctl = snd_pcm_lib_ioctl, @@ -804,8 +791,8 @@ static int loopback_pcm_new(struct loopback *loopback, substreams, substreams, &pcm); if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_capture_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_pcm_ops); pcm->private_data = loopback; pcm->info_flags = 0; -- 2.16.3

1 0

[alsa-devel] [PATCH] ASoC: AMD: make channel 1 dma as circular
by Akshu Agrawal 27 May '18

27 May '18

channel 1: SYSMEM<->ACP channel 2: ACP<->I2S Instead of waiting on period interrupt of ch 2 and then starting dma on ch1, we make ch1 dma as circular. This removes dependency of period granularity on hw pointer. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 72 +++++++-------------------------------------- 1 file changed, 10 insertions(+), 62 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index ac32dea..f1d8678 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -337,8 +337,7 @@ static void config_acp_dma(void __iomem *acp_mmio, } /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, - u16 ch_num, bool is_circular) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) { u32 dma_ctrl; @@ -369,11 +368,8 @@ static void acp_dma_start(void __iomem *acp_mmio, break; } - /* enable for ACP SRAM to/from I2S DMA channel */ - if (is_circular == true) - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; - else - dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* circular for both DMA channel */ + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -617,7 +613,6 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { - u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -634,33 +629,13 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(ACP_TO_I2S_DMA_CH_NUM)) != 0) { valid_irq = true; - if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_13) == - PLAYBACK_START_DMA_DESCR_CH13) - dscr_idx = PLAYBACK_END_DMA_DESCR_CH12; - else - dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; - config_acp_dma_channel(acp_mmio, SYSRAM_TO_ACP_CH_NUM, dscr_idx, - 1, 0); - acp_dma_start(acp_mmio, SYSRAM_TO_ACP_CH_NUM, false); - snd_pcm_period_elapsed(irq_data->play_i2ssp_stream); - acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; - if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_9) == - PLAYBACK_START_DMA_DESCR_CH9) - dscr_idx = PLAYBACK_END_DMA_DESCR_CH8; - else - dscr_idx = PLAYBACK_START_DMA_DESCR_CH8; - config_acp_dma_channel(acp_mmio, - SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, - dscr_idx, 1, 0); - acp_dma_start(acp_mmio, SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, - false); snd_pcm_period_elapsed(irq_data->play_i2sbt_stream); acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) << 16, @@ -669,38 +644,20 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; - if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_15) == - CAPTURE_START_DMA_DESCR_CH15) - dscr_idx = CAPTURE_END_DMA_DESCR_CH14; - else - dscr_idx = CAPTURE_START_DMA_DESCR_CH14; - config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, - 1, 0); - acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); - + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { valid_irq = true; - snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; - if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_11) == - CAPTURE_START_DMA_DESCR_CH11) - dscr_idx = CAPTURE_END_DMA_DESCR_CH10; - else - dscr_idx = CAPTURE_START_DMA_DESCR_CH10; - config_acp_dma_channel(acp_mmio, - ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, - dscr_idx, 1, 0); - acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, - false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -708,7 +665,6 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; - snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -1015,11 +971,9 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; - u32 loops = 4000; u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *prtd = substream->private_data; struct audio_substream_data *rtd = runtime->private_data; struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); @@ -1034,18 +988,12 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) if (rtd->bytescount == 0) rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_start(rtd->acp_mmio, rtd->ch1, false); - while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & - BIT(rtd->ch1)) { - if (!loops--) { - dev_err(component->dev, - "acp dma start timeout\n"); - return -ETIMEDOUT; - } - cpu_relax(); - } + acp_dma_start(rtd->acp_mmio, rtd->ch1); + acp_dma_start(rtd->acp_mmio, rtd->ch2); + } else { + acp_dma_start(rtd->acp_mmio, rtd->ch2); + acp_dma_start(rtd->acp_mmio, rtd->ch1); } - acp_dma_start(rtd->acp_mmio, rtd->ch2, true); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: -- 1.9.1

3 3

[alsa-devel] [PATCH 0/2] ALSA: dice: optimization for TC Electronic Studio Konnekt 48
by Takashi Sakamoto 27 May '18

27 May '18

Hi, This patchset adds optimization to current ALSA dice driver for Studio Konnekt 48 produced by TC Electronic 2007. My recent work adds hard-coded stream format parameters for all of models produced by TC Electronic with ASICs of Dice series: * Konnekt 24D * Konnect 8 * Studio Konnekt 48 * Konnekt Live * Desktop Konnekt 6 * Impact Twin * Digital Konnekt 32 (Series of PowerCore is not based on Dice.) Except for Konnekt Live, I actually test with these models, and realized that current ALSA dice driver brings several issues, mainly with noises but with detection of packet discontinuities and jumbo payload for some models and transmission frequencies. I guess they comes from timestamping quirks which I reported: [alsa-devel] Dice packet sequence quirk and ALSA firewire stack in Linux 4.6 http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107715.html I have never found any good solution for this issue. Takashi Sakamoto (2): ALSA: dice: fix stream format parameters for TC Electronic Studio Konnekt 48 ALSA: dice: unuse second stream for MIDI conformant data channel for TC Electronic models sound/firewire/dice/dice-tcelectronic.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) -- 2.14.1

2 3

[alsa-devel] [PATCH] ALSA: seq: fix spelling mistake "Unamed" -> "Unnamed"
by Colin King 27 May '18

27 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in string Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/seq/seq_ports.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index d21ece9f8d73..24d90abfc64d 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -669,7 +669,7 @@ int snd_seq_event_port_attach(int client, /* Set up the port */ memset(&portinfo, 0, sizeof(portinfo)); portinfo.addr.client = client; - strlcpy(portinfo.name, portname ? portname : "Unamed port", + strlcpy(portinfo.name, portname ? portname : "Unnamed port", sizeof(portinfo.name)); portinfo.capability = cap; -- 2.17.0

2 1

[alsa-devel] [RFC/RFT PATCH] ASoC: topology: Improve backwards compatibility with v4 topology files
by Guenter Roeck 25 May '18

25 May '18

From: Guenter Roeck <groeck(a)chromium.org> Commit dc31e741db49 ("ASoC: topology: ABI - Add the types for BE DAI") introduced sound topology files version 5. Initially, this change made the topology code incompatible with v4 topology files. Backwards compatibility with v4 configuration files was subsequently added with commit 288b8da7e992 ("ASoC: topology: Support topology file of ABI v4"). Unfortunately, backwards compatibility was never fully implemented. First, the manifest size in (Skylake) v4 configuration files is set to 0, which causes manifest_new_ver() to bail out with error messages similar to the following. snd_soc_skl 0000:00:1f.3: ASoC: invalid manifest size snd_soc_skl 0000:00:1f.3: tplg component load failed-22 snd_soc_skl 0000:00:1f.3: Failed to init topology! snd_soc_skl 0000:00:1f.3: ASoC: failed to probe component -22 skl_n88l25_m98357a skl_n88l25_m98357a: ASoC: failed to instantiate card -22 skl_n88l25_m98357a: probe of skl_n88l25_m98357a failed with error -22 After this problem is fixed, the following error message is seen instead. snd_soc_skl 0000:00:1f.3: ASoC: old version of manifest snd_soc_skl 0000:00:1f.3: Invalid descriptor token 1093938482 snd_soc_skl 0000:00:1f.3: ASoC: failed to load widget media0_in cpr 0 snd_soc_skl 0000:00:1f.3: tPlg component load failed-22 This message is seen because backwards compatibility for loading widgets was never implemented. Both problems have been widely reported. Various attempts were made to fix the problem, usually by providing v5 configuration files. However, all those attempts result in incomplete ASoC configuration. Let's implement backward compatibility properly to solve the problem for good. Signed-off-by: Guenter Roeck <groeck(a)chromium.org> --- Tested on Caroline (Samsung Chromebook Pro) running v4.17-rc6 plus this patch, with original (v4) configuration file. Also tested on several other Chromebooks with this patch on top of chromeos-4.14. Additional real world test coverage would be useful before moving forward. sound/soc/intel/skylake/skl-topology.c | 235 +++++++++++++++++++++++++ sound/soc/soc-topology.c | 7 +- 2 files changed, 240 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 3b1dca419883..cc81e200e6b7 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -19,6 +19,7 @@ #include <linux/slab.h> #include <linux/types.h> #include <linux/firmware.h> +#include <linux/uuid.h> #include <sound/soc.h> #include <sound/soc-topology.h> #include <uapi/sound/snd_sst_tokens.h> @@ -30,6 +31,79 @@ #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +/* v4 configuration data */ +struct skl_dfw_v4_module_pin { + u16 module_id; + u16 instance_id; +} __packed; + +struct skl_dfw_v4_module_fmt { + u32 channels; + u32 freq; + u32 bit_depth; + u32 valid_bit_depth; + u32 ch_cfg; + u32 interleaving_style; + u32 sample_type; + u32 ch_map; +} __packed; + +struct skl_dfw_v4_module_caps { + u32 set_params:2; + u32 rsvd:30; + u32 param_id; + u32 caps_size; + u32 caps[HDA_SST_CFG_MAX]; +}; + +struct skl_dfw_v4_pipe { + u8 pipe_id; + u8 pipe_priority; + u16 conn_type:4; + u16 rsvd:4; + u16 memory_pages:8; +} __packed; + +struct skl_dfw_v4_module { + char uuid[SKL_UUID_STR_SZ]; + + u16 module_id; + u16 instance_id; + u32 max_mcps; + u32 mem_pages; + u32 obs; + u32 ibs; + u32 vbus_id; + + u32 max_in_queue:8; + u32 max_out_queue:8; + u32 time_slot:8; + u32 core_id:4; + u32 rsvd1:4; + + u32 module_type:8; + u32 conn_type:4; + u32 dev_type:4; + u32 hw_conn_type:4; + u32 rsvd2:12; + + u32 params_fixup:8; + u32 converter:8; + u32 input_pin_type:1; + u32 output_pin_type:1; + u32 is_dynamic_in_pin:1; + u32 is_dynamic_out_pin:1; + u32 is_loadable:1; + u32 rsvd3:11; + + struct skl_dfw_v4_pipe pipe; + struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE]; + struct skl_dfw_v4_module_fmt out_fmt[MAX_OUT_QUEUE]; + struct skl_dfw_v4_module_pin in_pin[MAX_IN_QUEUE]; + struct skl_dfw_v4_module_pin out_pin[MAX_OUT_QUEUE]; + struct skl_dfw_v4_module_caps caps; +} __packed; + #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) #define SKL_FMT_FIXUP_MASK (1 << 2) @@ -2724,6 +2798,160 @@ static int skl_tplg_get_desc_blocks(struct device *dev, return -EINVAL; } +/* + * Add pipeline from topology binary into driver pipeline list + * + * If already added we return that instance + * Otherwise we create a new instance and add into driver list + */ +static int skl_tplg_add_pipe_v4(struct device *dev, + struct skl_module_cfg *mconfig, struct skl *skl, + struct skl_dfw_v4_pipe *dfw_pipe) +{ + struct skl_pipeline *ppl; + struct skl_pipe *pipe; + struct skl_pipe_params *params; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == dfw_pipe->pipe_id) { + mconfig->pipe = ppl->pipe; + return 0; + } + } + + ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL); + if (!ppl) + return -ENOMEM; + + pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL); + if (!pipe) + return -ENOMEM; + + params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + + pipe->ppl_id = dfw_pipe->pipe_id; + pipe->memory_pages = dfw_pipe->memory_pages; + pipe->pipe_priority = dfw_pipe->pipe_priority; + pipe->conn_type = dfw_pipe->conn_type; + pipe->state = SKL_PIPE_INVALID; + pipe->p_params = params; + INIT_LIST_HEAD(&pipe->w_list); + + ppl->pipe = pipe; + list_add(&ppl->node, &skl->ppl_list); + + mconfig->pipe = pipe; + mconfig->pipe->state = SKL_PIPE_INVALID; + + return 0; +} + +static void skl_fill_module_pin_info_v4(struct skl_dfw_v4_module_pin *dfw_pin, + struct skl_module_pin *m_pin, + bool is_dynamic, int max_pin) +{ + int i; + + for (i = 0; i < max_pin; i++) { + m_pin[i].id.module_id = dfw_pin[i].module_id; + m_pin[i].id.instance_id = dfw_pin[i].instance_id; + m_pin[i].in_use = false; + m_pin[i].is_dynamic = is_dynamic; + m_pin[i].pin_state = SKL_PIN_UNBIND; + } +} + +static void skl_tplg_fill_fmt_v4(struct skl_module_pin_fmt *dst_fmt, + struct skl_dfw_v4_module_fmt *src_fmt, + int pins) +{ + int i; + + for (i = 0; i < pins; i++) { + dst_fmt[i].fmt.channels = src_fmt[i].channels; + dst_fmt[i].fmt.s_freq = src_fmt[i].freq; + dst_fmt[i].fmt.bit_depth = src_fmt[i].bit_depth; + dst_fmt[i].fmt.valid_bit_depth = src_fmt[i].valid_bit_depth; + dst_fmt[i].fmt.ch_cfg = src_fmt[i].ch_cfg; + dst_fmt[i].fmt.ch_map = src_fmt[i].ch_map; + dst_fmt[i].fmt.interleaving_style = + src_fmt[i].interleaving_style; + dst_fmt[i].fmt.sample_type = src_fmt[i].sample_type; + } +} + +static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w, + struct skl *skl, struct device *dev, + struct skl_module_cfg *mconfig) +{ + struct skl_dfw_v4_module *dfw = + (struct skl_dfw_v4_module *)tplg_w->priv.data; + int ret; + + dev_dbg(dev, "Parsing Skylake v4 widget topology data\n"); + + ret = guid_parse(dfw->uuid, (guid_t *)mconfig->guid); + if (ret) + return ret; + mconfig->id.module_id = -1; + mconfig->id.instance_id = dfw->instance_id; + mconfig->module->resources[0].cps = dfw->max_mcps; + mconfig->module->resources[0].ibs = dfw->ibs; + mconfig->module->resources[0].obs = dfw->obs; + mconfig->core_id = dfw->core_id; + mconfig->module->max_input_pins = dfw->max_in_queue; + mconfig->module->max_output_pins = dfw->max_out_queue; + mconfig->module->loadable = dfw->is_loadable; + skl_tplg_fill_fmt_v4(mconfig->module->formats[0].inputs, dfw->in_fmt, + MAX_IN_QUEUE); + skl_tplg_fill_fmt_v4(mconfig->module->formats[0].outputs, dfw->out_fmt, + MAX_OUT_QUEUE); + + mconfig->params_fixup = dfw->params_fixup; + mconfig->converter = dfw->converter; + mconfig->m_type = dfw->module_type; + mconfig->vbus_id = dfw->vbus_id; + mconfig->module->resources[0].is_pages = dfw->mem_pages; + + ret = skl_tplg_add_pipe_v4(dev, mconfig, skl, &dfw->pipe); + if (ret) + return ret; + + mconfig->dev_type = dfw->dev_type; + mconfig->hw_conn_type = dfw->hw_conn_type; + mconfig->time_slot = dfw->time_slot; + mconfig->formats_config.caps_size = dfw->caps.caps_size; + + mconfig->m_in_pin = devm_kzalloc(dev, + MAX_IN_QUEUE * sizeof(*mconfig->m_in_pin), + GFP_KERNEL); + if (!mconfig->m_in_pin) + return -ENOMEM; + + mconfig->m_out_pin = devm_kzalloc(dev, + MAX_OUT_QUEUE * sizeof(*mconfig->m_out_pin), + GFP_KERNEL); + if (!mconfig->m_out_pin) + return -ENOMEM; + + skl_fill_module_pin_info_v4(dfw->in_pin, mconfig->m_in_pin, + dfw->is_dynamic_in_pin, + mconfig->module->max_input_pins); + skl_fill_module_pin_info_v4(dfw->out_pin, mconfig->m_out_pin, + dfw->is_dynamic_out_pin, + mconfig->module->max_output_pins); + + if (mconfig->formats_config.caps_size) { + dev_warn(dev, "caps size %d not supported, will be ignored\n", + mconfig->formats_config.caps_size); + mconfig->formats_config.caps_size = 0; + } + + return 0; +} + /* * Parse the private data for the token and corresponding value. * The private data can have multiple data blocks. So, a data block @@ -2739,6 +2967,13 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, char *data; int ret; + /* + * v4 configuration files have a valid UUID at the start of + * the widget's private data. + */ + if (uuid_is_valid((char *)tplg_w->priv.data)) + return skl_tplg_get_pvt_data_v4(tplg_w, skl, dev, mconfig); + /* Read the NUM_DATA_BLOCKS descriptor */ array = (struct snd_soc_tplg_vendor_array *)tplg_w->priv.data; ret = skl_tplg_get_desc_blocks(dev, array); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 986b8b2f90fb..d66b2e5ccd67 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2293,8 +2293,11 @@ static int manifest_new_ver(struct soc_tplg *tplg, *manifest = NULL; if (src->size != sizeof(*src_v4)) { - dev_err(tplg->dev, "ASoC: invalid manifest size\n"); - return -EINVAL; + dev_warn(tplg->dev, "ASoC: invalid manifest size %d\n", + src->size); + if (src->size) + return -EINVAL; + src->size = sizeof(*src_v4); } dev_warn(tplg->dev, "ASoC: old version of manifest\n"); -- 2.17.0.441.gb46fe60e1d-goog

6 29

[alsa-devel] [PATCH 1/3] ASoC: mediatek: add sub dai to mtk_base_afe
by KaiChieh Chuang 25 May '18

25 May '18

In MediaTek SoC chip we have multiple DAI, such as I2S, ADDA, PCM, etc. Organize each DAI in to one sub dai, with its dai driver, controls, widgets, routes. add mtk_afe_combine_sub_dai() to combine dai driver from each DAI. add mtk_afe_add_sub_dai_control() to register the control, widget, routes to component. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 78 ++++++++++++++++++++++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 5 ++ sound/soc/mediatek/common/mtk-base-afe.h | 18 +++++ 3 files changed, 101 insertions(+) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 8966ee1..c449188 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -13,6 +13,84 @@ #include "mtk-afe-platform-driver.h" #include "mtk-base-afe.h" +int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) +{ + struct snd_soc_dai_driver *sub_dai_drivers; + size_t num_dai_drivers = 0, dai_idx = 0; + int i; + + if (!afe->sub_dais) { + dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); + return -EINVAL; + } + + /* calcualte total dai driver size */ + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].dai_drivers && + afe->sub_dais[i].num_dai_drivers != 0) + num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + } + + dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); + + /* combine sub_dais */ + afe->num_dai_drivers = num_dai_drivers; + afe->dai_drivers = devm_kcalloc(afe->dev, + num_dai_drivers, + sizeof(struct snd_soc_dai_driver), + GFP_KERNEL); + if (!afe->dai_drivers) + return -ENOMEM; + + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].dai_drivers && + afe->sub_dais[i].num_dai_drivers != 0) { + sub_dai_drivers = afe->sub_dais[i].dai_drivers; + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + sub_dai_drivers, + afe->sub_dais[i].num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += afe->sub_dais[i].num_dai_drivers; + } + } + + return 0; +} + +int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) +{ + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); + int i; + + if (!afe->sub_dais) { + dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); + return -EINVAL; + } + + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].controls) + snd_soc_add_component_controls(component, + afe->sub_dais[i].controls, + afe->sub_dais[i].num_controls); + + if (afe->sub_dais[i].dapm_widgets) + snd_soc_dapm_new_controls(&component->dapm, + afe->sub_dais[i].dapm_widgets, + afe->sub_dais[i].num_dapm_widgets); + + if (afe->sub_dais[i].dapm_routes) + snd_soc_dapm_add_routes(&component->dapm, + afe->sub_dais[i].dapm_routes, + afe->sub_dais[i].num_dapm_routes); + } + + snd_soc_dapm_new_widgets(component->dapm.card); + + return 0; + +} + static snd_pcm_uframes_t mtk_afe_pcm_pointer (struct snd_pcm_substream *substream) { diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.h b/sound/soc/mediatek/common/mtk-afe-platform-driver.h index 1c81d91..0c31fa4 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.h +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.h @@ -12,5 +12,10 @@ #define AFE_PCM_NAME "mtk-afe-pcm" extern const struct snd_soc_component_driver mtk_afe_pcm_platform; +struct mtk_base_afe; +struct snd_soc_component; + +int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe); +int mtk_afe_add_sub_dai_control(struct snd_soc_component *component); #endif diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index c2c5a6c..bcf562f0 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -48,6 +48,7 @@ struct mtk_base_irq_data { struct device; struct mtk_base_afe_memif; struct mtk_base_afe_irq; +struct mtk_base_afe_dai; struct regmap; struct snd_pcm_substream; struct snd_soc_dai; @@ -71,6 +72,11 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; + struct mtk_base_afe_dai *sub_dais; + int num_sub_dais; + struct snd_soc_dai_driver *dai_drivers; + unsigned int num_dai_drivers; + const struct snd_pcm_hardware *mtk_afe_hardware; int (*memif_fs)(struct snd_pcm_substream *substream, unsigned int rate); @@ -94,5 +100,17 @@ struct mtk_base_afe_irq { int irq_occupyed; }; +struct mtk_base_afe_dai { + struct snd_soc_dai_driver *dai_drivers; + unsigned int num_dai_drivers; + + const struct snd_kcontrol_new *controls; + unsigned int num_controls; + const struct snd_soc_dapm_widget *dapm_widgets; + unsigned int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + unsigned int num_dapm_routes; +}; + #endif -- 1.9.1

3 5

[alsa-devel] [PATCH v2 1/3] ASoC: mediatek: add sub dai to mtk_base_afe
by KaiChieh Chuang 25 May '18

25 May '18

In MediaTek SoC chip we have multiple DAI, such as I2S, ADDA, PCM, etc. Organize each DAI in to one sub dai, with its dai driver, controls, widgets, routes. add mtk_afe_combine_sub_dai() to combine dai driver from each DAI. add mtk_afe_add_sub_dai_control() to register the control, widget, routes to component. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 78 ++++++++++++++++++++++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 5 ++ sound/soc/mediatek/common/mtk-base-afe.h | 18 +++++ 3 files changed, 101 insertions(+) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 8966ee1..c449188 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -13,6 +13,84 @@ #include "mtk-afe-platform-driver.h" #include "mtk-base-afe.h" +int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) +{ + struct snd_soc_dai_driver *sub_dai_drivers; + size_t num_dai_drivers = 0, dai_idx = 0; + int i; + + if (!afe->sub_dais) { + dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); + return -EINVAL; + } + + /* calcualte total dai driver size */ + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].dai_drivers && + afe->sub_dais[i].num_dai_drivers != 0) + num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + } + + dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); + + /* combine sub_dais */ + afe->num_dai_drivers = num_dai_drivers; + afe->dai_drivers = devm_kcalloc(afe->dev, + num_dai_drivers, + sizeof(struct snd_soc_dai_driver), + GFP_KERNEL); + if (!afe->dai_drivers) + return -ENOMEM; + + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].dai_drivers && + afe->sub_dais[i].num_dai_drivers != 0) { + sub_dai_drivers = afe->sub_dais[i].dai_drivers; + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + sub_dai_drivers, + afe->sub_dais[i].num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += afe->sub_dais[i].num_dai_drivers; + } + } + + return 0; +} + +int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) +{ + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); + int i; + + if (!afe->sub_dais) { + dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); + return -EINVAL; + } + + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].controls) + snd_soc_add_component_controls(component, + afe->sub_dais[i].controls, + afe->sub_dais[i].num_controls); + + if (afe->sub_dais[i].dapm_widgets) + snd_soc_dapm_new_controls(&component->dapm, + afe->sub_dais[i].dapm_widgets, + afe->sub_dais[i].num_dapm_widgets); + + if (afe->sub_dais[i].dapm_routes) + snd_soc_dapm_add_routes(&component->dapm, + afe->sub_dais[i].dapm_routes, + afe->sub_dais[i].num_dapm_routes); + } + + snd_soc_dapm_new_widgets(component->dapm.card); + + return 0; + +} + static snd_pcm_uframes_t mtk_afe_pcm_pointer (struct snd_pcm_substream *substream) { diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.h b/sound/soc/mediatek/common/mtk-afe-platform-driver.h index 1c81d91..0c31fa4 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.h +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.h @@ -12,5 +12,10 @@ #define AFE_PCM_NAME "mtk-afe-pcm" extern const struct snd_soc_component_driver mtk_afe_pcm_platform; +struct mtk_base_afe; +struct snd_soc_component; + +int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe); +int mtk_afe_add_sub_dai_control(struct snd_soc_component *component); #endif diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index c2c5a6c..bcf562f0 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -48,6 +48,7 @@ struct mtk_base_irq_data { struct device; struct mtk_base_afe_memif; struct mtk_base_afe_irq; +struct mtk_base_afe_dai; struct regmap; struct snd_pcm_substream; struct snd_soc_dai; @@ -71,6 +72,11 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; + struct mtk_base_afe_dai *sub_dais; + int num_sub_dais; + struct snd_soc_dai_driver *dai_drivers; + unsigned int num_dai_drivers; + const struct snd_pcm_hardware *mtk_afe_hardware; int (*memif_fs)(struct snd_pcm_substream *substream, unsigned int rate); @@ -94,5 +100,17 @@ struct mtk_base_afe_irq { int irq_occupyed; }; +struct mtk_base_afe_dai { + struct snd_soc_dai_driver *dai_drivers; + unsigned int num_dai_drivers; + + const struct snd_kcontrol_new *controls; + unsigned int num_controls; + const struct snd_soc_dapm_widget *dapm_widgets; + unsigned int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + unsigned int num_dapm_routes; +}; + #endif -- 1.9.1

2 3

[alsa-devel] [PATCH v3 0/8] slimbus: fixes and some helpers
by Srinivas Kandagatla 25 May '18

25 May '18

Hi Greg, Here is set of patches for slimbus, there are 6 fixes and two helper functions. Most of these issues were found while testing QCOM NGD SLIMBus controller with WCD9335 codec. Changes since v2 (https://lkml.org/lkml/2018/5/17/251) - removed unnecessary changes in slim_prepare_txn patch thanks, Srini Srinivas Kandagatla (8): slimbus: core: rearrange slim_eaddr structure slimbus: core: add need_tid flag to slim_msg_txn slimbus: messaging: pass correct wbuf slimbus: messaging: remove multiple calls to pm_runtime_mark_last_busy slimbus: messaging: initialize completion correctly slimbus: qcom: remove redundant depends in Kconfig slimbus: messaging: add slim_prepare_txn() helper function slimbus: core: add of_slim_device_get() helper drivers/slimbus/Kconfig | 1 - drivers/slimbus/core.c | 39 ++++++++++++++++++++ drivers/slimbus/messaging.c | 89 ++++++++++++++++++++++++++++++--------------- drivers/slimbus/slimbus.h | 4 ++ include/linux/slimbus.h | 14 ++++--- 5 files changed, 110 insertions(+), 37 deletions(-) -- 2.16.2

1 8

[alsa-devel] [PATCH v5 0/3] ASoC: Add Multi CPU DAI support
by Shreyas NC 25 May '18

25 May '18

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v5: - Addressed review comments from Charles in calculating delays, soc_pcm_init_runtime_hw() and other nits - rebased on latest for-next branch Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 6 + sound/soc/soc-core.c | 280 ++++++++++++++++++++--------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 489 ++++++++++++++++++++++++++++++++++----------------- 4 files changed, 574 insertions(+), 272 deletions(-) -- 2.7.4

3 11

[alsa-devel] [PATCH 1/6] ARM: OMAP1: ams-delta: add GPIO lookup tables
by Janusz Krzysztofik 24 May '18

24 May '18

Scope of the change is limited to GPIO pins used by board specific device drivers which will be updated by follow-up patches of the series. Those are some OMAP GPIO (gpio-0-15) and most of Amstrad Delta latch2 GPIO bank pins. Remaining pins of those banks, as well as Amstrad Delta latch1 pins, will be addressed later. Assign a label ("latch2") to the bank, enumerate its pins and put that information, together with OMAP GPIO bank pins, in GPIO lookup tables. Assign lookup tables to devices as soon as those devices are registered and their names can be obtained. A step froward in: - removal of hard-coded GPIO numbers from drivers, - removal of board mach includes from drivers, - switching to dynamically assigned GPIO numbers. Created and compile tested agains linux-4.17-rc3 Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- arch/arm/mach-omap1/board-ams-delta.c | 102 ++++++++++++++++++++++++++++++++++ 1 file changed, 102 insertions(+) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 52e8e53ca154..4b78e73f8bf7 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -12,6 +12,7 @@ * published by the Free Software Foundation. */ #include <linux/gpio/driver.h> +#include <linux/gpio/machine.h> #include <linux/gpio.h> #include <linux/kernel.h> #include <linux/init.h> @@ -202,7 +203,10 @@ static struct resource latch2_resources[] = { }, }; +#define LATCH2_LABEL "latch2" + static struct bgpio_pdata latch2_pdata = { + .label = LATCH2_LABEL, .base = AMS_DELTA_LATCH2_GPIO_BASE, .ngpio = AMS_DELTA_LATCH2_NGPIO, }; @@ -217,6 +221,23 @@ static struct platform_device latch2_gpio_device = { }, }; +#define LATCH2_PIN_LCD_VBLEN 0 +#define LATCH2_PIN_LCD_NDISP 1 +#define LATCH2_PIN_NAND_NCE 2 +#define LATCH2_PIN_NAND_NRE 3 +#define LATCH2_PIN_NAND_NWP 4 +#define LATCH2_PIN_NAND_NWE 5 +#define LATCH2_PIN_NAND_ALE 6 +#define LATCH2_PIN_NAND_CLE 7 +#define LATCH2_PIN_KEYBRD_PWR 8 +#define LATCH2_PIN_KEYBRD_DATAOUT 9 +#define LATCH2_PIN_SCARD_RSTIN 10 +#define LATCH2_PIN_SCARD_CMDVCC 11 +#define LATCH2_PIN_MODEM_NRESET 12 +#define LATCH2_PIN_MODEM_CODEC 13 +#define LATCH2_PIN_HOOKFLASH1 14 +#define LATCH2_PIN_HOOKFLASH2 15 + static const struct gpio latch_gpios[] __initconst = { { .gpio = LATCH1_GPIO_BASE + 6, @@ -323,6 +344,22 @@ static struct platform_device ams_delta_nand_device = { .resource = ams_delta_nand_resources, }; +#define OMAP_GPIO_LABEL "gpio-0-15" + +static struct gpiod_lookup_table ams_delta_nand_gpio_table = { + .table = { + GPIO_LOOKUP(OMAP_GPIO_LABEL, AMS_DELTA_GPIO_PIN_NAND_RB, "rdy", + 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_NAND_NCE, "nce", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_NAND_NRE, "nre", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_NAND_NWP, "nwp", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_NAND_NWE, "nwe", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_NAND_ALE, "ale", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_NAND_CLE, "cle", 0), + { }, + }, +}; + static struct resource ams_delta_kp_resources[] = { [0] = { .start = INT_KEYBOARD, @@ -358,6 +395,14 @@ static struct platform_device ams_delta_lcd_device = { .id = -1, }; +static struct gpiod_lookup_table ams_delta_lcd_gpio_table = { + .table = { + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_LCD_VBLEN, "vblen", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_LCD_NDISP, "ndisp", 0), + { }, + }, +}; + static const struct gpio_led gpio_leds[] __initconst = { { .name = "camera", @@ -449,11 +494,35 @@ static struct platform_device ams_delta_audio_device = { .id = -1, }; +static struct gpiod_lookup_table ams_delta_audio_gpio_table = { + .table = { + GPIO_LOOKUP(OMAP_GPIO_LABEL, AMS_DELTA_GPIO_PIN_HOOK_SWITCH, + "hook_switch", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_MODEM_CODEC, + "modem_codec", 0), + { }, + }, +}; + static struct platform_device cx20442_codec_device = { .name = "cx20442-codec", .id = -1, }; +static struct gpiod_lookup_table ams_delta_serio_gpio_table = { + .table = { + GPIO_LOOKUP(OMAP_GPIO_LABEL, AMS_DELTA_GPIO_PIN_KEYBRD_DATA, + "data", 0), + GPIO_LOOKUP(OMAP_GPIO_LABEL, AMS_DELTA_GPIO_PIN_KEYBRD_CLK, + "clock", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_KEYBRD_PWR, + "power", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_KEYBRD_DATAOUT, + "dataout", 0), + { }, + }, +}; + static struct platform_device *ams_delta_devices[] __initdata = { &latch1_gpio_device, &latch2_gpio_device, @@ -468,6 +537,16 @@ static struct platform_device *late_devices[] __initdata = { &cx20442_codec_device, }; +static struct gpiod_lookup_table *ams_delta_gpio_tables[] __initdata = { + &ams_delta_audio_gpio_table, + &ams_delta_serio_gpio_table, +}; + +static struct gpiod_lookup_table *late_gpio_tables[] __initdata = { + &ams_delta_lcd_gpio_table, + &ams_delta_nand_gpio_table, +}; + static void __init ams_delta_init(void) { /* mux pins for uarts */ @@ -500,6 +579,20 @@ static void __init ams_delta_init(void) gpio_led_register_device(-1, &leds_pdata); platform_add_devices(ams_delta_devices, ARRAY_SIZE(ams_delta_devices)); + /* + * As soon as devices have been registered, assign their dev_names + * to respective GPIO lookup tables before they are added. + */ + ams_delta_audio_gpio_table.dev_id = + dev_name(&ams_delta_audio_device.dev); + /* + * No device name is assigned to GPIO lookup table for serio device + * as long as serio driver is not converted to platform device driver. + */ + + gpiod_add_lookup_tables(ams_delta_gpio_tables, + ARRAY_SIZE(ams_delta_gpio_tables)); + ams_delta_init_fiq(); omap_writew(omap_readw(ARM_RSTCT1) | 0x0004, ARM_RSTCT1); @@ -570,6 +663,15 @@ static int __init late_init(void) platform_add_devices(late_devices, ARRAY_SIZE(late_devices)); + /* + * As soon as devices have been registered, assign their dev_names + * to respective GPIO lookup tables before they are added. + */ + ams_delta_lcd_gpio_table.dev_id = dev_name(&ams_delta_lcd_device.dev); + ams_delta_nand_gpio_table.dev_id = dev_name(&ams_delta_nand_device.dev); + + gpiod_add_lookup_tables(late_gpio_tables, ARRAY_SIZE(late_gpio_tables)); + err = platform_device_register(&modem_nreset_device); if (err) { pr_err("Couldn't register the modem regulator device\n"); -- 2.16.1

6 22

[alsa-devel] [PATCH] [RFC]ASoC: dapm: enable kcontrols for siggen and effect type widgets
by Ranjani Sridharan 24 May '18

24 May '18

This patch proposes to enable kcontrols for siggen and effect type dapm widgets. This will allow triggering the tone generating siggen component and also modifying the run-time attributes such as frequency, amplitude etc. Effect type widgets such as the equalizer could potentially need kcontrols to pass algorithm data from the userspace. This patch also renames the function dapm_new_pga() to dapm_new_kcontrol() so that it does not allude that it applies only to pga widgets. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-dapm.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0c0cb7713eb4..4577880f69cc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -852,6 +852,8 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w, kcname_in_long_name = true; } else { switch (w->id) { + case snd_soc_dapm_siggen: + case snd_soc_dapm_effect: case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_pga: @@ -1010,8 +1012,8 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return 0; } -/* create new dapm volume control */ -static int dapm_new_pga(struct snd_soc_dapm_widget *w) +/* create new dapm kcontrol */ +static int dapm_new_kcontrol(struct snd_soc_dapm_widget *w) { int i, ret; @@ -3046,9 +3048,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_demux: dapm_new_mux(w); break; + case snd_soc_dapm_siggen: + case snd_soc_dapm_effect: case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: - dapm_new_pga(w); + dapm_new_kcontrol(w); break; case snd_soc_dapm_dai_link: dapm_new_dai_link(w); -- 2.17.0

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix runtime PM
by Lukas Wunner 24 May '18

24 May '18

Before commit 3b5b899ca67d ("ALSA: hda: Make use of core codec functions to sync power state"), hda_set_power_state() returned the response to the Get Power State verb, a 32-bit unsigned integer whose expected value is 0x233 after transitioning a codec to D3, and 0x0 after transitioning it to D0. The response value is significant because hda_codec_runtime_suspend() does not clear the codec's bit in the codec_powered bitmask unless the AC_PWRST_CLK_STOP_OK bit (0x200) is set in the response value. That in turn prevents the HDA controller from runtime suspending because azx_runtime_idle() checks that the codec_powered bitmask is zero. Since commit 3b5b899ca67d, hda_set_power_state() only returns 0x0 or 0x1, thereby breaking runtime PM for any HDA controller. That's because an inline function introduced by the commit returns a bool instead of a 32-bit unsigned int. The change was likely erroneous and resulted from copying and pasting snd_hda_check_power_state(), which is immediately preceding the newly introduced inline function. Fix it. Link: https://bugs.freedesktop.org/show_bug.cgi?id=106597 Fixes: 3b5b899ca67d ("ALSA: hda: Make use of core codec functions to sync power state") Cc: Alex Deucher <alexander.deucher(a)amd.com> Cc: Abhijeet Kumar <abhijeet.kumar(a)intel.com> Reported-and-tested-by: Gunnar Krüger <taijian(a)posteo.de> Signed-off-by: Lukas Wunner <lukas(a)wunner.de> --- sound/pci/hda/hda_local.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 321e78baa63c..9bd935216c18 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -622,8 +622,10 @@ snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid, { return snd_hdac_check_power_state(&codec->core, nid, target_state); } -static inline bool snd_hda_sync_power_state(struct hda_codec *codec, - hda_nid_t nid, unsigned int target_state) + +static inline unsigned int snd_hda_sync_power_state(struct hda_codec *codec, + hda_nid_t nid, + unsigned int target_state) { return snd_hdac_sync_power_state(&codec->core, nid, target_state); } -- 2.17.0

3 2

[alsa-devel] Submitting a patch for tinycompress v1.1.5
by Anoush Khazeni 24 May '18

24 May '18

Submitting a patch to add the missing closing bracket for the extern "C" directive in "tinycompress.h" from release v1.1.5 tar file "tinycompress-1.1.5.tar.bz2"

1 0

[alsa-devel] [PATCH 0/3] ALSA: trivial cleanup of old ifdefs
by Takashi Iwai 24 May '18

24 May '18

Hi, this is a series of cleanup patches for removing the old ifndefs that were needed for external alsa-driver builds. No functional changes, just cleanups. Takashi === Takashi Iwai (3): ALSA: memalloc: Drop superfluous ifndef ALSA: usb-audio: Drop superfluous ifndef ALSA: echoaudio: Drop superfluous macro include/sound/memalloc.h | 2 -- sound/pci/echoaudio/echoaudio.c | 2 +- sound/pci/echoaudio/echoaudio.h | 6 ------ sound/usb/helper.h | 4 ---- 4 files changed, 1 insertion(+), 13 deletions(-) -- 2.16.3

1 3

Re: [alsa-devel] [PATCH 05/15] mtd: nand: pxa3xx: remove the dmaengine compat need
by Daniel Mack 23 May '18

23 May '18

Hi Robert, Thanks for this series. On Monday, April 02, 2018 04:26 PM, Robert Jarzmik wrote: > From: Robert Jarzmik <robert.jarzmik(a)renault.com> > > As the pxa architecture switched towards the dmaengine slave map, the > old compatibility mechanism to acquire the dma requestor line number and > priority are not needed anymore. > > This patch simplifies the dma resource acquisition, using the more > generic function dma_request_slave_channel(). > > Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> > --- > drivers/mtd/nand/pxa3xx_nand.c | 10 +--------- This driver was replaced by drivers/mtd/nand/raw/marvell_nand.c recently, so this patch can be dropped. I attached a version for the new driver which you can pick instead. Thanks, Daniel

1 1

[alsa-devel] [PATCH 0/2] slimbus: Add QCOM SLIMBus NGD driver
by Srinivas Kandagatla 23 May '18

23 May '18

This patchset adds support to basic version of Qualcomm NGD SLIMBus controller driver found SoCs from B family. This controller is light-weight SLIMBus controller driver responsible for communicating with slave HW directly over the bus using messaging interface, and communicating with master component residing on ADSP for bandwidth and data-channel management. Tested this patchset on DB820c with WCD9335 codec. I have pushed my working branch to [1] incase someone want to try. Thanks, srini [1] https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimbus-… Srinivas Kandagatla (2): slimbus: ngd: dt-bindings: Add slim ngd dt bindings slimbus: ngd: Add qcom SLIMBus NGD driver .../bindings/slimbus/slim-ngd-qcom-ctrl.txt | 70 ++ drivers/slimbus/Kconfig | 10 + drivers/slimbus/Makefile | 3 + drivers/slimbus/qcom-ngd-ctrl.c | 1298 ++++++++++++++++++++ drivers/slimbus/slimbus.h | 8 + 5 files changed, 1389 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt create mode 100644 drivers/slimbus/qcom-ngd-ctrl.c -- 2.16.2

5 13

[alsa-devel] [BUG] On Allwinner A20 Board Cubietruck distorted SPDIF opical output
by Markus Mitsch 23 May '18

23 May '18

Hello, i have a new problem. if i want to output sound from optical i get distorted sound. At 44.1 khz its like an old vinly, but at 48 khz and higher it is very bad. thanks in advance for your time. mfg

3 7

[alsa-devel] [PATCH v2 1/2] speaker-test: Support S24_3LE sample format
by Julian Scheel 23 May '18

23 May '18

Implement support signed 24 bit samples, packed in 3 bytes. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- speaker-test/speaker-test.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 65ab523..4804bcf 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -283,6 +283,8 @@ static const int supported_formats[] = { SND_PCM_FORMAT_S16_LE, SND_PCM_FORMAT_S16_BE, SND_PCM_FORMAT_FLOAT_LE, + SND_PCM_FORMAT_S24_3LE, + SND_PCM_FORMAT_S24_3BE, SND_PCM_FORMAT_S32_LE, SND_PCM_FORMAT_S32_BE, -1 @@ -325,6 +327,18 @@ static void do_generate(uint8_t *frames, int channel, int count, case SND_PCM_FORMAT_FLOAT_LE: *samp_f++ = res.f; break; + case SND_PCM_FORMAT_S24_3LE: + res.i >>= 8; + *samp8++ = LE_INT(res.i); + *samp8++ = LE_INT(res.i) >> 8; + *samp8++ = LE_INT(res.i) >> 16; + break; + case SND_PCM_FORMAT_S24_3BE: + res.i >>= 8; + *samp8++ = BE_INT(res.i); + *samp8++ = BE_INT(res.i) >> 8; + *samp8++ = BE_INT(res.i) >> 16; + break; case SND_PCM_FORMAT_S32_LE: *samp32++ = LE_INT(res.i); break; -- 2.17.0

2 2

[alsa-devel] [PATCH 1/4] ALSA: hda: Add Intel NUC7i3BNB to the power_save blacklist
by Hans de Goede 23 May '18

23 May '18

Power-saving is causing a humming sound when active on the Intel NUC7i3BNB, add it to the blacklist. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1520902 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9a16af179503..7cf4e1a96cab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2214,6 +2214,8 @@ static struct snd_pci_quirk power_save_blacklist[] = { SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */ SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */ + SND_PCI_QUIRK(0x8086, 0x2068, "Intel NUC7i3BNB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ -- 2.17.0

2 4

[alsa-devel] sound: ALSA PCM Timestamps
by Muni Sekhar 23 May '18

23 May '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, There are several time stamping functions in ALSA: snd_pcm_status_get_trigger_tstamp snd_pcm_status_get_trigger_htstamp snd_pcm_status_get_tstamp snd_pcm_status_get_htstamp snd_pcm_status_get_audio_htstamp snd_pcm_status_get_driver_htstamp I would like to understand what points in time the resulting functions represent and what is the corresponding kernel mode callback for those API’s. I'd be really grateful if someone took the time to help me clearing my doubt. -- Thanks, Sekhar

2 1

[alsa-devel] [PATCH -next] slimbus: qcom: fix potential NULL dereference in qcom_slim_prg_slew()
by Wei Yongjun 23 May '18

23 May '18

platform_get_resource() may fail and return NULL, so we should better check it's return value to avoid a NULL pointer dereference a bit later in the code. This is detected by Coccinelle semantic patch. @@ expression pdev, res, n, t, e, e1, e2; @@ res = platform_get_resource_byname(pdev, t, n); + if (!res) + return -EINVAL; ... when != res == NULL e = devm_ioremap(e1, res->start, e2); Fixes: ad7fcbc308b0 ("slimbus: qcom: Add Qualcomm Slimbus controller driver") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- drivers/slimbus/qcom-ctrl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index bb36a8f..caea3b9 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -478,6 +478,8 @@ static void qcom_slim_prg_slew(struct platform_device *pdev, /* SLEW RATE register for this SLIMbus */ slew_mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "slew"); + if (!slew_mem) + return; ctrl->slew_reg = devm_ioremap(&pdev->dev, slew_mem->start, resource_size(slew_mem)); if (!ctrl->slew_reg)

2 1

[alsa-devel] Applied "ASoC: rockchip: cdn-dp sound output use spdif" to the asoc tree
by Mark Brown 23 May '18

23 May '18

The patch ASoC: rockchip: cdn-dp sound output use spdif has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b18c6c3c7768b7b81c2250bbccd0c3c0aed6b71f Mon Sep 17 00:00:00 2001 From: Lin Huang <hl(a)rock-chips.com> Date: Tue, 22 May 2018 16:53:42 +0800 Subject: [PATCH] ASoC: rockchip: cdn-dp sound output use spdif some monitors care about the parity bit in the sub-frame of I2S, but the cdn-dp always set this bit to "1", so these monitors do not have sound output if use i2s, use spdif can fix this issue. Signed-off-by: Chris Zhong <zyw(a)rock-chips.com> Signed-off-by: Lin Huang <hl(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 46 +-------------------------- 1 file changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 9a10181a0811..f184168f9a41 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -220,45 +220,6 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int rockchip_sound_cdndp_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int mclk, ret; - - /* in bypass mode, the mclk has to be one of the frequencies below */ - switch (params_rate(params)) { - case 8000: - case 16000: - case 24000: - case 32000: - case 48000: - case 64000: - case 96000: - mclk = 12288000; - break; - case 11025: - case 22050: - case 44100: - case 88200: - mclk = 11289600; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret < 0) { - dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); - return ret; - } - - return 0; -} - static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -293,10 +254,6 @@ static const struct snd_soc_ops rockchip_sound_da7219_ops = { .hw_params = rockchip_sound_da7219_hw_params, }; -static const struct snd_soc_ops rockchip_sound_cdndp_ops = { - .hw_params = rockchip_sound_cdndp_hw_params, -}; - static const struct snd_soc_ops rockchip_sound_dmic_ops = { .hw_params = rockchip_sound_dmic_hw_params, }; @@ -323,8 +280,7 @@ static const struct snd_soc_dai_link rockchip_dais[] = { [DAILINK_CDNDP] = { .name = "DP", .stream_name = "DP PCM", - .codec_dai_name = "i2s-hifi", - .ops = &rockchip_sound_cdndp_ops, + .codec_dai_name = "spdif-hifi", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, }, -- 2.17.0

1 0

[alsa-devel] [PATCH] ASoC: Intel: cht_bsw_nau8824: Fix jack_type to include SND_JACK_MICROPHONE
by Hans de Goede 23 May '18

23 May '18

The nau8824 codec can detect whether a headset or plain headphones is inserted (as well as button presses on the headset) as such the jack_type passed to snd_soc_card_jack_new() should include SND_JACK_MICROPHONE. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 072e94b69e57..30c46977d53c 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -120,7 +120,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) * KEY_VOLUMEUP * KEY_VOLUMEDOWN */ - jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + jack_type = SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3; ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); -- 2.17.0

3 2

[alsa-devel] [PATCH v3 0/6] ALSA: xen-front: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 23 May '18

23 May '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Please note: this patch series depends on [3]. This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Corresponding backend, implemented as a user-space application, can be found at [2]. Thank you, Oleksandr Changes since v2: ***************** 1. Kconfig: select SND_PCM instead of depends 2. Changed spinlock to mutex for protecting the rings since all the interrupts are threaded. Make those per event channel, not a single global one. This also addresses Juergen's comment on calling xen_snd_front_alsa_handle_cur_pos. 3. Fixed page leak on error while allocating an event channel 4. Fixed comments to start from a capital letter 5. Performed XEN_PAGE_SIZE != PAGE_SIZE check after xen_domain and xen_has_pv_devices checks 6. Changed driver specific structures names not to mislead with ALSA structs 7. Added comments on ring counters and overflows 8. Added MAINTAINERS entry Changes since v1: ***************** 1. Moved driver from sound/drivers to sound/xen 2. Coding style changes to better meet Linux Kernel 3. Added explicit back and front synchronization In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events 4. Added explicit back and front parameter negotiation In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameters given: request passes desired parameter's intervals/masks and the response to this request returns allowed min/max intervals/masks to be used. [1] https://xenproject.org/ [2] https://github.com/xen-troops/snd_be [3] https://lkml.org/lkml/2018/4/12/522 Oleksandr Andrushchenko (6): ALSA: xen-front: Introduce Xen para-virtualized sound frontend driver ALSA: xen-front: Read sound driver configuration from Xen store ALSA: xen-front: Implement Xen event channel handling ALSA: xen-front: Implement handling of shared buffers ALSA: xen-front: Implement ALSA virtual sound driver MAINTAINERS: Add ALSA: xen-front: maintainer entry MAINTAINERS | 7 + sound/Kconfig | 2 + sound/Makefile | 2 +- sound/xen/Kconfig | 10 + sound/xen/Makefile | 9 + sound/xen/xen_snd_front.c | 397 +++++++++++++++ sound/xen/xen_snd_front.h | 54 ++ sound/xen/xen_snd_front_alsa.c | 821 ++++++++++++++++++++++++++++++ sound/xen/xen_snd_front_alsa.h | 23 + sound/xen/xen_snd_front_cfg.c | 517 +++++++++++++++++++ sound/xen/xen_snd_front_cfg.h | 46 ++ sound/xen/xen_snd_front_evtchnl.c | 496 ++++++++++++++++++ sound/xen/xen_snd_front_evtchnl.h | 95 ++++ sound/xen/xen_snd_front_shbuf.c | 193 +++++++ sound/xen/xen_snd_front_shbuf.h | 36 ++ 15 files changed, 2707 insertions(+), 1 deletion(-) create mode 100644 sound/xen/Kconfig create mode 100644 sound/xen/Makefile create mode 100644 sound/xen/xen_snd_front.c create mode 100644 sound/xen/xen_snd_front.h create mode 100644 sound/xen/xen_snd_front_alsa.c create mode 100644 sound/xen/xen_snd_front_alsa.h create mode 100644 sound/xen/xen_snd_front_cfg.c create mode 100644 sound/xen/xen_snd_front_cfg.h create mode 100644 sound/xen/xen_snd_front_evtchnl.c create mode 100644 sound/xen/xen_snd_front_evtchnl.h create mode 100644 sound/xen/xen_snd_front_shbuf.c create mode 100644 sound/xen/xen_snd_front_shbuf.h -- 2.17.0

4 18

[alsa-devel] [BUG] sound: soc: sunxi: sun4i-codec.c left control in right mixer
by Markus Mitsch 22 May '18

22 May '18

Hi, in linux/sound/sunxi/sun4i-codec.c there is a control for the left channel in the section of routes for the right channel at line number 770. I dont know if this is wanted. And, i dont know if it is related, but to get sound out of my left speaker i have to MUTE the "Power Amplifier Mixer" channel with alsamixer. Regards

2 1

[alsa-devel] [PATCH 31/33] ASoC: max98088: use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/soc/codecs/max98088.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 865f64c..7383f90 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1382,14 +1382,11 @@ static int max98088_set_bias_level(struct snd_soc_component *component, static int max98088_get_channel(struct snd_soc_component *component, const char *name) { - int i; - - for (i = 0; i < ARRAY_SIZE(eq_mode_name); i++) - if (strcmp(name, eq_mode_name[i]) == 0) - return i; + int ret = match_string(eq_mode_name, ARRAY_SIZE(eq_mode_name), name); /* Shouldn't happen */ - dev_err(component->dev, "Bad EQ channel name '%s'\n", name); + if (ret < 0) + dev_err(component->dev, "Bad EQ channel name '%s'\n", name); return -EINVAL; } -- 1.7.12.4

2 3

[alsa-devel] alsa plugins + slibtool
by orbea＠fredslev.dk 22 May '18

22 May '18

Hi, I am experiencing build issues with alsa-plugins when using slibtool instead of libtool. This build issue is because of a bug in alsa-plugins which libtool hides so I have attached a log showing the build error and a small patch to solve it. This patch will require regenerating the autotools files with a command like 'autoreconf -fi' to take effect. If this could be applied in the next alsa-plugins release I would be very grateful. Also note that I have tried to send this patch to the mailing list before and am unsure if it ever made it past moderation? For reference here is a link to the slibtool source. https://git.midipix.org/cgit.cgi/slibtool/ And to their README that explains the project. https://git.midipix.org/cgit.cgi/slibtool/plain/README Thanks, Hunter

2 1

[alsa-devel] Xonar U7
by Steve Banks 22 May '18

22 May '18

Hi, In the USB-Audio.conf file, the Xonar U7 requires the following definition inthe following section USB-Audio.pcm.iec958_device { "Xonar U7" 1 for the spdif to function correctly Thanks Steve Banks

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/conexant - Add fixup for HP Z2 G4 workstation
by Takashi Iwai 22 May '18

22 May '18

HP Z2 G4 requires the same workaround as other HP machines that have no mic-pin detection. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 093d2a9ece85..dad7e9d5e5a6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -964,6 +964,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", 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), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.16.3

1 1

[alsa-devel] [PATCH 1/2] ASoC: pxa-ssp: allow more flexible setup order
by Daniel Mack 22 May '18

22 May '18

The pxa-ssp driver currently assumes that .set_fmt() is called before .set_clkdiv(), .set_pll() etc. Commit a8bd0ee558714 ("ASoC: raumfeld: Use static DAI format setup") broke support for Raumfeld hardware (and possible other PXA based ones) because it effectively changed the order of these calls. Also, as the call to .set_fmt() is now done at probe time, the port clock is not yet enabled. To fix this, strip all hardware register access code from the .set_fmt() callback and memorize the desired value, so we can use it from the .hw_params() callback. Also make the .set_fmt() callback less destructive by reading all registers that it writes to in the beginning and only masking out the bits that it possibly fiddles with. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/pxa-ssp.c | 82 +++++++++++++++++++++++++++++++++++++------------ 1 file changed, 62 insertions(+), 20 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a0189b88f1d2..fbed87d824b8 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -43,7 +43,8 @@ struct ssp_priv { struct ssp_device *ssp; unsigned int sysclk; - int dai_fmt; + unsigned int dai_fmt; + unsigned int configured_dai_fmt; #ifdef CONFIG_PM uint32_t cr0; uint32_t cr1; @@ -432,36 +433,72 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, return 0; } +static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + break; + + default: + return -EINVAL; + } + + /* Settings will be applied in hw_params() */ + priv->dai_fmt = fmt; + + return 0; +} + /* * Set up the SSP DAI format. * The SSP Port must be inactive before calling this function as the * physical interface format is changed. */ -static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) +static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; u32 sscr0, sscr1, sspsp, scfr; /* check if we need to change anything at all */ - if (priv->dai_fmt == fmt) + if (priv->configured_dai_fmt == priv->dai_fmt) return 0; - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { - dev_err(&ssp->pdev->dev, - "can't change hardware dai format: stream is in use"); - return -EINVAL; - } - /* reset port settings */ sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); - sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); - sspsp = 0; + ~(SSCR0_PSP | SSCR0_MOD); + sscr1 = pxa_ssp_read_reg(ssp, SSCR1) & + ~(SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR | + SSCR1_RWOT | SSCR1_TRAIL | SSCR1_TFT | SSCR1_RFT); + sspsp = pxa_ssp_read_reg(ssp, SSPSP) & + ~(SSPSP_SFRMP | SSPSP_SCMODE(3)); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + sscr1 |= SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + + switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR; break; @@ -474,7 +511,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (priv->dai_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: sspsp |= SSPSP_SFRMP; break; @@ -490,7 +527,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; @@ -512,7 +549,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, pxa_ssp_write_reg(ssp, SSCR1, sscr1); pxa_ssp_write_reg(ssp, SSPSP, sspsp); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: case SND_SOC_DAIFMT_CBM_CFS: scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR; @@ -529,7 +566,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, * we have to defer some things until hw_params() where we * know parameters like the sample size. */ - priv->dai_fmt = fmt; + priv->configured_dai_fmt = priv->dai_fmt; return 0; } @@ -550,6 +587,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; struct snd_dmaengine_dai_dma_data *dma_data; + int ret; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -565,6 +603,10 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; + ret = pxa_ssp_configure_dai_fmt(priv); + if (ret < 0) + return ret; + /* clear selected SSP bits */ sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); -- 2.14.3

2 3

[alsa-devel] [PATCH 1/2] ASoC: wm8782: add device-tree matching table
by Daniel Mack 22 May '18

22 May '18

This is needed when the codec is instanciated from from a device tree. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/codecs/wm8782.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index 7949703f3d04..317db9a149a7 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -67,9 +67,18 @@ static int wm8782_probe(struct platform_device *pdev) &soc_component_dev_wm8782, &wm8782_dai, 1); } +#ifdef CONFIG_OF +static const struct of_device_id wm8782_of_match[] = { + { .compatible = "wlf,wm8782", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8782_of_match); +#endif + static struct platform_driver wm8782_codec_driver = { .driver = { .name = "wm8782", + .of_match_table = of_match_ptr(wm8782_of_match), }, .probe = wm8782_probe, }; -- 2.14.3

2 3

[alsa-devel] [PATCH] ASoC: codec: wolfson: Make the node name generic
by Fabio Estevam 22 May '18

22 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> According to Devicetree Specification v0.2 document: "The name of a node should be somewhat generic, reflecting the function of the device and not its precise programming model." Do as suggested in the bindings examples. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/wm8510.txt | 2 +- Documentation/devicetree/bindings/sound/wm8523.txt | 2 +- Documentation/devicetree/bindings/sound/wm8524.txt | 2 +- Documentation/devicetree/bindings/sound/wm8580.txt | 2 +- Documentation/devicetree/bindings/sound/wm8711.txt | 2 +- Documentation/devicetree/bindings/sound/wm8728.txt | 2 +- Documentation/devicetree/bindings/sound/wm8731.txt | 2 +- Documentation/devicetree/bindings/sound/wm8737.txt | 2 +- Documentation/devicetree/bindings/sound/wm8741.txt | 2 +- Documentation/devicetree/bindings/sound/wm8750.txt | 2 +- Documentation/devicetree/bindings/sound/wm8753.txt | 2 +- Documentation/devicetree/bindings/sound/wm8770.txt | 2 +- Documentation/devicetree/bindings/sound/wm8776.txt | 2 +- Documentation/devicetree/bindings/sound/wm8804.txt | 2 +- Documentation/devicetree/bindings/sound/wm8903.txt | 2 +- Documentation/devicetree/bindings/sound/wm8994.txt | 2 +- 16 files changed, 16 insertions(+), 16 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8510.txt b/Documentation/devicetree/bindings/sound/wm8510.txt index fa1a32b..e6b6cc0 100644 --- a/Documentation/devicetree/bindings/sound/wm8510.txt +++ b/Documentation/devicetree/bindings/sound/wm8510.txt @@ -12,7 +12,7 @@ Required properties: Example: -codec: wm8510@1a { +wm8510: codec@1a { compatible = "wlf,wm8510"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8523.txt b/Documentation/devicetree/bindings/sound/wm8523.txt index 0474618..f3a6485 100644 --- a/Documentation/devicetree/bindings/sound/wm8523.txt +++ b/Documentation/devicetree/bindings/sound/wm8523.txt @@ -10,7 +10,7 @@ Required properties: Example: -codec: wm8523@1a { +wm8523: codec@1a { compatible = "wlf,wm8523"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt index 0f05535..f6c0c26 100644 --- a/Documentation/devicetree/bindings/sound/wm8524.txt +++ b/Documentation/devicetree/bindings/sound/wm8524.txt @@ -10,7 +10,7 @@ Required properties: Example: -codec: wm8524 { +wm8524: codec { compatible = "wlf,wm8524"; wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8580.txt b/Documentation/devicetree/bindings/sound/wm8580.txt index 78fce9b..ff3f9f5 100644 --- a/Documentation/devicetree/bindings/sound/wm8580.txt +++ b/Documentation/devicetree/bindings/sound/wm8580.txt @@ -10,7 +10,7 @@ Required properties: Example: -codec: wm8580@1a { +wm8580: codec@1a { compatible = "wlf,wm8580"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8711.txt b/Documentation/devicetree/bindings/sound/wm8711.txt index 8ed9998..c30a138 100644 --- a/Documentation/devicetree/bindings/sound/wm8711.txt +++ b/Documentation/devicetree/bindings/sound/wm8711.txt @@ -12,7 +12,7 @@ Required properties: Example: -codec: wm8711@1a { +wm8711: codec@1a { compatible = "wlf,wm8711"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8728.txt b/Documentation/devicetree/bindings/sound/wm8728.txt index a8b5c36..a3608b4 100644 --- a/Documentation/devicetree/bindings/sound/wm8728.txt +++ b/Documentation/devicetree/bindings/sound/wm8728.txt @@ -12,7 +12,7 @@ Required properties: Example: -codec: wm8728@1a { +wm8728: codec@1a { compatible = "wlf,wm8728"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt index 236690e..f660d9b 100644 --- a/Documentation/devicetree/bindings/sound/wm8731.txt +++ b/Documentation/devicetree/bindings/sound/wm8731.txt @@ -12,7 +12,7 @@ Required properties: Example: -codec: wm8731@1a { +wm8731: codec@1a { compatible = "wlf,wm8731"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8737.txt b/Documentation/devicetree/bindings/sound/wm8737.txt index 4bc2cea..eda1ec6 100644 --- a/Documentation/devicetree/bindings/sound/wm8737.txt +++ b/Documentation/devicetree/bindings/sound/wm8737.txt @@ -12,7 +12,7 @@ Required properties: Example: -codec: wm8737@1a { +wm8737: codec@1a { compatible = "wlf,wm8737"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt index a133154..b69e196 100644 --- a/Documentation/devicetree/bindings/sound/wm8741.txt +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -21,7 +21,7 @@ Optional properties: Example: -codec: wm8741@1a { +wm8741: codec@1a { compatible = "wlf,wm8741"; reg = <0x1a>; diff --git a/Documentation/devicetree/bindings/sound/wm8750.txt b/Documentation/devicetree/bindings/sound/wm8750.txt index 8db239f..682f221 100644 --- a/Documentation/devicetree/bindings/sound/wm8750.txt +++ b/Documentation/devicetree/bindings/sound/wm8750.txt @@ -12,7 +12,7 @@ Required properties: Example: -codec: wm8750@1a { +wm8750: codec@1a { compatible = "wlf,wm8750"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8753.txt b/Documentation/devicetree/bindings/sound/wm8753.txt index 8eee612..eca9e5a 100644 --- a/Documentation/devicetree/bindings/sound/wm8753.txt +++ b/Documentation/devicetree/bindings/sound/wm8753.txt @@ -34,7 +34,7 @@ Pins on the device (for linking into audio routes): Example: -codec: wm8753@1a { +wm8753: codec@1a { compatible = "wlf,wm8753"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8770.txt b/Documentation/devicetree/bindings/sound/wm8770.txt index 866e00c..cac762a 100644 --- a/Documentation/devicetree/bindings/sound/wm8770.txt +++ b/Documentation/devicetree/bindings/sound/wm8770.txt @@ -10,7 +10,7 @@ Required properties: Example: -codec: wm8770@1 { +wm8770: codec@1 { compatible = "wlf,wm8770"; reg = <1>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8776.txt b/Documentation/devicetree/bindings/sound/wm8776.txt index 3b9ca49..0117336 100644 --- a/Documentation/devicetree/bindings/sound/wm8776.txt +++ b/Documentation/devicetree/bindings/sound/wm8776.txt @@ -12,7 +12,7 @@ Required properties: Example: -codec: wm8776@1a { +wm8776: codec@1a { compatible = "wlf,wm8776"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8804.txt b/Documentation/devicetree/bindings/sound/wm8804.txt index 6fd124b..2c1641c 100644 --- a/Documentation/devicetree/bindings/sound/wm8804.txt +++ b/Documentation/devicetree/bindings/sound/wm8804.txt @@ -19,7 +19,7 @@ Optional properties: Example: -codec: wm8804@1a { +wm8804: codec@1a { compatible = "wlf,wm8804"; reg = <0x1a>; }; diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt index afc51ca..6371c24 100644 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -57,7 +57,7 @@ Pins on the device (for linking into audio routes): Example: -codec: wm8903@1a { +wm8903: codec@1a { compatible = "wlf,wm8903"; reg = <0x1a>; interrupts = < 347 >; diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt index 68c4e8d..4a9dead 100644 --- a/Documentation/devicetree/bindings/sound/wm8994.txt +++ b/Documentation/devicetree/bindings/sound/wm8994.txt @@ -59,7 +59,7 @@ Optional properties: Example: -codec: wm8994@1a { +wm8994: codec@1a { compatible = "wlf,wm8994"; reg = <0x1a>; -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: codec: realtek: Make the node name generic
by Fabio Estevam 22 May '18

22 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> "The name of a node should be somewhat generic, reflecting the function of the device and not its precise programming model." Do as suggested in the bindings examples. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/rt274.txt | 2 +- Documentation/devicetree/bindings/sound/rt5514.txt | 2 +- Documentation/devicetree/bindings/sound/rt5616.txt | 2 +- Documentation/devicetree/bindings/sound/rt5645.txt | 2 +- Documentation/devicetree/bindings/sound/rt5651.txt | 2 +- Documentation/devicetree/bindings/sound/rt5663.txt | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt index e9a6178..791a1bd 100644 --- a/Documentation/devicetree/bindings/sound/rt274.txt +++ b/Documentation/devicetree/bindings/sound/rt274.txt @@ -26,7 +26,7 @@ Pins on the device (for linking into audio routes) for RT274: Example: -codec: rt274@1c { +rt274: codec@1c { compatible = "realtek,rt274"; reg = <0x1c>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 4f33b0d..b25ed08 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -32,7 +32,7 @@ Pins on the device (for linking into audio routes) for I2C: Example: -codec: rt5514@57 { +rt5514: codec@57 { compatible = "realtek,rt5514"; reg = <0x57>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt index e410858..540a4bf 100644 --- a/Documentation/devicetree/bindings/sound/rt5616.txt +++ b/Documentation/devicetree/bindings/sound/rt5616.txt @@ -26,7 +26,7 @@ Pins on the device (for linking into audio routes) for RT5616: Example: -codec: rt5616@1b { +rt5616: codec@1b { compatible = "realtek,rt5616"; reg = <0x1b>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5645.txt b/Documentation/devicetree/bindings/sound/rt5645.txt index 7cee1f51..a03f9a8 100644 --- a/Documentation/devicetree/bindings/sound/rt5645.txt +++ b/Documentation/devicetree/bindings/sound/rt5645.txt @@ -69,4 +69,4 @@ codec: rt5650@1a { realtek,dmic-en = "true"; realtek,en-jd-func = "true"; realtek,jd-mode = <3>; -}; \ No newline at end of file +}; diff --git a/Documentation/devicetree/bindings/sound/rt5651.txt b/Documentation/devicetree/bindings/sound/rt5651.txt index b852218..a41199a 100644 --- a/Documentation/devicetree/bindings/sound/rt5651.txt +++ b/Documentation/devicetree/bindings/sound/rt5651.txt @@ -50,7 +50,7 @@ Pins on the device (for linking into audio routes) for RT5651: Example: -codec: rt5651@1a { +rt5651: codec@1a { compatible = "realtek,rt5651"; reg = <0x1a>; realtek,dmic-en = "true"; diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 497bcfc..2338644 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -47,7 +47,7 @@ Pins on the device (for linking into audio routes) for RT5663: Example: -codec: rt5663@12 { +rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; -- 2.7.4

2 1

[alsa-devel] [PATCH 29/33] ALSA: oxfw: use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/firewire/oxfw/oxfw.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 413ab63..1e5b2c8 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -49,7 +49,6 @@ static bool detect_loud_models(struct fw_unit *unit) "Tapco LINK.firewire 4x6", "U.420"}; char model[32]; - unsigned int i; int err; err = fw_csr_string(unit->directory, CSR_MODEL, @@ -57,12 +56,7 @@ static bool detect_loud_models(struct fw_unit *unit) if (err < 0) return false; - for (i = 0; i < ARRAY_SIZE(models); i++) { - if (strcmp(models[i], model) == 0) - break; - } - - return (i < ARRAY_SIZE(models)); + return match_string(models, ARRAY_SIZE(models), model) >= 0; } static int name_card(struct snd_oxfw *oxfw) -- 1.7.12.4

3 2

[alsa-devel] [asoc:for-4.18 50/211] sound//soc/soc-topology.c:1776:27: error: 'snd_soc_new_compress' undeclared
by kbuild test robot 22 May '18

22 May '18

Hi Liam, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: a6f933f63f2ffdb211dbf203dd9750449af000d3 commit: 5db6aab6f36f7560dc95f7ca340d5632b7a3be6a [50/211] ASoC: topology: Add support for compressed PCMs config: x86_64-randconfig-s2-05221211 (attached as .config) compiler: gcc-6 (Debian 6.4.0-9) 6.4.0 20171026 reproduce: git checkout 5db6aab6f36f7560dc95f7ca340d5632b7a3be6a # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound//soc/soc-topology.c: In function 'soc_tplg_dai_create': >> sound//soc/soc-topology.c:1776:27: error: 'snd_soc_new_compress' undeclared (first use in this function) dai_drv->compress_new = snd_soc_new_compress; ^~~~~~~~~~~~~~~~~~~~ sound//soc/soc-topology.c:1776:27: note: each undeclared identifier is reported only once for each function it appears in vim +/snd_soc_new_compress +1776 sound//soc/soc-topology.c 1746 1747 static int soc_tplg_dai_create(struct soc_tplg *tplg, 1748 struct snd_soc_tplg_pcm *pcm) 1749 { 1750 struct snd_soc_dai_driver *dai_drv; 1751 struct snd_soc_pcm_stream *stream; 1752 struct snd_soc_tplg_stream_caps *caps; 1753 int ret; 1754 1755 dai_drv = kzalloc(sizeof(struct snd_soc_dai_driver), GFP_KERNEL); 1756 if (dai_drv == NULL) 1757 return -ENOMEM; 1758 1759 if (strlen(pcm->dai_name)) 1760 dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); 1761 dai_drv->id = pcm->dai_id; 1762 1763 if (pcm->playback) { 1764 stream = &dai_drv->playback; 1765 caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; 1766 set_stream_info(stream, caps); 1767 } 1768 1769 if (pcm->capture) { 1770 stream = &dai_drv->capture; 1771 caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; 1772 set_stream_info(stream, caps); 1773 } 1774 1775 if (pcm->compress) > 1776 dai_drv->compress_new = snd_soc_new_compress; 1777 1778 /* pass control to component driver for optional further init */ 1779 ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); 1780 if (ret < 0) { 1781 dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); 1782 kfree(dai_drv); 1783 return ret; 1784 } 1785 1786 dai_drv->dobj.index = tplg->index; 1787 dai_drv->dobj.ops = tplg->ops; 1788 dai_drv->dobj.type = SND_SOC_DOBJ_PCM; 1789 list_add(&dai_drv->dobj.list, &tplg->comp->dobj_list); 1790 1791 /* register the DAI to the component */ 1792 return snd_soc_register_dai(tplg->comp, dai_drv); 1793 } 1794 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 28/33] ALSA: dice use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/firewire/dice/dice.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 96bb01b..0639074 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -35,19 +35,13 @@ static bool force_two_pcm_support(struct fw_unit *unit) "SAFFIRE_PRO_40_1", }; char model[32]; - unsigned int i; int err; err = fw_csr_string(unit->directory, CSR_MODEL, model, sizeof(model)); if (err < 0) return false; - for (i = 0; i < ARRAY_SIZE(models); i++) { - if (strcmp(models[i], model) == 0) - break; - } - - return i < ARRAY_SIZE(models); + return match_string(models, ARRAY_SIZE(models), model) >= 0; } static int check_dice_category(struct fw_unit *unit) -- 1.7.12.4

2 1

[alsa-devel] [PATCH 32/33] ASoC: max98095: use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/soc/codecs/max98095.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 6bf2d0b..0ef42dc 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1634,15 +1634,12 @@ static void max98095_handle_eq_pdata(struct snd_soc_component *component) static int max98095_get_bq_channel(struct snd_soc_component *component, const char *name) { - int i; - - for (i = 0; i < ARRAY_SIZE(bq_mode_name); i++) - if (strcmp(name, bq_mode_name[i]) == 0) - return i; + int ret = match_string(bq_mode_name, ARRAY_SIZE(bq_mode_name), name); /* Shouldn't happen */ - dev_err(component->dev, "Bad biquad channel name '%s'\n", name); - return -EINVAL; + if (ret < 0) + dev_err(component->dev, "Bad biquad channel name '%s'\n", name); + return ret; } static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, -- 1.7.12.4

2 1

[alsa-devel] [PATCH 30/33] ALSA: oxygen: use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/pci/oxygen/oxygen_mixer.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 4ca1266..fbd8f5e 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -1052,10 +1052,9 @@ static int add_controls(struct oxygen *chip, [CONTROL_CD_CAPTURE_SWITCH] = "CD Capture Switch", [CONTROL_AUX_CAPTURE_SWITCH] = "Aux Capture Switch", }; - unsigned int i, j; struct snd_kcontrol_new template; struct snd_kcontrol *ctl; - int err; + int i, j, err; for (i = 0; i < count; ++i) { template = controls[i]; @@ -1086,11 +1085,11 @@ static int add_controls(struct oxygen *chip, err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; - for (j = 0; j < CONTROL_COUNT; ++j) - if (!strcmp(ctl->id.name, known_ctl_names[j])) { - chip->controls[j] = ctl; - ctl->private_free = oxygen_any_ctl_free; - } + j = match_string(known_ctl_names, CONTROL_COUNT, ctl->id.name); + if (j >= 0) { + chip->controls[j] = ctl; + ctl->private_free = oxygen_any_ctl_free; + } } return 0; } -- 1.7.12.4

2 1

[alsa-devel] [PATCH] ASoC: intel: skylake: fix spelling mistake: "Homogenous" -> "Homogeneous"
by Colin King 21 May '18

21 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in snprintf literal string Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/skl-debug.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index a016455a6ddb..302b40e63e6c 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -142,8 +142,8 @@ static ssize_t module_read(struct file *file, char __user *user_buf, mconfig->max_out_queue, ret, false); ret += snprintf(buf + ret, MOD_BUF - ret, - "Other:\n\tDomain %d\n\tHomogenous Input %s\n\t" - "Homogenous Output %s\n\tIn Queue Mask %d\n\t" + "Other:\n\tDomain %d\n\tHomogeneous Input %s\n\t" + "Homogeneous Output %s\n\tIn Queue Mask %d\n\t" "Out Queue Mask %d\n\tDMA ID %d\n\tMem Pages %d\n\t" "Module Type %d\n\tModule State %d\n", mconfig->domain, -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: fsl: Mark 'big-endian' property as optional
by Fabio Estevam 21 May '18

21 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Currently the 'big-endian' property is listed as required, which is not correct. i.MX SoCs do not need such property, so move it under 'Optional properties' entry instead. Also, fsl-sai.txt incorrectly referenced 'FTM_PWM registers', so change it to 'SAI registers', which is the intended description. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,asrc.txt | 10 ++++++---- Documentation/devicetree/bindings/sound/fsl,esai.txt | 2 ++ Documentation/devicetree/bindings/sound/fsl,spdif.txt | 2 ++ Documentation/devicetree/bindings/sound/fsl-sai.txt | 8 +++++--- 4 files changed, 15 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index f5a1411..1d4d9f9 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -31,14 +31,16 @@ Required properties: it. This property is optional depending on the SoC design. - - big-endian : If this property is absent, the little endian mode - will be in use as default. Otherwise, the big endian - mode will be in use for all the device registers. - - fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends. - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. +Optional properties: + + - big-endian : If this property is absent, the little endian mode + will be in use as default. Otherwise, the big endian + mode will be in use for all the device registers. + Example: asrc: asrc@2034000 { diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt index cacd18b..5b99143 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -42,6 +42,8 @@ Required properties: means all the settings for Receiving would be duplicated from Transmition related registers. +Optional properties: + - big-endian : If this property is absent, the native endian mode will be in use as default, or the big endian mode will be in use for all the device registers. diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index 38cfa75..8b324f8 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -33,6 +33,8 @@ Required properties: it. This property is optional depending on the SoC design. +Optional properties: + - big-endian : If this property is absent, the native endian mode will be in use as default, or the big endian mode will be in use for all the device registers. diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 740b467..dd9e597 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -28,9 +28,6 @@ Required properties: pinctrl-names. See ../pinctrl/pinctrl-bindings.txt for details of the property values. - - big-endian : Boolean property, required if all the FTM_PWM - registers are big-endian rather than little-endian. - - lsb-first : Configures whether the LSB or the MSB is transmitted first for the fifo data. If this property is absent, the MSB is transmitted first as default, or the LSB @@ -48,6 +45,11 @@ Required properties: receive data by following their own bit clocks and frame sync clocks separately. +Optional properties: + + - big-endian : Boolean property, required if all the SAI + registers are big-endian rather than little-endian. + Optional properties (for mx6ul): - fsl,sai-mclk-direction-output: This is a boolean property. If present, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: core: fix return code in error message
by Daniel Mack 21 May '18

21 May '18

Log the correct error code in case the .open() call to a component fails. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 68d9dc930096..abc1dc7074f5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -517,7 +517,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (__ret < 0) { dev_err(component->dev, "ASoC: can't open component %s: %d\n", - component->name, ret); + component->name, __ret); ret = __ret; } } -- 2.14.3

2 1

[alsa-devel] [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio
by Srinivas Kandagatla 21 May '18

21 May '18

Thankyou everyone for providing feedback and testing v8 patchset. Changes since v8 (https://lkml.org/lkml/2018/5/9/491) - Fixed usecase across afe, adm, asm where user can trigger manual unbind if dsp does not respond/responds very lately. Spotted by Mark B. - rebased patchset on top of(removed patches arleady applied) https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/?h=for-ne… This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is available in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.17-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Thanks, srini Srinivas Kandagatla (15): ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver .../devicetree/bindings/sound/qcom,apq8096.txt | 109 ++ sound/soc/qcom/Kconfig | 33 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/apq8096.c | 255 ++++ sound/soc/qcom/qdsp6/Makefile | 6 + sound/soc/qcom/qdsp6/q6adm.c | 646 +++++++++ sound/soc/qcom/qdsp6/q6adm.h | 27 + sound/soc/qcom/qdsp6/q6afe-dai.c | 748 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1068 +++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 624 +++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1399 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 + sound/soc/qcom/qdsp6/q6routing.c | 990 ++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 15 files changed, 6178 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.16.2

2 24

[alsa-devel] [PATCH V2 01/10] ASoC: amd: dma config parameters changes
by Vijendar Mukunda 21 May '18

21 May '18

Added dma configuration parameters to rtd structure. Moved dma configuration parameters initialization to hw_params callback. Removed hard coding in prepare and trigger callbacks. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1->v2 : Fixed capture stream wrong channel assignment added comments in dma trigger api sound/soc/amd/acp-pcm-dma.c | 103 ++++++++++++++++++-------------------------- sound/soc/amd/acp.h | 5 +++ 2 files changed, 48 insertions(+), 60 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9c026c4..e9736fe 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -321,19 +321,12 @@ static void config_acp_dma(void __iomem *acp_mmio, u32 asic_type) { u32 pte_offset, sram_bank; - u16 ch1, ch2, destination, dma_dscr_idx; if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { pte_offset = ACP_PLAYBACK_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; sram_bank = ACP_SHARED_RAM_BANK_1_ADDRESS; - destination = TO_ACP_I2S_1; - } else { pte_offset = ACP_CAPTURE_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; switch (asic_type) { case CHIP_STONEY: sram_bank = ACP_SHARED_RAM_BANK_3_ADDRESS; @@ -341,30 +334,19 @@ static void config_acp_dma(void __iomem *acp_mmio, default: sram_bank = ACP_SHARED_RAM_BANK_5_ADDRESS; } - destination = FROM_ACP_I2S_1; } - acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, pte_offset); - if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH14; - /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, - rtd->direction, pte_offset, ch1, - sram_bank, dma_dscr_idx, asic_type); - - if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH13; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH15; + rtd->direction, pte_offset, + rtd->ch1, sram_bank, + rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, rtd->direction, sram_bank, - destination, ch2, dma_dscr_idx, - asic_type); + rtd->destination, rtd->ch2, + rtd->dma_dscr_idx_2, asic_type); } /* Start a given DMA channel transfer */ @@ -804,6 +786,21 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->ch1 = SYSRAM_TO_ACP_CH_NUM; + rtd->ch2 = ACP_TO_I2S_DMA_CH_NUM; + rtd->destination = TO_ACP_I2S_1; + rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; + rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; + } else { + rtd->ch1 = ACP_TO_SYSRAM_CH_NUM; + rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM; + rtd->destination = FROM_ACP_I2S_1; + rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; + rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; + } + size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) @@ -898,21 +895,15 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH12, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH13, - NUM_DSCRS_PER_CHANNEL, 0); - } else { - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_SYSRAM_CH_NUM, - CAPTURE_START_DMA_DESCR_CH14, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM, - CAPTURE_START_DMA_DESCR_CH15, - NUM_DSCRS_PER_CHANNEL, 0); - } + + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch1, + rtd->dma_dscr_idx_1, + NUM_DSCRS_PER_CHANNEL, 0); + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch2, + rtd->dma_dscr_idx_2, + NUM_DSCRS_PER_CHANNEL, 0); return 0; } @@ -939,10 +930,9 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (rtd->i2ssp_renderbytescount == 0) rtd->i2ssp_renderbytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM, false); + acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & - BIT(SYSRAM_TO_ACP_CH_NUM)) { + BIT(rtd->ch1)) { if (!loops--) { dev_err(component->dev, "acp dma start timeout\n"); @@ -950,38 +940,31 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) } cpu_relax(); } - - acp_dma_start(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM, true); - } else { if (rtd->i2ssp_capturebytescount == 0) rtd->i2ssp_capturebytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM, true); } + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* - * Need to stop only circular DMA channels : - * ACP_TO_I2S_DMA_CH_NUM / I2S_TO_ACP_DMA_CH_NUM. Non-circular - * channels will stopped automatically after its transfer - * completes : SYSRAM_TO_ACP_CH_NUM / ACP_TO_SYSRAM_CH_NUM + /* For playback, non circular dma should be stopped first + * i.e Sysram to acp dma transfer channel(rtd->ch1) should be + * stopped before stopping cirular dma which is acp sram to i2s + * fifo dma transfer channel(rtd->ch2). Where as in Capture + * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped + * first before stopping acp sram to sysram which is circular + * dma(rtd->ch1). */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = acp_dma_stop(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM); + acp_dma_stop(rtd->acp_mmio, rtd->ch1); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); rtd->i2ssp_renderbytescount = 0; } else { - ret = acp_dma_stop(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_SYSRAM_CH_NUM); + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); rtd->i2ssp_capturebytescount = 0; } break; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 0e6089b..5e25428 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -85,6 +85,11 @@ struct audio_substream_data { unsigned int order; u16 num_of_pages; u16 direction; + u16 ch1; + u16 ch2; + u16 destination; + u16 dma_dscr_idx_1; + u16 dma_dscr_idx_2; uint64_t size; u64 i2ssp_renderbytescount; u64 i2ssp_capturebytescount; -- 2.7.4

5 21

[alsa-devel] [PATCH 01/11] ASoC: amd: rename audio_substream_data variable
by Vijendar Mukunda 21 May '18

21 May '18

In order to make audio_substream_data structure variable consistent throughout the code, changed the name from audio_config to rtd wherever applicable. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 5ffe2ef..9c026c4 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -317,13 +317,13 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, } static void config_acp_dma(void __iomem *acp_mmio, - struct audio_substream_data *audio_config, + struct audio_substream_data *rtd, u32 asic_type) { u32 pte_offset, sram_bank; u16 ch1, ch2, destination, dma_dscr_idx; - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { pte_offset = ACP_PLAYBACK_PTE_OFFSET; ch1 = SYSRAM_TO_ACP_CH_NUM; ch2 = ACP_TO_I2S_DMA_CH_NUM; @@ -344,25 +344,25 @@ static void config_acp_dma(void __iomem *acp_mmio, destination = FROM_ACP_I2S_1; } - acp_pte_config(acp_mmio, audio_config->pg, audio_config->num_of_pages, + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, pte_offset); - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; else dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH14; /* Configure System memory <-> ACP SRAM DMA descriptors */ - set_acp_sysmem_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, pte_offset, ch1, + set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, + rtd->direction, pte_offset, ch1, sram_bank, dma_dscr_idx, asic_type); - if (audio_config->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH13; else dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH15; /* Configure ACP SRAM <-> I2S DMA descriptors */ - set_acp_to_i2s_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, sram_bank, + set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, + rtd->direction, sram_bank, destination, ch2, dma_dscr_idx, asic_type); } -- 2.7.4

4 27

[alsa-devel] [PATCH V2 01/10] ASoC: amd: dma config parameters changes
by Vijendar Mukunda 21 May '18

21 May '18

Added dma configuration parameters to rtd structure. Moved dma configuration parameters initialization to hw_params callback. Removed hard coding in prepare and trigger callbacks. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Daniel Kurtz <djkurtz(a)chromium.org> --- v1->v2 : Fixed capture stream wrong channel assignment added comments in dma trigger api sound/soc/amd/acp-pcm-dma.c | 103 ++++++++++++++++++-------------------------- sound/soc/amd/acp.h | 5 +++ 2 files changed, 48 insertions(+), 60 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9c026c4..e9736fe 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -321,19 +321,12 @@ static void config_acp_dma(void __iomem *acp_mmio, u32 asic_type) { u32 pte_offset, sram_bank; - u16 ch1, ch2, destination, dma_dscr_idx; if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { pte_offset = ACP_PLAYBACK_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; sram_bank = ACP_SHARED_RAM_BANK_1_ADDRESS; - destination = TO_ACP_I2S_1; - } else { pte_offset = ACP_CAPTURE_PTE_OFFSET; - ch1 = SYSRAM_TO_ACP_CH_NUM; - ch2 = ACP_TO_I2S_DMA_CH_NUM; switch (asic_type) { case CHIP_STONEY: sram_bank = ACP_SHARED_RAM_BANK_3_ADDRESS; @@ -341,30 +334,19 @@ static void config_acp_dma(void __iomem *acp_mmio, default: sram_bank = ACP_SHARED_RAM_BANK_5_ADDRESS; } - destination = FROM_ACP_I2S_1; } - acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, pte_offset); - if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH14; - /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, - rtd->direction, pte_offset, ch1, - sram_bank, dma_dscr_idx, asic_type); - - if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) - dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH13; - else - dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH15; + rtd->direction, pte_offset, + rtd->ch1, sram_bank, + rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, rtd->direction, sram_bank, - destination, ch2, dma_dscr_idx, - asic_type); + rtd->destination, rtd->ch2, + rtd->dma_dscr_idx_2, asic_type); } /* Start a given DMA channel transfer */ @@ -804,6 +786,21 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->ch1 = SYSRAM_TO_ACP_CH_NUM; + rtd->ch2 = ACP_TO_I2S_DMA_CH_NUM; + rtd->destination = TO_ACP_I2S_1; + rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; + rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; + } else { + rtd->ch1 = ACP_TO_SYSRAM_CH_NUM; + rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM; + rtd->destination = FROM_ACP_I2S_1; + rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; + rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; + } + size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) @@ -898,21 +895,15 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH12, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH13, - NUM_DSCRS_PER_CHANNEL, 0); - } else { - config_acp_dma_channel(rtd->acp_mmio, ACP_TO_SYSRAM_CH_NUM, - CAPTURE_START_DMA_DESCR_CH14, - NUM_DSCRS_PER_CHANNEL, 0); - config_acp_dma_channel(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM, - CAPTURE_START_DMA_DESCR_CH15, - NUM_DSCRS_PER_CHANNEL, 0); - } + + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch1, + rtd->dma_dscr_idx_1, + NUM_DSCRS_PER_CHANNEL, 0); + config_acp_dma_channel(rtd->acp_mmio, + rtd->ch2, + rtd->dma_dscr_idx_2, + NUM_DSCRS_PER_CHANNEL, 0); return 0; } @@ -939,10 +930,9 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (rtd->i2ssp_renderbytescount == 0) rtd->i2ssp_renderbytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM, false); + acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & - BIT(SYSRAM_TO_ACP_CH_NUM)) { + BIT(rtd->ch1)) { if (!loops--) { dev_err(component->dev, "acp dma start timeout\n"); @@ -950,38 +940,31 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) } cpu_relax(); } - - acp_dma_start(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM, true); - } else { if (rtd->i2ssp_capturebytescount == 0) rtd->i2ssp_capturebytescount = bytescount; - acp_dma_start(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM, true); } + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* - * Need to stop only circular DMA channels : - * ACP_TO_I2S_DMA_CH_NUM / I2S_TO_ACP_DMA_CH_NUM. Non-circular - * channels will stopped automatically after its transfer - * completes : SYSRAM_TO_ACP_CH_NUM / ACP_TO_SYSRAM_CH_NUM + /* For playback, non circular dma should be stopped first + * i.e Sysram to acp dma transfer channel(rtd->ch1) should be + * stopped before stopping cirular dma which is acp sram to i2s + * fifo dma transfer channel(rtd->ch2). Where as in Capture + * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped + * first before stopping acp sram to sysram which is circular + * dma(rtd->ch1). */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = acp_dma_stop(rtd->acp_mmio, - SYSRAM_TO_ACP_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_I2S_DMA_CH_NUM); + acp_dma_stop(rtd->acp_mmio, rtd->ch1); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); rtd->i2ssp_renderbytescount = 0; } else { - ret = acp_dma_stop(rtd->acp_mmio, - I2S_TO_ACP_DMA_CH_NUM); - ret = acp_dma_stop(rtd->acp_mmio, - ACP_TO_SYSRAM_CH_NUM); + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); rtd->i2ssp_capturebytescount = 0; } break; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 0e6089b..5e25428 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -85,6 +85,11 @@ struct audio_substream_data { unsigned int order; u16 num_of_pages; u16 direction; + u16 ch1; + u16 ch2; + u16 destination; + u16 dma_dscr_idx_1; + u16 dma_dscr_idx_2; uint64_t size; u64 i2ssp_renderbytescount; u64 i2ssp_capturebytescount; -- 2.7.4

3 12

[alsa-devel] [PATCH v3 00/25] ASoC: qcom: Add support to QDSP based Audio
by srinivas.kandagatla＠linaro.org 21 May '18

21 May '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Here is my test branch incase somone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.16-rc… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | fedai | <------> | mixers | <-----> | bedai | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Changes since v2 (https://lwn.net/Articles/741472/) - fixed locking issues with ASM and APR. - Added support to MI2S and SLIMBus ports - moved dma device to apr. - added dt bindings for ASM, ADM, AFE. - Fixed dt bindings for msm8996 sound card. - added compatible strings to ASM, ADM, AFE modules. - Cleaned up common code as suggested in review. - converted sound card driver to proper apr client driver. - Fixed various issues spotted by Banajit and Kasam. - Tested on DB410c, DB820c for both analog and digital playbacks. - Exported q6core functions as suggested by Rohit. - Fixed various issues with dsp carsh/recover usecase. - Added multiple ASM streams as requested in review. Srinivas Kandagatla (25): dt-bindings: soc: qcom: Add bindings for APR bus soc: qcom: add support to APR bus driver ASoC: qcom: qdsp6: Add common qdsp6 helper functions dt-bindings: sound: qcom: Add bindings for q6afe ASoC: qcom: qdsp6: Add support to Q6AFE dt-bindings: sound: qcom: Add bindings for q6adm ASoC: qcom: qdsp6: Add support to Q6ADM dt-bindings: sound: qcom: Add bindings for q6asm ASoC: qcom: qdsp6: Add support to Q6ASM ASoC: qcom: q6asm: Add support to memory map and unmap ASoC: qcom: q6asm: add support to audio stream apis ASoC: qcom: qdsp6: Add support to Q6CORE ASoC: qcom: qdsp6: Add support to q6routing driver ASoC: qcom: qdsp6: Add support to q6afe dai driver ASoC: qcom: qdsp6: Add support to q6asm dai driver ASoC: qcom: q6afe: add SLIMBus port Support ASoC: qcom: q6afe-dai: add support to slim afe dais ASoC: qcom: q6routing: add support to all SLIMBus Mixers ASoC: qcom: q6afe: add support to MI2S ports ASoC: qcom: q6afe: add support to MI2S sysclks ASoC: qcom: q6afe-dai: add support to 4 MI2S ports ASoC: qcom: q6routing: add support to MI2S Mixers dt-bindings: sound: qcom: Add devicetree bindings for apq8096 ASoC: qcom: apq8096: Add db820c machine driver arm64: dts: msm8996: db820c: Add sound card support .../devicetree/bindings/soc/qcom/qcom,apr.txt | 83 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 89 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 31 + .../devicetree/bindings/sound/qcom,q6afe.txt | 38 + .../devicetree/bindings/sound/qcom,q6asm.txt | 38 + arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 44 +- arch/arm64/boot/dts/qcom/msm8996.dtsi | 62 ++ drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 381 +++++++ include/dt-bindings/soc/qcom,apr.h | 27 + include/dt-bindings/sound/qcom,q6afe.h | 33 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 131 +++ sound/soc/qcom/Kconfig | 42 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 173 ++++ sound/soc/qcom/qdsp6/Makefile | 5 + sound/soc/qcom/qdsp6/q6adm.c | 634 ++++++++++++ sound/soc/qcom/qdsp6/q6adm.h | 29 + sound/soc/qcom/qdsp6/q6afe-dai.c | 645 ++++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 875 ++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 74 ++ sound/soc/qcom/qdsp6/q6asm-dai.c | 621 ++++++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1063 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 66 ++ sound/soc/qcom/qdsp6/q6core.c | 235 +++++ sound/soc/qcom/qdsp6/q6core.h | 9 + sound/soc/qcom/qdsp6/q6dsp-common.c | 67 ++ sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 97 ++ sound/soc/qcom/qdsp6/q6routing.c | 758 ++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 34 files changed, 6430 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/dt-bindings/sound/qcom,q6afe.h create mode 100644 include/dt-bindings/sound/qcom,q6asm.h create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6core.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.15.1

5 62

[alsa-devel] [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio
by srinivas.kandagatla＠linaro.org 21 May '18

21 May '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Thankyou everyone for providing feedback on v3 patchset. This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Previous version of these patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.16-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Changes since v3 (https://lkml.org/lkml/2018/2/13/719) - Cleaned up apr and other dt bindings from comments from various reviewers. - Updated sndcard and dsp components to use component framework - Fixed various locking issues spotted by Mark B. - Removed Error to string lookup functions which can be done later. - cleaned up the subject headers suggested by Mark B - cleaned up Kconfig as suggested in review. - Fixed SPDX licence commenting style, suggested by Mark. - Made adsp command and response in sync - moved sound node out of apr bus. - Added more info into dsp versioning support - Fixed issues with MI2S SD-line selection, spotted by Rohit. - Added my self as co-maintainer for qcom audio stuff. - Fixed q6routing issue spotted by flto and rohit. Srinivas Kandagatla (24): soc: qcom dt-bindings: Add APR bus bindings soc: qcom: Add APR bus driver ASoC: qdsp6: q6common: Add qdsp6 helper functions ASoC: qdsp6: dt-bindings: Add q6afe dt bindings ASoC: qdsp6: dt-bindings: Add q6adm dt bindings ASoC: qdsp6: dt-bindings: Add q6asm dt bindings ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6core: Add q6core driver ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver arm64: dts: db820c: add sound card dt entry MAINTAINERS: Add myself as co-maintainer of qcom audio .../devicetree/bindings/soc/qcom/qcom,apr.txt | 85 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 80 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 34 + .../devicetree/bindings/sound/qcom,q6afe.txt | 107 ++ .../devicetree/bindings/sound/qcom,q6asm.txt | 34 + MAINTAINERS | 1 + arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 34 + arch/arm64/boot/dts/qcom/msm8996.dtsi | 75 +- drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 379 ++++++ include/dt-bindings/soc/qcom,apr.h | 27 + include/dt-bindings/sound/qcom,q6afe.h | 31 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 131 ++ sound/soc/qcom/Kconfig | 50 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 235 ++++ sound/soc/qcom/qdsp6/Makefile | 8 + sound/soc/qcom/qdsp6/q6adm.c | 664 ++++++++++ sound/soc/qcom/qdsp6/q6adm.h | 25 + sound/soc/qcom/qdsp6/q6afe-dai.c | 746 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1067 ++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 632 ++++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1310 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 ++ sound/soc/qcom/qdsp6/q6core.c | 381 ++++++ sound/soc/qcom/qdsp6/q6core.h | 15 + sound/soc/qcom/qdsp6/q6dsp-common.c | 66 + sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 51 + sound/soc/qcom/qdsp6/q6routing.c | 982 +++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 35 files changed, 7592 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/dt-bindings/sound/qcom,q6afe.h create mode 100644 include/dt-bindings/sound/qcom,q6asm.h create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6core.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.15.1

4 32

[alsa-devel] [PATCH v8 00/24] ASoC: qcom: Add support to QDSP based Audio
by Srinivas Kandagatla 21 May '18

21 May '18

Thankyou everyone for providing feedback and testing v7 patchset. This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.17-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Changes since v6 (https://lkml.org/lkml/2018/5/1/176) - Added Banajit's and Bjorn's reviewed by tags to patches. - Few minor typos spotted by Banajit. - Made read/write allocations to be atomic as spotted by Rohit. - fixed error path in db820c machine driver. - rebased patchset on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/?h=for-ne… Srinivas Kandagatla (24): soc: qcom dt-bindings: Add APR bus bindings soc: qcom: Add APR bus driver ASoC: qdsp6: dt-bindings: Add q6core dt bindings ASoC: qdsp6: dt-bindings: Add q6afe dt bindings ASoC: qdsp6: dt-bindings: Add q6adm dt bindings ASoC: qdsp6: dt-bindings: Add q6asm dt bindings ASoC: qdsp6: q6common: Add qdsp6 helper functions ASoC: qdsp6: q6core: Add q6core driver ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver MAINTAINERS: Add myself as co-maintainer of qcom audio .../devicetree/bindings/soc/qcom/qcom,apr.txt | 84 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 109 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 33 + .../devicetree/bindings/sound/qcom,q6afe.txt | 104 ++ .../devicetree/bindings/sound/qcom,q6asm.txt | 33 + .../devicetree/bindings/sound/qcom,q6core.txt | 21 + MAINTAINERS | 1 + drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 378 ++++++ include/dt-bindings/soc/qcom,apr.h | 28 + include/dt-bindings/sound/qcom,q6afe.h | 31 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 128 ++ sound/soc/qcom/Kconfig | 50 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 255 ++++ sound/soc/qcom/qdsp6/Makefile | 8 + sound/soc/qcom/qdsp6/q6adm.c | 633 +++++++++ sound/soc/qcom/qdsp6/q6adm.h | 25 + sound/soc/qcom/qdsp6/q6afe-dai.c | 748 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1046 +++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 624 +++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1398 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 + sound/soc/qcom/qdsp6/q6core.c | 380 ++++++ sound/soc/qcom/qdsp6/q6core.h | 15 + sound/soc/qcom/qdsp6/q6dsp-common.c | 66 + sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 51 + sound/soc/qcom/qdsp6/q6routing.c | 982 ++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 34 files changed, 7574 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/dt-bindings/sound/qcom,q6afe.h create mode 100644 include/dt-bindings/sound/qcom,q6asm.h create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6core.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.16.2

3 35

[alsa-devel] [PATCH v6 00/24] ASoC: qcom: Add support to QDSP based Audio
by srinivas.kandagatla＠linaro.org 21 May '18

21 May '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Thankyou everyone for providing feedback and testing v5 patchset. This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.17-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Changes since v5 (https://lkml.org/lkml/2018/4/18/553) - updated bindings as reviewed by Rob H. - added missing binding for q6core - fixed copp double count reported by Amit Pundir - Updated proper SPDX licence on headers - reorder patches to fix bisect errors reported by kbuild Srinivas Kandagatla (24): soc: qcom dt-bindings: Add APR bus bindings soc: qcom: Add APR bus driver ASoC: qdsp6: dt-bindings: Add q6core dt bindings ASoC: qdsp6: dt-bindings: Add q6afe dt bindings ASoC: qdsp6: dt-bindings: Add q6adm dt bindings ASoC: qdsp6: dt-bindings: Add q6asm dt bindings ASoC: qdsp6: q6common: Add qdsp6 helper functions ASoC: qdsp6: q6core: Add q6core driver ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver MAINTAINERS: Add myself as co-maintainer of qcom audio .../devicetree/bindings/soc/qcom/qcom,apr.txt | 84 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 109 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 33 + .../devicetree/bindings/sound/qcom,q6afe.txt | 88 ++ .../devicetree/bindings/sound/qcom,q6asm.txt | 33 + .../devicetree/bindings/sound/qcom,q6core.txt | 21 + MAINTAINERS | 1 + drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 384 ++++++ include/dt-bindings/soc/qcom,apr.h | 28 + include/dt-bindings/sound/qcom,q6afe.h | 31 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 130 ++ sound/soc/qcom/Kconfig | 50 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 238 ++++ sound/soc/qcom/qdsp6/Makefile | 8 + sound/soc/qcom/qdsp6/q6adm.c | 635 ++++++++++ sound/soc/qcom/qdsp6/q6adm.h | 25 + sound/soc/qcom/qdsp6/q6afe-dai.c | 752 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1071 ++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 632 ++++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1312 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 + sound/soc/qcom/qdsp6/q6core.c | 380 ++++++ sound/soc/qcom/qdsp6/q6core.h | 15 + sound/soc/qcom/qdsp6/q6dsp-common.c | 66 + sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 51 + sound/soc/qcom/qdsp6/q6routing.c | 982 +++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 34 files changed, 7502 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/dt-bindings/sound/qcom,q6afe.h create mode 100644 include/dt-bindings/sound/qcom,q6asm.h create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6core.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.16.2

6 44

[alsa-devel] [PATCH] ALSA: dice: add stream format parameters for TC Electronic Digital Konnekt x32
by Takashi Sakamoto 21 May '18

21 May '18

TC Electronic Digital Konnekt x32 is an application of WaveFront DiceII STD and doesn't support TCAT extended application protocol. For such devices, ALSA dice driver needs to have hard-coded parameters for stream formats. This commit adds stream format parameters for this model. Unfortunately, at sampling transmission frequencies of 88.2/96.0kHz, I confirmed that current ALSA dice driver doesn't drive the device appropriately due to detecting packet discontinuities. $ journalctl kernel: snd_dice fw1.0: Detect discontinuity of CIP: 90 80 At the frequencies, the device transfers 16 data blocks per packet and 16 data channels per data block, as a result one packet includes 1032 bytes if it's not NODATA. However, as long as I checked, the device often postpone packet transmission and continue with truncated payload than metadata in isochronous packet header. Below is a sample of sequence I got. sec cycle bytes CIP1 CIP2 37 3314 1032 0x01100090 0x900449E2 37 3315 8 0x011000A0 0x9004FFFF 37 3316 1032 0x011000A0 0x900461E2 37 3317 1032 0x011000B0 0x900475E2 37 3318 1032 0x011000C0 0x900489E2 37 3319 8 0x011000D0 0x9004FFFF 37 3320 1032 0x011000D0 0x9004A1E2 37 3321 1032 0x011000E0 0x9004B5E2 37 3322 1032 0x011000F0 0x9004C9E2 37 3323 8 0x01100000 0x9004FFFF 37 3324 1032 0x01100000 0x9004E1E2 37 3325 1032 0x01100010 0x9004F5E2 37 3326 1032 0x01100020 0x900409E2 37 3327 8 0x01100030 0x9004FFFF 37 3328 1032 0x01100030 0x900421E2 37 3329 1032 0x01100040 0x900435E2 37 3330 (skip) 37 3331 (skip) 37 3332 (skip) 37 3333 (skip) 37 3334 (skip) 37 3335 (skip) 37 3336 (skip) 37 3337 (skip) 37 3338 (skip) 37 3339 (skip) 37 3340 (skip) 37 3341 (skip) 37 3342 (skip) 37 3343 (skip) 37 3344 (skip) 37 3345 (skip) 37 3346 (skip) 37 3347 (skip) 37 3348 (skip) 37 3349 (skip) 37 3350 (skip) 37 3351 (skip) 37 3352 (skip) 37 3353 (skip) 37 3354 (skip) 37 3355 (skip) 37 3356 (skip) 37 3357 (skip) 37 3358 (skip) 37 3359 (skip) 37 3360 (skip) 37 3361 (skip) 37 3362 (skip) 37 3363 (skip) 37 3364 (skip) 37 3365 (skip) 37 3366 (skip) 37 3367 1032 0x01100050 0x900461E1 37 3368 1032 0x01100060 0x900475E1 37 3369 1032 0x01100070 0x9004A1E1 37 3370 1032 0x01100080 0x9004A1E1 but content of payload is truncated. 37 3371 (skip) 37 3371 1032 0x01100080 0x9004B5E0 detect discontinuity 37 3372 1032 0x01100090 0x9004C9E0 37 3373 1032 0x011000A0 0x9004E1E0 37 3374 1032 0x011000B0 0x9004F5E0 37 3375 1032 0x011000C0 0x900409E0 37 3376 1032 0x011000D0 0x900421E0 37 3377 1032 0x011000E0 0x900435E0 37 3378 1032 0x011000F0 0x900449DF 37 3379 8 0x01100000 0x9004FFFF 37 3380 1032 0x01100000 0x900461DF 37 3381 1032 0x01100010 0x900475DF 37 3382 1032 0x01100020 0x900489DF 37 3383 8 0x01100030 0x9004FFFF 37 3384 1032 0x01100030 0x9004A1DF 37 3385 1032 0x01100040 0x9004B5DF 37 3386 1032 0x01100050 0x9004C9DF 37 3387 8 0x01100060 0x9004FFFF I cannot confirm this quirks with Windows driver. ALSA dice driver has a cause if assumed differences between these two drivers are ways of timestampling to RX packets from the drivers to the device. I've already reported timestamping quirk of Dice-based devices and this might bring this issue. [alsa-devel] Dice packet sequence quirk and ALSA firewire stack in Linux 4.6 http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107715.html Well, nevertheless, I enable ALSA dice driver to work at the frequencies. This may brings inconvenience to users but I expect developers and users to fix it. $ cd linux-firewire-utils/src $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 040423bb bus_info_length 4, crc_length 4, crc 9147 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 00016604 company_id 000166 | 410 0c232c28 device_id 040c232c28 | EUI-64 000166040c232c28 root directory ----------------------------------------------------------------- 414 0006b6cb directory_length 6, crc 46795 418 03000166 vendor 41c 8100000a --> descriptor leaf at 444 420 17000030 model 424 8100000f --> descriptor leaf at 460 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 000476c2 directory_length 4, crc 30402 434 12000166 specifier id 438 13000001 version 43c 17000030 model 440 81000010 --> descriptor leaf at 480 descriptor leaf at 444 ----------------------------------------------------------------- 444 0006c490 leaf_length 6, crc 50320 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 54432045 "TC E" 454 6c656374 "lect" 458 726f6e69 "roni" 45c 63000000 "c" descriptor leaf at 460 ----------------------------------------------------------------- 460 000772b4 leaf_length 7, crc 29364 464 00000000 textual descriptor 468 00000000 minimal ASCII 46c 44696769 "Digi" 470 74616c4b "talK" 474 6f6e6e65 "onne" 478 6b747833 "ktx3" 47c 32000000 "2" descriptor leaf at 480 ----------------------------------------------------------------- 480 000772b4 leaf_length 7, crc 29364 484 00000000 textual descriptor 488 00000000 minimal ASCII 48c 44696769 "Digi" 490 74616c4b "talK" 494 6f6e6e65 "onne" 498 6b747833 "ktx3" 49c 32000000 "2" Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-tcelectronic.c | 7 +++++++ sound/firewire/dice/dice.c | 8 ++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c index e134a5110c6c..f9e0072deb81 100644 --- a/sound/firewire/dice/dice-tcelectronic.c +++ b/sound/firewire/dice/dice-tcelectronic.c @@ -49,6 +49,12 @@ static const struct dice_tc_spec studio_konnekt_48 = { .has_midi = true, }; +static const struct dice_tc_spec digital_konnekt_x32 = { + .tx_pcm_chs = {{16, 16, 4}, {0, 0, 0} }, + .rx_pcm_chs = {{16, 16, 4}, {0, 0, 0} }, + .has_midi = false, +}; + int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice) { static const struct { @@ -61,6 +67,7 @@ int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice) {0x00000023, &konnekt_live}, {0x00000024, &desktop_konnekt6}, {0x00000027, &impact_twin}, + {0x00000030, &digital_konnekt_x32}, }; struct fw_csr_iterator it; int key, val, model_id; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index beeef62581ba..774eb2205668 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -358,6 +358,14 @@ static const struct ieee1394_device_id dice_id_table[] = { .model_id = 0x000027, .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, }, + /* TC Electronic Digital Konnekt x32. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000030, + .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats, + }, /* Alesis iO14/iO26. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: imx-audmux: add RXFS/RXCLK defines for 6-wire connections
by Marco Felsch 20 May '18

20 May '18

From: Philipp Zabel <p.zabel(a)pengutronix.de> In asynchronous mode, a RxFS and RxClk connection needs to be made between two ports. Add a define for the bit to be set in the *SEL fields. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 751fe1416f95..6120696abe63 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -25,6 +25,10 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +/* TFSEL/TCSEL/RFSEL/RCSEL can be sourced from Rx/Tx clocks on i.MX51/53/6Q */ +#define IMX_AUDMUX_RXFS 0x8 +#define IMX_AUDMUX_RXCLK 0x8 + /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ #define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) #define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -- 2.17.0

2 1

[alsa-devel] [PATCH 1/2] conf/ucm: bytcr-rt5640: Add generic bytcr-rt5640 UCM profile
by Hans de Goede 19 May '18

19 May '18

This commit adds the generic UCM profile for bytcr-rt5640 boards from: https://github.com/plbossart/UCM, plus the fixes from this pull-req: https://github.com/plbossart/UCM/pull/31 The profile has been split up into separate per input / output files to allow for creation of long-name profiles with the specific input / output combinations found on a board without needing to copy and paste things. Note this profile exports all inputs and both stereo/mono speaker setups even though a typical device will not use all. Ideally a long-name based device specific profile made up of the various parts should be used instead. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- configure.ac | 3 + src/conf/ucm/Makefile.am | 3 + src/conf/ucm/bytcr-rt5640/HiFi.conf | 29 +++++ src/conf/ucm/bytcr-rt5640/Makefile.am | 4 + src/conf/ucm/bytcr-rt5640/bytcr-rt5640.conf | 8 ++ src/conf/ucm/bytcr/Makefile.am | 4 + src/conf/ucm/bytcr/PlatformEnableSeq.conf | 115 ++++++++++++++++++++ src/conf/ucm/rt5640/DigitalMics.conf | 31 ++++++ src/conf/ucm/rt5640/EnableSeq.conf | 41 +++++++ src/conf/ucm/rt5640/HeadPhones.conf | 41 +++++++ src/conf/ucm/rt5640/HeadsetMic.conf | 48 ++++++++ src/conf/ucm/rt5640/IN1-InternalMic.conf | 50 +++++++++ src/conf/ucm/rt5640/IN3-InternalMic.conf | 50 +++++++++ src/conf/ucm/rt5640/Makefile.am | 5 + src/conf/ucm/rt5640/MonoSpeaker.conf | 45 ++++++++ src/conf/ucm/rt5640/Speaker.conf | 45 ++++++++ 16 files changed, 522 insertions(+) create mode 100644 src/conf/ucm/bytcr-rt5640/HiFi.conf create mode 100644 src/conf/ucm/bytcr-rt5640/Makefile.am create mode 100644 src/conf/ucm/bytcr-rt5640/bytcr-rt5640.conf create mode 100644 src/conf/ucm/bytcr/Makefile.am create mode 100644 src/conf/ucm/bytcr/PlatformEnableSeq.conf create mode 100644 src/conf/ucm/rt5640/DigitalMics.conf create mode 100644 src/conf/ucm/rt5640/EnableSeq.conf create mode 100644 src/conf/ucm/rt5640/HeadPhones.conf create mode 100644 src/conf/ucm/rt5640/HeadsetMic.conf create mode 100644 src/conf/ucm/rt5640/IN1-InternalMic.conf create mode 100644 src/conf/ucm/rt5640/IN3-InternalMic.conf create mode 100644 src/conf/ucm/rt5640/Makefile.am create mode 100644 src/conf/ucm/rt5640/MonoSpeaker.conf create mode 100644 src/conf/ucm/rt5640/Speaker.conf diff --git a/configure.ac b/configure.ac index 693b5d53..f6ee6ca8 100644 --- a/configure.ac +++ b/configure.ac @@ -720,6 +720,8 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/broxton-rt298/Makefile \ + src/conf/ucm/bytcr/Makefile \ + src/conf/ucm/bytcr-rt5640/Makefile \ src/conf/ucm/bytcr-rt5651/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ @@ -734,6 +736,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/PandaBoard/Makefile \ src/conf/ucm/PandaBoardES/Makefile \ src/conf/ucm/PAZ00/Makefile \ + src/conf/ucm/rt5640/Makefile \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/skylake-rt286/Makefile \ src/conf/ucm/TECLAST-X80Pro-Defaultstring-CherryTrailCR/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 38fc6b25..dc2d6857 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -2,6 +2,8 @@ SUBDIRS=\ ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN \ broadwell-rt286 \ broxton-rt298 \ +bytcr \ +bytcr-rt5640 \ bytcr-rt5651 \ chtrt5645 \ chtrt5645-mono-speaker-analog-mic \ @@ -16,6 +18,7 @@ LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 \ PandaBoard \ PandaBoardES \ PAZ00 \ +rt5640 \ SDP4430 \ skylake-rt286 \ TECLAST-X80Pro-Defaultstring-CherryTrailCR \ diff --git a/src/conf/ucm/bytcr-rt5640/HiFi.conf b/src/conf/ucm/bytcr-rt5640/HiFi.conf new file mode 100644 index 00000000..130671ca --- /dev/null +++ b/src/conf/ucm/bytcr-rt5640/HiFi.conf @@ -0,0 +1,29 @@ +# Use case Configuration for bytcr-rt5640 + +<searchdir:ucm> + +SectionVerb { + EnableSequence [ + cdev "hw:bytcrrt5640" + + <bytcr/PlatformEnableSeq.conf> + <rt5640/EnableSeq.conf> + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:bytcrrt5640" + CapturePCM "hw:bytcrrt5640" + } +} + +<rt5640/Speaker.conf> +<rt5640/MonoSpeaker.conf> +<rt5640/HeadPhones.conf> + +<rt5640/DigitalMics.conf> +<rt5640/IN1-InternalMic.conf> +<rt5640/IN3-InternalMic.conf> +<rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640/Makefile.am b/src/conf/ucm/bytcr-rt5640/Makefile.am new file mode 100644 index 00000000..84ddf7b1 --- /dev/null +++ b/src/conf/ucm/bytcr-rt5640/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/bytcr-rt5640 +ucm_DATA = bytcr-rt5640.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/bytcr-rt5640/bytcr-rt5640.conf b/src/conf/ucm/bytcr-rt5640/bytcr-rt5640.conf new file mode 100644 index 00000000..676b4fd9 --- /dev/null +++ b/src/conf/ucm/bytcr-rt5640/bytcr-rt5640.conf @@ -0,0 +1,8 @@ +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Play HiFi quality Music" +} + +SectionDefaults [ + cdev "hw:bytcrrt5640" +] diff --git a/src/conf/ucm/bytcr/Makefile.am b/src/conf/ucm/bytcr/Makefile.am new file mode 100644 index 00000000..a0e5545e --- /dev/null +++ b/src/conf/ucm/bytcr/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/bytcr +ucm_DATA = PlatformEnableSeq.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/bytcr/PlatformEnableSeq.conf b/src/conf/ucm/bytcr/PlatformEnableSeq.conf new file mode 100644 index 00000000..b5ee2b41 --- /dev/null +++ b/src/conf/ucm/bytcr/PlatformEnableSeq.conf @@ -0,0 +1,115 @@ +# media mixer settings +# compress +cset "name='media0_in Gain 0 Switch' on" +cset "name='media0_in Gain 0 Volume' 0" + +# normal +cset "name='media1_in Gain 0 Switch' on" +cset "name='media1_in Gain 0 Volume' 0" +# swm loopback +cset "name='media2_in Gain 0 Switch' off" +cset "name='media2_in Gain 0 Volume' 0%" +# deep buffer +cset "name='media3_in Gain 0 Switch' on" +cset "name='media3_in Gain 0 Volume' 0" + +cset "name='media0_out mix 0 media0_in Switch' on" +cset "name='media0_out mix 0 media1_in Switch' on" +cset "name='media0_out mix 0 media2_in Switch' off" +cset "name='media0_out mix 0 media3_in Switch' on" + +cset "name='media1_out mix 0 media0_in Switch' off" +cset "name='media1_out mix 0 media1_in Switch' off" +cset "name='media1_out mix 0 media2_in Switch' off" +cset "name='media1_out mix 0 media3_in Switch' off" + +cset "name='pcm0_in Gain 0 Switch' on" +cset "name='pcm0_in Gain 0 Volume' 0" + +cset "name='pcm1_in Gain 0 Switch' off" +cset "name='pcm1_in Gain 0 Volume' 0%" + +# codec0_out settings (used if SSP2 is connected) +cset "name='codec_out0 mix 0 codec_in0 Switch' off" +cset "name='codec_out0 mix 0 codec_in1 Switch' off" +cset "name='codec_out0 mix 0 media_loop1_in Switch' off" +cset "name='codec_out0 mix 0 media_loop2_in Switch' off" +cset "name='codec_out0 mix 0 pcm0_in Switch' on" +cset "name='codec_out0 mix 0 pcm1_in Switch' off" +cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" +cset "name='codec_out0 Gain 0 Switch' on" +cset "name='codec_out0 Gain 0 Volume' 0" + +# modem_out settings (used if SSP0 is connected) +cset "name='modem_out mix 0 codec_in0 Switch' off" +cset "name='modem_out mix 0 codec_in1 Switch' off" +cset "name='modem_out mix 0 media_loop1_in Switch' off" +cset "name='modem_out mix 0 media_loop2_in Switch' off" +cset "name='modem_out mix 0 pcm0_in Switch' on" +cset "name='modem_out mix 0 pcm1_in Switch' off" +cset "name='modem_out mix 0 sprot_loop_in Switch' off" +cset "name='modem_out Gain 0 Switch' on" +cset "name='modem_out Gain 0 Volume' 0" + +# input settings + +# input used when SSP2 is connected +cset "name='codec_in0 Gain 0 Switch' on" +cset "name='codec_in0 Gain 0 Volume' 0" + +# input used when SSP0 is connected +cset "name='modem_in Gain 0 Switch' on" +cset "name='modem_in Gain 0 Volume' 0" + +# pcm1_out settings +cset "name='pcm1_out mix 0 codec_in0 Switch' on" +cset "name='pcm1_out mix 0 modem_in Switch' on" +cset "name='pcm1_out mix 0 codec_in1 Switch' off" +cset "name='pcm1_out mix 0 media_loop1_in Switch' off" +cset "name='pcm1_out mix 0 media_loop2_in Switch' off" +cset "name='pcm1_out mix 0 pcm0_in Switch' off" +cset "name='pcm1_out mix 0 pcm1_in Switch' off" +cset "name='pcm1_out mix 0 sprot_loop_in Switch' off" + +cset "name='pcm1_out Gain 0 Switch' on" +cset "name='pcm1_out Gain 0 Volume' 0" + +# disable codec_out1 +cset "name='codec_out1 mix 0 codec_in0 Switch' off" +cset "name='codec_out1 mix 0 codec_in1 Switch' off" +cset "name='codec_out1 mix 0 media_loop1_in Switch' off" +cset "name='codec_out1 mix 0 media_loop2_in Switch' off" +cset "name='codec_out1 mix 0 pcm0_in Switch' off" +cset "name='codec_out1 mix 0 pcm1_in Switch' off" +cset "name='codec_out1 mix 0 sprot_loop_in Switch' off" +cset "name='codec_out1 Gain 0 Switch' off" +cset "name='codec_out1 Gain 0 Volume' 0%" + +# disable codec_in1 +cset "name='codec_in1 Gain 0 Switch' off" +cset "name='codec_in1 Gain 0 Volume' 0%" + +# disable all loops +cset "name='media_loop1_out mix 0 codec_in0 Switch' off" +cset "name='media_loop1_out mix 0 codec_in1 Switch' off" +cset "name='media_loop1_out mix 0 media_loop1_in Switch' off" +cset "name='media_loop1_out mix 0 media_loop2_in Switch' off" +cset "name='media_loop1_out mix 0 pcm0_in Switch' off" +cset "name='media_loop1_out mix 0 pcm1_in Switch' off" +cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off" + +cset "name='media_loop2_out mix 0 codec_in0 Switch' off" +cset "name='media_loop2_out mix 0 codec_in1 Switch' off" +cset "name='media_loop2_out mix 0 media_loop1_in Switch' off" +cset "name='media_loop2_out mix 0 media_loop2_in Switch' off" +cset "name='media_loop2_out mix 0 pcm0_in Switch' off" +cset "name='media_loop2_out mix 0 pcm1_in Switch' off" +cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off" + +cset "name='sprot_loop_out mix 0 codec_in0 Switch' off" +cset "name='sprot_loop_out mix 0 codec_in1 Switch' off" +cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off" +cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off" +cset "name='sprot_loop_out mix 0 pcm0_in Switch' off" +cset "name='sprot_loop_out mix 0 pcm1_in Switch' off" +cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off" diff --git a/src/conf/ucm/rt5640/DigitalMics.conf b/src/conf/ucm/rt5640/DigitalMics.conf new file mode 100644 index 00000000..c7d54099 --- /dev/null +++ b/src/conf/ucm/rt5640/DigitalMics.conf @@ -0,0 +1,31 @@ +SectionDevice."DigitalMics" { + Comment "DigitalMics capture" + + ConflictingDevice [ + "IN1-InternalMics" + "IN3-InternalMics" + "HeadsetMic" + ] + + EnableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='Stereo ADC MIXL ADC2 Switch' on" + cset "name='Stereo ADC MIXR ADC2 Switch' on" + cset "name='Internal Mic Switch' on" + + ] + + DisableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='Stereo ADC MIXL ADC2 Switch' off" + cset "name='Stereo ADC MIXR ADC2 Switch' off" + cset "name='Internal Mic Switch' off" + + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/rt5640/EnableSeq.conf b/src/conf/ucm/rt5640/EnableSeq.conf new file mode 100644 index 00000000..62795fa6 --- /dev/null +++ b/src/conf/ucm/rt5640/EnableSeq.conf @@ -0,0 +1,41 @@ +# RT5640 default output routing +cset "name='OUT MIXL DAC L1 Switch' on" +cset "name='OUT MIXR DAC R1 Switch' on" + +# uncomment for loopback mic->speakers +# cset "name='SPOL MIX BST1 Switch' on" +# cset "name='SPOR MIX BST1 Switch' on" + +# uncomment for loopback playback -> capture +# cset "name='RECMIXL OUT MIXL Switch' on" +# cset "name='RECMIXR OUT MIXR Switch' on" + +# uncomment to enable swap between AIF1 and AIF2 +# warning: can only work with SSP0 firmware enabled +cset "name='SDI select' 0" +cset "name='DAI select' 0" +#cset "name='SDI select' 1" +#cset "name='DAI select' 1" + +cset "name='DAC2 Playback Switch' on" +cset "name='DIG MIXL DAC L2 Switch' on" + +# Input Configuration +cset "name='Stereo ADC1 Mux' ADC" +cset "name='Stereo ADC2 Mux' DMIC1" +cset "name='Mono ADC L1 Mux' 1" +cset "name='Mono ADC R1 Mux' 1" + +# 47=0dB, 0.375 dB/step, set it to 6 dB to help with soft mics +cset "name='ADC Capture Volume' 63" +# Set ADC Boost Gain to 3dB higher vals result in too much noise +cset "name='ADC Boost Gain' 1" +# Set IN1/IN3 internal mic boost to 8 (max) +# Set IN2 headset-mic boost to 1, headset mics are quite loud +cset "name='IN1 Boost' 8" +cset "name='IN2 Boost' 1" +cset "name='IN3 Boost' 8" + +cset "name='Internal Mic Switch' off" +cset "name='Headset Mic Switch' off" +cset "name='ADC Capture Switch' on" diff --git a/src/conf/ucm/rt5640/HeadPhones.conf b/src/conf/ucm/rt5640/HeadPhones.conf new file mode 100644 index 00000000..7d5948b4 --- /dev/null +++ b/src/conf/ucm/rt5640/HeadPhones.conf @@ -0,0 +1,41 @@ +SectionDevice."Headphones" { + Comment "Headphones playback" + + ConflictingDevice [ + "Speaker" + "MonoSpeaker" + ] + + EnableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='DAC MIXL INF1 Switch' on" + cset "name='DAC MIXR INF1 Switch' on" + cset "name='Stereo DAC MIXL DAC L1 Switch' on" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + set "name='Stereo DAC MIXL DAC L2 Switch' on" + cset "name='Stereo DAC MIXR DAC R2 Switch' on" + cset "name='HPO MIX HPVOL Switch' on" + cset "name='Headphone Switch' on" + cset "name='HP Channel Switch' on" + cset "name='Speaker Switch' off" + cset "name='HP L Playback Switch' on" + cset "name='HP R Playback Switch' on" + cset "name='HP Playback Volume' 29" + + ] + + DisableSequence [ + cdev "hw:bytcrrt5640" + cset "name='Headphone Switch' off" + cset "name='HP Channel Switch' off" + cset "name='HP L Playback Switch' off" + cset "name='HP R Playback Switch' off" + cset "name='HP Playback Volume' 0" + ] + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + } +} diff --git a/src/conf/ucm/rt5640/HeadsetMic.conf b/src/conf/ucm/rt5640/HeadsetMic.conf new file mode 100644 index 00000000..4d54eed0 --- /dev/null +++ b/src/conf/ucm/rt5640/HeadsetMic.conf @@ -0,0 +1,48 @@ +SectionDevice."HeadsetMic" { + Comment "Headset Mic capture" + + ConflictingDevice [ + "IN1-InternalMics" + "IN3-InternalMics" + "DigitalMics" + ] + + EnableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='Headset Mic Switch' on" + + cset "name='RECMIXL BST2 Switch' on" + cset "name='RECMIXR BST2 Switch' on" + + cset "name='Stereo ADC MIXL ADC1 Switch' on" + cset "name='Stereo ADC MIXR ADC1 Switch' on" + + cset "name='Mono ADC MIXL ADC1 Switch' on" + cset "name='Mono ADC MIXR ADC1 Switch' on" + + cset "name='Mono ADC Capture Switch' on" + ] + + DisableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='Stereo ADC MIXL ADC1 Switch' off" + cset "name='Stereo ADC MIXR ADC1 Switch' off" + + cset "name='RECMIXL BST2 Switch' off" + cset "name='RECMIXR BST2 Switch' off" + + cset "name='Headset Mic Switch' off" + + cset "name='Mono ADC MIXL ADC1 Switch' off" + cset "name='Mono ADC MIXR ADC1 Switch' off" + + cset "name='Mono ADC Capture Switch' off" + ] + + Value { + CaptureChannels 2 + JackControl "Headset Mic Jack" + } +} diff --git a/src/conf/ucm/rt5640/IN1-InternalMic.conf b/src/conf/ucm/rt5640/IN1-InternalMic.conf new file mode 100644 index 00000000..d18c21db --- /dev/null +++ b/src/conf/ucm/rt5640/IN1-InternalMic.conf @@ -0,0 +1,50 @@ +SectionDevice."IN1-InternalMics" { + Comment "Internal IN1 analog Mic capture" + + ConflictingDevice [ + "DigitalMics" + "HeadsetMic" + "IN3-InternalMics" + ] + + EnableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='Internal Mic Switch' on" + + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + cset "name='RECMIXL BST3 Switch' off" + cset "name='RECMIXR BST3 Switch' off" + + cset "name='Stereo ADC MIXL ADC1 Switch' on" + cset "name='Stereo ADC MIXR ADC1 Switch' on" + + cset "name='Mono ADC MIXL ADC1 Switch' on" + cset "name='Mono ADC MIXR ADC1 Switch' on" + + cset "name='Mono ADC Capture Switch' on" + ] + + DisableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='Stereo ADC MIXL ADC1 Switch' off" + cset "name='Stereo ADC MIXR ADC1 Switch' off" + + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + + cset "name='Internal Mic Switch' off" + + cset "name='Mono ADC MIXL ADC1 Switch' off" + cset "name='Mono ADC MIXR ADC1 Switch' off" + + cset "name='Mono ADC Capture Switch' off" + + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/rt5640/IN3-InternalMic.conf b/src/conf/ucm/rt5640/IN3-InternalMic.conf new file mode 100644 index 00000000..17d85229 --- /dev/null +++ b/src/conf/ucm/rt5640/IN3-InternalMic.conf @@ -0,0 +1,50 @@ +SectionDevice."IN3-InternalMics" { + Comment "Internal IN3 analog Mic capture" + + ConflictingDevice [ + "DigitalMics" + "HeadsetMic" + "IN1-InternalMics" + ] + + EnableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='Internal Mic Switch' on" + + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='RECMIXL BST3 Switch' on" + cset "name='RECMIXR BST3 Switch' on" + + cset "name='Stereo ADC MIXL ADC1 Switch' on" + cset "name='Stereo ADC MIXR ADC1 Switch' on" + + cset "name='Mono ADC MIXL ADC1 Switch' on" + cset "name='Mono ADC MIXR ADC1 Switch' on" + + cset "name='Mono ADC Capture Switch' on" + + ] + + DisableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='Stereo ADC MIXL ADC1 Switch' off" + cset "name='Stereo ADC MIXR ADC1 Switch' off" + + cset "name='RECMIXL BST3 Switch' off" + cset "name='RECMIXR BST3 Switch' off" + + cset "name='Internal Mic Switch' off" + + cset "name='Mono ADC MIXL ADC1 Switch' off" + cset "name='Mono ADC MIXR ADC1 Switch' off" + + cset "name='Mono ADC Capture Switch' off" + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/rt5640/Makefile.am b/src/conf/ucm/rt5640/Makefile.am new file mode 100644 index 00000000..a6fbdc57 --- /dev/null +++ b/src/conf/ucm/rt5640/Makefile.am @@ -0,0 +1,5 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/rt5640 +ucm_DATA = DigitalMics.conf EnableSeq.conf HeadPhones.conf HeadsetMic.conf \ + IN1-InternalMic.conf IN3-InternalMic.conf MonoSpeaker.conf Speaker.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/rt5640/MonoSpeaker.conf b/src/conf/ucm/rt5640/MonoSpeaker.conf new file mode 100644 index 00000000..0fb54030 --- /dev/null +++ b/src/conf/ucm/rt5640/MonoSpeaker.conf @@ -0,0 +1,45 @@ +SectionDevice."MonoSpeaker" { + Comment "MonoSpeaker playback" + + ConflictingDevice [ + "Headphones" + "Speaker" + ] + + EnableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='DAC MIXL INF1 Switch' on" + cset "name='DAC MIXR INF1 Switch' on" + cset "name='Stereo DAC MIXL DAC L1 Switch' on" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + cset "name='Stereo DAC MIXL DAC L2 Switch' on" + cset "name='Stereo DAC MIXR DAC R2 Switch' on" + cset "name='SPK MIXL DAC L1 Switch' on" + cset "name='SPK MIXR DAC R1 Switch' on" + cset "name='SPOL MIX SPKVOL L Switch' on" +# for mono speaker we apply left on right +# cset "name='SPOR MIX SPKVOL R Switch' on" + cset "name='SPOL MIX SPKVOL R Switch' on" + cset "name='Speaker Switch' on" + cset "name='Speaker Channel Switch' on" + cset "name='Headphone Switch' off" + cset "name='Speaker L Playback Switch' on" + cset "name='Speaker R Playback Switch' on" + cset "name='Speaker Playback Volume' 35" + + ] + + DisableSequence [ + cdev "hw:bytcrrt5640" + cset "name='Speaker Switch' off" + cset "name='Speaker Channel Switch' off" + cset "name='Speaker L Playback Switch' off" + cset "name='Speaker R Playback Switch' off" + cset "name='Speaker Playback Volume' 0" + ] + + Value { + PlaybackChannels "2" + } +} diff --git a/src/conf/ucm/rt5640/Speaker.conf b/src/conf/ucm/rt5640/Speaker.conf new file mode 100644 index 00000000..b745ed06 --- /dev/null +++ b/src/conf/ucm/rt5640/Speaker.conf @@ -0,0 +1,45 @@ +SectionDevice."Speaker" { + Comment "Speaker playback" + + ConflictingDevice [ + "Headphones" + "MonoSpeaker" + ] + + EnableSequence [ + cdev "hw:bytcrrt5640" + + cset "name='DAC MIXL INF1 Switch' on" + cset "name='DAC MIXR INF1 Switch' on" + cset "name='Stereo DAC MIXL DAC L1 Switch' on" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + cset "name='Stereo DAC MIXL DAC L2 Switch' on" + cset "name='Stereo DAC MIXR DAC R2 Switch' on" + cset "name='SPK MIXL DAC L1 Switch' on" + cset "name='SPK MIXR DAC R1 Switch' on" + cset "name='SPOL MIX SPKVOL L Switch' on" + cset "name='SPOR MIX SPKVOL R Switch' on" + # undo MonoSpeaker mixing of right channel to left speaker + cset "name='SPOL MIX SPKVOL R Switch' off" + cset "name='Speaker Switch' on" + cset "name='Speaker Channel Switch' on" + cset "name='Headphone Switch' off" + cset "name='Speaker L Playback Switch' on" + cset "name='Speaker R Playback Switch' on" + cset "name='Speaker Playback Volume' 35" + + ] + + DisableSequence [ + cdev "hw:bytcrrt5640" + cset "name='Speaker Switch' off" + cset "name='Speaker Channel Switch' off" + cset "name='Speaker L Playback Switch' off" + cset "name='Speaker R Playback Switch' off" + cset "name='Speaker Playback Volume' 0" + ] + + Value { + PlaybackChannels "2" + } +} -- 2.17.0

1 1

[alsa-devel] Submitting a patch for tinycompress v1.1.5
by Anoush Khazeni 19 May '18

19 May '18

Submitting a patch to add the missing closing bracket for the extern "C" directive in "tinycompress.h" from release v1.1.5 tar file "tinycompress-1.1.5.tar.bz2"

1 0

[alsa-devel] [PATCH] ASoC: pcm512x: Add ACPI support
by Pierre-Louis Bossart 18 May '18

18 May '18

HID is assumed to be made of TI PCI ID (0x104C) + part number, so all four 104C5121, 104C5122, 104C5141 104C5142 are valid. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/pcm512x-i2c.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index 5f9c069569d5..0fe5ced841a3 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -17,6 +17,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/acpi.h> #include "pcm512x.h" @@ -52,6 +53,7 @@ static const struct i2c_device_id pcm512x_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); +#if defined(CONFIG_OF) static const struct of_device_id pcm512x_of_match[] = { { .compatible = "ti,pcm5121", }, { .compatible = "ti,pcm5122", }, @@ -60,6 +62,18 @@ static const struct of_device_id pcm512x_of_match[] = { { } }; MODULE_DEVICE_TABLE(of, pcm512x_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id pcm512x_acpi_match[] = { + { "104C5121", 0 }, + { "104C5122", 0 }, + { "104C5141", 0 }, + { "104C5142", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, pcm512x_acpi_match); +#endif static struct i2c_driver pcm512x_i2c_driver = { .probe = pcm512x_i2c_probe, @@ -67,7 +81,8 @@ static struct i2c_driver pcm512x_i2c_driver = { .id_table = pcm512x_i2c_id, .driver = { .name = "pcm512x", - .of_match_table = pcm512x_of_match, + .of_match_table = of_match_ptr(pcm512x_of_match), + .acpi_match_table = ACPI_PTR(pcm512x_acpi_match), .pm = &pcm512x_pm_ops, }, }; -- 2.14.1

4 4

[alsa-devel] [PATCH 00/19] ASoC: rt5640: Add jack-detect and button-press support
by Hans de Goede 18 May '18

18 May '18

Hi All, This series has been long in the making, but it is finally ready now. This series adds jack-detect support for rt5640 using devices. This is modelled after the recent rt5651 codec- and bytcr-rt5651 machine- driver jack-detect changes (which were actually based on a WIP version of this series). As discussed on the list already unfortunately there are no really good defaults which work everywhere, so patch 18 adds quirks for 10 devices, I've decided to do this in one go rather then split this into 10 patches. Patch 19 makes the bytcr-rt5640 set a longname based on the speaker and input-map quirks, so that userspace can pick a correct UCM profile based on the longname with working jack-detect based input / output switching. I've matching patches adding UCM profiles for this to alsa-lib here: https://github.com/jwrdegoede/alsa-lib/commits/master I'm waiting with submitting those upstream until patch 19 is accepted for merging. Regards, Hans

4 48

[alsa-devel] [PATCH] ASoC: ssm2305: fix header layout
by Marco Felsch 18 May '18

18 May '18

Make C-header and SPDX-License-Identifier header uniform. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- sound/soc/codecs/ssm2305.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ssm2305.c b/sound/soc/codecs/ssm2305.c index 39d8b01cd852..2968959c4b75 100644 --- a/sound/soc/codecs/ssm2305.c +++ b/sound/soc/codecs/ssm2305.c @@ -1,9 +1,9 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * Analog Devices SSM2305 Amplifier Driver - * - * Copyright (C) 2018 Pengutronix, Marco Felsch <kernel(a)pengutronix.de> - */ +// +// Analog Devices SSM2305 Amplifier Driver +// +// Copyright (C) 2018 Pengutronix, Marco Felsch <kernel(a)pengutronix.de> +// #include <linux/gpio/consumer.h> #include <linux/module.h> -- 2.17.0

2 1

[alsa-devel] [PATCH v1] ASoC: rt5677: Convert I2C driver to ->probe_new()
by Andy Shevchenko 18 May '18

18 May '18

There is no platform code that uses i2c module table. Remove it altogether and adjust ->probe() to be ->probe_new(). Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/codecs/rt5677.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index bc1a23dd7c2d..8a0181a2db08 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5006,13 +5006,6 @@ static const struct regmap_config rt5677_regmap = { .num_ranges = ARRAY_SIZE(rt5677_ranges), }; -static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", RT5677 }, - { "rt5676", RT5676 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); - static const struct of_device_id rt5677_of_match[] = { { .compatible = "realtek,rt5677", RT5677 }, { } @@ -5130,8 +5123,7 @@ static void rt5677_free_irq(struct i2c_client *i2c) regmap_del_irq_chip(i2c->irq, rt5677->irq_data); } -static int rt5677_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int rt5677_i2c_probe(struct i2c_client *i2c) { struct rt5677_priv *rt5677; int ret; @@ -5278,9 +5270,8 @@ static struct i2c_driver rt5677_i2c_driver = { .of_match_table = rt5677_of_match, .acpi_match_table = ACPI_PTR(rt5677_acpi_match), }, - .probe = rt5677_i2c_probe, + .probe_new = rt5677_i2c_probe, .remove = rt5677_i2c_remove, - .id_table = rt5677_i2c_id, }; module_i2c_driver(rt5677_i2c_driver); -- 2.17.0

2 1

[alsa-devel] sound: latency and synchronization measurements
by Muni Sekhar 18 May '18

18 May '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, I am trying to characterize the audio synchronization between two systems. There is a mechanism to route audio between two systems, we run ‘aplay’ at one system and ‘arecord’ in other system. We need to measure the time difference when audio sample is out at one port and arriving at another port in different system. I plan to add printk’s in “struct snd_pcm_ops” call backs, but does these call backs tells when it received first and last audio frames\samples? Or else please suggest any other best method. -- Thanks, Sekhar

3 4

[alsa-devel] [PATCH] ALSA: emu10k1: Rate-limit error messages about page errors
by Takashi Iwai 18 May '18

18 May '18

The error messages at sanity checks of memory pages tend to repeat too many times once when it hits, and without the rate limit, it may flood and become unreadable. Replace such messages with the *_ratelimited() variant. Bugzilla: http://bugzilla.opensuse.org/show_bug.cgi?id=1093027 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/memory.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 5865f3b90b34..dbc7d8d0e1c4 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -248,13 +248,13 @@ search_empty(struct snd_emu10k1 *emu, int size) static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr) { if (addr & ~emu->dma_mask) { - dev_err(emu->card->dev, + dev_err_ratelimited(emu->card->dev, "max memory size is 0x%lx (addr = 0x%lx)!!\n", emu->dma_mask, (unsigned long)addr); return 0; } if (addr & (EMUPAGESIZE-1)) { - dev_err(emu->card->dev, "page is not aligned\n"); + dev_err_ratelimited(emu->card->dev, "page is not aligned\n"); return 0; } return 1; @@ -345,7 +345,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst else addr = snd_pcm_sgbuf_get_addr(substream, ofs); if (! is_valid_page(emu, addr)) { - dev_err(emu->card->dev, + dev_err_ratelimited(emu->card->dev, "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); return NULL; -- 2.16.3

2 2

[alsa-devel] [PATCH] ALSA: dice: fix a bounds check in snd_dice_detect_tcelectronic_formats()
by Dan Carpenter 18 May '18

18 May '18

The "entry" pointer is always non-NULL so this test for out of bounds won't work. Fixes: f1f0f330b1d0 ("ALSA: dice: add parameters of stream formats for models produced by TC Electronic") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c index af8203b9d1a6..e134a5110c6c 100644 --- a/sound/firewire/dice/dice-tcelectronic.c +++ b/sound/firewire/dice/dice-tcelectronic.c @@ -75,13 +75,12 @@ int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice) } } - entry = NULL; for (i = 0; i < ARRAY_SIZE(entries); ++i) { entry = entries + i; if (entry->model_id == model_id) break; } - if (!entry) + if (i == ARRAY_SIZE(entries)) return -ENODEV; memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs,

3 2

[alsa-devel] [PATCH] ASoC: wm8904: fix spelling mistake: "Caputure" -> "Capture"
by Colin King 18 May '18

18 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in SOC_ENUM literal string Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 20fdcae06c6b..f13ef334c0d7 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -596,7 +596,7 @@ static const struct snd_kcontrol_new wm8904_adc_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8904_ADC_DIGITAL_VOLUME_LEFT, WM8904_ADC_DIGITAL_VOLUME_RIGHT, 1, 119, 0, digital_tlv), -SOC_ENUM("Left Caputure Mode", lin_mode), +SOC_ENUM("Left Capture Mode", lin_mode), SOC_ENUM("Right Capture Mode", rin_mode), /* No TLV since it depends on mode */ -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: wm2200, wm5100: fix spelling mistake: "Coefficeints" -> "Coefficients"
by Colin King 18 May '18

18 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistakes in SND_SOC_BYTES literal strings Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wm2200.c | 4 ++-- sound/soc/codecs/wm5100.c | 8 ++++---- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index d5f4bbf27182..3663b9fd4d65 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1155,8 +1155,8 @@ SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_3L, SND_SOC_BYTES_MASK("EQL Coefficients", WM2200_EQL_1, 20, WM2200_EQL_ENA), SND_SOC_BYTES_MASK("EQR Coefficients", WM2200_EQR_1, 20, WM2200_EQR_ENA), -SND_SOC_BYTES("LHPF1 Coefficeints", WM2200_HPLPF1_2, 1), -SND_SOC_BYTES("LHPF2 Coefficeints", WM2200_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF1 Coefficients", WM2200_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", WM2200_HPLPF2_2, 1), SOC_SINGLE("OUT1 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_1L, WM2200_OUT1_OSR_SHIFT, 1, 0), diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 87f9a99ce978..ba89d9d711f7 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -573,10 +573,10 @@ SND_SOC_BYTES_MASK("EQ4 Coefficients", WM5100_EQ4_1, 20, WM5100_EQ4_ENA), SND_SOC_BYTES_MASK("DRC Coefficients", WM5100_DRC1_CTRL1, 5, WM5100_DRCL_ENA | WM5100_DRCR_ENA), -SND_SOC_BYTES("LHPF1 Coefficeints", WM5100_HPLPF1_2, 1), -SND_SOC_BYTES("LHPF2 Coefficeints", WM5100_HPLPF2_2, 1), -SND_SOC_BYTES("LHPF3 Coefficeints", WM5100_HPLPF3_2, 1), -SND_SOC_BYTES("LHPF4 Coefficeints", WM5100_HPLPF4_2, 1), +SND_SOC_BYTES("LHPF1 Coefficients", WM5100_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", WM5100_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", WM5100_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", WM5100_HPLPF4_2, 1), SOC_SINGLE("HPOUT1 High Performance Switch", WM5100_OUT_VOLUME_1L, WM5100_OUT1_OSR_SHIFT, 1, 0), -- 2.17.0

2 1

[alsa-devel] [PATCH v4] ASoC: ssm2305: Add amplifier driver
by Marco Felsch 18 May '18

18 May '18

The ssm2305 is a simple Class-D audio amplifier. A application can turn on/off the device by a gpio. It's also possible to hardwire the shutdown pin. Tested on a i.MX6 based custom board. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Reviewed-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Notes (changelog): v2: - fix copyright information - rename shutdown gpio and add some more documentation - mark shutdown gpio as required - rm registration information v3: - fix error handling v4: - fix license header .../devicetree/bindings/sound/adi,ssm2305.txt | 14 +++ sound/soc/codecs/Kconfig | 7 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2305.c | 104 ++++++++++++++++++ 4 files changed, 127 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2305.txt create mode 100644 sound/soc/codecs/ssm2305.c diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2305.txt b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt new file mode 100644 index 000000000000..a9c9d83c8a30 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt @@ -0,0 +1,14 @@ +Analog Devices SSM2305 Speaker Amplifier +======================================== + +Required properties: + - compatible : "adi,ssm2305" + - shutdown-gpios : The gpio connected to the shutdown pin. + The gpio signal is ACTIVE_LOW. + +Example: + +ssm2305: analog-amplifier { + compatible = "adi,ssm2305"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9548f63ca531..d81b0cb291a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -142,6 +142,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF + select SND_SOC_SSM2305 select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C @@ -883,6 +884,12 @@ config SND_SOC_SIRF_AUDIO_CODEC config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SSM2305 + tristate "Analog Devices SSM2305 Class-D Amplifier" + help + Enable support for Analog Devices SSM2305 filterless + high-efficiency mono Class-D audio power amplifiers. + config SND_SOC_SSM2518 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e849d1495308..7d75ac8f6716 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -153,6 +153,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o @@ -404,6 +405,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o +obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o diff --git a/sound/soc/codecs/ssm2305.c b/sound/soc/codecs/ssm2305.c new file mode 100644 index 000000000000..d17864c3e2a9 --- /dev/null +++ b/sound/soc/codecs/ssm2305.c @@ -0,0 +1,104 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices SSM2305 Amplifier Driver +// +// Copyright (C) 2018 Pengutronix, Marco Felsch <kernel(a)pengutronix.de> +// + +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <sound/soc.h> + +#define DRV_NAME "ssm2305" + +struct ssm2305 { + /* shutdown gpio */ + struct gpio_desc *gpiod_shutdown; +}; + +static int ssm2305_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kctrl, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct ssm2305 *data = snd_soc_component_get_drvdata(c); + + gpiod_set_value_cansleep(data->gpiod_shutdown, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget ssm2305_dapm_widgets[] = { + /* Stereo input/output */ + SND_SOC_DAPM_INPUT("L_IN"), + SND_SOC_DAPM_INPUT("R_IN"), + SND_SOC_DAPM_OUTPUT("L_OUT"), + SND_SOC_DAPM_OUTPUT("R_OUT"), + + SND_SOC_DAPM_SUPPLY("Power", SND_SOC_NOPM, 0, 0, ssm2305_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route ssm2305_dapm_routes[] = { + { "L_OUT", NULL, "L_IN" }, + { "R_OUT", NULL, "R_IN" }, + { "L_IN", NULL, "Power" }, + { "R_IN", NULL, "Power" }, +}; + +static const struct snd_soc_component_driver ssm2305_component_driver = { + .dapm_widgets = ssm2305_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2305_dapm_widgets), + .dapm_routes = ssm2305_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2305_dapm_routes), +}; + +static int ssm2305_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ssm2305 *priv; + int err; + + /* Allocate the private data */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + + /* Get shutdown gpio */ + priv->gpiod_shutdown = devm_gpiod_get(dev, "shutdown", + GPIOD_OUT_LOW); + if (IS_ERR(priv->gpiod_shutdown)) { + err = PTR_ERR(priv->gpiod_shutdown); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'shutdown' gpio: %d\n", + err); + return err; + } + + return devm_snd_soc_register_component(dev, &ssm2305_component_driver, + NULL, 0); +} + +#ifdef CONFIG_OF +static const struct of_device_id ssm2305_of_match[] = { + { .compatible = "adi,ssm2305", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2305_of_match); +#endif + +static struct platform_driver ssm2305_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(ssm2305_of_match), + }, + .probe = ssm2305_probe, +}; + +module_platform_driver(ssm2305_driver); + +MODULE_DESCRIPTION("ASoC SSM2305 amplifier driver"); +MODULE_AUTHOR("Marco Felsch <m.felsch(a)pengutronix.de>"); +MODULE_LICENSE("GPL v2"); -- 2.17.0

2 1

[alsa-devel] [PATCH v3] ASoC: ssm2305: Add amplifier driver
by Marco Felsch 18 May '18

18 May '18

The ssm2305 is a simple Class-D audio amplifier. A application can turn on/off the device by a gpio. It's also possible to hardwire the shutdown pin. Tested on a i.MX6 based custom board. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- Notes (changelog): v2: - fix copyright information - rename shutdown gpio and add some more documentation - mark shutdown gpio as required - rm registration information v3: - fix error handling .../devicetree/bindings/sound/adi,ssm2305.txt | 14 +++ sound/soc/codecs/Kconfig | 7 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2305.c | 104 ++++++++++++++++++ 4 files changed, 127 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2305.txt create mode 100644 sound/soc/codecs/ssm2305.c diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2305.txt b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt new file mode 100644 index 000000000000..a9c9d83c8a30 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt @@ -0,0 +1,14 @@ +Analog Devices SSM2305 Speaker Amplifier +======================================== + +Required properties: + - compatible : "adi,ssm2305" + - shutdown-gpios : The gpio connected to the shutdown pin. + The gpio signal is ACTIVE_LOW. + +Example: + +ssm2305: analog-amplifier { + compatible = "adi,ssm2305"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9548f63ca531..d81b0cb291a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -142,6 +142,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF + select SND_SOC_SSM2305 select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C @@ -883,6 +884,12 @@ config SND_SOC_SIRF_AUDIO_CODEC config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SSM2305 + tristate "Analog Devices SSM2305 Class-D Amplifier" + help + Enable support for Analog Devices SSM2305 filterless + high-efficiency mono Class-D audio power amplifiers. + config SND_SOC_SSM2518 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e849d1495308..7d75ac8f6716 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -153,6 +153,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o @@ -404,6 +405,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o +obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o diff --git a/sound/soc/codecs/ssm2305.c b/sound/soc/codecs/ssm2305.c new file mode 100644 index 000000000000..39d8b01cd852 --- /dev/null +++ b/sound/soc/codecs/ssm2305.c @@ -0,0 +1,104 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices SSM2305 Amplifier Driver + * + * Copyright (C) 2018 Pengutronix, Marco Felsch <kernel(a)pengutronix.de> + */ + +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <sound/soc.h> + +#define DRV_NAME "ssm2305" + +struct ssm2305 { + /* shutdown gpio */ + struct gpio_desc *gpiod_shutdown; +}; + +static int ssm2305_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kctrl, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct ssm2305 *data = snd_soc_component_get_drvdata(c); + + gpiod_set_value_cansleep(data->gpiod_shutdown, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget ssm2305_dapm_widgets[] = { + /* Stereo input/output */ + SND_SOC_DAPM_INPUT("L_IN"), + SND_SOC_DAPM_INPUT("R_IN"), + SND_SOC_DAPM_OUTPUT("L_OUT"), + SND_SOC_DAPM_OUTPUT("R_OUT"), + + SND_SOC_DAPM_SUPPLY("Power", SND_SOC_NOPM, 0, 0, ssm2305_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route ssm2305_dapm_routes[] = { + { "L_OUT", NULL, "L_IN" }, + { "R_OUT", NULL, "R_IN" }, + { "L_IN", NULL, "Power" }, + { "R_IN", NULL, "Power" }, +}; + +static const struct snd_soc_component_driver ssm2305_component_driver = { + .dapm_widgets = ssm2305_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2305_dapm_widgets), + .dapm_routes = ssm2305_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2305_dapm_routes), +}; + +static int ssm2305_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ssm2305 *priv; + int err; + + /* Allocate the private data */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + + /* Get shutdown gpio */ + priv->gpiod_shutdown = devm_gpiod_get(dev, "shutdown", + GPIOD_OUT_LOW); + if (IS_ERR(priv->gpiod_shutdown)) { + err = PTR_ERR(priv->gpiod_shutdown); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'shutdown' gpio: %d\n", + err); + return err; + } + + return devm_snd_soc_register_component(dev, &ssm2305_component_driver, + NULL, 0); +} + +#ifdef CONFIG_OF +static const struct of_device_id ssm2305_of_match[] = { + { .compatible = "adi,ssm2305", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2305_of_match); +#endif + +static struct platform_driver ssm2305_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(ssm2305_of_match), + }, + .probe = ssm2305_probe, +}; + +module_platform_driver(ssm2305_driver); + +MODULE_DESCRIPTION("ASoC SSM2305 amplifier driver"); +MODULE_AUTHOR("Marco Felsch <m.felsch(a)pengutronix.de>"); +MODULE_LICENSE("GPL v2"); -- 2.17.0

3 4

[alsa-devel] [PATCH] ALSA: timer: Fix pause event notification
by Ben Hutchings 18 May '18

18 May '18

Commit f65e0d299807 ("ALSA: timer: Call notifier in the same spinlock") combined the start/continue and stop/pause functions, and in doing so changed the event code for the pause case to SNDRV_TIMER_EVENT_CONTINUE. Change it back to SNDRV_TIMER_EVENT_PAUSE. Fixes: f65e0d299807 ("ALSA: timer: Call notifier in the same spinlock") Signed-off-by: Ben Hutchings <ben.hutchings(a)codethink.co.uk> Cc: stable(a)vger.kernel.org --- sound/core/timer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index dc87728c5b74..0ddcae495838 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -592,7 +592,7 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) else timeri->flags |= SNDRV_TIMER_IFLG_PAUSED; snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : - SNDRV_TIMER_EVENT_CONTINUE); + SNDRV_TIMER_EVENT_PAUSE); unlock: spin_unlock_irqrestore(&timer->lock, flags); return result; @@ -614,7 +614,7 @@ static int snd_timer_stop_slave(struct snd_timer_instance *timeri, bool stop) list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : - SNDRV_TIMER_EVENT_CONTINUE); + SNDRV_TIMER_EVENT_PAUSE); spin_unlock(&timeri->timer->lock); } spin_unlock_irqrestore(&slave_active_lock, flags); -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: usb: stream: fix potential memory leak during uac3 interface parsing
by Ruslan Bilovol 18 May '18

18 May '18

UAC3 channel map is created during interface parsing, and in some cases was not freed in failure paths. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> --- sound/usb/stream.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index bce3152..d16e1c2 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -982,13 +982,16 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, dev_err(&dev->dev, "%u:%d : bogus bTerminalLink %d\n", iface_no, altno, as->bTerminalLink); + kfree(chmap); return NULL; found_clock: fp = audio_format_alloc_init(chip, alts, UAC_VERSION_3, iface_no, altset_idx, altno, num_channels, clock); - if (!fp) + if (!fp) { + kfree(chmap); return ERR_PTR(-ENOMEM); + } fp->chmap = chmap; @@ -1009,6 +1012,7 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, iface_no); /* ok, let's parse further... */ if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) { + kfree(fp->chmap); kfree(fp->rate_table); kfree(fp); return NULL; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Boards: Add CNL RT274 I2S machine driver
by Guneshwor Singh 17 May '18

17 May '18

Add CNL I2S machine driver using Realtek RT274 codec in I2S mode configured to ssp0. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/boards/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cnl_rt274.c | 250 +++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/boards/cnl_rt274.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6f754708a48c..29066f0b744f 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -262,4 +262,17 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_CNL_RT274_MACH + tristate "ASoC Audio driver for Cannonlake with RT274 I2S mode" + depends on X86 && ACPI && I2C + select SND_SOC_INTEL_SST + depends on SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT274 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Cannonlake platform + with RT274 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 69d2dfaeb00c..a7a832c072ca 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -19,6 +19,7 @@ snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-cnl-rt274-objs := cnl_rt274.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -40,3 +41,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_CNL_RT274_MACH) += snd-soc-cnl-rt274.o diff --git a/sound/soc/intel/boards/cnl_rt274.c b/sound/soc/intel/boards/cnl_rt274.c new file mode 100644 index 000000000000..19458fb50f55 --- /dev/null +++ b/sound/soc/intel/boards/cnl_rt274.c @@ -0,0 +1,250 @@ +/* + * cnl_rt274.c - ASOC Machine driver for CNL + * + * Copyright (C) 2016-17 Intel Corp + * Author: Guneshwor Singh <guneshwor.o.singh(a)intel.com> + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/rt274.h" + +#define CNL_FREQ_OUT 24000000 +#define CNL_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai = + snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + int ret, ratio = 100; + + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, + CNL_FREQ_OUT, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + if (ret) { + dev_err(codec_dai->dev, + "set bclk ratio failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + CNL_BE_FIXUP_RATE * ratio, + CNL_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, + "enable PLL2 failed: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_jack cnl_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin cnl_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new cnl_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget cnl_rt274_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + cnl_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static int cnl_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static const struct snd_soc_dapm_route cnl_map[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + {"DMic", NULL, "SoC DMIC"}, + {"DMIC01 Rx", NULL, "Capture"}, + {"dmic01_hifi", NULL, "DMIC01 Rx"}, + + {"AIF1 Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "codec1_out"}, + {"ssp0 Tx", NULL, "codec0_out"}, + + {"ssp0 Rx", NULL, "AIF1 Capture"}, + {"codec0_in", NULL, "ssp0 Rx"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"Mic Jack", NULL, "Platform Clock"}, +}; + +static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + int ret; + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &cnl_headset, + cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins)); + if (ret) + return ret; + + ret = snd_soc_codec_set_jack(codec, &cnl_headset, NULL); + if (ret) + return ret; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int cnl_be_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = CNL_BE_FIXUP_RATE; + channels->min = channels->max = 2; + snd_mask_none(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT)); + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static struct snd_soc_dai_link cnl_rt274_dailink[] = { + { + .name = "SSP0-Codec", + .cpu_dai_name = "SSP0 Pin", + .codec_name = "i2c-INT34C2:00", + .codec_dai_name = "rt274-aif1", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_be_fixup, + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = cnl_rt274_init, + }, + { + .name = "dmic01", + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_dmic_fixup, + .no_pcm = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + }, +}; + +static int +cnl_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + link->platform_name = "0000:00:1f.3"; + link->nonatomic = 1; + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cnl = { + .name = "cnl-audio", + .dai_link = cnl_rt274_dailink, + .num_links = ARRAY_SIZE(cnl_rt274_dailink), + .dapm_widgets = cnl_rt274_widgets, + .num_dapm_widgets = ARRAY_SIZE(cnl_rt274_widgets), + .dapm_routes = cnl_map, + .num_dapm_routes = ARRAY_SIZE(cnl_map), + .controls = cnl_controls, + .num_controls = ARRAY_SIZE(cnl_controls), + .add_dai_link = cnl_add_dai_link, + .fully_routed = true, +}; + +static int snd_cnl_rt274_probe(struct platform_device *pdev) +{ + snd_soc_card_cnl.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cnl); +} + +static struct platform_driver snd_cnl_rt274_driver = { + .driver = { + .name = "cnl_rt274", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cnl_rt274_probe, +}; + +module_platform_driver(snd_cnl_rt274_driver); + +MODULE_AUTHOR("Guneshwor Singh <guneshwor.o.singh(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cnl_rt274"); -- 2.15.0

4 6

[alsa-devel] Applied "ASoC: ssm2305: Add amplifier driver" to the asoc tree
by Mark Brown 17 May '18

17 May '18

The patch ASoC: ssm2305: Add amplifier driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 74f24d8728ef12276d58e3d73283cc1d76db7507 Mon Sep 17 00:00:00 2001 From: Marco Felsch <m.felsch(a)pengutronix.de> Date: Thu, 17 May 2018 15:55:18 +0200 Subject: [PATCH] ASoC: ssm2305: Add amplifier driver The ssm2305 is a simple Class-D audio amplifier. A application can turn on/off the device by a gpio. It's also possible to hardwire the shutdown pin. Tested on a i.MX6 based custom board. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Reviewed-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/adi,ssm2305.txt | 14 +++ sound/soc/codecs/Kconfig | 7 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2305.c | 104 ++++++++++++++++++ 4 files changed, 127 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2305.txt create mode 100644 sound/soc/codecs/ssm2305.c diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2305.txt b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt new file mode 100644 index 000000000000..a9c9d83c8a30 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt @@ -0,0 +1,14 @@ +Analog Devices SSM2305 Speaker Amplifier +======================================== + +Required properties: + - compatible : "adi,ssm2305" + - shutdown-gpios : The gpio connected to the shutdown pin. + The gpio signal is ACTIVE_LOW. + +Example: + +ssm2305: analog-amplifier { + compatible = "adi,ssm2305"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 83c51c5a4d08..a3e7c44c6586 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -145,6 +145,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF + select SND_SOC_SSM2305 select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C @@ -896,6 +897,12 @@ config SND_SOC_SIRF_AUDIO_CODEC config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SSM2305 + tristate "Analog Devices SSM2305 Class-D Amplifier" + help + Enable support for Analog Devices SSM2305 filterless + high-efficiency mono Class-D audio power amplifiers. + config SND_SOC_SSM2518 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1ac49b9abec6..4c85f3391705 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -156,6 +156,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o @@ -410,6 +411,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o +obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o diff --git a/sound/soc/codecs/ssm2305.c b/sound/soc/codecs/ssm2305.c new file mode 100644 index 000000000000..39d8b01cd852 --- /dev/null +++ b/sound/soc/codecs/ssm2305.c @@ -0,0 +1,104 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices SSM2305 Amplifier Driver + * + * Copyright (C) 2018 Pengutronix, Marco Felsch <kernel(a)pengutronix.de> + */ + +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <sound/soc.h> + +#define DRV_NAME "ssm2305" + +struct ssm2305 { + /* shutdown gpio */ + struct gpio_desc *gpiod_shutdown; +}; + +static int ssm2305_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kctrl, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct ssm2305 *data = snd_soc_component_get_drvdata(c); + + gpiod_set_value_cansleep(data->gpiod_shutdown, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget ssm2305_dapm_widgets[] = { + /* Stereo input/output */ + SND_SOC_DAPM_INPUT("L_IN"), + SND_SOC_DAPM_INPUT("R_IN"), + SND_SOC_DAPM_OUTPUT("L_OUT"), + SND_SOC_DAPM_OUTPUT("R_OUT"), + + SND_SOC_DAPM_SUPPLY("Power", SND_SOC_NOPM, 0, 0, ssm2305_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route ssm2305_dapm_routes[] = { + { "L_OUT", NULL, "L_IN" }, + { "R_OUT", NULL, "R_IN" }, + { "L_IN", NULL, "Power" }, + { "R_IN", NULL, "Power" }, +}; + +static const struct snd_soc_component_driver ssm2305_component_driver = { + .dapm_widgets = ssm2305_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2305_dapm_widgets), + .dapm_routes = ssm2305_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2305_dapm_routes), +}; + +static int ssm2305_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ssm2305 *priv; + int err; + + /* Allocate the private data */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + + /* Get shutdown gpio */ + priv->gpiod_shutdown = devm_gpiod_get(dev, "shutdown", + GPIOD_OUT_LOW); + if (IS_ERR(priv->gpiod_shutdown)) { + err = PTR_ERR(priv->gpiod_shutdown); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'shutdown' gpio: %d\n", + err); + return err; + } + + return devm_snd_soc_register_component(dev, &ssm2305_component_driver, + NULL, 0); +} + +#ifdef CONFIG_OF +static const struct of_device_id ssm2305_of_match[] = { + { .compatible = "adi,ssm2305", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2305_of_match); +#endif + +static struct platform_driver ssm2305_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(ssm2305_of_match), + }, + .probe = ssm2305_probe, +}; + +module_platform_driver(ssm2305_driver); + +MODULE_DESCRIPTION("ASoC SSM2305 amplifier driver"); +MODULE_AUTHOR("Marco Felsch <m.felsch(a)pengutronix.de>"); +MODULE_LICENSE("GPL v2"); -- 2.17.0

1 0

[alsa-devel] [PATCH] ASoC: rt5670: improve PLL function's stability
by Bard Liao 17 May '18

17 May '18

Set PR-38 register to 0x1fe1 will make PLL function more stable. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index dc7df33..732ef92 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -71,7 +71,7 @@ static const struct regmap_range_cfg rt5670_ranges[] = { static const struct reg_sequence init_list[] = { { RT5670_PR_BASE + 0x14, 0x9a8a }, - { RT5670_PR_BASE + 0x38, 0x3ba1 }, + { RT5670_PR_BASE + 0x38, 0x1fe1 }, { RT5670_PR_BASE + 0x3d, 0x3640 }, { 0x8a, 0x0123 }, }; -- 2.7.4

2 1

[alsa-devel] [PATCH] soc: qcom: apr: fix invalid msg_type check
by Srinivas Kandagatla 17 May '18

17 May '18

Removed invalid msg_type check. This also fixes below static checker warning: apr.c:95:35: warning: comparison is always true due to limited range of data type [-Wtype-limits] warn: always true condition '(msg_type != 69864) => (0-u16max != 69864)' Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soc/qcom/apr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soc/qcom/apr.c b/drivers/soc/qcom/apr.c index 97f3622da535..57af8a537332 100644 --- a/drivers/soc/qcom/apr.c +++ b/drivers/soc/qcom/apr.c @@ -92,7 +92,7 @@ static int apr_callback(struct rpmsg_device *rpdev, void *buf, } msg_type = APR_HDR_FIELD_MT(hdr->hdr_field); - if (msg_type >= APR_MSG_TYPE_MAX && msg_type != APR_BASIC_RSP_RESULT) { + if (msg_type >= APR_MSG_TYPE_MAX) { dev_err(apr->dev, "APR: Wrong message type: %d\n", msg_type); return -EINVAL; } -- 2.16.2

2 1

[alsa-devel] [PATCH] MAINTAINERS: update sound/soc/intel maintainers
by Pierre-Louis Bossart 17 May '18

17 May '18

The information for Intel SoC drivers was not updated for several years. Add myself, Liam and Keyon (Jie) as maintainers to get notified of contributions and bug reports. As discussed with Mark and Takashi, I'll also monitor alsa-devel and ack Intel patches as necessary. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- MAINTAINERS | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index e83cee26e195..d70add4c253b 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7030,14 +7030,13 @@ L: linux-fbdev(a)vger.kernel.org S: Maintained F: drivers/video/fbdev/i810/ -INTEL ASoC BDW/HSW DRIVERS +INTEL ASoC DRIVERS +M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> +M: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> M: Jie Yang <yang.jie(a)linux.intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported -F: sound/soc/intel/common/sst-dsp* -F: sound/soc/intel/common/sst-firmware.c -F: sound/soc/intel/boards/broadwell.c -F: sound/soc/intel/haswell/ +F: sound/soc/intel/ INTEL C600 SERIES SAS CONTROLLER DRIVER M: Intel SCU Linux support <intel-linux-scu(a)intel.com> -- 2.14.1

3 2

[alsa-devel] [PATCH v2 0/2] ASoC: omap: sdma-pcm: modpost warning fix and include cleanup
by Peter Ujfalusi 17 May '18

17 May '18

Hi, Changes since v1: - Changed the MODULE_LICENSE from "GPL" to "GPL v2" to match the SPDX line. Since the sdma-pcm can be built as module it must have MODULE_LICENSE defined, whcih I have missed: After merging the sound-asoc tree, today's linux-next build (x86_64 allmodconfig) produced this warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o see include/linux/module.h for more information WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o see include/linux/module.h for more information The first patch also going to fix runtime issue with module loading: snd_soc_sdma: Unknown symbol devm_kmalloc (err 0) snd_soc_sdma: Unknown symbol omap_dma_filter_fn (err 0) snd_soc_sdma: Unknown symbol snd_dmaengine_pcm_prepare_slave_config (err 0) snd_soc_sdma: Unknown symbol devm_snd_dmaengine_pcm_register (err 0) As these symbols are EXPORT_SYMBOL_GPL()... Introduced by commit dde637f2daf1 ("ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm") The second patch is kust a small adjustement for the include files. Regards, Peter --- Peter Ujfalusi (2): ASoC: omap: sdma-pcm: Fix modpost warning ASoC: omap: sdma-pcm: Correction for the include files sound/soc/omap/sdma-pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 3

[alsa-devel] [PATCH 0/2] ASoC: omap: sdma-pcm: modpost warning fix and include cleanup
by Peter Ujfalusi 17 May '18

17 May '18

Hi, Since the sdma-pcm can be built as module it must have MODULE_LICENSE defined, whcih I have missed: After merging the sound-asoc tree, today's linux-next build (x86_64 allmodconfig) produced this warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o see include/linux/module.h for more information WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o see include/linux/module.h for more information Introduced by commit dde637f2daf1 ("ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm") The second patch is kust a small adjustement for the include files. Regards, Peter --- Peter Ujfalusi (2): ASoC: omap: sdma-pcm: Fix modpost warning ASoC: omap: sdma-pcm: Correction for the include files sound/soc/omap/sdma-pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 4

[alsa-devel] [PATCH] ASoC: rt5663: Use the set_jack() instead of the export function
by Oder Chiou 17 May '18

17 May '18

The patch replaces the export function with the new API set_jack(). Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 5 ++--- sound/soc/codecs/rt5663.h | 2 -- sound/soc/intel/boards/kbl_rt5663_max98927.c | 3 ++- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 4 files changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index f0cef52..eefa918 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1857,7 +1857,7 @@ static irqreturn_t rt5663_irq(int irq, void *data) } int rt5663_set_jack_detect(struct snd_soc_component *component, - struct snd_soc_jack *hs_jack) + struct snd_soc_jack *hs_jack, void *data) { struct rt5663_priv *rt5663 = snd_soc_component_get_drvdata(component); @@ -1867,7 +1867,6 @@ int rt5663_set_jack_detect(struct snd_soc_component *component, return 0; } -EXPORT_SYMBOL_GPL(rt5663_set_jack_detect); static bool rt5663_check_jd_status(struct snd_soc_component *component) { @@ -3244,10 +3243,10 @@ static const struct snd_soc_component_driver soc_component_dev_rt5663 = { .num_dapm_widgets = ARRAY_SIZE(rt5663_dapm_widgets), .dapm_routes = rt5663_dapm_routes, .num_dapm_routes = ARRAY_SIZE(rt5663_dapm_routes), + .set_jack = rt5663_set_jack_detect, .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, - }; static const struct regmap_config rt5663_v2_regmap = { diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h index 865203c..794cf3f 100644 --- a/sound/soc/codecs/rt5663.h +++ b/sound/soc/codecs/rt5663.h @@ -1125,8 +1125,6 @@ enum { RT5663_AD_STEREO_FILTER = 0x2, }; -int rt5663_set_jack_detect(struct snd_soc_component *component, - struct snd_soc_jack *hs_jack); int rt5663_sel_asrc_clk_src(struct snd_soc_component *component, unsigned int filter_mask, unsigned int clk_src); diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 513cc65..3a61252 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -299,7 +299,8 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - rt5663_set_jack_detect(component, &ctx->kabylake_headset); + snd_soc_component_set_jack(component, &ctx->kabylake_headset, NULL); + return ret; } diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index e8382d6..92f5fb2 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -200,7 +200,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - rt5663_set_jack_detect(component, &ctx->kabylake_headset); + snd_soc_component_set_jack(component, &ctx->kabylake_headset, NULL); ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "DMIC"); if (ret) -- 2.7.4

3 4

[alsa-devel] [PATCH v3 00/10] ASoC: rt5640: Add jack-detect and button-press support
by Hans de Goede 17 May '18

17 May '18

Hi Mark, et. al., Here is v3 of the remaining patches from my rt5640 jack-detect series. This time this is really based on your sound/for-next branch, so this should apply cleanly now. As mentioned with v2 I've also added a quirk for one more device to the: "ASoC: Intel: bytcr_rt5640: Add quirks for various devices" patch. Other then that there are no changes. Regards, Hans

2 11

[alsa-devel] [PATCH v2 00/10] ASoC: rt5640: Add jack-detect and button-press support
by Hans de Goede 17 May '18

17 May '18

Hi Mark, et. al., Here is v2 of the remaining patches from my rt5640 jack-detect series. As requested this is rebased on your sound/for-next branch, so this should apply cleanly now. I also added a quirk for one more device to the: "ASoC: Intel: bytcr_rt5640: Add quirks for various devices" patch. Other then that there are no changes. Regards, Hans

2 13

[alsa-devel] [PATCH 0/1] Handling of serializers
by Vishal Thanki 17 May '18

17 May '18

On a custom AM33xx based platform, we use 4 channels for audio output (2 * Left/Right) by making use of two TX serializers. The audio input path only uses 2 channels, hence one RX serializer. If the recording is started before the playback, the playback cannot make use of all 4 channels because of the imposed max number of channels by recording. And if playback is started before recording, the hw_params set during recording stream would disable an active TX serializer which causes audio distortion. I have tried to fix this unwanted behavior in following patch. Please comment if this should be done differently. Vishal Thanki (1): ASoC: davinci-mcasp: Only disable inactive serializer sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) -- 2.14.3

3 6

[alsa-devel] [PATCH v2] ASoC: ssm2305: Add amplifier driver
by Marco Felsch 17 May '18

17 May '18

The ssm2305 is a simple Class-D audio amplifier. A application can turn on/off the device by a gpio. It's also possible to hardwire the shutdown pin. Tested on a i.MX6 based custom board. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- Notes (changelog): v2: - fix copyright information - rename shutdown gpio and add some more documentation - mark shutdown gpio as required - rm registration information .../devicetree/bindings/sound/adi,ssm2305.txt | 14 +++ sound/soc/codecs/Kconfig | 7 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2305.c | 105 ++++++++++++++++++ 4 files changed, 128 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2305.txt create mode 100644 sound/soc/codecs/ssm2305.c diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2305.txt b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt new file mode 100644 index 000000000000..a9c9d83c8a30 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt @@ -0,0 +1,14 @@ +Analog Devices SSM2305 Speaker Amplifier +======================================== + +Required properties: + - compatible : "adi,ssm2305" + - shutdown-gpios : The gpio connected to the shutdown pin. + The gpio signal is ACTIVE_LOW. + +Example: + +ssm2305: analog-amplifier { + compatible = "adi,ssm2305"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9548f63ca531..d81b0cb291a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -142,6 +142,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF + select SND_SOC_SSM2305 select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C @@ -883,6 +884,12 @@ config SND_SOC_SIRF_AUDIO_CODEC config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SSM2305 + tristate "Analog Devices SSM2305 Class-D Amplifier" + help + Enable support for Analog Devices SSM2305 filterless + high-efficiency mono Class-D audio power amplifiers. + config SND_SOC_SSM2518 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e849d1495308..7d75ac8f6716 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -153,6 +153,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o @@ -404,6 +405,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o +obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o diff --git a/sound/soc/codecs/ssm2305.c b/sound/soc/codecs/ssm2305.c new file mode 100644 index 000000000000..06ff649a1fd2 --- /dev/null +++ b/sound/soc/codecs/ssm2305.c @@ -0,0 +1,105 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices SSM2305 Amplifier Driver + * + * Copyright (C) 2018 Pengutronix, Marco Felsch <kernel(a)pengutronix.de> + */ + +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <sound/soc.h> + +#define DRV_NAME "ssm2305" + +struct ssm2305 { + /* shutdown gpio */ + struct gpio_desc *gpiod_shutdown; +}; + +static int ssm2305_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kctrl, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct ssm2305 *data = snd_soc_component_get_drvdata(c); + + gpiod_set_value_cansleep(data->gpiod_shutdown, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget ssm2305_dapm_widgets[] = { + /* Stereo input/output */ + SND_SOC_DAPM_INPUT("L_IN"), + SND_SOC_DAPM_INPUT("R_IN"), + SND_SOC_DAPM_OUTPUT("L_OUT"), + SND_SOC_DAPM_OUTPUT("R_OUT"), + + SND_SOC_DAPM_SUPPLY("Power", SND_SOC_NOPM, 0, 0, ssm2305_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route ssm2305_dapm_routes[] = { + { "L_OUT", NULL, "L_IN" }, + { "R_OUT", NULL, "R_IN" }, + { "L_IN", NULL, "Power" }, + { "R_IN", NULL, "Power" }, +}; + +static const struct snd_soc_component_driver ssm2305_component_driver = { + .dapm_widgets = ssm2305_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2305_dapm_widgets), + .dapm_routes = ssm2305_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2305_dapm_routes), +}; + +static int ssm2305_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ssm2305 *priv; + int err; + + /* Allocate the private data */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + + /* Get shutdown gpio */ + priv->gpiod_shutdown = devm_gpiod_get(dev, "shutdown", + GPIOD_OUT_LOW); + if (IS_ERR(priv->gpiod_shutdown)) { + err = PTR_ERR(priv->gpiod_shutdown); + if (err != -EPROBE_DEFER) { + dev_err(dev, "Failed to get 'shutdown' gpio: %d\n", + err); + return err; + } + } + + return devm_snd_soc_register_component(dev, &ssm2305_component_driver, + NULL, 0); +} + +#ifdef CONFIG_OF +static const struct of_device_id ssm2305_of_match[] = { + { .compatible = "adi,ssm2305", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2305_of_match); +#endif + +static struct platform_driver ssm2305_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(ssm2305_of_match), + }, + .probe = ssm2305_probe, +}; + +module_platform_driver(ssm2305_driver); + +MODULE_DESCRIPTION("ASoC SSM2305 amplifier driver"); +MODULE_AUTHOR("Marco Felsch <m.felsch(a)pengutronix.de>"); +MODULE_LICENSE("GPL v2"); -- 2.17.0

2 2

[alsa-devel] [PATCH] ASoC: ssm2305: Add amplifier driver
by Marco Felsch 17 May '18

17 May '18

The ssm2305 is a simple Class-D audio amplifier. A application can turn on/off the device by a gpio. It's also possible to hardwire the shutdown pin. Tested on a i.MX6 based custom board. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- .../devicetree/bindings/sound/adi,ssm2305.txt | 15 +++ sound/soc/codecs/Kconfig | 7 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2305.c | 105 ++++++++++++++++++ 4 files changed, 129 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2305.txt create mode 100644 sound/soc/codecs/ssm2305.c diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2305.txt b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt new file mode 100644 index 000000000000..fc4c99225538 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2305.txt @@ -0,0 +1,15 @@ +Analog Devices SSM2305 Speaker Amplifier +======================================== + +Required properties: + - compatible : "adi,ssm2305" + +Optional properties: + - ssm2305,shutdown-gpio : the gpio connected to the shutdown pin + +Example: + +ssm2305: analog-amplifier { + compatible = "adi,ssm2305"; + ssm2305,shutdown-gpio = <&gpio3 20 GPIO_ACTIVE_LOW>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9548f63ca531..d81b0cb291a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -142,6 +142,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF + select SND_SOC_SSM2305 select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C @@ -883,6 +884,12 @@ config SND_SOC_SIRF_AUDIO_CODEC config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SSM2305 + tristate "Analog Devices SSM2305 Class-D Amplifier" + help + Enable support for Analog Devices SSM2305 filterless + high-efficiency mono Class-D audio power amplifiers. + config SND_SOC_SSM2518 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e849d1495308..7d75ac8f6716 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -153,6 +153,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o @@ -404,6 +405,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o +obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o diff --git a/sound/soc/codecs/ssm2305.c b/sound/soc/codecs/ssm2305.c new file mode 100644 index 000000000000..ce8d5a18c329 --- /dev/null +++ b/sound/soc/codecs/ssm2305.c @@ -0,0 +1,105 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices SSM2305 Amplifier Driver + * + * Copyright (C) 2018 Pengutronix, Fridolin Tux <kernel(a)pengutronix.de> + */ + +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <sound/soc.h> + +#define DRV_NAME "ssm2305" + +struct ssm2305 { + /* shutdown gpio */ + struct gpio_desc *gpiod_shutdown; +}; + +static int ssm2305_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kctrl, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct ssm2305 *data = snd_soc_component_get_drvdata(c); + + gpiod_set_value_cansleep(data->gpiod_shutdown, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget ssm2305_dapm_widgets[] = { + /* Stereo input/output */ + SND_SOC_DAPM_INPUT("L_IN"), + SND_SOC_DAPM_INPUT("R_IN"), + SND_SOC_DAPM_OUTPUT("L_OUT"), + SND_SOC_DAPM_OUTPUT("R_OUT"), + + SND_SOC_DAPM_SUPPLY("Power", SND_SOC_NOPM, 0, 0, ssm2305_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route ssm2305_dapm_routes[] = { + { "L_OUT", NULL, "L_IN" }, + { "R_OUT", NULL, "R_IN" }, + { "L_IN", NULL, "Power" }, + { "R_IN", NULL, "Power" }, +}; + +static const struct snd_soc_component_driver ssm2305_component_driver = { + .dapm_widgets = ssm2305_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2305_dapm_widgets), + .dapm_routes = ssm2305_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2305_dapm_routes), +}; + +static int ssm2305_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ssm2305 *priv; + int err; + + /* Allocate the private data */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + + /* Shutdown gpio is optional */ + priv->gpiod_shutdown = devm_gpiod_get(dev, "ssm2305,shutdown", + GPIOD_OUT_LOW); + if (IS_ERR(priv->gpiod_shutdown)) { + err = PTR_ERR(priv->gpiod_shutdown); + if (err != -EPROBE_DEFER) + dev_warn(dev, "Failed to get 'shutdown' gpio: %d\n", + err); + } + + dev_info(dev, "probed\n"); + + return devm_snd_soc_register_component(dev, &ssm2305_component_driver, + NULL, 0); +} + +#ifdef CONFIG_OF +static const struct of_device_id ssm2305_of_match[] = { + { .compatible = "adi,ssm2305", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2305_of_match); +#endif + +static struct platform_driver ssm2305_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(ssm2305_of_match), + }, + .probe = ssm2305_probe, +}; + +module_platform_driver(ssm2305_driver); + +MODULE_DESCRIPTION("ASoC SSM2305 amplifier driver"); +MODULE_AUTHOR("Marco Felsch <m.felsch(a)pengutronix.de>"); +MODULE_LICENSE("GPL v2"); -- 2.17.0

3 3

[alsa-devel] [PATCH v2 0/8] slimbus: fixes and some helpers
by Srinivas Kandagatla 17 May '18

17 May '18

Hi Greg, Here is set of patches for slimbus, there are 6 fixes and two helper functions. Most of these issues were found while testing QCOM NGD SLIMBus controller with WCD9335 codec. Changes since v1 (https://lkml.org/lkml/2018/5/16/436): - add need_tid to documentation reported by 0 day test. - remove redundant kconfig for qcom slim controller. thanks, Srini Srinivas Kandagatla (8): slimbus: core: add of_slim_device_get() helper slimbus: core: rearrange slim_eaddr structure slimbus: core: add need_tid flag to slim_msg_txn slimbus: messaging: pass correct wbuf slimbus: messaging: remove multiple calls to pm_runtime_mark_last_busy slimbus: messaging: add slim_prepare_txn() helper function slimbus: messaging: initialize completion correctly slimbus: qcom: remove redundant depends in Kconfig drivers/slimbus/Kconfig | 1 - drivers/slimbus/core.c | 39 +++++++++++++++++++ drivers/slimbus/messaging.c | 92 ++++++++++++++++++++++++++++++--------------- drivers/slimbus/qcom-ctrl.c | 7 +++- drivers/slimbus/slimbus.h | 4 ++ include/linux/slimbus.h | 14 ++++--- 6 files changed, 119 insertions(+), 38 deletions(-) -- 2.16.2

1 8

[alsa-devel] [PATCH 1/2] ALSA: timer: Simplify timer hw resolution calls
by Takashi Iwai 17 May '18

17 May '18

There multiple open-codes to get the hardware timer resolution. Make a local helper function snd_timer_hw_resolution() and call it from all relevant places. There is no functional change by this, just a preliminary work for the following timer resolution hardening patch. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 33 ++++++++++++++------------------- 1 file changed, 14 insertions(+), 19 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index dc87728c5b74..23b941df9c43 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -427,6 +427,14 @@ int snd_timer_close(struct snd_timer_instance *timeri) } EXPORT_SYMBOL(snd_timer_close); +static unsigned long snd_timer_hw_resolution(struct snd_timer *timer) +{ + if (timer->hw.c_resolution) + return timer->hw.c_resolution(timer); + else + return timer->hw.resolution; +} + unsigned long snd_timer_resolution(struct snd_timer_instance *timeri) { struct snd_timer * timer; @@ -434,11 +442,8 @@ unsigned long snd_timer_resolution(struct snd_timer_instance *timeri) if (timeri == NULL) return 0; timer = timeri->timer; - if (timer) { - if (timer->hw.c_resolution) - return timer->hw.c_resolution(timer); - return timer->hw.resolution; - } + if (timer) + return snd_timer_hw_resolution(timer); return 0; } EXPORT_SYMBOL(snd_timer_resolution); @@ -771,10 +776,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) spin_lock_irqsave(&timer->lock, flags); /* remember the current resolution */ - if (timer->hw.c_resolution) - resolution = timer->hw.c_resolution(timer); - else - resolution = timer->hw.resolution; + resolution = snd_timer_hw_resolution(timer); /* loop for all active instances * Here we cannot use list_for_each_entry because the active_list of a @@ -1014,12 +1016,8 @@ void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstam spin_lock_irqsave(&timer->lock, flags); if (event == SNDRV_TIMER_EVENT_MSTART || event == SNDRV_TIMER_EVENT_MCONTINUE || - event == SNDRV_TIMER_EVENT_MRESUME) { - if (timer->hw.c_resolution) - resolution = timer->hw.c_resolution(timer); - else - resolution = timer->hw.resolution; - } + event == SNDRV_TIMER_EVENT_MRESUME) + resolution = snd_timer_hw_resolution(timer); list_for_each_entry(ti, &timer->active_list_head, active_list) { if (ti->ccallback) ti->ccallback(ti, event, tstamp, resolution); @@ -1656,10 +1654,7 @@ static int snd_timer_user_gstatus(struct file *file, mutex_lock(&register_mutex); t = snd_timer_find(&tid); if (t != NULL) { - if (t->hw.c_resolution) - gstatus.resolution = t->hw.c_resolution(t); - else - gstatus.resolution = t->hw.resolution; + gstatus.resolution = snd_timer_hw_resolution(t); if (t->hw.precise_resolution) { t->hw.precise_resolution(t, &gstatus.resolution_num, &gstatus.resolution_den); -- 2.16.3

1 2

[alsa-devel] [PATCH] ALSA: emu10k1: fix spelling mistake: "Caputre" -> "Capture"
by Colin King 17 May '18

17 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistakes in audigy_outs arrays. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/emu10k1/emufx.c | 2 +- sound/pci/emu10k1/emuproc.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 608ff4857d70..b45a01bb73e5 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -170,7 +170,7 @@ static char *audigy_outs[32] = { /* 0x0f */ "Rear Right", /* 0x10 */ "AC97 Front Left", /* 0x11 */ "AC97 Front Right", - /* 0x12 */ "ADC Caputre Left", + /* 0x12 */ "ADC Capture Left", /* 0x13 */ "ADC Capture Right", /* 0x14 */ NULL, /* 0x15 */ NULL, diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index bde0d1954f56..055227caa7ca 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -135,7 +135,7 @@ static void snd_emu10k1_proc_read(struct snd_info_entry *entry, /* 15 */ "Rear Right", /* 16 */ "AC97 Front Left", /* 17 */ "AC97 Front Right", - /* 18 */ "ADC Caputre Left", + /* 18 */ "ADC Capture Left", /* 19 */ "ADC Capture Right", /* 20 */ "???", /* 21 */ "???", -- 2.17.0

2 1

[alsa-devel] [PATCH 43/45] sound/soc/zte: remove duplicate includes
by Pravin Shedge 17 May '18

17 May '18

These duplicate includes have been found with scripts/checkincludes.pl but they have been removed manually to avoid removing false positives. Signed-off-by: Pravin Shedge <pravin.shedge4linux(a)gmail.com> --- sound/soc/zte/zx-i2s.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 9a05659..93428e1 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -20,9 +20,6 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/initval.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> #define ZX_I2S_PROCESS_CTRL 0x04 #define ZX_I2S_TIMING_CTRL 0x08 -- 2.7.4

2 2

[alsa-devel] Is add uevent of controlC still a reliable synchronization point?
by Tzung-Bi Shih 17 May '18

17 May '18

Hi, We found an assumption in 78-sound-cards.rules and commit 289ca025ee1d78223e3368801fc2b984e5efbfc7 are conflict. The control device will not be the last one to register. The assumption in 78-sound-card.rules: > The control device node creation can be used as synchronization point. All other devices that belong to a card are created in the kernel before it. The commit: > commit 289ca025ee1d78223e3368801fc2b984e5efbfc7 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Wed Jan 29 15:53:35 2014 +0100 > > ALSA: Use priority list for managing device list > > Basically, the device type specifies the priority of the device to be > registered / freed, too. However, the priority value isn't well > utilized but only it's checked as a group. This results in > inconsistent register and free order (where each of them should be in > reversed direction). > > This patch simplifies the device list management code by simply > inserting a list entry at creation time in an incremental order for > the priority value. Since we can just follow the link for register, > disconnect and free calls, we don't have to specify the group; so the > whole enum definitions are also simplified as well. > > The visible change to outside is that the priorities of some object > types are revisited. For example, now the SNDRV_DEV_LOWLEVEL object > is registered before others (control, PCM, etc) and, in return, > released after others. Similarly, SNDRV_DEV_CODEC is in a lower > priority than SNDRV_DEV_BUS for ensuring the dependency. > > Also, the unused SNDRV_DEV_TOPLEVEL, SNDRV_DEV_LOWLEVEL_PRE and > SNDRV_DEV_LOWLEVEL_NORMAL are removed as a cleanup. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> The device types: > enum snd_device_type { > SNDRV_DEV_LOWLEVEL, > SNDRV_DEV_CONTROL, > SNDRV_DEV_INFO, > SNDRV_DEV_BUS, > SNDRV_DEV_CODEC, > SNDRV_DEV_SEQUENCER, > SNDRV_DEV_HWDEP, > SNDRV_DEV_JACK, > }; The commit sorts device types ascendantly and registers them from the list head. As a result, SNDRV_DEV_CONTROL devices will be registered before most other devices. We are writing to ask: Is add uevent of controlC still a reliable signal for indicating everything is ready for a sound card? Or is there now a better way to do so?

2 8

[alsa-devel] [PATCH v3] ASoC: da7219: read fmw property to get mclk for non-dts systems
by Akshu Agrawal 17 May '18

17 May '18

Non-dts based systems can use ACPI DSDT to pass on the mclk to da7219. This enables da7219 mclk to be linked to system clock. Enable/Disable of the mclk is already handled in the codec so platform drivers don't have to explicitly do handling of mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Fixed kbuild error v3: Add corresponding clk_put for clk_get include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 10 +++++++++- 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16..df7ddf4 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + const char *mclk_name; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8..ecd46fc 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1624,6 +1624,8 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone dev_warn(dev, "Using default clk name: %s\n", pdata->dai_clks_name); + device_property_read_string(dev, "dlg,mclk-name", &pdata->mclk_name); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -1905,7 +1907,10 @@ static int da7219_probe(struct snd_soc_component *component) da7219_handle_pdata(component); /* Check if MCLK provided */ - da7219->mclk = devm_clk_get(component->dev, "mclk"); + if (da7219->pdata->mclk_name) + da7219->mclk = clk_get(NULL, da7219->pdata->mclk_name); + if (!da7219->mclk) + da7219->mclk = devm_clk_get(component->dev, "mclk"); if (IS_ERR(da7219->mclk)) { if (PTR_ERR(da7219->mclk) != -ENOENT) { ret = PTR_ERR(da7219->mclk); @@ -1971,6 +1976,9 @@ static void da7219_remove(struct snd_soc_component *component) clkdev_drop(da7219->dai_clks_lookup); #endif + if (da7219->pdata->mclk_name) + clk_put(da7219->mclk); + /* Supplies */ regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies); } -- 1.9.1

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[alsa-devel] [PATCH v4 0/3] ASoC: Add Multi CPU DAI support
by Shreyas NC 17 May '18

17 May '18

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v4: - Addressed review comments from Charles regarding error handling in PCM operations and other nits - Rebased on the latest for-next branch Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 6 + sound/soc/soc-core.c | 283 ++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 487 ++++++++++++++++++++++++++++++++++----------------- 4 files changed, 573 insertions(+), 274 deletions(-) -- 2.7.4

2 5

[alsa-devel] [PATCH 0/7] slimbus: fixes and some helpers
by Srinivas Kandagatla 17 May '18

17 May '18

Hi Greg, Here is set of patches for slimbus, there are 5 fixes and two helper functions. Most of these issues were found while testing QCOM NGD SLIMBus controller with WCD9335 codec. thanks, Srini Srinivas Kandagatla (7): slimbus: core: add of_slim_device_get() helper slimbus: core: rearrange slim_eaddr structure slimbus: core: add need_tid flag to slim_msg_txn slimbus: messaging: pass correct wbuf slimbus: messaging: remove multiple calls to pm_runtime_mark_last_busy slimbus: messaging: add slim_prepare_txn() helper function slimbus: messaging: initialize completion correctly drivers/slimbus/core.c | 39 +++++++++++++++++++ drivers/slimbus/messaging.c | 92 ++++++++++++++++++++++++++++++--------------- drivers/slimbus/qcom-ctrl.c | 7 +++- drivers/slimbus/slimbus.h | 3 ++ include/linux/slimbus.h | 14 ++++--- 5 files changed, 118 insertions(+), 37 deletions(-) -- 2.16.2

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[alsa-devel] LAC2018 schedule and registration online!
by Linux Audio Conference 16 May '18

16 May '18

Dear all, we're happy to announce (and you might have noticed already), that the schedule for Linux Audio Conference 2018 is live! https://lac.linuxaudio.org/2018/pages/schedule Please note, that it will still be subject to slight changes! We have prepared a registration form for attendees: https://www.ak.tu-berlin.de/index.php?id=195758 Please sign up at your earliest convenience to let us know, if you're part of the conference. Also, we'd like to draw special attention to this year's keynote, which will be held by Gianfranco Ceccolini of MOD Devices: https://lac.linuxaudio.org/2018/pages/event/102/ Looking forward to seeing you in Berlin, -- Linux Audio Conference team

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Apply rate limit to warning messages in URB complete callback
by Takashi Iwai 16 May '18

16 May '18

retire_capture_urb() may print warning messages when the given URB doesn't align, and this may flood the system log easily. Put the rate limit to the message for avoiding it. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1093485 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index dc2dfec9effd..20bed1c7a312 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1303,7 +1303,7 @@ static void retire_capture_urb(struct snd_usb_substream *subs, if (bytes % (runtime->sample_bits >> 3) != 0) { int oldbytes = bytes; bytes = frames * stride; - dev_warn(&subs->dev->dev, + dev_warn_ratelimited(&subs->dev->dev, "Corrected urb data len. %d->%d\n", oldbytes, bytes); } -- 2.16.3

1 0

[alsa-devel] Sound is not working for my headphones although it works for Windows 10
by Faisal Ahmed Moshiur 16 May '18

16 May '18

http://www.alsa-project.org/db/?f=8864f5f90259b36ff18270e6242508f64cba4292 This contains my PC info.

2 1

[alsa-devel] Aplay Fails When Reading From stdin
by Jay Foster 16 May '18

16 May '18

I recently updated may alsa from 1.1.4.1 to 1.1.6. I now noticed that aplay does not work properly (the same as 1.1.4.1) when reading the sound file data from stdin. This might have something to do with the recent change with reading the sound file header. With the previous version of aplay, aplay would report: Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono With 1.1.6, aplay reports: Playing raw data 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono This results in static/noise output. If I explicitly add the '-f S16_LE -r 22050' options to aplay, then it does play correctly. Is this a known bug and is there a fix? Jay

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Re: [alsa-devel] [PATCH v2 09/11] docs: Fix some broken references
by Mathieu Poirier 16 May '18

16 May '18

On 9 May 2018 at 07:18, Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> wrote: > As we move stuff around, some doc references are broken. Fix some of > them via this script: > ./scripts/documentation-file-ref-check --fix-rst > > Manually checked if the produced result is valid, removing a few > false-positives. > > Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> > --- > > diff --git a/Documentation/trace/coresight.txt b/Documentation/trace/coresight.txt > index 1d74ad0202b6..efbc832146e7 100644 > --- a/Documentation/trace/coresight.txt > +++ b/Documentation/trace/coresight.txt > @@ -426,5 +426,5 @@ root@genericarmv8:~# > Details on how to use the generic STM API can be found here [2]. > > [1]. Documentation/ABI/testing/sysfs-bus-coresight-devices-stm > -[2]. Documentation/trace/stm.txt > +[2]. Documentation/trace/stm.rst > [3]. https://github.com/Linaro/perf-opencsd Acked-by: Mathieu Poirier <mathieu.poirier(a)linaro.org>

1 0

[alsa-devel] [bug report] ALSA: usb: add UAC3 BADD profiles support
by Dan Carpenter 16 May '18

16 May '18

[ Your patch didn't really introduce the bug, it just made it visible to static checkers. - dan ] Hello Ruslan Bilovol, The patch 17156f23e93c: "ALSA: usb: add UAC3 BADD profiles support" from May 4, 2018, leads to the following static checker warning: sound/usb/stream.c:984 snd_usb_get_audioformat_uac3() warn: possible memory leak of 'chmap' sound/usb/stream.c 864 865 chmap = kzalloc(sizeof(*chmap), GFP_KERNEL); 866 if (!chmap) 867 return ERR_PTR(-ENOMEM); 868 869 if (num_channels == 1) { 870 chmap->map[0] = SNDRV_CHMAP_MONO; 871 } else { 872 chmap->map[0] = SNDRV_CHMAP_FL; 873 chmap->map[1] = SNDRV_CHMAP_FR; 874 } 875 876 chmap->channels = num_channels; 877 clock = UAC3_BADD_CS_ID9; 878 goto found_clock; 879 } 880 881 as = snd_usb_find_csint_desc(alts->extra, alts->extralen, 882 NULL, UAC_AS_GENERAL); 883 if (!as) { 884 dev_err(&dev->dev, 885 "%u:%d : UAC_AS_GENERAL descriptor not found\n", 886 iface_no, altno); 887 return NULL; 888 } 889 890 if (as->bLength < sizeof(*as)) { 891 dev_err(&dev->dev, 892 "%u:%d : invalid UAC_AS_GENERAL desc\n", 893 iface_no, altno); 894 return NULL; 895 } 896 897 cluster_id = le16_to_cpu(as->wClusterDescrID); 898 if (!cluster_id) { 899 dev_err(&dev->dev, 900 "%u:%d : no cluster descriptor\n", 901 iface_no, altno); 902 return NULL; 903 } 904 905 /* 906 * Get number of channels and channel map through 907 * High Capability Cluster Descriptor 908 * 909 * First step: get High Capability header and 910 * read size of Cluster Descriptor 911 */ 912 err = snd_usb_ctl_msg(chip->dev, 913 usb_rcvctrlpipe(chip->dev, 0), 914 UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, 915 USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 916 cluster_id, 917 snd_usb_ctrl_intf(chip), 918 &hc_header, sizeof(hc_header)); 919 if (err < 0) 920 return ERR_PTR(err); 921 else if (err != sizeof(hc_header)) { 922 dev_err(&dev->dev, 923 "%u:%d : can't get High Capability descriptor\n", 924 iface_no, altno); 925 return ERR_PTR(-EIO); 926 } 927 928 /* 929 * Second step: allocate needed amount of memory 930 * and request Cluster Descriptor 931 */ 932 wLength = le16_to_cpu(hc_header.wLength); 933 cluster = kzalloc(wLength, GFP_KERNEL); 934 if (!cluster) 935 return ERR_PTR(-ENOMEM); 936 err = snd_usb_ctl_msg(chip->dev, 937 usb_rcvctrlpipe(chip->dev, 0), 938 UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, 939 USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 940 cluster_id, 941 snd_usb_ctrl_intf(chip), 942 cluster, wLength); 943 if (err < 0) { 944 kfree(cluster); 945 return ERR_PTR(err); 946 } else if (err != wLength) { 947 dev_err(&dev->dev, 948 "%u:%d : can't get Cluster Descriptor\n", 949 iface_no, altno); 950 kfree(cluster); 951 return ERR_PTR(-EIO); 952 } 953 954 num_channels = cluster->bNrChannels; 955 chmap = convert_chmap_v3(cluster); 956 kfree(cluster); 957 958 /* 959 * lookup the terminal associated to this interface 960 * to extract the clock 961 */ 962 input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, 963 as->bTerminalLink); 964 if (input_term) { 965 clock = input_term->bCSourceID; 966 goto found_clock; 967 } 968 969 output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, 970 as->bTerminalLink); 971 if (output_term) { 972 clock = output_term->bCSourceID; 973 goto found_clock; 974 } 975 976 dev_err(&dev->dev, "%u:%d : bogus bTerminalLink %d\n", 977 iface_no, altno, as->bTerminalLink); 978 return NULL; 979 980 found_clock: 981 fp = audio_format_alloc_init(chip, alts, UAC_VERSION_3, iface_no, 982 altset_idx, altno, num_channels, clock); 983 if (!fp) 984 return ERR_PTR(-ENOMEM); We should free "chmap" before returning. 985 986 fp->chmap = chmap; 987 988 if (badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) { 989 fp->attributes = 0; /* No attributes */ 990 991 fp->fmt_type = UAC_FORMAT_TYPE_I; 992 fp->formats = badd_formats; 993 994 fp->nr_rates = 0; /* SNDRV_PCM_RATE_CONTINUOUS */ 995 fp->rate_min = UAC3_BADD_SAMPLING_RATE; 996 fp->rate_max = UAC3_BADD_SAMPLING_RATE; 997 fp->rates = SNDRV_PCM_RATE_CONTINUOUS; 998 999 } else { 1000 fp->attributes = parse_uac_endpoint_attributes(chip, alts, 1001 UAC_VERSION_3, 1002 iface_no); 1003 /* ok, let's parse further... */ 1004 if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) { 1005 kfree(fp->rate_table); 1006 kfree(fp); 1007 return NULL; And here as well? I don't know. 1008 } 1009 } 1010 1011 return fp; 1012 } regards, dan carpenter

2 1

Re: [alsa-devel] [PATCH v2 09/11] docs: Fix some broken references
by Charles Keepax 16 May '18

16 May '18

On Wed, May 09, 2018 at 10:18:52AM -0300, Mauro Carvalho Chehab wrote: > As we move stuff around, some doc references are broken. Fix some of > them via this script: > ./scripts/documentation-file-ref-check --fix-rst > > Manually checked if the produced result is valid, removing a few > false-positives. > > Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> > --- > drivers/input/touchscreen/wm97xx-core.c | 2 +- Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Thanks, Charles

1 0

Re: [alsa-devel] [PATCH v2 09/11] docs: Fix some broken references
by Stephen Boyd 15 May '18

15 May '18

Quoting Mauro Carvalho Chehab (2018-05-09 06:18:52) > As we move stuff around, some doc references are broken. Fix some of > them via this script: > ./scripts/documentation-file-ref-check --fix-rst > > Manually checked if the produced result is valid, removing a few > false-positives. > > Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> > --- > drivers/clk/clk.c | 4 ++-- > drivers/clk/ingenic/cgu.h | 2 +- Acked-by: Stephen Boyd <sboyd(a)kernel.org>

1 0

[alsa-devel] [PATCH 0/4] ALSA: hda/ca0132: constify local symbols
by Takashi Sakamoto 15 May '18

15 May '18

Hi, This module has some local symbols which are read-only. This commit adds 'const' qualifier to the symbols and applies some code optimization. Takashi Sakamoto (4): ALSA: hda/ca0132: constify templates for control element set ALSA: hda/ca0132: constify read-only members of string array ALSA: hda/ca0132: merge strings just for printk ALSA: hda/ca0132: constify parameter table for effects sound/pci/hda/patch_ca0132.c | 34 ++++++++++++++++------------------ 1 file changed, 16 insertions(+), 18 deletions(-) -- 2.14.1

2 5

[alsa-devel] [alsa-lib][PATCH 0/6] control: optimization for position offset macro of TLV data
by Takashi Sakamoto 15 May '18

15 May '18

Hi, In development period for Linux v4.18, a series of SNDRV_CTL_TLVO_XXX macro was introduced to kernel stuffs for position offset of TLV data. This patchset backports the macros to this library as well as adds code optimization with them. Takashi Sakamoto (6): control: add a series of macro for offset of several types of TLV control: use position offset macro of TLV data hcontrol: use position offset macro of TLV data pcm: hw: use position offset macro of TLV data pcm: softvol: use position offset macro of TLV data test: use position offset macro of TLV data include/sound/tlv.h | 16 ++++++++++ src/control/control.c | 12 ++++---- src/control/control_hw.c | 6 ++-- src/control/control_local.h | 1 + src/control/hcontrol.c | 4 +-- src/control/tlv.c | 72 ++++++++++++++++++++++++--------------------- src/pcm/pcm_hw.c | 10 ++++--- src/pcm/pcm_softvol.c | 11 ++++--- test/user-ctl-element-set.c | 2 +- 9 files changed, 83 insertions(+), 51 deletions(-) -- 2.14.1

2 7

[alsa-devel] [PATCH V3] ASoC: amd: modified dma pointer callback implementation
by Vijendar Mukunda 15 May '18

15 May '18

Modified dma pointer callback implementation. hw ptr calculated based on System Memory to ACP SRAM dma channel transfer count in case of playback. In case of capture, hw ptr calculated based on dma transfer count for ACP SRAM to System Memory Dma channel. Added IOC Bit for Sysmem to ACP Dma channel for both controller instances. Invoked snd_pcm_period_elapsed() call when IOC is reported for System memory to ACP SRAM DMA channel. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v1->v2: removed hard coded values in acp dma pointer callback declared dma varaibles in rtd structure. v2->v3: modified code to invoke snd_pcm_period_elapsed() call when IOC reported for System Memory to ACP SRAM dma channel. This patch is dependent on [V4,10/10] ASoC: amd: dma driver changes for bt i2s instance https://patchwork.kernel.org/patch/10385381/ sound/soc/amd/acp-pcm-dma.c | 85 +++++++++++++++++++++++---------------------- sound/soc/amd/acp.h | 15 +++----- 2 files changed, 48 insertions(+), 52 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 906eb77..dfb37b5 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -207,11 +207,13 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, switch (asic_type) { case CHIP_STONEY: dmadscr[i].xfer_val |= + BIT(22) | (ACP_DMA_ATTR_DAGB_GARLIC_TO_SHAREDMEM << 16) | (size / 2); break; default: dmadscr[i].xfer_val |= + BIT(22) | (ACP_DMA_ATTR_DAGB_ONION_TO_SHAREDMEM << 16) | (size / 2); } @@ -356,9 +358,11 @@ static void acp_dma_start(void __iomem *acp_mmio, dma_ctrl |= ACP_DMA_CNTL_0__DMAChRun_MASK; switch (ch_num) { + case SYSRAM_TO_ACP_CH_NUM: case ACP_TO_I2S_DMA_CH_NUM: case ACP_TO_SYSRAM_CH_NUM: case I2S_TO_ACP_DMA_CH_NUM: + case SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM: case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: @@ -642,10 +646,14 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) config_acp_dma_channel(acp_mmio, SYSRAM_TO_ACP_CH_NUM, dscr_idx, 1, 0); acp_dma_start(acp_mmio, SYSRAM_TO_ACP_CH_NUM, false); + acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + if ((intr_flag & BIT(SYSRAM_TO_ACP_CH_NUM)) != 0) { + valid_irq = true; snd_pcm_period_elapsed(irq_data->play_i2ssp_stream); - - acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_CH_NUM)) << 16, + acp_reg_write((intr_flag & BIT(SYSRAM_TO_ACP_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } @@ -661,12 +669,19 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) dscr_idx, 1, 0); acp_dma_start(acp_mmio, SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, false); - snd_pcm_period_elapsed(irq_data->play_i2sbt_stream); acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } + if ((intr_flag & BIT(SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + snd_pcm_period_elapsed(irq_data->play_i2sbt_stream); + acp_reg_write((intr_flag & + BIT(SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_15) == @@ -854,10 +869,9 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_BLUETOOTH; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH8; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH9; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_8; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_8; + rtd->end_dma_dscr = PLAYBACK_END_DMA_DESCR_CH8; adata->play_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -875,10 +889,9 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_ACP_I2S_1; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_12; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_12; + rtd->end_dma_dscr = PLAYBACK_END_DMA_DESCR_CH12; adata->play_i2ssp_stream = substream; } } else { @@ -891,10 +904,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + rtd->dma_cntl = mmACP_DMA_CNTL_10; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_10; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_10; + rtd->end_dma_dscr = CAPTURE_END_DMA_DESCR_CH10; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -914,10 +927,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + rtd->dma_cntl = mmACP_DMA_CNTL_14; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_14; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_14; + rtd->end_dma_dscr = CAPTURE_END_DMA_DESCR_CH14; adata->capture_i2ssp_stream = substream; } } @@ -955,22 +968,13 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static u64 acp_get_byte_count(struct audio_substream_data *rtd) -{ - union acp_dma_count byte_count; - - byte_count.bcount.high = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_high_reg_offset); - byte_count.bcount.low = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_low_reg_offset); - return byte_count.bytescount; -} - static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) { u32 buffersize; u32 pos = 0; - u64 bytescount = 0; + u32 dma_count = 0; + u16 dscr; + u32 period_bytes; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -979,11 +983,15 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) return -EINVAL; buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + period_bytes = frames_to_bytes(runtime, runtime->period_size); + if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) && + acp_reg_read(rtd->acp_mmio, rtd->dma_cntl) == 0) + return 0; + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + dma_count = acp_reg_read(rtd->acp_mmio, rtd->cur_trans_cnt); + if (dscr == rtd->end_dma_dscr) + dma_count += period_bytes; + pos = dma_count % buffersize; return bytes_to_frames(runtime, pos); } @@ -1016,7 +1024,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; u32 loops = 4000; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -1030,9 +1037,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & @@ -1066,7 +1070,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_stop(rtd->acp_mmio, rtd->ch2); ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); } - rtd->bytescount = 0; break; default: ret = -EINVAL; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96..4740804 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -131,12 +131,13 @@ struct audio_substream_data { u16 destination; u16 dma_dscr_idx_1; u16 dma_dscr_idx_2; + u16 end_dma_dscr; u32 pte_offset; u32 sram_bank; - u32 byte_cnt_high_reg_offset; - u32 byte_cnt_low_reg_offset; + u32 dma_cntl; + u32 dma_curr_dscr; + u32 cur_trans_cnt; uint64_t size; - u64 bytescount; void __iomem *acp_mmio; }; @@ -157,14 +158,6 @@ struct acp_platform_info { u16 i2s_instance; }; -union acp_dma_count { - struct { - u32 low; - u32 high; - } bcount; - u64 bytescount; -}; - enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- 2.7.4

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[alsa-devel] [PATCH 0/4] ALSA: hda: Bug fixes
by Sriram Periyasamy 15 May '18

15 May '18

The series contains few bug fixes and debug updates Pardha Saradhi K (4): ALSA: hda: Make sure DMA is stopped by reading back the RUN bit ALSA: hda: Make sure DMA is started by reading back the RUN bit ALSA: hda: Log HDA Hardware related errors ALSA: hda: check if stream is stopped in snd_hdac_stream_clear sound/hda/hdac_stream.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) -- 2.7.4

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[alsa-devel] [PATCH v2 0/2] ASoC: omap: sdma-pcm: modpost warning fix and include cleanup
by Peter Ujfalusi 15 May '18

15 May '18

Hi, Changes since v1: - Changed the MODULE_LICENSE from "GPL" to "GPL v2" to match the SPDX line. Since the sdma-pcm can be built as module it must have MODULE_LICENSE defined, whcih I have missed: After merging the sound-asoc tree, today's linux-next build (x86_64 allmodconfig) produced this warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o see include/linux/module.h for more information WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o see include/linux/module.h for more information The first patch also going to fix runtime issue with module loading: snd_soc_sdma: Unknown symbol devm_kmalloc (err 0) snd_soc_sdma: Unknown symbol omap_dma_filter_fn (err 0) snd_soc_sdma: Unknown symbol snd_dmaengine_pcm_prepare_slave_config (err 0) snd_soc_sdma: Unknown symbol devm_snd_dmaengine_pcm_register (err 0) As these symbols are EXPORT_SYMBOL_GPL()... Introduced by commit dde637f2daf1 ("ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm") The second patch is kust a small adjustement for the include files. Regards, Peter --- Peter Ujfalusi (2): firmware: ti_sci: Correct the timeout type in ti_sci_do_xfer() ASoC: omap: sdma-pcm: Fix modpost warning drivers/firmware/ti_sci.c | 2 +- sound/soc/omap/sdma-pcm.c | 5 +++++ 2 files changed, 6 insertions(+), 1 deletion(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH 0/4] ALSA: usb: UAC3 new features.
by Jorge Sanjuan 15 May '18

15 May '18

Now that the UAC3 patch [1] has made it to linux-next I have some extra features to make a UAC3 device fully work in Linux. Including Jack insertion control that I have put on top of this other patch [2] for UAC2. Also adding support for the UAC3 Mixer Unit which is most likely to appear in most headset type devices. UAC3 devices also require to have a Basic Audio Device (BADD) in a separate config for which both Ruslan Bilovol and myself have submited different approaches[3][4] but I don't know what the final merge will be. Once there is official support for BADD, we'll need to test it with an actual UAC3 device to confirm it all wokrs. All this features are tested with an actual UAC3 device that is still in development. For this patch series, only the legacy config (#1. UAC1/UAC2) and the UAC3 config have been tested. The BADD config is only tested using and updated verison of [4]. [1]: https://patchwork.kernel.org/patch/10298179/ [2]: https://patchwork.kernel.org/patch/10305847/ [3]: https://patchwork.kernel.org/patch/10340851/ [4]: https://www.spinics.net/lists/alsa-devel/msg71617.html Based on linux-next tag: next-20180420 Jorge Sanjuan (3): ALSA: usb-audio: UAC3. Add support for mixer unit. ALSA: usb-audio: Use Class Specific EP for UAC3 devices. ALSA: usb-audio: UAC3 Add support for connector insertion. Michael Drake (1): ALSA: usb-audio: ADC3: Fix channel mapping conversion for ADC3. include/linux/usb/audio-v2.h | 7 ++ include/linux/usb/audio-v3.h | 14 ++++ include/uapi/linux/usb/audio.h | 13 ++- sound/usb/mixer.c | 175 ++++++++++++++++++++++++++++++++++++----- sound/usb/stream.c | 19 ++++- 5 files changed, 204 insertions(+), 24 deletions(-) -- 2.11.0

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[alsa-devel] [sound:for-next 62/68] sound/pci/hda/patch_ca0132.c:5062:50-51: WARNING: Use ARRAY_SIZE
by kbuild test robot 14 May '18

14 May '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 4844f51282b60f8a96b05d422f452a785df308b9 commit: 47cdf76e44e87d25b91c0f0d0539944c932941ba [62/68] ALSA: hda/ca0132: Add new control changes for SBZ + R3Di coccinelle warnings: (new ones prefixed by >>) >> sound/pci/hda/patch_ca0132.c:5062:50-51: WARNING: Use ARRAY_SIZE sound/pci/hda/patch_ca0132.c:5092:50-51: WARNING: Use ARRAY_SIZE Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 0/4] ALSA: control: optimization for position offset macro of TLV data
by Takashi Sakamoto 14 May '18

14 May '18

Hi, This patchset is a supplement for a commit 08f9f4485f21 ('ALSA: core api: define offsets for TLV items') for items below: - add position offset macro for db-minmax and db-linear types of TLV data - optimization with these macros Takashi Sakamoto (4): ALSA: control: complement TLV macro for db-minmax and db-linear types ALSA: vmaster: use position offset macro of TLV data ALSA: hda: use position offset macro of TLV data ALSA: isight: use position offset macro of TLV data include/uapi/sound/tlv.h | 8 ++++++++ sound/core/vmaster.c | 16 +++++++++------- sound/firewire/isight.c | 10 ++++++---- sound/pci/hda/hda_codec.c | 22 +++++++++++----------- sound/pci/hda/hda_generic.c | 2 +- 5 files changed, 35 insertions(+), 23 deletions(-) -- 2.14.1

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[alsa-devel] [PATCH] Clevo P950ER ALC1220 Fixup
by Jeremy Soller 14 May '18

14 May '18

This adds support for the P950ER, which has the same required fixup as the P950HR, but has a different PCI ID. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2dd34dd77447..01a6643fc7d4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2363,6 +2363,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530),

2 2

[alsa-devel] [PATCH v14 0/8] ASoC: sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 14 May '18

14 May '18

This patchset adds some mixer controls to sun4i-codec for the Allwinner A10 and the Allwinner A20. It also adds a mux for the capture source and the PGA for the MIC2 preamp. Where possible, it uses SOC_DAPM_DOUBLE in order to cut down on the number of distinct controls in alsamixer. v14 changes compared to v13 are: - Merged some of the patches together if it made sense to test them together. - Use snd_soc_component_driver. - Moved SUN4I_CODEC_DAC_ACTL_LFMS, SUN4I_CODEC_DAC_ACTL_RFMS to the correct patch in the series. - Kept "Left Mixer Left DAC Playback Switch", "Right Mixer Left DAC Playback Switch" and "Right Mixer Right DAC Playback Switch" unchanged compared to the released version - for backward compatibility. v13 changes compared to v12 are: - Added my "Signed-off-by". - Clarified some commit message text. v12 changes compared to v11 are: - Split up patchset in another way. - Renamed "Mic1 Capture Volume" to "Mic1 Boost Volume". - Renamed "Mic2 Capture Volume" to "Mic2 Boost Volume". - Renamed "Line Capture Volume" to "Line Boost Volume". - Renamed "Differential Line Capture Switch" to "Differential Line Source". v11 changes compared to v10 are: - Split up patchset. - Fixed typo in Differential Line Capture Switch. - Renamed "Non-Differential" value to "Stereo". - Removed duplicate PA Volume mixer control. v10 changes compared to v9 are: - Use SOC_DAPM_DOUBLE where possible and it makes sense in order to cut down on the number of controls. v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (8): ASoC: sun4i-codec: Add MIC2 Pre-Amplifier, Mic2, Mic Playback Volume ASoC: sun4i-codec: Add Mic1 Boost Volume, Mic2 Boost Volume ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls. ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch. ASoC: sun4i-codec: Add FM Playback Volume, FM Left, FM Right, FM Playback Switch ASoC: sun4i-codec: Add Line Playback Volume, Line Boost Volume, Line Right, Line Left, Line Playback Switch ASoC: sun4i-codec: Add Differential Line Source. ASoC: sun4i-codec: Add Left Capture Select, Right Capture Select. sound/soc/sunxi/sun4i-codec.c | 233 +++++++++++++++++++++++++++++++++++++----- 1 file changed, 209 insertions(+), 24 deletions(-)

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Re: [alsa-devel] [PATCH v2 09/11] docs: Fix some broken references
by Masami Hiramatsu 14 May '18

14 May '18

On Wed, 9 May 2018 10:18:52 -0300 Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> wrote: > As we move stuff around, some doc references are broken. Fix some of > them via this script: > ./scripts/documentation-file-ref-check --fix-rst > > Manually checked if the produced result is valid, removing a few > false-positives. The part of kprobes and trace_kprobe changes look good to me :) Acked-by: Masami Hiramatsu <mhiramat(a)kernel.org> Thank you, > > Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> > --- > .../admin-guide/kernel-parameters.txt | 4 ++-- > .../bindings/input/rotary-encoder.txt | 2 +- > Documentation/driver-api/gpio/consumer.rst | 2 +- > Documentation/kprobes.txt | 4 ++-- > Documentation/trace/coresight.txt | 2 +- > Documentation/trace/events.rst | 2 +- > Documentation/trace/ftrace-uses.rst | 2 +- > Documentation/trace/histogram.txt | 2 +- > Documentation/trace/intel_th.rst | 2 +- > Documentation/trace/tracepoint-analysis.rst | 6 +++--- > Documentation/translations/ja_JP/howto.rst | 4 ++-- > .../translations/zh_CN/magic-number.txt | 4 ++-- > .../zh_CN/video4linux/omap3isp.txt | 4 ++-- > MAINTAINERS | 20 +++++++++---------- > arch/Kconfig | 2 +- > arch/arm/include/asm/cacheflush.h | 2 +- > arch/arm64/include/asm/cacheflush.h | 2 +- > arch/microblaze/include/asm/cacheflush.h | 2 +- > arch/um/Kconfig.um | 2 +- > arch/unicore32/include/asm/cacheflush.h | 2 +- > arch/x86/entry/vsyscall/vsyscall_64.c | 2 +- > arch/xtensa/include/asm/cacheflush.h | 4 ++-- > block/Kconfig | 2 +- > certs/Kconfig | 2 +- > crypto/asymmetric_keys/asymmetric_type.c | 2 +- > crypto/asymmetric_keys/signature.c | 2 +- > drivers/char/Kconfig | 2 +- > drivers/clk/clk.c | 4 ++-- > drivers/clk/ingenic/cgu.h | 2 +- > drivers/gpu/vga/Kconfig | 2 +- > drivers/gpu/vga/vgaarb.c | 2 +- > drivers/input/joystick/Kconfig | 10 +++++----- > drivers/input/joystick/iforce/Kconfig | 2 +- > drivers/input/joystick/walkera0701.c | 2 +- > drivers/input/misc/Kconfig | 4 ++-- > drivers/input/misc/rotary_encoder.c | 2 +- > drivers/input/mouse/Kconfig | 6 +++--- > drivers/input/mouse/alps.c | 2 +- > drivers/input/touchscreen/wm97xx-core.c | 2 +- > drivers/lightnvm/pblk-rb.c | 2 +- > drivers/md/bcache/Kconfig | 2 +- > drivers/md/bcache/btree.c | 2 +- > drivers/md/bcache/extents.c | 2 +- > drivers/media/dvb-core/dvb_ringbuffer.c | 2 +- > drivers/media/pci/meye/Kconfig | 2 +- > drivers/media/platform/pxa_camera.c | 4 ++-- > .../soc_camera/sh_mobile_ceu_camera.c | 2 +- > drivers/media/radio/Kconfig | 2 +- > drivers/media/radio/si470x/Kconfig | 2 +- > drivers/media/usb/dvb-usb-v2/lmedm04.c | 2 +- > drivers/media/usb/zr364xx/Kconfig | 2 +- > drivers/parport/Kconfig | 6 +++--- > drivers/staging/media/bcm2048/TODO | 2 +- > include/keys/asymmetric-subtype.h | 2 +- > include/keys/asymmetric-type.h | 2 +- > include/linux/assoc_array.h | 2 +- > include/linux/assoc_array_priv.h | 2 +- > include/linux/circ_buf.h | 2 +- > include/linux/ftrace.h | 2 +- > include/linux/rculist_nulls.h | 2 +- > include/uapi/linux/prctl.h | 2 +- > include/xen/interface/io/kbdif.h | 2 +- > kernel/cgroup/cpuset.c | 2 +- > kernel/trace/Kconfig | 16 +++++++-------- > lib/Kconfig | 2 +- > security/selinux/hooks.c | 2 +- > sound/core/Kconfig | 4 ++-- > sound/drivers/Kconfig | 4 ++-- > sound/pci/Kconfig | 10 +++++----- > tools/include/uapi/linux/prctl.h | 2 +- > tools/lib/api/fs/fs.c | 2 +- > tools/memory-model/README | 10 +++++----- > tools/perf/util/bpf-prologue.c | 2 +- > .../config/custom-timeline-functions.cfg | 4 ++-- > 74 files changed, 120 insertions(+), 120 deletions(-) > -- Masami Hiramatsu <mhiramat(a)kernel.org>

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[alsa-devel] sound: aplay\arecord audio frame flow
by Muni Sekhar 14 May '18

14 May '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, Is there anyway in the kernel sound subsystem to know when the application calls ‘aplay’ and anything starts playing? Also for arecord\aplay, Is there any way from kernel's point of view to know when it receives for example the first and last audio frames? -- Thanks, Sekhar

1 0

[alsa-devel] [PATCH V2] ASoC: amd: modified dma pointer callback implementation
by Vijendar Mukunda 14 May '18

14 May '18

Modified dma pointer callback implementation. hw ptr calculated based on System Memory to ACP SRAM dma channel transfer count in case of playback. Increased min period size for playback to 4096 to avoid underruns. In case of capture, hw ptr calculated based on dma transfer count for ACP SRAM to System Memory Dma channel. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1->v2: removed hard coded values in acp dma pointer callback declared dma varaibles in rtd structure sound/soc/amd/acp-pcm-dma.c | 66 +++++++++++++++++++-------------------------- sound/soc/amd/acp.h | 15 +++-------- 2 files changed, 31 insertions(+), 50 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 906eb77..34a88ab 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -28,7 +28,7 @@ #define PLAYBACK_MIN_NUM_PERIODS 2 #define PLAYBACK_MAX_NUM_PERIODS 2 #define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 1024 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 #define CAPTURE_MIN_NUM_PERIODS 2 #define CAPTURE_MAX_NUM_PERIODS 2 #define CAPTURE_MAX_PERIOD_SIZE 16384 @@ -854,10 +854,9 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_BLUETOOTH; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH8; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH9; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_8; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_8; + rtd->end_dma_dscr = PLAYBACK_END_DMA_DESCR_CH8; adata->play_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -875,10 +874,9 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_ACP_I2S_1; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_12; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_12; + rtd->end_dma_dscr = PLAYBACK_END_DMA_DESCR_CH12; adata->play_i2ssp_stream = substream; } } else { @@ -891,10 +889,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + rtd->dma_cntl = mmACP_DMA_CNTL_10; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_10; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_10; + rtd->end_dma_dscr = CAPTURE_END_DMA_DESCR_CH10; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -914,10 +912,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + rtd->dma_cntl = mmACP_DMA_CNTL_14; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_14; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_14; + rtd->end_dma_dscr = CAPTURE_END_DMA_DESCR_CH14; adata->capture_i2ssp_stream = substream; } } @@ -955,22 +953,13 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static u64 acp_get_byte_count(struct audio_substream_data *rtd) -{ - union acp_dma_count byte_count; - - byte_count.bcount.high = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_high_reg_offset); - byte_count.bcount.low = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_low_reg_offset); - return byte_count.bytescount; -} - static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) { u32 buffersize; u32 pos = 0; - u64 bytescount = 0; + u32 dma_count = 0; + u16 dscr; + u32 period_bytes; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -979,11 +968,15 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) return -EINVAL; buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + period_bytes = frames_to_bytes(runtime, runtime->period_size); + if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) && + acp_reg_read(rtd->acp_mmio, rtd->dma_cntl) == 0) + return 0; + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + dma_count = acp_reg_read(rtd->acp_mmio, rtd->cur_trans_cnt); + if (dscr == rtd->end_dma_dscr) + dma_count += period_bytes; + pos = dma_count % buffersize; return bytes_to_frames(runtime, pos); } @@ -1016,7 +1009,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; u32 loops = 4000; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -1030,9 +1022,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & @@ -1066,7 +1055,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_stop(rtd->acp_mmio, rtd->ch2); ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); } - rtd->bytescount = 0; break; default: ret = -EINVAL; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96..4740804 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -131,12 +131,13 @@ struct audio_substream_data { u16 destination; u16 dma_dscr_idx_1; u16 dma_dscr_idx_2; + u16 end_dma_dscr; u32 pte_offset; u32 sram_bank; - u32 byte_cnt_high_reg_offset; - u32 byte_cnt_low_reg_offset; + u32 dma_cntl; + u32 dma_curr_dscr; + u32 cur_trans_cnt; uint64_t size; - u64 bytescount; void __iomem *acp_mmio; }; @@ -157,14 +158,6 @@ struct acp_platform_info { u16 i2s_instance; }; -union acp_dma_count { - struct { - u32 low; - u32 high; - } bcount; - u64 bytescount; -}; - enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- 2.7.4

1 0

[alsa-devel] [PATCH -next] ASoC: omap: add sdma-pcm.c MODULE_LICENSE
by Randy Dunlap 14 May '18

14 May '18

From: Randy Dunlap <rdunlap(a)infradead.org> ASoC: omap: add sdma-pcm.c MODULE_LICENSE Fixes this build warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/sdma-pcm.c | 3 +++ 1 file changed, 3 insertions(+) --- linux-next-20180510.orig/sound/soc/omap/sdma-pcm.c +++ linux-next-20180510/sound/soc/omap/sdma-pcm.c @@ -9,6 +9,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <linux/module.h> #include <linux/omap-dma.h> #include "sdma-pcm.h" @@ -66,3 +67,5 @@ int sdma_pcm_platform_register(struct de return devm_snd_dmaengine_pcm_register(dev, config, flags); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); + +MODULE_LICENSE("GPL v2");

2 1

Re: [alsa-devel] BUG: unable to handle kernel paging request in memset_erms (2)
by Eric Biggers 13 May '18

13 May '18

On Fri, Jan 19, 2018 at 01:58:01PM -0800, syzbot wrote: > Hello, > > syzbot hit the following crash on mmots commit > 2164355612187e55e8d60a28d2cc6b2337841a7e (Fri Jan 19 01:07:54 2018 +0000) > pci: test for unexpectedly disabled bridges > > So far this crash happened 2 times on mmots. > C reproducer is attached. > syzkaller reproducer is attached. > Raw console output is attached. > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached. > > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+29f08ad5cb6820798dfe(a)syzkaller.appspotmail.com > It will help syzbot understand when the bug is fixed. See footer for > details. > If you forward the report, please keep this part and the footer. > > BUG: unable to handle kernel paging request at ffffc90001691000 > IP: memset_erms+0x9/0x10 arch/x86/lib/memset_64.S:65 > PGD 1dad2c067 P4D 1dad2c067 PUD 1dad2d067 PMD 1c6a8f067 PTE 0 > Oops: 0002 [#1] SMP KASAN > Dumping ftrace buffer: > (ftrace buffer empty) > Modules linked in: > CPU: 1 PID: 5739 Comm: syzkaller592073 Not tainted 4.15.0-rc8-mm1+ #57 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS > Google 01/01/2011 > RIP: 0010:memset_erms+0x9/0x10 arch/x86/lib/memset_64.S:65 > RSP: 0018:ffff8801cbbdfb78 EFLAGS: 00010246 > RAX: fffff520002d3f00 RBX: ffffc90001691000 RCX: 000000000000ee51 > RDX: 000000000000ee51 RSI: 0000000000000000 RDI: ffffc90001691000 > RBP: ffff8801cbbdfb98 R08: fffff520002d3fcb R09: ffffc90001691000 > R10: 0000000000001dcb R11: fffff520002d3fca R12: 000000000000ee51 > R13: 0000000000000000 R14: 00007ffffffff000 R15: 000000002001be51 > FS: 00007f88ae7d7700(0000) GS:ffff8801db300000(0000) knlGS:0000000000000000 > CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 > CR2: ffffc90001691000 CR3: 00000001ccefa005 CR4: 00000000001606e0 > DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 > DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 > Call Trace: > memset include/linux/string.h:329 [inline] > _copy_from_user+0xe9/0x110 lib/usercopy.c:16 > copy_from_user include/linux/uaccess.h:147 [inline] > snd_pcm_oss_write1 sound/core/oss/pcm_oss.c:1347 [inline] > snd_pcm_oss_write+0x438/0x880 sound/core/oss/pcm_oss.c:2659 > __vfs_write+0xef/0x970 fs/read_write.c:480 > vfs_write+0x189/0x510 fs/read_write.c:544 > SYSC_write fs/read_write.c:589 [inline] > SyS_write+0xef/0x220 fs/read_write.c:581 > entry_SYSCALL_64_fastpath+0x29/0xa0 > RIP: 0033:0x44a559 > RSP: 002b:00007f88ae7d6da8 EFLAGS: 00000293 ORIG_RAX: 0000000000000001 > RAX: ffffffffffffffda RBX: 00000000006dcc24 RCX: 000000000044a559 > RDX: 000000000000fe51 RSI: 000000002000c000 RDI: 0000000000000003 > RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 > R10: 0000000000000000 R11: 0000000000000293 R12: 00000000006dcc20 > R13: 7073642f7665642f R14: 00800000c0045006 R15: 0000000000000001 > Code: 48 c1 e9 03 40 0f b6 f6 48 b8 01 01 01 01 01 01 01 01 48 0f af c6 f3 > 48 ab 89 d1 f3 aa 4c 89 c8 c3 90 49 89 f9 40 88 f0 48 89 d1 <f3> aa 4c 89 c8 > c3 90 49 89 fa 40 0f b6 ce 48 b8 01 01 01 01 01 > RIP: memset_erms+0x9/0x10 arch/x86/lib/memset_64.S:65 RSP: ffff8801cbbdfb78 > CR2: ffffc90001691000 > ---[ end trace 8f421641f3e10f44 ]--- > Kernel panic - not syncing: Fatal exception > Dumping ftrace buffer: > (ftrace buffer empty) > Kernel Offset: disabled > Rebooting in 86400 seconds.. > > > --- > This bug is generated by a dumb bot. It may contain errors. > See https://goo.gl/tpsmEJ for details. > Direct all questions to syzkaller(a)googlegroups.com. > > syzbot will keep track of this bug report. > If you forgot to add the Reported-by tag, once the fix for this bug is > merged > into any tree, please reply to this email with: > #syz fix: exact-commit-title The reproducer is unreliable, but this is another one that seems to have been fixed by commit 02a5d6925cd34: #syz fix: ALSA: pcm: Avoid potential races between OSS ioctls and read/write The reproducer is opening /dev/dsp1, then concurrently doing write() and ioctl(SNDCTL_DSP_CHANNELS). - Eric

1 0

[alsa-devel] [PATCH] ALSA: usb: mixer: volume quirk for CM102-A+/102S+
by Federico Cuello 13 May '18

13 May '18

Currently it's not possible to set volume lower than 26% (it just mutes). Also fixes this warning: Warning! Unlikely big volume range (=9472), cval->res is probably wrong. [13] FU [PCM Playback Volume] ch = 2, val = -9473/-1/1 , and volume works fine for full range. --- sound/usb/mixer.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 344d7b069d59..bb5ab7a7dfa5 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -967,6 +967,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; + case USB_ID(0x0d8c, 0x0103): + if (!strcmp(kctl->id.name, "PCM Playback Volume")) { + usb_audio_info(chip, + "set volume quirk for CM102-A+/102S+\n"); + cval->min = -256; + } + break; + case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): -- 2.17.0

2 3

[alsa-devel] [PATCH] ALSA: usb: mixer: volume quirk for CM102-A+/102S+
by Federico Cuello 13 May '18

13 May '18

Currently it's not possible to set volume lower than 26% (it just mutes). Also fixes this warning: Warning! Unlikely big volume range (=9472), cval->res is probably wrong. [13] FU [PCM Playback Volume] ch = 2, val = -9473/-1/1 , and volume works fine for full range. Signed-off-by: Federico Cuello <fedux(a)fedux.com.ar> --- sound/usb/mixer.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 344d7b069d59..bb5ab7a7dfa5 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -967,6 +967,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; + case USB_ID(0x0d8c, 0x0103): + if (!strcmp(kctl->id.name, "PCM Playback Volume")) { + usb_audio_info(chip, + "set volume quirk for CM102-A+/102S+\n"); + cval->min = -256; + } + break; + case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): -- 2.17.0

2 1

[alsa-devel] [PATCH] ALSA: hda: Add Lenovo C50 All in one to the power_save blacklist
by Hans de Goede 13 May '18

13 May '18

Power-saving is causing loud plops on the Lenovo C50 All in one, add it to the blacklist. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1572975 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0981c3c64d87..9a16af179503 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2214,6 +2214,8 @@ static struct snd_pci_quirk power_save_blacklist[] = { SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */ SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ + SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), {} -- 2.17.0

2 6

[alsa-devel] [PATCH] ALSA: usb-audio: add boot quirk for Axe-Fx III
by Alberto Aguirre 13 May '18

13 May '18

Wait for Axe-Fx III to fully bootup before initializing card. Signed-off-by: Alberto Aguirre <albaguirre(a)gmail.com> --- sound/usb/quirks.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5681767cc0d5..f4b69173682c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -851,6 +851,36 @@ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) return 0; /* Successful boot */ } +static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) +{ + int err; + + dev_dbg(&dev->dev, "Waiting for Axe-Fx III to boot up...\n"); + + /* If the Axe-Fx III has not fully booted, it will timeout when trying + * to enable the audio streaming interface. A more generous timeout is + * used here to detect when the Axe-Fx III has finished booting as the + * set interface message will be acked once it has + */ + err = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + USB_REQ_SET_INTERFACE, USB_RECIP_INTERFACE, + 1, 1, NULL, 0, 120000); + if (err < 0) { + dev_err(&dev->dev, + "failed waiting for Axe-Fx III to boot: %d\n", err); + return err; + } + + dev_dbg(&dev->dev, "Axe-Fx III is now ready\n"); + + err = usb_set_interface(dev, 1, 0); + if (err < 0) + dev_dbg(&dev->dev, + "error stopping Axe-Fx III interface: %d\n", err); + + return 0; +} + /* * Setup quirks */ @@ -1026,6 +1056,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_fasttrackpro_boot_quirk(dev); case USB_ID(0x047f, 0xc010): /* Plantronics Gamecom 780 */ return snd_usb_gamecon780_boot_quirk(dev); + case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ + return snd_usb_axefx3_boot_quirk(dev); } return 0; -- 2.17.0

2 1

[alsa-devel] [PATCH v4] ALSA: core api: define offsets for TLV items
by Ranjani Sridharan 13 May '18

13 May '18

Currently, there are no pre-defined accessors for the elements in topology TLV data. In the absence of such offsets, the tlv data will have to be decoded using hardwired offset numbers 0-N depending on the type of TLV. This patch defines accessor offsets for the type, length, min and mute/step items in TLV data for DB_SCALE type tlv's. These will be used by drivers to decode the TLV data while loading topology thereby improving code readability. The type and len offsets are common for all TLV types. The min and step/mute offsets are specific to DB_SCALE tlv type. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- Notes: v4: change placement of tlv accessor offsets include/uapi/sound/tlv.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index be5371f09a62..e3437e96519a 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -42,6 +42,10 @@ #define SNDRV_CTL_TLVD_LENGTH(...) \ ((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ })) +/* Accessor offsets for TLV data items */ +#define SNDRV_CTL_TLVO_TYPE 0 +#define SNDRV_CTL_TLVO_LEN 1 + #define SNDRV_CTL_TLVD_CONTAINER_ITEM(...) \ SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__) #define SNDRV_CTL_TLVD_DECLARE_CONTAINER(name, ...) \ @@ -61,6 +65,10 @@ SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \ } +/* Accessor offsets for min, mute and step items in dB scale type TLV */ +#define SNDRV_CTL_TLVO_DB_SCALE_MIN 2 +#define SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP 3 + /* dB scale specified with min/max values instead of step */ #define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \ SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB)) -- 2.14.1

3 2

[alsa-devel] [GIT PULL] soundwire updates
by Vinod 13 May '18

13 May '18

Hi Greg, As discussed here is the pull request containing SoundWire streaming patches and email update patch for me. I have added Mark's Ack on ASoC patch as well. The following changes since commit 60cc43fc888428bb2f18f08997432d426a243338: Linux 4.17-rc1 (2018-04-15 18:24:20 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git next for you to fetch changes up to c46302ec554c575e79d791c7c84fd4c795e97680: soundwire: intel: Add audio DAI ops (2018-05-11 21:48:07 +0530) ---------------------------------------------------------------- Sanyog Kale (7): Documentation: soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add support for port management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine soundwire: Add stream configuration APIs Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (5): soundwire: Update email address for Vinod soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 + Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 ++ Documentation/driver-api/soundwire/stream.rst | 372 +++++ MAINTAINERS | 2 +- drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 43 + drivers/soundwire/bus.h | 72 + drivers/soundwire/cadence_master.c | 451 +++++- drivers/soundwire/cadence_master.h | 151 ++ drivers/soundwire/intel.c | 525 ++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/stream.c | 1479 ++++++++++++++++++++ include/linux/soundwire/sdw.h | 332 ++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 23 + 18 files changed, 3634 insertions(+), 15 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/stream.c Thanks -- ~Vinod

2 3

[alsa-devel] KASAN: null-ptr-deref Write in linear_transfer
by syzbot 13 May '18

13 May '18

Hello, syzkaller hit the following crash on 30a7acd573899fd8b8ac39236eff6468b195ac7d git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master compiler: gcc (GCC) 7.1.1 20170620 .config is attached Raw console output is attached. C reproducer is attached syzkaller reproducer is attached. See https://goo.gl/kgGztJ for information about syzkaller reproducers IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+a8f5641f452c7e6abf03(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. ================================================================== BUG: KASAN: null-ptr-deref in memcpy include/linux/string.h:344 [inline] BUG: KASAN: null-ptr-deref in do_convert sound/core/oss/linear.c:52 [inline] BUG: KASAN: null-ptr-deref in convert sound/core/oss/linear.c:81 [inline] BUG: KASAN: null-ptr-deref in linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 Write of size 2 at addr (null) by task syzkaller360172/7860 CPU: 0 PID: 7860 Comm: syzkaller360172 Not tainted 4.15.0-rc6+ #155 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 kasan_report_error mm/kasan/report.c:349 [inline] kasan_report+0x13b/0x340 mm/kasan/report.c:409 check_memory_region_inline mm/kasan/kasan.c:260 [inline] check_memory_region+0x137/0x190 mm/kasan/kasan.c:267 memcpy+0x37/0x50 mm/kasan/kasan.c:303 memcpy include/linux/string.h:344 [inline] do_convert sound/core/oss/linear.c:52 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 snd_pcm_oss_sync1+0x1cc/0x550 sound/core/oss/pcm_oss.c:1530 snd_pcm_oss_sync+0x5b6/0x830 sound/core/oss/pcm_oss.c:1604 snd_pcm_oss_release+0x20b/0x280 sound/core/oss/pcm_oss.c:2431 __fput+0x327/0x7e0 fs/file_table.c:210 ____fput+0x15/0x20 fs/file_table.c:244 task_work_run+0x199/0x270 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x9bb/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 get_signal+0x73f/0x16c0 kernel/signal.c:2335 do_signal+0x90/0x1eb0 arch/x86/kernel/signal.c:809 exit_to_usermode_loop+0x214/0x310 arch/x86/entry/common.c:158 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:264 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:333 [inline] do_fast_syscall_32+0xbfd/0xf9d arch/x86/entry/common.c:389 entry_SYSENTER_compat+0x54/0x63 arch/x86/entry/entry_64_compat.S:129 RIP: 0023:0xf7fcac79 RSP: 002b:00000000f7f8412c EFLAGS: 00000292 ORIG_RAX: 00000000000000f0 RAX: fffffffffffffe00 RBX: 00000000080fd024 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 00000000080fd024 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 ================================================================== Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 7860 Comm: syzkaller360172 Tainted: G B 4.15.0-rc6+ #155 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 panic+0x1e4/0x41c kernel/panic.c:183 kasan_end_report+0x50/0x50 mm/kasan/report.c:176 kasan_report_error mm/kasan/report.c:356 [inline] kasan_report+0x144/0x340 mm/kasan/report.c:409 check_memory_region_inline mm/kasan/kasan.c:260 [inline] check_memory_region+0x137/0x190 mm/kasan/kasan.c:267 memcpy+0x37/0x50 mm/kasan/kasan.c:303 memcpy include/linux/string.h:344 [inline] do_convert sound/core/oss/linear.c:52 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 snd_pcm_oss_sync1+0x1cc/0x550 sound/core/oss/pcm_oss.c:1530 snd_pcm_oss_sync+0x5b6/0x830 sound/core/oss/pcm_oss.c:1604 snd_pcm_oss_release+0x20b/0x280 sound/core/oss/pcm_oss.c:2431 __fput+0x327/0x7e0 fs/file_table.c:210 ____fput+0x15/0x20 fs/file_table.c:244 task_work_run+0x199/0x270 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x9bb/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 get_signal+0x73f/0x16c0 kernel/signal.c:2335 do_signal+0x90/0x1eb0 arch/x86/kernel/signal.c:809 exit_to_usermode_loop+0x214/0x310 arch/x86/entry/common.c:158 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:264 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:333 [inline] do_fast_syscall_32+0xbfd/0xf9d arch/x86/entry/common.c:389 entry_SYSENTER_compat+0x54/0x63 arch/x86/entry/entry_64_compat.S:129 RIP: 0023:0xf7fcac79 RSP: 002b:00000000f7f8412c EFLAGS: 00000292 ORIG_RAX: 00000000000000f0 RAX: fffffffffffffe00 RBX: 00000000080fd024 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 00000000080fd024 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] [PATCH v5 00/13] ALSA: hda/ca0132: Patch Series for Recon3Di and Sound Blaster Z Support
by Connor McAdams 13 May '18

13 May '18

This patchset adds support for the Sound Blaster Z and the Recon3Di. In order to figure out how to get these cards to work, I made a program called QemuHDADump[1], which uses the trace function of qemu to see interactions with the memory mapped pci BAR space of the card being used in the virtual machine. With this, I obtain the CORB buffer location to get the command verbs, and then dump them each time the buffer rolls over. This program may be useful for fixing other HDA related driver issues where there is no documentation for the device. So far, I have been able to get all features supported on the Sound Blaster Z and the Recon3Di. All output and input effects work, all inputs and outputs work, and just about anything else I can think of. I have also added new controls in order to select the new inputs and outputs, as well as controls to change the effect levels and presets. I have also added the ability to use firmware taken from the Windows drivers of both the Sound Blaster Z and Recon3Di. I am trying to get into contact with Creative to get permission to redistribute these along with the current file included with the Chromebook, but they have not been very responsive. Luckily, the cards work with the Chromebook firmware just fine, although I believe there has to be a reason they have different firmware in Windows. I will not link to the firmwares here, but if you look up my thread on Creative Labs forums, you will find the link to download the firmwares there. I am willing to help get the other non-working cards such as the ZxR and the newer AE-5 working too, but I will need someone willing to run QemuHDADump in a virtual machine in order to get the commands. So, in summary: -This patchset makes the cards work better than they did before (they really didn't work before) -This patchset leaves the original chromebook related stuff alone. Thanks. [1] https://github.com/Conmanx360/QemuHDADump Bugs: ------------------------------------------------------------------------------- Recon3Di: (Reported by Mariusz Ceier) ******************* -Occasionally switching between rear and front mic breaks the input until computer is shutdown or put to sleep. -Surround Sound works, but is inconsistent. Sometimes, just updating the volume fixes it, and sometimes, it requires a restart. Sound Blaster Z: ******************* -none that I'm aware of. Version changes: ------------------------------------------------------------------------------- v1: ******************* -Massive patch formatting failure, please ignore v1. v2: ******************* -Fixed patch formatting failure. v3: ******************* -Fixed mem_base unmap, instead of checking for QUIRK_SBZ on exit, have it check if the area is mapped, and if it is, unmap it. Also make it unmap after all other commands are finished. -Change notification of failure to map mem_base from codec_dbg to codec_warn, and use codec_info to tell the user that their card might have been incorrectly identified as a Sound Blaster Z. -Remove commented out commands in sbz_exit_chip function, only reintroduce them when their functions are defined. v4: ******************* -Split patch into smaller pieces. -Added const to alt_out_presets array. -Fixed command that was commented out and only put it in when it was actually used. v5: ******************* -Fixed issue identified by kbuild test robot, where patch 12 didn't compile individually. Connor McAdams (13): ALSA: hda/ca0132: R3Di and SBZ quirk entires + alt firmware loading ALSA: hda/ca0132: Add pincfg for SBZ + R3Di, add fp hp auto-detect ALSA: hda/ca0132: Add PCI region2 iomap for SBZ ALSA: hda/ca0132: Add extra exit functions for R3Di and SBZ ALSA: hda/ca0132: add extra init functions for r3di + sbz ALSA: hda/ca0132: update core functions for sbz + r3di ALSA: hda/ca0132: add dsp setup related commands for the sbz ALSA: hda/ca0132: Add dsp setup + gpio functions for r3di ALSA: hda/ca0132: add the ability to set src_id on scp commands ALSA: hda/ca0132: add alt_select_in/out for R3Di + SBZ ALSA: hda/ca0132: Add DSP Volume set and New mixers for SBZ + R3Di ALSA: hda/ca0132: add ca0132_alt_set_vipsource ALSA: hda/ca0132: Add new control changes for SBZ + R3Di sound/pci/hda/patch_ca0132.c | 3057 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 2941 insertions(+), 116 deletions(-) -- 2.7.4

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Re: [alsa-devel] [PATCH v2 09/11] docs: Fix some broken references
by Takashi Iwai 13 May '18

13 May '18

On Wed, 09 May 2018 15:18:52 +0200, Mauro Carvalho Chehab wrote: > > As we move stuff around, some doc references are broken. Fix some of > them via this script: > ./scripts/documentation-file-ref-check --fix-rst > > Manually checked if the produced result is valid, removing a few > false-positives. > > Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> For sound stuff, feel free to take my ack (if it's not too late): Acked-by: Takashi Iwai <tiwai(a)suse.de> thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: control: fix a redundant-copy issue
by Wenwen Wang 13 May '18

13 May '18

In snd_ctl_elem_add_compat(), the fields of the struct 'data' need to be copied from the corresponding fields of the struct 'data32' in userspace. This is achieved by invoking copy_from_user() and get_user() functions. The problem here is that the 'type' field is copied twice. One is by copy_from_user() and one is by get_user(). Given that the 'type' field is not used between the two copies, the second copy is *completely* redundant and should be removed for better performance and cleanup. Also, these two copies can cause inconsistent data: as the struct 'data32' resides in userspace and a malicious userspace process can race to change the 'type' field between the two copies to cause inconsistent data. Depending on how the data is used in the future, such an inconsistency may cause potential security risks. For above reasons, we should take out the second copy. Signed-off-by: Wenwen Wang <wang6495(a)umn.edu> --- sound/core/control_compat.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index a848836..507fd52 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -396,8 +396,7 @@ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, if (copy_from_user(&data->id, &data32->id, sizeof(data->id)) || copy_from_user(&data->type, &data32->type, 3 * sizeof(u32))) goto error; - if (get_user(data->owner, &data32->owner) || - get_user(data->type, &data32->type)) + if (get_user(data->owner, &data32->owner)) goto error; switch (data->type) { case SNDRV_CTL_ELEM_TYPE_BOOLEAN: -- 2.7.4

2 1

[alsa-devel] [PATCH v2 0/7] USB Audio Device Class 3.0 BADD profiles support
by Ruslan Bilovol 13 May '18

13 May '18

This patchset adds BADD profiles support from the USB Audio Device Class 3.0 spec [1]. BADD profile support is defined as mandatory feature of UAC3-compliant device, it should be implemented as a separate USB configuration. Notable issue with BADD configuration is that it misses class-specific descriptors (and it's mandatory as per spec), so host should guess them from BADD profile number and parameters of endpoints (type, number of endpoints and max packet size) This patchset adds support of all known/existing BADD profiles from the UAC3 specification. First 5 patches are refactoring and improvements, and last 2 patches actually implement UAC3 BADD profiles support. It's an alternative implementation comparing to [2], and doesn't build usb descriptors on the host but instead initializes alsa-usb structures with known parameters, so we don't need to keep whole class-specific descriptors in the driver since we anyway need to have BADD-specific logic. I've picked one Jorge's UAC1 patch-improvement and updated it to v4.17 wich contais recently introduced header's sanity checks. Remaining part is to add interrupt endpoint support so we will be able to detect jack insertion in the Headset Adapter profile. This has been tested on ARM and x86-64 machines with custom UAC3 gadget which I'll post later to linux-usb Comments and testing are welcome. v2: - split refactoring patch to more atomic and bisectable changes as suggested by Takashi - renamed "Side Tone"->"Sidetone" as suggested by Andrew - revorked main BADD patch to have table lookup for checking BADD channels validity and to reduce identation as suggested by Takashi - also removed code duplication in BADD patch by reusing common part of build_feature_ctl() func v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-April/134412.html [1] http://www.usb.org/developers/docs/devclass_docs/USB_Audio_v3.0.zip [2] https://www.spinics.net/lists/alsa-devel/msg71614.html Jorge Sanjuan (1): ALSA: usb: Only get AudioControl header for UAC1 class. Ruslan Bilovol (6): ALSA: usb: stream: move audioformat alloc/init into separate function ALSA: usb: stream: refactor uac1/2 audio interface parsing ALSA: usb: stream: refactor uac3 audio interface parsing ALSA: usb: mixer: make string parsing independent of mixer_build state include: usb: audio-v3: add BADD-specific values ALSA: usb: add UAC3 BADD profiles support include/linux/usb/audio-v3.h | 26 ++ sound/usb/card.c | 53 ++-- sound/usb/clock.c | 9 +- sound/usb/mixer.c | 351 +++++++++++++++++++--- sound/usb/mixer_maps.c | 65 ++++ sound/usb/stream.c | 687 +++++++++++++++++++++++++------------------ sound/usb/usbaudio.h | 2 + 7 files changed, 849 insertions(+), 344 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2] ALSA: hda - Add missing HD-Audio codec information
by Shrikant Dharwadkar 13 May '18

13 May '18

This adds missing HD-Audio codec information such as GPIO info, power controls, volume capabilities from HDA spec. These additional info can be useful in debugging issues. Following are the info that are being added: Function group nodes -Add power state controls -Add subordinate node count parameters -Add function group type information -Add function group capabilities -Add unsolicited response controls -Add input converter SDI select information -Add GPIO enable control -Add GPIO direction information -Add GPIO wake enable information -Add GPIO unsolicited control information -Add GPIO sticky mask data -Add GPIO data sets associated with GPIO signals Widget nodes -Add pin widget controls -Add power state controls -Add converter stream and channel controls -Add processing state information -Add processing capabilities -Add volume knob capabilities -Add volume knob control information -Add digital converter control information -Add pin sense control information -Add EAPD/BTL enable controls -Add unsolicited response controls Signed-off-by: Shrikant Dharwadkar <shrikant.dharwadkar(a)intel.com> --- sound/hda/hdac_sysfs.c | 167 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 167 insertions(+) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index fb2aa34..b1acff2 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -37,6 +37,9 @@ static ssize_t type##_show(struct device *dev, \ } \ static DEVICE_ATTR_RO(type) +#define CODEC_PRINT(verb) sprintf(buf, "0x%08x\n", \ + snd_hdac_read_parm(codec, nid, verb)) + CODEC_ATTR(type); CODEC_ATTR(vendor_id); CODEC_ATTR(subsystem_id); @@ -173,6 +176,12 @@ static ssize_t pin_caps_show(struct hdac_device *codec, hda_nid_t nid, snd_hdac_read_parm(codec, nid, AC_PAR_PIN_CAP)); } +static ssize_t pin_widget_ctl_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PIN_WIDGET_CONTROL); +} + static ssize_t pin_cfg_show(struct hdac_device *codec, hda_nid_t nid, struct widget_attribute *attr, char *buf) { @@ -243,6 +252,14 @@ static ssize_t power_caps_show(struct hdac_device *codec, hda_nid_t nid, snd_hdac_read_parm(codec, nid, AC_PAR_POWER_STATE)); } +static ssize_t power_states_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (nid != codec->afg && !(get_wcaps(codec, nid) & AC_WCAP_POWER)) + return 0; + return CODEC_PRINT(AC_VERB_GET_POWER_STATE); +} + static ssize_t gpio_caps_show(struct hdac_device *codec, hda_nid_t nid, struct widget_attribute *attr, char *buf) { @@ -266,27 +283,166 @@ static ssize_t connections_show(struct hdac_device *codec, hda_nid_t nid, return ret; } +static ssize_t converter_stream_channel_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_CONV); +} + +static ssize_t processing_states_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PROC_STATE); +} + +static ssize_t subordinate_node_count_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_NODE_COUNT); +} + +static ssize_t function_group_type_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_FUNCTION_TYPE); +} + +static ssize_t processing_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_PROC_CAP); +} + +static ssize_t volume_knob_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_VOL_KNB_CAP); +} + +static ssize_t volume_knob_ctl_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_VOLUME_KNOB_CONTROL); +} + +static ssize_t digital_converter_ctl_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_DIGI_CONVERT_1); +} + +static ssize_t pin_sense_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PIN_SENSE); +} + +static ssize_t EAPD_BTL_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_EAPD_BTLENABLE); +} + +static ssize_t unsol_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_UNSOLICITED_RESPONSE); +} + +static ssize_t sdi_select_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_SDI_SELECT); +} + +static ssize_t gpio_enable_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_MASK); +} + +static ssize_t gpio_direction_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_DIRECTION); +} + +static ssize_t gpio_wake_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_WAKE_MASK); +} + +static ssize_t gpio_unsol_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK); +} + +static ssize_t gpio_sticky_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_STICKY_MASK); +} + +static ssize_t gpio_data_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_DATA); +} + static WIDGET_ATTR_RO(caps); static WIDGET_ATTR_RO(pin_caps); static WIDGET_ATTR_RO(pin_cfg); +static WIDGET_ATTR_RO(pin_widget_ctl); static WIDGET_ATTR_RO(pcm_caps); static WIDGET_ATTR_RO(pcm_formats); static WIDGET_ATTR_RO(amp_in_caps); static WIDGET_ATTR_RO(amp_out_caps); static WIDGET_ATTR_RO(power_caps); +static WIDGET_ATTR_RO(power_states); static WIDGET_ATTR_RO(gpio_caps); static WIDGET_ATTR_RO(connections); +static WIDGET_ATTR_RO(converter_stream_channel); +static WIDGET_ATTR_RO(processing_states); +static WIDGET_ATTR_RO(subordinate_node_count); +static WIDGET_ATTR_RO(function_group_type); +static WIDGET_ATTR_RO(processing_caps); +static WIDGET_ATTR_RO(volume_knob_caps); +static WIDGET_ATTR_RO(volume_knob_ctl); +static WIDGET_ATTR_RO(digital_converter_ctl); +static WIDGET_ATTR_RO(pin_sense); +static WIDGET_ATTR_RO(EAPD_BTL); +static WIDGET_ATTR_RO(unsol_caps); +static WIDGET_ATTR_RO(sdi_select); +static WIDGET_ATTR_RO(gpio_enable); +static WIDGET_ATTR_RO(gpio_direction); +static WIDGET_ATTR_RO(gpio_wake); +static WIDGET_ATTR_RO(gpio_unsol); +static WIDGET_ATTR_RO(gpio_sticky); +static WIDGET_ATTR_RO(gpio_data); static struct attribute *widget_node_attrs[] = { &wid_attr_caps.attr, &wid_attr_pin_caps.attr, + &wid_attr_pin_widget_ctl.attr, &wid_attr_pin_cfg.attr, &wid_attr_pcm_caps.attr, &wid_attr_pcm_formats.attr, &wid_attr_amp_in_caps.attr, &wid_attr_amp_out_caps.attr, &wid_attr_power_caps.attr, + &wid_attr_power_states.attr, &wid_attr_connections.attr, + &wid_attr_converter_stream_channel.attr, + &wid_attr_processing_states.attr, + &wid_attr_processing_caps.attr, + &wid_attr_volume_knob_caps.attr, + &wid_attr_volume_knob_ctl.attr, + &wid_attr_digital_converter_ctl.attr, + &wid_attr_pin_sense.attr, + &wid_attr_EAPD_BTL.attr, + &wid_attr_unsol_caps.attr, NULL, }; @@ -296,7 +452,18 @@ static struct attribute *widget_afg_attrs[] = { &wid_attr_amp_in_caps.attr, &wid_attr_amp_out_caps.attr, &wid_attr_power_caps.attr, + &wid_attr_power_states.attr, &wid_attr_gpio_caps.attr, + &wid_attr_subordinate_node_count.attr, + &wid_attr_function_group_type.attr, + &wid_attr_unsol_caps.attr, + &wid_attr_sdi_select.attr, + &wid_attr_gpio_enable.attr, + &wid_attr_gpio_direction.attr, + &wid_attr_gpio_wake.attr, + &wid_attr_gpio_unsol.attr, + &wid_attr_gpio_sticky.attr, + &wid_attr_gpio_data.attr, NULL, }; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: hisilicon: fix spelling mistake: "uknown" -> "unknown"
by Colin King 13 May '18

13 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/hisilicon/hi6210-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 07a57209e055..53344a3b7a60 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -498,7 +498,7 @@ static int hi6210_i2s_trigger(struct snd_pcm_substream *substream, int cmd, hi6210_i2s_txctrl(cpu_dai, 0); break; default: - dev_err(cpu_dai->dev, "uknown cmd\n"); + dev_err(cpu_dai->dev, "unknown cmd\n"); return -EINVAL; } return 0; -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Fix compile error
by Hans de Goede 13 May '18

13 May '18

Fix the compile error introduced by: "ASoC: Intel: bytcr_rt5640: Configure PLL1 before using it". Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Mark, feel free to squash this into the original commit --- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 493240194991..828be866fb00 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -210,14 +210,14 @@ static int byt_rt5640_prepare_and_enable_pll1(struct snd_soc_dai *codec_dai, } if (ret < 0) { - dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); + dev_err(codec_dai->component->dev, "can't set pll: %d\n", ret); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); + dev_err(codec_dai->component->dev, "can't set clock %d\n", ret); return ret; } -- 2.17.0

2 1

[alsa-devel] What is the meaning of #sound-dai-cells in a device tree?
by sdaau 11 May '18

11 May '18

Hi all, Apologies if this isn't the right forum (my thinking was, is that it is related to sound, and thus ALSA, more) - however, if it isn't, please let me know where I could forward this query (maybe devicetree mailing list ?) I have tried, but I really cannot find any documentation where I can read about, what does conceptually #sound-dai-cells in a device tree mean. Most I've found so far, is: linux_git$ grep -r 'sound-dai-cells' --exclude-dir='.git' arch/arm/boot/dts/sun7i-a20-cubietruck.dts: #sound-dai-cells = <0>; ... arch/arm/boot/dts/r8a7790.dtsi: * #sound-dai-cells is required arch/arm/boot/dts/r8a7790.dtsi: * Single DAI : #sound-dai-cells = <0>; <&rcar_sound>; arch/arm/boot/dts/r8a7790.dtsi: * Multi DAI : #sound-dai-cells = <1>; <&rcar_sound N>; ... Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt:- #sound-dai-cells: Should be set to 1 (for multi-dai) ... As far as I can tell: * if it is 1, then you can refer to "ports" (?) of a soundcard node, i.e. <&rcar_sound N>; * if it is 0, then you cannot refer to "ports", and you can only refer to <&rcar_sound> Also, it seems to appear only in these kernel source files: drivers/misc/atmel-ssc.c: if (!of_property_read_bool(np, "#sound-dai-cells")) drivers/misc/atmel-ssc.c: if (of_property_read_bool(ssc->pdev->dev.of_node, "#sound-dai-cells")) ... sound/soc/generic/simple-card.c:#define CELL "#sound-dai-cells" ... sound/soc/generic/simple-scu-card.c:#define CELL "#sound-dai-cells" ... sound/soc/soc-core.c: "#sound-dai-cells", 0, &args); sound/soc/soc-core.c: "#sound-dai-cells"); sound/soc/soc-core.c: "#sound-dai-cells", ... so, I guess, it only has meaning in what would be nodes under `target = <&sound>;` in a device tree - is that correct? But then - what is sound-dai-cells actually describing? Number of entries in `static struct snd_soc_dai_driver` of a (codec?) driver, if it is defined as an array? Or a number of DAI links? Could anyone offer a more proper conceptual description of what #sound-dai-cells is supposed to describe? Many thanks in advance for any pointers, s.

1 0

[alsa-devel] [PATCH v6 00/13] soundwire: Add stream support
by Vinod Koul 11 May '18

11 May '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Updates in v6: - Fix return error as pointed by Sanyog - Update stream state enum documentation as pointed by Pierre and few other nits Updated in v5: - Add todo for device-device changes in future and some comment updates based on feedback from Pierre Updates in v4: - Add more comments for sdw_release_master_stream() - Make sure ports are freed upon sdw_stream_remove_slave() - Clarify snd_soc_dai_set_sdw_stream() and setting of NULL upon release. Updates in v3: - Remove stream state transition checks and redefine stream states - Update bansk switch routines - Other nitpicks pointed out by Pierre. Updates in v2: - Make ASoC API inlined - Make stream states as states and not action - Update the direction enum - Fix some typos and comment updates *** SUBJECT HERE *** *** BLURB HERE *** Sanyog Kale (7): Documentation: soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add support for port management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine soundwire: Add stream configuration APIs Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (4): soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 + Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 ++ Documentation/driver-api/soundwire/stream.rst | 372 +++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 43 + drivers/soundwire/bus.h | 72 + drivers/soundwire/cadence_master.c | 451 +++++- drivers/soundwire/cadence_master.h | 151 ++ drivers/soundwire/intel.c | 525 ++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/stream.c | 1479 ++++++++++++++++++++ include/linux/soundwire/sdw.h | 332 ++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 23 + 17 files changed, 3633 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/stream.c -- 2.7.4

6 19

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Only disable inactive serializer
by Vishal Thanki 11 May '18

11 May '18

As a side effect of the following commit, the active TX serializer may get disabled which may result in distorted audio output. ASoC: davinci-mcasp: Add support for multichannel playback (2952b27e2e463b28d5c0f04000f96b968137ca42) For example, if a 4 channel I2S playback with two TX serializers is activated. Later on, if a recording of 2 channels, with only 1 RX serializer is started, which will also disable one of the TX serializer because max_active_serializers is only calculated for RX (recording) stream. This patch fixes this issue. Signed-off-by: Vishal Thanki <vishalthanki(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 79fc4aa..1f96c9d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -789,7 +789,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, rx_ser < max_active_serializers) { mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; - } else { + } else if (mcasp->serial_dir[i] == INACTIVE_MODE) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), SRMOD_INACTIVE, SRMOD_MASK); } -- 2.7.4

1 0

[alsa-devel] [asoc:for-4.18 139/159] sound/soc/intel/boards/bytcr_rt5640.c:177:20: error: 'struct snd_soc_dai' has no member named 'codec'
by kbuild test robot 11 May '18

11 May '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: 89027d9ebb5948392dd5a4ccc72d5a4eaa865537 commit: bcd9a325f0b0f407c4559779a94e802977c67274 [139/159] ASoC: Intel: bytcr_rt5640: Configure PLL1 before using it config: x86_64-federa-25 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout bcd9a325f0b0f407c4559779a94e802977c67274 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc/intel/boards/bytcr_rt5640.c: In function 'byt_rt5640_prepare_and_enable_pll1': >> sound/soc/intel/boards/bytcr_rt5640.c:177:20: error: 'struct snd_soc_dai' has no member named 'codec' dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); ^~ sound/soc/intel/boards/bytcr_rt5640.c:184:20: error: 'struct snd_soc_dai' has no member named 'codec' dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); ^~ vim +177 sound/soc/intel/boards/bytcr_rt5640.c 143 144 static int byt_rt5640_prepare_and_enable_pll1(struct snd_soc_dai *codec_dai, 145 int rate) 146 { 147 int ret; 148 149 /* Configure the PLL before selecting it */ 150 if (!(byt_rt5640_quirk & BYT_RT5640_MCLK_EN)) { 151 /* use bitclock as PLL input */ 152 if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || 153 (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { 154 /* 2x16 bit slots on SSP0 */ 155 ret = snd_soc_dai_set_pll(codec_dai, 0, 156 RT5640_PLL1_S_BCLK1, 157 rate * 32, rate * 512); 158 } else { 159 /* 2x15 bit slots on SSP2 */ 160 ret = snd_soc_dai_set_pll(codec_dai, 0, 161 RT5640_PLL1_S_BCLK1, 162 rate * 50, rate * 512); 163 } 164 } else { 165 if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) { 166 ret = snd_soc_dai_set_pll(codec_dai, 0, 167 RT5640_PLL1_S_MCLK, 168 25000000, rate * 512); 169 } else { 170 ret = snd_soc_dai_set_pll(codec_dai, 0, 171 RT5640_PLL1_S_MCLK, 172 19200000, rate * 512); 173 } 174 } 175 176 if (ret < 0) { > 177 dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); 178 return ret; 179 } 180 181 ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, 182 rate * 512, SND_SOC_CLOCK_IN); 183 if (ret < 0) { 184 dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); 185 return ret; 186 } 187 188 return 0; 189 } 190 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/4] ASoC: compress: Only assign compr->ops->copy once
by Charles Keepax 11 May '18

11 May '18

There are only one set of ops on the compressed stream so no need to reassign the copy callback repeatedly, stop after copy is seen to be necessary. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-compress.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ba56f87f96d4c..62875c6a93a14 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -973,6 +973,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) continue; compr->ops->copy = soc_compr_copy; + break; } -- 2.11.0

4 13

[alsa-devel] [PATCH v7 00/24] ASoC: qcom: Add support to QDSP based Audio
by Srinivas Kandagatla 11 May '18

11 May '18

Thankyou everyone for providing feedback and testing v6 patchset. This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.17-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Changes since v6 (https://lkml.org/lkml/2018/4/26/264) - Added relevant acks and reviewed-by tags. - Update apr_pkt_send and callbacks to take structures instead of void * suggested by Bjorn - Updated q6afe bindings to explain clearly on grandchild dai node bindings, suggested by Rob H. - Fixed few typos in APR commit msg spotted by Mark B. - rebased patchset on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/?h=for-ne… Srinivas Kandagatla (24): soc: qcom dt-bindings: Add APR bus bindings soc: qcom: Add APR bus driver ASoC: qdsp6: dt-bindings: Add q6core dt bindings ASoC: qdsp6: dt-bindings: Add q6afe dt bindings ASoC: qdsp6: dt-bindings: Add q6adm dt bindings ASoC: qdsp6: dt-bindings: Add q6asm dt bindings ASoC: qdsp6: q6common: Add qdsp6 helper functions ASoC: qdsp6: q6core: Add q6core driver ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver MAINTAINERS: Add myself as co-maintainer of qcom audio .../devicetree/bindings/soc/qcom/qcom,apr.txt | 84 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 109 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 33 + .../devicetree/bindings/sound/qcom,q6afe.txt | 104 ++ .../devicetree/bindings/sound/qcom,q6asm.txt | 33 + .../devicetree/bindings/sound/qcom,q6core.txt | 21 + MAINTAINERS | 1 + drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 378 ++++++ include/dt-bindings/soc/qcom,apr.h | 28 + include/dt-bindings/sound/qcom,q6afe.h | 31 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 128 ++ sound/soc/qcom/Kconfig | 50 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 238 ++++ sound/soc/qcom/qdsp6/Makefile | 8 + sound/soc/qcom/qdsp6/q6adm.c | 636 +++++++++ sound/soc/qcom/qdsp6/q6adm.h | 25 + sound/soc/qcom/qdsp6/q6afe-dai.c | 752 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1046 +++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 628 +++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1401 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 + sound/soc/qcom/qdsp6/q6core.c | 380 ++++++ sound/soc/qcom/qdsp6/q6core.h | 15 + sound/soc/qcom/qdsp6/q6dsp-common.c | 66 + sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 51 + sound/soc/qcom/qdsp6/q6routing.c | 983 ++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 34 files changed, 7572 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/dt-bindings/sound/qcom,q6afe.h create mode 100644 include/dt-bindings/sound/qcom,q6asm.h create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6core.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.16.2

6 61

[alsa-devel] Applied "ASoC: zx-i2s: fix spelling mistake: "timeing" -> "timing"" to the asoc tree
by Mark Brown 11 May '18

11 May '18

The patch ASoC: zx-i2s: fix spelling mistake: "timeing" -> "timing" has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d165b5a86fcad715aea1f164186224fe0c2f57f3 Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Thu, 10 May 2018 15:58:38 +0100 Subject: [PATCH] ASoC: zx-i2s: fix spelling mistake: "timeing" -> "timing" Trivial fix to spelling mistake in dev_err message text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/zte/zx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 9a0565937d1f..b6b056c607fa 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -197,7 +197,7 @@ static int zx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) val |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_LSB_JUSTIF); break; default: - dev_err(cpu_dai->dev, "Unknown i2s timeing\n"); + dev_err(cpu_dai->dev, "Unknown i2s timing\n"); return -EINVAL; } -- 2.17.0

1 0

[alsa-devel] Applied "ASoC: rt5640: Remove is_sys_clk_from_pll, it has ordering issues" to the asoc tree
by Mark Brown 11 May '18

11 May '18

The patch ASoC: rt5640: Remove is_sys_clk_from_pll, it has ordering issues has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8e7a1f1f177fef28cbd5edc663044b8946d08ab2 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Tue, 8 May 2018 17:35:46 +0200 Subject: [PATCH] ASoC: rt5640: Remove is_sys_clk_from_pll, it has ordering issues is_sys_clk_from_pll() is used as a snd_soc_dapm_route.connected callback, checking RT5640_GBL_CLK to determine if the sys-clk is PLL1 and thus the PWR_PLL bit in reg PWR_ANLG2 must be set. RT5640_GBL_CLK is changed by rt5640_set_dai_sysclk(), which gets called by the pre_pmu / post_pmd functions of the "Platform Clock" dapm-supply. This creates an ordering issue, during a dapm transition first all connected() callbacks are called to build a list of supplies to enable and then the complete list is walked to enable the supplies. Since the connected() check happens before enabling any supplies, is_sys_clk_from_pll() ends up deciding if the PWR_PLL bit should be set based on the state the "Platform Clock" supply had *before* the transition. This sometimes results in PWR_PLL being off, even though *after* the transition PLL1 is configured as sys-clk. This commit removes is_sys_clk_from_pll() instead simply setting / clearing PWR_PLL in rt5640_set_dai_sysclk() based on the selected sys-clk, which fixes this and as a bonus results in a nice cleanup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5640.c | 29 ++++------------------------- 1 file changed, 4 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 05567426f211..140bad07429e 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -476,20 +476,6 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, return idx; } -static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) -{ - struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm); - unsigned int val; - - val = snd_soc_component_read32(component, RT5640_GLB_CLK); - val &= RT5640_SCLK_SRC_MASK; - if (val == RT5640_SCLK_SRC_PLL1) - return 1; - else - return 0; -} - static int is_using_asrc(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -1071,9 +1057,6 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, } static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { - SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, - RT5640_PWR_PLL_BIT, 0, NULL, 0), - /* ASRC */ SND_SOC_DAPM_SUPPLY_S("Stereo Filter ASRC", 1, RT5640_ASRC_1, 15, 0, NULL, 0), @@ -1427,22 +1410,18 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Stereo ADC MIXL", "ADC1 Switch", "Stereo ADC L1 Mux"}, {"Stereo ADC MIXL", "ADC2 Switch", "Stereo ADC L2 Mux"}, {"Stereo ADC MIXL", NULL, "Stereo Filter"}, - {"Stereo Filter", NULL, "PLL1", is_sys_clk_from_pll}, {"Stereo ADC MIXR", "ADC1 Switch", "Stereo ADC R1 Mux"}, {"Stereo ADC MIXR", "ADC2 Switch", "Stereo ADC R2 Mux"}, {"Stereo ADC MIXR", NULL, "Stereo Filter"}, - {"Stereo Filter", NULL, "PLL1", is_sys_clk_from_pll}, {"Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux"}, {"Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux"}, {"Mono ADC MIXL", NULL, "Mono Left Filter"}, - {"Mono Left Filter", NULL, "PLL1", is_sys_clk_from_pll}, {"Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux"}, {"Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux"}, {"Mono ADC MIXR", NULL, "Mono Right Filter"}, - {"Mono Right Filter", NULL, "PLL1", is_sys_clk_from_pll}, {"IF2 ADC L", NULL, "Mono ADC MIXL"}, {"IF2 ADC R", NULL, "Mono ADC MIXR"}, @@ -1512,10 +1491,8 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, {"DAC L1", NULL, "Stereo DAC MIXL"}, - {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC L1", NULL, "DAC L1 Power"}, {"DAC R1", NULL, "Stereo DAC MIXR"}, - {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R1", NULL, "DAC R1 Power"}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, @@ -1622,10 +1599,8 @@ static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"DAC L2", NULL, "Mono DAC MIXL"}, - {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC L2", NULL, "DAC L2 Power"}, {"DAC R2", NULL, "Mono DAC MIXR"}, - {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R2", NULL, "DAC R2 Power"}, {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, @@ -1861,6 +1836,7 @@ static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, struct snd_soc_component *component = dai->component; struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + unsigned int pll_bit = 0; if (freq == rt5640->sysclk && clk_id == rt5640->sysclk_src) return 0; @@ -1871,6 +1847,7 @@ static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, break; case RT5640_SCLK_S_PLL1: reg_val |= RT5640_SCLK_SRC_PLL1; + pll_bit |= RT5640_PWR_PLL; break; case RT5640_SCLK_S_RCCLK: reg_val |= RT5640_SCLK_SRC_RCCLK; @@ -1879,6 +1856,8 @@ static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); return -EINVAL; } + snd_soc_component_update_bits(component, RT5640_PWR_ANLG2, + RT5640_PWR_PLL, pll_bit); snd_soc_component_update_bits(component, RT5640_GLB_CLK, RT5640_SCLK_SRC_MASK, reg_val); rt5640->sysclk = freq; -- 2.17.0

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[alsa-devel] Applied "ASoC: rt5640: Add devicetree-bindings for dmic, jack-detect" to the asoc tree
by Mark Brown 11 May '18

11 May '18

The patch ASoC: rt5640: Add devicetree-bindings for dmic, jack-detect has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e9e7a3bdcdf41efe9137dcf225b021e0b6fd2dc3 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Tue, 8 May 2018 17:35:47 +0200 Subject: [PATCH] ASoC: rt5640: Add devicetree-bindings for dmic, jack-detect Add devicetree-bindings for the dmic, jack-detect source and overcurrent- detect threshold settings. The dmic bindings mirror the existing bindings for the rt5645. The jd-src and ovcd bindings mirror the existing bindings for the rt5651. Cc devicetree(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/rt5640.txt | 35 +++++++++++++++++++ include/dt-bindings/sound/rt5640.h | 25 +++++++++++++ 2 files changed, 60 insertions(+) create mode 100644 include/dt-bindings/sound/rt5640.h diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index 57fe64643050..e40e4893eed8 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -22,6 +22,41 @@ Optional properties: - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN1P pin as dmic1 data pin + 2: using GPIO3 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN1N pin as dmic2 data pin + 2: using GPIO4 pin as dmic2 data pin + +- realtek,jack-detect-source + u32. Valid values: + 0: jack-detect is not used + 1: Use GPIO1 for jack-detect + 2: Use JD1_IN4P for jack-detect + 3: Use JD2_IN4N for jack-detect + 4: Use GPIO2 for jack-detect + 5: Use GPIO3 for jack-detect + 6: Use GPIO4 for jack-detect + +- realtek,jack-detect-not-inverted + bool. Normal jack-detect switches give an inverted signal, set this bool + in the rare case you've a jack-detect switch which is not inverted. + +- realtek,over-current-threshold-microamp + u32, micbias over-current detection threshold in µA, valid values are + 600, 1500 and 2000µA. + +- realtek,over-current-scale-factor + u32, micbias over-current detection scale-factor, valid values are: + 0: Scale current by 0.5 + 1: Scale current by 0.75 + 2: Scale current by 1.0 + 3: Scale current by 1.5 + Pins on the device (for linking into audio routes) for RT5639/RT5640: * DMIC1 diff --git a/include/dt-bindings/sound/rt5640.h b/include/dt-bindings/sound/rt5640.h new file mode 100644 index 000000000000..154c9b4414f2 --- /dev/null +++ b/include/dt-bindings/sound/rt5640.h @@ -0,0 +1,25 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_RT5640_H +#define __DT_RT5640_H + +#define RT5640_DMIC1_DATA_PIN_NONE 0 +#define RT5640_DMIC1_DATA_PIN_IN1P 1 +#define RT5640_DMIC1_DATA_PIN_GPIO3 2 + +#define RT5640_DMIC2_DATA_PIN_NONE 0 +#define RT5640_DMIC2_DATA_PIN_IN1N 1 +#define RT5640_DMIC2_DATA_PIN_GPIO4 2 + +#define RT5640_JD_SRC_GPIO1 1 +#define RT5640_JD_SRC_JD1_IN4P 2 +#define RT5640_JD_SRC_JD2_IN4N 3 +#define RT5640_JD_SRC_GPIO2 4 +#define RT5640_JD_SRC_GPIO3 5 +#define RT5640_JD_SRC_GPIO4 6 + +#define RT5640_OVCD_SF_0P5 0 +#define RT5640_OVCD_SF_0P75 1 +#define RT5640_OVCD_SF_1P0 2 +#define RT5640_OVCD_SF_1P5 3 + +#endif /* __DT_RT5640_H */ -- 2.17.0

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[alsa-devel] [PATCH v5 0/2] ASoC: omap: Deprecate omap-pcm for sdma-pcm
by Peter Ujfalusi 11 May '18

11 May '18

Hi, Changes since v4: - Really rebased on a clean tree - Dropped the already applied patches Changes since v3: - Rebased to make sure it applies cleanly (I had one local Kconfig patch) Changes since v2: - Return -ENODEV from sdma_pcm_platform_register() in case !CONFIG_SND_SDMA_SOC Changes since v1: - Rebased on the n810 patches (Kconfig conflict) - Delete the omap-pcm as the last step instead of keeping the .c/.h file. If there is a need it can be resurrected with a simple revert. - I left out the Acked-by from Jarkko for the last patch since I have changed it omap-pcm served us well over the years, but there is no reason to keep it around as we can rely on generic dmaengine_pcm code without the need to maintain custom code. The series has been tested on: omap2 n810, omap3 beagle-xm, omap4 PandaES, omap4 Blaze (SDP), am57xx beagle-x15, dra7 evm. McBSP, McASP, McPDM, DMIC and HDMI audio was tested and I have found no issue with them. I was not able to test on OMAP1 for various reasons, but mainly because the Nokia 770 audio is not supported in mainline... Regards, Peter --- Peter Ujfalusi (2): ASoC: davinci-mcasp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap: Delete the obsolete omap-pcm include/sound/omap-pcm.h | 30 ---- sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-mcasp.c | 8 +- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/Makefile | 2 - sound/soc/omap/omap-pcm.c | 262 ------------------------------ 6 files changed, 6 insertions(+), 301 deletions(-) delete mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 3

[alsa-devel] [PATCH v4 0/7] ASoC: omap: Deprecate omap-pcm for sdma-pcm
by Peter Ujfalusi 11 May '18

11 May '18

Hi, Changes since v3: - Rebased to make sure it applies cleanly (I had one local Kconfig patch) Changes since v2: - Return -ENODEV from sdma_pcm_platform_register() in case !CONFIG_SND_SDMA_SOC Changes since v1: - Rebased on the n810 patches (Kconfig conflict) - Delete the omap-pcm as the last step instead of keeping the .c/.h file. If there is a need it can be resurrected with a simple revert. - I left out the Acked-by from Jarkko for the last patch since I have changed it omap-pcm served us well over the years, but there is no reason to keep it around as we can rely on generic dmaengine_pcm code without the need to maintain custom code. The series has been tested on: omap2 n810, omap3 beagle-xm, omap4 PandaES, omap4 Blaze (SDP), am57xx beagle-x15, dra7 evm. McBSP, McASP, McPDM, DMIC and HDMI audio was tested and I have found no issue with them. I was not able to test on OMAP1 for various reasons, but mainly because the Nokia 770 audio is not supported in mainline... Regards, Peter --- Peter Ujfalusi (7): ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm ASoC: davinci-mcasp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-hdmi-audio: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-dmic: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcpdm: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcbsp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap: Delete the obsolete omap-pcm include/sound/omap-pcm.h | 30 ---- sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-mcasp.c | 8 +- sound/soc/omap/Kconfig | 27 +-- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-hdmi-audio.c | 5 +- sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/omap/omap-mcpdm.c | 4 +- sound/soc/omap/omap-pcm.c | 262 ------------------------------ sound/soc/omap/sdma-pcm.c | 68 ++++++++ sound/soc/omap/sdma-pcm.h | 21 +++ 12 files changed, 121 insertions(+), 318 deletions(-) delete mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.h -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 22

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add platform-data for Lenovo Ideapad Mixx 310
by Hans de Goede 11 May '18

11 May '18

The Lenovo Ideapad Mixx 310 has a differential internal analog mic, add platform-data for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc8d829ce45b..70f351c61d41 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3652,6 +3652,11 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { + .jd_mode = 3, + .in2_diff = true, +}; + static const struct rt5645_platform_data jd_mode3_platform_data = { .jd_mode = 3, }; @@ -3735,6 +3740,15 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&jd_mode3_platform_data, }, + { + .ident = "Lenovo Ideapad Miix 310", + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "80SG"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "MIIX 310-10ICR"), + }, + .driver_data = (void *)&lenovo_ideapad_miix_310_pdata, + }, { } }; -- 2.17.0

2 3

[alsa-devel] [PATCH 0/5] ASoC: cirrus: Series of fixes for ep93xx-i2s
by Alexander Sverdlin 11 May '18

11 May '18

This series is a result of long-running debugging of rarely screwed I2S stream out of EP9302 chip. Logic analyser shows that the stream is shifted by one byte (out of 32 bits per sample) which leads to noise intead of original stream. DMA is verified to work fine (no underrun). The issue is only reproducible under high load (CPU or AMBA bus -- not yet clear). This stream corruption happens at the same time I2S controller reports internal FIFO underrun (even though according to documentation, it should handle an underrun gracefully). First two patches are fixing unrelated issues in I2S driver found during debugging of the stream shift. Third one is simplification of the driver to prepare for workaround. Both controller configuration options for TX FIFO underrun handling were tested, both do not work reliable. Hence, this ugly watchdog in the last patch. The whole series is tested on a board similar to EDB9302 (identical in sound part). Alexander Sverdlin (5): ASoC: cirrus: i2s: Fix LRCLK configuration ASoC: cirrus: i2s: Fix {TX|RX}LinCtrlData setup ASoC: cirrus: i2s: Stop enabling I2S2 and I2S3 FIFOs ARM: ep93xx: i2s: Add IRQ to platform device resources ASoC: cirrus: i2s: IRQ-based stream watchdog arch/arm/mach-ep93xx/core.c | 1 + sound/soc/cirrus/Kconfig | 17 +++++++ sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 106 +++++++++++++++++++++++++++++++++-------- sound/soc/cirrus/snappercl15.c | 2 +- 5 files changed, 106 insertions(+), 22 deletions(-) -- 2.16.2

2 10

[alsa-devel] [PATCH 1/2] ASoC: nau8810: change input PGA mixer stage
by John Hsu 11 May '18

11 May '18

Organize the paths of the mixer, "Input Boost Stage", including the routes of the mixer. The control is not used correctly before. Besides, the driver changes the name of the mixer controls. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index ca2ba1c..bfd74b8 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -373,9 +373,11 @@ static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { }; /* PGA Mute */ -static const struct snd_kcontrol_new nau8810_inpga_mute[] = { +static const struct snd_kcontrol_new nau8810_pgaboost_mixer_controls[] = { SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, - NAU8810_PGAMT_SFT, 1, 0), + NAU8810_PGAMT_SFT, 1, 1), + SOC_DAPM_SINGLE("PMIC PGA Switch", NAU8810_REG_ADCBOOST, + NAU8810_PMICBSTGAIN_SFT, 0x7, 0), }; /* Input PGA */ @@ -386,11 +388,6 @@ static const struct snd_kcontrol_new nau8810_inpga[] = { NAU8810_PMICPGA_SFT, 1, 0), }; -/* Mic Input boost vol */ -static const struct snd_kcontrol_new nau8810_mic_boost_controls = - SOC_DAPM_SINGLE("Mic Volume", NAU8810_REG_ADCBOOST, - NAU8810_PMICBSTGAIN_SFT, 0x7, 0); - /* Loopback Switch */ static const struct snd_kcontrol_new nau8810_loopback = SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, @@ -429,8 +426,8 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { NAU8810_PGA_EN_SFT, 0, nau8810_inpga, ARRAY_SIZE(nau8810_inpga)), SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, - NAU8810_BST_EN_SFT, 0, nau8810_inpga_mute, - ARRAY_SIZE(nau8810_inpga_mute)), + NAU8810_BST_EN_SFT, 0, nau8810_pgaboost_mixer_controls, + ARRAY_SIZE(nau8810_pgaboost_mixer_controls)), SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), @@ -469,8 +466,8 @@ static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { /* Input Boost Stage */ {"ADC", NULL, "Input Boost Stage"}, {"ADC", NULL, "PLL", check_mclk_select_pll}, - {"Input Boost Stage", NULL, "Input PGA"}, - {"Input Boost Stage", NULL, "MICP"}, + {"Input Boost Stage", "PGA Mute Switch", "Input PGA"}, + {"Input Boost Stage", "PMIC PGA Switch", "MICP"}, /* Input PGA */ {"Input PGA", NULL, "Mic Bias"}, -- 2.6.4

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[alsa-devel] [PATCH] ASoC: zx-i2s: fix spelling mistake: "timeing" -> "timing"
by Colin King 10 May '18

10 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/zte/zx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 9a0565937d1f..b6b056c607fa 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -197,7 +197,7 @@ static int zx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) val |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_LSB_JUSTIF); break; default: - dev_err(cpu_dai->dev, "Unknown i2s timeing\n"); + dev_err(cpu_dai->dev, "Unknown i2s timing\n"); return -EINVAL; } -- 2.17.0

1 0

[alsa-devel] [PATCH] ASoC: amd: modified dma pointer callback implementation
by Vijendar Mukunda 10 May '18

10 May '18

Modified dma pointer callback implementation. hw ptr calculated based on System Memory to ACP SRAM dma channel transfer count in case of playback. Increased min period size for playback to 4096 to avoid underruns. In case of capture, hw ptr calculated based on dma transfer count for ACP SRAM to System Memory Dma channel. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- This patch is dependent on [V4,10/10] ASoC: amd: dma driver changes for bt i2s instance https://patchwork.kernel.org/patch/10385381/ sound/soc/amd/acp-pcm-dma.c | 94 ++++++++++++++++++++++++++------------------- sound/soc/amd/acp.h | 11 ------ 2 files changed, 55 insertions(+), 50 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 906eb77..969c46c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -28,7 +28,7 @@ #define PLAYBACK_MIN_NUM_PERIODS 2 #define PLAYBACK_MAX_NUM_PERIODS 2 #define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 1024 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 #define CAPTURE_MIN_NUM_PERIODS 2 #define CAPTURE_MAX_NUM_PERIODS 2 #define CAPTURE_MAX_PERIOD_SIZE 16384 @@ -854,10 +854,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_BLUETOOTH; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH8; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH9; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_LOW; adata->play_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -875,10 +871,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_ACP_I2S_1; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_LOW; adata->play_i2ssp_stream = substream; } } else { @@ -891,10 +883,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -914,10 +902,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_LOW; adata->capture_i2ssp_stream = substream; } } @@ -955,22 +939,14 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static u64 acp_get_byte_count(struct audio_substream_data *rtd) -{ - union acp_dma_count byte_count; - - byte_count.bcount.high = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_high_reg_offset); - byte_count.bcount.low = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_low_reg_offset); - return byte_count.bytescount; -} - static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) { u32 buffersize; u32 pos = 0; - u64 bytescount = 0; + u32 dma_count = 0; + u32 mul = 0; + u16 dscr; + u32 dma_config, period_bytes; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -979,11 +955,56 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) return -EINVAL; buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + period_bytes = frames_to_bytes(runtime, runtime->period_size); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + dscr = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_DSCR_8); + dma_count = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_TRANS_CNT_8); + if (dscr == PLAYBACK_END_DMA_DESCR_CH8) + mul = 1; + break; + case I2S_SP_INSTANCE: + default: + dscr = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_DSCR_12); + dma_count = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_TRANS_CNT_12); + if (dscr == PLAYBACK_END_DMA_DESCR_CH12) + mul = 1; + break; + } + } else { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + dma_config = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CNTL_10); + if (dma_config != 0) { + dscr = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_DSCR_10); + dma_count = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_TRANS_CNT_10); + if (dscr == CAPTURE_END_DMA_DESCR_CH10) + mul = 1; + } + break; + case I2S_SP_INSTANCE: + default: + dma_config = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CNTL_14); + if (dma_config != 0) { + dscr = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_DSCR_14); + dma_count = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_TRANS_CNT_14); + if (dscr == CAPTURE_END_DMA_DESCR_CH14) + mul = 1; + } + } + } + pos = ((mul * period_bytes) + dma_count) % buffersize; return bytes_to_frames(runtime, pos); } @@ -1016,7 +1037,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; u32 loops = 4000; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -1030,9 +1050,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & @@ -1066,7 +1083,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_stop(rtd->acp_mmio, rtd->ch2); ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); } - rtd->bytescount = 0; break; default: ret = -EINVAL; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96..b6d52ef 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -133,10 +133,7 @@ struct audio_substream_data { u16 dma_dscr_idx_2; u32 pte_offset; u32 sram_bank; - u32 byte_cnt_high_reg_offset; - u32 byte_cnt_low_reg_offset; uint64_t size; - u64 bytescount; void __iomem *acp_mmio; }; @@ -157,14 +154,6 @@ struct acp_platform_info { u16 i2s_instance; }; -union acp_dma_count { - struct { - u32 low; - u32 high; - } bcount; - u64 bytescount; -}; - enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- 2.7.4

1 0

[alsa-devel] [PATCH v3] ALSA: core api: define offsets for TLV items
by Ranjani Sridharan 10 May '18

10 May '18

Currently, there are no pre-defined accessors for the elements in topology TLV data. In the absence of such offsets, the tlv data will have to be decoded using hardwired offset numbers 0-N depending on the type of TLV. This patch defines accessor offsets for the type, length, min and mute/step items in TLV data for DB_SCALE type tlv's. These will be used by drivers to decode the TLV data while loading topology thereby improving code readability. The type and len offsets are common for all TLV types. The min and step/mute offsets are specific to DB_SCALE tlv type. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- Notes: v3: fix prefix for offset definitions include/uapi/sound/tlv.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index be5371f09a62..13f425c2abbd 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -98,4 +98,12 @@ #define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999 +/* Accessor offsets for TLV data items */ +#define SNDRV_CTL_TLVO_TYPE 0 +#define SNDRV_CTL_TLVO_LEN 1 + +/* Accessor offsets for min, mute and step items in dB scale type TLV */ +#define SNDRV_CTL_TLVO_DB_SCALE_MIN 2 +#define SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP 3 + #endif -- 2.14.1

2 2

[alsa-devel] [PATCH 1/3] ASoC: pcm_native: Use 'snd_pcm_format_t' type
by Fabio Estevam 09 May '18

09 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> 'snd_pcm_format_t' type is better suited for iterating through the SNDRV_PCM_FORMAT members. Also, use SNDRV_PCM_FORMAT_FIRST for the first element. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/core/pcm_native.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7585444..5d62b7c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2051,13 +2051,13 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - unsigned int k; + snd_pcm_format_t k; const struct snd_interval *i = hw_param_interval_c(params, rule->deps[0]); struct snd_mask m; struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_any(&m); - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + for (k = SNDRV_PCM_FORMAT_FIRST; k <= SNDRV_PCM_FORMAT_LAST; ++k) { int bits; if (! snd_mask_test(mask, k)) continue; @@ -2074,12 +2074,12 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; - unsigned int k; + snd_pcm_format_t k; t.min = UINT_MAX; t.max = 0; t.openmin = 0; t.openmax = 0; - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + for (k = SNDRV_PCM_FORMAT_FIRST; k <= SNDRV_PCM_FORMAT_LAST; ++k) { int bits; if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) continue; -- 2.7.4

5 9

[alsa-devel] [PATCH 0/7] ASoC: omap: Deprecate omap-pcm for sdma-pcm
by Peter Ujfalusi 09 May '18

09 May '18

Hi, omap-pcm served us well over the years, but there is no reason to keep it around as we can rely on generic dmaengine_pcm code without the need to maintain custom code. The series has been tested on: omap3 beagle-xm, omap4 PandaES, omap4 Blaze (SDP), am57xx beagle-x15, dra7 evm. McBSP, McASP, McPDM, DMIC and HDMI audio was tested and I have found no issue with them. I was not able to test on OMAP1 or OMAP2 for various reasons, but mainly because neither Nokia 770 or Nokia n810 audio probes with linux-next. It does not matter if I have omap-pcm or sdma-pcm... I would be grateful if someone can give tested-by or pointers on how to get 770/n810 audio working. Regards, Peter --- Peter Ujfalusi (7): ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm ASoC: davinci-mcasp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-hdmi-audio: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-dmic: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcpdm: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcbsp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap: Mark the omap-pcm as deprecated, but keep it in the tree sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-mcasp.c | 8 ++-- sound/soc/omap/Kconfig | 33 ++++++++------- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-hdmi-audio.c | 5 ++- sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/omap/omap-mcpdm.c | 4 +- sound/soc/omap/sdma-pcm.c | 68 +++++++++++++++++++++++++++++++ sound/soc/omap/sdma-pcm.h | 21 ++++++++++ 10 files changed, 124 insertions(+), 29 deletions(-) create mode 100644 sound/soc/omap/sdma-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.h -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

4 17

[alsa-devel] Prepare to covert Codec/Platform into Component
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark, Lars-Peter I will post all prepare patches to convert Codec/Platform into Component. It has 3 steps. 1) add new necessary functions 2) remove/don't use codec hw_write/read 3) remove/don't use codec reg_cache If these are accepted, all prepares were completed. Next, I can post "convert Codec/Platform into Component" patch bomb. These are too many convert patches, to avoid review issue, I will post few of them as sample. 4) sample of don't use rtd->codec 5) sample of convert Platform into Component 6) sample of convert Codec into Component I will post some of "remove" patches too for each steps. I will re-post all "remove" patches if all are accepted. For your review, please consider below. If these are not good, all covert patch doesn't work. 1. use devm_snd_soc_register_component() and remove snd_soc_unregister_component() My patches is using devm_ version function, and removed unregister function. This is not a big deal 2. New flags. Because "Codec" and "CPU/Platform" are using different features, we need to use some new flags. .idle_bias_off/on .ignore_pmdown_time/pmdown_time These are because "CPU/Platform" and "Codec" are using different default value. "Codec" side convert patches are using these. .endianness .non_legacy_dai_naming These are because "Codec" only needs fixup format, and not use "legacy" DAI name. "Codec" side convert patches are using these I added its flags info on each patch. 3. rtd->platform rtd->platform will be removed. Now we can find all rtd connected component by using rtdcom_lookup. rtd->platform will be replaced to snd_soc_rtdcom_lookup(rtd, DRV_NAME); Here, DRV_NAME is platform's component driver name. 4. rtd->codec rtd->codec will be removed. In soc_bind_dai_link(), It is rtd->codec = rtd->codec_dai->codec; And now, DAI has codec and component pointer. So, rtd->codec will be replaced into rtd->codec_dai->component. Best regards --- Kuninori Morimoto

2 104

[alsa-devel] [PATCH 0/5][RFC] ASoC: remove Codec/Platform code
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark Now, all Codec/Platform are replaced to Component. Codec/Platform related code are no longer needed, these patches remove these. But I added [RFC] on these patches because, 1) people want to check/test 2) Media related added new driver is using Codec/Platform style I'm posting replace patch for 2) to Media / ASoC ML, but not yet accepted. (= I think it needs Mark's Acked-by ? I don't know) I think this Codec/Platform remove patch should based on it, otherwise, compile error happen. I'm happy to re-post this remove patch if Media side replace was accepted. Subject: [alsa-devel] [PATCH v3] media: i2c: tda1997: replace codec to component Date: Fri, 6 Apr 2018 05:44:25 +0000 Best regards --- Kuninori Morimoto

3 8

[alsa-devel] Applied "ASoC: remove Codec related code" to the asoc tree
by Mark Brown 09 May '18

09 May '18

The patch ASoC: remove Codec related code has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 999f7f5af8eb7766f68d925a22bf296011abc84c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 8 May 2018 03:20:24 +0000 Subject: [PATCH] ASoC: remove Codec related code Now no one is using Codec related code. Let's remove all Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-dai.h | 1 - include/sound/soc.h | 265 -------------------------------- sound/soc/soc-core.c | 331 ++-------------------------------------- sound/soc/soc-io.c | 62 -------- sound/soc/soc-jack.c | 22 --- sound/soc/soc-pcm.c | 5 - 6 files changed, 11 insertions(+), 675 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 35ebb0be5114..568f6a72c974 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -313,7 +313,6 @@ struct snd_soc_dai { unsigned int sample_bits; /* parent platform/codec */ - struct snd_soc_codec *codec; struct snd_soc_component *component; /* CODEC TDM slot masks and params (for fixup) */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 131185563532..21861f366dcb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -402,8 +402,6 @@ struct snd_soc_pcm_runtime; struct snd_soc_dai; struct snd_soc_dai_driver; struct snd_soc_dai_link; -struct snd_soc_codec; -struct snd_soc_codec_driver; struct snd_soc_component; struct snd_soc_component_driver; struct soc_enum; @@ -428,13 +426,6 @@ enum snd_soc_card_subclass { SND_SOC_CARD_CLASS_RUNTIME = 1, }; -int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - int source, unsigned int freq, int dir); -int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, - unsigned int freq_in, unsigned int freq_out); -int snd_soc_codec_set_jack(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, void *data); - int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); @@ -453,10 +444,6 @@ static inline int snd_soc_resume(struct device *dev) } #endif int snd_soc_poweroff(struct device *dev); -int snd_soc_register_codec(struct device *dev, - const struct snd_soc_codec_driver *codec_drv, - struct snd_soc_dai_driver *dai_drv, int num_dai); -void snd_soc_unregister_codec(struct device *dev); int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -559,23 +546,7 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, } #endif -/* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, - unsigned int mask, unsigned int value); -int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned int reg, unsigned int mask, - unsigned int value); -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, - unsigned int mask, unsigned int value); - #ifdef CONFIG_SND_SOC_AC97_BUS -#define snd_soc_alloc_ac97_codec(codec) \ - snd_soc_alloc_ac97_component(&codec->component) -#define snd_soc_new_ac97_codec(codec, id, id_mask) \ - snd_soc_new_ac97_component(&codec->component, id, id_mask) -#define snd_soc_free_ac97_codec(ac97) \ - snd_soc_free_ac97_component(ac97) - struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); @@ -609,8 +580,6 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); int snd_soc_add_component_controls(struct snd_soc_component *component, const struct snd_kcontrol_new *controls, unsigned int num_controls); -int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_card_controls(struct snd_soc_card *soc_card, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_dai_controls(struct snd_soc_dai *dai, @@ -843,8 +812,6 @@ struct snd_soc_component { unsigned int active; - unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ - unsigned int registered_as_component:1; unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; @@ -856,9 +823,6 @@ struct snd_soc_component { struct list_head dai_list; int num_dai; - int (*read)(struct snd_soc_component *, unsigned int, unsigned int *); - int (*write)(struct snd_soc_component *, unsigned int, unsigned int); - struct regmap *regmap; int val_bytes; @@ -880,22 +844,6 @@ struct snd_soc_component { /* Don't use these, use snd_soc_component_get_dapm() */ struct snd_soc_dapm_context dapm; - struct snd_soc_codec *codec; - - int (*probe)(struct snd_soc_component *); - void (*remove)(struct snd_soc_component *); - int (*suspend)(struct snd_soc_component *); - int (*resume)(struct snd_soc_component *); - - int (*set_sysclk)(struct snd_soc_component *component, - int clk_id, int source, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_component *component, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out); - int (*set_jack)(struct snd_soc_component *component, - struct snd_soc_jack *jack, void *data); - int (*set_bias_level)(struct snd_soc_component *component, - enum snd_soc_bias_level level); - /* machine specific init */ int (*init)(struct snd_soc_component *component); @@ -916,50 +864,6 @@ snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, #define for_each_rtdcom_safe(rtd, rtdcom1, rtdcom2) \ list_for_each_entry_safe(rtdcom1, rtdcom2, &(rtd)->component_list, list) -/* SoC Audio Codec device */ -struct snd_soc_codec { - struct device *dev; - const struct snd_soc_codec_driver *driver; - - struct list_head list; - - /* component */ - struct snd_soc_component component; -}; - -/* codec driver */ -struct snd_soc_codec_driver { - - /* driver ops */ - int (*probe)(struct snd_soc_codec *); - int (*remove)(struct snd_soc_codec *); - int (*suspend)(struct snd_soc_codec *); - int (*resume)(struct snd_soc_codec *); - struct snd_soc_component_driver component_driver; - - /* codec wide operations */ - int (*set_sysclk)(struct snd_soc_codec *codec, - int clk_id, int source, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, - unsigned int freq_in, unsigned int freq_out); - int (*set_jack)(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, void *data); - - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - - /* codec bias level */ - int (*set_bias_level)(struct snd_soc_codec *, - enum snd_soc_bias_level level); - bool idle_bias_off; - bool suspend_bias_off; - - void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type, int); - - bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ -}; - struct snd_soc_dai_link_component { const char *name; struct device_node *of_node; @@ -1213,7 +1117,6 @@ struct snd_soc_pcm_runtime { /* runtime devices */ struct snd_pcm *pcm; struct snd_compr *compr; - struct snd_soc_codec *codec; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; @@ -1281,19 +1184,6 @@ struct soc_enum { struct snd_soc_dobj dobj; }; -/** - * snd_soc_component_to_codec() - Casts a component to the CODEC it is embedded in - * @component: The component to cast to a CODEC - * - * This function must only be used on components that are known to be CODECs. - * Otherwise the behavior is undefined. - */ -static inline struct snd_soc_codec *snd_soc_component_to_codec( - struct snd_soc_component *component) -{ - return container_of(component, struct snd_soc_codec, component); -} - /** * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is * embedded in @@ -1309,19 +1199,6 @@ static inline struct snd_soc_component *snd_soc_dapm_to_component( return container_of(dapm, struct snd_soc_component, dapm); } -/** - * snd_soc_dapm_to_codec() - Casts a DAPM context to the CODEC it is embedded in - * @dapm: The DAPM context to cast to the CODEC - * - * This function must only be used on DAPM contexts that are known to be part of - * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined. - */ -static inline struct snd_soc_codec *snd_soc_dapm_to_codec( - struct snd_soc_dapm_context *dapm) -{ - return snd_soc_component_to_codec(snd_soc_dapm_to_component(dapm)); -} - /** * snd_soc_component_get_dapm() - Returns the DAPM context associated with a * component @@ -1333,31 +1210,6 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( return &component->dapm; } -/** - * snd_soc_codec_get_dapm() - Returns the DAPM context for the CODEC - * @codec: The CODEC for which to get the DAPM context - * - * Note: Use this function instead of directly accessing the CODEC's dapm field - */ -static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm( - struct snd_soc_codec *codec) -{ - return snd_soc_component_get_dapm(&codec->component); -} - -/** - * snd_soc_dapm_init_bias_level() - Initialize CODEC DAPM bias level - * @codec: The CODEC for which to initialize the DAPM bias level - * @level: The DAPM level to initialize to - * - * Initializes the CODEC DAPM bias level. See snd_soc_dapm_init_bias_level(). - */ -static inline void snd_soc_codec_init_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - snd_soc_dapm_init_bias_level(snd_soc_codec_get_dapm(codec), level); -} - /** * snd_soc_component_init_bias_level() - Initialize COMPONENT DAPM bias level * @component: The COMPONENT for which to initialize the DAPM bias level @@ -1373,18 +1225,6 @@ snd_soc_component_init_bias_level(struct snd_soc_component *component, snd_soc_component_get_dapm(component), level); } -/** - * snd_soc_dapm_get_bias_level() - Get current CODEC DAPM bias level - * @codec: The CODEC for which to get the DAPM bias level - * - * Returns: The current DAPM bias level of the CODEC. - */ -static inline enum snd_soc_bias_level snd_soc_codec_get_bias_level( - struct snd_soc_codec *codec) -{ - return snd_soc_dapm_get_bias_level(snd_soc_codec_get_dapm(codec)); -} - /** * snd_soc_component_get_bias_level() - Get current COMPONENT DAPM bias level * @component: The COMPONENT for which to get the DAPM bias level @@ -1398,21 +1238,6 @@ snd_soc_component_get_bias_level(struct snd_soc_component *component) snd_soc_component_get_dapm(component)); } -/** - * snd_soc_codec_force_bias_level() - Set the CODEC DAPM bias level - * @codec: The CODEC for which to set the level - * @level: The level to set to - * - * Forces the CODEC bias level to a specific state. See - * snd_soc_dapm_force_bias_level(). - */ -static inline int snd_soc_codec_force_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - return snd_soc_dapm_force_bias_level(snd_soc_codec_get_dapm(codec), - level); -} - /** * snd_soc_component_force_bias_level() - Set the COMPONENT DAPM bias level * @component: The COMPONENT for which to set the level @@ -1430,19 +1255,6 @@ snd_soc_component_force_bias_level(struct snd_soc_component *component, level); } -/** - * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol - * @kcontrol: The kcontrol - * - * This function must only be used on DAPM contexts that are known to be part of - * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined. - */ -static inline struct snd_soc_codec *snd_soc_dapm_kcontrol_codec( - struct snd_kcontrol *kcontrol) -{ - return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); -} - /** * snd_soc_dapm_kcontrol_component() - Returns the component associated to a kcontrol * @kcontrol: The kcontrol @@ -1456,22 +1268,6 @@ static inline struct snd_soc_component *snd_soc_dapm_kcontrol_component( return snd_soc_dapm_to_component(snd_soc_dapm_kcontrol_dapm(kcontrol)); } -/* codec IO */ -unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); -int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val); - -/** - * snd_soc_cache_sync() - Sync the register cache with the hardware - * @codec: CODEC to sync - * - * Note: This function will call regcache_sync() - */ -static inline int snd_soc_cache_sync(struct snd_soc_codec *codec) -{ - return regcache_sync(codec->component.regmap); -} - /** * snd_soc_component_cache_sync() - Sync the register cache with the hardware * @component: COMPONENT to sync @@ -1514,37 +1310,6 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component, struct regmap *regmap); void snd_soc_component_exit_regmap(struct snd_soc_component *component); -/** - * snd_soc_codec_init_regmap() - Initialize regmap instance for the CODEC - * @codec: The CODEC for which to initialize the regmap instance - * @regmap: The regmap instance that should be used by the CODEC - * - * This function allows deferred assignment of the regmap instance that is - * associated with the CODEC. Only use this if the regmap instance is not yet - * ready when the CODEC is registered. The function must also be called before - * the first IO attempt of the CODEC. - */ -static inline void snd_soc_codec_init_regmap(struct snd_soc_codec *codec, - struct regmap *regmap) -{ - snd_soc_component_init_regmap(&codec->component, regmap); -} - -/** - * snd_soc_codec_exit_regmap() - De-initialize regmap instance for the CODEC - * @codec: The CODEC for which to de-initialize the regmap instance - * - * Calls regmap_exit() on the regmap instance associated to the CODEC and - * removes the regmap instance from the CODEC. - * - * This function should only be used if snd_soc_codec_init_regmap() was used to - * initialize the regmap instance. - */ -static inline void snd_soc_codec_exit_regmap(struct snd_soc_codec *codec) -{ - snd_soc_component_exit_regmap(&codec->component); -} - #endif /* device driver data */ @@ -1571,17 +1336,6 @@ static inline void *snd_soc_component_get_drvdata(struct snd_soc_component *c) return dev_get_drvdata(c->dev); } -static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec, - void *data) -{ - snd_soc_component_set_drvdata(&codec->component, data); -} - -static inline void *snd_soc_codec_get_drvdata(struct snd_soc_codec *codec) -{ - return snd_soc_component_get_drvdata(&codec->component); -} - static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) { INIT_LIST_HEAD(&card->widgets); @@ -1633,11 +1387,6 @@ static inline bool snd_soc_component_is_active( return component->active != 0; } -static inline bool snd_soc_codec_is_active(struct snd_soc_codec *codec) -{ - return snd_soc_component_is_active(&codec->component); -} - /** * snd_soc_kcontrol_component() - Returns the component that registered the * control @@ -1654,20 +1403,6 @@ static inline struct snd_soc_component *snd_soc_kcontrol_component( return snd_kcontrol_chip(kcontrol); } -/** - * snd_soc_kcontrol_codec() - Returns the CODEC that registered the control - * @kcontrol: The control for which to get the CODEC - * - * Note: This function will only work correctly if the control has been - * registered with snd_soc_add_codec_controls() or via table based setup of - * snd_soc_codec_driver. Otherwise the behavior is undefined. - */ -static inline struct snd_soc_codec *snd_soc_kcontrol_codec( - struct snd_kcontrol *kcontrol) -{ - return snd_soc_component_to_codec(snd_soc_kcontrol_component(kcontrol)); -} - int snd_soc_util_init(void); void snd_soc_util_exit(void); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3b78868969d5..a4cb141b8f38 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -56,7 +56,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); -static LIST_HEAD(codec_list); static LIST_HEAD(component_list); /* @@ -174,21 +173,6 @@ static void soc_cleanup_component_debugfs(struct snd_soc_component *component) debugfs_remove_recursive(component->debugfs_root); } -static int codec_list_show(struct seq_file *m, void *v) -{ - struct snd_soc_codec *codec; - - mutex_lock(&client_mutex); - - list_for_each_entry(codec, &codec_list, list) - seq_printf(m, "%s\n", codec->component.name); - - mutex_unlock(&client_mutex); - - return 0; -} -DEFINE_SHOW_ATTRIBUTE(codec_list); - static int dai_list_show(struct seq_file *m, void *v) { struct snd_soc_component *component; @@ -241,10 +225,6 @@ static void snd_soc_debugfs_init(void) return; } - if (!debugfs_create_file("codecs", 0444, snd_soc_debugfs_root, NULL, - &codec_list_fops)) - pr_warn("ASoC: Failed to create CODEC list debugfs file\n"); - if (!debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL, &dai_list_fops)) pr_warn("ASoC: Failed to create DAI list debugfs file\n"); @@ -536,8 +516,8 @@ int snd_soc_suspend(struct device *dev) } case SND_SOC_BIAS_OFF: - if (component->suspend) - component->suspend(component); + if (component->driver->suspend) + component->driver->suspend(component); component->suspended = 1; if (component->regmap) regcache_mark_dirty(component->regmap); @@ -608,8 +588,8 @@ static void soc_resume_deferred(struct work_struct *work) list_for_each_entry(component, &card->component_dev_list, card_list) { if (component->suspended) { - if (component->resume) - component->resume(component); + if (component->driver->resume) + component->driver->resume(component); component->suspended = 0; } } @@ -892,7 +872,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* Single codec links expect codec and codec_dai in runtime data */ rtd->codec_dai = codec_dais[0]; - rtd->codec = rtd->codec_dai->codec; /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; @@ -931,8 +910,8 @@ static void soc_remove_component(struct snd_soc_component *component) list_del(&component->card_list); - if (component->remove) - component->remove(component); + if (component->driver->remove) + component->driver->remove(component); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); @@ -1271,8 +1250,8 @@ static int soc_probe_component(struct snd_soc_card *card, } } - if (component->probe) { - ret = component->probe(component); + if (component->driver->probe) { + ret = component->driver->probe(component); if (ret < 0) { dev_err(component->dev, "ASoC: failed to probe component %d\n", ret); @@ -1663,8 +1642,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ /* the component which has non_legacy_dai_naming is Codec */ - if (cpu_dai->codec || - cpu_dai->component->driver->non_legacy_dai_naming) { + if (cpu_dai->component->driver->non_legacy_dai_naming) { unsigned int inv_dai_fmt; inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; @@ -2256,25 +2234,6 @@ int snd_soc_add_component_controls(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_add_component_controls); -/** - * snd_soc_add_codec_controls - add an array of controls to a codec. - * Convenience function to add a list of controls. Many codecs were - * duplicating this code. - * - * @codec: codec to add controls to - * @controls: array of controls to add - * @num_controls: number of elements in the array - * - * Return 0 for success, else error. - */ -int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, unsigned int num_controls) -{ - return snd_soc_add_component_controls(&codec->component, controls, - num_controls); -} -EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); - /** * snd_soc_add_card_controls - add an array of controls to a SoC card. * Convenience function to add a list of controls. @@ -2335,27 +2294,6 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); -/** - * snd_soc_codec_set_sysclk - configure CODEC system or master clock. - * @codec: CODEC - * @clk_id: DAI specific clock ID - * @source: Source for the clock - * @freq: new clock frequency in Hz - * @dir: new clock direction - input/output. - * - * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. - */ -int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - int source, unsigned int freq, int dir) -{ - if (codec->driver->set_sysclk) - return codec->driver->set_sysclk(codec, clk_id, source, - freq, dir); - else - return -ENOTSUPP; -} -EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); - /** * snd_soc_component_set_sysclk - configure COMPONENT system or master clock. * @component: COMPONENT @@ -2369,11 +2307,6 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir) { - /* will be removed */ - if (component->set_sysclk) - return component->set_sysclk(component, clk_id, source, - freq, dir); - if (component->driver->set_sysclk) return component->driver->set_sysclk(component, clk_id, source, freq, dir); @@ -2424,27 +2357,6 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, } EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); -/* - * snd_soc_codec_set_pll - configure codec PLL. - * @codec: CODEC - * @pll_id: DAI specific PLL ID - * @source: DAI specific source for the PLL - * @freq_in: PLL input clock frequency in Hz - * @freq_out: requested PLL output clock frequency in Hz - * - * Configures and enables PLL to generate output clock based on input clock. - */ -int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, - unsigned int freq_in, unsigned int freq_out) -{ - if (codec->driver->set_pll) - return codec->driver->set_pll(codec, pll_id, source, - freq_in, freq_out); - else - return -EINVAL; -} -EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); - /* * snd_soc_component_set_pll - configure component PLL. * @component: COMPONENT @@ -2459,11 +2371,6 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - /* will be removed */ - if (component->set_pll) - return component->set_pll(component, pll_id, source, - freq_in, freq_out); - if (component->driver->set_pll) return component->driver->set_pll(component, pll_id, source, freq_in, freq_out); @@ -2964,13 +2871,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->dev = dev; component->driver = driver; - component->probe = component->driver->probe; - component->remove = component->driver->remove; - component->suspend = component->driver->suspend; - component->resume = component->driver->resume; - component->set_sysclk = component->driver->set_sysclk; - component->set_pll = component->driver->set_pll; - component->set_jack = component->driver->set_jack; dapm = snd_soc_component_get_dapm(component); dapm->dev = dev; @@ -3041,7 +2941,7 @@ EXPORT_SYMBOL_GPL(snd_soc_component_exit_regmap); static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { - if (!component->write && !component->read) { + if (!component->driver->write && !component->driver->read) { if (!component->regmap) component->regmap = dev_get_regmap(component->dev, NULL); if (component->regmap) @@ -3123,9 +3023,6 @@ int snd_soc_add_component(struct device *dev, if (ret) goto err_free; - component->ignore_pmdown_time = true; - component->registered_as_component = true; - if (component_driver->endianness) { for (i = 0; i < num_dai; i++) { convert_endianness_formats(&dai_drv[i].playback); @@ -3180,8 +3077,7 @@ static int __snd_soc_unregister_component(struct device *dev) mutex_lock(&client_mutex); list_for_each_entry(component, &component_list, list) { - if (dev != component->dev || - !component->registered_as_component) + if (dev != component->dev) continue; snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL); @@ -3230,211 +3126,6 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_lookup_component); -static int snd_soc_codec_drv_probe(struct snd_soc_component *component) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return codec->driver->probe(codec); -} - -static void snd_soc_codec_drv_remove(struct snd_soc_component *component) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - codec->driver->remove(codec); -} - -static int snd_soc_codec_drv_suspend(struct snd_soc_component *component) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return codec->driver->suspend(codec); -} - -static int snd_soc_codec_drv_resume(struct snd_soc_component *component) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return codec->driver->resume(codec); -} - -static int snd_soc_codec_drv_write(struct snd_soc_component *component, - unsigned int reg, unsigned int val) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return codec->driver->write(codec, reg, val); -} - -static int snd_soc_codec_drv_read(struct snd_soc_component *component, - unsigned int reg, unsigned int *val) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - *val = codec->driver->read(codec, reg); - - return 0; -} - -static int snd_soc_codec_set_sysclk_(struct snd_soc_component *component, - int clk_id, int source, unsigned int freq, int dir) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return snd_soc_codec_set_sysclk(codec, clk_id, source, freq, dir); -} - -static int snd_soc_codec_set_pll_(struct snd_soc_component *component, - int pll_id, int source, unsigned int freq_in, - unsigned int freq_out) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return snd_soc_codec_set_pll(codec, pll_id, source, freq_in, freq_out); -} - -static int snd_soc_codec_set_jack_(struct snd_soc_component *component, - struct snd_soc_jack *jack, void *data) -{ - struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - - return snd_soc_codec_set_jack(codec, jack, data); -} - -static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); - - return codec->driver->set_bias_level(codec, level); -} - -/** - * snd_soc_register_codec - Register a codec with the ASoC core - * - * @dev: The parent device for this codec - * @codec_drv: Codec driver - * @dai_drv: The associated DAI driver - * @num_dai: Number of DAIs - */ -int snd_soc_register_codec(struct device *dev, - const struct snd_soc_codec_driver *codec_drv, - struct snd_soc_dai_driver *dai_drv, - int num_dai) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - int ret, i; - - dev_dbg(dev, "codec register %s\n", dev_name(dev)); - - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - codec->component.codec = codec; - - ret = snd_soc_component_initialize(&codec->component, - &codec_drv->component_driver, dev); - if (ret) - goto err_free; - - if (codec_drv->probe) - codec->component.probe = snd_soc_codec_drv_probe; - if (codec_drv->remove) - codec->component.remove = snd_soc_codec_drv_remove; - if (codec_drv->suspend) - codec->component.suspend = snd_soc_codec_drv_suspend; - if (codec_drv->resume) - codec->component.resume = snd_soc_codec_drv_resume; - if (codec_drv->write) - codec->component.write = snd_soc_codec_drv_write; - if (codec_drv->read) - codec->component.read = snd_soc_codec_drv_read; - if (codec_drv->set_sysclk) - codec->component.set_sysclk = snd_soc_codec_set_sysclk_; - if (codec_drv->set_pll) - codec->component.set_pll = snd_soc_codec_set_pll_; - if (codec_drv->set_jack) - codec->component.set_jack = snd_soc_codec_set_jack_; - codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - - dapm = snd_soc_codec_get_dapm(codec); - dapm->idle_bias_off = codec_drv->idle_bias_off; - dapm->suspend_bias_off = codec_drv->suspend_bias_off; - if (codec_drv->seq_notifier) - dapm->seq_notifier = codec_drv->seq_notifier; - if (codec_drv->set_bias_level) - dapm->set_bias_level = snd_soc_codec_set_bias_level; - codec->dev = dev; - codec->driver = codec_drv; - -#ifdef CONFIG_DEBUG_FS - codec->component.debugfs_prefix = "codec"; -#endif - - for (i = 0; i < num_dai; i++) { - convert_endianness_formats(&dai_drv[i].playback); - convert_endianness_formats(&dai_drv[i].capture); - } - - ret = snd_soc_register_dais(&codec->component, dai_drv, num_dai, false); - if (ret < 0) { - dev_err(dev, "ASoC: Failed to register DAIs: %d\n", ret); - goto err_cleanup; - } - - list_for_each_entry(dai, &codec->component.dai_list, list) - dai->codec = codec; - - mutex_lock(&client_mutex); - snd_soc_component_add_unlocked(&codec->component); - list_add(&codec->list, &codec_list); - mutex_unlock(&client_mutex); - - dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", - codec->component.name); - return 0; - -err_cleanup: - snd_soc_component_cleanup(&codec->component); -err_free: - kfree(codec); - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_register_codec); - -/** - * snd_soc_unregister_codec - Unregister a codec from the ASoC core - * - * @dev: codec to unregister - */ -void snd_soc_unregister_codec(struct device *dev) -{ - struct snd_soc_codec *codec; - - mutex_lock(&client_mutex); - list_for_each_entry(codec, &codec_list, list) { - if (dev == codec->dev) - goto found; - } - mutex_unlock(&client_mutex); - return; - -found: - list_del(&codec->list); - snd_soc_component_del_unlocked(&codec->component); - mutex_unlock(&client_mutex); - - dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", - codec->component.name); - - snd_soc_component_cleanup(&codec->component); - kfree(codec); -} -EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); - /* Retrieve a card's name from device tree */ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c92a04bac3c5..026cd5347e53 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -32,8 +32,6 @@ int snd_soc_component_read(struct snd_soc_component *component, if (component->regmap) ret = regmap_read(component->regmap, reg, val); - else if (component->read) - ret = component->read(component, reg, val); else if (component->driver->read) { *val = component->driver->read(component, reg); ret = 0; @@ -72,8 +70,6 @@ int snd_soc_component_write(struct snd_soc_component *component, { if (component->regmap) return regmap_write(component->regmap, reg, val); - else if (component->write) - return component->write(component, reg, val); else if (component->driver->write) return component->driver->write(component, reg, val); else @@ -209,61 +205,3 @@ int snd_soc_component_test_bits(struct snd_soc_component *component, return old != new; } EXPORT_SYMBOL_GPL(snd_soc_component_test_bits); - -unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg) -{ - unsigned int val; - int ret; - - ret = snd_soc_component_read(&codec->component, reg, &val); - if (ret < 0) - return -1; - - return val; -} -EXPORT_SYMBOL_GPL(snd_soc_read); - -int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - return snd_soc_component_write(&codec->component, reg, val); -} -EXPORT_SYMBOL_GPL(snd_soc_write); - -/** - * snd_soc_update_bits - update codec register bits - * @codec: audio codec - * @reg: codec register - * @mask: register mask - * @value: new value - * - * Writes new register value. - * - * Returns 1 for change, 0 for no change, or negative error code. - */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, - unsigned int mask, unsigned int value) -{ - return snd_soc_component_update_bits(&codec->component, reg, mask, - value); -} -EXPORT_SYMBOL_GPL(snd_soc_update_bits); - -/** - * snd_soc_test_bits - test register for change - * @codec: audio codec - * @reg: codec register - * @mask: register mask - * @value: new value - * - * Tests a register with a new value and checks if the new value is - * different from the old value. - * - * Returns 1 for change else 0. - */ -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, - unsigned int mask, unsigned int value) -{ - return snd_soc_component_test_bits(&codec->component, reg, mask, value); -} -EXPORT_SYMBOL_GPL(snd_soc_test_bits); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 99902ae1a2d9..b2b16044ae80 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -28,24 +28,6 @@ struct jack_gpio_tbl { struct snd_soc_jack_gpio *gpios; }; -/** - * snd_soc_codec_set_jack - configure codec jack. - * @codec: CODEC - * @jack: structure to use for the jack - * @data: can be used if codec driver need extra data for configuring jack - * - * Configures and enables jack detection function. - */ -int snd_soc_codec_set_jack(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, void *data) -{ - if (codec->driver->set_jack) - return codec->driver->set_jack(codec, jack, data); - else - return -ENOTSUPP; -} -EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); - /** * snd_soc_component_set_jack - configure component jack. * @component: COMPONENTs @@ -57,10 +39,6 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); int snd_soc_component_set_jack(struct snd_soc_component *component, struct snd_soc_jack *jack, void *data) { - /* will be removed */ - if (component->set_jack) - return component->set_jack(component, jack, data); - if (component->driver->set_jack) return component->driver->set_jack(component, jack, data); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3f6375499102..87c9af2158d0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -135,7 +135,6 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; - int i; bool ignore = true; if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) @@ -147,10 +146,6 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) ignore &= !component->driver->use_pmdown_time; } - /* this will be removed */ - for (i = 0; i < rtd->num_codecs; i++) - ignore &= rtd->codec_dais[i]->component->ignore_pmdown_time; - return ignore; } -- 2.17.0

1 0

[alsa-devel] Applied "ASoC: convert platform explanation to component" to the asoc tree
by Mark Brown 09 May '18

09 May '18

The patch ASoC: convert platform explanation to component has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d9f29e9ebe6ec7193c160ac3d38d5d1d6b55adda Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 8 May 2018 03:21:24 +0000 Subject: [PATCH] ASoC: convert platform explanation to component commit ef050bece1b55 ("ASoC: Remove platform code now everything is componentised") removed platform code, but it didn't care about platform documentation. This patch convert platform explanation to component Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/sound/soc/platform.rst | 30 ++++++++++++---------------- 1 file changed, 13 insertions(+), 17 deletions(-) diff --git a/Documentation/sound/soc/platform.rst b/Documentation/sound/soc/platform.rst index d5574904d981..cc817078daa6 100644 --- a/Documentation/sound/soc/platform.rst +++ b/Documentation/sound/soc/platform.rst @@ -23,30 +23,26 @@ The platform DMA driver optionally supports the following ALSA operations:- }; The platform driver exports its DMA functionality via struct -snd_soc_platform_driver:- +snd_soc_component_driver:- :: - struct snd_soc_platform_driver { - char *name; + struct snd_soc_component_driver { + const char *name; - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); - int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + ... + int (*probe)(struct snd_soc_component *); + void (*remove)(struct snd_soc_component *); + int (*suspend)(struct snd_soc_component *); + int (*resume)(struct snd_soc_component *); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); + int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); - /* - * For platform caused delay reporting. - * Optional. - */ - snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, - struct snd_soc_dai *); - - /* platform stream ops */ - struct snd_pcm_ops *pcm_ops; + ... + const struct snd_pcm_ops *ops; + const struct snd_compr_ops *compr_ops; + ... }; Please refer to the ALSA driver documentation for details of audio DMA. -- 2.17.0

1 0

[alsa-devel] [PATCH 00/10] ASoC: remove old platform/codec code
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark We are now using component base ASoC. Thus, old platform/codec code are no longer needed. These removes it. Then, I very surprised that platform code was already removed :) But, it seems this patch didn't care about documentation and platform specific callback (= .pcm_new/free). 4), 7) handle it. One concern is that it completely removed "platform_list_show" for debugfs. But, I think we want to keep it as "component_list_show" (My original patch did it). 6) reborned it. 8) - 10) are cleanup which can be done *after* removing platform/codec. Kuninori Morimoto (10): 1) ASoC: remove codec hw_write/control_data 2) ASoC: remove codec reg_cache 3) ASoC: remove .get_regmap 4) ASoC: remove unneeded .pcm_new/free 5) ASoC: remove Codec related code 6) ASoC: add component_list_show() 7) ASoC: convert platform explanation to component 8) ASoC: soc.h: merge CONFIG_DEBUG_FS 9) ASoC: soc-core: remove snd_soc_component_add_unlocked() 10) ASoC: soc-core: remove legacy_dai_naming from snd_soc_register_dais() Documentation/sound/soc/platform.rst | 30 +- include/sound/soc-dai.h | 1 - include/sound/soc.h | 289 +---------------- sound/soc/Makefile | 2 +- sound/soc/soc-cache.c | 53 ---- sound/soc/soc-core.c | 585 ++--------------------------------- sound/soc/soc-io.c | 62 ---- sound/soc/soc-jack.c | 22 -- sound/soc/soc-pcm.c | 13 +- 9 files changed, 50 insertions(+), 1007 deletions(-) delete mode 100644 sound/soc/soc-cache.c Best regards --- Kuninori Morimoto

3 25

[alsa-devel] [PATCH] ASoC: soc-core: remove legacy_dai_naming from snd_soc_register_dais()
by Kuninori Morimoto 09 May '18

09 May '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We can get legacy dai name flag from component driver. Thus, there is no need to have its parameter on snd_soc_register_dais(). Let's remove unneeded parameter Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1faebc04..9c65477 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3146,8 +3146,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count, - bool legacy_dai_naming) + struct snd_soc_dai_driver *dai_drv, size_t count) { struct device *dev = component->dev; struct snd_soc_dai *dai; @@ -3159,7 +3158,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, for (i = 0; i < count; i++) { dai = soc_add_dai(component, dai_drv + i, - count == 1 && legacy_dai_naming); + count == 1 && !component->driver->non_legacy_dai_naming); if (dai == NULL) { ret = -ENOMEM; goto err; @@ -3440,8 +3439,7 @@ int snd_soc_add_component(struct device *dev, } } - ret = snd_soc_register_dais(component, dai_drv, num_dai, - !component_driver->non_legacy_dai_naming); + ret = snd_soc_register_dais(component, dai_drv, num_dai); if (ret < 0) { dev_err(dev, "ASoC: Failed to register DAIs: %d\n", ret); goto err_cleanup; -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: uniphier: add digital output volume for UniPhier sound system
by Katsuhiro Suzuki 09 May '18

09 May '18

This patch adds controllers for digital volume of PCM output. Volume effects simply linear, not dB scale as follows: Gained PCM = Original * 0x4000 / Volume The value range of volume is from 0x0001 to 0xffff. 0x0000 works as mute. Initial value is 0x4000 (+0dB). Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 58 ++++++++++++++ sound/soc/uniphier/aio-cpu.c | 140 ++++++++++++++++++++++++++++++++++ sound/soc/uniphier/aio-reg.h | 33 ++++++-- sound/soc/uniphier/aio.h | 7 ++ 4 files changed, 233 insertions(+), 5 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index e37b80921abb..638cb3fc5f7b 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -659,6 +659,64 @@ void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable) } } +/** + * aio_port_get_volume - get volume of AIO port block + * @sub: the AIO substream pointer + * + * Return: current volume, range is 0x0000 - 0xffff + */ +int aio_port_get_volume(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 v; + + regmap_read(r, OPORTMXTYVOLGAINSTATUS(sub->swm->oport.map, 0), &v); + + return FIELD_GET(OPORTMXTYVOLGAINSTATUS_CUR_MASK, v); +} + +/** + * aio_port_set_volume - set volume of AIO port block + * @sub: the AIO substream pointer + * @vol: target volume, range is 0x0000 - 0xffff. + * + * Change digital volume and perfome fade-out/fade-in effect for specified + * output slot of port. Gained PCM value can calculate as the following: + * Gained = Original * vol / 0x4000 + */ +void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) +{ + struct regmap *r = sub->aio->chip->regmap; + int oport_map = sub->swm->oport.map; + int cur, diff, slope = 0, fs; + + if (sub->swm->dir == PORT_DIR_INPUT) + return; + + cur = aio_port_get_volume(sub); + diff = abs(vol - cur); + fs = params_rate(&sub->params); + if (fs) + slope = diff / AUD_VOL_FADE_TIME * 1000 / fs; + slope = max(1, slope); + + regmap_update_bits(r, OPORTMXTYVOLPARA1(oport_map, 0), + OPORTMXTYVOLPARA1_SLOPEU_MASK, slope << 16); + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_TARGET_MASK, vol); + + if (cur < vol) + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_FADE_MASK, + OPORTMXTYVOLPARA2_FADE_FADEIN); + else + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_FADE_MASK, + OPORTMXTYVOLPARA2_FADE_FADEOUT); + + regmap_write(r, AOUTFADECTR0, BIT(oport_map)); +} + /** * aio_if_set_param - set parameters of AIO DMA I/F block * @sub: the AIO substream pointer diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 00b6441bf195..80daec17be25 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -32,6 +32,35 @@ static bool is_valid_pll(struct uniphier_aio_chip *chip, int pll_id) return chip->plls[pll_id].enable; } +/** + * find_volume - find volume supported HW port by HW port number + * @chip: the AIO chip pointer + * @oport_hw: HW port number, one of AUD_HW_XXXX + * + * Find AIO device from device list by HW port number. Volume feature is + * available only in Output and PCM ports, this limitation comes from HW + * specifications. + * + * Return: The pointer of AIO substream if successful, otherwise NULL on error. + */ +static struct uniphier_aio_sub *find_volume(struct uniphier_aio_chip *chip, + int oport_hw) +{ + int i; + + for (i = 0; i < chip->num_aios; i++) { + struct uniphier_aio_sub *sub = &chip->aios[i].sub[0]; + + if (!sub->swm) + continue; + + if (sub->swm->oport.hw == oport_hw) + return sub; + } + + return NULL; +} + static bool match_spec(const struct uniphier_aio_spec *spec, const char *name, int dir) { @@ -287,6 +316,7 @@ static int uniphier_aio_hw_params(struct snd_pcm_substream *substream, sub->setting = 1; aio_port_reset(sub); + aio_port_set_volume(sub, sub->vol); aio_src_reset(sub); return 0; @@ -373,6 +403,8 @@ int uniphier_aio_dai_probe(struct snd_soc_dai *dai) sub->swm = &spec->swm; sub->spec = spec; + + sub->vol = AUD_VOL_INIT; } aio_iecout_set_enable(aio->chip, true); @@ -449,8 +481,116 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) } EXPORT_SYMBOL_GPL(uniphier_aio_dai_resume); +static int uniphier_aio_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = AUD_VOL_MAX; + + return 0; +} + +static int uniphier_aio_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct uniphier_aio_chip *chip = snd_soc_component_get_drvdata(comp); + struct uniphier_aio_sub *sub; + int oport_hw = kcontrol->private_value; + + sub = find_volume(chip, oport_hw); + if (!sub) + return 0; + + ucontrol->value.integer.value[0] = sub->vol; + + return 0; +} + +static int uniphier_aio_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct uniphier_aio_chip *chip = snd_soc_component_get_drvdata(comp); + struct uniphier_aio_sub *sub; + int oport_hw = kcontrol->private_value; + + sub = find_volume(chip, oport_hw); + if (!sub) + return 0; + + if (sub->vol == ucontrol->value.integer.value[0]) + return 0; + sub->vol = ucontrol->value.integer.value[0]; + + aio_port_set_volume(sub, sub->vol); + + return 0; +} + +static const struct snd_kcontrol_new uniphier_aio_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "HPCMOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_HPCMOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT2 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT2, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT3 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT3, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "HIECOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_HIECOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "IECOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_IECOUT1, + }, +}; + static const struct snd_soc_component_driver uniphier_aio_component = { .name = "uniphier-aio", + .controls = uniphier_aio_controls, + .num_controls = ARRAY_SIZE(uniphier_aio_controls), }; int uniphier_aio_probe(struct platform_device *pdev) diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 511ea3c01847..45fdc6ae358a 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -169,6 +169,7 @@ #define PBINMXPAUSECTR1(n) (0x20208 + 0x40 * (n)) /* AOUT */ +#define AOUTFADECTR0 0x40020 #define AOUTENCTR0 0x40040 #define AOUTENCTR1 0x40044 #define AOUTENCTR2 0x40048 @@ -179,6 +180,9 @@ #define AOUTSRCRSTCTR1 0x400c4 #define AOUTSRCRSTCTR2 0x400c8 +/* AOUT PCMOUT has 5 slots, slot0-3: D0-3, slot4: DMIX */ +#define OPORT_SLOT_MAX 5 + /* AOUT(PCMOUTN) */ #define OPORTMXCTR1(n) (0x42000 + 0x400 * (n)) #define OPORTMXCTR1_I2SLRSEL_MASK (0x11 << 10) @@ -359,11 +363,30 @@ #define OPORTMXMASK_XCKMSK_ON (0x0 << 0) #define OPORTMXMASK_XCKMSK_OFF (0x7 << 0) #define OPORTMXDEBUG(n) (0x420fc + 0x400 * (n)) -#define OPORTMXT0RSTCTR(n) (0x4211c + 0x400 * (n)) -#define OPORTMXT1RSTCTR(n) (0x4213c + 0x400 * (n)) -#define OPORTMXT2RSTCTR(n) (0x4215c + 0x400 * (n)) -#define OPORTMXT3RSTCTR(n) (0x4217c + 0x400 * (n)) -#define OPORTMXT4RSTCTR(n) (0x4219c + 0x400 * (n)) +#define OPORTMXTYVOLPARA1(n, m) (0x42100 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLPARA1_SLOPEU_MASK GENMASK(31, 16) +#define OPORTMXTYVOLPARA2(n, m) (0x42104 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLPARA2_FADE_MASK GENMASK(17, 16) +#define OPORTMXTYVOLPARA2_FADE_NOOP (0x0 << 16) +#define OPORTMXTYVOLPARA2_FADE_FADEOUT (0x1 << 16) +#define OPORTMXTYVOLPARA2_FADE_FADEIN (0x2 << 16) +#define OPORTMXTYVOLPARA2_TARGET_MASK GENMASK(15, 0) +#define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) +#define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT2 (0xa << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT3 (0xb << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT4 (0xc << 8) +#define OPORTMXT0SLOTCTR_MUTEOFF_MASK BIT(1) +#define OPORTMXT0SLOTCTR_MUTEOFF_MUTE (0x0 << 1) +#define OPORTMXT0SLOTCTR_MUTEOFF_UNMUTE (0x1 << 1) +#define OPORTMXTYRSTCTR(n, m) (0x4211c + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXT0RSTCTR_RST_MASK BIT(1) +#define OPORTMXT0RSTCTR_RST_OFF (0x0 << 1) +#define OPORTMXT0RSTCTR_RST_ON (0x1 << 1) #define SBF_(frame, shift) (((frame) * 2 - 1) << shift) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index 52670126084f..aa89c2f6fa24 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -130,6 +130,10 @@ enum IEC61937_PC { #define AUD_PLLDIV_1_1 2 #define AUD_PLLDIV_2_3 3 +#define AUD_VOL_INIT 0x4000 /* +0dB */ +#define AUD_VOL_MAX 0xffff /* +6dB */ +#define AUD_VOL_FADE_TIME 20 /* 20ms */ + #define AUD_RING_SIZE (128 * 1024) #define AUD_MIN_FRAGMENT 4 @@ -231,6 +235,7 @@ struct uniphier_aio_sub { /* For PCM audio */ struct snd_pcm_substream *substream; struct snd_pcm_hw_params params; + int vol; /* For compress audio */ struct snd_compr_stream *cstream; @@ -323,6 +328,8 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); +int aio_port_get_volume(struct uniphier_aio_sub *sub); +void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol); int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through); int aio_oport_set_stream_type(struct uniphier_aio_sub *sub, enum IEC61937_PC pc); -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: nau8824: fix spelling mistake: "semaphone" -> "semaphore"
by Colin King 09 May '18

09 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_warn messages Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/nau8824.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 76502c090654..6bd14453f06e 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -205,11 +205,11 @@ static int nau8824_sema_acquire(struct nau8824 *nau8824, long timeout) if (timeout) { ret = down_timeout(&nau8824->jd_sem, timeout); if (ret < 0) - dev_warn(nau8824->dev, "Acquire semaphone timeout\n"); + dev_warn(nau8824->dev, "Acquire semaphore timeout\n"); } else { ret = down_interruptible(&nau8824->jd_sem); if (ret < 0) - dev_warn(nau8824->dev, "Acquire semaphone fail\n"); + dev_warn(nau8824->dev, "Acquire semaphore fail\n"); } return ret; -- 2.17.0

2 2

[alsa-devel] [PATCH] ASoC: uniphier: evea: use DAPM to change source of line-in
by Katsuhiro Suzuki 09 May '18

09 May '18

This patch replaces mixer switch to DAPM one for changing audio source of line-in. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/evea.c | 55 +++++++++++++++++++++++---------------- 1 file changed, 33 insertions(+), 22 deletions(-) diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index 73fd6730095c..f9c10165fbc1 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -54,8 +54,21 @@ struct evea_priv { int switch_hp; }; +static const char * const linsw1_sel1_text[] = { + "LIN1", "LIN2", "LIN3" +}; + +static SOC_ENUM_SINGLE_DECL(linsw1_sel1_enum, + ALINSW1, ALINSW1_SEL1_SHIFT, + linsw1_sel1_text); + +static const struct snd_kcontrol_new linesw1_mux[] = { + SOC_DAPM_ENUM("Line In 1 Source", linsw1_sel1_enum), +}; + static const struct snd_soc_dapm_widget evea_widgets[] = { - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("Line In 1 Mux", SND_SOC_NOPM, 0, 0, linesw1_mux), SND_SOC_DAPM_INPUT("LIN1_LP"), SND_SOC_DAPM_INPUT("LIN1_RP"), SND_SOC_DAPM_INPUT("LIN2_LP"), @@ -63,7 +76,9 @@ static const struct snd_soc_dapm_widget evea_widgets[] = { SND_SOC_DAPM_INPUT("LIN3_LP"), SND_SOC_DAPM_INPUT("LIN3_RP"), - SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC HP", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC LO1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC LO2", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_OUTPUT("HP1_L"), SND_SOC_DAPM_OUTPUT("HP1_R"), SND_SOC_DAPM_OUTPUT("LO2_L"), @@ -71,17 +86,22 @@ static const struct snd_soc_dapm_widget evea_widgets[] = { }; static const struct snd_soc_dapm_route evea_routes[] = { - { "ADC", NULL, "LIN1_LP" }, - { "ADC", NULL, "LIN1_RP" }, - { "ADC", NULL, "LIN2_LP" }, - { "ADC", NULL, "LIN2_RP" }, - { "ADC", NULL, "LIN3_LP" }, - { "ADC", NULL, "LIN3_RP" }, - - { "HP1_L", NULL, "DAC" }, - { "HP1_R", NULL, "DAC" }, - { "LO2_L", NULL, "DAC" }, - { "LO2_R", NULL, "DAC" }, + { "Line In 1", NULL, "ADC" }, + { "ADC", NULL, "Line In 1 Mux" }, + { "Line In 1 Mux", "LIN1", "LIN1_LP" }, + { "Line In 1 Mux", "LIN1", "LIN1_RP" }, + { "Line In 1 Mux", "LIN2", "LIN2_LP" }, + { "Line In 1 Mux", "LIN2", "LIN2_RP" }, + { "Line In 1 Mux", "LIN3", "LIN3_LP" }, + { "Line In 1 Mux", "LIN3", "LIN3_RP" }, + + { "DAC HP", NULL, "Headphone 1" }, + { "DAC LO1", NULL, "Line Out 1" }, + { "DAC LO2", NULL, "Line Out 2" }, + { "HP1_L", NULL, "DAC HP" }, + { "HP1_R", NULL, "DAC HP" }, + { "LO2_L", NULL, "DAC LO2" }, + { "LO2_R", NULL, "DAC LO2" }, }; static void evea_set_power_state_on(struct evea_priv *evea) @@ -280,16 +300,7 @@ static int evea_set_switch_hp(struct snd_kcontrol *kcontrol, return evea_update_switch_hp(evea); } -static const char * const linsw1_sel1_text[] = { - "LIN1", "LIN2", "LIN3" -}; - -static SOC_ENUM_SINGLE_DECL(linsw1_sel1_enum, - ALINSW1, ALINSW1_SEL1_SHIFT, - linsw1_sel1_text); - static const struct snd_kcontrol_new evea_controls[] = { - SOC_ENUM("Line Capture Source", linsw1_sel1_enum), SOC_SINGLE_BOOL_EXT("Line Capture Switch", 0, evea_get_switch_lin, evea_set_switch_lin), SOC_SINGLE_BOOL_EXT("Line Playback Switch", 0, -- 2.17.0

2 2

[alsa-devel] [PATCH -next] ASoC: fix return value check in mt6351_codec_driver_probe()
by Wei Yongjun 09 May '18

09 May '18

In case of error, the function dev_get_regmap() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/mt6351.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mt6351.c b/sound/soc/codecs/mt6351.c index e739f07..f73dcd7 100644 --- a/sound/soc/codecs/mt6351.c +++ b/sound/soc/codecs/mt6351.c @@ -1472,8 +1472,8 @@ static int mt6351_codec_driver_probe(struct platform_device *pdev) priv->dev = &pdev->dev; priv->regmap = dev_get_regmap(pdev->dev.parent, NULL); - if (IS_ERR(priv->regmap)) - return PTR_ERR(priv->regmap); + if (!priv->regmap) + return -ENODEV; dev_dbg(priv->dev, "%s(), dev name %s\n", __func__, dev_name(&pdev->dev));

2 4

[alsa-devel] [PATCH v4 00/13] ALSA: hda/ca0132: Patch Series for Recon3Di and Sound Blaster Z Support
by Connor McAdams 08 May '18

08 May '18

This patchset adds support for the Sound Blaster Z and the Recon3Di. In order to figure out how to get these cards to work, I made a program called QemuHDADump[1], which uses the trace function of qemu to see interactions with the memory mapped pci BAR space of the card being used in the virtual machine. With this, I obtain the CORB buffer location to get the command verbs, and then dump them each time the buffer rolls over. This program may be useful for fixing other HDA related driver issues where there is no documentation for the device. So far, I have been able to get all features supported on the Sound Blaster Z and the Recon3Di. All output and input effects work, all inputs and outputs work, and just about anything else I can think of. I have also added new controls in order to select the new inputs and outputs, as well as controls to change the effect levels and presets. I have also added the ability to use firmware taken from the Windows drivers of both the Sound Blaster Z and Recon3Di. I am trying to get into contact with Creative to get permission to redistribute these along with the current file included with the Chromebook, but they have not been very responsive. Luckily, the cards work with the Chromebook firmware just fine, although I believe there has to be a reason they have different firmware in Windows. I will not link to the firmwares here, but if you look up my thread on Creative Labs forums, you will find the link to download the firmwares there. I am willing to help get the other non-working cards such as the ZxR and the newer AE-5 working too, but I will need someone willing to run QemuHDADump in a virtual machine in order to get the commands. So, in summary: -This patchset makes the cards work better than they did before (they really didn't work before) -This patchset leaves the original chromebook related stuff alone. Thanks. [1] https://github.com/Conmanx360/QemuHDADump Bugs: ------------------------------------------------------------------------------- Recon3Di: (Reported by Mariusz Ceier) ******************* -Occasionally switching between rear and front mic breaks the input until computer is shutdown or put to sleep. -Surround Sound works, but is inconsistent. Sometimes, just updating the volume fixes it, and sometimes, it requires a restart. Sound Blaster Z: ******************* -none that I'm aware of. Version changes: ------------------------------------------------------------------------------- v1: ******************* -Massive patch formatting failure, please ignore v1. v2: ******************* -Fixed patch formatting failure. v3: ******************* -Fixed mem_base unmap, instead of checking for QUIRK_SBZ on exit, have it check if the area is mapped, and if it is, unmap it. Also make it unmap after all other commands are finished. -Change notification of failure to map mem_base from codec_dbg to codec_warn, and use codec_info to tell the user that their card might have been incorrectly identified as a Sound Blaster Z. -Remove commented out commands in sbz_exit_chip function, only reintroduce them when their functions are defined. v4: ******************* -Split patch into smaller pieces. -Added const to alt_out_presets array. -Fixed command that was commented out and only put it in when it was actually used. Connor McAdams (13): ALSA: hda/ca0132: R3Di and SBZ quirk entires + alt firmware loading ALSA: hda/ca0132: Add pincfg for SBZ + R3Di, add fp hp auto-detect ALSA: hda/ca0132: Add PCI region2 iomap for SBZ ALSA: hda/ca0132: Add extra exit functions for R3Di and SBZ ALSA: hda/ca0132: add extra init functions for r3di + sbz ALSA: hda/ca0132: update core functions for sbz + r3di ALSA: hda/ca0132: add dsp setup related commands for the sbz ALSA: hda/ca0132: Add dsp setup + gpio functions for r3di ALSA: hda/ca0132: add the ability to set src_id on scp commands ALSA: hda/ca0132: add alt_select_in/out for R3Di + SBZ ALSA: hda/ca0132: Add DSP Volume set and New mixers for SBZ + R3Di ALSA: hda/ca0132: add ca0132_alt_set_vipsource ALSA: hda/ca0132: Add new control changes for SBZ + R3Di sound/pci/hda/patch_ca0132.c | 3057 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 2941 insertions(+), 116 deletions(-) -- 2.7.4

2 14

[alsa-devel] Help with pcm3168a device tree, and simple-audio-card/simple-scu-audio-card?
by sdaau 08 May '18

08 May '18

Hi all, I am experimenting with PCM3168A audio codec, connected to a Raspberry Pi 3 B running Raspbian 9. I have set up GPIO4 (BCM4) of the Broadcom BCM2837 SoC of the RPi3B as GPCLK0, and use that as SCKI (System clock input) of the PCM3168A; other than that, I have set up the I2S pins of the BCM2837 in I2S/PCM mode, and use the BCLK/LRCLK clock lines to drive both AD and DA (respective) clock lines; and also the PCM DIN and DOUT lines are connected: the PCM3168A codec has 8 (mono, differential) DACs and 6 (mono, differential) ADCs - but I can only test with 2 DACs for stereo out, and 2 ADCs for stereo in, since there is only one line for DIN and DOUT each on the RPi3B/BCM2837 I2S interface. The PCM3168A is controlled by RPi3B's SPI lines, chip select 1. I have had limited success in running this setup with a device tree overlay based on: * simple-audio-card driver: https://github.com/raspberrypi/linux/blob/rpi-4.14.y/sound/soc/generic/simp… * The already existing PCM3168A Linux kernel driver: https://github.com/raspberrypi/linux/blob/rpi-4.14.y/sound/soc/codecs/pcm31… ... however, I cannot get it to properly work, so I hope I can get some help filling in the blanks. First question: * Q1: In the above setup, the pcm3168a.c is a "codec class driver"; but is simple-card.c a platform class or a machine class driver? Also, would the device tree overlay act as a machine class driver, since it "acts as the glue that describes and binds the other component drivers together" (sound/soc/overview.html)? Here are the relevant device tree fragments of the somewhat working device tree overlay, cased on `simple-card.c`: ``` fragment@3 { target = <&spi0>; __overlay__ { #address-cells = <1>; #size-cells = <0>; status = "okay"; pcm3168a: codec@1 { #sound-dai-cells = <0>; reg = <1>; compatible = "ti,pcm3168a"; spi-max-frequency = <1000000>; clocks = <&clocks 38>; // * BCM2835_CLOCK_GP0 * / clock-names = "scki"; pinctrl-names = "default"; pinctrl-0 = <&gpclk0_gpio4>; VDD1-supply = <&vdd_3v3_reg>; VDD2-supply = <&vdd_3v3_reg>; VCCAD1-supply = <&vdd_5v0_reg>; VCCAD2-supply = <&vdd_5v0_reg>; VCCDA1-supply = <&vdd_5v0_reg>; VCCDA2-supply = <&vdd_5v0_reg>; status = "okay"; }; }; }; fragment@4 { /////// a working one: target = <&sound>; slave_sound_overlay: __overlay__ { compatible = "simple-audio-card"; i2s-controller = <&i2s>; status = "okay"; simple-audio-card,name = "my-sound-card"; simple-audio-card,widgets = "Line", "Line In", "Line", "Line Out"; simple-audio-card,routing = "AIN1L", "Line In", "AIN1R", "Line In", "Line Out", "AOUT1L", "Line Out", "AOUT1R"; simple-audio-card,format = "right_j"; //"i2s"; simple-audio-card,mclk-fs = <128>; simple-audio-card,cpu { sound-dai = <&i2s>; }; sound_master: simple-audio-card,codec { sound-dai = <&pcm3168a>; system-clock-frequency = <6144000>; // both these work: 12288000/48000 = 256; 6144000/48000 = 128 }; }; }; ``` When I load this device tree on Raspbian, the following drivers get loaded: ``` pi@raspberrypi:~ $ lsmod Module Size Used by snd_soc_bcm2835_i2s 16384 2 snd_soc_pcm3168a_spi 16384 1 snd_soc_pcm3168a 24576 1 snd_soc_pcm3168a_spi snd_soc_simple_card 16384 0 snd_soc_simple_card_utils 20480 1 snd_soc_simple_card snd_soc_core 204800 4 snd_soc_pcm3168a,snd_soc_simple_card_utils,snd_soc_bcm2835_i2s,snd_soc_simple_card spidev 16384 0 spi_bcm2835 16384 0 snd_compress 20480 1 snd_soc_core snd_pcm_dmaengine 16384 1 snd_soc_core ... snd_pcm 98304 5 snd_pcm_dmaengine,snd_soc_pcm3168a,snd_soc_bcm2835_i2s,snd_bcm2835,snd_soc_core snd_timer 32768 1 snd_pcm snd 69632 7 snd_compress,snd_timer,snd_bcm2835,snd_soc_core,snd_pcm ... ``` ... with relevant syslog messages: ``` kernel: [ 410.676653] bus: 'platform': add driver spi-bcm2835 ... kernel: [ 410.818605] bus: 'platform': add driver snd-soc-dummy kernel: [ 410.819136] bus: 'platform': add driver soc-audio ... kernel: [ 410.830593] bus: 'platform': add driver asoc-simple-card ... kernel: [ 410.833990] bus: 'spi': add driver pcm3168a kernel: [ 410.840144] bus: 'platform': add driver bcm2835-i2s ... kernel: [ 410.857249] asoc-simple-card soc:sound: pcm3168a-dac <-> 3f203000.i2s mapping ok ... systemd-udevd[921]: Execute 'load' 'platform:snd-soc-dummy' systemd-udevd[921]: No module matches 'platform:snd-soc-dummy' ... systemd-udevd[914]: created db file '/run/udev/data/+sound:card1' for '/devices/platform/soc/soc:sound/sound/card1' ... systemd[1]: sys-devices-platform-soc-soc:sound-sound-card1.device: Changed dead -> plugged ... systemd-udevd[921]: handling device node '/dev/snd/pcmC1D0p', devnum=c116:48, mode=0660, uid=0, gid=29 ... systemd-udevd[917]: handling device node '/dev/snd/controlC1', devnum=c116:32, mode=0660, uid=0, gid=29 ... ``` After this, I have a playback card device - but not a capture one: ``` pi@raspberrypi:~ $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ... card 1: mysoundcard [my-sound-card], device 0: bcm2835-i2s-pcm3168a-dac pcm3168a-dac-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 pi@raspberrypi:~ $ arecord -l **** List of CAPTURE Hardware Devices **** pi@raspberrypi:~ $ ``` Note that at this point, with drivers loaded, I do have "card controls" accessible by amixer - but no "simple controls": ``` pi@raspberrypi:~ $ amixer -d -D hw:CARD=mysoundcard info Card hw:CARD=mysoundcard 'mysoundcard'/'my-sound-card' Mixer name : '' Components : '' Controls : 45 amixer: Mixer load hw:CARD=mysoundcard error: Device or resource busy pi@raspberrypi:~ $ amixer -d -D hw:CARD=mysoundcard scontrols amixer: Mixer hw:CARD=mysoundcard load error: Device or resource busy pi@raspberrypi:~ $ amixer -d -D hw:CARD=mysoundcard controls numid=19,iface=MIXER,name='Master Playback Volume' numid=42,iface=MIXER,name='Master Capture Volume' numid=41,iface=MIXER,name='ADC Overflow Flag Polarity' ... numid=13,iface=MIXER,name='DAC4 Zero Flag' pi@raspberrypi:~ $ amixer -d -D hw:CARD=mysoundcard cset iface=MIXER,name='Master Playback Volume' 81% VERIFY ID: numid=0,iface=MIXER,name='Master Playback Volume' numid=19,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=201,step=0 : values=163 | dBscale-min=-100.50dB,step=0.50dB,mute=1 ``` The accessible controls are, in fact, defined in pcm3168a.c (in ...struct snd_kcontrol_new pcm3168a_snd_controls[]) - so, more questions: * Q2: Is the amixer "error: Device or resource busy" an indication that: *no* driver loaded by the device tree overlay, has a definition for any mixer "simple control"? If so, can I set that up via device tree overlay - or do I need to code it in a driver? If coding a driver, which driver should be patched - pcm3168a.c or simple-card.c? * Q3: In the above `amixer cset` of 'Master Playback Volume', I can see `mute=1` in the output. Does that mean master playback is muted? If so, how do I unmute it with amixer? * Q4: As far as I can see, `alsamixer` deals only with "simple controls", not "card controls" (it exits with `cannot load mixer controls: Device or resource busy` as soon as I choose mysoundcard). Is there any ALSA GUI software that can show and manipulate all card controls of a card? Note that pcm3168a.c has notions of a *stereo* DAC and ADC, so the accessible amixer controls refer to ADC1-ADC3 and DAC1-DAC4. Finally, I test this with: ``` pi@raspberrypi:~ $ speaker-test -D hw:CARD=mysoundcard,DEV=0 -c 2 -t sine -f 440 speaker-test 1.1.3 Playback device is hw:CARD=mysoundcard,DEV=0 Stream parameters are 48000Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 131072 Period size range from 64 to 65536 Using max buffer size 131072 Periods = 4 was set period_size = 32768 was set buffer_size = 131072 0 - Front Left 1 - Front Right Time per period = 2.781788 ... ``` This is what I observe when speaker-test runs: - I've captured I2S and SPI signals with Saleae Logic: GPCLK0 kicks in first, after about 20 ms, SPI sends on MOSI (and SCL) 14 bytes in about 1 ms - immediately after, BCLK starts; and after some 50 ms, LRCLK and DOUT start in synchrony. So, at least from the RPi3B/BCM2837 side, at first glance all seems fine. (I've also used I2S analyzer in the software, and can confirm there's a sinusoid in the I2S DOUT data, though I cannot decode it 100% yet, since I see some discontinuities in the plots) - I've also measured PCM3168A outputs with "analog" scope - unfortunately I had a massive 50 Hz hum where I've measured, so it's difficult to say much with certainty. But basically, when speaker-test runs, there is a significant HF noise appearing on the DAC outputs (which disappears when speaker-test stops). I cannot tell if this is output from the DAC, but faulty, because of misconfiguration of the PCM3168a chip - or if the DAC is actually mute, and this is just noise leaking from GPCLK0 and similar signals (I'll need to find a quieter lab first to re-do measurements) Here are some more questions: * Q5: How would I go about reading the internal configuration registers of the PCM3168A? I guess I could use printk's in pcm3168a.c - but is there an easy/generic way of reading those from user space? * Q6: Is it possible to set internal configuration registers of the PCM3168A from a device tree overlay - say, if I'd like to set up DAC1 and DAC2 (as per datasheet, which is = stereo DAC1 in pcm3168a.c) to work in differential (or single-ended) mode? Note that the device tree overlay sets "right_j" (right justified) digital data exchange (I2S/PCM) format; because whenever I try to set it to "i2s", running speaker-test fails with "Unable to set hw params for playback: Invalid argument" and these syslog messages: ``` pcm3168a spi0.1: 32-bit frames are supported only for slave mode using right justified pcm3168a spi0.1: ASoC: can't set pcm3168a-dac hw params: -22 ``` * Q6: Where did this "32-bit frames" come from, when using format "i2s"? As far as I see it, speaker-test requested S16_LE stereo, meaning 16-bits need to be clocked in when LRCLK is 'L', and 16-bits when LRCLK is 'R' - can't see a "32-bit frame" here anywhere? On the other hand, there is in the PCM3168A datasheet "Table 11. Audio Data Interface Formats and Sampling Rate, Bit Clock, and System Clock Restrictions", with "(1) BCK = 48 fS, 32 fS is supported only in slave mode; BCK = 32 fS is supported only for 16-bit data length." - Does this mean that, since I've requested 16-bit samples, earlier code decided that BCK should be 32 fS, resulting with this "32-bit frame", which then fails at the point it does ?! Can I explicitly override this frame length in the device tree overlay - if so, where? * Q7: I'm very confused about how the master/slave I2S relationship is established in a device tree with `simple-audio-card`. For instance, I've seen `simple-audio-card,bitclock-master` and `,frame-master`, but also `,bitclock-slave` and `,frame-slave`; and yet, my device tree fragment above, uses none of those - and still, at least some of the system works?! So some relationship must have been established implicitly - my guess is (ignoring the `sound_master` label) that since GPCLK0 is set as SCKI of pcm3168a, bcm2835-i2s becomes master, and pcm3168a (codec) becomes slave. Is that correct - if not, how would one derive the actual implied relationship? I should mention, that one of the biggest mysteries for me here is: * Q8: Why is there no capture interface with this usage of `simple-audio-card` in device tree? In fact, I can see nowhere in the above device tree fragment that there is an explicit setup for playback - and yet, I do get a playback interface (and partially, it works)? I would have thought, that the setup of `simple-audio-card,widgets` and `simple-audio-card,routing` (most of the other .dts examples I've seen, have this as the only reference to input/capture vs output/playback) would have been enough to infer that capture interface is needed too, since I refer to "AIN1L", "AIN1R" along with "AOUT1L", "AOUT1R" (all defined in pcm3168a.c) - but apparently, not? Only other example I've seen is `arch/arm/boot/dts/overlays/pibell-overlay.dts`, which sets up `capture_link: simple-audio-card,dai-link@0` and `playback_link: simple-audio-card,dai-link@1` in `target = <&sound>`; but then again, it uses different chips for input and output - is this approach preferable, also when a single chip de als with both playback and capture? Now, an interesting thing is that, even in my above device tree fragment, if I just change `compatible = "simple-audio-card";` to `compatible = "simple-scu-audio-card";` - drivers will typically load with the device tree, but speaker-test will fail with a segfault at very start. To be sure, I tried writing another device tree fragment for "simple-scu-audio-card": ``` fragment@4 { target = <&sound>; slave_sound_overlay: __overlay__ { compatible = "simple-scu-audio-card"; i2s-controller = <&i2s>; status = "okay"; simple-audio-card,name = "my-sound-card"; simple-audio-card,format = "i2s"; simple-audio-card,convert-rate = <48000>; // simple-scu-card.txt (simple-scu-audio-card); prob no effect with simple-card.c (simple-audio-card) only; //simple-audio-card,prefix = "pcm3168a"; // NOTE: this causes the // ASoC: no source widget found for AOUT1L! // without ,widgets: "ASoC: no sink widget found for Line Out/In" simple-audio-card,widgets = //~ "Line", "Line In", "Line", "Line Out"; simple-audio-card,routing = "Line Out","AOUT1L", "Line Out","AOUT1R"; //, //~ "AIN1L","Line In", //~ "AIN1R","Line In"; //"Playback","DAC1"; //passes without simple-audio-card,widgets - but segfaults sndcpu: simple-audio-card,cpu { sound-dai = <&i2s>; }; sndcodec: simple-audio-card,codec { sound-dai = <&pcm3168a>; system-clock-frequency = <12288000>; }; }; }; ``` The funny thing here is - even if the "Line In" references are commented in ,routing - still, both a playback and a capture interface appear, along with the load of device tree and corresponding drivers! However, speaker-test causes segfault with a stack trace like: ``` kernel: [20819.557733] Unable to handle kernel NULL pointer dereference at virtual address 000000b0 ... kernel: [20819.557997] Hardware name: BCM2835 kernel: [20819.558008] task: 9f31ad00 task.stack: 9f3c8000 kernel: [20819.558148] PC is at is_connected_output_ep+0x1c/0x1fc [snd_soc_core] kernel: [20819.558274] LR is at snd_soc_dapm_dai_get_connected_widgets+0x90/0x1e4 [snd_soc_core] ... [ 1190.725416] [<7f4d1e50>] (is_connected_output_ep [snd_soc_core]) from [<7f4d4cf4>] (snd_soc_dapm_dai_get_connected_widgets+0x90/0x1e4 [snd_soc_core]) ### sound/soc/soc-dapm.c [ 1190.725663] [<7f4d4cf4>] (snd_soc_dapm_dai_get_connected_widgets [snd_soc_core]) from [<7f4db064>] (dpcm_path_get+0x2c/0x8c [snd_soc_core]) ### sound/soc/soc-dapm.c [ 1190.725908] [<7f4db064>] (dpcm_path_get [snd_soc_core]) from [<7f4db55c>] (dpcm_fe_dai_open+0x74/0x688 [snd_soc_core]) ### sound/soc/soc-pcm.c [ 1190.726104] [<7f4db55c>] (dpcm_fe_dai_open [snd_soc_core]) from [<7f101e9c>] (snd_pcm_open_substream+0x6c/0x114 [snd_pcm]) ### sound/soc/soc-pcm.c [ 1190.726238] [<7f101e9c>] (snd_pcm_open_substream [snd_pcm]) from [<7f101ffc>] (snd_pcm_open+0xb8/0x218 [snd_pcm]) [ 1190.726370] [<7f101ffc>] (snd_pcm_open [snd_pcm]) from [<7f102214>] (snd_pcm_playback_open+0x4c/0x6c [snd_pcm]) [ 1190.726498] [<7f102214>] (snd_pcm_playback_open [snd_pcm]) from [<7f0d24f8>] (snd_open+0xa4/0x144 [snd]) [ 1190.726573] [<7f0d24f8>] (snd_open [snd]) from [<80288a54>] (chrdev_open+0xe0/0x1b4) ... ``` * Q9: I've added some extra prints to driver code - but as far as I can see, `simple-audio-card` never ends up calling `is_connected_output_ep`, but `simple-scu-audio-card` does; unless I've made a mistake during debugging, would there a reason for that? * Q10: How can I tell `simple-scu-audio-card` which DACs/ADCs should be used in playback and capture device - given that apparently ,widgets and ,routing alone are not enough to specify state (given that it creates a capture device, even if there are no references to AIN*/Line in there) ?! And finally, in general: * Q11: I generally want to use PCM3168A chip DAC1 and DAC2 as stereo OUT (driver: stereo DAC1), and chip ADC1 and ADC2 as stereo IN (driver: stereo ADC1). Am I better off basing myself on `simple-audio-card`, or on `simple-scu-audio-card`, in the device tree overlay? In fact, is it possible to set up this kind of operation using existing drivers in a device tree overlay - or would I have to add my own driver code (either as patch of existing drivers, or as "my own" driver, possibly started from an existing driver)? * Q12: I'd also want to use (driver: stereo DAC1) and (driver: stereo ADC1) in full-duplex mode. Can this be set up solely via `dai-tdm-slot-num`, `-width`, `-tx-mask` and `-rx-mask` under `simple-audio-card,cpu` in device tree overlay - or would I have to add my own driver code (either patching, or "my own")? Many thanks in advance for any pointers!

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[alsa-devel] [PATCH] ALSA: hda - Enable additional HD-Audio codec information
by Shrikant Dharwadkar 08 May '18

08 May '18

Enable additional HD-Audio codec information to be parsed using alsa-info script to generate codec information with CONFIG_SND_SOC enabled Function group nodes -Add power state controls -Add subordinate node count parameters -Add function group type information -Add function group capabilities -Add unsolicited response controls -Add input converter SDI select information -Add GPIO enable control -Add GPIO direction information -Add GPIO wake enable information -Add GPIO unsolicited control information -Add GPIO sticky mask data -Add GPIO data sets associated with GPIO signals Widget nodes -Add pin widget controls -Add power state controls -Add converter stream and channel controls -Add processing state information -Add processing capabilities -Add volume knob capabilities -Add volume knob control information -Add digital converter control information -Add pin sense control information -Add EAPD/BTL enable controls -Add unsolicited response controls Signed-off-by: Shrikant Dharwadkar <shrikant.dharwadkar(a)intel.com> --- sound/hda/hdac_sysfs.c | 167 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 167 insertions(+) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index fb2aa34..b1acff2 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -37,6 +37,9 @@ static ssize_t type##_show(struct device *dev, \ } \ static DEVICE_ATTR_RO(type) +#define CODEC_PRINT(verb) sprintf(buf, "0x%08x\n", \ + snd_hdac_read_parm(codec, nid, verb)) + CODEC_ATTR(type); CODEC_ATTR(vendor_id); CODEC_ATTR(subsystem_id); @@ -173,6 +176,12 @@ static ssize_t pin_caps_show(struct hdac_device *codec, hda_nid_t nid, snd_hdac_read_parm(codec, nid, AC_PAR_PIN_CAP)); } +static ssize_t pin_widget_ctl_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PIN_WIDGET_CONTROL); +} + static ssize_t pin_cfg_show(struct hdac_device *codec, hda_nid_t nid, struct widget_attribute *attr, char *buf) { @@ -243,6 +252,14 @@ static ssize_t power_caps_show(struct hdac_device *codec, hda_nid_t nid, snd_hdac_read_parm(codec, nid, AC_PAR_POWER_STATE)); } +static ssize_t power_states_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (nid != codec->afg && !(get_wcaps(codec, nid) & AC_WCAP_POWER)) + return 0; + return CODEC_PRINT(AC_VERB_GET_POWER_STATE); +} + static ssize_t gpio_caps_show(struct hdac_device *codec, hda_nid_t nid, struct widget_attribute *attr, char *buf) { @@ -266,27 +283,166 @@ static ssize_t connections_show(struct hdac_device *codec, hda_nid_t nid, return ret; } +static ssize_t converter_stream_channel_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_CONV); +} + +static ssize_t processing_states_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PROC_STATE); +} + +static ssize_t subordinate_node_count_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_NODE_COUNT); +} + +static ssize_t function_group_type_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_FUNCTION_TYPE); +} + +static ssize_t processing_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_PROC_CAP); +} + +static ssize_t volume_knob_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_VOL_KNB_CAP); +} + +static ssize_t volume_knob_ctl_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_VOLUME_KNOB_CONTROL); +} + +static ssize_t digital_converter_ctl_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_DIGI_CONVERT_1); +} + +static ssize_t pin_sense_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PIN_SENSE); +} + +static ssize_t EAPD_BTL_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_EAPD_BTLENABLE); +} + +static ssize_t unsol_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_UNSOLICITED_RESPONSE); +} + +static ssize_t sdi_select_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_SDI_SELECT); +} + +static ssize_t gpio_enable_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_MASK); +} + +static ssize_t gpio_direction_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_DIRECTION); +} + +static ssize_t gpio_wake_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_WAKE_MASK); +} + +static ssize_t gpio_unsol_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK); +} + +static ssize_t gpio_sticky_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_STICKY_MASK); +} + +static ssize_t gpio_data_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_DATA); +} + static WIDGET_ATTR_RO(caps); static WIDGET_ATTR_RO(pin_caps); static WIDGET_ATTR_RO(pin_cfg); +static WIDGET_ATTR_RO(pin_widget_ctl); static WIDGET_ATTR_RO(pcm_caps); static WIDGET_ATTR_RO(pcm_formats); static WIDGET_ATTR_RO(amp_in_caps); static WIDGET_ATTR_RO(amp_out_caps); static WIDGET_ATTR_RO(power_caps); +static WIDGET_ATTR_RO(power_states); static WIDGET_ATTR_RO(gpio_caps); static WIDGET_ATTR_RO(connections); +static WIDGET_ATTR_RO(converter_stream_channel); +static WIDGET_ATTR_RO(processing_states); +static WIDGET_ATTR_RO(subordinate_node_count); +static WIDGET_ATTR_RO(function_group_type); +static WIDGET_ATTR_RO(processing_caps); +static WIDGET_ATTR_RO(volume_knob_caps); +static WIDGET_ATTR_RO(volume_knob_ctl); +static WIDGET_ATTR_RO(digital_converter_ctl); +static WIDGET_ATTR_RO(pin_sense); +static WIDGET_ATTR_RO(EAPD_BTL); +static WIDGET_ATTR_RO(unsol_caps); +static WIDGET_ATTR_RO(sdi_select); +static WIDGET_ATTR_RO(gpio_enable); +static WIDGET_ATTR_RO(gpio_direction); +static WIDGET_ATTR_RO(gpio_wake); +static WIDGET_ATTR_RO(gpio_unsol); +static WIDGET_ATTR_RO(gpio_sticky); +static WIDGET_ATTR_RO(gpio_data); static struct attribute *widget_node_attrs[] = { &wid_attr_caps.attr, &wid_attr_pin_caps.attr, + &wid_attr_pin_widget_ctl.attr, &wid_attr_pin_cfg.attr, &wid_attr_pcm_caps.attr, &wid_attr_pcm_formats.attr, &wid_attr_amp_in_caps.attr, &wid_attr_amp_out_caps.attr, &wid_attr_power_caps.attr, + &wid_attr_power_states.attr, &wid_attr_connections.attr, + &wid_attr_converter_stream_channel.attr, + &wid_attr_processing_states.attr, + &wid_attr_processing_caps.attr, + &wid_attr_volume_knob_caps.attr, + &wid_attr_volume_knob_ctl.attr, + &wid_attr_digital_converter_ctl.attr, + &wid_attr_pin_sense.attr, + &wid_attr_EAPD_BTL.attr, + &wid_attr_unsol_caps.attr, NULL, }; @@ -296,7 +452,18 @@ static struct attribute *widget_afg_attrs[] = { &wid_attr_amp_in_caps.attr, &wid_attr_amp_out_caps.attr, &wid_attr_power_caps.attr, + &wid_attr_power_states.attr, &wid_attr_gpio_caps.attr, + &wid_attr_subordinate_node_count.attr, + &wid_attr_function_group_type.attr, + &wid_attr_unsol_caps.attr, + &wid_attr_sdi_select.attr, + &wid_attr_gpio_enable.attr, + &wid_attr_gpio_direction.attr, + &wid_attr_gpio_wake.attr, + &wid_attr_gpio_unsol.attr, + &wid_attr_gpio_sticky.attr, + &wid_attr_gpio_data.attr, NULL, }; -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: usb: mixer: volume quirk for CM102-A+/102S+
by Federico Cuello 07 May '18

07 May '18

Currently it's not possible to set volume lower than 26% (it just mutes). Also fixes this warning: Warning! Unlikely big volume range (=9472), cval->res is probably wrong. [13] FU [PCM Playback Volume] ch = 2, val = -9473/-1/1 , and volume works fine for full range. --- sound/usb/mixer.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 344d7b069d59..bb5ab7a7dfa5 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -967,6 +967,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; + case USB_ID(0x0d8c, 0x0103): + if (!strcmp(kctl->id.name, "PCM Playback Volume")) { + usb_audio_info(chip, + "set volume quirk for CM102-A+/102S+\n"); + cval->min = -256; + } + break; + case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): -- 2.17.0

1 0

[alsa-devel] [PATCH v3 0/9] ALSA: hda/ca0132: Patch Series for Recon3Di and Sound Blaster Z Support
by Connor McAdams 07 May '18

07 May '18

This patchset adds support for the Sound Blaster Z and the Recon3Di. In order to figure out how to get these cards to work, I made a program called QemuHDADump[1], which uses the trace function of qemu to see interactions with the memory mapped pci BAR space of the card being used in the virtual machine. With this, I obtain the CORB buffer location to get the command verbs, and then dump them each time the buffer rolls over. This program may be useful for fixing other HDA related driver issues where there is no documentation for the device. So far, I have been able to get all features supported on the Sound Blaster Z and the Recon3Di. All output and input effects work, all inputs and outputs work, and just about anything else I can think of. I have also added new controls in order to select the new inputs and outputs, as well as controls to change the effect levels and presets. I have also added the ability to use firmware taken from the Windows drivers of both the Sound Blaster Z and Recon3Di. I am trying to get into contact with Creative to get permission to redistribute these along with the current file included with the Chromebook, but they have not been very responsive. Luckily, the cards work with the Chromebook firmware just fine, although I believe there has to be a reason they have different firmware in Windows. I will not link to the firmwares here, but if you look up my thread on Creative Labs forums, you will find the link to download the firmwares there. I am willing to help get the other non-working cards such as the ZxR and the newer AE-5 working too, but I will need someone willing to run QemuHDADump in a virtual machine in order to get the commands. So, in summary: -This patchset makes the cards work better than they did before (they really didn't work before) -This patchset leaves the original chromebook related stuff alone. Thanks. [1] https://github.com/Conmanx360/QemuHDADump Bugs: ------------------------------------------------------------------------------- Recon3Di: (Reported by Mariusz Ceier) ******************* -Occasionally switching between rear and front mic breaks the input until computer is shutdown or put to sleep. -Surround Sound works, but is inconsistent. Sometimes, just updating the volume fixes it, and sometimes, it requires a restart. Sound Blaster Z: ******************* -none that I'm aware of. Version changes: ------------------------------------------------------------------------------- v1: ******************* -Massive patch formatting failure, please ignore v1. v2: ******************* -Fixed patch formatting failure. v3: ******************* -Fixed mem_base unmap, instead of checking for QUIRK_SBZ on exit, have it check if the area is mapped, and if it is, unmap it. Also make it unmap after all other commands are finished. -Change notification of failure to map mem_base from codec_dbg to codec_warn, and use codec_info to tell the user that their card might have been incorrectly identified as a Sound Blaster Z. -Remove commented out commands in sbz_exit_chip function, only reintroduce them when their functions are defined. Connor McAdams (9): ALSA: hda/ca0132: R3Di and SBZ quirk entires + alt firmware loading ALSA: hda/ca0132: Add pincfg for SBZ + R3Di, add fp hp auto-detect ALSA: hda/ca0132: Add PCI region2 iomap for SBZ ALSA: hda/ca0132: Add extra exit functions for R3Di and SBZ ALSA: hda/ca0132: add/change helper functions for R3Di and SBZ ALSA: hda/ca0132: add alt_select_in/out for R3Di + SBZ ALSA: hda/ca0132: Add DSP Volume set and New mixers for SBZ + R3Di ALSA: hda/ca0132: add ca0132_alt_set_vipsource ALSA: hda/ca0132: Add new control changes for SBZ + R3Di sound/pci/hda/patch_ca0132.c | 3055 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 2941 insertions(+), 114 deletions(-) -- 2.7.4

2 12

[alsa-devel] [PATCH v3 00/20] Allow compile-testing NO_DMA (drivers)
by Geert Uytterhoeven 07 May '18

07 May '18

Hi all, As of v4.17-rc1, patch series "[PATCH v2 0/5] Allow compile-testing NO_DMA (core)" (https://lkml.org/lkml/2018/3/16/435) has been included upstream, and drivers using the DMA API can be compile-tested on platforms selecting NO_DMA. This follow-up patch series removes dependencies on HAS_DMA for symbols that already have platform dependencies implying HAS_DMA, which increases compile-coverage. Please apply to your tree if appropriate. Changes compared to v2: - Add Acked-by, - Rebased to v4.17-rc1, dropping applied patch for scsi/hisi_sas, - Handle new VIDEO_RENESAS_CEU symbol, - Drop obsolete note about FSL_FMAN. Changes compared to v1: - Add Reviewed-by, Acked-by, - Drop dependency of SND_SOC_LPASS_IPQ806X on HAS_DMA, - Drop dependency of VIDEOBUF{,2}_DMA_{CONTIG,SG} on HAS_DMA, - Drop new dependencies of VIDEO_IPU3_CIO2, DVB_C8SECTPFE, and MTD_NAND_MARVELL on HAS_DMA, - Split in per-subsystem patches, - Split-off the core part in a separate series. This series is against v4.17-rc1. It can also be found at https://git.kernel.org/pub/scm/linux/kernel/git/geert/linux-m68k.git/log/?h… It has been compile-tested with allmodconfig and allyesconfig for m68k/sun3, and has received attention from the kbuild test robot. Thanks for applying! Geert Uytterhoeven (20): ASoC: Remove depends on HAS_DMA in case of platform dependency ata: Remove depends on HAS_DMA in case of platform dependency crypto: Remove depends on HAS_DMA in case of platform dependency fbdev: Remove depends on HAS_DMA in case of platform dependency firewire: Remove depends on HAS_DMA in case of platform dependency fpga: Remove depends on HAS_DMA in case of platform dependency i2c: Remove depends on HAS_DMA in case of platform dependency iio: adc: Remove depends on HAS_DMA in case of platform dependency iommu: Remove depends on HAS_DMA in case of platform dependency lightnvm: Remove depends on HAS_DMA in case of platform dependency mailbox: Remove depends on HAS_DMA in case of platform dependency media: Remove depends on HAS_DMA in case of platform dependency mmc: Remove depends on HAS_DMA in case of platform dependency mtd: Remove depends on HAS_DMA in case of platform dependency net: Remove depends on HAS_DMA in case of platform dependency remoteproc: Remove depends on HAS_DMA in case of platform dependency serial: Remove depends on HAS_DMA in case of platform dependency spi: Remove depends on HAS_DMA in case of platform dependency staging: vc04_services: Remove depends on HAS_DMA in case of platform dependency usb: Remove depends on HAS_DMA in case of platform dependency drivers/ata/Kconfig | 2 -- drivers/crypto/Kconfig | 14 +++------ drivers/firewire/Kconfig | 1 - drivers/fpga/Kconfig | 1 - drivers/i2c/busses/Kconfig | 3 -- drivers/iio/adc/Kconfig | 2 -- drivers/iommu/Kconfig | 5 ++- drivers/lightnvm/Kconfig | 2 +- drivers/mailbox/Kconfig | 2 -- drivers/media/common/videobuf2/Kconfig | 2 -- drivers/media/pci/dt3155/Kconfig | 1 - drivers/media/pci/intel/ipu3/Kconfig | 1 - drivers/media/pci/solo6x10/Kconfig | 1 - drivers/media/pci/sta2x11/Kconfig | 1 - drivers/media/pci/tw5864/Kconfig | 1 - drivers/media/pci/tw686x/Kconfig | 1 - drivers/media/platform/Kconfig | 42 +++++++++---------------- drivers/media/platform/am437x/Kconfig | 2 +- drivers/media/platform/atmel/Kconfig | 4 +-- drivers/media/platform/davinci/Kconfig | 6 ---- drivers/media/platform/marvell-ccic/Kconfig | 3 +- drivers/media/platform/rcar-vin/Kconfig | 2 +- drivers/media/platform/soc_camera/Kconfig | 3 +- drivers/media/platform/sti/c8sectpfe/Kconfig | 2 +- drivers/media/v4l2-core/Kconfig | 2 -- drivers/mmc/host/Kconfig | 10 ++---- drivers/mtd/nand/raw/Kconfig | 8 ++--- drivers/mtd/spi-nor/Kconfig | 2 +- drivers/net/ethernet/amd/Kconfig | 2 +- drivers/net/ethernet/apm/xgene-v2/Kconfig | 1 - drivers/net/ethernet/apm/xgene/Kconfig | 1 - drivers/net/ethernet/arc/Kconfig | 6 ++-- drivers/net/ethernet/broadcom/Kconfig | 2 -- drivers/net/ethernet/calxeda/Kconfig | 2 +- drivers/net/ethernet/hisilicon/Kconfig | 2 +- drivers/net/ethernet/marvell/Kconfig | 8 ++--- drivers/net/ethernet/mellanox/mlxsw/Kconfig | 2 +- drivers/net/ethernet/renesas/Kconfig | 2 -- drivers/net/wireless/broadcom/brcm80211/Kconfig | 1 - drivers/net/wireless/quantenna/qtnfmac/Kconfig | 2 +- drivers/remoteproc/Kconfig | 1 - drivers/spi/Kconfig | 12 ++----- drivers/staging/media/davinci_vpfe/Kconfig | 1 - drivers/staging/media/omap4iss/Kconfig | 1 - drivers/staging/vc04_services/Kconfig | 1 - drivers/tty/serial/Kconfig | 4 --- drivers/usb/gadget/udc/Kconfig | 4 +-- drivers/usb/mtu3/Kconfig | 2 +- drivers/video/fbdev/Kconfig | 3 +- sound/soc/bcm/Kconfig | 3 +- sound/soc/kirkwood/Kconfig | 1 - sound/soc/pxa/Kconfig | 1 - sound/soc/qcom/Kconfig | 7 ++--- 53 files changed, 56 insertions(+), 142 deletions(-) -- 2.7.4 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

8 28

[alsa-devel] [PATCH] soundwire: Update email address for Vinod
by Vinod Koul 06 May '18

06 May '18

Update the email address for SoundWire maintainer Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 715d054531af..5caaad0d465d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -12687,7 +12687,7 @@ F: include/uapi/sound/ F: sound/ SOUND - COMPRESSED AUDIO -M: Vinod Koul <vinod.koul(a)intel.com> +M: Vinod Koul <vkoul(a)kernel.org> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Supported -- 2.7.4

2 1

[alsa-devel] [PATCH v3] media: i2c: tda1997: replace codec to component
by Kuninori Morimoto 05 May '18

05 May '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Tim, Mauro This replace patch is needed for ALSA SoC, otherwise it can't probe anymore. Mark, I think it needs your acked-by ? v2 -> v3 - fixup driver name (= tda1997) drivers/media/i2c/tda1997x.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/drivers/media/i2c/tda1997x.c b/drivers/media/i2c/tda1997x.c index 3021913..33d7fcf 100644 --- a/drivers/media/i2c/tda1997x.c +++ b/drivers/media/i2c/tda1997x.c @@ -2444,7 +2444,7 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct tda1997x_state *state = snd_soc_dai_get_drvdata(dai); - struct snd_soc_codec *codec = dai->codec; + struct snd_soc_component *component = dai->component; struct snd_pcm_runtime *rtd = substream->runtime; int rate, err; @@ -2452,11 +2452,11 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, err = snd_pcm_hw_constraint_minmax(rtd, SNDRV_PCM_HW_PARAM_RATE, rate, rate); if (err < 0) { - dev_err(codec->dev, "failed to constrain samplerate to %dHz\n", + dev_err(component->dev, "failed to constrain samplerate to %dHz\n", rate); return err; } - dev_info(codec->dev, "set samplerate constraint to %dHz\n", rate); + dev_info(component->dev, "set samplerate constraint to %dHz\n", rate); return 0; } @@ -2479,20 +2479,22 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, .ops = &tda1997x_dai_ops, }; -static int tda1997x_codec_probe(struct snd_soc_codec *codec) +static int tda1997x_codec_probe(struct snd_soc_component *component) { return 0; } -static int tda1997x_codec_remove(struct snd_soc_codec *codec) +static void tda1997x_codec_remove(struct snd_soc_component *component) { - return 0; } -static struct snd_soc_codec_driver tda1997x_codec_driver = { - .probe = tda1997x_codec_probe, - .remove = tda1997x_codec_remove, - .reg_word_size = sizeof(u16), +static struct snd_soc_component_driver tda1997x_codec_driver = { + .probe = tda1997x_codec_probe, + .remove = tda1997x_codec_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static int tda1997x_probe(struct i2c_client *client, @@ -2737,7 +2739,7 @@ static int tda1997x_probe(struct i2c_client *client, else formats = SNDRV_PCM_FMTBIT_S16_LE; tda1997x_audio_dai.capture.formats = formats; - ret = snd_soc_register_codec(&state->client->dev, + ret = devm_snd_soc_register_component(&state->client->dev, &tda1997x_codec_driver, &tda1997x_audio_dai, 1); if (ret) { @@ -2782,7 +2784,6 @@ static int tda1997x_remove(struct i2c_client *client) struct tda1997x_platform_data *pdata = &state->pdata; if (pdata->audout_format) { - snd_soc_unregister_codec(&client->dev); mutex_destroy(&state->audio_lock); } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Optimize the power consumption
by Oder Chiou 05 May '18

05 May '18

The patch optimizes the power consumption. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 48 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 36 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 20c0aee..f0cef52 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,8 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0117, 0x0f28 }, + { 0x02fb, 0x8089 }, }; static const struct reg_default rt5663_v2_reg[] = { @@ -593,7 +595,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0113, 0x2000 }, { 0x0114, 0x0000 }, { 0x0116, 0x0000 }, - { 0x0117, 0x0f00 }, + { 0x0117, 0x0f28 }, { 0x0118, 0x0006 }, { 0x0125, 0x2424 }, { 0x0126, 0x5550 }, @@ -693,7 +695,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0251, 0x0000 }, { 0x0252, 0x028a }, { 0x02fa, 0x0000 }, - { 0x02fb, 0x00a4 }, + { 0x02fb, 0x8089 }, { 0x02fc, 0x0300 }, { 0x0300, 0x0000 }, { 0x03d0, 0x0000 }, @@ -1556,6 +1558,14 @@ static int rt5663_jack_detect(struct snd_soc_component *component, int jack_inse RT5663_PWR_MB_MASK | RT5663_LDO1_DVO_MASK | RT5663_AMP_HP_MASK, RT5663_PWR_MB | RT5663_LDO1_DVO_0_9V | RT5663_AMP_HP_3X); + snd_soc_component_update_bits(component, RT5663_PWR_ANLG_1, + RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK | + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_VREF1 | RT5663_PWR_VREF2); + msleep(20); + snd_soc_component_update_bits(component, RT5663_PWR_ANLG_1, + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_FV1 | RT5663_PWR_FV2); snd_soc_component_update_bits(component, RT5663_AUTO_1MRC_CLK, RT5663_IRQ_POW_SAV_MASK, RT5663_IRQ_POW_SAV_EN); snd_soc_component_update_bits(component, RT5663_IRQ_1, @@ -1613,7 +1623,10 @@ static int rt5663_jack_detect(struct snd_soc_component *component, int jack_inse break; default: rt5663->jack_type = SND_JACK_HEADPHONE; - + snd_soc_component_update_bits(component, + RT5663_PWR_ANLG_1, + RT5663_PWR_MB_MASK | RT5663_PWR_VREF1_MASK | + RT5663_PWR_VREF2_MASK, 0); if (rt5663->pdata.impedance_sensing_num) break; @@ -1638,6 +1651,9 @@ static int rt5663_jack_detect(struct snd_soc_component *component, int jack_inse if (rt5663->jack_type == SND_JACK_HEADSET) rt5663_enable_push_button_irq(component, false); rt5663->jack_type = 0; + snd_soc_component_update_bits(component, RT5663_PWR_ANLG_1, + RT5663_PWR_MB_MASK | RT5663_PWR_VREF1_MASK | + RT5663_PWR_VREF2_MASK, 0); } dev_dbg(component->dev, "jack_type = %d\n", rt5663->jack_type); @@ -2307,6 +2323,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_HP_SIG_SRC1_MASK, RT5663_HP_SIG_SRC1_SILENCE); } else { + snd_soc_component_update_bits(component, + RT5663_DACREF_LDO, 0x3e0e, 0x3a0a); snd_soc_component_write(component, RT5663_DEPOP_2, 0x3003); snd_soc_component_update_bits(component, RT5663_HP_CHARGE_PUMP_1, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_DIS); @@ -2332,6 +2350,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x0); snd_soc_component_update_bits(component, RT5663_HP_CHARGE_PUMP_1, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); + snd_soc_component_update_bits(component, + RT5663_DACREF_LDO, 0x3e0e, 0); } break; @@ -3086,9 +3106,17 @@ static int rt5663_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_OFF: - snd_soc_component_update_bits(component, RT5663_PWR_ANLG_1, - RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK | - RT5663_PWR_FV1 | RT5663_PWR_FV2, 0x0); + if (rt5663->jack_type != SND_JACK_HEADSET) + snd_soc_component_update_bits(component, + RT5663_PWR_ANLG_1, + RT5663_PWR_VREF1_MASK | RT5663_PWR_VREF2_MASK | + RT5663_PWR_FV1 | RT5663_PWR_FV2 | + RT5663_PWR_MB_MASK, 0); + else + snd_soc_component_update_bits(component, + RT5663_PWR_ANLG_1, + RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK, + RT5663_PWR_FV1 | RT5663_PWR_FV2); break; default: @@ -3310,6 +3338,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) regmap_write(rt5663->regmap, RT5663_HP_IMP_SEN_19, 0x000c); regmap_write(rt5663->regmap, RT5663_DUMMY_1, 0x0324); regmap_write(rt5663->regmap, RT5663_DIG_MISC, 0x8001); + regmap_write(rt5663->regmap, RT5663_VREFADJ_OP, 0x0f28); regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xa23b); msleep(30); regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xf23b); @@ -3344,6 +3373,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x8003); regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x018c); regmap_write(rt5663->regmap, RT5663_HP_CHARGE_PUMP_1, 0x1e32); + regmap_write(rt5663->regmap, RT5663_DUMMY_2, 0x8089); regmap_write(rt5663->regmap, RT5663_DACREF_LDO, 0x3b0b); msleep(40); regmap_write(rt5663->regmap, RT5663_STO_DAC_MIXER, 0x0000); @@ -3578,15 +3608,9 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, RT5663_GPIO1_TYPE_MASK, RT5663_GPIO1_TYPE_EN); regmap_write(rt5663->regmap, RT5663_VREF_RECMIX, 0x0032); - regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xa2be); - msleep(20); - regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xf2be); regmap_update_bits(rt5663->regmap, RT5663_GPIO_2, RT5663_GP1_PIN_CONF_MASK | RT5663_SEL_GPIO1_MASK, RT5663_GP1_PIN_CONF_OUTPUT | RT5663_SEL_GPIO1_EN); - /* DACREF LDO control */ - regmap_update_bits(rt5663->regmap, RT5663_DACREF_LDO, 0x3e0e, - 0x3a0a); regmap_update_bits(rt5663->regmap, RT5663_RECMIX, RT5663_RECMIX1_BST1_MASK, RT5663_RECMIX1_BST1_ON); regmap_update_bits(rt5663->regmap, RT5663_TDM_2, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: cs35l35: Add use_single_rw to regmap config
by Paul Handrigan 05 May '18

05 May '18

Add the use_single_rw flag to regmap config since the device does not support bulk transactions over i2c. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- sound/soc/codecs/cs35l35.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index a4a2cb1..bd6226b 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1105,6 +1105,7 @@ static struct regmap_config cs35l35_regmap = { .readable_reg = cs35l35_readable_register, .precious_reg = cs35l35_precious_register, .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, }; static irqreturn_t cs35l35_irq(int irq, void *data) -- 2.7.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: sgtl5000: Make the node name generic
by Fabio Estevam 05 May '18

05 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> According to Devicetree Specification v0.2 document: "The name of a node should be somewhat generic, reflecting the function of the device and not its precise programming model." Do as suggested in the binding example. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 9a36c7e..0f21445 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -39,7 +39,7 @@ VDDIO 1.8V 2.5V 3.3V Example: -codec: sgtl5000@a { +sgtl5000: codec@a { compatible = "fsl,sgtl5000"; reg = <0x0a>; #sound-dai-cells = <0>; -- 2.7.4

3 11

[alsa-devel] [PATCH v3 0/7] ASoC: omap: Deprecate omap-pcm for sdma-pcm
by Peter Ujfalusi 05 May '18

05 May '18

Hi, Changes since v2: - Return -ENODEV from sdma_pcm_platform_register() in case !CONFIG_SND_SDMA_SOC Changes since v1: - Rebased on the n810 patches (Kconfig conflict) - Delete the omap-pcm as the last step instead of keeping the .c/.h file. If there is a need it can be resurrected with a simple revert. - I left out the Acked-by from Jarkko for the last patch since I have changed it omap-pcm served us well over the years, but there is no reason to keep it around as we can rely on generic dmaengine_pcm code without the need to maintain custom code. The series has been tested on: omap2 n810, omap3 beagle-xm, omap4 PandaES, omap4 Blaze (SDP), am57xx beagle-x15, dra7 evm. McBSP, McASP, McPDM, DMIC and HDMI audio was tested and I have found no issue with them. I was not able to test on OMAP1 for various reasons, but mainly because the Nokia 770 audio is not supported in mainline... Regards, Peter --- Peter Ujfalusi (7): ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm ASoC: davinci-mcasp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-hdmi-audio: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-dmic: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcpdm: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcbsp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap: Delete the obsolete omap-pcm include/sound/omap-pcm.h | 30 ---- sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-mcasp.c | 8 +- sound/soc/omap/Kconfig | 33 ++-- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-hdmi-audio.c | 5 +- sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/omap/omap-mcpdm.c | 4 +- sound/soc/omap/omap-pcm.c | 262 ------------------------------ sound/soc/omap/sdma-pcm.c | 68 ++++++++ sound/soc/omap/sdma-pcm.h | 21 +++ 12 files changed, 124 insertions(+), 321 deletions(-) delete mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.h -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 9

Re: [alsa-devel] Buffer size calculation based on period size
by noman pouigt 04 May '18

04 May '18

>>noman pouigt wrote: > I am using tinycap to capture the audio and > setting below settings: > bitwidth = 16 > rate = 16000 > channel = 1 > period_size = 320 > period_count = 2 > > I am wondering how is below > period_size and buffer_size calculated? > > Shouldn't period_size of 320 bytes and > period_count of 2 fail? >From the tinyalsa source code: >>param_set_min(&params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, config->period_size); >>So it uses either the supplied value, or if that is too small, the smallest allowed value. Probably I was not clear; Why ALSA picked period_size of 2042 bytes when the minimum period bytes specified is 4084? Reason for this question is probably my wrong understanding. AFAIK: period_bytes_min specified in hw_params means the minimum bytes after which audio driver can expect an interrupt. So, total buffer size should be multiple of period_bytes_min or multiple of user provided period size if it is greater than period_bytes_min. So, if the above understanding is right then buffer size should have been 4084*2 (number of periods asked by user). Ultimately the question is how period size is calculated given min_period_bytes, max_period_bytes, buffer_bytes_max from pcm_hardware and user supplied period size and period count? I tried looking into the code but looks like there is lot of macro magic going on in pcm native layer. > cat sub0/hw_params > > access: RW_INTERLEAVED > format: S16_LE > subformat: STD > channels: 1 > rate: 16000 (16000/1) > period_size: 2042 > buffer_size: 4084 >>The actual values of period_size and period_count are returned by >>pcm_open().

2 1

[alsa-devel] [PATCH] ALSA: pcm: Hide local_irq_disable/enable() and local_irqsave/restore()
by Sebastian Andrzej Siewior 04 May '18

04 May '18

From: Anna-Maria Gleixner <anna-maria(a)linutronix.de> The snd_pcm_stream_lock_irq*() functions decouple disabling interrupts from the actual locking process. This does not work as expected if the locking primitives are replaced like on preempt-rt. Provide one function for locking which uses correct locking primitives. Signed-off-by: Anna-Maria Gleixner <anna-maria(a)linutronix.de> Signed-off-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> --- sound/core/pcm_native.c | 85 +++++++++++++++++++++++++++-------------- 1 file changed, 57 insertions(+), 28 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d18b3982548b..92c10fc838e9 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -99,6 +99,57 @@ static inline void down_write_nonblock(struct rw_semaphore *lock) cond_resched(); } +#define PCM_LOCK_DEFAULT 0 +#define PCM_LOCK_IRQ 1 +#define PCM_LOCK_IRQSAVE 2 + +static unsigned long __snd_pcm_stream_lock_mode(struct snd_pcm_substream *substream, + unsigned int mode) +{ + unsigned long flags = 0; + if (substream->pcm->nonatomic) { + down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING); + mutex_lock(&substream->self_group.mutex); + } else { + switch (mode) { + case PCM_LOCK_DEFAULT: + read_lock(&snd_pcm_link_rwlock); + break; + case PCM_LOCK_IRQ: + read_lock_irq(&snd_pcm_link_rwlock); + break; + case PCM_LOCK_IRQSAVE: + read_lock_irqsave(&snd_pcm_link_rwlock, flags); + break; + } + spin_lock(&substream->self_group.lock); + } + return flags; +} + +static void __snd_pcm_stream_unlock_mode(struct snd_pcm_substream *substream, + unsigned int mode, unsigned long flags) +{ + if (substream->pcm->nonatomic) { + mutex_unlock(&substream->self_group.mutex); + up_read(&snd_pcm_link_rwsem); + } else { + spin_unlock(&substream->self_group.lock); + + switch (mode) { + case PCM_LOCK_DEFAULT: + read_unlock(&snd_pcm_link_rwlock); + break; + case PCM_LOCK_IRQ: + read_unlock_irq(&snd_pcm_link_rwlock); + break; + case PCM_LOCK_IRQSAVE: + read_unlock_irqrestore(&snd_pcm_link_rwlock, flags); + break; + } + } +} + /** * snd_pcm_stream_lock - Lock the PCM stream * @substream: PCM substream @@ -109,13 +160,7 @@ static inline void down_write_nonblock(struct rw_semaphore *lock) */ void snd_pcm_stream_lock(struct snd_pcm_substream *substream) { - if (substream->pcm->nonatomic) { - down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING); - mutex_lock(&substream->self_group.mutex); - } else { - read_lock(&snd_pcm_link_rwlock); - spin_lock(&substream->self_group.lock); - } + __snd_pcm_stream_lock_mode(substream, PCM_LOCK_DEFAULT); } EXPORT_SYMBOL_GPL(snd_pcm_stream_lock); @@ -127,13 +172,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock); */ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream) { - if (substream->pcm->nonatomic) { - mutex_unlock(&substream->self_group.mutex); - up_read(&snd_pcm_link_rwsem); - } else { - spin_unlock(&substream->self_group.lock); - read_unlock(&snd_pcm_link_rwlock); - } + __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_DEFAULT, 0); } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock); @@ -147,9 +186,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock); */ void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream) { - if (!substream->pcm->nonatomic) - local_irq_disable(); - snd_pcm_stream_lock(substream); + __snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQ); } EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq); @@ -161,19 +198,13 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq); */ void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream) { - snd_pcm_stream_unlock(substream); - if (!substream->pcm->nonatomic) - local_irq_enable(); + __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQ, 0); } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irq); unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream) { - unsigned long flags = 0; - if (!substream->pcm->nonatomic) - local_irq_save(flags); - snd_pcm_stream_lock(substream); - return flags; + return __snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQSAVE); } EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); @@ -187,9 +218,7 @@ EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, unsigned long flags) { - snd_pcm_stream_unlock(substream); - if (!substream->pcm->nonatomic) - local_irq_restore(flags); + __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQSAVE, flags); } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore); -- 2.17.0

2 1

[alsa-devel] Buffer size calculation based on period size
by noman pouigt 04 May '18

04 May '18

I am at my wits end trying to figure out how period_size is being calculated in pcm_native layer based on period_size and period_count being set by the userspace. I am using tinycap to capture the audio and setting below settings: bitwidth = 16 rate = 16000 channel = 1 period_size = 320 period_count = 2 I am wondering how is below period_size and buffer_size calculated? Shouldn't period_size of 320 bytes and period_count of 2 fail? cat sub0/hw_params access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 1 rate: 16000 (16000/1) period_size: 2042 buffer_size: 4084 And this is my settings in pcm_hardware structure: static struct snd_pcm_hardware s1000_pcm_hw = { .buffer_bytes_max = 1024*1024, .period_bytes_min = 4084, .period_bytes_max = (1024*1024)/2, .periods_min = 2, .periods_max = 32, };

2 1

[alsa-devel] [PATCH] ALSA: dice: support transmission mode at 176.4/192.0 kHz for M-Audio 610/2626
by Takashi Sakamoto 04 May '18

04 May '18

At highest sampling transmission frequency, ALSA dice driver detects jumbo payload in tx packets from M-Audio 610/2626. $ dmesg snd_dice fw1.0: Detect jumbo payload: 0388 0348 Usual dice devices have a quirk called as dual-wire incompliant to IEC 61883-6 at highest sampling transmission frequency. On the other hand, M-Audio 610/2626 is compliant. This is a reason of the detection of jumbo payload. This commit adds support compliant transmission mode. For these devices, tx packets are correctly handled by this driver, while rx packets can't make these devices generate sound, unfortunately. The other factor affects it. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 6 +++++- sound/firewire/dice/dice.c | 9 +++++++++ sound/firewire/dice/dice.h | 2 ++ 3 files changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 422a1ab1ddf5..e3dbe357b4fa 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -204,7 +204,11 @@ static int keep_resources(struct snd_dice *dice, * For this quirk, blocking mode is required and PCM buffer size should * be aligned to SYT_INTERVAL. */ - double_pcm_frames = rate > 96000; + if (rate < 176400 || dice->no_dualwire) + double_pcm_frames = false; + else + double_pcm_frames = true; + if (double_pcm_frames) { rate /= 2; pcm_chs *= 2; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 40f7a32e4893..8baad7704bf0 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -236,6 +236,15 @@ static int dice_probe(struct fw_unit *unit, (snd_dice_detect_formats_t)entry->driver_data; } + /* + * M-Audio units are compliant to IEC 61883-6 at high sampling + * transmission frequency thus don't support 'dualwire' quirk. These + * units have 0x0100d1 in its version field in config ROM, against + * the value of DICE_INTERFACE. + */ + if (entry->vendor_id == OUI_MAUDIO) + dice->no_dualwire = true; + spin_lock_init(&dice->lock); mutex_init(&dice->mutex); init_completion(&dice->clock_accepted); diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 505b79fea6d9..66e7a277ab68 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -113,6 +113,8 @@ struct snd_dice { bool global_enabled; struct completion clock_accepted; unsigned int substreams_counter; + + bool no_dualwire; }; enum snd_dice_addr_type { -- 2.14.1

1 0

[alsa-devel] [PATCH 1/9] R3Di and SBZ quirk entires + alt firmware loading
by Connor McAdams 04 May '18

04 May '18

This patch adds PCI quirk ID's for the Sound Blaster Z and Recon3Di. Only the currently tested ID's have been added. This patch also adds the ability to load alternative firmwares for each card, the firmwares can be obtained from within the Windows driver. The Recon3Di uses "ctefx-r3di.bin" and the Sound Blaster Z uses "ctefx-sbz.bin". If the alternative firmware for the given quirk is not found, the original ctefx.bin will be used. This has been confirmed to work for both the R3Di and the SBZ. This patch also makes the character array *dirstr a const. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 61 +++++++++++++++++++++++++++++++++++++++----- 1 file changed, 55 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea86..8346100 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -72,12 +72,16 @@ #define SCP_GET 1 #define EFX_FILE "ctefx.bin" +#define SBZ_EFX_FILE "ctefx-sbz.bin" +#define R3DI_EFX_FILE "ctefx-r3di.bin" #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP MODULE_FIRMWARE(EFX_FILE); +MODULE_FIRMWARE(SBZ_EFX_FILE); +MODULE_FIRMWARE(R3DI_EFX_FILE); #endif -static char *dirstr[2] = { "Playback", "Capture" }; +static const char *dirstr[2] = { "Playback", "Capture" }; enum { SPEAKER_OUT, @@ -734,6 +738,7 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; + bool alt_firmware_present; /* mixer and effects related */ unsigned char dmic_ctl; @@ -762,6 +767,8 @@ struct ca0132_spec { enum { QUIRK_NONE, QUIRK_ALIENWARE, + QUIRK_SBZ, + QUIRK_R3DI, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -782,6 +789,10 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), + SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI), {} }; @@ -3207,7 +3218,7 @@ static int ca0132_select_out(struct hda_codec *codec) pin_ctl & ~PIN_HP); /* enable speaker node */ pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); snd_hda_set_pin_ctl(codec, spec->out_pins[0], pin_ctl | PIN_OUT); } else { @@ -4370,11 +4381,49 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k) static bool ca0132_download_dsp_images(struct hda_codec *codec) { bool dsp_loaded = false; + struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - - if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) - return false; + /* + * Alternate firmwares for different variants. The Recon3Di apparently + * can use the default firmware, but I'll leave the option in case + * it needs it again. + */ + switch (spec->quirk) { + case QUIRK_SBZ: + if (request_firmware(&fw_entry, SBZ_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "SBZ alt firmware not detected. "); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Sound Blaster Z firmware selected."); + spec->alt_firmware_present = true; + } + break; + case QUIRK_R3DI: + if (request_firmware(&fw_entry, R3DI_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "Recon3Di alt firmware not detected."); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Recon3Di firmware selected."); + spec->alt_firmware_present = true; + } + break; + default: + spec->alt_firmware_present = false; + break; + } + /* + * Use default ctefx.bin if no alt firmware is detected, or if none + * exists for your particular codec. + */ + if (!spec->alt_firmware_present) { + codec_dbg(codec, "Default firmware selected."); + if (request_firmware(&fw_entry, EFX_FILE, + codec->card->dev) != 0) + return false; + } dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { @@ -4476,7 +4525,7 @@ static struct hda_verb ca0132_base_exit_verbs[] = { {} }; -/* Other verbs tables. Sends after DSP download. */ +/* Other verbs tables. Sends after DSP download. */ static struct hda_verb ca0132_init_verbs0[] = { /* chip init verbs */ {0x15, 0x70D, 0xF0}, -- 2.7.4

3 11

[alsa-devel] [PATCH 1/9] R3Di and SBZ quirk entires + alt firmware loading
by Connor McAdams 04 May '18

04 May '18

This patch adds PCI quirk ID's for the Sound Blaster Z and Recon3Di. Only the currently tested ID's have been added. This patch also adds the ability to load alternative firmwares for each card, the firmwares can be obtained from within the Windows driver. The Recon3Di uses "ctefx-r3di.bin" and the Sound Blaster Z uses "ctefx-sbz.bin". If the alternative firmware for the given quirk is not found, the original ctefx.bin will be used. This has been confirmed to work for both the R3Di and the SBZ. This patch also makes the character array *dirstr a const. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 55 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 51 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea86..9a0286e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -72,12 +72,16 @@ #define SCP_GET 1 #define EFX_FILE "ctefx.bin" +#define SBZ_EFX_FILE "ctefx-sbz.bin" +#define R3DI_EFX_FILE "ctefx-r3di.bin" #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP MODULE_FIRMWARE(EFX_FILE); +MODULE_FIRMWARE(SBZ_EFX_FILE); +MODULE_FIRMWARE(R3DI_EFX_FILE); #endif -static char *dirstr[2] = { "Playback", "Capture" }; +static const char *dirstr[2] = { "Playback", "Capture" }; enum { SPEAKER_OUT, @@ -734,6 +738,7 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; + bool alt_firmware_present; /* mixer and effects related */ unsigned char dmic_ctl; @@ -762,6 +767,8 @@ struct ca0132_spec { enum { QUIRK_NONE, QUIRK_ALIENWARE, + QUIRK_SBZ, + QUIRK_R3DI, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -782,6 +789,10 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), + SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI), {} }; @@ -4370,11 +4381,47 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k) static bool ca0132_download_dsp_images(struct hda_codec *codec) { bool dsp_loaded = false; + struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - - if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) - return false; + /* + * Alternate firmwares for different variants. The Recon3Di apparently + * can use the default firmware, but I'll leave the option in case + * it needs it again. + */ + switch (spec->quirk) { + case QUIRK_SBZ: + if (request_firmware(&fw_entry, SBZ_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "SBZ alt firmware not detected. "); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Sound Blaster Z firmware selected."); + spec->alt_firmware_present = true; + } + break; + case QUIRK_R3DI: + if (request_firmware(&fw_entry, R3DI_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "Recon3Di alt firmware not detected."); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Recon3Di firmware selected."); + spec->alt_firmware_present = true; + } + break; + default: + spec->alt_firmware_present = false; + break; + } + /* Use default ctefx.bin if no alt firmware is detected, or if none + * exists for your particular codec. + */ + if (spec->alt_firmware_present == false) { + codec_dbg(codec, "Default firmware selected."); + if (request_firmware(&fw_entry, EFX_FILE, + codec->card->dev) != 0) + return false; + } dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { -- 2.7.4

2 9

[alsa-devel] Segmentation fault in libasound.so.2.0.0 (alsa-lib-1.1.6)
by Paul Bordukoff 04 May '18

04 May '18

Hello everyone, I am using ALSA (alsa-lib-1.1.6) on Gentoo laptop with x32 ABI (linux-4.16.7). My mpv (0.27.2) gets segfault each time I try to play video with sound: pashgan@light ~/temp $ mpv ./SampleVideo_1280x720_10mb.mp4 Playing: ./SampleVideo_1280x720_10mb.mp4 (+) Video --vid=1 (*) (h264 1280x720 25.000fps) (+) Audio --aid=1 --alang=und (*) (aac 6ch 48000Hz) AO: [alsa] 48000Hz stereo 2ch float VO: [opengl] 1280x720 yuv420p Segmentation fault dmesg shows: [12526.319909] traps: mpv/ao[13621] general protection ip:f28ce4fa sp:e46f2d60 error:0 in libasound.so.2.0.0[f2843000+fe000] gdb session: (gdb) bt #0 0xf29b14fa in mix_areas_32_smp (size=3935624896, dst=0x56a6e7f0, src=0x56a6e7f0, sum=0x1, dst_step=4, src_step=4, sum_step=4) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_dmix_x86_64.h:143 #1 0xf29b2361 in mix_areas (size=3935493824, dst_ofs=4056, src_ofs=0, dst_areas=0x56a6cd90, src_areas=0x56a6dc30, dmix=0x56a6c900) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_dmix.c:197 #2 snd_pcm_dmix_sync_area (pcm=pcm@entry=0x56a6d270) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_dmix.c:379 #3 0xf29b267f in snd_pcm_dmix_start (pcm=<optimized out>) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_dmix.c:611 #4 0xf2972349 in __snd_pcm_start (pcm=0x56a6d270, pcm=0x56a6d270) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_local.h:434 #5 snd_pcm_start (pcm=<optimized out>) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm.c:1226 #6 0xf2972349 in __snd_pcm_start (pcm=0x56a6d8b0, pcm=0x56a6d8b0) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_local.h:434 #7 snd_pcm_start (pcm=<optimized out>) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm.c:1226 #8 0xf2979a78 in __snd_pcm_start (pcm=0x56a52e60, pcm=0x56a52e60) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_local.h:434 #9 snd1_pcm_write_areas (pcm=pcm@entry=0x56a52e60, areas=areas@entry=0xe47f2f00, offset=offset@entry=0, size=<optimized out>, size@entry=12288, func=func@entry=0xf2988770 <snd_pcm_plugin_write_areas>) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm.c:7421 #10 0xf29890e7 in snd_pcm_plugin_writei (pcm=<optimized out>, buffer=<optimized out>, size=12288) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_plugin.c:358 #11 0x56590280 in ?? () #12 0x56594846 in ?? () #13 0xf1b9808b in start_thread () from /libx32/libpthread.so.0 #14 0xf18d4e2f in clone () from /libx32/libc.so.6 (gdb) Same bug in Gentoo bug tracker: https://bugs.gentoo.org/654754 I need help to investigate pcm_dmix behaviour, I see asm code there (pcm_dmix_x86_64.h:143) but have no idea how can I debug it (my asm experience is far from ok). Can somebody guide me?

1 0

[alsa-devel] [PATCH v4] ASoC: tas6424: Allow disabling auto diagnostics for faster power-on
by Jean-Jacques Hiblot 04 May '18

04 May '18

The TAS6424 incorporates both DC-load and AC-load diagnostics which are used to determine the status of the load. The DC diagnostics runs when any channel is directed to leave the Hi-Z state and enter the MUTE or PLAY state. The DC diagnostics are turned on by default but, if a fast startup without diagnostics is required, the diagnostics can be disabled using a dedicated ALSA control. Signed-off-by: Jean-Jacques Hiblot <jjhiblot(a)ti.com> --- changes in v4: - rename the control "Auto Diagnostics Swicth" with a capital S on the Switch sound/soc/codecs/tas6424.c | 13 +++++++++++-- sound/soc/codecs/tas6424.h | 4 ++++ 2 files changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 89fd0c1..14999b9 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -64,6 +64,8 @@ static const struct snd_kcontrol_new tas6424_snd_controls[] = { TAS6424_CH3_VOL_CTRL, 0, 0xff, 0, dac_tlv), SOC_SINGLE_TLV("Speaker Driver CH4 Playback Volume", TAS6424_CH4_VOL_CTRL, 0, 0xff, 0, dac_tlv), + SOC_SINGLE_STROBE("Auto Diagnostics Switch", TAS6424_DC_DIAG_CTRL1, + TAS6424_LDGBYPASS_SHIFT, 1), }; static int tas6424_dac_event(struct snd_soc_dapm_widget *w, @@ -297,6 +299,11 @@ static int tas6424_power_on(struct snd_soc_component *component) struct tas6424_data *tas6424 = snd_soc_component_get_drvdata(component); int ret; u8 chan_states; + int no_auto_diags = 0; + unsigned int reg_val; + + if (!regmap_read(tas6424->regmap, TAS6424_DC_DIAG_CTRL1, &reg_val)) + no_auto_diags = reg_val & TAS6424_LDGBYPASS_MASK; ret = regulator_bulk_enable(ARRAY_SIZE(tas6424->supplies), tas6424->supplies); @@ -327,9 +334,11 @@ static int tas6424_power_on(struct snd_soc_component *component) snd_soc_component_write(component, TAS6424_CH_STATE_CTRL, chan_states); /* any time we come out of HIZ, the output channels automatically run DC - * load diagnostics, wait here until this completes + * load diagnostics if autodiagnotics are enabled. wait here until this + * completes. */ - msleep(230); + if (!no_auto_diags) + msleep(230); return 0; } diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h index 4305883..b5958c4 100644 --- a/sound/soc/codecs/tas6424.h +++ b/sound/soc/codecs/tas6424.h @@ -111,6 +111,10 @@ TAS6424_CH3_STATE_DIAG | \ TAS6424_CH4_STATE_DIAG) +/* TAS6424_DC_DIAG_CTRL1 */ +#define TAS6424_LDGBYPASS_SHIFT 0 +#define TAS6424_LDGBYPASS_MASK BIT(TAS6424_LDGBYPASS_SHIFT) + /* TAS6424_GLOB_FAULT1_REG */ #define TAS6424_FAULT_CLOCK BIT(4) #define TAS6424_FAULT_PVDD_OV BIT(3) -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Clean up mixer element list traverse
by Takashi Iwai 03 May '18

03 May '18

Introduce a new macro for iterating over mixer element list for avoiding the open codes in many places. Also the open-coded container_of() and the forced cast to struct usb_mixer_elem_info are replaced with another simple macro, too. No functional changes but just readability improvement. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 20 +++++++++----------- sound/usb/mixer.h | 6 ++++++ sound/usb/mixer_quirks.c | 2 +- sound/usb/mixer_scarlett.c | 6 ++---- 4 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index bb203b3684fc..265258b0e74c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2598,9 +2598,9 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info = - (struct usb_mixer_elem_info *)list; + mixer_elem_list_to_info(list); /* invalidate cache, so the value is read from the device */ info->cached = 0; snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, @@ -2611,7 +2611,7 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16"}; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " @@ -2637,8 +2637,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { - for (list = mixer->id_elems[unitid]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { snd_iprintf(buffer, " Unit: %i\n", list->id); if (list->kctl) snd_iprintf(buffer, @@ -2668,19 +2667,19 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for_each_mixer_elem(list, mixer, unitid) count++; if (count == 0) return; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info; if (!list->kctl) continue; - info = (struct usb_mixer_elem_info *)list; + info = mixer_elem_list_to_info(list); if (count > 1 && info->control != control) continue; @@ -2946,7 +2945,7 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) static int restore_mixer_value(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); int c, err, idx; if (cval->cmask) { @@ -2982,8 +2981,7 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) if (reset_resume) { /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { - for (list = mixer->id_elems[id]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, id) { if (list->resume) { err = list->resume(list); if (err < 0) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ba27f7ade670..e02653465e29 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -53,6 +53,12 @@ struct usb_mixer_elem_list { usb_mixer_elem_resume_func_t resume; }; +/* iterate over mixer element list of the given unit id */ +#define for_each_mixer_elem(list, mixer, id) \ + for ((list) = (mixer)->id_elems[id]; (list); (list) = (list)->next_id_elem) +#define mixer_elem_list_to_info(list) \ + container_of(list, struct usb_mixer_elem_info, head) + struct usb_mixer_elem_info { struct usb_mixer_elem_list head; unsigned int control; /* CS or ICN (high byte) */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 4377374affd3..1b94387e18b6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1172,7 +1172,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, int unitid = 12; /* SamleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = (struct usb_mixer_elem_info *)mixer->id_elems[unitid]; + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); if (cval) { snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index c33e2378089d..4aeb9488a0c9 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -287,8 +287,7 @@ static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl, static int scarlett_ctl_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); int i; for (i = 0; i < elem->channels; i++) @@ -447,8 +446,7 @@ static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl, static int scarlett_ctl_enum_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); if (elem->cached) snd_usb_set_cur_mix_value(elem, 0, 0, *elem->cache_val); -- 2.16.3

1 0

[alsa-devel] Rationale for BYTCR defaults in the kernel's bytcr_rt5640 machine driver ?
by Hans de Goede 03 May '18

03 May '18

Hi Pierre-Louis, I'm finally wrapping up my rt5640 jack-detect work, as such I'm currently testing speaker + headphones + internal mic + headset-mic + jack-detect functionality on the 10 different x86 devices with a rt5640 codec which I've gathered. 2 of them stand out in that they use the BYTCR SoC, but don't have the ACPI table for detecting if SSP0 AIF1 or AIF2 should be used. Currently the driver defaults to AIF2 in this case. As the somewhat wildcard-ish DMI quirk for boards where the sys_vendor is "Insyde" (which applies to a lot of generic designs) shows: { .callback = byt_rt5640_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), }, .driver_data = (void *)(BYT_RT5640_IN3_MAP | BYT_RT5640_MCLK_EN | BYT_RT5640_SSP0_AIF1), }, AIF1 seems to be a better default both models without the ACPI table which I've here, a HP pavilion X2 and a Toshiba Click Mini L9W-B need a quirk to use AIF1. So I was wondering if there was a specific rationale for the AIF2 default and if it would not be better to change the default to AIF1 (which will unfortunately bring a chance of regressions) ? Regards, Hans

2 3

[alsa-devel] [BUG - REPORT] ALC892 not working
by Xristos Boul 03 May '18

03 May '18

Hi everyone, I'm having a problem with my line-out port not being working properly. Specifically it can't play audio when the 'alsamixer' auto-mute mode is enabled, so sound is coming only from the headphones.However, when auto-mute is disabled the sound is coming from both ports simultaneously. I'm using the Asus Z170-A mobo with ALC892 codec, OS: Arch I ran alsa-info.sh on both cases and i saw that the Pin-Complex 0x14 had 'Pin-ctls: OUT' when auto-mute was disabled whereas when auto-mute was enabled it was 'Pin-ctls: 0x00' (as it currenly is). So i ran HDA-analyzer tool and changed that widget setting to 'OUT' and obviously i produced the same behaviour like when i disabled 'auto-mute' (again sound coming from both ports) and also in all cases PulseAudio lists the 'Line-out' port as 'unplugged'. Also 2 more things: 1st: In BIOS when i changed the driver for the Front panel from 'HD-Audio' to 'AC97' and the ports worked correctly, however no 'auto-mute' mode was available so i had to mute the ports manually when i plugged/unplugged the headphones for e.x I tried uninstalling PulseAudio to see if it was the case but the problem persisted so i think it really is some codec problem since i also have a similar problem in Windows OS. 2nd: When i unplugged the Front-panel connection from the mobo the port played correctly. So is it maybe a mixer problem or something related? Finally i tried every 'model' parameter from HD-Audio-Models.txt for ALC892 but got nothing. In any case i'm attaching the current alsa-info.sh. If you would like i could also send alsa-info.sh for all those cases above. I don't think i'm alone in this issue but i would greatly appreciate the help! Thanks in advance. Link to alsa-info.txt -> http://www.alsa-project.org/db/?f=1020241a74addd837a3625d3f1c6be374996aa71

1 0

[alsa-devel] [PATCH] ALSA: sparc: Use GFP_KERNEL for non-atomic allocation
by Takashi Iwai 03 May '18

03 May '18

dbri driver allocates a resource with GFP_ATOMIC unnecessarily in its probe function. Replace it with the standard GFP_KERNEL for avoiding the bogus allocation failures. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/sparc/dbri.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index abc7bd5055eb..f0e713527e91 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2542,7 +2542,7 @@ static int snd_dbri_create(struct snd_card *card, dbri->irq = irq; dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma), - &dbri->dma_dvma, GFP_ATOMIC); + &dbri->dma_dvma, GFP_KERNEL); if (!dbri->dma) return -ENOMEM; -- 2.16.3

1 0

[alsa-devel] [PATCH v2] ASoC: da7219: read fmw property to get mclk for non-dts systems
by Akshu Agrawal 03 May '18

03 May '18

Non-dts based systems can use ACPI DSDT to pass on the mclk to da7219. This enables da7219 mclk to be linked to system clock. Enable/Disable of the mclk is already handled in the codec so platform drivers don't have to explicitly do handling of mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Fixed kbuild error include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 7 ++++++- 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16..df7ddf4 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + const char *mclk_name; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8..aed68a4 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1624,6 +1624,8 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone dev_warn(dev, "Using default clk name: %s\n", pdata->dai_clks_name); + device_property_read_string(dev, "dlg,mclk-name", &pdata->mclk_name); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -1905,7 +1907,10 @@ static int da7219_probe(struct snd_soc_component *component) da7219_handle_pdata(component); /* Check if MCLK provided */ - da7219->mclk = devm_clk_get(component->dev, "mclk"); + if (da7219->pdata->mclk_name) + da7219->mclk = clk_get(NULL, da7219->pdata->mclk_name); + if (!da7219->mclk) + da7219->mclk = devm_clk_get(component->dev, "mclk"); if (IS_ERR(da7219->mclk)) { if (PTR_ERR(da7219->mclk) != -ENOENT) { ret = PTR_ERR(da7219->mclk); -- 1.9.1

5 9

[alsa-devel] [PATCH alsa-lib] conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio
by Takashi Iwai 03 May '18

03 May '18

USB-audio device on Dell WD15 docking station provides two individual PCM streams, one for headphone and another for line out. A UCM profile gives the proper roles for these. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is the alsa-lib part to correspond to the recent Dell WD15 dock patchset. src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf | 5 +++++ src/conf/ucm/Dell-WD15-Dock/HiFi.conf | 26 +++++++++++++++++++++++++ src/conf/ucm/Dell-WD15-Dock/Makefile.am | 4 ++++ src/conf/ucm/Makefile.am | 1 + 4 files changed, 36 insertions(+) create mode 100644 src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf create mode 100644 src/conf/ucm/Dell-WD15-Dock/HiFi.conf create mode 100644 src/conf/ucm/Dell-WD15-Dock/Makefile.am diff --git a/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf b/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf new file mode 100644 index 000000000000..f4f0a8d97e73 --- /dev/null +++ b/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf @@ -0,0 +1,5 @@ +Comment "USB-audio on Dell WD15 docking station" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} diff --git a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf new file mode 100644 index 000000000000..b07f34400086 --- /dev/null +++ b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf @@ -0,0 +1,26 @@ +SectionDevice."Headphone" { + Comment "Headphone" + + Value { + PlaybackChannels "2" + PlaybackPCM "hw:WD15Dock,0" + } +} + +SectionDevice."LineOut" { + Comment "Line Out" + + Value { + PlaybackChannels "2" + PlaybackPCM "hw:WD15Dock,1" + } +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CaptureChannels "2" + CapturePCM "hw:WD15Dock,0" + } +} diff --git a/src/conf/ucm/Dell-WD15-Dock/Makefile.am b/src/conf/ucm/Dell-WD15-Dock/Makefile.am new file mode 100644 index 000000000000..7ab58730dc26 --- /dev/null +++ b/src/conf/ucm/Dell-WD15-Dock/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/Dell-WD15-Dock +ucm_DATA = Dell-WD15-Dock.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 02df25f01b25..38fc6b252bcb 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -7,6 +7,7 @@ chtrt5645 \ chtrt5645-mono-speaker-analog-mic \ DAISY-I2S \ DB410c \ +Dell-WD15-Dock \ GoogleNyan \ gpd-win-pocket-rt5645 \ HDAudio-Gigabyte-ALC1220DualCodecs \ -- 2.16.3

2 4

[alsa-devel] [PATCH] conf/ucm: Fix build error for UCM Dell WD15 Dock USB-audio
by Xiuli Pan 03 May '18

03 May '18

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> Miss oneline in configure.ac in provious patch conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> --- configure.ac | 1 + 1 file changed, 1 insertion(+) diff --git a/configure.ac b/configure.ac index 3ee989e..693b5d5 100644 --- a/configure.ac +++ b/configure.ac @@ -725,6 +725,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ src/conf/ucm/DAISY-I2S/Makefile \ src/conf/ucm/DB410c/Makefile \ + src/conf/ucm/Dell-WD15-Dock/Makefile \ src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/gpd-win-pocket-rt5645/Makefile \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt565: fix missing assignment to ret_val
by Colin King 03 May '18

03 May '18

From: Colin Ian King <colin.king(a)canonical.com> Currently, the check that ret_val is not -ENOENT is always true and the quirk bit BYT_RY5651_MCLK_EN is never being cleared because ret_val is always zero at this point from a previous assignment earlier on. I believe that ret_val should actually be assigned to the return from devm_clk_get() as this can return -ENOENT (from a deeper call to clk_get_sys) and that was the original intention to check this. Detected by CoverityScan, CID#1460228 ("Logically dead code") Fixes: 02c0a3b3047f ("ASoC: Intel: bytcr_rt5651: add MCLK, quirks and cleanups") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 3c7d93520c52..469e2695d121 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -856,9 +856,10 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { + ret_val = PTR_ERR(priv->mclk); dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(priv->mclk)); + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret_val); /* * Fall back to bit clock usage for -ENOENT (clock not * available likely due to missing dependencies), bail -- 2.17.0

3 2

[alsa-devel] [PATCH v2 1/4] ASoC: fsl_ssi: Switch to SPDX identifier
by Fabio Estevam 03 May '18

03 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Retain the copyright statements in fsl_ssi.h (Timur) sound/soc/fsl/fsl_ssi.c | 57 +++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 6 ++--- sound/soc/fsl/fsl_ssi_dbg.c | 18 ++++++-------- 3 files changed, 35 insertions(+), 46 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 1544166..0a64822 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1,34 +1,29 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver - * - * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * - * Some notes why imx-pcm-fiq is used instead of DMA on some boards: - * - * The i.MX SSI core has some nasty limitations in AC97 mode. While most - * sane processor vendors have a FIFO per AC97 slot, the i.MX has only - * one FIFO which combines all valid receive slots. We cannot even select - * which slots we want to receive. The WM9712 with which this driver - * was developed with always sends GPIO status data in slot 12 which - * we receive in our (PCM-) data stream. The only chance we have is to - * manually skip this data in the FIQ handler. With sampling rates different - * from 48000Hz not every frame has valid receive data, so the ratio - * between pcm data and GPIO status data changes. Our FIQ handler is not - * able to handle this, hence this driver only works with 48000Hz sampling - * rate. - * Reading and writing AC97 registers is another challenge. The core - * provides us status bits when the read register is updated with *another* - * value. When we read the same register two times (and the register still - * contains the same value) these status bits are not set. We work - * around this by not polling these bits but only wait a fixed delay. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver +// +// Author: Timur Tabi <timur(a)freescale.com> +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. +// +// Some notes why imx-pcm-fiq is used instead of DMA on some boards: +// +// The i.MX SSI core has some nasty limitations in AC97 mode. While most +// sane processor vendors have a FIFO per AC97 slot, the i.MX has only +// one FIFO which combines all valid receive slots. We cannot even select +// which slots we want to receive. The WM9712 with which this driver +// was developed with always sends GPIO status data in slot 12 which +// we receive in our (PCM-) data stream. The only chance we have is to +// manually skip this data in the FIQ handler. With sampling rates different +// from 48000Hz not every frame has valid receive data, so the ratio +// between pcm data and GPIO status data changes. Our FIQ handler is not +// able to handle this, hence this driver only works with 48000Hz sampling +// rate. +// Reading and writing AC97 registers is another challenge. The core +// provides us status bits when the read register is updated with *another* +// value. When we read the same register two times (and the register still +// contains the same value) these status bits are not set. We work +// around this by not polling these bits but only wait a fixed delay. #include <linux/init.h> #include <linux/io.h> diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 18f8dd5..0bdda60 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -1,12 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 and i.MX SoC * * Author: Timur Tabi <timur(a)freescale.com> * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. + * Copyright 2007-2008 Freescale Semiconductor, Inc. */ #ifndef _MPC8610_I2S_H diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 7aac63e..1bacfa2 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -1,14 +1,10 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions - * - * Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix - * - * Splitted from fsl_ssi.c - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions +// +// Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix +// +// Split from fsl_ssi.c #include <linux/debugfs.h> #include <linux/device.h> -- 2.7.4

2 4

[alsa-devel] [PATCH 1/4] ASoC: fsl_ssi: Switch to SPDX identifier
by Fabio Estevam 03 May '18

03 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 57 +++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 6 +---- sound/soc/fsl/fsl_ssi_dbg.c | 18 ++++++-------- 3 files changed, 34 insertions(+), 47 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 1544166..0a64822 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1,34 +1,29 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver - * - * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * - * Some notes why imx-pcm-fiq is used instead of DMA on some boards: - * - * The i.MX SSI core has some nasty limitations in AC97 mode. While most - * sane processor vendors have a FIFO per AC97 slot, the i.MX has only - * one FIFO which combines all valid receive slots. We cannot even select - * which slots we want to receive. The WM9712 with which this driver - * was developed with always sends GPIO status data in slot 12 which - * we receive in our (PCM-) data stream. The only chance we have is to - * manually skip this data in the FIQ handler. With sampling rates different - * from 48000Hz not every frame has valid receive data, so the ratio - * between pcm data and GPIO status data changes. Our FIQ handler is not - * able to handle this, hence this driver only works with 48000Hz sampling - * rate. - * Reading and writing AC97 registers is another challenge. The core - * provides us status bits when the read register is updated with *another* - * value. When we read the same register two times (and the register still - * contains the same value) these status bits are not set. We work - * around this by not polling these bits but only wait a fixed delay. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver +// +// Author: Timur Tabi <timur(a)freescale.com> +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. +// +// Some notes why imx-pcm-fiq is used instead of DMA on some boards: +// +// The i.MX SSI core has some nasty limitations in AC97 mode. While most +// sane processor vendors have a FIFO per AC97 slot, the i.MX has only +// one FIFO which combines all valid receive slots. We cannot even select +// which slots we want to receive. The WM9712 with which this driver +// was developed with always sends GPIO status data in slot 12 which +// we receive in our (PCM-) data stream. The only chance we have is to +// manually skip this data in the FIQ handler. With sampling rates different +// from 48000Hz not every frame has valid receive data, so the ratio +// between pcm data and GPIO status data changes. Our FIQ handler is not +// able to handle this, hence this driver only works with 48000Hz sampling +// rate. +// Reading and writing AC97 registers is another challenge. The core +// provides us status bits when the read register is updated with *another* +// value. When we read the same register two times (and the register still +// contains the same value) these status bits are not set. We work +// around this by not polling these bits but only wait a fixed delay. #include <linux/init.h> #include <linux/io.h> diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 18f8dd5..146981f 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -1,12 +1,8 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 and i.MX SoC * * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. */ #ifndef _MPC8610_I2S_H diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 7aac63e..ddf2f64 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -1,14 +1,10 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions - * - * Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix - * - * Splitted from fsl_ssi.c - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions +// +// Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix +// +// Split from fsl_ssi.c #include <linux/debugfs.h> #include <linux/device.h> -- 2.7.4

3 9

[alsa-devel] [PATCH 0/6] usb-audio: quirks for Dell WD15 dock and some extensions
by Takashi Iwai 02 May '18

02 May '18

Hi, here is a patch set for adding quirks and some new extensions for Dell WD15 dock. The first patch is the resend of the previous post, and the rest are new. Basically it adds a new "Keep Interface" control for the card to skip resetting the USB interface per stream manipulation. It helps for reducing the long pause before each start of stream on Dell WD15 dock. OTOH, this was confirmed to be harmless on some dock early firmware version, so we can't set it statically in the driver. So, instead, the feature is exposed as a control element so that user can flip dynamically. This feature may be useful for other USB audio devices as well. Another extension is to allow the quirk entry to override the card longname field. This is needed to give a proper UCM profile. WD15 dock provides two streams, one for the headphone and another for the line out. It's a good example for using a UCM profile. Takashi === Takashi Iwai (6): ALSA: usb-audio: Initialize Dell Dock playback volumes ALSA: usb-audio: Avoid superfluous usb_set_interface() calls ALSA: usb-audio: Add keep_iface flag ALSA: usb-audio: Add "Keep Interface" control ALSA: usb-audio: Allow to override the longname string ALSA: usb-audio: Give proper vendor/product name for Dell WD15 Dock sound/usb/card.c | 146 ++++++++++++++++++++++++++++------------------- sound/usb/mixer.c | 47 +++++++++++++++ sound/usb/mixer_quirks.c | 34 +++++++++++ sound/usb/mixer_quirks.h | 4 ++ sound/usb/pcm.c | 28 ++++----- sound/usb/quirks-table.h | 10 ++++ sound/usb/usbaudio.h | 4 ++ 7 files changed, 200 insertions(+), 73 deletions(-) -- 2.16.3

1 6

[alsa-devel] [PATCH v3 00/13] ALSA: dice: use cache of stream formats to generate PCM rules/constraints
by Takashi Sakamoto 02 May '18

02 May '18

Hi, This patchset updates my previous one: [alsa-devel] [PATCH v2 00/11] ALSA: dice: use cache of stream formats to generate PCM rules/constraints http://mailman.alsa-project.org/pipermail/alsa-devel/2018-April/135330.html Once, ALSA dice driver cached stream formats to generate PCM constraints/rules. The cache is created by changing current sampling transmission frequency step by step directly at unit probing process. This feature was dropped because it changes state of clock on target device thus it's against users' expectation[1]. However this is not necessarily convenient because reconfiguration of sampling rate is not available. TC Applied Technologies (TCAT) has extended protocol for DICE interface, a.k.a EAP (Extended Application Protocol). In this extension, drivers can retrieve information of stream formats without changing state of target unit. Fortunately, this protocol is supported widely by many models. This patchset regain the cache to generate PCM constraints/rules by using this protocol. Userspace applications can reconfigure sampling rate via ALSA PCM interface. For units which doesn't support the extension, available set of sampling rate is still restricted at current sampling transmission frequency. This is also inconvenient and this patchset introduces a solution. Structure for unit entry (struct ieee1394_device_id) has a member for unit-specific data. This patchset uses it for callback function to handle hard-coded stream format. In this time, I adopt this way for some known models produced by TC Electronic and Alesis. If users find their models still have restrictions in a point of available sampling transmission frequency, it's preferable to work for this way. Changes from v2: - use robust representation to calcurate mode of sampling transmission frequency. [1] [alsa-devel] [PATCH 0/6 v2] ALSA: dice: constrain PCM substreams to current sampling transfer frequency http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104065.h… Takashi Sakamoto (13): ALSA: dice: add cache of stream formats ALSA: dice: add 'firewire' directory for proc nodes ALSA: dice: add proc node for stream formation ALSA: dice: cache stream formats at current mode of sampling transmission frequency ALSA: dice: add parameters of stream formats for models produced by TC Electronic ALSA: dice: add parameters of stream formats for models produced by Alesis ALSA: dice: use extended protocol to detect available stream formats ALSA: dice: use cache of stream format to check running stream ALSA: dice: add a helper function to restart all of available streams ALSA: dice: enable to change current sampling transmission frequency ALSA: dice: use stream formats to add MIDI substreams ALSA: dice: use cache for PCM constraints and rules ALSA: dice: remove local frag of force_two_pcms sound/firewire/dice/Makefile | 3 +- sound/firewire/dice/dice-alesis.c | 52 ++++++ sound/firewire/dice/dice-extension.c | 172 +++++++++++++++++++ sound/firewire/dice/dice-midi.c | 23 +-- sound/firewire/dice/dice-pcm.c | 233 ++++++++++++++++++-------- sound/firewire/dice/dice-proc.c | 70 +++++++- sound/firewire/dice/dice-stream.c | 283 ++++++++++++++++++++++++-------- sound/firewire/dice/dice-tcelectronic.c | 100 +++++++++++ sound/firewire/dice/dice.c | 139 ++++++++++------ sound/firewire/dice/dice.h | 24 ++- 10 files changed, 891 insertions(+), 208 deletions(-) create mode 100644 sound/firewire/dice/dice-alesis.c create mode 100644 sound/firewire/dice/dice-extension.c create mode 100644 sound/firewire/dice/dice-tcelectronic.c -- 2.14.1

2 14

[alsa-devel] [PATCH] ALSA: sc6000: fix spelling mistake: "iomaped" -> "iomapped"
by Colin King 02 May '18

02 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in KERN_ERR error messages Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/sc6000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index c09d9b914efe..a985e9183be9 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -592,7 +592,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev) *vport = devm_ioport_map(devptr, port[dev], 0x10); if (*vport == NULL) { snd_printk(KERN_ERR PFX - "I/O port cannot be iomaped.\n"); + "I/O port cannot be iomapped.\n"); err = -EBUSY; goto err_unmap1; } @@ -607,7 +607,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev) vmss_port = devm_ioport_map(devptr, mss_port[dev], 4); if (!vmss_port) { snd_printk(KERN_ERR PFX - "MSS port I/O cannot be iomaped.\n"); + "MSS port I/O cannot be iomapped.\n"); err = -EBUSY; goto err_unmap2; } -- 2.17.0

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132: fix build failure when a local macro is defined
by Takashi Sakamoto 02 May '18

02 May '18

DECLARE_TLV_DB_SCALE (alias of SNDRV_CTL_TLVD_DECLARE_DB_SCALE) is used but tlv.h is not included. This causes build failure when local macro is defined by comment-out. This commit fixes the bug. At the same time, the alias macro is replaced with a destination macro added at a commit 46e860f76804 ("ALSA: rename TLV-related macros so that they're friendly to user applications") Reported-by: Connor McAdams <conmanx360(a)gmail.com> Fixes: 44f0c9782cc6 ('ALSA: hda/ca0132: Add tuning controls') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/pci/hda/patch_ca0132.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea8651993..84261ef02c93 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -39,6 +39,10 @@ /* Enable this to see controls for tuning purpose. */ /*#define ENABLE_TUNING_CONTROLS*/ +#ifdef ENABLE_TUNING_CONTROLS +#include <sound/tlv.h> +#endif + #define FLOAT_ZERO 0x00000000 #define FLOAT_ONE 0x3f800000 #define FLOAT_TWO 0x40000000 @@ -3068,8 +3072,8 @@ static int equalizer_ctl_put(struct snd_kcontrol *kcontrol, return 1; } -static const DECLARE_TLV_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); -static const DECLARE_TLV_DB_SCALE(eq_db_scale, -2400, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(eq_db_scale, -2400, 100, 0); static int add_tuning_control(struct hda_codec *codec, hda_nid_t pnid, hda_nid_t nid, -- 2.14.1

2 1

[alsa-devel] linux-sound: timestamps for audio frames
by Muni Sekhar 02 May '18

02 May '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, >From sound driver, I am wondering if we could get the timestamps when It receives for example the first and last audio frames. Is there any way we can extract or expose that information to user mode? -- Thanks, Sekhar

1 0

[alsa-devel] [PATCH v3 0/3] ASoC: Add Multi CPU DAI support
by Vinod Koul 02 May '18

02 May '18

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html Changes in v3: - Rebased on asoc for-next Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add Multi CPU DAI support for PCM ops ASoC: Add Multi CPU DAI support in DAPM include/sound/soc.h | 6 + sound/soc/soc-core.c | 283 ++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 483 +++++++++++++++++++++++++++++++++------------------ 4 files changed, 565 insertions(+), 278 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 0/5] ALSA: xen-front: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 02 May '18

02 May '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Please note: this patch series depends on [3]. This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Corresponding backend, implemented as a user-space application, can be found at [2]. Thank you, Oleksandr Changes since v1: ***************** 1. Moved driver from sound/drivers to sound/xen 2. Coding style changes to better meet Linux Kernel 3. Added explicit back and front synchronization In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events 4. Added explicit back and front parameter negotiation In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameters given: request passes desired parameter's intervals/masks and the response to this request returns allowed min/max intervals/masks to be used. [1] https://xenproject.org/ [2] https://github.com/xen-troops/snd_be [3] https://lkml.org/lkml/2018/4/12/522 Oleksandr Andrushchenko (5): ALSA: xen-front: Introduce Xen para-virtualized sound frontend driver ALSA: xen-front: Read sound driver configuration from Xen store ALSA: xen-front: Implement Xen event channel handling ALSA: xen-front: Implement handling of shared buffers ALSA: xen-front: Implement ALSA virtual sound driver sound/Kconfig | 2 + sound/Makefile | 2 +- sound/xen/Kconfig | 10 + sound/xen/Makefile | 9 + sound/xen/xen_snd_front.c | 410 +++++++++++++++++++ sound/xen/xen_snd_front.h | 57 +++ sound/xen/xen_snd_front_alsa.c | 830 ++++++++++++++++++++++++++++++++++++++ sound/xen/xen_snd_front_alsa.h | 23 ++ sound/xen/xen_snd_front_cfg.c | 517 ++++++++++++++++++++++++ sound/xen/xen_snd_front_cfg.h | 46 +++ sound/xen/xen_snd_front_evtchnl.c | 478 ++++++++++++++++++++++ sound/xen/xen_snd_front_evtchnl.h | 92 +++++ sound/xen/xen_snd_front_shbuf.c | 193 +++++++++ sound/xen/xen_snd_front_shbuf.h | 36 ++ 14 files changed, 2704 insertions(+), 1 deletion(-) create mode 100644 sound/xen/Kconfig create mode 100644 sound/xen/Makefile create mode 100644 sound/xen/xen_snd_front.c create mode 100644 sound/xen/xen_snd_front.h create mode 100644 sound/xen/xen_snd_front_alsa.c create mode 100644 sound/xen/xen_snd_front_alsa.h create mode 100644 sound/xen/xen_snd_front_cfg.c create mode 100644 sound/xen/xen_snd_front_cfg.h create mode 100644 sound/xen/xen_snd_front_evtchnl.c create mode 100644 sound/xen/xen_snd_front_evtchnl.h create mode 100644 sound/xen/xen_snd_front_shbuf.c create mode 100644 sound/xen/xen_snd_front_shbuf.h -- 2.16.2

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[alsa-devel] [PATCH v4 1/3] dell-led: Change dell-led.h to dell-common.h
by Kai-Heng Feng 02 May '18

02 May '18

This header will be used for more than just led. Change it to a more generic name. Cc: Mario Limonciello <mario.limonciello(a)dell.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v4: Change the commit message to clarify there's no more runtime pm warning. Also skip the check for thunderbolt attached devices. v3: Simplify dell_switchable_gfx_is_enabled() by returning bool instead of error code. Use DMI_DEV_TYPE_OEM_STRING to match Dell System. v2: Mario suggested to squash the HDA part into the same series. drivers/platform/x86/dell-laptop.c | 2 +- include/linux/{dell-led.h => dell-common.h} | 4 ++-- sound/pci/hda/dell_wmi_helper.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) rename include/linux/{dell-led.h => dell-common.h} (61%) diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index c52c6723374b..8ba820e6c3d0 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -29,7 +29,7 @@ #include <linux/mm.h> #include <linux/i8042.h> #include <linux/debugfs.h> -#include <linux/dell-led.h> +#include <linux/dell-common.h> #include <linux/seq_file.h> #include <acpi/video.h> #include "dell-rbtn.h" diff --git a/include/linux/dell-led.h b/include/linux/dell-common.h similarity index 61% rename from include/linux/dell-led.h rename to include/linux/dell-common.h index 92521471517f..37e4b614dd74 100644 --- a/include/linux/dell-led.h +++ b/include/linux/dell-common.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ -#ifndef __DELL_LED_H__ -#define __DELL_LED_H__ +#ifndef __DELL_COMMON_H__ +#define __DELL_COMMON_H__ int dell_micmute_led_set(int on); diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..56050cc3c0ee 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -4,7 +4,7 @@ */ #if IS_ENABLED(CONFIG_DELL_LAPTOP) -#include <linux/dell-led.h> +#include <linux/dell-common.h> enum { MICMUTE_LED_ON, -- 2.17.0

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[alsa-devel] [PATCH] ALSA: usb-audio: Initialize Dell Dock playback volumes
by Takashi Iwai 02 May '18

02 May '18

In the early commit adcdd0d5a1cb ("ALSA: usb-audio: Skip volume controls triggers hangup on Dell USB Dock"), we add the mixer quirks for Dell dock to skip two mixer FU's for playback. This supposed that the device has always the proper initial volume, but it doesn't seem always correct. This patch adds the explicit initialization of the volumes to the fixed 0dB at the device probe time. Also, such a fixup is needed after the resume, so a new function is hooked to the resume callback as well. Bugzilla: http://bugzilla.suse.com/show_bug.cgi?id=1089467 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 2 ++ sound/usb/mixer_quirks.c | 34 ++++++++++++++++++++++++++++++++++ sound/usb/mixer_quirks.h | 4 ++++ 3 files changed, 40 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 344d7b069d59..76fabc4b72b5 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2948,6 +2948,8 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) } } + snd_usb_mixer_resume_quirk(mixer); + return snd_usb_mixer_activate(mixer); } #endif diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 56537a156580..4377374affd3 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1799,6 +1799,26 @@ static int snd_soundblaster_e1_switch_create(struct usb_mixer_interface *mixer) NULL); } +static void dell_dock_init_vol(struct snd_usb_audio *chip, int ch, int id) +{ + u16 buf = 0; + + snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + ch, snd_usb_ctrl_intf(chip) | (id << 8), + &buf, 2); +} + +static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) +{ + /* fix to 0dB playback volumes */ + dell_dock_init_vol(mixer->chip, 1, 16); + dell_dock_init_vol(mixer->chip, 2, 16); + dell_dock_init_vol(mixer->chip, 1, 19); + dell_dock_init_vol(mixer->chip, 2, 19); + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1884,11 +1904,25 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; + case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ + err = dell_dock_mixer_init(mixer); + break; } return err; } +#ifdef CONFIG_PM +void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer) +{ + switch (mixer->chip->usb_id) { + case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ + dell_dock_mixer_init(mixer); + break; + } +} +#endif + void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid) { diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index b5abd328a361..52be26db558f 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -14,5 +14,9 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *cval, int unitid, struct snd_kcontrol *kctl); +#ifdef CONFIG_PM +void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer); +#endif + #endif /* SND_USB_MIXER_QUIRKS_H */ -- 2.16.3

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[alsa-devel] [PATCH] ALSA: hda - Enforce CONFIG_SND_DYNAMIC_MINORS for HDMI/DP codec
by Takashi Iwai 02 May '18

02 May '18

The DP-MST support requires more PCM streams than usual, hence CONFIG_SND_DYNAMIC_MINORS is almost mandatory. Currently the driver just warns and continues even if streams are missing, but it doesn't seem to enough convince users to switch to the modern setup. This patch adds the enforced selection of CONFIG_SND_DYNAMIC_MINORS from CONFIG_SND_HDA_CODEC_HDMI for covering that. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index f7a492c382d9..4235907b7858 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -127,11 +127,15 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_HDMI tristate "Build HDMI/DisplayPort HD-audio codec support" + select SND_DYNAMIC_MINORS help Say Y or M here to include HDMI and DisplayPort HD-audio codec support in snd-hda-intel driver. This includes all AMD/ATI, Intel and Nvidia HDMI/DisplayPort codecs. + Note that this option mandatorily enables CONFIG_SND_DYNAMIC_MINORS + to assure the multiple streams for DP-MST support. + comment "Set to Y if you want auto-loading the codec driver" depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m -- 2.16.3

1 0

[alsa-devel] [PATCH 0/5] conf/ucm: chtrt5645: Various updates
by Hans de Goede 02 May '18

02 May '18

Hi All, Here is a series of chtrt5645 UCM updates, the first 2 patches bring in some fixes for the generic config from: https://github.com/plbossart/UCM The 3th patch adds some analog mic input fixes, together these 3 patches bring the generic chtrt5645 config in sync with the newer chtrt5645-mono-speaker-analog-mic config, but then for stereo speakers. The 4th and the 5th patch add long-name using board/model specific configs for the Asus T100HA and Lenovo Miix 320, these use a digital mic connected to the DMIC1 resp. DMIC2. input. The generic chtrt5645 config has been tested on a Lenovo Miix 310 and a Kazam Vision tablet, both of which feature stereo speakers and an analog microphone. The Asus T100HA and Lenovo Miix 320 configs have been tested on the resp. models they apply to. Regards, Hans

2 6

[alsa-devel] WARNING in __snd_pcm_lib_xfer
by syzbot 02 May '18

02 May '18

Hello, syzbot found the following crash on: HEAD commit: fff75eb2a08c Merge tag 'errseq-v4.17' of git://git.kernel.o... git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?id=6530078334779392 kernel config: https://syzkaller.appspot.com/x/.config?id=6493557782959164711 dashboard link: https://syzkaller.appspot.com/bug?extid=1dac3a4f6bc9c1c675d4 compiler: gcc (GCC) 8.0.1 20180413 (experimental) userspace arch: i386 syzkaller repro:https://syzkaller.appspot.com/x/repro.syz?id=5691583680741376 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+1dac3a4f6bc9c1c675d4(a)syzkaller.appspotmail.com random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) IPVS: ftp: loaded support on port[0] = 21 WARNING: CPU: 1 PID: 4520 at sound/core/pcm_lib.c:2057 pcm_sanity_check sound/core/pcm_lib.c:2057 [inline] WARNING: CPU: 1 PID: 4520 at sound/core/pcm_lib.c:2057 __snd_pcm_lib_xfer+0x7c3/0x1d10 sound/core/pcm_lib.c:2122 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 4520 Comm: syz-executor0 Not tainted 4.17.0-rc3+ #52 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1b9/0x294 lib/dump_stack.c:113 panic+0x22f/0x4de kernel/panic.c:184 __warn.cold.8+0x163/0x1b3 kernel/panic.c:536 report_bug+0x252/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1de/0x490 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:315 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:992 RIP: 0010:pcm_sanity_check sound/core/pcm_lib.c:2057 [inline] RIP: 0010:__snd_pcm_lib_xfer+0x7c3/0x1d10 sound/core/pcm_lib.c:2122 RSP: 0018:ffff8801d97b7898 EFLAGS: 00010293 RAX: ffff8801ad00e0c0 RBX: ffff8801d965e100 RCX: 0000000000000000 RDX: 0000000000000000 RSI: ffffffff85c3d333 RDI: ffff8801d965e418 RBP: ffff8801d97b7b38 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000003 R11: 0000000000000000 R12: ffffffff884945c0 R13: 0000000000000000 R14: ffff8801d965e100 R15: ffff8801cdd2c940 snd_pcm_lib_readv include/sound/pcm.h:1113 [inline] snd_pcm_ioctl_xfern_compat+0x266/0x330 sound/core/pcm_compat.c:448 snd_pcm_ioctl_compat+0xc96/0x1c90 sound/core/pcm_compat.c:716 __do_compat_sys_ioctl fs/compat_ioctl.c:1461 [inline] __se_compat_sys_ioctl fs/compat_ioctl.c:1407 [inline] __ia32_compat_sys_ioctl+0x221/0x640 fs/compat_ioctl.c:1407 do_syscall_32_irqs_on arch/x86/entry/common.c:323 [inline] do_fast_syscall_32+0x345/0xf9b arch/x86/entry/common.c:394 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f51cb9 RSP: 002b:00000000ffafd6ac EFLAGS: 00000282 ORIG_RAX: 0000000000000036 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 00000000800c4153 RDX: 0000000020000040 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000292 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] [alsa-lib][PATCH 1/2] ucm: adding the folder of card_long_name when finding verb conf file
by Hui Wang 02 May '18

02 May '18

The board configuration file and verb conf file are allowed to be in the folder named of card_long_name, so when finding the verb conf file, we need to check if it is in the folder of card_long_name or card_name. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- src/ucm/parser.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 2d76152..219edb9 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -1056,6 +1056,7 @@ static int parse_verb_file(snd_use_case_mgr_t *uc_mgr, char filename[MAX_FILE]; char *env = getenv(ALSA_CONFIG_UCM_VAR); int err; + char *folder_name; /* allocate verb */ verb = calloc(1, sizeof(struct use_case_verb)); @@ -1082,12 +1083,17 @@ static int parse_verb_file(snd_use_case_mgr_t *uc_mgr, } /* open Verb file for reading */ + if (!strncmp(uc_mgr->conf_file_name, uc_mgr->card_long_name, MAX_CARD_LONG_NAME)) + folder_name = uc_mgr->card_long_name; + else + folder_name = uc_mgr->card_name; + if (env) snprintf(filename, sizeof(filename), "%s/%s/%s", - env, uc_mgr->card_name, file); + env, folder_name, file); else snprintf(filename, sizeof(filename), "%s/ucm/%s/%s", - snd_config_topdir(), uc_mgr->card_name, file); + snd_config_topdir(), folder_name, file); filename[sizeof(filename)-1] = '\0'; err = uc_mgr_config_load(filename, &cfg); -- 2.7.4

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