May 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: gus: remove unused local flag
by Takashi Sakamoto 23 May '17

23 May '17

ALSA driver series for devices of Gravis Ultra Sound includes local variable 'snd_gf1_pcm_use_dma'. Although this is a flag to change behaviours of local implementations for 'struct snd_pcm_ops.copy' and 'struct snd_pcm_ops.silence', it's invariable during module lifetime. This commit removes this local variable and the relevant operations. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/isa/gus/gus_pcm.c | 49 ++++++++++++++++++++++--------------------------- 1 file changed, 22 insertions(+), 27 deletions(-) diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index 33c1891f469a..0cc3f272edf1 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -61,8 +61,6 @@ struct gus_pcm_private { int final_volume; }; -static int snd_gf1_pcm_use_dma = 1; - static void snd_gf1_pcm_block_change_ack(struct snd_gus_card * gus, void *private_data) { struct gus_pcm_private *pcmp = private_data; @@ -363,7 +361,9 @@ static int snd_gf1_pcm_playback_copy(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct gus_pcm_private *pcmp = runtime->private_data; + struct snd_gus_card *gus = pcmp->gus; unsigned int bpos, len; + int w16, invert; bpos = samples_to_bytes(runtime, pos) + (voice * (pcmp->dma_size / 2)); len = samples_to_bytes(runtime, count); @@ -373,18 +373,14 @@ static int snd_gf1_pcm_playback_copy(struct snd_pcm_substream *substream, return -EIO; if (copy_from_user(runtime->dma_area + bpos, src, len)) return -EFAULT; - if (snd_gf1_pcm_use_dma && len > 32) { - return snd_gf1_pcm_block_change(substream, bpos, pcmp->memory + bpos, len); - } else { - struct snd_gus_card *gus = pcmp->gus; - int err, w16, invert; - - w16 = (snd_pcm_format_width(runtime->format) == 16); - invert = snd_pcm_format_unsigned(runtime->format); - if ((err = snd_gf1_pcm_poke_block(gus, runtime->dma_area + bpos, pcmp->memory + bpos, len, w16, invert)) < 0) - return err; - } - return 0; + if (len > 32) + return snd_gf1_pcm_block_change(substream, bpos, + pcmp->memory + bpos, len); + + w16 = (snd_pcm_format_width(runtime->format) == 16); + invert = snd_pcm_format_unsigned(runtime->format); + return snd_gf1_pcm_poke_block(gus, runtime->dma_area + bpos, + pcmp->memory + bpos, len, w16, invert); } static int snd_gf1_pcm_playback_silence(struct snd_pcm_substream *substream, @@ -394,7 +390,9 @@ static int snd_gf1_pcm_playback_silence(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct gus_pcm_private *pcmp = runtime->private_data; + struct snd_gus_card *gus = pcmp->gus; unsigned int bpos, len; + int w16, invert; bpos = samples_to_bytes(runtime, pos) + (voice * (pcmp->dma_size / 2)); len = samples_to_bytes(runtime, count); @@ -402,19 +400,16 @@ static int snd_gf1_pcm_playback_silence(struct snd_pcm_substream *substream, return -EIO; if (snd_BUG_ON(bpos + len > pcmp->dma_size)) return -EIO; - snd_pcm_format_set_silence(runtime->format, runtime->dma_area + bpos, count); - if (snd_gf1_pcm_use_dma && len > 32) { - return snd_gf1_pcm_block_change(substream, bpos, pcmp->memory + bpos, len); - } else { - struct snd_gus_card *gus = pcmp->gus; - int err, w16, invert; - - w16 = (snd_pcm_format_width(runtime->format) == 16); - invert = snd_pcm_format_unsigned(runtime->format); - if ((err = snd_gf1_pcm_poke_block(gus, runtime->dma_area + bpos, pcmp->memory + bpos, len, w16, invert)) < 0) - return err; - } - return 0; + snd_pcm_format_set_silence(runtime->format, runtime->dma_area + bpos, + count); + if (len > 32) + return snd_gf1_pcm_block_change(substream, bpos, + pcmp->memory + bpos, len); + + w16 = (snd_pcm_format_width(runtime->format) == 16); + invert = snd_pcm_format_unsigned(runtime->format); + return snd_gf1_pcm_poke_block(gus, runtime->dma_area + bpos, + pcmp->memory + bpos, len, w16, invert); } static int snd_gf1_pcm_playback_hw_params(struct snd_pcm_substream *substream, -- 2.11.0

2 1

[alsa-devel] [RESEND PATCH] ASoC: da7218: Fix incorrect usage of bitwise '&' operator for SRM check
by Adam Thomson 23 May '17

23 May '17

In the SRM lock check section of code the '&' bitwise operator is used as part of checking lock status. Functionally the code works as intended, but the conditional statement is a boolean comparison so should really use '&&' logical operator instead. This commit rectifies this discrepancy. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/soc/codecs/da7218.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index d256ebf..6e1940e 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1457,7 +1457,7 @@ static int da7218_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(DA7218_SRM_CHECK_DELAY); } - } while ((i < DA7218_SRM_CHECK_TRIES) & (!success)); + } while ((i < DA7218_SRM_CHECK_TRIES) && (!success)); if (!success) dev_warn(codec->dev, "SRM failed to lock\n"); -- 1.9.1

1 0

[alsa-devel] [PATCH 0/4] ALSA: Extend PCM buffer-copy helpers
by Takashi Iwai 23 May '17

23 May '17

Hop, step... before jump. This is a patch series based on the previous one ("[PATCH 00/16] ALSA: Convert to new copy_silence PCM ops"), as the second step for killing set_fs(). It extends the PCM buffer-copy helper functions to allow in-kernel buffer copy via the new PCM ops, as well as some code refactoring. Once when these are implemented, the set_fs() can be eliminated by simply replacing with these new helpers. The latest patches are found in topic/kill-set_fs branch in sound git tree. Takashi === Takashi Iwai (4): ALSA: pcm: Shuffle code ALSA: pcm: Call directly the common read/write helpers ALSA: pcm: Provide read/write helpers for in-kernel buffer transfer ALSA: pcm: Split PCM transfer codes include/sound/pcm.h | 81 ++++++++-- sound/core/pcm_lib.c | 410 ++++++++++++++++++++++++++------------------------- 2 files changed, 282 insertions(+), 209 deletions(-) -- 2.13.0

1 4

[alsa-devel] [PATCH] ALSA: hda - apply STAC_9200_DELL_M22 quirk for Dell Latitude D430
by Alexander Tsoy 23 May '17

23 May '17

This model is actually called 92XXM2-8 in Windows driver. But since pin configs for M22 and M28 are identical, just reuse M22 quirk. Fixes external microphone (tested) and probably docking station ports (not tested). Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index faa3d38bac0b..6cefdf6c0b75 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1559,6 +1559,8 @@ static const struct snd_pci_quirk stac9200_fixup_tbl[] = { "Dell Inspiron 1501", STAC_9200_DELL_M26), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f6, "unknown Dell", STAC_9200_DELL_M26), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0201, + "Dell Latitude D430", STAC_9200_DELL_M22), /* Panasonic */ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_9200_PANASONIC), /* Gateway machines needs EAPD to be set on resume */ -- 2.13.0

2 1

[alsa-devel] [PATCH] ALSA: pcm: Remove set_fs() in PCM core code
by Takashi Iwai 23 May '17

23 May '17

PCM core code has a few usages of set_fs(), mostly for two codepaths: - The DELAY ioctl call from pcm_compat.c - The ioctl wrapper in kernel context for PCM OSS and other This patch removes the set_fs() usage in these places by a slight code refactoring. For the former point, snd_pcm_delay() is changed to return the value directly instead of putting the value to the given address. Each caller stores the result in an appropriate manner. For fixing the latter, snd_pcm_lib_kernel_ioctl() is changed to call the functions directly as well. For achieving it, now the function accepts only the limited set of ioctls that have been used, so far. The primary user of this function is the PCM OSS layer, and the only other user is USB UAC1 gadget driver. Both drivers don't need the full set of ioctls. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_compat.c | 12 ++---- sound/core/pcm_native.c | 102 ++++++++++++++++++++++++++++++------------------ 2 files changed, 67 insertions(+), 47 deletions(-) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 1f64ab0c2a95..8a0f8d51e95d 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -27,17 +27,13 @@ static int snd_pcm_ioctl_delay_compat(struct snd_pcm_substream *substream, s32 __user *src) { snd_pcm_sframes_t delay; - mm_segment_t fs; - int err; - fs = snd_enter_user(); - err = snd_pcm_delay(substream, &delay); - snd_leave_user(fs); - if (err < 0) - return err; + delay = snd_pcm_delay(substream); + if (delay < 0) + return delay; if (put_user(delay, src)) return -EFAULT; - return err; + return 0; } static int snd_pcm_ioctl_rewind_compat(struct snd_pcm_substream *substream, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f3a3580eb44c..c6de8976fbd6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -181,20 +181,6 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore); -static inline mm_segment_t snd_enter_user(void) -{ - mm_segment_t fs = get_fs(); - set_fs(get_ds()); - return fs; -} - -static inline void snd_leave_user(mm_segment_t fs) -{ - set_fs(fs); -} - - - int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) { struct snd_pcm_runtime *runtime; @@ -1081,6 +1067,7 @@ static const struct action_ops snd_pcm_action_start = { * @substream: the PCM substream instance * * Return: Zero if successful, or a negative error code. + * The stream lock must be acquired before calling this function. */ int snd_pcm_start(struct snd_pcm_substream *substream) { @@ -1088,6 +1075,13 @@ int snd_pcm_start(struct snd_pcm_substream *substream) SNDRV_PCM_STATE_RUNNING); } +/* take the stream lock and start the streams */ +static int snd_pcm_start_lock_irq(struct snd_pcm_substream *substream) +{ + return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, + SNDRV_PCM_STATE_RUNNING); +} + /* * stop callbacks */ @@ -2658,8 +2652,7 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream) return err; } -static int snd_pcm_delay(struct snd_pcm_substream *substream, - snd_pcm_sframes_t __user *res) +static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; int err; @@ -2693,10 +2686,7 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, break; } snd_pcm_stream_unlock_irq(substream); - if (!err) - if (put_user(n, res)) - err = -EFAULT; - return err; + return err < 0 ? err : n; } static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, @@ -2784,7 +2774,7 @@ static int snd_pcm_common_ioctl1(struct file *file, case SNDRV_PCM_IOCTL_RESET: return snd_pcm_reset(substream); case SNDRV_PCM_IOCTL_START: - return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, SNDRV_PCM_STATE_RUNNING); + return snd_pcm_start_lock_irq(substream); case SNDRV_PCM_IOCTL_LINK: return snd_pcm_link(substream, (int)(unsigned long) arg); case SNDRV_PCM_IOCTL_UNLINK: @@ -2796,7 +2786,16 @@ static int snd_pcm_common_ioctl1(struct file *file, case SNDRV_PCM_IOCTL_HWSYNC: return snd_pcm_hwsync(substream); case SNDRV_PCM_IOCTL_DELAY: - return snd_pcm_delay(substream, arg); + { + snd_pcm_sframes_t delay = snd_pcm_delay(substream); + snd_pcm_sframes_t __user *res = arg; + + if (delay < 0) + return delay; + if (put_user(delay, res)) + return -EFAULT; + return 0; + } case SNDRV_PCM_IOCTL_SYNC_PTR: return snd_pcm_sync_ptr(substream, arg); #ifdef CONFIG_SND_SUPPORT_OLD_API @@ -3010,30 +3009,55 @@ static long snd_pcm_capture_ioctl(struct file *file, unsigned int cmd, (void __user *)arg); } +/** + * snd_pcm_kernel_ioctl - Execute PCM ioctl in the kernel-space + * @substream: PCM substream + * @cmd: IOCTL cmd + * @arg: IOCTL argument + * + * The function is provided primarily for OSS layer and USB gadget drivers, + * and it allows only the limited set of ioctls (hw_params, sw_params, + * prepare, start, drain, drop, forward). + */ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - mm_segment_t fs; - int result; + snd_pcm_uframes_t *frames = arg; + snd_pcm_sframes_t result; - fs = snd_enter_user(); - switch (substream->stream) { - case SNDRV_PCM_STREAM_PLAYBACK: - result = snd_pcm_playback_ioctl1(NULL, substream, cmd, - (void __user *)arg); - break; - case SNDRV_PCM_STREAM_CAPTURE: - result = snd_pcm_capture_ioctl1(NULL, substream, cmd, - (void __user *)arg); - break; + switch (cmd) { + case SNDRV_PCM_IOCTL_FORWARD: + { + /* provided only for OSS; capture-only and no value returned */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + result = snd_pcm_capture_forward(substream, *frames); + return result < 0 ? result : 0; + } + case SNDRV_PCM_IOCTL_HW_PARAMS: + return snd_pcm_hw_params(substream, arg); + case SNDRV_PCM_IOCTL_SW_PARAMS: + return snd_pcm_sw_params(substream, arg); + case SNDRV_PCM_IOCTL_PREPARE: + return snd_pcm_prepare(substream, NULL); + case SNDRV_PCM_IOCTL_START: + return snd_pcm_start_lock_irq(substream); + case SNDRV_PCM_IOCTL_DRAIN: + return snd_pcm_drain(substream, NULL); + case SNDRV_PCM_IOCTL_DROP: + return snd_pcm_drop(substream); + case SNDRV_PCM_IOCTL_DELAY: + { + result = snd_pcm_delay(substream); + if (result < 0) + return result; + *frames = result; + return 0; + } default: - result = -EINVAL; - break; + return -EINVAL; } - snd_leave_user(fs); - return result; } - EXPORT_SYMBOL(snd_pcm_kernel_ioctl); static ssize_t snd_pcm_read(struct file *file, char __user *buf, size_t count, -- 2.13.0

2 1

[alsa-devel] [PATCH] ALSA: control: Simplify snd_ctl_elem_list() implementation
by Takashi Iwai 22 May '17

22 May '17

This patch simplifies the code of snd_ctl_elem_list() in the following ways: - Avoid a vmalloc() temporary buffer but do copy in each iteration; the vmalloc buffer was introduced at the time we took the spinlock for the ctl element management. - Use the standard list_for_each_entry() macro - Merge two loops into one; it used to be a loop for skipping until offset becomes zero and another loop to copy the data. They can be folded into a single loop easily. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 66 +++++++++++++++++++--------------------------------- 1 file changed, 24 insertions(+), 42 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index c109b82eef4b..47080da8451a 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -747,11 +747,11 @@ static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, static int snd_ctl_elem_list(struct snd_card *card, struct snd_ctl_elem_list __user *_list) { - struct list_head *plist; struct snd_ctl_elem_list list; struct snd_kcontrol *kctl; - struct snd_ctl_elem_id *dst, *id; + struct snd_ctl_elem_id id; unsigned int offset, space, jidx; + int err = 0; if (copy_from_user(&list, _list, sizeof(list))) return -EFAULT; @@ -760,52 +760,34 @@ static int snd_ctl_elem_list(struct snd_card *card, /* try limit maximum space */ if (space > 16384) return -ENOMEM; + down_read(&card->controls_rwsem); + list.count = card->controls_count; + list.used = 0; if (space > 0) { - /* allocate temporary buffer for atomic operation */ - dst = vmalloc(space * sizeof(struct snd_ctl_elem_id)); - if (dst == NULL) - return -ENOMEM; - down_read(&card->controls_rwsem); - list.count = card->controls_count; - plist = card->controls.next; - while (plist != &card->controls) { - if (offset == 0) - break; - kctl = snd_kcontrol(plist); - if (offset < kctl->count) - break; - offset -= kctl->count; - plist = plist->next; - } - list.used = 0; - id = dst; - while (space > 0 && plist != &card->controls) { - kctl = snd_kcontrol(plist); - for (jidx = offset; space > 0 && jidx < kctl->count; jidx++) { - snd_ctl_build_ioff(id, kctl, jidx); - id++; - space--; + list_for_each_entry(kctl, &card->controls, list) { + if (offset >= kctl->count) { + offset -= kctl->count; + continue; + } + for (jidx = offset; jidx < kctl->count; jidx++) { + snd_ctl_build_ioff(&id, kctl, jidx); + if (copy_to_user(list.pids + list.used, &id, + sizeof(id))) { + err = -EFAULT; + goto out; + } list.used++; + if (!--space) + goto out; } - plist = plist->next; offset = 0; } - up_read(&card->controls_rwsem); - if (list.used > 0 && - copy_to_user(list.pids, dst, - list.used * sizeof(struct snd_ctl_elem_id))) { - vfree(dst); - return -EFAULT; - } - vfree(dst); - } else { - down_read(&card->controls_rwsem); - list.count = card->controls_count; - up_read(&card->controls_rwsem); } - if (copy_to_user(_list, &list, sizeof(list))) - return -EFAULT; - return 0; + out: + up_read(&card->controls_rwsem); + if (!err && copy_to_user(_list, &list, sizeof(list))) + err = -EFAULT; + return err; } static bool validate_element_member_dimension(struct snd_ctl_elem_info *info) -- 2.13.0

2 2

[alsa-devel] [PATCH v1] ACPI: Switch to use generic UUID API
by Andy Shevchenko 22 May '17

22 May '17

acpi_evaluate_dsm() and friends take a pointer to a raw buffer of 16 bytes. Instead we convert them to use uuid_le type. At the same time we convert current users. acpi_str_to_uuid() becomes useless after the conversion and it's safe to get rid of it. The conversion fixes a potential bug in int340x_thermal as well since we have to use memcmp() on binary data. Cc: Rafael J. Wysocki <rjw(a)rjwysocki.net> Cc: Mika Westerberg <mika.westerberg(a)linux.intel.com> Cc: Borislav Petkov <bp(a)suse.de> Cc: Dan Williams <dan.j.williams(a)intel.com> Cc: Amir Goldstein <amir73il(a)gmail.com> Cc: Jarkko Sakkinen <jarkko.sakkinen(a)linux.intel.com> Cc: Jani Nikula <jani.nikula(a)linux.intel.com> Cc: Ben Skeggs <bskeggs(a)redhat.com> Cc: Benjamin Tissoires <benjamin.tissoires(a)redhat.com> Cc: Joerg Roedel <joro(a)8bytes.org> Cc: Adrian Hunter <adrian.hunter(a)intel.com> Cc: Yisen Zhuang <yisen.zhuang(a)huawei.com> Cc: Bjorn Helgaas <bhelgaas(a)google.com> Cc: Zhang Rui <rui.zhang(a)intel.com> Cc: Felipe Balbi <balbi(a)kernel.org> Cc: Mathias Nyman <mathias.nyman(a)intel.com> Cc: Heikki Krogerus <heikki.krogerus(a)linux.intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- drivers/acpi/acpi_extlog.c | 10 +++--- drivers/acpi/bus.c | 29 ++-------------- drivers/acpi/nfit/core.c | 40 +++++++++++----------- drivers/acpi/nfit/nfit.h | 3 +- drivers/acpi/utils.c | 4 +-- drivers/char/tpm/tpm_crb.c | 9 +++-- drivers/char/tpm/tpm_ppi.c | 20 +++++------ drivers/gpu/drm/i915/intel_acpi.c | 14 +++----- drivers/gpu/drm/nouveau/nouveau_acpi.c | 20 +++++------ drivers/gpu/drm/nouveau/nvkm/subdev/mxm/base.c | 9 +++-- drivers/hid/i2c-hid/i2c-hid.c | 9 +++-- drivers/iommu/dmar.c | 11 +++--- drivers/mmc/host/sdhci-pci-core.c | 9 +++-- drivers/net/ethernet/hisilicon/hns/hns_dsaf_misc.c | 15 ++++---- drivers/pci/pci-acpi.c | 11 +++--- drivers/pci/pci-label.c | 4 +-- drivers/thermal/int340x_thermal/int3400_thermal.c | 8 ++--- drivers/usb/dwc3/dwc3-pci.c | 6 ++-- drivers/usb/host/xhci-pci.c | 9 +++-- drivers/usb/misc/ucsi.c | 2 +- drivers/usb/typec/typec_wcove.c | 4 +-- include/acpi/acpi_bus.h | 9 ++--- include/linux/acpi.h | 4 +-- include/linux/pci-acpi.h | 2 +- sound/soc/intel/skylake/skl-nhlt.c | 7 ++-- tools/testing/nvdimm/test/iomap.c | 2 +- tools/testing/nvdimm/test/nfit.c | 2 +- 27 files changed, 116 insertions(+), 156 deletions(-) diff --git a/drivers/acpi/acpi_extlog.c b/drivers/acpi/acpi_extlog.c index 502ea4dc2080..69d6140b6afa 100644 --- a/drivers/acpi/acpi_extlog.c +++ b/drivers/acpi/acpi_extlog.c @@ -182,17 +182,17 @@ static int extlog_print(struct notifier_block *nb, unsigned long val, static bool __init extlog_get_l1addr(void) { - u8 uuid[16]; + uuid_le uuid; acpi_handle handle; union acpi_object *obj; - acpi_str_to_uuid(extlog_dsm_uuid, uuid); - + if (uuid_le_to_bin(extlog_dsm_uuid, &uuid)) + return false; if (ACPI_FAILURE(acpi_get_handle(NULL, "\\_SB", &handle))) return false; - if (!acpi_check_dsm(handle, uuid, EXTLOG_DSM_REV, 1 << EXTLOG_FN_ADDR)) + if (!acpi_check_dsm(handle, &uuid, EXTLOG_DSM_REV, 1 << EXTLOG_FN_ADDR)) return false; - obj = acpi_evaluate_dsm_typed(handle, uuid, EXTLOG_DSM_REV, + obj = acpi_evaluate_dsm_typed(handle, &uuid, EXTLOG_DSM_REV, EXTLOG_FN_ADDR, NULL, ACPI_TYPE_INTEGER); if (!obj) { return false; diff --git a/drivers/acpi/bus.c b/drivers/acpi/bus.c index 784bda663d16..e8130a4873e9 100644 --- a/drivers/acpi/bus.c +++ b/drivers/acpi/bus.c @@ -196,42 +196,19 @@ static void acpi_print_osc_error(acpi_handle handle, pr_debug("\n"); } -acpi_status acpi_str_to_uuid(char *str, u8 *uuid) -{ - int i; - static int opc_map_to_uuid[16] = {6, 4, 2, 0, 11, 9, 16, 14, 19, 21, - 24, 26, 28, 30, 32, 34}; - - if (strlen(str) != 36) - return AE_BAD_PARAMETER; - for (i = 0; i < 36; i++) { - if (i == 8 || i == 13 || i == 18 || i == 23) { - if (str[i] != '-') - return AE_BAD_PARAMETER; - } else if (!isxdigit(str[i])) - return AE_BAD_PARAMETER; - } - for (i = 0; i < 16; i++) { - uuid[i] = hex_to_bin(str[opc_map_to_uuid[i]]) << 4; - uuid[i] |= hex_to_bin(str[opc_map_to_uuid[i] + 1]); - } - return AE_OK; -} -EXPORT_SYMBOL_GPL(acpi_str_to_uuid); - acpi_status acpi_run_osc(acpi_handle handle, struct acpi_osc_context *context) { acpi_status status; struct acpi_object_list input; union acpi_object in_params[4]; union acpi_object *out_obj; - u8 uuid[16]; + uuid_le uuid; u32 errors; struct acpi_buffer output = {ACPI_ALLOCATE_BUFFER, NULL}; if (!context) return AE_ERROR; - if (ACPI_FAILURE(acpi_str_to_uuid(context->uuid_str, uuid))) + if (uuid_le_to_bin(context->uuid_str, &uuid)) return AE_ERROR; context->ret.length = ACPI_ALLOCATE_BUFFER; context->ret.pointer = NULL; @@ -241,7 +218,7 @@ acpi_status acpi_run_osc(acpi_handle handle, struct acpi_osc_context *context) input.pointer = in_params; in_params[0].type = ACPI_TYPE_BUFFER; in_params[0].buffer.length = 16; - in_params[0].buffer.pointer = uuid; + in_params[0].buffer.pointer = (u8 *)&uuid; in_params[1].type = ACPI_TYPE_INTEGER; in_params[1].integer.value = context->rev; in_params[2].type = ACPI_TYPE_INTEGER; diff --git a/drivers/acpi/nfit/core.c b/drivers/acpi/nfit/core.c index 0f7982a1caaf..bd3e45ede056 100644 --- a/drivers/acpi/nfit/core.c +++ b/drivers/acpi/nfit/core.c @@ -74,11 +74,11 @@ struct nfit_table_prev { struct list_head flushes; }; -static u8 nfit_uuid[NFIT_UUID_MAX][16]; +static uuid_le nfit_uuid[NFIT_UUID_MAX]; -const u8 *to_nfit_uuid(enum nfit_uuids id) +const uuid_le *to_nfit_uuid(enum nfit_uuids id) { - return nfit_uuid[id]; + return &nfit_uuid[id]; } EXPORT_SYMBOL(to_nfit_uuid); @@ -207,7 +207,7 @@ int acpi_nfit_ctl(struct nvdimm_bus_descriptor *nd_desc, struct nvdimm *nvdimm, u32 offset, fw_status = 0; acpi_handle handle; unsigned int func; - const u8 *uuid; + const uuid_le *uuid; int rc, i; func = cmd; @@ -394,7 +394,7 @@ int nfit_spa_type(struct acpi_nfit_system_address *spa) int i; for (i = 0; i < NFIT_UUID_MAX; i++) - if (memcmp(to_nfit_uuid(i), spa->range_guid, 16) == 0) + if (!uuid_le_cmp_pp(to_nfit_uuid(i), (uuid_le *)spa->range_guid)) return i; return -1; } @@ -1400,7 +1400,7 @@ static int acpi_nfit_add_dimm(struct acpi_nfit_desc *acpi_desc, struct acpi_device *adev, *adev_dimm; struct device *dev = acpi_desc->dev; unsigned long dsm_mask; - const u8 *uuid; + const uuid_le *uuid = to_nfit_uuid(NFIT_DEV_DIMM); int i; int family = -1; @@ -1596,7 +1596,7 @@ static int acpi_nfit_register_dimms(struct acpi_nfit_desc *acpi_desc) static void acpi_nfit_init_dsms(struct acpi_nfit_desc *acpi_desc) { struct nvdimm_bus_descriptor *nd_desc = &acpi_desc->nd_desc; - const u8 *uuid = to_nfit_uuid(NFIT_DEV_BUS); + const uuid_le *uuid = to_nfit_uuid(NFIT_DEV_BUS); struct acpi_device *adev; int i; @@ -3036,19 +3036,19 @@ static __init int nfit_init(void) BUILD_BUG_ON(sizeof(struct acpi_nfit_control_region) != 80); BUILD_BUG_ON(sizeof(struct acpi_nfit_data_region) != 40); - acpi_str_to_uuid(UUID_VOLATILE_MEMORY, nfit_uuid[NFIT_SPA_VOLATILE]); - acpi_str_to_uuid(UUID_PERSISTENT_MEMORY, nfit_uuid[NFIT_SPA_PM]); - acpi_str_to_uuid(UUID_CONTROL_REGION, nfit_uuid[NFIT_SPA_DCR]); - acpi_str_to_uuid(UUID_DATA_REGION, nfit_uuid[NFIT_SPA_BDW]); - acpi_str_to_uuid(UUID_VOLATILE_VIRTUAL_DISK, nfit_uuid[NFIT_SPA_VDISK]); - acpi_str_to_uuid(UUID_VOLATILE_VIRTUAL_CD, nfit_uuid[NFIT_SPA_VCD]); - acpi_str_to_uuid(UUID_PERSISTENT_VIRTUAL_DISK, nfit_uuid[NFIT_SPA_PDISK]); - acpi_str_to_uuid(UUID_PERSISTENT_VIRTUAL_CD, nfit_uuid[NFIT_SPA_PCD]); - acpi_str_to_uuid(UUID_NFIT_BUS, nfit_uuid[NFIT_DEV_BUS]); - acpi_str_to_uuid(UUID_NFIT_DIMM, nfit_uuid[NFIT_DEV_DIMM]); - acpi_str_to_uuid(UUID_NFIT_DIMM_N_HPE1, nfit_uuid[NFIT_DEV_DIMM_N_HPE1]); - acpi_str_to_uuid(UUID_NFIT_DIMM_N_HPE2, nfit_uuid[NFIT_DEV_DIMM_N_HPE2]); - acpi_str_to_uuid(UUID_NFIT_DIMM_N_MSFT, nfit_uuid[NFIT_DEV_DIMM_N_MSFT]); + uuid_le_to_bin(UUID_VOLATILE_MEMORY, &nfit_uuid[NFIT_SPA_VOLATILE]); + uuid_le_to_bin(UUID_PERSISTENT_MEMORY, &nfit_uuid[NFIT_SPA_PM]); + uuid_le_to_bin(UUID_CONTROL_REGION, &nfit_uuid[NFIT_SPA_DCR]); + uuid_le_to_bin(UUID_DATA_REGION, &nfit_uuid[NFIT_SPA_BDW]); + uuid_le_to_bin(UUID_VOLATILE_VIRTUAL_DISK, &nfit_uuid[NFIT_SPA_VDISK]); + uuid_le_to_bin(UUID_VOLATILE_VIRTUAL_CD, &nfit_uuid[NFIT_SPA_VCD]); + uuid_le_to_bin(UUID_PERSISTENT_VIRTUAL_DISK, &nfit_uuid[NFIT_SPA_PDISK]); + uuid_le_to_bin(UUID_PERSISTENT_VIRTUAL_CD, &nfit_uuid[NFIT_SPA_PCD]); + uuid_le_to_bin(UUID_NFIT_BUS, &nfit_uuid[NFIT_DEV_BUS]); + uuid_le_to_bin(UUID_NFIT_DIMM, &nfit_uuid[NFIT_DEV_DIMM]); + uuid_le_to_bin(UUID_NFIT_DIMM_N_HPE1, &nfit_uuid[NFIT_DEV_DIMM_N_HPE1]); + uuid_le_to_bin(UUID_NFIT_DIMM_N_HPE2, &nfit_uuid[NFIT_DEV_DIMM_N_HPE2]); + uuid_le_to_bin(UUID_NFIT_DIMM_N_MSFT, &nfit_uuid[NFIT_DEV_DIMM_N_MSFT]); nfit_wq = create_singlethread_workqueue("nfit"); if (!nfit_wq) diff --git a/drivers/acpi/nfit/nfit.h b/drivers/acpi/nfit/nfit.h index 58fb7d68e04a..2f233b28709f 100644 --- a/drivers/acpi/nfit/nfit.h +++ b/drivers/acpi/nfit/nfit.h @@ -18,7 +18,6 @@ #include <linux/libnvdimm.h> #include <linux/ndctl.h> #include <linux/types.h> -#include <linux/uuid.h> #include <linux/acpi.h> #include <acpi/acuuid.h> @@ -237,7 +236,7 @@ static inline struct acpi_nfit_desc *to_acpi_desc( return container_of(nd_desc, struct acpi_nfit_desc, nd_desc); } -const u8 *to_nfit_uuid(enum nfit_uuids id); +const uuid_le *to_nfit_uuid(enum nfit_uuids id); int acpi_nfit_init(struct acpi_nfit_desc *acpi_desc, void *nfit, acpi_size sz); void acpi_nfit_shutdown(void *data); void __acpi_nfit_notify(struct device *dev, acpi_handle handle, u32 event); diff --git a/drivers/acpi/utils.c b/drivers/acpi/utils.c index 27d0dcfcf47d..bbe8a950e508 100644 --- a/drivers/acpi/utils.c +++ b/drivers/acpi/utils.c @@ -625,7 +625,7 @@ acpi_status acpi_evaluate_lck(acpi_handle handle, int lock) * some old BIOSes do expect a buffer or an integer etc. */ union acpi_object * -acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, u64 rev, u64 func, +acpi_evaluate_dsm(acpi_handle handle, const uuid_le *uuid, u64 rev, u64 func, union acpi_object *argv4) { acpi_status ret; @@ -674,7 +674,7 @@ EXPORT_SYMBOL(acpi_evaluate_dsm); * functions. Currently only support 64 functions at maximum, should be * enough for now. */ -bool acpi_check_dsm(acpi_handle handle, const u8 *uuid, u64 rev, u64 funcs) +bool acpi_check_dsm(acpi_handle handle, const uuid_le *uuid, u64 rev, u64 funcs) { int i; u64 mask = 0; diff --git a/drivers/char/tpm/tpm_crb.c b/drivers/char/tpm/tpm_crb.c index b917b9d5f710..f789f7e5a17d 100644 --- a/drivers/char/tpm/tpm_crb.c +++ b/drivers/char/tpm/tpm_crb.c @@ -27,10 +27,9 @@ #define ACPI_SIG_TPM2 "TPM2" -static const u8 CRB_ACPI_START_UUID[] = { - /* 0000 */ 0xAB, 0x6C, 0xBF, 0x6B, 0x63, 0x54, 0x14, 0x47, - /* 0008 */ 0xB7, 0xCD, 0xF0, 0x20, 0x3C, 0x03, 0x68, 0xD4 -}; +static const uuid_le crb_acpi_start_uuid = + UUID_LE(0x6BBF6CAB, 0x5463, 0x4714, + 0xB7, 0xCD, 0xF0, 0x20, 0x3C, 0x03, 0x68, 0xD4); enum crb_defaults { CRB_ACPI_START_REVISION_ID = 1, @@ -266,7 +265,7 @@ static int crb_do_acpi_start(struct tpm_chip *chip) int rc; obj = acpi_evaluate_dsm(chip->acpi_dev_handle, - CRB_ACPI_START_UUID, + &crb_acpi_start_uuid, CRB_ACPI_START_REVISION_ID, CRB_ACPI_START_INDEX, NULL); diff --git a/drivers/char/tpm/tpm_ppi.c b/drivers/char/tpm/tpm_ppi.c index 692a2c6ae036..7cf682426361 100644 --- a/drivers/char/tpm/tpm_ppi.c +++ b/drivers/char/tpm/tpm_ppi.c @@ -32,20 +32,16 @@ #define PPI_VS_REQ_START 128 #define PPI_VS_REQ_END 255 -static const u8 tpm_ppi_uuid[] = { - 0xA6, 0xFA, 0xDD, 0x3D, - 0x1B, 0x36, - 0xB4, 0x4E, - 0xA4, 0x24, - 0x8D, 0x10, 0x08, 0x9D, 0x16, 0x53 -}; +static const uuid_le tpm_ppi_uuid = + UUID_LE(0x3DDDFAA6, 0x361B, 0x4EB4, + 0xA4, 0x24, 0x8D, 0x10, 0x08, 0x9D, 0x16, 0x53); static inline union acpi_object * tpm_eval_dsm(acpi_handle ppi_handle, int func, acpi_object_type type, union acpi_object *argv4) { BUG_ON(!ppi_handle); - return acpi_evaluate_dsm_typed(ppi_handle, tpm_ppi_uuid, + return acpi_evaluate_dsm_typed(ppi_handle, &tpm_ppi_uuid, TPM_PPI_REVISION_ID, func, argv4, type); } @@ -107,7 +103,7 @@ static ssize_t tpm_store_ppi_request(struct device *dev, * is updated with function index from SUBREQ to SUBREQ2 since PPI * version 1.1 */ - if (acpi_check_dsm(chip->acpi_dev_handle, tpm_ppi_uuid, + if (acpi_check_dsm(chip->acpi_dev_handle, &tpm_ppi_uuid, TPM_PPI_REVISION_ID, 1 << TPM_PPI_FN_SUBREQ2)) func = TPM_PPI_FN_SUBREQ2; @@ -268,7 +264,7 @@ static ssize_t show_ppi_operations(acpi_handle dev_handle, char *buf, u32 start, "User not required", }; - if (!acpi_check_dsm(dev_handle, tpm_ppi_uuid, TPM_PPI_REVISION_ID, + if (!acpi_check_dsm(dev_handle, &tpm_ppi_uuid, TPM_PPI_REVISION_ID, 1 << TPM_PPI_FN_GETOPR)) return -EPERM; @@ -341,12 +337,12 @@ void tpm_add_ppi(struct tpm_chip *chip) if (!chip->acpi_dev_handle) return; - if (!acpi_check_dsm(chip->acpi_dev_handle, tpm_ppi_uuid, + if (!acpi_check_dsm(chip->acpi_dev_handle, &tpm_ppi_uuid, TPM_PPI_REVISION_ID, 1 << TPM_PPI_FN_VERSION)) return; /* Cache PPI version string. */ - obj = acpi_evaluate_dsm_typed(chip->acpi_dev_handle, tpm_ppi_uuid, + obj = acpi_evaluate_dsm_typed(chip->acpi_dev_handle, &tpm_ppi_uuid, TPM_PPI_REVISION_ID, TPM_PPI_FN_VERSION, NULL, ACPI_TYPE_STRING); if (obj) { diff --git a/drivers/gpu/drm/i915/intel_acpi.c b/drivers/gpu/drm/i915/intel_acpi.c index eb638a1e69d2..72bfe6ceadf8 100644 --- a/drivers/gpu/drm/i915/intel_acpi.c +++ b/drivers/gpu/drm/i915/intel_acpi.c @@ -15,13 +15,9 @@ static struct intel_dsm_priv { acpi_handle dhandle; } intel_dsm_priv; -static const u8 intel_dsm_guid[] = { - 0xd3, 0x73, 0xd8, 0x7e, - 0xd0, 0xc2, - 0x4f, 0x4e, - 0xa8, 0x54, - 0x0f, 0x13, 0x17, 0xb0, 0x1c, 0x2c -}; +static const uuid_le intel_dsm_guid = + UUID_LE(0x7ed873d3, 0xc2d0, 0x4e4f, + 0xa8, 0x54, 0x0f, 0x13, 0x17, 0xb0, 0x1c, 0x2c); static char *intel_dsm_port_name(u8 id) { @@ -80,7 +76,7 @@ static void intel_dsm_platform_mux_info(void) int i; union acpi_object *pkg, *connector_count; - pkg = acpi_evaluate_dsm_typed(intel_dsm_priv.dhandle, intel_dsm_guid, + pkg = acpi_evaluate_dsm_typed(intel_dsm_priv.dhandle, &intel_dsm_guid, INTEL_DSM_REVISION_ID, INTEL_DSM_FN_PLATFORM_MUX_INFO, NULL, ACPI_TYPE_PACKAGE); if (!pkg) { @@ -118,7 +114,7 @@ static bool intel_dsm_pci_probe(struct pci_dev *pdev) if (!dhandle) return false; - if (!acpi_check_dsm(dhandle, intel_dsm_guid, INTEL_DSM_REVISION_ID, + if (!acpi_check_dsm(dhandle, &intel_dsm_guid, INTEL_DSM_REVISION_ID, 1 << INTEL_DSM_FN_PLATFORM_MUX_INFO)) { DRM_DEBUG_KMS("no _DSM method for intel device\n"); return false; diff --git a/drivers/gpu/drm/nouveau/nouveau_acpi.c b/drivers/gpu/drm/nouveau/nouveau_acpi.c index 39468c218027..faea23276d4a 100644 --- a/drivers/gpu/drm/nouveau/nouveau_acpi.c +++ b/drivers/gpu/drm/nouveau/nouveau_acpi.c @@ -60,15 +60,13 @@ bool nouveau_is_v1_dsm(void) { } #ifdef CONFIG_VGA_SWITCHEROO -static const char nouveau_dsm_muid[] = { - 0xA0, 0xA0, 0x95, 0x9D, 0x60, 0x00, 0x48, 0x4D, - 0xB3, 0x4D, 0x7E, 0x5F, 0xEA, 0x12, 0x9F, 0xD4, -}; +static const uuid_le nouveau_dsm_muid = + UUID_LE(0x9D95A0A0, 0x0060, 0x4D48, + 0xB3, 0x4D, 0x7E, 0x5F, 0xEA, 0x12, 0x9F, 0xD4); -static const char nouveau_op_dsm_muid[] = { - 0xF8, 0xD8, 0x86, 0xA4, 0xDA, 0x0B, 0x1B, 0x47, - 0xA7, 0x2B, 0x60, 0x42, 0xA6, 0xB5, 0xBE, 0xE0, -}; +static const uuid_le nouveau_op_dsm_muid = + UUID_LE(0xA486D8F8, 0x0BDA, 0x471B, + 0xA7, 0x2B, 0x60, 0x42, 0xA6, 0xB5, 0xBE, 0xE0); static int nouveau_optimus_dsm(acpi_handle handle, int func, int arg, uint32_t *result) { @@ -86,7 +84,7 @@ static int nouveau_optimus_dsm(acpi_handle handle, int func, int arg, uint32_t * args_buff[i] = (arg >> i * 8) & 0xFF; *result = 0; - obj = acpi_evaluate_dsm_typed(handle, nouveau_op_dsm_muid, 0x00000100, + obj = acpi_evaluate_dsm_typed(handle, &nouveau_op_dsm_muid, 0x00000100, func, &argv4, ACPI_TYPE_BUFFER); if (!obj) { acpi_handle_info(handle, "failed to evaluate _DSM\n"); @@ -138,7 +136,7 @@ static int nouveau_dsm(acpi_handle handle, int func, int arg) .integer.value = arg, }; - obj = acpi_evaluate_dsm_typed(handle, nouveau_dsm_muid, 0x00000102, + obj = acpi_evaluate_dsm_typed(handle, &nouveau_dsm_muid, 0x00000102, func, &argv4, ACPI_TYPE_INTEGER); if (!obj) { acpi_handle_info(handle, "failed to evaluate _DSM\n"); @@ -259,7 +257,7 @@ static void nouveau_dsm_pci_probe(struct pci_dev *pdev, acpi_handle *dhandle_out if (!acpi_has_method(dhandle, "_DSM")) return; - supports_mux = acpi_check_dsm(dhandle, nouveau_dsm_muid, 0x00000102, + supports_mux = acpi_check_dsm(dhandle, &nouveau_dsm_muid, 0x00000102, 1 << NOUVEAU_DSM_POWER); optimus_funcs = nouveau_dsm_get_optimus_functions(dhandle); diff --git a/drivers/gpu/drm/nouveau/nvkm/subdev/mxm/base.c b/drivers/gpu/drm/nouveau/nvkm/subdev/mxm/base.c index e3e2f5e83815..cc95b8150a86 100644 --- a/drivers/gpu/drm/nouveau/nvkm/subdev/mxm/base.c +++ b/drivers/gpu/drm/nouveau/nvkm/subdev/mxm/base.c @@ -81,10 +81,9 @@ mxm_shadow_dsm(struct nvkm_mxm *mxm, u8 version) { struct nvkm_subdev *subdev = &mxm->subdev; struct nvkm_device *device = subdev->device; - static char muid[] = { - 0x00, 0xA4, 0x04, 0x40, 0x7D, 0x91, 0xF2, 0x4C, - 0xB8, 0x9C, 0x79, 0xB6, 0x2F, 0xD5, 0x56, 0x65 - }; + static uuid_le muid = + UUID_LE(0x4004A400, 0x917D, 0x4CF2, + 0xB8, 0x9C, 0x79, 0xB6, 0x2F, 0xD5, 0x56, 0x65); u32 mxms_args[] = { 0x00000000 }; union acpi_object argv4 = { .buffer.type = ACPI_TYPE_BUFFER, @@ -105,7 +104,7 @@ mxm_shadow_dsm(struct nvkm_mxm *mxm, u8 version) * unless you pass in exactly the version it supports.. */ rev = (version & 0xf0) << 4 | (version & 0x0f); - obj = acpi_evaluate_dsm(handle, muid, rev, 0x00000010, &argv4); + obj = acpi_evaluate_dsm(handle, &muid, rev, 0x00000010, &argv4); if (!obj) { nvkm_debug(subdev, "DSM MXMS failed\n"); return false; diff --git a/drivers/hid/i2c-hid/i2c-hid.c b/drivers/hid/i2c-hid/i2c-hid.c index 8daa8ce64ebb..f83bd717cdd5 100644 --- a/drivers/hid/i2c-hid/i2c-hid.c +++ b/drivers/hid/i2c-hid/i2c-hid.c @@ -872,10 +872,9 @@ static int i2c_hid_fetch_hid_descriptor(struct i2c_hid *ihid) static int i2c_hid_acpi_pdata(struct i2c_client *client, struct i2c_hid_platform_data *pdata) { - static u8 i2c_hid_guid[] = { - 0xF7, 0xF6, 0xDF, 0x3C, 0x67, 0x42, 0x55, 0x45, - 0xAD, 0x05, 0xB3, 0x0A, 0x3D, 0x89, 0x38, 0xDE, - }; + static uuid_le i2c_hid_guid = + UUID_LE(0x3CDFF6F7, 0x4267, 0x4555, + 0xAD, 0x05, 0xB3, 0x0A, 0x3D, 0x89, 0x38, 0xDE); union acpi_object *obj; struct acpi_device *adev; acpi_handle handle; @@ -884,7 +883,7 @@ static int i2c_hid_acpi_pdata(struct i2c_client *client, if (!handle || acpi_bus_get_device(handle, &adev)) return -ENODEV; - obj = acpi_evaluate_dsm_typed(handle, i2c_hid_guid, 1, 1, NULL, + obj = acpi_evaluate_dsm_typed(handle, &i2c_hid_guid, 1, 1, NULL, ACPI_TYPE_INTEGER); if (!obj) { dev_err(&client->dev, "device _DSM execution failed\n"); diff --git a/drivers/iommu/dmar.c b/drivers/iommu/dmar.c index cbf7763d8091..420d51b286ad 100644 --- a/drivers/iommu/dmar.c +++ b/drivers/iommu/dmar.c @@ -1808,10 +1808,9 @@ IOMMU_INIT_POST(detect_intel_iommu); * for Directed-IO Architecture Specifiction, Rev 2.2, Section 8.8 * "Remapping Hardware Unit Hot Plug". */ -static u8 dmar_hp_uuid[] = { - /* 0000 */ 0xA6, 0xA3, 0xC1, 0xD8, 0x9B, 0xBE, 0x9B, 0x4C, - /* 0008 */ 0x91, 0xBF, 0xC3, 0xCB, 0x81, 0xFC, 0x5D, 0xAF -}; +static uuid_le dmar_hp_uuid = + UUID_LE(0xD8C1A3A6, 0xBE9B, 0x4C9B, + 0x91, 0xBF, 0xC3, 0xCB, 0x81, 0xFC, 0x5D, 0xAF); /* * Currently there's only one revision and BIOS will not check the revision id, @@ -1824,7 +1823,7 @@ static u8 dmar_hp_uuid[] = { static inline bool dmar_detect_dsm(acpi_handle handle, int func) { - return acpi_check_dsm(handle, dmar_hp_uuid, DMAR_DSM_REV_ID, 1 << func); + return acpi_check_dsm(handle, &dmar_hp_uuid, DMAR_DSM_REV_ID, 1 << func); } static int dmar_walk_dsm_resource(acpi_handle handle, int func, @@ -1843,7 +1842,7 @@ static int dmar_walk_dsm_resource(acpi_handle handle, int func, if (!dmar_detect_dsm(handle, func)) return 0; - obj = acpi_evaluate_dsm_typed(handle, dmar_hp_uuid, DMAR_DSM_REV_ID, + obj = acpi_evaluate_dsm_typed(handle, &dmar_hp_uuid, DMAR_DSM_REV_ID, func, NULL, ACPI_TYPE_BUFFER); if (!obj) return -ENODEV; diff --git a/drivers/mmc/host/sdhci-pci-core.c b/drivers/mmc/host/sdhci-pci-core.c index 92fc3f7c538d..262b8c320d7c 100644 --- a/drivers/mmc/host/sdhci-pci-core.c +++ b/drivers/mmc/host/sdhci-pci-core.c @@ -404,10 +404,9 @@ struct intel_host { bool d3_retune; }; -const u8 intel_dsm_uuid[] = { - 0xA5, 0x3E, 0xC1, 0xF6, 0xCD, 0x65, 0x1F, 0x46, - 0xAB, 0x7A, 0x29, 0xF7, 0xE8, 0xD5, 0xBD, 0x61, -}; +const uuid_le intel_dsm_uuid = + UUID_LE(0xF6C13EA5, 0x65CD, 0x461F, + 0xAB, 0x7A, 0x29, 0xF7, 0xE8, 0xD5, 0xBD, 0x61); static int __intel_dsm(struct intel_host *intel_host, struct device *dev, unsigned int fn, u32 *result) @@ -416,7 +415,7 @@ static int __intel_dsm(struct intel_host *intel_host, struct device *dev, int err = 0; size_t len; - obj = acpi_evaluate_dsm(ACPI_HANDLE(dev), intel_dsm_uuid, 0, fn, NULL); + obj = acpi_evaluate_dsm(ACPI_HANDLE(dev), &intel_dsm_uuid, 0, fn, NULL); if (!obj) return -EOPNOTSUPP; diff --git a/drivers/net/ethernet/hisilicon/hns/hns_dsaf_misc.c b/drivers/net/ethernet/hisilicon/hns/hns_dsaf_misc.c index e13aa064a8e9..02842fe7f1d0 100644 --- a/drivers/net/ethernet/hisilicon/hns/hns_dsaf_misc.c +++ b/drivers/net/ethernet/hisilicon/hns/hns_dsaf_misc.c @@ -29,10 +29,9 @@ enum _dsm_rst_type { HNS_ROCE_RESET_FUNC = 0x7, }; -const u8 hns_dsaf_acpi_dsm_uuid[] = { - 0x1A, 0xAA, 0x85, 0x1A, 0x93, 0xE2, 0x5E, 0x41, - 0x8E, 0x28, 0x8D, 0x69, 0x0A, 0x0F, 0x82, 0x0A -}; +const uuid_le hns_dsaf_acpi_dsm_uuid = + UUID_LE(0x1A85AA1A, 0xE293, 0x415E, + 0x8E, 0x28, 0x8D, 0x69, 0x0A, 0x0F, 0x82, 0x0A); static void dsaf_write_sub(struct dsaf_device *dsaf_dev, u32 reg, u32 val) { @@ -151,7 +150,7 @@ static void hns_dsaf_acpi_srst_by_port(struct dsaf_device *dsaf_dev, u8 op_type, argv4.package.elements = obj_args; obj = acpi_evaluate_dsm(ACPI_HANDLE(dsaf_dev->dev), - hns_dsaf_acpi_dsm_uuid, 0, op_type, &argv4); + &hns_dsaf_acpi_dsm_uuid, 0, op_type, &argv4); if (!obj) { dev_warn(dsaf_dev->dev, "reset port_type%d port%d fail!", port_type, port); @@ -434,7 +433,7 @@ static phy_interface_t hns_mac_get_phy_if_acpi(struct hns_mac_cb *mac_cb) argv4.package.elements = &obj_args, obj = acpi_evaluate_dsm(ACPI_HANDLE(mac_cb->dev), - hns_dsaf_acpi_dsm_uuid, 0, + &hns_dsaf_acpi_dsm_uuid, 0, HNS_OP_GET_PORT_TYPE_FUNC, &argv4); if (!obj || obj->type != ACPI_TYPE_INTEGER) @@ -474,7 +473,7 @@ int hns_mac_get_sfp_prsnt_acpi(struct hns_mac_cb *mac_cb, int *sfp_prsnt) argv4.package.elements = &obj_args, obj = acpi_evaluate_dsm(ACPI_HANDLE(mac_cb->dev), - hns_dsaf_acpi_dsm_uuid, 0, + &hns_dsaf_acpi_dsm_uuid, 0, HNS_OP_GET_SFP_STAT_FUNC, &argv4); if (!obj || obj->type != ACPI_TYPE_INTEGER) @@ -565,7 +564,7 @@ hns_mac_config_sds_loopback_acpi(struct hns_mac_cb *mac_cb, bool en) argv4.package.elements = obj_args; obj = acpi_evaluate_dsm(ACPI_HANDLE(mac_cb->dsaf_dev->dev), - hns_dsaf_acpi_dsm_uuid, 0, + &hns_dsaf_acpi_dsm_uuid, 0, HNS_OP_SERDES_LP_FUNC, &argv4); if (!obj) { dev_warn(mac_cb->dsaf_dev->dev, "set port%d serdes lp fail!", diff --git a/drivers/pci/pci-acpi.c b/drivers/pci/pci-acpi.c index 001860361434..eb612123e0dd 100644 --- a/drivers/pci/pci-acpi.c +++ b/drivers/pci/pci-acpi.c @@ -24,10 +24,9 @@ * The UUID is defined in the PCI Firmware Specification available here: * https://www.pcisig.com/members/downloads/pcifw_r3_1_13Dec10.pdf */ -const u8 pci_acpi_dsm_uuid[] = { - 0xd0, 0x37, 0xc9, 0xe5, 0x53, 0x35, 0x7a, 0x4d, - 0x91, 0x17, 0xea, 0x4d, 0x19, 0xc3, 0x43, 0x4d -}; +const uuid_le pci_acpi_dsm_uuid = + UUID_LE(0xe5c937d0, 0x3553, 0x4d7a, + 0x91, 0x17, 0xea, 0x4d, 0x19, 0xc3, 0x43, 0x4d); #if defined(CONFIG_PCI_QUIRKS) && defined(CONFIG_ARM64) static int acpi_get_rc_addr(struct acpi_device *adev, struct resource *res) @@ -680,7 +679,7 @@ void acpi_pci_add_bus(struct pci_bus *bus) if (!pci_is_root_bus(bus)) return; - obj = acpi_evaluate_dsm(ACPI_HANDLE(bus->bridge), pci_acpi_dsm_uuid, 3, + obj = acpi_evaluate_dsm(ACPI_HANDLE(bus->bridge), &pci_acpi_dsm_uuid, 3, RESET_DELAY_DSM, NULL); if (!obj) return; @@ -745,7 +744,7 @@ static void pci_acpi_optimize_delay(struct pci_dev *pdev, if (bridge->ignore_reset_delay) pdev->d3cold_delay = 0; - obj = acpi_evaluate_dsm(handle, pci_acpi_dsm_uuid, 3, + obj = acpi_evaluate_dsm(handle, &pci_acpi_dsm_uuid, 3, FUNCTION_DELAY_DSM, NULL); if (!obj) return; diff --git a/drivers/pci/pci-label.c b/drivers/pci/pci-label.c index 51357377efbc..a2c04229f1dc 100644 --- a/drivers/pci/pci-label.c +++ b/drivers/pci/pci-label.c @@ -172,7 +172,7 @@ static int dsm_get_label(struct device *dev, char *buf, if (!handle) return -1; - obj = acpi_evaluate_dsm(handle, pci_acpi_dsm_uuid, 0x2, + obj = acpi_evaluate_dsm(handle, &pci_acpi_dsm_uuid, 0x2, DEVICE_LABEL_DSM, NULL); if (!obj) return -1; @@ -212,7 +212,7 @@ static bool device_has_dsm(struct device *dev) if (!handle) return false; - return !!acpi_check_dsm(handle, pci_acpi_dsm_uuid, 0x2, + return !!acpi_check_dsm(handle, &pci_acpi_dsm_uuid, 0x2, 1 << DEVICE_LABEL_DSM); } diff --git a/drivers/thermal/int340x_thermal/int3400_thermal.c b/drivers/thermal/int340x_thermal/int3400_thermal.c index 9413c4abf0b9..c0eb3bb19b23 100644 --- a/drivers/thermal/int340x_thermal/int3400_thermal.c +++ b/drivers/thermal/int340x_thermal/int3400_thermal.c @@ -23,7 +23,7 @@ enum int3400_thermal_uuid { INT3400_THERMAL_MAXIMUM_UUID, }; -static u8 *int3400_thermal_uuids[INT3400_THERMAL_MAXIMUM_UUID] = { +static const char *int3400_thermal_uuids[INT3400_THERMAL_MAXIMUM_UUID] = { "42A441D6-AE6A-462b-A84B-4A8CE79027D3", "3A95C389-E4B8-4629-A526-C52C88626BAE", "97C68AE7-15FA-499c-B8C9-5DA81D606E0A", @@ -141,10 +141,10 @@ static int int3400_thermal_get_uuids(struct int3400_thermal_priv *priv) } for (j = 0; j < INT3400_THERMAL_MAXIMUM_UUID; j++) { - u8 uuid[16]; + uuid_le u; - acpi_str_to_uuid(int3400_thermal_uuids[j], uuid); - if (!strncmp(uuid, objb->buffer.pointer, 16)) { + uuid_le_to_bin(int3400_thermal_uuids[j], &u); + if (!uuid_le_cmp(*(uuid_le *)objb->buffer.pointer), u) { priv->uuid_bitmap |= (1 << j); break; } diff --git a/drivers/usb/dwc3/dwc3-pci.c b/drivers/usb/dwc3/dwc3-pci.c index a15ec71d0423..6b5284ec76df 100644 --- a/drivers/usb/dwc3/dwc3-pci.c +++ b/drivers/usb/dwc3/dwc3-pci.c @@ -56,7 +56,7 @@ struct dwc3_pci { struct platform_device *dwc3; struct pci_dev *pci; - u8 uuid[16]; + uuid_le uuid; unsigned int has_dsm_for_pm:1; }; @@ -118,7 +118,7 @@ static int dwc3_pci_quirks(struct dwc3_pci *dwc) if (pdev->device == PCI_DEVICE_ID_INTEL_BXT || pdev->device == PCI_DEVICE_ID_INTEL_BXT_M) { - acpi_str_to_uuid(PCI_INTEL_BXT_DSM_UUID, dwc->uuid); + uuid_le_to_bin(PCI_INTEL_BXT_DSM_UUID, &dwc->uuid); dwc->has_dsm_for_pm = true; } @@ -288,7 +288,7 @@ static int dwc3_pci_dsm(struct dwc3_pci *dwc, int param) tmp.type = ACPI_TYPE_INTEGER; tmp.integer.value = param; - obj = acpi_evaluate_dsm(ACPI_HANDLE(&dwc->pci->dev), dwc->uuid, + obj = acpi_evaluate_dsm(ACPI_HANDLE(&dwc->pci->dev), &dwc->uuid, 1, PCI_INTEL_BXT_FUNC_PMU_PWR, &argv4); if (!obj) { dev_err(&dwc->pci->dev, "failed to evaluate _DSM\n"); diff --git a/drivers/usb/host/xhci-pci.c b/drivers/usb/host/xhci-pci.c index 7b86508ac8cf..93b4f0de9418 100644 --- a/drivers/usb/host/xhci-pci.c +++ b/drivers/usb/host/xhci-pci.c @@ -210,13 +210,12 @@ static void xhci_pci_quirks(struct device *dev, struct xhci_hcd *xhci) #ifdef CONFIG_ACPI static void xhci_pme_acpi_rtd3_enable(struct pci_dev *dev) { - static const u8 intel_dsm_uuid[] = { - 0xb7, 0x0c, 0x34, 0xac, 0x01, 0xe9, 0xbf, 0x45, - 0xb7, 0xe6, 0x2b, 0x34, 0xec, 0x93, 0x1e, 0x23, - }; + static const uuid_le intel_dsm_uuid = + UUID_LE(0xac340cb7, 0xe901, 0x45bf, + 0xb7, 0xe6, 0x2b, 0x34, 0xec, 0x93, 0x1e, 0x23); union acpi_object *obj; - obj = acpi_evaluate_dsm(ACPI_HANDLE(&dev->dev), intel_dsm_uuid, 3, 1, + obj = acpi_evaluate_dsm(ACPI_HANDLE(&dev->dev), &intel_dsm_uuid, 3, 1, NULL); ACPI_FREE(obj); } diff --git a/drivers/usb/misc/ucsi.c b/drivers/usb/misc/ucsi.c index 07397bddefa3..49e32ffbe59f 100644 --- a/drivers/usb/misc/ucsi.c +++ b/drivers/usb/misc/ucsi.c @@ -61,7 +61,7 @@ static int ucsi_acpi_cmd(struct ucsi *ucsi, struct ucsi_control *ctrl) ucsi->data->ctrl.raw_cmd = ctrl->raw_cmd; - obj = acpi_evaluate_dsm(ACPI_HANDLE(ucsi->dev), uuid.b, 1, 1, NULL); + obj = acpi_evaluate_dsm(ACPI_HANDLE(ucsi->dev), &uuid, 1, 1, NULL); if (!obj) { dev_err(ucsi->dev, "%s: failed to evaluate _DSM\n", __func__); return -EIO; diff --git a/drivers/usb/typec/typec_wcove.c b/drivers/usb/typec/typec_wcove.c index d5a7b21fa3f1..5ce93e0a15ca 100644 --- a/drivers/usb/typec/typec_wcove.c +++ b/drivers/usb/typec/typec_wcove.c @@ -118,7 +118,7 @@ static int wcove_typec_func(struct wcove_typec *wcove, tmp.type = ACPI_TYPE_INTEGER; tmp.integer.value = param; - obj = acpi_evaluate_dsm(ACPI_HANDLE(wcove->dev), uuid.b, 1, func, + obj = acpi_evaluate_dsm(ACPI_HANDLE(wcove->dev), &uuid, 1, func, &argv4); if (!obj) { dev_err(wcove->dev, "%s: failed to evaluate _DSM\n", __func__); @@ -314,7 +314,7 @@ static int wcove_typec_probe(struct platform_device *pdev) if (ret) return ret; - if (!acpi_check_dsm(ACPI_HANDLE(&pdev->dev), uuid.b, 0, 0x1f)) { + if (!acpi_check_dsm(ACPI_HANDLE(&pdev->dev), &uuid, 0, 0x1f)) { dev_err(&pdev->dev, "Missing _DSM functions\n"); return -ENODEV; } diff --git a/include/acpi/acpi_bus.h b/include/acpi/acpi_bus.h index 197f3fffc9a7..0682942d6a76 100644 --- a/include/acpi/acpi_bus.h +++ b/include/acpi/acpi_bus.h @@ -61,13 +61,14 @@ bool acpi_ata_match(acpi_handle handle); bool acpi_bay_match(acpi_handle handle); bool acpi_dock_match(acpi_handle handle); -bool acpi_check_dsm(acpi_handle handle, const u8 *uuid, u64 rev, u64 funcs); -union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, +bool acpi_check_dsm(acpi_handle handle, const uuid_le *uuid, u64 rev, u64 funcs); +union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const uuid_le *uuid, u64 rev, u64 func, union acpi_object *argv4); static inline union acpi_object * -acpi_evaluate_dsm_typed(acpi_handle handle, const u8 *uuid, u64 rev, u64 func, - union acpi_object *argv4, acpi_object_type type) +acpi_evaluate_dsm_typed(acpi_handle handle, const uuid_le *uuid, u64 rev, + u64 func, union acpi_object *argv4, + acpi_object_type type) { union acpi_object *obj; diff --git a/include/linux/acpi.h b/include/linux/acpi.h index 137e4a3d89c5..66d135003780 100644 --- a/include/linux/acpi.h +++ b/include/linux/acpi.h @@ -39,6 +39,7 @@ #include <linux/dynamic_debug.h> #include <linux/module.h> #include <linux/mutex.h> +#include <linux/uuid.h> #include <acpi/acpi_bus.h> #include <acpi/acpi_drivers.h> @@ -457,7 +458,6 @@ struct acpi_osc_context { struct acpi_buffer ret; /* free by caller if success */ }; -acpi_status acpi_str_to_uuid(char *str, u8 *uuid); acpi_status acpi_run_osc(acpi_handle handle, struct acpi_osc_context *context); /* Indexes into _OSC Capabilities Buffer (DWORDs 2 & 3 are device-specific) */ @@ -741,7 +741,7 @@ static inline bool acpi_driver_match_device(struct device *dev, } static inline union acpi_object *acpi_evaluate_dsm(acpi_handle handle, - const u8 *uuid, + const uuid_le *uuid, int rev, int func, union acpi_object *argv4) { diff --git a/include/linux/pci-acpi.h b/include/linux/pci-acpi.h index 7a4e83a8c89c..917a6ad0f24d 100644 --- a/include/linux/pci-acpi.h +++ b/include/linux/pci-acpi.h @@ -105,7 +105,7 @@ static inline void acpiphp_remove_slots(struct pci_bus *bus) { } static inline void acpiphp_check_host_bridge(struct acpi_device *adev) { } #endif -extern const u8 pci_acpi_dsm_uuid[]; +extern const uuid_le pci_acpi_dsm_uuid; #define DEVICE_LABEL_DSM 0x07 #define RESET_DELAY_DSM 0x08 #define FUNCTION_DELAY_DSM 0x09 diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index e3f06672fd6d..b95f659dced5 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -21,8 +21,9 @@ #include "skl.h" /* Unique identification for getting NHLT blobs */ -static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, - 0xA4, 0x1F, 0x7B, 0x5D, 0xCE, 0x24, 0xC5, 0x53}; +static uuid_le osc_uuid = + UUID_LE(0xA69F886E, 0x6CEB, 0x4594, + 0xA4, 0x1F, 0x7B, 0x5D, 0xCE, 0x24, 0xC5, 0x53); struct nhlt_acpi_table *skl_nhlt_init(struct device *dev) { @@ -37,7 +38,7 @@ struct nhlt_acpi_table *skl_nhlt_init(struct device *dev) return NULL; } - obj = acpi_evaluate_dsm(handle, OSC_UUID, 1, 1, NULL); + obj = acpi_evaluate_dsm(handle, &osc_uuid, 1, 1, NULL); if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; nhlt_table = (struct nhlt_acpi_table *) diff --git a/tools/testing/nvdimm/test/iomap.c b/tools/testing/nvdimm/test/iomap.c index 64cae1a5deff..f190f22c53dd 100644 --- a/tools/testing/nvdimm/test/iomap.c +++ b/tools/testing/nvdimm/test/iomap.c @@ -370,7 +370,7 @@ acpi_status __wrap_acpi_evaluate_object(acpi_handle handle, acpi_string path, } EXPORT_SYMBOL(__wrap_acpi_evaluate_object); -union acpi_object * __wrap_acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, +union acpi_object * __wrap_acpi_evaluate_dsm(acpi_handle handle, const uuid_le *uuid, u64 rev, u64 func, union acpi_object *argv4) { union acpi_object *obj = ERR_PTR(-ENXIO); diff --git a/tools/testing/nvdimm/test/nfit.c b/tools/testing/nvdimm/test/nfit.c index c2187178fb13..145f6ee0234f 100644 --- a/tools/testing/nvdimm/test/nfit.c +++ b/tools/testing/nvdimm/test/nfit.c @@ -1559,7 +1559,7 @@ static unsigned long nfit_ctl_handle; union acpi_object *result; static union acpi_object *nfit_test_evaluate_dsm(acpi_handle handle, - const u8 *uuid, u64 rev, u64 func, union acpi_object *argv4) + const uuid_le *uuid, u64 rev, u64 func, union acpi_object *argv4) { if (handle != &nfit_ctl_handle) return ERR_PTR(-ENXIO); -- 2.11.0

13 17

[alsa-devel] [PATCH] ALSA: info: Use kvzalloc() for a temporary write buffer
by Takashi Iwai 22 May '17

22 May '17

We used to use kmalloc (more exactly, krealloc()) for creating and growing the temporary buffer for text proc write. It can grow up to 16kB, and it's already a bit doubtful whether it's always safe to use kmalloc(). With the recent addition of kvmalloc(), we can have a better chance for succeed of memory allocation, so let's switch to that new API. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/info.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index 8ab72e0f5932..fc14ebe751d8 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -344,12 +344,12 @@ static ssize_t snd_info_text_entry_write(struct file *file, } } if (next > buf->len) { - char *nbuf = krealloc(buf->buffer, PAGE_ALIGN(next), - GFP_KERNEL | __GFP_ZERO); + char *nbuf = kvzalloc(PAGE_ALIGN(next), GFP_KERNEL); if (!nbuf) { err = -ENOMEM; goto error; } + kvfree(buf->buffer); buf->buffer = nbuf; buf->len = PAGE_ALIGN(next); } @@ -427,7 +427,7 @@ static int snd_info_text_entry_release(struct inode *inode, struct file *file) single_release(inode, file); kfree(data->rbuffer); if (data->wbuffer) { - kfree(data->wbuffer->buffer); + kvfree(data->wbuffer->buffer); kfree(data->wbuffer); } -- 2.13.0

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - Apply dual-codec quirk for MSI Z270-Gaming mobo
by Takashi Iwai 22 May '17

22 May '17

MSI Z270-Gamin mobo has also two ALC1220 codecs like Gigabyte AZ370- Gaming mobo. Apply the same quirk to this one. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 9c22ad694534..3fdd5af190a4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2328,6 +2328,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), + SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 2.13.0

1 2

[alsa-devel] [PATCH] ALSA: firewire: remove support for 16 bit PCM samples in playback substream
by Takashi Sakamoto 22 May '17

22 May '17

In IEC 61883-6, AM824 is described as format of data block. In this format, one data block consists of several data channels, which is aligned to 32 bit. One data channel has 8 bit label field and 24 bit data field. PCM frames are transferred in Multi Bit Linear Audio (MBLA) data channel. This channel can include 16/20/24 bit PCM sample. As long as I know, models which support IEC 61883-1/6 doesn't allow to switch bit length of PCM sample in MBLA data channel. They always transmit/receive PCM frames of 24 bit length. This can be seen for the other models which support protocols similar to IEC 61883-1/6. On the other hand, current drivers for these protocols supports 16 bit length PCM sample in playback substream. In this case, PCM sample is put into the MBLA data channel with 8 bit padding in LSB side. Although 16 bit PCM sample is major because it's in CD format, this doesn't represent device capability as is. This commit removes support for 16 bit PCM samples in playback substream. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-am824.c | 71 +------------------------------- sound/firewire/amdtp-am824.h | 6 +-- sound/firewire/bebob/bebob_pcm.c | 4 -- sound/firewire/dice/dice-pcm.c | 6 --- sound/firewire/digi00x/amdtp-dot.c | 63 +--------------------------- sound/firewire/digi00x/digi00x-pcm.c | 7 +--- sound/firewire/digi00x/digi00x.h | 1 - sound/firewire/fireworks/fireworks_pcm.c | 4 -- sound/firewire/oxfw/oxfw-pcm.c | 4 -- sound/firewire/tascam/amdtp-tascam.c | 62 +--------------------------- sound/firewire/tascam/tascam-pcm.c | 7 +--- sound/firewire/tascam/tascam.h | 1 - 12 files changed, 9 insertions(+), 227 deletions(-) diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index bebddc60fde8..23ccddb20de1 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -38,10 +38,6 @@ struct amdtp_am824 { u8 pcm_positions[AM824_MAX_CHANNELS_FOR_PCM]; u8 midi_position; - void (*transfer_samples)(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames); - unsigned int frame_multiplier; }; @@ -177,32 +173,6 @@ static void write_pcm_s32(struct amdtp_stream *s, } } -static void write_pcm_s16(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) -{ - struct amdtp_am824 *p = s->protocol; - struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; - const u16 *src; - - channels = p->pcm_channels; - src = (void *)runtime->dma_area + - frames_to_bytes(runtime, s->pcm_buffer_pointer); - remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; - - for (i = 0; i < frames; ++i) { - for (c = 0; c < channels; ++c) { - buffer[p->pcm_positions[c]] = - cpu_to_be32((*src << 8) | 0x42000000); - src++; - } - buffer += s->data_block_quadlets; - if (--remaining_frames == 0) - src = (void *)runtime->dma_area; - } -} - static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, __be32 *buffer, unsigned int frames) @@ -242,43 +212,6 @@ static void write_pcm_silence(struct amdtp_stream *s, } /** - * amdtp_am824_set_pcm_format - set the PCM format - * @s: the AMDTP stream to configure - * @format: the format of the ALSA PCM device - * - * The sample format must be set after the other parameters (rate/PCM channels/ - * MIDI) and before the stream is started, and must not be changed while the - * stream is running. - */ -void amdtp_am824_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format) -{ - struct amdtp_am824 *p = s->protocol; - - if (WARN_ON(amdtp_stream_pcm_running(s))) - return; - - switch (format) { - default: - WARN_ON(1); - /* fall through */ - case SNDRV_PCM_FORMAT_S16: - if (s->direction == AMDTP_OUT_STREAM) { - p->transfer_samples = write_pcm_s16; - break; - } - WARN_ON(1); - /* fall through */ - case SNDRV_PCM_FORMAT_S32: - if (s->direction == AMDTP_OUT_STREAM) - p->transfer_samples = write_pcm_s32; - else - p->transfer_samples = read_pcm_s32; - break; - } -} -EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_format); - -/** * amdtp_am824_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream for AM824 data block, must be initialized. * @runtime: the PCM substream runtime @@ -407,7 +340,7 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffe unsigned int pcm_frames; if (pcm) { - p->transfer_samples(s, pcm, buffer, data_blocks); + write_pcm_s32(s, pcm, buffer, data_blocks); pcm_frames = data_blocks * p->frame_multiplier; } else { write_pcm_silence(s, buffer, data_blocks); @@ -428,7 +361,7 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffe unsigned int pcm_frames; if (pcm) { - p->transfer_samples(s, pcm, buffer, data_blocks); + read_pcm_s32(s, pcm, buffer, data_blocks); pcm_frames = data_blocks * p->frame_multiplier; } else { pcm_frames = 0; diff --git a/sound/firewire/amdtp-am824.h b/sound/firewire/amdtp-am824.h index 73b07b3109db..b56e61fc997d 100644 --- a/sound/firewire/amdtp-am824.h +++ b/sound/firewire/amdtp-am824.h @@ -8,8 +8,7 @@ #define AM824_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32 -#define AM824_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \ - SNDRV_PCM_FMTBIT_S32) +#define AM824_OUT_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32 /* * This module supports maximum 64 PCM channels for one PCM stream @@ -41,9 +40,6 @@ void amdtp_am824_set_midi_position(struct amdtp_stream *s, int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime); -void amdtp_am824_set_pcm_format(struct amdtp_stream *s, - snd_pcm_format_t format); - void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port, struct snd_rawmidi_substream *midi); diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 9e27eb8e1dd4..e2f023f3cd9f 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -224,8 +224,6 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&bebob->mutex); } - amdtp_am824_set_pcm_format(&bebob->tx_stream, params_format(hw_params)); - return 0; } static int @@ -246,8 +244,6 @@ pcm_playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&bebob->mutex); } - amdtp_am824_set_pcm_format(&bebob->rx_stream, params_format(hw_params)); - return 0; } diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 6074fe1f00f7..b633805a885f 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -146,7 +146,6 @@ static int capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; - struct amdtp_stream *stream = &dice->tx_stream[substream->pcm->device]; int err; err = snd_pcm_lib_alloc_vmalloc_buffer(substream, @@ -160,15 +159,12 @@ static int capture_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&dice->mutex); } - amdtp_am824_set_pcm_format(stream, params_format(hw_params)); - return 0; } static int playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; - struct amdtp_stream *stream = &dice->rx_stream[substream->pcm->device]; int err; err = snd_pcm_lib_alloc_vmalloc_buffer(substream, @@ -182,8 +178,6 @@ static int playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&dice->mutex); } - amdtp_am824_set_pcm_format(stream, params_format(hw_params)); - return 0; } diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index a4688545339c..1453c34ce99f 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -48,10 +48,6 @@ struct amdtp_dot { struct snd_rawmidi_substream *midi[MAX_MIDI_PORTS]; int midi_fifo_used[MAX_MIDI_PORTS]; int midi_fifo_limit; - - void (*transfer_samples)(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames); }; /* @@ -173,32 +169,6 @@ static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, } } -static void write_pcm_s16(struct amdtp_stream *s, struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) -{ - struct amdtp_dot *p = s->protocol; - struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; - const u16 *src; - - channels = p->pcm_channels; - src = (void *)runtime->dma_area + - frames_to_bytes(runtime, s->pcm_buffer_pointer); - remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; - - buffer++; - for (i = 0; i < frames; ++i) { - for (c = 0; c < channels; ++c) { - buffer[c] = cpu_to_be32((*src << 8) | 0x40000000); - dot_encode_step(&p->state, &buffer[c]); - src++; - } - buffer += s->data_block_quadlets; - if (--remaining_frames == 0) - src = (void *)runtime->dma_area; - } -} - static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, __be32 *buffer, unsigned int frames) { @@ -351,33 +321,6 @@ int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s, return amdtp_stream_add_pcm_hw_constraints(s, runtime); } -void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format) -{ - struct amdtp_dot *p = s->protocol; - - if (WARN_ON(amdtp_stream_pcm_running(s))) - return; - - switch (format) { - default: - WARN_ON(1); - /* fall through */ - case SNDRV_PCM_FORMAT_S16: - if (s->direction == AMDTP_OUT_STREAM) { - p->transfer_samples = write_pcm_s16; - break; - } - WARN_ON(1); - /* fall through */ - case SNDRV_PCM_FORMAT_S32: - if (s->direction == AMDTP_OUT_STREAM) - p->transfer_samples = write_pcm_s32; - else - p->transfer_samples = read_pcm_s32; - break; - } -} - void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port, struct snd_rawmidi_substream *midi) { @@ -392,13 +335,12 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s, unsigned int data_blocks, unsigned int *syt) { - struct amdtp_dot *p = (struct amdtp_dot *)s->protocol; struct snd_pcm_substream *pcm; unsigned int pcm_frames; pcm = ACCESS_ONCE(s->pcm); if (pcm) { - p->transfer_samples(s, pcm, buffer, data_blocks); + read_pcm_s32(s, pcm, buffer, data_blocks); pcm_frames = data_blocks; } else { pcm_frames = 0; @@ -414,13 +356,12 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s, unsigned int data_blocks, unsigned int *syt) { - struct amdtp_dot *p = (struct amdtp_dot *)s->protocol; struct snd_pcm_substream *pcm; unsigned int pcm_frames; pcm = ACCESS_ONCE(s->pcm); if (pcm) { - p->transfer_samples(s, pcm, buffer, data_blocks); + write_pcm_s32(s, pcm, buffer, data_blocks); pcm_frames = data_blocks; } else { write_pcm_silence(s, buffer, data_blocks); diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 68d1c52db051..6379f67416d7 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -88,8 +88,7 @@ static int pcm_init_hw_params(struct snd_dg00x *dg00x, substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; s = &dg00x->tx_stream; } else { - substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16 | - SNDRV_PCM_FMTBIT_S32; + substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; s = &dg00x->rx_stream; } @@ -184,8 +183,6 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&dg00x->mutex); } - amdtp_dot_set_pcm_format(&dg00x->tx_stream, params_format(hw_params)); - return 0; } @@ -206,8 +203,6 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&dg00x->mutex); } - amdtp_dot_set_pcm_format(&dg00x->rx_stream, params_format(hw_params)); - return 0; } diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 1275a50956c0..4dd1bbf2ed3c 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -121,7 +121,6 @@ int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate, void amdtp_dot_reset(struct amdtp_stream *s); int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime); -void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format); void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port, struct snd_rawmidi_substream *midi); diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 9171702f7d0b..f10aec117998 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -257,8 +257,6 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&efw->mutex); } - amdtp_am824_set_pcm_format(&efw->tx_stream, params_format(hw_params)); - return 0; } static int pcm_playback_hw_params(struct snd_pcm_substream *substream, @@ -278,8 +276,6 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&efw->mutex); } - amdtp_am824_set_pcm_format(&efw->rx_stream, params_format(hw_params)); - return 0; } diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index f3530f89a025..d4594f7115ae 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -244,8 +244,6 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&oxfw->mutex); } - amdtp_am824_set_pcm_format(&oxfw->tx_stream, params_format(hw_params)); - return 0; } static int pcm_playback_hw_params(struct snd_pcm_substream *substream, @@ -265,8 +263,6 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&oxfw->mutex); } - amdtp_am824_set_pcm_format(&oxfw->rx_stream, params_format(hw_params)); - return 0; } diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c index 9dd0fccd5ccc..6aff1fc1c72d 100644 --- a/sound/firewire/tascam/amdtp-tascam.c +++ b/sound/firewire/tascam/amdtp-tascam.c @@ -14,10 +14,6 @@ struct amdtp_tscm { unsigned int pcm_channels; - - void (*transfer_samples)(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames); }; int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate) @@ -62,31 +58,6 @@ static void write_pcm_s32(struct amdtp_stream *s, } } -static void write_pcm_s16(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) -{ - struct amdtp_tscm *p = s->protocol; - struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; - const u16 *src; - - channels = p->pcm_channels; - src = (void *)runtime->dma_area + - frames_to_bytes(runtime, s->pcm_buffer_pointer); - remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; - - for (i = 0; i < frames; ++i) { - for (c = 0; c < channels; ++c) { - buffer[c] = cpu_to_be32(*src << 16); - src++; - } - buffer += s->data_block_quadlets; - if (--remaining_frames == 0) - src = (void *)runtime->dma_area; - } -} - static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, __be32 *buffer, unsigned int frames) @@ -146,44 +117,16 @@ int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s, return amdtp_stream_add_pcm_hw_constraints(s, runtime); } -void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format) -{ - struct amdtp_tscm *p = s->protocol; - - if (WARN_ON(amdtp_stream_pcm_running(s))) - return; - - switch (format) { - default: - WARN_ON(1); - /* fall through */ - case SNDRV_PCM_FORMAT_S16: - if (s->direction == AMDTP_OUT_STREAM) { - p->transfer_samples = write_pcm_s16; - break; - } - WARN_ON(1); - /* fall through */ - case SNDRV_PCM_FORMAT_S32: - if (s->direction == AMDTP_OUT_STREAM) - p->transfer_samples = write_pcm_s32; - else - p->transfer_samples = read_pcm_s32; - break; - } -} - static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer, unsigned int data_blocks, unsigned int *syt) { - struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol; struct snd_pcm_substream *pcm; pcm = ACCESS_ONCE(s->pcm); if (data_blocks > 0 && pcm) - p->transfer_samples(s, pcm, buffer, data_blocks); + read_pcm_s32(s, pcm, buffer, data_blocks); /* A place holder for control messages. */ @@ -195,7 +138,6 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s, unsigned int data_blocks, unsigned int *syt) { - struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol; struct snd_pcm_substream *pcm; /* This field is not used. */ @@ -203,7 +145,7 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s, pcm = ACCESS_ONCE(s->pcm); if (pcm) - p->transfer_samples(s, pcm, buffer, data_blocks); + write_pcm_s32(s, pcm, buffer, data_blocks); else write_pcm_silence(s, buffer, data_blocks); diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index f5dd6ce6b6f1..6207588d7c73 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -48,8 +48,7 @@ static int pcm_init_hw_params(struct snd_tscm *tscm, stream = &tscm->tx_stream; pcm_channels = tscm->spec->pcm_capture_analog_channels; } else { - runtime->hw.formats = - SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32; + runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; stream = &tscm->rx_stream; pcm_channels = tscm->spec->pcm_playback_analog_channels; } @@ -125,8 +124,6 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&tscm->mutex); } - amdtp_tscm_set_pcm_format(&tscm->tx_stream, params_format(hw_params)); - return 0; } @@ -147,8 +144,6 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&tscm->mutex); } - amdtp_tscm_set_pcm_format(&tscm->rx_stream, params_format(hw_params)); - return 0; } diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 08ecfae5c584..a5bd167eb5d9 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -131,7 +131,6 @@ int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit, int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate); int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime); -void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format); int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate); int snd_tscm_stream_get_clock(struct snd_tscm *tscm, -- 2.11.0

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[alsa-devel] [PATCH 00/16] ALSA: Convert to new copy_silence PCM ops
by Takashi Iwai 22 May '17

22 May '17

Hi, this is a part of the previous RFC patchset, and it's preliminary for eliminating set_fs() usages in the rest ALSA codes. This patchset itself converts the existing copy and silence PCM ops to a new single copy_silence ops. The new callback takes in_kernel flag for allowing in-kernel buffer copy, so that the PCM drivers can pass the buffer in kernel-space later directly without set_fs() hackery. The latest codes are found in topic/kill-set_fs branch of sound git tree. The media people are Cc'ed for solo6x10 changes. Takashi === Takashi Iwai (16): ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops .../sound/kernel-api/writing-an-alsa-driver.rst | 110 ++++++++++-------- drivers/media/pci/solo6x10/solo6x10-g723.c | 13 ++- include/sound/pcm.h | 8 +- sound/core/pcm_lib.c | 68 ++++++----- sound/drivers/dummy.c | 13 +-- sound/isa/gus/gus_pcm.c | 43 ++----- sound/isa/sb/emu8000_pcm.c | 99 +++++----------- sound/pci/es1938.c | 11 +- sound/pci/korg1212/korg1212.c | 128 ++++++--------------- sound/pci/nm256/nm256.c | 35 +++--- sound/pci/rme32.c | 49 ++++---- sound/pci/rme96.c | 52 ++++----- sound/pci/rme9652/hdsp.c | 44 ++++--- sound/pci/rme9652/rme9652.c | 46 ++++---- sound/sh/sh_dac_audio.c | 40 ++----- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 ++- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 +++----- sound/soc/soc-pcm.c | 3 +- 19 files changed, 340 insertions(+), 492 deletions(-) -- 2.13.0

3 18

[alsa-devel] HDA Intel ALC889A optical spdif input on MacBooks
by René Rebe 22 May '17

22 May '17

Hi, recently I wanted to connect my low noise mic pre-amp via optical spdif to a spare "10 year” old MacBook or MacBookPro. Those indeed have a optical input in the 3.5mm headphone jack. On macOS this also works and works as expected. On Linux, however, no matter what I do I only get zeros when recording the -D iec958. In /proc I see 3 capture sub-devices, where the 2nd is labeled: # cat /proc/asound/card0/pcm1c/info ... id: ALC889A Digital name: ALC889A Digital ... The behavior is the same on booth the MacBook as well as the MacBook Pro with either of those PCI devices: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) I have the feeling there could be some auto sensing be involved, as in the macOS system preferences I see the input switching from Mic to Optical Input as soon as I plugin in an active fibre. As I’m not familiar with this Intel HD stuff, any hint how to start analyzing and poking this mystery would be highly appreciated. Best regards, René -- ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin http://exactcode.com | http://exactscan.com | http://ocrkit.com | http://t2-project.org | http://rene.rebe.de

3 10

[alsa-devel] [PATCH] ALSA: sound/isa: constify snd_kcontrol_new structures
by Bhumika Goyal 22 May '17

22 May '17

Declare snd_kcontrol_new structures as const as they are only passed an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/isa/gus/gus_main.c | 2 +- sound/isa/gus/gus_pcm.c | 4 ++-- sound/isa/sb/sb16_csp.c | 4 ++-- sound/isa/sb/sb16_main.c | 2 +- sound/isa/sscape.c | 2 +- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 4490ee4..3cf9b13 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -82,7 +82,7 @@ static int snd_gus_joystick_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el return change; } -static struct snd_kcontrol_new snd_gus_joystick_control = { +static const struct snd_kcontrol_new snd_gus_joystick_control = { .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = "Joystick Speed", .info = snd_gus_joystick_info, diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index 0650599..33c1891 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -809,7 +809,7 @@ static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ return change; } -static struct snd_kcontrol_new snd_gf1_pcm_volume_control = +static const struct snd_kcontrol_new snd_gf1_pcm_volume_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", @@ -818,7 +818,7 @@ static struct snd_kcontrol_new snd_gf1_pcm_volume_control = .put = snd_gf1_pcm_volume_put }; -static struct snd_kcontrol_new snd_gf1_pcm_volume_control1 = +static const struct snd_kcontrol_new snd_gf1_pcm_volume_control1 = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "GPCM Playback Volume", diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 48da227..fa5780b 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -1029,7 +1029,7 @@ static int snd_sb_qsound_space_put(struct snd_kcontrol *kcontrol, struct snd_ctl return change; } -static struct snd_kcontrol_new snd_sb_qsound_switch = { +static const struct snd_kcontrol_new snd_sb_qsound_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "3D Control - Switch", .info = snd_sb_qsound_switch_info, @@ -1037,7 +1037,7 @@ static struct snd_kcontrol_new snd_sb_qsound_switch = { .put = snd_sb_qsound_switch_put }; -static struct snd_kcontrol_new snd_sb_qsound_space = { +static const struct snd_kcontrol_new snd_sb_qsound_space = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "3D Control - Space", .info = snd_sb_qsound_space_info, diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 8b2d6c6..4be1350 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -737,7 +737,7 @@ static int snd_sb16_dma_control_put(struct snd_kcontrol *kcontrol, struct snd_ct return change; } -static struct snd_kcontrol_new snd_sb16_dma_control = { +static const struct snd_kcontrol_new snd_sb16_dma_control = { .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = "16-bit DMA Allocation", .info = snd_sb16_dma_control_info, diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 54f5758a..1cd2908 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -671,7 +671,7 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, return change; } -static struct snd_kcontrol_new midi_mixer_ctl = { +static const struct snd_kcontrol_new midi_mixer_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "MIDI", .info = sscape_midi_info, -- 2.7.4

3 7

[alsa-devel] [PATCH 1/2] ALSA: pcm: Use a common helper for PCM state check and hwsync
by Takashi Iwai 21 May '17

21 May '17

The mostly same codes for checking the current PCM state and calling hwsync are found in a few places. This patch simplifies them by creating a common helper function. It also fixes a couple of cases where we missed the proper state check (e.g. PAUSED state wasn't handled in rewind and snd_pcm_hwsync()), too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 153 +++++++++++------------------------------------- 1 file changed, 35 insertions(+), 118 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f3a3580eb44c..93bd2c662c1d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2431,6 +2431,30 @@ static int snd_pcm_release(struct inode *inode, struct file *file) return 0; } +/* check and update PCM state; return 0 or a negative error + * call this inside PCM lock + */ +static int do_pcm_hwsync(struct snd_pcm_substream *substream) +{ + switch (substream->runtime->status->state) { + case SNDRV_PCM_STATE_DRAINING: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return -EBADFD; + /* Fall through */ + case SNDRV_PCM_STATE_RUNNING: + return snd_pcm_update_hw_ptr(substream); + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: + return 0; + case SNDRV_PCM_STATE_SUSPENDED: + return -ESTRPIPE; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; + default: + return -EBADFD; + } +} + static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *substream, snd_pcm_uframes_t frames) { @@ -2443,25 +2467,9 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst return 0; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { - case SNDRV_PCM_STATE_PREPARED: - break; - case SNDRV_PCM_STATE_DRAINING: - case SNDRV_PCM_STATE_RUNNING: - if (snd_pcm_update_hw_ptr(substream) >= 0) - break; - /* Fall through */ - case SNDRV_PCM_STATE_XRUN: - ret = -EPIPE; - goto __end; - case SNDRV_PCM_STATE_SUSPENDED: - ret = -ESTRPIPE; - goto __end; - default: - ret = -EBADFD; + ret = do_pcm_hwsync(substream); + if (ret < 0) goto __end; - } - hw_avail = snd_pcm_playback_hw_avail(runtime); if (hw_avail <= 0) { ret = 0; @@ -2491,25 +2499,9 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr return 0; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { - case SNDRV_PCM_STATE_PREPARED: - case SNDRV_PCM_STATE_DRAINING: - break; - case SNDRV_PCM_STATE_RUNNING: - if (snd_pcm_update_hw_ptr(substream) >= 0) - break; - /* Fall through */ - case SNDRV_PCM_STATE_XRUN: - ret = -EPIPE; + ret = do_pcm_hwsync(substream); + if (ret < 0) goto __end; - case SNDRV_PCM_STATE_SUSPENDED: - ret = -ESTRPIPE; - goto __end; - default: - ret = -EBADFD; - goto __end; - } - hw_avail = snd_pcm_capture_hw_avail(runtime); if (hw_avail <= 0) { ret = 0; @@ -2539,26 +2531,9 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs return 0; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { - case SNDRV_PCM_STATE_PREPARED: - case SNDRV_PCM_STATE_PAUSED: - break; - case SNDRV_PCM_STATE_DRAINING: - case SNDRV_PCM_STATE_RUNNING: - if (snd_pcm_update_hw_ptr(substream) >= 0) - break; - /* Fall through */ - case SNDRV_PCM_STATE_XRUN: - ret = -EPIPE; - goto __end; - case SNDRV_PCM_STATE_SUSPENDED: - ret = -ESTRPIPE; + ret = do_pcm_hwsync(substream); + if (ret < 0) goto __end; - default: - ret = -EBADFD; - goto __end; - } - avail = snd_pcm_playback_avail(runtime); if (avail <= 0) { ret = 0; @@ -2588,26 +2563,9 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst return 0; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { - case SNDRV_PCM_STATE_PREPARED: - case SNDRV_PCM_STATE_DRAINING: - case SNDRV_PCM_STATE_PAUSED: - break; - case SNDRV_PCM_STATE_RUNNING: - if (snd_pcm_update_hw_ptr(substream) >= 0) - break; - /* Fall through */ - case SNDRV_PCM_STATE_XRUN: - ret = -EPIPE; - goto __end; - case SNDRV_PCM_STATE_SUSPENDED: - ret = -ESTRPIPE; + ret = do_pcm_hwsync(substream); + if (ret < 0) goto __end; - default: - ret = -EBADFD; - goto __end; - } - avail = snd_pcm_capture_avail(runtime); if (avail <= 0) { ret = 0; @@ -2627,33 +2585,10 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst static int snd_pcm_hwsync(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; int err; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { - case SNDRV_PCM_STATE_DRAINING: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - goto __badfd; - /* Fall through */ - case SNDRV_PCM_STATE_RUNNING: - if ((err = snd_pcm_update_hw_ptr(substream)) < 0) - break; - /* Fall through */ - case SNDRV_PCM_STATE_PREPARED: - err = 0; - break; - case SNDRV_PCM_STATE_SUSPENDED: - err = -ESTRPIPE; - break; - case SNDRV_PCM_STATE_XRUN: - err = -EPIPE; - break; - default: - __badfd: - err = -EBADFD; - break; - } + err = do_pcm_hwsync(substream); snd_pcm_stream_unlock_irq(substream); return err; } @@ -2666,31 +2601,13 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t n = 0; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { - case SNDRV_PCM_STATE_DRAINING: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - goto __badfd; - /* Fall through */ - case SNDRV_PCM_STATE_RUNNING: - if ((err = snd_pcm_update_hw_ptr(substream)) < 0) - break; - /* Fall through */ - case SNDRV_PCM_STATE_PREPARED: - case SNDRV_PCM_STATE_SUSPENDED: - err = 0; + err = do_pcm_hwsync(substream); + if (!err) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) n = snd_pcm_playback_hw_avail(runtime); else n = snd_pcm_capture_avail(runtime); n += runtime->delay; - break; - case SNDRV_PCM_STATE_XRUN: - err = -EPIPE; - break; - default: - __badfd: - err = -EBADFD; - break; } snd_pcm_stream_unlock_irq(substream); if (!err) -- 2.13.0

2 3

[alsa-devel] [PATCH] ASoC: remove NULL pointer check for clk_disable_unprepare
by Masahiro Yamada 20 May '17

20 May '17

After long term efforts of fixing non-common clock implementations, clk_disable() is a no-op for a NULL pointer input, and this is now tree-wide consistent. All clock consumers can safely call clk_disable(_unprepare) without NULL pointer check. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- sound/soc/codecs/adau17x1.c | 3 +-- sound/soc/codecs/da7213.c | 3 +-- sound/soc/codecs/da7218.c | 3 +-- sound/soc/codecs/da7219-aad.c | 5 ++--- sound/soc/codecs/da7219.c | 3 +-- sound/soc/codecs/es8328.c | 3 +-- sound/soc/codecs/wm8731.c | 3 +-- sound/soc/davinci/davinci-evm.c | 3 +-- sound/soc/fsl/imx-audmux.c | 6 ++---- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 3 +-- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 ++---- sound/soc/samsung/i2s.c | 3 +-- 13 files changed, 16 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 2c1bd27..f50739b 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -977,8 +977,7 @@ void adau17x1_remove(struct device *dev) struct adau *adau = dev_get_drvdata(dev); snd_soc_unregister_codec(dev); - if (adau->mclk) - clk_disable_unprepare(adau->mclk); + clk_disable_unprepare(adau->mclk); } EXPORT_SYMBOL_GPL(adau17x1_remove); diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 6dd7578..130e16a 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1516,8 +1516,7 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, DA7213_VMID_EN | DA7213_BIAS_EN); } else { /* Remove MCLK */ - if (da7213->mclk) - clk_disable_unprepare(da7213->mclk); + clk_disable_unprepare(da7213->mclk); } break; case SND_SOC_BIAS_OFF: diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index d256ebf..eeb0bb6 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2611,8 +2611,7 @@ static int da7218_set_bias_level(struct snd_soc_codec *codec, DA7218_LDO_EN_MASK); } else { /* Remove MCLK */ - if (da7218->mclk) - clk_disable_unprepare(da7218->mclk); + clk_disable_unprepare(da7218->mclk); } break; case SND_SOC_BIAS_OFF: diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 6274d79..31518df 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -302,9 +302,8 @@ static void da7219_aad_hptest_work(struct work_struct *work) snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK, DA7219_HP_R_AMP_OE_MASK); - /* Remove MCLK, if previously enabled */ - if (da7219->mclk) - clk_disable_unprepare(da7219->mclk); + /* Remove MCLK */ + clk_disable_unprepare(da7219->mclk); mutex_unlock(&da7219->lock); snd_soc_dapm_mutex_unlock(dapm); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 9960162..307039e 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1630,8 +1630,7 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec, if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE) { /* Remove MCLK */ - if (da7219->mclk) - clk_disable_unprepare(da7219->mclk); + clk_disable_unprepare(da7219->mclk); } break; case SND_SOC_BIAS_OFF: diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index ed7cc42..848e0b3 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -812,8 +812,7 @@ static int es8328_remove(struct snd_soc_codec *codec) es8328 = snd_soc_codec_get_drvdata(codec); - if (es8328->clk) - clk_disable_unprepare(es8328->clk); + clk_disable_unprepare(es8328->clk); regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), es8328->supplies); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4f9a1eb..302ed88 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -517,8 +517,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - if (wm8731->mclk) - clk_disable_unprepare(wm8731->mclk); + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7a369e0..07a91c7 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -49,8 +49,7 @@ static void evm_shutdown(struct snd_pcm_substream *substream) struct snd_soc_card_drvdata_davinci *drvdata = snd_soc_card_get_drvdata(soc_card); - if (drvdata->mclk) - clk_disable_unprepare(drvdata->mclk); + clk_disable_unprepare(drvdata->mclk); } static int evm_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index fc57da3..a7561e3 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -79,8 +79,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) @@ -245,8 +244,7 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); return 0; } diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4a76b09..8c2a7a8 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -199,7 +199,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, 48000 * 512, SND_SOC_CLOCK_IN); if (!ret) { - if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && priv->mclk) + if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) clk_disable_unprepare(priv->mclk); } } diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 5bcde01..cb6f587 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -122,8 +122,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return ret; } - if (ctx->mclk) - clk_disable_unprepare(ctx->mclk); + clk_disable_unprepare(ctx->mclk); } return 0; diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 8a643a3..8857c08 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -294,10 +294,8 @@ static int mt8173_afe_dais_set_clks(struct mtk_base_afe *afe, static void mt8173_afe_dais_disable_clks(struct mtk_base_afe *afe, struct clk *m_ck, struct clk *b_ck) { - if (m_ck) - clk_disable_unprepare(m_ck); - if (b_ck) - clk_disable_unprepare(b_ck); + clk_disable_unprepare(m_ck); + clk_disable_unprepare(b_ck); } static int mt8173_afe_i2s_startup(struct snd_pcm_substream *substream, diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d..3a06acd 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1122,8 +1122,7 @@ static int i2s_runtime_suspend(struct device *dev) i2s->suspend_i2scon = readl(i2s->addr + I2SCON); i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); - if (i2s->op_clk) - clk_disable_unprepare(i2s->op_clk); + clk_disable_unprepare(i2s->op_clk); clk_disable_unprepare(i2s->clk); return 0; -- 2.7.4

2 3

[alsa-devel] Bug with alsamixer: ALSA lib simple_none.c:1550:(simple_add1) helem (MIXER, 'Master Playback Switch', 0, 1, 0) appears twice or more
by Rudolf Schmidt 20 May '17

20 May '17

Hi guys, I want to report a bug or some kind of problem with alsa $ alsamixer ALSA lib simple_none.c:1550:(simple_add1) helem (MIXER,'Master Playback Switch',0,1,0) appears twice or more cannot load mixer controls: Invalid argument I tried to figure out what is going on, tried a lot of things, removing packages, installing packages, removing pulseaudio etc. Nothing helped. I found a gentoo user who reported to had the same issue, upgrading to kernel 4.12 helped, in particular that patch: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=t… So I also upgraded to kernel 4.12 but in my case I have still the same problem. Here is my config: http://www.alsa-project.org/db/?f=9d35ec5d71ac60e210daf91b6c311f72fc8a98a0 Hope you can help me. -- Best Regards, Rudolf Schmidt

2 1

[alsa-devel] [PATCH v2 0/3] Add I2S driver
by olivier moysan 19 May '17

19 May '17

This patch-set handles the SPI/I2S IP on STM32 platforms. It applies to STM32 platforms implementing version 2 of SPI/I2S IP like STM32H7 series. The SPI/I2S block is a serial peripheral interface (SPI), which can also be configured to work on I2S/PCM mode. ASoC STM32 I2S driver only supports this I2S/PCM mode. The I2S/PCM interface can either support full duplex or half-duplex communication (transmitter or receiver only). The change of mode requires to stop the interface. This means that the two audio paths cannot be basically managed independently. The driver supports playback and capture through a single bidirectionnal DAI. In the first version of the driver, playback and capture paths are made mutually exclusive. A second patch introduces support of full duplex mode. In this implementation the interface is systematically configured in full-duplex mode from hardware point of view. So, when managing a single audio path, the interface is not configured as transmit or receive only, as it should be. This restriction is handled by masking underrun/overrun errors from the unused path. The I2S/PCM interface supports four audio standards: I2S Philips standard, MSB justified standard, LSB justified standard, PCM standard PCM standard is declined in two versions, PCM short and long, according frame synchronization duration. These standards can be mapped to ASoC standards as follows: - I2S: i2s - MSB justified standard: left_j - LSB justified standard: right_j - PCM short: dsp_a PCM long format is not exposed as it does not match supported ASoC standards. olivier moysan (3): dt-bindings: Document STM32 I2S bindings ASoC: stm32: Add I2S driver ASoC: stm32: Add full duplex support to i2s .../devicetree/bindings/sound/st,stm32-i2s.txt | 68 ++ sound/soc/stm/Kconfig | 2 +- sound/soc/stm/Makefile | 4 + sound/soc/stm/stm32_i2s.c | 947 +++++++++++++++++++++ 4 files changed, 1020 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 sound/soc/stm/stm32_i2s.c -- 1.9.1

2 4

[alsa-devel] [PATCH v3 0/3] Add I2S driver
by olivier moysan 19 May '17

19 May '17

This patch-set handles the SPI/I2S IP on STM32 platforms. It applies to STM32 platforms implementing version 2 of SPI/I2S IP like STM32H7 series. The SPI/I2S block is a serial peripheral interface (SPI), which can also be configured to work on I2S/PCM mode. ASoC STM32 I2S driver only supports this I2S/PCM mode. The I2S/PCM interface can either support full duplex or half-duplex communication (transmitter or receiver only). The change of mode requires to stop the interface. This means that the two audio paths cannot be basically managed independently. The driver supports playback and capture through a single bidirectionnal DAI. In the first version of the driver, playback and capture paths are made mutually exclusive. A second patch introduces support of full duplex mode. In this implementation the interface is systematically configured in full-duplex mode from hardware point of view. So, when managing a single audio path, the interface is not configured as transmit or receive only, as it should be. This restriction is handled by masking underrun/overrun errors from the unused path. The I2S/PCM interface supports four audio standards: I2S Philips standard, MSB justified standard, LSB justified standard, PCM standard PCM standard is declined in two versions, PCM short and long, according frame synchronization duration. These standards can be mapped to ASoC standards as follows: - I2S: i2s - MSB justified standard: left_j - LSB justified standard: right_j - PCM short: dsp_a PCM long format is not exposed as it does not match supported ASoC standards. v2 -> v3 - fix spinlock handling olivier moysan (3): dt-bindings: Document STM32 I2S bindings ASoC: stm32: Add I2S driver ASoC: stm32: Add full duplex support to i2s .../devicetree/bindings/sound/st,stm32-i2s.txt | 68 ++ sound/soc/stm/Kconfig | 2 +- sound/soc/stm/Makefile | 4 + sound/soc/stm/stm32_i2s.c | 946 +++++++++++++++++++++ 4 files changed, 1019 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 sound/soc/stm/stm32_i2s.c -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: rsnd: HDMI sound support
by Kuninori Morimoto 19 May '17

19 May '17

Hi Mark These supports HDMI sound on Renesas R-Car driver. These are based on of-graph base sound card. Kuninori Morimoto (3): ASoC: rsnd: add support graph base DT phase 1 ASoC: rsnd: add support graph base DT phase 2 ASoC: rsnd: add HDMI output support sound/soc/sh/Kconfig | 2 +- sound/soc/sh/rcar/core.c | 195 +++++++++++++++++++++++++++++------------------ sound/soc/sh/rcar/gen.c | 2 + sound/soc/sh/rcar/rsnd.h | 11 ++- sound/soc/sh/rcar/ssi.c | 58 ++++++++++++++ sound/soc/sh/rcar/ssiu.c | 37 +++++++++ 6 files changed, 230 insertions(+), 75 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2 1/2] ASoC: add es8316 codec driver
by Daniel Drake 19 May '17

19 May '17

Add a codec driver for the Everest ES8316, based on code provided by David Yang from Everest Semi. I limited the functionality to items where the vendor code was clear, and things that can be tested on the Weibu F3C (Intel Cherry Trail). As a result the initial implementation only supports running in slave mode at single speed (up to 48kHz sample rate) using I2S. HPD is not supported. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> [drake(a)endlessm.com: significant cleanups and simplifications, remove dead/unclear code] Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- Notes: v2: - Add sign-off from David Yang - Only support whole number rates calculated with mclk ratios sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8316.c | 646 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8316.h | 122 +++++++++ 4 files changed, 774 insertions(+) create mode 100644 sound/soc/codecs/es8316.c create mode 100644 sound/soc/codecs/es8316.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883ed4c8a551..c6286e5ba511 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 select SND_SOC_DMIC + select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 @@ -543,6 +544,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC" +config SND_SOC_ES8316 + tristate "Everest Semi ES8316 CODEC" + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28a63fdaf982..e878306ce46e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -300,6 +301,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c new file mode 100644 index 000000000000..35fe32070285 --- /dev/null +++ b/sound/soc/codecs/es8316.c @@ -0,0 +1,646 @@ +/* + * es8316.c -- es8316 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com>, + * Daniel Drake <drake(a)endlessm.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/acpi.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/regmap.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8316.h" + +/* In slave mode at single speed, the codec is documented as accepting 5 + * MCLK/LRCK ratios, but we also add ratio 400, which is commonly used on + * Intel Cherry Trail platforms (19.2MHz MCLK, 48kHz LRCK). + */ +#define NR_SUPPORTED_MCLK_LRCK_RATIOS 6 +static const unsigned int supported_mclk_lrck_ratios[] = { + 256, 384, 400, 512, 768, 1024 +}; + +struct es8316_priv { + unsigned int sysclk; + unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS]; + struct snd_pcm_hw_constraint_list sysclk_constraints; +}; + +/* + * ES8316 controls + */ +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const DECLARE_TLV_DB_SCALE(mic_bst_tlv, 0, 1200, 0); + +static unsigned int linin_pga_tlv[] = { + TLV_DB_RANGE_HEAD(12), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(300, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(600, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(900, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(1200, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(1500, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(1800, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(2100, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(2400, 0, 0), +}; +static unsigned int hpout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 3, TLV_DB_SCALE_ITEM(-4800, 1200, 0), +}; +static const char * const alc_func_txt[] = { "Off", "On" }; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(ES8316_ADC_ALC1, 6, 2, alc_func_txt); + +static const char * const ng_type_txt[] = + { "Constant PGA Gain", "Mute ADC Output"}; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(ES8316_ADC_ALC6, 6, 2, ng_type_txt); + +static const char * const adcpol_txt[] = { "Normal", "Invert" }; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(ES8316_ADC_MUTE, 1, 2, adcpol_txt); +static const char *const dacpol_txt[] = + { "Normal", "R Invert", "L Invert", "L + R Invert" }; +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8316_DAC_SET1, 0, 4, dacpol_txt); + +static const struct snd_kcontrol_new es8316_snd_controls[] = { + SOC_DOUBLE_TLV("HP Playback Volume", ES8316_CPHP_ICAL_VOL, + 4, 0, 0, 1, hpout_vol_tlv), + SOC_DOUBLE_TLV("HPMixer Gain", ES8316_HPMIX_VOL, + 0, 4, 7, 0, hpmixer_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, + ES8316_DAC_VOLR, 0, 0xC0, 1, dac_vol_tlv), + + SOC_SINGLE("Enable DAC Soft Ramp", ES8316_DAC_SET1, 4, 1, 1), + SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0), + + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE("DAC Notch Filter", ES8316_DAC_SET2, 6, 1, 0), + SOC_SINGLE("DAC Double Fs Mode", ES8316_DAC_SET2, 7, 1, 0), + SOC_SINGLE("DAC Volume Control-LeR", ES8316_DAC_SET2, 2, 1, 0), + SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0), + + /* +20dB D2SE PGA Control */ + SOC_SINGLE_TLV("MIC Boost", ES8316_ADC_D2SEPGA, + 0, 1, 0, mic_bst_tlv), + + /* 0-+24dB Lineinput PGA Control */ + SOC_SINGLE_TLV("Input PGA", ES8316_ADC_PGAGAIN, + 4, 8, 0, linin_pga_tlv), + + SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME, + 0, 0xc0, 1, adc_vol_tlv), + SOC_SINGLE("ADC Soft Ramp", ES8316_ADC_MUTE, 4, 1, 0), + SOC_ENUM("Capture Polarity", adcpol), + SOC_SINGLE("ADC Double FS Mode", ES8316_ADC_DMIC, 4, 1, 0), + + SOC_SINGLE("ALC Capture Target Volume", + ES8316_ADC_ALC3, 4, 10, 0), + SOC_SINGLE("ALC Capture Max PGA", ES8316_ADC_ALC1, 0, 28, 0), + SOC_SINGLE("ALC Capture Min PGA", ES8316_ADC_ALC2, 0, 28, 0), + SOC_ENUM("ALC Capture Function", alc_func), + SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0), + SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack Time", ES8316_ADC_ALC4, 0, 10, 0), + SOC_SINGLE("ALC Capture NG Threshold", ES8316_ADC_ALC6, 0, 31, 0), + SOC_ENUM("ALC Capture NG Type", ng_type), + SOC_SINGLE("ALC Capture NG Switch", ES8316_ADC_ALC6, 5, 1, 0), +}; + +/* Analog Input Mux */ +static const char * const es8316_analog_in_txt[] = { + "lin1-rin1", + "lin2-rin2", + "lin1-rin1 with 20db Boost", + "lin2-rin2 with 20db Boost" +}; +static const unsigned int es8316_analog_in_values[] = { 0, 1, 2, 3 }; +static const struct soc_enum es8316_analog_input_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_PDN_LINSEL, 4, 3, + ARRAY_SIZE(es8316_analog_in_txt), + es8316_analog_in_txt, + es8316_analog_in_values); +static const struct snd_kcontrol_new es8316_analog_in_mux_controls = + SOC_DAPM_ENUM("Route", es8316_analog_input_enum); + +static const char * const es8316_dmic_txt[] = { + "dmic disable", + "dmic data at high level", + "dmic data at low level", +}; +static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const struct soc_enum es8316_dmic_src_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, + ARRAY_SIZE(es8316_dmic_txt), + es8316_dmic_txt, + es8316_dmic_values); +static const struct snd_kcontrol_new es8316_dmic_src_controls = + SOC_DAPM_ENUM("Route", es8316_dmic_src_enum); + +/* hp mixer mux */ +static const char * const es8316_hpmux_texts[] = { + "lin1-rin1", + "lin2-rin2", + "lin-rin with Boost", + "lin-rin with Boost and PGA" +}; + +static const unsigned int es8316_hpmux_values[] = { 0, 1, 2, 3 }; + +static SOC_ENUM_SINGLE_DECL(es8316_left_hpmux_enum, ES8316_HPMIX_SEL, + 4, es8316_hpmux_texts); + +static const struct snd_kcontrol_new es8316_left_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_left_hpmux_enum); + +static SOC_ENUM_SINGLE_DECL(es8316_right_hpmux_enum, ES8316_HPMIX_SEL, + 0, es8316_hpmux_texts); + +static const struct snd_kcontrol_new es8316_right_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_right_hpmux_enum); + +/* headphone Output Mixer */ +static const struct snd_kcontrol_new es8316_out_left_mix[] = { + SOC_DAPM_SINGLE("LLIN Switch", ES8316_HPMIX_SWITCH, 6, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", ES8316_HPMIX_SWITCH, 7, 1, 0), +}; +static const struct snd_kcontrol_new es8316_out_right_mix[] = { + SOC_DAPM_SINGLE("RLIN Switch", ES8316_HPMIX_SWITCH, 2, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", ES8316_HPMIX_SWITCH, 3, 1, 0), +}; + +/* DAC data source mux */ +static const char * const es8316_dacsrc_texts[] = { + "LDATA TO LDAC, RDATA TO RDAC", + "LDATA TO LDAC, LDATA TO RDAC", + "RDATA TO LDAC, RDATA TO RDAC", + "RDATA TO LDAC, LDATA TO RDAC", +}; + +static const unsigned int es8316_dacsrc_values[] = { 0, 1, 2, 3 }; + +static SOC_ENUM_SINGLE_DECL(es8316_dacsrc_mux_enum, ES8316_DAC_SET1, + 6, es8316_dacsrc_texts); + +static const struct snd_kcontrol_new es8316_dacsrc_mux_controls = + SOC_DAPM_ENUM("Route", es8316_dacsrc_mux_enum); + +static const struct snd_soc_dapm_widget es8316_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_MICBIAS("micbias", SND_SOC_NOPM, 0, 0), + + /* Input Mux */ + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8316_analog_in_mux_controls), + + SND_SOC_DAPM_PGA("Line input PGA", ES8316_ADC_PDN_LINSEL, + 7, 1, NULL, 0), + SND_SOC_DAPM_ADC("Mono ADC", NULL, ES8316_ADC_PDN_LINSEL, 6, 1), + SND_SOC_DAPM_MUX("Digital Mic Mux", SND_SOC_NOPM, 0, 0, + &es8316_dmic_src_controls), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 1, + ES8316_SERDATA_ADC, 6, 1), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC SRC Mux", SND_SOC_NOPM, 0, 0, + &es8316_dacsrc_mux_controls), + + SND_SOC_DAPM_DAC("Right DAC", NULL, SND_SOC_NOPM, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, SND_SOC_NOPM, 4, 1), + + /* Headphone Output Side */ + SND_SOC_DAPM_MUX("Left Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_left_hpmux_controls), + SND_SOC_DAPM_MUX("Right Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_right_hpmux_controls), + SND_SOC_DAPM_MIXER("Left Hp mixer", SND_SOC_NOPM, + 4, 1, &es8316_out_left_mix[0], + ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Hp mixer", SND_SOC_NOPM, + 0, 1, &es8316_out_right_mix[0], + ARRAY_SIZE(es8316_out_right_mix)), + + /* Output charge pump */ + SND_SOC_DAPM_PGA("HPCP L", SND_SOC_NOPM, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPCP R", SND_SOC_NOPM, 2, 0, NULL, 0), + + /* Output Driver */ + SND_SOC_DAPM_PGA("HPVOL L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPVOL R", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8316_dapm_routes[] = { + /* Recording */ + {"MIC1", NULL, "micbias"}, + {"MIC2", NULL, "micbias"}, + + {"Differential Mux", "lin1-rin1", "MIC1"}, + {"Differential Mux", "lin2-rin2", "MIC2"}, + {"Line input PGA", NULL, "Differential Mux"}, + + {"Mono ADC", NULL, "Line input PGA"}, + + {"Digital Mic Mux", "dmic disable", "Mono ADC"}, + + {"I2S OUT", NULL, "Digital Mic Mux"}, + + /* Playback */ + {"DAC SRC Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + + {"Left DAC", NULL, "DAC SRC Mux"}, + {"Right DAC", NULL, "DAC SRC Mux"}, + + {"Left Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + {"Right Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Left Hp mixer", "LLIN Switch", "Left Hp mux"}, + {"Left Hp mixer", "Left DAC Switch", "Left DAC"}, + + {"Right Hp mixer", "RLIN Switch", "Right Hp mux"}, + {"Right Hp mixer", "Right DAC Switch", "Right DAC"}, + + {"HPCP L", NULL, "Left Hp mixer"}, + {"HPCP R", NULL, "Right Hp mixer"}, + + {"HPVOL L", NULL, "HPCP L"}, + {"HPVOL R", NULL, "HPCP R"}, + + {"HPOL", NULL, "HPVOL L"}, + {"HPOR", NULL, "HPVOL R"}, +}; + +static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + int i; + int count = 0; + + es8316->sysclk = freq; + + if (freq == 0) + return 0; + + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (freq % ratio == 0) + es8316->allowed_rates[count++] = freq / ratio; + } + + es8316->sysclk_constraints.list = es8316->allowed_rates; + es8316->sysclk_constraints.count = count; + + return 0; +} + +static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 serdata1 = 0; + u8 serdata2 = 0; + u8 mask; + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Codec driver only supports slave mode\n"); + return -EINVAL; + } + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) { + dev_err(codec->dev, "Codec driver only supports I2S format\n"); + return -EINVAL; + } + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + serdata1 |= ES8316_SERDATA1_BCLK_INV; + serdata2 |= ES8316_SERDATA2_ADCLRP; + break; + case SND_SOC_DAIFMT_IB_NF: + serdata1 |= ES8316_SERDATA1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + serdata2 |= ES8316_SERDATA2_ADCLRP; + break; + default: + return -EINVAL; + } + + mask = ES8316_SERDATA1_MASTER | ES8316_SERDATA1_BCLK_INV; + snd_soc_update_bits(codec, ES8316_SERDATA1, mask, serdata1); + + mask = ES8316_SERDATA2_FMT_MASK | ES8316_SERDATA2_ADCLRP; + snd_soc_update_bits(codec, ES8316_SERDATA_ADC, mask, serdata2); + snd_soc_update_bits(codec, ES8316_SERDATA_DAC, mask, serdata2); + + return 0; +} + +static int es8316_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + + if (es8316->sysclk == 0) { + dev_err(codec->dev, "No sysclk provided\n"); + return -EINVAL; + } + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC. + */ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &es8316->sysclk_constraints); + + return 0; +} + +static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + u8 wordlen = 0; + + if (!es8316->sysclk) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wordlen = ES8316_SERDATA2_LEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wordlen = ES8316_SERDATA2_LEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wordlen = ES8316_SERDATA2_LEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wordlen = ES8316_SERDATA2_LEN_32; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ES8316_SERDATA_DAC, + ES8316_SERDATA2_LEN_MASK, wordlen); + snd_soc_update_bits(codec, ES8316_SERDATA_ADC, + ES8316_SERDATA2_LEN_MASK, wordlen); + return 0; +} + +static int es8316_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ES8316_DAC_SET1, 0x20, + mute ? 0x20 : 0); + return 0; +} + +static int es8316_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_write(codec, ES8316_RESET, 0xC0); + snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x66); + snd_soc_write(codec, ES8316_DAC_PDN, 0x00); + snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x30); + snd_soc_write(codec, ES8316_CPHP_PDN2, 0x10); + snd_soc_write(codec, ES8316_CPHP_PDN1, 0x03); + snd_soc_write(codec, ES8316_HPMIX_PDN, 0x00); + snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0x00); + snd_soc_write(codec, ES8316_SYS_PDN, 0x00); + snd_soc_write(codec, ES8316_SYS_LP1, 0x04); + snd_soc_write(codec, ES8316_SYS_LP2, 0x0c); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x00); + snd_soc_write(codec, ES8316_DAC_PDN, 0x11); + snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x03); + snd_soc_write(codec, ES8316_CPHP_PDN2, 0x22); + snd_soc_write(codec, ES8316_CPHP_PDN1, 0x06); + snd_soc_write(codec, ES8316_HPMIX_PDN, 0x33); + snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0xc0); + snd_soc_write(codec, ES8316_SYS_PDN, 0x3f); + snd_soc_write(codec, ES8316_SYS_LP1, 0x3f); + snd_soc_write(codec, ES8316_SYS_LP2, 0x1f); + snd_soc_write(codec, ES8316_RESET, 0x00); + break; + } + + return 0; +} + +#define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops es8316_ops = { + .startup = es8316_pcm_startup, + .hw_params = es8316_pcm_hw_params, + .set_fmt = es8316_set_dai_fmt, + .set_sysclk = es8316_set_dai_sysclk, + .digital_mute = es8316_mute, +}; + +static struct snd_soc_dai_driver es8316_dai = { + .name = "ES8316 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ES8316_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ES8316_FORMATS, + }, + .ops = &es8316_ops, + .symmetric_rates = 1, +}; + +static int es8316_probe(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ES8316_RESET, 0x3f); + usleep_range(5000, 5500); + snd_soc_write(codec, ES8316_RESET, 0x00); + snd_soc_write(codec, ES8316_SYS_VMIDSEL, 0xFF); + msleep(30); + + snd_soc_write(codec, ES8316_CLKMGR_CLKSEL, 0x08); + snd_soc_write(codec, ES8316_CLKMGR_ADCOSR, 0x32); + snd_soc_write(codec, ES8316_CLKMGR_ADCDIV1, 0x11); + snd_soc_write(codec, ES8316_CLKMGR_ADCDIV2, 0x00); + snd_soc_write(codec, ES8316_CLKMGR_DACDIV1, 0x11); + snd_soc_write(codec, ES8316_CLKMGR_DACDIV2, 0x00); + snd_soc_write(codec, ES8316_CLKMGR_CPDIV, 0x00); + snd_soc_write(codec, ES8316_SERDATA1, 0x04); + snd_soc_write(codec, ES8316_CLKMGR_CLKSW, 0x7f); + snd_soc_write(codec, ES8316_CAL_TYPE, 0x0F); + snd_soc_write(codec, ES8316_CAL_HPLIV, 0x90); + snd_soc_write(codec, ES8316_CAL_HPRIV, 0x90); + snd_soc_write(codec, ES8316_ADC_VOLUME, 0x00); + snd_soc_write(codec, ES8316_ADC_D2SEPGA, 0x00); + snd_soc_write(codec, ES8316_ADC_DMIC, 0x08); + snd_soc_write(codec, ES8316_DAC_SET2, 0x00); + snd_soc_write(codec, ES8316_DAC_SET3, 0x00); + snd_soc_write(codec, ES8316_DAC_VOLL, 0x02); + snd_soc_write(codec, ES8316_DAC_VOLR, 0x02); + snd_soc_write(codec, ES8316_SERDATA_ADC, 0x00); + snd_soc_write(codec, ES8316_SERDATA_DAC, 0x00); + snd_soc_write(codec, ES8316_SYS_VMIDLOW, 0x11); + snd_soc_write(codec, ES8316_SYS_VSEL, 0xfc); + snd_soc_write(codec, ES8316_SYS_REF, 0x28); + snd_soc_write(codec, ES8316_HPMIX_SEL, 0x00); + snd_soc_write(codec, ES8316_HPMIX_SWITCH, 0x88); + snd_soc_write(codec, ES8316_HPMIX_VOL, 0x88); + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL, 0x00); + snd_soc_write(codec, ES8316_GPIO_SEL, 0x00); + snd_soc_write(codec, ES8316_GPIO_DEBUNCE_INT, 0x02); + snd_soc_write(codec, ES8316_TESTMODE, 0xa0); + snd_soc_write(codec, ES8316_TEST1, 0x00); + snd_soc_write(codec, ES8316_TEST2, 0x04); + snd_soc_write(codec, ES8316_RESET, 0xc0); + msleep(50); + snd_soc_write(codec, ES8316_ADC_PGAGAIN, 0x60); + snd_soc_write(codec, ES8316_ADC_D2SEPGA, 0x01); + + /* ADC DS mode, HPF enable */ + snd_soc_write(codec, ES8316_ADC_DMIC, 0x08); + snd_soc_write(codec, ES8316_ADC_ALC1, 0xcd); + snd_soc_write(codec, ES8316_ADC_ALC2, 0x08); + snd_soc_write(codec, ES8316_ADC_ALC3, 0xa0); + snd_soc_write(codec, ES8316_ADC_ALC4, 0x05); + snd_soc_write(codec, ES8316_ADC_ALC5, 0x06); + snd_soc_write(codec, ES8316_ADC_ALC6, 0x61); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_es8316 = { + .probe = es8316_probe, + .set_bias_level = es8316_set_bias_level, + .idle_bias_off = true, + + .component_driver = { + .controls = es8316_snd_controls, + .num_controls = ARRAY_SIZE(es8316_snd_controls), + .dapm_widgets = es8316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8316_dapm_widgets), + .dapm_routes = es8316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes), + }, +}; + +static const struct regmap_config es8316_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x53, + .cache_type = REGCACHE_RBTREE, +}; + +static int es8316_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct es8316_priv *es8316; + struct regmap *regmap; + + es8316 = devm_kzalloc(&i2c_client->dev, sizeof(struct es8316_priv), + GFP_KERNEL); + if (es8316 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, es8316); + + regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_es8316, + &es8316_dai, 1); +} + +static int es8316_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id es8316_i2c_id[] = { + {"ESSX8316", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8316_i2c_id); + +static const struct of_device_id es8316_of_match[] = { + { .compatible = "everest,es8316", }, + {}, +}; +MODULE_DEVICE_TABLE(of, es8316_of_match); + +static const struct acpi_device_id es8316_acpi_match[] = { + {"ESSX8316", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8316_acpi_match); + +static struct i2c_driver es8316_i2c_driver = { + .driver = { + .name = "es8316", + .acpi_match_table = ACPI_PTR(es8316_acpi_match), + .of_match_table = of_match_ptr(es8316_of_match), + }, + .probe = es8316_i2c_probe, + .remove = es8316_i2c_remove, + .id_table = es8316_i2c_id, +}; +module_i2c_driver(es8316_i2c_driver); + +MODULE_DESCRIPTION("Everest Semi ES8316 ALSA SoC Codec Driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h new file mode 100644 index 000000000000..702bebc69ca8 --- /dev/null +++ b/sound/soc/codecs/es8316.h @@ -0,0 +1,122 @@ +/* + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang <yangxiaohua(a)everest-semi.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef _ES8316_H +#define _ES8316_H + +/* + * ES8316 register space + */ + +/* Reset Control */ +#define ES8316_RESET 0x00 + +/* Clock Management */ +#define ES8316_CLKMGR_CLKSW 0x01 +#define ES8316_CLKMGR_CLKSEL 0x02 +#define ES8316_CLKMGR_ADCOSR 0x03 +#define ES8316_CLKMGR_ADCDIV1 0x04 +#define ES8316_CLKMGR_ADCDIV2 0x05 +#define ES8316_CLKMGR_DACDIV1 0x06 +#define ES8316_CLKMGR_DACDIV2 0x07 +#define ES8316_CLKMGR_CPDIV 0x08 + +/* Serial Data Port Control */ +#define ES8316_SERDATA1 0x09 +#define ES8316_SERDATA_ADC 0x0a +#define ES8316_SERDATA_DAC 0x0b + +/* System Control */ +#define ES8316_SYS_VMIDSEL 0x0c +#define ES8316_SYS_PDN 0x0d +#define ES8316_SYS_LP1 0x0e +#define ES8316_SYS_LP2 0x0f +#define ES8316_SYS_VMIDLOW 0x10 +#define ES8316_SYS_VSEL 0x11 +#define ES8316_SYS_REF 0x12 + +/* Headphone Mixer */ +#define ES8316_HPMIX_SEL 0x13 +#define ES8316_HPMIX_SWITCH 0x14 +#define ES8316_HPMIX_PDN 0x15 +#define ES8316_HPMIX_VOL 0x16 + +/* Charge Pump Headphone driver */ +#define ES8316_CPHP_OUTEN 0x17 +#define ES8316_CPHP_ICAL_VOL 0x18 +#define ES8316_CPHP_PDN1 0x19 +#define ES8316_CPHP_PDN2 0x1a +#define ES8316_CPHP_LDOCTL 0x1b + +/* Calibration */ +#define ES8316_CAL_TYPE 0x1c +#define ES8316_CAL_SET 0x1d +#define ES8316_CAL_HPLIV 0x1e +#define ES8316_CAL_HPRIV 0x1f +#define ES8316_CAL_HPLMV 0x20 +#define ES8316_CAL_HPRMV 0x21 + +/* ADC Control */ +#define ES8316_ADC_PDN_LINSEL 0x22 +#define ES8316_ADC_PGAGAIN 0x23 +#define ES8316_ADC_D2SEPGA 0x24 +#define ES8316_ADC_DMIC 0x25 +#define ES8316_ADC_MUTE 0x26 +#define ES8316_ADC_VOLUME 0x27 +#define ES8316_ADC_ALC1 0x29 +#define ES8316_ADC_ALC2 0x2a +#define ES8316_ADC_ALC3 0x2b +#define ES8316_ADC_ALC4 0x2c +#define ES8316_ADC_ALC5 0x2d +#define ES8316_ADC_ALC6 0x2e + +/* DAC Control */ +#define ES8316_DAC_PDN 0x2f +#define ES8316_DAC_SET1 0x30 +#define ES8316_DAC_SET2 0x31 +#define ES8316_DAC_SET3 0x32 +#define ES8316_DAC_VOLL 0x33 +#define ES8316_DAC_VOLR 0x34 + +/* GPIO */ +#define ES8316_GPIO_SEL 0x4d +#define ES8316_GPIO_DEBUNCE_INT 0x4e +#define ES8316_GPIO_FLAG 0x4f + +/* Test mode */ +#define ES8316_TESTMODE 0x50 +#define ES8316_TEST1 0x51 +#define ES8316_TEST2 0x52 +#define ES8316_TEST3 0x53 + +/* + * Field definitions + */ + +/* ES8316_SERDATA1 */ +#define ES8316_SERDATA1_MASTER 0x80 +#define ES8316_SERDATA1_BCLK_INV 0x20 + +/* ES8316_SERDATA_ADC and _DAC */ +#define ES8316_SERDATA2_FMT_MASK 0x3 +#define ES8316_SERDATA2_FMT_I2S 0x00 +#define ES8316_SERDATA2_FMT_LEFTJ 0x01 +#define ES8316_SERDATA2_FMT_RIGHTJ 0x02 +#define ES8316_SERDATA2_FMT_PCM 0x03 +#define ES8316_SERDATA2_ADCLRP 0x20 +#define ES8316_SERDATA2_LEN_MASK 0x1c +#define ES8316_SERDATA2_LEN_24 0x00 +#define ES8316_SERDATA2_LEN_20 0x04 +#define ES8316_SERDATA2_LEN_18 0x08 +#define ES8316_SERDATA2_LEN_16 0x0c +#define ES8316_SERDATA2_LEN_32 0x10 + +#endif -- 2.11.0

2 2

[alsa-devel] [PATCH] ASoC: da7213: Update driver to use device_property* FW functions
by Adam Thomson 19 May '17

19 May '17

The driver now supports ACPI based initialisation as well as DT and old pdata methods. However the FW data handling still uses DT specific calls to read firmware data (of_property*) so for ACPI based initialisation the FW data will only be set to default values. This patch updates the FW handling to use device_property* calls instead so that both ACPI and DT are handled as expected. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7213.c | 37 +++++++++++++++++++------------------ 1 file changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 024d83f..c3e1189 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -13,6 +13,8 @@ */ #include <linux/acpi.h> +#include <linux/of_device.h> +#include <linux/property.h> #include <linux/clk.h> #include <linux/delay.h> #include <linux/i2c.h> @@ -1606,12 +1608,12 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, } static struct da7213_platform_data - *da7213_of_to_pdata(struct snd_soc_codec *codec) + *da7213_fw_to_pdata(struct snd_soc_codec *codec) { - struct device_node *np = codec->dev->of_node; + struct device *dev = codec->dev; struct da7213_platform_data *pdata; - const char *of_str; - u32 of_val32; + const char *fw_str; + u32 fw_val32; pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); if (!pdata) { @@ -1619,29 +1621,29 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, return NULL; } - if (of_property_read_u32(np, "dlg,micbias1-lvl", &of_val32) >= 0) - pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, of_val32); + if (device_property_read_u32(dev, "dlg,micbias1-lvl", &fw_val32) >= 0) + pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, fw_val32); else pdata->micbias1_lvl = DA7213_MICBIAS_2_2V; - if (of_property_read_u32(np, "dlg,micbias2-lvl", &of_val32) >= 0) - pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, of_val32); + if (device_property_read_u32(dev, "dlg,micbias2-lvl", &fw_val32) >= 0) + pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, fw_val32); else pdata->micbias2_lvl = DA7213_MICBIAS_2_2V; - if (!of_property_read_string(np, "dlg,dmic-data-sel", &of_str)) - pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, of_str); + if (!device_property_read_string(dev, "dlg,dmic-data-sel", &fw_str)) + pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, fw_str); else pdata->dmic_data_sel = DA7213_DMIC_DATA_LRISE_RFALL; - if (!of_property_read_string(np, "dlg,dmic-samplephase", &of_str)) + if (!device_property_read_string(dev, "dlg,dmic-samplephase", &fw_str)) pdata->dmic_samplephase = - da7213_of_dmic_samplephase(codec, of_str); + da7213_of_dmic_samplephase(codec, fw_str); else pdata->dmic_samplephase = DA7213_DMIC_SAMPLE_ON_CLKEDGE; - if (of_property_read_u32(np, "dlg,dmic-clkrate", &of_val32) >= 0) - pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, of_val32); + if (device_property_read_u32(dev, "dlg,dmic-clkrate", &fw_val32) >= 0) + pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, fw_val32); else pdata->dmic_clk_rate = DA7213_DMIC_CLK_3_0MHZ; @@ -1713,10 +1715,9 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE); /* Handle DT/Platform data */ - if (codec->dev->of_node) - da7213->pdata = da7213_of_to_pdata(codec); - else - da7213->pdata = dev_get_platdata(codec->dev); + da7213->pdata = dev_get_platdata(codec->dev); + if (!da7213->pdata) + da7213->pdata = da7213_fw_to_pdata(codec); /* Set platform data values */ if (da7213->pdata) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: fix spelling mistake: "missmatch" -> "mismatch"
by Colin King 19 May '17

19 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/generic/audio-graph-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 07e010d38596..e0c26380921e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -107,7 +107,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, int ret; if (rcpu_ep != cpu_ep) { - dev_err(dev, "remote-endpoint missmatch (%s/%s/%s)\n", + dev_err(dev, "remote-endpoint mismatch (%s/%s/%s)\n", cpu_ep->name, codec_ep->name, rcpu_ep->name); ret = -EINVAL; goto dai_link_of_err; -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: simple-card: fix mic jack initialization
by Stefan Agner 19 May '17

19 May '17

Initialize asoc_simple_card_init_mic with the correct struct asoc_simple_jack. Signed-off-by: Stefan Agner <stefan(a)agner.ch> Fixes: 9eac361877b3 ("ASoC: simple-card: add new asoc_simple_jack and use it") --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2c9dedab5184..bc136d2bd7cd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -202,7 +202,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = asoc_simple_card_init_mic(rtd->card, &priv->hp_jack, PREFIX); + ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX); if (ret < 0) return ret; -- 2.13.0

3 2

[alsa-devel] [PATCH] ASoC: max98927: Adding support for TDM mode and Envelop tracking control
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes * Added code to support TDM mode. * Added "EnvTrack Switch" and "EnvTrack Headroom" controls. * Added six more registers into volatile register list. * Added missing \n to end of dev_err messages. * Removed an obsolete variable from 'max98927_priv'. sound/soc/codecs/max98927.c | 173 ++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98927.h | 17 +++-- 2 files changed, 152 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b5ee294..9b864e2 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -159,7 +160,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) mode = MAX98927_PCM_MASTER_MODE_MASTER; break; default: - dev_err(codec->dev, "DAI clock mode unsupported"); + dev_err(codec->dev, "DAI clock mode unsupported\n"); return -EINVAL; } @@ -175,7 +176,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) invert = MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE; break; default: - dev_err(codec->dev, "DAI invert mode unsupported"); + dev_err(codec->dev, "DAI invert mode unsupported\n"); return -EINVAL; } @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +236,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + + } return 0; } @@ -245,6 +252,21 @@ static const int rate_table[] = { 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,19 +292,17 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: + if ((max98927->iface == SND_SOC_DAIFMT_DSP_A) || + (max98927->iface == SND_SOC_DAIFMT_DSP_B)) + return 0; + + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "BCLK %d not supported\n", blr_clk_ratio); return -EINVAL; } + regmap_update_bits(max98927->regmap, MAX98927_R0022_PCM_CLK_SETUP, MAX98927_PCM_CLK_SETUP_BSEL_MASK, @@ -311,7 +331,7 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; break; default: - dev_err(codec->dev, "format unsupported %d", + dev_err(codec->dev, "format unsupported %d\n", params_format(params)); goto err; } @@ -386,6 +406,76 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +495,7 @@ static const struct snd_soc_dai_ops max98927_dai_ops = { .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, @@ -495,6 +586,12 @@ static bool max98927_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + case MAX98927_R004C_MEAS_ADC_CH0_READ: + case MAX98927_R004D_MEAS_ADC_CH1_READ: + case MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS: + case MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R01FF_REV_ID: return true; default: return false; @@ -523,6 +620,18 @@ static SOC_ENUM_SINGLE_DECL(max98927_current_limit, MAX98927_R0042_BOOST_CTRL1, 1, max98927_current_limit_text); +static const char * const max98927_env_track_headroom_text[] = { + "0.000V", "0.125V", "0.250V", "0.375V", "0.500V", "0.625V", + "0.750V", "0.875V", "1.000V", "1.125V", "1.250V", "1.375V", + "1.500V", "1.625V", "1.750V", "1.875V", "2.000V", "2.125V", + "2.250V", "2.375V", "2.500V", "2.625V", "2.750V", "2.875V", + "3.000V", "3.125V", "3.250V", "3.375V", "3.500V" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_env_track_headroom, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0, + max98927_env_track_headroom_text); + static const struct snd_kcontrol_new max98927_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", MAX98927_R003C_SPK_GAIN, 0, 6, 0, @@ -540,6 +649,9 @@ static const struct snd_kcontrol_new max98927_snd_controls[] = { MAX98927_AMP_VOL_SEL_SHIFT, 1, 0), SOC_ENUM("Boost Output Voltage", max98927_boost_voltage), SOC_ENUM("Current Limit", max98927_current_limit), + SOC_SINGLE("EnvTrack Switch", MAX98927_R0086_ENV_TRACK_CTRL, + MAX98927_ENV_TRACKER_EN_SHIFT, 1, 0), + SOC_ENUM("EnvTrack Headroom", max98927_env_track_headroom), }; static const struct snd_soc_dapm_route max98927_audio_map[] = { @@ -635,13 +747,10 @@ static int max98927_probe(struct snd_soc_codec *codec) /* Envelope Tracking configuration */ regmap_write(max98927->regmap, MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, - 0x08); + 0x0A); regmap_write(max98927->regmap, MAX98927_R0086_ENV_TRACK_CTRL, 0x01); - regmap_write(max98927->regmap, - MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, - 0x10); /* voltage, current slot configuration */ regmap_write(max98927->regmap, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..43d3c74 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) @@ -194,7 +196,7 @@ /* MAX98927_R0024_PCM_SR_SETUP2 */ #define MAX98927_PCM_SR_SET2_SR_MASK (0xF << 4) #define MAX98927_PCM_SR_SET2_SR_SHIFT (4) -#define MAX98927_PCM_SR_SET2_IVADC_SR_MASK (0xf << 0) +#define MAX98927_PCM_SR_SET2_IVADC_SR_MASK (0xF << 0) /* MAX98927_R0025_PCM_TO_SPK_MONOMIX_A */ #define MAX98927_PCM_TO_SPK_MONOMIX_CFG_MASK (0x3 << 6) @@ -207,7 +209,7 @@ #define MAX98927_AMP_VOL_SEL (0x1 << 7) #define MAX98927_AMP_VOL_SEL_WIDTH (1) #define MAX98927_AMP_VOL_SEL_SHIFT (7) -#define MAX98927_AMP_VOL_MASK (0x7f << 0) +#define MAX98927_AMP_VOL_MASK (0x7F << 0) #define MAX98927_AMP_VOL_WIDTH (7) #define MAX98927_AMP_VOL_SHIFT (0) @@ -238,7 +240,7 @@ #define MAX98927_MEAS_I_EN (0x1 << 1) /* MAX98927_R0040_BOOST_CTRL0 */ -#define MAX98927_BOOST_CTRL0_VOUT_MASK (0x1f << 0) +#define MAX98927_BOOST_CTRL0_VOUT_MASK (0x1F << 0) #define MAX98927_BOOST_CTRL0_PVDD_MASK (0x1 << 7) #define MAX98927_BOOST_CTRL0_PVDD_EN_SHIFT (7) @@ -248,6 +250,10 @@ #define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) #define MAX98927_BROWNOUT_DSP_SHIFT (2) +/* MAX98927_R0086_ENV_TRACK_CTRL */ +#define MAX98927_ENV_TRACKER_EN (0x1 << 0) +#define MAX98927_ENV_TRACKER_EN_SHIFT (0) + /* MAX98927_R0100_SOFT_RESET */ #define MAX98927_SOFT_RESET (0x1 << 0) @@ -257,7 +263,6 @@ struct max98927_priv { struct regmap *regmap; struct snd_soc_codec *codec; - struct max98927_pdata *pdata; unsigned int spk_gain; unsigned int sysclk; unsigned int v_l_slot; -- 2.7.4

2 2

[alsa-devel] [PATCH 5/5] staging: max98927: Removed an obsolete variable
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index 3069a09..3551e7d 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -263,7 +263,6 @@ struct max98927_priv { struct regmap *regmap; struct snd_soc_codec *codec; - struct max98927_pdata *pdata; unsigned int spk_gain; unsigned int sysclk; unsigned int v_l_slot; -- 2.7.4

1 0

[alsa-devel] [PATCH 3/5] staging: max98927: Updated volatile register list
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index cdee3a3..e787a03 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -586,6 +586,12 @@ static bool max98927_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + case MAX98927_R004C_MEAS_ADC_CH0_READ: + case MAX98927_R004D_MEAS_ADC_CH1_READ: + case MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS: + case MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R01FF_REV_ID: return true; default: return false; -- 2.7.4

1 0

[alsa-devel] [PATCH 2/5] staging: max98927: Added controls for Envelope tracking
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 20 ++++++++++++++++---- sound/soc/codecs/max98927.h | 4 ++++ 2 files changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index 99d6e41..cdee3a3 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -614,6 +614,18 @@ static SOC_ENUM_SINGLE_DECL(max98927_current_limit, MAX98927_R0042_BOOST_CTRL1, 1, max98927_current_limit_text); +static const char * const max98927_env_track_headroom_text[] = { + "0.000V", "0.125V", "0.250V", "0.375V", "0.500V", "0.625V", + "0.750V", "0.875V", "1.000V", "1.125V", "1.250V", "1.375V", + "1.500V", "1.625V", "1.750V", "1.875V", "2.000V", "2.125V", + "2.250V", "2.375V", "2.500V", "2.625V", "2.750V", "2.875V", + "3.000V", "3.125V", "3.250V", "3.375V", "3.500V" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_env_track_headroom, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0, + max98927_env_track_headroom_text); + static const struct snd_kcontrol_new max98927_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", MAX98927_R003C_SPK_GAIN, 0, 6, 0, @@ -631,6 +643,9 @@ static const struct snd_kcontrol_new max98927_snd_controls[] = { MAX98927_AMP_VOL_SEL_SHIFT, 1, 0), SOC_ENUM("Boost Output Voltage", max98927_boost_voltage), SOC_ENUM("Current Limit", max98927_current_limit), + SOC_SINGLE("EnvTrack Switch", MAX98927_R0086_ENV_TRACK_CTRL, + MAX98927_ENV_TRACKER_EN_SHIFT, 1, 0), + SOC_ENUM("EnvTrack Headroom", max98927_env_track_headroom), }; static const struct snd_soc_dapm_route max98927_audio_map[] = { @@ -726,13 +741,10 @@ static int max98927_probe(struct snd_soc_codec *codec) /* Envelope Tracking configuration */ regmap_write(max98927->regmap, MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, - 0x08); + 0x0A); regmap_write(max98927->regmap, MAX98927_R0086_ENV_TRACK_CTRL, 0x01); - regmap_write(max98927->regmap, - MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, - 0x10); /* voltage, current slot configuration */ regmap_write(max98927->regmap, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index bf7a6f92..3069a09 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -250,6 +250,10 @@ #define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) #define MAX98927_BROWNOUT_DSP_SHIFT (2) +/* MAX98927_R0086_ENV_TRACK_CTRL */ +#define MAX98927_ENV_TRACKER_EN (0x1 << 0) +#define MAX98927_ENV_TRACKER_EN_SHIFT (0) + /* MAX98927_R0100_SOFT_RESET */ #define MAX98927_SOFT_RESET (0x1 << 0) -- 2.7.4

1 0

[alsa-devel] [PATCH 1/5] staging: max98927: Added TDM support
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 141 ++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98927.h | 6 +- 2 files changed, 120 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b5ee294..99d6e41 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +236,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + + } return 0; } @@ -245,6 +252,21 @@ static const int rate_table[] = { 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,19 +292,17 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: + if ((max98927->iface == SND_SOC_DAIFMT_DSP_A) || + (max98927->iface == SND_SOC_DAIFMT_DSP_B)) + return 0; + + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "BCLK %d not supported\n", blr_clk_ratio); return -EINVAL; } + regmap_update_bits(max98927->regmap, MAX98927_R0022_PCM_CLK_SETUP, MAX98927_PCM_CLK_SETUP_BSEL_MASK, @@ -386,6 +406,76 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +495,7 @@ static const struct snd_soc_dai_ops max98927_dai_ops = { .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..bf7a6f92 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) -- 2.7.4

1 0

[alsa-devel] [bug report] ASoC: add audio-graph-card support
by Dan Carpenter 19 May '17

19 May '17

Hello Kuninori Morimoto, The patch 2692c1c63c29: "ASoC: add audio-graph-card support" from Apr 20, 2017, leads to the following static checker warning: sound/soc/generic/audio-graph-card.c:275 asoc_graph_card_probe() info: return a literal instead of 'ret' sound/soc/generic/audio-graph-card.c 268 goto err; 269 } 270 271 snd_soc_card_set_drvdata(card, priv); 272 273 ret = devm_snd_soc_register_card(dev, card); 274 if (ret >= 0) 275 return ret; This code is confusing. I'm pretty sure that devm_snd_soc_register_card() returns zero on success or negative error codes but why are we checking >= 0 here? It's better to do error handling than success handling. People often get a bit creative with the last condition in a function... :/ In other words, it's cleaner like this: ret = devm_snd_soc_register_card(dev, card); if (ret) goto err; return 0; err: asoc_simple_card_clean_reference(card); return ret; 276 err: 277 asoc_simple_card_clean_reference(card); 278 279 return ret; 280 } regards, dan carpenter

2 1

Re: [alsa-devel] WM5102 - Help to make baytrail machine driver work
by Paulo Sergio 18 May '17

18 May '17

Adding alsa list + intel devs (Sorry for that ) Inline answers for Charles Keepax questions bellow: 2017-05-08 5:24 GMT-03:00 Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com>: > On Sun, May 07, 2017 at 10:42:31PM -0300, Paulo Sergio wrote: >> Hi, >> >> >> I'm trying to make wm5102 work on Lenovo Yoga 2 1051F (Baytrail) again >> and would like some tips. >> >> With a revised machine driver code [1] ( based on bytcr_rt5640 and >> some ports from Lenovo kernel [2]) I can initialize >> and setting routes with alsa mixer [3], I still don't have sound. >> >> However now if I connect 'HPOUT1' to 'Tone Generator 1' I can hear the 1k tone. >> Question: When I do this association, it is using my route? (in my >> case CODEC OUT -> SSP0 TX -> AIF1RX) >> Or this internal tone generator has a direct link to HPOUTs? >> >> Here's some lines from my dmesg output while attemptig to play >> something (complete dmesg - see [4]): >> >> [ 211.801176] arizona spi-WM510204:00: FLL1: Fref=25000000 Fout=49152000 >> [ 211.801184] arizona spi-WM510204:00: FLL1: Fvco=98304000Hz >> [ 211.801190] arizona spi-WM510204:00: FLL1: GCD=4000 >> [ 211.801195] arizona spi-WM510204:00: FLL1: N=7 THETA=a8d LAMBDA=c35 >> [ 211.801200] arizona spi-WM510204:00: FLL1: FRATIO=0(0) OUTDIV=2 REFCLK_DIV=1 >> [ 211.801204] arizona spi-WM510204:00: FLL1: GAIN=4 >> [ 211.801210] arizona spi-WM510204:00: 173 <= a8d >> [ 211.801227] arizona spi-WM510204:00: 174 <= c35 >> [ 211.801234] arizona spi-WM510204:00: 176 <= 40 >> [ 211.801241] arizona spi-WM510204:00: 179 <= 10 >> [ 211.813301] arizona spi-WM510204:00: 172 <= 8007 >> [ 211.813353] arizona spi-WM510204:00: 171 <= 1 >> [ 211.813366] arizona spi-WM510204:00: FLL1: Waiting for FLL lock... >> [ 211.814098] arizona spi-WM510204:00: d23 => 100 >> [ 211.814641] arizona spi-WM510204:00: d23 => 101 >> [ 211.814651] arizona spi-WM510204:00: FLL1: FLL locked (1 polls) >> [ 211.814660] arizona spi-WM510204:00: SYSCLK set to 49152000Hz >> [ 211.814668] arizona spi-WM510204:00: SYSCLK set to 49152000Hz >> [ 211.814675] arizona spi-WM510204:00: 101 <= 300 >> [ 211.814882] wm5102-codec wm5102-codec: AIF1: BCLK 1536000Hz LRCLK 48000Hz >> [ 211.814892] arizona spi-WM510204:00: 80 <= 3 >> [ 211.815191] arizona spi-WM510204:00: 4dc <= 0 >> [ 211.815339] arizona spi-WM510204:00: 4dd <= 0 >> [ 211.815460] arizona spi-WM510204:00: 80 <= 0 >> [ 211.815662] sst-mfld-platform sst-mfld-platform: Enter: enable=1 >> port_name=ssp0-port >> [ 211.816833] wm5102-codec wm5102-codec: AIF1: BCLK 1536000Hz LRCLK 48000Hz >> [ 211.816845] arizona spi-WM510204:00: 80 <= 3 >> [ 211.817030] arizona spi-WM510204:00: 4dc <= 0 >> [ 211.817287] arizona spi-WM510204:00: 4dd <= 0 >> >> Thanks in advance! >> >> >> [1] - https://github.com/pstglia/linux/tree/lenovo_yoga2_returns >> [2] - https://github.com/lenovo-yt2-dev/android_kernel_lenovo_baytrail/blob/cm-12… >> [3] - https://drive.google.com/file/d/0BxO6THtB865fQnQyZkx6c05Nbjg/view?usp=shari… >> [4] - https://drive.google.com/file/d/0BxO6THtB865fRy0xV3F0elRrUHM/view?usp=shari… > > OK good news you are getting sound from the Tone generator, that > likely implies things are setup sensibly on the CODEC side. My > suspicion would be that data is not making it over the I2S. > Probalby worth copying the Intel guys on the email, as I don't > know know much about that side of things. Copied alsa list and Intel guys (hit reply instead of reply all - my apologies) > > How are you setting up the I2S? As in which side is master, what > format are you using? Also do you see any over/underflow error, > does the system complain with "Error playing sample". The only part where I2S is explicity used on machine driver is this setting on backend DAI (SSP0-Codec) dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, And after setting the routes using routes mentioned on [3], I have no errors, but no sound Note: Based on Lenovo code, I concluded this device uses ssp0, so I'm using a fw file "fw_sst_0f28_ssp0.bin" posted another thread. Could use the hacks used on kernel 4.11, but as I'm still testing with kernel 4.4, using this fw would make things easier x86_64:/data # alsa_aplay -v TNT.mp3 Playing raw data 'TNT.mp3' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Rate conversion PCM (48000, sformat=U8) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 262144000 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 8 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 Slave: Hardware PCM card 0 'baytrailcraudio' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 appl_ptr : 0 hw_ptr : 0 Regards, Pstglia

3 10

[alsa-devel] [PATCH 1/2] ALSA: pcm: use "do {} while (0)" for empty macro
by Arnd Bergmann 18 May '17

18 May '17

Recent compilers produce a harmless warning for the new pcm_call_notify() macro when CONFIG_SND_PCM_OSS is disabled: sound/core/pcm.c: In function 'snd_pcm_free': sound/core/pcm.c:905:37: error: suggest braces around empty body in an 'if' statement [-Werror=empty-body] This turns the empty macro into a 'do {} while (0)' statement to avoid the warning. Fixes: 58f30d650c7f ("ALSA: pcm: Build pcm notifier code conditionally") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/core/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 24acbfb5c531..d30dba0ee688 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -894,7 +894,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) _notify->call(pcm); \ } while (0) #else -#define pcm_call_notify(pcm, call) /* NOP */ +#define pcm_call_notify(pcm, call) do {} while (0) #endif static int snd_pcm_free(struct snd_pcm *pcm) -- 2.9.0

3 3

[alsa-devel] Question about tsched=0 on PulseAudio
by Kuninori Morimoto 18 May '17

18 May '17

Hi ALSA ML We noticed that our PulseAudio has crackling / skipping noise which doesn't happen on normal aplay. And it was removed if we do below load-module module-hal-detect tsched=0 According to google, it say tsched Since 0.9.11. Use system-timer based model (aka glitch-free). Defaults to 1 (enabled). If your hardware does not return accurate timing information (e.g. Creative sound cards) you can try to set tsched=0 to enable the interupt based timing which was used in 0.9.10 and before. Here, what does this "return accurate timing information" mean ? Is it struct snd_pcm_ops::pointer ?? Best regards --- Kuninori Morimoto

3 4

[alsa-devel] [PATCH] ALSA: mixart: constify snd_kcontrol_new structures
by Bhumika Goyal 17 May '17

17 May '17

Declare snd_kcontrol_new structures as const as they are only passed an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/mixart/mixart_mixer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 4a4616a..2b9496a 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -404,7 +404,7 @@ static int mixart_analog_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e static const DECLARE_TLV_DB_SCALE(db_scale_analog, -9600, 50, 0); -static struct snd_kcontrol_new mixart_control_analog_level = { +static const struct snd_kcontrol_new mixart_control_analog_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -897,7 +897,7 @@ static int mixart_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static const DECLARE_TLV_DB_SCALE(db_scale_digital, -10950, 50, 0); -static struct snd_kcontrol_new snd_mixart_pcm_vol = +static const struct snd_kcontrol_new snd_mixart_pcm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -951,7 +951,7 @@ static int mixart_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ return changed; } -static struct snd_kcontrol_new mixart_control_pcm_switch = { +static const struct snd_kcontrol_new mixart_control_pcm_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* name will be filled later */ .count = MIXART_PLAYBACK_STREAMS, -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: cs46xx: constify snd_kcontrol_new structures
by Bhumika Goyal 17 May '17

17 May '17

Declare snd_kcontrol_new structures as const as they are only passed an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index e4cf318..00fa52e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2371,7 +2371,7 @@ static int snd_cs46xx_front_dup_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] ? 0 : 0x200); } -static struct snd_kcontrol_new snd_cs46xx_front_dup_ctl = { +static const struct snd_kcontrol_new snd_cs46xx_front_dup_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Duplicate Front", .info = snd_mixer_boolean_info, -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: ice1712: constify snd_kcontrol_new structures
by Bhumika Goyal 17 May '17

17 May '17

Declare snd_kcontrol_new structures as const as they are only passed an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 6 +++--- sound/pci/ice1712/ice1724.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 1d8612c..6b650ab 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1401,7 +1401,7 @@ static struct snd_kcontrol_new snd_ice1712_multi_playback_ctrls[] = { }, }; -static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_switch = { +static const struct snd_kcontrol_new snd_ice1712_multi_capture_analog_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "H/W Multi Capture Switch", .info = snd_ice1712_pro_mixer_switch_info, @@ -1420,7 +1420,7 @@ static const struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch = { .count = 2, }; -static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume = { +static const struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -2165,7 +2165,7 @@ static int snd_ice1712_pro_route_spdif_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ice1712_mixer_pro_analog_route = { +static const struct snd_kcontrol_new snd_ice1712_mixer_pro_analog_route = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "H/W Playback Route", .info = snd_ice1712_pro_route_info, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 9cd6e55..7851881 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2142,7 +2142,7 @@ static int snd_vt1724_pro_route_spdif_put(struct snd_kcontrol *kcontrol, digital_route_shift(idx)); } -static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route = +static const struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "H/W Playback Route", -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: rt5665: Fix uninitialized warning in rt5665_i2s_pin_event()
by Geert Uytterhoeven 17 May '17

17 May '17

With gcc 4.1.2: sound/soc/codecs/rt5665.c: In function ‘rt5665_i2s_pin_event’: sound/soc/codecs/rt5665.c:2610: warning: ‘mask1’ may be used uninitialized in this function sound/soc/codecs/rt5665.c:2610: warning: ‘val2’ may be used uninitialized in this function sound/soc/codecs/rt5665.c:2610: warning: ‘val1’ may be used uninitialized in this function The first one is currently a false positive, as rt5665_i2s_pin_event() is never called with snd_soc_dapm_widget.shift set to a value not handled by the switch() statement. But that may change, so preinitialize mask1 to fix this, like is already done for mask2. The last two are false-positives, the compiler is just not smart enough to notice the mask and val variables are always used together. Fixes: 9b5d3865b3b410d2 ("ASoC: rt5665: set i2s pin share configuration") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/codecs/rt5665.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 8cd22307f5b6e6ab..5e42f4ee51ba5b20 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2607,7 +2607,7 @@ static int rt5665_i2s_pin_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int val1, val2, mask1, mask2 = 0; + unsigned int val1, val2, mask1 = 0, mask2 = 0; switch (w->shift) { case RT5665_PWR_I2S2_1_BIT: @@ -2635,13 +2635,17 @@ static int rt5665_i2s_pin_event(struct snd_soc_dapm_widget *w, } switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, mask1, val1); + if (mask1) + snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, + mask1, val1); if (mask2) snd_soc_update_bits(codec, RT5665_GPIO_CTRL_2, mask2, val2); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, mask1, 0); + if (mask1) + snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, + mask1, 0); if (mask2) snd_soc_update_bits(codec, RT5665_GPIO_CTRL_2, mask2, 0); -- 2.7.4

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[alsa-devel] [PATCH] base: export symbol of_phandle_iterator_init/next
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> of_for_each_phandle() uses of_phandle_iterator_init/next but these aren't exported. So kernel module complile will say ERROR: "of_phandle_iterator_init" [xxx.ko] undefined! ERROR: "of_phandle_iterator_next" [xxx.ko] undefined! This patch solves this issue Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch is related to this patch-set Subject: [PATCH v5 0/9] ASoC: add OF-graph base simple-card Date: Tue, 21 Mar 2017 14:17:03 +0900 drivers/of/base.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/of/base.c b/drivers/of/base.c index 812edb9..bc42f91 100644 --- a/drivers/of/base.c +++ b/drivers/of/base.c @@ -1570,6 +1570,7 @@ int of_phandle_iterator_init(struct of_phandle_iterator *it, return 0; } +EXPORT_SYMBOL_GPL(of_phandle_iterator_init); int of_phandle_iterator_next(struct of_phandle_iterator *it) { @@ -1639,6 +1640,7 @@ int of_phandle_iterator_next(struct of_phandle_iterator *it) return -EINVAL; } +EXPORT_SYMBOL_GPL(of_phandle_iterator_next); int of_phandle_iterator_args(struct of_phandle_iterator *it, uint32_t *args, -- 1.9.1

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[alsa-devel] [PATCH 0/7] of_graph: prepare for ALSA graph support
by Kuninori Morimoto 17 May '17

17 May '17

Hi Rob, Mark Cc Guennadi, Laurent Now OF graph is mainly used by V4L2 SoC, and ALSA SoC is using different style for SoC <-> Codec binding. But, for example, HDMI case, V4L2 <-> ALSA need to collaborate, and then ALSA SoC needs to adjust to OF graph. OTOH, V4L2's "OF graph" position is same as ALSA SoC "sound card" position. And ALSA SoC side want to keep existing supported feature on new OF graph style. I'm posting this on ALSA SoC ML now. Now, current of_graph is indicating port/endpoint, but there is no way to understand that it is for video port ? or sound port ? or other device port ? For example, HDMI has video port, and sound port. Because of this reason, ALSA SoC side can't handle OF graph correctly. Thus, this patch-set tries to add new "type" on OF graph. And this patch-set includes small feature which are useful for ALSA SoC side OF graph support. Kuninori Morimoto (7): Documentation: of: add type property of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_port/endpoint_type_is() of_graph: add of_graph_get_endpoint_count() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_top_port() of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() Documentation/devicetree/bindings/graph.txt | 26 ++++ drivers/of/base.c | 196 ++++++++++++++++++++++++++-- include/linux/of_graph.h | 67 ++++++++++ 3 files changed, 279 insertions(+), 10 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v4 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Rob, Mark These are v4 of OF-graph base audio card patch-set. Main difference from v3 is that it doesn't have prefix on DT, and have new "label", "dai-format". 1) - 3) : new OF-graph helper functions 4) - 5) : expand "label", "dai-format" 6) - 9) : OF-graph base simple-card Kuninori Morimoto (9): 1) of_graph: add of_graph_get_remote_endpoint() 2) of_graph: add of_graph_get_port_parent() 3) of_graph: add of_graph_get_endpoint_count() 4) ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name 5) ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() 6) ASoC: add snd_soc_get_dai_id() 7) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 8) ASoC: add audio-graph-card document 9) ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ .../devicetree/bindings/sound/simple-card.txt | 3 +- .../devicetree/bindings/sound/simple-scu-card.txt | 4 +- drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 301 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 12 files changed, 602 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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[alsa-devel] [PATCH v6 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark, Rob These are v6 of OF-graph sound card support. Main change is asoc_simple_card_get_dai_id() becames simple-card-utils's static function. 1) - 4) : OF-graph base functions 5) - 9) : OF-graph + simple-card (= audio_graph_card) Kuninori Morimoto (9): 1) of-graph: export symbol of_phandle_iterator_init/next 2) of_graph: add of_graph_get_remote_endpoint() 3) of_graph: add of_graph_get_port_parent() 4) of_graph: add of_graph_get_endpoint_count() 5) ASoC: soc-core: enable "dai-format" on snd_soc_of_parse_daifmt() 6) ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name 7) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 8) ASoC: add audio-graph-card document 9) ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 62 ++++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 75 ++++- sound/soc/soc-core.c | 10 +- 9 files changed, 603 insertions(+), 17 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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[alsa-devel] [PATCH v7 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark, Rob These are v7 of OF-graph sound card support. Main change from v6 "label" doesn't allow [prefix]label (in [6/9]) 1) - 4) : OF-graph base functions 5) - 9) : OF-graph + simple-card (= audio_graph_card) Kuninori Morimoto (9): of-graph: export symbol of_phandle_iterator_init/next of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: soc-core: enable "dai-format" on snd_soc_of_parse_daifmt() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 62 ++++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 73 ++++- sound/soc/soc-core.c | 10 +- 9 files changed, 600 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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[alsa-devel] [PATCH v5 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Rob, Mark These are v5 of OF-graph base audio card patch-set. It fixed small things which was pointed by Rob. 1) - 3) : new OF-graph helper functions 4) - 5) : expand "label", "dai-format" 6) - 9) : OF-graph base simple-card Kuninori Morimoto (9): of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() ASoC: add snd_soc_get_dai_id() ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 10 files changed, 604 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: rsnd: tidyup Document
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark Cc Geert These patches tidyup Document for Renesas sound driver Kuninori Morimoto (2): ASoC: rsnd: move "renesas,rsrc-card" to "simple-scu-audio-card" on Document ASoC: rsnd: add missing clocks/clock-names on Document .../devicetree/bindings/sound/renesas,rsnd.txt | 34 ++++++++++++---------- 1 file changed, 19 insertions(+), 15 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: hdmi-codec: fix spelling mistake: "deteced" -> "detected"
by Colin King 17 May '17

17 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8c5ae1fc23a9..a3f15149afcf 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -819,7 +819,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) mutex_unlock(&hdmi_mutex); if (hd->cnt >= ARRAY_SIZE(hdmi_dai_name)) { - dev_err(dev, "too many hdmi codec are deteced\n"); + dev_err(dev, "too many hdmi codec are detected\n"); return -EINVAL; } -- 2.11.0

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[alsa-devel] [PATCH] ASoC: rsnd: remove un-used OUF_SRCO/I macro
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 20b5b2e..a8ab26d 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -12,10 +12,6 @@ #define SRC_NAME "src" -/* SRCx_STATUS */ -#define OUF_SRCO ((1 << 12) | (1 << 13)) -#define OUF_SRCI ((1 << 9) | (1 << 8)) - /* SCU_SYSTEM_STATUS0/1 */ #define OUF_SRC(id) ((1 << (id + 16)) | (1 << id)) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: remove mismatch explain comment from src.c
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Before, src.c cares SRC and SSIU, but now SSIU is cared by ssiu.c by commit c7f69ab5364d ("ASoC: rsnd: use mod base common method on SSIU"). This patch removes mismatched explanation for SRC/SSIU from src.c Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a8ab26d..e4bde0c 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -51,20 +51,6 @@ struct rsnd_src { * */ -/* - * src.c is caring... - * - * Gen1 - * - * [mem] -> [SRU] -> [SSI] - * |--------| - * - * Gen2 - * - * [mem] -> [SRC] -> [SSIU] -> [SSI] - * |-----------------| - */ - static void rsnd_src_activation(struct rsnd_mod *mod) { rsnd_mod_write(mod, SRC_SWRSR, 0); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: clkout-lr-asynchronous is AUDIO_CLKOUT related property
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> clkout-lr-asynchronous is AUDIO_CLKOUT related property, thus, it should be handled by rsnd_adg_get_clkout(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 66203d1..cb4eab0 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -480,6 +480,9 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, if (req_rate[0] % 48000 == 0) adg->flags = AUDIO_OUT_48; + if (of_get_property(np, "clkout-lr-asynchronous", NULL)) + adg->flags = LRCLK_ASYNC; + /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC * have 44.1kHz or 48kHz base clocks for now. @@ -578,7 +581,6 @@ int rsnd_adg_probe(struct rsnd_priv *priv) { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); - struct device_node *np = dev->of_node; int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); @@ -595,9 +597,6 @@ int rsnd_adg_probe(struct rsnd_priv *priv) rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); - if (of_get_property(np, "clkout-lr-asynchronous", NULL)) - adg->flags = LRCLK_ASYNC; - priv->adg = adg; rsnd_adg_clk_enable(priv); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: remove error pointer for adg->clkout[i]
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit d7f298197a22f11b38059f257842dac7c30a564c ("ASoC: rcar: fixup of_clk_add_provider() usage for multi clkout") added adg->clkout[i] = ERR_PTR(-ENOENT), but, adg->clkout[i] are assumed NULL or clk pointer. This patch fix it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index cb4eab0..36bece8 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -556,7 +556,6 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, req_rate[0]); - adg->clkout[i] = ERR_PTR(-ENOENT); if (!IS_ERR(clk)) adg->clkout[i] = clk; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: need to call nolock_stop if nolock_start was failed
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_dai_call() is counting the called number of start/stop functions. nolock_stop should be called if nolock_start was failed. Otherwise, nolock_start doesn't called in 2nd play. This patch solved this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8c1f4e2..e16f723 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -766,11 +766,16 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + int ret; /* * call rsnd_dai_call without spinlock */ - return rsnd_dai_call(nolock_start, io, priv); + ret = rsnd_dai_call(nolock_start, io, priv); + if (ret < 0) + rsnd_dai_call(nolock_stop, io, priv); + + return ret; } static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, -- 1.9.1

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Re: [alsa-devel] ASoC: rsnd: fix sound route path when using SRC6/SRC9
by Geert Uytterhoeven 17 May '17

17 May '17

Hi Morimoto-san, On Fri, Mar 31, 2017 at 9:32 PM, Linux Kernel Mailing List <linux-kernel(a)vger.kernel.org> wrote: > Web: https://git.kernel.org/torvalds/c/a1c2ff53726907aff5feb37e4cfd45c1ff626431 > Commit: a1c2ff53726907aff5feb37e4cfd45c1ff626431 > Parent: 4b30eebfc35c67771b5f58d9274d3e321b72d7a8 > Refname: refs/heads/master > Author: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> > AuthorDate: Wed Mar 1 03:51:00 2017 +0000 > Committer: Mark Brown <broonie(a)kernel.org> > CommitDate: Mon Mar 13 12:58:07 2017 +0000 > > ASoC: rsnd: fix sound route path when using SRC6/SRC9 > > This patch fixes the problem that the missing value of the route path > setting table and incorrect values are set in the CMD_ROUTE_SELECT > register. > > Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> > [Kuninori: shared data on MIX and non-MIX case] > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Signed-off-by: Mark Brown <broonie(a)kernel.org> > --- > sound/soc/sh/rcar/cmd.c | 36 ++++++++++++++++++++---------------- > 1 file changed, 20 insertions(+), 16 deletions(-) > > diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c > index abb5eaac854a..7d92a24b7cfa 100644 > --- a/sound/soc/sh/rcar/cmd.c > +++ b/sound/soc/sh/rcar/cmd.c > @@ -31,23 +31,24 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, > struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); > struct device *dev = rsnd_priv_to_dev(priv); > u32 data; > + u32 path[] = { > + [1] = 1 << 0, > + [5] = 1 << 8, > + [6] = 1 << 12, > + [9] = 1 << 15, > + }; > > if (!mix && !dvc) > return 0; > > + if (ARRAY_SIZE(path) < rsnd_mod_id(mod) + 1) > + return -ENXIO; > + > if (mix) { > struct rsnd_dai *rdai; > struct rsnd_mod *src; > struct rsnd_dai_stream *tio; > int i; > - u32 path[] = { > - [0] = 0, > - [1] = 1 << 0, > - [2] = 0, > - [3] = 0, > - [4] = 0, > - [5] = 1 << 8 > - }; > > /* > * it is assuming that integrater is well understanding about > @@ -70,16 +71,19 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, > } else { > struct rsnd_mod *src = rsnd_io_to_mod_src(io); > > - u32 path[] = { > - [0] = 0x30000, > - [1] = 0x30001, > - [2] = 0x40000, > - [3] = 0x10000, > - [4] = 0x20000, > - [5] = 0x40100 > + u8 cmd_case[] = { > + [0] = 0x3, > + [1] = 0x3, > + [2] = 0x4, > + [3] = 0x1, > + [4] = 0x2, > + [5] = 0x4, > + [6] = 0x1, > + [9] = 0x2, > }; > > - data = path[rsnd_mod_id(src)]; > + data = path[rsnd_mod_id(src)] | With gcc 4.9.0: sound/soc/sh/rcar/cmd.c: In function 'rsnd_cmd_init': sound/soc/sh/rcar/cmd.c:85:14: warning: array subscript is below array bounds [-Warray-bounds] data = path[rsnd_mod_id(src)] | > + cmd_case[rsnd_mod_id(src)] << 16; > } > > dev_dbg(dev, "ctu/mix path = 0x%08x", data); 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

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