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

[alsa-devel] [PATCH v3 0/2] ASoC: Add support for rt5640 sound card found on HP Pavilion x2 10-p000 tablet.
by Alexandrov Stansilav 16 Jan '17

16 Jan '17

Inside DSDT table there is record for this soundcard: Device (RTKC) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "10EC3276") // _HID: Hardware ID Name (_CID, "10EC3276") // _CID: Compatible ID Name (_DDN, "ALC3276") // _DDN: DOS Device Name Name (_SUB, "103C827C") // _SUB: Subsystem ID Name (_PR0, Package (0x01) // _PR0: Power Resources for D0 original bugreport: https://bugzilla.kernel.org/show_bug.cgi?id=187621 v3: Fixed patch descriptions v2: Splited patch to codec and sst parts Alexandrov Stansilav (2): ASoc: rt5640: Add "10EC3276" ACPI ID ASoC: Intel: Atom: Add HP Pavilion x2 10 10-p000 machine entry sound/soc/codecs/rt5640.c | 1 + sound/soc/intel/atom/sst/sst_acpi.c | 2 ++ 2 files changed, 3 insertions(+) -- 2.10.2

4 5

[alsa-devel] [PATCH] pcm: direct: Fix deadlock in poll_descriptors
by Takashi Iwai 16 Jan '17

16 Jan '17

The recent change in PCM direct plugins to check XRUN in poll_descriptors callback caused a regression; as consequence, the whole playback hangs up. The culprit is a mutex dead lock by the call in snd_pcm_state() inside the new snd_pcm_direct_poll_descriptors(). The poll_descriptors code path is protected with pcm mutex, thus an unlocked version (__snd_pcm_state()) has to be used inside the callback instead. Fixes: 789ee39727a1 ("pcm: direct: check state before enter poll on timer") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index f4d7c3f2f284..e4bef7860d01 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -676,7 +676,8 @@ int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, } /* this will also evaluate slave state and enter xrun if necessary */ - switch (snd_pcm_state(pcm)) { + /* using __snd_pcm_state() since this function is called inside lock */ + switch (__snd_pcm_state(pcm)) { case SND_PCM_STATE_XRUN: return -EPIPE; default: -- 2.11.0

1 0

[alsa-devel] [PATCH v2 0/4] ASoC: Dell IoT Gateway audio support
by Shrirang Bagul 16 Jan '17

16 Jan '17

This work is based on for-next branch of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git These patches add basic audio support on new Dell Edge IoT Gateways. RT5660 codec is connected to SSP2 port on Intel Atom Baytrail E3815 SoC. The machine driver is based on bytcr_rt5640. These devices have line-in and line-out jacks with line-out mute enable gpio described in ACPI tables. Since v1: * rt5660: Use enumerations to define GPIO's described ACPI DSDT * Initialize rt5660 controls from card struct * Re-use existing bytcr_rt5640 machine driver to support rt5660 codec based audio Shrirang Bagul (4): ASoC: rt5660: Add ACPI support ASoC: Intel: bytcr_rt5640: move codec clks to card driver data ASoC: Intel: Support rt5660 codec for Baytrail ASoC: Intel: bytcr_rt5640: Support line-out mute gpio sound/soc/codecs/rt5660.c | 32 ++++++ sound/soc/intel/Kconfig | 11 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 + sound/soc/intel/boards/bytcr_rt5640.c | 211 ++++++++++++++++++++++++++++++---- 4 files changed, 228 insertions(+), 28 deletions(-) -- 2.9.3

4 9

[alsa-devel] [PATCH v3] ALSA: mips: avoid potential uninitialized variable use
by Arnd Bergmann 16 Jan '17

16 Jan '17

MIPS allmodconfig results in this warning: sound/mips/hal2.c: In function 'hal2_gain_get': sound/mips/hal2.c:224:35: error: 'r' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c:223:35: error: 'l' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c: In function 'hal2_gain_put': sound/mips/hal2.c:260:13: error: 'new' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c:260:13: error: 'old' may be used uninitialized in this function [-Werror=maybe-uninitialized] Returning an error for all unexpected cases shuts up the warning Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v3: actually send the correct patch, sorry for the mixup v2: return an error instead trying to handle gracefully sound/mips/hal2.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index ede449f0b50d..00fc9241d266 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -219,6 +219,8 @@ static int hal2_gain_get(struct snd_kcontrol *kcontrol, l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15; r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15; break; + default: + return -EINVAL; } ucontrol->value.integer.value[0] = l; ucontrol->value.integer.value[1] = r; @@ -256,6 +258,8 @@ static int hal2_gain_put(struct snd_kcontrol *kcontrol, new |= (r << H2I_C2_R_GAIN_SHIFT); hal2_i_write32(hal2, H2I_ADC_C2, new); break; + default: + return -EINVAL; } return old != new; } -- 2.9.0

2 1

[alsa-devel] [PATCH] ALSA: mips: avoid potential uninitialized variable use
by Arnd Bergmann 16 Jan '17

16 Jan '17

MIPS allmodconfig results in this warning: sound/mips/hal2.c: In function 'hal2_gain_get': sound/mips/hal2.c:224:35: error: 'r' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c:223:35: error: 'l' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c: In function 'hal2_gain_put': sound/mips/hal2.c:260:13: error: 'new' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c:260:13: error: 'old' may be used uninitialized in this function [-Werror=maybe-uninitialized] It's easy enough to avoid by adding a 'default' clause to the switch statements here. I assume that in practice no other case can happen, but adding a default puts us on the safe side and shuts up the warnings. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/mips/hal2.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index ede449f0b50d..0af4f7a9b3e7 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -219,6 +219,9 @@ static int hal2_gain_get(struct snd_kcontrol *kcontrol, l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15; r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15; break; + default: + l = 0; + r = 0; } ucontrol->value.integer.value[0] = l; ucontrol->value.integer.value[1] = r; @@ -256,6 +259,9 @@ static int hal2_gain_put(struct snd_kcontrol *kcontrol, new |= (r << H2I_C2_R_GAIN_SHIFT); hal2_i_write32(hal2, H2I_ADC_C2, new); break; + default: + new = 0; + old = 0; } return old != new; } -- 2.9.0

2 2

[alsa-devel] [PATCH v2] ALSA: mips: avoid potential uninitialized variable use
by Arnd Bergmann 16 Jan '17

16 Jan '17

MIPS allmodconfig results in this warning: sound/mips/hal2.c: In function 'hal2_gain_get': sound/mips/hal2.c:224:35: error: 'r' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c:223:35: error: 'l' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c: In function 'hal2_gain_put': sound/mips/hal2.c:260:13: error: 'new' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/mips/hal2.c:260:13: error: 'old' may be used uninitialized in this function [-Werror=maybe-uninitialized] Returning an error for all unexpected cases shuts up the warning Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: return an error instead trying to handle gracefully --- sound/mips/hal2.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index ede449f0b50d..0af4f7a9b3e7 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -219,6 +219,9 @@ static int hal2_gain_get(struct snd_kcontrol *kcontrol, l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15; r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15; break; + default: + l = 0; + r = 0; } ucontrol->value.integer.value[0] = l; ucontrol->value.integer.value[1] = r; @@ -256,6 +259,9 @@ static int hal2_gain_put(struct snd_kcontrol *kcontrol, new |= (r << H2I_C2_R_GAIN_SHIFT); hal2_i_write32(hal2, H2I_ADC_C2, new); break; + default: + new = 0; + old = 0; } return old != new; } -- 2.9.0

1 0

[alsa-devel] [PATCH v5 0/3] ucm: Automatically load the best config file based on the card long name
by mengdong.lin＠linux.intel.com 16 Jan '17

16 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Intel DSP platform drivers are used by many different devices. For user space to differentiate them, now ASoC provide API for machine drivers to include the DMI info (vendor, product and board) in card long name. This series will add UCM support for loading best device-specific configuration based on the card long name. Mengdong Lin (2): ucm: Load device-specific configuration file based on the card long name ucm: Add command 'get _file' to get the config file name of the opened card History: v2: Request the device-specific file name to match the card long name, no long use automatic key word matching and scoring to avoid error and uncertainty. And add command to check the name of actually loaded configuration file. v3: Fix num in strncpy of the long name, which should be MAX_CARD_LONG_NAME - 1, not MAX_CARD_LONG_NAME (80). v4: No code change. Just update on comments because the kernel v4 patches add DMI product version to the card long name, and use "-" as the separator between DMI fields. v5: Revise the comments. Add patch to assure the input card name not to exceed the maximum card long name supported by kernel to avoid use strncpy and strncmp later. Mengdong Lin (3): ucm: Assure the user input card name not to exceed max size of card long name ucm: Load device-specific configuration file based on the card long name ucm: Add command 'get _file' to get the config file name of the opened card include/use-case.h | 1 + src/ucm/main.c | 14 +++++++ src/ucm/parser.c | 105 ++++++++++++++++++++++++++++++++++++++++++++++++++-- src/ucm/ucm_local.h | 3 ++ 4 files changed, 119 insertions(+), 4 deletions(-) -- 2.7.4

1 3

[alsa-devel] Add support for ALC1220 codec
by Kailang 16 Jan '17

16 Jan '17

Hi Takashi, Attach patch is support new codec of ALC1220. Many Thanks, Kailang

2 1

[alsa-devel] [PATCH 0/4] Add support for es8388 on the rock2
by Romain Perier 16 Jan '17

16 Jan '17

This set of patches adds a machine driver for rockchip boards that use ES8388 codecs. It also adds slave mode to the es8328 driver that currently only supported the master mode. Then, it adds the required DT definitions to link rockchip-i2s to the es8388 analog output. This work is based on the initial work that was done by Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> with some improvements, changes and more commits. Romain Perier (4): ASoC: es8328-i2c: Add compatible for ES8388 ASoC: es8328: Add support for slave mode ASoC: rockchip: Add machine driver for ES8388 codecs arm: dts: Add support for ES8388 to the Radxa Rock 2 Documentation/devicetree/bindings/sound/es8328.txt | 2 +- .../devicetree/bindings/sound/rockchip-es8388.txt | 28 +++ arch/arm/boot/dts/rk3288-rock2-square.dts | 37 +++ sound/soc/codecs/es8328-i2c.c | 3 +- sound/soc/codecs/es8328.c | 28 ++- sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_es8388.c | 270 +++++++++++++++++++++ 8 files changed, 371 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-es8388.txt create mode 100644 sound/soc/rockchip/rockchip_es8388.c -- 2.9.3

3 8

[alsa-devel] [PATCH] ASoC: tegra: constify tegra30_ahub_soc_data structures
by Bhumika Goyal 15 Jan '17

15 Jan '17

The tegra30_ahub_soc_data structures are only stored in the data field of of_device_id structures. This field is of type const, so tegra30_ahub_soc_data structures having the same property can be declared const too. File size before: sound/soc/tegra/tegra30_ahub.o text data bss dec hex filename 7079 240 8 7327 1c9f sound/soc/tegra/tegra30_ahub.o File size after: sound/soc/tegra/tegra30_ahub.o text data bss dec hex filename 7143 192 8 7343 1caf sound/soc/tegra/tegra30_ahub.o Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/tegra/tegra30_ahub.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index fef3b9a..e63ef8d 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -493,17 +493,17 @@ static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) .cache_type = REGCACHE_FLAT, }; -static struct tegra30_ahub_soc_data soc_data_tegra30 = { +static const struct tegra30_ahub_soc_data soc_data_tegra30 = { .mod_list_mask = MOD_LIST_MASK_TEGRA30, .set_audio_cif = tegra30_ahub_set_cif, }; -static struct tegra30_ahub_soc_data soc_data_tegra114 = { +static const struct tegra30_ahub_soc_data soc_data_tegra114 = { .mod_list_mask = MOD_LIST_MASK_TEGRA114, .set_audio_cif = tegra30_ahub_set_cif, }; -static struct tegra30_ahub_soc_data soc_data_tegra124 = { +static const struct tegra30_ahub_soc_data soc_data_tegra124 = { .mod_list_mask = MOD_LIST_MASK_TEGRA124, .set_audio_cif = tegra124_ahub_set_cif, }; -- 1.9.1

1 0

[alsa-devel] [PATCH v5 0/4] SoC: Intel: clean up Kconfig
by Andy Shevchenko 14 Jan '17

14 Jan '17

Clean up Kconfig here: - select DW_DMAC_CORE because it's mandatory for some modules - remove redundant selects and dependencies - rename MFLD_PLATFORM to ATOM_HIFI2_PLATFORM to show that it covers more than *Field Since v4: - append Vinod's and Liam's ACKs Since v3: - use proposed name as SND_SST_ATOM_HIFI2_PLATFORM (Pierre-Louis) - append Keyon's tag Since v2: - add three more patches - rebase on top of latest linux-next Andy Shevchenko (4): ASoC: Intel: select DW_DMAC_CORE since it's mandatory ASoC: Intel: remove redundant select SND_SOC_INTEL_SST ASoC: Intel: remove ignored dependencies ASoC: Intel: rename SND_SST_MFLD_PLATFORM to SND_SST_ATOM_HIFI2_PLATFORM sound/soc/intel/Kconfig | 46 +++++++++------------------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/Makefile | 7 +++-- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 + 4 files changed, 21 insertions(+), 35 deletions(-) -- 2.11.0

4 9

[alsa-devel] [PATCH] ALSA: oxfw: enclose identifiers referred by single function for scs1x feature
by Takashi Sakamoto 14 Jan '17

14 Jan '17

Some identifiers are referred just by one functions. In this case, they can be put into the function definition. This brings two merits; readers can easily follow codes related to the identifiers, developers are free from name conflict. This commit moves such identifiers to each function definition. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-scs1x.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c index 7940058..93209eb 100644 --- a/sound/firewire/oxfw/oxfw-scs1x.c +++ b/sound/firewire/oxfw/oxfw-scs1x.c @@ -338,12 +338,6 @@ static void midi_playback_drain(struct snd_rawmidi_substream *stream) wait_event(scs->idle_wait, scs->output_idle); } -static const struct snd_rawmidi_ops midi_playback_ops = { - .open = midi_playback_open, - .close = midi_playback_close, - .trigger = midi_playback_trigger, - .drain = midi_playback_drain, -}; static int register_address(struct snd_oxfw *oxfw) { struct fw_scs1x *scs = oxfw->spec; @@ -369,6 +363,12 @@ void snd_oxfw_scs1x_update(struct snd_oxfw *oxfw) int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw) { + static const struct snd_rawmidi_ops midi_playback_ops = { + .open = midi_playback_open, + .close = midi_playback_close, + .trigger = midi_playback_trigger, + .drain = midi_playback_drain, + }; struct snd_rawmidi *rmidi; struct fw_scs1x *scs; int err; -- 2.9.3

2 1

[alsa-devel] [PATCH] uapi: use wildcards to list files
by Nicolas Dichtel 13 Jan '17

13 Jan '17

Regularly, when a new header is created in include/uapi/, the developer forgets to add it in the corresponding Kbuild file. This error is usually detected after the release is out. In fact, all headers under include/uapi/ should be exported, so let's use wildcards. After this patch, the following files, which were not exported, are now exported: drm/vgem_drm.h drm/armada_drm.h drm/omap_drm.h drm/etnaviv_drm.h rdma/qedr-abi.h linux/bcache.h linux/kfd_ioctl.h linux/cryptouser.h linux/kcm.h linux/kcov.h linux/seg6_iptunnel.h linux/stm.h linux/seg6.h linux/auto_dev-ioctl.h linux/userio.h linux/pr.h linux/wil6210_uapi.h linux/nilfs2_ondisk.h linux/hash_info.h linux/seg6_genl.h linux/seg6_hmac.h linux/batman_adv.h linux/nsfs.h linux/qrtr.h linux/btrfs_tree.h linux/coresight-stm.h linux/dma-buf.h linux/module.h linux/lightnvm.h linux/nilfs2_api.h Signed-off-by: Nicolas Dichtel <nicolas.dichtel(a)6wind.com> --- This patch is built against linus tree. I don't know if it should be done against antoher tree. Comments are welcomed, Nicolas include/uapi/asm-generic/Kbuild | 36 +-- include/uapi/drm/Kbuild | 22 +- include/uapi/linux/Kbuild | 463 +---------------------------- include/uapi/linux/android/Kbuild | 2 +- include/uapi/linux/byteorder/Kbuild | 3 +- include/uapi/linux/caif/Kbuild | 3 +- include/uapi/linux/can/Kbuild | 6 +- include/uapi/linux/dvb/Kbuild | 9 +- include/uapi/linux/hdlc/Kbuild | 2 +- include/uapi/linux/hsi/Kbuild | 2 +- include/uapi/linux/iio/Kbuild | 3 +- include/uapi/linux/isdn/Kbuild | 2 +- include/uapi/linux/mmc/Kbuild | 2 +- include/uapi/linux/netfilter/Kbuild | 88 +----- include/uapi/linux/netfilter/ipset/Kbuild | 5 +- include/uapi/linux/netfilter_arp/Kbuild | 3 +- include/uapi/linux/netfilter_bridge/Kbuild | 18 +- include/uapi/linux/netfilter_ipv4/Kbuild | 10 +- include/uapi/linux/netfilter_ipv6/Kbuild | 13 +- include/uapi/linux/nfsd/Kbuild | 6 +- include/uapi/linux/raid/Kbuild | 3 +- include/uapi/linux/spi/Kbuild | 2 +- include/uapi/linux/sunrpc/Kbuild | 2 +- include/uapi/linux/tc_act/Kbuild | 15 +- include/uapi/linux/tc_ematch/Kbuild | 5 +- include/uapi/linux/usb/Kbuild | 12 +- include/uapi/linux/wimax/Kbuild | 2 +- include/uapi/misc/Kbuild | 2 +- include/uapi/mtd/Kbuild | 6 +- include/uapi/rdma/Kbuild | 17 +- include/uapi/rdma/hfi/Kbuild | 2 +- include/uapi/scsi/Kbuild | 5 +- include/uapi/scsi/fc/Kbuild | 5 +- include/uapi/sound/Kbuild | 16 +- include/uapi/video/Kbuild | 4 +- include/uapi/xen/Kbuild | 5 +- 36 files changed, 47 insertions(+), 754 deletions(-) diff --git a/include/uapi/asm-generic/Kbuild b/include/uapi/asm-generic/Kbuild index b73de7bb7a62..8e52cdc3d941 100644 --- a/include/uapi/asm-generic/Kbuild +++ b/include/uapi/asm-generic/Kbuild @@ -1,36 +1,2 @@ # UAPI Header export list -header-y += auxvec.h -header-y += bitsperlong.h -header-y += errno-base.h -header-y += errno.h -header-y += fcntl.h -header-y += int-l64.h -header-y += int-ll64.h -header-y += ioctl.h -header-y += ioctls.h -header-y += ipcbuf.h -header-y += kvm_para.h -header-y += mman-common.h -header-y += mman.h -header-y += msgbuf.h -header-y += param.h -header-y += poll.h -header-y += posix_types.h -header-y += resource.h -header-y += sembuf.h -header-y += setup.h -header-y += shmbuf.h -header-y += shmparam.h -header-y += siginfo.h -header-y += signal-defs.h -header-y += signal.h -header-y += socket.h -header-y += sockios.h -header-y += stat.h -header-y += statfs.h -header-y += swab.h -header-y += termbits.h -header-y += termios.h -header-y += types.h -header-y += ucontext.h -header-y += unistd.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/asm-generic/*.h)) diff --git a/include/uapi/drm/Kbuild b/include/uapi/drm/Kbuild index 9355dd8eff3b..75f4cde6d9ba 100644 --- a/include/uapi/drm/Kbuild +++ b/include/uapi/drm/Kbuild @@ -1,22 +1,2 @@ # UAPI Header export list -header-y += drm.h -header-y += drm_fourcc.h -header-y += drm_mode.h -header-y += drm_sarea.h -header-y += amdgpu_drm.h -header-y += exynos_drm.h -header-y += i810_drm.h -header-y += i915_drm.h -header-y += mga_drm.h -header-y += nouveau_drm.h -header-y += qxl_drm.h -header-y += r128_drm.h -header-y += radeon_drm.h -header-y += savage_drm.h -header-y += sis_drm.h -header-y += tegra_drm.h -header-y += via_drm.h -header-y += vmwgfx_drm.h -header-y += msm_drm.h -header-y += vc4_drm.h -header-y += virtgpu_drm.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/drm/*.h)) diff --git a/include/uapi/linux/Kbuild b/include/uapi/linux/Kbuild index a8b93e685239..9d2d4ebc1e5c 100644 --- a/include/uapi/linux/Kbuild +++ b/include/uapi/linux/Kbuild @@ -25,458 +25,19 @@ header-y += wimax/ genhdr-y += version.h -ifneq ($(wildcard $(srctree)/arch/$(SRCARCH)/include/uapi/asm/a.out.h \ - $(srctree)/arch/$(SRCARCH)/include/asm/a.out.h),) -header-y += a.out.h -endif - -header-y += acct.h -header-y += adb.h -header-y += adfs_fs.h -header-y += affs_hardblocks.h -header-y += agpgart.h -header-y += aio_abi.h -header-y += am437x-vpfe.h -header-y += apm_bios.h -header-y += arcfb.h -header-y += atalk.h -header-y += atmapi.h -header-y += atmarp.h -header-y += atmbr2684.h -header-y += atmclip.h -header-y += atmdev.h -header-y += atm_eni.h -header-y += atm.h -header-y += atm_he.h -header-y += atm_idt77105.h -header-y += atmioc.h -header-y += atmlec.h -header-y += atmmpc.h -header-y += atm_nicstar.h -header-y += atmppp.h -header-y += atmsap.h -header-y += atmsvc.h -header-y += atm_tcp.h -header-y += atm_zatm.h -header-y += audit.h -header-y += auto_fs4.h -header-y += auto_fs.h -header-y += auxvec.h -header-y += ax25.h -header-y += b1lli.h -header-y += baycom.h -header-y += bcm933xx_hcs.h -header-y += bfs_fs.h -header-y += binfmts.h -header-y += blkpg.h -header-y += blktrace_api.h -header-y += blkzoned.h -header-y += bpf_common.h -header-y += bpf_perf_event.h -header-y += bpf.h -header-y += bpqether.h -header-y += bsg.h -header-y += bt-bmc.h -header-y += btrfs.h -header-y += can.h -header-y += capability.h -header-y += capi.h -header-y += cciss_defs.h -header-y += cciss_ioctl.h -header-y += cdrom.h -header-y += cec.h -header-y += cec-funcs.h -header-y += cgroupstats.h -header-y += chio.h -header-y += cm4000_cs.h -header-y += cn_proc.h -header-y += coda.h -header-y += coda_psdev.h -header-y += coff.h -header-y += connector.h -header-y += const.h -header-y += cramfs_fs.h -header-y += cuda.h -header-y += cyclades.h -header-y += cycx_cfm.h -header-y += dcbnl.h -header-y += dccp.h -header-y += devlink.h -header-y += dlmconstants.h -header-y += dlm_device.h -header-y += dlm.h -header-y += dlm_netlink.h -header-y += dlm_plock.h -header-y += dm-ioctl.h -header-y += dm-log-userspace.h -header-y += dn.h -header-y += dqblk_xfs.h -header-y += edd.h -header-y += efs_fs_sb.h -header-y += elfcore.h -header-y += elf-em.h -header-y += elf-fdpic.h -header-y += elf.h -header-y += errno.h -header-y += errqueue.h -header-y += ethtool.h -header-y += eventpoll.h -header-y += fadvise.h -header-y += falloc.h -header-y += fanotify.h -header-y += fb.h -header-y += fcntl.h -header-y += fd.h -header-y += fdreg.h -header-y += fib_rules.h -header-y += fiemap.h -header-y += filter.h -header-y += firewire-cdev.h -header-y += firewire-constants.h -header-y += flat.h -header-y += fou.h -header-y += fs.h -header-y += fsl_hypervisor.h -header-y += fuse.h -header-y += futex.h -header-y += gameport.h -header-y += genetlink.h -header-y += gen_stats.h -header-y += gfs2_ondisk.h -header-y += gigaset_dev.h -header-y += gpio.h -header-y += gsmmux.h -header-y += gtp.h -header-y += hdlcdrv.h -header-y += hdlc.h -header-y += hdreg.h -header-y += hiddev.h -header-y += hid.h -header-y += hidraw.h -header-y += hpet.h -header-y += hsr_netlink.h -header-y += hyperv.h -header-y += hysdn_if.h -header-y += i2c-dev.h -header-y += i2c.h -header-y += i2o-dev.h -header-y += i8k.h -header-y += icmp.h -header-y += icmpv6.h -header-y += if_addr.h -header-y += if_addrlabel.h -header-y += if_alg.h -header-y += if_arcnet.h -header-y += if_arp.h -header-y += if_bonding.h -header-y += if_bridge.h -header-y += if_cablemodem.h -header-y += if_eql.h -header-y += if_ether.h -header-y += if_fc.h -header-y += if_fddi.h -header-y += if_frad.h -header-y += if.h -header-y += if_hippi.h -header-y += if_infiniband.h -header-y += if_link.h -header-y += if_ltalk.h -header-y += if_macsec.h -header-y += if_packet.h -header-y += if_phonet.h -header-y += if_plip.h -header-y += if_ppp.h -header-y += if_pppol2tp.h -header-y += if_pppox.h -header-y += if_slip.h -header-y += if_team.h -header-y += if_tun.h -header-y += if_tunnel.h -header-y += if_vlan.h -header-y += if_x25.h -header-y += igmp.h -header-y += ila.h -header-y += in6.h -header-y += inet_diag.h -header-y += in.h -header-y += inotify.h -header-y += input.h -header-y += input-event-codes.h -header-y += in_route.h -header-y += ioctl.h -header-y += ip6_tunnel.h -header-y += ipc.h -header-y += ip.h -header-y += ipmi.h -header-y += ipmi_msgdefs.h -header-y += ipsec.h -header-y += ipv6.h -header-y += ipv6_route.h -header-y += ip_vs.h -header-y += ipx.h -header-y += irda.h -header-y += irqnr.h -header-y += isdn_divertif.h -header-y += isdn.h -header-y += isdnif.h -header-y += isdn_ppp.h -header-y += iso_fs.h -header-y += ivtvfb.h -header-y += ivtv.h -header-y += ixjuser.h -header-y += jffs2.h -header-y += joystick.h -header-y += kcmp.h -header-y += kdev_t.h -header-y += kd.h -header-y += kernelcapi.h -header-y += kernel.h -header-y += kernel-page-flags.h -header-y += kexec.h -header-y += keyboard.h -header-y += keyctl.h +tmphdr-y := $(notdir $(wildcard $(srctree)/include/uapi/linux/*.h)) -ifneq ($(wildcard $(srctree)/arch/$(SRCARCH)/include/uapi/asm/kvm.h \ - $(srctree)/arch/$(SRCARCH)/include/asm/kvm.h),) -header-y += kvm.h +ifeq ($(wildcard $(srctree)/arch/$(SRCARCH)/include/uapi/asm/a.out.h \ + $(srctree)/arch/$(SRCARCH)/include/asm/a.out.h),) +tmphdr-y = $(filter-out a.out.h $(tmphdr-y)) endif - - -ifneq ($(wildcard $(srctree)/arch/$(SRCARCH)/include/uapi/asm/kvm_para.h \ - $(srctree)/arch/$(SRCARCH)/include/asm/kvm_para.h),) -header-y += kvm_para.h +ifeq ($(wildcard $(srctree)/arch/$(SRCARCH)/include/uapi/asm/kvm.h \ + $(srctree)/arch/$(SRCARCH)/include/asm/kvm.h),) +tmphdr-y = $(filter-out kvm.h $(tmphdr-y)) +endif +ifeq ($(wildcard $(srctree)/arch/$(SRCARCH)/include/uapi/asm/kvm_para.h \ + $(srctree)/arch/$(SRCARCH)/include/asm/kvm_para.h),) +tmphdr-y = $(filter-out kvm_para.h $(tmphdr-y)) endif -header-y += hw_breakpoint.h -header-y += l2tp.h -header-y += libc-compat.h -header-y += lirc.h -header-y += limits.h -header-y += llc.h -header-y += loop.h -header-y += lp.h -header-y += lwtunnel.h -header-y += magic.h -header-y += major.h -header-y += map_to_7segment.h -header-y += matroxfb.h -header-y += mdio.h -header-y += media.h -header-y += media-bus-format.h -header-y += mei.h -header-y += membarrier.h -header-y += memfd.h -header-y += mempolicy.h -header-y += meye.h -header-y += mic_common.h -header-y += mic_ioctl.h -header-y += mii.h -header-y += minix_fs.h -header-y += mman.h -header-y += mmtimer.h -header-y += mpls.h -header-y += mpls_iptunnel.h -header-y += mqueue.h -header-y += mroute6.h -header-y += mroute.h -header-y += msdos_fs.h -header-y += msg.h -header-y += mtio.h -header-y += nbd.h -header-y += ncp_fs.h -header-y += ncp.h -header-y += ncp_mount.h -header-y += ncp_no.h -header-y += ndctl.h -header-y += neighbour.h -header-y += netconf.h -header-y += netdevice.h -header-y += net_dropmon.h -header-y += netfilter_arp.h -header-y += netfilter_bridge.h -header-y += netfilter_decnet.h -header-y += netfilter.h -header-y += netfilter_ipv4.h -header-y += netfilter_ipv6.h -header-y += net.h -header-y += netlink_diag.h -header-y += netlink.h -header-y += netrom.h -header-y += net_namespace.h -header-y += net_tstamp.h -header-y += nfc.h -header-y += nfs2.h -header-y += nfs3.h -header-y += nfs4.h -header-y += nfs4_mount.h -header-y += nfsacl.h -header-y += nfs_fs.h -header-y += nfs.h -header-y += nfs_idmap.h -header-y += nfs_mount.h -header-y += nl80211.h -header-y += n_r3964.h -header-y += nubus.h -header-y += nvme_ioctl.h -header-y += nvram.h -header-y += omap3isp.h -header-y += omapfb.h -header-y += oom.h -header-y += openvswitch.h -header-y += packet_diag.h -header-y += param.h -header-y += parport.h -header-y += patchkey.h -header-y += pci.h -header-y += pci_regs.h -header-y += perf_event.h -header-y += personality.h -header-y += pfkeyv2.h -header-y += pg.h -header-y += phantom.h -header-y += phonet.h -header-y += pktcdvd.h -header-y += pkt_cls.h -header-y += pkt_sched.h -header-y += pmu.h -header-y += poll.h -header-y += posix_acl.h -header-y += posix_acl_xattr.h -header-y += posix_types.h -header-y += ppdev.h -header-y += ppp-comp.h -header-y += ppp_defs.h -header-y += ppp-ioctl.h -header-y += pps.h -header-y += prctl.h -header-y += psci.h -header-y += ptp_clock.h -header-y += ptrace.h -header-y += qnx4_fs.h -header-y += qnxtypes.h -header-y += quota.h -header-y += radeonfb.h -header-y += random.h -header-y += raw.h -header-y += rds.h -header-y += reboot.h -header-y += reiserfs_fs.h -header-y += reiserfs_xattr.h -header-y += resource.h -header-y += rfkill.h -header-y += rio_cm_cdev.h -header-y += rio_mport_cdev.h -header-y += romfs_fs.h -header-y += rose.h -header-y += route.h -header-y += rtc.h -header-y += rtnetlink.h -header-y += scc.h -header-y += sched.h -header-y += scif_ioctl.h -header-y += screen_info.h -header-y += sctp.h -header-y += sdla.h -header-y += seccomp.h -header-y += securebits.h -header-y += selinux_netlink.h -header-y += sem.h -header-y += serial_core.h -header-y += serial.h -header-y += serial_reg.h -header-y += serio.h -header-y += shm.h -header-y += signalfd.h -header-y += signal.h -header-y += smiapp.h -header-y += snmp.h -header-y += sock_diag.h -header-y += socket.h -header-y += sockios.h -header-y += sonet.h -header-y += sonypi.h -header-y += soundcard.h -header-y += sound.h -header-y += stat.h -header-y += stddef.h -header-y += string.h -header-y += suspend_ioctls.h -header-y += swab.h -header-y += synclink.h -header-y += sync_file.h -header-y += sysctl.h -header-y += sysinfo.h -header-y += target_core_user.h -header-y += taskstats.h -header-y += tcp.h -header-y += tcp_metrics.h -header-y += telephony.h -header-y += termios.h -header-y += thermal.h -header-y += time.h -header-y += times.h -header-y += timex.h -header-y += tiocl.h -header-y += tipc_config.h -header-y += tipc_netlink.h -header-y += tipc.h -header-y += toshiba.h -header-y += tty_flags.h -header-y += tty.h -header-y += types.h -header-y += udf_fs_i.h -header-y += udp.h -header-y += uhid.h -header-y += uinput.h -header-y += uio.h -header-y += uleds.h -header-y += ultrasound.h -header-y += un.h -header-y += unistd.h -header-y += unix_diag.h -header-y += usbdevice_fs.h -header-y += usbip.h -header-y += utime.h -header-y += utsname.h -header-y += uuid.h -header-y += uvcvideo.h -header-y += v4l2-common.h -header-y += v4l2-controls.h -header-y += v4l2-dv-timings.h -header-y += v4l2-mediabus.h -header-y += v4l2-subdev.h -header-y += veth.h -header-y += vfio.h -header-y += vhost.h -header-y += videodev2.h -header-y += virtio_9p.h -header-y += virtio_balloon.h -header-y += virtio_blk.h -header-y += virtio_config.h -header-y += virtio_console.h -header-y += virtio_gpu.h -header-y += virtio_ids.h -header-y += virtio_input.h -header-y += virtio_net.h -header-y += virtio_pci.h -header-y += virtio_ring.h -header-y += virtio_rng.h -header-y += virtio_scsi.h -header-y += virtio_types.h -header-y += virtio_vsock.h -header-y += virtio_crypto.h -header-y += vm_sockets.h -header-y += vt.h -header-y += vtpm_proxy.h -header-y += wait.h -header-y += wanrouter.h -header-y += watchdog.h -header-y += wimax.h -header-y += wireless.h -header-y += x25.h -header-y += xattr.h -header-y += xfrm.h -header-y += xilinx-v4l2-controls.h -header-y += zorro.h -header-y += zorro_ids.h -header-y += userfaultfd.h +header-y += $(tmphdr-y) diff --git a/include/uapi/linux/android/Kbuild b/include/uapi/linux/android/Kbuild index ca011eec252a..37a629f4746a 100644 --- a/include/uapi/linux/android/Kbuild +++ b/include/uapi/linux/android/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += binder.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/android/*.h)) diff --git a/include/uapi/linux/byteorder/Kbuild b/include/uapi/linux/byteorder/Kbuild index 619225b9ff2e..d6585c79597c 100644 --- a/include/uapi/linux/byteorder/Kbuild +++ b/include/uapi/linux/byteorder/Kbuild @@ -1,3 +1,2 @@ # UAPI Header export list -header-y += big_endian.h -header-y += little_endian.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/byteorder/*.h)) diff --git a/include/uapi/linux/caif/Kbuild b/include/uapi/linux/caif/Kbuild index 43396612d3a3..0deed17a523c 100644 --- a/include/uapi/linux/caif/Kbuild +++ b/include/uapi/linux/caif/Kbuild @@ -1,3 +1,2 @@ # UAPI Header export list -header-y += caif_socket.h -header-y += if_caif.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/caif/*.h)) diff --git a/include/uapi/linux/can/Kbuild b/include/uapi/linux/can/Kbuild index 21c91bf25a29..0b7f01ea62d6 100644 --- a/include/uapi/linux/can/Kbuild +++ b/include/uapi/linux/can/Kbuild @@ -1,6 +1,2 @@ # UAPI Header export list -header-y += bcm.h -header-y += error.h -header-y += gw.h -header-y += netlink.h -header-y += raw.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/can/*.h)) diff --git a/include/uapi/linux/dvb/Kbuild b/include/uapi/linux/dvb/Kbuild index d40942cfc627..6845c2b87161 100644 --- a/include/uapi/linux/dvb/Kbuild +++ b/include/uapi/linux/dvb/Kbuild @@ -1,9 +1,2 @@ # UAPI Header export list -header-y += audio.h -header-y += ca.h -header-y += dmx.h -header-y += frontend.h -header-y += net.h -header-y += osd.h -header-y += version.h -header-y += video.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/dvb/*.h)) diff --git a/include/uapi/linux/hdlc/Kbuild b/include/uapi/linux/hdlc/Kbuild index 8c1d2cb75e33..529c2c839277 100644 --- a/include/uapi/linux/hdlc/Kbuild +++ b/include/uapi/linux/hdlc/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += ioctl.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/hdlc/*.h)) diff --git a/include/uapi/linux/hsi/Kbuild b/include/uapi/linux/hsi/Kbuild index a16a00544258..8f59b8f5f8e5 100644 --- a/include/uapi/linux/hsi/Kbuild +++ b/include/uapi/linux/hsi/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += hsi_char.h cs-protocol.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/hsi/*.h)) diff --git a/include/uapi/linux/iio/Kbuild b/include/uapi/linux/iio/Kbuild index 86f76d84c44f..aa6f9887cc4e 100644 --- a/include/uapi/linux/iio/Kbuild +++ b/include/uapi/linux/iio/Kbuild @@ -1,3 +1,2 @@ # UAPI Header export list -header-y += events.h -header-y += types.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/iio/*.h)) diff --git a/include/uapi/linux/isdn/Kbuild b/include/uapi/linux/isdn/Kbuild index 89e52850bf29..1e842431b0a8 100644 --- a/include/uapi/linux/isdn/Kbuild +++ b/include/uapi/linux/isdn/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += capicmd.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/isdn/*.h)) diff --git a/include/uapi/linux/mmc/Kbuild b/include/uapi/linux/mmc/Kbuild index 8c1d2cb75e33..02d0c0605d1a 100644 --- a/include/uapi/linux/mmc/Kbuild +++ b/include/uapi/linux/mmc/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += ioctl.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/mmc/*.h)) diff --git a/include/uapi/linux/netfilter/Kbuild b/include/uapi/linux/netfilter/Kbuild index 03f194aeadc5..a27c332c489c 100644 --- a/include/uapi/linux/netfilter/Kbuild +++ b/include/uapi/linux/netfilter/Kbuild @@ -1,89 +1,3 @@ # UAPI Header export list header-y += ipset/ -header-y += nf_conntrack_common.h -header-y += nf_conntrack_ftp.h -header-y += nf_conntrack_sctp.h -header-y += nf_conntrack_tcp.h -header-y += nf_conntrack_tuple_common.h -header-y += nf_log.h -header-y += nf_tables.h -header-y += nf_tables_compat.h -header-y += nf_nat.h -header-y += nfnetlink.h -header-y += nfnetlink_acct.h -header-y += nfnetlink_compat.h -header-y += nfnetlink_conntrack.h -header-y += nfnetlink_cthelper.h -header-y += nfnetlink_cttimeout.h -header-y += nfnetlink_log.h -header-y += nfnetlink_queue.h -header-y += x_tables.h -header-y += xt_AUDIT.h -header-y += xt_CHECKSUM.h -header-y += xt_CLASSIFY.h -header-y += xt_CONNMARK.h -header-y += xt_CONNSECMARK.h -header-y += xt_CT.h -header-y += xt_DSCP.h -header-y += xt_HMARK.h -header-y += xt_IDLETIMER.h -header-y += xt_LED.h -header-y += xt_LOG.h -header-y += xt_MARK.h -header-y += xt_NFLOG.h -header-y += xt_NFQUEUE.h -header-y += xt_RATEEST.h -header-y += xt_SECMARK.h -header-y += xt_SYNPROXY.h -header-y += xt_TCPMSS.h -header-y += xt_TCPOPTSTRIP.h -header-y += xt_TEE.h -header-y += xt_TPROXY.h -header-y += xt_addrtype.h -header-y += xt_bpf.h -header-y += xt_cgroup.h -header-y += xt_cluster.h -header-y += xt_comment.h -header-y += xt_connbytes.h -header-y += xt_connlabel.h -header-y += xt_connlimit.h -header-y += xt_connmark.h -header-y += xt_conntrack.h -header-y += xt_cpu.h -header-y += xt_dccp.h -header-y += xt_devgroup.h -header-y += xt_dscp.h -header-y += xt_ecn.h -header-y += xt_esp.h -header-y += xt_hashlimit.h -header-y += xt_helper.h -header-y += xt_ipcomp.h -header-y += xt_iprange.h -header-y += xt_ipvs.h -header-y += xt_l2tp.h -header-y += xt_length.h -header-y += xt_limit.h -header-y += xt_mac.h -header-y += xt_mark.h -header-y += xt_multiport.h -header-y += xt_nfacct.h -header-y += xt_osf.h -header-y += xt_owner.h -header-y += xt_physdev.h -header-y += xt_pkttype.h -header-y += xt_policy.h -header-y += xt_quota.h -header-y += xt_rateest.h -header-y += xt_realm.h -header-y += xt_recent.h -header-y += xt_rpfilter.h -header-y += xt_sctp.h -header-y += xt_set.h -header-y += xt_socket.h -header-y += xt_state.h -header-y += xt_statistic.h -header-y += xt_string.h -header-y += xt_tcpmss.h -header-y += xt_tcpudp.h -header-y += xt_time.h -header-y += xt_u32.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/netfilter/*.h)) diff --git a/include/uapi/linux/netfilter/ipset/Kbuild b/include/uapi/linux/netfilter/ipset/Kbuild index d2680423d9ab..b6dc00483ac2 100644 --- a/include/uapi/linux/netfilter/ipset/Kbuild +++ b/include/uapi/linux/netfilter/ipset/Kbuild @@ -1,5 +1,2 @@ # UAPI Header export list -header-y += ip_set.h -header-y += ip_set_bitmap.h -header-y += ip_set_hash.h -header-y += ip_set_list.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/netfilter/ipset/*.h)) diff --git a/include/uapi/linux/netfilter_arp/Kbuild b/include/uapi/linux/netfilter_arp/Kbuild index 62d5637cc0ac..63b09feaf660 100644 --- a/include/uapi/linux/netfilter_arp/Kbuild +++ b/include/uapi/linux/netfilter_arp/Kbuild @@ -1,3 +1,2 @@ # UAPI Header export list -header-y += arp_tables.h -header-y += arpt_mangle.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/netfilter_arp/*.h)) diff --git a/include/uapi/linux/netfilter_bridge/Kbuild b/include/uapi/linux/netfilter_bridge/Kbuild index 0fbad8ef96de..6bbfc5a91965 100644 --- a/include/uapi/linux/netfilter_bridge/Kbuild +++ b/include/uapi/linux/netfilter_bridge/Kbuild @@ -1,18 +1,2 @@ # UAPI Header export list -header-y += ebt_802_3.h -header-y += ebt_among.h -header-y += ebt_arp.h -header-y += ebt_arpreply.h -header-y += ebt_ip.h -header-y += ebt_ip6.h -header-y += ebt_limit.h -header-y += ebt_log.h -header-y += ebt_mark_m.h -header-y += ebt_mark_t.h -header-y += ebt_nat.h -header-y += ebt_nflog.h -header-y += ebt_pkttype.h -header-y += ebt_redirect.h -header-y += ebt_stp.h -header-y += ebt_vlan.h -header-y += ebtables.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/netfilter_bridge/*.h)) diff --git a/include/uapi/linux/netfilter_ipv4/Kbuild b/include/uapi/linux/netfilter_ipv4/Kbuild index ecb291df390e..273777606d02 100644 --- a/include/uapi/linux/netfilter_ipv4/Kbuild +++ b/include/uapi/linux/netfilter_ipv4/Kbuild @@ -1,10 +1,2 @@ # UAPI Header export list -header-y += ip_tables.h -header-y += ipt_CLUSTERIP.h -header-y += ipt_ECN.h -header-y += ipt_LOG.h -header-y += ipt_REJECT.h -header-y += ipt_TTL.h -header-y += ipt_ah.h -header-y += ipt_ecn.h -header-y += ipt_ttl.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/netfilter_ipv4/*.h)) diff --git a/include/uapi/linux/netfilter_ipv6/Kbuild b/include/uapi/linux/netfilter_ipv6/Kbuild index 75a668ca2353..2d3507f5b0aa 100644 --- a/include/uapi/linux/netfilter_ipv6/Kbuild +++ b/include/uapi/linux/netfilter_ipv6/Kbuild @@ -1,13 +1,2 @@ # UAPI Header export list -header-y += ip6_tables.h -header-y += ip6t_HL.h -header-y += ip6t_LOG.h -header-y += ip6t_NPT.h -header-y += ip6t_REJECT.h -header-y += ip6t_ah.h -header-y += ip6t_frag.h -header-y += ip6t_hl.h -header-y += ip6t_ipv6header.h -header-y += ip6t_mh.h -header-y += ip6t_opts.h -header-y += ip6t_rt.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/netfilter_ipv6/*.h)) diff --git a/include/uapi/linux/nfsd/Kbuild b/include/uapi/linux/nfsd/Kbuild index c11bc404053c..b66c1120b54f 100644 --- a/include/uapi/linux/nfsd/Kbuild +++ b/include/uapi/linux/nfsd/Kbuild @@ -1,6 +1,2 @@ # UAPI Header export list -header-y += cld.h -header-y += debug.h -header-y += export.h -header-y += nfsfh.h -header-y += stats.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/nfsd/*.h)) diff --git a/include/uapi/linux/raid/Kbuild b/include/uapi/linux/raid/Kbuild index e2c3d25405d7..409aa2ecd82f 100644 --- a/include/uapi/linux/raid/Kbuild +++ b/include/uapi/linux/raid/Kbuild @@ -1,3 +1,2 @@ # UAPI Header export list -header-y += md_p.h -header-y += md_u.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/raid/*.h)) diff --git a/include/uapi/linux/spi/Kbuild b/include/uapi/linux/spi/Kbuild index 0cc747eff165..8d269bd5e31a 100644 --- a/include/uapi/linux/spi/Kbuild +++ b/include/uapi/linux/spi/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += spidev.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/spi/*.h)) diff --git a/include/uapi/linux/sunrpc/Kbuild b/include/uapi/linux/sunrpc/Kbuild index 8e02e47c20fb..4a880e5aeba3 100644 --- a/include/uapi/linux/sunrpc/Kbuild +++ b/include/uapi/linux/sunrpc/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += debug.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/sunrpc/*.h)) diff --git a/include/uapi/linux/tc_act/Kbuild b/include/uapi/linux/tc_act/Kbuild index e3db7403296f..24f8b1d9a4ed 100644 --- a/include/uapi/linux/tc_act/Kbuild +++ b/include/uapi/linux/tc_act/Kbuild @@ -1,15 +1,2 @@ # UAPI Header export list -header-y += tc_csum.h -header-y += tc_defact.h -header-y += tc_gact.h -header-y += tc_ipt.h -header-y += tc_mirred.h -header-y += tc_nat.h -header-y += tc_pedit.h -header-y += tc_skbedit.h -header-y += tc_vlan.h -header-y += tc_bpf.h -header-y += tc_connmark.h -header-y += tc_ife.h -header-y += tc_tunnel_key.h -header-y += tc_skbmod.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/tc_act/*.h)) diff --git a/include/uapi/linux/tc_ematch/Kbuild b/include/uapi/linux/tc_ematch/Kbuild index 53fca3925535..909ef0d196b1 100644 --- a/include/uapi/linux/tc_ematch/Kbuild +++ b/include/uapi/linux/tc_ematch/Kbuild @@ -1,5 +1,2 @@ # UAPI Header export list -header-y += tc_em_cmp.h -header-y += tc_em_meta.h -header-y += tc_em_nbyte.h -header-y += tc_em_text.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/tc_ematch/*.h)) diff --git a/include/uapi/linux/usb/Kbuild b/include/uapi/linux/usb/Kbuild index 4cc4d6e7e523..4a5eb3f2b704 100644 --- a/include/uapi/linux/usb/Kbuild +++ b/include/uapi/linux/usb/Kbuild @@ -1,12 +1,2 @@ # UAPI Header export list -header-y += audio.h -header-y += cdc.h -header-y += cdc-wdm.h -header-y += ch11.h -header-y += ch9.h -header-y += functionfs.h -header-y += g_printer.h -header-y += gadgetfs.h -header-y += midi.h -header-y += tmc.h -header-y += video.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/usb/*.h)) diff --git a/include/uapi/linux/wimax/Kbuild b/include/uapi/linux/wimax/Kbuild index 1c97be49971f..f9f41558ebbe 100644 --- a/include/uapi/linux/wimax/Kbuild +++ b/include/uapi/linux/wimax/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += i2400m.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/linux/wimax/*.h)) diff --git a/include/uapi/misc/Kbuild b/include/uapi/misc/Kbuild index e96cae7d58c9..c4ad43dfbcfa 100644 --- a/include/uapi/misc/Kbuild +++ b/include/uapi/misc/Kbuild @@ -1,2 +1,2 @@ # misc Header export list -header-y += cxl.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/misc/*.h)) diff --git a/include/uapi/mtd/Kbuild b/include/uapi/mtd/Kbuild index 5a691e10cd0e..0fbbdecf0302 100644 --- a/include/uapi/mtd/Kbuild +++ b/include/uapi/mtd/Kbuild @@ -1,6 +1,2 @@ # UAPI Header export list -header-y += inftl-user.h -header-y += mtd-abi.h -header-y += mtd-user.h -header-y += nftl-user.h -header-y += ubi-user.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/mtd/*.h)) diff --git a/include/uapi/rdma/Kbuild b/include/uapi/rdma/Kbuild index 82bdf5626859..0c51ce6905fb 100644 --- a/include/uapi/rdma/Kbuild +++ b/include/uapi/rdma/Kbuild @@ -1,18 +1,3 @@ # UAPI Header export list -header-y += ib_user_cm.h -header-y += ib_user_mad.h -header-y += ib_user_sa.h -header-y += ib_user_verbs.h -header-y += rdma_netlink.h -header-y += rdma_user_cm.h header-y += hfi/ -header-y += rdma_user_rxe.h -header-y += cxgb3-abi.h -header-y += cxgb4-abi.h -header-y += mlx4-abi.h -header-y += mlx5-abi.h -header-y += mthca-abi.h -header-y += nes-abi.h -header-y += ocrdma-abi.h -header-y += hns-abi.h -header-y += vmw_pvrdma-abi.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/rdma/*.h)) diff --git a/include/uapi/rdma/hfi/Kbuild b/include/uapi/rdma/hfi/Kbuild index ef23c294fc71..4fef4d891000 100644 --- a/include/uapi/rdma/hfi/Kbuild +++ b/include/uapi/rdma/hfi/Kbuild @@ -1,2 +1,2 @@ # UAPI Header export list -header-y += hfi1_user.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/rdma/hfi/*.h)) diff --git a/include/uapi/scsi/Kbuild b/include/uapi/scsi/Kbuild index d791e0ad509d..f3f1df5e24dd 100644 --- a/include/uapi/scsi/Kbuild +++ b/include/uapi/scsi/Kbuild @@ -1,6 +1,3 @@ # UAPI Header export list header-y += fc/ -header-y += scsi_bsg_fc.h -header-y += scsi_netlink.h -header-y += scsi_netlink_fc.h -header-y += cxlflash_ioctl.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/scsi/*.h)) diff --git a/include/uapi/scsi/fc/Kbuild b/include/uapi/scsi/fc/Kbuild index 5ead9fac265c..1b84093983c5 100644 --- a/include/uapi/scsi/fc/Kbuild +++ b/include/uapi/scsi/fc/Kbuild @@ -1,5 +1,2 @@ # UAPI Header export list -header-y += fc_els.h -header-y += fc_fs.h -header-y += fc_gs.h -header-y += fc_ns.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/scsi/fc/*.h)) diff --git a/include/uapi/sound/Kbuild b/include/uapi/sound/Kbuild index 9578d8bdbf31..d2eb3b2aedf3 100644 --- a/include/uapi/sound/Kbuild +++ b/include/uapi/sound/Kbuild @@ -1,16 +1,2 @@ # UAPI Header export list -header-y += asequencer.h -header-y += asoc.h -header-y += asound.h -header-y += asound_fm.h -header-y += compress_offload.h -header-y += compress_params.h -header-y += emu10k1.h -header-y += firewire.h -header-y += hdsp.h -header-y += hdspm.h -header-y += sb16_csp.h -header-y += sfnt_info.h -header-y += tlv.h -header-y += usb_stream.h -header-y += snd_sst_tokens.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/sound/*.h)) diff --git a/include/uapi/video/Kbuild b/include/uapi/video/Kbuild index ac7203bb32cc..cd6d03c1e7ff 100644 --- a/include/uapi/video/Kbuild +++ b/include/uapi/video/Kbuild @@ -1,4 +1,2 @@ # UAPI Header export list -header-y += edid.h -header-y += sisfb.h -header-y += uvesafb.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/video/*.h)) diff --git a/include/uapi/xen/Kbuild b/include/uapi/xen/Kbuild index 5c459628e8c7..32feafaaa784 100644 --- a/include/uapi/xen/Kbuild +++ b/include/uapi/xen/Kbuild @@ -1,5 +1,2 @@ # UAPI Header export list -header-y += evtchn.h -header-y += gntalloc.h -header-y += gntdev.h -header-y += privcmd.h +header-y += $(notdir $(wildcard $(srctree)/include/uapi/xen/*.h)) -- 2.8.1

13 37

Re: [alsa-devel] [PATCH v3 4/8] x86: stop exporting msr-index.h to userland
by Nicolas Dichtel 13 Jan '17

13 Jan '17

Le 13/01/2017 à 16:43, David Howells a écrit : >> -header-y += msr-index.h > > I see it on my desktop as /usr/include/asm/msr-index.h and it's been there at > least four years - and as such it's part of the UAPI. I don't think you can > remove it unless you can guarantee there are no userspace users. I keep it in the v2 of the series, but the maintainer, Borislav Petkov, asks me to un-export it. I will follow the maintainer decision. Regards, Nicolas

2 1

Re: [alsa-devel] [PATCH v3 1/8] arm: put types.h in uapi
by Nicolas Dichtel 13 Jan '17

13 Jan '17

Please, do not remove the email subject when you reply. I restore it to ease the thread follow-up. Le 13/01/2017 à 16:36, David Howells a écrit : > Nicolas Dichtel <nicolas.dichtel(a)6wind.com> wrote: > >> This header file is exported, thus move it to uapi. > > Exported how? It is listed in include/uapi/asm-generic/Kbuild.asm, which is included by arch/arm/include/uapi/asm/Kbuild. You can also have a look at patch #5 to see why it was exported even if it was not in an uapi directory. Regards, Nicolas

2 1

[alsa-devel] [GIT PULL] intel: audio firmware updates
by Vinod Koul 13 Jan '17

13 Jan '17

Hi, Please pull to receive the updated versions of skylake and broxton firmware binaries. The following changes since commit 4b9559fb0565f582a21a4070a81a3b6a7d77610b: linux-firmware: update firmware file for atusb (2016-12-07 01:01:04 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to 4529fb4b320ba8e71a213723d0ad029058b15912: linux-firmware: intel: Update Broxton audio firmware (2017-01-12 14:58:44 +0530) ---------------------------------------------------------------- Jeeja KP (1): linux-firmware: intel: Update Skylake audio firmware Subhransu S. Prusty (1): linux-firmware: intel: Update Broxton audio firmware WHENCE | 8 ++++++-- intel/dsp_fw_bxtn.bin | 2 +- intel/dsp_fw_bxtn_v2010.bin | Bin 0 -> 505608 bytes intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v969.bin | Bin 0 -> 221184 bytes 5 files changed, 8 insertions(+), 4 deletions(-) create mode 100644 intel/dsp_fw_bxtn_v2010.bin create mode 100644 intel/dsp_fw_release_v969.bin -- ~Vinod

2 1

[alsa-devel] [PATCH] conf/ucm: broxton: add broxton-rt298 conf files
by Vinod Koul 13 Jan '17

13 Jan '17

This adds the UCM conf files for broxton enabling with rt298 codec on I2S audio, HDMI and DMIC ports. Signed-off-by: Nishit Sharma <nishitx.sharma(a)intel.com> Signed-off-by: G Kranthi <gudishax.kranthikumar(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/broxton-rt298/Hdmi1 | 22 +++++ src/conf/ucm/broxton-rt298/Hdmi2 | 14 +++ src/conf/ucm/broxton-rt298/HiFi | 129 ++++++++++++++++++++++++++ src/conf/ucm/broxton-rt298/Makefile.am | 4 + src/conf/ucm/broxton-rt298/broxton-rt298.conf | 58 ++++++++++++ 7 files changed, 229 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/broxton-rt298/Hdmi1 create mode 100644 src/conf/ucm/broxton-rt298/Hdmi2 create mode 100644 src/conf/ucm/broxton-rt298/HiFi create mode 100644 src/conf/ucm/broxton-rt298/Makefile.am create mode 100644 src/conf/ucm/broxton-rt298/broxton-rt298.conf diff --git a/configure.ac b/configure.ac index 71bd85271009..d645ab6c3e4c 100644 --- a/configure.ac +++ b/configure.ac @@ -684,6 +684,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/topology/bxtrt298/Makefile \ alsalisp/Makefile aserver/Makefile \ test/Makefile test/lsb/Makefile \ + src/conf/ucm/broxton-rt298/Makefile \ utils/Makefile utils/alsa-lib.spec utils/alsa.pc) dnl Create asoundlib.h dynamically according to configure options diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index a11d324e785b..ea9b3850e2dd 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c broxton-rt298 diff --git a/src/conf/ucm/broxton-rt298/Hdmi1 b/src/conf/ucm/broxton-rt298/Hdmi1 new file mode 100644 index 000000000000..48514f47efaa --- /dev/null +++ b/src/conf/ucm/broxton-rt298/Hdmi1 @@ -0,0 +1,22 @@ +# Usecase for device HDMI1/Display Port stereo playback on Intel SKYLAKE/KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi1" { + Comment "HDMI/Display Port 1 Stereo" + + EnableSequence [ + exec "echo Hdmi1 EnableSequnece" + ] + + DisableSequence [ + exec "echo Hdmi1 DisableSequnece" + ] + + Value { + PlaybackPCM "hw:broxtonrt298,4" + PlaybackChannels "2" + PlaybackPriority "3" + JackControl "HDMI/DP, pcm=4 Jack" + } +} + diff --git a/src/conf/ucm/broxton-rt298/Hdmi2 b/src/conf/ucm/broxton-rt298/Hdmi2 new file mode 100644 index 000000000000..764f8b9995f8 --- /dev/null +++ b/src/conf/ucm/broxton-rt298/Hdmi2 @@ -0,0 +1,14 @@ +# Usecase for device HDMI2/Display Port stereo playback on Intel Broxton platforms +# For Audio in I2S mode + +SectionDevice."Hdmi2" { + Comment "HDMI/Display Port 2 Stereo" + + Value { + PlaybackPCM "hw:broxtonrt298,5" + PlaybackChannels "2" + PlaybackPriority "4" + JackControl "HDMI/DP, pcm=5 Jack" + } +} + diff --git a/src/conf/ucm/broxton-rt298/HiFi b/src/conf/ucm/broxton-rt298/HiFi new file mode 100644 index 000000000000..2e2d6fec0cc6 --- /dev/null +++ b/src/conf/ucm/broxton-rt298/HiFi @@ -0,0 +1,129 @@ +# Usecase for stereo playback Speaker and Headset, Recording on DMIC and Headset MIC. +# For Audio in I2S mode on Intel Broxton platforms + +SectionVerb { + + EnableSequence [ + cdev "hw:broxtonrt298" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + ] + + DisableSequence [ + cdev "hw:broxtonrt298" + ] + + Value { + TQ "HiFi" + CapturePCM "hw:broxtonrt298,1" + PlaybackPCM "hw:broxtonrt298,0" + } +} + +SectionDevice."dmiccap" { + Comment "DMIC Stereo" + + ConflictingDevice [ + "Headset" + ] + + EnableSequence [ + + cdev "hw:broxtonrt298" + exec "echo broxtonrt298 dmiccap called" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 0" + cset "name='Mic Jack Switch' 0" + + ] + + DisableSequence [ + cdev "hw:broxtonrt298" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 0" + cset "name='Mic Jack Switch' 1" + cset "name='ADC 0 Mux' 0" + ] + Value { + CaptureChannels "2" + CapturePriority "2" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + ] + EnableSequence [ + cdev "hw:broxtonrt298" + exec "echo broxtonrt298 Headphone called" + cset "name='HPO L Switch' 1" + cset "name='HPO R Switch' 1" + cset "name='Headphone Jack Switch' 1" + cset "name='Speaker Playback Switch' 0,0" + ] + + DisableSequence [ + cdev "hw:broxtonrt298" + ] + Value { + PlaybackChannels "2" + PlaybackPriority "1" + JackControl "Headphone Jack" + JackHWMute "Speaker" + } +} + +SectionDevice."Speaker" { + Comment "Speaker" + + ConflictingDevice [ + "Headphones" + ] + EnableSequence [ + cdev "hw:broxtonrt298" + exec "echo broxtonrt298 speaker called" + cset "name='SPO Switch' 1" + cset "name='Speaker Playback Switch' 1,1" + cset "name='Speaker Switch' 1" + cset "name='HPO L Switch' 0" + cset "name='HPO R Switch' 0" + ] + + DisableSequence [ + ] + Value { + PlaybackChannels "2" + PlaybackPriority "1" + JackHWMute "Headphones" + } +} + +SectionDevice."Headset" { + Comment "Headset Mic" + + ConflictingDevice [ + "dmiccap" + ] + EnableSequence [ + cdev "hw:broxtonrt298" + exec "echo broxtonrt298 Headset called" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='ADC0 Capture Volume' 105,105" + cset "name='ADC 0 Mux' 0" + cset "name='Mic Jack Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 0" + ] + + DisableSequence [ + cdev "hw:broxtonrt298" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 0" + ] + Value { + CaptureChannels "2" + CapturePriority "2" + JackControl "Mic Jack" + } +} diff --git a/src/conf/ucm/broxton-rt298/Makefile.am b/src/conf/ucm/broxton-rt298/Makefile.am new file mode 100644 index 000000000000..80f480ec4222 --- /dev/null +++ b/src/conf/ucm/broxton-rt298/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/broxton-rt298 +ucm_DATA = broxton-rt298.conf HiFi Hdmi1 Hdmi2 +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/broxton-rt298/broxton-rt298.conf b/src/conf/ucm/broxton-rt298/broxton-rt298.conf new file mode 100644 index 000000000000..3f3d0be6fdf2 --- /dev/null +++ b/src/conf/ucm/broxton-rt298/broxton-rt298.conf @@ -0,0 +1,58 @@ +# UCM for Intel Broxton platforms +# For Audio in I2S mode + +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play and record HiFi quality Music" +} + +SectionUseCase."Hdmi1" { + File "Hdmi1" + Comment "Play on Hdmi/DP 1" +} + +SectionUseCase."Hdmi2" { + File "Hdmi2" + Comment "Play on Hdmi/DP 2" +} + +ValueDefaults { + PlaybackCTL "hw:broxtonrt298" + CaptureCTL "hw:broxtonrt298" + CaptureChannels "2" + CapturePriority "2" +} + +SectionDefaults [ + cdev "hw:broxtonrt298" + exec "echo broxtonrt298 CONF Defaults" + cset "name='Headphone Jack Switch' 1" + cset "name='Speaker Switch' 1" + cset "name='Speaker Playback Switch' 0,0" + cset "name='Front DAC Switch' 1" + cset "name='Front RECMIX Switch' 0" + cset "name='Mic Jack Switch' 1" + cset "name='ADC 0 Mux' 2" + cset "name='ADC 1 Mux' 0" + cset "name='DAC0 Playback Volume' 100,100" + cset "name='HPO L Switch' 0" + cset "name='HPO Mux' 0" + cset "name='HPO R Switch' 0" + cset "name='Pin 5 Mux' 1" + cset "name='Pin 6 Mux' 2" + cset "name='Pin 7 Mux' 3" + cset "name='RECMIX Beep Switch' 0" + cset "name='RECMIX Line1 Switch' 0" + cset "name='RECMIX Mic1 Switch' 0" + cset "name='SPK Mux' 0" + cset "name='SPO Switch' 1" + cset "name='codec0_out mo codec0_in mi Switch' 0" + cset "name='codec0_out mo dmic01_hifi_in mi Switch' 0" + cset "name='codec0_out mo media0_in mi Switch' 1" + cset "name='codec1_out mo codec0_in mi Switch' 0" + cset "name='codec1_out mo dmic01_hifi_in mi Switch' 0" + cset "name='codec1_out mo media0_in mi Switch' 0" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo media0_in mi Switch' 0" +] -- 1.9.1

3 3

[alsa-devel] [PATCH v6] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Matt Ranostay 13 Jan '17

13 Jan '17

We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Basing the QoS latency calculation on the FIFO size, threshold, sample rate, and channels. Based on the original patch by Tony Lindgren Link: https://patchwork.kernel.org/patch/9305867/ Cc: Tony Lindgren <tony(a)atomide.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> --- Changes from v1: * add calculations for latency per number of FIFO locations Changes from v2: * add missing mcbsp.h header change Changes from v3: * base the latency calculations on threshold, buffer size, sample rate, and channels Changes from v4: * using Peter Ujfalusi's suggestions for restoring a higher latency on audio stream completion, or if not applicable remove the QoS request Changes from v5: * clean up latency checking logic * move logic to .prepare and .shutdown to avoid functions that can sleep Changes from v6: * move QoS removal to asoc_mcbsp_remove from omap_mcbsp_cleanup sound/soc/omap/mcbsp.c | 1 + sound/soc/omap/mcbsp.h | 3 +++ sound/soc/omap/omap-mcbsp.c | 47 ++++++++++++++++++++++++++++++++++++++++++++- 3 files changed, 50 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 06fec5699cc8..8fdb949a266e 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 61e93b1c185d..46ae1269a698 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -323,8 +323,11 @@ struct omap_mcbsp { unsigned int fmt; unsigned int in_freq; + unsigned int latency[2]; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d018e966e533..3b8c9724d496 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -157,6 +157,16 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + + mcbsp->latency[stream1] = 0; + if (mcbsp->latency[stream2]) + pm_qos_update_request(&mcbsp->pm_qos_req, + mcbsp->latency[stream2]); + else + pm_qos_remove_request(&mcbsp->pm_qos_req); if (!cpu_dai->active) { omap_mcbsp_free(mcbsp); @@ -164,6 +174,28 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, } } +static int omap_mcbsp_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct pm_qos_request *pm_qos_req = &mcbsp->pm_qos_req; + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + int latency = mcbsp->latency[stream2]; + + /* Prevent omap hardware from hitting off between FIFO fills */ + if (!latency || mcbsp->latency[stream1] < latency) + latency = mcbsp->latency[stream1]; + + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, latency); + else if (latency) + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency); + + return 0; +} + static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -226,6 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int wlen, channels, wpf; int pkt_size = 0; unsigned int format, div, framesize, master; + unsigned int buffer_size = mcbsp->pdata->buffer_size; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); channels = params_channels(params); @@ -240,7 +273,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (mcbsp->pdata->buffer_size) { + if (buffer_size) { + int latency; + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; int divider = 0; @@ -271,6 +306,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Use packet mode for non mono streams */ pkt_size = channels; } + + latency = ((((buffer_size - pkt_size) / channels) * 1000) + / (params->rate_num / params->rate_den)); + + mcbsp->latency[substream->stream] = latency; + omap_mcbsp_set_threshold(substream, pkt_size); } @@ -554,6 +595,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, static const struct snd_soc_dai_ops mcbsp_dai_ops = { .startup = omap_mcbsp_dai_startup, .shutdown = omap_mcbsp_dai_shutdown, + .prepare = omap_mcbsp_dai_prepare, .trigger = omap_mcbsp_dai_trigger, .delay = omap_mcbsp_dai_delay, .hw_params = omap_mcbsp_dai_hw_params, @@ -835,6 +877,9 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); + if (pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + omap_mcbsp_cleanup(mcbsp); clk_put(mcbsp->fclk); -- 2.10.2

2 2

[alsa-devel] [PATCH v2 0/2] ASoC: Intel: Skylake: Driver updates on DSP firmware
by jeeja.kp＠intel.com 12 Jan '17

12 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides update on DSP firmware download by optimizing o updating the dsp register poll method to use accurate timeout. o optimizing ROM init retries. o saving the firmware/library context at boot and use this ctx for downloading the firmware to DSP memory. o cleanup to store the library manifest data o release the firmware in cleanup routine. Changes in v2: - changed the title to remove fix as this is optimization. - removed already merged 4 patchs Jeeja KP (2): ASoC: Intel: bxtn: Store the FW/Library context at boot ASoC: Intel: bxtn: optimize ROM init retries sound/soc/intel/skylake/bxt-sst.c | 82 +++++++++++++++++++++++++-------------- 1 file changed, 52 insertions(+), 30 deletions(-) -- 2.5.0

3 4

[alsa-devel] [PATCH] ASoC: rt5651: use msleep for large delays
by Nicholas Mc Guire 12 Jan '17

12 Jan '17

ulseep_range() uses hrtimers and provides no advantage over msleep() for larger delays. Fix up the 75/85ms delays here to use msleep() and reduce the load on the hrtimer subsystem. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- As these are non-atomic regions and a delay of 75/85 ms implies that there is almost certainty multiple context switches occurring in the sleep time and thus relatively high jitter must be assumed with respect to actual wakeup time implying that there is little point in using high-resolution timers here. Patch was compile tested with: x86_64_defconfig + SND_SOC=m SND_SOC_INTEL_BYTCR_RT5651_MACH=m (implies CONFIG_SND_SOC_RT5651) Patch is aginast 4.10-rc3 (localversion-next is next-20170111) sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index f5d3415..fb592b0 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -802,7 +802,7 @@ static int rt5651_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: rt5651->hp_mute = 1; - usleep_range(70000, 75000); + msleep(75); break; default: @@ -822,7 +822,7 @@ static int rt5651_hp_post_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: if (!rt5651->hp_mute) - usleep_range(80000, 85000); + msleep(85); break; -- 2.1.4

4 7

[alsa-devel] [PATCH] ASoC: rt5659: drop double const
by Nicholas Mc Guire 12 Jan '17

12 Jan '17

Drop the const qualifier as it is being added by SOC_ENUM_DOUBLE_DECL() already which is called by SOC_ENUM_SINGLE_DECL() as well as the double const by calls to SOC_VALUE_ENUM_SINGLE_DECL() via SOC_VALUE_ENUM_DOUBLE_DECL). Fixes: commit d3cb2de2479b ("ASoC: rt5659: add rt5659 codec driver") Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Problem found by sparse: sound/soc/codecs/rt5659.c:1153:14: warning: duplicate const sound/soc/codecs/rt5659.c:1156:14: warning: duplicate const sound/soc/codecs/rt5659.c:1159:14: warning: duplicate const sound/soc/codecs/rt5659.c:1162:14: warning: duplicate const sound/soc/codecs/rt5659.c:1165:14: warning: duplicate const sound/soc/codecs/rt5659.c:1168:14: warning: duplicate const sound/soc/codecs/rt5659.c:1171:14: warning: duplicate const sound/soc/codecs/rt5659.c:1174:14: warning: duplicate const sound/soc/codecs/rt5659.c:1210:14: warning: duplicate const sound/soc/codecs/rt5659.c:1214:14: warning: duplicate const sound/soc/codecs/rt5659.c:1218:14: warning: duplicate const sound/soc/codecs/rt5659.c:1222:14: warning: duplicate const sound/soc/codecs/rt5659.c:1226:14: warning: duplicate const sound/soc/codecs/rt5659.c:1230:14: warning: duplicate const sound/soc/codecs/rt5659.c:1234:14: warning: duplicate const sound/soc/codecs/rt5659.c:1933:14: warning: duplicate const sound/soc/codecs/rt5659.c:1940:14: warning: duplicate const sound/soc/codecs/rt5659.c:1954:14: warning: duplicate const sound/soc/codecs/rt5659.c:1967:14: warning: duplicate const sound/soc/codecs/rt5659.c:1980:14: warning: duplicate const sound/soc/codecs/rt5659.c:1993:14: warning: duplicate const sound/soc/codecs/rt5659.c:2007:14: warning: duplicate const sound/soc/codecs/rt5659.c:2021:14: warning: duplicate const sound/soc/codecs/rt5659.c:2034:14: warning: duplicate const sound/soc/codecs/rt5659.c:2041:14: warning: duplicate const sound/soc/codecs/rt5659.c:2054:14: warning: duplicate const sound/soc/codecs/rt5659.c:2067:14: warning: duplicate const sound/soc/codecs/rt5659.c:2080:14: warning: duplicate const sound/soc/codecs/rt5659.c:2093:14: warning: duplicate const sound/soc/codecs/rt5659.c:2107:14: warning: duplicate const sound/soc/codecs/rt5659.c:2114:14: warning: duplicate const sound/soc/codecs/rt5659.c:2127:14: warning: duplicate const sound/soc/codecs/rt5659.c:2134:14: warning: duplicate const sound/soc/codecs/rt5659.c:2147:14: warning: duplicate const sound/soc/codecs/rt5659.c:2154:14: warning: duplicate const sound/soc/codecs/rt5659.c:2167:14: warning: duplicate const sound/soc/codecs/rt5659.c:2174:14: warning: duplicate const sound/soc/codecs/rt5659.c:2187:14: warning: duplicate const sound/soc/codecs/rt5659.c:2200:14: warning: duplicate const sound/soc/codecs/rt5659.c:2213:14: warning: duplicate const sound/soc/codecs/rt5659.c:2220:14: warning: duplicate const sound/soc/codecs/rt5659.c:2233:14: warning: duplicate const sound/soc/codecs/rt5659.c:2253:14: warning: duplicate const which is due to: static const SOC_ENUM_SINGLE_DECL(... which via: #define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \ SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) expands to: #define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \ const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ ARRAY_SIZE(xtexts), xtexts) and that already carries the const qualifier so that results in the "duplicate const" from sparse. Likewise static const SOC_VALUE_ENUM_SINGLE_DECL(... via: #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) which expands to: #define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ARRAY_SIZE(xtexts), xtexts, xvalues) results in a double const qualifier. The fix is simply to drop the first const. sound/soc/codecs/rt5659.c | 86 ++++++++++++++++++++++----------------------- 1 file changed, 43 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index db54550..80672d5 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -1150,28 +1150,28 @@ static const char * const rt5659_data_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static const SOC_ENUM_SINGLE_DECL(rt5659_if1_01_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5659_if1_01_adc_enum, RT5659_TDM_CTRL_2, RT5659_DS_ADC_SLOT01_SFT, rt5659_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5659_if1_23_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5659_if1_23_adc_enum, RT5659_TDM_CTRL_2, RT5659_DS_ADC_SLOT23_SFT, rt5659_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5659_if1_45_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5659_if1_45_adc_enum, RT5659_TDM_CTRL_2, RT5659_DS_ADC_SLOT45_SFT, rt5659_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5659_if1_67_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5659_if1_67_adc_enum, RT5659_TDM_CTRL_2, RT5659_DS_ADC_SLOT67_SFT, rt5659_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5659_if2_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5659_if2_dac_enum, RT5659_DIG_INF23_DATA, RT5659_IF2_DAC_SEL_SFT, rt5659_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5659_if2_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5659_if2_adc_enum, RT5659_DIG_INF23_DATA, RT5659_IF2_ADC_SEL_SFT, rt5659_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5659_if3_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5659_if3_dac_enum, RT5659_DIG_INF23_DATA, RT5659_IF3_DAC_SEL_SFT, rt5659_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5659_if3_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5659_if3_adc_enum, RT5659_DIG_INF23_DATA, RT5659_IF3_ADC_SEL_SFT, rt5659_data_select); static const struct snd_kcontrol_new rt5659_if1_01_adc_swap_mux = @@ -1207,31 +1207,31 @@ static unsigned int rt5659_asrc_clk_map_values[] = { 0, 1, 2, 3, 5, 6, }; -static const SOC_VALUE_ENUM_SINGLE_DECL( +static SOC_VALUE_ENUM_SINGLE_DECL( rt5659_da_sto_asrc_enum, RT5659_ASRC_2, RT5659_DA_STO_T_SFT, 0x7, rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); -static const SOC_VALUE_ENUM_SINGLE_DECL( +static SOC_VALUE_ENUM_SINGLE_DECL( rt5659_da_monol_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_L_T_SFT, 0x7, rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); -static const SOC_VALUE_ENUM_SINGLE_DECL( +static SOC_VALUE_ENUM_SINGLE_DECL( rt5659_da_monor_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_R_T_SFT, 0x7, rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); -static const SOC_VALUE_ENUM_SINGLE_DECL( +static SOC_VALUE_ENUM_SINGLE_DECL( rt5659_ad_sto1_asrc_enum, RT5659_ASRC_2, RT5659_AD_STO1_T_SFT, 0x7, rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); -static const SOC_VALUE_ENUM_SINGLE_DECL( +static SOC_VALUE_ENUM_SINGLE_DECL( rt5659_ad_sto2_asrc_enum, RT5659_ASRC_3, RT5659_AD_STO2_T_SFT, 0x7, rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); -static const SOC_VALUE_ENUM_SINGLE_DECL( +static SOC_VALUE_ENUM_SINGLE_DECL( rt5659_ad_monol_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_L_T_SFT, 0x7, rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); -static const SOC_VALUE_ENUM_SINGLE_DECL( +static SOC_VALUE_ENUM_SINGLE_DECL( rt5659_ad_monor_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_R_T_SFT, 0x7, rt5659_asrc_clk_src, rt5659_asrc_clk_map_values); @@ -1930,14 +1930,14 @@ static const char * const rt5659_dac2_src[] = { "IF1 DAC2", "IF2 DAC", "IF3 DAC", "Mono ADC MIX" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_dac_l2_enum, RT5659_DAC_CTRL, RT5659_DAC_L2_SEL_SFT, rt5659_dac2_src); static const struct snd_kcontrol_new rt5659_dac_l2_mux = SOC_DAPM_ENUM("DAC L2 Source", rt5659_dac_l2_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_dac_r2_enum, RT5659_DAC_CTRL, RT5659_DAC_R2_SEL_SFT, rt5659_dac2_src); @@ -1951,7 +1951,7 @@ static const char * const rt5659_sto1_adc1_src[] = { "DAC MIX", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_sto1_adc1_enum, RT5659_STO1_ADC_MIXER, RT5659_STO1_ADC1_SRC_SFT, rt5659_sto1_adc1_src); @@ -1964,7 +1964,7 @@ static const char * const rt5659_sto1_adc_src[] = { "ADC1", "ADC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_sto1_adc_enum, RT5659_STO1_ADC_MIXER, RT5659_STO1_ADC_SRC_SFT, rt5659_sto1_adc_src); @@ -1977,7 +1977,7 @@ static const char * const rt5659_sto1_adc2_src[] = { "DAC MIX", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_sto1_adc2_enum, RT5659_STO1_ADC_MIXER, RT5659_STO1_ADC2_SRC_SFT, rt5659_sto1_adc2_src); @@ -1990,7 +1990,7 @@ static const char * const rt5659_sto1_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_sto1_dmic_enum, RT5659_STO1_ADC_MIXER, RT5659_STO1_DMIC_SRC_SFT, rt5659_sto1_dmic_src); @@ -2004,7 +2004,7 @@ static const char * const rt5659_mono_adc_l2_src[] = { "Mono DAC MIXL", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_mono_adc_l2_enum, RT5659_MONO_ADC_MIXER, RT5659_MONO_ADC_L2_SRC_SFT, rt5659_mono_adc_l2_src); @@ -2018,7 +2018,7 @@ static const char * const rt5659_mono_adc_l1_src[] = { "Mono DAC MIXL", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_mono_adc_l1_enum, RT5659_MONO_ADC_MIXER, RT5659_MONO_ADC_L1_SRC_SFT, rt5659_mono_adc_l1_src); @@ -2031,14 +2031,14 @@ static const char * const rt5659_mono_adc_src[] = { "ADC1 L", "ADC1 R", "ADC2 L", "ADC2 R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_mono_adc_l_enum, RT5659_MONO_ADC_MIXER, RT5659_MONO_ADC_L_SRC_SFT, rt5659_mono_adc_src); static const struct snd_kcontrol_new rt5659_mono_adc_l_mux = SOC_DAPM_ENUM("Mono ADC L Source", rt5659_mono_adc_l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_mono_adcr_enum, RT5659_MONO_ADC_MIXER, RT5659_MONO_ADC_R_SRC_SFT, rt5659_mono_adc_src); @@ -2051,7 +2051,7 @@ static const char * const rt5659_mono_dmic_l_src[] = { "DMIC1 L", "DMIC2 L" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_mono_dmic_l_enum, RT5659_MONO_ADC_MIXER, RT5659_MONO_DMIC_L_SRC_SFT, rt5659_mono_dmic_l_src); @@ -2064,7 +2064,7 @@ static const char * const rt5659_mono_adc_r2_src[] = { "Mono DAC MIXR", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_mono_adc_r2_enum, RT5659_MONO_ADC_MIXER, RT5659_MONO_ADC_R2_SRC_SFT, rt5659_mono_adc_r2_src); @@ -2077,7 +2077,7 @@ static const char * const rt5659_mono_adc_r1_src[] = { "Mono DAC MIXR", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_mono_adc_r1_enum, RT5659_MONO_ADC_MIXER, RT5659_MONO_ADC_R1_SRC_SFT, rt5659_mono_adc_r1_src); @@ -2090,7 +2090,7 @@ static const char * const rt5659_mono_dmic_r_src[] = { "DMIC1 R", "DMIC2 R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_mono_dmic_r_enum, RT5659_MONO_ADC_MIXER, RT5659_MONO_DMIC_R_SRC_SFT, rt5659_mono_dmic_r_src); @@ -2104,14 +2104,14 @@ static const char * const rt5659_dac1_src[] = { "IF1 DAC1", "IF2 DAC", "IF3 DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_dac_r1_enum, RT5659_AD_DA_MIXER, RT5659_DAC1_R_SEL_SFT, rt5659_dac1_src); static const struct snd_kcontrol_new rt5659_dac_r1_mux = SOC_DAPM_ENUM("DAC R1 Source", rt5659_dac_r1_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_dac_l1_enum, RT5659_AD_DA_MIXER, RT5659_DAC1_L_SEL_SFT, rt5659_dac1_src); @@ -2124,14 +2124,14 @@ static const char * const rt5659_dig_dac_mix_src[] = { "Stereo DAC Mixer", "Mono DAC Mixer" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_dig_dac_mixl_enum, RT5659_DIG_MIXER, RT5659_DAC_MIX_L_SFT, rt5659_dig_dac_mix_src); static const struct snd_kcontrol_new rt5659_dig_dac_mixl_mux = SOC_DAPM_ENUM("DAC Digital Mixer L Source", rt5659_dig_dac_mixl_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_dig_dac_mixr_enum, RT5659_DIG_MIXER, RT5659_DAC_MIX_R_SFT, rt5659_dig_dac_mix_src); @@ -2144,14 +2144,14 @@ static const char * const rt5659_alg_dac1_src[] = { "DAC", "Stereo DAC Mixer" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_alg_dac_l1_enum, RT5659_A_DAC_MUX, RT5659_A_DACL1_SFT, rt5659_alg_dac1_src); static const struct snd_kcontrol_new rt5659_alg_dac_l1_mux = SOC_DAPM_ENUM("Analog DACL1 Source", rt5659_alg_dac_l1_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_alg_dac_r1_enum, RT5659_A_DAC_MUX, RT5659_A_DACR1_SFT, rt5659_alg_dac1_src); @@ -2164,14 +2164,14 @@ static const char * const rt5659_alg_dac2_src[] = { "Stereo DAC Mixer", "Mono DAC Mixer" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_alg_dac_l2_enum, RT5659_A_DAC_MUX, RT5659_A_DACL2_SFT, rt5659_alg_dac2_src); static const struct snd_kcontrol_new rt5659_alg_dac_l2_mux = SOC_DAPM_ENUM("Analog DAC L2 Source", rt5659_alg_dac_l2_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_alg_dac_r2_enum, RT5659_A_DAC_MUX, RT5659_A_DACR2_SFT, rt5659_alg_dac2_src); @@ -2184,7 +2184,7 @@ static const char * const rt5659_if2_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "DAC_REF", "IF_ADC3" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_if2_adc_in_enum, RT5659_DIG_INF23_DATA, RT5659_IF2_ADC_IN_SFT, rt5659_if2_adc_in_src); @@ -2197,7 +2197,7 @@ static const char * const rt5659_if3_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "DAC_REF", "Stereo2_ADC_L/R" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_if3_adc_in_enum, RT5659_DIG_INF23_DATA, RT5659_IF3_ADC_IN_SFT, rt5659_if3_adc_in_src); @@ -2210,14 +2210,14 @@ static const char * const rt5659_pdm_src[] = { "Mono DAC", "Stereo DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_pdm_l_enum, RT5659_PDM_OUT_CTRL, RT5659_PDM1_L_SFT, rt5659_pdm_src); static const struct snd_kcontrol_new rt5659_pdm_l_mux = SOC_DAPM_ENUM("PDM L Source", rt5659_pdm_l_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_pdm_r_enum, RT5659_PDM_OUT_CTRL, RT5659_PDM1_R_SFT, rt5659_pdm_src); @@ -2230,7 +2230,7 @@ static const char * const rt5659_spdif_src[] = { "IF1_DAC1", "IF1_DAC2", "IF2_DAC", "IF3_DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_spdif_enum, RT5659_SPDIF_CTRL, RT5659_SPDIF_SEL_SFT, rt5659_spdif_src); @@ -2250,7 +2250,7 @@ static const char * const rt5659_rx_adc_data_src[] = { "NUL:AD2:DAC:AD1", "NUL:DAC:DAC:AD2", "NUL:DAC:AD2:DAC" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5659_rx_adc_data_enum, RT5659_TDM_CTRL_2, RT5659_ADCDAT_SRC_SFT, rt5659_rx_adc_data_src); -- 2.1.4

1 0

[alsa-devel] [PATCH] ASoC: rt5659: declare rt5659_i2c_driver static
by Nicholas Mc Guire 12 Jan '17

12 Jan '17

Declar rt5659_i2c_driver, which is only being passed to module_i2c_driver(rt5659_i2c_driver), static. Fixes: commit d3cb2de2479b ("ASoC: rt5659: add rt5659 codec driver") Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Problem found by sparse: sound/soc/codecs/rt5659.c:4233:19: warning: symbol 'rt5659_i2c_driver' was not declared. Should it be static? Patch was compile tested with: x86_64_defconfig + CONFIG_COMPILE_TEST=y SND_SOC=m + SND_SOC_ALL_CODECS=m (implies SND_SOC_RT5659=m) Patch is aginast 4.10-rc3 (localversion-next is next-20170112) sound/soc/codecs/rt5659.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index db54550..90b5cc6 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4230,7 +4230,7 @@ static struct acpi_device_id rt5659_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5659_acpi_match); #endif -struct i2c_driver rt5659_i2c_driver = { +static struct i2c_driver rt5659_i2c_driver = { .driver = { .name = "rt5659", .owner = THIS_MODULE, -- 2.1.4

1 0

[alsa-devel] [PATCH] ASoC: rt5659: use msleep() for long delays
by Nicholas Mc Guire 12 Jan '17

12 Jan '17

ulseep_range() uses hrtimers and provides no advantage over msleep() for larger delays. For this large delay msleep() is preferable. Fixes: commit d3cb2de2479b ("ASoC: rt5659: add rt5659 codec driver") Link: http://lkml.org/lkml/2017/1/11/377 Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Problem was found by cocinelle script. Patch was compile tested with: x86_64_defconfig + CONFIG_COMPILE_TEST=y SND_SOC=m + SND_SOC_ALL_CODECS=m (implies SND_SOC_RT5659=m) (44 sparse warnings - fixed in a seperate patches) Patch is aginast 4.10-rc3 (localversion-next is next-20170112) sound/soc/codecs/rt5659.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index db54550..a57f67b 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4018,7 +4018,7 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, GPIOD_OUT_HIGH); /* Sleep for 300 ms miniumum */ - usleep_range(300000, 350000); + msleep(300); rt5659->regmap = devm_regmap_init_i2c(i2c, &rt5659_regmap); if (IS_ERR(rt5659->regmap)) { -- 2.1.4

1 0

[alsa-devel] [PATCH] ASoC: rt5640: use msleep() for long delays
by Nicholas Mc Guire 12 Jan '17

12 Jan '17

ulseep_range() uses hrtimers and provides no advantage over msleep() for larger delays. Fix up the 70/80ms delays here passing the "min" value to msleep(). This reduces the load on the hrtimer subsystem. Link: http://lkml.org/lkml/2017/1/11/377 Fixes: commit 246693ba7b0b ("ASoC: rt5640: change widget sequence for depop") Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Problem located by coccinelle script Patch was compile tested with: x86_64_defconfig + SND_SOC=m SND_SOC_INTEL_BYTCR_RT5640_MACH=m (implies CONFIG_SND_SOC_RT5640) Patch is aginast 4.10-rc3 (localversion-next is next-20170112) sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0f1b216..33e080f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -995,7 +995,7 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: rt5640->hp_mute = 1; - usleep_range(70000, 75000); + msleep(70); break; default: @@ -1059,7 +1059,7 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: if (!rt5640->hp_mute) - usleep_range(80000, 85000); + msleep(80); break; -- 2.1.4

1 0

[alsa-devel] [PATCH 0/7] Constify snd_rawmidi_ops
by Takashi Iwai 12 Jan '17

12 Jan '17

Hi, this is a small patchset to constify snd_rawmidi_ops references, as already done for many other ops. Takashi == Takashi Iwai (7): ALSA: rawmidi: Add const to snd_rawmidi_ops ALSA: seq: Constify snd_rawmidi_ops ALSA: drivers: Constify snd_rawmidi_ops ALSA: isa: Constify snd_rawmidi_ops ALSA: pci: Constify snd_rawmidi_ops ALSA: firewire: Constify snd_rawmidi_ops ALSA: usb: Constify snd_rawmidi_ops include/sound/rawmidi.h | 4 ++-- include/sound/snd_wavefront.h | 4 ++-- sound/core/rawmidi.c | 2 +- sound/core/seq/seq_virmidi.c | 4 ++-- sound/drivers/mpu401/mpu401_uart.c | 4 ++-- sound/drivers/mtpav.c | 4 ++-- sound/drivers/mts64.c | 4 ++-- sound/drivers/portman2x4.c | 4 ++-- sound/drivers/serial-u16550.c | 4 ++-- sound/firewire/bebob/bebob_midi.c | 4 ++-- sound/firewire/dice/dice-midi.c | 4 ++-- sound/firewire/digi00x/digi00x-midi.c | 8 ++++---- sound/firewire/fireworks/fireworks_midi.c | 4 ++-- sound/firewire/oxfw/oxfw-midi.c | 4 ++-- sound/firewire/oxfw/oxfw-scs1x.c | 4 ++-- sound/firewire/tascam/tascam-midi.c | 4 ++-- sound/isa/gus/gus_uart.c | 4 ++-- sound/isa/msnd/msnd_midi.c | 2 +- sound/isa/sb/sb8_midi.c | 4 ++-- sound/isa/wavefront/wavefront_midi.c | 4 ++-- sound/pci/ca0106/ca_midi.c | 4 ++-- sound/pci/cs4281.c | 4 ++-- sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/echoaudio/midi.c | 4 ++-- sound/pci/emu10k1/emu10k1x.c | 4 ++-- sound/pci/emu10k1/emumpu401.c | 4 ++-- sound/pci/ens1370.c | 4 ++-- sound/pci/ice1712/ice1724.c | 4 ++-- sound/pci/rme9652/hdsp.c | 4 ++-- sound/pci/rme9652/hdspm.c | 4 ++-- sound/usb/6fire/midi.c | 4 ++-- sound/usb/bcd2000/bcd2000.c | 4 ++-- sound/usb/caiaq/midi.c | 4 ++-- sound/usb/line6/midi.c | 4 ++-- sound/usb/midi.c | 4 ++-- 35 files changed, 70 insertions(+), 70 deletions(-) -- 2.11.0

2 8

[alsa-devel] [PATCH] ASoC: Fixup some small kernel-doc typos
by Charles Keepax 12 Jan '17

12 Jan '17

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-ac97.c | 2 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-ops.c | 2 +- sound/soc/soc-topology.c | 6 +++--- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 6c8b0b0..36dae41 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -251,7 +251,7 @@ EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); /** * snd_soc_free_ac97_codec - free AC97 codec device - * @codec: audio codec + * @ac97: snd_ac97 device to be freed * * Frees AC97 codec device resources. */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4f6f682..616fee5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -979,7 +979,7 @@ EXPORT_SYMBOL_GPL(snd_soc_find_dai); * @card: soc card * @id: DAI link ID to match * @name: DAI link name to match, optional - * @stream name: DAI link stream name to match, optional + * @stream_name: DAI link stream name to match, optional * * This function will search all existing DAI links of the soc card to * find the link of the same ID. Since DAI links may not have their diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 9fc1a7b..500f98c 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -120,7 +120,7 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); /** - * snd_soc_read_signed - Read a codec register and interprete as signed value + * snd_soc_read_signed - Read a codec register and interpret as signed value * @component: component * @reg: Register to read * @mask: Mask to use after shifting the register value diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 87757c2a..dc1f041 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1918,7 +1918,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, /** * set_link_hw_format - Set the HW audio format of the physical DAI link. - * @tplg: topology context + * @link: &snd_soc_dai_link which should be updated * @cfg: physical link configs. * * Topology context contains a list of supported HW formats (configs) and @@ -1969,7 +1969,7 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, /** * link_new_ver - Create a new physical link config from the old * version of source. - * @toplogy: topology context + * @tplg: topology context * @src: old version of phyical link config as a source * @link: latest version of physical link config created from the source * @@ -2211,7 +2211,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, /** * manifest_new_ver - Create a new version of manifest from the old version * of source. - * @toplogy: topology context + * @tplg: topology context * @src: old version of manifest as a source * @manifest: latest version of manifest created from the source * -- 2.1.4

1 0

[alsa-devel] [PATCH] ASoC: rt5660: remove double const
by Nicholas Mc Guire 12 Jan '17

12 Jan '17

Drop the const qualifier as it is being added by SOC_ENUM_DOUBLE_DECL() already which is called by SOC_ENUM_SINGLE_DECL() here. Fixes: commit 2b26dd4c1fc5 ("ASoC: rt5660: add rt5660 codec driver") Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- sparse throws the warning: sound/soc/codecs/rt5660.c:529:14: warning: duplicate const sound/soc/codecs/rt5660.c:532:14: warning: duplicate const this is due to: static const SOC_ENUM_SINGLE_DECL(rt5660_if1_dac_enum,... which via: #define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \ SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) expands to: #define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \ const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ ARRAY_SIZE(xtexts), xtexts) and that already carries the const qualifier so that results in the "duplicate const" from sparse. The fix is simply to drop the first const. Patch was compile tested with: x86_64_defconfig + CONFIG_COMPILE_TEST=y SND_SOC=m + SND_SOC_ALL_CODECS=m (implies SND_SOC_RT5660=m) Patch is aginast 4.10-rc3 (localversion-next is next-20170112) sound/soc/codecs/rt5660.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 76cf76a..296b7b0 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -526,10 +526,10 @@ static const char * const rt5660_data_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static const SOC_ENUM_SINGLE_DECL(rt5660_if1_dac_enum, +static SOC_ENUM_SINGLE_DECL(rt5660_if1_dac_enum, RT5660_DIG_INF1_DATA, RT5660_IF1_DAC_IN_SFT, rt5660_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5660_if1_adc_enum, +static SOC_ENUM_SINGLE_DECL(rt5660_if1_adc_enum, RT5660_DIG_INF1_DATA, RT5660_IF1_ADC_IN_SFT, rt5660_data_select); static const struct snd_kcontrol_new rt5660_if1_dac_swap_mux = -- 2.1.4

1 0

[alsa-devel] [PATCH] ASoC: rt5660: use msleep() for long delay
by Nicholas Mc Guire 12 Jan '17

12 Jan '17

ulseep_range() uses hrtimers and provides no advantage over msleep() for larger delays. For this large delay msleep() is preferable. Link: http://lkml.org/lkml/2017/1/11/377 Fixes: commit 2b26dd4c1fc5 ("ASoC: rt5660: add rt5660 codec driver") Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Problem was found by cocinelle script. Patch was compile tested with: x86_64_defconfig + CONFIG_COMPILE_TEST=y SND_SOC=m + SND_SOC_ALL_CODECS=m (implies SND_SOC_RT5660=m) (2 sparse warnings - fix in a seperate patch) Patch is aginast 4.10-rc3 (localversion-next is next-20170112) sound/soc/codecs/rt5660.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 76cf76a..78a77bc 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1152,7 +1152,7 @@ static int rt5660_resume(struct snd_soc_codec *codec) struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); if (rt5660->pdata.poweroff_codec_in_suspend) - usleep_range(350000, 400000); + msleep(350); regcache_cache_only(rt5660->regmap, false); regcache_sync(rt5660->regmap); -- 2.1.4

1 0

[alsa-devel] [PATCH V2] ASoC: rt5651: use msleep for large delays
by Nicholas Mc Guire 11 Jan '17

11 Jan '17

ulseep_range() uses hrtimers and provides no advantage over msleep() for larger delays. Fix up the 70/80ms delays here to use msleep() and reduce the load on the hrtimer subsystem. Link: http://lkml.org/lkml/2017/1/11/377 Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- V2: Mark Brown <broonie(a)kernel.org> suggested to take the lower time stamp for such conversions as this is the behavior developers expect rather than the upper that would be closer to the current (somewhat odd) usleep_range() behavior. As these are non-atomic regions and a delay of 70/80 ms implies that there is almost certainty multiple context switches occurring in the sleep time and thus relatively high jitter must be assumed with respect to actual wakeup time implying that there is little point in using high-resolution timers here. Patch was compile tested with: x86_64_defconfig + SND_SOC=m SND_SOC_INTEL_BYTCR_RT5651_MACH=m (implies CONFIG_SND_SOC_RT5651) Patch is aginast 4.10-rc3 (localversion-next is next-20170111) sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index f5d3415..fb592b0 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -802,7 +802,7 @@ static int rt5651_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: rt5651->hp_mute = 1; - usleep_range(70000, 75000); + msleep(70); break; default: @@ -822,7 +822,7 @@ static int rt5651_hp_post_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: if (!rt5651->hp_mute) - usleep_range(80000, 85000); + msleep(80); break; -- 2.1.4

1 0

[alsa-devel] [GIT PULL] ASoC fixes for v4.10
by Mark Brown 11 Jan '17

11 Jan '17

The following changes since commit a121103c922847ba5010819a3f250f1f7fc84ab8: Linux 4.10-rc3 (2017-01-08 14:18:17 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.10-rc3 for you to fetch changes up to 42e0ebdef53d91891939a475c6ef23970829ff99: Merge remote-tracking branches 'asoc/fix/nau8825', 'asoc/fix/rt5645', 'asoc/fix/tlv320aic3x' and 'asoc/fix/topology' into asoc-linus (2017-01-10 10:47:54 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.10 As well as the usual smattering of driver specific fixes collected since the merge window this has one particularly important fix to the core for handling of aux_devs which was broken during the merge window by some of the componentization refactoring. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt5645: set sel_i2s_pre_div1 to 2 Caleb Crome (1): ASoC: fsl_ssi: set fifo watermark to more reliable value Colin Ian King (2): ASoC: rsnd: don't double free kctrl ASoC: topology: kfree kcontrol->private_value before freeing kcontrol G Kranthi (1): ASoC: Intel: Skylake: Fix to fail safely if module not available in path Jeeja KP (1): ASoC: Intel: Skylake: Release FW ctx in cleanup John Hsu (2): ASoC: nau8825: correct the function name of register ASoC: nau8825: fix invalid configuration in Pre-Scalar of FLL Jose Abreu (1): ASoC: dwc: Fix PIO mode initialization Mark Brown (5): Merge remote-tracking branch 'asoc/fix/component' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/dpcm', 'asoc/fix/dwc', 'asoc/fix/fsl-ssi' and 'asoc/fix/hdmi-codec' into asoc-linus Merge remote-tracking branches 'asoc/fix/nau8825', 'asoc/fix/rt5645', 'asoc/fix/tlv320aic3x' and 'asoc/fix/topology' into asoc-linus Patrick Lai (1): ASoC: dpcm: Avoid putting stream state to STOP when FE stream is paused Peter Ujfalusi (1): ASoC: tlv320aic3x: Mark the RESET register as volatile Pierre-Louis Bossart (2): ASoC: Intel: bytcr_rt5640: fallback mechanism if MCLK is not enabled ASoC: Intel: bytcr-rt5640: fix settings in internal clock mode Richard Fitzgerald (1): ASoC: wm_adsp: Don't overrun firmware file buffer when reading region data Sylwester Nawrocki (1): ASoC: Fix binding and probing of auxiliary components Takashi Sakamoto (1): ASoC: hdmi-codec: use unsigned type to structure members with bit-field include/sound/hdmi-codec.h | 8 ++-- include/sound/soc.h | 3 ++ sound/soc/codecs/nau8825.c | 9 +++-- sound/soc/codecs/nau8825.h | 7 ++-- sound/soc/codecs/rt5645.c | 3 ++ sound/soc/codecs/tlv320aic3x.c | 13 ++++++ sound/soc/codecs/wm_adsp.c | 25 +++++++++++- sound/soc/dwc/designware_i2s.c | 25 ++++++------ sound/soc/fsl/fsl_ssi.c | 74 +++++++++++++++++++++++++---------- sound/soc/intel/boards/bytcr_rt5640.c | 18 +++++++-- sound/soc/intel/skylake/skl-pcm.c | 3 ++ sound/soc/intel/skylake/skl-sst.c | 3 ++ sound/soc/sh/rcar/core.c | 4 +- sound/soc/soc-core.c | 10 ++--- sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 3 +- 16 files changed, 150 insertions(+), 62 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132 - fix possible NULL pointer use
by Arnd Bergmann 11 Jan '17

11 Jan '17

gcc-7 caught what it considers a NULL pointer dereference: sound/pci/hda/patch_ca0132.c: In function 'dspio_scp.constprop': sound/pci/hda/patch_ca0132.c:1487:4: error: argument 1 null where non-null expected [-Werror=nonnull] This is plausible from looking at the function, as we compare 'reply' to NULL earlier in it. I have not tried to analyze if there are constraints that make it impossible to hit the bug, but adding another NULL check in the end kills the warning and makes the function more robust. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/hda/patch_ca0132.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index bd7c29f8ab1e..07a9deb17477 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1482,6 +1482,9 @@ static int dspio_scp(struct hda_codec *codec, } else if (ret_size != reply_data_size) { codec_dbg(codec, "RetLen and HdrLen .NE.\n"); return -EINVAL; + } else if (!reply) { + codec_dbg(codec, "NULL reply\n"); + return -EINVAL; } else { *reply_len = ret_size*sizeof(unsigned int); memcpy(reply, scp_reply.data, *reply_len); -- 2.9.0

2 1

[alsa-devel] [PATCH v3 0/2] ASoC: Use DMI name in sound card long name
by mengdong.lin＠linux.intel.com 11 Jan '17

11 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series is based on Liam's idea to use DMI name as the sound card short name in the audio workshop. Now we add DMI info to the card long name: - Not change the card name (short name), keep it simple. So the card ID won't change as well. - Use DMI name to make the card long name. Machine drivers can use a new API to get this feature. In user space, when the sound server ask UCM to open a sound card, UCM will try to find the best machine-specific configuration file based on the card long name, and only if not avaiable, fallback to the default configuration file that matches the card name (short name). Please see patch: ucm: Automatically load the best config file based on the card long name History: v1 from RFC: Give up using a table to map DMI info to a simple string, since we cannot maintain such a table for each machine. Just do verbatim copy of DMI info and let user space decide how to use the DMI info in card long name to find the best UCM config file. If device driver adds a flavor string, just append it after the DMI info fields to the card long name. v2: Exclude card name and only keep DMI name in the long name, to save the space since the long name cannot exceed 80 characters. Trim special characters like SPACE in the long name, because UCM configuration files & directories may use the same long name and Autoconf cannot handle special characters. v3: Also include DMI product version in the long name, since vendors like Lenovo may only put self-explanatory name in this field. Liam Girdwood (1): ASoC: core: Add API to use DMI name in sound card long name Mengdong Lin (1): ASoC: Intel: Use DMI name for sound card long name in Broadwell machine driver include/sound/soc.h | 4 ++ sound/soc/intel/boards/broadwell.c | 2 + sound/soc/soc-core.c | 122 +++++++++++++++++++++++++++++++++++++ 3 files changed, 128 insertions(+) -- 2.5.0

4 8

Re: [alsa-devel] [PATCH 2/2] ASoC: cx2072x Add driver for CX2072X CODEC
by Mark Brown 11 Jan '17

11 Jan '17

On Sat, Dec 17, 2016 at 03:52:33PM +0800, simon.ho.cnxt(a)gmail.com wrote: > +/*FIXME: need to move the default settings to device tree*/ No, we'd expect things to be configured from userspace via a binary control. > +static unsigned char cx2072x_eq_coeff_array[MAX_EQ_BAND][MAC_EQ_COEFF] = { > + {0x77, 0x26, 0x13, 0xb3, 0x76, 0x26, 0x0a, 0x3d, 0xd4, 0xe2, 0x04}, > + {0x97, 0x3e, 0xb3, 0x86, 0xc2, 0x3b, 0x4d, 0x79, 0xa7, 0xc5, 0x03}, > + {0x0f, 0x39, 0x76, 0xa3, 0x1b, 0x2b, 0x89, 0x5c, 0xd7, 0xdb, 0x03}, > + {0x21, 0x46, 0xfe, 0xa6, 0xec, 0x24, 0x01, 0x59, 0xf4, 0xd4, 0x03}, > + {0xe9, 0x78, 0x9c, 0xb0, 0x8a, 0x56, 0x64, 0x4f, 0x8d, 0xb0, 0x02}, > + {0x60, 0x6e, 0x57, 0xee, 0xec, 0x18, 0xa8, 0x11, 0xb5, 0xf8, 0x02}, > + {0x5a, 0x14, 0x68, 0xe9, 0x1d, 0x06, 0xb9, 0x5f, 0x68, 0xdc, 0x03}, > +}; > + > +static unsigned char cx2072x_drc_array[MAX_DRC_REGS] = { > + 0x65, 0x55, 0x3C, 0x01, 0x05, 0x39, 0x76, 0x1A, 0x00 > +}; Just use the chip defaults, this avoids any confusion or debate about the values in the kernel. > +#define get_cx2072x_priv(_codec_) \ > + ((struct cx2072x_priv *) snd_soc_codec_get_drvdata(_codec_)) There is no need to cast void pointers, this will at most hide actual errors, and there's no real need for the macro at all really. > + { 36864000, 7 },/* Don't use div 7*/ Does something enforce that? Also coding style, missing space. > +static const struct reg_default cx2072x_reg_defaults[] = { > + { 0x0414, 0x00000003 }, /*2072X_AFG_POWER_STATE */ We have defines for the register names which are helpfully commented here, why not just actually use the defines? > + if (reg == CX2072X_UM_INTERRUPT_CRTL_E) { > + /* Update the MSB byte only */ > + reg += 3; > + size = 1; > + value >>= 24; > + } Use a switch here in case you find any more magic registers. > + ret = i2c_master_send(client, buf, size + 2); > + if (ret == size + 2) { > + ret = 0; > + } else if (ret < 0) { > + dev_err(dev, > + "I2C write address failed\n"); Please print the error code, it makes life easier for people looking at logs with errors. > +static unsigned int get_div_from_mclk(unsigned int mclk) > +{ > + unsigned int div = 8; > + int i = 0; > + > + for (i = 0; i < ARRAY_SIZE(MCLK_PRE_DIV); i++) { > + if (mclk <= MCLK_PRE_DIV[i].mclk) { > + div = MCLK_PRE_DIV[i].div; > + break; > + } > + } > + return div; > +} Why is this function in the middle of all the register I/O functions? > +static int cx2072x_config_headset_det(struct cx2072x_priv *cx2072x) > +{ > + const int interrupt_gpio_pin = 1; > + > + dev_dbg(cx2072x->dev, > + "Configure interrupt pin: %d\n", interrupt_gpio_pin); > + /*No-sticky input type*/ Coding style. There's *lots* of really obvious and bad coding style problems throughout the driver, this is really not good - as well as making things harder to read poor coding style is normally a good indicator that there are other problems. > + /* support both nokia and apple headset set. Monitor time = 275 ms*/ > + regmap_write(cx2072x->regmap, CX2072X_DIGITAL_TEST15, 0x73); > + > + /* Disable TIP detection*/ > + regmap_write(cx2072x->regmap, CX2072X_ANALOG_TEST12, 0x300); > + > + /* Switch MusicD3Live pin to GPIO */ > + regmap_write(cx2072x->regmap, CX2072X_DIGITAL_TEST1, 0); These look awfully like they should be configured by the system integrator via DT. > + regmap_write(cx2072x->regmap, CX2072X_ANALOG_TEST8, > + (unsigned char)int_div & 0xffff); This cast is worrying, why is it needed? > + > + /* clock inversion */ > + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { > + case SND_SOC_DAIFMT_NB_NF: > + is_frame_inv = is_i2s ? 1 : 0; > + is_bclk_inv = is_i2s ? 1 : 0; Please don't abuse the ternery operator like this, write normal logic statements with straightforward assignments. Readers will thank you. > + regmap_write(cx2072x->regmap, CX2072X_I2SPCM_CONTROL1, reg1.ulVal); Please don't use Hungarian notation. > + /*Enables bclk and EAPD pin*/ > + if (cx2072x->rev_id == CX2072X_REV_A2) > + regmap_update_bits(cx2072x->regmap, CX2072X_DIGITAL_BIOS_TEST2, > + 0x84, 0xFF); > + else > + regmap_write(cx2072x->regmap, CX2072X_DIGITAL_BIOS_TEST2, > + regDBT2.ulVal); Again, switch statement. > +static int portg_power_ev(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) > +{ > + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); > + > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + cx2072x_update_dsp(codec); > + break; > + default: > + break; > + } > + return 0; > +} > +static int cx2072x_plbk_dsp_info(struct snd_kcontrol *kcontrol, Blank lines between functions. > + mutex_lock(&cx2072x->eq_coeff_lock); > + memcpy(cache, param, CX2072X_PLBK_EQ_COEF_LEN); > + cx2072x->plbk_dsp_changed = true; Shouldn't this involve a memcmp() to check if the new value is different? > +static const struct snd_kcontrol_new cx2072x_snd_controls[] = { > + > + SOC_DOUBLE_R_TLV("PortD Boost", CX2072X_PORTD_GAIN_LEFT, > + CX2072X_PORTD_GAIN_RIGHT, 0, 3, 0, boost_tlv), All volume controls should end in Volume so userspace knows how to handle them. Also please use normal kernel indentation for the continuation lines. > + SOC_DOUBLE_R("DAC1 Mute", CX2072X_DAC1_AMP_GAIN_LEFT, > + CX2072X_DAC1_AMP_GAIN_RIGHT, 7, 1, 0), All boolean controls should end in Switch so userspace knows how to handle them. > + CX2072X_PLBK_EQ_COEF("DACL EQ 0", 0, 0), > + CX2072X_PLBK_EQ_COEF("DACL EQ 1", 0, 1), > + CX2072X_PLBK_EQ_COEF("DACL EQ 2", 0, 2), > + CX2072X_PLBK_EQ_COEF("DACL EQ 3", 0, 3), > + CX2072X_PLBK_EQ_COEF("DACL EQ 4", 0, 4), > + CX2072X_PLBK_EQ_COEF("DACL EQ 5", 0, 5), > + CX2072X_PLBK_EQ_COEF("DACL EQ 6", 0, 6), > + CX2072X_PLBK_EQ_COEF("DACR EQ 0", 1, 0), > + CX2072X_PLBK_EQ_COEF("DACR EQ 1", 1, 1), > + CX2072X_PLBK_EQ_COEF("DACR EQ 2", 1, 2), > + CX2072X_PLBK_EQ_COEF("DACR EQ 3", 1, 3), > + CX2072X_PLBK_EQ_COEF("DACR EQ 4", 1, 4), > + CX2072X_PLBK_EQ_COEF("DACR EQ 5", 1, 5), > + CX2072X_PLBK_EQ_COEF("DACR EQ 6", 1, 6), This looks like it should be two binary controls, one per EQ. > +int cx2072x_hs_jack_report(struct snd_soc_codec *codec) > +{ > +} > +EXPORT_SYMBOL_GPL(cx2072x_hs_jack_report); We have a standard jack detection interface in the kernel, the driver should be using it rather than writing its own. > +static int cx2072x_set_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 cx2072x_priv *cx2072x = snd_soc_codec_get_drvdata(codec); > + > + if (slots == 0) > + goto out; > + > + > + switch (rx_mask) { > + case 1 ... 0xf: > + default: > + return -EINVAL; > + } So all possible values are invalid? That doesn't seemw right... > + switch (clk_id) { > + case CX2072X_MCLK_EXTERNAL_PLL: > + if (cx2072x->mclk && clk_set_rate(cx2072x->mclk, freq)) > + return -EINVAL; Please write sensible error handling, log what went wrong if you get an error and pass back error codes you get from other APIs. This will make it a lot easier for users to figure out what's happened. > +#define CX2072X_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, wmask, won_val, \ > + woff_val, wevent, wflags) \ > + {.id = snd_soc_dapm_supply, .name = wname, .kcontrol_news = NULL, \ > + .num_kcontrols = 0, .reg = wreg, .shift = wshift, .mask = wmask, \ > + .on_val = won_val, .off_val = woff_val, \ > + .subseq = wsubseq, .event = wevent, .event_flags = wflags} There appears to be nothing device specific in here, why is this not being added as a generic supply type? > +static void cx2072x_sw_reset(struct cx2072x_priv *cx2072x) > +{ > + > + regmap_write(cx2072x->regmap, CX2072X_AFG_FUNCTION_RESET, 0x01); Coding style, random blank line hre. > + /* reduce the jack monitor time*/ > + regmap_update_bits(cx2072x->regmap, CX2072X_DIGITAL_TEST15, > + 0x00, 0x06); Coding style, please intent contiuation lines... > + cx2072x_config_headset_det(cx2072x); > + /* configre PortC as input device */ > + regmap_update_bits(cx2072x->regmap, CX2072X_PORTC_PIN_CTRL, > + 0x20, 0x20); ...this isn't even consistent with the rest of the same function :( > + switch (level) { > + case SND_SOC_BIAS_ON: > + /* Enable Headset Mic Bias */ > + if (cx2072x->is_biason == 0) > + cx2072x->is_biason = 1; > + break; This looks worrying but since nothing else uses that driver data I'm guessing it doesn't actually do anything. > + case SND_SOC_BIAS_PREPARE: > + if (old_level == SND_SOC_BIAS_STANDBY) { > + if (cx2072x->mclk) { > + dev_dbg(cx2072x->dev, "Turn on MCLK\n"); > + ret = clk_prepare_enable(cx2072x->mclk); > + if (ret) > + return ret; > + } > + } > + break; > + case SND_SOC_BIAS_STANDBY: > + > + if (old_level == SND_SOC_BIAS_OFF) { > + if (cx2072x->mclk) { > + dev_dbg(cx2072x->dev, "Turn on MCLK\n"); > + ret = clk_prepare_enable(cx2072x->mclk); > + if (ret) > + return ret; > + } So we enable MCLK both when coming out of _OFF and when coming out of _STANDBY? Why (and note that we're missing some disables here... > + regmap_read(cx2072x->regmap, CX2072X_VENDOR_ID, &ven_id); > + regmap_read(cx2072x->regmap, CX2072X_REVISION_ID, &cx2072x->rev_id); > + dev_dbg(codec->dev, "codec version: %08x,%08x\n", > + ven_id, cx2072x->rev_id); Log this at info level. > +#ifdef CONFIG_PM > +static int cx2072x_runtime_suspend(struct device *dev) > +{ > + struct cx2072x_priv *cx2072x = dev_get_drvdata(dev); > + > + cx2072x_set_bias_level(cx2072x->codec, SND_SOC_BIAS_OFF); Let the framework take care of managing bias level for you. > +static bool cx2072x_readable_register(struct device *dev, unsigned int reg) > +{ > + switch (reg) { Why is this nowhere near the rest of the regmap code? > +static int cx2072x_i2c_probe(struct i2c_client *i2c, > + const struct i2c_device_id *id) > +{ > + int ret = -1; > + struct cx2072x_priv *cx2072x; > + > + cx2072x = (struct cx2072x_priv *)devm_kzalloc( > + &i2c->dev, sizeof(struct cx2072x_priv), GFP_KERNEL); No need to cast away from void. > + if (cx2072x == NULL) { > + dev_err(&i2c->dev, "Out of memory!\n"); kzalloc() will already be very noisy if this happens. > +static const struct i2c_device_id cx2072x_i2c_id[] = { > + { "cx20721", 0 }, > + { "cx20723", 0 }, > + { "14F10720", 0 }, > + {} > +}; > +MODULE_DEVICE_TABLE(i2c, cx2072x_i2c_id); One of those looks like an ACPI ID not an I2C ID. > + > +static const struct of_device_id cx2072x_of_match[] = { > + { .compatible = "cnxt,cx20721", }, > + { .compatible = "cnxt,cx20723", }, > + { .compatible = "cnxt,cx7601", }, > + {} Why is cx7601 not an I2C ID? > +#ifdef CONFIG_ACPI > + .acpi_match_table = ACPI_PTR(cx2072x_acpi_match), > +#endif No need for the defines here.

3 4

[alsa-devel] [PATCH v4 0/3] support DP MST audio
by libin.yang＠intel.com 11 Jan '17

11 Jan '17

From: Libin Yang <libin.yang(a)intel.com> This patchset starts to support DP MST audio. This patchset is based on drm-tip on git://anongit.freedesktop.org/drm-tip change log: v4: refine some comments in 0001 and 0002 patches. refine the 0003 patch for the new ALSA-Configuration document Libin Yang (3): ALSA: hda - add DP mst verb support ALSA: hda - add DP MST audio support ALSA: Documentation about HDA DP MST pin init and connection Documentation/sound/hd-audio/dp-mst.rst | 17 +++ sound/pci/hda/hda_codec.c | 76 +++++++++- sound/pci/hda/hda_codec.h | 3 + sound/pci/hda/patch_hdmi.c | 245 +++++++++++++++++++++++++------- 4 files changed, 285 insertions(+), 56 deletions(-) -- 2.7.4

4 15

[alsa-devel] [PATCH v6] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Matt Ranostay 11 Jan '17

11 Jan '17

We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Basing the QoS latency calculation on the FIFO size, threshold, sample rate, and channels. Based on the original patch by Tony Lindgren Link: https://patchwork.kernel.org/patch/9305867/ Cc: Tony Lindgren <tony(a)atomide.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> --- hanges from v1: * add calculations for latency per number of FIFO locations Changes from v2: * add missing mcbsp.h header change Changes from v3: * base the latency calculations on threshold, buffer size, sample rate, and channels Changes from v4: * using Peter Ujfalusi's suggestions for restoring a higher latency on audio stream completion, or if not applicable remove the QoS request Changes from v5: * clean up latency checking logic * move logic to .prepare and .shutdown to avoid functions that can sleep sound/soc/omap/mcbsp.c | 4 ++++ sound/soc/omap/mcbsp.h | 3 +++ sound/soc/omap/omap-mcbsp.c | 44 +++++++++++++++++++++++++++++++++++++++++++- 3 files changed, 50 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 06fec5699cc8..e1c3d21dc5ed 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -1098,6 +1099,9 @@ int omap_mcbsp_init(struct platform_device *pdev) void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 61e93b1c185d..46ae1269a698 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -323,8 +323,11 @@ struct omap_mcbsp { unsigned int fmt; unsigned int in_freq; + unsigned int latency[2]; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d018e966e533..3814a234ef61 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -157,6 +157,16 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + + mcbsp->latency[stream1] = 0; + if (mcbsp->latency[stream2]) + pm_qos_update_request(&mcbsp->pm_qos_req, + mcbsp->latency[stream2]); + else + pm_qos_remove_request(&mcbsp->pm_qos_req); if (!cpu_dai->active) { omap_mcbsp_free(mcbsp); @@ -164,6 +174,28 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, } } +static int omap_mcbsp_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct pm_qos_request *pm_qos_req = &mcbsp->pm_qos_req; + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + int latency = mcbsp->latency[stream2]; + + /* Prevent omap hardware from hitting off between FIFO fills */ + if (!latency || mcbsp->latency[stream1] < latency) + latency = mcbsp->latency[stream1]; + + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, latency); + else if (latency) + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency); + + return 0; +} + static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -226,6 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int wlen, channels, wpf; int pkt_size = 0; unsigned int format, div, framesize, master; + unsigned int buffer_size = mcbsp->pdata->buffer_size; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); channels = params_channels(params); @@ -240,7 +273,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (mcbsp->pdata->buffer_size) { + if (buffer_size) { + int latency; + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; int divider = 0; @@ -271,6 +306,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Use packet mode for non mono streams */ pkt_size = channels; } + + latency = ((((buffer_size - pkt_size) / channels) * 1000) + / (params->rate_num / params->rate_den)); + + mcbsp->latency[substream->stream] = latency; + omap_mcbsp_set_threshold(substream, pkt_size); } @@ -554,6 +595,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, static const struct snd_soc_dai_ops mcbsp_dai_ops = { .startup = omap_mcbsp_dai_startup, .shutdown = omap_mcbsp_dai_shutdown, + .prepare = omap_mcbsp_dai_prepare, .trigger = omap_mcbsp_dai_trigger, .delay = omap_mcbsp_dai_delay, .hw_params = omap_mcbsp_dai_hw_params, -- 2.10.2

2 2

[alsa-devel] [PATCH 0/6] ASoC: platform cleanup
by Kuninori Morimoto 11 Jan '17

11 Jan '17

Hi Mark, Lars-Peter These removes un-used platform feature, and move last platform feature to component. If these are OK, all platform feature goes to component. Last platform piece is platform_list on soc-core.c Kuninori Morimoto (6): ASoC: remove .delay from snd_soc_platform_driver ASoC: remove .bespoke_trigger from snd_soc_platform_driver ASoC: remove snd_soc_platform_trigger() ASoC: add for_each_component{_safe} ASoC: add Component level struct snd_compr_ops ASoC: add Component level struct snd_pcm_ops include/sound/soc.h | 23 +++-- sound/soc/soc-compress.c | 228 ++++++++++++++++++++++++++++--------------- sound/soc/soc-core.c | 14 +-- sound/soc/soc-pcm.c | 244 +++++++++++++++++++++++++++++++++-------------- 4 files changed, 339 insertions(+), 170 deletions(-) -- 1.9.1

3 18

[alsa-devel] [PATCH v3] ASoC: ak4642: Replace mdelay function to msleep
by Kuninori Morimoto 11 Jan '17

11 Jan '17

From: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Replace mdelay to msleep to avoid busy loop on ak4642_lout_event(). Otherwise, sometimes playback doesn't work correctly when pulseaudio was used. Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - based on latest kernel sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2609f95..23ab964 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -189,7 +189,7 @@ static int ak4642_lout_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_POST_PMD: /* Power save mode OFF */ - mdelay(300); + msleep(300); snd_soc_update_bits(codec, SG_SL2, LOPS, 0); break; } -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: mpc5200_psc_ac97: Remove unused DAI ID defines
by Lars-Peter Clausen 10 Jan '17

10 Jan '17

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. This also means the mpc5200_psc_ac97.h file no longer has any content and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/efika-audio-fabric.c | 1 - sound/soc/fsl/mpc5200_psc_ac97.c | 1 - sound/soc/fsl/mpc5200_psc_ac97.h | 13 ------------- 3 files changed, 15 deletions(-) delete mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index f200d1c..667f421 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -26,7 +26,6 @@ #include <sound/soc.h> #include "mpc5200_dma.h" -#include "mpc5200_psc_ac97.h" #define DRV_NAME "efika-audio-fabric" diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 243700c..07ee355 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -25,7 +25,6 @@ #include <asm/mpc52xx_psc.h> #include "mpc5200_dma.h" -#include "mpc5200_psc_ac97.h" #define DRV_NAME "mpc5200-psc-ac97" diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h deleted file mode 100644 index e881e78..0000000 --- a/sound/soc/fsl/mpc5200_psc_ac97.h +++ /dev/null @@ -1,13 +0,0 @@ -/* - * Freescale MPC5200 PSC in AC97 mode - * ALSA SoC Digital Audio Interface (DAI) driver - * - */ - -#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ -#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ - -#define MPC5200_AC97_NORMAL 0 -#define MPC5200_AC97_SPDIF 1 - -#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ -- 2.1.4

1 0

[alsa-devel] [PATCH v5] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Matt Ranostay 10 Jan '17

10 Jan '17

We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Basing the QoS latency calculation on the FIFO size, threshold, sample rate, and channels. Based on the original patch by Tony Lindgren Link: https://patchwork.kernel.org/patch/9305867/ Cc: Tony Lindgren <tony(a)atomide.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> --- Changes from v1: * add calculations for latency per number of FIFO locations Changes from v2: * add missing mcbsp.h header change Changes from v3: * base the latency calculations on threshold, buffer size, sample rate, and channels Changes from v4: * using Peter Ujfalusi's suggestions for restoring a higher latency on audio stream completion, or if not applicable remove the QoS request sound/soc/omap/mcbsp.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcbsp.h | 3 +++ sound/soc/omap/omap-mcbsp.c | 11 ++++++++++- 3 files changed, 50 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 06fec5699cc8..f079650b71e3 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -640,9 +641,28 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) */ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) { + struct pm_qos_request *pm_qos_req = &mcbsp->pm_qos_req; + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; int enable_srg = 0; + int latency; u16 w; + /* Prevent omap hardware from hitting off between FIFO fills */ + if (!mcbsp->latency[stream2] || + mcbsp->latency[stream1] < mcbsp->latency[stream2]) + latency = mcbsp->latency[stream1]; + else + latency = mcbsp->latency[stream2]; + + if (latency) { + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, latency); + else + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + latency); + } + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -731,6 +751,20 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); + + if (pm_qos_request_active(&mcbsp->pm_qos_req)) { + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : + SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : + SNDRV_PCM_STREAM_PLAYBACK; + + mcbsp->latency[stream1] = 0; + if (mcbsp->latency[stream2]) + pm_qos_update_request(&mcbsp->pm_qos_req, + mcbsp->latency[stream2]); + else + pm_qos_remove_request(&mcbsp->pm_qos_req); + } } int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) @@ -1098,6 +1132,9 @@ int omap_mcbsp_init(struct platform_device *pdev) void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 61e93b1c185d..46ae1269a698 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -323,8 +323,11 @@ struct omap_mcbsp { unsigned int fmt; unsigned int in_freq; + unsigned int latency[2]; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d018e966e533..7e00938271c6 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -226,6 +226,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int wlen, channels, wpf; int pkt_size = 0; unsigned int format, div, framesize, master; + unsigned int buffer_size = mcbsp->pdata->buffer_size; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); channels = params_channels(params); @@ -240,7 +241,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (mcbsp->pdata->buffer_size) { + if (buffer_size) { + int latency; + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; int divider = 0; @@ -271,6 +274,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Use packet mode for non mono streams */ pkt_size = channels; } + + latency = ((((buffer_size - pkt_size) / channels) * 1000) + / (params->rate_num / params->rate_den)); + + mcbsp->latency[substream->stream] = latency; + omap_mcbsp_set_threshold(substream, pkt_size); } -- 2.10.2

3 9

[alsa-devel] [PATCH] ASoC: tm2_wm5110: include gpio/consumer.h
by Arnd Bergmann 10 Jan '17

10 Jan '17

gpiod_set_value_cansleep is declared in linux/gpio/consumer.h, but that is not always included implicitly, so we have to include both gpio.h and gpio/consumer.h here: sound/soc/samsung/tm2_wm5110.c: In function 'tm2_mic_bias': sound/soc/samsung/tm2_wm5110.c:220:3: error: implicit declaration of function 'gpiod_set_value_cansleep';did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] Fixes: 1bfbc260a5b4 ("ASoC: samsung: Add machine driver for Exynos5433 based TM2 board") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/samsung/tm2_wm5110.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 5cdf7d19b87f..24cc9d63ce87 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -12,6 +12,7 @@ #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <sound/pcm_params.h> -- 2.9.0

2 3

[alsa-devel] [PATCH v5 0/3] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 10 Jan '17

10 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Changes in v2: - Addressed Mark's comment on patch 1, split this into 2 as this had additional changes other than configuring the DMA. - Addressed Takashi's comment on the usage of snd_hdac_updatel() Changes in v3: - Addressed Takashi's comment on superfluous shift used in reading the value of PPCTL register and also cleaned the condition for updating the reg value. Changes in v4: - Acked by Takashi for patch "ALSA: hda: check stream decoupled register state". - Removed compilation warning in patch "ASoC: hdac_hdmi: Enable pin and converter in prepare". - Remove 4 patches which are already applied. Changes in v5: - Removed the SSP MCLK patchs from this series, will rework and posted later based on Marks' comments. Jeeja KP (3): ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/soc/codecs/hdac_hdmi.c | 37 +++++++++++++++++-------------------- sound/soc/intel/skylake/skl-pcm.c | 6 +++--- 2 files changed, 20 insertions(+), 23 deletions(-) -- 2.5.0

1 3

[alsa-devel] [PATCH] ASoC: ak4642: Replace mdelay function to msleep
by Kuninori Morimoto 10 Jan '17

10 Jan '17

From: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Replace mdelay to msleep to avoid busy loop on ak4642_lout_event(). Otherwise, sometimes playback doesn't work correctly when pulseaudio was used. Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 40500cd..0205ae1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -186,7 +186,7 @@ static int ak4642_lout_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: /* after widget power up */ /* Power save mode OFF */ - mdelay(popup_wait); + msleep(popup_wait); snd_soc_update_bits(codec, SG_SL2, LOPS, 0); break; case SND_SOC_DAPM_PRE_PMD: /* before widget power down */ -- 2.9.2

2 1

[alsa-devel] [PATCH] Add a quirk for Plantronics BT600
by denk 10 Jan '17

10 Jan '17

Plantronics BT600 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x1" and "cannot get freq at ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu <denk(a)post.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b3fd238..9d747a9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1136,6 +1136,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ + case USB_ID(0x047F, 0x02F7): /* Plantronics BT-600 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ ---

2 1

[alsa-devel] [PATCH 1/4] alsa-lib:pcm: returning semop error code for semaphore up/down failures.
by sutar.mounesh＠gmail.com 10 Jan '17

10 Jan '17

From: Mounesh Sutar <sutar.mounesh(a)gmail.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 91e816c..aee38c5 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -263,7 +263,10 @@ static inline int snd_pcm_direct_semaphore_down(snd_pcm_direct_t *dmix, int sem_ { struct sembuf op[2] = { { sem_num, 0, 0 }, { sem_num, 1, SEM_UNDO } }; int err = semop(dmix->semid, op, 2); - if (err == 0) dmix->locked[sem_num]++; + if (err == 0) + dmix->locked[sem_num]++; + else if (err == -1) + err = -errno; return err; } @@ -271,7 +274,10 @@ static inline int snd_pcm_direct_semaphore_up(snd_pcm_direct_t *dmix, int sem_nu { struct sembuf op = { sem_num, -1, SEM_UNDO | IPC_NOWAIT }; int err = semop(dmix->semid, &op, 1); - if (err == 0) dmix->locked[sem_num]--; + if (err == 0) + dmix->locked[sem_num]--; + else if (err == -1) + err = -errno; return err; } -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] alsa-utils: improve alsa-info.sh
by Pierre-Louis Bossart 10 Jan '17

10 Jan '17

Provide additional DMI information and filter ACPI/HID device status to help with platform quirks Pierre-Louis Bossart (2): alsa-info: provide more DMI information alsa-info: add ACPI device status alsa-info/alsa-info.sh | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) -- 2.7.4

2 3

[alsa-devel] [PATCH 4/4] alsa-lib:pcm: don't return bogus buffer levels in xrun state
by sutar.mounesh＠gmail.com 10 Jan '17

10 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 2e7d9d2..1948fa0 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -882,6 +882,9 @@ static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) return err; } + if (dmix->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 09d165d..cdd1521 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -570,6 +570,9 @@ static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) return err; } + if (dshare->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 2b0df89..60a5843 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -454,6 +454,9 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_avail_update(snd_pcm_t *pcm) if (err < 0) return err; } + if (dsnoop->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_capture_avail(pcm); } -- 2.7.4

1 0

[alsa-devel] [PATCH 3/4] alsa-lib:pcm: check state before enter poll on timer.
by sutar.mounesh＠gmail.com 10 Jan '17

10 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> To avoid the chances of timeout, we need to check the enter poll in state xrun. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 2bb2d4a..0642852 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -645,6 +645,46 @@ int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) return 0; } +/* + * This is the only operation guaranteed to be called before entering poll(). + * Direct plugins use fd of snd_timer to poll on, these timers do NOT check + * state of substream in kernel by intention. + * Only the enter to xrun might be notified once (SND_TIMER_EVENT_MSTOP). + * If xrun event was not correctly handled or was ignored it will never be + * evaluated again afterwards. + * This will result in snd_pcm_wait() always returning timeout. + * In contrast poll() on pcm hardware checks ALSA state and will immediately + * return POLLERR on XRUN. + * + * To prevent timeout and applications endlessly spinning without xrun + * detected we add a state check here which may trigger the xrun sequence. + * + * return count of filled descriptors or negative error code + */ +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, + unsigned int space) +{ + if (pcm->poll_fd < 0) { + SNDMSG("poll_fd < 0"); + return -EIO; + } + if (space >= 1 && pfds) { + pfds->fd = pcm->poll_fd; + pfds->events = pcm->poll_events | POLLERR | POLLNVAL; + } else { + return 0; + } + + /* this will also evaluate slave state and enter xrun if necessary */ + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } + return 1; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { snd_pcm_direct_t *dmix = pcm->private_data; diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index d48f6b4..66107ec 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -307,6 +307,8 @@ int snd_pcm_direct_parse_bindings(snd_pcm_direct_t *dmix, snd_config_t *cfg); int snd_pcm_direct_nonblock(snd_pcm_t *pcm, int nonblock); int snd_pcm_direct_async(snd_pcm_t *pcm, int sig, pid_t pid); +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, + unsigned int space); int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); int snd_pcm_direct_info(snd_pcm_t *pcm, snd_pcm_info_t * info); int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 258d4de..2e7d9d2 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -462,17 +462,22 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dmix->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dmix, pcm); if (dmix->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dmix->state; @@ -968,7 +973,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .avail_update = snd_pcm_dmix_avail_update, .mmap_commit = snd_pcm_dmix_mmap_commit, .htimestamp = snd_pcm_dmix_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_dmix_poll_revents, }; diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 570f180..09d165d 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -255,17 +255,22 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dshare->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dshare->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dshare, pcm); if (dshare->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dshare->state; @@ -646,7 +651,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .avail_update = snd_pcm_dshare_avail_update, .mmap_commit = snd_pcm_dshare_mmap_commit, .htimestamp = snd_pcm_dshare_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 6b721ec..2b0df89 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -208,17 +208,22 @@ static int snd_pcm_dsnoop_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dsnoop->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dsnoop->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dsnoop, pcm); return dsnoop->state; } @@ -531,7 +536,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .avail_update = snd_pcm_dsnoop_avail_update, .mmap_commit = snd_pcm_dsnoop_mmap_commit, .htimestamp = snd_pcm_dsnoop_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; -- 2.7.4

1 0

[alsa-devel] [PATCH 2/4] alsa-lib: Fix for sync issue on xrun recover
by sutar.mounesh＠gmail.com 10 Jan '17

10 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> If using very short periods, DSHARE/DSNOOP/DMIX may report underruns while in status 'prepared'. This prohibits correct recovery. Now slave xrun conditions for DSHARE/DSNOOP/DMIX are being handled properly. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 0770abc..2bb2d4a 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -550,6 +550,101 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) return 0; } +/* + * Recover slave on XRUN. + * Even if direct plugins disable xrun detection, there might be an xrun + * raised directly by some drivers. + * The first client recovers slave pcm. + * Each client needs to execute sw xrun handling afterwards + */ +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) +{ + int ret; + int semerr; + + semerr = snd_pcm_direct_semaphore_down(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMDOWN FAILED with err %d", semerr); + return semerr; + } + + if (snd_pcm_state(direct->spcm) != SND_PCM_STATE_XRUN) { + /* ignore... someone else already did recovery */ + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return 0; + } + + ret = snd_pcm_prepare(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to prepare slave"); + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + + if (direct->type == SND_PCM_TYPE_DSHARE) { + const snd_pcm_channel_area_t *dst_areas; + dst_areas = snd_pcm_mmap_areas(direct->spcm); + snd_pcm_areas_silence(dst_areas, 0, direct->spcm->channels, + direct->spcm->buffer_size, + direct->spcm->format); + } + + ret = snd_pcm_start(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to start slave"); + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + direct->shmptr->recoveries++; + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return 0; +} + +/* + * enter xrun state, if slave xrun occured + * @return: 0 - no xrun >0: xrun happened + */ +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) +{ + if (direct->shmptr->recoveries != direct->recoveries) { + /* no matter how many xruns we missed - + * so don't increment but just update to actual counter + */ + direct->recoveries = direct->shmptr->recoveries; + pcm->fast_ops->drop(pcm); + /* trigger_tstamp update is missing in drop callbacks */ + gettimestamp(&direct->trigger_tstamp, pcm->tstamp_type); + /* no timer clear: + * if slave already entered xrun again the event is lost. + * snd_pcm_direct_clear_timer_queue(direct); + */ + direct->state = SND_PCM_STATE_XRUN; + return 1; + } + return 0; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { snd_pcm_direct_t *dmix = pcm->private_data; @@ -572,6 +667,12 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in } switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + /* recover slave and update client state to xrun + * before returning POLLERR + */ + snd_pcm_direct_slave_recover(dmix); + snd_pcm_direct_client_chk_xrun(dmix, pcm); + /* fallthrough */ case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SETUP: events |= POLLERR; @@ -1382,6 +1483,7 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm dmix->slave_buffer_size = spcm->buffer_size; dmix->slave_period_size = dmix->shmptr->s.period_size; dmix->slave_boundary = spcm->boundary; + dmix->recoveries = dmix->shmptr->recoveries; ret = snd_pcm_mmap(spcm); if (ret < 0) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index aee38c5..d48f6b4 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -66,6 +66,7 @@ typedef struct { char socket_name[256]; /* name of communication socket */ snd_pcm_type_t type; /* PCM type (currently only hw) */ int use_server; + unsigned int recoveries; /* no of executed recoveries on slave*/ struct { unsigned int format; snd_interval_t rate; @@ -157,6 +158,7 @@ struct snd_pcm_direct { int var_periodsize; /* allow variable period size if max_periods is != -1*/ unsigned int channels; /* client's channels */ unsigned int *bindings; + unsigned int recoveries; /* mirror of executed recoveries on slave */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -324,7 +326,8 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm snd_pcm_chmap_query_t **snd_pcm_direct_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_direct_get_chmap(snd_pcm_t *pcm); int snd_pcm_direct_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); - +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct); +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm); int snd_timer_async(snd_timer_t *timer, int sig, pid_t pid); struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 0ab7323..258d4de 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -434,15 +434,21 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_DISCONNECTED: dmix->state = SND_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (dmix->slowptr) snd_pcm_hwsync(dmix->spcm); @@ -828,12 +834,16 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -841,8 +851,10 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dmix_start_timer(pcm, dmix)) < 0) return err; } else if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } if (dmix->state == SND_PCM_STATE_RUNNING || dmix->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -858,10 +870,13 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index a1fed5d..570f180 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -162,7 +162,7 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + old_slave_hw_ptr = dshare->slave_hw_ptr; dshare->slave_hw_ptr = slave_hw_ptr; diff = slave_hw_ptr - old_slave_hw_ptr; @@ -202,15 +202,21 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_DISCONNECTED: dshare->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (dshare->slowptr) snd_pcm_hwsync(dshare->spcm); @@ -516,12 +522,16 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -529,8 +539,10 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dshare_start_timer(dshare)) < 0) return err; } else if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } if (dshare->state == SND_PCM_STATE_RUNNING || dshare->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -546,10 +558,13 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 85f0ff4..6b721ec 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -132,14 +132,21 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) snd_pcm_direct_t *dsnoop = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + int err; + switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_DISCONNECTED: dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->slowptr) snd_pcm_hwsync(dsnoop->spcm); old_slave_hw_ptr = dsnoop->slave_hw_ptr; @@ -410,12 +417,16 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->state == SND_PCM_STATE_RUNNING) { err = snd_pcm_dsnoop_sync_ptr(pcm); if (err < 0) -- 2.7.4

1 0

[alsa-devel] [PATCH 0/7] Move dell-led to drivers/platform/x86
by Michał Kępień 10 Jan '17

10 Jan '17

This patch series moves the dell-led driver from the LED subsystem to the x86 platform driver subsystem. I decided to also CC the sound subsystem contacts for the whole series as sound/pci/hda/dell_wmi_helper.c is also affected. The original motivation behind this effort was to move all code using the dell-smbios module to the x86 platform driver subsystem. While I was investigating the possibilites to do that, it quickly emerged that dell-led can and in fact should be moved to the x86 platform driver subsystem in its entirety. dell-led consists of two major parts: - the part exposing a microphone mute LED interface, introduced in db6d8cc ("dell-led: add mic mute led interface"); this interface is used by sound/pci/hda/dell_wmi_helper.c; while the original implementation used a WMI interface, it was changed to use dell-smbios in cf0d7ea ("dell-led: use dell_smbios_find_token() for finding mic DMI tokens") and 0c41a08 ("dell-led: use dell_smbios_send_request() for performing SMBIOS calls"), - the part handling an activity LED present in Dell Latitude 2100 netbooks, introduced in 72dcd8d ("leds: Add Dell Business Class Netbook LED driver"); it binds to a specific WMI GUID and then registers a LED device which is controlled using WMI (i.e. it is basically a WMI driver). Patches 1-4 clean up the microphone mute LED interface to minimize the amount of code moved around. Patch 5 moves the microphone mute LED interface to drivers/platform/x86/dell-laptop.c, effectively causing sound/pci/hda/dell_wmi_helper.c to depend on CONFIG_DELL_LAPTOP instead of CONFIG_LEDS_DELL_NETBOOKS. Patch 6 reverts dell-led to the state it was in after its initial commit 72dcd8d ("leds: Add Dell Business Class Netbook LED driver") by removing all remnants of the microphone mute LED handling code. Patch 7 moves all that is left of dell-led (i.e. the activity LED part, as originally implemented), to a new module which is placed in drivers/platform/x86/dell-wmi-led.c. This patch series is based on linux-leds/for-4.11 as the LED subsystem is affected by all patches except patch 3. If anyone reading this has access to a Dell device which has an activity LED and/or a microphone mute LED currently supported by dell-led, I would love to hear from you as I do not have the hardware needed to practically test this patch series. drivers/leds/Kconfig | 9 --- drivers/leds/Makefile | 1 - drivers/platform/x86/Kconfig | 8 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/dell-laptop.c | 28 ++++++++ .../dell-led.c => platform/x86/dell-wmi-led.c} | 75 +++------------------- include/linux/dell-led.h | 6 +- sound/pci/hda/dell_wmi_helper.c | 18 +++--- 8 files changed, 55 insertions(+), 91 deletions(-) rename drivers/{leds/dell-led.c => platform/x86/dell-wmi-led.c} (73%) -- 2.10.2

5 24

[alsa-devel] [PATCH] ASoC: ak4642: Replace mdelay function to msleep
by Kuninori Morimoto 10 Jan '17

10 Jan '17

From: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Replace mdelay to msleep to avoid busy loop on ak4642_lout_event(). Otherwise, sometimes playback doesn't work correctly when pulseaudio was used. Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 40500cd..0205ae1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -186,7 +186,7 @@ static int ak4642_lout_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: /* after widget power up */ /* Power save mode OFF */ - mdelay(popup_wait); + msleep(popup_wait); snd_soc_update_bits(codec, SG_SL2, LOPS, 0); break; case SND_SOC_DAPM_PRE_PMD: /* before widget power down */ -- 2.9.2

2 2

[alsa-devel] [PATCH] ASoC: dpcm: Avoid putting stream state to STOP when FE stream is paused
by Patrick Lai 09 Jan '17

09 Jan '17

When multiple front-ends are using the same back-end, putting state of a front-end to STOP state upon receiving pause command will result in backend stream getting released by DPCM framework unintentionally. In order to avoid backend to be released when another active front-end stream is present, put the stream state to PAUSED state instead of STOP state. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7a1eaa..6aba140 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2184,9 +2184,11 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; } out: -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,\na Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5645: set sel_i2s_pre_div1 to 2
by Bard Liao 09 Jan '17

09 Jan '17

The i2s clock pre-divider 1 is used for both i2s1 and sysclk. The i2s1 is usually used for the main i2s and the pre-divider will be set in hw_params function. However, if i2s2 is used, the pre-divider is not set in the hw_params function and the default value of i2s clock pre-divider 1 is too high for sysclk and DMIC usage. Fix by overriding default divider value to 2. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681 Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 10c2a56..1ac96ef 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3833,6 +3833,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } + regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, + RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2); + if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH 1/2] ASoC: dwc: Add record capability in PIO mode
by Jose Abreu 09 Jan '17

09 Jan '17

Up until now PIO mode offered only playback support. With this patch we add support for record mode. The PCM was refactored so that we could reuse the existing infrastructure without many changes. We have support for 16 and 32 bits of sample size using only 2 channels. Tested in a x86_64 platform and in ARC AXS101 SDP platform. Signed-off-by: Jose Abreu <joabreu(a)synopsys.com> Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/dwc/designware_i2s.c | 9 +++- sound/soc/dwc/designware_pcm.c | 97 +++++++++++++++++++++++++++++++++--------- sound/soc/dwc/local.h | 9 +++- 3 files changed, 92 insertions(+), 23 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index bdf8398..9c46e41 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -121,9 +121,14 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) irq_valid = true; } - /* Data available. Record mode not supported in PIO mode */ - if (isr[i] & ISR_RXDA) + /* + * Data available. Retrieve samples from FIFO + * NOTE: Only two channels supported + */ + if ((isr[i] & ISR_RXDA) && (i == 0) && dev->use_pio) { + dw_pcm_pop_rx(dev); irq_valid = true; + } /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { diff --git a/sound/soc/dwc/designware_pcm.c b/sound/soc/dwc/designware_pcm.c index 4a83a22..b063c86 100644 --- a/sound/soc/dwc/designware_pcm.c +++ b/sound/soc/dwc/designware_pcm.c @@ -41,10 +41,33 @@ return tx_ptr; \ } +#define dw_pcm_rx_fn(sample_bits) \ +static unsigned int dw_pcm_rx_##sample_bits(struct dw_i2s_dev *dev, \ + struct snd_pcm_runtime *runtime, unsigned int rx_ptr, \ + bool *period_elapsed) \ +{ \ + u##sample_bits (*p)[2] = (void *)runtime->dma_area; \ + unsigned int period_pos = rx_ptr % runtime->period_size; \ + int i; \ +\ + for (i = 0; i < dev->fifo_th; i++) { \ + p[rx_ptr][0] = ioread32(dev->i2s_base + LRBR_LTHR(0)); \ + p[rx_ptr][1] = ioread32(dev->i2s_base + RRBR_RTHR(0)); \ + period_pos++; \ + if (++rx_ptr >= runtime->buffer_size) \ + rx_ptr = 0; \ + } \ + *period_elapsed = period_pos >= runtime->period_size; \ + return rx_ptr; \ +} + dw_pcm_tx_fn(16); dw_pcm_tx_fn(32); +dw_pcm_rx_fn(16); +dw_pcm_rx_fn(32); #undef dw_pcm_tx_fn +#undef dw_pcm_rx_fn static const struct snd_pcm_hardware dw_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | @@ -68,27 +91,51 @@ .fifo_size = 16, }; -void dw_pcm_push_tx(struct dw_i2s_dev *dev) +static void dw_pcm_transfer(struct dw_i2s_dev *dev, bool push) { - struct snd_pcm_substream *tx_substream; - bool tx_active, period_elapsed; + struct snd_pcm_substream *substream; + bool active, period_elapsed; rcu_read_lock(); - tx_substream = rcu_dereference(dev->tx_substream); - tx_active = tx_substream && snd_pcm_running(tx_substream); - if (tx_active) { - unsigned int tx_ptr = READ_ONCE(dev->tx_ptr); - unsigned int new_tx_ptr = dev->tx_fn(dev, tx_substream->runtime, - tx_ptr, &period_elapsed); - cmpxchg(&dev->tx_ptr, tx_ptr, new_tx_ptr); + if (push) + substream = rcu_dereference(dev->tx_substream); + else + substream = rcu_dereference(dev->rx_substream); + active = substream && snd_pcm_running(substream); + if (active) { + unsigned int ptr; + unsigned int new_ptr; + + if (push) { + ptr = READ_ONCE(dev->tx_ptr); + new_ptr = dev->tx_fn(dev, substream->runtime, ptr, + &period_elapsed); + cmpxchg(&dev->tx_ptr, ptr, new_ptr); + } else { + ptr = READ_ONCE(dev->rx_ptr); + new_ptr = dev->rx_fn(dev, substream->runtime, ptr, + &period_elapsed); + cmpxchg(&dev->rx_ptr, ptr, new_ptr); + } if (period_elapsed) - snd_pcm_period_elapsed(tx_substream); + snd_pcm_period_elapsed(substream); } rcu_read_unlock(); } + +void dw_pcm_push_tx(struct dw_i2s_dev *dev) +{ + dw_pcm_transfer(dev, true); +} EXPORT_SYMBOL_GPL(dw_pcm_push_tx); +void dw_pcm_pop_rx(struct dw_i2s_dev *dev) +{ + dw_pcm_transfer(dev, false); +} +EXPORT_SYMBOL_GPL(dw_pcm_pop_rx); + static int dw_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -126,20 +173,17 @@ static int dw_pcm_hw_params(struct snd_pcm_substream *substream, switch (params_format(hw_params)) { case SNDRV_PCM_FORMAT_S16_LE: dev->tx_fn = dw_pcm_tx_16; + dev->rx_fn = dw_pcm_rx_16; break; case SNDRV_PCM_FORMAT_S32_LE: dev->tx_fn = dw_pcm_tx_32; + dev->rx_fn = dw_pcm_rx_32; break; default: dev_err(dev->dev, "invalid format\n"); return -EINVAL; } - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) { - dev_err(dev->dev, "only playback is available\n"); - return -EINVAL; - } - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) @@ -163,13 +207,21 @@ static int dw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - WRITE_ONCE(dev->tx_ptr, 0); - rcu_assign_pointer(dev->tx_substream, substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + WRITE_ONCE(dev->tx_ptr, 0); + rcu_assign_pointer(dev->tx_substream, substream); + } else { + WRITE_ONCE(dev->rx_ptr, 0); + rcu_assign_pointer(dev->rx_substream, substream); + } break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - rcu_assign_pointer(dev->tx_substream, NULL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rcu_assign_pointer(dev->tx_substream, NULL); + else + rcu_assign_pointer(dev->rx_substream, NULL); break; default: ret = -EINVAL; @@ -183,7 +235,12 @@ static snd_pcm_uframes_t dw_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct dw_i2s_dev *dev = runtime->private_data; - snd_pcm_uframes_t pos = READ_ONCE(dev->tx_ptr); + snd_pcm_uframes_t pos; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + pos = READ_ONCE(dev->tx_ptr); + else + pos = READ_ONCE(dev->rx_ptr); return pos < runtime->buffer_size ? pos : 0; } diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 68afd75..91dc70a 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -105,20 +105,27 @@ struct dw_i2s_dev { struct i2s_clk_config_data config; int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); - /* data related to PIO transfers (TX) */ + /* data related to PIO transfers */ bool use_pio; struct snd_pcm_substream __rcu *tx_substream; + struct snd_pcm_substream __rcu *rx_substream; unsigned int (*tx_fn)(struct dw_i2s_dev *dev, struct snd_pcm_runtime *runtime, unsigned int tx_ptr, bool *period_elapsed); + unsigned int (*rx_fn)(struct dw_i2s_dev *dev, + struct snd_pcm_runtime *runtime, unsigned int rx_ptr, + bool *period_elapsed); unsigned int tx_ptr; + unsigned int rx_ptr; }; #if IS_ENABLED(CONFIG_SND_DESIGNWARE_PCM) void dw_pcm_push_tx(struct dw_i2s_dev *dev); +void dw_pcm_pop_rx(struct dw_i2s_dev *dev); int dw_pcm_register(struct platform_device *pdev); #else void dw_pcm_push_tx(struct dw_i2s_dev *dev) { } +void dw_pcm_pop_rx(struct dw_i2s_dev *dev) { } int dw_pcm_register(struct platform_device *pdev) { return -EINVAL; -- 1.9.1

2 3

[alsa-devel] [PATCH v3 0/6] ASoC: sun4i-codec: Add support for A31 Codec
by Chen-Yu Tsai 09 Jan '17

09 Jan '17

Hi everyone, This is v3 of my Allwinner A31 audio codec support series. These are the remaining driver patches not yet merged, with 1 new fix, and the dts file patches. The A31's internal codec is similar (in terms of DMA, interface and control layouts) to the one found in the A10/A13/A20 SoCs. However it has more external inputs and outputs, the mixer controls are now stereo (left/right separated), and it also has some audio processing features (not supported yet). Changes since v2: - Added patch """ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route""" which fixes line out when all the sources have their left channel muted. - Dropped unused codec_pa_pin from A31 hummingbird dts patch. In the past we always provided a pinmux setting for GPIOs used, to avoid accidentally muxing a pin that was already used for GPIO, or vice versa. The pinctrl subsystem now supports exclusive use of a pin for GPIO, which the sunxi platform wants to migrate to. This drops the requirement of the explicit pinmux setting for GPIOs. - Added Maxime's Acked-by for the two existing patches from v2. After these are merged I plan to submit patches to support the audio codec found in Allwinner's A23 and H3 SoCs. Regards ChenYu Chen-Yu Tsai (6): ASoC: sun4i-codec: Add support for optional reset control to quirks ASoC: sun4i-codec: Add support for A31 ADC capture path ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route ARM: dts: sun6i: Add audio codec device node ARM: dts: sun6i: hummingbird: Enable internal audio codec ARM: dts: sun6i: sina31s: Enable internal audio codec arch/arm/boot/dts/sun6i-a31-hummingbird.dts | 13 ++++ arch/arm/boot/dts/sun6i-a31.dtsi | 13 ++++ arch/arm/boot/dts/sun6i-a31s-sina31s.dts | 8 +++ sound/soc/sunxi/sun4i-codec.c | 94 ++++++++++++++++++++++++++++- 4 files changed, 127 insertions(+), 1 deletion(-) -- 2.10.2

3 10

[alsa-devel] [PATCH] ASoC: nau8825: Add Class G function options
by Ben Zhang 09 Jan '17

09 Jan '17

Add a DAPM_MUX for userspace to control the Class G function: "Switching": Use +-1.8V or +-0.9V supply, based on signal amplitude. "1.8V": Always use +-1.8V. The default is "Switching" which is the existing behavior without this patch, so no effect on current users. On some devices, the switching may cause a clicking noise at low volume. "1.8V" eliminates the noise at the expense of a few additional mW. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index efe3a44..ddd15a0 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1062,6 +1062,10 @@ static const char * const nau8825_dac_src[] = { "DACL", "DACR", }; +static const char * const nau8825_class_g_src[] = { + "Switching", "1.8V" +}; + static SOC_ENUM_SINGLE_DECL( nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); @@ -1070,12 +1074,17 @@ static SOC_ENUM_SINGLE_DECL( nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); +static SOC_ENUM_SINGLE_DECL( + nau8825_class_g_enum, SND_SOC_NOPM, 0, nau8825_class_g_src); + static const struct snd_kcontrol_new nau8825_dacl_mux = SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); static const struct snd_kcontrol_new nau8825_dacr_mux = SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); +static const struct snd_kcontrol_new nau8825_class_g_mux = + SOC_DAPM_ENUM("Class G Source", nau8825_class_g_enum); static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, @@ -1154,9 +1163,16 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, NAU8825_REG_BOOST, 9, 1, NULL, 0), - /* Class G operation control*/ - SND_SOC_DAPM_PGA_S("Class G", 10, + /* Class G operation control. "Class G" MUX options: + * "Switching": Use +-1.8V or +-0.9V supply, based on signal amplitude. + * "1.8V": Always use +-1.8V. + * The PGA_S widget is used to ensure the Class G control bit is set at + * the correct time w.r.t the other widgets. + */ + SND_SOC_DAPM_PGA_S("Class G Driver", 10, NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("Class G", SND_SOC_NOPM, 0, 0, + &nau8825_class_g_mux), SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), @@ -1197,7 +1213,9 @@ static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { {"HPOR Pulldown", NULL, "Output DACR"}, {"HP Boost Driver", NULL, "HPOL Pulldown"}, {"HP Boost Driver", NULL, "HPOR Pulldown"}, - {"Class G", NULL, "HP Boost Driver"}, + {"Class G Driver", NULL, "HP Boost Driver"}, + {"Class G", "1.8V", "HP Boost Driver"}, + {"Class G", "Switching", "Class G Driver"}, {"HPOL", NULL, "Class G"}, {"HPOR", NULL, "Class G"}, }; -- 2.8.0.rc3.226.g39d4020

2 3

[alsa-devel] Input sound doesn't work without power
by Gustavo Duarte 09 Jan '17

09 Jan '17

Hi all, We are facing a problem with audio intput (mic), on a laptop with sound device ALC283. When the laptop boots connected to power line, all works fine. When the laptop boots without power line connection, sound input doesn't work. No audio can be recorder, using arecord or qarecord. I got the output of "arecord -l" comand in each case. Follwoing are the results. With power line connection: **** Lista de CAPTURE dispositivos hardware **** tarjeta 0: PCH [HDA Intel PCH], dispositivo 0: ALC283 Analog [ALC283 Analog] Subdispositivos: 1/1 Subdispositivo #0: subdevice #0 Without power line connection: **** Lista de CAPTURE dispositivos hardware **** tarjeta 0: PCH [HDA Intel PCH], dispositivo 0: ALC283 Analog [ALC283 Analog] Subdispositivos: 0/1 Subdispositivo #0: subdevice #0 As yo can see, the difference is on, Subdispositivos :1/1 (with power line) and Subdispositivos: 0/1 without power line. Anybody can help me resolve this ? Do you need some additional info ? Thanks advance.

1 0

[alsa-devel] Building a new out-of-tree ALSA module on Ubuntu Xenial (was gitcompile fail)
by Mark Brand 09 Jan '17

09 Jan '17

Hi all My abundant thanks to Clemens and Takashi for your kind and patient assistance. May I now answer my own question by offering the following step-by-step procedure in case anyone comes looking for it? Any improvements or suggestions would be greatly appreciated too, thank you. $ sudo apt-get install linux-headers-$(uname -r) $ sudo mkdir /lib/modules/$(uname -r)/updates $ sudo apt-get build-dep linux-image-$(uname -r) $ mkdir $HOME/src; cd $HOME/src $ apt-get source linux-image-$(uname -r) $ cd linux-$(uname -r | awk -F - '{print $1}') Now I am at the root of the kernel source tree. In my case, I want to add a new driver for an as-yet unsupported device. I find a supported device that appears substantially similar to what I need (sound/drivers/mtpav.c) and base my hack on that - producing sound/drivers/mq8portse.c. I then add the following lines to the sound/drivers/Makefile (each mirroring entries found in respect of mtpav.c) snd-mq8portse-objs := mq8portse.o obj-$(CONFIG_SND_MQ8PORTSE) += snd-mq8portse.o Now (from the root of the kernel source tree): $ cd sound/drivers/ $ CONFIG_SND_MQ8PORTSE=m make -C /lib/modules/$(uname -r)/build M=$(pwd) snd-mq8portse.ko This produces my module in the same folder, which I copy to the updates folder. $ sudo cp snd-mq8portse.ko /lib/modules/$(uname -r)/updates/ $ sudo depmod -a Now, because my particular device is attached to the LPT port, I need to unload the lp driver before snd-mq8portse will load. $ sudo modprobe -r lp $ sudo modprobe snd-mq8portse $ modinfo snd-mq8portse I try it out in my ALSA client ... then remove ... $ sudo modprobe -r snd-mq8portse The MusicQuest 8Port/SE is attached to the LPT port, so a more permanent setup requires that I blacklist all typical parport stuff. I add the following as /etc/modprobe.d/blacklist-parport.conf: # keep LPT port open for mq8portse alias parport_pc off alias parport off blacklist parport_pc blacklist parport blacklist ppdev blacklist lp I also comment out autoloading of lp in /etc/modules. I can then add mq8portse to /etc/modules for automatic loading, unless I want to defer to manual modprobe-ing. Once again, thank you ever so much, one and all. Next I need to find out how to submit this driver for inclusion in ALSA. *Mark Brand* Lecturer in Music Technology School of Music, Art & Design Summerstrand Campus (South) Tel: +27 (0)41 5042250 Fax: +27 (0)41 5042507

1 0

[alsa-devel] [PATCH v2 0/2] ASoC: dwc: Add support for record mode
by Jose Abreu 09 Jan '17

09 Jan '17

This patchset adds support for record mode in I2S PIO pcm and enables 24bit transaction. With this patchset we now have PIO support for playback and record using only two channels, at sample rate 32k, 44.1k and 48k and sample width 16/24 and 32 bit. Tested in a x86_64 platform and in ARC AXS101 SDP platform. Jose Abreu (2): ASoC: dwc: Add record capability in PIO mode ASoC: dwc: Enable 24 bit sample size in PIO mode sound/soc/dwc/designware_i2s.c | 9 +++- sound/soc/dwc/designware_pcm.c | 99 +++++++++++++++++++++++++++++++++--------- sound/soc/dwc/local.h | 9 +++- 3 files changed, 94 insertions(+), 23 deletions(-) -- Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org v2: - Added missing cover letter 1.9.1

1 2

[alsa-devel] [PATCH 2/6 v4] alsa-lib: Fix for sync issue on xrun recover
by sutar.mounesh＠gmail.com 09 Jan '17

09 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> If using very short periods, DSHARE/DSNOOP/DMIX may report underruns while in status 'prepared'. This prohibits correct recovery. Now slave xrun conditions for DSHARE/DSNOOP/DMIX are being handled properly. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 0770abc..2bb2d4a 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -550,6 +550,101 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) return 0; } +/* + * Recover slave on XRUN. + * Even if direct plugins disable xrun detection, there might be an xrun + * raised directly by some drivers. + * The first client recovers slave pcm. + * Each client needs to execute sw xrun handling afterwards + */ +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) +{ + int ret; + int semerr; + + semerr = snd_pcm_direct_semaphore_down(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMDOWN FAILED with err %d", semerr); + return semerr; + } + + if (snd_pcm_state(direct->spcm) != SND_PCM_STATE_XRUN) { + /* ignore... someone else already did recovery */ + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return 0; + } + + ret = snd_pcm_prepare(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to prepare slave"); + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + + if (direct->type == SND_PCM_TYPE_DSHARE) { + const snd_pcm_channel_area_t *dst_areas; + dst_areas = snd_pcm_mmap_areas(direct->spcm); + snd_pcm_areas_silence(dst_areas, 0, direct->spcm->channels, + direct->spcm->buffer_size, + direct->spcm->format); + } + + ret = snd_pcm_start(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to start slave"); + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + direct->shmptr->recoveries++; + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr < 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return 0; +} + +/* + * enter xrun state, if slave xrun occured + * @return: 0 - no xrun >0: xrun happened + */ +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) +{ + if (direct->shmptr->recoveries != direct->recoveries) { + /* no matter how many xruns we missed - + * so don't increment but just update to actual counter + */ + direct->recoveries = direct->shmptr->recoveries; + pcm->fast_ops->drop(pcm); + /* trigger_tstamp update is missing in drop callbacks */ + gettimestamp(&direct->trigger_tstamp, pcm->tstamp_type); + /* no timer clear: + * if slave already entered xrun again the event is lost. + * snd_pcm_direct_clear_timer_queue(direct); + */ + direct->state = SND_PCM_STATE_XRUN; + return 1; + } + return 0; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { snd_pcm_direct_t *dmix = pcm->private_data; @@ -572,6 +667,12 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in } switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + /* recover slave and update client state to xrun + * before returning POLLERR + */ + snd_pcm_direct_slave_recover(dmix); + snd_pcm_direct_client_chk_xrun(dmix, pcm); + /* fallthrough */ case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SETUP: events |= POLLERR; @@ -1382,6 +1483,7 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm dmix->slave_buffer_size = spcm->buffer_size; dmix->slave_period_size = dmix->shmptr->s.period_size; dmix->slave_boundary = spcm->boundary; + dmix->recoveries = dmix->shmptr->recoveries; ret = snd_pcm_mmap(spcm); if (ret < 0) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index a7c0439..5f51021 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -66,6 +66,7 @@ typedef struct { char socket_name[256]; /* name of communication socket */ snd_pcm_type_t type; /* PCM type (currently only hw) */ int use_server; + unsigned int recoveries; /* no of executed recoveries on slave*/ struct { unsigned int format; snd_interval_t rate; @@ -157,6 +158,7 @@ struct snd_pcm_direct { int var_periodsize; /* allow variable period size if max_periods is != -1*/ unsigned int channels; /* client's channels */ unsigned int *bindings; + unsigned int recoveries; /* mirror of executed recoveries on slave */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -328,7 +330,8 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm snd_pcm_chmap_query_t **snd_pcm_direct_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_direct_get_chmap(snd_pcm_t *pcm); int snd_pcm_direct_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); - +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct); +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm); int snd_timer_async(snd_timer_t *timer, int sig, pid_t pid); struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 0ab7323..258d4de 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -434,15 +434,21 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_DISCONNECTED: dmix->state = SND_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (dmix->slowptr) snd_pcm_hwsync(dmix->spcm); @@ -828,12 +834,16 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -841,8 +851,10 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dmix_start_timer(pcm, dmix)) < 0) return err; } else if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } if (dmix->state == SND_PCM_STATE_RUNNING || dmix->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -858,10 +870,13 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index a1fed5d..570f180 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -162,7 +162,7 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + old_slave_hw_ptr = dshare->slave_hw_ptr; dshare->slave_hw_ptr = slave_hw_ptr; diff = slave_hw_ptr - old_slave_hw_ptr; @@ -202,15 +202,21 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_DISCONNECTED: dshare->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (dshare->slowptr) snd_pcm_hwsync(dshare->spcm); @@ -516,12 +522,16 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -529,8 +539,10 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dshare_start_timer(dshare)) < 0) return err; } else if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } if (dshare->state == SND_PCM_STATE_RUNNING || dshare->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -546,10 +558,13 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 85f0ff4..6b721ec 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -132,14 +132,21 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) snd_pcm_direct_t *dsnoop = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + int err; + switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_DISCONNECTED: dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->slowptr) snd_pcm_hwsync(dsnoop->spcm); old_slave_hw_ptr = dsnoop->slave_hw_ptr; @@ -410,12 +417,16 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->state == SND_PCM_STATE_RUNNING) { err = snd_pcm_dsnoop_sync_ptr(pcm); if (err < 0) -- 2.7.4

3 2

[alsa-devel] [PATCH] pcm: returning semop error code for semaphore up/down failures.
by sutar.mounesh＠gmail.com 09 Jan '17

09 Jan '17

From: Mounesh Sutar <sutar.mounesh(a)gmail.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 91e816c..a7c0439 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -263,7 +263,12 @@ static inline int snd_pcm_direct_semaphore_down(snd_pcm_direct_t *dmix, int sem_ { struct sembuf op[2] = { { sem_num, 0, 0 }, { sem_num, 1, SEM_UNDO } }; int err = semop(dmix->semid, op, 2); - if (err == 0) dmix->locked[sem_num]++; + if (err == 0) { + dmix->locked[sem_num]++; + } + else if (err == -1) { + err = -errno; + } return err; } @@ -271,7 +276,12 @@ static inline int snd_pcm_direct_semaphore_up(snd_pcm_direct_t *dmix, int sem_nu { struct sembuf op = { sem_num, -1, SEM_UNDO | IPC_NOWAIT }; int err = semop(dmix->semid, &op, 1); - if (err == 0) dmix->locked[sem_num]--; + if (err == 0) { + dmix->locked[sem_num]--; + } + else if (err == -1) { + err = -errno; + } return err; } -- 2.7.4

2 1

[alsa-devel] [PATCH] Revert "ASoC: core: replace aux_comp_list to component_dev_list"
by Nikita Yushchenko 09 Jan '17

09 Jan '17

This reverts commit 1a653aa44725668590b36bbe2d7fe4736a69f055 ("ASoC: core: replace aux_comp_list to component_dev_list"). That commit tries to remove card->aux_comp_list, using flagged entries in card->component_dev_list instead. However, components are added to card->component_dev_list in soc_probe_component(), which is called for aux devices by soc_probe_aux_devices(). Before that commit, it traversed card->aux_comp_list and found aux devices added by soc_bind_aux_dev(). After that commit, it traverses card->component_dev_list and finds no aux devices, because soc_probe_component() was not yet called for them. Thus all aux devices are lost and any setup that needs them no longer works. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- Fixed lost line in commit message ;) include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 17 +++++------------ 2 files changed, 8 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6cc6d0..8172a512632a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -809,10 +809,10 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; - unsigned int auxiliary:1; /* for auxiliary component of the card */ unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1152,6 +1152,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1547,6 +1548,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb1466e..15657a65e4ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1747,8 +1747,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) } component->init = aux_dev->init; - component->auxiliary = 1; - + list_add(&component->list_aux, &card->aux_comp_list); return 0; err_defer: @@ -1764,10 +1763,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1791,14 +1787,11 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { list_for_each_entry_safe(comp, _comp, - &card->component_dev_list, card_list) { - - if (!comp->auxiliary) - continue; - + &card->aux_comp_list, list_aux) { if (comp->driver->remove_order == order) { soc_remove_component(comp); - comp->auxiliary = 0; + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); } } } -- 2.1.4

4 8

[alsa-devel] [PATCH v2 0/2] ASoC: Use DMI name in sound card long name
by mengdong.lin＠linux.intel.com 09 Jan '17

09 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series is based on Liam's idea to use DMI name as the sound card short name in the audio workshop. Now we add DMI info to the card long name: - Not change the card name (short name), keep it simple. So the card ID won't change as well. - Use DMI name to make the card long name. Machine drivers can use a new API to get this feature. In user space, when the sound server ask UCM to open a sound card, UCM will try to find the best machine-specific configuration file based on the card long name, and only if not avaiable, fallback to the default configuration file that matches the card name (short name). Please see patch: ucm: Automatically load the best config file based on the card long name History: v1 from RFC: Give up using a table to map DMI info to a simple string, since we cannot maintain such a table for each machine. Just do verbatim copy of DMI info and let user space decide how to use the DMI info in card long name to find the best UCM config file. If device driver adds a flavor string, just append it after the DMI info fields to the card long name. v2: Exclude card name and only keep DMI name in the long name, to save the space since the long name cannot exceed 80 characters. Trim special characters like SPACE in the long name, because UCM configuration files & directories may use the same long name and Autoconf cannot handle special characters. Liam Girdwood (1): ASoC: core: Add API to use DMI name in sound card long name Mengdong Lin (1): ASoC: Intel: Use DMI name for sound card long name in Broadwell machine driver include/sound/soc.h | 4 ++ sound/soc/intel/boards/broadwell.c | 2 + sound/soc/soc-core.c | 99 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 105 insertions(+) -- 2.5.0

5 10

[alsa-devel] How to build alsa-utils from git?
by Paul Menzel 09 Jan '17

09 Jan '17

Dear ALSA folks, Could you please help me building alsa-utils from git. ``` $ git describe --tag origin/master v1.1.3-2-ge9a6d42 $ aclocal --version aclocal (GNU automake) 1.15 Copyright (C) 2014 Free Software Foundation, Inc. License GPLv2+: GNU GPL version 2 or later <http://gnu.org/licenses/gpl-2.0.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Written by Tom Tromey <tromey(a)redhat.com> and Alexandre Duret-Lutz <adl(a)gnu.org>. $ aclocal $ gettextize Copying file ABOUT-NLS Copying file config.rpath Not copying intl/ directory. Copying file po/Makefile.in.in Copying file po/boldquot.sed Copying file po/en(a)boldquot.header Copying file po/en(a)quot.header Copying file po/insert-header.sin Copying file po/Makevars.template Copying file po/quot.sed Copying file po/remove-potcdate.sin Copying file po/Rules-quot Creating po/ChangeLog Copying file m4/gettext.m4 Copying file m4/iconv.m4 Copying file m4/lib-ld.m4 Copying file m4/lib-link.m4 Copying file m4/lib-prefix.m4 Copying file m4/nls.m4 Copying file m4/po.m4 Copying file m4/progtest.m4 Updating EXTRA_DIST in m4/Makefile.am (backup is in m4/Makefile.am~) Updating Makefile.am (backup is in Makefile.am~) Updating configure.ac (backup is in configure.ac~) Adding an entry to ChangeLog (backup is in ChangeLog~) Please run 'aclocal -I m4' to regenerate the aclocal.m4 file. You need aclocal from GNU automake 1.9 (or newer) to do this. Then run 'autoconf' to regenerate the configure file. You will also need config.guess and config.sub, which you can get from the CVS of the 'config' project at http://savannah.gnu.org/. The commands to fetch them are $ wget 'http://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.gue…' $ wget 'http://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.sub' You might also want to copy the convenience header file gettext.h from the /usr/share/gettext directory into your package. It is a wrapper around <libintl.h> that implements the configure --disable-nls option. Press Return to acknowledge the previous three paragraphs. $ gettextize --version /usr/bin/gettextize (GNU gettext-tools) 0.19.8.1 Copyright (C) 1995-1998, 2000-2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Written by Ulrich Drepper $ autoheader configure.ac:418: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ../../lib/autoconf/status.m4:288: AC_CONFIG_FILES is expanded from... configure.ac:418: the top level autom4te: /usr/bin/m4 failed with exit status: 1 autoheader: '/usr/bin/autom4te' failed with exit status: 1 ``` Could you please share the correct commands? Thanks, Paul

3 2

[alsa-devel] [PATCH] ASoC: wm8731: Adjust clk definitions so that simple card can work
by Jörg Krause 08 Jan '17

08 Jan '17

When trying to use simple card with wm8962 the following probe error happens: wm8731 0-001a: simple-card: set_sysclk error In simple-card.c the snd_soc_dai_set_sysclk() function is called with clk_id as 0, which is an invalid clock for wm8731. Adjust the clocks source definitions in wm8731.h so that the simple card driver can work successfully Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- sound/soc/codecs/wm8731.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index e9c0c76ab73b..c7c6f15b0e42 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -31,8 +31,8 @@ #define WM8731_CACHEREGNUM 10 +#define WM8731_SYSCLK_MCLK 0 #define WM8731_SYSCLK_XTAL 1 -#define WM8731_SYSCLK_MCLK 2 #define WM8731_DAI 0 -- 2.11.0

1 0

[alsa-devel] gitcompile fail
by Mark Brand 08 Jan '17

08 Jan '17

Hi I am unable to compile ALSA from fresh git clones. Please would you advise me? My procedure: mkdir $HOME/Projects/tiwai cd $HOME/Projects/tiwai git clone git://github.com/tiwai/alsa-driver-build.git git clone git://github.com/tiwai/sound.git cd alsa-driver-build git checkout build cd alsa ./utils/setup-alsa-kernel -c ../../sound ./gitcompile --with-debug=full --enable-dynamic-minors --with-moddir=updates Problems start with setup-alsa-kernel: mark@erudite:~/Projects/tiwai/alsa-driver-build/alsa$ ./utils/setup-alsa-kernel -c ../../sound cp: cannot stat '../../sound/Documentation/sound/alsa': No such file or directory cp: cannot stat '../../sound/Documentation/DocBook/alsa-driver-api.tmpl': No such file or directory cp: cannot stat '../../sound/Documentation/DocBook/writing-an-alsa-driver.tmpl': No such file or directory cp: cannot create hard link 'alsa-kernel/Documentation/DocBook' to '../../sound/Documentation/DocBook/stylesheet.xsl': No such file or directory So I try to gitcompile anyway: mark@erudite:~/Projects/tiwai/alsa-driver-build/alsa$ ./gitcompile --with-debug=full --enable-dynamic-minors --with-moddir=updates gcc utils/mod-deps.c -o utils/mod-deps utils/mod-deps.c: In function ‘output_card_list’: utils/mod-deps.c:1086:4: warning: format not a string literal and no format arguments [-Wformat-security] printf(card_name); ^ utils/mod-deps --basedir /home/mark/Projects/tiwai/alsa-driver-build/alsa/alsa-kernel --hiddendir /home/mark/Projects/tiwai/alsa-driver-build/alsa --versiondep /home/mark/Projects/tiwai/alsa-driver-build/alsa/kconfig-vers --makeconf > toplevel.config.in can't handle 'default SND_SEQUENCER && SND_RAWMIDI', translating as depends on can't handle 'default SND_SEQUENCER && SND_OPL3_LIB', translating as depends on can't handle 'default SND_SEQUENCER && SND_OPL4_LIB', translating as depends on can't handle 'default SND_SEQUENCER && SND_SBAWE', translating as depends on can't handle 'default SND_SEQUENCER && SND_EMU10K1', translating as depends on can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_ES1968 properly, supposing it's OK can't handle word VIDEO_V4L2=y properly, supposing it's OK can't handle word VIDEO_V4L2=SND_ES1968 properly, supposing it's OK can't handle word VIDEO_V4L2=y properly, supposing it's OK can't handle word VIDEO_V4L2=SND_FM801 properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_MAESTRO3 properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_HDA properly, supposing it's OK can't handle word SND_HDA_INPUT_BEEP=y properly, supposing it's OK can't handle word INPUT=y properly, supposing it's OK can't handle word INPUT=SND_USB_CAIAQ properly, supposing it's OK Wrong condition = Makefile:152: recipe for target 'toplevel.config.in' failed make: *** [toplevel.config.in] Error 1 mark@erudite:~/Projects/tiwai/alsa-driver-build/alsa$ I have successfully compiled ALSA at various times in the past (years ago), and have intermittently worked on developing a driver for an old device. I'd very much like to finish that off and submit it now, but I'm struggling to get out of the starting blocks! You kind assistance is greatly appreciated. Regards Mark Brand

3 6

[alsa-devel] Alsamixer: missing Speakers entry in analog output choice (chipset AV200)
by Germano Massullo 08 Jan '17

08 Jan '17

I have a Asus Xonar Essence ST soundboard (Chip: AV200) on Fedora 23 kernel 4.4.5-300.fc23.x86_64 The audiocard has 3 different outputs - Speakers - Headphones - FP Headphones that have mutual exclusion, so that when you are using one of them, you cannot use the others. Sometimes the Speakers output disappears from alsamixer. By restarting the computer, the system gets back to its normal behaviour. I attach 3 logs taken when the problem was occurring. The problem happened also in past years, but in last a 1-2 years it happened less frequently

3 3

[alsa-devel] [PATCH v2] ASoC: pcm5102a: Provide a menu selection text
by Jörg Krause 07 Jan '17

07 Jan '17

Provide a menu selection text so that users can enable, disable or mark it as module in menuconfig. This is necessary for using the codec with the simple-card framework. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- Changes v2: - improve commit log (suggested by Takashi Iwai) --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e1718a8cb1c..534c8c2b5da1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -642,7 +642,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.11.0

1 0

[alsa-devel] ASoC: Using mxs-saif with simple-card
by Jörg Krause 07 Jan '17

07 Jan '17

Hi, I am trying to use the simple-card framework on a custom i.MX28 board. There are two SAIF modules (saif0 and saif1) instantiated on the cpu. A reference for the platform driver is mxs-sgtl5000. The two cpu dais saif0 and saif1 are connected to the audio codec sgtl5000. saif0 is used to playback and saif1 is used to capture. Unfortunately, I do not have a sgtl5000 codec, but a pcm5102a, a wm8524 and wm8731 to play with. For all those codec I copied the mxs-sgtl5000 driver, e.g. mxs- pcm5102a, mxs-wm8524, mxs-wm8731. I will refer to these drivers as mxs- <codec> later. The only difference was in setting the mclk. That's why I'm interested in using the simple-card driver for this platform. With the help of Mans Rullgard I came up with these parts of the device tree node when using the wm8731 codec: sound { compatible = "simple-audio-card"; simple-audio-card,name = "hbm10-wm8731"; simple-audio-card,widgets = "Headphone", "Headphone Jack", "Microphone", "Microphone Jack", "Line", "Line Jack"; simple-audio-card,routing = "Headphone Jack", "RHPOUT", "Headphone Jack", "LHPOUT", "LLINEIN", "Line Jack", "MICIN", "Mic Bias", "Mic Bias", "Microphone Jack"; simple-audio-card,dai-link@0 { format = "i2s"; bitclock-master = <&dai0_master>; frame-master = <&dai0_master>; mclk-fs = <256>; dai0_master: cpu { sound-dai = <&saif0>; }; codec { sound-dai = <&wm8731>; clocks = <&saif0>; }; }; simple-audio-card,dai-link@1 { format = "i2s"; bitclock-master = <&dai1_master>; frame-master = <&dai1_master>; mclk-fs = <256>; dai1_master: cpu { sound-dai = <&saif1>; }; codec { sound-dai = <&wm8731>; }; }; }; &saif0 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&saif0_pins_a>; assigned-clocks = <&clks 53>; assigned-clock-rates = <12288000>; status = "okay"; }; &saif1 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&saif1_pins_a>; fsl,saif-master = <&saif0>; status = "okay"; }; &i2c0 { pinctrl-names = "default"; pinctrl-0 = <&i2c0_pins_a>; status = "okay"; wm8731: wm8731@1a { #sound-dai-cells = <0>; compatible = "wlf,wm8731"; reg = <0x1a>; }; }; >From my understanding of the simple-card framework this should connect saif0 (cpu dai) <-> wm8731 (codec) and saif1 (cpu dai) <-> wm8731 (codec) as it is done in the probe function of the mxs-<codec> driver, right? However, this gives me a warning while bringing up the board: sysfs: cannot create duplicate filename '/devices/soc0/sound/mxs-saif- wm8731-hifi'. Note, that the simple-card framework works fine for mxs-saif if using only the dai0 link (playback) and leaving the dai1 link (capture). Am I doing something wrong here? Is it possible to map the two dai links using the current simple-card framework? Is anyone using the simple-card framework on mxs-saif using both playback and capture? Best regards, Jörg Krause

1 0

[alsa-devel] [PATCH resend v3] ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route
by Chen-Yu Tsai 07 Jan '17

07 Jan '17

The mono differential output for "Line Out" downmixes the stereo audio from the mixer, instead of just taking the left channel. Add a route from the "Right Mixer" to "Line Out Source Playback Route" through the "Mono Differential" path, so DAPM doesn't shut down everything if the left channel is muted. Fixes: 0f909f98d7cb ("ASoC: sun4i-codec: Add support for A31 Line Out playback") Signed-off-by: Chen-Yu Tsai <wens(a)csie.org> Acked-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- Last sent 2016/11/24. Resending with Maxime's ack. --- sound/soc/sunxi/sun4i-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 848af01692a0..c3aab10fa085 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1058,6 +1058,7 @@ static const struct snd_soc_dapm_route sun6i_codec_codec_dapm_routes[] = { { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, { "LINEOUT", NULL, "Line Out Source Playback Route" }, /* ADC Routes */ -- 2.11.0

1 0

[alsa-devel] [PATCH 2/6 v3] alsa-lib: Fix for sync issue on xrun recover
by sutar.mounesh＠gmail.com 07 Jan '17

07 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> If using very short periods, DSHARE/DSNOOP/DMIX may report underruns while in status 'prepared'. This prohibits correct recovery. Now slave xrun conditions for DSHARE/DSNOOP/DMIX are being handled properly. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 0770abc..4e7fda8 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -550,6 +550,101 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) return 0; } +/* + * Recover slave on XRUN. + * Even if direct plugins disable xrun detection, there might be an xrun + * raised directly by some drivers. + * The first client recovers slave pcm. + * Each client needs to execute sw xrun handling afterwards + */ +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) +{ + int ret; + int semerr; + + semerr = snd_pcm_direct_semaphore_down(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMDOWN FAILED with err %d", semerr); + return semerr; + } + + if (snd_pcm_state(direct->spcm) != SND_PCM_STATE_XRUN) { + /* ignore... someone else already did recovery */ + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + + ret = snd_pcm_prepare(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to prepare slave"); + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + + if (direct->type == SND_PCM_TYPE_DSHARE) { + const snd_pcm_channel_area_t *dst_areas; + dst_areas = snd_pcm_mmap_areas(direct->spcm); + snd_pcm_areas_silence(dst_areas, 0, direct->spcm->channels, + direct->spcm->buffer_size, + direct->spcm->format); + } + + ret = snd_pcm_start(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to start slave"); + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return ret; + } + direct->shmptr->recoveries++; + semerr = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT); + if (semerr > 0) { + SNDERR("SEMUP FAILED with err %d", semerr); + return semerr; + } + return 0; +} + +/* + * enter xrun state, if slave xrun occured + * @return: 0 - no xrun >0: xrun happened + */ +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) +{ + if (direct->shmptr->recoveries != direct->recoveries) { + /* no matter how many xruns we missed - + * so don't increment but just update to actual counter + */ + direct->recoveries = direct->shmptr->recoveries; + pcm->fast_ops->drop(pcm); + /* trigger_tstamp update is missing in drop callbacks */ + gettimestamp(&direct->trigger_tstamp, pcm->tstamp_type); + /* no timer clear: + * if slave already entered xrun again the event is lost. + * snd_pcm_direct_clear_timer_queue(direct); + */ + direct->state = SND_PCM_STATE_XRUN; + return 1; + } + return 0; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { snd_pcm_direct_t *dmix = pcm->private_data; @@ -572,6 +667,12 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in } switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + /* recover slave and update client state to xrun + * before returning POLLERR + */ + snd_pcm_direct_slave_recover(dmix); + snd_pcm_direct_client_chk_xrun(dmix, pcm); + /* fallthrough */ case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SETUP: events |= POLLERR; @@ -1382,6 +1483,7 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm dmix->slave_buffer_size = spcm->buffer_size; dmix->slave_period_size = dmix->shmptr->s.period_size; dmix->slave_boundary = spcm->boundary; + dmix->recoveries = dmix->shmptr->recoveries; ret = snd_pcm_mmap(spcm); if (ret < 0) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 91e816c..569d2be 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -66,6 +66,7 @@ typedef struct { char socket_name[256]; /* name of communication socket */ snd_pcm_type_t type; /* PCM type (currently only hw) */ int use_server; + unsigned int recoveries; /* no of executed recoveries on slave*/ struct { unsigned int format; snd_interval_t rate; @@ -157,6 +158,7 @@ struct snd_pcm_direct { int var_periodsize; /* allow variable period size if max_periods is != -1*/ unsigned int channels; /* client's channels */ unsigned int *bindings; + unsigned int recoveries; /* mirror of executed recoveries on slave */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -318,7 +320,8 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm snd_pcm_chmap_query_t **snd_pcm_direct_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_direct_get_chmap(snd_pcm_t *pcm); int snd_pcm_direct_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); - +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct); +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm); int snd_timer_async(snd_timer_t *timer, int sig, pid_t pid); struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 0ab7323..258d4de 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -434,15 +434,21 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_DISCONNECTED: dmix->state = SND_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (dmix->slowptr) snd_pcm_hwsync(dmix->spcm); @@ -828,12 +834,16 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -841,8 +851,10 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dmix_start_timer(pcm, dmix)) < 0) return err; } else if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } if (dmix->state == SND_PCM_STATE_RUNNING || dmix->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -858,10 +870,13 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index a1fed5d..570f180 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -162,7 +162,7 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + old_slave_hw_ptr = dshare->slave_hw_ptr; dshare->slave_hw_ptr = slave_hw_ptr; diff = slave_hw_ptr - old_slave_hw_ptr; @@ -202,15 +202,21 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_DISCONNECTED: dshare->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (dshare->slowptr) snd_pcm_hwsync(dshare->spcm); @@ -516,12 +522,16 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -529,8 +539,10 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dshare_start_timer(dshare)) < 0) return err; } else if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } if (dshare->state == SND_PCM_STATE_RUNNING || dshare->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -546,10 +558,13 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 85f0ff4..6b721ec 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -132,14 +132,21 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) snd_pcm_direct_t *dsnoop = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + int err; + switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_DISCONNECTED: dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->slowptr) snd_pcm_hwsync(dsnoop->spcm); old_slave_hw_ptr = dsnoop->slave_hw_ptr; @@ -410,12 +417,16 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->state == SND_PCM_STATE_RUNNING) { err = snd_pcm_dsnoop_sync_ptr(pcm); if (err < 0) -- 2.7.4

2 1

[alsa-devel] what is exactly pcm.boundary?
by Ricardo Biehl 07 Jan '17

07 Jan '17

Hello. After some thoughts in ALSA internals (hw_ptr, appl_ptr, mmap'ed buffer, ioctls) I got me in many doubts. I would love if you help me understanding this. Well, in src/pcm/pcm_local.h line ~469: 1 static inline snd_pcm_uframes_t snd_pcm_mmap_playback_avail(snd_pcm_t *pcm) 2 { 3 snd_pcm_sframes_t avail; 4 avail = *pcm->hw.ptr + pcm->buffer_size - *pcm->appl.ptr; 5 if (avail < 0) 6 avail += pcm->boundary; 7 else if ((snd_pcm_uframes_t) avail >= pcm->boundary) 8 avail -= pcm->boundary; 9 return avail; 10 } First, I assumed that both hw_ptr and appl_ptr always points to addresses inside a REGION of memory that is exactly the size of buffer = (period_size * periods). I've also tried to imagine line 4 as: The forward distance between appl_ptr and hw_ptr where our application can write new samples: |s0|s1|s2|s3|s4|s5|s6|s7|s8|s9| buffer_size = 10 For example, if hw_ptr is on 's1' and appl_ptr is on 's7' we'll have ( 1 + 10 - 7 = 4 ) [s8,s9,s0,s1] available to write, right? hw_ptr = Last sample that sound device has eaten up in ring buffer. appl_ptr = Last sample that application has written in ring buffer. Ok, the problem comes, for example, when hw_ptr is on 's9' and appl_ptr is on 's4', so the sum is ( 9 + 10 - 4 = 15 ) -- a number greater than buffer_size. Although I expected pcm->boundary is equal buffer_size and then the test in 7th line would correct it: (9 + 10 - 4 = 15) (15 >= 10) 7th line condition test (15 - buffer_size = __5__) resulting in five [s5,s6,s7,s8,s9] samples available. But it's not. Why? And what is exactly 'boundary' ? I read this <http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg29891.html> but doesn't help me Another question ... can I thrust in hw_ptr value after a HWSYNC? will it be the exactly value of the current processed sample? And to end ... is the latency of HWSYNC operation insignificant, or it could block or something that increases latency may happen? Thank you very much! Cheers :-) -- Ricardo Biehl Pasquali

4 6

[alsa-devel] [PATCH V2 0/7] Add support for Legacy HDMI audio drivers
by Jerome Anand 07 Jan '17

07 Jan '17

Legacy (CherryTrail/ Baytrail) HDMI audio drivers added Legacy hdmi audio-Gfx interaction/ interfacing is updated to use irq chip framework Jerome Anand (7): drm/i915: setup bridge for HDMI LPE audio driver drm/i915: Add support for audio driver notifications ALSA: add shell for Intel HDMI LPE audio driver ALSA: x86: hdmi: Add audio support for BYT and CHT ALSA: x86: hdmi: Improve position reporting ALSA: x86: hdmi: Fixup some monitor ALSA: x86: hdmi: continue playback even when display resolution changes Documentation/gpu/i915.rst | 9 + drivers/gpu/drm/i915/Makefile | 3 + drivers/gpu/drm/i915/i915_drv.c | 8 +- drivers/gpu/drm/i915/i915_drv.h | 18 + drivers/gpu/drm/i915/i915_irq.c | 16 + drivers/gpu/drm/i915/i915_reg.h | 3 + drivers/gpu/drm/i915/intel_audio.c | 8 + drivers/gpu/drm/i915/intel_hdmi.c | 1 + drivers/gpu/drm/i915/intel_lpe_audio.c | 401 +++++++ include/drm/intel_lpe_audio.h | 46 + sound/Kconfig | 2 + sound/Makefile | 2 +- sound/x86/Kconfig | 16 + sound/x86/Makefile | 8 + sound/x86/intel_hdmi_audio.c | 1928 ++++++++++++++++++++++++++++++++ sound/x86/intel_hdmi_audio.h | 201 ++++ sound/x86/intel_hdmi_audio_if.c | 551 +++++++++ sound/x86/intel_hdmi_lpe_audio.c | 627 +++++++++++ sound/x86/intel_hdmi_lpe_audio.h | 685 ++++++++++++ 19 files changed, 4530 insertions(+), 3 deletions(-) create mode 100644 drivers/gpu/drm/i915/intel_lpe_audio.c create mode 100644 include/drm/intel_lpe_audio.h create mode 100644 sound/x86/Kconfig create mode 100644 sound/x86/Makefile create mode 100644 sound/x86/intel_hdmi_audio.c create mode 100644 sound/x86/intel_hdmi_audio.h create mode 100644 sound/x86/intel_hdmi_audio_if.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.h -- 2.9.3 base-commit: 362e5ebefe40f1f00a7075ad2e472ef44a36a7da

3 18

[alsa-devel] [PATCH 1/7] ASoC: wm9705: Remove unused DAI ID defines
by Lars-Peter Clausen 06 Jan '17

06 Jan '17

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. This also means the wm9705.h file no longer has any content and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm9705.c | 2 -- sound/soc/codecs/wm9705.h | 11 ----------- sound/soc/pxa/e740_wm9705.c | 1 - sound/soc/pxa/e750_wm9705.c | 1 - 4 files changed, 15 deletions(-) delete mode 100644 sound/soc/codecs/wm9705.h diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index dcdd055..266b6ec 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -20,8 +20,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include "wm9705.h" - #define WM9705_VENDOR_ID 0x574d4c05 #define WM9705_VENDOR_ID_MASK 0xffffffff diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h deleted file mode 100644 index 23ea9ce..0000000 --- a/sound/soc/codecs/wm9705.h +++ /dev/null @@ -1,11 +0,0 @@ -/* - * wm9705.h -- WM9705 Soc Audio driver - */ - -#ifndef _WM9705_H -#define _WM9705_H - -#define WM9705_DAI_AC97_HIFI 0 -#define WM9705_DAI_AC97_AUX 1 - -#endif diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 1de8765..086c37a 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -22,7 +22,6 @@ #include <asm/mach-types.h> -#include "../codecs/wm9705.h" #include "pxa2xx-ac97.h" diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index b7eb7cd..7823278 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -22,7 +22,6 @@ #include <asm/mach-types.h> -#include "../codecs/wm9705.h" #include "pxa2xx-ac97.h" static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, -- 2.1.4

4 12

[alsa-devel] [PATCH 1/2] ASoC: Intel: bytcr-rt5640: fix settings in internal clock mode
by Pierre-Louis Bossart 06 Jan '17

06 Jan '17

Frequency value of zero did not make sense, use same 24.576MHz setting and only change the clock source in idle mode Suggested-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e33e477..8d2fb2d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -142,7 +142,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, * for Jack detection and button press */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_RCCLK, - 0, + 48000 * 512, SND_SOC_CLOCK_IN); if (!ret) { if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && priv->mclk) -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: wm_adsp: Add mechanism to preload firmware on a core
by Charles Keepax 06 Jan '17

06 Jan '17

As requirements to bring up audio paths are continuous getting tighter and the DSP download to most ADSP devices happens over an external bus it can become an important factor in the path bring up time. As such sometimes it is a reasonable trade off to download the firmware ahead of when it will be required and take a small hit on power consumption for keeping the core powered up. This "preloading" adds an additional control for each DSP core "DSPx Preload Switch" that when set to true will power up the DSP core and download the firmware currently selected in the "DSPx Firmware" control. Whilst the core is preloaded the current firmware can not be changed and the CODEC will be kept powered up and SYSCLK held on. Although future improvements may allow the SYSCLK to be powered down as well because the hardware only requires SYSCLK whilst the download is actually taking place, but this is not covered in this series. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/cs47l24.c | 3 +++ sound/soc/codecs/wm5102.c | 2 ++ sound/soc/codecs/wm5110.c | 5 +++++ sound/soc/codecs/wm_adsp.c | 45 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 11 +++++++++++ 6 files changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 5670786..1822e3b 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -192,6 +192,7 @@ extern unsigned int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; #define ARIZONA_DSP_ROUTES(name) \ { name, NULL, name " Preloader"}, \ { name " Preloader", NULL, "SYSCLK" }, \ + { name " Preload", NULL, name " Preloader"}, \ { name, NULL, name " Aux 1" }, \ { name, NULL, name " Aux 2" }, \ { name, NULL, name " Aux 3" }, \ diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 73559ae..0dd721e 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -173,6 +173,9 @@ SOC_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), SOC_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), SOC_ENUM("ASRC RATE 1", arizona_asrc_rate1), +WM_ADSP2_PRELOAD_SWITCH("DSP2", 2), +WM_ADSP2_PRELOAD_SWITCH("DSP3", 3), + ARIZONA_MIXER_CONTROLS("DSP2L", ARIZONA_DSP2LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e7ab37d..9edd239 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -855,6 +855,8 @@ ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), +WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), + ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 585fc70..01b5b6c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -778,6 +778,11 @@ SOC_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), SOC_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), SOC_ENUM("ASRC RATE 1", arizona_asrc_rate1), +WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), +WM_ADSP2_PRELOAD_SWITCH("DSP2", 2), +WM_ADSP2_PRELOAD_SWITCH("DSP3", 3), +WM_ADSP2_PRELOAD_SWITCH("DSP4", 4), + ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP2L", ARIZONA_DSP2LMIX_INPUT_1_SOURCE), diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d72ccef..73dbe4c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2523,6 +2523,43 @@ static void wm_adsp2_set_dspclk(struct wm_adsp *dsp, unsigned int freq) adsp_err(dsp, "Failed to set clock rate: %d\n", ret); } +int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = dsp->preloaded; + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_preloader_get); + +int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + char preload[32]; + + snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", mc->shift); + + dsp->preloaded = ucontrol->value.integer.value[0]; + + if (ucontrol->value.integer.value[0]) + snd_soc_dapm_force_enable_pin(dapm, preload); + else + snd_soc_dapm_disable_pin(dapm, preload); + + snd_soc_dapm_sync(dapm); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); + int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event, unsigned int freq) @@ -2654,10 +2691,16 @@ EXPORT_SYMBOL_GPL(wm_adsp2_event); int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) { - dsp->codec = codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + char preload[32]; + + snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num); + snd_soc_dapm_disable_pin(dapm, preload); wm_adsp2_init_debugfs(dsp, codec); + dsp->codec = codec; + return snd_soc_add_codec_controls(codec, &wm_adsp_fw_controls[dsp->num - 1], 1); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 411d062..3706b11 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -62,6 +62,7 @@ struct wm_adsp { int fw; int fw_ver; + bool preloaded; bool booted; bool running; @@ -86,7 +87,12 @@ struct wm_adsp { SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) +#define WM_ADSP2_PRELOAD_SWITCH(wname, num) \ + SOC_SINGLE_EXT(wname " Preload Switch", SND_SOC_NOPM, num, 1, 0, \ + wm_adsp2_preloader_get, wm_adsp2_preloader_put) + #define WM_ADSP2(wname, num, event_fn) \ + SND_SOC_DAPM_SPK(wname " Preload", NULL), \ { .id = snd_soc_dapm_supply, .name = wname " Preloader", \ .reg = SND_SOC_NOPM, .shift = num, .event = event_fn, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD, \ @@ -110,6 +116,11 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream); int wm_adsp_compr_free(struct snd_compr_stream *stream); int wm_adsp_compr_set_params(struct snd_compr_stream *stream, -- 2.1.4

2 1

[alsa-devel] [PATCH v3 00/13] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 06 Jan '17

06 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Changes in v2: - Addressed Mark's comment on patch 1, split this into 2 as this had additional changes other than configuring the DMA. - Addressed Takashi's comment on the usage of snd_hdac_updatel() Changes in v3: - Addressed Takashi's comment on superfluous shit used in reading the value of PPCTL register and also cleaned the condition for updating the reg value. Jeeja KP (13): ASoC: Intel: Skylake: Update link_index and format in pipe params ASoC: Intel: Skylake: Add helper function to setup host/link dma ASoC: Intel: Skylake: Configure DMA in PRE_PMD handler of Mixer ASoC: Intel: Skylake: Removed unused skl_get_format() ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Add set_tristate DAI ops to enable SSP MCLK ASoC: Intel: Skylake: Remove unused SSP BE prepare DAI ops ASoC: Intel: Skylake: Add supply widget as non DSP widget ASoC: Intel: Skylake: Add supply widget in skl_nau_max machine ASoC: Intel: Skylake: Add supply widget in bxt_da_max machine ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/hda/ext/hdac_ext_stream.c | 15 +- sound/soc/codecs/hdac_hdmi.c | 28 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 ++++ sound/soc/intel/skylake/skl-pcm.c | 210 +++++++++++++----------- sound/soc/intel/skylake/skl-topology.c | 27 ++- sound/soc/intel/skylake/skl-topology.h | 6 + 7 files changed, 240 insertions(+), 114 deletions(-) -- 2.5.0

4 17

[alsa-devel] ubi_io_read: error -74 (ECC error)
by Anurag Raghavan (RBEI/ETW11) 06 Jan '17

06 Jan '17

Hi All, My appdata partition could not be mounted or where the partition was not able to be used. Anyone can help me to find out the root cause of this. What are the possibilities of this ubifs corruption. Any patched are available to fix this issue. I am using the kernel version-30.3.5 Any kernel patches are available to solve this issue...?? Error logs: [ 1.797141] UBI error: ubi_io_read: error -74 (ECC error) while reading 253952 bytes from PEB 445:8192, read 253952 bytes [ 1.808274] UBIFS error (pid 491): ubifs_scan: corrupt empty space at LEB 489:233760 [ 1.816037] UBIFS error (pid 491): ubifs_scanned_corruption: corruption at LEB 489:233760 [ 1.828660] UBIFS error (pid 491): ubifs_scan: LEB 489 scanning failed [ 1.835215] UBIFS warning (pid 491): ubifs_ro_mode: switched to read-only mode, error -117 [ 1.843502] UBIFS error (pid 491): make_reservation: cannot reserve 58 bytes in jhead 2, error -117 [ 1.852569] UBIFS error (pid 491): do_writepage: cannot write page 0 of inode 76584, error -117 dpkg: error: unable to sync new file '/var/lib/dpkg/arch-new': Structure needs cleaning Best regards Raghavan Anurag RBEI/ETW1 Tel. +91(422)667-4001 | Mobil 9986968950

2 2

[alsa-devel] [PATCH 6/6 v3] alsa-lib:pcm: don't return bogus buffer levels in xrun state
by sutar.mounesh＠gmail.com 06 Jan '17

06 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 2e7d9d2..1948fa0 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -882,6 +882,9 @@ static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) if ((err = snd_pcm_dmix_sync_ptr(pcm)) < 0) return err; } + if (dmix->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 09d165d..cdd1521 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -570,6 +570,9 @@ static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) return err; } + if (dshare->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 2b0df89..60a5843 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -454,6 +454,9 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_avail_update(snd_pcm_t *pcm) if (err < 0) return err; } + if (dsnoop->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_capture_avail(pcm); } -- 2.7.4

1 0

[alsa-devel] [PATCH 5/6 v3] alsa-lib:pcm: check state before enter poll on timer.
by sutar.mounesh＠gmail.com 06 Jan '17

06 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> To avoid the chances of timeout, we need to check the enter poll in state xrun. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 4e7fda8..db0381f 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -645,6 +645,46 @@ int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) return 0; } +/* + * This is the only operation guaranteed to be called before entering poll(). + * Direct plugins use fd of snd_timer to poll on, these timers do NOT check + * state of substream in kernel by intention. + * Only the enter to xrun might be notified once (SND_TIMER_EVENT_MSTOP). + * If xrun event was not correctly handled or was ignored it will never be + * evaluated again afterwards. + * This will result in snd_pcm_wait() always returning timeout. + * In contrast poll() on pcm hardware checks ALSA state and will immediately + * return POLLERR on XRUN. + * + * To prevent timeout and applications endlessly spinning without xrun + * detected we add a state check here which may trigger the xrun sequence. + * + * return count of filled descriptors or negative error code + */ +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, + unsigned int space) +{ + if (pcm->poll_fd < 0) { + SNDMSG("poll_fd < 0"); + return -EIO; + } + if (space >= 1 && pfds) { + pfds->fd = pcm->poll_fd; + pfds->events = pcm->poll_events | POLLERR | POLLNVAL; + } else { + return 0; + } + + /* this will also evaluate slave state and enter xrun if necessary */ + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } + return 1; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { snd_pcm_direct_t *dmix = pcm->private_data; diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 569d2be..deb0690 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -301,6 +301,8 @@ int snd_pcm_direct_parse_bindings(snd_pcm_direct_t *dmix, snd_config_t *cfg); int snd_pcm_direct_nonblock(snd_pcm_t *pcm, int nonblock); int snd_pcm_direct_async(snd_pcm_t *pcm, int sig, pid_t pid); +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, + unsigned int space); int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); int snd_pcm_direct_info(snd_pcm_t *pcm, snd_pcm_info_t * info); int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 258d4de..2e7d9d2 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -462,17 +462,22 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dmix->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dmix, pcm); if (dmix->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dmix->state; @@ -968,7 +973,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .avail_update = snd_pcm_dmix_avail_update, .mmap_commit = snd_pcm_dmix_mmap_commit, .htimestamp = snd_pcm_dmix_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_dmix_poll_revents, }; diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 570f180..09d165d 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -255,17 +255,22 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dshare->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dshare->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dshare, pcm); if (dshare->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dshare->state; @@ -646,7 +651,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .avail_update = snd_pcm_dshare_avail_update, .mmap_commit = snd_pcm_dshare_mmap_commit, .htimestamp = snd_pcm_dshare_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 6b721ec..2b0df89 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -208,17 +208,22 @@ static int snd_pcm_dsnoop_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dsnoop->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dsnoop->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dsnoop, pcm); return dsnoop->state; } @@ -531,7 +536,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .avail_update = snd_pcm_dsnoop_avail_update, .mmap_commit = snd_pcm_dsnoop_mmap_commit, .htimestamp = snd_pcm_dsnoop_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: pcm5102a: Provide a menu selection text
by Jörg Krause 06 Jan '17

06 Jan '17

Provide a menu selection text so that users can enable, disable or mark it as module in menuconfig. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 084c2aa..9ee9591 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -624,7 +624,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.10.2

2 6

[alsa-devel] [PATCH v2 0/6] ALSA: firewire: enclose identifiers reffered by single function
by Takashi Sakamoto 05 Jan '17

05 Jan '17

Hi, This patchset update my previous one: [alsa-devel] [PATCH 0/6] ALSA: firewire: simplify symbol space for local linkage http://mailman.alsa-project.org/pipermail/alsa-devel/2017-January/116180.ht… Changes from v1: - Rewrite commit comments with proper explanation of its aim. Takashi Sakamoto (6): ALSA: bebob: enclose identifiers referred by single function ALSA: fireworks: enclose identifiers referred by single function ALSA: oxfw: enclose identifiers referred by single function ALSA: dice: enclose identifiers referred by single function ALSA: firewire-digi00x: enclose identifiers referred by single function ALSA: firewire-tascam: enclose identifiers referred by single function sound/firewire/bebob/bebob_hwdep.c | 17 +++++----- sound/firewire/bebob/bebob_midi.c | 26 +++++++-------- sound/firewire/bebob/bebob_pcm.c | 51 ++++++++++++++--------------- sound/firewire/dice/dice-midi.c | 22 ++++++------- sound/firewire/digi00x/digi00x-hwdep.c | 17 +++++----- sound/firewire/digi00x/digi00x-midi.c | 52 ++++++++++++++---------------- sound/firewire/digi00x/digi00x-pcm.c | 52 ++++++++++++++---------------- sound/firewire/fireworks/fireworks_hwdep.c | 19 ++++++----- sound/firewire/fireworks/fireworks_midi.c | 26 +++++++-------- sound/firewire/fireworks/fireworks_pcm.c | 52 ++++++++++++++---------------- sound/firewire/oxfw/oxfw-midi.c | 26 +++++++-------- sound/firewire/tascam/tascam-hwdep.c | 17 +++++----- sound/firewire/tascam/tascam-midi.c | 26 +++++++-------- sound/firewire/tascam/tascam-pcm.c | 52 ++++++++++++++---------------- 14 files changed, 215 insertions(+), 240 deletions(-) -- 2.9.3

2 7

[alsa-devel] [PATCH 6/6 v2] alsa-lib:pcm: don't return bogus buffer levels in xrun state
by sutar.mounesh＠gmail.com 05 Jan '17

05 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index dc1b032..3e1dc0a 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -882,6 +882,9 @@ static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) if (( err = snd_pcm_dmix_sync_ptr(pcm)) < 0) return err; } + if (dmix->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 90ac6ed..8bdb4bc 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -569,6 +569,9 @@ static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) return err; } + if (dshare->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 2b0df89..60a5843 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -454,6 +454,9 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_avail_update(snd_pcm_t *pcm) if (err < 0) return err; } + if (dsnoop->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_capture_avail(pcm); } -- 2.7.4

2 1

[alsa-devel] [PATCH 5/6 v2] alsa-lib:pcm: check state before enter poll on timer.
by sutar.mounesh＠gmail.com 05 Jan '17

05 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> To avoid the chances of timeout, we need to check the enter poll in state xrun. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 0770abc..e4fbdb4 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -550,6 +550,46 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) return 0; } +/* + * This is the only operation guaranteed to be called before entering poll(). + * Direct plugins use fd of snd_timer to poll on, these timers do NOT check + * state of substream in kernel by intention. + * Only the enter to xrun might be notified once (SND_TIMER_EVENT_MSTOP). + * If xrun event was not correctly handled or was ignored it will never be + * evaluated again afterwards. + * This will result in snd_pcm_wait() always returning timeout. + * In contrast poll() on pcm hardware checks ALSA state and will immediately + * return POLLERR on XRUN. + * + * To prevent timeout and applications endlessly spinning without xrun + * detected we add a state check here which may trigger the xrun sequence. + * + * return count of filled descriptors or negative error code + */ +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, + unsigned int space) +{ + if (pcm->poll_fd < 0) { + SNDMSG("poll_fd < 0"); + return -EIO; + } + if (space >= 1 && pfds) { + pfds->fd = pcm->poll_fd; + pfds->events = pcm->poll_events | POLLERR | POLLNVAL; + } else { + return 0; + } + + /*this will also evaluate slave state and enter xrun if necessary*/ + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } + return 1; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { snd_pcm_direct_t *dmix = pcm->private_data; diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 91e816c..e729fc0 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -299,6 +299,8 @@ int snd_pcm_direct_parse_bindings(snd_pcm_direct_t *dmix, snd_config_t *cfg); int snd_pcm_direct_nonblock(snd_pcm_t *pcm, int nonblock); int snd_pcm_direct_async(snd_pcm_t *pcm, int sig, pid_t pid); +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, + unsigned int space); int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); int snd_pcm_direct_info(snd_pcm_t *pcm, snd_pcm_info_t * info); int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 0ab7323..f074744 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -456,17 +456,22 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dmix->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dmix, pcm); if (dmix->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dmix->state; @@ -953,7 +958,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .avail_update = snd_pcm_dmix_avail_update, .mmap_commit = snd_pcm_dmix_mmap_commit, .htimestamp = snd_pcm_dmix_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_dmix_poll_revents, }; diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index a1fed5d..fdd2efb 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -249,17 +249,22 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dshare->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dshare->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dshare, pcm); if (dshare->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dshare->state; @@ -631,7 +636,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .avail_update = snd_pcm_dshare_avail_update, .mmap_commit = snd_pcm_dshare_mmap_commit, .htimestamp = snd_pcm_dshare_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 85f0ff4..0c8dcbf 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -201,17 +201,22 @@ static int snd_pcm_dsnoop_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dsnoop->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dsnoop->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dsnoop, pcm); return dsnoop->state; } @@ -520,7 +525,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .avail_update = snd_pcm_dsnoop_avail_update, .mmap_commit = snd_pcm_dsnoop_mmap_commit, .htimestamp = snd_pcm_dsnoop_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; -- 2.7.4

2 1

[alsa-devel] [PATCH 2/6 v2] alsa-lib: Fix for sync issue on xrun recover
by sutar.mounesh＠gmail.com 05 Jan '17

05 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> If using very short periods, DSHARE/DSNOOP/DMIX may report underruns while in status 'prepared'. This prohibits correct recovery. Now slave xrun conditions for DSHARE/DSNOOP/DMIX are being handled properly. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 0770abc..1056308 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -550,6 +550,87 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) return 0; } +/* + * Recover slave on XRUN. + * Even if direct plugins disable xrun detection, there might be an xrun + * raised directly by some drivers. + * The first client recovers slave pcm. + * Each client needs to execute sw xrun handling afterwards + */ +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) +{ + int ret = 0; + + if (ret=snd_pcm_direct_semaphore_down(direct, DIRECT_IPC_SEM_CLIENT)) { + SNDERR("SEMDOWN FAILED with err %d", ret); + return ret; + } + + if (snd_pcm_state(direct->spcm) != SND_PCM_STATE_XRUN) { + /*ignore... someone else already did recovery*/ + if (ret = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT)) { + SNDERR("SEMUP FAILED with err %d", ret); + } + return ret; + } + + ret = snd_pcm_prepare(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to prepare slave"); + if (ret = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT)) { + SNDERR("SEMUP FAILED with err %d", ret); + } + return ret; + } + + if (direct->type == SND_PCM_TYPE_DSHARE) { + const snd_pcm_channel_area_t *dst_areas; + dst_areas = snd_pcm_mmap_areas(direct->spcm); + snd_pcm_areas_silence(dst_areas, 0, direct->spcm->channels, + direct->spcm->buffer_size, + direct->spcm->format); + } + + ret = snd_pcm_start(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to start slave"); + if (ret = snd_pcm_direct_semaphore_up(direct, + DIRECT_IPC_SEM_CLIENT)) { + SNDERR("SEMUP FAILED with err %d", ret); + } + return ret; + } + direct->shmptr->recoveries++; + if (ret = snd_pcm_direct_semaphore_up(direct, DIRECT_IPC_SEM_CLIENT)) { + SNDERR("SEMUP FAILED with err %d", ret); + } + return ret; +} + +/* + * enter xrun state, if slave xrun occured + * @return: 0 - no xrun >0: xrun happened + */ +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) +{ + if (direct->shmptr->recoveries != direct->recoveries) { + /* no matter how many xruns we missed - + so don't increment but just update to actual counter*/ + direct->recoveries = direct->shmptr->recoveries; + pcm->fast_ops->drop(pcm); + /*trigger_tstamp update is missing in drop callbacks*/ + gettimestamp(&direct->trigger_tstamp, pcm->tstamp_type); + /*no timer clear: + if slave already entered xrun again the event is lost...*/ + /*snd_pcm_direct_clear_timer_queue(direct);*/ + direct->state = SND_PCM_STATE_XRUN; + return 1; + } + return 0; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { snd_pcm_direct_t *dmix = pcm->private_data; @@ -572,6 +653,10 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in } switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + /*recover slave and update client state to xrun + before returning POLLERR*/ + snd_pcm_direct_slave_recover(dmix); + snd_pcm_direct_client_chk_xrun(dmix, pcm); case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SETUP: events |= POLLERR; @@ -1382,6 +1467,7 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm dmix->slave_buffer_size = spcm->buffer_size; dmix->slave_period_size = dmix->shmptr->s.period_size; dmix->slave_boundary = spcm->boundary; + dmix->recoveries = dmix->shmptr->recoveries; ret = snd_pcm_mmap(spcm); if (ret < 0) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 91e816c..569d2be 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -66,6 +66,7 @@ typedef struct { char socket_name[256]; /* name of communication socket */ snd_pcm_type_t type; /* PCM type (currently only hw) */ int use_server; + unsigned int recoveries; /* no of executed recoveries on slave*/ struct { unsigned int format; snd_interval_t rate; @@ -157,6 +158,7 @@ struct snd_pcm_direct { int var_periodsize; /* allow variable period size if max_periods is != -1*/ unsigned int channels; /* client's channels */ unsigned int *bindings; + unsigned int recoveries; /* mirror of executed recoveries on slave */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -318,7 +320,8 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm snd_pcm_chmap_query_t **snd_pcm_direct_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_direct_get_chmap(snd_pcm_t *pcm); int snd_pcm_direct_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); - +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct); +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm); int snd_timer_async(snd_timer_t *timer, int sig, pid_t pid); struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 0ab7323..b1c846b 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -434,15 +434,21 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_DISCONNECTED: dmix->state = SND_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (dmix->slowptr) snd_pcm_hwsync(dmix->spcm); @@ -828,12 +834,16 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -841,8 +851,10 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dmix_start_timer(pcm, dmix)) < 0) return err; } else if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if (( err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } if (dmix->state == SND_PCM_STATE_RUNNING || dmix->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -858,10 +870,13 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if (( err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index a1fed5d..ef1e6c1 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -162,7 +162,6 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - old_slave_hw_ptr = dshare->slave_hw_ptr; dshare->slave_hw_ptr = slave_hw_ptr; diff = slave_hw_ptr - old_slave_hw_ptr; @@ -202,15 +201,21 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_DISCONNECTED: dshare->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (dshare->slowptr) snd_pcm_hwsync(dshare->spcm); @@ -516,12 +521,16 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -529,8 +538,10 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dshare_start_timer(dshare)) < 0) return err; } else if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } if (dshare->state == SND_PCM_STATE_RUNNING || dshare->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -546,10 +557,13 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 85f0ff4..6b721ec 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -132,14 +132,21 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) snd_pcm_direct_t *dsnoop = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - + int err; + switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_DISCONNECTED: dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->slowptr) snd_pcm_hwsync(dsnoop->spcm); old_slave_hw_ptr = dsnoop->slave_hw_ptr; @@ -410,12 +417,16 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->state == SND_PCM_STATE_RUNNING) { err = snd_pcm_dsnoop_sync_ptr(pcm); if (err < 0) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: pxa2xx-ac97: Remove unused DAI ID defines
by Lars-Peter Clausen 05 Jan '17

05 Jan '17

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. This also means the pxa2xx-ac97.h file no longer has any content and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Acked-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Tested-by: Robert Jarzmik <robert.jarzmik(a)free.fr> (for mioa701_wm9713) --- sound/soc/pxa/e740_wm9705.c | 3 --- sound/soc/pxa/e750_wm9705.c | 2 -- sound/soc/pxa/e800_wm9712.c | 2 -- sound/soc/pxa/em-x270.c | 2 -- sound/soc/pxa/mioa701_wm9713.c | 1 - sound/soc/pxa/palm27x.c | 2 -- sound/soc/pxa/pxa2xx-ac97.c | 2 -- sound/soc/pxa/pxa2xx-ac97.h | 17 ----------------- sound/soc/pxa/tosa.c | 2 -- sound/soc/pxa/zylonite.c | 1 - 10 files changed, 34 deletions(-) delete mode 100644 sound/soc/pxa/pxa2xx-ac97.h diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 086c37a..8ab7032 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -22,9 +22,6 @@ #include <asm/mach-types.h> -#include "pxa2xx-ac97.h" - - #define E740_AUDIO_OUT 1 #define E740_AUDIO_IN 2 diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 7823278..fdcd94a 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -22,8 +22,6 @@ #include <asm/mach-types.h> -#include "pxa2xx-ac97.h" - static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 07b9c6e..2df714f 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -21,8 +21,6 @@ #include <mach/audio.h> #include <mach/eseries-gpio.h> -#include "pxa2xx-ac97.h" - static int e800_spk_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index 966163d..6f2020f 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -30,8 +30,6 @@ #include <asm/mach-types.h> #include <mach/audio.h> -#include "pxa2xx-ac97.h" - static struct snd_soc_dai_link em_x270_dai[] = { { .name = "AC97", diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 0fe0abe..8760a66 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -53,7 +53,6 @@ #include <sound/initval.h> #include <sound/ac97_codec.h> -#include "pxa2xx-ac97.h" #include "../codecs/wm9713.h" #define AC97_GPIO_PULL 0x58 diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 387492d..9716704 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -27,8 +27,6 @@ #include <mach/audio.h> #include <linux/platform_data/asoc-palm27x.h> -#include "pxa2xx-ac97.h" - static struct snd_soc_jack hs_jack; /* Headphones jack detection DAPM pins */ diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9615e6d..2e2fb18 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -27,8 +27,6 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -#include "pxa2xx-ac97.h" - static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) { pxa2xx_ac97_try_warm_reset(ac97); diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h deleted file mode 100644 index a49c21b..0000000 --- a/sound/soc/pxa/pxa2xx-ac97.h +++ /dev/null @@ -1,17 +0,0 @@ -/* - * linux/sound/soc/pxa/pxa2xx-ac97.h - * - * 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 _PXA2XX_AC97_H -#define _PXA2XX_AC97_H - -/* pxa2xx DAI ID's */ -#define PXA2XX_DAI_AC97_HIFI 0 -#define PXA2XX_DAI_AC97_AUX 1 -#define PXA2XX_DAI_AC97_MIC 2 - -#endif diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2e312c6..e022b2a 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -31,8 +31,6 @@ #include <mach/tosa.h> #include <mach/audio.h> -#include "pxa2xx-ac97.h" - #define TOSA_HP 0 #define TOSA_MIC_INT 1 #define TOSA_HEADSET 2 diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 8f301c7..6fbcdf0 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -22,7 +22,6 @@ #include <sound/soc.h> #include "../codecs/wm9713.h" -#include "pxa2xx-ac97.h" #include "pxa-ssp.h" /* -- 2.1.4

1 0

[alsa-devel] [PATCH] Revert "ALSA: firewire-lib: change structure member with proper type"
by Takashi Sakamoto 05 Jan '17

05 Jan '17

This reverts commit 6b7e95d1336b9eb0d4c6db190ce756480496bd13. This commit is based on a concern about value of the given parameter. It's expected to be ORed value with some enumeration-constants, thus often it can not be one of the enumeration-constants. I understood that this is out of specification and causes implementation-dependent issues. In C language specification, enumerated type can be interpreted as an integer type, in which all of enumeration-constants in corresponding enumerator-list can be stored. Implementations can select one of char, signed int and unsigned int as its type, and this selection is implementation-dependent. In GCC, a signed integer is selected when at least one of enumeration-constants has negative value, else an unsigned integer is selected. This behaviour can be switched by -fshort-enums to short type. Anyway, the type can be decided after scanning all of enumeration-constants. Totally, there's no rules to constrain the value of enumerated type to be one of enumeration-constants. In short, in enumerated type, decision of actual type for the type is the most important and enumeration-constants are just used for the decision, thus it's permitted to have an integer value in a range of enumeration-constants. In our case, actual type for the type is currently deterministic to be either char or unsigned int. Under GCC, it's unsigned int. --- sound/firewire/amdtp-stream.c | 2 +- sound/firewire/amdtp-stream.h | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 8ce93cd..00060c4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -69,7 +69,7 @@ static void pcm_period_tasklet(unsigned long data); * @protocol_size: the size to allocate newly for protocol */ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, - enum amdtp_stream_direction dir, int flags, + enum amdtp_stream_direction dir, enum cip_flags flags, unsigned int fmt, amdtp_stream_process_data_blocks_t process_data_blocks, unsigned int protocol_size) diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 7be2142..c1bc7fa 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -93,7 +93,7 @@ typedef unsigned int (*amdtp_stream_process_data_blocks_t)( unsigned int *syt); struct amdtp_stream { struct fw_unit *unit; - int flags; + enum cip_flags flags; enum amdtp_stream_direction direction; struct mutex mutex; @@ -137,7 +137,7 @@ struct amdtp_stream { }; int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, - enum amdtp_stream_direction dir, int flags, + enum amdtp_stream_direction dir, enum cip_flags flags, unsigned int fmt, amdtp_stream_process_data_blocks_t process_data_blocks, unsigned int protocol_size); -- 2.9.3

2 3

[alsa-devel] [PATCH v3 0/2] ALSA: Fix usb-audio races
by Ioan-Adrian Ratiu 05 Jan '17

05 Jan '17

Changes since v2: * Fixed snd_usb_*lock_shutdown imbalance caused by an early return in snd_usb_pcm_prepare() Ioan-Adrian Ratiu (2): ALSA: usb-audio: Fix irq/process data synchronization ALSA: usb-audio: test EP_FLAG_RUNNING at urb completion sound/usb/endpoint.c | 20 ++++++++++---------- sound/usb/endpoint.h | 2 +- sound/usb/pcm.c | 10 +++++----- 3 files changed, 16 insertions(+), 16 deletions(-) -- 2.11.0

2 3

Re: [alsa-devel] Stream format conversion in SoC driver
by Clemens Ladisch 05 Jan '17

05 Jan '17

Giedrius Trainavičius wrote: >It uses I2S for sending / receiving audio. However, the master clock on our >sound card is generated by the ADC, which supports only 24 bit data, so >getting other sample formats is not possible. The I2S sample format need not be the same as what the DMA controller reads/writes from/to memory. >A question on 'plug' - it accepts 'nearest' as the argument for format, >rate, what happens in case the slave pcm stream supports the exact same >format as requested by user software? The plug plugin instantiates the actual conversion plugins only as needed; with no conversion, it vanishes entirely. Regards, Clemens

2 4

[alsa-devel] [PATCH v6 0/3] Add platform clock for BayTrail platforms
by Irina Tirdea 05 Jan '17

05 Jan '17

These patches specifically enable the audio MCLK required by Baytrail CR devices. It is the remaining part of a bigger set of patches (already merged in Mark's tree) that enable sound for Baytrail CR devices (especially Asus T100TAF) [1]. They include the clock driver and enabling the clock in the pmc_atom code (along with moving of the non-architectural pmc_atom driver code into drivers/platform/x86). [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111704.html Changes from v5: - fix build error reported by kbuild test robot - split the clk driver code from x86 platform changes Changes from v4: - move the pmc_atom driver from arch/x86/platform/atom to drivers/platform/x86 Changes from v3: - replace devm_kzalloc with devm_kcalloc - add x86 architecture maintainers Changes from v2: - move clk platform data structures to a separate include file - store clk_hw pointer for the fixed rate clocks Changes from v1: - register the clk device as child of pmc device - pass iomem pointer from pmc driver to clk driver to avoid using pmc_atom_read()/write() and use readl/writel API instead - use devm_clk_hw_register/clkdev_hw_create instead of clk_register/clkdev_create Irina Tirdea (3): clk: x86: Add Atom PMC platform clocks arch/x86/platform/atom: Move pmc_atom to drivers/platform/x86 platform/x86: Enable Atom PMC platform clocks arch/x86/Kconfig | 4 - arch/x86/platform/atom/Makefile | 1 - drivers/acpi/acpi_lpss.c | 2 +- drivers/clk/x86/Makefile | 3 + drivers/clk/x86/clk-byt-plt.c | 380 +++++++++++++++++++++ drivers/platform/x86/Kconfig | 5 + drivers/platform/x86/Makefile | 1 + .../atom => drivers/platform/x86}/pmc_atom.c | 81 ++++- include/linux/platform_data/x86/clk-byt-plt.h | 31 ++ .../linux/platform_data/x86}/pmc_atom.h | 3 + 10 files changed, 500 insertions(+), 11 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c rename {arch/x86/platform/atom => drivers/platform/x86}/pmc_atom.c (88%) create mode 100644 include/linux/platform_data/x86/clk-byt-plt.h rename {arch/x86/include/asm => include/linux/platform_data/x86}/pmc_atom.h (98%) -- 1.9.1

6 29

[alsa-devel] [PATCH 0/3] ALSA: firewire: fix minor bugs
by Takashi Sakamoto 05 Jan '17

05 Jan '17

Hi, This patchset fixes some minor regressions and mis-programming. These bugs has quite less influences, thus they're not worth for stable maintenance. I'd like to apply them into kernel 4.10 or later. Takashi Sakamoto (3): ALSA: fireworks: fix asymmetric API call at unit removal ALSA: firewire-tascam: Fix to handle error from initialization of stream data ALSA: firewire-lib: change structure member with proper type sound/firewire/amdtp-stream.c | 2 +- sound/firewire/amdtp-stream.h | 4 ++-- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/tascam/tascam-stream.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) -- 2.9.3

3 6

[alsa-devel] [PATCH] ALSA: hda - Apply asus-mode8 fixup to ASUS X71SL
by Takashi Iwai 04 Jan '17

04 Jan '17

Although the old quirk table showed ASUS X71SL with ALC663 codec being compatible with asus-mode3 fixup, the bugzilla reporter explained that asus-model8 fits better for the dual headphone controls. So be it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=191781 Cc: <stable(a)vger.kernel.org> 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 d43f396450f2..7d660ee1d5e8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6984,6 +6984,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), -- 2.11.0

1 0

[alsa-devel] Better support for Focusrite saffire pro 6 usb 1
by Juho Jokelainen 04 Jan '17

04 Jan '17

After getting said device, I was bummed it didn't work well in linux (or windows for that matter). Fortunately I fould an earlier post about it. http://alsa-devel.alsa-project.narkive.com/pr5aHEeM/rfc-patch-0-2-add-suppo… For whatever reason, that patch has not made it into kernel. I can vouch that this patch (with a few line change to support kernel changes in the last 3 years) does work and my device now supports both audio input and output. Unfortunately I have no experience in kernel development nor do I really understand the alsa system. Since the code has been kept out of kernel, I suspect there is something fishy about it. If in any way possible I would still like to see code that effectively does the same thing as this patch upstreamed. The slightly modified patch (for kernel 4.8): --- sound/usb/pcm.c | 44 ++++++++++++++++++++++++++++++++++++-------- sound/usb/quirks-table.h | 24 ++++++++++++++++++++++++ 2 files changed, 60 insertions(+), 8 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 44d178e..19e319d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -483,7 +483,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; struct usb_interface *iface; - int err; + int err, need_init, i; iface = usb_ifnum_to_if(dev, fmt->iface); if (WARN_ON(!iface)) @@ -498,12 +498,26 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) /* close the old interface */ if (subs->interface >= 0 && subs->interface != fmt->iface) { - err = usb_set_interface(subs->dev, subs->interface, 0); - if (err < 0) { - dev_err(&dev->dev, - "%d:%d: return to setting 0 failed (%d)\n", - fmt->iface, fmt->altsetting, err); - return -EIO; + + need_init = iface->cur_altsetting != alts; + + if (need_init) { + err = usb_set_interface(dev, fmt->iface, fmt->altsetting); + if (err < 0) { + dev_err(&dev->dev, + "%d:%d: usb_set_interface failed (%d)\n", + fmt->iface, fmt->altsetting, err); + return -EIO; + } + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + fmt->iface, fmt->altsetting); + } + + for (i = 0; i < alts->desc.bNumEndpoints; ++i) { + if (alts->endpoint[i].desc.bEndpointAddress == fmt->endpoint) + alts->endpoint[i].enabled = 1; + else if (need_init) + alts->endpoint[i].enabled = 0; } subs->interface = -1; subs->altset_idx = 0; @@ -1241,12 +1255,26 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) { struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; + struct usb_interface *iface; + struct usb_host_interface *cur_alts; + int i, ep_active = 0; stop_endpoints(subs, true); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { - usb_set_interface(subs->dev, subs->interface, 0); + iface = usb_ifnum_to_if(subs->dev, subs->interface); + cur_alts = iface->cur_altsetting; + for (i = 0; i < cur_alts->desc.bNumEndpoints; ++i) { + if (cur_alts->endpoint[i].desc.bEndpointAddress == subs->ep_num) + cur_alts->endpoint[i].enabled = 0; + + ep_active += cur_alts->endpoint[i].enabled; + } + + if (!ep_active) + usb_set_interface(subs->dev, subs->interface, 0); + subs->interface = -1; snd_usb_unlock_shutdown(subs->stream->chip); } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c60a776..2670498 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2701,6 +2701,30 @@ YAMAHA_DEVICE(0x7010, "UB99"), .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x01, .ep_attr = USB_ENDPOINT_XFER_ISOC, + .maxpacksize = 0x024c, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .maxpacksize = 0x0126, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .rate_min = 44100, -- 2.10.2 my asoundrc for testing purposes: pcm.frontroute { type route slave.pcm "dmix:USB" ttable.0.0 1 ttable.1.1 1 } pcm.frontusb { type plug slave.pcm "frontroute" hint { show on description "USB front channels" } } pcm.rearroute { type route slave.pcm "dmix:USB" ttable.0.2 1 ttable.1.3 1 } pcm.rearusb { type plug slave.pcm "rearroute" hint { show on description "USB rear channels" } } pcm.micaroute { type route slave.pcm "dsnoop:USB" ttable.0.0 1 } pcm.mica { type plug slave.pcm "micaroute" hint { show on description "Focusrite Saffire Input 1" } } pcm.micbroute { type route slave.pcm "dsnoop:USB" ttable.0.1 1 } pcm.micb { type plug slave.pcm "micbroute" hint { show on description "Focusrite Saffire Input 2" } } pcm.!default { type plug slave { pcm "hw:1,0" } } ctl.!default { type hw card 1 } <https://gist.github.com/shiona/def87540b87466fff315c4ac805c6766>

2 2

[alsa-devel] [PATCH] ASoC: rt5640: move DAC2 Power to rt5640_dapm_widgets
by Bard Liao 04 Jan '17

04 Jan '17

"DAC L2 Power" and "DAC R2 Power" are used by both rt5639 and rt5640. But it was defined in rt5640_specific_dapm_widgets[]. Move them to rt5640_dapm_widgets will let both rt5639 and rt5640 can use it. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e29a6de..0f1b216 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1227,6 +1227,10 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, RT5640_PWR_DAC_L1_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5640_PWR_DIG1, RT5640_PWR_DAC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC L2 Power", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Power", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), @@ -1322,10 +1326,6 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC L2 Power", RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC R2 Power", RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] fsl_ssi: set fifo watermark to more reliable value
by Caleb Crome 04 Jan '17

04 Jan '17

From: Caleb Crome <caleb(a)crome.org> The fsl_ssi fifo watermark is by default set to 2 free spaces (i.e. activate DMA on FIFO when only 2 spaces are left.) This means the DMA must service the fifo within 2 audio samples, which is just not enough time for many use cases with high data rate. In many configurations the audio channel slips (causing l/r swap in stereo configurations, or channel slipping in multi-channel configurations). This patch gives more breathing room and allows the SSI to operate reliably by changing the fifio refill watermark to 8. There is no change in behavior for older chips (with an 8-deep fifo). Only the newer chips with a 15-deep fifo get the new behavior. I suspect a new fifo depth setting could be optimized on the older chips too, but I have not tested. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- sound/soc/fsl/fsl_ssi.c | 74 +++++++++++++++++++++++++++++++++++-------------- 1 file changed, 53 insertions(+), 21 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bedec4a..56245e6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -224,6 +224,12 @@ struct fsl_ssi_soc_data { * @dbg_stats: Debugging statistics * * @soc: SoC specific data + * + * @fifo_watermark: the FIFO watermark setting. Notifies DMA when + * there are @fifo_watermark or fewer words in TX fifo or + * @fifo_watermark or more empty words in RX fifo. + * @dma_maxburst: max number of words to transfer in one go. So far, + * this is always the same as fifo_watermark. */ struct fsl_ssi_private { struct regmap *regs; @@ -263,6 +269,9 @@ struct fsl_ssi_private { const struct fsl_ssi_soc_data *soc; struct device *dev; + + u32 fifo_watermark; + u32 dma_maxburst; }; /* @@ -1051,21 +1060,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, regmap_write(regs, CCSR_SSI_SRCR, srcr); regmap_write(regs, CCSR_SSI_SCR, scr); - /* - * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't - * use FIFO 1. We program the transmit water to signal a DMA transfer - * if there are only two (or fewer) elements left in the FIFO. Two - * elements equals one frame (left channel, right channel). This value, - * however, depends on the depth of the transmit buffer. - * - * We set the watermark on the same level as the DMA burstsize. For - * fiq it is probably better to use the biggest possible watermark - * size. - */ - if (ssi_private->use_dma) - wm = ssi_private->fifo_depth - 2; - else - wm = ssi_private->fifo_depth; + wm = ssi_private->fifo_watermark; regmap_write(regs, CCSR_SSI_SFCSR, CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) | @@ -1373,12 +1368,8 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, dev_dbg(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); - /* - * We have burstsize be "fifo_depth - 2" to match the SSI - * watermark setting in fsl_ssi_startup(). - */ - ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2; - ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2; + ssi_private->dma_params_tx.maxburst = ssi_private->dma_maxburst; + ssi_private->dma_params_rx.maxburst = ssi_private->dma_maxburst; ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; @@ -1543,6 +1534,47 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + /* + * Set the watermark for transmit FIFO 0 and receive FIFO 0. We don't + * use FIFO 1 but set the watermark appropriately nontheless. + * We program the transmit water to signal a DMA transfer + * if there are N elements left in the FIFO. For chips with 15-deep + * FIFOs, set watermark to 8. This allows the SSI to operate at a + * high data rate without channel slipping. Behavior is unchanged + * for the older chips with a fifo depth of only 8. A value of 4 + * might be appropriate for the older chips, but is left at + * fifo_depth-2 until sombody has a chance to test. + * + * We set the watermark on the same level as the DMA burstsize. For + * fiq it is probably better to use the biggest possible watermark + * size. + */ + switch (ssi_private->fifo_depth) { + case 15: + /* + * 2 samples is not enough when running at high data + * rates (like 48kHz @ 16 bits/channel, 16 channels) + * 8 seems to split things evenly and leave enough time + * for the DMA to fill the FIFO before it's over/under + * run. + */ + ssi_private->fifo_watermark = 8; + ssi_private->dma_maxburst = 8; + break; + case 8: + default: + /* + * maintain old behavior for older chips. + * Keeping it the same because I don't have an older + * board to test with. + * I suspect this could be changed to be something to + * leave some more space in the fifo. + */ + ssi_private->fifo_watermark = ssi_private->fifo_depth - 2; + ssi_private->dma_maxburst = ssi_private->fifo_depth - 2; + break; + } + dev_set_drvdata(&pdev->dev, ssi_private); if (ssi_private->soc->imx) { -- 2.7.4

4 5

[alsa-devel] [PATCH 1/2] ALSA: vx: Fix possible transfer overflow
by Takashi Iwai 04 Jan '17

04 Jan '17

The pseudo DMA transfer codes in VX222 and VX-pocket driver have a slight bug where they check the buffer boundary wrongly, and may overflow. Also, the zero sample count might be handled badly for the playback (although it shouldn't happen in theory). This patch addresses these issues. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=141541 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/vx/vx_pcm.c | 6 ++++-- sound/pci/vx222/vx222_ops.c | 12 ++++++------ sound/pcmcia/vx/vxp_ops.c | 12 ++++++------ 3 files changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 11467272089e..41a617886cc7 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -1048,8 +1048,10 @@ static void vx_pcm_capture_update(struct vx_core *chip, struct snd_pcm_substream /* ok, let's accelerate! */ int align = pipe->align * 3; space = (count / align) * align; - vx_pseudo_dma_read(chip, runtime, pipe, space); - count -= space; + if (space > 0) { + vx_pseudo_dma_read(chip, runtime, pipe, space); + count -= space; + } } /* read the rest of bytes */ while (count > 0) { diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index af83b3b38052..8e457ea27f89 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -269,12 +269,12 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, /* Transfer using pseudo-dma. */ - if (offset + count > pipe->buffer_bytes) { + if (offset + count >= pipe->buffer_bytes) { int length = pipe->buffer_bytes - offset; count -= length; length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ - while (length-- > 0) { + for (; length > 0; length--) { outl(cpu_to_le32(*addr), port); addr++; } @@ -284,7 +284,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, pipe->hw_ptr += count; count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ - while (count-- > 0) { + for (; count > 0; count--) { outl(cpu_to_le32(*addr), port); addr++; } @@ -307,12 +307,12 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, vx2_setup_pseudo_dma(chip, 0); /* Transfer using pseudo-dma. */ - if (offset + count > pipe->buffer_bytes) { + if (offset + count >= pipe->buffer_bytes) { int length = pipe->buffer_bytes - offset; count -= length; length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ - while (length-- > 0) + for (; length > 0; length--) *addr++ = le32_to_cpu(inl(port)); addr = (u32 *)runtime->dma_area; pipe->hw_ptr = 0; @@ -320,7 +320,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, pipe->hw_ptr += count; count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ - while (count-- > 0) + for (; count > 0; count--) *addr++ = le32_to_cpu(inl(port)); vx2_release_pseudo_dma(chip); diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 281972913c32..56aa1ba73ccc 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -369,12 +369,12 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, unsigned short *addr = (unsigned short *)(runtime->dma_area + offset); vx_setup_pseudo_dma(chip, 1); - if (offset + count > pipe->buffer_bytes) { + if (offset + count >= pipe->buffer_bytes) { int length = pipe->buffer_bytes - offset; count -= length; length >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ - while (length-- > 0) { + for (; length > 0; length--) { outw(cpu_to_le16(*addr), port); addr++; } @@ -384,7 +384,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, pipe->hw_ptr += count; count >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ - while (count-- > 0) { + for (; count > 0; count--) { outw(cpu_to_le16(*addr), port); addr++; } @@ -411,12 +411,12 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, if (snd_BUG_ON(count % 2)) return; vx_setup_pseudo_dma(chip, 0); - if (offset + count > pipe->buffer_bytes) { + if (offset + count >= pipe->buffer_bytes) { int length = pipe->buffer_bytes - offset; count -= length; length >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ - while (length-- > 0) + for (; length > 0; length--) *addr++ = le16_to_cpu(inw(port)); addr = (unsigned short *)runtime->dma_area; pipe->hw_ptr = 0; @@ -424,7 +424,7 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, pipe->hw_ptr += count; count >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ - while (count-- > 1) + for (; count > 1; count--) *addr++ = le16_to_cpu(inw(port)); /* Disable DMA */ pchip->regDIALOG &= ~VXP_DLG_DMAREAD_SEL_MASK; -- 2.11.0

1 1

[alsa-devel] [PATCH] ALSA: hda - Fix up GPIO for ASUS ROG Ranger
by Takashi Iwai 04 Jan '17

04 Jan '17

ASUS ROG Ranger VIII with ALC1150 codec requires the extra GPIO pin to up for the front panel. Just use the existing fixup for setting up the GPIO pins. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189411 Cc: <stable(a)vger.kernel.org> 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 9448daff9d8b..d43f396450f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2230,6 +2230,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), + SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), -- 2.11.0

1 0

[alsa-devel] ASoC: fsl_sai: Independent TX and RX master slave configuration
by Vishal Thanki 04 Jan '17

04 Jan '17

Hi All, I am working on a custom board based on i.MX7d SoC from NXP. The hardware uses SAI (Synchronous Audio Interface) Port 1 of i.MX7d for playback and recording use cases. The required system configuration is: SoC should act as I2S master for TX part, and slave for RX part. However, I could not figure out a clean way to instruct CPU dai driver (sound/soc/fsl/fsl_sai.c) about different roles for TX and RX of a same DAI. The .set_fmt DAI op configures the TX and RX registers in the same manner. In order to achieve the required configuration (TX master, RX slave), I added a device tree property (i.e. "fsl,sai-is-rx-slave") and based on that make necessary changes in .set_fmt DAI operation. Attached is the patch for the same. Is this the right way to go? Any suggestions would be helpful. Thanks in advance, Vishal

1 0

[alsa-devel] [PATCH] drm: sti: allow audio playback on HDMI even if disabled.
by Arnaud Pouliquen 04 Jan '17

04 Jan '17

This fix allows to play audio while HDMI is disconnected. When HDMI is disable, audio configuration is stored and samples are dropped (by HDMI IP). When HDMI is enabled, audio HDMI configuration is applied and samples are outputted on HDMI wire. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- drivers/gpu/drm/sti/sti_hdmi.c | 205 ++++++++++++++++++++--------------------- 1 file changed, 101 insertions(+), 104 deletions(-) diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c index 376b076..9c0025e 100644 --- a/drivers/gpu/drm/sti/sti_hdmi.c +++ b/drivers/gpu/drm/sti/sti_hdmi.c @@ -788,6 +788,95 @@ static void sti_hdmi_disable(struct drm_bridge *bridge) hdmi->enabled = false; } +/** + * sti_hdmi_audio_get_non_coherent_n() - get N parameter for non-coherent + * clocks. None-coherent clocks means that audio and TMDS clocks have not the + * same source (drifts between clocks). In this case assumption is that CTS is + * automatically calculated by hardware. + * + * @audio_fs: audio frame clock frequency in Hz + * + * Values computed are based on table described in HDMI specification 1.4b + * + * Returns n value. + */ +static int sti_hdmi_audio_get_non_coherent_n(unsigned int audio_fs) +{ + unsigned int n; + + switch (audio_fs) { + case 32000: + n = 4096; + break; + case 44100: + n = 6272; + break; + case 48000: + n = 6144; + break; + case 88200: + n = 6272 * 2; + break; + case 96000: + n = 6144 * 2; + break; + case 176400: + n = 6272 * 4; + break; + case 192000: + n = 6144 * 4; + break; + default: + /* Not pre-defined, recommended value: 128 * fs / 1000 */ + n = (audio_fs * 128) / 1000; + } + + return n; +} + +static int hdmi_audio_configure(struct sti_hdmi *hdmi) +{ + int audio_cfg, n; + struct hdmi_audio_params *params = &hdmi->audio; + struct hdmi_audio_infoframe *info = &params->cea; + + DRM_DEBUG_DRIVER("\n"); + + if (!hdmi->enabled) + return 0; + + /* update N parameter */ + n = sti_hdmi_audio_get_non_coherent_n(params->sample_rate); + + DRM_DEBUG_DRIVER("Audio rate = %d Hz, TMDS clock = %d Hz, n = %d\n", + params->sample_rate, hdmi->mode.clock * 1000, n); + hdmi_write(hdmi, n, HDMI_AUDN); + + /* update HDMI registers according to configuration */ + audio_cfg = HDMI_AUD_CFG_SPDIF_DIV_2 | HDMI_AUD_CFG_DTS_INVALID | + HDMI_AUD_CFG_ONE_BIT_INVALID; + + switch (info->channels) { + case 8: + audio_cfg |= HDMI_AUD_CFG_CH78_VALID; + case 6: + audio_cfg |= HDMI_AUD_CFG_CH56_VALID; + case 4: + audio_cfg |= HDMI_AUD_CFG_CH34_VALID | HDMI_AUD_CFG_8CH; + case 2: + audio_cfg |= HDMI_AUD_CFG_CH12_VALID; + break; + default: + DRM_ERROR("ERROR: Unsupported number of channels (%d)!\n", + info->channels); + return -EINVAL; + } + + hdmi_write(hdmi, audio_cfg, HDMI_AUDIO_CFG); + + return hdmi_audio_infoframe_config(hdmi); +} + static void sti_hdmi_pre_enable(struct drm_bridge *bridge) { struct sti_hdmi *hdmi = bridge->driver_private; @@ -826,9 +915,12 @@ static void sti_hdmi_pre_enable(struct drm_bridge *bridge) if (hdmi_avi_infoframe_config(hdmi)) DRM_ERROR("Unable to configure AVI infoframe\n"); - /* Program AUDIO infoframe */ - if (hdmi_audio_infoframe_config(hdmi)) - DRM_ERROR("Unable to configure AUDIO infoframe\n"); + if (hdmi->audio.enabled) { + if (hdmi_audio_configure(hdmi)) + DRM_ERROR("Unable to configure audio\n"); + } else { + hdmi_audio_infoframe_config(hdmi); + } /* Program VS infoframe */ if (hdmi_vendor_infoframe_config(hdmi)) @@ -1078,97 +1170,6 @@ static struct drm_encoder *sti_hdmi_find_encoder(struct drm_device *dev) return NULL; } -/** - * sti_hdmi_audio_get_non_coherent_n() - get N parameter for non-coherent - * clocks. None-coherent clocks means that audio and TMDS clocks have not the - * same source (drifts between clocks). In this case assumption is that CTS is - * automatically calculated by hardware. - * - * @audio_fs: audio frame clock frequency in Hz - * - * Values computed are based on table described in HDMI specification 1.4b - * - * Returns n value. - */ -static int sti_hdmi_audio_get_non_coherent_n(unsigned int audio_fs) -{ - unsigned int n; - - switch (audio_fs) { - case 32000: - n = 4096; - break; - case 44100: - n = 6272; - break; - case 48000: - n = 6144; - break; - case 88200: - n = 6272 * 2; - break; - case 96000: - n = 6144 * 2; - break; - case 176400: - n = 6272 * 4; - break; - case 192000: - n = 6144 * 4; - break; - default: - /* Not pre-defined, recommended value: 128 * fs / 1000 */ - n = (audio_fs * 128) / 1000; - } - - return n; -} - -static int hdmi_audio_configure(struct sti_hdmi *hdmi, - struct hdmi_audio_params *params) -{ - int audio_cfg, n; - struct hdmi_audio_infoframe *info = &params->cea; - - DRM_DEBUG_DRIVER("\n"); - - if (!hdmi->enabled) - return 0; - - /* update N parameter */ - n = sti_hdmi_audio_get_non_coherent_n(params->sample_rate); - - DRM_DEBUG_DRIVER("Audio rate = %d Hz, TMDS clock = %d Hz, n = %d\n", - params->sample_rate, hdmi->mode.clock * 1000, n); - hdmi_write(hdmi, n, HDMI_AUDN); - - /* update HDMI registers according to configuration */ - audio_cfg = HDMI_AUD_CFG_SPDIF_DIV_2 | HDMI_AUD_CFG_DTS_INVALID | - HDMI_AUD_CFG_ONE_BIT_INVALID; - - switch (info->channels) { - case 8: - audio_cfg |= HDMI_AUD_CFG_CH78_VALID; - case 6: - audio_cfg |= HDMI_AUD_CFG_CH56_VALID; - case 4: - audio_cfg |= HDMI_AUD_CFG_CH34_VALID | HDMI_AUD_CFG_8CH; - case 2: - audio_cfg |= HDMI_AUD_CFG_CH12_VALID; - break; - default: - DRM_ERROR("ERROR: Unsupported number of channels (%d)!\n", - info->channels); - return -EINVAL; - } - - hdmi_write(hdmi, audio_cfg, HDMI_AUDIO_CFG); - - hdmi->audio = *params; - - return hdmi_audio_infoframe_config(hdmi); -} - static void hdmi_audio_shutdown(struct device *dev, void *data) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); @@ -1192,17 +1193,9 @@ static int hdmi_audio_hw_params(struct device *dev, { struct sti_hdmi *hdmi = dev_get_drvdata(dev); int ret; - struct hdmi_audio_params audio = { - .sample_width = params->sample_width, - .sample_rate = params->sample_rate, - .cea = params->cea, - }; DRM_DEBUG_DRIVER("\n"); - if (!hdmi->enabled) - return 0; - if ((daifmt->fmt != HDMI_I2S) || daifmt->bit_clk_inv || daifmt->frame_clk_inv || daifmt->bit_clk_master || daifmt->frame_clk_master) { @@ -1213,9 +1206,13 @@ static int hdmi_audio_hw_params(struct device *dev, return -EINVAL; } - audio.enabled = true; + hdmi->audio.sample_width = params->sample_width; + hdmi->audio.sample_rate = params->sample_rate; + hdmi->audio.cea = params->cea; + + hdmi->audio.enabled = true; - ret = hdmi_audio_configure(hdmi, &audio); + ret = hdmi_audio_configure(hdmi); if (ret < 0) return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC : fsl_ssi : Correct the condition to check AC97 mode
by Harisangam, Sharvari (S.) 04 Jan '17

04 Jan '17

Corrected the condition to check if ssi is configured for AC97 mode. Other modes like dsp_a also satisfy the ANDing condition. Signed-off-by: Sharvari Harisangam <sharvari.harisangam(a)visteon.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ca6f6b1..4fbb7a2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -317,7 +317,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -1016,7 +1017,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_AC97) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 2.7.4

5 5

[alsa-devel] [PATCH - alsa-lib 1/1] conf/cards: add support for pistachio-card.
by manohar.narkhede＠imgtec.com 04 Jan '17

04 Jan '17

From: Manohar Narkhede <Manohar.Narkhede(a)imgtec.com> The data sheet of the chip and technical reference manual can be found at https://docs.creatordev.io/ci40/guides/hardwaredocs/cXT200_datasheet2.pdf and https://docs.creatordev.io/ci40/guides/hardwaredocs/MIPS_Creator_cXT200_Tec…. The additional information about the cards can be found in src/conf/cards/pistachio-card.conf file. Signed-off-by: Manohar Narkhede <Manohar.Narkhede(a)imgtec.com> diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index 008d399..60b5309 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -39,6 +39,7 @@ cfg_files = aliases.conf \ Maestro3.conf \ NFORCE.conf \ PC-Speaker.conf \ + pistachio-card.conf \ PMac.conf \ PMacToonie.conf \ PS3.conf \ diff --git a/src/conf/cards/aliases.conf b/src/conf/cards/aliases.conf index 4a92fb2..60f9d26 100644 --- a/src/conf/cards/aliases.conf +++ b/src/conf/cards/aliases.conf @@ -55,6 +55,7 @@ AV100 cards.CMI8788 AV200 cards.CMI8788 CMI8786 cards.CMI8788 CMI8787 cards.CMI8788 +pistachio cards.pistachio-card <confdir:pcm/default.conf> <confdir:pcm/dmix.conf> diff --git a/src/conf/cards/pistachio-card.conf b/src/conf/cards/pistachio-card.conf new file mode 100644 index 0000000..80e8013 --- /dev/null +++ b/src/conf/cards/pistachio-card.conf @@ -0,0 +1,58 @@ +# +# Configuration for the pistachio chip. +# +# The data sheet of the chip and technical reference manual can be found at https://docs.creatordev.io/ci40/guides/hardwaredocs/cXT200_datasheet2.pdf +# and https://docs.creatordev.io/ci40/guides/hardwaredocs/MIPS_Creator_cXT200_Tec…. +# +# The list of hardware devices is as per below: +# +# root@OpenWrt:/# arecord -l +# **** List of CAPTURE Hardware Devices **** card 0: pistachiocard [pistachio-card], device 1: pistachio-spdif-in snd-soc-dummy-dai-1 [] +# Subdevices: 1/1 +# Subdevice #0: subdevice #0 +# card 0: pistachiocard [pistachio-card], device 4: pistachio-i2s-in-0 snd-soc-dummy-dai-4 [] +# Subdevices: 1/1 +# Subdevice #0: subdevice #0 +# +# root@OpenWrt:/# aplay -l +# **** List of PLAYBACK Hardware Devices **** card 0: pistachiocard [pistachio-card], device 0: pistachio-spdif-out snd-soc-dummy-dai-0 [] +# Subdevices: 1/1 +# Subdevice #0: subdevice #0 +# card 0: pistachiocard [pistachio-card], device 2: pistachio-parallel-out pistachio_internal_dac-2 [] +# Subdevices: 1/1 +# Subdevice #0: subdevice #0 +# card 0: pistachiocard [pistachio-card], device 3: pistachio-i2s-out snd-soc-dummy-dai-3 [] +# Subdevices: 1/1 +# Subdevice #0: subdevice #0 +# + +pistachio-card.pcm.default{ + @args [ CARD ] + @args.CARD { + type string + default "pistachio" + } + @args.DEVICE { + type integer + default 2 + } + + type asym + capture.pcm { + type multi + slaves.a.pcm "hw:0,4" + slaves.a.channels 12 + bindings.0.slave a + bindings.0.channel 4 + bindings.1.slave a + bindings.1.channel 5 + } + + playback.pcm { + type hw + card $CARD + device $DEVICE + + } + +} -- 2.6.2

1 0

[alsa-devel] [PATCH 0/6] ALSA: firewire: simplify symbol space for local linkage
by Takashi Sakamoto 04 Jan '17

04 Jan '17

Hi, This patchset is just for code improvements, without adding features. Modules in ALSA firewire stack have some implementations for operation structures which sound subsystem defines. All of them are referred just by each local function to register corresponding operations. In this case, it's better to put them into the function local so that symbol space in local linkage of translation unit is kept to be simple. Takashi Sakamoto (6): ALSA: bebob: simplify symbol space for local linkage ALSA: fireworks: simplify symbol space for local linkage ALSA: oxfw: simplify symbol space for local linkage ALSA: dice: simplify symbol space for local linkage ALSA: firewire-digi00x: simplify symbol space for local linkage ALSA: firewire-tascam: simplify symbol space for local linkage sound/firewire/bebob/bebob_hwdep.c | 17 +++++----- sound/firewire/bebob/bebob_midi.c | 26 +++++++-------- sound/firewire/bebob/bebob_pcm.c | 51 ++++++++++++++--------------- sound/firewire/dice/dice-midi.c | 22 ++++++------- sound/firewire/digi00x/digi00x-hwdep.c | 17 +++++----- sound/firewire/digi00x/digi00x-midi.c | 52 ++++++++++++++---------------- sound/firewire/digi00x/digi00x-pcm.c | 52 ++++++++++++++---------------- sound/firewire/fireworks/fireworks_hwdep.c | 19 ++++++----- sound/firewire/fireworks/fireworks_midi.c | 26 +++++++-------- sound/firewire/fireworks/fireworks_pcm.c | 52 ++++++++++++++---------------- sound/firewire/oxfw/oxfw-midi.c | 26 +++++++-------- sound/firewire/tascam/tascam-hwdep.c | 17 +++++----- sound/firewire/tascam/tascam-midi.c | 26 +++++++-------- sound/firewire/tascam/tascam-pcm.c | 52 ++++++++++++++---------------- 14 files changed, 215 insertions(+), 240 deletions(-) -- 2.9.3

2 9

[alsa-devel] [PATCH v2 0/2] ALSA: Fix usb-audio races
by Ioan-Adrian Ratiu 04 Jan '17

04 Jan '17

Many thanks to Takashi Iwai & Sakamoto for their awesome feedback. Changes since v1: * Rebased my fix on top of tiwai's revert and integrated the changes from the original fix into this. * Dropped the stop_endpoints() call inside snd_usb_pcm_prepare() and kept the previously existing snd_usb_endpoint_sync_pending_stop() call. * Retained the deactivate_urbs() call in snd_usb_pcm_prepare(), I only removed the can_sleep logic. * Split the EP_FLAG_RUNNING check in a separate commit to keep the log clean since this is not part of the race fix. Ioan-Adrian Ratiu (2): ALSA: usb-audio: Fix irq/process data synchronization ALSA: usb-audio: test EP_FLAG_RUNNING at urb completion sound/usb/endpoint.c | 20 ++++++++++---------- sound/usb/endpoint.h | 2 +- sound/usb/pcm.c | 10 +++++----- 3 files changed, 16 insertions(+), 16 deletions(-) -- 2.11.0

2 7

[alsa-devel] Add ALC299 support
by Kailang 04 Jan '17

04 Jan '17

Hi Takashi, We had new codec ALC299. It was similar as ALC225. Attach was the patch. BR, Kailang

2 1

[alsa-devel] [PATCH] ASoC: intel: rt5640: add support for sound card found on HP Pavilion x2 10-p000
by Alexandrov Stanislav 04 Jan '17

04 Jan '17

Add support for rt5640 sound card found on HP Pavilion x2 10-p000 tablet. Inside DSDT table there is record for this soundcard: Device (RTKC) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "10EC3276") // _HID: Hardware ID Name (_CID, "10EC3276") // _CID: Compatible ID Name (_DDN, "ALC3276") // _DDN: DOS Device Name Name (_SUB, "103C827C") // _SUB: Subsystem ID Name (_PR0, Package (0x01) // _PR0: Power Resources for D0 original bugreport: https://bugzilla.kernel.org/show_bug.cgi?id=187621 --- diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e29a6de..c5234b9 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2315,6 +2315,7 @@ static const struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { "10EC5640", 0 }, { "10EC5642", 0 }, + { "10EC3276", 0 }, { "INTCCFFD", 0 }, { }, }; diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index f4d92bb..896ced2 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -463,6 +463,8 @@ static struct sst_acpi_mach sst_acpi_chv[] = { /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ {"10EC5640", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", cht_quirk, &chv_platform_data }, + {"10EC3276", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", NULL, + &chv_platform_data }, {}, };

2 1

[alsa-devel] [PATCH] ASoC: intel: rt5640: add support for sound card found on HP Pavilion x2 10-p000
by Alexandrov Stanislav 03 Jan '17

03 Jan '17

Add support for rt5640 sound card found on HP Pavilion x2 10-p000 tablet. original bugreport: https://bugzilla.kernel.org/show_bug.cgi?id=187621 Signed-off-by: Alexandrov Stanislav <neko(a)nya.ai> --- sound/soc/codecs/rt5640.c | 1 + sound/soc/intel/atom/sst/sst_acpi.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e29a6de..c5234b9 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2315,6 +2315,7 @@ static const struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { "10EC5640", 0 }, { "10EC5642", 0 }, + { "10EC3276", 0 }, { "INTCCFFD", 0 }, { }, }; diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index f4d92bb..896ced2 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -463,6 +463,8 @@ static struct sst_acpi_mach sst_acpi_chv[] = { /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ {"10EC5640", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", cht_quirk, &chv_platform_data }, + {"10EC3276", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", NULL, + &chv_platform_data }, {}, }; -- 2.10.2

1 0

[alsa-devel] [PATCH - seq 1/1] seq: improve documentation about new get pid/card functions
by Adam Goode 03 Jan '17

03 Jan '17

Document the technique for determining if the running kernel supports the new snd_seq_client_info_get_pid and snd_seq_client_info_get_card functions. Also add a little information about how to use these functions and add some cross references. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/src/seq/seq.c b/src/seq/seq.c index 9279830..b206e2f 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1530,7 +1530,25 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) * \param info client_info container * \return card number or -1 if value is not available. * - * Only available for SND_SEQ_KERNEL_CLIENT clients. + * Only available for #SND_SEQ_KERNEL_CLIENT clients. + * + * The card number can be used to query state about the hardware + * device providing this client, by concatenating <code>"hw:CARD="</code> + * with the card number and using it as the <code>name</code> parameter + * to #snd_ctl_open(). + * + * \note + * The return value of -1 is returned for two different conditions: when the + * running kernel does not support this operation, and when the client + * does not have a hardware card attached. See + * #snd_seq_client_info_get_pid() for a way to determine if the + * currently running kernel has support for this operation. + * + * \sa snd_seq_client_info_get_pid(), + * snd_card_get_name(), + * snd_card_get_longname(), + * snd_ctl_open(), + * snd_ctl_card_info() */ int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) { @@ -1543,7 +1561,29 @@ int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) * \param info client_info container * \return pid or -1 if value is not available. * - * Only available for SND_SEQ_USER_CLIENT clients. + * Only available for #SND_SEQ_USER_CLIENT clients. + * + * \note + * The functionality for getting a client's PID and getting a + * client's card was added to the kernel at the same time, so you can + * use this function to determine if the running kernel + * supports reporting these values. If your own client has a valid + * PID as reported by this function, then the running kernel supports + * both #snd_seq_client_info_get_card() and #snd_seq_client_info_get_pid(). + * + * \note + * Example code for determining kernel support: + * \code + * int is_get_card_or_pid_supported(snd_seq_t *seq) + * { + * snd_seq_client_info_t *my_client_info; + * snd_seq_client_info_alloca(&my_client_info); + * snd_seq_get_client_info(seq, my_client_info); + * return snd_seq_client_info_get_pid(my_client_info) != -1; + * } + * \endcode + * + * \sa snd_seq_client_info_get_card() */ int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info) { -- 2.8.0.rc3.226.g39d4020

2 1

[alsa-devel] [RFC PATCH] ucm: Add support for device positions
by mengdong.lin＠linux.intel.com 03 Jan '17

03 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Users can provide prosition info of audio devices as a device value. This will help the sound server to choose an audio devices from some candidates based on the their locations and the status of the machine. The value name is the "Postion", and its value should be a composition of "Top", "Bottom", "Left", "Right", "Front" and "Back". For example, a speaker may have a position like "Front Top". The postion value can be got by either of the two APIs: - snd_use_case_get(), via identifier 'Postion/device', to get the original position string defined by the user. - snd_use_case_geti(), via identifier '_devpos/device', to get the integer value of the position. If the user has not define a position, 0 will be returned that means an unknown position. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/use-case.h b/include/use-case.h index 8911645..b3b451b 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -165,6 +165,28 @@ extern "C" { #define SND_USE_CASE_TQ_VOICE "Voice" /**< Voice Tone Quality */ #define SND_USE_CASE_TQ_TONES "Tones" /**< Tones Tone Quality */ + +/** + * Device Position bits + * + * 0 is reserved for unknown position in each dimesion. Users can get the + * integer value of the device postion by API snd_use_case_geti() via + * identifier '_devpos/device', and the returned value is a compostion of + * three dimesions. + * + * NOTE: Users can also get the position string like "Top Left" by API + * snd_use_case_get() via identifier 'Position/device". + */ +#define SND_USE_CASE_POS_LEFT (1<<0) +#define SND_USE_CASE_POS_RIGHT (2<<0) +#define SND_USE_CASE_POS_HORIZON_MASK (0xffff<<0) +#define SND_USE_CASE_POS_TOP (1<<4) +#define SND_USE_CASE_POS_BOTTOM (2<<4) +#define SND_USE_CASE_POS_VERTICAL_MASK (0xffff<<4) +#define SND_USE_CASE_POS_FRONT (1<<8) +#define SND_USE_CASE_POS_BACK (2<<8) +#define SND_USE_CASE_DEPTH_MASK (0xffff<<8) + /** use case container */ typedef struct snd_use_case_mgr snd_use_case_mgr_t; @@ -319,6 +341,10 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * trick upper software layers to e.g. automatically mute speakers when * headphones are plugged in, but that's application policy * configuration that doesn't belong to UCM configuration files. + * - Position + * Position of the device, a composition of "Left", "Right", "Top", + * "Bottom", "Front" and "Back". For example, a device speaker may + * have a position like "Front Top". */ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, diff --git a/src/ucm/main.c b/src/ucm/main.c index 2d33886..d196a57 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -801,6 +801,26 @@ long device_status(snd_use_case_mgr_t *uc_mgr, return 0; } +/* get the device postion */ +long device_position(snd_use_case_mgr_t *uc_mgr, const char *device_name) +{ + struct use_case_device *dev; + struct list_head *pos, *base; + + if (uc_mgr->active_verb == NULL) + return -EINVAL; + + base = &uc_mgr->active_verb->device_list; + list_for_each(pos, base) { + dev = list_entry(pos, struct use_case_device, list); + if (strcmp(dev->name, device_name) == 0) + return dev->position; + } + + /* cannot find the device */ + return -1; +} + long modifier_status(snd_use_case_mgr_t *uc_mgr, const char *modifier_name) { @@ -1645,6 +1665,19 @@ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, } else if (identifier[0] == '_') err = -ENOENT; #endif + } else if (check_identifier(identifier, "_devpos")) { + if (!str) { + err = -EINVAL; + goto __end; + } + + err = device_position(uc_mgr, str); + if (err >= 0) { + *value = err; + err = 0; + } else { + err = -EINVAL; + } } else err = -ENOENT; if (str) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 4dc35c3..fc035c3 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -55,6 +55,22 @@ static const char * const component_dir[] = { NULL, /* terminator */ }; +struct position { + int val; + const char *id; +}; + +/* device positions */ +static struct position positions[] = { + { SND_USE_CASE_POS_LEFT, "Left" }, + { SND_USE_CASE_POS_RIGHT, "Right" }, + { SND_USE_CASE_POS_TOP, "Top" }, + { SND_USE_CASE_POS_BOTTOM, "Bottom" }, + { SND_USE_CASE_POS_FRONT, "Front" }, + { SND_USE_CASE_POS_BACK, "Back" }, + { 0, NULL }, /* terminator */ +}; + static int filename_filter(const struct dirent *dirent); static int is_component_directory(const char *dir); @@ -62,6 +78,30 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr, struct list_head *base, snd_config_t *cfg); +static unsigned int parse_position(struct list_head *value_list) +{ + struct list_head *pos; + struct ucm_value *val; + int i, position = 0; + + list_for_each(pos, value_list) { + val = list_entry(pos, struct ucm_value, list); + if (strncmp("Position", val->name, MAX_IDENTIFIER) == 0) { + i = 0; + while (positions[i].id) { + if (strstr(val->data, positions[i].id)) + position |= positions[i].val; + i++; + } + + return position; + } + } + + /* unknown position */ + return 0; +} + /* * Parse string */ @@ -870,6 +910,7 @@ static int parse_device(snd_use_case_mgr_t *uc_mgr, uc_error("error: failed to parse Value"); return err; } + device->position = parse_position(&device->value_list); continue; } } diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 299a5b9..2537a1d 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -42,6 +42,7 @@ #define MAX_FILE 256 #define MAX_CARD_LONG_NAME 80 +#define MAX_IDENTIFIER 32 /* Max length of identifier string */ #define ALSA_USE_CASE_DIR ALSA_CONFIG_DIR "/ucm" #define SEQUENCE_ELEMENT_TYPE_CDEV 1 @@ -139,6 +140,7 @@ struct use_case_device { char *name; char *comment; + int position; /* bits of SND_USE_CASE_POS_, 0 for unknown position */ /* device enable and disable sequences */ struct list_head enable_list; -- 2.7.4

6 6

[alsa-devel] [PATCH - alsa-lib 1/1] Change card/pid get functions to return -ENOSYS if the kernel is too old
by Adam Goode 03 Jan '17

03 Jan '17

When trying to get the pid or card of a client, the get functions will return -1 if there is no pid or card for a client. Clients have zero or one of these, so -1 is correct for these cases. But we also need to detect the case where the kernel cannot tell us if there is a card or pid, so that userspace can fallback to probing this information in the old way. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/src/seq/seq.c b/src/seq/seq.c index 4405e68..7de1e81 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,9 +1522,10 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** - * \brief Get the sound card number. + * \brief Get the sound card number if the kernel supports this. * \param info client_info container - * \return card number or -1 if value is not available. + * \return card number, -1 if there is no card for this client, + * or \c -ENOSYS if the kernel does not have support for this operation * * Only available for SND_SEQ_KERNEL_CLIENT clients. */ @@ -1535,9 +1536,10 @@ int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) } /** - * \brief Get the owning PID. + * \brief Get the owning PID if the kernel supports this. * \param info client_info container - * \return pid or -1 if value is not available. + * \return pid, -1 if there is no PID for this client, + * or \c -ENOSYS if the kernel does not have support for this operation * * Only available for SND_SEQ_USER_CLIENT clients. */ diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index 578ef12..a1d1e4a 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -102,8 +102,8 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in return -errno; } if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { - info->card = -1; - info->pid = -1; + info->card = -ENOSYS; + info->pid = -ENOSYS; } return 0; } @@ -374,8 +374,8 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i return -errno; } if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { - info->card = -1; - info->pid = -1; + info->card = -ENOSYS; + info->pid = -ENOSYS; } return 0; } -- 2.8.0.rc3.226.g39d4020

5 16

[alsa-devel] [PATCH] ALSA: dice: ensure transmission speed for transmitted packets
by Takashi Sakamoto 03 Jan '17

03 Jan '17

As of kernel 4.10, ALSA dice driver is expected to be used in default speed. In most cases, it's S400. While, IEEE 1394 specification describes the other speed such as S800. According to 'TCD30XX User Guide', its link layer controller supports several transmission speed up to S800[0]. In Dice software interface, transmission speed in output direction can be configured by asynchronous transaction to 'TX_SPEED' offset in its address space. S800 may be available. This commit improves configuration of transmission unit before starting packet streaming for this purpose. The value of 'max_speed' in 'fw_device' data structure has available maximum speed decided in bus arbitration, thus it's within capacity of the unit. [0] TCD3xx User Guide - TCAT 1394 LLC, Revision 0.9.0-41360 (TC Applied Technologies, May 6 2015) http://www.tctechnologies.tc/index.php/support/support-hardware/dice-iii-de… Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-interface.h | 1 + sound/firewire/dice/dice-stream.c | 12 +++++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index 27b044f..47f2c0a 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -251,6 +251,7 @@ /* * The speed at which the packets are sent, SCODE_100-_400; read/write. + * SCODE_800 is only available in Dice III. */ #define TX_SPEED 0x014 diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index ec4db3a..8573289 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -195,6 +195,7 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, unsigned int i, pcm_chs, midi_ports; struct amdtp_stream *streams; struct fw_iso_resources *resources; + struct fw_device *fw_dev = fw_parent_device(dice->unit); int err = 0; if (dir == AMDTP_IN_STREAM) { @@ -237,8 +238,17 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, if (err < 0) return err; + if (dir == AMDTP_IN_STREAM) { + reg[0] = cpu_to_be32(fw_dev->max_speed); + err = snd_dice_transaction_write_tx(dice, + params->size * i + TX_SPEED, + reg, sizeof(reg[0])); + if (err < 0) + return err; + } + err = amdtp_stream_start(&streams[i], resources[i].channel, - fw_parent_device(dice->unit)->max_speed); + fw_dev->max_speed); if (err < 0) return err; } -- 2.9.3

2 1

[alsa-devel] [PATCH] ALSA: oxfw: add support for Mackie Onyx 1640i
by Takashi Sakamoto 03 Jan '17

03 Jan '17

This is for Linux 4.11 development. As of kernel 4.10, ALSA OXFW driver has no entry for Onyx 1640i produced by Mackie (Loud Technologies). This commit supplement it. I note that there're two models produced by Mackie (Loud Technologies), which have the same name 'Onyx 1640i'. The former model based on OXFW970, the latter model based on Dice. This is probably due to low quality of communication of OXFW series. Additionally, the tester reports his or her experiences to get unexpected result at higher sampling transmission frequency as 88.2/96.0 kHz. We didn't have further investigation yet[0]. $ ./linux-firewire-utils/src/crpp < config_rom ROM header and bus information block ----------------------------------------------------------------- 400 042525ce bus_info_length 4, crc_length 37, crc 9678 404 31333934 bus_name "1394" 408 20ff5003 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 5 (64) 40c 000ff205 company_id 000ff2 | 410 00000fcf device_id 0500000fcf | EUI-64 000ff20500000fcf root directory ----------------------------------------------------------------- 414 0006c1b7 directory_length 6, crc 49591 418 03000ff2 vendor 41c 8100000a --> descriptor leaf at 444 420 17001640 model 424 81000011 --> descriptor leaf at 468 428 0c0083c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 00040b97 directory_length 4, crc 2967 434 1200a02d specifier id: 1394 TA 438 13010001 version: AV/C 43c 17001640 model 440 81000010 --> descriptor leaf at 480 descriptor leaf at 444 ----------------------------------------------------------------- 444 0008a886 leaf_length 8, crc 43142 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4c6f7564 "Loud" 454 20546563 " Tec" 458 686e6f6c "hnol" 45c 6f676965 "ogie" 460 7320496e "s In" 464 632e0000 "c." descriptor leaf at 468 ----------------------------------------------------------------- 468 00059fcf leaf_length 5, crc 40911 46c 00000000 textual descriptor 470 00000000 minimal ASCII 474 4f6e7978 "Onyx" 478 20313634 " 164" 47c 30690000 "0i" descriptor leaf at 480 ----------------------------------------------------------------- 480 00059fcf leaf_length 5, crc 40911 484 00000000 textual descriptor 488 00000000 minimal ASCII 48c 4f6e7978 "Onyx" 490 20313634 " 164" 494 30690000 "0i" [0]: [FFADO-user] Mackie 1640i issues (finer details) https://sourceforge.net/p/ffado/mailman/message/35229260/ Tested-by: Seth O'Bannion <saobannion(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/oxfw/oxfw.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index ab894ed..9f00696 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -34,6 +34,7 @@ config SND_OXFW * LaCie Firewire Speakers * Behringer F-Control Audio 202 * Mackie(Loud) Onyx-i series (former models) + * Mackie(Loud) Onyx 1640i (former model) * Mackie(Loud) Onyx Satellite * Mackie(Loud) Tapco Link.Firewire * Mackie(Loud) d.2 pro/d.4 pro diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index e629b88..74d7fb6 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -43,6 +43,7 @@ static bool detect_loud_models(struct fw_unit *unit) const char *const models[] = { "Onyxi", "Onyx-i", + "Onyx 1640i", "d.Pro", "Mackie Onyx Satellite", "Tapco LINK.firewire 4x6", -- 2.9.3

2 1

[alsa-devel] [PATCH v3 0/2] ucm: Automatically load the best config file based on the card long name
by mengdong.lin＠linux.intel.com 03 Jan '17

03 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Intel DSP platform drivers are used by many different devices. For user space to differentiate them, now ASoC provide API for machine drivers to include the DMI info (vendor, product and board) in card long name. This series will add UCM support for loading best device-specific configuration based on the card long name. For the ASoC kernel patch, please see ASoC: core: Add API to use DMI name in sound card long name (v2) Mengdong Lin (2): ucm: Load device-specific configuration file based on the card long name ucm: Add command 'get _file' to get the config file name of the opened card History: v2: Request the device-specific file name to match the card long name, no long use automatic key word matching and scoring to avoid error and uncertainty. And add command to check the name of actually loaded configuration file. v3: Fix num in strncpy of the long name, which should be MAX_CARD_LONG_NAME - 1, not MAX_CARD_LONG_NAME (80). Mengdong Lin (2): ucm: Load device-specific configuration file based on the card long name ucm: Add command 'get _file' to get the config file name of the opened card include/use-case.h | 1 + src/ucm/main.c | 14 ++++++++ src/ucm/parser.c | 101 +++++++++++++++++++++++++++++++++++++++++++++++++--- src/ucm/ucm_local.h | 3 ++ 4 files changed, 115 insertions(+), 4 deletions(-) -- 2.7.4

1 2

[alsa-devel] [PATCH v2 0/2] ucm: Automatically load the best config file based on the card long name
by mengdong.lin＠linux.intel.com 03 Jan '17

03 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Intel DSP platform drivers are used by many different devices. For user space to differentiate them, now ASoC provide API for machine drivers to include the DMI info (vendor, product and board) in card long name. This series will add UCM support for loading best device-specific configuration based on the card long name. For the ASoC kernel patch, please see ASoC: core: Add API to use DMI name in sound card long name (v2) Mengdong Lin (2): ucm: Load device-specific configuration file based on the card long name ucm: Add command 'get _file' to get the config file name of the opened card History: v2: Request the device-specific file name to match the card long name, no long use automatic key word matching and scoring to avoid error and uncertainty. And add command to check the name of actually loaded configuration file. include/use-case.h | 2 +- src/ucm/main.c | 14 ++++++++ src/ucm/parser.c | 101 +++++++++++++++++++++++++++++++++++++++++++++++++--- src/ucm/ucm_local.h | 3 ++ 4 files changed, 115 insertions(+), 5 deletions(-) -- 2.7.4

1 2

[alsa-devel] [PATCH v4] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Matt Ranostay 03 Jan '17

03 Jan '17

We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Basing the QoS latency calculation on the FIFO size, threshold, sample rate, and channels. Based on the original patch by Tony Lindgren Link: https://patchwork.kernel.org/patch/9305867/ Cc: Tony Lindgren <tony(a)atomide.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> --- Changes from v1: * add calculations for latency per number of FIFO locations Changes from v2: * add missing mcbsp.h header change Changes from v3: * base the latency calculations on threshold, buffer size, sample rate, and channels sound/soc/omap/mcbsp.c | 19 +++++++++++++++++++ sound/soc/omap/mcbsp.h | 3 +++ sound/soc/omap/omap-mcbsp.c | 21 ++++++++++++++++++++- 3 files changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 06fec5699cc8..8f792e714894 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -640,9 +641,19 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) */ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) { + struct pm_qos_request *pm_qos_req = &mcbsp->pm_qos_req; int enable_srg = 0; u16 w; + /* Prevent omap hardware from hitting off between fifo fills */ + if (mcbsp->latency) { + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, mcbsp->latency); + else + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + mcbsp->latency); + } + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -731,6 +742,11 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); + + if (mcbsp->active == 1 && pm_qos_request_active(&mcbsp->pm_qos_req)) { + pm_qos_remove_request(&mcbsp->pm_qos_req); + mcbsp->latency = 0; + } } int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) @@ -1098,6 +1114,9 @@ int omap_mcbsp_init(struct platform_device *pdev) void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 61e93b1c185d..d312e72f7979 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -323,8 +323,11 @@ struct omap_mcbsp { unsigned int fmt; unsigned int in_freq; + unsigned int latency; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d018e966e533..ddaec37fb4c3 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -226,6 +226,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int wlen, channels, wpf; int pkt_size = 0; unsigned int format, div, framesize, master; + unsigned int buffer_size = mcbsp->pdata->buffer_size; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); channels = params_channels(params); @@ -240,7 +241,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (mcbsp->pdata->buffer_size) { + if (buffer_size) { + int latency; + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; int divider = 0; @@ -271,6 +274,22 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Use packet mode for non mono streams */ pkt_size = channels; } + + latency = ((((buffer_size - pkt_size) / channels) * 1000) + / (params->rate_num / params->rate_den)); + + /* + * Use the lowest QoS latency value for active streams, but + * this has the disadvantage of using lower than needed latency + * if an stream with lower QoS ends before one with a higher + * one. Latency setting only gets reset when all streams + * complete. + */ + if (latency <= mcbsp->latency) + mcbsp->latency = latency; + else + mcbsp->latency = 0; + omap_mcbsp_set_threshold(substream, pkt_size); } -- 2.7.4

2 2

[alsa-devel] [PATCH][alsa-lib] topology: fix unused-const-variable warning
by Takashi Sakamoto 02 Jan '17

02 Jan '17

Last year, unused static const variable was added, then compiler generates a below warning. dapm.c:43:30: warning: ‘widget_control_map’ defined but not used [-Wunused-const-variable=] static const struct map_elem widget_control_map[] = { ^~~~~~~~~~~~~~~~~~ This commit removes it. Fixes: 01a0e1a1c219 ("topology: Add DAPM object parser") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/topology/dapm.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/src/topology/dapm.c b/src/topology/dapm.c index e830751..8c585a7 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -39,14 +39,6 @@ static const struct map_elem widget_map[] = { {"dai_link", SND_SOC_TPLG_DAPM_DAI_LINK}, }; -/* mapping of widget kcontrol text names to types */ -static const struct map_elem widget_control_map[] = { - {"volsw", SND_SOC_TPLG_DAPM_CTL_VOLSW}, - {"enum_double", SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE}, - {"enum_virt", SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT}, - {"enum_value", SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE}, -}; - static int lookup_widget(const char *w) { unsigned int i; -- 2.9.3

4 8

[alsa-devel] [PATCH][alsa-lib] topology: fix sign-compare warning introduced to set_link_hw_config() and tplg_add_link_object()
by Takashi Sakamoto 02 Jan '17

02 Jan '17

Some functions recently introduced to this library include comparison between variables with different types, then compiler generates below warnings. pcm.c: In function ‘set_link_hw_config’: pcm.c:1179:16: warning: comparison between signed and unsigned integer expressions [-Wsign-compare] for (i = 0; i < cfg->tx_channels; i++) ^ pcm.c:1183:16: warning: comparison between signed and unsigned integer expressions [-Wsign-compare] for (i = 0; i < cfg->rx_channels; i++) ^ pcm.c: In function ‘tplg_add_link_object’: pcm.c:1223:16: warning: comparison between signed and unsigned integer expressions [-Wsign-compare] for (i = 0; i < link->num_streams; i++) ^ pcm.c:1231:16: warning: comparison between signed and unsigned integer expressions [-Wsign-compare] for (i = 0; i < link->num_hw_configs; i++) This commit suppresses them. Fixes: 6b4d775b9752 ("topology: Parse HW configurations of physical DAI links defined by C API") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/topology/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 8f8a703..1de8655 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -1147,7 +1147,7 @@ int tplg_add_pcm_object(snd_tplg_t *tplg, snd_tplg_obj_template_t *t) static int set_link_hw_config(struct snd_soc_tplg_hw_config *cfg, struct snd_tplg_hw_config_template *tpl) { - int i; + unsigned int i; cfg->size = sizeof(*cfg); cfg->id = tpl->id; @@ -1190,7 +1190,7 @@ int tplg_add_link_object(snd_tplg_t *tplg, snd_tplg_obj_template_t *t) struct snd_tplg_link_template *link_tpl = t->link; struct snd_soc_tplg_link_config *link, *_link; struct tplg_elem *elem; - int i; + unsigned int i; if (t->type != SND_TPLG_TYPE_LINK && t->type != SND_TPLG_TYPE_BE && t->type != SND_TPLG_TYPE_CC) -- 2.9.3

3 3

[alsa-devel] [PATCH 0/2 v2] alsa-lib: dynamically adapt the avail_min on the slave
by Jiada Wang 02 Jan '17

02 Jan '17

When configuring avail_min to multiple of slave period size it can happen that user waits one slave period longer than needed for available data. Root cause is implicit grabbing of slave samples in avail_update operation. On next entering poll, the slave will wait for the avail_min threshold reached again, as he is not aware that there are already pending samples in the above layer which are not yet provided to user. Solution is to dynamically adapt the avail_min on the slave. Andreas Pape (2): plugin: dynamically update avail_min on slave rate: dynamic update avail_min on slave src/pcm/pcm_plugin.c | 64 +++++++++++++++++++++++++++++++++++++++++++++++++++- src/pcm/pcm_plugin.h | 3 +++ src/pcm/pcm_rate.c | 2 +- 3 files changed, 67 insertions(+), 2 deletions(-) -- 2.9.3

2 3

[alsa-devel] [PATCH 3/6] alsa-lib: pcm:dshare enable silence
by sutar.mounesh＠gmail.com 02 Jan '17

02 Jan '17

From: Alexander Jahn <ajahn(a)de.adit-jv.com> This issue depends on system load - if the process using dshare is scheduled fast enough, then there is no noise. A delay of e.g >~2ms produces hearable noise. Reproduction with instrumented aplay(sleep every 100th period for a given time): During the sleep time of 2000000us (2s) the hardware plays old samples in a loop before xrun is detected and recovered after the sleep. This is resolved by placing it in silence, in case of dshare plugin. Signed-off-by: Alexander Jahn <ajahn(a)de.adit-jv.com> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 4234d66..779845c 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1240,7 +1240,7 @@ int snd_pcm_direct_initialize_slave(snd_pcm_direct_t *dmix, snd_pcm_t *spcm, str return ret; } - if (dmix->type != SND_PCM_TYPE_DMIX) + if ((dmix->type != SND_PCM_TYPE_DMIX)&&(dmix->type != SND_PCM_TYPE_DSHARE)) goto __skip_silencing; ret = snd_pcm_sw_params_set_silence_threshold(spcm, &sw_params, 0); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 2/6] alsa-lib: Fix for sync issue on xrun recover
by sutar.mounesh＠gmail.com 02 Jan '17

02 Jan '17

From: Andreas Pape <apape(a)de.adit-jv.com> If using very short periods, DSHARE/DSNOOP/DMIX may report underruns while in status 'prepared'. This prohibits correct recovery. Now slave xrun conditions for DSHARE/DSNOOP/DMIX are being handled properly. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 8f42b19..4234d66 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -572,6 +572,9 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in } switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + /*recover slave and update client state to xrun before returning POLLERR*/ + snd_pcm_direct_slave_recover(dmix); + snd_pcm_direct_client_chk_xrun(dmix, pcm); case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SETUP: events |= POLLERR; @@ -965,6 +968,83 @@ static void save_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) #undef COPY_SLAVE +#define direct_sem_down_chk_ret(d, id) {\ + int semerr = snd_pcm_direct_semaphore_down(d, DIRECT_IPC_SEM_CLIENT); \ + if (semerr) {\ + SNDERR("SEMDOWN FAILED with err %d", semerr);\ + return semerr;\ + }\ +} + +#define direct_sem_up_chk_ret(d, id) {\ + int semerr = snd_pcm_direct_semaphore_up(d, DIRECT_IPC_SEM_CLIENT); \ + if (semerr) {\ + SNDERR("SEMUP FAILED with err %d", semerr);\ + return semerr;\ + }\ +} + +/* + * Recover slave on XRUN. + * Even if direct plugins disable xrun detection, there might be an xrun raised directly by some drivers. + * The first client recovers slave pcm. + * Each client needs to execute sw xrun handling afterwards + */ +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) +{ + int ret = 0; + + direct_sem_down_chk_ret(direct, DIRECT_IPC_SEM_CLIENT); + if (snd_pcm_state(direct->spcm) != SND_PCM_STATE_XRUN) { + /*ignore... someone else already did recovery*/ + direct_sem_up_chk_ret(direct, DIRECT_IPC_SEM_CLIENT); + return ret; + } + + ret = snd_pcm_prepare(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to prepare slave"); + direct_sem_up_chk_ret(direct, DIRECT_IPC_SEM_CLIENT); + return ret; + } + + if (direct->type == SND_PCM_TYPE_DSHARE) { + const snd_pcm_channel_area_t *dst_areas; + dst_areas = snd_pcm_mmap_areas(direct->spcm); + snd_pcm_areas_silence(dst_areas, 0, direct->spcm->channels, direct->spcm->buffer_size, direct->spcm->format); + } + + ret = snd_pcm_start(direct->spcm); + if (ret < 0) { + SNDERR("recover: unable to start slave"); + direct_sem_up_chk_ret(direct, DIRECT_IPC_SEM_CLIENT); + return ret; + } + direct->shmptr->recoveries++; + direct_sem_up_chk_ret(direct, DIRECT_IPC_SEM_CLIENT); + return 0; +} + +/* + * enter xrun state, if slave xrun occured + * @return: 0 - no xrun >0: xrun happened + */ +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) +{ + if (direct->shmptr->recoveries != direct->recoveries) { + /* no matter how many xruns we missed - so don't increment but just update to actual counter*/ + direct->recoveries = direct->shmptr->recoveries; + pcm->fast_ops->drop(pcm); + /*trigger_tstamp update is missing in drop callbacks*/ + gettimestamp(&direct->trigger_tstamp, pcm->tstamp_type); + /*no timer clear: if slave already entered xrun again the event is lost...*/ + /*snd_pcm_direct_clear_timer_queue(direct);*/ + direct->state = SND_PCM_STATE_XRUN; + return 1; + } + return 0; +} + /* * this function initializes hardware and starts playback operation with * no stop threshold (it operates all time without xrun checking) @@ -1380,6 +1460,7 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm dmix->slave_buffer_size = spcm->buffer_size; dmix->slave_period_size = dmix->shmptr->s.period_size; dmix->slave_boundary = spcm->boundary; + dmix->recoveries = dmix->shmptr->recoveries; ret = snd_pcm_mmap(spcm); if (ret < 0) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 91e816c..569d2be 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -66,6 +66,7 @@ typedef struct { char socket_name[256]; /* name of communication socket */ snd_pcm_type_t type; /* PCM type (currently only hw) */ int use_server; + unsigned int recoveries; /* no of executed recoveries on slave*/ struct { unsigned int format; snd_interval_t rate; @@ -157,6 +158,7 @@ struct snd_pcm_direct { int var_periodsize; /* allow variable period size if max_periods is != -1*/ unsigned int channels; /* client's channels */ unsigned int *bindings; + unsigned int recoveries; /* mirror of executed recoveries on slave */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -318,7 +320,8 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm snd_pcm_chmap_query_t **snd_pcm_direct_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_direct_get_chmap(snd_pcm_t *pcm); int snd_pcm_direct_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); - +int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct); +int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm); int snd_timer_async(snd_timer_t *timer, int sig, pid_t pid); struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 0ab7323..5c3eb2d 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -396,6 +396,7 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr snd_pcm_direct_t *dmix = pcm->private_data; snd_pcm_uframes_t old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; + int err; old_slave_hw_ptr = dmix->slave_hw_ptr; dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; @@ -434,15 +435,21 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_DISCONNECTED: dmix->state = SND_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (dmix->slowptr) snd_pcm_hwsync(dmix->spcm); @@ -828,12 +835,16 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dmix, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -841,8 +852,10 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dmix_start_timer(pcm, dmix)) < 0) return err; } else if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if (( err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } if (dmix->state == SND_PCM_STATE_RUNNING || dmix->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -858,10 +871,13 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; if (dmix->state == SND_PCM_STATE_RUNNING || - dmix->state == SND_PCM_STATE_DRAINING) - snd_pcm_dmix_sync_ptr(pcm); + dmix->state == SND_PCM_STATE_DRAINING) { + if (( err = snd_pcm_dmix_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index a1fed5d..707e2c2 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -202,15 +202,21 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_DISCONNECTED: dshare->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) <0) + return err; + break; default: break; } - + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (dshare->slowptr) snd_pcm_hwsync(dshare->spcm); @@ -516,12 +522,16 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dshare, pcm)) + return -EPIPE; if (! size) return 0; snd_pcm_mmap_appl_forward(pcm, size); @@ -529,8 +539,10 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, if ((err = snd_pcm_dshare_start_timer(dshare)) < 0) return err; } else if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } if (dshare->state == SND_PCM_STATE_RUNNING || dshare->state == SND_PCM_STATE_DRAINING) { /* ok, we commit the changes after the validation of area */ @@ -546,10 +558,13 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; if (dshare->state == SND_PCM_STATE_RUNNING || - dshare->state == SND_PCM_STATE_DRAINING) - snd_pcm_dshare_sync_ptr(pcm); + dshare->state == SND_PCM_STATE_DRAINING) { + if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) + return err; + } return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 85f0ff4..512e957 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -132,14 +132,21 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) snd_pcm_direct_t *dsnoop = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; + int err; switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_DISCONNECTED: dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->slowptr) snd_pcm_hwsync(dsnoop->spcm); old_slave_hw_ptr = dsnoop->slave_hw_ptr; @@ -410,12 +417,16 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_XRUN: - return -EPIPE; + if ((err = snd_pcm_direct_slave_recover(dsnoop)) <0) + return err; + break; case SND_PCM_STATE_SUSPENDED: return -ESTRPIPE; default: break; } + if (snd_pcm_direct_client_chk_xrun(dsnoop, pcm)) + return -EPIPE; if (dsnoop->state == SND_PCM_STATE_RUNNING) { err = snd_pcm_dsnoop_sync_ptr(pcm); if (err < 0) -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/6] alsa-lib:pcm: allow users to configure different period sizes.
by sutar.mounesh＠gmail.com 02 Jan '17

02 Jan '17

From: Joshua Frkuska joshua_frkuska(a)mentor.com This patch allows the effective period size to be a multiple of the slave-pcm period size. Allowing only exact multiple of original period size is achieved by borrowing code from the kernel hwrules implemention. This patch is intended to save cpu workload when for example, the slave operates with very small periods but a user does not need that small periods. This feature is enabled by default and can be disabled by adding config option 'var_periodsize 0'. Signed-off-by: Alexander Jahn <ajahn(a)de.adit-jv.com> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 6434983..8f42b19 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -660,6 +660,28 @@ static int hw_param_interval_refine_minmax(snd_pcm_hw_params_t *params, return hw_param_interval_refine_one(params, var, &t); } +/* this code is used 'as-is' from the alsa kernel code */ +static int snd_interval_step(struct snd_interval *i, unsigned int min, unsigned int step) +{ + unsigned int n; + int changed = 0; + n = (i->min - min) % step; + if (n != 0 || i->openmin) { + i->min += step - n; + changed = 1; + } + n = (i->max - min) % step; + if (n != 0 || i->openmax) { + i->max -= n; + changed = 1; + } + if (snd_interval_checkempty(i)) { + i->empty = 1; + return -EINVAL; + } + return changed; +} + #undef REFINE_DEBUG int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) @@ -710,15 +732,16 @@ int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) &dshare->shmptr->hw.rate); if (err < 0) return err; - err = hw_param_interval_refine_one(params, SND_PCM_HW_PARAM_PERIOD_SIZE, - &dshare->shmptr->hw.period_size); - if (err < 0) - return err; - err = hw_param_interval_refine_one(params, SND_PCM_HW_PARAM_PERIOD_TIME, - &dshare->shmptr->hw.period_time); - if (err < 0) - return err; + if (dshare->max_periods < 0) { + err = hw_param_interval_refine_one(params, SND_PCM_HW_PARAM_PERIOD_SIZE, + &dshare->shmptr->hw.period_size); + if (err < 0) + return err; + err = hw_param_interval_refine_one(params, SND_PCM_HW_PARAM_PERIOD_TIME, + &dshare->shmptr->hw.period_time); + if (err < 0) + return err; err = hw_param_interval_refine_one(params, SND_PCM_HW_PARAM_BUFFER_SIZE, &dshare->shmptr->hw.buffer_size); if (err < 0) @@ -730,11 +753,38 @@ int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } else if (params->rmask & ((1<<SND_PCM_HW_PARAM_PERIODS)| (1<<SND_PCM_HW_PARAM_BUFFER_BYTES)| (1<<SND_PCM_HW_PARAM_BUFFER_SIZE)| - (1<<SND_PCM_HW_PARAM_BUFFER_TIME))) { + (1<<SND_PCM_HW_PARAM_BUFFER_TIME)| + (1<<SND_PCM_HW_PARAM_PERIOD_TIME)| + (1<<SND_PCM_HW_PARAM_PERIOD_SIZE)| + (1<<SND_PCM_HW_PARAM_PERIOD_BYTES))) { + snd_interval_t period_size = dshare->shmptr->hw.period_size; + snd_interval_t period_time = dshare->shmptr->hw.period_time; int changed; unsigned int max_periods = dshare->max_periods; if (max_periods < 2) max_periods = dshare->slave_buffer_size / dshare->slave_period_size; + + /*make sure buffer size does not exceed slave buffer size*/ + err = hw_param_interval_refine_minmax(params, SND_PCM_HW_PARAM_BUFFER_SIZE, + 2 * dshare->slave_period_size, dshare->slave_buffer_size); + if (err < 0) + return err; + if (dshare->var_periodsize) { + /*more tolerant settings...*/ + if((dshare->shmptr->hw.buffer_size.max / 2) > period_size.max) + period_size.max = dshare->shmptr->hw.buffer_size.max / 2; + if((dshare->shmptr->hw.buffer_time.max / 2) > period_time.max) + period_time.max = dshare->shmptr->hw.buffer_time.max / 2; + } + + err = hw_param_interval_refine_one(params, SND_PCM_HW_PARAM_PERIOD_SIZE, + &period_size); + if (err < 0) + return err; + err = hw_param_interval_refine_one(params, SND_PCM_HW_PARAM_PERIOD_TIME, + &period_time); + if (err < 0) + return err; do { changed = 0; err = hw_param_interval_refine_minmax(params, SND_PCM_HW_PARAM_PERIODS, @@ -746,8 +796,16 @@ int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) if (err < 0) return err; changed |= err; + err = snd_interval_step(hw_param_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE), + 0, dshare->slave_period_size); + if (err < 0) + return err; + changed |= err; + if (err) + params->rmask |= (1<<SND_PCM_HW_PARAM_PERIOD_SIZE); } while (changed); } + dshare->timer_ticks = hw_param_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE)->max/dshare->slave_period_size; params->info = dshare->shmptr->s.info; #ifdef REFINE_DEBUG snd_output_puts(log, "DMIX REFINE (end):\n"); @@ -1183,6 +1241,7 @@ int snd_pcm_direct_initialize_poll_fd(snd_pcm_direct_t *dmix) dmix->tread = 1; dmix->timer_need_poll = 0; + dmix->timer_ticks = 1; ret = snd_pcm_info(dmix->spcm, &info); if (ret < 0) { SNDERR("unable to info for slave pcm"); @@ -1366,7 +1425,7 @@ int snd_pcm_direct_set_timer_params(snd_pcm_direct_t *dmix) snd_timer_params_set_auto_start(&params, 1); if (dmix->type != SND_PCM_TYPE_DSNOOP) snd_timer_params_set_early_event(&params, 1); - snd_timer_params_set_ticks(&params, 1); + snd_timer_params_set_ticks(&params, dmix->timer_ticks); if (dmix->tread) { filter = (1<<SND_TIMER_EVENT_TICK) | dmix->timer_events; @@ -1656,6 +1715,7 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, rec->ipc_gid = -1; rec->slowptr = 1; rec->max_periods = 0; + rec->var_periodsize = 1; /* read defaults */ if (snd_config_search(root, "defaults.pcm.dmix_max_periods", &n) >= 0) { @@ -1762,6 +1822,13 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, rec->max_periods = val; continue; } + if (strcmp(id, "var_periodsize") == 0) { + err = snd_config_get_bool(n); + if (err < 0) + return err; + rec->var_periodsize = err; + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 611ad29..91e816c 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -147,12 +147,14 @@ struct snd_pcm_direct { int tread: 1; int timer_need_poll: 1; unsigned int timer_events; + unsigned int timer_ticks; int server_fd; pid_t server_pid; snd_timer_t *timer; /* timer used as poll_fd */ int interleaved; /* we have interleaved buffer */ int slowptr; /* use slow but more precise ptr updates */ int max_periods; /* max periods (-1 = fixed periods, 0 = max buffer size) */ + int var_periodsize; /* allow variable period size if max_periods is != -1*/ unsigned int channels; /* client's channels */ unsigned int *bindings; union { @@ -326,6 +328,7 @@ struct snd_pcm_direct_open_conf { int ipc_gid; int slowptr; int max_periods; + int var_periodsize; snd_config_t *slave; snd_config_t *bindings; }; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 825677f..0ab7323 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -1040,6 +1040,7 @@ int snd_pcm_dmix_open(snd_pcm_t **pcmp, const char *name, dmix->state = SND_PCM_STATE_OPEN; dmix->slowptr = opts->slowptr; dmix->max_periods = opts->max_periods; + dmix->var_periodsize = opts->var_periodsize; dmix->sync_ptr = snd_pcm_dmix_sync_ptr; retry: diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 29cd6c6..a1fed5d 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -725,6 +725,7 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, dshare->state = SND_PCM_STATE_OPEN; dshare->slowptr = opts->slowptr; dshare->max_periods = opts->max_periods; + dshare->var_periodsize = opts->var_periodsize; dshare->sync_ptr = snd_pcm_dshare_sync_ptr; retry: diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 1aedf3c..85f0ff4 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -606,6 +606,7 @@ int snd_pcm_dsnoop_open(snd_pcm_t **pcmp, const char *name, dsnoop->state = SND_PCM_STATE_OPEN; dsnoop->slowptr = opts->slowptr; dsnoop->max_periods = opts->max_periods; + dsnoop->var_periodsize = opts->var_periodsize; dsnoop->sync_ptr = snd_pcm_dsnoop_sync_ptr; retry: -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/3] ASoC: tlv320aic3x: Pop noise reductions
by Peter Ujfalusi 02 Jan '17

02 Jan '17

Hi, When we don't have any active component between the output of the codec and the speaker/jack the codec emmits pop noise due to several factors: - RESET register is toggled by regcache_sync (patch 1) - After regcache_sync we need to wait a bit to allow the codec to settle down: the codec will be soft reseted when it is powered down, so when we power it on and run regcache_sync, we need to wait for the flushed settings to be valid. (patch 2) - The HP ouptut power-down mode can be changed to be driven weakly to VCM, this can reduce the pop noise as well, with slight increase in powere consumption. These factors were not taken care as in n810 and n900 we have external active amplifiers and they will remove any artifacts coming from the codec, but on boards w/o active external components they can reach the output. Regards, Peter --- Peter Ujfalusi (3): ASoC: tlv320aic3x: Mark the RESET register as volatile ASoC: tlv320aic3x: Add delay after power on and register sync ASoC: tlv320aic3x: Add controls for selecting HP power down modes sound/soc/codecs/tlv320aic3x.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) -- 2.11.0

3 9

[alsa-devel] snd_hda_intel/snd_hda_codec_hdmi module load/unload race
by Imre Deak 02 Jan '17

02 Jan '17

Hi, I got the trace below while trying to unload (unbind) snd_hda_intel, while its still loading the HDMI codec driver. IIUC what happens is: Task1 Task2 Task3 modprobe snd_hda_intel schedule(azx_probe_work) unbind snd_hda_intel via sysfs device_release_driver() device_lock(snd_hda_intel) azx_remove() cancel_work_sync(azx_probe_work) azx_probe_work() request_module(snd-hda-codec-hdmi) hdmi_driver_init() __driver_attach() device_lock(snd_hda_intel) Deadlock, since azx_probe_work() will never finish and the snd_hda_intel device lock will never get released. --Imre [ 246.805268] INFO: task drv_module_relo:1684 blocked for more than 120 seconds. [ 246.805834] Tainted: G W 4.9.0-CI-CI_DRM_1946+ #158 [ 246.806284] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 246.806779] drv_module_relo D 0 1684 1597 0x00000000 [ 246.807171] ffff8802703cbac0 ffff8802643926c0 000000000000479d ffff8802713b0040 [ 246.807706] ffff88027fc98118 ffffc90003cffa58 ffffffff8181d429 0000000000000046 [ 246.808217] ffffffff827d2470 0000000000000001 ffff88027fc98118 ffff88027fc98100 [ 246.808739] Call Trace: [ 246.808945] [<ffffffff8181d429>] ? __schedule+0x2c9/0xa30 [ 246.809281] [<ffffffff8181dbcb>] schedule+0x3b/0x90 [ 246.809618] [<ffffffff8182227b>] schedule_timeout+0x25b/0x4c0 [ 246.810021] [<ffffffff810d6c9f>] ? mark_held_locks+0x6f/0xa0 [ 246.810370] [<ffffffff81823987>] ? _raw_spin_unlock_irq+0x27/0x50 [ 246.810759] [<ffffffff810d6df2>] ? trace_hardirqs_on_caller+0x122/0x1b0 [ 246.811963] [<ffffffff8181e730>] wait_for_common+0xf0/0x160 [ 246.812463] [<ffffffff810af5d0>] ? wake_up_q+0x70/0x70 [ 246.812795] [<ffffffff8181e7b8>] wait_for_completion+0x18/0x20 [ 246.813389] [<ffffffff8109b9e3>] flush_work+0x203/0x2e0 [ 246.813714] [<ffffffff8109ba86>] ? flush_work+0x2a6/0x2e0 [ 246.814088] [<ffffffff81099410>] ? flush_workqueue_prep_pwqs+0x220/0x220 [ 246.814790] [<ffffffff8109d290>] __cancel_work_timer+0x110/0x1d0 [ 246.815346] [<ffffffff810d6df2>] ? trace_hardirqs_on_caller+0x122/0x1b0 [ 246.815742] [<ffffffff810d6e8d>] ? trace_hardirqs_on+0xd/0x10 [ 246.816096] [<ffffffff8109d35b>] cancel_work_sync+0xb/0x10 [ 246.816566] [<ffffffffa00d4fa4>] azx_remove+0x24/0x30 [snd_hda_intel] [ 246.816956] [<ffffffff81485864>] pci_device_remove+0x34/0xb0 [ 246.817373] [<ffffffff81584cfc>] __device_release_driver+0x9c/0x150 [ 246.817717] [<ffffffff81584dd0>] device_release_driver+0x20/0x30 [ 246.818077] [<ffffffff815836aa>] unbind_store+0x10a/0x150 [ 246.818549] [<ffffffff81582d70>] drv_attr_store+0x20/0x30 [ 246.818885] [<ffffffff812775a0>] sysfs_kf_write+0x40/0x50 [ 246.819290] [<ffffffff812768e0>] kernfs_fop_write+0x130/0x1b0 [ 246.819639] [<ffffffff811f6383>] __vfs_write+0x23/0x120 [ 246.819963] [<ffffffff810f48c0>] ? rcu_read_lock_sched_held+0x70/0x80 [ 246.820476] [<ffffffff810f4caa>] ? rcu_sync_lockdep_assert+0x2a/0x50 [ 246.820859] [<ffffffff811f9a68>] ? __sb_start_write+0xf8/0x200 [ 246.821292] [<ffffffff811f7129>] ? vfs_write+0x169/0x1b0 [ 246.821597] [<ffffffff811f7073>] vfs_write+0xb3/0x1b0 [ 246.821907] [<ffffffff810d6df2>] ? trace_hardirqs_on_caller+0x122/0x1b0 [ 246.822405] [<ffffffff811f83b4>] SyS_write+0x44/0xb0 [ 246.822684] [<ffffffff8182412e>] entry_SYSCALL_64_fastpath+0x1c/0xb1 [ 246.823064] Showing all locks held in the system: [ 246.823568] 2 locks held by kworker/u8:0/6: [ 246.823817] #0: ("events_unbound"){.+.+.+}, at: [<ffffffff8109c1f6>] process_one_work+0x166/0x6b0 [ 246.824400] #1: ((&sub_info->work)){+.+.+.}, at: [<ffffffff8109c1f6>] process_one_work+0x166/0x6b0 [ 246.824940] 2 locks held by khungtaskd/34: [ 246.825210] #0: (rcu_read_lock){......}, at: [<ffffffff811329cc>] watchdog+0x9c/0x600 [ 246.825726] #1: (tasklist_lock){.+.+..}, at: [<ffffffff810d536d>] debug_show_all_locks+0x3d/0x1a0 [ 246.826303] 2 locks held by kworker/3:1/50: [ 246.826535] #0: ("events"){++++.+}, at: [<ffffffff8109c1f6>] process_one_work+0x166/0x6b0 [ 246.827104] #1: ((&hda->probe_work)){+.+.+.}, at: [<ffffffff8109c1f6>] process_one_work+0x166/0x6b0 [ 246.827706] 5 locks held by drv_module_relo/1684: [ 246.827963] #0: (sb_writers#6){.+.+.+}, at: [<ffffffff811f7129>] vfs_write+0x169/0x1b0 [ 246.828507] #1: (&of->mutex){+.+.+.}, at: [<ffffffff812768ac>] kernfs_fop_write+0xfc/0x1b0 [ 246.829083] #2: (s_active#186){++++.+}, at: [<ffffffff812768b4>] kernfs_fop_write+0x104/0x1b0 [ 246.829663] #3: (&dev->mutex){......}, at: [<ffffffff8158369a>] unbind_store+0xfa/0x150 [ 246.830204] #4: (&dev->mutex){......}, at: [<ffffffff81584dc8>] device_release_driver+0x18/0x30 [ 246.830772] 1 lock held by modprobe/1685: [ 246.830992] #0: (&dev->mutex){......}, at: [<ffffffff8158561a>] __driver_attach+0x5a/0xe0 [ 246.831575] 2 locks held by bash/1727: [ 246.831782] #0: (&tty->ldisc_sem){++++.+}, at: [<ffffffff81822d3d>] ldsem_down_read+0x2d/0x40 [ 246.832344] #1: (&ldata->atomic_read_lock){+.+...}, at: [<ffffffff81506529>] n_tty_read+0xa9/0x880 [ 246.833050] ============================================= [ 246.833436] INFO: task modprobe:1685 blocked for more than 120 seconds. [ 246.833829] Tainted: G W 4.9.0-CI-CI_DRM_1946+ #158 [ 246.834221] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 246.834680] modprobe D 0 1685 6 0x00000000 [ 246.835029] ffff8802703c9a40 ffff8802760e4d40 00000000000091ba ffff8802559e26c0 [ 246.835522] ffff88027fd18118 ffffc900032bbb70 ffffffff8181d429 0000000000000000 [ 246.835967] ffffc900032bbb98 0000000000000292 ffff88027fd18118 ffff88027fd18100 [ 246.836462] Call Trace: [ 246.836628] [<ffffffff8181d429>] ? __schedule+0x2c9/0xa30 [ 246.836924] [<ffffffff8181dbcb>] schedule+0x3b/0x90 [ 246.837243] [<ffffffff8181e043>] schedule_preempt_disabled+0x13/0x20 [ 246.837629] [<ffffffff8181ecbc>] mutex_lock_nested+0x16c/0x400 [ 246.837946] [<ffffffff8158561a>] ? __driver_attach+0x5a/0xe0 [ 246.838298] [<ffffffff815855c0>] ? driver_probe_device+0x430/0x430 [ 246.838676] [<ffffffff8158561a>] __driver_attach+0x5a/0xe0 [ 246.839008] [<ffffffff815855c0>] ? driver_probe_device+0x430/0x430 [ 246.839354] [<ffffffff81582fc1>] bus_for_each_dev+0x61/0xa0 [ 246.839704] [<ffffffff81584b89>] driver_attach+0x19/0x20 [ 246.840029] [<ffffffff815846cb>] bus_add_driver+0x1fb/0x270 [ 246.840346] [<ffffffffa014a000>] ? 0xffffffffa014a000 [ 246.840645] [<ffffffff8158629b>] driver_register+0x5b/0xd0 [ 246.840977] [<ffffffffa048f0d5>] __hda_codec_driver_register+0x55/0x60 [snd_hda_codec] [ 246.841417] [<ffffffffa014a01e>] hdmi_driver_init+0x1e/0x20 [snd_hda_codec_hdmi] [ 246.841861] [<ffffffff81000428>] do_one_initcall+0x38/0x150 [ 246.842177] [<ffffffff810f48c0>] ? rcu_read_lock_sched_held+0x70/0x80 [ 246.842566] [<ffffffff811d7804>] ? kmem_cache_alloc_trace+0x274/0x2e0 [ 246.842951] [<ffffffff81176161>] ? do_init_module+0x22/0x1d1 [ 246.843307] [<ffffffff81176199>] do_init_module+0x5a/0x1d1 [ 246.843617] [<ffffffff8111cb15>] load_module+0x2275/0x27d0 [ 246.843951] [<ffffffff81119e60>] ? symbol_put_addr+0x60/0x60 [ 246.844306] [<ffffffff811fd7d5>] ? kernel_read_file+0x105/0x190 [ 246.844637] [<ffffffff8111d29c>] SyS_finit_module+0xbc/0xf0 [ 246.844979] [<ffffffff8182412e>] entry_SYSCALL_64_fastpath+0x1c/0xb1 [ 246.845368] Showing all locks held in the system: [ 246.845745] 2 locks held by kworker/u8:0/6: [ 246.845974] #0: ("events_unbound"){.+.+.+}, at: [<ffffffff8109c1f6>] process_one_work+0x166/0x6b0 [ 246.846553] #1: ((&sub_info->work)){+.+.+.}, at: [<ffffffff8109c1f6>] process_one_work+0x166/0x6b0 [ 246.847133] 2 locks held by khungtaskd/34: [ 246.847399] #0: (rcu_read_lock){......}, at: [<ffffffff811329cc>] watchdog+0x9c/0x600 [ 246.847913] #1: (tasklist_lock){.+.+..}, at: [<ffffffff810d536d>] debug_show_all_locks+0x3d/0x1a0 [ 246.848488] 2 locks held by kworker/3:1/50: [ 246.848747] #0: ("events"){++++.+}, at: [<ffffffff8109c1f6>] process_one_work+0x166/0x6b0 [ 246.849286] #1: ((&hda->probe_work)){+.+.+.}, at: [<ffffffff8109c1f6>] process_one_work+0x166/0x6b0 [ 246.849875] 1 lock held by in:imklog/491: [ 246.850094] #0: (&f->f_pos_lock){+.+.+.}, at: [<ffffffff8121854e>] __fdget_pos+0x3e/0x50 [ 246.850657] 5 locks held by drv_module_relo/1684: [ 246.850931] #0: (sb_writers#6){.+.+.+}, at: [<ffffffff811f7129>] vfs_write+0x169/0x1b0 [ 246.851474] #1: (&of->mutex){+.+.+.}, at: [<ffffffff812768ac>] kernfs_fop_write+0xfc/0x1b0 [ 246.852013] #2: (s_active#186){++++.+}, at: [<ffffffff812768b4>] kernfs_fop_write+0x104/0x1b0 [ 246.852577] #3: (&dev->mutex){......}, at: [<ffffffff8158369a>] unbind_store+0xfa/0x150 [ 246.853103] #4: (&dev->mutex){......}, at: [<ffffffff81584dc8>] device_release_driver+0x18/0x30 [ 246.853714] 1 lock held by modprobe/1685: [ 246.853934] #0: (&dev->mutex){......}, at: [<ffffffff8158561a>] __driver_attach+0x5a/0xe0 [ 246.854473] 2 locks held by bash/1727: [ 246.854713] #0: (&tty->ldisc_sem){++++.+}, at: [<ffffffff81822d3d>] ldsem_down_read+0x2d/0x40 [ 246.855272] #1: (&ldata->atomic_read_lock){+.+...}, at: [<ffffffff81506529>] n_tty_read+0xa9/0x880 Stack trace of kworker/50: [<ffffffff810979e3>] call_usermodehelper_exec+0x133/0x1d0 [<ffffffff81097c0f>] __request_module+0x18f/0x360 [<ffffffffa048f182>] request_codec_module+0x52/0x60 [snd_hda_codec] [<ffffffffa048f21d>] snd_hda_codec_configure+0x5d/0x170 [snd_hda_codec] [<ffffffffa0499693>] azx_codec_configure+0x23/0x40 [snd_hda_codec] [<ffffffffa00d6700>] azx_probe_work+0x5d0/0x8d0 [snd_hda_intel] [<ffffffff8109c27c>] process_one_work+0x1ec/0x6b0 [<ffffffff8109c789>] worker_thread+0x49/0x490 [<ffffffff810a2aab>] kthread+0xeb/0x110 [<ffffffff818243a7>] ret_from_fork+0x27/0x40 [<ffffffffffffffff>] 0xffffffffffffffff

3 3

[alsa-devel] [PATCH 0/5] Runtime PM support for Exynos I2S driver
by Marek Szyprowski 31 Dec '16

31 Dec '16

Exynos I2S audio controller was missing proper runtime PM implementation and this patchset addresses this issue. This patchset is a another step to add support for audio power domain on Exynos5 SoCs. Audio power domain on Exynos5 SoCs contains following hardware modules: 1. clock controller 2. pin controller 3. PL330 DMA controller 4. I2S audio controller The other patches related to enabling full support for audio power domain can be found here: 1. PL330 ADMA controller non-irqsafe runtime PM: https://www.spinics.net/lists/arm-kernel/msg550008.html 2. Runtime PM for clock controllers (Exynos Audio subsystem will be added in v4 soon): https://www.spinics.net/lists/arm-kernel/msg538122.html 3. Runtime PM support for pin controller: https://www.spinics.net/lists/linux-samsung-soc/msg56674.html Patches are based on linux-next from 2016.12.24. Tested on Odroid U3 (Exynos4412), Odroid XU3 (Exynos5422) and TM2 (Exynos5433) boards. Best regards Marek Szyprowski Samsung R&D Institute Poland Patch summary: Marek Szyprowski (5): ASoC: samsung: i2s: Remove virtual device for secondary DAI ASoC: samsung: i2s: Ensure proper runtime PM state of I2S device ASoC: samsung: i2s: Move saving and restoring regs to runtime pm operations ASoC: samsung: i2s: Let runtime PM operations to control op_clk too ASoC: samsung: i2s: Provide I2S device for registered clocks sound/soc/samsung/i2s.c | 193 +++++++++++++++++++++++------------------------- 1 file changed, 92 insertions(+), 101 deletions(-) -- 1.9.1

2 11

[alsa-devel] [PATCH 0/2] ASoC: sun4i-spdif: Changes in preparation of supporting new SoCs
by codekipper＠gmail.com 31 Dec '16

31 Dec '16

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached two patches. First being a cleanup of something that shouldn't be in the code and the second adds the mechanism for supporting newer Allwiner SoCs. There was a third patch in the series and that was bringing in support of SPDIF on the Allwinner H3 but I've not been able to test it so that I hear audio. As soon as I get some hardware with a SPDIF connection I will confirm and push. BR, CK Marcus Cooper (2): ASoC: sun4i-spdif: remove legacy dapm components ASoC: sun4i-spdif: Add quirks to the spdif driver sound/soc/sunxi/sun4i-spdif.c | 36 +++++++++++++++++++++++++++--------- 1 file changed, 27 insertions(+), 9 deletions(-) -- 2.11.0

3 6

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix to fail safely if module not available in path
by Subhransu S. Prusty 31 Dec '16

31 Dec '16

From: G Kranthi <gudishax.kranthikumar(a)intel.com> If a module is not available in a pipeline, fail safely rather than causing oops. Signed-off-by: G Kranthi <gudishax.kranthikumar(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 10fa10d..7e97ae6 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -254,6 +254,9 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, snd_pcm_set_sync(substream); mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); + if (!mconfig) + return -EINVAL; + skl_tplg_d0i3_get(skl, mconfig->d0i3_caps); return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: nau8825: fix invalid configuration in Pre-Scalar of FLL
by John Hsu 31 Dec '16

31 Dec '16

The clk_ref_div is not configured in the correct position of the register. The patch fixes that clk_ref_div, Pre-Scalar, is assigned the wrong value. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 3 ++- sound/soc/codecs/nau8825.h | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f69f738..e4fb017 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2078,7 +2078,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_INTEGER_MASK, fll_param->fll_int); /* FLL pre-scaler */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4, - NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); + NAU8825_FLL_REF_DIV_MASK, + fll_param->clk_ref_div << NAU8825_FLL_REF_DIV_SFT); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index fa571ec..5732ce1 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -137,7 +137,8 @@ #define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ -#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) +#define NAU8825_FLL_REF_DIV_SFT 10 +#define NAU8825_FLL_REF_DIV_MASK (0x3 << NAU8825_FLL_REF_DIV_SFT) /* FLL5 (0x08) */ #define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) -- 2.6.4

2 1

Re: [alsa-devel] [PATCH] ALSA: snd-usb: fix IRQ triggered NULL pointer dereference
by Takashi Iwai 30 Dec '16

30 Dec '16

On Wed, 21 Dec 2016 11:38:54 +0100, Ioan-Adrian Ratiu wrote: > > >> > Please take the time to fully analyze my patch and let's have a > >> > discussion based on it, not reject it outright and replace it with > >> > a quick and dirty delay hack. > >> > >> OK. I'll deliberately check it again so that I have no overlooks. I > >> with this thread also catch the other developers enough helpful to > >> you. (I just eventually caught your patch in LKML and not developer > >> for this category of devices.) > > > > Sorry for the late reply, as I've been (still) off and had bad net > > connections. > > > > About your fix: Sakamoto-san is right, it's no good way to fix this > > kind or problem. The easiest option right now is just to revert my > > previous fix, as it obviously introduces another regression. The > > correct fix will be taken after that. > > > > I'm going to prepare a revert patch and ask Linus to take it before > > rc1. > > I agree with reverting the initial commit decision because my problem > disappears with it. > > But can you please state a reason for why my patch is "no good way to > fix"? Being too intrusive is not a good reason. "Being too intrusive" is the exact reason why it's not good as a "regression fix" like this case. The logic you've implemented in the patch itself looks good (although the code introduces a bug, the unbalance of snd_usb_*lock_shutdown()). The only point I couldn't take it is that it's rather a fundamental change, not a quick fix for a regression. What's the worst case scenario in a regression fix? It's when a fix introduces yet another regression. It'd be much worse if the new regression is deeper. The complicated logical change has a potential risk of such. Thus an intrusive change is not always good as a "regression fix". In general, if the change were trivial, it's obviously OK to take as a regression fix -- where the logic is also trivial in most cases, too. But when the fix becomes complex, we need to rethink. Especially when the original buggy commit is small, reverting it is often better as a clear cut. Think in that way: you're addressing a deeper problem that was revealed accidentally by my previous bad fix. Writing the change as if it were merely a regression fix gives the wrong understanding to readers :) That said, I'd appreciate if you respin the fix again with a combination of my previous fix and brush up the code a bit as a whole, and document it not as a regression fix but as a complete fix of the existing races. I can write it in my side quickly, but it'd be almost in the same form as you posted, I suppose. thanks, Takashi

2 1

[alsa-devel] [PATCH] alsactl: Check existence of `alsactl` in udev rule
by Paul Menzel 30 Dec '16

30 Dec '16

From: Jordi Mallach <jordi(a)debian.org> Date: Thu, 10 Jan 2013 00:17:58 +0000 Include the line `TEST=="/usr/sbin/alsactl"` in the udev rule, to properly fix the state restoring for users with split `/usr` filesystems [1]. Upstream the patch from the Debian package [2]. [1] https://bugs.debian.org/670490 "Debian Bug report logs - #670490 alsa-utils: Restore sound volume in udev" [2] https://sources.debian.net/src/alsa-utils/1.1.2-1/debian/patches/udev_test_… Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> CC: Jordi Mallach <jordi(a)debian.org> --- Please apply with `git am saved-messages.mbox`. alsactl/90-alsa-restore.rules.in | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in index c0c1b23..f190b85 100644 --- a/alsactl/90-alsa-restore.rules.in +++ b/alsactl/90-alsa-restore.rules.in @@ -2,7 +2,7 @@ ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", TEST== GOTO="alsa_restore_end" LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{device/number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{device/number}" +TEST!="@daemonswitch@", TEST=="@sbindir@/alsactl", RUN+="@sbindir@/alsactl restore $attr{device/number}" +TEST=="@daemonswitch@", TEST=="@sbindir@/alsactl", RUN+="@sbindir@/alsactl nrestore $attr{device/number}" LABEL="alsa_restore_end" -- 2.11.0

2 3

[alsa-devel] [PATCH 6/6] alsa-lib:pcm: don't return bogus buffer levels in xrun state
by sutar.mounesh＠gmail.com 30 Dec '16

30 Dec '16

From: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 320c948..127b06f 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -884,6 +884,9 @@ static snd_pcm_sframes_t snd_pcm_dmix_avail_update(snd_pcm_t *pcm) if (( err = snd_pcm_dmix_sync_ptr(pcm)) < 0) return err; } + if (dmix->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index f4fa32b..5b897f5 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -570,6 +570,9 @@ static snd_pcm_sframes_t snd_pcm_dshare_avail_update(snd_pcm_t *pcm) if ((err = snd_pcm_dshare_sync_ptr(pcm)) < 0) return err; } + if (dshare->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_playback_avail(pcm); } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 0729a5a..61579e8 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -454,6 +454,9 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_avail_update(snd_pcm_t *pcm) if (err < 0) return err; } + if (dsnoop->state == SND_PCM_STATE_XRUN) + return -EPIPE; + return snd_pcm_mmap_capture_avail(pcm); } -- 1.7.9.5

1 0

[alsa-devel] [PATCH 5/6] alsa-lib:pcm: check state before enter poll on timer.
by sutar.mounesh＠gmail.com 30 Dec '16

30 Dec '16

From: Andreas Pape <apape(a)de.adit-jv.com> To avoid the chances of timeout, we need to check the enter poll in state xrun. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 779845c..2928512 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -549,6 +549,42 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) snd_timer_stop(dmix->timer); return 0; } +/* + * This is the only operation guaranteed to be called before entering poll(). + * Direct plugins use fd of snd_timer to poll on, these timers do NOT check state of substream in kernel by intention. + * Only the enter to xrun might be notified once (SND_TIMER_EVENT_MSTOP). + * If xrun event was not correctly handled or was ignored it will never be evaluated again afterwards. + * This will result in snd_pcm_wait() always returning timeout. + * In contrast poll() on pcm hardware checks ALSA state and will immediately return POLLERR on XRUN. + * + * To prevent timeout and applications endlessly spinning without xrun detected we add a state check + * here which may trigger the xrun sequence. + * + * return count of filled descriptors or negative error code + */ +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space) +{ + if (pcm->poll_fd < 0) { + SNDMSG("poll_fd < 0"); + return -EIO; + } + if (space >= 1 && pfds) { + pfds->fd = pcm->poll_fd; + pfds->events = pcm->poll_events | POLLERR | POLLNVAL; + } else { + return 0; + } + + /*this will also evaluate slave state and enter xrun if necessary*/ + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } + return 1; +} + int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 569d2be..3a1c1c0 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -301,6 +301,7 @@ int snd_pcm_direct_parse_bindings(snd_pcm_direct_t *dmix, snd_config_t *cfg); int snd_pcm_direct_nonblock(snd_pcm_t *pcm, int nonblock); int snd_pcm_direct_async(snd_pcm_t *pcm, int sig, pid_t pid); +int snd_pcm_direct_poll_descriptors(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space); int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); int snd_pcm_direct_info(snd_pcm_t *pcm, snd_pcm_info_t * info); int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 5c3eb2d..320c948 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -463,17 +463,23 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dmix->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dmix, pcm); + if (dmix->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dmix->state; @@ -969,7 +975,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .avail_update = snd_pcm_dmix_avail_update, .mmap_commit = snd_pcm_dmix_mmap_commit, .htimestamp = snd_pcm_dmix_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_dmix_poll_revents, }; diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 707e2c2..f4fa32b 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -255,17 +255,22 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dshare->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dshare->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dshare, pcm); if (dshare->state == STATE_RUN_PENDING) return SNDRV_PCM_STATE_RUNNING; return dshare->state; @@ -646,7 +651,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .avail_update = snd_pcm_dshare_avail_update, .mmap_commit = snd_pcm_dshare_mmap_commit, .htimestamp = snd_pcm_dshare_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 512e957..0729a5a 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -208,17 +208,22 @@ static int snd_pcm_dsnoop_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; + int err; snd_pcm_state_t state; state = snd_pcm_state(dsnoop->spcm); switch (state) { - case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dsnoop->state = state; return state; + case SND_PCM_STATE_XRUN: + if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) + return err; + break; default: break; } + snd_pcm_direct_client_chk_xrun(dsnoop, pcm); return dsnoop->state; } @@ -531,7 +536,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .avail_update = snd_pcm_dsnoop_avail_update, .mmap_commit = snd_pcm_dsnoop_mmap_commit, .htimestamp = snd_pcm_dsnoop_htimestamp, - .poll_descriptors = NULL, + .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, }; -- 1.7.9.5

1 0

[alsa-devel] [PATCH] Revert "ASoC: core: replace aux_comp_list to component_dev_list"
by Nikita Yushchenko 29 Dec '16

29 Dec '16

This reverts commit 1a653aa44725668590b36bbe2d7fe4736a69f055 ("ASoC: core: replace aux_comp_list to component_dev_list"). That commit tries to remove card->aux_comp_list, using flagged entries in card->component_dev_list instead. However, components are added to card->component_dev_list in soc_probe_component(). Before that commit, it traversed card->aux_comp_list and found aux devices added by soc_bind_aux_dev(). After that commit, it traverses card->component_dev_list and finds no aux devices, because soc_probe_component() was not yet called for them. Thus all aux devices are lost and any setup that needs them no longer works. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 17 +++++------------ 2 files changed, 8 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6cc6d0..8172a512632a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -809,10 +809,10 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; - unsigned int auxiliary:1; /* for auxiliary component of the card */ unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1152,6 +1152,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1547,6 +1548,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb1466e..15657a65e4ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1747,8 +1747,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) } component->init = aux_dev->init; - component->auxiliary = 1; - + list_add(&component->list_aux, &card->aux_comp_list); return 0; err_defer: @@ -1764,10 +1763,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1791,14 +1787,11 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { list_for_each_entry_safe(comp, _comp, - &card->component_dev_list, card_list) { - - if (!comp->auxiliary) - continue; - + &card->aux_comp_list, list_aux) { if (comp->driver->remove_order == order) { soc_remove_component(comp); - comp->auxiliary = 0; + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); } } } -- 2.1.4

1 0

[alsa-devel] [PATCH] ASoC: Fix binding and probing of auxiliary components
by Sylwester Nawrocki 29 Dec '16

29 Dec '16

Currently binding of auxiliary devices doesn't work as in soc_bind_aux_dev() function a bound component is not being added to any list and in soc_probe_aux_devices() we are trying to walk the component_dev_list list to probe auxiliary components but at that time this list doesn't contain any auxiliary components since they are being added to the card only in soc_probe_component(). This patch adds a list to the card where are stored bound but not probed auxiliary devices, so that all aux devices can be probed. Fixes: 1a653aa44725 "ASoC: core: replace aux_comp_list to component_dev_list" Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- I'm not sure if this is a correct approach, without this patch the amplifier component doesn't get bound to the card on the TM2 board. --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 10 +++++----- 2 files changed, 8 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6..b86168a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -813,6 +813,7 @@ struct snd_soc_component { unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; + struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1152,6 +1153,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1547,6 +1549,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb..baa1afa 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1748,6 +1748,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) component->init = aux_dev->init; component->auxiliary = 1; + list_add(&component->card_aux_list, &card->aux_comp_list); return 0; @@ -1758,16 +1759,14 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_component *comp; + struct snd_soc_component *comp, *tmp; int order; int ret; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry_safe(comp, tmp, &card->aux_comp_list, + card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1776,6 +1775,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) comp->name, ret); return ret; } + list_del(&comp->card_aux_list); } } } -- 1.9.1

1 0

[alsa-devel] Stream format conversion in SoC driver
by Giedrius Trainavičius 28 Dec '16

28 Dec '16

Hi, We are developing a soundcard for Raspberry Pi, you may find the current driver source code here: https://github.com/raspberrypi/linux/blob/rpi-4.4.y/sound/soc/bcm/pisound.c The card natively supports 24 and 32 bit PCM streams at 48kHz, 96kHz and 192kHz, symmetric capture and playback. Is it possible to have ALSA do the necessary format conversions for the PCM streams in natively unsupported formats to the closest natively supported format? What would be required to enable it? Thank you, Giedrius.

2 5

Re: [alsa-devel] [PATCH] ASoC : fsl_ssi : Correct the condition to check AC97 mode
by Andreas Schwab 28 Dec '16

28 Dec '16

On Dez 28 2016, Timur Tabi <timur(a)tabi.org> wrote: > Andreas Schwab wrote: >>> > + return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97) == >>> > + SND_SOC_DAIFMT_AC97; >> This is never true. > > I think the right parenthesis should be at the end of the expression. That would still test the wrong bitfield. And all that is obsolete anyway since commit 5b64c173cd (of Sep 2015). Andreas. -- Andreas Schwab, schwab(a)linux-m68k.org GPG Key fingerprint = 58CA 54C7 6D53 942B 1756 01D3 44D5 214B 8276 4ED5 "And now for something completely different."

1 0

[alsa-devel] [PATCH] alsactl: Remove standard output definition in systemd unit
by Paul Menzel 28 Dec '16

28 Dec '16

Date: Tue, 8 Jul 2014 07:23:06 +0000 `/lib/systemd/system/alsa-restore.service` specifies `StandardOutput=syslog`. This overrides the `DefaultStandardOutput` setting from `/etc/systemd/system.conf`, which the system administrator can use to specify how output gets logged. In particular, the sysadmin may want output to go to the journal, or to syslog, or nowhere at all [1]. This patch removes the definition entirely, so the units can use the system default. Upstream the patch from the Debian package [2]. [1] https://bugs.debian.org/741123 "systemd services should not use StandardOutput=syslog; should rely on DefaultStandardOutput" [2] https://sources.debian.net/src/alsa-utils/1.1.2-1/debian/patches/systemd_st… Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> CC: Jordi Mallach <jordi(a)debian.org> --- alsactl/alsa-restore.service.in | 1 - 1 file changed, 1 deletion(-) diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index f3cd86b..c092c2f 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -15,4 +15,3 @@ Type=oneshot RemainAfterExit=true ExecStart=-@sbindir@/alsactl restore ExecStop=-@sbindir@/alsactl store -StandardOutput=syslog -- 2.11.0

2 1

[alsa-devel] [PATCH] alsactl: Check for state file existence in systemd unit
by Paul Menzel 28 Dec '16

28 Dec '16

Date: Sat, 19 Oct 2013 10:12:19 +0000 Check for the existence of the state file, by default `/var/lib/alsa/asound.state`, before doing anything in the systemd service unit `alsa-restore.service`. If the state file is not present, `alsactl` will fail with exit code 99. Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> CC: Jordi Mallach <jordi(a)debian.org> --- Please apply with `git am saved-messages.mbox`. alsactl/Makefile.am | 1 + alsactl/alsa-restore.service.in | 1 + 2 files changed, 2 insertions(+) diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index cac8094..b691dfa 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -40,6 +40,7 @@ endif edit = \ $(SED) -r -e 's,@sbindir\@,$(sbindir),g' \ -e 's,@mydatadir\@,$(mydatadir),g' \ + -e 's,@asoundstatedir\@,$(ASOUND_STATE_DIR),g' \ -e 's,@daemonswitch\@,$(ALSACTL_DAEMONSWITCH),g' \ < $< > $@ || rm $@ diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index 2884098..f3cd86b 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -7,6 +7,7 @@ Description=Save/Restore Sound Card State ConditionPathExists=!@daemonswitch@ ConditionPathExistsGlob=/dev/snd/control* +ConditionPathExists=@asoundstatedir@/asound.state After=alsa-state.service [Service] -- 2.11.0

2 1

[alsa-devel] [PATCH] alsactl: Make homedir configurable
by Paul Menzel 28 Dec '16

28 Dec '16

Date: Sat, 19 Oct 2013 10:12:19 +0000 Add new configure switch to set alsactl's homedir. This is used to avoid that a directory `.pulse` is created by the init script, when PulseAudio is installed [1]. [1] https://bugs.debian.org/712980 "alsa-utils startup script creates /.pulse files ("Failed to create secure directory") when pulseaudio is installed" CC: Jordi Mallach <jordi(a)debian.org> Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> --- Please apply with `git am saved-messages.mbox`. alsactl/90-alsa-restore.rules.in | 4 ++-- alsactl/Makefile.am | 1 + alsactl/alsa-restore.service.in | 4 ++-- alsactl/alsa-state.service.in | 4 ++-- configure.ac | 6 ++++++ 5 files changed, 13 insertions(+), 6 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in index c0c1b23..95b26c4 100644 --- a/alsactl/90-alsa-restore.rules.in +++ b/alsactl/90-alsa-restore.rules.in @@ -2,7 +2,7 @@ ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", TEST== GOTO="alsa_restore_end" LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{device/number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{device/number}" +TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl -E HOME=@alsactlhome@ restore $attr{device/number}" +TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl -E HOME=@alsactlhome@ nrestore $attr{device/number}" LABEL="alsa_restore_end" diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index cac8094..d03137b 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -40,6 +40,7 @@ endif edit = \ $(SED) -r -e 's,@sbindir\@,$(sbindir),g' \ -e 's,@mydatadir\@,$(mydatadir),g' \ + -e 's,@alsactlhome\@,$(ALSACTL_HOME_DIR),g' \ -e 's,@daemonswitch\@,$(ALSACTL_DAEMONSWITCH),g' \ < $< > $@ || rm $@ diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index 2884098..23dece8 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -12,6 +12,6 @@ After=alsa-state.service [Service] Type=oneshot RemainAfterExit=true -ExecStart=-@sbindir@/alsactl restore -ExecStop=-@sbindir@/alsactl store +ExecStart=-@sbindir@/alsactl -E HOME=@alsactlhome@ restore +ExecStop=-@sbindir@/alsactl -E HOME=@alsactlhome@ store StandardOutput=syslog diff --git a/alsactl/alsa-state.service.in b/alsactl/alsa-state.service.in index f1321d6..28e3043 100644 --- a/alsactl/alsa-state.service.in +++ b/alsactl/alsa-state.service.in @@ -10,5 +10,5 @@ After=sysinit.target [Service] Type=simple -ExecStart=-@sbindir@/alsactl -s -n 19 -c rdaemon -ExecStop=-@sbindir@/alsactl -s kill save_and_quit +ExecStart=-@sbindir@/alsactl -E HOME=@alsactlhome@ -s -n 19 -c rdaemon +ExecStop=-@sbindir@/alsactl -E HOME=@alsactlhome@ -s kill save_and_quit diff --git a/configure.ac b/configure.ac index 69ff2ab..8cbf798 100644 --- a/configure.ac +++ b/configure.ac @@ -409,6 +409,12 @@ AC_ARG_WITH([alsactl-pidfile-dir], [ALSACTL_PIDFILE_DIR="/var/run"]) AC_SUBST(ALSACTL_PIDFILE_DIR) +AC_ARG_WITH([alsactl-home-dir], + AS_HELP_STRING([--with-alsactl-home-dir=DIR], [Directory to use as homedir for alsactl]), + [ALSACTL_HOME_DIR="$withval"], + [ALSACTL_HOME_DIR="/var/run/alsa"]) +AC_SUBST(ALSACTL_HOME_DIR) + AC_ARG_WITH([alsactl-daemonswitch], AS_HELP_STRING([--with-alsactl-daemonswitch=FILE], [File to test for the daemon mode]), [ALSACTL_DAEMONSWITCH="$withval"], -- 2.11.0

2 1

[alsa-devel] [PATCH] alsactl: Fix spelling mistakes
by Paul Menzel 28 Dec '16

28 Dec '16

Date: Sun, 8 May 2016 16:50:23 +0000 Upstream the patch from the Debian package [1]. [1] https://sources.debian.net/src/alsa-utils/1.1.2-1/debian/patches/spellfixes… CC: Elimar Riesebieter <riesebie(a)lxtec.de> Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> --- The renaming from `ressize` to `resize` might not be correct, if it’s the size of the variable *res*. Please tell me, if I should remove that. alsactl/alsactl.c | 2 +- alsactl/init_parse.c | 10 +++++----- alsactl/init_utils_run.c | 14 +++++++------- alsactl/state.c | 2 +- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/alsactl/alsactl.c b/alsactl/alsactl.c index bcbc187..f0060e9 100644 --- a/alsactl/alsactl.c +++ b/alsactl/alsactl.c @@ -90,7 +90,7 @@ static struct arg args[] = { { FILEARG | 'e', "pid-file", "pathname for the process id (daemon mode)" }, { HEADER, NULL, "Available init options:" }, { ENVARG | 'E', "env", "set environment variable for init phase (NAME=VALUE)" }, -{ FILEARG | 'i', "initfile", "main configuation file for init phase" }, +{ FILEARG | 'i', "initfile", "main configuration file for init phase" }, { 0, NULL, " (default " DATADIR "/init/00main)" }, { 'b', "background", "run daemon in background" }, { 's', "syslog", "use syslog for messages" }, diff --git a/alsactl/init_parse.c b/alsactl/init_parse.c index e81174f..f2a598c 100644 --- a/alsactl/init_parse.c +++ b/alsactl/init_parse.c @@ -939,7 +939,7 @@ static int get_key(char **line, char **key, enum key_op *op, char **value) } /* extract possible KEY{attr} */ -static char *get_key_attribute(struct space *space, char *str, char *res, size_t ressize) +static char *get_key_attribute(struct space *space, char *str, char *res, size_t resize) { char *pos; char *attr; @@ -953,7 +953,7 @@ static char *get_key_attribute(struct space *space, char *str, char *res, size_t return NULL; } pos[0] = '\0'; - strlcpy(res, attr, ressize); + strlcpy(res, attr, resize); pos[0] = '}'; dbg("attribute='%s'", res); return res; @@ -1209,7 +1209,7 @@ found: /* possibly truncate to format-char specified length */ if (len != -1) { head[len] = '\0'; - dbg("truncate to %i chars, subtitution string becomes '%s'", len, head); + dbg("truncate to %i chars, substitution string becomes '%s'", len, head); } strlcat(string, temp, maxsize); } @@ -1242,7 +1242,7 @@ found: static int run_program1(struct space *space, const char *command0, char *result, - size_t ressize, size_t *reslen, int log) + size_t resize, size_t *reslen, int log) { if (strncmp(command0, "__ctl_search", 12) == 0) { const char *res = elemid_get(space, "do_search"); @@ -1254,7 +1254,7 @@ int run_program1(struct space *space, const char *res = elemid_get(space, "do_count"); if (res == NULL || strcmp(res, "0") == 0) return EXIT_FAILURE; - strlcpy(result, res, ressize); + strlcpy(result, res, resize); return EXIT_SUCCESS; } Perror(space, "unknown buildin command '%s'", command0); diff --git a/alsactl/init_utils_run.c b/alsactl/init_utils_run.c index dde490b..ffa0e23 100644 --- a/alsactl/init_utils_run.c +++ b/alsactl/init_utils_run.c @@ -24,12 +24,12 @@ static int run_program1(struct space *space, const char *command0, char *result, - size_t ressize, size_t *reslen, int log); + size_t resize, size_t *reslen, int log); static int run_program0(struct space *space, const char *command0, char *result, - size_t ressize, size_t *reslen, int log) + size_t resize, size_t *reslen, int log) { int retval = 0; int status; @@ -175,11 +175,11 @@ int run_program0(struct space *space, /* store result for rule processing */ if (result) { - if (respos + count < ressize) { + if (respos + count < resize) { memcpy(&result[respos], inbuf, count); respos += count; } else { - Perror(space, "ressize %ld too short", (long)ressize); + Perror(space, "resize %ld too short", (long)resize); retval = -1; } } @@ -239,9 +239,9 @@ int run_program0(struct space *space, static int run_program(struct space *space, const char *command0, char *result, - size_t ressize, size_t *reslen, int log) + size_t resize, size_t *reslen, int log) { if (command0[0] == '_' && command0[1] == '_') - return run_program1(space, command0, result, ressize, reslen, log); - return run_program0(space, command0, result, ressize, reslen, log); + return run_program1(space, command0, result, resize, reslen, log); + return run_program0(space, command0, result, resize, reslen, log); } diff --git a/alsactl/state.c b/alsactl/state.c index 3908ec4..d2d6dac 100644 --- a/alsactl/state.c +++ b/alsactl/state.c @@ -1160,7 +1160,7 @@ static int restore_config_value(snd_ctl_t *handle, snd_ctl_elem_info_t *info, case SND_CTL_ELEM_TYPE_IEC958: break; default: - cerror(doit, "Unknow control type: %d", type); + cerror(doit, "Unknown control type: %d", type); return -EINVAL; } return 0; -- 2.11.0

3 2

[alsa-devel] [PATCH] ucm: parser needs limits.h
by Gustavo Zacarias 28 Dec '16

28 Dec '16

It's using PATH_MAX which is defined there, otherwise the build fails on musl libc. Signed-off-by: Gustavo Zacarias <gustavo(a)zacarias.com.ar> --- src/ucm/parser.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index c98373a..f520abc 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -32,6 +32,7 @@ #include "ucm_local.h" #include <dirent.h> +#include <limits.h> /** The name of the environment variable containing the UCM directory */ #define ALSA_CONFIG_UCM_VAR "ALSA_CONFIG_UCM" -- 2.10.2

2 1

[alsa-devel] [PATCH - alsa-lib 1/1] conf/cards: add support for pistachio-card.
by manohar.narkhede＠imgtec.com 28 Dec '16

28 Dec '16

From: Manohar Narkhede <Manohar.Narkhede(a)imgtec.com> Signed-off-by: Manohar Narkhede <Manohar.Narkhede(a)imgtec.com> diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index 008d399..60b5309 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -39,6 +39,7 @@ cfg_files = aliases.conf \ Maestro3.conf \ NFORCE.conf \ PC-Speaker.conf \ + pistachio-card.conf \ PMac.conf \ PMacToonie.conf \ PS3.conf \ diff --git a/src/conf/cards/aliases.conf b/src/conf/cards/aliases.conf index 4a92fb2..60f9d26 100644 --- a/src/conf/cards/aliases.conf +++ b/src/conf/cards/aliases.conf @@ -55,6 +55,7 @@ AV100 cards.CMI8788 AV200 cards.CMI8788 CMI8786 cards.CMI8788 CMI8787 cards.CMI8788 +pistachio cards.pistachio-card <confdir:pcm/default.conf> <confdir:pcm/dmix.conf> diff --git a/src/conf/cards/pistachio-card.conf b/src/conf/cards/pistachio-card.conf new file mode 100644 index 0000000..9c0318d --- /dev/null +++ b/src/conf/cards/pistachio-card.conf @@ -0,0 +1,34 @@ +# +# Configuration for the pistachio chip +# + +pistachio-card.pcm.default{ + @args [ CARD ] + @args.CARD { + type string + default "pistachio" + } + @args.DEVICE { + type integer + default 2 + } + + type asym + capture.pcm { + type multi + slaves.a.pcm "hw:0,4" + slaves.a.channels 12 + bindings.0.slave a + bindings.0.channel 4 + bindings.1.slave a + bindings.1.channel 5 + } + + playback.pcm { + type hw + card $CARD + device $DEVICE + + } + +} -- 2.1.0

3 3

[alsa-devel] [PATCH 2/2] ucm: Add ATTRIBUTE_UNUSED for unused parameters of execute_component_seq()
by mengdong.lin＠linux.intel.com 28 Dec '16

28 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> To fix the following warnings: main.c: In function ‘execute_component_seq’: main.c:489:24: warning: unused parameter ‘value_list1’ [-Wunused-parameter] struct list_head *value_list1, ^ main.c:490:24: warning: unused parameter ‘value_list2’ [-Wunused-parameter] struct list_head *value_list2, ^ main.c:491:24: warning: unused parameter ‘value_list3’ [-Wunused-parameter] struct list_head *value_list3, ^ Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 750e65d..38a5e81 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -486,9 +486,9 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, */ static int execute_component_seq(snd_use_case_mgr_t *uc_mgr, struct component_sequence *cmpt_seq, - struct list_head *value_list1, - struct list_head *value_list2, - struct list_head *value_list3, + struct list_head *value_list1 ATTRIBUTE_UNUSED, + struct list_head *value_list2 ATTRIBUTE_UNUSED, + struct list_head *value_list3 ATTRIBUTE_UNUSED, char *cdev) { struct use_case_device *device = cmpt_seq->device; -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ucm: Enable all warnings for automake C preprocessor
by mengdong.lin＠linux.intel.com 28 Dec '16

28 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> To display warnings for us to fix them, add '-Wall' to AM_CPPFLAGS. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/ucm/Makefile.am b/src/ucm/Makefile.am index 9d66b24..79e57d2 100644 --- a/src/ucm/Makefile.am +++ b/src/ucm/Makefile.am @@ -7,4 +7,4 @@ noinst_HEADERS = ucm_local.h all: libucm.la -AM_CPPFLAGS=-I$(top_srcdir)/include +AM_CPPFLAGS=-I$(top_srcdir)/include -Wall -- 2.7.4

3 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Apply ALC269_FIXUP_NO_SHUTUP on HDA_FIXUP_ACT_PROBE
by Gabriele Mazzotta 28 Dec '16

28 Dec '16

Setting shutup when the action is HDA_FIXUP_ACT_PRE_PROBE might not have the desired effect since it could be overridden by another more generic shutup function. Prevent this by setting the more specific shutup function on HDA_FIXUP_ACT_PROBE. Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ea81c08ddc7a..d30cc49512e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4400,7 +4400,7 @@ static void alc_no_shutup(struct hda_codec *codec) static void alc_fixup_no_shutup(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action == HDA_FIXUP_ACT_PRE_PROBE) { + if (action == HDA_FIXUP_ACT_PROBE) { struct alc_spec *spec = codec->spec; spec->shutup = alc_no_shutup; } -- 2.11.0

2 2

[alsa-devel] [PATCH] ALSA: mixart: fix a comment typo
by Geliang Tang 28 Dec '16

28 Dec '16

Fix a comment typo in mixart.h. Signed-off-by: Geliang Tang <geliangtang(a)gmail.com> --- sound/pci/mixart/mixart.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h index 0cc17e0..4267438 100644 --- a/sound/pci/mixart/mixart.h +++ b/sound/pci/mixart/mixart.h @@ -86,7 +86,7 @@ struct mixart_mgr { u32 msg_fifo[MSG_FIFO_SIZE]; int msg_fifo_readptr; int msg_fifo_writeptr; - atomic_t msg_processed; /* number of messages to be processed in takslet */ + atomic_t msg_processed; /* number of messages to be processed in tasklet */ struct mutex lock; /* interrupt lock */ struct mutex msg_lock; /* mailbox lock */ -- 2.9.3

2 1

[alsa-devel] [PATCH] Support both DSD LE/BE Amanero firmware versions
by Jussi Laako 28 Dec '16

28 Dec '16

Add DSD support for both little endian (DSD_U32_LE) and big endian (DSD_U32_BE) version of the Amanero firmware. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b3fd238..f5c68ea 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1359,6 +1359,21 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + + /* Amanero Combo384 USB interface with native DSD support */ + case USB_ID(0x16d0, 0x071a): + if (fp->altsetting == 2) { + switch (chip->dev->descriptor.bcdDevice) { + case 0x199: + return SNDRV_PCM_FMTBIT_DSD_U32_LE; + case 0x19b: + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + default: + break; + } + } + break; + default: break; } -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: use designated initializers
by Kees Cook 28 Dec '16

28 Dec '16

Prepare to mark sensitive kernel structures for randomization by making sure they're using designated initializers. These were identified during allyesconfig builds of x86, arm, and arm64, with most initializer fixes extracted from grsecurity. Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/synth/emux/emux_seq.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index a0209204ae48..55579f6b8cb2 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -33,13 +33,13 @@ static int snd_emux_unuse(void *private_data, struct snd_seq_port_subscribe *inf * MIDI emulation operators */ static struct snd_midi_op emux_ops = { - snd_emux_note_on, - snd_emux_note_off, - snd_emux_key_press, - snd_emux_terminate_note, - snd_emux_control, - snd_emux_nrpn, - snd_emux_sysex, + .note_on = snd_emux_note_on, + .note_off = snd_emux_note_off, + .key_press = snd_emux_key_press, + .note_terminate = snd_emux_terminate_note, + .control = snd_emux_control, + .nrpn = snd_emux_nrpn, + .sysex = snd_emux_sysex, }; -- 2.7.4 -- Kees Cook Nexus Security

3 2

Re: [alsa-devel] [PATCH] ALSA: cs5535audio: fix unused warnings on resume/suspend
by Takashi Iwai 28 Dec '16

28 Dec '16

On Sat, 17 Dec 2016 01:25:27 +0100, Jérémy Lefaure wrote: > > When CONFIG_PM_SLEEP is disabled, SIMPLE_DEV_PM_OPS does not use > snd_cs5535audio_resume and snd_cs5535audio_suspend functions: > > sound/pci/cs5535audio/cs5535audio_pm.c:77:12: warning: ‘snd_cs5535audio_resume’ defined but not used [-Wunused-function] > static int snd_cs5535audio_resume(struct device *dev) > ^~~~~~~~~~~~~~~~~~~~~~ > sound/pci/cs5535audio/cs5535audio_pm.c:58:12: warning: ‘snd_cs5535audio_suspend’ defined but not used [-Wunused-function] > static int snd_cs5535audio_suspend(struct device *dev) > ^~~~~~~~~~~~~~~~~~~~~~~ > > Adding __maybe_unused to the declaration of these functions removes the > warnings. > > Signed-off-by: Jérémy Lefaure <jeremy.lefaure(a)lse.epita.fr> Applied, thanks. Takashi > --- > sound/pci/cs5535audio/cs5535audio_pm.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c > index 06ac5d8da362..82bd10b68a77 100644 > --- a/sound/pci/cs5535audio/cs5535audio_pm.c > +++ b/sound/pci/cs5535audio/cs5535audio_pm.c > @@ -55,7 +55,7 @@ static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au) > > } > > -static int snd_cs5535audio_suspend(struct device *dev) > +static int __maybe_unused snd_cs5535audio_suspend(struct device *dev) > { > struct snd_card *card = dev_get_drvdata(dev); > struct cs5535audio *cs5535au = card->private_data; > @@ -74,7 +74,7 @@ static int snd_cs5535audio_suspend(struct device *dev) > return 0; > } > > -static int snd_cs5535audio_resume(struct device *dev) > +static int __maybe_unused snd_cs5535audio_resume(struct device *dev) > { > struct snd_card *card = dev_get_drvdata(dev); > struct cs5535audio *cs5535au = card->private_data; > -- > 2.11.0 > >

1 0

[alsa-devel] [PATCH 0/2] ASoC: Dell IoT Gateway audio support
by Shrirang Bagul 27 Dec '16

27 Dec '16

These patches add basic audio support on new Dell Edge IoT Gateways. RT5660 codec is connected to SSP2 port on Intel Atom Baytrail E3815 SoC. The machine driver is based on bytcr_rt5640. These devices have line-in and line-out jacks with line-out mute enable gpio described in ACPI tables. Shrirang Bagul (2): ASoC: rt5660: Add ACPI support ASoC: Intel: boards: Add Baytrail RT5660 machine driver sound/soc/codecs/rt5660.c | 26 ++ sound/soc/intel/Kconfig | 13 + sound/soc/intel/atom/sst/sst_acpi.c | 2 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcr_rt5660.c | 573 ++++++++++++++++++++++++++++++++++ 5 files changed, 616 insertions(+) create mode 100644 sound/soc/intel/boards/bytcr_rt5660.c -- 2.9.3

3 9

[alsa-devel] [snd-usb-audio] BUG: NULL pointer dereference at 0000000000000070
by Dave Young 25 Dec '16

25 Dec '16

Hi, With recent mainline kernel, I see a BUG, it is easy to reproduce, just plugging the usb microphone, bisected the first bad commit is: 16200948d8353fe29a473a394d7d26790deae0e7 is the first bad commit commit 16200948d8353fe29a473a394d7d26790deae0e7 Author: Takashi Iwai <tiwai(a)suse.de> Date: Mon Dec 5 11:19:38 2016 +0100 ALSA: usb-audio: Fix race at stopping the stream We've got a kernel crash report showing like: Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = a1d7c000 [snip] The BUG dmesg itself is below: [54029.102610] input: Samson Technologies Samson Meteor Mic as /devices/pci0000:00/0000:00:14.0/usb2/2-3/2-3:1.3/0003:17A0:0310.0003/input/input19 [54029.154424] hid-generic 0003:17A0:0310.0003: input: USB HID v1.00 Device [Samson Technologies Samson Meteor Mic] on usb-0000:00:14.0-3/input3 [54029.202035] usbcore: registered new interface driver snd-usb-audio [54029.242705] BUG: unable to handle kernel NULL pointer dereference at 0000000000000070 [54029.271667] IP: retire_playback_urb+0x5/0xd0 [snd_usb_audio] [54029.300462] PGD 0 [54029.300462] [54029.355691] Oops: 0000 [#1] SMP [54029.383215] Modules linked in: snd_usb_audio snd_usbmidi_lib snd_rawmidi macvtap macvlan tun ccm rfcomm fuse snd_hda_codec_hdmi cmac bnep kvm_intel kvm irqbypass i915 arc4 intel_gtt drm_kms_helper syscopyarea sysfillrect sysimgblt fb_sys_fops drm snd_hda_codec_realtek snd_hda_codec_generic iwlmvm mac80211 rtsx_pci_sdmmc iwlwifi snd_hda_intel snd_hda_codec snd_hwdep snd_hda_core cfg80211 snd_seq snd_seq_device btusb btrtl thinkpad_acpi btbcm pcspkr input_leds btintel serio_raw bluetooth snd_pcm e1000e snd_timer ptp rtsx_pci snd i2c_i801 pps_core rfkill mfd_core soundcore video nfsd auth_rpcgss nfs_acl lockd grace sunrpc [54029.480514] CPU: 1 PID: 0 Comm: swapper/1 Tainted: G W 4.9.0+ #209 [54029.514169] Hardware name: LENOVO 20ARS1BJ02/20ARS1BJ02, BIOS GJET91WW (2.41 ) 09/21/2016 [54029.548395] task: ffff9c2894a18bc0 task.stack: ffffa69dc0cd0000 [54029.582630] RIP: 0010:retire_playback_urb+0x5/0xd0 [snd_usb_audio] [54029.617049] RSP: 0018:ffff9c289f243cd0 EFLAGS: 00010086 [54029.651439] RAX: ffffffffc031bac0 RBX: ffff9c2868a8a000 RCX: 0000000000000001 [54029.686222] RDX: 0000000000000000 RSI: ffff9c288e3e3a00 RDI: 0000000000000000 [54029.721046] RBP: ffff9c289f243d00 R08: 0000000000000001 R09: ffff9c289e803b00 [54029.755850] R10: ffff9c28848f3380 R11: ffff9c289038d0b0 R12: ffff9c2868a8a140 [54029.790482] R13: ffff9c288e3e3a00 R14: 0000000000000000 R15: ffff9c288e3e0390 [54029.824649] FS: 0000000000000000(0000) GS:ffff9c289f240000(0000) knlGS:0000000000000000 [54029.859458] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [54029.894609] CR2: 0000000000000070 CR3: 000000021e008000 CR4: 00000000001406e0 [54029.929476] Call Trace: [54029.963968] <IRQ> [54029.998291] ? snd_complete_urb+0x80/0x260 [snd_usb_audio] [54030.033359] __usb_hcd_giveback_urb+0x76/0x100 [54030.068352] usb_hcd_giveback_urb+0x3c/0xc0 [54030.103167] xhci_giveback_urb_in_irq.isra.23+0x6f/0xa0 [54030.138683] finish_td.constprop.39+0x175/0x260 [54030.173733] xhci_irq+0x9f0/0x1450 [54030.208790] ? try_to_wake_up+0x1f2/0x390 [54030.243696] ? usb_hcd_poll_rh_status+0x190/0x190 [54030.278521] xhci_msi_irq+0x11/0x20 [54030.313376] __handle_irq_event_percpu+0x7e/0x1a0 [54030.348615] handle_irq_event_percpu+0x32/0x80 [54030.383917] handle_irq_event+0x2c/0x50 [54030.419012] handle_edge_irq+0x9f/0x120 [54030.454042] handle_irq+0x73/0x130 [54030.488522] ? _local_bh_enable+0x21/0x50 [54030.522777] do_IRQ+0x46/0xd0 [54030.556882] common_interrupt+0x90/0x90 [54030.591095] RIP: 0010:cpuidle_enter_state+0x134/0x2a0 [54030.625661] RSP: 0018:ffffa69dc0cd3e60 EFLAGS: 00000246 ORIG_RAX: ffffffffffffff2c [54030.660438] RAX: 0000000000000000 RBX: 00003123a9d13159 RCX: 000000000000001f [54030.695705] RDX: 00003123a9d13159 RSI: ffff9c289f254f98 RDI: 0000000000000000 [54030.731113] RBP: ffffa69dc0cd3e98 R08: cccccccccccccccd R09: 0000000000000018 [54030.766539] R10: 000000000000019c R11: 00000000000000a7 R12: 0000000000000004 [54030.802207] R13: 0000000000000004 R14: ffff9c289f25db08 R15: 00003123a9c9b583 [54030.837897] </IRQ> [54030.873227] cpuidle_enter+0x17/0x20 [54030.908827] call_cpuidle+0x23/0x40 [54030.944343] do_idle+0x189/0x200 [54030.979754] cpu_startup_entry+0x71/0x80 [54031.015166] start_secondary+0x142/0x160 [54031.050630] start_cpu+0x14/0x14 [54031.085944] Code: e9 03 41 5e 5d f7 f1 89 c0 c3 41 8b 76 64 4c 89 e7 e8 f0 fe ff ff eb c4 0f 1f 40 00 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 44 00 00 <48> 8b 4f 70 31 d2 8b 86 88 00 00 00 f7 b1 98 15 00 00 85 c0 75 [54031.124608] RIP: retire_playback_urb+0x5/0xd0 [snd_usb_audio] RSP: ffff9c289f243cd0 [54031.162852] CR2: 0000000000000070

1 0

[alsa-devel] How to keep track of the amount of queued samples?
by Jan Deinhard 23 Dec '16

23 Dec '16

Hi, I would like to understand the following quote which I copied from https://www.kernel.org/doc/Documentation/sound/alsa/timestamping.txt "The ALSA API provides two basic pieces of information, avail and delay, which combined with the trigger and current system timestamps allow for applications to keep track of the 'fullness' of the ring buffer and the amount of queued samples." Querying avail and delay is straight forward. But I have no idea how to use these values in combination with trigger_htstamp and the current system timestamp to calculate the fullness of the ring buffer. Can someone provide some insight? What exactly is meant by "current system timestamp"? Is it provied by ALSA (htstamp?) or should I use clock_gettime(CLOCK_MONOTONIC, &system_time); to get that timestamp? Thanks, Jan

2 1

[alsa-devel] [PATCH 1/2] ucm: Automatically load the best config file based on the card long name
by mengdong.lin＠linux.intel.com 23 Dec '16

23 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Intel DSP platform drivers are used by many different devices. For user space to differentiate them, ASoC machine driver may include the DMI info (vendor, product and board) in card long name. Possible card long names are: broadwell-rt286-Dell Inc.-XPS 13 9343-0310JH broadwell-rt286-Intel Corp.-Broadwell Client platform-Wilson Beach SDS bytcr-rt5640-ASUSTeK COMPUTER INC.-T100TA-T100TA bytcr-rt5651-Circuitco-Minnowboard Max D0 PLATFORM-MinnowBoard MAX ... And user space can define configuration files including fields separated by '.' as below: broadwell-rt286 broadwell-rt286.Dell.XPS bytcr-rt5640 bytcr-rt5640.ASUS.T100 bytcr-rt5651.MinnowboardMax ... When being asked to load configuration file of a card, UCM will try to find the card long name from the local machine, and then scan all available configuration file names, search every field of config file name in the card long name. The more characters match, the higher score the file has. Finally, the file with the highest score will be loaded to configure the sound card. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/ucm/parser.c b/src/ucm/parser.c index c98373a..ff75da7 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -55,6 +55,9 @@ static const char * const component_dir[] = { NULL, /* terminator */ }; +static int filename_filter(const struct dirent *dirent); +static int is_component_directory(const char *dir); + static int parse_sequence(snd_use_case_mgr_t *uc_mgr, struct list_head *base, snd_config_t *cfg); @@ -1328,6 +1331,210 @@ static int parse_master_file(snd_use_case_mgr_t *uc_mgr, snd_config_t *cfg) return 0; } +/* trim SPACE (0x20) from the string */ +static void trim_space(char *str) +{ + int i, j = 0; + + for (i = 0; str[i] && i < MAX_FILE; i++) { + if (str[i] != 0x20) + str[j++] = str[i]; + } + + str[j] = '\0'; +} + +/* find and store the card long name if the card is in this machine */ +static int get_card_long_name(snd_use_case_mgr_t *mgr) +{ + const char *card_name = mgr->card_name; + snd_ctl_t *handle; + int card, err, dev, idx; + snd_ctl_card_info_t *info; + const char *_name, *_long_name; + + snd_ctl_card_info_alloca(&info); + + card = -1; + if (snd_card_next(&card) < 0 || card < 0) { + uc_error("no soundcards found..."); + return -1; + } + + while (card >= 0) { + char name[32]; + + sprintf(name, "hw:%d", card); + err = snd_ctl_open(&handle, name, 0); + if (err < 0) { + uc_error("control open (%i): %s", card, + snd_strerror(err)); + goto next_card; + } + + err = snd_ctl_card_info(handle, info); + if (err < 0) { + uc_error("control hardware info (%i): %s", card, + snd_strerror(err)); + snd_ctl_close(handle); + goto next_card; + } + + _name = snd_ctl_card_info_get_name(info); + if (!strncmp(card_name, _name, 32)) { + _long_name = snd_ctl_card_info_get_longname(info); + strncpy(mgr->card_long_name, _long_name, + MAX_CARD_LONG_NAME); + snd_ctl_close(handle); + return 0; + } + + snd_ctl_close(handle); +next_card: + if (snd_card_next(&card) < 0) { + uc_error("snd_card_next"); + break; + } + } + + return -1; +} + +/* This function will find the best device-specific configuration file based + * on the sound card long name. + * Different devices may share the same sound driver and thus the same sound + * card name (short name), but they may still need different device-specific + * configurations. For user space to differentiate them, kernel drivers may + * include the DMI info (vendor, product and board) in the card long name. + * And user space can define configuration file names appending DMI keywords + * to the card name, like: + * bytcr-rt5640.ASUS.T100 + * bytcr-rt5651.MinnowBoard + * + * When being asked to load the configuration file for a card, this function + * will try to find the card long name from the local machine, and then scan + * all available configuration file names, search every field of the config + * file name in the card long name. The more characters match, the higher + * score the file has. Finally, the file with the highest score will be loaded. + */ +static void find_best_config_file(snd_use_case_mgr_t *mgr) +{ + const char *card_name = mgr->card_name; + char name[MAX_FILE]; + char *env = getenv(ALSA_CONFIG_UCM_VAR); + struct dirent **namelist; + char card_long_name[MAX_CARD_LONG_NAME]; + char *lpos, *lpos_next, *lname_end; + char *d_name, *dpos, *dpos_next, *d_name_end; + int d_name_len, field_len, long_name_len; + int i, cnt, err; + int score, score_max = 0, d_index = -1; + int is_first_field; + + if (get_card_long_name(mgr) < 0) { + /* cannot get long name, use card name as the conf file name */ + strcpy(mgr->conf_file_name, mgr->card_name); + return; + } + + strncpy(card_long_name, mgr->card_long_name, MAX_CARD_LONG_NAME); + trim_space(card_long_name); /* trim SPACE for matching */ + lname_end = card_long_name + strlen(card_long_name); + + /* get list of card directories under /usr/share/alsa/ucm */ + snprintf(name, sizeof(name)-1, + "%s", env ? env : ALSA_USE_CASE_DIR); + name[MAX_FILE-1] = '\0'; + + err = scandir(name, &namelist, filename_filter, alphasort); + if (err < 0) { + err = -errno; + uc_error("error: could not scan directory %s: %s", + name, strerror(-err)); + return; + } + cnt = err; + + /* scan the card directory names */ + for (i = 0; i < cnt; i++) { + + /* Skip the directories for component devices */ + if (is_component_directory(namelist[i]->d_name)) + continue; + + score = 0; + lpos = card_long_name; + + d_name = namelist[i]->d_name; + d_name_len = strlen(d_name); + d_name_end = d_name + d_name_len; + is_first_field = 1; + dpos = d_name; + + /* scan fields in the card directory name, separated by '.' */ + while (1) { + dpos_next = strchr(dpos, '.'); + if (dpos_next == dpos) /* start with '.' */ + goto next_field; + + if (!dpos_next) /* last field */ + field_len = d_name_end - dpos; + else + field_len = dpos_next - dpos; + + memcpy(name, dpos, field_len); + name[field_len] = '\0'; + + /* search the field in the given card long name */ + lpos_next = strstr(lpos, name); + if (!lpos_next) { + /* first field is the card name, must match */ + if (is_first_field) + break; + goto next_field; + } else { + /* The more characters match, + * the higher the score is. + */ + score += field_len; + if (lpos_next + field_len >= lname_end) + break; /* reach end of long name */ + + /* Skip the matched field, + * for searching next field. + */ + lpos = lpos_next + field_len; + } + +next_field: + if (!dpos_next) + break; /* no more fields */ + + dpos = dpos_next + 1; /* skip the separator '.' */ + is_first_field = 0; + if (dpos >= d_name_end) + break; + } + + if (score > score_max) { + score_max = score; + d_index = i; + } else if (score == 0 && score_max > 0) { + /* passed the card directories that can match, + * since directories are in alphabetical order. + */ + break; + } + } + + if (d_index >= 0) + strncpy(mgr->conf_file_name, namelist[d_index]->d_name, + MAX_CARD_LONG_NAME); + else + strncpy(mgr->conf_file_name, mgr->card_name, + MAX_CARD_LONG_NAME); +} + static int load_master_config(const char *card_name, snd_config_t **cfg) { char filename[MAX_FILE]; @@ -1355,7 +1562,9 @@ int uc_mgr_import_master_config(snd_use_case_mgr_t *uc_mgr) snd_config_t *cfg; int err; - err = load_master_config(uc_mgr->card_name, &cfg); + find_best_config_file(uc_mgr); + + err = load_master_config(uc_mgr->conf_file_name, &cfg); if (err < 0) return err; err = parse_master_file(uc_mgr, cfg); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 6d3302f..299a5b9 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -41,6 +41,7 @@ #include "use-case.h" #define MAX_FILE 256 +#define MAX_CARD_LONG_NAME 80 #define ALSA_USE_CASE_DIR ALSA_CONFIG_DIR "/ucm" #define SEQUENCE_ELEMENT_TYPE_CDEV 1 @@ -190,6 +191,8 @@ struct use_case_verb { */ struct snd_use_case_mgr { char *card_name; + char card_long_name[MAX_CARD_LONG_NAME]; + char conf_file_name[MAX_CARD_LONG_NAME]; char *comment; /* use case verb, devices and modifier configs parsed from files */ -- 2.7.4

5 11

[alsa-devel] [PATCH 0/2] ASoC: Use DMI name in sound card long name
by mengdong.lin＠linux.intel.com 22 Dec '16

22 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series is based on Liam's idea to use DMI name as the sound card short name in the audio workshop. Now we add DMI info to the card long name: - Not change the card name (short name), keep it simple. So the card ID won't change as well. - Use short name and DMI name together to make a readable and unqique long name. Machine drivers can use a new API to get this feature. In user space, when the sound server ask UCM to open a sound card, UCM will try to find the best configuration file based on the card long name, and only if not avaiable, fallback to the default configuration file that matches the card name (short name). Please see patch: ucm: Automatically load the best config file based on the card long name History: v1 from RFC: Give up using a table to map DMI info to a simple string, since we cannot maintain such a table for each machine. Just do verbatim copy of DMI info and let user space decide how to use the DMI info in card long name to find the best UCM config file. If device driver adds a flavor string, just append it after the DMI info fields to the card long name. Liam Girdwood (1): ASoC: core: Add API to use DMI name in sound card long name Mengdong Lin (1): ASoC: Intel: Use DMI name for sound card long name in Broadwell machine driver include/sound/soc.h | 4 ++ sound/soc/intel/boards/broadwell.c | 2 + sound/soc/soc-core.c | 81 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 87 insertions(+) -- 2.5.0

5 8

[alsa-devel] [PATCH] Revert "ALSA: usb-audio: Fix race at stopping the stream"
by Takashi Iwai 21 Dec '16

21 Dec '16

This reverts commit 16200948d8353fe29a473a394d7d26790deae0e7. The commit was intended to cover the race condition, but it introduced yet another regression for devices with the implicit feedback, leading to a kernel panic due to NULL-dereference in an irq context. As the race condition that was addressed by the commit is very rare and the regression is much worse, let's revert the commit for rc1, and fix the issue properly in a later patch. Fixes: 16200948d835 ("ALSA: usb-audio: Fix race at stopping the stream") Reported-by: Ioan-Adrian Ratiu <adi(a)adirat.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Linus, could you take this manually for rc1? I'm traveling and having a difficulty to set up a git pull request right now. Sorry for inconvenience. Thanks! Takashi sound/usb/endpoint.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a2cdf3370afe..15d1d5c63c3c 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -384,9 +384,6 @@ static void snd_complete_urb(struct urb *urb) if (unlikely(atomic_read(&ep->chip->shutdown))) goto exit_clear; - if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) - goto exit_clear; - if (usb_pipeout(ep->pipe)) { retire_outbound_urb(ep, ctx); /* can be stopped during retire callback */ @@ -537,11 +534,6 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) alive, ep->ep_num); clear_bit(EP_FLAG_STOPPING, &ep->flags); - ep->data_subs = NULL; - ep->sync_slave = NULL; - ep->retire_data_urb = NULL; - ep->prepare_data_urb = NULL; - return 0; } @@ -1028,6 +1020,10 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep) if (--ep->use_count == 0) { deactivate_urbs(ep, false); + ep->data_subs = NULL; + ep->sync_slave = NULL; + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; set_bit(EP_FLAG_STOPPING, &ep->flags); } } -- 2.11.0

1 0

Re: [alsa-devel] [PATCH] ALSA: snd-usb: fix IRQ triggered NULL pointer dereference
by Takashi Iwai 21 Dec '16

21 Dec '16

On Tue, 20 Dec 2016 14:04:56 +0100, Takashi Sakamoto wrote: > > Hi, > > On Dec 20 2016 20:47, Ioan-Adrian Ratiu wrote: > > On Tue, 20 Dec 2016, Takashi Sakamoto <o-takashi(a)sakamocchi.jp> wrote: > >> --- > >> > >> Hi, > >> > >>> Commit 16200948d835 ("ALSA: usb-audio: Fix race at stopping the stream") > >>> fixes a race-codition but it also introduces another really nasty data race > >>> regression which makes my usb sound card [1] completely useless, throwing > >>> the kernel into a panic if anything from userspace tries to start playback. > >>> > >>> The problem is this: ep->data_subs is now set to NULL every time inside > >>> wait_clear_urbs(). ep->data_subs is initalized only in one place in > >>> start_endpoints(), then it is immediately wiped by the pre-existing call to > >>> wait_clear_urbs() inside snd_usb_endpoint_start(). > >>> > >>> To ilustrate, this is what happens in the non-irq part of the code: > >>> > >>> Step 1 (inside start_endpoints): ep->data_subs = subs; > >>> Step 2 (inside start_endpoints): snd_usb_endpoint_start(ep, can_sleep); > >>> Step 3 (inside snd_usb_endpoint_start): wait_clear_urbs(ep); > >>> Step 4 (inside wait_clear_urbs): ep->data_subs = NULL; > >>> > >>> Here's a simplified call stack for the IRQ part (full one at the end): > >>> > >>> (NULL dereference, param "subs" is passed NULL value of ep->data_subs) > >>> retire_playback_urb > >>> retire_outbound_urb > >>> snd_complete_urb > >>> (...) > >>> xhci_irq > >>> > >>> Looking at the git log there seems to be quite a lot of history in this > >>> part of the codebase, dating back to 2012 or earlier. My fix is based on > >>> 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start") and > >>> e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture > >>> stream") with a few modifications of my own. My idea is to do the > >>> ep->data_subs wiping before endpoint initialization in a manner similar > >>> to these older commits by using stop_endpoints() in snd_usb_pcm_prepare() > >>> and at the same time keep the ep->data_subs = NULL in wait_clear_urbs() to > >>> not trigger the recently fixed stream stopping race again. > >>> > >>> Full call stack: > >>> > >>> [ 131.093240] BUG: unable to handle kernel NULL pointer dereference at 0000000000000010 > >>> [ 131.094313] IP: retire_playback_urb+0x1b/0x160 [snd_usb_audio] > >>> [ 131.095046] PGD 0 > >>> [ 131.095047] > >>> [ 131.096509] Oops: 0000 [#1] PREEMPT SMP > >>> [ 131.097255] Modules linked in: fuse snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_device ctr ccm arc4 ath9k intel_rapl ath9k_common x86_pkg_temp_thermal ath9k_hw intel_powerclamp coretemp joydev mousedev ath kvm_intel mac80211 kvm > >>> input_leds hid_generic psmouse irqbypass usbhid hid crct10dif_pclmul crc32_pclmul serio_raw crc32c_intel atkbd ghash_clmulni_intel snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel pcbc aesni_intel libps2 cfg8 > >>> 0211 aes_x86_64 crypto_simd dell_laptop glue_helper r8169 dell_smbios snd_hda_codec sr_mod cryptd led_class mii rfkill snd_hwdep cdrom snd_hda_core ac dcdbas i8042 xhci_pci serio xhci_hcd snd_pcm tpm_tis battery tpm_tis_core lpc_ich tpm > >>> snd_timer evdev shpchp i2c_i801 sch_fq_codel > >>> [ 131.105551] CPU: 2 PID: 165 Comm: irq/29-xhci_hcd Not tainted 4.9.0-gd824cdc58ba0 #10 > >>> [ 131.107516] Hardware name: Dell Inc. Inspiron 3521/018DYG, BIOS A14 07/31/2015 > >>> [ 131.109592] task: ffff880154a70000 task.stack: ffffc90000f48000 > >>> [ 131.111746] RIP: 0010:retire_playback_urb+0x1b/0x160 [snd_usb_audio] > >>> [ 131.113899] RSP: 0018:ffffc90000f4bc10 EFLAGS: 00010082 > >>> [ 131.116080] RAX: ffffffffa04cabe0 RBX: 0000000000000000 RCX: 0000000000000000 > >>> [ 131.118284] RDX: 0000000000000000 RSI: ffff8801435a4c00 RDI: 0000000000000000 > >>> [ 131.120505] RBP: ffffc90000f4bc40 R08: 0000000000000001 R09: 0000000000000001 > >>> [ 131.122807] R10: 0000000000000001 R11: ffffffff82f60d6d R12: ffff8801432a0238 > >>> [ 131.125265] R13: ffff8801435a4c00 R14: 0000000000000000 R15: ffff8801535c83b8 > >>> [ 131.127723] FS: 0000000000000000(0000) GS:ffff88015a000000(0000) knlGS:0000000000000000 > >>> [ 131.130217] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 > >>> [ 131.132568] CR2: 0000000000000010 CR3: 0000000001e0b000 CR4: 00000000001406e0 > >>> [ 131.135007] Call Trace: > >>> [ 131.137329] snd_complete_urb+0x190/0x2b0 [snd_usb_audio] > >>> [ 131.139526] __usb_hcd_giveback_urb+0x8e/0x160 > >>> [ 131.141948] usb_hcd_giveback_urb+0x43/0x110 > >>> [ 131.144104] xhci_giveback_urb_in_irq.isra.22+0x7d/0xb0 [xhci_hcd] > >>> [ 131.146521] finish_td.constprop.38+0x1de/0x2f0 [xhci_hcd] > >>> [ 131.148736] xhci_irq+0x13a2/0x1ca0 [xhci_hcd] > >>> [ 131.150972] ? trace_hardirqs_on+0xd/0x10 > >>> [ 131.153209] ? _raw_spin_unlock_irq+0x2c/0x50 > >>> [ 131.155390] ? irq_thread+0xb5/0x1d0 > >>> [ 131.157771] xhci_msi_irq+0x11/0x20 [xhci_hcd] > >>> [ 131.159938] irq_forced_thread_fn+0x2f/0x70 > >>> [ 131.162072] ? irq_thread+0xb5/0x1d0 > >>> [ 131.164141] irq_thread+0x149/0x1d0 > >>> [ 131.166132] ? irq_finalize_oneshot.part.2+0xe0/0xe0 > >>> [ 131.168142] ? wake_threads_waitq+0x30/0x30 > >>> [ 131.170149] kthread+0x10f/0x150 > >>> [ 131.172144] ? irq_thread_dtor+0xc0/0xc0 > >>> [ 131.174139] ? kthread_create_on_node+0x40/0x40 > >>> [ 131.176116] ret_from_fork+0x2a/0x40 > >>> [ 131.178074] Code: 8b 77 64 4c 89 e7 e8 e5 fe ff ff eb c3 0f 1f 00 0f 1f 44 00 00 55 31 d2 48 89 e5 41 57 41 56 41 55 41 54 53 48 89 fb 48 83 ec 08 <48> 8b 47 10 48 8b 4f 70 48 c7 c7 88 7b 4d a0 4c 8b a0 78 01 00 > >>> [ 131.182382] RIP: retire_playback_urb+0x1b/0x160 [snd_usb_audio] RSP: ffffc90000f4bc10 > >>> [ 131.184562] CR2: 0000000000000010 > >>> > >>> [1] 041e:3232 Creative Technology SoundBlaster X-FI HD > >>> [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-December/115425.h… > >> > >> I can regenerate this bug with my EMU 0404 USB. My understanding of this bug is > >> quite similar to your perspective. This bug is quite critical. In my > >> understanding, we encounter this bug in all of cases in which snd-usb-audio is > >> applied for PCM playback substream. > >> > >> Below workaround can also suppress this bug. > >> > >> diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > >> index a2cdf33..9b34f76 100644 > >> --- a/sound/usb/endpoint.c > >> +++ b/sound/usb/endpoint.c > >> @@ -537,11 +537,14 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) > >> alive, ep->ep_num); > >> clear_bit(EP_FLAG_STOPPING, &ep->flags); > >> > >> - ep->data_subs = NULL; > >> ep->sync_slave = NULL; > >> ep->retire_data_urb = NULL; > >> ep->prepare_data_urb = NULL; > >> > >> + msleep(200); > >> + > >> + ep->data_subs = NULL; > >> + > >> return 0; > >> } > >> > >> If initialization of 'struct snd_usb_endpoint.data_subs' can be done after all > >> of queued URBs are processed and corresponding complete interrupts are cought, > >> we can solve this critical bug. > >> > >>> Signed-off-by: Ioan-Adrian Ratiu <adi(a)adirat.com> > >>> --- > >>> sound/usb/endpoint.c | 11 ++--------- > >>> sound/usb/endpoint.h | 2 +- > >>> sound/usb/pcm.c | 13 ++++++------- > >>> 3 files changed, 9 insertions(+), 17 deletions(-) > >>> > >>> diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > >>> index a2cdf3370afe..4465f324c2c2 100644 > >>> --- a/sound/usb/endpoint.c > >>> +++ b/sound/usb/endpoint.c > >>> @@ -920,9 +920,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, > >>> /** > >>> * snd_usb_endpoint_start: start an snd_usb_endpoint > >>> * > >>> - * @ep: the endpoint to start > >>> - * @can_sleep: flag indicating whether the operation is executed in > >>> - * non-atomic context > >>> + * @ep: the endpoint to start > >>> * > >>> * A call to this function will increment the use count of the endpoint. > >>> * In case it is not already running, the URBs for this endpoint will be > >>> @@ -932,7 +930,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, > >>> * > >>> * Returns an error if the URB submission failed, 0 in all other cases. > >>> */ > >>> -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep) > >>> +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) > >>> { > >>> int err; > >>> unsigned int i; > >>> @@ -944,11 +942,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep) > >>> if (++ep->use_count != 1) > >>> return 0; > >>> > >>> - /* just to be sure */ > >>> - deactivate_urbs(ep, false); > >>> - if (can_sleep) > >>> - wait_clear_urbs(ep); > >>> - > >>> ep->active_mask = 0; > >>> ep->unlink_mask = 0; > >>> ep->phase = 0; > >>> diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h > >>> index 6428392d8f62..584f295d7c77 100644 > >>> --- a/sound/usb/endpoint.h > >>> +++ b/sound/usb/endpoint.h > >>> @@ -18,7 +18,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, > >>> struct audioformat *fmt, > >>> struct snd_usb_endpoint *sync_ep); > >>> > >>> -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep); > >>> +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); > >>> void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep); > >>> void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); > >>> int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); > >>> diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > >>> index 34c6d4f2c0b6..db26f767f851 100644 > >>> --- a/sound/usb/pcm.c > >>> +++ b/sound/usb/pcm.c > >>> @@ -218,7 +218,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, > >>> } > >>> } > >>> > >>> -static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) > >>> +static int start_endpoints(struct snd_usb_substream *subs) > >>> { > >>> int err; > >>> > >>> @@ -231,7 +231,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) > >>> dev_dbg(&subs->dev->dev, "Starting data EP @%p\n", ep); > >>> > >>> ep->data_subs = subs; > >>> - err = snd_usb_endpoint_start(ep, can_sleep); > >>> + err = snd_usb_endpoint_start(ep); > >>> if (err < 0) { > >>> clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); > >>> return err; > >>> @@ -260,7 +260,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) > >>> dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); > >>> > >>> ep->sync_slave = subs->data_endpoint; > >>> - err = snd_usb_endpoint_start(ep, can_sleep); > >>> + err = snd_usb_endpoint_start(ep); > >>> if (err < 0) { > >>> clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); > >>> return err; > >>> @@ -809,8 +809,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) > >>> goto unlock; > >>> } > >>> > >>> - snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); > >>> - snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); > >>> + stop_endpoints(subs, true); > >>> > >>> ret = set_format(subs, subs->cur_audiofmt); > >>> if (ret < 0) > >>> @@ -850,7 +849,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) > >>> /* for playback, submit the URBs now; otherwise, the first hwptr_done > >>> * updates for all URBs would happen at the same time when starting */ > >>> if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) > >>> - ret = start_endpoints(subs, true); > >>> + return start_endpoints(subs); > >>> > >>> unlock: > >>> snd_usb_unlock_shutdown(subs->stream->chip); > >>> @@ -1666,7 +1665,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream > >>> > >>> switch (cmd) { > >>> case SNDRV_PCM_TRIGGER_START: > >>> - err = start_endpoints(subs, false); > >>> + err = start_endpoints(subs); > >>> if (err < 0) > >>> return err; > >> > >> This patch works better, but not the best. It's a bit intrusive. > > > > I disagree. Being intrusive is not a good reason to reject a patch, > > especially if the alternative is to insert random magic number delays > > in hopes of synchronizing process and irq contexts to avoid critical > > errors. > > I apologize if you got some negative opinions in my previous message. > > My intention to show the ugly workaround is not to merge it instead of > yours, but clear what causes this bug. > > For example, we can also apply below patch for this bug, to solve it > roughly. > > diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > index a2cdf33..6ec4eb1 100644 > --- a/sound/usb/endpoint.c > +++ b/sound/usb/endpoint.c > @@ -162,7 +162,7 @@ int snd_usb_endpoint_next_packet_size(struct > snd_usb_endpoint *ep) > static void retire_outbound_urb(struct snd_usb_endpoint *ep, > struct snd_urb_ctx *urb_ctx) > { > - if (ep->retire_data_urb) > + if (ep->retire_data_urb && ep->data_subs) > ep->retire_data_urb(ep->data_subs, urb_ctx->urb); > } > > But this is just a workaround and can't fix it thoroughly. The main > issue is synchronization between interrupt/process contexts. > > > Please take the time to fully analyze my patch and let's have a > > discussion based on it, not reject it outright and replace it with > > a quick and dirty delay hack. > > OK. I'll deliberately check it again so that I have no overlooks. I > with this thread also catch the other developers enough helpful to > you. (I just eventually caught your patch in LKML and not developer > for this category of devices.) Sorry for the late reply, as I've been (still) off and had bad net connections. About your fix: Sakamoto-san is right, it's no good way to fix this kind or problem. The easiest option right now is just to revert my previous fix, as it obviously introduces another regression. The correct fix will be taken after that. I'm going to prepare a revert patch and ask Linus to take it before rc1. thanks, Takashi > > > Ionel > > > >> > >> What we need to synchronize process contexts to interrupt context till all of > >> queued URBs are handled. This seems to be a main purpose of wait_clear_bits, > >> while current implementation of snd_complete_urb()/wait_clear_bits() is not > >> enough. > >> > >> Thanks for your reporting. > >> > >> Takashi Sakamoto > >> -- > >> 2.9.3 > > > Thanks > > Takashi Sakamoto >

1 0

[alsa-devel] [PATCH] ASoC: sun4i-i2s: Add quirks for newer SoCs
by codekipper＠gmail.com 20 Dec '16

20 Dec '16

From: Marcus Cooper <codekipper(a)gmail.com> Newer SoCs have additional functionality so a quirks structure has been added to handle them. So far we've seen the use of a reset controller, a different address for the TXFIFO and varying register changes. This patch prepares the driver for these changes and adds the reset specifier. Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> --- .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 47 ++++++++++++++++++++-- 2 files changed, 45 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt index 7a2c0945fd22..494a881ccd21 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt @@ -18,6 +18,8 @@ Required properties: - "apb" : clock for the I2S bus interface - "mod" : module clock for the I2S controller - #sound-dai-cells : Must be equal to 0 +- resets: reset specifier for the ahb reset (A31 and newer only) + Example: diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index f24d19526603..80fe4f1d6e3b 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -14,9 +14,11 @@ #include <linux/clk.h> #include <linux/dmaengine.h> #include <linux/module.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> @@ -92,6 +94,7 @@ struct sun4i_i2s { struct clk *bus_clk; struct clk *mod_clk; struct regmap *regmap; + struct reset_control *rst; unsigned int mclk_freq; @@ -104,6 +107,13 @@ struct sun4i_i2s_clk_div { u8 val; }; +struct sun4i_i2s_quirks { + unsigned int reg_dac_txdata; /* TX FIFO offset for DMA config */ + bool has_reset; + const struct regmap_config *sun4i_i2s_regmap; + const struct snd_soc_dai_ops *ops; +}; + static const struct sun4i_i2s_clk_div sun4i_i2s_bclk_div[] = { { .div = 2, .val = 0 }, { .div = 4, .val = 1 }, @@ -541,7 +551,6 @@ static struct snd_soc_dai_driver sun4i_i2s_dai = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &sun4i_i2s_dai_ops, .symmetric_rates = 1, }; @@ -655,6 +664,7 @@ static int sun4i_i2s_probe(struct platform_device *pdev) { struct sun4i_i2s *i2s; struct resource *res; + const struct sun4i_i2s_quirks *quirks; void __iomem *regs; int irq, ret; @@ -680,8 +690,14 @@ static int sun4i_i2s_probe(struct platform_device *pdev) return PTR_ERR(i2s->bus_clk); } + quirks = of_device_get_match_data(&pdev->dev); + if (quirks == NULL) { + dev_err(&pdev->dev, "Failed to determine the quirks to use\n"); + return -ENODEV; + } + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, - &sun4i_i2s_regmap_config); + quirks->sun4i_i2s_regmap); if (IS_ERR(i2s->regmap)) { dev_err(&pdev->dev, "Regmap initialisation failed\n"); return PTR_ERR(i2s->regmap); @@ -692,13 +708,25 @@ static int sun4i_i2s_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Can't get our mod clock\n"); return PTR_ERR(i2s->mod_clk); } + - i2s->playback_dma_data.addr = res->start + SUN4I_I2S_FIFO_TX_REG; + i2s->playback_dma_data.addr = res->start + quirks->reg_dac_txdata; i2s->playback_dma_data.maxburst = 4; i2s->capture_dma_data.addr = res->start + SUN4I_I2S_FIFO_RX_REG; i2s->capture_dma_data.maxburst = 4; + if (quirks->has_reset) { + i2s->rst = devm_reset_control_get_optional(&pdev->dev, NULL); + if (IS_ERR(i2s->rst) && PTR_ERR(i2s->rst) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + dev_err(&pdev->dev, "Failed to get reset: %d\n", ret); + goto err_pm_disable; + } + if (!IS_ERR(i2s->rst)) + reset_control_deassert(i2s->rst); + } + pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { ret = sun4i_i2s_runtime_resume(&pdev->dev); @@ -706,6 +734,8 @@ static int sun4i_i2s_probe(struct platform_device *pdev) goto err_pm_disable; } + /* Register ops with dai */ + sun4i_i2s_dai.ops = quirks->ops; ret = devm_snd_soc_register_component(&pdev->dev, &sun4i_i2s_component, &sun4i_i2s_dai, 1); @@ -742,8 +772,17 @@ static int sun4i_i2s_remove(struct platform_device *pdev) return 0; } +static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = { + .reg_dac_txdata = SUN4I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, + .ops = &sun4i_i2s_dai_ops, +}; + static const struct of_device_id sun4i_i2s_match[] = { - { .compatible = "allwinner,sun4i-a10-i2s", }, + { + .compatible = "allwinner,sun4i-a10-i2s", + .data = &sun4i_a10_i2s_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun4i_i2s_match); -- 2.11.0

3 2

[alsa-devel] [PATCH v3 0/4] Generic HDMI codec: Add channel mapping control
by Arnaud Pouliquen 20 Dec '16

20 Dec '16

Aim of this patch is to add 'Playback Channel Map' control to export audio capabilities in term of HDMI sink speakers allocation. V3: - "ASoC: hdmi-codec: add channel mapping control": => Minor fixes: - select stereo speaker config by default if HDMI cable unplugged - fix compilation warning. - "DRM: add help to get ELD speaker allocation" => No delta vs V2. - "ALSA pcm: allow non constant snd_pcm_chmap_elem" => No delta vs V2. - "ASoC: core: add optional pcm_new callback for DAI driver" => No delta vs V2. V2: In this version I use chmap helper functions from pcm_lib.c. It is quite tricky to use it from ASoC due to the relation chip of the controls with the pcm runtime. After several tries, my conclusion is that it must be handled in ASoC DAI driver. Main reason is that the DAI driver can not provide snd_pcm_chmap struct through kcontrol structure. But this induces that soc-core provides pcm runtime structure to DAI driver during probe. Base on this conclusion. I reworked my patches by adding 2 new patches in patch-set 1) ALSA pcm: allow non constant snd_pcm_chmap_elem This patch allows to handle non constant channel mapping. As ELD information can change during runtime it is mandatory to properly handle the feature. In term of compatibility with legacy it should be straightforward, as update just consists in suppressing the 'const' constraint. 2) ASoC: core: add optional pcm_new callback for DAI driver This is the only way i found to be able to use snd_pcm_add_chmap_ctls and associated controls helper functions. From my point of view, this is the more simple way to add relationship between DAI and associated pcm devices. Notice that this patch, if accepted, makes the following one obsolete, as it also answer to the associated topic: [PATCH v5 0/2] ALSA controls management using index/device/sub-devices fields (http://www.spinics.net/lists/alsa-devel/msg57639.html). If you estimate that this it not reasonable i will come back to my first version. V1: This patch follows discussion initiate here: [RFC] ASOC: HDMI audio info frame speaker allocation http://www.spinics.net/lists/alsa-devel/msg57363.html The code is fully inspired from HDA driver. On hw_param, HDMI sink speaker capabilities are exported via TLV ops and a CEA allocation is choson, based on ELD information and the number of channels requested by user. Mains differences with HDA implementation are: - Control is read only - Channel swap is not supported. Consequence is that unused channel in the mid of CEA audio infoframe channel mapping are considered as active. example for channel allocation 0x02: FL, FR, 0, FC) This configuration is only available for a 4 channels stream. - Channel allocation table has been reordered and HDMI 2.0 is not supported. Arnaud Pouliquen (4): DRM: add help to get ELD speaker allocation ALSA pcm: allow non constant snd_pcm_chmap_elem ASoC: core: add optional pcm_new callback for DAI driver ASoC: hdmi-codec: add channel mapping control include/drm/drm_edid.h | 13 ++ include/sound/pcm.h | 4 +- include/sound/soc-dai.h | 3 + sound/core/pcm_lib.c | 6 +- sound/soc/codecs/hdmi-codec.c | 313 +++++++++++++++++++++++++++++++++++++++++- sound/soc/soc-core.c | 28 ++++ 6 files changed, 361 insertions(+), 6 deletions(-) -- 1.9.1

4 8

[alsa-devel] [PATCH v2] ASoC: wm_adsp: Don't overrun firmware file buffer when reading region data
by Richard Fitzgerald 20 Dec '16

20 Dec '16

Protect against corrupt firmware files by ensuring that the length we get for the data in a region actually lies within the available firmware file data buffer. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 593b7d1..d72ccef 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1551,7 +1551,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wmfw_region *region; const struct wm_adsp_region *mem; const char *region_name; - char *file, *text; + char *file, *text = NULL; struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; @@ -1700,10 +1700,21 @@ static int wm_adsp_load(struct wm_adsp *dsp) regions, le32_to_cpu(region->len), offset, region_name); + if ((pos + le32_to_cpu(region->len) + sizeof(*region)) > + firmware->size) { + adsp_err(dsp, + "%s.%d: %s region len %d bytes exceeds file length %zu\n", + file, regions, region_name, + le32_to_cpu(region->len), firmware->size); + ret = -EINVAL; + goto out_fw; + } + if (text) { memcpy(text, region->data, le32_to_cpu(region->len)); adsp_info(dsp, "%s: %s\n", file, text); kfree(text); + text = NULL; } if (reg) { @@ -1748,6 +1759,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) regmap_async_complete(regmap); wm_adsp_buf_free(&buf_list); release_firmware(firmware); + kfree(text); out: kfree(file); @@ -2233,6 +2245,17 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { + if ((pos + le32_to_cpu(blk->len) + sizeof(*blk)) > + firmware->size) { + adsp_err(dsp, + "%s.%d: %s region len %d bytes exceeds file length %zu\n", + file, blocks, region_name, + le32_to_cpu(blk->len), + firmware->size); + ret = -EINVAL; + goto out_fw; + } + buf = wm_adsp_buf_alloc(blk->data, le32_to_cpu(blk->len), &buf_list); -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] ALSA: snd-usb: fix IRQ triggered NULL pointer dereference
by Takashi Sakamoto 20 Dec '16

20 Dec '16

--- Hi, > Commit 16200948d835 ("ALSA: usb-audio: Fix race at stopping the stream") > fixes a race-codition but it also introduces another really nasty data race > regression which makes my usb sound card [1] completely useless, throwing > the kernel into a panic if anything from userspace tries to start playback. > > The problem is this: ep->data_subs is now set to NULL every time inside > wait_clear_urbs(). ep->data_subs is initalized only in one place in > start_endpoints(), then it is immediately wiped by the pre-existing call to > wait_clear_urbs() inside snd_usb_endpoint_start(). > > To ilustrate, this is what happens in the non-irq part of the code: > > Step 1 (inside start_endpoints): ep->data_subs = subs; > Step 2 (inside start_endpoints): snd_usb_endpoint_start(ep, can_sleep); > Step 3 (inside snd_usb_endpoint_start): wait_clear_urbs(ep); > Step 4 (inside wait_clear_urbs): ep->data_subs = NULL; > > Here's a simplified call stack for the IRQ part (full one at the end): > > (NULL dereference, param "subs" is passed NULL value of ep->data_subs) > retire_playback_urb > retire_outbound_urb > snd_complete_urb > (...) > xhci_irq > > Looking at the git log there seems to be quite a lot of history in this > part of the codebase, dating back to 2012 or earlier. My fix is based on > 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start") and > e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture > stream") with a few modifications of my own. My idea is to do the > ep->data_subs wiping before endpoint initialization in a manner similar > to these older commits by using stop_endpoints() in snd_usb_pcm_prepare() > and at the same time keep the ep->data_subs = NULL in wait_clear_urbs() to > not trigger the recently fixed stream stopping race again. > > Full call stack: > > [ 131.093240] BUG: unable to handle kernel NULL pointer dereference at 0000000000000010 > [ 131.094313] IP: retire_playback_urb+0x1b/0x160 [snd_usb_audio] > [ 131.095046] PGD 0 > [ 131.095047] > [ 131.096509] Oops: 0000 [#1] PREEMPT SMP > [ 131.097255] Modules linked in: fuse snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_device ctr ccm arc4 ath9k intel_rapl ath9k_common x86_pkg_temp_thermal ath9k_hw intel_powerclamp coretemp joydev mousedev ath kvm_intel mac80211 kvm > input_leds hid_generic psmouse irqbypass usbhid hid crct10dif_pclmul crc32_pclmul serio_raw crc32c_intel atkbd ghash_clmulni_intel snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel pcbc aesni_intel libps2 cfg8 > 0211 aes_x86_64 crypto_simd dell_laptop glue_helper r8169 dell_smbios snd_hda_codec sr_mod cryptd led_class mii rfkill snd_hwdep cdrom snd_hda_core ac dcdbas i8042 xhci_pci serio xhci_hcd snd_pcm tpm_tis battery tpm_tis_core lpc_ich tpm > snd_timer evdev shpchp i2c_i801 sch_fq_codel > [ 131.105551] CPU: 2 PID: 165 Comm: irq/29-xhci_hcd Not tainted 4.9.0-gd824cdc58ba0 #10 > [ 131.107516] Hardware name: Dell Inc. Inspiron 3521/018DYG, BIOS A14 07/31/2015 > [ 131.109592] task: ffff880154a70000 task.stack: ffffc90000f48000 > [ 131.111746] RIP: 0010:retire_playback_urb+0x1b/0x160 [snd_usb_audio] > [ 131.113899] RSP: 0018:ffffc90000f4bc10 EFLAGS: 00010082 > [ 131.116080] RAX: ffffffffa04cabe0 RBX: 0000000000000000 RCX: 0000000000000000 > [ 131.118284] RDX: 0000000000000000 RSI: ffff8801435a4c00 RDI: 0000000000000000 > [ 131.120505] RBP: ffffc90000f4bc40 R08: 0000000000000001 R09: 0000000000000001 > [ 131.122807] R10: 0000000000000001 R11: ffffffff82f60d6d R12: ffff8801432a0238 > [ 131.125265] R13: ffff8801435a4c00 R14: 0000000000000000 R15: ffff8801535c83b8 > [ 131.127723] FS: 0000000000000000(0000) GS:ffff88015a000000(0000) knlGS:0000000000000000 > [ 131.130217] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 > [ 131.132568] CR2: 0000000000000010 CR3: 0000000001e0b000 CR4: 00000000001406e0 > [ 131.135007] Call Trace: > [ 131.137329] snd_complete_urb+0x190/0x2b0 [snd_usb_audio] > [ 131.139526] __usb_hcd_giveback_urb+0x8e/0x160 > [ 131.141948] usb_hcd_giveback_urb+0x43/0x110 > [ 131.144104] xhci_giveback_urb_in_irq.isra.22+0x7d/0xb0 [xhci_hcd] > [ 131.146521] finish_td.constprop.38+0x1de/0x2f0 [xhci_hcd] > [ 131.148736] xhci_irq+0x13a2/0x1ca0 [xhci_hcd] > [ 131.150972] ? trace_hardirqs_on+0xd/0x10 > [ 131.153209] ? _raw_spin_unlock_irq+0x2c/0x50 > [ 131.155390] ? irq_thread+0xb5/0x1d0 > [ 131.157771] xhci_msi_irq+0x11/0x20 [xhci_hcd] > [ 131.159938] irq_forced_thread_fn+0x2f/0x70 > [ 131.162072] ? irq_thread+0xb5/0x1d0 > [ 131.164141] irq_thread+0x149/0x1d0 > [ 131.166132] ? irq_finalize_oneshot.part.2+0xe0/0xe0 > [ 131.168142] ? wake_threads_waitq+0x30/0x30 > [ 131.170149] kthread+0x10f/0x150 > [ 131.172144] ? irq_thread_dtor+0xc0/0xc0 > [ 131.174139] ? kthread_create_on_node+0x40/0x40 > [ 131.176116] ret_from_fork+0x2a/0x40 > [ 131.178074] Code: 8b 77 64 4c 89 e7 e8 e5 fe ff ff eb c3 0f 1f 00 0f 1f 44 00 00 55 31 d2 48 89 e5 41 57 41 56 41 55 41 54 53 48 89 fb 48 83 ec 08 <48> 8b 47 10 48 8b 4f 70 48 c7 c7 88 7b 4d a0 4c 8b a0 78 01 00 > [ 131.182382] RIP: retire_playback_urb+0x1b/0x160 [snd_usb_audio] RSP: ffffc90000f4bc10 > [ 131.184562] CR2: 0000000000000010 > > [1] 041e:3232 Creative Technology SoundBlaster X-FI HD > [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-December/115425.h… I can regenerate this bug with my EMU 0404 USB. My understanding of this bug is quite similar to your perspective. This bug is quite critical. In my understanding, we encounter this bug in all of cases in which snd-usb-audio is applied for PCM playback substream. Below workaround can also suppress this bug. diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a2cdf33..9b34f76 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -537,11 +537,14 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) alive, ep->ep_num); clear_bit(EP_FLAG_STOPPING, &ep->flags); - ep->data_subs = NULL; ep->sync_slave = NULL; ep->retire_data_urb = NULL; ep->prepare_data_urb = NULL; + msleep(200); + + ep->data_subs = NULL; + return 0; } If initialization of 'struct snd_usb_endpoint.data_subs' can be done after all of queued URBs are processed and corresponding complete interrupts are cought, we can solve this critical bug. > Signed-off-by: Ioan-Adrian Ratiu <adi(a)adirat.com> > --- > sound/usb/endpoint.c | 11 ++--------- > sound/usb/endpoint.h | 2 +- > sound/usb/pcm.c | 13 ++++++------- > 3 files changed, 9 insertions(+), 17 deletions(-) > > diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > index a2cdf3370afe..4465f324c2c2 100644 > --- a/sound/usb/endpoint.c > +++ b/sound/usb/endpoint.c > @@ -920,9 +920,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, > /** > * snd_usb_endpoint_start: start an snd_usb_endpoint > * > - * @ep: the endpoint to start > - * @can_sleep: flag indicating whether the operation is executed in > - * non-atomic context > + * @ep: the endpoint to start > * > * A call to this function will increment the use count of the endpoint. > * In case it is not already running, the URBs for this endpoint will be > @@ -932,7 +930,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, > * > * Returns an error if the URB submission failed, 0 in all other cases. > */ > -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep) > +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) > { > int err; > unsigned int i; > @@ -944,11 +942,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep) > if (++ep->use_count != 1) > return 0; > > - /* just to be sure */ > - deactivate_urbs(ep, false); > - if (can_sleep) > - wait_clear_urbs(ep); > - > ep->active_mask = 0; > ep->unlink_mask = 0; > ep->phase = 0; > diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h > index 6428392d8f62..584f295d7c77 100644 > --- a/sound/usb/endpoint.h > +++ b/sound/usb/endpoint.h > @@ -18,7 +18,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, > struct audioformat *fmt, > struct snd_usb_endpoint *sync_ep); > > -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep); > +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); > void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep); > void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); > int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index 34c6d4f2c0b6..db26f767f851 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -218,7 +218,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, > } > } > > -static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) > +static int start_endpoints(struct snd_usb_substream *subs) > { > int err; > > @@ -231,7 +231,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) > dev_dbg(&subs->dev->dev, "Starting data EP @%p\n", ep); > > ep->data_subs = subs; > - err = snd_usb_endpoint_start(ep, can_sleep); > + err = snd_usb_endpoint_start(ep); > if (err < 0) { > clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); > return err; > @@ -260,7 +260,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) > dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); > > ep->sync_slave = subs->data_endpoint; > - err = snd_usb_endpoint_start(ep, can_sleep); > + err = snd_usb_endpoint_start(ep); > if (err < 0) { > clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); > return err; > @@ -809,8 +809,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) > goto unlock; > } > > - snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); > - snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); > + stop_endpoints(subs, true); > > ret = set_format(subs, subs->cur_audiofmt); > if (ret < 0) > @@ -850,7 +849,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) > /* for playback, submit the URBs now; otherwise, the first hwptr_done > * updates for all URBs would happen at the same time when starting */ > if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) > - ret = start_endpoints(subs, true); > + return start_endpoints(subs); > > unlock: > snd_usb_unlock_shutdown(subs->stream->chip); > @@ -1666,7 +1665,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream > > switch (cmd) { > case SNDRV_PCM_TRIGGER_START: > - err = start_endpoints(subs, false); > + err = start_endpoints(subs); > if (err < 0) > return err; This patch works better, but not the best. It's a bit intrusive. What we need to synchronize process contexts to interrupt context till all of queued URBs are handled. This seems to be a main purpose of wait_clear_bits, while current implementation of snd_complete_urb()/wait_clear_bits() is not enough. Thanks for your reporting. Takashi Sakamoto -- 2.9.3

1 0

[alsa-devel] [PATCH v3 2/2] ASoC: cs43130: Add devicetree bindings for CS43130
by Li Xu 19 Dec '16

19 Dec '16

Add devicetree bindings documentation file for Cirrus Logic CS43130 codec. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs43130.txt | 41 ++++++++++++++++++++++ 1 file changed, 41 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs43130.txt diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt new file mode 100644 index 0000000..1af6b78 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs43130.txt @@ -0,0 +1,41 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : Active low GPIO used to reset the device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + +Example: + +cs43130: audio-codec@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 1>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; +}; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 2/2] ASoC: cs35l35: Add device tree documentation for CS35L35
by Li Xu 19 Dec '16

19 Dec '16

Add device tree documentation for Cirrus Logic CS35L35 speaker amplifier Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l35.txt | 172 +++++++++++++++++++++ 1 file changed, 172 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l35.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt new file mode 100644 index 0000000..8b13f67 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -0,0 +1,172 @@ +CS35L35 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l35" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - interrupt-parent : Specifies the phandle of the interrupt controller to + which the IRQs from CS35L35 are delivered to. + - interrupts : IRQ line info CS35L35. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + +Optional properties: + - cirrus,reset-gpios : Active low GPIO used to reset the amplifier + + - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a + Stereo configuration + + - cirrus,audio-channel : Set Location of Audio Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal + if 2 Devices share Boost BST_CTL + + - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength + Table 3-10 of the datasheet lists drive-strength specifications + 0 = 1x (Default) + 1 = .5x + + - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control + powers down with a rectification FET On or Off. If VSPK is supplied + externally then FET is off. + + - cirrus,boost-ctl-millivolt : Boost Converter control word. Step Size of 100mV + 0x00 = (Default) VP + 0x01 = 2600mV + 0x41 = 9000mV + + - cirrus,boost-ipk-milliamp : Boost-converter peak current limit. + Configures the peak current by monitoring the current through the boost FET. + Step size: 112mA + 0x00 = 1680mA + 0x19 = 4480mA + + - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change + zero-cross + +Optional H/G Algorithm sub-node: + + The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node + that will disable automatic control of the internal H/G Algorithm. + + It is strongly recommended that the Datasheet be referenced when adjusting + or using these Class H Algorithm controls over the internal Algorithm. + Serious damage can occur to the Device and surrounding components. + + - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm + See Section 4.3 Internal Class H Algorithm in the Datasheet. + If not used, the device manages the ClassH Algorithm internally. + +Optional properties for the "cirrus,classh-internal-algo" Sub-node + + Section 7.29 Class H Control + - cirrus,classh-bst-overide : Boolean + - cirrus,classh-bst-max-limit + - cirrus,classh-mem-depth + + Section 7.30 Class H Headroom Control + - cirrus,classh-headroom + + Section 7.31 Class H Release Rate + - cirrus,classh-release-rate + + Section 7.32 Class H Weak FET Drive Control + - cirrus,classh-wk-fet-disable + - cirrus,classh-wk-fet-delay + - cirrus,classh-wk-fet-thld + + Section 7.34 Class H VP Control + - cirrus,classh-vpch-auto + - cirrus,classh-vpch-rate + - cirrus,classh-vpch-man + +Optional Monitor Signal Format sub-node: + + The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node + for adjusting the Depth, Location and Frame of the Monitoring Signals + for Algorithms. + + See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet + + -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formating + on the I2S Port. Each of the 3 8 bit values in the array contain the settings + for depth, location, and frame. + + If not used, the defaults for the 6 monitor signals is used. + + Sections 7.44 - 7.53 lists values for the depth, location, and frame + for each monitoring signal. + + - cirrus,imon : 3 8 bit values to set the depth, location, and frame + of the IMON monitor signal. + + - cirrus,vmon : 3 8 bit values to set the depth, location, and frame + of the VMON monitor signal. + + - cirrus,vpmon : 3 8 bit values to set the depth, location, and frame + of the VPMON monitor signal. + + - cirrus,vbstmon : 3 8 bit values to set the depth, location, and frame + of the VBSTMON monitor signal + + - cirrus,vpbrstat : 3 8 bit values to set the depth, location, and frame + of the VPBRSTAT monitor signal + + - cirrus,zerofill : 3 8 bit values to set the depth, location, and frame\ + of the ZEROFILL packet in the monitor signal + +Example: + +cs35l35: audio-codec@20 { + compatible = "cirrus,cs35l35"; + reg = <0x20>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&axi_gpio 54 1>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + cirrus,boost-ctl = <0x41>; + + cirrus,stereo-config { + cirrus,audio-channel = <0x00>; + cirrus,advisory-channel = <0x01>; + cirrus,shared-boost; + }; + + cirrus,classh-internal-algo { + cirrus,classh-bst-overide; + cirrus,classh-bst-max-limit = <0x01>; + cirrus,classh-mem-depth = <0x01>; + cirrus,classh-release-rate = <0x08>; + cirrus,classh-headroom-millivolt = <0x0B>; + cirrus,classh-wk-fet-disable = <0x01>; + cirrus,classh-wk-fet-delay = <0x04>; + cirrus,classh-wk-fet-thld = <0x01>; + cirrus,classh-vpch-auto = <0x01>; + cirrus,classh-vpch-rate = <0x02>; + cirrus,classh-vpch-man = <0x05>; + }; + + /* Depth, Location, Frame */ + cirrus,monitor-signal-format { + cirrus,imon = /bits/ 8 <0x03 0x00 0x01>; + cirrus,vmon = /bits/ 8 <0x03 0x00 0x00>; + cirrus,vpmon = /bits/ 8 <0x03 0x04 0x00>; + cirrus,vbstmon = /bits/ 8 <0x03 0x04 0x01>; + cirrus,vpbrstat = /bits/ 8 <0x00 0x04 0x00>; + cirrus,zerofill = /bits/ 8 <0x00 0x00 0x00>; + }; + +}; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Don't overrun firmware file buffer when reading region data
by Richard Fitzgerald 19 Dec '16

19 Dec '16

Protect against corrupt firmware files by ensuring that the length we get for the data in a region actually lies within the available firmware file data buffer. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 593b7d1..91ccf62 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1551,7 +1551,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wmfw_region *region; const struct wm_adsp_region *mem; const char *region_name; - char *file, *text; + char *file, *text = NULL; struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; @@ -1700,10 +1700,21 @@ static int wm_adsp_load(struct wm_adsp *dsp) regions, le32_to_cpu(region->len), offset, region_name); + if ((pos + le32_to_cpu(region->len) + sizeof(*region)) > + firmware->size) { + adsp_err(dsp, + "%s.%d: %s region len %d bytes exceeds file length %d\n", + file, regions, region_name, + le32_to_cpu(region->len), firmware->size); + ret = -EINVAL; + goto out_fw; + } + if (text) { memcpy(text, region->data, le32_to_cpu(region->len)); adsp_info(dsp, "%s: %s\n", file, text); kfree(text); + text = NULL; } if (reg) { @@ -1748,6 +1759,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) regmap_async_complete(regmap); wm_adsp_buf_free(&buf_list); release_firmware(firmware); + kfree(text); out: kfree(file); @@ -2233,6 +2245,17 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { + if ((pos + le32_to_cpu(blk->len) + sizeof(*blk)) > + firmware->size) { + adsp_err(dsp, + "%s.%d: %s region len %d bytes exceeds file length %d\n", + file, blocks, region_name, + le32_to_cpu(blk->len), + firmware->size); + ret = -EINVAL; + goto out_fw; + } + buf = wm_adsp_buf_alloc(blk->data, le32_to_cpu(blk->len), &buf_list); -- 1.9.1

1 0

[alsa-devel] [PATCH v3] ASoC: Intel: bytcr_rt5640: fallback mechanism if MCLK is not enabled
by Pierre-Louis Bossart 19 Dec '16

19 Dec '16

Commit df1a2776a795 ("ASoC: Intel: bytcr_rt5640: add MCLK support") was merged but the corresponding clock framework patches have not, after being bumped from audio to clock to x86 domains. The missing clock-related patches result in a regression starting with 4.9 with the audio card not being created. Rather than reverting this commit and all following updates already queued up for 4.10, handle run-time dependency on MCLK and fall back to the previous bit-clock mode. This provides the same functionality as in 4.8 for Baytrail devices. On Baytrail-CR most devices remain silent with this fallback but additional patches are needed anyway. As suggested by Mark Brown, the fallback is only allowed with -ENOENT, all other run-time errors, including -EPROBE_DEFER, will stop the probe with no sound card registered. This patch should be applied to -stable as well as ASoC 4.10 fixes Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Notes: v3: fixed printf format warning reported by 0-day sound/soc/intel/boards/bytcr_rt5640.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 507a86a..e33e477 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -825,10 +825,20 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && (is_valleyview())) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { + ret_val = PTR_ERR(priv->mclk); + dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(priv->mclk)); - return PTR_ERR(priv->mclk); + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret_val); + + /* + * Fall back to bit clock usage for -ENOENT (clock not + * available likely due to missing dependencies), bail + * for all other errors, including -EPROBE_DEFER + */ + if (ret_val != -ENOENT) + return ret_val; + byt_rt5640_quirk &= ~BYT_RT5640_MCLK_EN; } } -- 2.7.4

2 1

Re: [alsa-devel] [PATCH 4/4] hda: thinkpad_helper: Add support for hid-lenovo LED control
by Dennis Wassenberg 19 Dec '16

19 Dec '16

Hi Jiri, On 19.12.2016 11:53, Jiri Kosina wrote: > On Mon, 19 Dec 2016, Dennis Wassenberg wrote: > >> "led_set_func_hid_lenovo" is set to "hid_lenovo_led_set" >> >> +#if IS_ENABLED(CONFIG_HID_LENOVO) >> +static int hda_fixup_thinkpad_hid_prepare(struct hda_codec *codec) >> +{ >> + struct hda_gen_spec *spec = codec->spec; >> + int ret = 0; >> + >> + if (!is_thinkpad(codec)) >> + return -ENODEV; >> + if (!led_set_func_hid_lenovo) >> + led_set_func_hid_lenovo = symbol_request(hid_lenovo_led_set); >> >> "hid_lenovo_led_set" is introduced in "[PATCH v2 2/2] hda: >> thinkpad_helper: Add support for hid-lenovo LED control". This means >> that "[PATCH 4/4] hda: thinkpad_helper: Add support for hid-lenovo LED >> control" can only work with applied "[PATCH 1/4] hid-lenovo: Add support >> for X1 Tablet cover special keys" and "[PATCH v2 2/2] hda: >> thinkpad_helper: Add support for hid-lenovo LED control". > > Ah, okay, I wasn't aware of the other patchset. These two probably should > be merged together. Is the other 2-patch set merged in any tree already? > Where has it been submitted to? > No 1/4, 2/4 and 3/4 are currently not merged to any tree. I reworked 1/4, 2/4 and 3/4 because of some comments from Benjamin Tissoires (mail from 14th September 2016). The patches are submitted to all recipients at the To, CC-Line (e.g. linux-input mailing list). Maybe I can merge 2/4 and 4/4 together. This is all for LED control. 1/4 is for key/input handling and 3/4 is a special FNLock handling where I should use a separate patch. Currently I have no feedback regarding the reworked linux-input specific code. I hope I will get it soon :) Best regards, Dennis

1 0

Re: [alsa-devel] [PATCH 4/4] hda: thinkpad_helper: Add support for hid-lenovo LED control
by Dennis Wassenberg 19 Dec '16

19 Dec '16

Hi Jiri, "led_set_func_hid_lenovo" is set to "hid_lenovo_led_set" +#if IS_ENABLED(CONFIG_HID_LENOVO) +static int hda_fixup_thinkpad_hid_prepare(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + int ret = 0; + + if (!is_thinkpad(codec)) + return -ENODEV; + if (!led_set_func_hid_lenovo) + led_set_func_hid_lenovo = symbol_request(hid_lenovo_led_set); "hid_lenovo_led_set" is introduced in "[PATCH v2 2/2] hda: thinkpad_helper: Add support for hid-lenovo LED control". This means that "[PATCH 4/4] hda: thinkpad_helper: Add support for hid-lenovo LED control" can only work with applied "[PATCH 1/4] hid-lenovo: Add support for X1 Tablet cover special keys" and "[PATCH v2 2/2] hda: thinkpad_helper: Add support for hid-lenovo LED control". Best regards, Dennis On 19.12.2016 11:09, Jiri Kosina wrote: > On Fri, 9 Dec 2016, Dennis Wassenberg wrote: > >> +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) >> + if (led_set_func_tpacpi) >> + led_set_func_tpacpi(TPACPI_LED_MUTE, !enabled); >> +#endif >> +#if IS_ENABLED(CONFIG_HID_LENOVO) >> + if (led_set_func_hid_lenovo) >> + led_set_func_hid_lenovo(HID_LENOVO_LED_MUTE, !enabled); > > Where does led_set_func_hid_lenovo() come from? I don't see it in Linus' > tree as of today. Is that something else than tpacpi_led_set()? > > Thanks, >

1 0

Re: [alsa-devel] [PATCH 2/4] hid-lenovo: Add support for X1 Tablet cover LED control
by Dennis Wassenberg 19 Dec '16

19 Dec '16

Hi Jiri, Thanks for your comments. I added comments in-lined. Best regards, Dennis On 19.12.2016 10:54, Jiri Kosina wrote: > Hi Dennis, > > thanks a lot for the patch. > > On Fri, 9 Dec 2016, Dennis Wassenberg wrote: > >> +int hid_lenovo_led_set(enum hid_lenovo_led_type led, bool on) >> +{ >> + struct led_classdev *dev = NULL; >> + struct lenovo_led_list_entry *entry; >> + >> + if (led >= HID_LENOVO_LED_MAX) >> + return -EINVAL; >> + >> + hid_lenovo_initial_leds[led] = on ? LED_FULL : LED_OFF; >> + >> + list_for_each_entry(entry, &hid_lenovo_leds, list) { >> + if (entry->type == led) { >> + dev = entry->dev; >> + break; >> + } >> + } > > How exactly is this synchronized against lenovo_remove_tpx1cover()? > In case of that the tpx1cover is disconnected it will be removed from hid_lenovo_leds list. That means not included at the hid_lenovo_leds list. In this case dev is still NULL and the function will return -ENODEV. The static array hid_lenovo_initial_leds is still set to store the current state of a LED type. This is used to set the LED appropriately if the tpx1cover is replugged. Maybe I should add a mutex to protect the hid_lenovo_leds list operations in hid-lenovo.c to fix the case if a unplug occurred concurrently to setting an led?! >> + >> + if (!dev) >> + return -ENODEV; >> + >> + if (!dev->brightness_set) >> + return -ENODEV; >> + >> + dev->brightness_set(dev, on ? LED_FULL : LED_OFF); >> + >> + return 0; >> +} >> +EXPORT_SYMBOL_GPL(hid_lenovo_led_set); > > Does this really need to be exported to the whole universe? (I guess this > will be further discussed in 4/4). No, it is sufficient if thinkpad-helper can access it. >

1 0

[alsa-devel] [PATCH 0/2] ASoC: Add driver for CX2072X CODEC
by simon.ho.cnxt＠gmail.com 17 Dec '16

17 Dec '16

From: Simon Ho <simon.ho(a)conexant.com> Re-send the patch because my previous email was sent by a non-member email account and are being held. So please disregard my previous email if you saw it. Sorry if I caused any trouble. This patch adds support for Conexant CX2072X codec driver. The CX2072X is a ultra low power stereo audio codec supports I2S/TDM host interface with EQ, DRC features in playback mode. Featues of CX2072X codec: * Two 24 bits DACs and DACs. * Stereo Headphone AMP. * 2.8W per channel class-D output. * Integrated seven bands per channel EQ and DRC. * Fully integrated headset support with detect/switch. * Stereo digital microphone for array applications. * S/PDIF output. * Bi-directional GPIOs. * Support analog and digital PC Beeep. * One headset button support. * Supports a wide variety of host interfaces. -wide variety of I2S and similar bit-stream formats with word lengths of up to 24 bits. -TDM stream supports up to 4 channels. * AEC loopback support. Simon Ho (2): ASoC: cx2072x: Add DT bingings documentation for CX2072X CODEC ASoC: cx2072x Add driver for CX2072X CODEC .../devicetree/bindings/sound/cx2072x.txt | 36 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2080 ++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 322 +++ 5 files changed, 2445 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cx2072x.txt create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h -- 2.7.4

1 1

[alsa-devel] USB Driver Development Sinn 7 Status 24|96 (0x200c:0x1006)
by Marc Streckfuß 17 Dec '16

17 Dec '16

Hello Guys, I apologize in advance if this is the wrong mailing list (In general I am new to Mailing Lists), but I thought since I plan to develop a driver this might fit, I'm just not completely done. == Device Information The Device in Question is from the Brand "Sinn 7" with the name "Status 24|96", it's a relatively unknown recording interface design to be used in DAWs on Windows using Steinberg's Proprietary ASIO Interface. This is to provide a low-latency interface on Windows but it seems to be support a regular interface as well (at least you can use it as regular windows speaker and have different options there). It seems to be manufactured by Reloop, since that's the vendorId 0x200c. I tried the snd-usb-audio quirk for the Reloop Play already, but it didn't work (Applying the Quirk seemed to fail, due to endpoint mismatch, etc). >From reverse engineering the windows driver a bit I found the name " usb-audio.de". It's a website of the Pylotec GmbH which seems to provide the base boards for USB Audio Hardware used in many Interfaces (They have a page where they mention what is using their usb audio driver) The interesting lsusb output is this (I omitted the "empty" altsetting 0 for both interfaces, all classes are 0xFF == Vendor Specific): Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 3 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 == My Findings So after days of trying out different quirks, building my own version of the snd-usb-audio and endless research I found out some stuff. A) I have captured the init sequence and I am able to redo it using libusb B) It turns out that there are 3 permanent communication types, even when I disable the device in Windows: There are writes to the Isochronous Endpoint 0x2 containing 32 Sub-Packets, so that is 128 Bytes of 0x0. Oddly enough that isn't the Audio data!! There are writes to the Bulk Endpoint 0x5 containing a variable size ranging roughly from ~16kB to 20kB. These are filled mostly with 0x0, but periodically (every 200h Bytes?) I had a 0xFD 0xFF Sequence. Those actually contain the Audio Data. I was able to produce some knocking sound when sending random values (either 0x0 or 0x1!) here. [I was expecting a white noise, though] => This is where I am stuck. I understand the idea of PCM in General, but I would expect it to send [0, 0xFF] values and not just 0x0 or 0x1, since that is a bandwidth waste. This and I would also expect a constant packet size. If it is variable, does that mean that the Sample Rate changes? (Note: I confirmed the 0x0..0x1 and the variable packet size both: By using libusb/pyUSB and by sniffing the traffic of a virtual windows box with Wireshark) There are reads from the Bulk Endpoint 0x86 (Note: Different InterfaceNumber, but it seems to work without changing the interface??) which result in an array of values ranging from 0x0 to 0x3. (Maybe that would apply to the output as well and during sniffing I just had the test sound too low gained?). Most of the time their size seems to be in line with the output bulk packets. Some of the sent sizes (data fields only) are: 19'986Bytes, 16'384 Bytes, 12'800 Bytes however most of the time it's 16k, whereas the output tends to go to 19k and the input to 12k (I have nothing attached to the input connectors). Now what I actually want to know: Does that protocol sound familiar to you? Do you have a clue on how to patch it in as a simple quirk? I was thinking maybe I can use a setup quirk and then let the FIXED_ENDPOINT do the rest? However I am stuck with the Format, I don't know whether that really is a known format or if I need some conversion and how it is with some synchronization clock and such. Basically I need information on how to proceed. - Marc Streckfuß

1 0

[alsa-devel] 1.1.3 release?
by Takashi Iwai 16 Dec '16

16 Dec '16

Hi Jaroslav, shall we release v1.1.3 now? We've gathered quite a lot of changes since the last release. thanks, Takashi

4 5

[alsa-devel] [alsa-lib][PATCH] ASoC: hdmi-codec: use unsigned type to structure members with bit-field instead of signed type
by Takashi Sakamoto 16 Dec '16

16 Dec '16

This is a fix for Linux 4.10-rc1. In C language specification, a bit-field is interpreted as a signed or unsigned integer type consisting of the specified number of bits. In GCC manual, the range of a signed bit field of N bits is from -(2^N) / 2 to ((2^N) / 2) - 1 https://www.gnu.org/software/gnu-c-manual/gnu-c-manual.html#Bit-Fields Therefore, when defined as 1 bit-field with signed type, variables can represents -1 and 0. The snd-soc-hdmi-codec module includes a structure which has signed type members with bit-fields. Codes of this module assign 0 and 1 to the members. This seems to result in implementation-dependent behaviours. As of v4.10-rc1 merge window, outside of sound subsystem, this structure is referred by below GPU modules. - tda998x - sti-drm - mediatek-drm-hdmi - msm As long as I review their codes relevant to the structure, the structure members are used just for condition statements and printk formats. My proposal of change is a bit intrusive to the printk formats but this may be acceptable. Totally, it's reasonable to use unsigned type for the structure members. This bug is detected by Sparse, static code analyzer with below warnings. ./include/sound/hdmi-codec.h:39:26: error: dubious one-bit signed bitfield ./include/sound/hdmi-codec.h:40:28: error: dubious one-bit signed bitfield ./include/sound/hdmi-codec.h:41:29: error: dubious one-bit signed bitfield ./include/sound/hdmi-codec.h:42:31: error: dubious one-bit signed bitfield CC: Jyri Sarha <jsarha(a)ti.com> CC: Jie Qiu <jie.qiu(a)mediatek.com> CC: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> CC: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> CC: dri-devel(a)lists.freedesktop.org CC: stable(a)vger.kernel.org Fixes: 09184118a8ab ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/hdmi-codec.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 530c57b..915c435 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -36,10 +36,10 @@ struct hdmi_codec_daifmt { HDMI_AC97, HDMI_SPDIF, } fmt; - int bit_clk_inv:1; - int frame_clk_inv:1; - int bit_clk_master:1; - int frame_clk_master:1; + unsigned int bit_clk_inv:1; + unsigned int frame_clk_inv:1; + unsigned int bit_clk_master:1; + unsigned int frame_clk_master:1; }; /* -- 2.9.3

3 4

[alsa-devel] [PATCH] ASoC: da7218: Set DAI output pin high impedance when not in use
by Adam Thomson 16 Dec '16

16 Dec '16

For TDM mode, the I2S data out line can be shared between mutliple codecs. In this scenario, only the active codec should be using the line, and all others should be high impedance. However, currently in the driver this configuration isn't set when capture is inactive, and the line remains driven. This patch updates the AIF_OUT widget to set the DAI output pin of the device as high impedance when not in use. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7218.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index c69e976..d256ebf 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1634,7 +1634,8 @@ static int da7218_hp_pga_event(struct snd_soc_dapm_widget *w, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* DAI */ - SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, DA7218_DAI_TDM_CTRL, + DA7218_DAI_OE_SHIFT, DA7218_NO_INVERT), SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0), /* Output Mixers */ -- 1.9.3

2 1

[alsa-devel] [PATCH] tinycompress: rename crec usages
by Vinod Koul 16 Dec '16

16 Dec '16

The creord utility help listed crec as the utility name, change that to crecord and also update the program explanation text. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- src/utils/crecord.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/src/utils/crecord.c b/src/utils/crecord.c index 8d5b7b0b2c30..1ba54d70e89c 100644 --- a/src/utils/crecord.c +++ b/src/utils/crecord.c @@ -1,7 +1,7 @@ /* * BSD LICENSE * - * crec command line recorder for compress audio record in alsa + * crecord command line recorder for compress audio record in alsa * Copyright (c) 2011-2012, Intel Corporation * Copyright (c) 2013-2014, Wolfson Microelectronic Ltd. * All rights reserved. @@ -35,7 +35,7 @@ * * LGPL LICENSE * - * crec command line recorder for compress audio record in alsa + * crecord command line recorder for compress audio record in alsa * Copyright (c) 2011-2012, Intel Corporation * Copyright (c) 2013-2014, Wolfson Microelectronic Ltd. * @@ -153,7 +153,7 @@ static void size_wave_header(struct wave_header *header, uint32_t size) static void usage(void) { - fprintf(stderr, "usage: crec [OPTIONS] [filename]\n" + fprintf(stderr, "usage: crecord [OPTIONS] [filename]\n" "-c\tcard number\n" "-d\tdevice node\n" "-b\tbuffer size\n" @@ -167,8 +167,8 @@ static void usage(void) "If filename is not given the output is\n" "written to stdout\n\n" "Example:\n" - "\tcrec -c 1 -d 2 test.wav\n" - "\tcrec -f 5 test.wav\n", + "\tcrecord -c 1 -d 2 test.wav\n" + "\tcrecord -f 5 test.wav\n", DEFAULT_CHANNELS, DEFAULT_RATE); exit(EXIT_FAILURE); -- 1.9.1

1 0

[alsa-devel] [PATCH] tinycompress: rename crec.c
by Vinod Koul 16 Dec '16

16 Dec '16

Since the compress recorder utility is called crecord, it makes it apt to have src file named crecord.c. Reported-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- src/utils/Makefile.am | 2 +- src/utils/{crec.c => crecord.c} | 0 2 files changed, 1 insertion(+), 1 deletion(-) rename src/utils/{crec.c => crecord.c} (100%) diff --git a/src/utils/Makefile.am b/src/utils/Makefile.am index 5f685e3b2ffe..1b996d430f21 100644 --- a/src/utils/Makefile.am +++ b/src/utils/Makefile.am @@ -1,7 +1,7 @@ bin_PROGRAMS = cplay crecord cplay_SOURCES = cplay.c -crecord_SOURCES = crec.c +crecord_SOURCES = crecord.c cplay_CFLAGS = -I$(top_srcdir)/include crecord_CFLAGS = -I$(top_srcdir)/include diff --git a/src/utils/crec.c b/src/utils/crecord.c similarity index 100% rename from src/utils/crec.c rename to src/utils/crecord.c -- 1.9.1

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: bytcr_rt5640: fallback mechanism if MCLK is not enabled
by Pierre-Louis Bossart 16 Dec '16

16 Dec '16

Commit df1a2776a795 ("ASoC: Intel: bytcr_rt5640: add MCLK support") was merged but the corresponding clock framework patches have not, after being bumped from audio to clock to x86 domains. The missing clock-related patches result in a regression starting with 4.9 with the audio card not being created. Rather than reverting this commit and all following updates already queued up for 4.10, handle run-time dependency on MCLK and fall back to the previous bit-clock mode. This provides the same functionality as in 4.8 for Baytrail devices. On Baytrail-CR most devices remain silent with this fallback but additional patches are needed anyway. As suggested by Mark Brown, the fallback is only allowed with -ENOENT, all other run-time errors, including -EPROBE_DEFER, will stop the probe with no sound card registered. This patch should be applied to -stable as well as ASoC 4.10 fixes Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 507a86a..7d66127 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -825,10 +825,20 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && (is_valleyview())) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { + ret_val = PTR_ERR(priv->mclk); + dev_err(&pdev->dev, "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(priv->mclk)); - return PTR_ERR(priv->mclk); + ret_val); + + /* + * Fall back to bit clock usage for -ENOENT (clock not + * available likely due to missing dependencies), bail + * for all other errors, including -EPROBE_DEFER + */ + if (ret_val != -ENOENT) + return ret_val; + byt_rt5640_quirk &= ~BYT_RT5640_MCLK_EN; } } -- 2.7.4

2 2

[alsa-devel] [bug report] ASoC: rsnd: call missing snd_ctl_remove()
by Dan Carpenter 16 Dec '16

16 Dec '16

Hello Kuninori Morimoto, The patch d1f83d6ef43b: "ASoC: rsnd: call missing snd_ctl_remove()" from Feb 2, 2015, leads to the following static checker warning: sound/soc/sh/rcar/core.c:1034 __rsnd_kctrl_new() error: double free of 'kctrl' sound/soc/sh/rcar/core.c 1026 int ret; 1027 1028 kctrl = snd_ctl_new1(&knew, mod); 1029 if (!kctrl) 1030 return -ENOMEM; 1031 1032 ret = snd_ctl_add(card, kctrl); 1033 if (ret < 0) { 1034 snd_ctl_free_one(kctrl); The snd_ctl_add() function frees kctrl on error. It's a confusing layering violation IMHO but there are a lot of callers so I'm too lazy to clean this up myself... 1035 return ret; 1036 } 1037 regards, dan carpenter

2 1

Re: [alsa-devel] [PATCH] fix get_stop_threshold alsa-lib 0.9 alias
by Takashi Iwai 15 Dec '16

15 Dec '16

On Tue, 13 Dec 2016 13:54:26 +0100, Rene Rebe wrote: > > Hi there, > > resurrecting some decade old code I got some crashes and noticed a typo in > an old 0.9 legacy symver alias: > > Signed-off-by: René Rebe <rene(a)exactcode.com> A good catch, applied now. Thanks. Takashi > > --- alsa-lib-1.1.2/include/pcm_old.h.vanilla 2016-12-12 19:27:34.119596843 +0100 > +++ alsa-lib-1.1.2/include/pcm_old.h 2016-12-12 19:28:02.039596717 +0100 > @@ -206,7 +206,7 @@ > asm(".symver snd_pcm_sw_params_get_avail_min,snd_pcm_sw_params_get_avail_min(a)ALSA_0.9"); > asm(".symver snd_pcm_sw_params_get_xfer_align,snd_pcm_sw_params_get_xfer_align(a)ALSA_0.9"); > asm(".symver snd_pcm_sw_params_get_start_threshold,snd_pcm_sw_params_get_start_threshold(a)ALSA_0.9"); > -asm(".symver snd_pcm_sw_params_get_stop_threshold,snd_pcm_sw_params_set_stop_threshold(a)ALSA_0.9"); > +asm(".symver snd_pcm_sw_params_get_stop_threshold,snd_pcm_sw_params_get_stop_threshold(a)ALSA_0.9"); > asm(".symver snd_pcm_sw_params_get_silence_threshold,snd_pcm_sw_params_get_silence_threshold(a)ALSA_0.9"); > asm(".symver snd_pcm_sw_params_get_silence_size,snd_pcm_sw_params_get_silence_size(a)ALSA_0.9"); > > > -- > René Rebe, ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin > http://exactcode.com | http://t2-project.org | http://rene.rebe.de >

1 0

[alsa-devel] [PATCH 0/7] Add support for Legacy HDMI audio drivers
by Jerome Anand 15 Dec '16

15 Dec '16

Legacy (CherryTrail/ Baytrail) HDMI audio drivers added Legacy hdmi audio-Gfx interaction/ interfacing is updated to use irq chip framework Jerome Anand (7): drm/i915: setup bridge for HDMI LPE audio driver drm/i915: Add support for audio driver notifications ALSA: add shell for Intel HDMI LPE audio driver ALSA: x86: hdmi: Add audio support for BYT and CHT ALSA: x86: hdmi: Improve position reporting ALSA: x86: hdmi: Fixup some monitor ALSA: x86: hdmi: continue playback even when display resolution changes Documentation/gpu/i915.rst | 9 + drivers/gpu/drm/i915/Makefile | 3 + drivers/gpu/drm/i915/i915_drv.c | 8 +- drivers/gpu/drm/i915/i915_drv.h | 18 + drivers/gpu/drm/i915/i915_irq.c | 27 + drivers/gpu/drm/i915/i915_reg.h | 3 + drivers/gpu/drm/i915/intel_audio.c | 8 + drivers/gpu/drm/i915/intel_hdmi.c | 1 + drivers/gpu/drm/i915/intel_lpe_audio.c | 409 +++++++ include/drm/intel_lpe_audio.h | 46 + sound/Kconfig | 2 + sound/Makefile | 2 +- sound/x86/Kconfig | 16 + sound/x86/Makefile | 10 + sound/x86/intel_hdmi_audio.c | 1932 ++++++++++++++++++++++++++++++++ sound/x86/intel_hdmi_audio.h | 201 ++++ sound/x86/intel_hdmi_audio_if.c | 551 +++++++++ sound/x86/intel_hdmi_lpe_audio.c | 626 +++++++++++ sound/x86/intel_hdmi_lpe_audio.h | 692 ++++++++++++ 19 files changed, 4561 insertions(+), 3 deletions(-) create mode 100644 drivers/gpu/drm/i915/intel_lpe_audio.c create mode 100644 include/drm/intel_lpe_audio.h create mode 100644 sound/x86/Kconfig create mode 100644 sound/x86/Makefile create mode 100644 sound/x86/intel_hdmi_audio.c create mode 100644 sound/x86/intel_hdmi_audio.h create mode 100644 sound/x86/intel_hdmi_audio_if.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.h -- 2.9.3 base-commit: f6a248e2507f98d7eb1f4fec8cfcbf685a33d289

7 37

[alsa-devel] [PATCH 0/5] ASoC/arizona: Ensure pin searches use widget name prefix
by Richard Fitzgerald 15 Dec '16

15 Dec '16

The name of a codec pin can have an optional prefix string, which is defined by the audio machine driver. The snd_soc_dapm_x_pin functions take the fully-specified name including the prefix and so the existing code would fail to find the pin if the audio machine driver had added a prefix. This patch chain adds new helper functions that take a non-prefixed name for a specific ASoC component and internally add that component's prefix. The other patches update the arizona drivers to use these new functions. Richard Fitzgerald (5): ASoC: core: Add component pin control functions ASoC: arizona: Use component pin control functions regulator: arizona-micsupp: Use SoC component pin control functions extcon: arizona: Use SoC component pin control functions Input: arizona-haptics - Use SoC component pin control functions drivers/extcon/extcon-arizona.c | 8 +- drivers/input/misc/arizona-haptics.c | 13 ++- drivers/regulator/arizona-micsupp.c | 6 +- include/sound/soc.h | 25 +++++ sound/soc/codecs/arizona.c | 13 ++- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/soc-utils.c | 199 +++++++++++++++++++++++++++++++++++ 10 files changed, 254 insertions(+), 18 deletions(-) -- 1.9.1

4 11

[alsa-devel] [PATCH] ASoC: add Component level pcm_new/pcm_free
by Kuninori Morimoto 15 Dec '16

15 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> In current ALSA SoC, Platform only has pcm_new/pcm_free feature, but it should be supported on Component level. This patch adds it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 21 +++++++++++++++++++++ sound/soc/soc-pcm.c | 32 +++++++++++++++++++++++--------- 3 files changed, 50 insertions(+), 9 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6..e580a67 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -785,6 +785,10 @@ struct snd_soc_component_driver { int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); + /* pcm creation and destruction */ + int (*pcm_new)(struct snd_soc_pcm_runtime *); + void (*pcm_free)(struct snd_pcm *); + /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, @@ -858,6 +862,8 @@ struct snd_soc_component { void (*remove)(struct snd_soc_component *); int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); + int (*pcm_new)(struct snd_soc_pcm_runtime *); + void (*pcm_free)(struct snd_pcm *); /* machine specific init */ int (*init)(struct snd_soc_component *component); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb..981443e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2976,6 +2976,8 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->remove = component->driver->remove; component->suspend = component->driver->suspend; component->resume = component->driver->resume; + component->pcm_new = component->driver->pcm_new; + component->pcm_free= component->driver->pcm_free; dapm = &component->dapm; dapm->dev = dev; @@ -3158,6 +3160,21 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } +static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_platform *platform = rtd->platform; + + return platform->driver->pcm_new(rtd); +} + +static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_soc_platform *platform = rtd->platform; + + platform->driver->pcm_free(pcm); +} + /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -3181,6 +3198,10 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; + if (platform_drv->pcm_new) + platform->component.pcm_new = snd_soc_platform_drv_pcm_new; + if (platform_drv->pcm_free) + platform->component.pcm_free = snd_soc_platform_drv_pcm_free; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7a1eaa..a9ef8ae 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2640,12 +2640,25 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } +static void soc_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_soc_component *component; + + list_for_each_entry(component, &rtd->card->component_dev_list, + card_list) { + if (component->pcm_free) + component->pcm_free(pcm); + } +} + /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_component *component; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; @@ -2754,17 +2767,18 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); - if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd); - if (ret < 0) { - dev_err(platform->dev, - "ASoC: pcm constructor failed: %d\n", - ret); - return ret; + list_for_each_entry(component, &rtd->card->component_dev_list, card_list) { + if (component->pcm_new) { + ret = component->pcm_new(rtd); + if (ret < 0) { + dev_err(component->dev, + "ASoC: pcm constructor failed: %d\n", + ret); + return ret; + } } } - - pcm->private_free = platform->driver->pcm_free; + pcm->private_free = soc_pcm_free; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: wm0010: Remove unneeded linux/miscdevice.h include
by Corentin Labbe 15 Dec '16

15 Dec '16

sound/soc/codecs/wm0010.c does not use any miscdevice so this patch remove this unnecessary inclusion. Signed-off-by: Corentin Labbe <clabbe.montjoie(a)gmail.com> --- sound/soc/codecs/wm0010.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 0eb5dcf..4f5f571 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -21,7 +21,6 @@ #include <linux/firmware.h> #include <linux/delay.h> #include <linux/fs.h> -#include <linux/miscdevice.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> #include <linux/mutex.h> -- 2.10.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5514-spi: Remove unneeded linux/miscdevice.h include
by Corentin Labbe 15 Dec '16

15 Dec '16

sound/soc/codecs/rt5514-spi.c does not use any miscdevice so this patch remove this unnecessary inclusion. Signed-off-by: Corentin Labbe <clabbe.montjoie(a)gmail.com> --- sound/soc/codecs/rt5514-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 0901e25..7ed62e8 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -21,7 +21,6 @@ #include <linux/gpio.h> #include <linux/sched.h> #include <linux/uaccess.h> -#include <linux/miscdevice.h> #include <linux/regulator/consumer.h> #include <linux/pm_qos.h> #include <linux/sysfs.h> -- 2.10.2

1 1

Re: [alsa-devel] [PATCH 0/7] Move dell-led to drivers/platform/x86
by Michał Kępień 15 Dec '16

15 Dec '16

> On 12 December 2016 at 01:13, <Mario.Limonciello(a)dell.com> wrote: > > > add Anthony Wong @ Canonical. > > > > I'm about to go on break and won't have access to hardware for a while. > > > > Anthony, > > > > Your team has this hardware more readily available than I do. Would you be > > able to have someone on the team verify this series? > > > Sorry, just back from vacation. > > Yes, we can test it here some time next week. That is great news, thanks! I will try to post a v2 of this patch series within the next few days, so hopefully it will be ready before you get around to testing. I will keep you in the loop. -- Best regards, Michał Kępień

1 0

[alsa-devel] [PATCH] ASoC: rsnd: don't double free kctrl
by Colin King 15 Dec '16

15 Dec '16

From: Colin Ian King <colin.king(a)canonical.com> On an error, snd_ctl_add already free's kctrl, so calling snd_ctl_free_one to free it again leads to a double free error. Fix this by removing the extraneous snd_ctl_free_one call. Issue found using static analysis with CoverityScan, CID 1372908 Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rcar/core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4bd68de..99b5b08 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1030,10 +1030,8 @@ static int __rsnd_kctrl_new(struct rsnd_mod *mod, return -ENOMEM; ret = snd_ctl_add(card, kctrl); - if (ret < 0) { - snd_ctl_free_one(kctrl); + if (ret < 0) return ret; - } cfg->update = update; cfg->card = card; -- 2.10.2

3 2

[alsa-devel] [PATCH 00/14] ASoC: Compile fixes with W=1
by Vinod Koul 15 Dec '16

15 Dec '16

We seem to have few warnings in ASoC (I have tried only one config today) when we compile with W=1. So this is first set to fix these warning, which predominantly are unused but initialized variables. Few instances actually helped remove a bit more code as well. Vinod Koul (14): ASoC: Intel: Skylake: remove unused 'ret' ASoC: Intel: sst: remove unused 'msg_high' ASoC: Intel: sst: remove unused 'ops' ASoC: topology: remove unused 'err' ASoC: hdac_hdmi: remove unused 'dai_map' ASoC: max98090: remove superflous check for 'micbias' ASoC: AMD: remove unused ‘dma_buffer’ ASoC: adau17x1: remove unused ‘ret’ ASoC: max9867: remove unused ‘ret’ ASoC: pcm3168a: remove unused ‘format’ ASoC: img: remove unused ‘format’ ASoC: Intel: sst: remove unused ‘ret_val’ ASoC: samsung: smdk_wm8580: remove unused ‘bfs’ ASoC: zx296702-i2s: remove unused ‘format’ sound/soc/amd/acp-pcm-dma.c | 3 --- sound/soc/codecs/adau17x1.c | 3 +-- sound/soc/codecs/hdac_hdmi.c | 5 ----- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max9867.c | 5 ++--- sound/soc/codecs/pcm3168a.c | 2 -- sound/soc/img/img-parallel-out.c | 2 -- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 ++-- sound/soc/intel/atom/sst/sst_ipc.c | 2 -- sound/soc/intel/atom/sst/sst_stream.c | 2 -- sound/soc/intel/skylake/skl-topology.c | 3 +-- sound/soc/samsung/smdk_wm8580.c | 5 +---- sound/soc/soc-topology.c | 4 ++-- sound/soc/zte/zx296702-i2s.c | 5 +---- 14 files changed, 11 insertions(+), 36 deletions(-) -- 1.9.1

4 25

[alsa-devel] [PATCH] ASoC: topology: kfree kcontrol->private_value before freeing kcontrol
by Colin King 15 Dec '16

15 Dec '16

From: Colin Ian King <colin.king(a)canonical.com> kcontrol->private_value is being kfree'd after kcontrol has been freed (in previous call to snd_ctl_remove). Instead, fix this by kfreeing the private_value before kcontrol. CoverityScan CID#1388311 "Read from pointer after free" Fixes: eea3dd4f1247a ("ASoC: topology: Only free TLV for volume mixers of a widget") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 11feb19..b5472e0 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -514,13 +514,12 @@ static void remove_widget(struct snd_soc_component *comp, == SND_SOC_TPLG_TYPE_MIXER) kfree(kcontrol->tlv.p); - snd_ctl_remove(card, kcontrol); - /* Private value is used as struct soc_mixer_control * for volume mixers or soc_bytes_ext for bytes * controls. */ kfree((void *)kcontrol->private_value); + snd_ctl_remove(card, kcontrol); } kfree(w->kcontrol_news); } -- 2.10.2

2 1

[alsa-devel] [PATCH] ASoC: rt5665: Make SND_SOC_RT5665 entry sort in Kconfig and Makefile
by Axel Lin 15 Dec '16

15 Dec '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 10 +++++----- sound/soc/codecs/Makefile | 4 ++-- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e1718a..cfc108e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -117,8 +117,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5651 if I2C select SND_SOC_RT5659 if I2C select SND_SOC_RT5660 if I2C - select SND_SOC_RT5665 if I2C select SND_SOC_RT5663 if I2C + select SND_SOC_RT5665 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -668,8 +668,8 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y default y if SND_SOC_RT5660=y - default y if SND_SOC_RT5665=y default y if SND_SOC_RT5663=y + default y if SND_SOC_RT5665=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -679,8 +679,8 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5651=m default m if SND_SOC_RT5659=m default m if SND_SOC_RT5660=m - default m if SND_SOC_RT5665=m default m if SND_SOC_RT5663=m + default m if SND_SOC_RT5665=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -728,10 +728,10 @@ config SND_SOC_RT5659 config SND_SOC_RT5660 tristate -config SND_SOC_RT5665 +config SND_SOC_RT5663 tristate -config SND_SOC_RT5663 +config SND_SOC_RT5665 tristate config SND_SOC_RT5670 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7e1dad7..2624c73 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -118,8 +118,8 @@ snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5659-objs := rt5659.o snd-soc-rt5660-objs := rt5660.o -snd-soc-rt5665-objs := rt5665.o snd-soc-rt5663-objs := rt5663.o +snd-soc-rt5665-objs := rt5665.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -346,8 +346,8 @@ obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5659) += snd-soc-rt5659.o obj-$(CONFIG_SND_SOC_RT5660) += snd-soc-rt5660.o -obj-$(CONFIG_SND_SOC_RT5665) += snd-soc-rt5665.o obj-$(CONFIG_SND_SOC_RT5663) += snd-soc-rt5663.o +obj-$(CONFIG_SND_SOC_RT5665) += snd-soc-rt5665.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o -- 2.9.3

2 1

[alsa-devel] [PATCH 1/1 linux-next] ASoC: samsung: include gpio consumer.h
by Fabian Frederick 15 Dec '16

15 Dec '16

Fix the following build errors sound/soc/samsung/tm2_wm5110.c:220:3: error: implicit declaration of function 'gpiod_set_value_cansleep' [-Werror=implicit-function-declaration] sound/soc/samsung/tm2_wm5110.c:438:24: error: implicit declaration of function 'devm_gpiod_get' [-Werror=implicit-function-declaration] Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/soc/samsung/tm2_wm5110.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 5cdf7d1..24cc9d6 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -12,6 +12,7 @@ #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <sound/pcm_params.h> -- 2.7.4

3 4

[alsa-devel] [PATCH V2 2/2] ASoC: samsung: add GPIOLIB dependency
by Fabian Frederick 15 Dec '16

15 Dec '16

Both SND_SOC_SMARTQ and SND_SOC_SAMSUNG_TM2_WM5110 use gpio/consumer.h This patch adds GPIOLIB || COMPILE_TEST to Kconfig entries to avoid any build errors. See commit 638f958baeaf ("extcon: Allow compile test of GPIO consumers if !GPIOLIB") for similar problem and explanations. Reported-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/soc/samsung/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7c42315..f1f1d79 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -111,6 +111,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" depends on MACH_SMARTQ || COMPILE_TEST + depends on GPIOLIB || COMPILE_TEST depends on I2C select SND_SAMSUNG_I2S select SND_SOC_WM8750 @@ -193,6 +194,7 @@ config SND_SOC_ARNDALE_RT5631_ALC5631 config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on GPIOLIB || COMPILE_TEST select SND_SOC_MAX98504 select SND_SOC_WM5110 select SND_SAMSUNG_I2S -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: samsung: Remove tests of member address
by Krzysztof Kozlowski 15 Dec '16

15 Dec '16

The driver was checking for non-NULL address of struct's members: - s3c_audio_pdata->type (union), - s3c_audio_pdata->type.i2s (embedded struct). This is pointless as these will be always non-NULL. The 's3c_audio_pdata' is always initialized in static memory so it will be zeroed. Additionally the 'type' member was an union with only one member. It is safe to reorganize the structures to get rid of useless union and checks for addresses to fix the coccinelle warning: >> sound/soc/samsung/i2s.c:1270:2-4: ERROR: test of a variable/field address Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Not tested on the hardware. --- arch/arm/mach-s3c64xx/dev-audio.c | 4 +--- include/linux/platform_data/asoc-s3c.h | 6 ++---- sound/soc/samsung/i2s.c | 10 ++-------- 3 files changed, 5 insertions(+), 15 deletions(-) diff --git a/arch/arm/mach-s3c64xx/dev-audio.c b/arch/arm/mach-s3c64xx/dev-audio.c index b57783371d52..247dcc0b691e 100644 --- a/arch/arm/mach-s3c64xx/dev-audio.c +++ b/arch/arm/mach-s3c64xx/dev-audio.c @@ -106,9 +106,7 @@ static struct s3c_audio_pdata i2sv4_pdata = { .dma_playback = DMACH_HSI_I2SV40_TX, .dma_capture = DMACH_HSI_I2SV40_RX, .type = { - .i2s = { - .quirks = QUIRK_PRI_6CHAN, - }, + .quirks = QUIRK_PRI_6CHAN, }, }; diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 15bf56ee8af7..90641a5daaf0 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -18,7 +18,7 @@ extern void s3c64xx_ac97_setup_gpio(int); -struct samsung_i2s { +struct samsung_i2s_type { /* If the Primary DAI has 5.1 Channels */ #define QUIRK_PRI_6CHAN (1 << 0) /* If the I2S block has a Stereo Overlay Channel */ @@ -47,7 +47,5 @@ struct s3c_audio_pdata { void *dma_capture; void *dma_play_sec; void *dma_capture_mic; - union { - struct samsung_i2s i2s; - } type; + struct samsung_i2s_type type; }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index e00974bc5616..d55326289a4a 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1218,7 +1218,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) { struct i2s_dai *pri_dai, *sec_dai = NULL; struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; - struct samsung_i2s *i2s_cfg = NULL; struct resource *res; u32 regs_base, quirks = 0, idma_addr = 0; struct device_node *np = pdev->dev.of_node; @@ -1267,13 +1266,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.filter_data = i2s_pdata->dma_capture; pri_dai->filter = i2s_pdata->dma_filter; - if (&i2s_pdata->type) - i2s_cfg = &i2s_pdata->type.i2s; - - if (i2s_cfg) { - quirks = i2s_cfg->quirks; - idma_addr = i2s_cfg->idma_addr; - } + quirks = i2s_pdata->type.quirks; + idma_addr = i2s_pdata->type.idma_addr; } else { quirks = i2s_dai_data->quirks; if (of_property_read_u32(np, "samsung,idma-addr", -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: wm8753: Add control to allow swapping HiFi DAC channels
by Alexander Shiyan 15 Dec '16

15 Dec '16

This patch adds a control to allow swapping HiFi DAC Left/Right channels. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/wm8753.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 9bdf544..d05d76e 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -280,6 +280,7 @@ static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0); static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); static const struct snd_kcontrol_new wm8753_snd_controls[] = { +SOC_SINGLE("Hi-Fi DAC Left/Right channel Swap", WM8753_HIFI, 5, 1, 0), SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv), SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0, @@ -1087,7 +1088,7 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec *codec, { u16 ioctl, hifi; - hifi = snd_soc_read(codec, WM8753_HIFI) & 0x011f; + hifi = snd_soc_read(codec, WM8753_HIFI) & 0x013f; ioctl = snd_soc_read(codec, WM8753_IOCTL) & 0x00ae; /* set master/slave audio interface */ -- 2.4.9

2 1

[alsa-devel] [PATCH 0/2] register atmel-ssc as sound DAI w/o platform driver
by Peter Rosin 15 Dec '16

15 Dec '16

Hi! The Atmel SSC is currently not usable as an audio DAI unless someone registers it with ASoC. This is currently delegated to a platform driver for every possible audio use, and prevents the SSC from being used as a cpu DAI with the simple-audio-card driver. The first patch fixes this. The second patch simplifies one of these platform drivers, since it can now rely on the SSC to register itself with ASoC. However, this may not be a possible simplification for other, older, drivers since it also requires device tree changes. Cheers, Peter Peter Rosin (2): misc: atmel-ssc: register as sound DAI if #sound-dai-cells is present ASoC: atmel: tse850: rely on the ssc to register as a cpu dai by itself .../devicetree/bindings/misc/atmel-ssc.txt | 2 + .../bindings/sound/axentia,tse850-pcm5142.txt | 5 +-- drivers/misc/atmel-ssc.c | 50 ++++++++++++++++++++++ include/linux/atmel-ssc.h | 1 + sound/soc/atmel/tse850-pcm5142.c | 23 ++-------- 5 files changed, 58 insertions(+), 23 deletions(-) -- 2.1.4

3 5

[alsa-devel] [PATCH v2 0/2] register atmel-ssc as sound DAI w/o platform driver
by Peter Rosin 15 Dec '16

15 Dec '16

Hi! v1 -> v2 changes - add ack from Rob Herring on patch 1/2. - add reasons why breaking compatibility is ok for patch 2/2 (requested by Rob). - add #sound-dai-cells to the ssc0 node in the devcietree example. The Atmel SSC is currently not usable as an audio DAI unless someone registers it with ASoC. This is currently delegated to a platform driver for every possible audio use, and prevents the SSC from being used as a cpu DAI with the simple-audio-card driver. The first patch fixes this. The second patch simplifies one of these platform drivers, since it can now rely on the SSC to register itself with ASoC. However, this may not be a possible simplification for other, older, drivers since it also requires device tree changes. Cheers, Peter Peter Rosin (2): misc: atmel-ssc: register as sound DAI if #sound-dai-cells is present ASoC: atmel: tse850: rely on the ssc to register as a cpu dai by itself .../devicetree/bindings/misc/atmel-ssc.txt | 2 + .../bindings/sound/axentia,tse850-pcm5142.txt | 11 +++-- drivers/misc/atmel-ssc.c | 50 ++++++++++++++++++++++ include/linux/atmel-ssc.h | 1 + sound/soc/atmel/tse850-pcm5142.c | 23 ++-------- 5 files changed, 64 insertions(+), 23 deletions(-) -- 2.1.4

4 5

[alsa-devel] [PATCH v2 00/13] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 15 Dec '16

15 Dec '16

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Changes in v2: - Addressed Mark's comment on patch 1, split this into 2 as this had additional changes other than configuring the DMA. - Addressed Takashi's comment on the usage of snd_hdac_updatel() Jeeja KP (13): ASoC: Intel: Skylake: Update link_index and format in pipe params ASoC: Intel: Skylake: Add helper function to setup host/link dma ASoC: Intel: Skylake: Configure DMA in PRE_PMD handler of Mixer ASoC: Intel: Skylake: Removed unused skl_get_format() ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Add set_tristate DAI ops to enable SSP MCLK ASoC: Intel: Skylake: Remove unused SSP BE prepare DAI ops ASoC: Intel: Skylake: Add supply widget as non DSP widget ASoC: Intel: Skylake: Add supply widget in skl_nau_max machine ASoC: Intel: Skylake: Add supply widget in bxt_da_max machine ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/hda/ext/hdac_ext_stream.c | 15 +- sound/soc/codecs/hdac_hdmi.c | 28 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 ++++ sound/soc/intel/skylake/skl-pcm.c | 210 +++++++++++++----------- sound/soc/intel/skylake/skl-topology.c | 27 ++- sound/soc/intel/skylake/skl-topology.h | 6 + 7 files changed, 240 insertions(+), 114 deletions(-) -- 2.5.0

4 17

[alsa-devel] [PATCH v3 1/2] ASoC: cs43130: Add support for CS43130 codec
by Li Xu 15 Dec '16

15 Dec '16

Add support for Cirrus Logic CS43130 codec. Support I2C control and I2S audio playback. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs43130.c | 1153 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs43130.h | 268 ++++++++++ 4 files changed, 1429 insertions(+) create mode 100644 sound/soc/codecs/cs43130.c create mode 100644 sound/soc/codecs/cs43130.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e1718a..d6ede2b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS43130 if I2C select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 select SND_SOC_CS53L30 if I2C @@ -473,6 +474,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS43130 HiFi DAC +config SND_SOC_CS43130 + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C + # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7e1dad7..2f15228 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs43130-objs := cs43130.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o snd-soc-cs53l30-objs := cs53l30.o @@ -281,6 +282,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c new file mode 100644 index 0000000..016d1b3 --- /dev/null +++ b/sound/soc/codecs/cs43130.c @@ -0,0 +1,1153 @@ +/* + * cs43130.c -- CS43130 ALSA Soc Audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Authors: Li Xu <li.xu(a)cirrus.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/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_runtime.h> +#include <linux/of_irq.h> + +#include "cs43130.h" + + +static const struct reg_default cs43130_reg_defaults[] = { + { CS43130_SYS_CLK_CTL_1, 0x06 }, + { CS43130_SP_SRATE, 0x01 }, + { CS43130_SP_BITSIZE, 0x05 }, + { CS43130_PAD_INT_CFG, 0x03 }, + { CS43130_PWDN_CTL, 0xFE }, + { CS43130_CRYSTAL_SET, 0x04 }, + { CS43130_PLL_SET_1, 0x00 }, + { CS43130_PLL_SET_2, 0x00 }, + { CS43130_PLL_SET_3, 0x00 }, + { CS43130_PLL_SET_4, 0x00 }, + { CS43130_PLL_SET_5, 0x40 }, + { CS43130_PLL_SET_6, 0x10 }, + { CS43130_PLL_SET_7, 0x80 }, + { CS43130_PLL_SET_8, 0x03 }, + { CS43130_PLL_SET_9, 0x02 }, + { CS43130_PLL_SET_10, 0x02 }, + { CS43130_CLKOUT_CTL, 0x00 }, + { CS43130_ASP_NUM_1, 0x01 }, + { CS43130_ASP_NUM_2, 0x00 }, + { CS43130_ASP_DENOM_1, 0x08 }, + { CS43130_ASP_DENOM_2, 0x00 }, + { CS43130_ASP_LRCK_HI_TIME_1, 0x1F }, + { CS43130_ASP_LRCK_HI_TIME_2, 0x00 }, + { CS43130_ASP_LRCK_PERIOD_1, 0x3F }, + { CS43130_ASP_LRCK_PERIOD_2, 0x00 }, + { CS43130_ASP_CLOCK_CONF, 0x0C }, + { CS43130_ASP_FRAME_CONF, 0x0A }, + { CS43130_XSP_NUM_1, 0x01 }, + { CS43130_XSP_NUM_2, 0x00 }, + { CS43130_XSP_DENOM_1, 0x02 }, + { CS43130_XSP_DENOM_2, 0x00 }, + { CS43130_XSP_LRCK_HI_TIME_1, 0x1F }, + { CS43130_XSP_LRCK_HI_TIME_2, 0x00 }, + { CS43130_XSP_LRCK_PERIOD_1, 0x3F }, + { CS43130_XSP_LRCK_PERIOD_2, 0x00 }, + { CS43130_XSP_CLOCK_CONF, 0x0C }, + { CS43130_XSP_FRAME_CONF, 0x0A }, + { CS43130_ASP_CH_1_LOC, 0x00 }, + { CS43130_ASP_CH_2_LOC, 0x00 }, + { CS43130_ASP_CH_1_SZ_EN, 0x06 }, + { CS43130_ASP_CH_2_SZ_EN, 0x0E }, + { CS43130_XSP_CH_1_LOC, 0x00 }, + { CS43130_XSP_CH_2_LOC, 0x00 }, + { CS43130_XSP_CH_1_SZ_EN, 0x06 }, + { CS43130_XSP_CH_2_SZ_EN, 0x0E }, + { CS43130_DSD_VOL_B, 0x78 }, + { CS43130_DSD_VOL_A, 0x78 }, + { CS43130_DSD_PATH_CTL_1, 0xA8 }, + { CS43130_DSD_INT_CFG, 0x00 }, + { CS43130_DSD_PATH_CTL_2, 0x00 }, + { CS43130_DSD_PCM_MIX_CTL, 0x00 }, + { CS43130_DSD_PATH_CTL_3, 0x40 }, + { CS43130_HP_OUT_CTL_1, 0x30 }, + { CS43130_PCM_FILT_OPT, 0x02 }, + { CS43130_PCM_VOL_B, 0x78 }, + { CS43130_PCM_VOL_A, 0x78 }, + { CS43130_PCM_PATH_CTL_1, 0xA8 }, + { CS43130_PCM_PATH_CTL_2, 0x00 }, + { CS43130_CLASS_H_CTL, 0x1E }, + { CS43130_HP_DETECT, 0x04 }, + { CS43130_HP_LOAD_1, 0x00 }, + { CS43130_HP_MEAS_LOAD_1, 0x00 }, + { CS43130_HP_MEAS_LOAD_2, 0x00 }, + { CS43130_INT_MASK_1, 0xFF }, + { CS43130_INT_MASK_2, 0xFF }, + { CS43130_INT_MASK_3, 0xFF }, + { CS43130_INT_MASK_4, 0xFF }, + { CS43130_INT_MASK_5, 0xFF }, +}; + +static bool cs43130_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + return true; + default: + return false; + } +} + +static bool cs43130_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_DEVID_AB ... CS43130_SYS_CLK_CTL_1: + case CS43130_SP_SRATE ... CS43130_PAD_INT_CFG: + case CS43130_DXD1: + case CS43130_PWDN_CTL: + case CS43130_DXD2: + case CS43130_CRYSTAL_SET: + case CS43130_PLL_SET_1 ... CS43130_PLL_SET_5: + case CS43130_PLL_SET_6: + case CS43130_PLL_SET_7: + case CS43130_PLL_SET_8: + case CS43130_PLL_SET_9: + case CS43130_PLL_SET_10: + case CS43130_CLKOUT_CTL: + case CS43130_ASP_NUM_1 ... CS43130_ASP_FRAME_CONF: + case CS43130_XSP_NUM_1 ... CS43130_XSP_FRAME_CONF: + case CS43130_ASP_CH_1_LOC: + case CS43130_ASP_CH_2_LOC: + case CS43130_ASP_CH_1_SZ_EN: + case CS43130_ASP_CH_2_SZ_EN: + case CS43130_XSP_CH_1_LOC: + case CS43130_XSP_CH_2_LOC: + case CS43130_XSP_CH_1_SZ_EN: + case CS43130_XSP_CH_2_SZ_EN: + case CS43130_DSD_VOL_B ... CS43130_DSD_PATH_CTL_3: + case CS43130_HP_OUT_CTL_1: + case CS43130_PCM_FILT_OPT ... CS43130_PCM_PATH_CTL_2: + case CS43130_CLASS_H_CTL: + case CS43130_HP_DETECT: + case CS43130_HP_STATUS: + case CS43130_HP_LOAD_1: + case CS43130_HP_MEAS_LOAD_1: + case CS43130_HP_MEAS_LOAD_2: + case CS43130_HP_DC_STAT_1: + case CS43130_HP_DC_STAT_2: + case CS43130_HP_AC_STAT_1: + case CS43130_HP_AC_STAT_2: + case CS43130_HP_LOAD_STAT: + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + case CS43130_INT_MASK_1 ... CS43130_INT_MASK_5: + return true; + default: + return false; + } +} + +static bool cs43130_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + return true; + default: + return false; + } +} + +struct cs43130_pll_params { + u32 pll_in; + u8 mclk_int; + u8 sclk_prediv; + u8 pll_div_int; + u32 pll_div_frac; + u8 pll_mode; + u8 pll_divout; + u32 pll_out; + u8 pll_cal_ratio; +}; + +static const struct cs43130_pll_params pll_ratio_table[] = { + { 9600000, 1, 0x02, 0x49, 0x800000, 0x00, 0x08, 22579200, 151 }, + { 9600000, 0, 0x02, 0x50, 0x000000, 0x00, 0x08, 24576000, 164 }, + + { 11289600, 1, 0x02, 0X40, 0, 0x01, 0x08, 22579200, 128 }, + { 11289600, 0, 0x02, 0x44, 0x06F700, 0x0, 0x08, 24576000, 139 }, + + { 12000000, 1, 0x02, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120 }, + { 12000000, 0, 0x02, 0x40, 0x000000, 0x00, 0x08, 24576000, 131 }, + + { 12288000, 1, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118 }, + { 12288000, 0, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128 }, + + { 13000000, 1, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118 }, + { 13000000, 0, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128 }, + + { 19200000, 1, 0x02, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111 }, + { 19200000, 0, 0x02, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121 }, + + { 22579200, 1, 0, 0, 0, 0, 0, 22579200, 0 }, + { 22579200, 0, 0x03, 0x44, 0x06F700, 0x00, 0x08, 24576000, 139 }, + + { 24000000, 1, 0x03, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120 }, + { 24000000, 0, 0x03, 0x40, 0x000000, 0x00, 0x08, 24576000, 131 }, + + { 24576000, 1, 0x03, 0x49, 0x800000, 0x01, 0x0A, 22579200, 128 }, + { 24576000, 0, 0, 0, 0, 0, 0, 24576000, 0 }, + + { 26000000, 1, 0x03, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111 }, + { 26000000, 0, 0x03, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121 }, +}; + +static int cs43130_pll_config(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "%s: cs43130->mclk = %d, cs43130->pll_out = %d", + __func__, cs43130->mclk, cs43130->pll_out); + for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { + if (pll_ratio_table[i].pll_in == cs43130->mclk && + pll_ratio_table[i].pll_out == cs43130->pll_out) { + + cs43130->mclk_int = pll_ratio_table[i].mclk_int; + + if (pll_ratio_table[i].pll_cal_ratio == 0) { + if (cs43130->xtal_ibias > 0) { + usleep_range(1000, 1050); + /*PDN_XTAL = 0,enable*/ + regmap_update_bits(cs43130->regmap, + CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, + 0 << CS43130_PDN_XTAL_SHIFT); + } + + /* PLL_START = 0, disable PLL_START */ + regmap_update_bits(cs43130->regmap, + CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, + 0 << CS43130_PLL_START_MASK); + + cs43130->pll_bypass = true; + return 0; + } + + /* PDN_PLL= 0,enable */ + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, + 0 << CS43130_PDN_PLL_SHIFT); + + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_9, + CS43130_PLL_REF_PREDIV_MASK, + pll_ratio_table[i].sclk_prediv); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_5, + pll_ratio_table[i].pll_div_int); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_2, + pll_ratio_table[i].pll_div_frac & + CS43130_7_0_MASK); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_3, + (pll_ratio_table[i].pll_div_frac & + CS43130_15_8_MASK) >> 8); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_4, + (pll_ratio_table[i].pll_div_frac & + CS43130_23_16_MASK) >> 16); + + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_8, + CS43130_PLL_MODE_MASK, + pll_ratio_table[i].pll_mode + << CS43130_PLL_MODE_SHIFT); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_6, + pll_ratio_table[i].pll_divout); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_7, + pll_ratio_table[i].pll_cal_ratio); + + /* PLL_START = 1, enable PLL_START */ + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, CS43130_PLL_START_MASK); + cs43130->pll_bypass = false; + return 0; + } + } + return -EINVAL; +} + +static int cs43130_format_config(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + int period_size = 0; + int pulse_width = 0; + int asp_fsd; + int asp_stp; + int bick_inv; + int asp_m = 0; + int asp_sprate = 0; + int ret = 0; + unsigned int bitwidth_sclk = (cs43130->sclk / cs43130->fs) / 2; + unsigned int bitwidth_dai = (cs43130->dai_bit + 1) * 8; + + if (cs43130->fs) { + if (bitwidth_sclk < bitwidth_dai) { + dev_err(codec->dev, "Format not supported\n"); + return -EINVAL; + } + period_size = cs43130->sclk / cs43130->fs; + pulse_width = period_size/2; + + if (cs43130->sclk != 0) + asp_m = cs43130->pll_out / cs43130->sclk; + } + dev_dbg(codec->dev, "%s: cs43130->sclk = %d, cs43130->fs = %d, cs43130->dai_bit = %d", + __func__, cs43130->sclk, cs43130->fs, cs43130->dai_bit); + dev_dbg(codec->dev, "%s: period_size = %d, pulse_width = %d, asp_m = %d", + __func__, period_size, pulse_width, asp_m); + + if (cs43130->dai_format) { + /*MSB*/ + bick_inv = 0; + asp_fsd = 0; + asp_stp = 1; + + } else { + /*I2S*/ + bick_inv = 1; + asp_fsd = 2; /* one bick delay */ + asp_stp = 0; + } + dev_dbg(codec->dev, + "%s: cs43130->dai_format = %d, bick_inv = %d, asp_fsd = %d, asp_stp = %d", + __func__, cs43130->dai_format, bick_inv, asp_fsd, asp_stp); + + switch (cs43130->fs) { + case 32000: + asp_sprate = CS43130_ASP_SPRATE_32K; + break; + case 44100: + asp_sprate = CS43130_ASP_SPRATE_44_1K; + break; + case 48000: + asp_sprate = CS43130_ASP_SPRATE_48K; + break; + case 88200: + asp_sprate = CS43130_ASP_SPRATE_88_2K; + break; + case 96000: + asp_sprate = CS43130_ASP_SPRATE_96K; + break; + case 176400: + asp_sprate = CS43130_ASP_SPRATE_176_4K; + break; + case 192000: + asp_sprate = CS43130_ASP_SPRATE_192K; + break; + case 352800: + asp_sprate = CS43130_ASP_SPRATE_352_8K; + break; + case 384000: + asp_sprate = CS43130_ASP_SPRATE_384K; + break; + default: + dev_err(codec->dev, "sample rate(%d) not supported\n", + cs43130->fs); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: asp_sprate = %d, cs43130->asp_size = %d", + __func__, asp_sprate, cs43130->asp_size); + + /* ASP_SPRATE = fs*/ + regmap_write(cs43130->regmap, CS43130_SP_SRATE, asp_sprate); + /*ASP_SPSIZE*/ + regmap_update_bits(cs43130->regmap, CS43130_SP_BITSIZE, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->asp_size); + + + /* Set up slave mode */ + /*BICK = ASP_N/ASP_M * PLL_OUT */ + /* ASP_N = 1 */ + regmap_write(cs43130->regmap, CS43130_ASP_NUM_1, 1); + regmap_write(cs43130->regmap, CS43130_ASP_NUM_2, 0); + + /*ASP_M*/ + regmap_write(cs43130->regmap, CS43130_ASP_DENOM_1, asp_m & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_DENOM_2, (asp_m >> 8) & 0x3f); + + + /* H / L ratio of LRCK*/ + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_1, + (pulse_width-1) & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_2, + ((pulse_width-1) >> 8) & 0xff); + + /* the period of LRCK*/ + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_1, + (period_size-1) & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_2, + ((period_size-1) >> 8) & 0xff); + + /*resolution*/ + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_1_SZ_EN, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->dai_bit); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_2_SZ_EN, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->dai_bit); + + regmap_update_bits(cs43130->regmap, CS43130_ASP_FRAME_CONF, + CS43130_ASP_FSD_MASK, asp_fsd << CS43130_ASP_FSD_SHIFT); + + regmap_update_bits(cs43130->regmap, CS43130_ASP_FRAME_CONF, + CS43130_ASP_STP_MASK, asp_stp << CS43130_ASP_STP_SHIFT); + + /* set clk master/slave */ + dev_dbg(codec->dev, "%s: cs43130->dai_mode = %d", + __func__, cs43130->dai_mode); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CLOCK_CONF, + CS43130_ASP_MODE_MASK, cs43130->dai_mode << CS43130_ASP_MODE_SHIFT); + + return ret; +} + +static int cs43130_change_clksrc(struct snd_soc_codec *codec, + enum cs43130_mclk_src_sel src) +{ + int ret = 0; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, cs43130->mclk_int << CS43130_MCLK_INT_SHIFT); + + usleep_range(150, 200); + + return ret; +} + +static int cs43130_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + unsigned int bitwidth; + int ret = 0; + + cs43130->fs = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_8; + cs43130->asp_size = CS43130_SP_BIT_SIZE_32; + break; + case SNDRV_PCM_FORMAT_S16_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_16; + cs43130->asp_size = CS43130_SP_BIT_SIZE_24; + break; + case SNDRV_PCM_FORMAT_S24_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_24; + cs43130->asp_size = CS43130_SP_BIT_SIZE_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_32; + cs43130->asp_size = CS43130_SP_BIT_SIZE_8; + break; + default: + dev_err(codec->dev, "Format(%d) not supported", + params_format(params)); + return -EINVAL; + } + + bitwidth = (cs43130->dai_bit+1)*8; + dev_dbg(codec->dev, "(data bit)%d: (rate)%d", + bitwidth, cs43130->fs); + + ret = cs43130_format_config(codec); + return ret; +} + +static const DECLARE_TLV_DB_SCALE(pcm_vol_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new cs43130_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS43130_PCM_VOL_A, CS43130_PCM_VOL_B, 0, 0xFF, 1, + pcm_vol_tlv), + SOC_SINGLE("Swap L/R", CS43130_PCM_PATH_CTL_2, 1, 1, 0), + SOC_SINGLE("Copy L/R", CS43130_PCM_PATH_CTL_2, 0, 1, 0), +}; + +static int cs43130_aspin_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); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_XTAL); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + + usleep_range(10000, 10050); + /* ASP_3ST = 0 in master mode */ + if (cs43130->dai_mode) + regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG, + 0x01, 0x00); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_CLKOUT_MASK, 0 + << CS43130_PDN_CLKOUT_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + break; + default: + dev_err(codec->dev, "Invalid ASPOUT event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static int cs43130_dac_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); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, 1 + << CS43130_PDN_XTAL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, 1 + << CS43130_PDN_PLL_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAC event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static int cs43130_hpin_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); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + regmap_write(cs43130->regmap, CS43130_DXD1, 0x99); + regmap_update_bits(cs43130->regmap, CS43130_HP_OUT_CTL_1, + CS43130_HP_IN_EN_MASK, 0 << CS43130_HP_IN_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_DXD2, 0x00); + regmap_write(cs43130->regmap, CS43130_DXD1, 0x00); + break; + case SND_SOC_DAPM_POST_PMU: + regmap_write(cs43130->regmap, CS43130_DXD1, 0x99); + regmap_write(cs43130->regmap, CS43130_DXD2, 0x01); + regmap_update_bits(cs43130->regmap, CS43130_HP_OUT_CTL_1, + CS43130_HP_IN_EN_MASK, 1 << CS43130_HP_IN_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_DXD1, 0x00); + break; + default: + dev_err(codec->dev, "Invalid HPIN event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct snd_soc_dapm_widget cs43130_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + + SND_SOC_DAPM_AIF_IN_E("ASPIN", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_ASP_SHIFT, 1, cs43130_aspin_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_DAC_E("HiFi DAC", + NULL, CS43130_PWDN_CTL, CS43130_PDN_HP_SHIFT, 1, + cs43130_dac_event, + (SND_SOC_DAPM_PRE_PMD) + ), + + SND_SOC_DAPM_LINE("Analog Playback", cs43130_hpin_event), +}; + +static const struct snd_soc_dapm_route cs43130_routes[] = { + {"ASPIN", NULL, "DAC Playback"}, + {"HiFi DAC", NULL, "ASPIN"}, + + {"HPOUTA", NULL, "HiFi DAC"}, + {"HPOUTB", NULL, "HiFi DAC"}, + {"HPOUTA", NULL, "Analog Playback"}, + {"HPOUTB", NULL, "Analog Playback"}, +}; + +static const unsigned int cs43130_src_rates[] = { + 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs43130_constraints = { + .count = ARRAY_SIZE(cs43130_src_rates), + .list = cs43130_src_rates, +}; + +static int cs43130_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs43130_constraints); + return 0; +} + +static int cs43130_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs43130->dai_mode = CS43130_SLAVE_MODE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs43130->dai_mode = CS43130_MASTER_MODE; + break; + default: + dev_err(codec->dev, "unsupported i2s master mode\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs43130->dai_format = 0; + break; + case SND_SOC_DAIFMT_LEFT_J: + cs43130->dai_format = 1; + break; + default: + dev_err(codec->dev, "unsupported audio format except I2S and MSB\n"); + return -EINVAL; + } + + /* BICK/LRCK pority */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + cs43130->bick_invert = false; + cs43130->lrck_invert = false; + break; + case SND_SOC_DAIFMT_IB_NF: + cs43130->bick_invert = true; + cs43130->lrck_invert = false; + break; + case SND_SOC_DAIFMT_NB_IF: + cs43130->bick_invert = false; + cs43130->lrck_invert = true; + break; + case SND_SOC_DAIFMT_IB_IF: + cs43130->bick_invert = true; + cs43130->lrck_invert = true; + break; + default: + dev_err(codec->dev, "unsupported audio polarity\n"); + return -EINVAL; + } + + return 0; +} + + +static int cs43130_set_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + unsigned int reg; + u8 mute_reg; + + regmap_read(cs43130->regmap, CS43130_PCM_PATH_CTL_1, &reg); + mute_reg = reg & 0xfc; + if (mute) + regmap_write(cs43130->regmap, CS43130_PCM_PATH_CTL_1, + mute_reg | 0x03); + else + regmap_write(cs43130->regmap, CS43130_PCM_PATH_CTL_1, mute_reg); + + return ret; +} + +static int cs43130_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + + switch (div_id) { + case CS43130_AIF_BICK_RATE: + cs43130->bick = div; + break; + default: + dev_err(codec->dev, + "Unsupported divide value: div_id = %d", div_id); + return -EINVAL; + } + return 0; +} + +static int cs43130_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + int ret = 0; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + if (freq_in < 9600000 || freq_in > 26000000) { + dev_err(codec->dev, + "unsupported pll input reference clock:%d\n", freq_in); + return -EINVAL; + } + + switch (freq_in) { + case 9600000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 19200000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs43130->mclk = freq_in; + break; + default: + dev_err(codec->dev, + "unsupported pll input reference clock:%d\n", freq_in); + return -EINVAL; + } + + switch (freq_out) { + case 22579200: + cs43130->pll_out = freq_out; + cs43130->mclk_int = 1; + break; + case 24576000: + cs43130->pll_out = freq_out; + cs43130->mclk_int = 0; + break; + default: + dev_err(codec->dev, + "unsupported pll output reference clock:%d\n", + freq_out); + return -EINVAL; + } + + ret = cs43130_pll_config(codec); + dev_dbg(codec->dev, "%s: cs43130->pll_bypass = %d", + __func__, cs43130->pll_bypass); + return ret; +} + +static int cs43130_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: clk_id = %d, freq = %d, dir = %d", + __func__, clk_id, freq, dir); + cs43130->sclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops cs43130_dai_ops = { + .startup = cs43130_pcm_startup, + .hw_params = cs43130_pcm_hw_params, + .set_sysclk = cs43130_dai_set_sysclk, + .set_fmt = cs43130_set_dai_fmt, + .digital_mute = cs43130_set_mute, + .set_clkdiv = cs43130_set_clkdiv, +}; + +static struct snd_soc_dai_driver cs43130_dai[] = { + { + .name = "cs43130_hifi", + .id = 0, + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_ASP_FORMATS, + }, + .ops = &cs43130_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-xsp", + .id = 1, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_XSP_FORMATS, + }, + .symmetric_rates = 1, + }, +}; + +static int cs43130_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + /* 24576000 is not supported */ + unsigned int mclk_int_freq = 22579200; + + dev_dbg(codec->dev, "%s: clk_id = %d, source = %d, freq = %d, dir = %d\n", + __func__, clk_id, source, freq, dir); + /* + * freq is external mclk freq + * if freq == mclk_int_freq, pll is bypassed + * modify mclk_int_freq as needed for application + */ + cs43130_set_pll(codec, 0, 0, freq, mclk_int_freq); + return 0; +} + +static irqreturn_t cs43130_irq_thread(int irq, void *data) +{ + struct cs43130_private *cs43130 = + (struct cs43130_private *)data; + struct snd_soc_codec *codec = cs43130->codec; + unsigned int stickies[CS43130_NUM_INT]; + unsigned int masks[CS43130_NUM_INT]; + unsigned int i; + + /* Read all INT status and mask reg */ + regmap_bulk_read(cs43130->regmap, CS43130_INT_STATUS_1, + stickies, CS43130_NUM_INT * sizeof(unsigned int)); + regmap_bulk_read(cs43130->regmap, CS43130_INT_MASK_1, + masks, CS43130_NUM_INT * sizeof(unsigned int)); + + for (i = 0; i < ARRAY_SIZE(stickies); i++) + stickies[i] = stickies[i] & (~masks[i]); + + if (stickies[0] & CS43130_XTAL_RDY_INT) + dev_dbg(codec->dev, "%s: Crystal ready", __func__); + + if (stickies[0] & CS43130_XTAL_ERR_INT) + dev_err(codec->dev, "%s: Crystal err", __func__); + + return IRQ_HANDLED; +} + +static int cs43130_probe(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + cs43130->codec = codec; + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs43130 = { + .probe = cs43130_probe, + .component_driver = { + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + .dapm_widgets = cs43130_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs43130_dapm_widgets), + .dapm_routes = cs43130_routes, + .num_dapm_routes = ARRAY_SIZE(cs43130_routes), + }, + .set_sysclk = cs43130_codec_set_sysclk, + .set_pll = cs43130_set_pll, +}; + +static const struct regmap_config cs43130_regmap = { + .reg_bits = 24, + .pad_bits = 8, + .val_bits = 8, + + .max_register = CS43130_LASTREG, + .reg_defaults = cs43130_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs43130_reg_defaults), + .readable_reg = cs43130_readable_register, + .precious_reg = cs43130_precious_register, + .volatile_reg = cs43130_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs43130_handle_device_data( + struct i2c_client *i2c_client, struct cs43130_private *cs43130) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + int ret = 0; + + of_property_read_u32(np, "cirrus,xtal-ibias", &val); + switch (val) { + case 1: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_7_5UA; + break; + case 2: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_12_5UA; + break; + case 3: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_15UA; + break; + default: + dev_info(&i2c_client->dev, + "cirrus,xtal-ibias value or xtal unused %d", + val); + } + return ret; +} + +static int cs43130_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs43130_private *cs43130; + int ret; + unsigned int devid = 0; + unsigned int reg; + int i; + + cs43130 = devm_kzalloc(&client->dev, sizeof(*cs43130), GFP_KERNEL); + if (cs43130 == NULL) + return -ENOMEM; + + i2c_set_clientdata(client, cs43130); + + cs43130->regmap = devm_regmap_init_i2c(client, &cs43130_regmap); + if (IS_ERR(cs43130->regmap)) { + ret = PTR_ERR(cs43130->regmap); + return ret; + } + + if (client->dev.of_node) { + ret = cs43130_handle_device_data(client, cs43130); + if (ret != 0) + return ret; + } + for (i = 0; i < ARRAY_SIZE(cs43130->supplies); i++) + cs43130->supplies[i].supply = cs43130_supply_names[i]; + + ret = devm_regulator_bulk_get(&client->dev, + ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, + "Failed to request supplies: %d\n", ret); + return ret; + } + ret = regulator_bulk_enable(ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + cs43130->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs43130->reset_gpio)) + return PTR_ERR(cs43130->reset_gpio); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 1); + + usleep_range(2000, 2050); + + /* initialize codec */ + ret = regmap_read(cs43130->regmap, CS43130_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + switch (devid) { + case CS43130_CHIP_ID: + break; + case CS4399_CHIP_ID: + break; + default: + dev_err(&client->dev, + "CS43130 Device ID (%X). Expected ID %X or %X\n", + devid, CS43130_CHIP_ID, CS4399_CHIP_ID); + ret = -ENODEV; + goto err; + } + + cs43130->dev_id = devid; + ret = regmap_read(cs43130->regmap, CS43130_REV_ID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + goto err; + } + + dev_info(&client->dev, + "Cirrus Logic CS43130 (%x), Revision: %02X\n", devid, + reg & 0xFF); + + /* Enable interrupt handler */ + ret = devm_request_threaded_irq(&client->dev, + client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(&client->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + /* Unmask INT */ + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT | CS43130_XTAL_ERR_INT, 0); + + regmap_write(cs43130->regmap, + CS43130_CRYSTAL_SET, cs43130->xtal_ibias); + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs43130, cs43130_dai, + ARRAY_SIZE(cs43130_dai)); + + if (ret < 0) { + dev_err(&client->dev, + "%s: snd_soc_register_codec failed with ret = %d\n", + __func__, ret); + goto err; + } + return 0; +err: + return ret; + +} + +static int cs43130_i2c_remove(struct i2c_client *client) +{ + struct cs43130_private *cs43130 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs43130->reset_gpio) + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(CS43130_NUM_SUPPLIES, + cs43130->supplies); + + return 0; +} + +#ifdef CONFIG_PM +static int cs43130_runtime_suspend(struct device *dev) +{ + struct cs43130_private *cs43130 = dev_get_drvdata(dev); + + regcache_cache_only(cs43130->regmap, true); + regcache_mark_dirty(cs43130->regmap); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + + regulator_bulk_disable(CS43130_NUM_SUPPLIES, + cs43130->supplies); + return 0; +} + +static int cs43130_runtime_resume(struct device *dev) +{ + struct cs43130_private *cs43130 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(CS43130_NUM_SUPPLIES, + cs43130->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + regcache_cache_only(cs43130->regmap, false); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 1); + + usleep_range(2000, 2050); + + ret = regcache_sync(cs43130->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + return 0; +err: + regcache_cache_only(cs43130->regmap, true); + regulator_bulk_disable(CS43130_NUM_SUPPLIES, + cs43130->supplies); + + return ret; +} +#endif + +static const struct dev_pm_ops cs43130_runtime_pm = { + SET_RUNTIME_PM_OPS(cs43130_runtime_suspend, cs43130_runtime_resume, + NULL) +}; + +static const struct of_device_id cs43130_of_match[] = { + { .compatible = "cirrus,cs43130", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs43130_of_match); + +static const struct i2c_device_id cs43130_i2c_id[] = { + {"cs43130", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs43130_i2c_id); + +static struct i2c_driver cs43130_i2c_driver = { + .driver = { + .name = "cs43130", + .of_match_table = cs43130_of_match, + }, + .id_table = cs43130_i2c_id, + .probe = cs43130_i2c_probe, + .remove = cs43130_i2c_remove, +}; + +module_i2c_driver(cs43130_i2c_driver); + +MODULE_AUTHOR("Li Xu <li.xu(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS43130 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h new file mode 100644 index 0000000..bceae76 --- /dev/null +++ b/sound/soc/codecs/cs43130.h @@ -0,0 +1,268 @@ +/* + * ALSA SoC CS43130 codec driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Li Xu <li.xu(a)cirrus.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS43130_H__ +#define __CS43130_H__ + +/* CS43130 registers addresses */ +/* all reg address is shifted by a byte for control byte to be LSB */ +#define CS43130_FIRSTREG 0x010000 +#define CS43130_LASTREG 0x0F0014 +#define CS43130_CHIP_ID 0x00043130 +#define CS4399_CHIP_ID 0x00043990 +#define CS43130_DEVID_AB 0x010000 /*Device ID A & B [RO]*/ +#define CS43130_DEVID_CD 0x010001 /*Device ID C & D [RO]*/ +#define CS43130_DEVID_E 0x010002 /*Device ID E [RO]*/ +#define CS43130_FAB_ID 0x010003 /*Fab ID [RO]*/ +#define CS43130_REV_ID 0x010004 /*Revision ID [RO]*/ +#define CS43130_SUBREV_ID 0x010005 /*Subrevision ID*/ +#define CS43130_SYS_CLK_CTL_1 0x010006 /*System Clocking Ctl 1*/ +#define CS43130_SP_SRATE 0x01000B /*Serial Port Sample Rate*/ +#define CS43130_SP_BITSIZE 0x01000C /*Serial Port Bit Size*/ +#define CS43130_PAD_INT_CFG 0x01000D /*Pad Interface Config*/ +#define CS43130_DXD1 0x010010 /*DXD1*/ +#define CS43130_PWDN_CTL 0x020000 /*Power Down Ctl*/ +#define CS43130_DXD2 0x020019 /*DXD2*/ +#define CS43130_CRYSTAL_SET 0x020052 /*Crystal Setting*/ +#define CS43130_PLL_SET_1 0x030001 /*PLL Setting 1*/ +#define CS43130_PLL_SET_2 0x030002 /*PLL Setting 2*/ +#define CS43130_PLL_SET_3 0x030003 /*PLL Setting 3*/ +#define CS43130_PLL_SET_4 0x030004 /*PLL Setting 4*/ +#define CS43130_PLL_SET_5 0x030005 /*PLL Setting 5*/ +#define CS43130_PLL_SET_6 0x030008 /*PLL Setting 6*/ +#define CS43130_PLL_SET_7 0x03000A /*PLL Setting 7*/ +#define CS43130_PLL_SET_8 0x03001B /*PLL Setting 8*/ +#define CS43130_PLL_SET_9 0x040002 /*PLL Setting 9*/ +#define CS43130_PLL_SET_10 0x040003 /*PLL Setting 10*/ +#define CS43130_CLKOUT_CTL 0x040004 /*CLKOUT Ctl*/ +#define CS43130_ASP_NUM_1 0x040010 /*ASP Numerator 1*/ +#define CS43130_ASP_NUM_2 0x040011 /*ASP Numerator 2*/ +#define CS43130_ASP_DENOM_1 0x040012 /*ASP Denominator 1*/ +#define CS43130_ASP_DENOM_2 0x040013 /*ASP Denominator 2*/ +#define CS43130_ASP_LRCK_HI_TIME_1 0x040014 /*ASP LRCK High Time 1*/ +#define CS43130_ASP_LRCK_HI_TIME_2 0x040015 /*ASP LRCK High Time 2*/ +#define CS43130_ASP_LRCK_PERIOD_1 0x040016 /*ASP LRCK Period 1*/ +#define CS43130_ASP_LRCK_PERIOD_2 0x040017 /*ASP LRCK Period 2*/ +#define CS43130_ASP_CLOCK_CONF 0x040018 /*ASP Clock Config*/ +#define CS43130_ASP_FRAME_CONF 0x040019 /*ASP Frame Config*/ +#define CS43130_XSP_NUM_1 0x040020 /*XSP Numerator 1*/ +#define CS43130_XSP_NUM_2 0x040021 /*XSP Numerator 2*/ +#define CS43130_XSP_DENOM_1 0x040022 /*XSP Denominator 1*/ +#define CS43130_XSP_DENOM_2 0x040023 /*XSP Denominator 2*/ +#define CS43130_XSP_LRCK_HI_TIME_1 0x040024 /*XSP LRCK High Time 1*/ +#define CS43130_XSP_LRCK_HI_TIME_2 0x040025 /*XSP LRCK High Time 2*/ +#define CS43130_XSP_LRCK_PERIOD_1 0x040026 /*XSP LRCK Period 1*/ +#define CS43130_XSP_LRCK_PERIOD_2 0x040027 /*XSP LRCK Period 2*/ +#define CS43130_XSP_CLOCK_CONF 0x040028 /*XSP Clock Config*/ +#define CS43130_XSP_FRAME_CONF 0x040029 /*XSP Frame Config*/ +#define CS43130_ASP_CH_1_LOC 0x050000 /*ASP Chan 1 Location*/ +#define CS43130_ASP_CH_2_LOC 0x050001 /*ASP Chan 2 Location*/ +#define CS43130_ASP_CH_1_SZ_EN 0x05000A /*ASP Chan 1 Size, Enable*/ +#define CS43130_ASP_CH_2_SZ_EN 0x05000B /*ASP Chan 2 Size, Enable*/ +#define CS43130_XSP_CH_1_LOC 0x060000 /*XSP Chan 1 Location*/ +#define CS43130_XSP_CH_2_LOC 0x060001 /*XSP Chan 2 Location*/ +#define CS43130_XSP_CH_1_SZ_EN 0x06000A /*XSP Chan 1 Size, Enable*/ +#define CS43130_XSP_CH_2_SZ_EN 0x06000B /*XSP Chan 2 Size, Enable*/ +#define CS43130_DSD_VOL_B 0x070000 /*DSD Volume B*/ +#define CS43130_DSD_VOL_A 0x070001 /*DSD Volume A*/ +#define CS43130_DSD_PATH_CTL_1 0x070002 /*DSD Proc Path Sig Ctl 1*/ +#define CS43130_DSD_INT_CFG 0x070003 /*DSD Interface Config*/ +#define CS43130_DSD_PATH_CTL_2 0x070004 /*DSD Proc Path Sig Ctl 2*/ +#define CS43130_DSD_PCM_MIX_CTL 0x070005 /*DSD and PCM Mixing Ctl*/ +#define CS43130_DSD_PATH_CTL_3 0x070006 /*DSD Proc Path Sig Ctl 3*/ +#define CS43130_HP_OUT_CTL_1 0x080000 /*HP Output Ctl 1*/ +#define CS43130_PCM_FILT_OPT 0x090000 /*PCM Filter Option*/ +#define CS43130_PCM_VOL_B 0x090001 /*PCM Volume B*/ +#define CS43130_PCM_VOL_A 0x090002 /*PCM Volume A*/ +#define CS43130_PCM_PATH_CTL_1 0x090003 /*PCM Path Signal Ctl 1*/ +#define CS43130_PCM_PATH_CTL_2 0x090004 /*PCM Path Signal Ctl 2*/ +#define CS43130_CLASS_H_CTL 0x0B0000 /*Class H Ctl*/ +#define CS43130_HP_DETECT 0x0D0000 /*HP Detect*/ +#define CS43130_HP_STATUS 0x0D0001 /*HP Status [RO]*/ +#define CS43130_HP_LOAD_1 0x0E0000 /*HP Load 1*/ +#define CS43130_HP_MEAS_LOAD_1 0x0E0003 /*HP Load Measurement 1*/ +#define CS43130_HP_MEAS_LOAD_2 0x0E0004 /*HP Load Measurement 2*/ +#define CS43130_HP_DC_STAT_1 0x0E000D /*HP DC Load Status 0 [RO]*/ +#define CS43130_HP_DC_STAT_2 0x0E000E /*HP DC Load Status 1 [RO]*/ +#define CS43130_HP_AC_STAT_1 0x0E0010 /*HP AC Load Status 0 [RO]*/ +#define CS43130_HP_AC_STAT_2 0x0E0011 /*HP AC Load Status 1 [RO]*/ +#define CS43130_HP_LOAD_STAT 0x0E001A /*HP Load Status [RO]*/ +#define CS43130_INT_STATUS_1 0x0F0000 /*Interrupt Status 1*/ +#define CS43130_INT_STATUS_2 0x0F0001 /*Interrupt Status 2*/ +#define CS43130_INT_STATUS_3 0x0F0002 /*Interrupt Status 3*/ +#define CS43130_INT_STATUS_4 0x0F0003 /*Interrupt Status 4*/ +#define CS43130_INT_STATUS_5 0x0F0004 /*Interrupt Status 5*/ +#define CS43130_INT_MASK_1 0x0F0010 /*Interrupt Mask 1*/ +#define CS43130_INT_MASK_2 0x0F0011 /*Interrupt Mask 2*/ +#define CS43130_INT_MASK_3 0x0F0012 /*Interrupt Mask 3*/ +#define CS43130_INT_MASK_4 0x0F0013 /*Interrupt Mask 4*/ +#define CS43130_INT_MASK_5 0x0F0014 /*Interrupt Mask 5*/ + +#define CS43130_MCLK_SRC_SEL_MASK 0x03 +#define CS43130_MCLK_SRC_SEL_SHIFT 0 +#define CS43130_MCLK_INT_MASK 0x04 +#define CS43130_MCLK_INT_SHIFT 2 +#define CS43130_SP_SRATE_MASK 0x0F +#define CS43130_SP_SRATE_SHIFT 0 +#define CS43130_SP_BITSIZE_ASP_MASK 0x03 +#define CS43130_SP_BITSIZE_ASP_SHIFT 0 +#define CS43130_HP_DETECT_CTRL_SHIFT 6 +#define CS43130_HP_DETECT_CTRL_MASK (0x03 << CS43130_HP_DETECT_CTRL_SHIFT) +#define CS43130_HP_DETECT_INV_SHIFT 5 +#define CS43130_HP_DETECT_INV_MASK (1 << CS43130_HP_DETECT_INV_SHIFT) + +/* CS43130_INT_MASK_1 */ +#define CS43130_HP_PLUG_INT_SHIFT 6 +#define CS43130_HP_PLUG_INT (1 << CS43130_HP_PLUG_INT_SHIFT) +#define CS43130_HP_UNPLUG_INT_SHIFT 5 +#define CS43130_HP_UNPLUG_INT (1 << CS43130_HP_UNPLUG_INT_SHIFT) +#define CS43130_XTAL_RDY_INT_SHIFT 4 +#define CS43130_XTAL_RDY_INT (1 << CS43130_XTAL_RDY_INT_SHIFT) +#define CS43130_XTAL_ERR_INT_SHIFT 3 +#define CS43130_XTAL_ERR_INT (1 << CS43130_XTAL_ERR_INT_SHIFT) + +/*Reg CS43130_SP_BITSIZE*/ +#define CS43130_SP_BIT_SIZE_8 0x00 +#define CS43130_SP_BIT_SIZE_16 0x01 +#define CS43130_SP_BIT_SIZE_24 0x02 +#define CS43130_SP_BIT_SIZE_32 0x03 + +/*PLL*/ +#define CS43130_PLL_START_MASK (0x1<<0) +#define CS43130_PLL_MODE_MASK 0x02 +#define CS43130_PLL_MODE_SHIFT 1 + +#define CS43130_PLL_REF_PREDIV_MASK 0x3 + +#define CS43130_ASP_STP_MASK 0x10 +#define CS43130_ASP_STP_SHIFT 4 +#define CS43130_ASP_5050_MASK 0x08 +#define CS43130_ASP_5050_SHIFT 3 +#define CS43130_ASP_FSD_MASK 0x07 +#define CS43130_ASP_FSD_SHIFT 0 + +#define CS43130_ASP_MODE_MASK 0x10 +#define CS43130_ASP_MODE_SHIFT 4 +#define CS43130_ASP_SCPOL_OUT_MASK 0x08 +#define CS43130_ASP_SCPOL_OUT_SHIFT 3 +#define CS43130_ASP_SCPOL_IN_MASK 0x04 +#define CS43130_ASP_SCPOL_IN_SHIFT 2 +#define CS43130_ASP_LCPOL_OUT_MASK 0x02 +#define CS43130_ASP_LCPOL_OUT_SHIFT 1 +#define CS43130_ASP_LCPOL_IN_MASK 0x01 +#define CS43130_ASP_LCPOL_IN_SHIFT 0 + +/*Reg CS43130_PWDN_CTL*/ +#define CS43130_PDN_XSP_MASK 0x80 +#define CS43130_PDN_XSP_SHIFT 7 +#define CS43130_PDN_ASP_MASK 0x40 +#define CS43130_PDN_ASP_SHIFT 6 +#define CS43130_PDN_DSPIF_MASK 0x20 +#define CS43130_PDN_DSDIF_SHIFT 5 +#define CS43130_PDN_HP_MASK 0x10 +#define CS43130_PDN_HP_SHIFT 4 +#define CS43130_PDN_XTAL_MASK 0x08 +#define CS43130_PDN_XTAL_SHIFT 3 +#define CS43130_PDN_PLL_MASK 0x04 +#define CS43130_PDN_PLL_SHIFT 2 +#define CS43130_PDN_CLKOUT_MASK 0x02 +#define CS43130_PDN_CLKOUT_SHIFT 1 + +#define CS43130_7_0_MASK 0xFF +#define CS43130_15_8_MASK 0xFF00 +#define CS43130_23_16_MASK 0xFF0000 + +/* Reg CS43130_HP_OUT_CTL_1 */ +#define CS43130_HP_IN_EN_SHIFT 3 +#define CS43130_HP_IN_EN_MASK 0x08 + +#define CS43130_ASP_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define CS43130_XSP_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +enum cs43130_asp_rate { + CS43130_ASP_SPRATE_32K = 0, + CS43130_ASP_SPRATE_44_1K, + CS43130_ASP_SPRATE_48K, + CS43130_ASP_SPRATE_88_2K, + CS43130_ASP_SPRATE_96K, + CS43130_ASP_SPRATE_176_4K, + CS43130_ASP_SPRATE_192K, + CS43130_ASP_SPRATE_352_8K, + CS43130_ASP_SPRATE_384K, +}; + +enum cs43130_mclk_src_sel { + CS43130_MCLK_SRC_XTAL = 0, + CS43130_MCLK_SRC_PLL, + CS43130_MCLK_SRC_RCO +}; + +enum cs43130_mode { + CS43130_SLAVE_MODE = 0, + CS43130_MASTER_MODE +}; + +enum cs43130_xtal_ibias { + CS43130_XTAL_IBIAS_15UA = 2, + CS43130_XTAL_IBIAS_12_5UA = 4, + CS43130_XTAL_IBIAS_7_5UA = 6, +}; + +#define CS43130_AIF_BICK_RATE 1 +#define CS43130_SYSCLK_MCLK 1 +#define CS43130_NUM_SUPPLIES 5 +static const char *const cs43130_supply_names[CS43130_NUM_SUPPLIES] = { + "VA", + "VP", + "VCP", + "VD", + "VL", +}; + +#define CS43130_NUM_INT 5 /* number of interrupt status reg */ + +struct cs43130_private { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct regulator_bulk_data supplies[CS43130_NUM_SUPPLIES]; + /* codec device ID */ + unsigned int dev_id; + int mclk; + int sclk; + int xtal_ibias; + + bool pll_bypass; + int pll_out; + int mclk_int; + int dai_format; + int dai_mode; + int dai_bit; + int asp_size; + int fs; + bool bick_invert; + bool lrck_invert; + int bick; + struct gpio_desc *reset_gpio; +}; + +#endif /* __CS43130_H__ */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: fallback mechanism if MCLK is not enabled
by Pierre-Louis Bossart 15 Dec '16

15 Dec '16

Commit df1a2776a795 ("ASoC: Intel: bytcr_rt5640: add MCLK support") was merged but the corresponding clock framework patches have not, after being bumped from audio to clock to x86 domains. The missing clock-related patches result in a regression starting with 4.9 with the audio card not being created. Rather than reverting this commit and all following updates already queued up for 4.10, handle run-time dependency on MCLK and fall back to the previous bit-clock mode. This provides the same functionality as in 4.8 for Baytrail devices. On Baytrail-CR most devices remain silent with this fallback but additional patches are needed anyway. This patch should be applied to -stable as well as ASoC 4.10 fixes Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 507a86a..4a8b509 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -828,7 +828,9 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Failed to get MCLK from pmc_plt_clk_3: %ld\n", PTR_ERR(priv->mclk)); - return PTR_ERR(priv->mclk); + + /* Fall back to bitclock only */ + byt_rt5640_quirk &= ~BYT_RT5640_MCLK_EN; } } -- 2.9.3

3 5

[alsa-devel] [PATCH] ASoC: dwc: Fix PIO mode initialization
by Jose Abreu 14 Dec '16

14 Dec '16

We can no longer rely on the return value of devm_snd_dmaengine_pcm_register(...) to check if the DMA handle is declared in the DT. Previously this check activated PIO mode but currently dma_request_chan returns either a valid channel or -EPROBE_DEFER. In order to activate PIO mode check instead if the interrupt line is declared. This reflects better what is documented in the DT bindings (see Documentation/devicetree/bindings/sound/ designware-i2s.txt). Also, initialize use_pio variable which was never being set causing PIO mode to never work. Signed-off-by: Jose Abreu <joabreu(a)synopsys.com> Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/dwc/designware_i2s.c | 25 +++++++++++-------------- 1 file changed, 11 insertions(+), 14 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 2998954..bdf8398 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -681,22 +681,19 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (!pdata) { - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); - if (ret == -EPROBE_DEFER) { - dev_err(&pdev->dev, - "failed to register PCM, deferring probe\n"); - return ret; - } else if (ret) { - dev_err(&pdev->dev, - "Could not register DMA PCM: %d\n" - "falling back to PIO mode\n", ret); + if (irq >= 0) { ret = dw_pcm_register(pdev); - if (ret) { - dev_err(&pdev->dev, - "Could not register PIO PCM: %d\n", + dev->use_pio = true; + } else { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + 0); + dev->use_pio = false; + } + + if (ret) { + dev_err(&pdev->dev, "could not register pcm: %d\n", ret); - goto err_clk_disable; - } + goto err_clk_disable; } } -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] ASoC: nau8825: Add Class G function options
by Mark Brown 14 Dec '16

14 Dec '16

On Wed, Dec 14, 2016 at 04:40:15PM +0800, John Hsu wrote: > On 12/13/2016 2:12 PM, Ben Zhang wrote: > > +static SOC_ENUM_SINGLE_DECL( > > + nau8825_class_g_enum, SND_SOC_NOPM, 0, nau8825_class_g_src); > > + > The register SND_SOC_NOPM can't use at the enum put and get function > in the older kernel. Ex. kernel 3.14. The driver has to modify the > DAPM in the platform with older kernel. Older kernels are not relevant upstream. If you need to backport that should be handled separately.

1 0

[alsa-devel] [PATCH] alsa-lib: update linked hw_ptr and appl_ptr
by sutar.mounesh＠gmail.com 14 Dec '16

14 Dec '16

From: Andreas Pape <apape(a)de.adit-jv.com> Plugin file provides no private hw_ptr and appl_ptr but instead links them to the slave pcm. If the slave pcm itself changes its hw_ptr or app_prt this needs to be done in file plugin, too. Plugin 'plug' is such a candidate changing the hw_ptr and app_ptr in hw_params call dependent on the automatically inserted plugins. ALSA unfortunately has no support for automatically updating chained pointers. A notification on pointer change seems to be prepared inside the snd_pcm_set_ptr() routine via rbptr->changed(), but it is not (yet) implemented so that we need to care for it manually. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> --- src/pcm/pcm_file.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index fec76eb..1e3d35b 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -634,6 +634,14 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) return err; } } + + /* pointer may have changed - e.g if plug is used. */ + snd_pcm_unlink_hw_ptr(pcm, file->gen.slave); + snd_pcm_unlink_appl_ptr(pcm, file->gen.slave); + + snd_pcm_link_hw_ptr(pcm, file->gen.slave); + snd_pcm_link_appl_ptr(pcm, file->gen.slave); + return 0; } -- 1.7.9.5

3 3

[alsa-devel] [PATCH v2 2/2] ASoC: cs43130: Add devicetree bindings for CS43130
by Li Xu 13 Dec '16

13 Dec '16

Add devicetree bindings documentation file for Cirrus Logic CS43130 codec. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs43130.txt | 41 ++++++++++++++++++++++ 1 file changed, 41 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs43130.txt diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt new file mode 100644 index 0000000..1af6b78 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs43130.txt @@ -0,0 +1,41 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : Active low GPIO used to reset the device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + +Example: + +cs43130: audio-codec@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 1>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; +}; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Add support for CS43130 codec
by Li Xu 13 Dec '16

13 Dec '16

Add support for Cirrus Logic CS43130 codec. Support I2C control and I2S audio playback. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs43130.c | 1106 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs43130.h | 268 +++++++++++ 4 files changed, 1382 insertions(+) create mode 100644 sound/soc/codecs/cs43130.c create mode 100644 sound/soc/codecs/cs43130.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e1718a..c07bc0d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 select SND_SOC_CS53L30 if I2C + select SND_SOC_CS43130 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -486,6 +487,11 @@ config SND_SOC_CS53L30 tristate "Cirrus Logic CS53L30 CODEC" depends on I2C +# Cirrus Logic CS43130 HiFi DAC +config SND_SOC_CS43130 + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7e1dad7..7e41526 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o snd-soc-cs53l30-objs := cs53l30.o +snd-soc-cs43130-objs := cs43130.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -284,6 +285,7 @@ obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o +obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c new file mode 100644 index 0000000..cf1c60c --- /dev/null +++ b/sound/soc/codecs/cs43130.c @@ -0,0 +1,1106 @@ +/* + * cs43130.c -- CS43130 ALSA Soc Audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Authors: Li Xu <li.xu(a)cirrus.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/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/of_irq.h> + +#include "cs43130.h" + + +static const struct reg_default cs43130_reg_defaults[] = { + { CS43130_SYS_CLK_CTL_1, 0x06 }, + { CS43130_SP_SRATE, 0x01 }, + { CS43130_SP_BITSIZE, 0x05 }, + { CS43130_PAD_INT_CFG, 0x03 }, + { CS43130_PWDN_CTL, 0xFE }, + { CS43130_CRYSTAL_SET, 0x04 }, + { CS43130_PLL_SET_1, 0x00 }, + { CS43130_PLL_SET_2, 0x00 }, + { CS43130_PLL_SET_3, 0x00 }, + { CS43130_PLL_SET_4, 0x00 }, + { CS43130_PLL_SET_5, 0x40 }, + { CS43130_PLL_SET_6, 0x10 }, + { CS43130_PLL_SET_7, 0x80 }, + { CS43130_PLL_SET_8, 0x03 }, + { CS43130_PLL_SET_9, 0x02 }, + { CS43130_PLL_SET_10, 0x02 }, + { CS43130_CLKOUT_CTL, 0x00 }, + { CS43130_ASP_NUM_1, 0x01 }, + { CS43130_ASP_NUM_2, 0x00 }, + { CS43130_ASP_DENOM_1, 0x08 }, + { CS43130_ASP_DENOM_2, 0x00 }, + { CS43130_ASP_LRCK_HI_TIME_1, 0x1F }, + { CS43130_ASP_LRCK_HI_TIME_2, 0x00 }, + { CS43130_ASP_LRCK_PERIOD_1, 0x3F }, + { CS43130_ASP_LRCK_PERIOD_2, 0x00 }, + { CS43130_ASP_CLOCK_CONF, 0x0C }, + { CS43130_ASP_FRAME_CONF, 0x0A }, + { CS43130_XSP_NUM_1, 0x01 }, + { CS43130_XSP_NUM_2, 0x00 }, + { CS43130_XSP_DENOM_1, 0x02 }, + { CS43130_XSP_DENOM_2, 0x00 }, + { CS43130_XSP_LRCK_HI_TIME_1, 0x1F }, + { CS43130_XSP_LRCK_HI_TIME_2, 0x00 }, + { CS43130_XSP_LRCK_PERIOD_1, 0x3F }, + { CS43130_XSP_LRCK_PERIOD_2, 0x00 }, + { CS43130_XSP_CLOCK_CONF, 0x0C }, + { CS43130_XSP_FRAME_CONF, 0x0A }, + { CS43130_ASP_CH_1_LOC, 0x00 }, + { CS43130_ASP_CH_2_LOC, 0x00 }, + { CS43130_ASP_CH_1_SZ_EN, 0x06 }, + { CS43130_ASP_CH_2_SZ_EN, 0x0E }, + { CS43130_XSP_CH_1_LOC, 0x00 }, + { CS43130_XSP_CH_2_LOC, 0x00 }, + { CS43130_XSP_CH_1_SZ_EN, 0x06 }, + { CS43130_XSP_CH_2_SZ_EN, 0x0E }, + { CS43130_DSD_VOL_B, 0x78 }, + { CS43130_DSD_VOL_A, 0x78 }, + { CS43130_DSD_PATH_CTL_1, 0xA8 }, + { CS43130_DSD_INT_CFG, 0x00 }, + { CS43130_DSD_PATH_CTL_2, 0x00 }, + { CS43130_DSD_PCM_MIX_CTL, 0x00 }, + { CS43130_DSD_PATH_CTL_3, 0x40 }, + { CS43130_HP_OUT_CTL_1, 0x30 }, + { CS43130_PCM_FILT_OPT, 0x02 }, + { CS43130_PCM_VOL_B, 0x78 }, + { CS43130_PCM_VOL_A, 0x78 }, + { CS43130_PCM_PATH_CTL_1, 0xA8 }, + { CS43130_PCM_PATH_CTL_2, 0x00 }, + { CS43130_CLASS_H_CTL, 0x1E }, + { CS43130_HP_DETECT, 0x04 }, + { CS43130_HP_LOAD_1, 0x00 }, + { CS43130_HP_MEAS_LOAD_1, 0x00 }, + { CS43130_HP_MEAS_LOAD_2, 0x00 }, + { CS43130_INT_MASK_1, 0xFF }, + { CS43130_INT_MASK_2, 0xFF }, + { CS43130_INT_MASK_3, 0xFF }, + { CS43130_INT_MASK_4, 0xFF }, + { CS43130_INT_MASK_5, 0xFF }, +}; + +static bool cs43130_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_DEVID_AB ... CS43130_SUBREV_ID: + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + return true; + default: + return false; + } +} + +static bool cs43130_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_DEVID_AB ... CS43130_SYS_CLK_CTL_1: + case CS43130_SP_SRATE ... CS43130_PAD_INT_CFG: + case CS43130_DXD1: + case CS43130_PWDN_CTL: + case CS43130_DXD2: + case CS43130_CRYSTAL_SET: + case CS43130_PLL_SET_1 ... CS43130_PLL_SET_5: + case CS43130_PLL_SET_6: + case CS43130_PLL_SET_7: + case CS43130_PLL_SET_8: + case CS43130_PLL_SET_9: + case CS43130_PLL_SET_10: + case CS43130_CLKOUT_CTL: + case CS43130_ASP_NUM_1 ... CS43130_ASP_FRAME_CONF: + case CS43130_XSP_NUM_1 ... CS43130_XSP_FRAME_CONF: + case CS43130_ASP_CH_1_LOC: + case CS43130_ASP_CH_2_LOC: + case CS43130_ASP_CH_1_SZ_EN: + case CS43130_ASP_CH_2_SZ_EN: + case CS43130_XSP_CH_1_LOC: + case CS43130_XSP_CH_2_LOC: + case CS43130_XSP_CH_1_SZ_EN: + case CS43130_XSP_CH_2_SZ_EN: + case CS43130_DSD_VOL_B ... CS43130_DSD_PATH_CTL_3: + case CS43130_HP_OUT_CTL_1: + case CS43130_PCM_FILT_OPT ... CS43130_PCM_PATH_CTL_2: + case CS43130_CLASS_H_CTL: + case CS43130_HP_DETECT: + case CS43130_HP_STATUS: + case CS43130_HP_LOAD_1: + case CS43130_HP_MEAS_LOAD_1: + case CS43130_HP_MEAS_LOAD_2: + case CS43130_HP_DC_STAT_1: + case CS43130_HP_DC_STAT_2: + case CS43130_HP_AC_STAT_1: + case CS43130_HP_AC_STAT_2: + case CS43130_HP_LOAD_STAT: + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + case CS43130_INT_MASK_1 ... CS43130_INT_MASK_5: + return true; + default: + return false; + } +} + +static bool cs43130_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + return true; + default: + return false; + } +} + +struct cs43130_pll_params { + u32 pll_in; + u8 mclk_int; + u8 sclk_prediv; + u8 pll_div_int; + u32 pll_div_frac; + u8 pll_mode; + u8 pll_divout; + u32 pll_out; + u8 pll_cal_ratio; +}; + +static const struct cs43130_pll_params pll_ratio_table[] = { + { 9600000, 1, 0x02, 0x49, 0x800000, 0x00, 0x08, 22579200, 151 }, + { 9600000, 0, 0x02, 0x50, 0x000000, 0x00, 0x08, 24576000, 164 }, + + { 11289600, 1, 0x02, 0X40, 0, 0x01, 0x08, 22579200, 128 }, + { 11289600, 0, 0x02, 0x44, 0x06F700, 0x0, 0x08, 24576000, 139 }, + + { 12000000, 1, 0x02, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120 }, + { 12000000, 0, 0x02, 0x40, 0x000000, 0x00, 0x08, 24576000, 131 }, + + { 12288000, 1, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118 }, + { 12288000, 0, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128 }, + + { 13000000, 1, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118 }, + { 13000000, 0, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128 }, + + { 19200000, 1, 0x02, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111 }, + { 19200000, 0, 0x02, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121 }, + + { 22579200, 1, 0, 0, 0, 0, 0, 22579200, 0 }, + { 22579200, 0, 0x03, 0x44, 0x06F700, 0x00, 0x08, 24576000, 139 }, + + { 24000000, 1, 0x03, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120 }, + { 24000000, 0, 0x03, 0x40, 0x000000, 0x00, 0x08, 24576000, 131 }, + + { 24576000, 1, 0x03, 0x49, 0x800000, 0x01, 0x0A, 22579200, 128 }, + { 24576000, 0, 0, 0, 0, 0, 0, 24576000, 0 }, + + { 26000000, 1, 0x03, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111 }, + { 26000000, 0, 0x03, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121 }, +}; + +static int cs43130_pll_config(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "%s: cs43130->mclk = %d, cs43130->pll_out = %d", + __func__, cs43130->mclk, cs43130->pll_out); + for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { + if (pll_ratio_table[i].pll_in == cs43130->mclk && + pll_ratio_table[i].pll_out == cs43130->pll_out) { + + cs43130->mclk_int = pll_ratio_table[i].mclk_int; + + if (pll_ratio_table[i].pll_cal_ratio == 0) { + if (cs43130->xtal_ibias > 0) { + usleep_range(1000, 1050); + /*PDN_XTAL = 0,enable*/ + regmap_update_bits(cs43130->regmap, + CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, 0 + << CS43130_PDN_XTAL_SHIFT); + } + + /* PLL_START = 0, disable PLL_START */ + regmap_update_bits(cs43130->regmap, + CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, + 0 << CS43130_PLL_START_MASK); + + cs43130->pll_bypass = true; + return 0; + } + + /*PDN_PLL= 0,enable*/ + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, 0 + << CS43130_PDN_PLL_SHIFT); + + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_9, + CS43130_PLL_REF_PREDIV_MASK, + pll_ratio_table[i].sclk_prediv + ); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_5, + pll_ratio_table[i].pll_div_int); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_2, + pll_ratio_table[i].pll_div_frac & + CS43130_7_0_MASK); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_3, + (pll_ratio_table[i].pll_div_frac & + CS43130_15_8_MASK) >> 8); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_4, + (pll_ratio_table[i].pll_div_frac & + CS43130_23_16_MASK) >> 16); + + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_8, + CS43130_PLL_MODE_MASK, + pll_ratio_table[i].pll_mode + << CS43130_PLL_MODE_SHIFT); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_6, + pll_ratio_table[i].pll_divout); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_7, + pll_ratio_table[i].pll_cal_ratio); + + /* PLL_START = 1, enable PLL_START */ + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, + CS43130_PLL_START_MASK); + cs43130->pll_bypass = false; + return 0; + } + } + return -EINVAL; +} + +static int cs43130_format_config(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + int period_size = 0; + int pulse_width = 0; + int asp_fsd; + int asp_stp; + int bick_inv; + int asp_m = 0; + int asp_sprate = 0; + int ret = 0; + unsigned int bitwidth_sclk = (cs43130->sclk / cs43130->fs) / 2; + unsigned int bitwidth_dai = (cs43130->dai_bit + 1) * 8; + + if (cs43130->fs) { + if (bitwidth_sclk < bitwidth_dai) { + dev_err(codec->dev, "Format not supported\n"); + return -EINVAL; + } + period_size = cs43130->sclk / cs43130->fs; + pulse_width = period_size/2; + + if (cs43130->sclk != 0) + asp_m = cs43130->pll_out / cs43130->sclk; + } + dev_dbg(codec->dev, "%s: cs43130->sclk = %d, cs43130->fs = %d, cs43130->dai_bit = %d", + __func__, cs43130->sclk, cs43130->fs, cs43130->dai_bit); + dev_dbg(codec->dev, "%s: period_size = %d, pulse_width = %d, asp_m = %d", + __func__, period_size, pulse_width, asp_m); + + if (cs43130->dai_format) { + /*MSB*/ + bick_inv = 0; + asp_fsd = 0; + asp_stp = 1; + + } else { + /*I2S*/ + bick_inv = 1; + asp_fsd = 2; /* one bick delay */ + asp_stp = 0; + } + dev_dbg(codec->dev, + "%s: cs43130->dai_format = %d, bick_inv = %d, asp_fsd = %d, asp_stp = %d", + __func__, cs43130->dai_format, bick_inv, asp_fsd, asp_stp); + + switch (cs43130->fs) { + case 32000: + asp_sprate = CS43130_ASP_SPRATE_32K; + break; + case 44100: + asp_sprate = CS43130_ASP_SPRATE_44_1K; + break; + case 48000: + asp_sprate = CS43130_ASP_SPRATE_48K; + break; + case 88200: + asp_sprate = CS43130_ASP_SPRATE_88_2K; + break; + case 96000: + asp_sprate = CS43130_ASP_SPRATE_96K; + break; + case 176400: + asp_sprate = CS43130_ASP_SPRATE_176_4K; + break; + case 192000: + asp_sprate = CS43130_ASP_SPRATE_192K; + break; + case 352800: + asp_sprate = CS43130_ASP_SPRATE_352_8K; + break; + case 384000: + asp_sprate = CS43130_ASP_SPRATE_384K; + break; + default: + dev_err(codec->dev, "sample rate(%d) not supported\n", + cs43130->fs); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: asp_sprate = %d, cs43130->asp_size = %d", + __func__, asp_sprate, cs43130->asp_size); + + /* ASP_SPRATE = fs*/ + regmap_write(cs43130->regmap, CS43130_SP_SRATE, asp_sprate); + /*ASP_SPSIZE*/ + regmap_update_bits(cs43130->regmap, CS43130_SP_BITSIZE, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->asp_size); + + + /* Set up slave mode */ + /*BICK = ASP_N/ASP_M * PLL_OUT */ + /* ASP_N = 1 */ + regmap_write(cs43130->regmap, CS43130_ASP_NUM_1, 1); + regmap_write(cs43130->regmap, CS43130_ASP_NUM_2, 0); + + /*ASP_M*/ + regmap_write(cs43130->regmap, CS43130_ASP_DENOM_1, asp_m & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_DENOM_2, (asp_m >> 8) & 0x3f); + + + /* H / L ratio of LRCK*/ + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_1, + (pulse_width-1) & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_2, + ((pulse_width-1) >> 8) & 0xff); + + /* the period of LRCK*/ + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_1, + (period_size-1) & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_2, + ((period_size-1) >> 8) & 0xff); + + /*resolution*/ + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_1_SZ_EN, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->dai_bit); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_2_SZ_EN, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->dai_bit); + + regmap_update_bits(cs43130->regmap, CS43130_ASP_FRAME_CONF, + CS43130_ASP_FSD_MASK, asp_fsd << CS43130_ASP_FSD_SHIFT); + + regmap_update_bits(cs43130->regmap, CS43130_ASP_FRAME_CONF, + CS43130_ASP_STP_MASK, asp_stp << CS43130_ASP_STP_SHIFT); + + /* set clk master/slave */ + dev_dbg(codec->dev, "%s: cs43130->dai_mode = %d", + __func__, cs43130->dai_mode); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CLOCK_CONF, + CS43130_ASP_MODE_MASK, cs43130->dai_mode << CS43130_ASP_MODE_SHIFT); + + return ret; +} + +static int cs43130_change_clksrc(struct snd_soc_codec *codec, + enum cs43130_mclk_src_sel src) +{ + int ret = 0; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, cs43130->mclk_int << CS43130_MCLK_INT_SHIFT); + + usleep_range(150, 200); + + return ret; +} + +static int cs43130_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + unsigned int bitwidth; + int ret = 0; + + cs43130->fs = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_8; + cs43130->asp_size = CS43130_SP_BIT_SIZE_32; + break; + case SNDRV_PCM_FORMAT_S16_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_16; + cs43130->asp_size = CS43130_SP_BIT_SIZE_24; + break; + case SNDRV_PCM_FORMAT_S24_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_24; + cs43130->asp_size = CS43130_SP_BIT_SIZE_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_32; + cs43130->asp_size = CS43130_SP_BIT_SIZE_8; + break; + default: + dev_err(codec->dev, "Format(%d) not supported", + params_format(params)); + return -EINVAL; + } + + bitwidth = (cs43130->dai_bit+1)*8; + dev_dbg(codec->dev, "(data bit)%d: (rate)%d", + bitwidth, cs43130->fs); + + ret = cs43130_format_config(codec); + return ret; +} + +static const DECLARE_TLV_DB_SCALE(pcm_vol_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new cs43130_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS43130_PCM_VOL_A, CS43130_PCM_VOL_B, 0, 0xFF, 1, + pcm_vol_tlv), + SOC_SINGLE("Swap L/R", CS43130_PCM_PATH_CTL_2, 1, 1, 0), + SOC_SINGLE("Copy L/R", CS43130_PCM_PATH_CTL_2, 0, 1, 0), +}; + +static int cs43130_aspin_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); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_XTAL); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + + usleep_range(10000, 10050); + /*ASP_3ST = 0 in master mode*/ + if (cs43130->dai_mode) + regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG, + 0x01, 0x00); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_CLKOUT_MASK, 0 + << CS43130_PDN_CLKOUT_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + break; + default: + dev_err(codec->dev, "Invalid ASPOUT event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static int cs43130_dac_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); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, 1 + << CS43130_PDN_XTAL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, 1 + << CS43130_PDN_PLL_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAC event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static int cs43130_hpin_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); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + regmap_write(cs43130->regmap, CS43130_DXD1, 0x99); + regmap_update_bits(cs43130->regmap, CS43130_HP_OUT_CTL_1, + CS43130_HP_IN_EN_MASK, 0 << CS43130_HP_IN_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_DXD2, 0x00); + regmap_write(cs43130->regmap, CS43130_DXD1, 0x00); + break; + case SND_SOC_DAPM_POST_PMU: + regmap_write(cs43130->regmap, CS43130_DXD1, 0x99); + regmap_write(cs43130->regmap, CS43130_DXD2, 0x01); + regmap_update_bits(cs43130->regmap, CS43130_HP_OUT_CTL_1, + CS43130_HP_IN_EN_MASK, 1 << CS43130_HP_IN_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_DXD1, 0x00); + break; + default: + dev_err(codec->dev, "Invalid HPIN event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct snd_soc_dapm_widget cs43130_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + + SND_SOC_DAPM_AIF_IN_E("ASPIN", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_ASP_SHIFT, 1, cs43130_aspin_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_DAC_E("HiFi DAC", + NULL, CS43130_PWDN_CTL, CS43130_PDN_HP_SHIFT, 1, + cs43130_dac_event, + (SND_SOC_DAPM_PRE_PMD) + ), + + SND_SOC_DAPM_LINE("Analog Playback", cs43130_hpin_event), +}; + +static const struct snd_soc_dapm_route cs43130_routes[] = { + {"ASPIN", NULL, "DAC Playback"}, + {"HiFi DAC", NULL, "ASPIN"}, + + {"HPOUTA", NULL, "HiFi DAC"}, + {"HPOUTB", NULL, "HiFi DAC"}, + {"HPOUTA", NULL, "Analog Playback"}, + {"HPOUTB", NULL, "Analog Playback"}, +}; + +static const unsigned int cs43130_src_rates[] = { + 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs43130_constraints = { + .count = ARRAY_SIZE(cs43130_src_rates), + .list = cs43130_src_rates, +}; + +static int cs43130_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs43130_constraints); + return 0; +} + +static int cs43130_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs43130->dai_mode = CS43130_SLAVE_MODE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs43130->dai_mode = CS43130_MASTER_MODE; + break; + default: + dev_err(codec->dev, "unsupported i2s master mode\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs43130->dai_format = 0; + break; + case SND_SOC_DAIFMT_LEFT_J: + cs43130->dai_format = 1; + break; + default: + dev_err(codec->dev, "unsupported audio format except I2S and MSB\n"); + return -EINVAL; + } + + /* BICK/LRCK pority */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + cs43130->bick_invert = false; + cs43130->lrck_invert = false; + break; + case SND_SOC_DAIFMT_IB_NF: + cs43130->bick_invert = true; + cs43130->lrck_invert = false; + break; + case SND_SOC_DAIFMT_NB_IF: + cs43130->bick_invert = false; + cs43130->lrck_invert = true; + break; + case SND_SOC_DAIFMT_IB_IF: + cs43130->bick_invert = true; + cs43130->lrck_invert = true; + break; + default: + dev_err(codec->dev, "unsupported audio polarity\n"); + return -EINVAL; + } + + return 0; +} + + +static int cs43130_set_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + unsigned int reg; + u8 mute_reg; + + regmap_read(cs43130->regmap, CS43130_PCM_PATH_CTL_1, &reg); + mute_reg = reg & 0xfc; + if (mute) + regmap_write(cs43130->regmap, CS43130_PCM_PATH_CTL_1, + mute_reg | 0x03); + else + regmap_write(cs43130->regmap, CS43130_PCM_PATH_CTL_1, mute_reg); + + return ret; +} + +static int cs43130_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + + switch (div_id) { + case CS43130_AIF_BICK_RATE: + cs43130->bick = div; + break; + default: + dev_err(codec->dev, + "Unsupported divide value: div_id = %d", div_id); + return -EINVAL; + } + return 0; +} + +static int cs43130_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + int ret = 0; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + if (freq_in < 9600000 || freq_in > 26000000) { + dev_err(codec->dev, + "unsupported pll input reference clock:%d\n", freq_in); + return -EINVAL; + } + + switch (freq_in) { + case 9600000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 19200000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs43130->mclk = freq_in; + break; + default: + dev_err(codec->dev, + "unsupported pll input reference clock:%d\n", freq_in); + return -EINVAL; + } + + switch (freq_out) { + case 22579200: + cs43130->pll_out = freq_out; + cs43130->mclk_int = 1; + break; + case 24576000: + cs43130->pll_out = freq_out; + cs43130->mclk_int = 0; + break; + default: + dev_err(codec->dev, + "unsupported pll output reference clock:%d\n", + freq_out); + return -EINVAL; + } + + ret = cs43130_pll_config(codec); + dev_dbg(codec->dev, "%s: cs43130->pll_bypass = %d", + __func__, cs43130->pll_bypass); + return ret; +} + +static int cs43130_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: clk_id = %d, freq = %d, dir = %d", + __func__, clk_id, freq, dir); + cs43130->sclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops cs43130_dai_ops = { + .startup = cs43130_pcm_startup, + .hw_params = cs43130_pcm_hw_params, + .set_sysclk = cs43130_dai_set_sysclk, + .set_fmt = cs43130_set_dai_fmt, + .digital_mute = cs43130_set_mute, + .set_clkdiv = cs43130_set_clkdiv, +}; + +static struct snd_soc_dai_driver cs43130_dai[] = { + { + .name = "cs43130_hifi", + .id = 0, + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_ASP_FORMATS, + }, + .ops = &cs43130_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-xsp", + .id = 1, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_XSP_FORMATS, + }, + .symmetric_rates = 1, + }, +}; + +static int cs43130_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + /* 24576000 is not supported */ + unsigned int mclk_int_freq = 22579200; + + dev_dbg(codec->dev, "%s: clk_id = %d, source = %d, freq = %d, dir = %d\n", + __func__, clk_id, source, freq, dir); + /* + * freq is external mclk freq + * if freq == mclk_int_freq, pll is bypassed + * modify mclk_int_freq as needed for application + */ + cs43130_set_pll(codec, 0, 0, freq, mclk_int_freq); + return 0; +} + +static irqreturn_t cs43130_irq_thread(int irq, void *data) +{ + struct cs43130_private *cs43130 = + (struct cs43130_private *)data; + unsigned int stickies[CS43130_NUM_INT]; + unsigned int masks[CS43130_NUM_INT]; + unsigned int i; + + /* Read all INT status and mask reg */ + regmap_bulk_read(cs43130->regmap, CS43130_INT_STATUS_1, + stickies, CS43130_NUM_INT * sizeof(unsigned int)); + regmap_bulk_read(cs43130->regmap, CS43130_INT_MASK_1, + masks, CS43130_NUM_INT * sizeof(unsigned int)); + + for (i = 0; i < ARRAY_SIZE(stickies); i++) + stickies[i] = stickies[i] & (~masks[i]); + + if (stickies[0] & CS43130_XTAL_RDY_INT) + pr_debug("%s: Crystal ready\n", __func__); + + if (stickies[0] & CS43130_XTAL_ERR_INT) + pr_debug("%s: Crystal err\n", __func__); + + if (stickies[0] & CS43130_HP_PLUG_INT) + pr_debug("%s: HP plugged\n", __func__); + + if (stickies[0] & CS43130_HP_UNPLUG_INT) + pr_debug("%s: HP unplugged\n", __func__); + + return IRQ_HANDLED; +} +static struct snd_soc_codec_driver soc_codec_dev_cs43130 = { + .component_driver = { + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + .dapm_widgets = cs43130_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs43130_dapm_widgets), + .dapm_routes = cs43130_routes, + .num_dapm_routes = ARRAY_SIZE(cs43130_routes), + }, + .set_sysclk = cs43130_codec_set_sysclk, + .set_pll = cs43130_set_pll, +}; + +static const struct regmap_config cs43130_regmap = { + .reg_bits = 24, + .pad_bits = 8, + .val_bits = 8, + + .max_register = CS43130_LASTREG, + .reg_defaults = cs43130_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs43130_reg_defaults), + .readable_reg = cs43130_readable_register, + .precious_reg = cs43130_precious_register, + .volatile_reg = cs43130_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs43130_handle_device_data( + struct i2c_client *i2c_client, struct cs43130_private *cs43130) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + int ret = 0; + + of_property_read_u32(np, "cirrus,xtal-ibias", &val); + switch (val) { + case 1: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_7_5UA; + break; + case 2: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_12_5UA; + break; + case 3: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_15UA; + break; + default: + dev_info(&i2c_client->dev, + "cirrus,xtal-ibias value or xtal unused %d", + val); + } + return ret; +} + +static int cs43130_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs43130_private *cs43130; + int ret; + unsigned int devid = 0; + unsigned int reg; + int i; + + cs43130 = devm_kzalloc(&client->dev, sizeof(*cs43130), GFP_KERNEL); + if (cs43130 == NULL) + return -ENOMEM; + + i2c_set_clientdata(client, cs43130); + + cs43130->regmap = devm_regmap_init_i2c(client, &cs43130_regmap); + if (IS_ERR(cs43130->regmap)) { + ret = PTR_ERR(cs43130->regmap); + return ret; + } + + if (client->dev.of_node) { + ret = cs43130_handle_device_data(client, cs43130); + if (ret != 0) + return ret; + } + for (i = 0; i < ARRAY_SIZE(cs43130->supplies); i++) + cs43130->supplies[i].supply = cs43130_supply_names[i]; + + ret = devm_regulator_bulk_get(&client->dev, + ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, + "Failed to request supplies: %d\n", ret); + return ret; + } + ret = regulator_bulk_enable(ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + cs43130->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs43130->reset_gpio)) + return PTR_ERR(cs43130->reset_gpio); + + usleep_range(2000, 2050); + + /* initialize codec */ + ret = regmap_read(cs43130->regmap, CS43130_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + switch (devid) { + case CS43130_CHIP_ID: + break; + case CS4399_CHIP_ID: + break; + default: + dev_err(&client->dev, + "CS43130 Device ID (%X). Expected ID %X or %X\n", + devid, CS43130_CHIP_ID, CS4399_CHIP_ID); + ret = -ENODEV; + goto err; + } + + cs43130->dev_id = devid; + ret = regmap_read(cs43130->regmap, CS43130_REV_ID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + goto err; + } + + dev_info(&client->dev, + "Cirrus Logic CS43130 (%x), Revision: %02X\n", devid, + reg & 0xFF); + + /* Enable interrupt handler */ + ret = devm_request_threaded_irq(&client->dev, + client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(&client->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + /* Unmask INT */ + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT | CS43130_XTAL_ERR_INT | + CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT, 0); + + /* Enable HP detect */ + regmap_update_bits(cs43130->regmap, CS43130_HP_DETECT, + CS43130_HP_DETECT_CTRL_MASK, CS43130_HP_DETECT_CTRL_MASK); + + regmap_write(cs43130->regmap, + CS43130_CRYSTAL_SET, cs43130->xtal_ibias); + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs43130, cs43130_dai, + ARRAY_SIZE(cs43130_dai)); + + if (ret < 0) { + dev_err(&client->dev, + "%s: snd_soc_register_codec failed with ret = %d\n", + __func__, ret); + goto err; + } + return 0; +err: + return ret; + +} + +static int cs43130_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +#ifdef CONFIG_PM +static int cs43130_runtime_suspend(struct device *dev) +{ + return 0; +} + +static int cs43130_runtime_resume(struct device *dev) +{ + return 0; +} +#endif + +static const struct dev_pm_ops cs43130_runtime_pm = { + SET_RUNTIME_PM_OPS(cs43130_runtime_suspend, cs43130_runtime_resume, + NULL) +}; + +static const struct of_device_id cs43130_of_match[] = { + { .compatible = "cirrus,cs43130", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs43130_of_match); + +static const struct i2c_device_id cs43130_i2c_id[] = { + {"cs43130", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs43130_i2c_id); + +static struct i2c_driver cs43130_i2c_driver = { + .driver = { + .name = "cs43130", + .of_match_table = cs43130_of_match, + }, + .id_table = cs43130_i2c_id, + .probe = cs43130_i2c_probe, + .remove = cs43130_i2c_remove, +}; + +module_i2c_driver(cs43130_i2c_driver); + +MODULE_AUTHOR("Li Xu <li.xu(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS43130 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h new file mode 100644 index 0000000..bceae76 --- /dev/null +++ b/sound/soc/codecs/cs43130.h @@ -0,0 +1,268 @@ +/* + * ALSA SoC CS43130 codec driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Li Xu <li.xu(a)cirrus.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS43130_H__ +#define __CS43130_H__ + +/* CS43130 registers addresses */ +/* all reg address is shifted by a byte for control byte to be LSB */ +#define CS43130_FIRSTREG 0x010000 +#define CS43130_LASTREG 0x0F0014 +#define CS43130_CHIP_ID 0x00043130 +#define CS4399_CHIP_ID 0x00043990 +#define CS43130_DEVID_AB 0x010000 /*Device ID A & B [RO]*/ +#define CS43130_DEVID_CD 0x010001 /*Device ID C & D [RO]*/ +#define CS43130_DEVID_E 0x010002 /*Device ID E [RO]*/ +#define CS43130_FAB_ID 0x010003 /*Fab ID [RO]*/ +#define CS43130_REV_ID 0x010004 /*Revision ID [RO]*/ +#define CS43130_SUBREV_ID 0x010005 /*Subrevision ID*/ +#define CS43130_SYS_CLK_CTL_1 0x010006 /*System Clocking Ctl 1*/ +#define CS43130_SP_SRATE 0x01000B /*Serial Port Sample Rate*/ +#define CS43130_SP_BITSIZE 0x01000C /*Serial Port Bit Size*/ +#define CS43130_PAD_INT_CFG 0x01000D /*Pad Interface Config*/ +#define CS43130_DXD1 0x010010 /*DXD1*/ +#define CS43130_PWDN_CTL 0x020000 /*Power Down Ctl*/ +#define CS43130_DXD2 0x020019 /*DXD2*/ +#define CS43130_CRYSTAL_SET 0x020052 /*Crystal Setting*/ +#define CS43130_PLL_SET_1 0x030001 /*PLL Setting 1*/ +#define CS43130_PLL_SET_2 0x030002 /*PLL Setting 2*/ +#define CS43130_PLL_SET_3 0x030003 /*PLL Setting 3*/ +#define CS43130_PLL_SET_4 0x030004 /*PLL Setting 4*/ +#define CS43130_PLL_SET_5 0x030005 /*PLL Setting 5*/ +#define CS43130_PLL_SET_6 0x030008 /*PLL Setting 6*/ +#define CS43130_PLL_SET_7 0x03000A /*PLL Setting 7*/ +#define CS43130_PLL_SET_8 0x03001B /*PLL Setting 8*/ +#define CS43130_PLL_SET_9 0x040002 /*PLL Setting 9*/ +#define CS43130_PLL_SET_10 0x040003 /*PLL Setting 10*/ +#define CS43130_CLKOUT_CTL 0x040004 /*CLKOUT Ctl*/ +#define CS43130_ASP_NUM_1 0x040010 /*ASP Numerator 1*/ +#define CS43130_ASP_NUM_2 0x040011 /*ASP Numerator 2*/ +#define CS43130_ASP_DENOM_1 0x040012 /*ASP Denominator 1*/ +#define CS43130_ASP_DENOM_2 0x040013 /*ASP Denominator 2*/ +#define CS43130_ASP_LRCK_HI_TIME_1 0x040014 /*ASP LRCK High Time 1*/ +#define CS43130_ASP_LRCK_HI_TIME_2 0x040015 /*ASP LRCK High Time 2*/ +#define CS43130_ASP_LRCK_PERIOD_1 0x040016 /*ASP LRCK Period 1*/ +#define CS43130_ASP_LRCK_PERIOD_2 0x040017 /*ASP LRCK Period 2*/ +#define CS43130_ASP_CLOCK_CONF 0x040018 /*ASP Clock Config*/ +#define CS43130_ASP_FRAME_CONF 0x040019 /*ASP Frame Config*/ +#define CS43130_XSP_NUM_1 0x040020 /*XSP Numerator 1*/ +#define CS43130_XSP_NUM_2 0x040021 /*XSP Numerator 2*/ +#define CS43130_XSP_DENOM_1 0x040022 /*XSP Denominator 1*/ +#define CS43130_XSP_DENOM_2 0x040023 /*XSP Denominator 2*/ +#define CS43130_XSP_LRCK_HI_TIME_1 0x040024 /*XSP LRCK High Time 1*/ +#define CS43130_XSP_LRCK_HI_TIME_2 0x040025 /*XSP LRCK High Time 2*/ +#define CS43130_XSP_LRCK_PERIOD_1 0x040026 /*XSP LRCK Period 1*/ +#define CS43130_XSP_LRCK_PERIOD_2 0x040027 /*XSP LRCK Period 2*/ +#define CS43130_XSP_CLOCK_CONF 0x040028 /*XSP Clock Config*/ +#define CS43130_XSP_FRAME_CONF 0x040029 /*XSP Frame Config*/ +#define CS43130_ASP_CH_1_LOC 0x050000 /*ASP Chan 1 Location*/ +#define CS43130_ASP_CH_2_LOC 0x050001 /*ASP Chan 2 Location*/ +#define CS43130_ASP_CH_1_SZ_EN 0x05000A /*ASP Chan 1 Size, Enable*/ +#define CS43130_ASP_CH_2_SZ_EN 0x05000B /*ASP Chan 2 Size, Enable*/ +#define CS43130_XSP_CH_1_LOC 0x060000 /*XSP Chan 1 Location*/ +#define CS43130_XSP_CH_2_LOC 0x060001 /*XSP Chan 2 Location*/ +#define CS43130_XSP_CH_1_SZ_EN 0x06000A /*XSP Chan 1 Size, Enable*/ +#define CS43130_XSP_CH_2_SZ_EN 0x06000B /*XSP Chan 2 Size, Enable*/ +#define CS43130_DSD_VOL_B 0x070000 /*DSD Volume B*/ +#define CS43130_DSD_VOL_A 0x070001 /*DSD Volume A*/ +#define CS43130_DSD_PATH_CTL_1 0x070002 /*DSD Proc Path Sig Ctl 1*/ +#define CS43130_DSD_INT_CFG 0x070003 /*DSD Interface Config*/ +#define CS43130_DSD_PATH_CTL_2 0x070004 /*DSD Proc Path Sig Ctl 2*/ +#define CS43130_DSD_PCM_MIX_CTL 0x070005 /*DSD and PCM Mixing Ctl*/ +#define CS43130_DSD_PATH_CTL_3 0x070006 /*DSD Proc Path Sig Ctl 3*/ +#define CS43130_HP_OUT_CTL_1 0x080000 /*HP Output Ctl 1*/ +#define CS43130_PCM_FILT_OPT 0x090000 /*PCM Filter Option*/ +#define CS43130_PCM_VOL_B 0x090001 /*PCM Volume B*/ +#define CS43130_PCM_VOL_A 0x090002 /*PCM Volume A*/ +#define CS43130_PCM_PATH_CTL_1 0x090003 /*PCM Path Signal Ctl 1*/ +#define CS43130_PCM_PATH_CTL_2 0x090004 /*PCM Path Signal Ctl 2*/ +#define CS43130_CLASS_H_CTL 0x0B0000 /*Class H Ctl*/ +#define CS43130_HP_DETECT 0x0D0000 /*HP Detect*/ +#define CS43130_HP_STATUS 0x0D0001 /*HP Status [RO]*/ +#define CS43130_HP_LOAD_1 0x0E0000 /*HP Load 1*/ +#define CS43130_HP_MEAS_LOAD_1 0x0E0003 /*HP Load Measurement 1*/ +#define CS43130_HP_MEAS_LOAD_2 0x0E0004 /*HP Load Measurement 2*/ +#define CS43130_HP_DC_STAT_1 0x0E000D /*HP DC Load Status 0 [RO]*/ +#define CS43130_HP_DC_STAT_2 0x0E000E /*HP DC Load Status 1 [RO]*/ +#define CS43130_HP_AC_STAT_1 0x0E0010 /*HP AC Load Status 0 [RO]*/ +#define CS43130_HP_AC_STAT_2 0x0E0011 /*HP AC Load Status 1 [RO]*/ +#define CS43130_HP_LOAD_STAT 0x0E001A /*HP Load Status [RO]*/ +#define CS43130_INT_STATUS_1 0x0F0000 /*Interrupt Status 1*/ +#define CS43130_INT_STATUS_2 0x0F0001 /*Interrupt Status 2*/ +#define CS43130_INT_STATUS_3 0x0F0002 /*Interrupt Status 3*/ +#define CS43130_INT_STATUS_4 0x0F0003 /*Interrupt Status 4*/ +#define CS43130_INT_STATUS_5 0x0F0004 /*Interrupt Status 5*/ +#define CS43130_INT_MASK_1 0x0F0010 /*Interrupt Mask 1*/ +#define CS43130_INT_MASK_2 0x0F0011 /*Interrupt Mask 2*/ +#define CS43130_INT_MASK_3 0x0F0012 /*Interrupt Mask 3*/ +#define CS43130_INT_MASK_4 0x0F0013 /*Interrupt Mask 4*/ +#define CS43130_INT_MASK_5 0x0F0014 /*Interrupt Mask 5*/ + +#define CS43130_MCLK_SRC_SEL_MASK 0x03 +#define CS43130_MCLK_SRC_SEL_SHIFT 0 +#define CS43130_MCLK_INT_MASK 0x04 +#define CS43130_MCLK_INT_SHIFT 2 +#define CS43130_SP_SRATE_MASK 0x0F +#define CS43130_SP_SRATE_SHIFT 0 +#define CS43130_SP_BITSIZE_ASP_MASK 0x03 +#define CS43130_SP_BITSIZE_ASP_SHIFT 0 +#define CS43130_HP_DETECT_CTRL_SHIFT 6 +#define CS43130_HP_DETECT_CTRL_MASK (0x03 << CS43130_HP_DETECT_CTRL_SHIFT) +#define CS43130_HP_DETECT_INV_SHIFT 5 +#define CS43130_HP_DETECT_INV_MASK (1 << CS43130_HP_DETECT_INV_SHIFT) + +/* CS43130_INT_MASK_1 */ +#define CS43130_HP_PLUG_INT_SHIFT 6 +#define CS43130_HP_PLUG_INT (1 << CS43130_HP_PLUG_INT_SHIFT) +#define CS43130_HP_UNPLUG_INT_SHIFT 5 +#define CS43130_HP_UNPLUG_INT (1 << CS43130_HP_UNPLUG_INT_SHIFT) +#define CS43130_XTAL_RDY_INT_SHIFT 4 +#define CS43130_XTAL_RDY_INT (1 << CS43130_XTAL_RDY_INT_SHIFT) +#define CS43130_XTAL_ERR_INT_SHIFT 3 +#define CS43130_XTAL_ERR_INT (1 << CS43130_XTAL_ERR_INT_SHIFT) + +/*Reg CS43130_SP_BITSIZE*/ +#define CS43130_SP_BIT_SIZE_8 0x00 +#define CS43130_SP_BIT_SIZE_16 0x01 +#define CS43130_SP_BIT_SIZE_24 0x02 +#define CS43130_SP_BIT_SIZE_32 0x03 + +/*PLL*/ +#define CS43130_PLL_START_MASK (0x1<<0) +#define CS43130_PLL_MODE_MASK 0x02 +#define CS43130_PLL_MODE_SHIFT 1 + +#define CS43130_PLL_REF_PREDIV_MASK 0x3 + +#define CS43130_ASP_STP_MASK 0x10 +#define CS43130_ASP_STP_SHIFT 4 +#define CS43130_ASP_5050_MASK 0x08 +#define CS43130_ASP_5050_SHIFT 3 +#define CS43130_ASP_FSD_MASK 0x07 +#define CS43130_ASP_FSD_SHIFT 0 + +#define CS43130_ASP_MODE_MASK 0x10 +#define CS43130_ASP_MODE_SHIFT 4 +#define CS43130_ASP_SCPOL_OUT_MASK 0x08 +#define CS43130_ASP_SCPOL_OUT_SHIFT 3 +#define CS43130_ASP_SCPOL_IN_MASK 0x04 +#define CS43130_ASP_SCPOL_IN_SHIFT 2 +#define CS43130_ASP_LCPOL_OUT_MASK 0x02 +#define CS43130_ASP_LCPOL_OUT_SHIFT 1 +#define CS43130_ASP_LCPOL_IN_MASK 0x01 +#define CS43130_ASP_LCPOL_IN_SHIFT 0 + +/*Reg CS43130_PWDN_CTL*/ +#define CS43130_PDN_XSP_MASK 0x80 +#define CS43130_PDN_XSP_SHIFT 7 +#define CS43130_PDN_ASP_MASK 0x40 +#define CS43130_PDN_ASP_SHIFT 6 +#define CS43130_PDN_DSPIF_MASK 0x20 +#define CS43130_PDN_DSDIF_SHIFT 5 +#define CS43130_PDN_HP_MASK 0x10 +#define CS43130_PDN_HP_SHIFT 4 +#define CS43130_PDN_XTAL_MASK 0x08 +#define CS43130_PDN_XTAL_SHIFT 3 +#define CS43130_PDN_PLL_MASK 0x04 +#define CS43130_PDN_PLL_SHIFT 2 +#define CS43130_PDN_CLKOUT_MASK 0x02 +#define CS43130_PDN_CLKOUT_SHIFT 1 + +#define CS43130_7_0_MASK 0xFF +#define CS43130_15_8_MASK 0xFF00 +#define CS43130_23_16_MASK 0xFF0000 + +/* Reg CS43130_HP_OUT_CTL_1 */ +#define CS43130_HP_IN_EN_SHIFT 3 +#define CS43130_HP_IN_EN_MASK 0x08 + +#define CS43130_ASP_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define CS43130_XSP_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +enum cs43130_asp_rate { + CS43130_ASP_SPRATE_32K = 0, + CS43130_ASP_SPRATE_44_1K, + CS43130_ASP_SPRATE_48K, + CS43130_ASP_SPRATE_88_2K, + CS43130_ASP_SPRATE_96K, + CS43130_ASP_SPRATE_176_4K, + CS43130_ASP_SPRATE_192K, + CS43130_ASP_SPRATE_352_8K, + CS43130_ASP_SPRATE_384K, +}; + +enum cs43130_mclk_src_sel { + CS43130_MCLK_SRC_XTAL = 0, + CS43130_MCLK_SRC_PLL, + CS43130_MCLK_SRC_RCO +}; + +enum cs43130_mode { + CS43130_SLAVE_MODE = 0, + CS43130_MASTER_MODE +}; + +enum cs43130_xtal_ibias { + CS43130_XTAL_IBIAS_15UA = 2, + CS43130_XTAL_IBIAS_12_5UA = 4, + CS43130_XTAL_IBIAS_7_5UA = 6, +}; + +#define CS43130_AIF_BICK_RATE 1 +#define CS43130_SYSCLK_MCLK 1 +#define CS43130_NUM_SUPPLIES 5 +static const char *const cs43130_supply_names[CS43130_NUM_SUPPLIES] = { + "VA", + "VP", + "VCP", + "VD", + "VL", +}; + +#define CS43130_NUM_INT 5 /* number of interrupt status reg */ + +struct cs43130_private { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct regulator_bulk_data supplies[CS43130_NUM_SUPPLIES]; + /* codec device ID */ + unsigned int dev_id; + int mclk; + int sclk; + int xtal_ibias; + + bool pll_bypass; + int pll_out; + int mclk_int; + int dai_format; + int dai_mode; + int dai_bit; + int asp_size; + int fs; + bool bick_invert; + bool lrck_invert; + int bick; + struct gpio_desc *reset_gpio; +}; + +#endif /* __CS43130_H__ */ -- 1.9.1

3 4

[alsa-devel] [PATCH v3 2/2] ASoC: cs35l35: Add device tree documentation for CS35L35
by Li Xu 13 Dec '16

13 Dec '16

Add device tree documentation for Cirrus Logic CS35L35 speaker amplifier Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l35.txt | 172 +++++++++++++++++++++ 1 file changed, 172 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l35.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt new file mode 100644 index 0000000..80a1ed7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -0,0 +1,172 @@ +CS35L35 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l35" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - interrupt-parent : Specifies the phandle of the interrupt controller to + which the IRQs from CS35L35 are delivered to. + - interrupts : IRQ line info CS35L35. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + +Optional properties: + - cirrus,reset-gpios : Active low GPIO used to reset the amplifier + + - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a + Stereo configuration + + - cirrus,audio-channel : Set Location of Audio Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal + if 2 Devices share Boost BST_CTL + + - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength + Table 3-10 of the datasheet lists drive-strength specifications + 0 = 1x (Default) + 1 = .5x + + - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control + powers down with a rectification FET On or Off. If VSPK is supplied + externally then FET is off. + + - cirrus,boost-ctl-millivolt : Boost Converter control word. Step Size of 100mV + 0x00 = (Default) VP + 0x01 = 2600mV + 0x41 = 9000mV + + - cirrus,boost-ipk-milliamp : Boost-converter peak current limit. + Configures the peak current by monitoring the current through the boost FET. + Step size: 112mA + 0x00 = 1680mA + 0x19 = 4480mA + + - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change + zero-cross + +Optional H/G Algorithm sub-node: + + The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node + that will disable automatic control of the internal H/G Algorithm. + + It is strongly recommended that the Datasheet be referenced when adjusting + or using these Class H Algorithm controls over the internal Algorithm. + Serious damage can occur to the Device and surrounding components. + + - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm + See Section 4.3 Internal Class H Algorithm in the Datasheet. + If not used, the device manages the ClassH Algorithm internally. + +Optional properties for the "cirrus,classh-internal-algo" Sub-node + + Section 7.29 Class H Control + - cirrus,classh-bst-overide : Boolean + - cirrus,classh-bst-max-limit + - cirrus,classh-mem-depth + + Section 7.30 Class H Headroom Control + - cirrus,classh-headroom + + Section 7.31 Class H Release Rate + - cirrus,classh-release-rate + + Section 7.32 Class H Weak FET Drive Control + - cirrus,classh-wk-fet-disable + - cirrus,classh-wk-fet-delay + - cirrus,classh-wk-fet-thld + + Section 7.34 Class H VP Control + - cirrus,classh-vpch-auto + - cirrus,classh-vpch-rate + - cirrus,classh-vpch-man + +Optional Monitor Signal Format sub-node: + + The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node + for adjusting the Depth, Location and Frame of the Monitoring Signals + for Algorithms. + + See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet + + -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formating + on the I2S Port. Each of the 3 8 bit values in the array contain the settings + for depth, location, and frame. + + If not used, the defaults for the 6 monitor signals is used. + + Sections 7.44 - 7.53 lists values for the depth, location, and frame + for each monitoring signal. + + - cirrus,imon : 3 8 bit values to set the depth, location, and frame + of the IMON monitor signal. + + - cirrus,vmon : 3 8 bit values to set the depth, location, and frame + of the VMON monitor signal. + + - cirrus,vpmon : 3 8 bit values to set the depth, location, and frame + of the VPMON monitor signal. + + - cirrus,vbstmon : 3 8 bit values to set the depth, location, and frame + of the VBSTMON monitor signal + + - cirrus,vpbrstat : 3 8 bit values to set the depth, location, and frame + of the VPBRSTAT monitor signal + + - cirrus,zerofill : 3 8 bit values to set the depth, location, and frame\ + of the ZEROFILL packet in the monitor signal + +Example: + +cs35l35: audio-codec@40 { + compatible = "cirrus,cs35l35"; + reg = <0x20>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&axi_gpio 54 1>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + cirrus,boost-ctl = <0x41>; + + cirrus,stereo-config { + cirrus,audio-channel = <0x00>; + cirrus,advisory-channel = <0x01>; + cirrus,shared-boost; + }; + + cirrus,classh-internal-algo { + cirrus,classh-bst-overide; + cirrus,classh-bst-max-limit = <0x01>; + cirrus,classh-mem-depth = <0x01>; + cirrus,classh-release-rate = <0x08>; + cirrus,classh-headroom-millivolt = <0x0B>; + cirrus,classh-wk-fet-disable = <0x01>; + cirrus,classh-wk-fet-delay = <0x04>; + cirrus,classh-wk-fet-thld = <0x01>; + cirrus,classh-vpch-auto = <0x01>; + cirrus,classh-vpch-rate = <0x02>; + cirrus,classh-vpch-man = <0x05>; + }; + + /* Depth, Location, Frame */ + cirrus,monitor-signal-format { + cirrus,imon = /bits/ 8 <0x03 0x00 0x01>; + cirrus,vmon = /bits/ 8 <0x03 0x00 0x00>; + cirrus,vpmon = /bits/ 8 <0x03 0x04 0x00>; + cirrus,vbstmon = /bits/ 8 <0x03 0x04 0x01>; + cirrus,vpbrstat = /bits/ 8 <0x00 0x04 0x00>; + cirrus,zerofill = /bits/ 8 <0x00 0x00 0x00>; + }; + +}; -- 1.9.1

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: cs35l35: Add support for Cirrus CS35L35 Amplifier
by Li Xu 13 Dec '16

13 Dec '16

Add driver support for Cirrus Logic CS35L35 boosted speaker amplifier Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- include/sound/cs35l35.h | 104 ++++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l35.c | 1357 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l35.h | 285 ++++++++++ 5 files changed, 1753 insertions(+) create mode 100644 include/sound/cs35l35.h create mode 100644 sound/soc/codecs/cs35l35.c create mode 100644 sound/soc/codecs/cs35l35.h diff --git a/include/sound/cs35l35.h b/include/sound/cs35l35.h new file mode 100644 index 0000000..005a813 --- /dev/null +++ b/include/sound/cs35l35.h @@ -0,0 +1,104 @@ +/* + * linux/sound/cs35l35.h -- Platform data for CS35l35 + * + * Copyright (c) 2016 Cirrus Logic Inc. + * + * 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 __CS35L35_H +#define __CS35L35_H + +struct classh_cfg { + /* + * Class H Algorithm Control Variables + * You can either have it done + * automatically or you can adjust + * these variables for tuning + * + * if you do not enable the internal algorithm + * you will get a set of mixer controls for + * Class H tuning + * + * Section 4.3 of the datasheet + */ + /* Internal ClassH Algorithm */ + bool classh_bst_override; + bool classh_algo_enable; + int classh_bst_max_limit; + int classh_mem_depth; + int classh_release_rate; + int classh_headroom; + int classh_wk_fet_disable; + int classh_wk_fet_delay; + int classh_wk_fet_thld; + int classh_vpch_auto; + int classh_vpch_rate; + int classh_vpch_man; +}; + +struct monitor_cfg { + /* + * Signal Monitor Data + * highly configurable signal monitoring + * data positioning and different types of + * monitoring data. + * + * Section 4.8.2 - 4.8.4 of the datasheet + */ + bool is_present; + bool imon_specs; + bool vmon_specs; + bool vpmon_specs; + bool vbstmon_specs; + bool vpbrstat_specs; + bool zerofill_specs; + u8 imon_dpth; + u8 imon_loc; + u8 imon_frm; + u8 vmon_dpth; + u8 vmon_loc; + u8 vmon_frm; + u8 vpmon_dpth; + u8 vpmon_loc; + u8 vpmon_frm; + u8 vbstmon_dpth; + u8 vbstmon_loc; + u8 vbstmon_frm; + u8 vpbrstat_dpth; + u8 vpbrstat_loc; + u8 vpbrstat_frm; + u8 zerofill_dpth; + u8 zerofill_loc; + u8 zerofill_frm; +}; + +struct cs35l35_platform_data { + + /* Stereo (2 Device) */ + bool stereo; + /* serial port drive strength */ + int sp_drv_str; + /* Boost Power Down with FET */ + bool bst_pdn_fet_on; + /* Boost Voltage : used if ClassH Algo Enabled */ + int bst_vctl; + /* Boost Converter Peak Current CTRL */ + int bst_ipk; + /* Amp Gain Zero Cross */ + bool gain_zc; + /* Audio Input Location */ + int aud_channel; + /* Advisory Input Location */ + int adv_channel; + /* Shared Boost for stereo */ + bool shared_bst; + /* ClassH Algorithm */ + struct classh_cfg classh_algo; + /* Monitor Config */ + struct monitor_cfg mon_cfg; +}; + +#endif /* __CS35L35_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e1718a..3fd0a08 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C + select SND_SOC_CS35L35 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -407,6 +408,10 @@ config SND_SOC_CS35L34 tristate "Cirrus Logic CS35L34 CODEC" depends on I2C +config SND_SOC_CS35L35 + tristate "Cirrus Logic CS35L35 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7e1dad7..a5622f6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -39,6 +39,7 @@ snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o +snd-soc-cs35l35-objs := cs35l35.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -268,6 +269,7 @@ obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o +obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c new file mode 100644 index 0000000..ea84430 --- /dev/null +++ b/sound/soc/codecs/cs35l35.c @@ -0,0 +1,1357 @@ +/* + * cs35l35.c -- CS35L35 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Brian Austin <brian.austin(a)cirrus.com> + * Li Xu <li.xu(a)cirrus.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/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l35.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l35.h" + +static const struct reg_default cs35l35_reg[] = { + {CS35L35_PWRCTL1, 0x01}, + {CS35L35_PWRCTL2, 0x11}, + {CS35L35_PWRCTL3, 0x00}, + {CS35L35_CLK_CTL1, 0x04}, + {CS35L35_CLK_CTL2, 0x10}, + {CS35L35_CLK_CTL3, 0xCF}, + {CS35L35_SP_FMT_CTL1, 0x20}, + {CS35L35_SP_FMT_CTL2, 0x00}, + {CS35L35_SP_FMT_CTL3, 0x02}, + {CS35L35_MAG_COMP_CTL, 0x00}, + {CS35L35_AMP_INP_DRV_CTL, 0x01}, + {CS35L35_AMP_DIG_VOL_CTL, 0x12}, + {CS35L35_AMP_DIG_VOL, 0x00}, + {CS35L35_ADV_DIG_VOL, 0x00}, + {CS35L35_PROTECT_CTL, 0x06}, + {CS35L35_AMP_GAIN_AUD_CTL, 0x13}, + {CS35L35_AMP_GAIN_PDM_CTL, 0x00}, + {CS35L35_AMP_GAIN_ADV_CTL, 0x00}, + {CS35L35_GPI_CTL, 0x00}, + {CS35L35_BST_CVTR_V_CTL, 0x00}, + {CS35L35_BST_PEAK_I, 0x07}, + {CS35L35_BST_RAMP_CTL, 0x85}, + {CS35L35_BST_CONV_COEF_1, 0x20}, + {CS35L35_BST_CONV_COEF_2, 0x20}, + {CS35L35_BST_CONV_SLOPE_COMP, 0x47}, + {CS35L35_BST_CONV_SW_FREQ, 0x04}, + {CS35L35_CLASS_H_CTL, 0x0B}, + {CS35L35_CLASS_H_HEADRM_CTL, 0x0B}, + {CS35L35_CLASS_H_RELEASE_RATE, 0x08}, + {CS35L35_CLASS_H_FET_DRIVE_CTL, 0x41}, + {CS35L35_CLASS_H_VP_CTL, 0xC5}, + {CS35L35_VPBR_CTL, 0x0A}, + {CS35L35_VPBR_VOL_CTL, 0x09}, + {CS35L35_VPBR_TIMING_CTL, 0x6A}, + {CS35L35_VPBR_MODE_VOL_CTL, 0x00}, + {CS35L35_SPKR_MON_CTL, 0xC0}, + {CS35L35_IMON_SCALE_CTL, 0x30}, + {CS35L35_AUDIN_RXLOC_CTL, 0x00}, + {CS35L35_ADVIN_RXLOC_CTL, 0x80}, + {CS35L35_VMON_TXLOC_CTL, 0x00}, + {CS35L35_IMON_TXLOC_CTL, 0x80}, + {CS35L35_VPMON_TXLOC_CTL, 0x04}, + {CS35L35_VBSTMON_TXLOC_CTL, 0x84}, + {CS35L35_VPBR_STATUS_TXLOC_CTL, 0x04}, + {CS35L35_ZERO_FILL_LOC_CTL, 0x00}, + {CS35L35_AUDIN_DEPTH_CTL, 0x0F}, + {CS35L35_SPKMON_DEPTH_CTL, 0x0F}, + {CS35L35_SUPMON_DEPTH_CTL, 0x0F}, + {CS35L35_ZEROFILL_DEPTH_CTL, 0x00}, + {CS35L35_MULT_DEV_SYNCH1, 0x02}, + {CS35L35_MULT_DEV_SYNCH2, 0x80}, + {CS35L35_PROT_RELEASE_CTL, 0x00}, + {CS35L35_DIAG_MODE_REG_LOCK, 0x00}, + {CS35L35_DIAG_MODE_CTL_1, 0x40}, + {CS35L35_DIAG_MODE_CTL_2, 0x00}, + {CS35L35_INT_MASK_1, 0xFF}, + {CS35L35_INT_MASK_2, 0xFF}, + {CS35L35_INT_MASK_3, 0xFF}, + {CS35L35_INT_MASK_4, 0xFF}, + +}; + +static bool cs35l35_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_DEVID_AB ... CS35L35_REV_ID: + case CS35L35_INT_STATUS_1: + case CS35L35_INT_STATUS_2: + case CS35L35_INT_STATUS_3: + case CS35L35_INT_STATUS_4: + case CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l35_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_DEVID_AB ... CS35L35_PWRCTL3: + case CS35L35_CLK_CTL1 ... CS35L35_SP_FMT_CTL3: + case CS35L35_MAG_COMP_CTL ... CS35L35_AMP_GAIN_AUD_CTL: + case CS35L35_AMP_GAIN_PDM_CTL ... CS35L35_BST_PEAK_I: + case CS35L35_BST_RAMP_CTL ... CS35L35_BST_CONV_SW_FREQ: + case CS35L35_CLASS_H_CTL ... CS35L35_CLASS_H_VP_CTL: + case CS35L35_CLASS_H_STATUS: + case CS35L35_VPBR_CTL ... CS35L35_VPBR_MODE_VOL_CTL: + case CS35L35_VPBR_ATTEN_STATUS: + case CS35L35_SPKR_MON_CTL: + case CS35L35_IMON_SCALE_CTL ... CS35L35_ZEROFILL_DEPTH_CTL: + case CS35L35_MULT_DEV_SYNCH1 ... CS35L35_PROT_RELEASE_CTL: + case CS35L35_DIAG_MODE_REG_LOCK ... CS35L35_DIAG_MODE_CTL_2: + case CS35L35_INT_MASK_1 ... CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l35_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_INT_STATUS_1: + case CS35L35_INT_STATUS_2: + case CS35L35_INT_STATUS_3: + case CS35L35_INT_STATUS_4: + case CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static int cs35l35_sdin_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); + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MCLK_DIS_MASK, 0 << CS35L35_MCLK_DIS_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_DISCHG_FILT_MASK, 0 << CS35L35_DISCHG_FILT_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL_MASK, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL_MASK, 1); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_DISCHG_FILT_MASK, 1 << CS35L35_DISCHG_FILT_SHIFT); + + ret = wait_for_completion_timeout(&cs35l35->pdn_done, + msecs_to_jiffies(100)); + if (ret == 0) { + pr_err("TIMEOUT PDN_DONE did not complete in 100ms\n"); + ret = -ETIMEDOUT; + } + + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MCLK_DIS_MASK, 1 << CS35L35_MCLK_DIS_SHIFT); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + ret = -EINVAL; + } + return ret; +} + +static int cs35l35_main_amp_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); + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + unsigned int reg[4]; + int i; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 0 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 0 << CS35L35_PDN_BST_FETOFF_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 0 << CS35L35_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMU: + usleep_range(5000, 5100); + /* If PDM mode we must use VP + * for Voltage control + */ + if (cs35l35->pdm_mode) + regmap_update_bits(cs35l35->regmap, + CS35L35_BST_CVTR_V_CTL, CS35L35_BST_CTL_MASK, + 0 << CS35L35_BST_CTL_SHIFT); + for (i = 0; i < 2; i++) + regmap_bulk_read(cs35l35->regmap, CS35L35_INT_STATUS_1, + &reg, ARRAY_SIZE(reg)); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 1 << CS35L35_AMP_MUTE_SHIFT); + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 1 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 1 << CS35L35_PDN_BST_FETOFF_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + usleep_range(5000, 5100); + /* If PDM mode we should switch back to pdata value + * for Voltage control when we go down + */ + if (cs35l35->pdm_mode) + regmap_update_bits(cs35l35->regmap, + CS35L35_BST_CVTR_V_CTL, CS35L35_BST_CTL_MASK, + cs35l35->pdata.bst_vctl << CS35L35_BST_CTL_SHIFT); + + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l35_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital Audio Volume", CS35L35_AMP_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("AMP Audio Gain", CS35L35_AMP_GAIN_AUD_CTL, 0, 19, 0, + amp_gain_tlv), + SOC_SINGLE_TLV("AMP PDM Gain", CS35L35_AMP_GAIN_PDM_CTL, 0, 19, 0, + amp_gain_tlv), +}; + +static const struct snd_kcontrol_new cs35l35_adv_controls[] = { + SOC_SINGLE_SX_TLV("Digital Advisory Volume", CS35L35_ADV_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("AMP Advisory Gain", CS35L35_AMP_GAIN_ADV_CTL, 0, 19, 0, + amp_gain_tlv), +}; + +static const struct snd_soc_dapm_widget cs35l35_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L35_PWRCTL3, 1, 1, + cs35l35_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L35_PWRCTL3, 2, 1), + + SND_SOC_DAPM_OUTPUT("SPK"), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L35_PWRCTL2, 7, 1), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L35_PWRCTL2, 6, 1), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L35_PWRCTL3, 3, 1), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L35_PWRCTL3, 4, 1), + SND_SOC_DAPM_ADC("CLASS H", NULL, CS35L35_PWRCTL2, 5, 1), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L35_PWRCTL2, 0, 1, NULL, 0, + cs35l35_main_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l35_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "ISENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + {"SDOUT", NULL, "IMON ADC"}, + {"SDOUT", NULL, "VMON ADC"}, + {"SDOUT", NULL, "VBSTMON ADC"}, + {"SDOUT", NULL, "VPMON ADC"}, + {"AMP Capture", NULL, "SDOUT"}, + + {"SDIN", NULL, "AMP Playback"}, + {"CLASS H", NULL, "SDIN"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l35_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MS_MASK, 1 << CS35L35_MS_SHIFT); + cs35l35->slave_mode = false; + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MS_MASK, 0 << CS35L35_MS_SHIFT); + cs35l35->slave_mode = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs35l35->i2s_mode = true; + cs35l35->pdm_mode = false; + break; + case SND_SOC_DAIFMT_PDM: + cs35l35->pdm_mode = true; + cs35l35->i2s_mode = false; + break; + default: + return -EINVAL; + } + + return 0; +} + +struct cs35l35_sysclk_config { + int sysclk; + int srate; + u8 clk_cfg; +}; + +static struct cs35l35_sysclk_config cs35l35_clk_ctl[] = { + + /* SYSCLK, Sample Rate, Serial Port Cfg */ + {5644800, 44100, 0x00}, + {5644800, 88200, 0x40}, + {6144000, 48000, 0x10}, + {6144000, 96000, 0x50}, + {11289600, 44100, 0x01}, + {11289600, 88200, 0x41}, + {11289600, 176400, 0x81}, + {12000000, 44100, 0x03}, + {12000000, 48000, 0x13}, + {12000000, 88200, 0x43}, + {12000000, 96000, 0x53}, + {12000000, 176400, 0x83}, + {12000000, 192000, 0x93}, + {12288000, 48000, 0x11}, + {12288000, 96000, 0x51}, + {12288000, 192000, 0x91}, + {13000000, 44100, 0x07}, + {13000000, 48000, 0x17}, + {13000000, 88200, 0x47}, + {13000000, 96000, 0x57}, + {13000000, 176400, 0x87}, + {13000000, 192000, 0x97}, + {22579200, 44100, 0x02}, + {22579200, 88200, 0x42}, + {22579200, 176400, 0x82}, + {24000000, 44100, 0x0B}, + {24000000, 48000, 0x1B}, + {24000000, 88200, 0x4B}, + {24000000, 96000, 0x5B}, + {24000000, 176400, 0x8B}, + {24000000, 192000, 0x9B}, + {24576000, 48000, 0x12}, + {24576000, 96000, 0x52}, + {24576000, 192000, 0x92}, + {26000000, 44100, 0x0F}, + {26000000, 48000, 0x1F}, + {26000000, 88200, 0x4F}, + {26000000, 96000, 0x5F}, + {26000000, 176400, 0x8F}, + {26000000, 192000, 0x9F}, +}; + +static int cs35l35_get_clk_config(int sysclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l35_clk_ctl); i++) { + if (cs35l35_clk_ctl[i].sysclk == sysclk && + cs35l35_clk_ctl[i].srate == srate) + return cs35l35_clk_ctl[i].clk_cfg; + } + return -EINVAL; +} + +static int cs35l35_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + struct classh_cfg *classh = &cs35l35->pdata.classh_algo; + int srate = params_rate(params); + int ret = 0; + u8 sp_sclks; + int audin_format; + int errata_chk; + + int clk_ctl = cs35l35_get_clk_config(cs35l35->sysclk, srate); + + if (clk_ctl < 0) { + dev_err(codec->dev, "Invalid CLK:Rate %d:%d\n", + cs35l35->sysclk, srate); + return -EINVAL; + } + + ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL2, + CS35L35_CLK_CTL2_MASK, clk_ctl); + if (ret != 0) { + dev_err(codec->dev, "Failed to set port config %d\n", ret); + return ret; + } + + /* Rev A0 Errata + * + * When configured for the weak-drive detection path (CH_WKFET_DIS = 0) + * the Class H algorithm does not enable weak-drive operation for + * nonzero values of CH_WKFET_DELAY if SP_RATE = 01 or 10 + * + */ + errata_chk = clk_ctl & CS35L35_SP_RATE_MASK; + + if (classh->classh_wk_fet_disable == 0x00 && + (errata_chk == 0x01 || errata_chk == 0x03)) { + ret = regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, CS35L35_CH_WKFET_DEL_MASK, + 0 << CS35L35_CH_WKFET_DEL_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fet config %d\n", + ret); + return ret; + } + } + +/* + * You can pull more Monitor data from the SDOUT pin than going to SDIN + * Just make sure your SCLK is fast enough to fill the frame + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (params_width(params)) { + case 8: + audin_format = CS35L35_SDIN_DEPTH_8; + break; + case 16: + audin_format = CS35L35_SDIN_DEPTH_16; + break; + case 24: + audin_format = CS35L35_SDIN_DEPTH_24; + break; + default: + dev_err(codec->dev, "Unsupported Width %d\n", + params_width(params)); + } + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_DEPTH_CTL, CS35L35_AUDIN_DEPTH_MASK, + audin_format << CS35L35_AUDIN_DEPTH_SHIFT); + if (cs35l35->pdata.stereo) { + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_DEPTH_CTL, CS35L35_ADVIN_DEPTH_MASK, + audin_format << CS35L35_ADVIN_DEPTH_SHIFT); + } + } +/* We have to take the SCLK to derive num sclks + * to configure the CLOCK_CTL3 register correctly + */ + if ((cs35l35->sclk / srate) % 4) { + dev_err(codec->dev, "Unsupported sclk/fs ratio %d:%d\n", + cs35l35->sclk, srate); + return -EINVAL; + } + sp_sclks = ((cs35l35->sclk / srate) / 4) - 1; + + if (cs35l35->i2s_mode) { + /* Only certain ratios are supported in I2S Slave Mode */ + if (cs35l35->slave_mode) { + switch (sp_sclks) { + case CS35L35_SP_SCLKS_32FS: + case CS35L35_SP_SCLKS_48FS: + case CS35L35_SP_SCLKS_64FS: + break; + default: + dev_err(codec->dev, "ratio not supported\n"); + return -EINVAL; + }; + } else { + /* Only certain ratios supported in I2S MASTER Mode */ + switch (sp_sclks) { + case CS35L35_SP_SCLKS_32FS: + case CS35L35_SP_SCLKS_64FS: + break; + default: + dev_err(codec->dev, "ratio not supported\n"); + return -EINVAL; + }; + } + ret = regmap_update_bits(cs35l35->regmap, + CS35L35_CLK_CTL3, CS35L35_SP_SCLKS_MASK, + sp_sclks << CS35L35_SP_SCLKS_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fsclk %d\n", ret); + return ret; + } + } + if (cs35l35->pdm_mode) { + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, + CS35L35_PDM_MODE_MASK, 1 << CS35L35_PDM_MODE_SHIFT); + } else { + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, + CS35L35_PDM_MODE_MASK, 0 << CS35L35_PDM_MODE_SHIFT); + } + return ret; +} + +static const unsigned int cs35l35_src_rates[] = { + 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l35_constraints = { + .count = ARRAY_SIZE(cs35l35_src_rates), + .list = cs35l35_src_rates, +}; + +static int cs35l35_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l35_constraints); + return 0; +} + +static const unsigned int cs35l35_pdm_rates[] = { + 44100, 48000, 88200, 96000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l35_pdm_constraints = { + .count = ARRAY_SIZE(cs35l35_pdm_rates), + .list = cs35l35_pdm_rates, +}; + +static int cs35l35_pdm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l35_pdm_constraints); + return 0; +} + +static int cs35l35_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + /* Need the SCLK Frequency */ + cs35l35->sclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops cs35l35_ops = { + .startup = cs35l35_pcm_startup, + .set_fmt = cs35l35_set_dai_fmt, + .hw_params = cs35l35_pcm_hw_params, + .set_sysclk = cs35l35_dai_set_sysclk, +}; + +static const struct snd_soc_dai_ops cs35l35_pdm_ops = { + .startup = cs35l35_pdm_startup, + .set_fmt = cs35l35_set_dai_fmt, + .hw_params = cs35l35_pcm_hw_params, + .set_sysclk = cs35l35_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l35_dai[] = { + { + .name = "cs35l35-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .ops = &cs35l35_ops, + .symmetric_rates = 1, + }, + { + .name = "cs35l35-pdm", + .id = 1, + .playback = { + .stream_name = "PDM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .ops = &cs35l35_pdm_ops, + }, +}; + +static int cs35l35_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + int clksrc; + int ret = 0; + + switch (clk_id) { + case 0: + clksrc = CS35L35_CLK_SOURCE_MCLK; + break; + case 1: + clksrc = CS35L35_CLK_SOURCE_SCLK; + break; + case 2: + clksrc = CS35L35_CLK_SOURCE_PDM; + break; + default: + dev_err(codec->dev, "Invalid CLK Source\n"); + return -EINVAL; + }; + + switch (freq) { + case 5644800: + case 6144000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs35l35->sysclk = freq; + break; + default: + dev_err(codec->dev, "Invalid CLK Frequency\n"); + return -EINVAL; + } + + ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_CLK_SOURCE_MASK, clksrc << CS35L35_CLK_SOURCE_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set sysclk %d\n", ret); + return ret; + } + + return ret; +} + +static int cs35l35_codec_probe(struct snd_soc_codec *codec) +{ + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + struct classh_cfg *classh = &cs35l35->pdata.classh_algo; + struct monitor_cfg *monitor_config = &cs35l35->pdata.mon_cfg; + int ret; + + /* Set Platform Data */ + if (cs35l35->pdata.bst_vctl) + regmap_update_bits(cs35l35->regmap, CS35L35_BST_CVTR_V_CTL, + CS35L35_BST_CTL_MASK, cs35l35->pdata.bst_vctl); + + if (cs35l35->pdata.bst_ipk) + regmap_update_bits(cs35l35->regmap, CS35L35_BST_PEAK_I, + CS35L35_BST_IPK_MASK, + cs35l35->pdata.bst_ipk << CS35L35_BST_IPK_SHIFT); + + if (cs35l35->pdata.gain_zc) + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_GAIN_ZC_MASK, + cs35l35->pdata.gain_zc << CS35L35_AMP_GAIN_ZC_SHIFT); + + if (cs35l35->pdata.aud_channel) + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_RXLOC_CTL, + CS35L35_AUD_IN_LR_MASK, + cs35l35->pdata.aud_channel << CS35L35_AUD_IN_LR_SHIFT); + + if (cs35l35->pdata.stereo) { + regmap_update_bits(cs35l35->regmap, + CS35L35_ADVIN_RXLOC_CTL, CS35L35_ADV_IN_LR_MASK, + cs35l35->pdata.adv_channel << CS35L35_ADV_IN_LR_SHIFT); + if (cs35l35->pdata.shared_bst) + regmap_update_bits(cs35l35->regmap, CS35L35_CLASS_H_CTL, + CS35L35_CH_STEREO_MASK, 1 << CS35L35_CH_STEREO_SHIFT); + ret = snd_soc_add_codec_controls(codec, cs35l35_adv_controls, + ARRAY_SIZE(cs35l35_adv_controls)); + if (ret) + return ret; + } + + if (cs35l35->pdata.sp_drv_str) + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_SP_DRV_MASK, + cs35l35->pdata.sp_drv_str << CS35L35_SP_DRV_SHIFT); + + if (classh->classh_algo_enable) { + if (classh->classh_bst_override) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, CS35L35_CH_BST_OVR_MASK, + classh->classh_bst_override << CS35L35_CH_BST_OVR_SHIFT); + if (classh->classh_bst_max_limit) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, CS35L35_CH_BST_LIM_MASK, + classh->classh_bst_max_limit << CS35L35_CH_BST_LIM_SHIFT); + if (classh->classh_mem_depth) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, CS35L35_CH_MEM_DEPTH_MASK, + classh->classh_mem_depth << CS35L35_CH_MEM_DEPTH_SHIFT); + if (classh->classh_headroom) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_HEADRM_CTL, CS35L35_CH_HDRM_CTL_MASK, + classh->classh_headroom << CS35L35_CH_HDRM_CTL_SHIFT); + if (classh->classh_release_rate) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_RELEASE_RATE, CS35L35_CH_REL_RATE_MASK, + classh->classh_release_rate << CS35L35_CH_REL_RATE_SHIFT); + if (classh->classh_wk_fet_disable) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, CS35L35_CH_WKFET_DIS_MASK, + classh->classh_wk_fet_disable << CS35L35_CH_WKFET_DIS_SHIFT); + if (classh->classh_wk_fet_delay) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, CS35L35_CH_WKFET_DEL_MASK, + classh->classh_wk_fet_delay << CS35L35_CH_WKFET_DEL_SHIFT); + if (classh->classh_wk_fet_thld) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, CS35L35_CH_WKFET_THLD_MASK, + classh->classh_wk_fet_thld << CS35L35_CH_WKFET_THLD_SHIFT); + if (classh->classh_vpch_auto) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, CS35L35_CH_VP_AUTO_MASK, + classh->classh_vpch_auto << CS35L35_CH_VP_AUTO_SHIFT); + if (classh->classh_vpch_rate) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, CS35L35_CH_VP_RATE_MASK, + classh->classh_vpch_rate << CS35L35_CH_VP_RATE_SHIFT); + if (classh->classh_vpch_man) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, CS35L35_CH_VP_MAN_MASK, + classh->classh_vpch_man << CS35L35_CH_VP_MAN_SHIFT); + } + + if (monitor_config->is_present) { + if (monitor_config->vmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SPKMON_DEPTH_CTL, CS35L35_VMON_DEPTH_MASK, + monitor_config->vmon_dpth << CS35L35_VMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VMON_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->vmon_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VMON_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->vmon_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->imon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SPKMON_DEPTH_CTL, CS35L35_IMON_DEPTH_MASK, + monitor_config->imon_dpth << CS35L35_IMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->imon_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->imon_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vpmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, CS35L35_VPMON_DEPTH_MASK, + monitor_config->vpmon_dpth << CS35L35_VPMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPMON_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->vpmon_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPMON_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->vpmon_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vbstmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, CS35L35_VBSTMON_DEPTH_MASK, + monitor_config->vpmon_dpth << CS35L35_VBSTMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VBSTMON_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->vbstmon_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VBSTMON_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->vbstmon_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vpbrstat_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, CS35L35_VPBRSTAT_DEPTH_MASK, + monitor_config->vpbrstat_dpth << CS35L35_VPBRSTAT_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPBR_STATUS_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->vpbrstat_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPBR_STATUS_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->vpbrstat_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->zerofill_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, CS35L35_ZEROFILL_DEPTH_MASK, + monitor_config->zerofill_dpth << CS35L35_ZEROFILL_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_ZERO_FILL_LOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->zerofill_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_ZERO_FILL_LOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->zerofill_frm << CS35L35_MON_FRM_SHIFT); + } + } + + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l35 = { + .probe = cs35l35_codec_probe, + .set_sysclk = cs35l35_codec_set_sysclk, + .component_driver = { + .controls = cs35l35_aud_controls, + .num_controls = ARRAY_SIZE(cs35l35_aud_controls), + .dapm_widgets = cs35l35_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l35_dapm_widgets), + + .dapm_routes = cs35l35_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l35_audio_map), + }, +}; + +static struct regmap_config cs35l35_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L35_MAX_REGISTER, + .reg_defaults = cs35l35_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l35_reg), + .volatile_reg = cs35l35_volatile_register, + .readable_reg = cs35l35_readable_register, + .precious_reg = cs35l35_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l35_irq(int irq, void *data) +{ + struct cs35l35_private *cs35l35 = data; + unsigned int sticky1, sticky2, sticky3, sticky4; + unsigned int mask1, mask2, mask3, mask4, current1; + + /* ack the irq by reading all status registers */ + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_4, &sticky4); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_3, &sticky3); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_2, &sticky2); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_1, &sticky1); + + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_4, &mask4); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_3, &mask3); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_2, &mask2); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_1, &mask1); + + /* Check to see if unmasked bits are active */ + if (!(sticky1 & ~mask1) && !(sticky2 & ~mask2) && !(sticky3 & ~mask3) + && !(sticky4 & ~mask4)) + return IRQ_NONE; + + if (sticky2 & CS35L35_PDN_DONE) + complete(&cs35l35->pdn_done); + + /* read the current values */ + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_1, &current1); + + /* handle the interrupts */ + if (sticky1 & CS35L35_CAL_ERR) { + pr_err("%s : Calibration Error\n", __func__); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_CAL_ERR)) { + pr_debug("%s : Cal error release\n", __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_CAL_ERR_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_CAL_ERR_RLS, + CS35L35_CAL_ERR_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_CAL_ERR_RLS, 0); + } + } + + if (sticky1 & CS35L35_AMP_SHORT) { + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_AMP_SHORT)) { + pr_debug("%s :Amp short error release\n", __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_SHORT_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_SHORT_RLS, + CS35L35_SHORT_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_SHORT_RLS, 0); + } + } + + if (sticky1 & CS35L35_OTW) { + pr_err("%s : Over temperature warning\n", __func__); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_OTW)) { + pr_debug("%s : Over temperature warning release\n", + __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTW_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTW_RLS, + CS35L35_OTW_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTW_RLS, 0); + } + } + + if (sticky1 & CS35L35_OTE) { + pr_crit("%s : Over temperature error\n", __func__); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_OTE)) { + pr_debug("%s : Over temperature error release\n", + __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTE_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTE_RLS, + CS35L35_OTE_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTE_RLS, 0); + } + } + + if (sticky3 & CS35L35_BST_HIGH) { + pr_crit("%s : VBST error: powering off!\n", __func__); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_AMP, CS35L35_PDN_AMP); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL, CS35L35_PDN_ALL); + } + + if (sticky3 & CS35L35_LBST_SHORT) { + pr_crit("%s : LBST error: powering off!\n", __func__); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_AMP, CS35L35_PDN_AMP); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL, CS35L35_PDN_ALL); + } + + if (sticky2 & CS35L35_VPBR_ERR) + pr_err("%s : Error: Reactive Brownout\n", __func__); + + if (sticky4 & CS35L35_VMON_OVFL) + pr_err("%s : Error: VMON overflow\n", __func__); + + if (sticky4 & CS35L35_IMON_OVFL) + pr_err("%s : Error: IMON overflow\n", __func__); + + return IRQ_HANDLED; +} + + +static int cs35l35_handle_of_data(struct i2c_client *i2c_client, + struct cs35l35_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct device_node *classh, *signal_format; + struct classh_cfg *classh_config = &pdata->classh_algo; + struct monitor_cfg *monitor_config = &pdata->mon_cfg; + unsigned int val32 = 0; + u8 monitor_array[3]; + int ret = 0; + + if (!np) + return 0; + + pdata->bst_pdn_fet_on = of_property_read_bool(np, + "cirrus,boost-pdn-fet-on"); + + if (of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val32) >= 0) + pdata->bst_vctl = val32; + + if (of_property_read_u32(np, "cirrus,boost-ipk-milliamp", &val32) >= 0) + pdata->bst_ipk = val32; + + if (of_property_read_u32(np, "cirrus,sp-drv-strength", &val32) >= 0) + pdata->sp_drv_str = val32; + + pdata->stereo = of_property_read_bool(np, "cirrus,stereo-config"); + + if (pdata->stereo) { + if (of_property_read_u32(np, "cirrus,audio-channel", &val32) >= 0) + pdata->aud_channel = val32; + if (of_property_read_u32(np, "cirrus,advisory-channel", + &val32) >= 0) + pdata->adv_channel = val32; + pdata->shared_bst = of_property_read_bool(np, + "cirrus,shared-boost"); + } + + pdata->gain_zc = of_property_read_bool(np, "cirrus,amp-gain-zc"); + + classh = of_get_child_by_name(np, "cirrus,classh-internal-algo"); + classh_config->classh_algo_enable = classh ? true : false; + + if (classh_config->classh_algo_enable) { + classh_config->classh_bst_override = + of_property_read_bool(np, "cirrus,classh-bst-overide"); + + if (of_property_read_u32(classh, "cirrus,classh-bst-max-limit", + &val32) >= 0) + classh_config->classh_bst_max_limit = val32; + if (of_property_read_u32(classh, "cirrus,classh-mem-depth", + &val32) >= 0) + classh_config->classh_mem_depth = val32; + if (of_property_read_u32(classh, "cirrus,classh-release-rate", + &val32) >= 0) + classh_config->classh_release_rate = val32; + if (of_property_read_u32(classh, "cirrus,classh-headroom", + &val32) >= 0) + classh_config->classh_headroom = val32; + if (of_property_read_u32(classh, "cirrus,classh-wk-fet-disable", + &val32) >= 0) + classh_config->classh_wk_fet_disable = val32; + if (of_property_read_u32(classh, "cirrus,classh-wk-fet-delay", + &val32) >= 0) + classh_config->classh_wk_fet_delay = val32; + if (of_property_read_u32(classh, "cirrus,classh-wk-fet-thld", + &val32) >= 0) + classh_config->classh_wk_fet_thld = val32; + if (of_property_read_u32(classh, "cirrus,classh-vpch-auto", + &val32) >= 0) + classh_config->classh_vpch_auto = val32; + if (of_property_read_u32(classh, "cirrus,classh-vpch-rate", + &val32) >= 0) + classh_config->classh_vpch_rate = val32; + if (of_property_read_u32(classh, "cirrus,classh-vpch-man", + &val32) >= 0) + classh_config->classh_vpch_man = val32; + } + of_node_put(classh); + + /* frame depth location */ + signal_format = of_get_child_by_name(np, "cirrus,monitor-signal-format"); + monitor_config->is_present = signal_format ? true : false; + if (monitor_config->is_present) { + ret = of_property_read_u8_array(signal_format, "cirrus,imon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->imon_specs = true; + monitor_config->imon_dpth = monitor_array[0]; + monitor_config->imon_loc = monitor_array[1]; + monitor_config->imon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vmon_specs = true; + monitor_config->vmon_dpth = monitor_array[0]; + monitor_config->vmon_loc = monitor_array[1]; + monitor_config->vmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vpmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vpmon_specs = true; + monitor_config->vpmon_dpth = monitor_array[0]; + monitor_config->vpmon_loc = monitor_array[1]; + monitor_config->vpmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vbstmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vbstmon_specs = true; + monitor_config->vbstmon_dpth = monitor_array[0]; + monitor_config->vbstmon_loc = monitor_array[1]; + monitor_config->vbstmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vpbrstat", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vpbrstat_specs = true; + monitor_config->vpbrstat_dpth = monitor_array[0]; + monitor_config->vpbrstat_loc = monitor_array[1]; + monitor_config->vpbrstat_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,zerofill", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->zerofill_specs = true; + monitor_config->zerofill_dpth = monitor_array[0]; + monitor_config->zerofill_loc = monitor_array[1]; + monitor_config->zerofill_frm = monitor_array[2]; + } + } + of_node_put(signal_format); + + return 0; +} + +/* Errata Rev A0 */ +static const struct reg_sequence cs35l35_errata_patch[] = { + + { 0x7F, 0x99 }, + { 0x00, 0x99 }, + { 0x52, 0x22 }, + { 0x04, 0x14 }, + { 0x6D, 0x44 }, + { 0x24, 0x10 }, + { 0x58, 0xC4 }, + { 0x00, 0x98 }, + { 0x18, 0x08 }, + { 0x00, 0x00 }, + { 0x7F, 0x00 }, +}; + +static int cs35l35_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l35_private *cs35l35; + struct cs35l35_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int i; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs35l35 = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l35_private), + GFP_KERNEL); + if (!cs35l35) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs35l35); + cs35l35->regmap = devm_regmap_init_i2c(i2c_client, &cs35l35_regmap); + if (IS_ERR(cs35l35->regmap)) { + ret = PTR_ERR(cs35l35->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(cs35l35_supplies); i++) + cs35l35->supplies[i].supply = cs35l35_supplies[i]; + cs35l35->num_supplies = ARRAY_SIZE(cs35l35_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l35->num_supplies, + cs35l35->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l35->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l35_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, + "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + ret = cs35l35_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + cs35l35->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l35->num_supplies, + cs35l35->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l35->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l35->reset_gpio)) + return PTR_ERR(cs35l35->reset_gpio); + + if (cs35l35->reset_gpio) + gpiod_set_value_cansleep(cs35l35->reset_gpio, 1); + + init_completion(&cs35l35->pdn_done); + + ret = regmap_register_patch(cs35l35->regmap, cs35l35_errata_patch, + ARRAY_SIZE(cs35l35_errata_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to apply errata patch\n"); + return ret; + } + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l35_irq, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l35", cs35l35); + if (ret != 0) { + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + /* initialize codec */ + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L35_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L35 Device ID (%X). Expected ID %X\n", + devid, CS35L35_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l35->regmap, CS35L35_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L35 (%x), Revision: %02X\n", devid, + ret & 0xFF); + + /* Set the INT Masks for critical errors */ + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_1, CS35L35_INT1_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_2, CS35L35_INT2_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_3, CS35L35_INT3_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_4, CS35L35_INT4_CRIT_MASK); + + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PWR2_PDN_MASK, CS35L35_PWR2_PDN_MASK); + + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 1 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 1 << CS35L35_PDN_BST_FETOFF_SHIFT); + + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL3, + CS35L35_PWR3_PDN_MASK, CS35L35_PWR3_PDN_MASK); + + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 1 << CS35L35_AMP_MUTE_SHIFT); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l35, cs35l35_dai, + ARRAY_SIZE(cs35l35_dai)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "%s: Register codec failed\n", __func__); + goto err; + } + +err: + regulator_bulk_disable(cs35l35->num_supplies, + cs35l35->supplies); + return ret; +} + +static int cs35l35_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct of_device_id cs35l35_of_match[] = { + {.compatible = "cirrus,cs35l35"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l35_of_match); + +static const struct i2c_device_id cs35l35_id[] = { + {"cs35l35", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l35_id); + +static struct i2c_driver cs35l35_i2c_driver = { + .driver = { + .name = "cs35l35", + .of_match_table = cs35l35_of_match, + }, + .id_table = cs35l35_id, + .probe = cs35l35_i2c_probe, + .remove = cs35l35_i2c_remove, +}; + +module_i2c_driver(cs35l35_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L35 driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_AUTHOR("Li Xu, Cirrus Logic Inc, <li.xu(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h new file mode 100644 index 0000000..767227e --- /dev/null +++ b/sound/soc/codecs/cs35l35.h @@ -0,0 +1,285 @@ +/* + * cs35l35.h -- CS35L35 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Brian Austin <brian.austin(a)cirrus.com> + * Li Xu <li.xu(a)cirrus.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 __CS35L35_H__ +#define __CS35L35_H__ + +#define CS35L35_FIRSTREG 0x01 +#define CS35L35_LASTREG 0x7E +#define CS35L35_CHIP_ID 0x00035A35 +#define CS35L35_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L35_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L35_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L35_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L35_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L35_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L35_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L35_PWRCTL3 0x08 /* Power Ctl 3 */ +#define CS35L35_CLK_CTL1 0x0A /* Clocking Ctl 1 */ +#define CS35L35_CLK_CTL2 0x0B /* Clocking Ctl 2 */ +#define CS35L35_CLK_CTL3 0x0C /* Clocking Ctl 3 */ +#define CS35L35_SP_FMT_CTL1 0x0D /* Serial Port Format CTL1 */ +#define CS35L35_SP_FMT_CTL2 0x0E /* Serial Port Format CTL2 */ +#define CS35L35_SP_FMT_CTL3 0x0F /* Serial Port Format CTL3 */ +#define CS35L35_MAG_COMP_CTL 0x13 /* Magnitude Comp CTL */ +#define CS35L35_AMP_INP_DRV_CTL 0x14 /* Amp Input Drive Ctl */ +#define CS35L35_AMP_DIG_VOL_CTL 0x15 /* Amplifier Dig Volume Ctl */ +#define CS35L35_AMP_DIG_VOL 0x16 /* Amplifier Dig Volume */ +#define CS35L35_ADV_DIG_VOL 0x17 /* Advisory Digital Volume */ +#define CS35L35_PROTECT_CTL 0x18 /* Amp Gain - Prot Ctl Param */ +#define CS35L35_AMP_GAIN_AUD_CTL 0x19 /* Amp Serial Port Gain Ctl */ +#define CS35L35_AMP_GAIN_PDM_CTL 0x1A /* Amplifier Gain PDM Ctl */ +#define CS35L35_AMP_GAIN_ADV_CTL 0x1B /* Amplifier Gain Ctl */ +#define CS35L35_GPI_CTL 0x1C /* GPI Ctl */ +#define CS35L35_BST_CVTR_V_CTL 0x1D /* Boost Conv Voltage Ctl */ +#define CS35L35_BST_PEAK_I 0x1E /* Boost Conv Peak Current */ +#define CS35L35_BST_RAMP_CTL 0x20 /* Boost Conv Soft Ramp Ctl */ +#define CS35L35_BST_CONV_COEF_1 0x21 /* Boost Conv Coefficients 1 */ +#define CS35L35_BST_CONV_COEF_2 0x22 /* Boost Conv Coefficients 2 */ +#define CS35L35_BST_CONV_SLOPE_COMP 0x23 /* Boost Conv Slope Comp */ +#define CS35L35_BST_CONV_SW_FREQ 0x24 /* Boost Conv L BST SW Freq */ +#define CS35L35_CLASS_H_CTL 0x30 /* CLS H Control */ +#define CS35L35_CLASS_H_HEADRM_CTL 0x31 /* CLS H Headroom Ctl */ +#define CS35L35_CLASS_H_RELEASE_RATE 0x32 /* CLS H Release Rate */ +#define CS35L35_CLASS_H_FET_DRIVE_CTL 0x33 /* CLS H Weak FET Drive Ctl */ +#define CS35L35_CLASS_H_VP_CTL 0x34 /* CLS H VP Ctl */ +#define CS35L35_CLASS_H_STATUS 0x38 /* CLS H Status */ +#define CS35L35_VPBR_CTL 0x3A /* VPBR Ctl */ +#define CS35L35_VPBR_VOL_CTL 0x3B /* VPBR Volume Ctl */ +#define CS35L35_VPBR_TIMING_CTL 0x3C /* VPBR Timing Ctl */ +#define CS35L35_VPBR_MODE_VOL_CTL 0x3D /* VPBR Mode/Attack Vol Ctl */ +#define CS35L35_VPBR_ATTEN_STATUS 0x4B /* VPBR Attenuation Status */ +#define CS35L35_SPKR_MON_CTL 0x4E /* Speaker Monitoring Ctl */ +#define CS35L35_IMON_SCALE_CTL 0x51 /* IMON Scale Ctl */ +#define CS35L35_AUDIN_RXLOC_CTL 0x52 /* Audio Input RX Loc Ctl */ +#define CS35L35_ADVIN_RXLOC_CTL 0x53 /* Advisory Input RX Loc Ctl */ +#define CS35L35_VMON_TXLOC_CTL 0x54 /* VMON TX Loc Ctl */ +#define CS35L35_IMON_TXLOC_CTL 0x55 /* IMON TX Loc Ctl */ +#define CS35L35_VPMON_TXLOC_CTL 0x56 /* VPMON TX Loc Ctl */ +#define CS35L35_VBSTMON_TXLOC_CTL 0x57 /* VBSTMON TX Loc Ctl */ +#define CS35L35_VPBR_STATUS_TXLOC_CTL 0x58 /* VPBR Status TX Loc Ctl */ +#define CS35L35_ZERO_FILL_LOC_CTL 0x59 /* Zero Fill Loc Ctl */ +#define CS35L35_AUDIN_DEPTH_CTL 0x5A /* Audio Input Depth Ctl */ +#define CS35L35_SPKMON_DEPTH_CTL 0x5B /* SPK Mon Output Depth Ctl */ +#define CS35L35_SUPMON_DEPTH_CTL 0x5C /* Supply Mon Out Depth Ctl */ +#define CS35L35_ZEROFILL_DEPTH_CTL 0x5D /* Zero Fill Mon Output Ctl */ +#define CS35L35_MULT_DEV_SYNCH1 0x62 /* Multidevice Synch */ +#define CS35L35_MULT_DEV_SYNCH2 0x63 /* Multidevice Synch 2 */ +#define CS35L35_PROT_RELEASE_CTL 0x64 /* Protection Release Ctl */ +#define CS35L35_DIAG_MODE_REG_LOCK 0x68 /* Diagnostic Mode Reg Lock */ +#define CS35L35_DIAG_MODE_CTL_1 0x69 /* Diagnostic Mode Ctl 1 */ +#define CS35L35_DIAG_MODE_CTL_2 0x6A /* Diagnostic Mode Ctl 2 */ +#define CS35L35_INT_MASK_1 0x70 /* Interrupt Mask 1 */ +#define CS35L35_INT_MASK_2 0x71 /* Interrupt Mask 2 */ +#define CS35L35_INT_MASK_3 0x72 /* Interrupt Mask 3 */ +#define CS35L35_INT_MASK_4 0x73 /* Interrupt Mask 4 */ +#define CS35L35_INT_STATUS_1 0x74 /* Interrupt Status 1 */ +#define CS35L35_INT_STATUS_2 0x75 /* Interrupt Status 2 */ +#define CS35L35_INT_STATUS_3 0x76 /* Interrupt Status 3 */ +#define CS35L35_INT_STATUS_4 0x77 /* Interrupt Status 4 */ +#define CS35L35_PLL_STATUS 0x78 /* PLL Status */ +#define CS35L35_OTP_TRIM_STATUS 0x7E /* OTP Trim Status */ + +#define CS35L35_MAX_REGISTER 0x7F + +/* CS35L35_PWRCTL1 */ +#define CS35L35_SFT_RST 0x80 +#define CS35L35_DISCHG_FLT 0x02 +#define CS35L35_PDN_ALL 0x01 + +/* CS35L35_PWRCTL2 */ +#define CS35L35_PDN_VMON 0x80 +#define CS35L35_PDN_IMON 0x40 +#define CS35L35_PDN_CLASSH 0x20 +#define CS35L35_PDN_VPBR 0x10 +#define CS35L35_PDN_BST 0x04 +#define CS35L35_PDN_AMP 0x01 + +/* CS35L35_PWRCTL3 */ +#define CS35L35_PDN_VBSTMON_OUT 0x10 +#define CS35L35_PDN_VMON_OUT 0x08 + +#define CS35L35_AUDIN_DEPTH_MASK 0x03 +#define CS35L35_AUDIN_DEPTH_SHIFT 0 +#define CS35L35_ADVIN_DEPTH_MASK 0x12 +#define CS35L35_ADVIN_DEPTH_SHIFT 2 +#define CS35L35_SDIN_DEPTH_8 0x01 +#define CS35L35_SDIN_DEPTH_16 0x02 +#define CS35L35_SDIN_DEPTH_24 0x03 + +#define CS35L35_SDOUT_DEPTH_8 0x01 +#define CS35L35_SDOUT_DEPTH_12 0x02 +#define CS35L35_SDOUT_DEPTH_16 0x03 + +#define CS35L35_AUD_IN_LR_MASK 0x80 +#define CS35L35_AUD_IN_LR_SHIFT 7 +#define CS35L35_ADV_IN_LR_MASK 0x80 +#define CS35L35_ADV_IN_LR_SHIFT 7 +#define CS35L35_AUD_IN_LOC_MASK 0x0F +#define CS35L35_AUD_IN_LOC_SHIFT 0 +#define CS35L35_ADV_IN_LOC_MASK 0x0F +#define CS35L35_ADV_IN_LOC_SHIFT 0 + +#define CS35L35_IMON_DEPTH_MASK 0x03 +#define CS35L35_IMON_DEPTH_SHIFT 0 +#define CS35L35_VMON_DEPTH_MASK 0x0C +#define CS35L35_VMON_DEPTH_SHIFT 2 +#define CS35L35_VBSTMON_DEPTH_MASK 0x03 +#define CS35L35_VBSTMON_DEPTH_SHIFT 0 +#define CS35L35_VPMON_DEPTH_MASK 0x0C +#define CS35L35_VPMON_DEPTH_SHIFT 2 +#define CS35L35_VPBRSTAT_DEPTH_MASK 0x18 +#define CS35L35_VPBRSTAT_DEPTH_SHIFT 4 +#define CS35L35_ZEROFILL_DEPTH_MASK 0x03 +#define CS35L35_ZEROFILL_DEPTH_SHIFT 0x00 + +#define CS35L35_MON_TXLOC_MASK 0x3F +#define CS35L35_MON_TXLOC_SHIFT 0 +#define CS35L35_MON_FRM_MASK 0x80 +#define CS35L35_MON_FRM_SHIFT 7 + +#define CS35L35_MS_MASK 0x80 +#define CS35L35_MS_SHIFT 7 +#define CS35L35_SPMODE_MASK 0x40 +#define CS35L35_SP_DRV_MASK 0x10 +#define CS35L35_SP_DRV_SHIFT 4 +#define CS35L35_CLK_CTL2_MASK 0xFF +#define CS35L35_PDM_MODE_MASK 0x40 +#define CS35L35_PDM_MODE_SHIFT 6 +#define CS35L35_CLK_SOURCE_MASK 0x03 +#define CS35L35_CLK_SOURCE_SHIFT 0 +#define CS35L35_CLK_SOURCE_MCLK 0 +#define CS35L35_CLK_SOURCE_SCLK 1 +#define CS35L35_CLK_SOURCE_PDM 2 + +#define CS35L35_SP_SCLKS_MASK 0x0F +#define CS35L35_SP_SCLKS_SHIFT 0x00 +#define CS35L35_SP_SCLKS_16FS 0x03 +#define CS35L35_SP_SCLKS_32FS 0x07 +#define CS35L35_SP_SCLKS_48FS 0x0B +#define CS35L35_SP_SCLKS_64FS 0x0F +#define CS35L35_SP_RATE_MASK 0xC0 + +#define CS35L35_PDN_BST_MASK 0x06 +#define CS35L35_PDN_BST_FETON_SHIFT 1 +#define CS35L35_PDN_BST_FETOFF_SHIFT 2 +#define CS35L35_PWR2_PDN_MASK 0xE0 +#define CS35L35_PWR3_PDN_MASK 0x1E +#define CS35L35_PDN_ALL_MASK 0x01 +#define CS35L35_DISCHG_FILT_MASK 0x02 +#define CS35L35_DISCHG_FILT_SHIFT 1 +#define CS35L35_MCLK_DIS_MASK 0x04 +#define CS35L35_MCLK_DIS_SHIFT 2 + +#define CS35L35_BST_CTL_MASK 0x7F +#define CS35L35_BST_CTL_SHIFT 0 +#define CS35L35_BST_IPK_MASK 0x1F +#define CS35L35_BST_IPK_SHIFT 0 +#define CS35L35_AMP_MUTE_MASK 0x20 +#define CS35L35_AMP_MUTE_SHIFT 5 +#define CS35L35_AMP_GAIN_ZC_MASK 0x10 +#define CS35L35_AMP_GAIN_ZC_SHIFT 4 + +/* Class H Algorithm Control */ +#define CS35L35_CH_STEREO_MASK 0x40 +#define CS35L35_CH_STEREO_SHIFT 6 +#define CS35L35_CH_BST_OVR_MASK 0x04 +#define CS35L35_CH_BST_OVR_SHIFT 2 +#define CS35L35_CH_BST_LIM_MASK 0x08 +#define CS35L35_CH_BST_LIM_SHIFT 3 +#define CS35L35_CH_MEM_DEPTH_MASK 0x01 +#define CS35L35_CH_MEM_DEPTH_SHIFT 0 +#define CS35L35_CH_HDRM_CTL_MASK 0x3F +#define CS35L35_CH_HDRM_CTL_SHIFT 0 +#define CS35L35_CH_REL_RATE_MASK 0xFF +#define CS35L35_CH_REL_RATE_SHIFT 0 +#define CS35L35_CH_WKFET_DIS_MASK 0x80 +#define CS35L35_CH_WKFET_DIS_SHIFT 7 +#define CS35L35_CH_WKFET_DEL_MASK 0x70 +#define CS35L35_CH_WKFET_DEL_SHIFT 4 +#define CS35L35_CH_WKFET_THLD_MASK 0x0F +#define CS35L35_CH_WKFET_THLD_SHIFT 0 +#define CS35L35_CH_VP_AUTO_MASK 0x80 +#define CS35L35_CH_VP_AUTO_SHIFT 7 +#define CS35L35_CH_VP_RATE_MASK 0x60 +#define CS35L35_CH_VP_RATE_SHIFT 5 +#define CS35L35_CH_VP_MAN_MASK 0x1F +#define CS35L35_CH_VP_MAN_SHIFT 0 + +/* CS35L35_PROT_RELEASE_CTL */ +#define CS35L35_CAL_ERR_RLS 0x80 +#define CS35L35_SHORT_RLS 0x04 +#define CS35L35_OTW_RLS 0x02 +#define CS35L35_OTE_RLS 0x01 + +/* INT Mask Registers */ +#define CS35L35_INT1_CRIT_MASK 0x38 +#define CS35L35_INT2_CRIT_MASK 0xEF +#define CS35L35_INT3_CRIT_MASK 0xEE +#define CS35L35_INT4_CRIT_MASK 0xFF + +/* PDN DONE Masks */ +#define CS35L35_M_PDN_DONE_SHIFT 4 +#define CS35L35_M_PDN_DONE_MASK 0x10 + +/* CS35L35_INT_1 */ +#define CS35L35_CAL_ERR 0x80 +#define CS35L35_OTP_ERR 0x40 +#define CS35L35_LRCLK_ERR 0x20 +#define CS35L35_SPCLK_ERR 0x10 +#define CS35L35_MCLK_ERR 0x08 +#define CS35L35_AMP_SHORT 0x04 +#define CS35L35_OTW 0x02 +#define CS35L35_OTE 0x01 + +/* CS35L35_INT_2 */ +#define CS35L35_PDN_DONE 0x10 +#define CS35L35_VPBR_ERR 0x02 +#define CS35L35_VPBR_CLR 0x01 + +/* CS35L35_INT_3 */ +#define CS35L35_BST_HIGH 0x10 +#define CS35L35_BST_HIGH_FLAG 0x08 +#define CS35L35_BST_IPK_FLAG 0x04 +#define CS35L35_LBST_SHORT 0x01 + +/* CS35L35_INT_4 */ +#define CS35L35_VMON_OVFL 0x08 +#define CS35L35_IMON_OVFL 0x04 + +#define CS35L35_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct cs35l35_private { + struct snd_soc_codec *codec; + struct cs35l35_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int sysclk; + int sclk; + bool pdm_mode; + bool i2s_mode; + bool slave_mode; + /* GPIO for /RST */ + struct gpio_desc *reset_gpio; + struct completion pdn_done; +}; + +static const char * const cs35l35_supplies[] = { + "VA", + "VP", +}; + +#endif -- 1.9.1

1 0