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[PATCH 0/2] soundwire: add DMI quirks to work-around broken _ADR
by Pierre-Louis Bossart 08 Feb '21

08 Feb '21

Unfortunately the DSDT often exposes _ADR information which reports the presence of the wrong codecs, which prevents drivers from being probed, or they are reported on the wrong link, which breaks the ASoC machine driver. This patchset suggests a device-specific remapping to the expected values reported by codecs, controlled by a DMI check. Adding such quirks in the kernel code is ugly, but for commercial devices it's certainly better than asking every user to override the DSDT. Pierre-Louis Bossart (2): soundwire: slave: introduce DMI quirks for HP Spectre x360 Convertible soundwire: slave: add DMI quirk for Dell SKU 0A3E drivers/soundwire/slave.c | 79 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 79 insertions(+) -- 2.25.1

2 6

[PATCH 1/2] ALSA: hda: intel-dsp-config: Add FLAG_BYT_FIRST / _SECOND defines
by Hans de Goede 08 Feb '21

08 Feb '21

Instead of hardcording the SST driver having the highest prio, add FLAG_BYT_FIRST and FLAG_BYT_SECOND defines, which get set like this when both drivers are enabled: #define FLAG_BYT_FIRST FLAG_SST #define FLAG_BYT_SECOND FLAG_SOF And when only 1 driver is enabled then FLAG_BYT_FIRST gets set to the flag for that driver. This is a preparation patch for making which driver is preferred configurable through Kconfig. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/hda/intel-dsp-config.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index c45686172517..2201a1e6944e 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -452,6 +452,16 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) } EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); +/* FLAG_BYT_FIRST / _SECOND determine which driver is preferred on BYT/CHT when both are enabled */ +#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) && IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) +#define FLAG_BYT_FIRST FLAG_SST +#define FLAG_BYT_SECOND FLAG_SOF +#elif IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) +#define FLAG_BYT_FIRST FLAG_SST +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) +#define FLAG_BYT_FIRST FLAG_SOF +#endif + /* * configuration table * - the order of similar ACPI ID entries is important! @@ -459,28 +469,28 @@ EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); */ static const struct config_entry acpi_config_table[] = { /* BayTrail */ -#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) +#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) || IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) { - .flags = FLAG_SST, + .flags = FLAG_BYT_FIRST, .acpi_hid = "80860F28", }, #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) +#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) && IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) { - .flags = FLAG_SOF, + .flags = FLAG_BYT_SECOND, .acpi_hid = "80860F28", }, #endif /* CherryTrail */ -#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) +#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) || IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) { - .flags = FLAG_SST, + .flags = FLAG_BYT_FIRST, .acpi_hid = "808622A8", }, #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) +#if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) && IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) { - .flags = FLAG_SOF, + .flags = FLAG_BYT_SECOND, .acpi_hid = "808622A8", }, #endif -- 2.30.0

2 6

[PATCH v2 0/9] ALSA: add virtio sound driver
by Anton Yakovlev 08 Feb '21

08 Feb '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. The series is based (and was actually tested) on Linus's master branch [2], on top of commit 1e2a199f6ccd ("Merge tag 'spi-fix-v5.11-rc4' of ...") As a device part was used OpenSynergy proprietary implementation. Any comments are very welcome. v1->v2 changes: 1. For some reason, in the previous patch series, several patches were squashed. Fixed this issue to make the review easier. 2. Added mst(a)redhat.com to the MAINTAINERS. 3. When creating virtqueues, now only the event virtqueue is disabled. It's enabled only after successful initialization of the device. 4. Added additional comments to the reset worker function: [2/9] virtio_card.c:virtsnd_reset_fn() 5. Added check that VIRTIO_F_VERSION_1 feature bit is set. 6. Added additional comments to the device removing function: [2/9] virtio_card.c:virtsnd_remove() 7. Added additional comments to the tx/rx interrupt handler: [5/9] virtio_pcm_msg.c:virtsnd_pcm_msg_complete() 8. Added additional comments to substream release wait function. [6/9] virtio_pcm_ops.c:virtsnd_pcm_released() [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html [2] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git Anton Yakovlev (9): uapi: virtio_ids: add a sound device type ID from OASIS spec ALSA: virtio: add virtio sound driver ALSA: virtio: handling control messages ALSA: virtio: build PCM devices and substream hardware descriptors ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce PCM channel map support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 9 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 361 ++++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 577 +++++++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 121 +++++++ sound/virtio/virtio_chmap.c | 237 +++++++++++++ sound/virtio/virtio_ctl_msg.c | 293 ++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 122 +++++++ sound/virtio/virtio_jack.c | 255 ++++++++++++++ sound/virtio/virtio_pcm.c | 582 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 132 ++++++++ sound/virtio/virtio_pcm_msg.c | 325 ++++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 528 +++++++++++++++++++++++++++++ 17 files changed, 3570 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.0

3 29

[PATCH] ASoC: SOF: debug: Fix a potential issue on string buffer termination
by Hui Wang 08 Feb '21

08 Feb '21

The function simple_write_to_buffer() doesn't add string termination at the end of buf, we need to add it on our own if calling that function to write the size of count chars to buf. This change refers to the function tokenize_input() in debug.c and the function sof_dfsentry_trace_filter_write() in trace.c. We didn't find this potential issue in the past because sometimes we are very lucky, we kzalloc the size of count buf, the kernel not only returns a buf with buf[0 ... (count - 1)] = 0 but buf[count] = 0, with this luck, this issue will not be exposed. Fixes: 091c12e1f50c ("ASoC: SOF: debug: add new debugfs entries for IPC flood test") Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/sof/debug.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 30213a1beaaa..edd4893119dd 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -352,9 +352,10 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, char *string; int ret; - string = kzalloc(count, GFP_KERNEL); + string = kzalloc(count+1, GFP_KERNEL); if (!string) return -ENOMEM; + string[count] = '\0'; size = simple_write_to_buffer(string, count, ppos, buffer, count); ret = size; -- 2.25.1

2 2

[sound:for-next 76/104] powerpc-linux-ld: warning: orphan section `.data..LASANLOC27' from `sound/core/init.o' being placed in section `.data..LASANLOC27'
by kernel test robot 08 Feb '21

08 Feb '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 3c4ab49ec59b94651dea7c7b0104c781c79c62b5 commit: 75b1a8f9d62e50f05d0e4e9f3c8bcde32527ffc1 [76/104] ALSA: Convert strlcpy to strscpy when return value is unused config: powerpc64-randconfig-r032-20210207 (attached as .config) compiler: powerpc-linux-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-next git checkout 75b1a8f9d62e50f05d0e4e9f3c8bcde32527ffc1 # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=powerpc64 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `drivers/tee/tee_core.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `drivers/tee/tee_core.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `drivers/tee/tee_core.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/tee/tee_shm.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/tee/tee_shm.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/tee/tee_shm_pool.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/tee/tee_shm_pool.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/mux/core.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/mux/core.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `drivers/mux/core.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `drivers/mux/core.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `drivers/mux/core.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `drivers/mux/core.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `drivers/mux/core.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/mux/adg792a.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/mux/adg792a.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `drivers/mux/adg792a.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `drivers/mux/adg792a.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `drivers/mux/adg792a.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/mux/adgs1408.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/mux/adgs1408.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `drivers/mux/adgs1408.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `drivers/mux/adgs1408.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `drivers/mux/adgs1408.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/mux/mmio.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/mux/mmio.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `drivers/mux/mmio.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `drivers/mux/mmio.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/counter/counter.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `drivers/counter/counter.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC12' powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `drivers/counter/ti-eqep.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC12' powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `drivers/counter/microchip-tcb-capture.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/sound_core.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/sound_core.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/sound_core.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/sound.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/sound.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/sound.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/sound.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/sound.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/sound.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/sound.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/sound.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/sound.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/sound.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/sound.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/init.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/init.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/init.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/core/init.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `sound/core/init.o' being placed in section `.data..LASANLOC12' powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `sound/core/init.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC14' from `sound/core/init.o' being placed in section `.data..LASANLOC14' powerpc-linux-ld: warning: orphan section `.data..LASANLOC15' from `sound/core/init.o' being placed in section `.data..LASANLOC15' powerpc-linux-ld: warning: orphan section `.data..LASANLOC16' from `sound/core/init.o' being placed in section `.data..LASANLOC16' powerpc-linux-ld: warning: orphan section `.data..LASANLOC17' from `sound/core/init.o' being placed in section `.data..LASANLOC17' powerpc-linux-ld: warning: orphan section `.data..LASANLOC18' from `sound/core/init.o' being placed in section `.data..LASANLOC18' powerpc-linux-ld: warning: orphan section `.data..LASANLOC19' from `sound/core/init.o' being placed in section `.data..LASANLOC19' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/init.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC20' from `sound/core/init.o' being placed in section `.data..LASANLOC20' powerpc-linux-ld: warning: orphan section `.data..LASANLOC21' from `sound/core/init.o' being placed in section `.data..LASANLOC21' powerpc-linux-ld: warning: orphan section `.data..LASANLOC22' from `sound/core/init.o' being placed in section `.data..LASANLOC22' powerpc-linux-ld: warning: orphan section `.data..LASANLOC23' from `sound/core/init.o' being placed in section `.data..LASANLOC23' powerpc-linux-ld: warning: orphan section `.data..LASANLOC24' from `sound/core/init.o' being placed in section `.data..LASANLOC24' powerpc-linux-ld: warning: orphan section `.data..LASANLOC25' from `sound/core/init.o' being placed in section `.data..LASANLOC25' powerpc-linux-ld: warning: orphan section `.data..LASANLOC26' from `sound/core/init.o' being placed in section `.data..LASANLOC26' >> powerpc-linux-ld: warning: orphan section `.data..LASANLOC27' from `sound/core/init.o' being placed in section `.data..LASANLOC27' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/init.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/init.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/init.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/init.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/init.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/init.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/init.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/memory.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/control.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/control.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/control.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/core/control.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/control.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/control.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/control.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/control.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/control.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/control.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/control.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/control.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/info.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/info.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/info.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/info.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/info.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/info.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/info.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/info.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/timer.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/timer.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/timer.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/core/timer.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `sound/core/timer.o' being placed in section `.data..LASANLOC12' powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `sound/core/timer.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC14' from `sound/core/timer.o' being placed in section `.data..LASANLOC14' powerpc-linux-ld: warning: orphan section `.data..LASANLOC15' from `sound/core/timer.o' being placed in section `.data..LASANLOC15' powerpc-linux-ld: warning: orphan section `.data..LASANLOC16' from `sound/core/timer.o' being placed in section `.data..LASANLOC16' powerpc-linux-ld: warning: orphan section `.data..LASANLOC17' from `sound/core/timer.o' being placed in section `.data..LASANLOC17' powerpc-linux-ld: warning: orphan section `.data..LASANLOC18' from `sound/core/timer.o' being placed in section `.data..LASANLOC18' powerpc-linux-ld: warning: orphan section `.data..LASANLOC19' from `sound/core/timer.o' being placed in section `.data..LASANLOC19' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/timer.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC20' from `sound/core/timer.o' being placed in section `.data..LASANLOC20' powerpc-linux-ld: warning: orphan section `.data..LASANLOC21' from `sound/core/timer.o' being placed in section `.data..LASANLOC21' powerpc-linux-ld: warning: orphan section `.data..LASANLOC22' from `sound/core/timer.o' being placed in section `.data..LASANLOC22' powerpc-linux-ld: warning: orphan section `.data..LASANLOC23' from `sound/core/timer.o' being placed in section `.data..LASANLOC23' powerpc-linux-ld: warning: orphan section `.data..LASANLOC24' from `sound/core/timer.o' being placed in section `.data..LASANLOC24' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/timer.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/timer.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/timer.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/timer.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/timer.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/timer.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/timer.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/hrtimer.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/hrtimer.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/hrtimer.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/hrtimer.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/hrtimer.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/pcm.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/pcm.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/pcm.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/core/pcm.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `sound/core/pcm.o' being placed in section `.data..LASANLOC12' powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `sound/core/pcm.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC14' from `sound/core/pcm.o' being placed in section `.data..LASANLOC14' powerpc-linux-ld: warning: orphan section `.data..LASANLOC15' from `sound/core/pcm.o' being placed in section `.data..LASANLOC15' powerpc-linux-ld: warning: orphan section `.data..LASANLOC16' from `sound/core/pcm.o' being placed in section `.data..LASANLOC16' powerpc-linux-ld: warning: orphan section `.data..LASANLOC17' from `sound/core/pcm.o' being placed in section `.data..LASANLOC17' >> powerpc-linux-ld: warning: orphan section `.data..LASANLOC18' from `sound/core/pcm.o' being placed in section `.data..LASANLOC18' >> powerpc-linux-ld: warning: orphan section `.data..LASANLOC19' from `sound/core/pcm.o' being placed in section `.data..LASANLOC19' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/pcm.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/pcm.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/pcm.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/pcm.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/pcm.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/pcm.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/pcm.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/pcm.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/pcm_native.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC12' powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC14' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC14' powerpc-linux-ld: warning: orphan section `.data..LASANLOC15' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC15' powerpc-linux-ld: warning: orphan section `.data..LASANLOC16' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC16' powerpc-linux-ld: warning: orphan section `.data..LASANLOC17' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC17' powerpc-linux-ld: warning: orphan section `.data..LASANLOC18' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC18' powerpc-linux-ld: warning: orphan section `.data..LASANLOC19' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC19' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC20' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC20' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/pcm_native.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/pcm_lib.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/pcm_lib.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/pcm_lib.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/pcm_lib.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/pcm_lib.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/pcm_misc.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/pcm_misc.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/pcm_memory.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/pcm_memory.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/pcm_memory.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/pcm_memory.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/pcm_memory.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/pcm_memory.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/pcm_memory.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/memalloc.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq_device.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq_device.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq_device.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq_device.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/rawmidi.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC12' >> powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/rawmidi.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC12' powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC14' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC14' powerpc-linux-ld: warning: orphan section `.data..LASANLOC15' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC15' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/seq/seq.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC12' powerpc-linux-ld: warning: orphan section `.data..LASANLOC13' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC13' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/seq/seq_clientmgr.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_memory.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_memory.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_memory.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_queue.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_queue.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_queue.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq_queue.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/seq/seq_queue.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/seq/seq_queue.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/seq/seq_queue.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_fifo.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_fifo.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_fifo.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_prioq.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_prioq.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_timer.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_timer.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_system.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_system.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_system.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq_system.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/seq/seq_system.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_ports.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_ports.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_ports.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq_ports.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/seq/seq_ports.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_info.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_info.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_info.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq_info.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASANLOC9' from `sound/core/seq/seq_midi.o' being placed in section `.data..LASANLOC9' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_midi_event.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_midi_event.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_midi_event.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq_midi_event.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/seq/seq_midi_event.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/seq/seq_midi_event.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/seq/seq_midi_event.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC2' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASANLOC2' powerpc-linux-ld: warning: orphan section `.data..LASANLOC3' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASANLOC3' powerpc-linux-ld: warning: orphan section `.data..LASANLOC4' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASANLOC4' powerpc-linux-ld: warning: orphan section `.data..LASANLOC5' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASANLOC5' powerpc-linux-ld: warning: orphan section `.data..LASANLOC6' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASANLOC6' powerpc-linux-ld: warning: orphan section `.data..LASANLOC7' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASANLOC7' powerpc-linux-ld: warning: orphan section `.data..LASANLOC8' from `sound/core/seq/seq_virmidi.o' being placed in section `.data..LASANLOC8' powerpc-linux-ld: warning: orphan section `.data..LASAN0' from `sound/drivers/dummy.o' being placed in section `.data..LASAN0' powerpc-linux-ld: warning: orphan section `.data..LASANLOC1' from `sound/drivers/dummy.o' being placed in section `.data..LASANLOC1' powerpc-linux-ld: warning: orphan section `.data..LASANLOC10' from `sound/drivers/dummy.o' being placed in section `.data..LASANLOC10' powerpc-linux-ld: warning: orphan section `.data..LASANLOC11' from `sound/drivers/dummy.o' being placed in section `.data..LASANLOC11' powerpc-linux-ld: warning: orphan section `.data..LASANLOC12' from `sound/drivers/dummy.o' being placed in section `.data..LASANLOC12' .. --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

3 2

[PATCH 0/7] Add audio driver base on rpmsg on i.MX platform
by Shengjiu Wang 07 Feb '21

07 Feb '21

On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core, Linux is running on A core, RTOS is running on M core. The audio hardware device can be controlled by Cortex-M device, So audio playback/capture can be handled by M core. Rpmsg is the interface for sending and receiving msg to and from M core, that we can create a virtual sound on Cortex-A core side. A core will tell the Cortex-M core sound format/rate/channel, where is the data buffer, what is the period size, when to start, when to stop and when suspend or resume happen, each of this behavior there is defined rpmsg command. Especially we designed the low power audio case, that is to allocate a large buffer and fill the data, then Cortex-A core can go to sleep mode, Cortex-M core continue to play the sound, when the buffer is consumed, Cortex-M core will trigger the Cortex-A core to wakeup to fill data. Shengjiu Wang (7): ASoC: soc-component: Add snd_soc_pcm_component_ack ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver .../devicetree/bindings/sound/fsl,rpmsg.yaml | 80 ++ .../bindings/sound/imx-audio-rpmsg.yaml | 48 + include/sound/soc-component.h | 3 + sound/soc/fsl/Kconfig | 28 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl_rpmsg.c | 258 +++++ sound/soc/fsl/fsl_rpmsg.h | 38 + sound/soc/fsl/imx-audio-rpmsg.c | 142 +++ sound/soc/fsl/imx-pcm-rpmsg.c | 898 ++++++++++++++++++ sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++ sound/soc/fsl/imx-rpmsg.c | 148 +++ sound/soc/soc-component.c | 14 + sound/soc/soc-pcm.c | 2 + 13 files changed, 2177 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml create mode 100644 sound/soc/fsl/fsl_rpmsg.c create mode 100644 sound/soc/fsl/fsl_rpmsg.h create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h create mode 100644 sound/soc/fsl/imx-rpmsg.c -- 2.27.0

3 13

[PATCH] ALSA: pci: Assign boolean values to a bool variable
by Jiapeng Chong 07 Feb '21

07 Feb '21

Fix the following coccicheck warnings: ./sound/pci/azt3328.c:2451:2-16: WARNING: Assignment of 0/1 to bool variable. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/pci/azt3328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 77c7030..37f8fc4 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2448,7 +2448,7 @@ static void snd_azf3328_mixer_reg_decode(struct azf3328_mixer_reg *r, unsigned l /* shutdown codecs to reduce power / noise */ /* have ...ctrl_codec_activity() act properly */ - codec->running = 1; + codec->running = true; snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); spin_lock_irq(codec->lock); -- 1.8.3.1

2 1

[PATCH 0/4] ALSA: pcm: sync_stop fixes
by Takashi Iwai 06 Feb '21

06 Feb '21

Hi, here is a patch set to cover the missing handling of the recently introduced sync_stop PCM ops at suspend and disconnect states. The last patch is a pure cleanup. Takashi === Takashi Iwai (4): ALSA: pcm: Call sync_stop at disconnection ALSA: pcm: Assure sync with the pending stop operation at suspend ALSA: pcm: Don't call sync_stop if it hasn't been stopped ALSA: pcm: Use for_each_pcm_substream() macro sound/core/init.c | 4 +++ sound/core/pcm.c | 27 +++++++++++---------- sound/core/pcm_local.h | 7 ++++++ sound/core/pcm_memory.c | 12 +++------ sound/core/pcm_native.c | 54 +++++++++++++++++++++-------------------- 5 files changed, 57 insertions(+), 47 deletions(-) -- 2.26.2

1 4

[PATCH 0/3] ALSA: a few more USB-audio fixes
by Takashi Iwai 06 Feb '21

06 Feb '21

Hi, here is a small patch set for USB-audio. The goal of this patch set is to cover a small race window at the endpoint state transition, and the preparation for the upcoming patch set for PCM sync_stop fixes. Takashi === Takashi Iwai (3): ALSA: usb-audio: Handle invalid running state at releasing EP ALSA: usb-audio: More strict state change in EP ALSA: usb-audio: Don't avoid stopping the stream at disconnection sound/usb/card.h | 2 +- sound/usb/endpoint.c | 85 +++++++++++++++++++++++++------------------- sound/usb/pcm.c | 5 +-- 3 files changed, 51 insertions(+), 41 deletions(-) -- 2.26.2

1 3

BUG: KASAN: use-after-free in snd_complete_urb+0x109e/0x1740 [snd_usb_audio] (5.11-rc6)
by Mikhail Gavrilov 06 Feb '21

06 Feb '21

Hi folks. On 5.11-rc6 (git 3aaf0a27ffc2) I caught a new issue. For unknown reason sound disappeared in my headset Hyperx orbit s. But after reconnecting to another USB port headset stopped being detected as USB device in dmesg and in log appears a record about bug KASAN: use-after-free. [37920.370627] xhci_hcd 0000:06:00.3: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x0013 address=0xfe836480 flags=0x0000] [37920.370720] xhci_hcd 0000:06:00.3: WARNING: Host System Error [37925.514865] xhci_hcd 0000:06:00.3: xHCI host not responding to stop endpoint command. [37925.514875] xhci_hcd 0000:06:00.3: USBSTS: HCHalted HSE EINT HCE [37925.514883] xhci_hcd 0000:06:00.3: xHCI host controller not responding, assume dead [37925.515700] xhci_hcd 0000:06:00.3: HC died; cleaning up [37925.515760] usb 3-1: USB disconnect, device number 2 [37925.515765] usb 3-1.1: USB disconnect, device number 5 [37925.515896] ================================================================== [37925.515899] BUG: KASAN: use-after-free in snd_complete_urb+0x109e/0x1740 [snd_usb_audio] [37925.515919] Read of size 8 at addr ffff88813a264260 by task kworker/19:0/649395 [37925.515926] CPU: 19 PID: 649395 Comm: kworker/19:0 Tainted: G W --------- --- 5.11.0-0.rc6.20210203git3aaf0a27ffc2.144.fc34.x86_64 #1 [37925.515930] Hardware name: System manufacturer System Product Name/ROG STRIX X570-I GAMING, BIOS 3402 01/13/2021 [37925.515934] Workqueue: usb_hub_wq hub_event [37925.515940] Call Trace: [37925.515943] <IRQ> [37925.515945] dump_stack+0xae/0xe5 [37925.515955] print_address_description.constprop.0+0x18/0x160 [37925.515960] ? snd_complete_urb+0x109e/0x1740 [snd_usb_audio] [37925.515976] kasan_report.cold+0x7f/0x10e [37925.515981] ? snd_complete_urb+0x109e/0x1740 [snd_usb_audio] [37925.515997] snd_complete_urb+0x109e/0x1740 [snd_usb_audio] [37925.516013] ? usb_hcd_unmap_urb_for_dma+0x22/0x660 [37925.516020] ? xhci_unmap_urb_for_dma+0x21/0x3f0 [37925.516025] __usb_hcd_giveback_urb+0x24b/0x400 [37925.516029] usb_giveback_urb_bh+0x239/0x3f0 [37925.516034] ? __usb_hcd_giveback_urb+0x400/0x400 [37925.516037] ? ktime_get+0xbd/0x180 [37925.516042] ? sched_clock+0x5/0x10 [37925.516046] tasklet_action_common.constprop.0+0x166/0x320 [37925.516052] __do_softirq+0x1f6/0x9e9 [37925.516058] asm_call_irq_on_stack+0x12/0x20 [37925.516062] </IRQ> [37925.516064] do_softirq_own_stack+0x85/0xa0 [37925.516068] __irq_exit_rcu+0x201/0x2e0 [37925.516071] irq_exit_rcu+0xa/0x20 [37925.516074] sysvec_apic_timer_interrupt+0x48/0xa0 [37925.516079] asm_sysvec_apic_timer_interrupt+0x12/0x20 [37925.516083] RIP: 0010:__list_del_entry_valid+0xad/0xf0 [37925.516088] Code: 02 00 75 51 49 8b 14 24 48 39 ea 0f 85 55 d1 2a 01 48 b8 00 00 00 00 00 fc ff df 49 8d 7d 08 48 89 fa 48 c1 ea 03 80 3c 02 00 <75> 22 49 8b 55 08 48 39 ea 0f 85 66 d1 2a 01 5d b8 01 00 00 00 41 [37925.516092] RSP: 0018:ffffc9001b9f77f8 EFLAGS: 00000246 [37925.516096] RAX: dffffc0000000000 RBX: 0000000000000001 RCX: ffffffffb0464036 [37925.516099] RDX: 1ffff1102569fa0f RSI: 0000000000000004 RDI: ffff88812b4fd078 [37925.516101] RBP: ffff88812ac8b070 R08: 0000000000000001 R09: ffff88812ac8b083 [37925.516104] R10: ffffed1025591610 R11: 0000000000000001 R12: ffff888129d63048 [37925.516106] R13: ffff88812b4fd070 R14: ffff88812ac8b078 R15: dffffc0000000000 [37925.516114] ? klist_put+0x76/0x170 [37925.516119] klist_release+0x4e/0x450 [37925.516123] ? __device_link_free_srcu+0xe0/0xe0 [37925.516128] klist_put+0xa9/0x170 [37925.516132] device_del+0x22c/0xb50 [37925.516137] ? __device_link_del+0x350/0x350 [37925.516141] ? lock_acquire+0x5ab/0x790 [37925.516144] ? mutex_lock_io_nested+0x1010/0x1010 [37925.516147] ? lock_release+0x579/0xd20 [37925.516151] device_unregister+0x16/0xa0 [37925.516155] usb_remove_ep_devs+0x37/0x70 [37925.516158] usb_disable_device+0x25c/0x5f0 [37925.516163] usb_disconnect.cold+0x20c/0x683 [37925.516172] usb_disconnect.cold+0x107/0x683 [37925.516176] ? trace_hardirqs_on+0x1c/0x160 [37925.516181] hub_quiesce+0x148/0x2d0 [37925.516186] hub_event+0xcb1/0x3880 [37925.516189] ? lock_acquire+0x5ab/0x790 [37925.516192] ? lock_acquire+0x5ab/0x790 [37925.516196] ? lock_acquire+0x5ab/0x790 [37925.516199] ? lock_release+0x579/0xd20 [37925.516204] ? lock_downgrade+0x6b0/0x6b0 [37925.516207] ? lock_contended+0xbe0/0xbe0 [37925.516209] ? do_raw_spin_lock+0x250/0x250 [37925.516214] ? hub_port_debounce+0x2e0/0x2e0 [37925.516217] ? lock_acquire+0x5ab/0x790 [37925.516220] ? lock_release+0xd20/0xd20 [37925.516223] ? lock_downgrade+0x6b0/0x6b0 [37925.516226] ? do_raw_spin_trylock+0xa5/0x160 [37925.516230] ? lock_contended+0xbe0/0xbe0 [37925.516237] process_one_work+0x89f/0x1450 [37925.516243] ? pwq_dec_nr_in_flight+0x260/0x260 [37925.516247] ? lock_acquired+0x5d4/0xaf0 [37925.516251] worker_thread+0x590/0xf80 [37925.516258] ? process_one_work+0x1450/0x1450 [37925.516262] kthread+0x368/0x440 [37925.516265] ? _raw_spin_unlock_irq+0x24/0x40 [37925.516268] ? __kthread_bind_mask+0x90/0x90 [37925.516272] ret_from_fork+0x22/0x30 [37925.516282] Allocated by task 662343: [37925.516285] kasan_save_stack+0x1b/0x40 [37925.516292] ____kasan_kmalloc.constprop.0+0x84/0xa0 [37925.516295] snd_usb_add_endpoint+0xf9/0x4f0 [snd_usb_audio] [37925.516311] __snd_usb_parse_audio_interface+0xab3/0x35e0 [snd_usb_audio] [37925.516328] snd_usb_parse_audio_interface+0x5f/0xb0 [snd_usb_audio] [37925.516343] snd_usb_create_stream.isra.0+0x1be/0x430 [snd_usb_audio] [37925.516358] usb_audio_probe+0x731/0x21d0 [snd_usb_audio] [37925.516376] usb_probe_interface+0x26e/0x750 [37925.516379] really_probe+0x224/0xcc0 [37925.516382] driver_probe_device+0x1f2/0x380 [37925.516384] bus_for_each_drv+0x119/0x180 [37925.516388] __device_attach+0x26d/0x3d0 [37925.516390] bus_probe_device+0x1a5/0x260 [37925.516393] device_add+0x8c6/0x19e0 [37925.516398] usb_set_configuration+0x9df/0x1550 [37925.516401] usb_generic_driver_probe+0x79/0xa0 [37925.516404] usb_probe_device+0xa3/0x250 [37925.516407] really_probe+0x224/0xcc0 [37925.516409] driver_probe_device+0x1f2/0x380 [37925.516412] bus_for_each_drv+0x119/0x180 [37925.516415] __device_attach+0x26d/0x3d0 [37925.516422] bus_probe_device+0x1a5/0x260 [37925.516424] device_add+0x8c6/0x19e0 [37925.516427] usb_new_device.cold+0x536/0xc06 [37925.516430] hub_event+0x219b/0x3880 [37925.516433] process_one_work+0x89f/0x1450 [37925.516436] worker_thread+0x590/0xf80 [37925.516440] kthread+0x368/0x440 [37925.516443] ret_from_fork+0x22/0x30 [37925.516450] Last potentially related work creation: [37925.516452] kasan_save_stack+0x1b/0x40 [37925.516455] kasan_record_aux_stack+0xbc/0xe0 [37925.516459] call_rcu+0x162/0x1100 [37925.516462] amdgpu_bo_list_put+0x16c/0x1b0 [amdgpu] [37925.516650] amdgpu_cs_ioctl+0x2117/0x4d10 [amdgpu] [37925.516839] drm_ioctl_kernel+0x1af/0x240 [drm] [37925.516870] drm_ioctl+0x43b/0x8e0 [drm] [37925.516898] amdgpu_drm_ioctl+0xce/0x1b0 [amdgpu] [37925.517074] __x64_sys_ioctl+0x127/0x190 [37925.517078] do_syscall_64+0x33/0x40 [37925.517083] entry_SYSCALL_64_after_hwframe+0x44/0xa9 [37925.517089] Second to last potentially related work creation: [37925.517091] kasan_save_stack+0x1b/0x40 [37925.517093] kasan_record_aux_stack+0xbc/0xe0 [37925.517096] call_rcu+0x162/0x1100 [37925.517101] amdgpu_bo_list_put+0x16c/0x1b0 [amdgpu] [37925.517282] amdgpu_cs_ioctl+0x2117/0x4d10 [amdgpu] [37925.517461] drm_ioctl_kernel+0x1af/0x240 [drm] [37925.517489] drm_ioctl+0x43b/0x8e0 [drm] [37925.517518] amdgpu_drm_ioctl+0xce/0x1b0 [amdgpu] [37925.517694] __x64_sys_ioctl+0x127/0x190 [37925.517700] do_syscall_64+0x33/0x40 [37925.517704] entry_SYSCALL_64_after_hwframe+0x44/0xa9 [37925.517709] The buggy address belongs to the object at ffff88813a264000 which belongs to the cache kmalloc-8k of size 8192 [37925.517712] The buggy address is located 608 bytes inside of 8192-byte region [ffff88813a264000, ffff88813a266000) [37925.517716] The buggy address belongs to the page: [37925.517718] page:00000000a07d7675 refcount:1 mapcount:0 mapping:0000000000000000 index:0xffff88813a264000 pfn:0x13a260 [37925.517726] head:00000000a07d7675 order:3 compound_mapcount:0 compound_pincount:0 [37925.517729] flags: 0x17ffffc0010200(slab|head) [37925.517735] raw: 0017ffffc0010200 0000000000000000 0000000100000001 ffff888100042280 [37925.517738] raw: ffff88813a264000 0000000080020001 00000001ffffffff ffff8882fa23cca1 [37925.517740] page dumped because: kasan: bad access detected [37925.517742] pages's memcg:ffff8882fa23cca1 [37925.517746] Memory state around the buggy address: [37925.517751] ffff88813a264100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb [37925.517754] ffff88813a264180: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb [37925.517756] >ffff88813a264200: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb [37925.517758] ^ [37925.517761] ffff88813a264280: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb [37925.517763] ffff88813a264300: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb [37925.517765] ================================================================== [37925.521109] usb 3-1.2: USB disconnect, device number 9 [37925.521116] usb 3-1.2.4: USB disconnect, device number 21 [37925.522472] usb 4-1: USB disconnect, device number 2 [37925.522477] usb 4-1.1: USB disconnect, device number 3 [37925.525597] usb 4-1.2: USB disconnect, device number 4 $ eu-addr2line -e /lib/debug/lib/modules/5.11.0-0.rc6.20210203git3aaf0a27ffc2.144.fc34.x86_64/kernel/sound/usb/snd-usb-audio.ko.debug snd_complete_urb+0x109e sound/usb/endpoint.c:444:27 $ cat -s -n /usr/src/debug/kernel-20210204git61556703b610/linux-5.11.0-0.rc6.20210204git61556703b610.144.fc34.x86_64/sound/usb/endpoint.c | head -454 | tail -20 435 } 436 } 437 438 /* 439 * complete callback for urbs 440 */ 441 static void snd_complete_urb(struct urb *urb) 442 { 443 struct snd_urb_ctx *ctx = urb->context; 444 struct snd_usb_endpoint *ep = ctx->ep; 445 unsigned long flags; 446 int err; 447 448 if (unlikely(urb->status == -ENOENT || /* unlinked */ 449 urb->status == -ENODEV || /* device removed */ 450 urb->status == -ECONNRESET || /* unlinked */ 451 urb->status == -ESHUTDOWN)) /* device disabled */ 452 goto exit_clear; 453 /* device disconnected */ 454 if (unlikely(atomic_read(&ep->chip->shutdown))) $ git blame sound/usb/endpoint.c -L435,454 Blaming lines: 1% (20/1631), done. 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 435) } 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 436) } 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 437) 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 438) /* 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 439) * complete callback for urbs 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 440) */ 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 441) static void snd_complete_urb(struct urb *urb) 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 442) { 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 443) struct snd_urb_ctx *ctx = urb->context; 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 444) struct snd_usb_endpoint *ep = ctx->ep; 67e225009bb15 (Takashi Iwai 2014-11-06 13:04:49 +0100 445) unsigned long flags; 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 446) int err; 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 447) 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 448) if (unlikely(urb->status == -ENOENT || /* unlinked */ 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 449) urb->status == -ENODEV || /* device removed */ 8fdff6a319e7d (Daniel Mack 2012-04-12 13:51:11 +0200 450) urb->status == -ECONNRESET || /* unlinked */ 47ab154593827 (Takashi Iwai 2015-08-25 16:09:00 +0200 451) urb->status == -ESHUTDOWN)) /* device disabled */ 47ab154593827 (Takashi Iwai 2015-08-25 16:09:00 +0200 452) goto exit_clear; 47ab154593827 (Takashi Iwai 2015-08-25 16:09:00 +0200 453) /* device disconnected */ 47ab154593827 (Takashi Iwai 2015-08-25 16:09:00 +0200 454) if (unlikely(atomic_read(&ep->chip->shutdown))) Full kernel log here: https://pastebin.com/dRrF8Q8B -- Best Regards, Mike Gavrilov.

3 5

[PATCH] soundwire: debugfs: use controller id instead of link_id
by Srinivas Kandagatla 06 Feb '21

06 Feb '21

link_id can be zero and if we have multiple controller instances in a system like Qualcomm debugfs will end-up with duplicate namespace resulting in incorrect debugfs entries. Using id should give a unique debugfs directory entry and should fix below warning too. "debugfs: Directory 'master-0' with parent 'soundwire' already present!" Fixes: bf03473d5bcc ("soundwire: add debugfs support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/debugfs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/debugfs.c b/drivers/soundwire/debugfs.c index b6cad0d59b7b..5f9efa42bb25 100644 --- a/drivers/soundwire/debugfs.c +++ b/drivers/soundwire/debugfs.c @@ -19,7 +19,7 @@ void sdw_bus_debugfs_init(struct sdw_bus *bus) return; /* create the debugfs master-N */ - snprintf(name, sizeof(name), "master-%d", bus->link_id); + snprintf(name, sizeof(name), "master-%d", bus->id); bus->debugfs = debugfs_create_dir(name, sdw_debugfs_root); } -- 2.21.0

3 14

Re: BUG: KASAN: use-after-free in snd_complete_urb+0x109e/0x1740 [snd_usb_audio] (5.11-rc6)
by Hillf Danton 06 Feb '21

06 Feb '21

On Sat, 6 Feb 2021 Takashi Iwai wrote: > > Due to the reconnecting key word mentioned, no fix to > > d0f09d1e4a88 ("ALSA: usb-audio: Refactoring endpoint URB deactivation") > > will be added. > > > > What is added is to capture EP_FLAG_STOPPING and remove the one > > second wait limit if the reconnecting acts may make it easier to > > repro the uaf. The diff is only for idea show. > > If my understanding is right, this won't change. The problem is > rather the lack of this function call itself, i.e. the missing > synchronization for the stream stop. Thanks for taking a look at it. > > It worked casually in the past because the endpoint resource is > released at a later point that is after all streams are really closed. > Now it's released earlier and hitting the UAF. And add it if I dont misread you. Hillf --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -832,24 +832,14 @@ void snd_usb_endpoint_suspend(struct snd */ static int wait_clear_urbs(struct snd_usb_endpoint *ep) { - unsigned long end_time = jiffies + msecs_to_jiffies(1000); - int alive; - - if (!test_bit(EP_FLAG_STOPPING, &ep->flags)) - return 0; - + WARN_ON_ONCE(!test_bit(EP_FLAG_STOPPING, &ep->flags)); do { - alive = bitmap_weight(&ep->active_mask, ep->nurbs); - if (!alive) + if (!bitmap_weight(&ep->active_mask, ep->nurbs)) break; schedule_timeout_uninterruptible(1); - } while (time_before(jiffies, end_time)); + } while (1); - if (alive) - usb_audio_err(ep->chip, - "timeout: still %d active urbs on EP #%x\n", - alive, ep->ep_num); clear_bit(EP_FLAG_STOPPING, &ep->flags); ep->sync_sink = NULL; @@ -1433,7 +1423,7 @@ void snd_usb_endpoint_stop(struct snd_us WRITE_ONCE(ep->sync_source->sync_sink, NULL); if (!atomic_dec_return(&ep->running)) - stop_and_unlink_urbs(ep, false, false); + stop_and_unlink_urbs(ep, false, true); } /**

2 1

[PATCH] ALSA: usb-audio: add mixer quirks for Pioneer DJM-900NXS2
by Fabian Lesniak 05 Feb '21

05 Feb '21

This commit adds mixer quirks for the Pioneer DJM-900NXS2 mixer. This device has 6 capture channels, 5 of them allow setting the signal source. This adds controls for these, similar to the DJM-250Mk2. However, playpack channels are not controllable via software like on the 250Mk2, as they can only be set manually on the mixing console. Read-only controls showing the currently selected playback channels are omitted. This patch is built on top of the DJM750 mixer quirk support and requires it to be applied first. Signed-off-by: Fabian Lesniak <fabian(a)lesniak-it.de> --- sound/usb/mixer_quirks.c | 73 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 72 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 10e472feb757..6688a38321a4 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2777,9 +2777,76 @@ static const struct snd_pioneer_djm_option_group snd_pioneer_djm750_option_group snd_pioneer_djm_option_group_item("CH4 Input", djm750_capture_ch4, 0), }; + +/* DJM-900NXS2 */ +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm900nxs2_capture_ch1[] = { + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone DITITAL", .wValue = 0x0102, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm900nxs2_capture_ch2[] = { + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone DITITAL", .wValue = 0x0202, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm900nxs2_capture_ch3[] = { + { .name = "CH3 Control Tone LINE", .wValue = 0x0300, .wIndex = 0x8002 }, + { .name = "CH3 Control Tone DITITAL", .wValue = 0x0302, .wIndex = 0x8002 }, + { .name = "CH3 Control Tone PHONO", .wValue = 0x0303, .wIndex = 0x8002 }, + { .name = "Post CH3 Fader", .wValue = 0x0306, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm900nxs2_capture_ch4[] = { + { .name = "CH4 Control Tone LINE", .wValue = 0x0400, .wIndex = 0x8002 }, + { .name = "CH4 Control Tone DITITAL", .wValue = 0x0402, .wIndex = 0x8002 }, + { .name = "CH4 Control Tone PHONO", .wValue = 0x0403, .wIndex = 0x8002 }, + { .name = "Post CH4 Fader", .wValue = 0x0406, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0407, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0408, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0409, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x040a, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm900nxs2_capture_ch5[] = { + { .name = "Cross Fader A", .wValue = 0x0507, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0508, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0509, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x050a, .wIndex = 0x8002 }, + { .name = "Post Fader A", .wValue = 0x050b, .wIndex = 0x8002 }, + { .name = "Post Fader B", .wValue = 0x050c, .wIndex = 0x8002 }, + { .name = "Post Fader C", .wValue = 0x050d, .wIndex = 0x8002 }, + { .name = "Post Fader D", .wValue = 0x050e, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option_group snd_pioneer_djm900nxs2_option_groups[] = { + snd_pioneer_djm_option_group_item("Capture Level", capture_level, 0), + snd_pioneer_djm_option_group_item("CH1 Input", djm900nxs2_capture_ch1, 2), + snd_pioneer_djm_option_group_item("CH2 Input", djm900nxs2_capture_ch2, 2), + snd_pioneer_djm_option_group_item("CH3 Input", djm900nxs2_capture_ch3, 2), + snd_pioneer_djm_option_group_item("CH4 Input", djm900nxs2_capture_ch4, 2), + snd_pioneer_djm_option_group_item("CH5 Input", djm900nxs2_capture_ch5, 3), +}; + static const struct snd_pioneer_djm_device snd_pioneer_djm_devices[] = { { .name = "DJM-250Mk2", .controls = snd_pioneer_djm250mk2_option_groups, .ncontrols = 7}, - { .name = "DJM-750", .controls = snd_pioneer_djm750_option_groups, .ncontrols = 5} + { .name = "DJM-750", .controls = snd_pioneer_djm750_option_groups, .ncontrols = 5}, + { .name = "DJM-900NXS2", .controls = snd_pioneer_djm900nxs2_option_groups, .ncontrols = 6} }; // layout of the kcontrol->private_value: @@ -3020,6 +3087,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */ err = snd_pioneer_djm_controls_create(mixer, 0x01); break; + case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + err = snd_pioneer_djm_controls_create(mixer, 0x02); + break; + } return err; -- 2.30.0

2 6

[PATCH] ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk
by Olivia Mackintosh 05 Feb '21

05 Feb '21

This allows for N different devices to use the pioneer mixer quirk for setting capture/record type and recording level. The impementation has not changed much with the exception of an additional mask on private_value to allow storing of a device index: DEVICE MASK 0xff000000 GROUP_MASK 0x00ff0000 VALUE_MASK 0x0000ffff There is perhaps room for removing the duplication in the lookup tables (name, wValue, wIndex) and deriving these. See the code header comment to understand how this can be achieved. Feedback is very much appreciated as I'm not the most proficient C programmer but am learning as I go. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/mixer_quirks.c | 195 +++++++++++++++++++++++++++++---------- 1 file changed, 148 insertions(+), 47 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index abad1d61a536..8518691509d9 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2603,28 +2603,68 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) } /* - * Pioneer DJ DJM-250MK2 and maybe other DJM models + * Pioneer DJ DJM Mixers * - * For playback, no duplicate mapping should be set. - * There are three mixer stereo channels (CH1, CH2, AUX) - * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6). - * Each channel should be mapped just once to one source. - * If mapped multiple times, only one source will play on given channel - * (sources are not mixed together). + * These devices generally have options for soft-switching the playback and + * capture sources in addition to the recording level. Although different + * devices have different configurations, there seems to be canonical values + * for specific capture/playback types: * - * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * Capture | Playback + * ================================|=========== + * Control Tone Line 0x00 | Ch1 0x00 + * Control Tone CD/Line 0x01 | Ch2 0x01 + * Control Tone Phono 0x03 | Aux 0x04 + * Post Fader 0x06 | + * Cross Fader A 0x07 | + * Cross Fader B 0x08 | + * Mic 0x09 | + * Aux 0x0d | + * Rec_Out 0x0a | + * None 0x0f | + * Ch1 Post Fader 0x11 | + * Ch2 Post Fader 0x12 | * - * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * The wValue for configuring playback/capture type also contains the channel + * the input type should apply to: + * CHANNEL NUMBER 0xff00 + * INPUT TYPE 0x00ff + * e.g. Channel 2 set to 'Control Tone Phono' will have wValue 0x0203 + * + * The wIndex values are below: + * CAPTURE 0x8002 + * CAPTURE LEVEL 0x8003 + * PLAYBACK 0x8016 * - * See also notes in the quirks-table.h file. */ +#define snd_pioneer_djm_option_group_item(_name, suffix, _default_value) { \ + .name = _name, \ + .options = snd_pioneer_djm_options_##suffix, \ + .count = ARRAY_SIZE(snd_pioneer_djm_options_##suffix), \ + .default_value = _default_value } + +struct snd_pioneer_djm_device { + char *name; + const struct snd_pioneer_djm_option_group *controls; + const size_t ncontrols; +}; + struct snd_pioneer_djm_option { const u16 wIndex; const u16 wValue; const char *name; }; +struct snd_pioneer_djm_option_group { + const char *name; + const struct snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +}; + +/* Common Options + */ static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_level[] = { { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, @@ -2632,6 +2672,8 @@ static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_level { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } }; +/* DJM250MK2 Options + */ static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch12[] = { { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, @@ -2682,20 +2724,8 @@ static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_56[] { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } }; -struct snd_pioneer_djm_option_group { - const char *name; - const struct snd_pioneer_djm_option *options; - const size_t count; - const u16 default_value; -}; - -#define snd_pioneer_djm_option_group_item(_name, suffix, _default_value) { \ - .name = _name, \ - .options = snd_pioneer_djm_options_##suffix, \ - .count = ARRAY_SIZE(snd_pioneer_djm_options_##suffix), \ - .default_value = _default_value } -static const struct snd_pioneer_djm_option_group snd_pioneer_djm_option_groups[] = { +static const struct snd_pioneer_djm_option_group snd_pioneer_djm250mk2_option_groups[] = { snd_pioneer_djm_option_group_item("Master Capture Level Capture Switch", capture_level, 0), snd_pioneer_djm_option_group_item("Capture 1-2 Capture Switch", capture_ch12, 2), snd_pioneer_djm_option_group_item("Capture 3-4 Capture Switch", capture_ch34, 2), @@ -2705,22 +2735,77 @@ static const struct snd_pioneer_djm_option_group snd_pioneer_djm_option_groups[] snd_pioneer_djm_option_group_item("Playback 5-6 Playback Switch", playback_56, 2) }; + +/* DJM750 */ +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm750_capture_ch1[] = { + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm750_capture_ch2[] = { + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone CDLINE",.wValue = 0x0201, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm750_capture_ch3[] = { + { .name = "CH3 Control Tone LINE", .wValue = 0x0300, .wIndex = 0x8002 }, + { .name = "CH3 Control Tone CDLINE",.wValue = 0x0301, .wIndex = 0x8002 }, + { .name = "Post CH3 Fader", .wValue = 0x0306, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_djm750_capture_ch4[] = { + { .name = "CH4 Control Tone LINE", .wValue = 0x0400, .wIndex = 0x8002 }, + { .name = "CH4 Control Tone PHONO", .wValue = 0x0403, .wIndex = 0x8002 }, + { .name = "Post CH4 Fader", .wValue = 0x0406, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0407, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0408, .wIndex = 0x8002 }, +}; + +static const struct snd_pioneer_djm_option_group snd_pioneer_djm750_option_groups[] = { + snd_pioneer_djm_option_group_item("Capture Level", capture_level, 0), + snd_pioneer_djm_option_group_item("CH1 Input", djm750_capture_ch1, 2), + snd_pioneer_djm_option_group_item("CH2 Input", djm750_capture_ch2, 2), + snd_pioneer_djm_option_group_item("CH3 Input", djm750_capture_ch3, 0), + snd_pioneer_djm_option_group_item("CH4 Input", djm750_capture_ch4, 0), +}; + +static const struct snd_pioneer_djm_device snd_pioneer_djm_devices[] = { + { .name = "DJM-250Mk2", .controls = snd_pioneer_djm250mk2_option_groups, .ncontrols = 7}, + { .name = "DJM-750", .controls = snd_pioneer_djm750_option_groups, .ncontrols = 5} +}; + // layout of the kcontrol->private_value: #define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff -#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_MASK 0x00ff0000 +#define SND_PIONEER_DJM_DEVICE_MASK 0xff000000 #define SND_PIONEER_DJM_GROUP_SHIFT 16 +#define SND_PIONEER_DJM_DEVICE_SHIFT 24 + static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info) { - u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + unsigned long private_value = kctl->private_value; + u8 device_idx = (private_value & SND_PIONEER_DJM_DEVICE_MASK) >> SND_PIONEER_DJM_DEVICE_SHIFT; + u8 group_idx = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + + const struct snd_pioneer_djm_device device = snd_pioneer_djm_devices[device_idx]; + size_t count; const char *name; const struct snd_pioneer_djm_option_group *group; - if (group_index >= ARRAY_SIZE(snd_pioneer_djm_option_groups)) + if (group_idx >= device.ncontrols) return -EINVAL; - group = &snd_pioneer_djm_option_groups[group_index]; + group = &device.controls[group_idx]; count = group->count; if (info->value.enumerated.item >= count) info->value.enumerated.item = count - 1; @@ -2732,12 +2817,13 @@ static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_c return 0; } -static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, + u8 device_idx, u8 group, u16 value) { int err; + const struct snd_pioneer_djm_device device = snd_pioneer_djm_devices[device_idx]; - if (group >= ARRAY_SIZE(snd_pioneer_djm_option_groups) - || value >= snd_pioneer_djm_option_groups[group].count) + if ((group >= device.ncontrols) || value >= device.controls[group].count) return -EINVAL; err = snd_usb_lock_shutdown(mixer->chip); @@ -2747,9 +2833,8 @@ static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u1 err = snd_usb_ctl_msg( mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), USB_REQ_SET_FEATURE, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, - snd_pioneer_djm_option_groups[group].options[value].wValue, - snd_pioneer_djm_option_groups[group].options[value].wIndex, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, device.controls[group].options[value].wValue, + device.controls[group].options[value].wIndex, NULL, 0); snd_usb_unlock_shutdown(mixer->chip); @@ -2767,27 +2852,34 @@ static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ct struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); struct usb_mixer_interface *mixer = list->mixer; unsigned long private_value = kctl->private_value; - u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + + u8 device = (private_value & SND_PIONEER_DJM_DEVICE_MASK) >> SND_PIONEER_DJM_DEVICE_SHIFT; + u8 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; u16 value = elem->value.enumerated.item[0]; - kctl->private_value = (group << SND_PIONEER_DJM_GROUP_SHIFT) | value; + kctl->private_value = ((device << SND_PIONEER_DJM_DEVICE_SHIFT) | + (group << SND_PIONEER_DJM_GROUP_SHIFT) | + value); - return snd_pioneer_djm_controls_update(mixer, group, value); + return snd_pioneer_djm_controls_update(mixer, device, group, value); } static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) { unsigned long private_value = list->kctl->private_value; - u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u8 device = (private_value & SND_PIONEER_DJM_DEVICE_MASK) >> SND_PIONEER_DJM_DEVICE_SHIFT; + u8 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK); - - return snd_pioneer_djm_controls_update(list->mixer, group, value); + return snd_pioneer_djm_controls_update(list->mixer, device, group, value); } -static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer, + const u8 device_idx) { int err, i; - const struct snd_pioneer_djm_option_group *group; + + const struct snd_pioneer_djm_device device = snd_pioneer_djm_devices[device_idx]; + struct snd_kcontrol_new knew = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -2797,11 +2889,17 @@ static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) .put = snd_pioneer_djm_controls_put }; - for (i = 0; i < ARRAY_SIZE(snd_pioneer_djm_option_groups); i++) { - group = &snd_pioneer_djm_option_groups[i]; - knew.name = group->name; - knew.private_value = (i << SND_PIONEER_DJM_GROUP_SHIFT) | group->default_value; - err = snd_pioneer_djm_controls_update(mixer, i, group->default_value); + u16 value; + for (i = 0; i < device.ncontrols; i++) { + value = device.controls[i].default_value; + + knew.name = device.controls[i].name; + knew.private_value = ( + (device_idx << SND_PIONEER_DJM_DEVICE_SHIFT) | + (i << SND_PIONEER_DJM_GROUP_SHIFT) | + value); + + err = snd_pioneer_djm_controls_update(mixer, device_idx, i, value); if (err) return err; err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, @@ -2917,7 +3015,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_bbfpro_controls_create(mixer); break; case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ - err = snd_pioneer_djm_controls_create(mixer); + err = snd_pioneer_djm_controls_create(mixer, 0x00); + break; + case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */ + err = snd_pioneer_djm_controls_create(mixer, 0x01); break; } -- 2.30.0

4 16

[PATCH v2] ASoC: qcom: lpass: Fix i2s ctl register bit map
by Jun Nie 05 Feb '21

05 Feb '21

Fix bitwidth mapping in i2s ctl register per APQ8016 document. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Stephan Gerhold <stephan(a)gerhold.net> Signed-off-by: Jun Nie <jun.nie(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 8507ef8f6679..3efa133d1c64 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -250,7 +250,7 @@ static struct lpass_variant apq8016_data = { .micmode = REG_FIELD_ID(0x1000, 4, 7, 4, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 4, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 4, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 4, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 4, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0x8400, 12, 12, 2, 0x1000), .rdma_bursten = REG_FIELD_ID(0x8400, 11, 11, 2, 0x1000), -- 2.25.1

2 1

[PATCH] ASoC: dt-bindings: renesas, rsnd: Group tuples in playback/capture properties
by Geert Uytterhoeven 05 Feb '21

05 Feb '21

To improve human readability and enable automatic validation, the tuples in "playback" and "capture" properties in sound device nodes should be grouped using angle brackets. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 0fd37aa84947aa89..2e1046513603ed01 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -404,7 +404,7 @@ examples: /* DAI base */ rcar_sound,dai { dai0 { - playback = <&ssi5 &src5>; + playback = <&ssi5>, <&src5>; capture = <&ssi6>; }; dai1 { @@ -430,8 +430,8 @@ examples: bitclock-master = <&rsnd_endpoint0>; frame-master = <&rsnd_endpoint0>; - playback = <&ssi0 &src0 &dvc0>; - capture = <&ssi1 &src1 &dvc1>; + playback = <&ssi0>, <&src0>, <&dvc0>; + capture = <&ssi1>, <&src1>, <&dvc1>; }; }; }; -- 2.25.1

2 1

[RESEND PATH] ASoC: dmaengine_pcm: add peripheral configuration
by Shengjiu Wang 05 Feb '21

05 Feb '21

The commit e7bbb7acabf4 ("dmaengine: add peripheral configuration") adds peripheral configuration for dma_slave_config. This configuration is useful for some audio peripherals, for example, the peripheral supports multi fifos, we can let the DMA know which fifos are selected. So also add this configuration for snd_dmaengine_dai_dma_data. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- include/sound/dmaengine_pcm.h | 5 +++++ sound/core/pcm_dmaengine.c | 2 ++ 2 files changed, 7 insertions(+) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 8c5e38180fb0..96666efddb39 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -66,6 +66,9 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now + * @peripheral_config: peripheral configuration for programming peripheral + * for dmaengine transfer + * @peripheral_size: peripheral configuration buffer size */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -76,6 +79,8 @@ struct snd_dmaengine_dai_dma_data { const char *chan_name; unsigned int fifo_size; unsigned int flags; + void *peripheral_config; + size_t peripheral_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 4d0e8fe535a1..1fc2fa077574 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -125,6 +125,8 @@ void snd_dmaengine_pcm_set_config_from_dai_data( } slave_config->slave_id = dma_data->slave_id; + slave_config->peripheral_config = dma_data->peripheral_config; + slave_config->peripheral_size = dma_data->peripheral_size; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); -- 2.27.0

2 1

[PATCH v2 0/3] Rename audio graph export functions
by Sameer Pujar 05 Feb '21

05 Feb '21

This series renames exported functions from audio graph for a better global visibility. In doing so update the references in audio graph and Tegra audio graph card drivers. Changelog ========= v1 -> v2: --------- [Patch v2 1/3]: - Rename graph_parse_of() to audio_graph_parse_of() and graph_card_probe() to audio_graph_card_probe() as well. - Update above references in audio graph driver. [Patch v2 2/3]: New patch - Update references for audio_graph_parse_of() and audio_graph_card_probe() in Tegra graph driver. - Add "Depends-on" tag [Patch v2 3/3]: - Update commit message to add "Depends-on" tag. Sameer Pujar (3): ASoC: audio-graph: Rename functions needed for export ASoC: tegra: Update references of audio graph helpers ASoC: tegra: Add driver remove() callback include/sound/graph_card.h | 6 ++++-- sound/soc/generic/audio-graph-card.c | 17 +++++++++-------- sound/soc/tegra/tegra_audio_graph_card.c | 9 +++++---- 3 files changed, 18 insertions(+), 14 deletions(-) -- 2.7.4

3 5

[PATCH] ASoC: mediatek: mt8183-mt6358: set playback and capture constraints
by Jiaxin Yu 05 Feb '21

05 Feb '21

Sets playback and capture constraints to S16_LE, stereo, 48kHz. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 41 ++++++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 3cf8aea..271413e 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -118,6 +118,45 @@ static int mt8183_rt1015_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, } static int +mt8183_mt6358_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 48000, + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const unsigned int channels[] = { + 2, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + runtime->hw.channels_max = 2; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + return 0; +} + +static const struct snd_soc_ops mt8183_mt6358_ops = { + .startup = mt8183_mt6358_startup, +}; + +static int mt8183_mt6358_ts3a227_max98357_bt_sco_startup( struct snd_pcm_substream *substream) { @@ -356,6 +395,7 @@ static void mt8183_mt6358_tdm_shutdown(struct snd_pcm_substream *substream) SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8183_mt6358_ops, SND_SOC_DAILINK_REG(playback1), }, { @@ -403,6 +443,7 @@ static void mt8183_mt6358_tdm_shutdown(struct snd_pcm_substream *substream) SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8183_mt6358_ops, SND_SOC_DAILINK_REG(capture3), }, { -- 1.8.1.1.dirty

2 1

[RESEND PATCH 0/2] ASoC: stm32: i2s: add master clock provider
by Olivier Moysan 05 Feb '21

05 Feb '21

Add master clock generation support in STM32 I2S driver. Resend of patch https://lkml.org/lkml/2020/9/11/264 Olivier Moysan (2): ASoC: dt-bindings: add mclk provider support to stm32 i2s ASoC: stm32: i2s: add master clock provider .../bindings/sound/st,stm32-i2s.yaml | 4 + sound/soc/stm/stm32_i2s.c | 310 +++++++++++++++--- 2 files changed, 270 insertions(+), 44 deletions(-) -- 2.17.1

2 3

[PATCH 0/4] ASoC: codecs: Realtek/SoundWire: fix remove/suspend issues
by Pierre-Louis Bossart 05 Feb '21

05 Feb '21

These issues and fixes were identified during our module load/unload and suspend/resume stress tests. Bard Liao (1): ASoC: rt5682: do nothing in rt5682_suspend/resume in sdw mode Pierre-Louis Bossart (3): ASoC: rt700-sdw: use cancel_work_sync() in .remove as well as .suspend ASoC: rt711-sdw: use cancel_work_sync() for .remove ASoC: rt5682-sdw: cancel_work_sync() in .remove and .suspend sound/soc/codecs/rt5682-sdw.c | 4 +++- sound/soc/codecs/rt5682.c | 6 ++++++ sound/soc/codecs/rt700-sdw.c | 4 ++-- sound/soc/codecs/rt711-sdw.c | 4 ++-- 4 files changed, 13 insertions(+), 5 deletions(-) -- 2.25.1

2 5

[PATCH] ALSA: usb-audio: Fix PCM buffer allocation in non-vmalloc mode
by Takashi Iwai 05 Feb '21

05 Feb '21

The commit f274baa49be6 ("ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers") introduced the mode to allocate coherent pages for PCM buffers, and it used bus->controller device as its DMA device. It turned out, however, that bus->sysdev is a more appropriate device to be used for DMA mapping in HCD code. This patch corrects the device reference accordingly. Note that, on most platforms, both point to the very same device, hence this patch doesn't change anything practically. But on platforms like xhcd-plat hcd, the change becomes effective. Fixes: f274baa49be6 ("ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 078bb4c94033..dcadf8f164b2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1558,7 +1558,7 @@ void snd_usb_preallocate_buffer(struct snd_usb_substream *subs) { struct snd_pcm *pcm = subs->stream->pcm; struct snd_pcm_substream *s = pcm->streams[subs->direction].substream; - struct device *dev = subs->dev->bus->controller; + struct device *dev = subs->dev->bus->sysdev; if (snd_usb_use_vmalloc) snd_pcm_set_managed_buffer(s, SNDRV_DMA_TYPE_VMALLOC, -- 2.26.2

1 0

[PATCH] coresight: etm4x: Fix merge resolution for amba rework
by Uwe Kleine-König 05 Feb '21

05 Feb '21

This was non-trivial to get right because commits c23bc382ef0e ("coresight: etm4x: Refactor probing routine") and 5214b563588e ("coresight: etm4x: Add support for sysreg only devices") changed the code flow considerably. With this change the driver can be built again. Fixes: 0573d3fa4864 ("Merge branch 'devel-stable' of git://git.armlinux.org.uk/~rmk/linux-arm into char-misc-next") Signed-off-by: Uwe Kleine-König <uwe(a)kleine-koenig.org> --- On Fri, Feb 05, 2021 at 12:07:09PM +0100, Greg Kroah-Hartman wrote: > On Fri, Feb 05, 2021 at 11:56:15AM +0100, Uwe Kleine-König wrote: > > I didn't compile test, but I'm willing to bet your resolution is wrong. > > You have no return statement in etm4_remove_dev() but its return type is > > int and etm4_remove_amba() still returns int but should return void. > > Can you send a patch to fix this up? Sure, here it comes. As I'm unsure if you want to squash it into the merge or want to keep it separate I crafted a commit message. If you prefer squashing feel free to do so. This change corresponds to the merge resolution I suggested before. Best regards Uwe drivers/hwtracing/coresight/coresight-etm4x-core.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/hwtracing/coresight/coresight-etm4x-core.c b/drivers/hwtracing/coresight/coresight-etm4x-core.c index bc55b261af23..c8ecd91e289e 100644 --- a/drivers/hwtracing/coresight/coresight-etm4x-core.c +++ b/drivers/hwtracing/coresight/coresight-etm4x-core.c @@ -1906,15 +1906,16 @@ static int __exit etm4_remove_dev(struct etmv4_drvdata *drvdata) cpus_read_unlock(); coresight_unregister(drvdata->csdev); + + return 0; } -static int __exit etm4_remove_amba(struct amba_device *adev) +static void __exit etm4_remove_amba(struct amba_device *adev) { struct etmv4_drvdata *drvdata = dev_get_drvdata(&adev->dev); if (drvdata) - return etm4_remove_dev(drvdata); - return 0; + etm4_remove_dev(drvdata); } static int __exit etm4_remove_platform_dev(struct platform_device *pdev) -- 2.29.2

2 1

[PATCH v3 0/5] amba: minor fix and various cleanups
by Uwe Kleine-König 05 Feb '21

05 Feb '21

From: Uwe Kleine-König <u.kleine-koenig.org(a)pengutronix.de Hello, Changes since v2 sent with Message-Id: 20201124133139.3072124-1-uwe(a)kleine-koenig.org: - Rebase to v5.11-rc1 (which resulted in a few conflicts in drivers/hwtracing). - Add various Acks. - Send to more maintainers directly (which I think is one of the reasons why there are so few Acks). For my taste patch 4 needs some more acks (drivers/char/hw_random, drivers/dma, drivers/gpu/drm/pl111, drivers/i2c, drivers/mmc, drivers/vfio, drivers/watchdog and sound/arm have no maintainer feedback yet). My suggestion is to let this series go in via Russell King (who cares for amba). Once enough Acks are there I can also provide a tag for merging into different trees. Just tell me if you prefer this solution. Would be great if this could make it for v5.12, but I'm aware it's already late in the v5.11 cycle so it might have to wait for v5.13. Best regards Uwe Uwe Kleine-König (5): amba: Fix resource leak for drivers without .remove amba: reorder functions vfio: platform: simplify device removal amba: Make the remove callback return void amba: Make use of bus_type functions drivers/amba/bus.c | 234 +++++++++--------- drivers/char/hw_random/nomadik-rng.c | 3 +- drivers/dma/pl330.c | 3 +- drivers/gpu/drm/pl111/pl111_drv.c | 4 +- drivers/hwtracing/coresight/coresight-catu.c | 3 +- .../hwtracing/coresight/coresight-cpu-debug.c | 4 +- .../hwtracing/coresight/coresight-cti-core.c | 4 +- drivers/hwtracing/coresight/coresight-etb10.c | 4 +- .../coresight/coresight-etm3x-core.c | 4 +- .../coresight/coresight-etm4x-core.c | 4 +- .../hwtracing/coresight/coresight-funnel.c | 4 +- .../coresight/coresight-replicator.c | 4 +- drivers/hwtracing/coresight/coresight-stm.c | 4 +- .../hwtracing/coresight/coresight-tmc-core.c | 4 +- drivers/hwtracing/coresight/coresight-tpiu.c | 4 +- drivers/i2c/busses/i2c-nomadik.c | 4 +- drivers/input/serio/ambakmi.c | 3 +- drivers/memory/pl172.c | 4 +- drivers/memory/pl353-smc.c | 4 +- drivers/mmc/host/mmci.c | 4 +- drivers/rtc/rtc-pl030.c | 4 +- drivers/rtc/rtc-pl031.c | 4 +- drivers/spi/spi-pl022.c | 5 +- drivers/tty/serial/amba-pl010.c | 4 +- drivers/tty/serial/amba-pl011.c | 3 +- drivers/vfio/platform/vfio_amba.c | 15 +- drivers/video/fbdev/amba-clcd.c | 4 +- drivers/watchdog/sp805_wdt.c | 4 +- include/linux/amba/bus.h | 2 +- sound/arm/aaci.c | 4 +- 30 files changed, 157 insertions(+), 198 deletions(-) base-commit: 5c8fe583cce542aa0b84adc939ce85293de36e5e -- 2.29.2

11 26

[PATCH] sound: x86: Trivial spell fixes throughout the file intel_hdmi_audio.c
by Bhaskar Chowdhury 05 Feb '21

05 Feb '21

s/confgiuration/configuration/ s/Calculte/Calculate/ s/unreference/unreferenced/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/x86/intel_hdmi_audio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 9f9fcd2749f2..89ddd9dd8fde 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -342,7 +342,7 @@ static int had_prog_status_reg(struct snd_pcm_substream *substream, /* * function to initialize audio - * registers and buffer confgiuration registers + * registers and buffer configuration registers * This function is called in the prepare callback */ static int had_init_audio_ctrl(struct snd_pcm_substream *substream, @@ -611,7 +611,7 @@ static void had_prog_dip(struct snd_pcm_substream *substream, frame2.regx.chnl_cnt = substream->runtime->channels - 1; frame3.regx.chnl_alloc = ca; - /* Calculte the byte wide checksum for all valid DIP words */ + /* Calculate the byte wide checksum for all valid DIP words */ for (i = 0; i < BYTES_PER_WORD; i++) checksum += (info_frame >> (i * 8)) & 0xff; for (i = 0; i < BYTES_PER_WORD; i++) @@ -1109,7 +1109,7 @@ static int had_pcm_close(struct snd_pcm_substream *substream) intelhaddata = snd_pcm_substream_chip(substream); - /* unreference and sync with the pending PCM accesses */ + /* unreferenced and sync with the pending PCM accesses */ spin_lock_irq(&intelhaddata->had_spinlock); intelhaddata->stream_info.substream = NULL; intelhaddata->stream_info.substream_refcount--; -- 2.30.0

2 2

[PATCH] ALSA: usb-audio: Correct document for snd_usb_endpoint_free_all()
by Takashi Iwai 05 Feb '21

05 Feb '21

The kerndoc comment for the new function snd_usb_endpoint_free_all() had a typo wrt the argument name. Fix it. Fixes: 00272c61827e ("ALSA: usb-audio: Avoid unnecessary interface re-setup") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8e568823c992..4d1c678a0d80 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1451,7 +1451,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep) /** * snd_usb_endpoint_free_all: Free the resources of an snd_usb_endpoint - * @card: The chip + * @chip: The chip * * This free all endpoints and those resources */ -- 2.26.2

1 0

[PATCH v5 0/5] ALSA: usb-audio: Fixes for Pioneer devices
by Takashi Iwai 05 Feb '21

05 Feb '21

Hi, here is the revised patches for addressing the regressions reported on the Pioneer USB-audio devices. Those have the special configuration where both playback and capture endpoints share the same interface with the vendor-spec descriptors. It required the extension of the endpoint management code and the setup of the implicit feedback mode. v4->v5: - Fix the implicit feedback detection - Correct the maxpacksize for ep_idx=1 case Takashi === Takashi Iwai (5): ALSA: usb-audio: Fix the missing endpoints creations for quirks ALSA: usb-audio: Choose audioformat of a counter-part substream ALSA: usb-audio: Avoid unnecessary interface re-setup ALSA: usb-audio: Annotate the endpoint index in audioformat ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices sound/usb/card.c | 5 +-- sound/usb/card.h | 3 ++ sound/usb/endpoint.c | 84 ++++++++++++++++++++++++++++++++++------ sound/usb/endpoint.h | 2 +- sound/usb/implicit.c | 57 ++++++++++++++++++++------- sound/usb/quirks-table.h | 6 +++ sound/usb/quirks.c | 58 +++++++++++++++++++++------ sound/usb/usbaudio.h | 1 + 8 files changed, 173 insertions(+), 43 deletions(-) -- 2.26.2

2 7

ucm2: sof-hda-dsp: DMICS may not be present
by GitHub issues - opened 05 Feb '21

05 Feb '21

alsa-project/alsa-ucm-conf issue #80 was opened from plbossart: UCM files assume that when SOF is used, the number of DMICs is between 1 and 4. That's an invalid assumption in the following two cases: a) the user plays with kernel parameters to select SOF on a platform without DMIC options snd_intel_dspcfg dsp_driver=3 b) the user plays with kernel parameters to work-around invalid dmic information, e.g. for UpExtreme the NHLT information reports 4 microphones but none are attached to the connector by default options snd_sof_intel_hda_common dmic_num=0 In those two cases, the only solution is to comment out the entire 'Mic1' section in ucm2/sof-hda-dsp/HiFi.conf. It'd be nicer to detect this case with the component information The machine driver currently only provides the cfg-dmics component string when the real or overridden number of dmics is > 0. The component string is as follows ```` amixer -Dhw:0 info Card hw:0 'sofhdadsp'/'AAEON-UP_WHL01-V1.0' Mixer name : 'Realtek ALC892' Components : 'HDA:8086280b,80860101,00100000 HDA:10ec0892,160d0892,00100302' Controls : 40 Simple ctrls : 15 ```` Reported by @worldgeek and confirmed on my platform. @perexg @kv2019i @libinyang FYI Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/80 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3 0/3] ASoC: mediatek: mt8192: apply some cleanup
by Tzung-Bi Shih 04 Feb '21

04 Feb '21

The 1st and 2nd patch refactor to use asoc_substream_to_rtd(). The 3rd patch simplifies ops of Capture1 DAI link. Changes from v2[1]: - Fix typo in 3rd patch's title, s/simply/simplify/. Changes from v1[2]: - Separated from v1. [1]: https://patchwork.kernel.org/project/alsa-devel/list/?series=420919 [2]: https://patchwork.kernel.org/project/alsa-devel/list/?series=419769 Tzung-Bi Shih (3): ASoC: mediatek: mt8192-mt6359: use asoc_substream_to_rtd() ASoC: mediatek: mt8192: use asoc_substream_to_rtd() ASoC: mediatek: mt8192-mt6359: simplify ops for Capture1 DAI link sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 4 ++-- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 24 ++++++++----------- 2 files changed, 12 insertions(+), 16 deletions(-) -- 2.30.0.365.g02bc693789-goog

2 4

[PATCH] drm: bridge: adv7511: Add set_jack handler
by Jun Nie 04 Feb '21

04 Feb '21

With commit 55c5cc63ab, the hdmi_codec_set_jack() will report unsupport failure if set_jack handler is missing. Add set_jack handler to resolve this failure. Signed-off-by: Jun Nie <jun.nie(a)linaro.org> --- .../gpu/drm/bridge/adv7511/adv7511_audio.c | 27 ++++++++++++++----- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c b/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c index f101dd2819b5..16de1a8ca7a0 100644 --- a/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c +++ b/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c @@ -217,22 +217,35 @@ static int adv7511_hdmi_i2s_get_dai_id(struct snd_soc_component *component, return -EINVAL; } +static int adv7511_hdmi_i2s_hook_plugged_cb(struct device *dev, void *data, + hdmi_codec_plugged_cb fn, + struct device *codec_dev) +{ + struct adv7511 *adv7511 = data; + bool plugged = adv7511->connector.status == connector_status_connected; + + fn(codec_dev, plugged); + return 0; +} + static const struct hdmi_codec_ops adv7511_codec_ops = { .hw_params = adv7511_hdmi_hw_params, .audio_shutdown = audio_shutdown, .audio_startup = audio_startup, .get_dai_id = adv7511_hdmi_i2s_get_dai_id, -}; - -static const struct hdmi_codec_pdata codec_data = { - .ops = &adv7511_codec_ops, - .max_i2s_channels = 2, - .i2s = 1, - .spdif = 1, + .hook_plugged_cb = adv7511_hdmi_i2s_hook_plugged_cb, }; int adv7511_audio_init(struct device *dev, struct adv7511 *adv7511) { + struct hdmi_codec_pdata codec_data = { + .ops = &adv7511_codec_ops, + .max_i2s_channels = 2, + .i2s = 1, + .spdif = 1, + .data = adv7511, + }; + adv7511->audio_pdev = platform_device_register_data(dev, HDMI_CODEC_DRV_NAME, PLATFORM_DEVID_AUTO, -- 2.25.1

2 2

[PATCH] ASoC: rt5682: Fix panic in rt5682_jack_detect_handler happening during system shutdown
by Ranjani Sridharan 04 Feb '21

04 Feb '21

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> During Coldboot stress tests, system encountered the following panic. Panic logs depicts rt5682_i2c_shutdown() happened first and then later jack detect handler workqueue function triggered. This situation causes panic as rt5682_i2c_shutdown() resets codec. Fix this panic by cancelling properly delayed work. Panic log: [ 20.936124] sof_pci_shutdown [ 20.940248] snd_sof_device_shutdown [ 20.945023] snd_sof_shutdown [ 21.126849] rt5682_i2c_shutdown [ 21.286053] rt5682_jack_detect_handler [ 21.291235] BUG: kernel NULL pointer dereference, address: 000000000000037c [ 21.299302] #PF: supervisor read access in kernel mode [ 21.305254] #PF: error_code(0x0000) - not-present page [ 21.311218] PGD 0 P4D 0 [ 21.314155] Oops: 0000 [#1] PREEMPT SMP NOPTI [ 21.319206] CPU: 2 PID: 123 Comm: kworker/2:3 Tainted: G U 5.4.68 #10 [ 21.333687] ACPI: Preparing to enter system sleep state S5 [ 21.337669] Workqueue: events_power_efficient rt5682_jack_detect_handler [snd_soc_rt5682] [ 21.337671] RIP: 0010:rt5682_jack_detect_handler+0x6c/0x279 [snd_soc_rt5682] Fixes: a50067d4f3c1d ('ASoC: rt5682: split i2c driver into separate module') Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/codecs/rt5682-i2c.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 37d13120f5ba..6effd2611e3c 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -273,6 +273,8 @@ static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); + cancel_delayed_work_sync(&rt5682->jack_detect_work); + rt5682_reset(rt5682); } -- 2.25.1

2 2

hdspmixer: Hardware output loopback toggle buttons
by GitHub pull_request - reopened 04 Feb '21

04 Feb '21

alsa-project/alsa-tools pull request #4 was reopened from pfandl: Hi, as promised in the [sound/pci/rme9652 patch](https://github.com/tiwai/sound/commit/da2a040ee7cfe1dd57d5bec7906cb9… here are the user land changes which provide the user the ability to toggle the hardware output loopback functionality per output channel. I tried to hide the buttons for devices which do not expose the snd controls yet, but due to probably unnecessary struggle i cancelled that. Please let me know though should you want them removed for not yet enabled devices. Request URL : https://github.com/alsa-project/alsa-tools/pull/4 Patch URL : https://github.com/alsa-project/alsa-tools/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[RFC PATCH] ALSA: hda: call ext hda codec link up/down if available
by Kai Vehmanen 04 Feb '21

04 Feb '21

The extended HDA bus (hdac_ext) provides interfaces for more fine-grained control of individual links than what plain HDA provides for. Links can be powered off when they are not used and if all links are released, controller can shut down the command DMA. These interfaces are currently not used by common HDA codec drivers. When a HDA codec is runtime suspended, it calls snd_hdac_codec_link_down(), but there is no link to the HDA extended bus, and links are shut down only when all codecs are suspended. To support more fine-grained control of link power, this patch implements new helper functions for codec drivers to turn codec links up and down. The HDA common suspend/resume code is modified to use the new functions. This allows to fully reuse the driver code both for plain and extended HDA controllers. For compatibility, code reverts to plain HDA implementation if HDA extended bus is disabled in the kernel build, or if the controller is a plain HDA one. Co-developed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/hdaudio_ext.h | 15 +++++++++++++++ sound/hda/ext/hdac_ext_controller.c | 28 ++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.c | 5 +++-- 3 files changed, 46 insertions(+), 2 deletions(-) NOTES: - RFC status due to new usage of hdaudio_ext.h - Not fully tested! Validation still ongoing... diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 7abf74c1c474..151922fa3c1e 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -131,6 +131,21 @@ void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link); int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *link); +#if IS_ENABLED(CONFIG_SND_HDA_EXT_CORE) +void snd_hdac_ext_codec_link_up(struct hdac_device *codec); +void snd_hdac_ext_codec_link_down(struct hdac_device *codec); +#else +static inline void snd_hdac_ext_codec_link_up(struct hdac_device *codec) +{ + snd_hdac_codec_link_up(codec); +} + +static inline void snd_hdac_ext_codec_link_down(struct hdac_device *codec) +{ + snd_hdac_codec_link_down(codec); +} +#endif + /* update register macro */ #define snd_hdac_updatel(addr, reg, mask, val) \ writel(((readl(addr + reg) & ~(mask)) | (val)), \ diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index b0c0ef824d7d..fd03b5d3a10e 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -332,3 +332,31 @@ int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, return ret; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put); + +void snd_hdac_ext_codec_link_up(struct hdac_device *codec) +{ + snd_hdac_codec_link_up(codec); + + if (codec->type == HDA_DEV_ASOC) { + const char *devname = dev_name(&codec->dev); + struct hdac_ext_link *hlink = + snd_hdac_ext_bus_get_link(codec->bus, devname); + if (hlink) + snd_hdac_ext_bus_link_get(codec->bus, hlink); + } +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_codec_link_up); + +void snd_hdac_ext_codec_link_down(struct hdac_device *codec) +{ + snd_hdac_codec_link_down(codec); + + if (codec->type == HDA_DEV_ASOC) { + const char *devname = dev_name(&codec->dev); + struct hdac_ext_link *hlink = + snd_hdac_ext_bus_get_link(codec->bus, devname); + if (hlink) + snd_hdac_ext_bus_link_put(codec->bus, hlink); + } +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_codec_link_down); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9b755062d841..6113f787b494 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -14,6 +14,7 @@ #include <linux/pm_runtime.h> #include <sound/core.h> #include <sound/hda_codec.h> +#include <sound/hdaudio_ext.h> #include <sound/asoundef.h> #include <sound/tlv.h> #include <sound/initval.h> @@ -2948,7 +2949,7 @@ static int hda_codec_runtime_suspend(struct device *dev) if (codec->link_down_at_suspend || (codec_has_clkstop(codec) && codec_has_epss(codec) && (state & AC_PWRST_CLK_STOP_OK))) - snd_hdac_codec_link_down(&codec->core); + snd_hdac_ext_codec_link_down(&codec->core); codec_display_power(codec, false); return 0; } @@ -2962,7 +2963,7 @@ static int hda_codec_runtime_resume(struct device *dev) return 0; codec_display_power(codec, true); - snd_hdac_codec_link_up(&codec->core); + snd_hdac_ext_codec_link_up(&codec->core); hda_call_codec_resume(codec); pm_runtime_mark_last_busy(dev); return 0; base-commit: 307cfa506ef4e6d7fe9430e513db2353925fb440 -- 2.29.2

2 4

[PATCH v4 00/13] MFD/extcon/ASoC: Rework arizona codec jack-detect support
by Hans de Goede 04 Feb '21

04 Feb '21

Hi all, Here is v4 of my series to rework the arizona codec jack-detect support to use the snd_soc_jack helpers instead of direct extcon reporting. This is done by reworking the extcon driver into an arizona-jackdet library and then modifying the codec drivers to use that directly, replacing the old separate extcon child-devices and extcon-driver. This brings the arizona-codec jack-detect handling inline with how all other ASoC codec driver do this. This was developed and tested on a Lenovo Yoga Tablet 1051L with a WM5102 codec. There are various interdependencies between the patches in this series, so IMHO it would be best if this entire series would be merged through the MFD tree. Note this series applies on top of my "[PATCH v4 0/5] MFD/ASoC: Add support for Intel Bay Trail boards with WM5102 codec" series. Changes in v4: - Add sound/soc/codecs/arizona-jack.c to the WOLFSON MICROELECTRONICS DRIVERS MAINTAINERS section - Small codying style tweaks to the "ASoC: arizona-jack: Cleanup logging" patch Changes in v3: - Move the bugfix patches to earlier in the series so that they apply to drivers/extcon/extcon-arizona.c so that they can be cherry-picked into the stable series - Split runtime_pm_get -> runtime_pm_get_sync changes out into their own patch - Simply move drivers/extcon/extcon-arizona.c to sound/soc/codecs/arizona-jack.c instead of first adding arizona-jack.c as a copy and then later removing extcon-arizona.c - Some other small tweaks, see individual patch changelogs Regards, Hans Hans de Goede (13): mfd: arizona: Drop arizona-extcon cells extcon: arizona: Fix some issues when HPDET IRQ fires after the jack has been unplugged extcon: arizona: Fix various races on driver unbind extcon: arizona: Fix flags parameter to the gpiod_get("wlf,micd-pol") call extcon: arizona: Always use pm_runtime_get_sync() when we need the device to be awake ASoC/extcon: arizona: Move arizona jack code to sound/soc/codecs/arizona-jack.c ASoC: arizona-jack: Move jack-detect variables to struct arizona_priv ASoC: arizona-jack: Use arizona->dev for runtime-pm ASoC: arizona-jack: convert into a helper library for codec drivers ASoC: arizona-jack: Use snd_soc_jack to report jack events ASoC: arizona-jack: Cleanup logging ASoC: arizona: Make the wm5102, wm5110, wm8997 and wm8998 drivers use the new jack library ASoC: Intel: bytcr_wm5102: Add jack detect support MAINTAINERS | 3 +- drivers/extcon/Kconfig | 8 - drivers/extcon/Makefile | 1 - drivers/mfd/arizona-core.c | 20 - sound/soc/codecs/Makefile | 2 +- .../soc/codecs/arizona-jack.c | 577 +++++++----------- sound/soc/codecs/arizona.h | 44 ++ sound/soc/codecs/wm5102.c | 12 +- sound/soc/codecs/wm5110.c | 12 +- sound/soc/codecs/wm8997.c | 14 +- sound/soc/codecs/wm8998.c | 9 + sound/soc/intel/boards/bytcr_wm5102.c | 28 +- 12 files changed, 325 insertions(+), 405 deletions(-) rename drivers/extcon/extcon-arizona.c => sound/soc/codecs/arizona-jack.c (76%) -- 2.29.2

4 32

[PATCH] media: atomisp: Remove unneeded return variable
by Yang Li 04 Feb '21

04 Feb '21

This patch removes unneeded return variables, using only '0' instead. It fixes the following warning detected by coccinelle: ./drivers/staging/media/atomisp/pci/sh_css_mipi.c:39:5-8: Unneeded variable: "err". Return "0" on line 44 Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- drivers/staging/media/atomisp/pci/sh_css_mipi.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/drivers/staging/media/atomisp/pci/sh_css_mipi.c b/drivers/staging/media/atomisp/pci/sh_css_mipi.c index d5ae7f0..de8ee45 100644 --- a/drivers/staging/media/atomisp/pci/sh_css_mipi.c +++ b/drivers/staging/media/atomisp/pci/sh_css_mipi.c @@ -36,12 +36,10 @@ int ia_css_mipi_frame_specify(const unsigned int size_mem_words, const bool contiguous) { - int err = 0; - my_css.size_mem_words = size_mem_words; (void)contiguous; - return err; + return 0; } /* -- 1.8.3.1

1 0

[PATCH] ALSA: usb-audio: endpoint : remove redundant check before usb_free_coherent()
by Xu Wang 04 Feb '21

04 Feb '21

usb_free_coherent() is safe with NULL addr and this check is not required. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/usb/endpoint.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8e568823c992..d5ed4ddfd451 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -82,10 +82,9 @@ static inline unsigned get_usb_high_speed_rate(unsigned int rate) */ static void release_urb_ctx(struct snd_urb_ctx *u) { - if (u->buffer_size) - usb_free_coherent(u->ep->chip->dev, u->buffer_size, - u->urb->transfer_buffer, - u->urb->transfer_dma); + usb_free_coherent(u->ep->chip->dev, u->buffer_size, + u->urb->transfer_buffer, + u->urb->transfer_dma); usb_free_urb(u->urb); u->urb = NULL; } -- 2.17.1

2 1

[PATCH v3] arm64: defconfig: Enable RT5659
by Sameer Pujar 04 Feb '21

04 Feb '21

Enable the RT5659 audio codec driver. Jetson AGX Xavier has RT5658 codec which is compatible with this driver. This enables user to test external audio. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Bard Liao <bardliao(a)realtek.com> Acked-by: Jon Hunter <jonathanh(a)nvidia.com> --- This was part of original series, https://lore.kernel.org/alsa-devel/1611944866-29373-1-git-send-email-spujar… Since other patches from above series are already applied, sending this separately now. Also merging emails/groups from the resend version, https://lore.kernel.org/linux-arm-kernel/1612260252-31582-1-git-send-email-… Changelog ========= v2 -> v3: - Updated commit message to reflect usage of Jetson AGX Xavier - Added 'Acked-by' tag from Jon arch/arm64/configs/defconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/arch/arm64/configs/defconfig b/arch/arm64/configs/defconfig index d628125..496fcee 100644 --- a/arch/arm64/configs/defconfig +++ b/arch/arm64/configs/defconfig @@ -767,6 +767,7 @@ CONFIG_SND_SOC_ES7134=m CONFIG_SND_SOC_ES7241=m CONFIG_SND_SOC_GTM601=m CONFIG_SND_SOC_PCM3168A_I2C=m +CONFIG_SND_SOC_RT5659=m CONFIG_SND_SOC_SIMPLE_AMPLIFIER=m CONFIG_SND_SOC_TAS571X=m CONFIG_SND_SOC_WCD934X=m -- 2.7.4

1 0

[PATCH 0/2] Add remove path for Tegra audio graph card
by Sameer Pujar 04 Feb '21

04 Feb '21

Audio graph driver cleans up phandle references of CPU/CODEC nodes in graph_remove() path. As Tegra graph card driver is based on this populate remove callback for it. Sameer Pujar (2): ASoC: audio-graph: Export graph_remove() function ASoC: tegra: Add driver remove() callback include/sound/graph_card.h | 2 ++ sound/soc/generic/audio-graph-card.c | 3 ++- sound/soc/tegra/tegra_audio_graph_card.c | 3 ++- 3 files changed, 6 insertions(+), 2 deletions(-) -- 2.7.4

3 9

[PATCH v1 0/2] ASoC: tegra: Add RT5631 machine driver
by Ion Agorria 03 Feb '21

03 Feb '21

From: Ion Agorria <ion(a)agorria.com> Adds support for Tegra SoC devices with RT5631 sound codec. Playback to speakers, headphones and internal mic recording works. This driver is used for ASUS Transformer TF201, TF700T and others Tegra based devices containing RT5631. Svyatoslav Ryhel (2): ASoC: dt-bindings: tegra: Add binding for RT5631 ASoC: tegra: Add RT5631 machine driver .../sound/nvidia,tegra-audio-rt5631.yaml | 134 +++++++++ sound/soc/tegra/Kconfig | 8 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5631.c | 261 ++++++++++++++++++ 4 files changed, 405 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml create mode 100644 sound/soc/tegra/tegra_rt5631.c -- 2.26.2

4 11

[PATCH 0/5] ASoC: soc-pcm: cleanup soc_new_pcm() and bugfix
by Kuninori Morimoto 03 Feb '21

03 Feb '21

Hi Mark These are soc-pcm cleanup patchset. 1) - 3) : cleanup soc_new_pcm() function 4) : cleanup dpcm_runtime_merge_xxx() function 5) : bugfix of snd_pcm_limit_hw_rates() order Kuninori Morimoto (5): 1) ASoC: soc-pcm: tidyup pcm setting 2) ASoC: soc-pcm: add soc_get_playback_capture() and simplify soc_new_pcm() 3) ASoC: soc-pcm: add soc_create_pcm() and simplify soc_new_pcm() 4) ASoC: soc-pcm: use snd_pcm_hardware at dpcm_runtime_merge_xxx() 5) ASoC: soc-pcm: fixup snd_pcm_limit_hw_rates() timing sound/soc/soc-pcm.c | 124 +++++++++++++++++++++++++++----------------- 1 file changed, 75 insertions(+), 49 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: Intel: catpt: remove unneeded semicolon
by Yang Li 03 Feb '21

03 Feb '21

Eliminate the following coccicheck warning: ./sound/soc/intel/catpt/pcm.c:355:2-3: Unneeded semicolon Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/intel/catpt/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index e5d54bb..88a0879 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -352,7 +352,7 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, break; default: return 0; - }; + } list_for_each_entry(pos, &component->card->snd_card->controls, list) { if (pos->private_data == component && -- 1.8.3.1

3 3

ucm2: Separate the configuration lookups (hw based)
by GitHub pull_request - opened 03 Feb '21

03 Feb '21

alsa-project/alsa-ucm-conf pull request #79 was opened from perexg: Introduce ucm2/conf.d/ tree with symlinks to the real hardware configurations. In this way, we do not rely to create the configuration paths based on simple driver / device identification, but we can store the configurations more logically to make the maintenance (code reuse, multiple changes) more easy. BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/70 BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/76 BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/78 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/79 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/79.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Extend use case search paths to use kernel driver name
by GitHub pull_request - opened 03 Feb '21

03 Feb '21

alsa-project/alsa-ucm-conf pull request #78 was opened from lumag: This is a companion (or replacement) for #76 . This adds lookup for the use case search path using the in-kernel driver name. This allows easy lookup both for the setups which have the sound card/platform driver built as a module or builtin into the kernel. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/78 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/78.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [PATCH 0/2] Add mediatek MT6359 accdet driver
by Argus Lin 03 Feb '21

03 Feb '21

Dear maintainers: Can you reply to me if any opinion? thanks B.R. Argus On Wed, 2021-01-06 at 20:19 +0800, Argus Lin wrote: > MT6359 audio codec support accessory detect features, the general features > are jack plug detection and key detection. > > All of 3-pole and 4-pole jack are supported. > > change since v1: > - adds mt6359 accdet binding document > - adds mt6359 accdet driver > > > Argus Lin (2): > dt-bindings: mediatek: mt6359: add ASoC mt6359 accdet document > ASoC: mediatek: mt6359: add MT6359 accdet driver > > .../devicetree/bindings/sound/mt6359-accdet.yaml | 142 ++ > sound/soc/codecs/Kconfig | 7 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/mt6359-accdet.c | 1951 ++++++++++++++++++++ > sound/soc/codecs/mt6359-accdet.h | 136 ++ > sound/soc/codecs/mt6359.h | 1863 +++++++++++++++++-- > 6 files changed, 3995 insertions(+), 106 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/mt6359-accdet.yaml > create mode 100644 sound/soc/codecs/mt6359-accdet.c > create mode 100644 sound/soc/codecs/mt6359-accdet.h > > -- > 1.8.1.1.dirty >

2 1

question about ALSA devices not showing up in buildroot build
by Bert Schiettecatte 03 Feb '21

03 Feb '21

Hi all I am working on getting my ALSA machine driver and codecs working. I ported over my code which was based on 4.11, and I'm using 5.9.12 at the moment. At first sight everything seems to be working, but the hw: , plughw: etc devices are missing and when I list the PCMs with acrecord -L, I don't see any, even though arecord -l does show the card. I am using buildroot for my rootfs. So I am wondering if this is an issue with incorrectly defining the structs in my driver code, or if this is an issue with missing package in my buildroot config, or if this is due to a missing configuration file for ALSA in the rootfs. Previously I was using an off the shelf linaro rootfs when I was still on kernel version 4.11, and there the PCMs were showing up correctly and I could see the multiple devices such as hw:, plughw:, dmix: etc. Thanks! Bert

3 5

sof-soundwire: use the ${find-device} lookup for the dmic device
by GitHub pull_request - opened 03 Feb '21

03 Feb '21

alsa-project/alsa-ucm-conf pull request #77 was opened from perexg: The PCM device number for the internal digital microphone is not fixed. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/77 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/77.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] soundwire: bus: Make sdw_nwrite() data pointer argument const
by Richard Fitzgerald 03 Feb '21

03 Feb '21

Idiomatically, write functions should take const pointers to the data buffer, as they don't change the data. They are also likely to be called from functions that receive a const data pointer. Internally the pointer is passed to function/structs shared with the read functions, requiring a cast, but this is an implementation detail that should be hidden by the public API. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 6 +++--- include/linux/soundwire/sdw.h | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index d9deafdcf495..4c76358416c5 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -394,13 +394,13 @@ sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) } static int -sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { struct sdw_msg msg; int ret; ret = sdw_fill_msg(&msg, slave, addr, count, - slave->dev_num, SDW_MSG_FLAG_WRITE, val); + slave->dev_num, SDW_MSG_FLAG_WRITE, (u8 *)val); if (ret < 0) return ret; @@ -524,7 +524,7 @@ EXPORT_SYMBOL(sdw_nread); * @count: length * @val: Buffer for values to be read */ -int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index f0b01b728640..18f5979d9a60 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1006,6 +1006,6 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus); int sdw_read(struct sdw_slave *slave, u32 addr); int sdw_write(struct sdw_slave *slave, u32 addr, u8 value); int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); -int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); +int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); #endif /* __SOUNDWIRE_H */ -- 2.20.1

1 0

[PATCH 0/3] soundwire: clear bus clash/parity interrupt before the mask is enabled
by Bard Liao 03 Feb '21

03 Feb '21

The SoundWire specification allows a Slave device to report a bus clash or parity error with the in-band interrupt mechanism. Unfortunately, on some platforms, these errors are randomly reported and don't seem to be valid. This series suggests the addition of a Master level quirk to discard such interrupts. The quirk should in theory have been added at the Slave level, but since the problem was detected with different generations of Slave devices it's hard to point to a specific IP. The problem might also be board-dependent and hence dealing with a Master quirk is simpler. Bard Liao (2): soundwire: bus: clear bus clash interrupt before the mask is enabled soundwire: intel: add SDW_MASTER_QUIRKS_CLEAR_INITIAL_CLASH quirk Pierre-Louis Bossart (1): soundwire: bus: clear parity interrupt before the mask is enabled drivers/soundwire/bus.c | 19 +++++++++++++++++++ drivers/soundwire/intel.c | 3 +++ include/linux/soundwire/sdw.h | 5 +++++ 3 files changed, 27 insertions(+) -- 2.17.1

3 16

Re: [PATCH 2/2] ASoC: mediatek: mt6359: add MT6359 accdet driver
by Argus Lin 03 Feb '21

03 Feb '21

Dear maintainers: Can you reply to me if any opinion? thanks B.R. Argus On Wed, 2021-01-06 at 20:19 +0800, Argus Lin wrote: > MT6359 audio codec support accessory detect features, adds MT6359 > accdet driver to support plug detection and key detection. > > Signed-off-by: Argus Lin <argus.lin(a)mediatek.com> > --- > sound/soc/codecs/Kconfig | 7 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/mt6359-accdet.c | 1951 ++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/mt6359-accdet.h | 136 +++ > sound/soc/codecs/mt6359.h | 1863 +++++++++++++++++++++++++++++++++--- > 5 files changed, 3853 insertions(+), 106 deletions(-) > create mode 100644 sound/soc/codecs/mt6359-accdet.c > create mode 100644 sound/soc/codecs/mt6359-accdet.h > > diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig > index ba4eb54..dbf3141 100644 > --- a/sound/soc/codecs/Kconfig > +++ b/sound/soc/codecs/Kconfig > @@ -1774,6 +1774,13 @@ config SND_SOC_MT6359 > help > Enable support for the platform which uses MT6359 as > external codec device. > +config SND_SOC_MT6359_ACCDET > + tristate "MediaTek MT6359 ACCDET driver" > + depends on MTK_PMIC_WRAP > + help > + ACCDET means Accessory Detection technology, MediaTek develop it > + for ASoC codec soc-jack detection mechanism. > + Select N if you don't have jack on board. > > config SND_SOC_MT6660 > tristate "Mediatek MT6660 Speaker Amplifier" > diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile > index d277f03..f77d057 100644 > --- a/sound/soc/codecs/Makefile > +++ b/sound/soc/codecs/Makefile > @@ -133,6 +133,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o > snd-soc-mt6351-objs := mt6351.o > snd-soc-mt6358-objs := mt6358.o > snd-soc-mt6359-objs := mt6359.o > +snd-soc-mt6359-accdet-objs := mt6359-accdet.o > snd-soc-mt6660-objs := mt6660.o > snd-soc-nau8315-objs := nau8315.o > snd-soc-nau8540-objs := nau8540.o > @@ -448,6 +449,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o > obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o > obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o > obj-$(CONFIG_SND_SOC_MT6359) += snd-soc-mt6359.o > +obj-$(CONFIG_SND_SOC_MT6359_ACCDET) += mt6359-accdet.o > obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o > obj-$(CONFIG_SND_SOC_NAU8315) += snd-soc-nau8315.o > obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o > diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c > new file mode 100644 > index 0000000..ff90e4e > --- /dev/null > +++ b/sound/soc/codecs/mt6359-accdet.c > @@ -0,0 +1,1951 @@ > +// SPDX-License-Identifier: GPL-2.0 > +/* > + * Copyright (C) 2021 MediaTek Inc. > + * Author: Argus Lin <argus.lin(a)mediatek.com> > + */ > + > +#include <linux/of_gpio.h> > +#include <linux/of.h> > +#include <linux/of_irq.h> > +#include <linux/of_device.h> > +#include <linux/of_address.h> > +#include <linux/input.h> > +#include <linux/kthread.h> > +#include <linux/io.h> > +#include <linux/sched/clock.h> > +#include <linux/workqueue.h> > +#include <linux/timer.h> > +#include <linux/delay.h> > +#include <linux/module.h> > +#include <linux/platform_device.h> > +#include <linux/init.h> > +#include <linux/irqdomain.h> > +#include <linux/irq.h> > +#include <linux/regmap.h> > +#include <sound/soc.h> > +#include <sound/jack.h> > +#include <linux/mfd/mt6397/core.h> > +#include "mt6359-accdet.h" > +#include "mt6359.h" > +/* grobal variable definitions */ > +#define REGISTER_VAL(x) ((x) - 1) > +#define HAS_CAP(_c, _x) \ > + ({typeof(_c)c = (_c); \ > + typeof(_x)x = (_x); \ > + (((c) & (x)) == (x)); }) > +#define ACCDET_PMIC_EINT_IRQ BIT(0) > +#define ACCDET_AP_GPIO_EINT BIT(1) > + > +#define ACCDET_PMIC_EINT0 BIT(2) > +#define ACCDET_PMIC_EINT1 BIT(3) > +#define ACCDET_PMIC_BI_EINT BIT(4) > + > +#define ACCDET_PMIC_GPIO_TRIG_EINT BIT(5) > +#define ACCDET_PMIC_INVERTER_TRIG_EINT BIT(6) > +#define ACCDET_PMIC_RSV_EINT BIT(7) > + > +#define ACCDET_THREE_KEY BIT(8) > +#define ACCDET_FOUR_KEY BIT(9) > +#define ACCDET_TRI_KEY_CDD BIT(10) > +#define ACCDET_RSV_KEY BIT(11) > + > +#define ACCDET_ANALOG_FASTDISCHARGE BIT(12) > +#define ACCDET_DIGITAL_FASTDISCHARGE BIT(13) > +#define ACCDET_AD_FASTDISCHRAGE BIT(14) > + > +#define RET_LT_5K (-1) > +#define RET_GT_5K (0) > + > +/* Used to let accdet know if the pin has been fully plugged-in */ > +#define EINT_PLUG_OUT (0) > +#define EINT_PLUG_IN (1) > + > +struct mt63xx_accdet_data { > + struct snd_soc_jack jack; > + struct platform_device *pdev; > + struct device *dev; > + struct accdet_priv *data; > + atomic_t init_once; > + struct regmap *regmap; > + int accdet_irq; > + int accdet_eint0; > + int accdet_eint1; > + struct wakeup_source *wake_lock; > + struct wakeup_source *timer_lock; > + struct mutex res_lock; /* lock to protect flag */ > + bool eint_sync_flag; > + /* accdet FSM State & lock*/ > + unsigned int cur_eint_state; > + unsigned int eint0_state; > + unsigned int eint1_state; > + unsigned int accdet_status; > + unsigned int cable_type; > + unsigned int cur_key; > + unsigned int cali_voltage; > + int auxadc_offset; > + unsigned int eint_id; > + /* when caps include ACCDET_AP_GPIO_EINT */ > + struct pinctrl *pinctrl; > + struct pinctrl_state *pins_eint; > + unsigned int gpiopin; > + unsigned int gpio_hp_deb; > + unsigned int gpioirq; > + unsigned int accdet_eint_type; > + /* when MICBIAS_DISABLE_TIMER timeout, queue work: dis_micbias_work */ > + struct work_struct delay_init_work; > + struct workqueue_struct *delay_init_workqueue; > + struct work_struct dis_micbias_work; > + struct workqueue_struct *dis_micbias_workqueue; > + struct work_struct accdet_work; > + struct workqueue_struct *accdet_workqueue; > + /* when eint issued, queue work: eint_work */ > + struct work_struct eint_work; > + struct workqueue_struct *eint_workqueue; > +}; > + > +static struct mt63xx_accdet_data *accdet; > + > +static struct head_dts_data accdet_dts; > +struct pwm_deb_settings *cust_pwm_deb; > + > +struct accdet_priv { > + unsigned int caps; > +}; > + > +static struct accdet_priv mt6359_accdet[] = { > + { > + .caps = 0, > + }, > +}; > + > +const struct of_device_id mt6359_accdet_of_match[] = { > + { > + .compatible = "mediatek,mt6359-accdet", > + .data = &mt6359_accdet, > + }, { > + /* sentinel */ > + }, > +}; > + > +static struct platform_driver mt6359_accdet_driver; > +static const struct snd_soc_component_driver mt6359_accdet_soc_driver; > + > +#define ACCDET_INIT_WAIT_TIMER (10 * HZ) > +static struct timer_list accdet_init_timer; > +static void delay_init_timerhandler(struct timer_list *t); > +/* micbias_timer: disable micbias if no accdet irq after eint, > + * timeout: 6 seconds > + * timerHandler: dis_micbias_timerhandler() > + */ > +#define MICBIAS_DISABLE_TIMER (6 * HZ) > +static struct timer_list micbias_timer; > +static void dis_micbias_timerhandler(struct timer_list *t); > + > +static unsigned int button_press_debounce = 0x400; > +static unsigned int button_press_debounce_01 = 0x800; > + > +/* local function declaration */ > +static void accdet_init_once(void); > +static inline void accdet_init(void); > +static void accdet_init_debounce(void); > +static unsigned int adjust_eint_analog_setting(void); > +static unsigned int adjust_eint_setting(unsigned int eintsts); > +static void config_digital_init_by_mode(void); > +static void config_eint_init_by_mode(void); > +static unsigned int get_triggered_eint(void); > +static void recover_eint_analog_setting(void); > +static void recover_eint_digital_setting(void); > +static void recover_eint_setting(unsigned int eintsts); > +static void send_status_event(unsigned int cable_type, unsigned int status); > +static void accdet_set_debounce(int state, unsigned int debounce); > + > +/* global function declaration */ > + > +static u64 mt6359_accdet_get_current_time(void) > +{ > + return sched_clock(); > +} > + > +static bool mt6359_accdet_timeout_ns(u64 start_time_ns, u64 timeout_time_ns) > +{ > + u64 cur_time = 0; > + u64 elapse_time = 0; > + > + /* get current tick, ns */ > + cur_time = mt6359_accdet_get_current_time(); > + if (cur_time < start_time_ns) { > + start_time_ns = cur_time; > + /* 400us */ > + timeout_time_ns = 400 * 1000; > + } > + elapse_time = cur_time - start_time_ns; > + > + /* check if timeout */ > + if (timeout_time_ns <= elapse_time) > + return false; > + > + return true; > +} > + > +static unsigned int check_key(unsigned int v) > +{ > + if (HAS_CAP(accdet->data->caps, ACCDET_FOUR_KEY)) { > + if (v < accdet_dts.four_key.down && v >= accdet_dts.four_key.up) > + return DW_KEY; > + if (v < accdet_dts.four_key.up && v >= accdet_dts.four_key.voice) > + return UP_KEY; > + if (v < accdet_dts.four_key.voice && v >= accdet_dts.four_key.mid) > + return AS_KEY; > + if (v < accdet_dts.four_key.mid) > + return MD_KEY; > + } else { > + if (v < accdet_dts.three_key.down && v >= accdet_dts.three_key.up) > + return DW_KEY; > + if (v < accdet_dts.three_key.up && v >= accdet_dts.three_key.mid) > + return UP_KEY; > + if (v < accdet_dts.three_key.mid) > + return MD_KEY; > + } > + return NO_KEY; > +} > + > +static void send_key_event(unsigned int keycode, unsigned int flag) > +{ > + int report = 0; > + > + switch (keycode) { > + case DW_KEY: > + if (flag != 0) > + report = SND_JACK_BTN_1; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_BTN_1); > + break; > + case UP_KEY: > + if (flag != 0) > + report = SND_JACK_BTN_2; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_BTN_2); > + break; > + case MD_KEY: > + if (flag != 0) > + report = SND_JACK_BTN_0; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_BTN_0); > + break; > + case AS_KEY: > + if (flag != 0) > + report = SND_JACK_BTN_3; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_BTN_3); > + break; > + } > +} > + > +static void send_status_event(unsigned int cable_type, unsigned int status) > +{ > + int report = 0; > + > + switch (cable_type) { > + case HEADSET_NO_MIC: > + if (status) > + report = SND_JACK_HEADPHONE; > + else > + report = 0; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_HEADPHONE); > + /* when plug 4-pole out, if both AB=3 AB=0 happen,3-pole plug > + * in will be incorrectly reported, then 3-pole plug-out is > + * reported,if no mantory 4-pole plug-out, icon would be > + * visible. > + */ > + if (status == 0) { > + report = 0; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_MICROPHONE); > + } > + pr_info("accdet HEADPHONE(3-pole) %s\n", > + status ? "PlugIn" : "PlugOut"); > + break; > + case HEADSET_MIC: > + /* when plug 4-pole out, 3-pole plug out should also be > + * reported for slow plug-in case > + */ > + if (status == 0) { > + report = 0; > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_HEADPHONE); > + } > + if (status) > + report = SND_JACK_MICROPHONE; > + else > + report = 0; > + > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_MICROPHONE); > + pr_info("accdet MICROPHONE(4-pole) %s\n", > + status ? "PlugIn" : "PlugOut"); > + /* when press key for a long time then plug in > + * even recoginized as 4-pole > + * disable micbias timer still timeout after 6s > + * it check AB=00(because keep to press key) then disable > + * micbias, it will cause key no response > + */ > + del_timer_sync(&micbias_timer); > + break; > + case LINE_OUT_DEVICE: > + if (status) > + report = SND_JACK_LINEOUT; > + else > + report = 0; > + > + snd_soc_jack_report(&accdet->jack, report, SND_JACK_LINEOUT); > + pr_info("accdet LineOut %s\n", > + status ? "PlugIn" : "PlugOut"); > + break; > + default: > + pr_info("%s Invalid cableType\n", __func__); > + } > +} > + > +static void multi_key_detection(unsigned int cur_AB) > +{ > + if (cur_AB == ACCDET_STATE_AB_00) > + accdet->cur_key = check_key(accdet->cali_voltage); > + > + /* delay to wait eint been flaged in register. > + * or eint handler issued. accdet->cur_eint_state == PLUG_OUT > + */ > + usleep_range(10000, 12000); > + > + if (HAS_CAP(accdet->data->caps, ACCDET_AP_GPIO_EINT)) { > + if (accdet->cur_eint_state == EINT_PLUG_IN) > + send_key_event(accdet->cur_key, !cur_AB); > + else > + accdet->cur_key = NO_KEY; > + } else { > + bool irq_bit = false; > + unsigned int val = 0; > + > + regmap_read(accdet->regmap, ACCDET_IRQ_ADDR, &val); > + irq_bit = !(val & ACCDET_EINT_IRQ_B2_B3); > + /* send key when: > + * no eint is flaged in reg, and now eint PLUG_IN > + */ > + if (irq_bit && accdet->cur_eint_state == EINT_PLUG_IN) > + send_key_event(accdet->cur_key, !cur_AB); > + else > + accdet->cur_key = NO_KEY; > + } > + > + if (cur_AB) > + accdet->cur_key = NO_KEY; > +} > + > +static inline void clear_accdet_int(void) > +{ > + /* it is safe by using polling to adjust when to clear IRQ_CLR_BIT */ > + regmap_update_bits(accdet->regmap, ACCDET_IRQ_ADDR, > + BIT(ACCDET_IRQ_CLR_SFT), BIT(ACCDET_IRQ_CLR_SFT)); > +} > + > +static inline void clear_accdet_int_check(void) > +{ > + u64 cur_time = mt6359_accdet_get_current_time(); > + unsigned int val = 0; > + > + regmap_read(accdet->regmap, ACCDET_IRQ_ADDR, &val); > + while (val & ACCDET_IRQ_MASK_SFT && > + mt6359_accdet_timeout_ns(cur_time, ACCDET_TIME_OUT)) > + ; > + /* clear accdet int, modify for fix interrupt trigger twice error */ > + regmap_update_bits(accdet->regmap, ACCDET_IRQ_ADDR, > + BIT(ACCDET_IRQ_CLR_SFT), 0); > + regmap_update_bits(accdet->regmap, RG_INT_STATUS_ACCDET_ADDR, > + BIT(RG_INT_STATUS_ACCDET_SFT), > + BIT(RG_INT_STATUS_ACCDET_SFT)); > +} > + > +static inline void clear_accdet_eint(unsigned int eintid) > +{ > + if ((eintid & PMIC_EINT0) == PMIC_EINT0) { > + regmap_update_bits(accdet->regmap, ACCDET_IRQ_ADDR, > + BIT(ACCDET_EINT0_IRQ_CLR_SFT), > + BIT(ACCDET_EINT0_IRQ_CLR_SFT)); > + } > + if ((eintid & PMIC_EINT1) == PMIC_EINT1) { > + regmap_update_bits(accdet->regmap, ACCDET_IRQ_ADDR, > + BIT(ACCDET_EINT1_IRQ_CLR_SFT), > + BIT(ACCDET_EINT1_IRQ_CLR_SFT)); > + } > +} > + > +static inline void clear_accdet_eint_check(unsigned int eintid) > +{ > + u64 cur_time = mt6359_accdet_get_current_time(); > + unsigned int val = 0; > + > + regmap_read(accdet->regmap, ACCDET_IRQ_ADDR, &val); > + > + if ((eintid & PMIC_EINT0) == PMIC_EINT0) { > + while (val & ACCDET_EINT0_IRQ_MASK_SFT && > + mt6359_accdet_timeout_ns(cur_time, ACCDET_TIME_OUT)) > + ; > + regmap_update_bits(accdet->regmap, ACCDET_IRQ_ADDR, > + BIT(ACCDET_EINT0_IRQ_CLR_SFT), 0); > + regmap_update_bits(accdet->regmap, > + RG_INT_STATUS_ACCDET_ADDR, > + BIT(RG_INT_STATUS_ACCDET_EINT0_SFT), > + BIT(RG_INT_STATUS_ACCDET_EINT0_SFT)); > + } > + if ((eintid & PMIC_EINT1) == PMIC_EINT1) { > + while (val & ACCDET_EINT1_IRQ_MASK_SFT && > + mt6359_accdet_timeout_ns(cur_time, ACCDET_TIME_OUT)) > + ; > + regmap_update_bits(accdet->regmap, ACCDET_IRQ_ADDR, > + BIT(ACCDET_EINT1_IRQ_CLR_SFT), 0); > + regmap_update_bits(accdet->regmap, > + RG_INT_STATUS_ACCDET_ADDR, > + BIT(RG_INT_STATUS_ACCDET_EINT1_SFT), > + BIT(RG_INT_STATUS_ACCDET_EINT1_SFT)); > + } > +} > + > +static unsigned int adjust_eint_analog_setting(void) > +{ > + if (accdet_dts.eint_detect_mode == 0x3 || > + accdet_dts.eint_detect_mode == 0x4) { > + /* ESD switches off */ > + regmap_update_bits(accdet->regmap, > + RG_ACCDETSPARE_ADDR, 1 << 8, 0); > + } > + if (accdet_dts.eint_detect_mode == 0x4) { > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + /* enable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0CONFIGACCDET_ADDR, > + BIT(RG_EINT0CONFIGACCDET_SFT), > + BIT(RG_EINT0CONFIGACCDET_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + /* enable RG_EINT1CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT1CONFIGACCDET_ADDR, > + BIT(RG_EINT1CONFIGACCDET_SFT), > + BIT(RG_EINT1CONFIGACCDET_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + /* enable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0CONFIGACCDET_ADDR, > + BIT(RG_EINT0CONFIGACCDET_SFT), > + BIT(RG_EINT0CONFIGACCDET_SFT)); > + /* enable RG_EINT1CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT1CONFIGACCDET_ADDR, > + BIT(RG_EINT1CONFIGACCDET_SFT), > + BIT(RG_EINT1CONFIGACCDET_SFT)); > + } > + if (accdet_dts.eint_use_ext_res == 0x3 || > + accdet_dts.eint_use_ext_res == 0x4) { > + /*select 500k, use internal resistor */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0HIRENB_ADDR, > + BIT(RG_EINT0HIRENB_SFT), > + BIT(RG_EINT0HIRENB_SFT)); > + } > + } > + return 0; > +} > + > +static unsigned int adjust_eint_digital_setting(void) > +{ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + /* disable inverter */ > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT), 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + /* disable inverter */ > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT), 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + /* disable inverter */ > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT), 0); > + /* disable inverter */ > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT), 0); > + } > + > + if (accdet_dts.eint_detect_mode == 0x4) { > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + /* set DA stable signal */ > + regmap_update_bits(accdet->regmap, > + ACCDET_DA_STABLE_ADDR, > + BIT(ACCDET_EINT0_CEN_STABLE_SFT), 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + /* set DA stable signal */ > + regmap_update_bits(accdet->regmap, > + ACCDET_DA_STABLE_ADDR, > + BIT(ACCDET_EINT1_CEN_STABLE_SFT), 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + /* set DA stable signal */ > + regmap_update_bits(accdet->regmap, > + ACCDET_DA_STABLE_ADDR, > + BIT(ACCDET_EINT0_CEN_STABLE_SFT), 0); > + /* set DA stable signal */ > + regmap_update_bits(accdet->regmap, > + ACCDET_DA_STABLE_ADDR, > + BIT(ACCDET_EINT1_CEN_STABLE_SFT), 0); > + } > + } > + return 0; > +} > + > +static unsigned int adjust_eint_setting(unsigned int eintsts) > +{ > + if (eintsts == M_PLUG_IN) { > + /* adjust digital setting */ > + adjust_eint_digital_setting(); > + /* adjust analog setting */ > + adjust_eint_analog_setting(); > + } else if (eintsts == M_PLUG_OUT) { > + /* set debounce to 1ms */ > + accdet_set_debounce(eint_state000, > + accdet_dts.pwm_deb.eint_debounce0); > + } else { > + pr_debug("should not be here %s()\n", __func__); > + } > + > + return 0; > +} > + > +static void recover_eint_analog_setting(void) > +{ > + if (accdet_dts.eint_detect_mode == 0x3 || > + accdet_dts.eint_detect_mode == 0x4) { > + /* ESD switches on */ > + regmap_update_bits(accdet->regmap, RG_ACCDETSPARE_ADDR, > + 1 << 8, 1 << 8); > + } > + if (accdet_dts.eint_detect_mode == 0x4) { > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + /* disable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0CONFIGACCDET_ADDR, > + BIT(RG_EINT0CONFIGACCDET_SFT), 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + /* disable RG_EINT1CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT1CONFIGACCDET_ADDR, > + BIT(RG_EINT1CONFIGACCDET_SFT), 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + /* disable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0CONFIGACCDET_ADDR, > + BIT(RG_EINT0CONFIGACCDET_SFT), 0); > + /* disable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT1CONFIGACCDET_ADDR, > + BIT(RG_EINT1CONFIGACCDET_SFT), 0); > + } > + regmap_update_bits(accdet->regmap, RG_EINT0HIRENB_ADDR, > + BIT(RG_EINT0HIRENB_SFT), 0); > + } > +} > + > +static void recover_eint_digital_setting(void) > +{ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_M_SW_EN_ADDR, > + BIT(ACCDET_EINT0_M_SW_EN_SFT), 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_M_SW_EN_ADDR, > + BIT(ACCDET_EINT1_M_SW_EN_SFT), 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_M_SW_EN_ADDR, > + BIT(ACCDET_EINT0_M_SW_EN_SFT), 0); > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_M_SW_EN_ADDR, > + BIT(ACCDET_EINT1_M_SW_EN_SFT), 0); > + } > + if (accdet_dts.eint_detect_mode == 0x4) { > + /* enable eint0cen */ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + /* enable eint0cen */ > + regmap_update_bits(accdet->regmap, > + ACCDET_DA_STABLE_ADDR, > + BIT(ACCDET_EINT0_CEN_STABLE_SFT), > + BIT(ACCDET_EINT0_CEN_STABLE_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + /* enable eint1cen */ > + regmap_update_bits(accdet->regmap, > + ACCDET_DA_STABLE_ADDR, > + BIT(ACCDET_EINT1_CEN_STABLE_SFT), > + BIT(ACCDET_EINT1_CEN_STABLE_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + /* enable eint0cen */ > + regmap_update_bits(accdet->regmap, > + ACCDET_DA_STABLE_ADDR, > + BIT(ACCDET_EINT0_CEN_STABLE_SFT), > + BIT(ACCDET_EINT0_CEN_STABLE_SFT)); > + /* enable eint1cen */ > + regmap_update_bits(accdet->regmap, > + ACCDET_DA_STABLE_ADDR, > + BIT(ACCDET_EINT1_CEN_STABLE_SFT), > + BIT(ACCDET_EINT1_CEN_STABLE_SFT)); > + } > + } > + > + if (accdet_dts.eint_detect_mode != 0x1) { > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + /* enable inverter */ > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT), > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + /* enable inverter */ > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT), > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + /* enable inverter */ > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT), > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT)); > + /* enable inverter */ > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT), > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT)); > + } > + } > +} > + > +static void recover_eint_setting(unsigned int eintsts) > +{ > + if (eintsts == M_PLUG_OUT) { > + recover_eint_analog_setting(); > + recover_eint_digital_setting(); > + } > +} > + > +static unsigned int get_triggered_eint(void) > +{ > + unsigned int eint_ID = NO_PMIC_EINT; > + unsigned int irq_status = 0; > + > + regmap_read(accdet->regmap, ACCDET_IRQ_ADDR, &irq_status); > + > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + if ((irq_status & ACCDET_EINT0_IRQ_MASK_SFT) == > + ACCDET_EINT0_IRQ_MASK_SFT) > + eint_ID = PMIC_EINT0; > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + if ((irq_status & ACCDET_EINT1_IRQ_MASK_SFT) == > + ACCDET_EINT1_IRQ_MASK_SFT) > + eint_ID = PMIC_EINT1; > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + if ((irq_status & ACCDET_EINT0_IRQ_MASK_SFT) == > + ACCDET_EINT0_IRQ_MASK_SFT) > + eint_ID |= PMIC_EINT0; > + if ((irq_status & ACCDET_EINT1_IRQ_MASK_SFT) == > + ACCDET_EINT1_IRQ_MASK_SFT) > + eint_ID |= PMIC_EINT1; > + } > + return eint_ID; > +} > + > +static inline void enable_accdet(unsigned int state_swctrl) > +{ > + /* enable ACCDET unit */ > + regmap_update_bits(accdet->regmap, ACCDET_SW_EN_ADDR, > + BIT(ACCDET_SW_EN_SFT), BIT(ACCDET_SW_EN_SFT)); > +} > + > +static inline void disable_accdet(void) > +{ > + clear_accdet_int(); > + usleep_range(200, 300); > + mutex_lock(&accdet->res_lock); > + clear_accdet_int_check(); > + mutex_unlock(&accdet->res_lock); > + > + /* recover accdet debounce0,3 */ > + accdet_set_debounce(accdet_state000, cust_pwm_deb->debounce0); > + accdet_set_debounce(accdet_state001, cust_pwm_deb->debounce1); > + accdet_set_debounce(accdet_state011, cust_pwm_deb->debounce3); > +} > + > +static inline void headset_plug_out(void) > +{ > + send_status_event(accdet->cable_type, 0); > + accdet->accdet_status = PLUG_OUT; > + accdet->cable_type = NO_DEVICE; > + > + if (accdet->cur_key != 0) { > + send_key_event(accdet->cur_key, 0); > + accdet->cur_key = 0; > + } > + pr_info("accdet %s, set cable_type = NO_DEVICE\n", __func__); > +} > + > +static void dis_micbias_timerhandler(struct timer_list *t) > +{ > + int ret = 0; > + > + ret = queue_work(accdet->dis_micbias_workqueue, > + &accdet->dis_micbias_work); > + if (!ret) > + pr_notice("Error: %s (%d)\n", __func__, ret); > +} > + > +static void dis_micbias_work_callback(struct work_struct *work) > +{ > + unsigned int cur_AB = 0, eint = 0; > + unsigned int val = 0; > + > + /* check EINT0 status, if plug out, > + * not need to disable accdet here > + */ > + regmap_read(accdet->regmap, ACCDET_EINT0_MEM_IN_ADDR, &val); > + eint = ((val >> ACCDET_EINT0_MEM_IN_SFT) & ACCDET_EINT0_MEM_IN_MASK); > + if (eint == M_PLUG_OUT) { > + pr_notice("%s Plug-out, no need disable micbias\n", __func__); > + return; > + } > + > + regmap_read(accdet->regmap, ACCDET_MEM_IN_ADDR, &val); > + cur_AB = (val >> ACCDET_STATE_MEM_IN_OFFSET) & ACCDET_STATE_AB_MASK; > + > + /* if 3pole disable accdet > + * if <20k + 4pole, disable accdet will disable accdet > + * plug out interrupt. The behavior will same as 3pole > + */ > + if (accdet->cable_type == HEADSET_NO_MIC || > + cur_AB == ACCDET_STATE_AB_00 || > + cur_AB == ACCDET_STATE_AB_11) { > + regmap_update_bits(accdet->regmap, ACCDET_SW_EN_ADDR, > + BIT(ACCDET_SW_EN_SFT), 0); > + disable_accdet(); > + } > +} > + > +static void eint_work_callback(struct work_struct *work) > +{ > + if (accdet->cur_eint_state == EINT_PLUG_IN) { > + /* disable vusb LP */ > + regmap_write(accdet->regmap, RG_LDO_VUSB_HW0_OP_EN_ADDR, 0x8000); > + > + mutex_lock(&accdet->res_lock); > + accdet->eint_sync_flag = true; > + mutex_unlock(&accdet->res_lock); > + __pm_wakeup_event(accdet->timer_lock, > + jiffies_to_msecs(7 * HZ)); > + > + accdet_init(); > + > + enable_accdet(0); > + } else { > + mutex_lock(&accdet->res_lock); > + accdet->eint_sync_flag = false; > + mutex_unlock(&accdet->res_lock); > + del_timer_sync(&micbias_timer); > + > + regmap_update_bits(accdet->regmap, ACCDET_SW_EN_ADDR, > + BIT(ACCDET_SW_EN_SFT), 0); > + disable_accdet(); > + headset_plug_out(); > + } > + > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT_IRQ)) > + recover_eint_setting(accdet->eint_id); > + else if (HAS_CAP(accdet->data->caps, ACCDET_AP_GPIO_EINT)) > + enable_irq(accdet->gpioirq); > +} > + > +static void accdet_set_debounce(int state, unsigned int debounce) > +{ > + switch (state) { > + case accdet_state000: > + /* set ACCDET debounce value = debounce/32 ms */ > + regmap_write(accdet->regmap, ACCDET_DEBOUNCE0_ADDR, debounce); > + break; > + case accdet_state001: > + regmap_write(accdet->regmap, ACCDET_DEBOUNCE1_ADDR, debounce); > + break; > + case accdet_state010: > + regmap_write(accdet->regmap, ACCDET_DEBOUNCE2_ADDR, debounce); > + break; > + case accdet_state011: > + regmap_write(accdet->regmap, ACCDET_DEBOUNCE3_ADDR, debounce); > + break; > + case accdet_auxadc: > + /* set auxadc debounce:0x42(2ms) */ > + regmap_write(accdet->regmap, > + ACCDET_CONNECT_AUXADC_TIME_DIG_ADDR, debounce); > + break; > + case eint_state000: > + regmap_update_bits(accdet->regmap, ACCDET_EINT_DEBOUNCE0_ADDR, > + 0xF << ACCDET_EINT_DEBOUNCE0_SFT, > + debounce << ACCDET_EINT_DEBOUNCE0_SFT); > + break; > + case eint_state001: > + regmap_update_bits(accdet->regmap, ACCDET_EINT_DEBOUNCE1_ADDR, > + 0xF << ACCDET_EINT_DEBOUNCE1_SFT, > + debounce << ACCDET_EINT_DEBOUNCE1_SFT); > + break; > + case eint_state010: > + regmap_update_bits(accdet->regmap, ACCDET_EINT_DEBOUNCE2_ADDR, > + 0xF << ACCDET_EINT_DEBOUNCE2_SFT, > + debounce << ACCDET_EINT_DEBOUNCE2_SFT); > + break; > + case eint_state011: > + regmap_update_bits(accdet->regmap, ACCDET_EINT_DEBOUNCE3_ADDR, > + 0xF << ACCDET_EINT_DEBOUNCE3_SFT, > + debounce << ACCDET_EINT_DEBOUNCE3_SFT); > + break; > + case eint_inverter_state000: > + regmap_write(accdet->regmap, ACCDET_EINT_INVERTER_DEBOUNCE_ADDR, > + debounce); > + break; > + default: > + pr_notice("Error: %s error state (%d)\n", __func__, state); > + break; > + } > +} > + > +static inline void check_cable_type(void) > +{ > + unsigned int cur_AB = 0; > + unsigned int val = 0; > + > + regmap_read(accdet->regmap, ACCDET_MEM_IN_ADDR, &val); > + > + cur_AB = (val >> ACCDET_STATE_MEM_IN_OFFSET) & ACCDET_STATE_AB_MASK; > + > + switch (accdet->accdet_status) { > + case PLUG_OUT: > + if (cur_AB == ACCDET_STATE_AB_00) { > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) { > + accdet->cable_type = HEADSET_NO_MIC; > + accdet->accdet_status = HOOK_SWITCH; > + } else { > + pr_notice("accdet hp has been plug-out\n"); > + } > + mutex_unlock(&accdet->res_lock); > + /* for IOT HP */ > + accdet_set_debounce(eint_state011, > + accdet_dts.pwm_deb.eint_debounce3); > + } else if (cur_AB == ACCDET_STATE_AB_01) { > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) { > + accdet->accdet_status = MIC_BIAS; > + accdet->cable_type = HEADSET_MIC; > + } else { > + pr_notice("accdet hp has been plug-out\n"); > + } > + mutex_unlock(&accdet->res_lock); > + /* adjust hook switch debounce time > + * for fast key press condition, avoid to miss key > + */ > + accdet_set_debounce(accdet_state000, > + button_press_debounce); > + > + /* adjust debounce1 to original 0x800(64ms), > + * to fix miss key issue when fast press double key. > + */ > + accdet_set_debounce(accdet_state001, > + button_press_debounce_01); > + accdet_set_debounce(eint_state011, 0x1); > + } else if (cur_AB == ACCDET_STATE_AB_11) { > + /* accdet PLUG_OUT state not change */ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT_IRQ)) { > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) { > + accdet->accdet_status = PLUG_OUT; > + accdet->cable_type = NO_DEVICE; > + } else { > + pr_notice("accdet hp been plug-out\n"); > + } > + mutex_unlock(&accdet->res_lock); > + } > + } else { > + pr_notice("accdet %s Invalid AB.\n", __func__); > + } > + break; > + case MIC_BIAS: > + if (cur_AB == ACCDET_STATE_AB_00) { > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) { > + accdet->accdet_status = HOOK_SWITCH; > + multi_key_detection(cur_AB); > + } else { > + pr_notice("accdet hp has been plug-out\n"); > + } > + mutex_unlock(&accdet->res_lock); > + } else if (cur_AB == ACCDET_STATE_AB_01) { > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) { > + accdet->accdet_status = MIC_BIAS; > + accdet->cable_type = HEADSET_MIC; > + /* accdet MIC_BIAS state not change */ > + } else { > + pr_notice("accdet hp has been plug-out\n"); > + } > + mutex_unlock(&accdet->res_lock); > + accdet_set_debounce(eint_state011, 0x1); > + } else if (cur_AB == ACCDET_STATE_AB_11) { > + /* accdet Don't send plug out in MIC_BIAS */ > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) > + accdet->accdet_status = PLUG_OUT; > + else > + pr_notice("accdet hp has been plug-out\n"); > + mutex_unlock(&accdet->res_lock); > + } else { > + pr_notice("accdet %s Invalid AB.\n", __func__); > + } > + break; > + case HOOK_SWITCH: > + if (cur_AB == ACCDET_STATE_AB_00) { > + } else if (cur_AB == ACCDET_STATE_AB_01) { > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) { > + multi_key_detection(cur_AB); > + accdet->accdet_status = MIC_BIAS; > + accdet->cable_type = HEADSET_MIC; > + } else { > + pr_notice("accdet hp has been plug-out\n"); > + } > + mutex_unlock(&accdet->res_lock); > + > + /* adjust debounce0 and debounce1 to fix miss key issue. > + */ > + accdet_set_debounce(accdet_state000, > + button_press_debounce); > + accdet_set_debounce(accdet_state001, > + button_press_debounce_01); > + /* for IOT HP */ > + accdet_set_debounce(eint_state011, 0x1); > + } else if (cur_AB == ACCDET_STATE_AB_11) { > + /* accdet Don't send plugout in HOOK_SWITCH */ > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) > + accdet->accdet_status = PLUG_OUT; > + else > + pr_notice("accdet hp has been plug-out\n"); > + mutex_unlock(&accdet->res_lock); > + } else { > + pr_notice("accdet %s Invalid AB.\n", __func__); > + } > + break; > + case STAND_BY: > + /* accdet %s STANDBY state.Err!Do nothing */ > + break; > + default: > + /* accdet %s Error state.Do nothing */ > + break; > + } > +} > + > +static void accdet_work_callback(struct work_struct *work) > +{ > + unsigned int pre_cable_type = accdet->cable_type; > + > + __pm_stay_awake(accdet->wake_lock); > + check_cable_type(); > + > + mutex_lock(&accdet->res_lock); > + if (accdet->eint_sync_flag) { > + if (pre_cable_type != accdet->cable_type) > + send_status_event(accdet->cable_type, 1); > + } > + mutex_unlock(&accdet->res_lock); > + if (accdet->cable_type != NO_DEVICE) { > + /* enable vusb LP */ > + regmap_write(accdet->regmap, RG_LDO_VUSB_HW0_OP_EN_ADDR, 0x8005); > + } > + > + __pm_relax(accdet->wake_lock); > +} > + > +static void accdet_queue_work(void) > +{ > + int ret; > + > + if (accdet->accdet_status == MIC_BIAS) > + accdet->cali_voltage = 10; > + > + ret = queue_work(accdet->accdet_workqueue, &accdet->accdet_work); > + if (!ret) > + pr_notice("Error: %s (%d)\n", __func__, ret); > +} > + > +static int pmic_eint_queue_work(int eint) > +{ > + int ret = 0; > + > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + if (eint == PMIC_EINT0) { > + if (accdet->cur_eint_state == EINT_PLUG_IN) { > + accdet_set_debounce(accdet_state011, > + cust_pwm_deb->debounce3); > + accdet->cur_eint_state = EINT_PLUG_OUT; > + } else { > + if (accdet->eint_id != M_PLUG_OUT) { > + accdet->cur_eint_state = EINT_PLUG_IN; > + mod_timer(&micbias_timer, > + jiffies + MICBIAS_DISABLE_TIMER); > + } > + } > + ret = queue_work(accdet->eint_workqueue, > + &accdet->eint_work); > + } else { > + pr_notice("%s invalid EINT ID!\n", __func__); > + } > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + if (eint == PMIC_EINT1) { > + if (accdet->cur_eint_state == EINT_PLUG_IN) { > + accdet_set_debounce(accdet_state011, > + cust_pwm_deb->debounce3); > + accdet->cur_eint_state = EINT_PLUG_OUT; > + } else { > + if (accdet->eint_id != M_PLUG_OUT) { > + accdet->cur_eint_state = EINT_PLUG_IN; > + mod_timer(&micbias_timer, > + jiffies + MICBIAS_DISABLE_TIMER); > + } > + } > + ret = queue_work(accdet->eint_workqueue, > + &accdet->eint_work); > + } else { > + pr_notice("%s invalid EINT ID!\n", __func__); > + } > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + if ((eint & PMIC_EINT0) == PMIC_EINT0) { > + if (accdet->eint0_state == EINT_PLUG_IN) { > + accdet_set_debounce(accdet_state011, > + cust_pwm_deb->debounce3); > + accdet->eint0_state = EINT_PLUG_OUT; > + } else { > + if (accdet->eint_id != M_PLUG_OUT) > + accdet->eint0_state = EINT_PLUG_IN; > + } > + } > + if ((eint & PMIC_EINT1) == PMIC_EINT1) { > + if (accdet->eint1_state == EINT_PLUG_IN) { > + accdet_set_debounce(accdet_state011, > + cust_pwm_deb->debounce3); > + accdet->eint1_state = EINT_PLUG_OUT; > + } else { > + if (accdet->eint_id != M_PLUG_OUT) > + accdet->eint1_state = EINT_PLUG_IN; > + } > + } > + > + /* bi_eint trigger issued current state, may */ > + if (accdet->cur_eint_state == EINT_PLUG_OUT) { > + accdet->cur_eint_state = > + accdet->eint0_state & accdet->eint1_state; > + if (accdet->cur_eint_state == EINT_PLUG_IN) { > + mod_timer(&micbias_timer, > + jiffies + MICBIAS_DISABLE_TIMER); > + ret = queue_work(accdet->eint_workqueue, > + &accdet->eint_work); > + } > + } else if (accdet->cur_eint_state == EINT_PLUG_IN) { > + if ((accdet->eint0_state | accdet->eint1_state) == EINT_PLUG_OUT) { > + clear_accdet_eint_check(PMIC_EINT0); > + clear_accdet_eint_check(PMIC_EINT1); > + } else if ((accdet->eint0_state & accdet->eint1_state) > + == EINT_PLUG_OUT) { > + accdet->cur_eint_state = EINT_PLUG_OUT; > + ret = queue_work(accdet->eint_workqueue, > + &accdet->eint_work); > + } > + } > + } > + return ret; > +} > + > +void accdet_irq_handle(void) > +{ > + unsigned int eint = 0; > + unsigned int irq_status = 0, acc_sts = 0, eint_sts = 0; > + unsigned int val = 0; > + > + eint = get_triggered_eint(); > + regmap_read(accdet->regmap, ACCDET_IRQ_ADDR, &irq_status); > + regmap_read(accdet->regmap, ACCDET_MEM_IN_ADDR, &acc_sts); > + regmap_read(accdet->regmap, ACCDET_EINT0_MEM_IN_ADDR, &eint_sts); > + > + if ((irq_status & ACCDET_IRQ_MASK_SFT) && eint == 0) { > + clear_accdet_int(); > + accdet_queue_work(); > + clear_accdet_int_check(); > + } else if (eint != NO_PMIC_EINT) { > + /* check EINT0 status */ > + regmap_read(accdet->regmap, ACCDET_EINT0_MEM_IN_ADDR, &val); > + accdet->eint_id = ((val >> ACCDET_EINT0_MEM_IN_SFT) & > + ACCDET_EINT0_MEM_IN_MASK); > + /* adjust eint digital/analog setting */ > + adjust_eint_setting(accdet->eint_id); > + clear_accdet_eint(eint); > + clear_accdet_eint_check(eint); > + pmic_eint_queue_work(eint); > + } else { > + pr_notice("%s no interrupt detected!\n", __func__); > + } > +} > + > +static irqreturn_t mtk_accdet_irq_handler_thread(int irq, void *data) > +{ > + accdet_irq_handle(); > + > + return IRQ_HANDLED; > +} > + > +static irqreturn_t ex_eint_handler(int irq, void *data) > +{ > + int ret = 0; > + > + if (accdet->cur_eint_state == EINT_PLUG_IN) { > + /* To trigger EINT when the headset was plugged in > + * We set the polarity back as we initialed. > + */ > + if (accdet->accdet_eint_type == IRQ_TYPE_LEVEL_HIGH) > + irq_set_irq_type(accdet->gpioirq, IRQ_TYPE_LEVEL_HIGH); > + else > + irq_set_irq_type(accdet->gpioirq, IRQ_TYPE_LEVEL_LOW); > + gpio_set_debounce(accdet->gpiopin, accdet->gpio_hp_deb); > + > + accdet->cur_eint_state = EINT_PLUG_OUT; > + } else { > + /* To trigger EINT when the headset was plugged out > + * We set the opposite polarity to what we initialed. > + */ > + if (accdet->accdet_eint_type == IRQ_TYPE_LEVEL_HIGH) > + irq_set_irq_type(accdet->gpioirq, IRQ_TYPE_LEVEL_LOW); > + else > + irq_set_irq_type(accdet->gpioirq, IRQ_TYPE_LEVEL_HIGH); > + > + gpio_set_debounce(accdet->gpiopin, > + accdet_dts.plugout_deb * 1000); > + > + accdet->cur_eint_state = EINT_PLUG_IN; > + > + mod_timer(&micbias_timer, > + jiffies + MICBIAS_DISABLE_TIMER); > + } > + > + disable_irq_nosync(accdet->gpioirq); > + ret = queue_work(accdet->eint_workqueue, &accdet->eint_work); > + return IRQ_HANDLED; > +} > + > +static inline int ext_eint_setup(struct platform_device *platform_device) > +{ > + int ret = 0; > + unsigned int ints[4] = { 0 }; > + struct device_node *node = NULL; > + struct pinctrl_state *pins_default = NULL; > + > + accdet->pinctrl = devm_pinctrl_get(&platform_device->dev); > + if (IS_ERR(accdet->pinctrl)) { > + ret = PTR_ERR(accdet->pinctrl); > + return ret; > + } > + > + pins_default = pinctrl_lookup_state(accdet->pinctrl, "default"); > + if (IS_ERR(pins_default)) > + ret = PTR_ERR(pins_default); > + > + accdet->pins_eint = pinctrl_lookup_state(accdet->pinctrl, > + "state_eint_as_int"); > + if (IS_ERR(accdet->pins_eint)) { > + ret = PTR_ERR(accdet->pins_eint); > + return ret; > + } > + pinctrl_select_state(accdet->pinctrl, accdet->pins_eint); > + > + node = of_find_matching_node(node, mt6359_accdet_of_match); > + if (!node) > + return -1; > + > + accdet->gpiopin = of_get_named_gpio(node, "deb-gpios", 0); > + ret = of_property_read_u32(node, "debounce", > + &accdet->gpio_hp_deb); > + if (ret < 0) > + return ret; > + > + gpio_set_debounce(accdet->gpiopin, accdet->gpio_hp_deb); > + > + accdet->gpioirq = irq_of_parse_and_map(node, 0); > + ret = of_property_read_u32_array(node, "interrupts", ints, > + ARRAY_SIZE(ints)); > + if (ret) > + return ret; > + > + accdet->accdet_eint_type = ints[1]; > + ret = request_irq(accdet->gpioirq, ex_eint_handler, IRQF_TRIGGER_NONE, > + "accdet-eint", NULL); > + if (ret) > + return ret; > + > + return 0; > +} > + > +static int mt6359_accdet_parse_dt(void) > +{ > + int ret = 0; > + struct device_node *node = NULL; > + int pwm_deb[15] = {0}; > + int three_key[4] = {0}; > + unsigned int tmp = 0; > + > + node = of_find_matching_node(node, mt6359_accdet_of_match); > + if (!node) { > + pr_notice("Error: %s can't find dts node\n", __func__); > + return -1; > + } > + > + ret = of_property_read_u32(node, "eint_use_ext_res", > + &accdet_dts.eint_use_ext_res); > + if (ret) { > + /* eint use internal resister */ > + accdet_dts.eint_use_ext_res = 0x0; > + } > + ret = of_property_read_u32(node, "eint_detect_mode", > + &accdet_dts.eint_detect_mode); > + if (ret) { > + /* eint detection mode equals to EINT HW Mode */ > + accdet_dts.eint_detect_mode = 0x4; > + } > + > + ret = of_property_read_u32(node, > + "accdet-mic-vol", &accdet_dts.mic_vol); > + if (ret) > + accdet_dts.mic_vol = 8; > + > + ret = of_property_read_u32(node, "accdet-plugout-debounce", > + &accdet_dts.plugout_deb); > + if (ret) > + accdet_dts.plugout_deb = 1; > + > + ret = of_property_read_u32(node, > + "accdet-mic-mode", &accdet_dts.mic_mode); > + if (ret) > + accdet_dts.mic_mode = 2; > + > + ret = of_property_read_u32_array(node, "headset-mode-setting", pwm_deb, > + ARRAY_SIZE(pwm_deb)); > + /* debounce8(auxadc debounce) is default, needn't get from dts */ > + if (!ret) > + memcpy(&accdet_dts.pwm_deb, pwm_deb, sizeof(pwm_deb)); > + > + cust_pwm_deb = &accdet_dts.pwm_deb; > + > + ret = of_property_read_u32(node, "headset-eint-level-pol", > + &accdet_dts.eint_pol); > + if (ret) > + accdet_dts.eint_pol = 8; > + > + ret = of_property_read_u32(node, "headset-use-ap-eint", &tmp); > + if (ret) > + tmp = 0; > + if (tmp == 0) > + accdet->data->caps |= ACCDET_PMIC_EINT_IRQ; > + else if (tmp == 1) > + accdet->data->caps |= ACCDET_AP_GPIO_EINT; > + > + ret = of_property_read_u32(node, "headset-eint-num", &tmp); > + if (ret) > + tmp = 0; > + if (tmp == 0) > + accdet->data->caps |= ACCDET_PMIC_EINT0; > + else if (tmp == 1) > + accdet->data->caps |= ACCDET_PMIC_EINT1; > + else if (tmp == 2) > + accdet->data->caps |= ACCDET_PMIC_BI_EINT; > + > + ret = of_property_read_u32(node, "headset-eint-trig-mode", &tmp); > + if (ret) > + tmp = 0; > + if (tmp == 0) > + accdet->data->caps |= ACCDET_PMIC_GPIO_TRIG_EINT; > + else if (tmp == 1) > + accdet->data->caps |= ACCDET_PMIC_INVERTER_TRIG_EINT; > + > + ret = of_property_read_u32(node, "headset-key-mode", &tmp); > + if (ret) > + tmp = 0; > + if (tmp == 0) > + accdet->data->caps |= ACCDET_THREE_KEY; > + else if (tmp == 1) > + accdet->data->caps |= ACCDET_FOUR_KEY; > + else if (tmp == 2) > + accdet->data->caps |= ACCDET_TRI_KEY_CDD; > + > + if (HAS_CAP(accdet->data->caps, ACCDET_FOUR_KEY)) { > + int four_key[5]; > + > + ret = of_property_read_u32_array(node, > + "headset-four-key-threshold", > + four_key, ARRAY_SIZE(four_key)); > + if (!ret) { > + memcpy(&accdet_dts.four_key, four_key + 1, > + sizeof(struct four_key_threshold)); > + } else { > + pr_notice("accdet no 4-key-thrsh dts, use efuse\n"); > + } > + } else { > + if (HAS_CAP(accdet->data->caps, ACCDET_THREE_KEY)) { > + ret = of_property_read_u32_array(node, > + "headset-three-key-threshold", > + three_key, ARRAY_SIZE(three_key)); > + } else { > + ret = of_property_read_u32_array(node, > + "headset-three-key-threshold-CDD", > + three_key, > + ARRAY_SIZE(three_key)); > + } > + if (!ret) > + memcpy(&accdet_dts.three_key, three_key + 1, > + sizeof(struct three_key_threshold)); > + } > + > + ret = of_property_read_u32(node, "eint_use_ext_res", > + &accdet_dts.eint_use_ext_res); > + if (ret) { > + /* eint use internal resister */ > + accdet_dts.eint_use_ext_res = 0x0; > + } > + pr_info("accdet caps=%x\n", accdet->data->caps); > + > + return 0; > +} > + > +static void config_digital_init_by_mode(void) > +{ > + /* enable eint cmpmem pwm */ > + regmap_write(accdet->regmap, ACCDET_EINT_CMPMEN_PWM_THRESH_ADDR, > + (accdet_dts.pwm_deb.eint_pwm_width << 4 | > + accdet_dts.pwm_deb.eint_pwm_thresh)); > + /* DA signal stable */ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + regmap_write(accdet->regmap, ACCDET_DA_STABLE_ADDR, > + ACCDET_EINT0_STABLE_VAL); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + regmap_write(accdet->regmap, ACCDET_DA_STABLE_ADDR, > + ACCDET_EINT1_STABLE_VAL); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + regmap_write(accdet->regmap, ACCDET_DA_STABLE_ADDR, > + ACCDET_EINT0_STABLE_VAL); > + regmap_write(accdet->regmap, ACCDET_DA_STABLE_ADDR, > + ACCDET_EINT1_STABLE_VAL); > + } > + /* after receive n+1 number, interrupt issued. now is 2 times */ > + regmap_update_bits(accdet->regmap, ACCDET_EINT_M_PLUG_IN_NUM_ADDR, > + BIT(ACCDET_EINT_M_PLUG_IN_NUM_SFT), > + BIT(ACCDET_EINT_M_PLUG_IN_NUM_SFT)); > + /* setting HW mode, enable digital fast discharge > + * if use EINT0 & EINT1 detection, please modify > + * ACCDET_HWMODE_EN_ADDR[2:1] > + */ > + regmap_write(accdet->regmap, ACCDET_HWMODE_EN_ADDR, 0x100); > + > + regmap_update_bits(accdet->regmap, ACCDET_EINT_M_DETECT_EN_ADDR, > + BIT(ACCDET_EINT_M_DETECT_EN_SFT), 0); > + > + /* enable PWM */ > + regmap_write(accdet->regmap, ACCDET_CMP_PWM_EN_ADDR, 0x67); > + /* enable inverter detection */ > + if (accdet_dts.eint_detect_mode == 0x1) { > + /* disable inverter detection */ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT), > + 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT), > + 0); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT), > + 0); > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT), > + 0); > + } > + } else { > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT), > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT), > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT0_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT), > + BIT(ACCDET_EINT0_INVERTER_SW_EN_SFT)); > + regmap_update_bits(accdet->regmap, > + ACCDET_EINT1_INVERTER_SW_EN_ADDR, > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT), > + BIT(ACCDET_EINT1_INVERTER_SW_EN_SFT)); > + } > + } > +} > + > +static void config_eint_init_by_mode(void) > +{ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + regmap_update_bits(accdet->regmap, RG_EINT0EN_ADDR, > + BIT(RG_EINT0EN_SFT), BIT(RG_EINT0EN_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + regmap_update_bits(accdet->regmap, RG_EINT1EN_ADDR, > + BIT(RG_EINT1EN_SFT), BIT(RG_EINT1EN_SFT)); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + regmap_update_bits(accdet->regmap, RG_EINT0EN_ADDR, > + BIT(RG_EINT0EN_SFT), BIT(RG_EINT0EN_SFT)); > + regmap_update_bits(accdet->regmap, RG_EINT1EN_ADDR, > + BIT(RG_EINT1EN_SFT), BIT(RG_EINT1EN_SFT)); > + } > + /* ESD switches on */ > + regmap_update_bits(accdet->regmap, RG_ACCDETSPARE_ADDR, > + 1 << 8, 1 << 8); > + /* before playback, set NCP pull low before nagative voltage */ > + regmap_update_bits(accdet->regmap, RG_NCP_PDDIS_EN_ADDR, > + BIT(RG_NCP_PDDIS_EN_SFT), BIT(RG_NCP_PDDIS_EN_SFT)); > + > + if (accdet_dts.eint_detect_mode == 0x1 || > + accdet_dts.eint_detect_mode == 0x2 || > + accdet_dts.eint_detect_mode == 0x3) { > + if (accdet_dts.eint_use_ext_res == 0x1) { > + /* select VTH to 2v and 500k, use external resitance */ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + /* disable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0CONFIGACCDET_ADDR, > + BIT(RG_EINT0CONFIGACCDET_SFT), > + 0); > + } else if (HAS_CAP(accdet->data->caps, > + ACCDET_PMIC_EINT1)) { > + /* disable RG_EINT1CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT1CONFIGACCDET_ADDR, > + BIT(RG_EINT1CONFIGACCDET_SFT), > + 0); > + } else if (HAS_CAP(accdet->data->caps, > + ACCDET_PMIC_BI_EINT)) { > + /* disable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0CONFIGACCDET_ADDR, > + BIT(RG_EINT0CONFIGACCDET_SFT), > + 0); > + /* disable RG_EINT1CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT1CONFIGACCDET_ADDR, > + BIT(RG_EINT1CONFIGACCDET_SFT), > + 0); > + } > + } else { > + /* select VTH to 2v and 500k, use internal resitance */ > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + /* enable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0CONFIGACCDET_ADDR, > + BIT(RG_EINT0CONFIGACCDET_SFT), > + BIT(RG_EINT0CONFIGACCDET_SFT)); > + } else if (HAS_CAP(accdet->data->caps, > + ACCDET_PMIC_EINT1)) { > + /* enable RG_EINT1CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT1CONFIGACCDET_ADDR, > + BIT(RG_EINT1CONFIGACCDET_SFT), > + BIT(RG_EINT1CONFIGACCDET_SFT)); > + } else if (HAS_CAP(accdet->data->caps, > + ACCDET_PMIC_BI_EINT)) { > + /* enable RG_EINT0CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT0CONFIGACCDET_ADDR, > + BIT(RG_EINT0CONFIGACCDET_SFT), > + BIT(RG_EINT0CONFIGACCDET_SFT)); > + /* enable RG_EINT1CONFIGACCDET */ > + regmap_update_bits(accdet->regmap, > + RG_EINT1CONFIGACCDET_ADDR, > + BIT(RG_EINT1CONFIGACCDET_SFT), > + BIT(RG_EINT1CONFIGACCDET_SFT)); > + } > + } > + } > + > + if (accdet_dts.eint_detect_mode != 0x1) { > + /* current detect set 0.25uA */ > + regmap_update_bits(accdet->regmap, RG_ACCDETSPARE_ADDR, > + 0x3 << RG_ACCDETSPARE_SFT, > + 0x3 << RG_ACCDETSPARE_SFT); > + } > +} > + > +static void accdet_init_once(void) > +{ > + unsigned int reg = 0; > + > + /* reset the accdet unit */ > + regmap_update_bits(accdet->regmap, RG_ACCDET_RST_ADDR, > + BIT(RG_ACCDET_RST_SFT), BIT(RG_ACCDET_RST_SFT)); > + regmap_update_bits(accdet->regmap, RG_ACCDET_RST_ADDR, > + BIT(RG_ACCDET_RST_SFT), 0); > + > + /* clear high micbias1 voltage setting */ > + regmap_update_bits(accdet->regmap, RG_AUDPWDBMICBIAS1_ADDR, > + 0x3 << RG_AUDMICBIAS1HVEN_SFT, 0); > + /* clear micbias1 voltage */ > + regmap_update_bits(accdet->regmap, RG_AUDPWDBMICBIAS1_ADDR, > + 0x7 << RG_AUDMICBIAS1VREF_SFT, 0); > + > + /* init pwm frequency, duty & rise/falling delay */ > + regmap_write(accdet->regmap, ACCDET_PWM_WIDTH_ADDR, > + REGISTER_VAL(cust_pwm_deb->pwm_width)); > + regmap_write(accdet->regmap, ACCDET_PWM_THRESH_ADDR, > + REGISTER_VAL(cust_pwm_deb->pwm_thresh)); > + regmap_write(accdet->regmap, ACCDET_RISE_DELAY_ADDR, > + (cust_pwm_deb->fall_delay << 15 | cust_pwm_deb->rise_delay)); > + > + /* config micbias voltage, micbias1 vref is only controlled by accdet > + * if we need 2.8V, config [12:13] > + */ > + regmap_read(accdet->regmap, RG_AUDPWDBMICBIAS1_ADDR, &reg); > + if (accdet_dts.mic_vol <= 7) { > + /* micbias1 <= 2.7V */ > + regmap_write(accdet->regmap, RG_AUDPWDBMICBIAS1_ADDR, > + reg | (accdet_dts.mic_vol << RG_AUDMICBIAS1VREF_SFT) | > + RG_AUDMICBIAS1LOWPEN_MASK_SFT); > + } else if (accdet_dts.mic_vol == 8) { > + /* micbias1 = 2.8v */ > + regmap_write(accdet->regmap, RG_AUDPWDBMICBIAS1_ADDR, > + reg | (3 << RG_AUDMICBIAS1HVEN_SFT) | > + RG_AUDMICBIAS1LOWPEN_MASK_SFT); > + } else if (accdet_dts.mic_vol == 9) { > + /* micbias1 = 2.85v */ > + regmap_write(accdet->regmap, RG_AUDPWDBMICBIAS1_ADDR, > + reg | (1 << RG_AUDMICBIAS1HVEN_SFT) | > + RG_AUDMICBIAS1LOWPEN_MASK_SFT); > + } > + /* mic mode setting */ > + regmap_read(accdet->regmap, RG_AUDACCDETMICBIAS0PULLLOW_ADDR, &reg); > + if (accdet_dts.mic_mode == HEADSET_MODE_1) { > + /* ACC mode*/ > + regmap_write(accdet->regmap, RG_AUDACCDETMICBIAS0PULLLOW_ADDR, > + reg | RG_ACCDET_MODE_ANA11_MODE1); > + /* enable analog fast discharge */ > + regmap_update_bits(accdet->regmap, RG_ANALOGFDEN_ADDR, > + BIT(RG_ANALOGFDEN_SFT), > + BIT(RG_ANALOGFDEN_SFT)); > + regmap_update_bits(accdet->regmap, RG_ACCDETSPARE_ADDR, > + 0x3 << 11, 0x3 << 11); > + } else if (accdet_dts.mic_mode == HEADSET_MODE_2) { > + /* DCC mode Low cost mode without internal bias*/ > + regmap_write(accdet->regmap, RG_AUDACCDETMICBIAS0PULLLOW_ADDR, > + reg | RG_ACCDET_MODE_ANA11_MODE2); > + /* enable analog fast discharge */ > + regmap_update_bits(accdet->regmap, RG_ANALOGFDEN_ADDR, > + 0x3 << RG_ANALOGFDEN_SFT, > + 0x3 << RG_ANALOGFDEN_SFT); > + } else if (accdet_dts.mic_mode == HEADSET_MODE_6) { > + /* DCC mode Low cost mode with internal bias, > + * bit8 = 1 to use internal bias > + */ > + regmap_write(accdet->regmap, RG_AUDACCDETMICBIAS0PULLLOW_ADDR, > + reg | RG_ACCDET_MODE_ANA11_MODE6); > + regmap_update_bits(accdet->regmap, RG_AUDPWDBMICBIAS1_ADDR, > + BIT(RG_AUDMICBIAS1DCSW1PEN_SFT), > + BIT(RG_AUDMICBIAS1DCSW1PEN_SFT)); > + /* enable analog fast discharge */ > + regmap_update_bits(accdet->regmap, RG_ANALOGFDEN_ADDR, > + 0x3 << RG_ANALOGFDEN_SFT, > + 0x3 << RG_ANALOGFDEN_SFT); > + } > + > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT_IRQ)) { > + config_eint_init_by_mode(); > + config_digital_init_by_mode(); > + } else if (HAS_CAP(accdet->data->caps, ACCDET_AP_GPIO_EINT)) { > + /* set pull low pads and DCC mode */ > + regmap_write(accdet->regmap, RG_AUDACCDETMICBIAS0PULLLOW_ADDR, > + 0x8F); > + /* disconnect configaccdet */ > + regmap_write(accdet->regmap, RG_EINT1CONFIGACCDET_ADDR, > + 0x0); > + /* disable eint comparator */ > + regmap_write(accdet->regmap, RG_EINT0CMPEN_ADDR, 0x0); > + /* enable PWM */ > + regmap_write(accdet->regmap, ACCDET_CMP_PWM_EN_ADDR, 0x7); > + /* enable accdet sw mode */ > + regmap_write(accdet->regmap, ACCDET_HWMODE_EN_ADDR, 0x0); > + /* set DA signal to stable */ > + regmap_write(accdet->regmap, ACCDET_DA_STABLE_ADDR, 0x1); > + /* disable eint/inverter/sw_en */ > + regmap_write(accdet->regmap, ACCDET_SW_EN_ADDR, 0x0); > + } > +} > + > +static void accdet_init_debounce(void) > +{ > + /* set debounce to 1ms */ > + accdet_set_debounce(eint_state000, accdet_dts.pwm_deb.eint_debounce0); > + /* set debounce to 128ms */ > + accdet_set_debounce(eint_state011, accdet_dts.pwm_deb.eint_debounce3); > +} > + > +static inline void accdet_init(void) > +{ > + /* set and clear initial bit every eint interrutp */ > + regmap_update_bits(accdet->regmap, ACCDET_SEQ_INIT_ADDR, > + BIT(ACCDET_SEQ_INIT_SFT), BIT(ACCDET_SEQ_INIT_SFT)); > + mdelay(2); > + regmap_update_bits(accdet->regmap, ACCDET_SEQ_INIT_ADDR, > + BIT(ACCDET_SEQ_INIT_SFT), 0); > + mdelay(1); > + /* init the debounce time (debounce/32768)sec */ > + accdet_set_debounce(accdet_state000, cust_pwm_deb->debounce0); > + accdet_set_debounce(accdet_state001, cust_pwm_deb->debounce1); > + accdet_set_debounce(accdet_state011, cust_pwm_deb->debounce3); > + /* auxadc:2ms */ > + accdet_set_debounce(accdet_auxadc, cust_pwm_deb->debounce4); > + accdet_set_debounce(eint_state001, accdet_dts.pwm_deb.eint_debounce1); > + accdet_set_debounce(eint_inverter_state000, > + accdet_dts.pwm_deb.eint_inverter_debounce); > +} > + > +/* late init for DC trim, and this API Will be called by audio */ > +void mt6359_accdet_late_init(unsigned long data) > +{ > + if (atomic_cmpxchg(&accdet->init_once, 1, 0)) { > + del_timer_sync(&accdet_init_timer); > + accdet_init(); > + accdet_init_debounce(); > + accdet_init_once(); > + } else { > + pr_info("%s fail\n", __func__); > + } > +} > +EXPORT_SYMBOL(mt6359_accdet_late_init); > + > +static void delay_init_work_callback(struct work_struct *work) > +{ > + accdet_init(); > + accdet_init_debounce(); > + accdet_init_once(); > +} > + > +static void delay_init_timerhandler(struct timer_list *t) > +{ > + int ret = 0; > + > + ret = queue_work(accdet->delay_init_workqueue, > + &accdet->delay_init_work); > + if (!ret) > + pr_err("Error: %s (%d)\n", __func__, ret); > +} > + > +int mt6359_accdet_init(struct snd_soc_component *component, > + struct snd_soc_card *card) > +{ > + int ret = 0; > + struct mt63xx_accdet_data *priv = > + snd_soc_card_get_drvdata(component->card); > + > + /* Enable Headset and 4 Buttons Jack detection */ > + ret = snd_soc_card_jack_new(card, > + "Headset Jack", > + SND_JACK_HEADSET | > + SND_JACK_LINEOUT | > + SND_JACK_MECHANICAL, > + &priv->jack, > + NULL, 0); > + if (ret) { > + dev_err(card->dev, "Can't new Headset Jack %x\n", ret); > + return ret; > + } > + > + accdet->jack.jack->input_dev->id.bustype = BUS_HOST; > + snd_jack_set_key(accdet->jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); > + snd_jack_set_key(accdet->jack.jack, SND_JACK_BTN_1, KEY_VOLUMEDOWN); > + snd_jack_set_key(accdet->jack.jack, SND_JACK_BTN_2, KEY_VOLUMEUP); > + snd_jack_set_key(accdet->jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); > + > + return ret; > +} > +EXPORT_SYMBOL_GPL(mt6359_accdet_init); > + > +int mt6359_accdet_set_drvdata(struct snd_soc_card *card) > +{ > + snd_soc_card_set_drvdata(card, accdet); > + > + return 0; > +} > +EXPORT_SYMBOL_GPL(mt6359_accdet_set_drvdata); > + > +static int mt6359_accdet_probe(struct platform_device *pdev) > +{ > + int ret = 0; > + struct resource *res; > + struct mt6397_chip *mt6397_chip = dev_get_drvdata(pdev->dev.parent); > + const struct of_device_id *of_id = > + of_match_device(mt6359_accdet_of_match, &pdev->dev); > + if (!of_id) { > + dev_err(&pdev->dev, "Error: No device match found\n"); > + return -ENODEV; > + } > + > + accdet = devm_kzalloc(&pdev->dev, sizeof(*accdet), GFP_KERNEL); > + if (!accdet) > + return -ENOMEM; > + > + accdet->data = (struct accdet_priv *)of_id->data; > + accdet->pdev = pdev; > + > + /* parse dts attributes */ > + ret = mt6359_accdet_parse_dt(); > + if (ret) { > + dev_err(&pdev->dev, "Error: Get dts data failed (%d)\n", ret); > + return ret; > + } > + /* init lock */ > + accdet->wake_lock = wakeup_source_register(NULL, "accdet_wake_lock"); > + if (!accdet->wake_lock) { > + dev_err(&pdev->dev, "Error: get accdet_wake_lock fail\n"); > + return -ENOMEM; > + } > + accdet->timer_lock = wakeup_source_register(NULL, "accdet_timer_lock"); > + if (!accdet->timer_lock) { > + dev_err(&pdev->dev, "Error: get accdet_timer_lock fail\n"); > + return -ENOMEM; > + } > + mutex_init(&accdet->res_lock); > + > + platform_set_drvdata(pdev, accdet); > + ret = devm_snd_soc_register_component(&pdev->dev, > + &mt6359_accdet_soc_driver, > + NULL, 0); > + if (ret) { > + dev_err(&pdev->dev, "Error: register component fail\n"); > + return ret; > + } > + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + accdet->regmap = mt6397_chip->regmap; > + accdet->dev = &pdev->dev; > + accdet->auxadc_offset = 0; > + > + /* register pmic interrupt */ > + accdet->accdet_irq = platform_get_irq(pdev, 0); > + if (accdet->accdet_irq < 0) { > + dev_err(&pdev->dev, > + "Error: Get accdet irq failed (%d)\n", > + accdet->accdet_irq); > + return accdet->accdet_irq; > + } > + ret = devm_request_threaded_irq(&pdev->dev, accdet->accdet_irq, > + NULL, mtk_accdet_irq_handler_thread, > + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, > + "ACCDET_IRQ", accdet); > + if (ret) { > + dev_err(&pdev->dev, > + "Error: Get thread irq request failed (%d) (%d)\n", > + accdet->accdet_irq, ret); > + return ret; > + } > + > + if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT0)) { > + accdet->accdet_eint0 = platform_get_irq(pdev, 1); > + if (accdet->accdet_eint0 < 0) { > + dev_err(&pdev->dev, > + "Error: Get eint0 irq failed (%d)\n", > + accdet->accdet_eint0); > + return accdet->accdet_eint0; > + } > + ret = devm_request_threaded_irq(&pdev->dev, > + accdet->accdet_eint0, > + NULL, mtk_accdet_irq_handler_thread, > + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, > + "ACCDET_EINT0", accdet); > + if (ret) { > + dev_err(&pdev->dev, > + "Error: Get eint0 irq request failed (%d)\n", > + ret); > + return ret; > + } > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_EINT1)) { > + accdet->accdet_eint1 = platform_get_irq(pdev, 2); > + if (accdet->accdet_eint1 < 0) { > + dev_err(&pdev->dev, > + "Error: Get eint1 irq failed (%d)\n", > + accdet->accdet_eint1); > + return accdet->accdet_eint1; > + } > + ret = devm_request_threaded_irq(&pdev->dev, > + accdet->accdet_eint1, > + NULL, > + mtk_accdet_irq_handler_thread, > + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, > + "ACCDET_EINT1", accdet); > + if (ret) { > + dev_err(&pdev->dev, > + "Error: Get eint1 irq request failed (%d)\n", > + ret); > + return ret; > + } > + } else if (HAS_CAP(accdet->data->caps, ACCDET_PMIC_BI_EINT)) { > + accdet->accdet_eint0 = platform_get_irq(pdev, 1); > + if (accdet->accdet_eint0 < 0) { > + dev_err(&pdev->dev, > + "Error: Get eint0 irq failed (%d)\n", > + accdet->accdet_eint0); > + return accdet->accdet_eint0; > + } > + ret = devm_request_threaded_irq(&pdev->dev, > + accdet->accdet_eint0, > + NULL, > + mtk_accdet_irq_handler_thread, > + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, > + "ACCDET_EINT0", accdet); > + if (ret) { > + dev_err(&pdev->dev, > + "Error: Get eint0 irq request failed (%d)\n", > + ret); > + return ret; > + } > + accdet->accdet_eint1 = platform_get_irq(pdev, 2); > + if (accdet->accdet_eint1 < 0) { > + dev_err(&pdev->dev, > + "Error: Get eint1 irq failed (%d)\n", > + accdet->accdet_eint1); > + return accdet->accdet_eint1; > + } > + ret = devm_request_threaded_irq(&pdev->dev, > + accdet->accdet_eint1, > + NULL, > + mtk_accdet_irq_handler_thread, > + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, > + "ACCDET_EINT1", accdet); > + if (ret) { > + dev_err(&pdev->dev, > + "Error: Get eint1 irq request failed (%d)\n", > + ret); > + return ret; > + } > + } > + > + /* setup timer */ > + timer_setup(&micbias_timer, dis_micbias_timerhandler, 0); > + micbias_timer.expires = jiffies + MICBIAS_DISABLE_TIMER; > + timer_setup(&accdet_init_timer, delay_init_timerhandler, 0); > + accdet_init_timer.expires = jiffies + ACCDET_INIT_WAIT_TIMER; > + > + /* Create workqueue */ > + accdet->delay_init_workqueue = > + create_singlethread_workqueue("delay_init"); > + INIT_WORK(&accdet->delay_init_work, delay_init_work_callback); > + if (!accdet->delay_init_workqueue) { > + dev_err(&pdev->dev, "Error: Create dinit workqueue failed\n"); > + ret = -1; > + goto err_delay_init_wq; > + } > + accdet->accdet_workqueue = create_singlethread_workqueue("accdet"); > + INIT_WORK(&accdet->accdet_work, accdet_work_callback); > + if (!accdet->accdet_workqueue) { > + dev_err(&pdev->dev, "Error: Create accdet workqueue failed\n"); > + ret = -1; > + goto err_accdet_wq; > + } > + > + accdet->dis_micbias_workqueue = > + create_singlethread_workqueue("dismicQueue"); > + INIT_WORK(&accdet->dis_micbias_work, dis_micbias_work_callback); > + if (!accdet->dis_micbias_workqueue) { > + dev_err(&pdev->dev, "Error: Create dismic workqueue failed\n"); > + ret = -1; > + goto err_dis_micbias_wq; > + } > + accdet->eint_workqueue = create_singlethread_workqueue("accdet_eint"); > + INIT_WORK(&accdet->eint_work, eint_work_callback); > + if (!accdet->eint_workqueue) { > + dev_err(&pdev->dev, "Error: Create eint workqueue failed\n"); > + ret = -1; > + goto err_eint_wq; > + } > + if (HAS_CAP(accdet->data->caps, ACCDET_AP_GPIO_EINT)) { > + accdet->accdet_eint_type = IRQ_TYPE_LEVEL_LOW; > + ret = ext_eint_setup(pdev); > + if (ret) > + destroy_workqueue(accdet->eint_workqueue); > + } > + > + atomic_set(&accdet->init_once, 1); > + mod_timer(&accdet_init_timer, (jiffies + ACCDET_INIT_WAIT_TIMER)); > + > + return 0; > + > +err_eint_wq: > + destroy_workqueue(accdet->dis_micbias_workqueue); > +err_dis_micbias_wq: > + destroy_workqueue(accdet->accdet_workqueue); > +err_accdet_wq: > + destroy_workqueue(accdet->delay_init_workqueue); > +err_delay_init_wq: > + pr_notice("%s error. now exit.!\n", __func__); > + return ret; > +} > + > +static int mt6359_accdet_remove(struct platform_device *pdev) > +{ > + destroy_workqueue(accdet->eint_workqueue); > + destroy_workqueue(accdet->dis_micbias_workqueue); > + destroy_workqueue(accdet->accdet_workqueue); > + destroy_workqueue(accdet->delay_init_workqueue); > + devm_kfree(&pdev->dev, accdet); > + return 0; > +} > + > +static struct platform_driver mt6359_accdet_driver = { > + .probe = mt6359_accdet_probe, > + .remove = mt6359_accdet_remove, > + .driver = { > + .name = "pmic-codec-accdet", > + .of_match_table = mt6359_accdet_of_match, > + }, > +}; > + > +static int __init mt6359_accdet_soc_init(void) > +{ > + int ret = 0; > + > + ret = platform_driver_register(&mt6359_accdet_driver); > + if (ret) > + return -ENODEV; > + return 0; > +} > + > +static void __exit mt6359_accdet_soc_exit(void) > +{ > + platform_driver_unregister(&mt6359_accdet_driver); > +} > +module_init(mt6359_accdet_soc_init); > +module_exit(mt6359_accdet_soc_exit); > + > +/* Module information */ > +MODULE_DESCRIPTION("MT6359 ALSA SoC accdet driver"); > +MODULE_AUTHOR("Argus Lin <argus.lin(a)mediatek.com>"); > +MODULE_LICENSE("GPL v2"); > diff --git a/sound/soc/codecs/mt6359-accdet.h b/sound/soc/codecs/mt6359-accdet.h > new file mode 100644 > index 0000000..d081d69 > --- /dev/null > +++ b/sound/soc/codecs/mt6359-accdet.h > @@ -0,0 +1,136 @@ > +/* SPDX-License-Identifier: GPL-2.0 */ > +/* > + * Copyright (C) 2021 MediaTek Inc. > + * Author: Argus Lin <argus.lin(a)mediatek.com> > + */ > + > +#ifndef _ACCDET_H_ > +#define _ACCDET_H_ > + > +#include <linux/ctype.h> > +#include <linux/string.h> > + > +#define ACCDET_DEVNAME "accdet" > +/* SW ENV define */ > +#define NO_KEY (0x0) > +#define UP_KEY (0x01) > +#define MD_KEY (0x02) > +#define DW_KEY (0x04) > +#define AS_KEY (0x08) > + > +#define HEADSET_MODE_1 (1) > +#define HEADSET_MODE_2 (2) > +#define HEADSET_MODE_6 (6) > + > +/* IOCTL */ > +#define ACCDET_IOC_MAGIC 'A' > +#define ACCDET_INIT _IO(ACCDET_IOC_MAGIC, 0) > +#define SET_CALL_STATE _IO(ACCDET_IOC_MAGIC, 1) > +#define GET_BUTTON_STATUS _IO(ACCDET_IOC_MAGIC, 2) > + > +/* 400us, Accdet irq clear timeout */ > +#define ACCDET_TIME_OUT 0x61A80 > + > +/* cable type recognized by accdet, and report to WiredAccessoryManager */ > +enum accdet_report_state { > + NO_DEVICE = 0, > + HEADSET_MIC = 1, > + HEADSET_NO_MIC = 2, > + HEADSET_FIVE_POLE = 3, > + LINE_OUT_DEVICE = 4, > +}; > + > +/* accdet status got from accdet FSM */ > +enum accdet_status { > + PLUG_OUT = 0, > + MIC_BIAS = 1, > + HOOK_SWITCH = 2, > + BI_MIC_BIAS = 3, > + LINE_OUT = 4, > + STAND_BY = 5 > +}; > + > +enum accdet_eint_ID { > + NO_PMIC_EINT = 0, > + PMIC_EINT0 = 1, > + PMIC_EINT1 = 2, > + PMIC_BIEINT = 3, > +}; > + > +/* EINT state when moisture enable */ > +enum eint_moisture_status { > + M_PLUG_IN = 0, > + M_WATER_IN = 1, > + M_HP_PLUG_IN = 2, > + M_PLUG_OUT = 3, > + M_NO_ACT = 4, > + M_UNKNOWN = 5, > +}; > + > +struct three_key_threshold { > + unsigned int mid; > + unsigned int up; > + unsigned int down; > +}; > + > +struct four_key_threshold { > + unsigned int mid; > + unsigned int voice; > + unsigned int up; > + unsigned int down; > +}; > + > +struct pwm_deb_settings { > + unsigned int pwm_width; > + unsigned int pwm_thresh; > + unsigned int fall_delay; > + unsigned int rise_delay; > + unsigned int debounce0; > + unsigned int debounce1; > + unsigned int debounce3; > + unsigned int debounce4; > + unsigned int eint_pwm_width; > + unsigned int eint_pwm_thresh; > + unsigned int eint_debounce0; > + unsigned int eint_debounce1; > + unsigned int eint_debounce2; > + unsigned int eint_debounce3; > + unsigned int eint_inverter_debounce; > + > +}; > + > +struct head_dts_data { > + unsigned int mic_vol; > + unsigned int mic_mode; > + unsigned int plugout_deb; > + unsigned int eint_pol; > + struct pwm_deb_settings pwm_deb; > + struct three_key_threshold three_key; > + struct four_key_threshold four_key; > + unsigned int moisture_detect_enable; > + unsigned int eint_detect_mode; > + unsigned int eint_use_ext_res; > + unsigned int moisture_detect_mode; > + unsigned int moisture_comp_vth; > + unsigned int moisture_comp_vref2; > + unsigned int moisture_use_ext_res; > +}; > + > +enum { > + accdet_state000 = 0, > + accdet_state001, > + accdet_state010, > + accdet_state011, > + accdet_auxadc, > + eint_state000, > + eint_state001, > + eint_state010, > + eint_state011, > + eint_inverter_state000, > +}; > + > +/* just be called by audio module for DC trim */ > +void mt6359_accdet_late_init(unsigned long data); > +int mt6359_accdet_init(struct snd_soc_component *component, struct snd_soc_card *card); > +int mt6359_accdet_set_drvdata(struct snd_soc_card *card); > +#endif > diff --git a/sound/soc/codecs/mt6359.h b/sound/soc/codecs/mt6359.h > index 35f806b..9f9a94a 100644 > --- a/sound/soc/codecs/mt6359.h > +++ b/sound/soc/codecs/mt6359.h > @@ -8,128 +8,1779 @@ > #define _MT6359_H_ > > /*************Register Bit Define*************/ > -#define PMIC_ACCDET_IRQ_SHIFT 0 > -#define PMIC_ACCDET_EINT0_IRQ_SHIFT 2 > -#define PMIC_ACCDET_EINT1_IRQ_SHIFT 3 > -#define PMIC_ACCDET_IRQ_CLR_SHIFT 8 > -#define PMIC_ACCDET_EINT0_IRQ_CLR_SHIFT 10 > -#define PMIC_ACCDET_EINT1_IRQ_CLR_SHIFT 11 > -#define PMIC_RG_INT_STATUS_ACCDET_SHIFT 5 > -#define PMIC_RG_INT_STATUS_ACCDET_EINT0_SHIFT 6 > -#define PMIC_RG_INT_STATUS_ACCDET_EINT1_SHIFT 7 > -#define PMIC_RG_EINT0CONFIGACCDET_SHIFT 11 > -#define PMIC_RG_EINT1CONFIGACCDET_SHIFT 0 > -#define PMIC_ACCDET_EINT0_INVERTER_SW_EN_SHIFT 6 > -#define PMIC_ACCDET_EINT1_INVERTER_SW_EN_SHIFT 8 > -#define PMIC_RG_MTEST_EN_SHIFT 8 > -#define PMIC_RG_MTEST_SEL_SHIFT 9 > -#define PMIC_ACCDET_EINT0_M_SW_EN_SHIFT 10 > -#define PMIC_ACCDET_EINT1_M_SW_EN_SHIFT 11 > -#define PMIC_ACCDET_EINT0_CEN_STABLE_SHIFT 5 > -#define PMIC_ACCDET_EINT1_CEN_STABLE_SHIFT 10 > -#define PMIC_ACCDET_DA_STABLE_SHIFT 0 > -#define PMIC_ACCDET_EINT0_EN_STABLE_SHIFT 1 > -#define PMIC_ACCDET_EINT0_CMPEN_STABLE_SHIFT 2 > -#define PMIC_ACCDET_EINT1_EN_STABLE_SHIFT 6 > -#define PMIC_ACCDET_EINT1_CMPEN_STABLE_SHIFT 7 > -#define PMIC_ACCDET_EINT_CTURBO_SEL_SHIFT 7 > -#define PMIC_ACCDET_EINT0_CTURBO_SW_SHIFT 7 > -#define PMIC_RG_EINTCOMPVTH_SHIFT 4 > -#define PMIC_RG_EINT0HIRENB_SHIFT 12 > -#define PMIC_RG_EINT0NOHYS_SHIFT 10 > -#define PMIC_ACCDET_SW_EN_SHIFT 0 > -#define PMIC_ACCDET_EINT0_MEM_IN_SHIFT 6 > -#define PMIC_ACCDET_MEM_IN_SHIFT 6 > -#define PMIC_ACCDET_EINT_DEBOUNCE0_SHIFT 0 > -#define PMIC_ACCDET_EINT_DEBOUNCE1_SHIFT 4 > -#define PMIC_ACCDET_EINT_DEBOUNCE2_SHIFT 8 > -#define PMIC_ACCDET_EINT_DEBOUNCE3_SHIFT 12 > -#define PMIC_RG_ACCDET2AUXSWEN_SHIFT 14 > -#define PMIC_AUDACCDETAUXADCSWCTRL_SEL_SHIFT 9 > -#define PMIC_AUDACCDETAUXADCSWCTRL_SW_SHIFT 10 > -#define PMIC_RG_EINT0CTURBO_SHIFT 5 > -#define PMIC_RG_EINT1CTURBO_SHIFT 13 > -#define PMIC_ACCDET_EINT_M_PLUG_IN_NUM_SHIFT 12 > -#define PMIC_ACCDET_EINT_M_DETECT_EN_SHIFT 12 > -#define PMIC_ACCDET_EINT0_SW_EN_SHIFT 2 > -#define PMIC_ACCDET_EINT1_SW_EN_SHIFT 4 > -#define PMIC_ACCDET_EINT_CMPMOUT_SEL_SHIFT 12 > -#define PMIC_ACCDET_EINT_CMPMEN_SEL_SHIFT 6 > -#define PMIC_RG_HPLOUTPUTSTBENH_VAUDP32_SHIFT 0 > -#define PMIC_RG_HPROUTPUTSTBENH_VAUDP32_SHIFT 4 > -#define PMIC_RG_EINT0EN_SHIFT 2 > -#define PMIC_RG_EINT1EN_SHIFT 10 > -#define PMIC_RG_NCP_PDDIS_EN_SHIFT 0 > -#define PMIC_RG_ACCDETSPARE_SHIFT 0 > -#define PMIC_RG_ACCDET_RST_SHIFT 1 > -#define PMIC_RG_AUDMICBIAS1HVEN_SHIFT 12 > -#define PMIC_RG_AUDMICBIAS1VREF_SHIFT 4 > -#define PMIC_RG_ANALOGFDEN_SHIFT 12 > -#define PMIC_RG_AUDMICBIAS1DCSW1PEN_SHIFT 8 > -#define PMIC_RG_AUDMICBIAS1LOWPEN_SHIFT 2 > -#define PMIC_ACCDET_SEQ_INIT_SHIFT 1 > -#define PMIC_RG_EINTCOMPVTH_MASK 0xf > -#define PMIC_ACCDET_EINT0_MEM_IN_MASK 0x3 > -#define PMIC_ACCDET_EINT_DEBOUNCE0_MASK 0xf > -#define PMIC_ACCDET_EINT_DEBOUNCE1_MASK 0xf > -#define PMIC_ACCDET_EINT_DEBOUNCE2_MASK 0xf > -#define PMIC_ACCDET_EINT_DEBOUNCE3_MASK 0xf > -#define PMIC_ACCDET_EINT0_IRQ_SHIFT 2 > -#define PMIC_ACCDET_EINT1_IRQ_SHIFT 3 > - > -/* AUDENC_ANA_CON16: */ > -#define RG_AUD_MICBIAS1_LOWP_EN BIT(PMIC_RG_AUDMICBIAS1LOWPEN_SHIFT) > - > +#define MT6359_TOP0_ID 0x0 > +#define MT6359_SMT_CON1 0x32 > +#define MT6359_DRV_CON2 0x3c > +#define MT6359_DRV_CON3 0x3e > +#define MT6359_DRV_CON4 0x40 > +#define MT6359_TOP_CKPDN_CON0 0x10c > +#define MT6359_TOP_CKPDN_CON0_SET 0x10e > +#define MT6359_TOP_CKPDN_CON0_CLR 0x110 > +#define MT6359_AUXADC_RQST0 0x1108 > +#define MT6359_AUXADC_CON10 0x11a0 > +#define MT6359_AUXADC_ACCDET 0x11ba > +#define MT6359_LDO_VUSB_OP_EN 0x1d0c > +#define MT6359_LDO_VUSB_OP_EN_SET 0x1d0e > +#define MT6359_LDO_VUSB_OP_EN_CLR 0x1d10 > +#define MT6359_AUD_TOP_CKPDN_CON0 0x230c > +#define MT6359_AUD_TOP_CKPDN_CON0_SET 0x230e > +#define MT6359_AUD_TOP_CKPDN_CON0_CLR 0x2310 > +#define MT6359_AUD_TOP_RST_CON0 0x2320 > +#define MT6359_AUD_TOP_RST_CON0_SET 0x2322 > +#define MT6359_AUD_TOP_RST_CON0_CLR 0x2324 > +#define MT6359_AUD_TOP_INT_CON0 0x2328 > +#define MT6359_AUD_TOP_INT_CON0_SET 0x232a > +#define MT6359_AUD_TOP_INT_CON0_CLR 0x232c > +#define MT6359_AUD_TOP_INT_MASK_CON0 0x232e > +#define MT6359_AUD_TOP_INT_MASK_CON0_SET 0x2330 > +#define MT6359_AUD_TOP_INT_MASK_CON0_CLR 0x2332 > +#define MT6359_AUD_TOP_INT_STATUS0 0x2334 > +#define MT6359_AFE_NCP_CFG2 0x24e2 > +#define MT6359_AUDENC_DSN_ID 0x2500 > +#define MT6359_AUDENC_DSN_REV0 0x2502 > +#define MT6359_AUDENC_DSN_DBI 0x2504 > +#define MT6359_AUDENC_DSN_FPI 0x2506 > +#define MT6359_AUDENC_ANA_CON0 0x2508 > +#define MT6359_AUDENC_ANA_CON1 0x250a > +#define MT6359_AUDENC_ANA_CON2 0x250c > +#define MT6359_AUDENC_ANA_CON3 0x250e > +#define MT6359_AUDENC_ANA_CON4 0x2510 > +#define MT6359_AUDENC_ANA_CON5 0x2512 > +#define MT6359_AUDENC_ANA_CON6 0x2514 > +#define MT6359_AUDENC_ANA_CON7 0x2516 > +#define MT6359_AUDENC_ANA_CON8 0x2518 > +#define MT6359_AUDENC_ANA_CON9 0x251a > +#define MT6359_AUDENC_ANA_CON10 0x251c > +#define MT6359_AUDENC_ANA_CON11 0x251e > +#define MT6359_AUDENC_ANA_CON12 0x2520 > +#define MT6359_AUDENC_ANA_CON13 0x2522 > +#define MT6359_AUDENC_ANA_CON14 0x2524 > +#define MT6359_AUDENC_ANA_CON15 0x2526 > +#define MT6359_AUDENC_ANA_CON16 0x2528 > +#define MT6359_AUDENC_ANA_CON17 0x252a > +#define MT6359_AUDENC_ANA_CON18 0x252c > +#define MT6359_AUDENC_ANA_CON19 0x252e > +#define MT6359_AUDENC_ANA_CON20 0x2530 > +#define MT6359_AUDENC_ANA_CON21 0x2532 > +#define MT6359_AUDENC_ANA_CON22 0x2534 > +#define MT6359_AUDENC_ANA_CON23 0x2536 > +#define MT6359_AUDDEC_DSN_ID 0x2580 > +#define MT6359_AUDDEC_DSN_REV0 0x2582 > +#define MT6359_AUDDEC_DSN_DBI 0x2584 > +#define MT6359_AUDDEC_DSN_FPI 0x2586 > +#define MT6359_AUDDEC_ANA_CON0 0x2588 > +#define MT6359_AUDDEC_ANA_CON1 0x258a > +#define MT6359_AUDDEC_ANA_CON2 0x258c > +#define MT6359_AUDDEC_ANA_CON3 0x258e > +#define MT6359_AUDDEC_ANA_CON4 0x2590 > +#define MT6359_AUDDEC_ANA_CON5 0x2592 > +#define MT6359_AUDDEC_ANA_CON6 0x2594 > +#define MT6359_AUDDEC_ANA_CON7 0x2596 > +#define MT6359_AUDDEC_ANA_CON8 0x2598 > +#define MT6359_AUDDEC_ANA_CON9 0x259a > +#define MT6359_AUDDEC_ANA_CON10 0x259c > +#define MT6359_AUDDEC_ANA_CON11 0x259e > +#define MT6359_AUDDEC_ANA_CON12 0x25a0 > +#define MT6359_AUDDEC_ANA_CON13 0x25a2 > +#define MT6359_AUDDEC_ANA_CON14 0x25a4 > +#define MT6359_ACCDET_DSN_DIG_ID 0x2680 > +#define MT6359_ACCDET_DSN_DIG_REV0 0x2682 > +#define MT6359_ACCDET_DSN_DBI 0x2684 > +#define MT6359_ACCDET_DSN_FPI 0x2686 > +#define MT6359_ACCDET_CON0 0x2688 > +#define MT6359_ACCDET_CON1 0x268a > +#define MT6359_ACCDET_CON2 0x268c > +#define MT6359_ACCDET_CON3 0x268e > +#define MT6359_ACCDET_CON4 0x2690 > +#define MT6359_ACCDET_CON5 0x2692 > +#define MT6359_ACCDET_CON6 0x2694 > +#define MT6359_ACCDET_CON7 0x2696 > +#define MT6359_ACCDET_CON8 0x2698 > +#define MT6359_ACCDET_CON9 0x269a > +#define MT6359_ACCDET_CON10 0x269c > +#define MT6359_ACCDET_CON11 0x269e > +#define MT6359_ACCDET_CON12 0x26a0 > +#define MT6359_ACCDET_CON13 0x26a2 > +#define MT6359_ACCDET_CON14 0x26a4 > +#define MT6359_ACCDET_CON15 0x26a6 > +#define MT6359_ACCDET_CON16 0x26a8 > +#define MT6359_ACCDET_CON17 0x26aa > +#define MT6359_ACCDET_CON18 0x26ac > +#define MT6359_ACCDET_CON19 0x26ae > +#define MT6359_ACCDET_CON20 0x26b0 > +#define MT6359_ACCDET_CON21 0x26b2 > +#define MT6359_ACCDET_CON22 0x26b4 > +#define MT6359_ACCDET_CON23 0x26b6 > +#define MT6359_ACCDET_CON24 0x26b8 > +#define MT6359_ACCDET_CON25 0x26ba > +#define MT6359_ACCDET_CON26 0x26bc > +#define MT6359_ACCDET_CON27 0x26be > +#define MT6359_ACCDET_CON28 0x26c0 > +#define MT6359_ACCDET_CON29 0x26c2 > +#define MT6359_ACCDET_CON30 0x26c4 > +#define MT6359_ACCDET_CON31 0x26c6 > +#define MT6359_ACCDET_CON32 0x26c8 > +#define MT6359_ACCDET_CON33 0x26ca > +#define MT6359_ACCDET_CON34 0x26cc > +#define MT6359_ACCDET_CON35 0x26ce > +#define MT6359_ACCDET_CON36 0x26d0 > +#define MT6359_ACCDET_CON37 0x26d2 > +#define MT6359_ACCDET_CON38 0x26d4 > +#define MT6359_ACCDET_CON39 0x26d6 > +#define MT6359_ACCDET_CON40 0x26d8 > + > +#define TOP0_ANA_ID_ADDR \ > + MT6359_TOP0_ID > +#define TOP0_ANA_ID_SFT 0 > +#define TOP0_ANA_ID_MASK 0xFF > +#define TOP0_ANA_ID_MASK_SFT (0xFF << 0) > +#define AUXADC_RQST_CH0_ADDR \ > + MT6359_AUXADC_RQST0 > +#define AUXADC_RQST_CH0_SFT 0 > +#define AUXADC_RQST_CH0_MASK 0x1 > +#define AUXADC_RQST_CH0_MASK_SFT (0x1 << 0) > +#define AUXADC_ACCDET_ANASWCTRL_EN_ADDR \ > + MT6359_AUXADC_CON15 > +#define AUXADC_ACCDET_ANASWCTRL_EN_SFT 6 > +#define AUXADC_ACCDET_ANASWCTRL_EN_MASK 0x1 > +#define AUXADC_ACCDET_ANASWCTRL_EN_MASK_SFT (0x1 << 6) > + > +#define AUXADC_ACCDET_AUTO_SPL_ADDR \ > + MT6359_AUXADC_ACCDET > +#define AUXADC_ACCDET_AUTO_SPL_SFT 0 > +#define AUXADC_ACCDET_AUTO_SPL_MASK 0x1 > +#define AUXADC_ACCDET_AUTO_SPL_MASK_SFT (0x1 << 0) > +#define AUXADC_ACCDET_AUTO_RQST_CLR_ADDR \ > + MT6359_AUXADC_ACCDET > +#define AUXADC_ACCDET_AUTO_RQST_CLR_SFT 1 > +#define AUXADC_ACCDET_AUTO_RQST_CLR_MASK 0x1 > +#define AUXADC_ACCDET_AUTO_RQST_CLR_MASK_SFT (0x1 << 1) > +#define AUXADC_ACCDET_DIG1_RSV0_ADDR \ > + MT6359_AUXADC_ACCDET > +#define AUXADC_ACCDET_DIG1_RSV0_SFT 2 > +#define AUXADC_ACCDET_DIG1_RSV0_MASK 0x3F > +#define AUXADC_ACCDET_DIG1_RSV0_MASK_SFT (0x3F << 2) > +#define AUXADC_ACCDET_DIG0_RSV0_ADDR \ > + MT6359_AUXADC_ACCDET > +#define AUXADC_ACCDET_DIG0_RSV0_SFT 8 > +#define AUXADC_ACCDET_DIG0_RSV0_MASK 0xFF > +#define AUXADC_ACCDET_DIG0_RSV0_MASK_SFT (0xFF << 8) > + > +#define RG_ACCDET_CK_PDN_ADDR \ > + MT6359_AUD_TOP_CKPDN_CON0 > +#define RG_ACCDET_CK_PDN_SFT 0 > +#define RG_ACCDET_CK_PDN_MASK 0x1 > +#define RG_ACCDET_CK_PDN_MASK_SFT (0x1 << 0) > + > +#define RG_ACCDET_RST_ADDR \ > + MT6359_AUD_TOP_RST_CON0 > +#define RG_ACCDET_RST_SFT 1 > +#define RG_ACCDET_RST_MASK 0x1 > +#define RG_ACCDET_RST_MASK_SFT (0x1 << 1) > +#define BANK_ACCDET_SWRST_ADDR \ > + MT6359_AUD_TOP_RST_BANK_CON0 > +#define BANK_ACCDET_SWRST_SFT 0 > +#define BANK_ACCDET_SWRST_MASK 0x1 > +#define BANK_ACCDET_SWRST_MASK_SFT (0x1 << 0) > + > +#define RG_INT_EN_ACCDET_ADDR \ > + MT6359_AUD_TOP_INT_CON0 > +#define RG_INT_EN_ACCDET_SFT 5 > +#define RG_INT_EN_ACCDET_MASK 0x1 > +#define RG_INT_EN_ACCDET_MASK_SFT (0x1 << 5) > +#define RG_INT_EN_ACCDET_EINT0_ADDR \ > + MT6359_AUD_TOP_INT_CON0 > +#define RG_INT_EN_ACCDET_EINT0_SFT 6 > +#define RG_INT_EN_ACCDET_EINT0_MASK 0x1 > +#define RG_INT_EN_ACCDET_EINT0_MASK_SFT (0x1 << 6) > +#define RG_INT_EN_ACCDET_EINT1_ADDR \ > + MT6359_AUD_TOP_INT_CON0 > +#define RG_INT_EN_ACCDET_EINT1_SFT 7 > +#define RG_INT_EN_ACCDET_EINT1_MASK 0x1 > +#define RG_INT_EN_ACCDET_EINT1_MASK_SFT (0x1 << 7) > + > +#define RG_INT_MASK_ACCDET_ADDR \ > + MT6359_AUD_TOP_INT_MASK_CON0 > +#define RG_INT_MASK_ACCDET_SFT 5 > +#define RG_INT_MASK_ACCDET_MASK 0x1 > +#define RG_INT_MASK_ACCDET_MASK_SFT (0x1 << 5) > +#define RG_INT_MASK_ACCDET_EINT0_ADDR \ > + MT6359_AUD_TOP_INT_MASK_CON0 > +#define RG_INT_MASK_ACCDET_EINT0_SFT 6 > +#define RG_INT_MASK_ACCDET_EINT0_MASK 0x1 > +#define RG_INT_MASK_ACCDET_EINT0_MASK_SFT (0x1 << 6) > +#define RG_INT_MASK_ACCDET_EINT1_ADDR \ > + MT6359_AUD_TOP_INT_MASK_CON0 > +#define RG_INT_MASK_ACCDET_EINT1_SFT 7 > +#define RG_INT_MASK_ACCDET_EINT1_MASK 0x1 > +#define RG_INT_MASK_ACCDET_EINT1_MASK_SFT (0x1 << 7) > + > +#define RG_INT_STATUS_ACCDET_ADDR \ > + MT6359_AUD_TOP_INT_STATUS0 > +#define RG_INT_STATUS_ACCDET_SFT 5 > +#define RG_INT_STATUS_ACCDET_MASK 0x1 > +#define RG_INT_STATUS_ACCDET_MASK_SFT (0x1 << 5) > +#define RG_INT_STATUS_ACCDET_EINT0_ADDR \ > + MT6359_AUD_TOP_INT_STATUS0 > +#define RG_INT_STATUS_ACCDET_EINT0_SFT 6 > +#define RG_INT_STATUS_ACCDET_EINT0_MASK 0x1 > +#define RG_INT_STATUS_ACCDET_EINT0_MASK_SFT (0x1 << 6) > +#define RG_INT_STATUS_ACCDET_EINT1_ADDR \ > + MT6359_AUD_TOP_INT_STATUS0 > +#define RG_INT_STATUS_ACCDET_EINT1_SFT 7 > +#define RG_INT_STATUS_ACCDET_EINT1_MASK 0x1 > +#define RG_INT_STATUS_ACCDET_EINT1_MASK_SFT (0x1 << 7) > + > +#define RG_INT_RAW_STATUS_ACCDET_ADDR \ > + MT6359_AUD_TOP_INT_RAW_STATUS0 > +#define RG_INT_RAW_STATUS_ACCDET_SFT 5 > +#define RG_INT_RAW_STATUS_ACCDET_MASK 0x1 > +#define RG_INT_RAW_STATUS_ACCDET_MASK_SFT (0x1 << 5) > +#define RG_INT_RAW_STATUS_ACCDET_EINT0_ADDR \ > + MT6359_AUD_TOP_INT_RAW_STATUS0 > +#define RG_INT_RAW_STATUS_ACCDET_EINT0_SFT 6 > +#define RG_INT_RAW_STATUS_ACCDET_EINT0_MASK 0x1 > +#define RG_INT_RAW_STATUS_ACCDET_EINT0_MASK_SFT (0x1 << 6) > +#define RG_INT_RAW_STATUS_ACCDET_EINT1_ADDR \ > + MT6359_AUD_TOP_INT_RAW_STATUS0 > +#define RG_INT_RAW_STATUS_ACCDET_EINT1_SFT 7 > +#define RG_INT_RAW_STATUS_ACCDET_EINT1_MASK 0x1 > +#define RG_INT_RAW_STATUS_ACCDET_EINT1_MASK_SFT (0x1 << 7) > + > +#define RG_AUDACCDETMICBIAS0PULLLOW_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_AUDACCDETMICBIAS0PULLLOW_SFT 0 > +#define RG_AUDACCDETMICBIAS0PULLLOW_MASK 0x1 > +#define RG_AUDACCDETMICBIAS0PULLLOW_MASK_SFT (0x1 << 0) > +#define RG_AUDACCDETMICBIAS1PULLLOW_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_AUDACCDETMICBIAS1PULLLOW_SFT 1 > +#define RG_AUDACCDETMICBIAS1PULLLOW_MASK 0x1 > +#define RG_AUDACCDETMICBIAS1PULLLOW_MASK_SFT (0x1 << 1) > +#define RG_AUDACCDETMICBIAS2PULLLOW_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_AUDACCDETMICBIAS2PULLLOW_SFT 2 > +#define RG_AUDACCDETMICBIAS2PULLLOW_MASK 0x1 > +#define RG_AUDACCDETMICBIAS2PULLLOW_MASK_SFT (0x1 << 2) > +#define RG_AUDACCDETVIN1PULLLOW_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_AUDACCDETVIN1PULLLOW_SFT 3 > +#define RG_AUDACCDETVIN1PULLLOW_MASK 0x1 > +#define RG_AUDACCDETVIN1PULLLOW_MASK_SFT (0x1 << 3) > +#define RG_AUDACCDETVTHACAL_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_AUDACCDETVTHACAL_SFT 4 > +#define RG_AUDACCDETVTHACAL_MASK 0x1 > +#define RG_AUDACCDETVTHACAL_MASK_SFT (0x1 << 4) > +#define RG_AUDACCDETVTHBCAL_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_AUDACCDETVTHBCAL_SFT 5 > +#define RG_AUDACCDETVTHBCAL_MASK 0x1 > +#define RG_AUDACCDETVTHBCAL_MASK_SFT (0x1 << 5) > +#define RG_AUDACCDETTVDET_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_AUDACCDETTVDET_SFT 6 > +#define RG_AUDACCDETTVDET_MASK 0x1 > +#define RG_AUDACCDETTVDET_MASK_SFT (0x1 << 6) > +#define RG_ACCDETSEL_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_ACCDETSEL_SFT 7 > +#define RG_ACCDETSEL_MASK 0x1 > +#define RG_ACCDETSEL_MASK_SFT (0x1 << 7) > + > +#define RG_AUDPWDBMICBIAS1_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_AUDPWDBMICBIAS1_SFT 0 > +#define RG_AUDPWDBMICBIAS1_MASK 0x1 > +#define RG_AUDPWDBMICBIAS1_MASK_SFT (0x1 << 0) > +#define RG_AUDMICBIAS1BYPASSEN_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_AUDMICBIAS1BYPASSEN_SFT 1 > +#define RG_AUDMICBIAS1BYPASSEN_MASK 0x1 > +#define RG_AUDMICBIAS1BYPASSEN_MASK_SFT (0x1 << 1) > +#define RG_AUDMICBIAS1LOWPEN_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_AUDMICBIAS1LOWPEN_SFT 2 > +#define RG_AUDMICBIAS1LOWPEN_MASK 0x1 > +#define RG_AUDMICBIAS1LOWPEN_MASK_SFT (0x1 << 2) > +#define RG_AUDMICBIAS1VREF_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_AUDMICBIAS1VREF_SFT 4 > +#define RG_AUDMICBIAS1VREF_MASK 0x7 > +#define RG_AUDMICBIAS1VREF_MASK_SFT (0x7 << 4) > +#define RG_AUDMICBIAS1DCSW1PEN_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_AUDMICBIAS1DCSW1PEN_SFT 8 > +#define RG_AUDMICBIAS1DCSW1PEN_MASK 0x1 > +#define RG_AUDMICBIAS1DCSW1PEN_MASK_SFT (0x1 << 8) > +#define RG_AUDMICBIAS1DCSW1NEN_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_AUDMICBIAS1DCSW1NEN_SFT 9 > +#define RG_AUDMICBIAS1DCSW1NEN_MASK 0x1 > +#define RG_AUDMICBIAS1DCSW1NEN_MASK_SFT (0x1 << 9) > +#define RG_BANDGAPGEN_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_BANDGAPGEN_SFT 10 > +#define RG_BANDGAPGEN_MASK 0x1 > +#define RG_BANDGAPGEN_MASK_SFT (0x1 << 10) > +#define RG_AUDMICBIAS1HVEN_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_AUDMICBIAS1HVEN_SFT 12 > +#define RG_AUDMICBIAS1HVEN_MASK 0x1 > +#define RG_AUDMICBIAS1HVEN_MASK_SFT (0x1 << 12) > +#define RG_AUDMICBIAS1HVVREF_ADDR \ > + MT6359_AUDENC_ANA_CON16 > +#define RG_AUDMICBIAS1HVVREF_SFT 13 > +#define RG_AUDMICBIAS1HVVREF_MASK 0x1 > +#define RG_AUDMICBIAS1HVVREF_MASK_SFT (0x1 << 13) > + > +#define RG_EINT0NOHYS_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_EINT0NOHYS_SFT 10 > +#define RG_EINT0NOHYS_MASK 0x1 > +#define RG_EINT0NOHYS_MASK_SFT (0x1 << 10) > +#define RG_EINT0CONFIGACCDET_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_EINT0CONFIGACCDET_SFT 11 > +#define RG_EINT0CONFIGACCDET_MASK 0x1 > +#define RG_EINT0CONFIGACCDET_MASK_SFT (0x1 << 11) > +#define RG_EINT0HIRENB_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_EINT0HIRENB_SFT 12 > +#define RG_EINT0HIRENB_MASK 0x1 > +#define RG_EINT0HIRENB_MASK_SFT (0x1 << 12) > +#define RG_ACCDET2AUXRESBYPASS_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_ACCDET2AUXRESBYPASS_SFT 13 > +#define RG_ACCDET2AUXRESBYPASS_MASK 0x1 > +#define RG_ACCDET2AUXRESBYPASS_MASK_SFT (0x1 << 13) > +#define RG_ACCDET2AUXSWEN_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_ACCDET2AUXSWEN_SFT 14 > +#define RG_ACCDET2AUXSWEN_MASK 0x1 > +#define RG_ACCDET2AUXSWEN_MASK_SFT (0x1 << 14) > +#define RG_AUDACCDETMICBIAS3PULLLOW_ADDR \ > + MT6359_AUDENC_ANA_CON18 > +#define RG_AUDACCDETMICBIAS3PULLLOW_SFT 15 > +#define RG_AUDACCDETMICBIAS3PULLLOW_MASK 0x1 > +#define RG_AUDACCDETMICBIAS3PULLLOW_MASK_SFT (0x1 << 15) > +#define RG_EINT1CONFIGACCDET_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_EINT1CONFIGACCDET_SFT 0 > +#define RG_EINT1CONFIGACCDET_MASK 0x1 > +#define RG_EINT1CONFIGACCDET_MASK_SFT (0x1 << 0) > +#define RG_EINT1HIRENB_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_EINT1HIRENB_SFT 1 > +#define RG_EINT1HIRENB_MASK 0x1 > +#define RG_EINT1HIRENB_MASK_SFT (0x1 << 1) > +#define RG_EINT1NOHYS_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_EINT1NOHYS_SFT 2 > +#define RG_EINT1NOHYS_MASK 0x1 > +#define RG_EINT1NOHYS_MASK_SFT (0x1 << 2) > +#define RG_EINTCOMPVTH_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_MTEST_EN_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_MTEST_EN_SFT 8 > +#define RG_MTEST_EN_MASK 0x1 > +#define RG_MTEST_EN_MASK_SFT (0x1 << 8) > +#define RG_MTEST_SEL_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_MTEST_SEL_SFT 9 > +#define RG_MTEST_SEL_MASK 0x1 > +#define RG_MTEST_SEL_MASK_SFT (0x1 << 9) > +#define RG_MTEST_CURRENT_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_MTEST_CURRENT_SFT 10 > +#define RG_MTEST_CURRENT_MASK 0x1 > +#define RG_MTEST_CURRENT_MASK_SFT (0x1 << 10) > +#define RG_ANALOGFDEN_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_ANALOGFDEN_SFT 12 > +#define RG_ANALOGFDEN_MASK 0x1 > +#define RG_ANALOGFDEN_MASK_SFT (0x1 << 12) > +#define RG_FDVIN1PPULLLOW_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_FDVIN1PPULLLOW_SFT 13 > +#define RG_FDVIN1PPULLLOW_MASK 0x1 > +#define RG_FDVIN1PPULLLOW_MASK_SFT (0x1 << 13) > +#define RG_FDEINT0TYPE_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_FDEINT0TYPE_SFT 14 > +#define RG_FDEINT0TYPE_MASK 0x1 > +#define RG_FDEINT0TYPE_MASK_SFT (0x1 << 14) > +#define RG_FDEINT1TYPE_ADDR \ > + MT6359_AUDENC_ANA_CON19 > +#define RG_FDEINT1TYPE_SFT 15 > +#define RG_FDEINT1TYPE_MASK 0x1 > +#define RG_FDEINT1TYPE_MASK_SFT (0x1 << 15) > +#define RG_EINT0CMPEN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT0CMPEN_SFT 0 > +#define RG_EINT0CMPEN_MASK 0x1 > +#define RG_EINT0CMPEN_MASK_SFT (0x1 << 0) > +#define RG_EINT0CMPMEN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT0CMPMEN_SFT 1 > +#define RG_EINT0CMPMEN_MASK 0x1 > +#define RG_EINT0CMPMEN_MASK_SFT (0x1 << 1) > +#define RG_EINT0EN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT0EN_SFT 2 > +#define RG_EINT0EN_MASK 0x1 > +#define RG_EINT0EN_MASK_SFT (0x1 << 2) > +#define RG_EINT0CEN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT0CEN_SFT 3 > +#define RG_EINT0CEN_MASK 0x1 > +#define RG_EINT0CEN_MASK_SFT (0x1 << 3) > +#define RG_EINT0INVEN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT0INVEN_SFT 4 > +#define RG_EINT0INVEN_MASK 0x1 > +#define RG_EINT0INVEN_MASK_SFT (0x1 << 4) > +#define RG_EINT0CTURBO_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT0CTURBO_SFT 5 > +#define RG_EINT0CTURBO_MASK 0x7 > +#define RG_EINT0CTURBO_MASK_SFT (0x7 << 5) > +#define RG_EINT1CMPEN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT1CMPEN_SFT 8 > +#define RG_EINT1CMPEN_MASK 0x1 > +#define RG_EINT1CMPEN_MASK_SFT (0x1 << 8) > +#define RG_EINT1CMPMEN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT1CMPMEN_SFT 9 > +#define RG_EINT1CMPMEN_MASK 0x1 > +#define RG_EINT1CMPMEN_MASK_SFT (0x1 << 9) > +#define RG_EINT1EN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT1EN_SFT 10 > +#define RG_EINT1EN_MASK 0x1 > +#define RG_EINT1EN_MASK_SFT (0x1 << 10) > +#define RG_EINT1CEN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT1CEN_SFT 11 > +#define RG_EINT1CEN_MASK 0x1 > +#define RG_EINT1CEN_MASK_SFT (0x1 << 11) > +#define RG_EINT1INVEN_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT1INVEN_SFT 12 > +#define RG_EINT1INVEN_MASK 0x1 > +#define RG_EINT1INVEN_MASK_SFT (0x1 << 12) > +#define RG_EINT1CTURBO_ADDR \ > + MT6359_AUDENC_ANA_CON20 > +#define RG_EINT1CTURBO_SFT 13 > +#define RG_EINT1CTURBO_MASK 0x7 > +#define RG_EINT1CTURBO_MASK_SFT (0x7 << 13) > +#define RG_ACCDETSPARE_ADDR \ > + MT6359_AUDENC_ANA_CON21 > + > +#define ACCDET_ANA_ID_ADDR \ > + MT6359_ACCDET_DSN_DIG_ID > +#define ACCDET_ANA_ID_SFT 0 > +#define ACCDET_ANA_ID_MASK 0xFF > +#define ACCDET_ANA_ID_MASK_SFT (0xFF << 0) > +#define ACCDET_DIG_ID_ADDR \ > + MT6359_ACCDET_DSN_DIG_ID > +#define ACCDET_DIG_ID_SFT 8 > +#define ACCDET_DIG_ID_MASK 0xFF > +#define ACCDET_DIG_ID_MASK_SFT (0xFF << 8) > +#define ACCDET_ANA_MINOR_REV_ADDR \ > + MT6359_ACCDET_DSN_DIG_REV0 > +#define ACCDET_ANA_MINOR_REV_SFT 0 > +#define ACCDET_ANA_MINOR_REV_MASK 0xF > +#define ACCDET_ANA_MINOR_REV_MASK_SFT (0xF << 0) > +#define ACCDET_ANA_MAJOR_REV_ADDR \ > + MT6359_ACCDET_DSN_DIG_REV0 > +#define ACCDET_ANA_MAJOR_REV_SFT 4 > +#define ACCDET_ANA_MAJOR_REV_MASK 0xF > +#define ACCDET_ANA_MAJOR_REV_MASK_SFT (0xF << 4) > +#define ACCDET_DIG_MINOR_REV_ADDR \ > + MT6359_ACCDET_DSN_DIG_REV0 > +#define ACCDET_DIG_MINOR_REV_SFT 8 > +#define ACCDET_DIG_MINOR_REV_MASK 0xF > +#define ACCDET_DIG_MINOR_REV_MASK_SFT (0xF << 8) > +#define ACCDET_DIG_MAJOR_REV_ADDR \ > + MT6359_ACCDET_DSN_DIG_REV0 > +#define ACCDET_DIG_MAJOR_REV_SFT 12 > +#define ACCDET_DIG_MAJOR_REV_MASK 0xF > +#define ACCDET_DIG_MAJOR_REV_MASK_SFT (0xF << 12) > +#define ACCDET_DSN_CBS_ADDR \ > + MT6359_ACCDET_DSN_DBI > +#define ACCDET_DSN_CBS_SFT 0 > +#define ACCDET_DSN_CBS_MASK 0x3 > +#define ACCDET_DSN_CBS_MASK_SFT (0x3 << 0) > +#define ACCDET_DSN_BIX_ADDR \ > + MT6359_ACCDET_DSN_DBI > +#define ACCDET_DSN_BIX_SFT 2 > +#define ACCDET_DSN_BIX_MASK 0x3 > +#define ACCDET_DSN_BIX_MASK_SFT (0x3 << 2) > +#define ACCDET_ESP_ADDR \ > + MT6359_ACCDET_DSN_DBI > +#define ACCDET_ESP_SFT 8 > +#define ACCDET_ESP_MASK 0xFF > +#define ACCDET_ESP_MASK_SFT (0xFF << 8) > +#define ACCDET_DSN_FPI_ADDR \ > + MT6359_ACCDET_DSN_FPI > +#define ACCDET_DSN_FPI_SFT 0 > +#define ACCDET_DSN_FPI_MASK 0xFF > +#define ACCDET_DSN_FPI_MASK_SFT (0xFF << 0) > +#define ACCDET_AUXADC_SEL_ADDR \ > + MT6359_ACCDET_CON0 > +#define ACCDET_AUXADC_SEL_SFT 0 > +#define ACCDET_AUXADC_SEL_MASK 0x1 > +#define ACCDET_AUXADC_SEL_MASK_SFT (0x1 << 0) > +#define ACCDET_AUXADC_SW_ADDR \ > + MT6359_ACCDET_CON0 > +#define ACCDET_AUXADC_SW_SFT 1 > +#define ACCDET_AUXADC_SW_MASK 0x1 > +#define ACCDET_AUXADC_SW_MASK_SFT (0x1 << 1) > +#define ACCDET_TEST_AUXADC_ADDR \ > + MT6359_ACCDET_CON0 > +#define ACCDET_TEST_AUXADC_SFT 2 > +#define ACCDET_TEST_AUXADC_MASK 0x1 > +#define ACCDET_TEST_AUXADC_MASK_SFT (0x1 << 2) > +#define ACCDET_AUXADC_ANASWCTRL_SEL_ADDR \ > + MT6359_ACCDET_CON0 > +#define ACCDET_AUXADC_ANASWCTRL_SEL_SFT 8 > +#define ACCDET_AUXADC_ANASWCTRL_SEL_MASK 0x1 > +#define ACCDET_AUXADC_ANASWCTRL_SEL_MASK_SFT (0x1 << 8) > +#define AUDACCDETAUXADCSWCTRL_SEL_ADDR \ > + MT6359_ACCDET_CON0 > +#define AUDACCDETAUXADCSWCTRL_SEL_SFT 9 > +#define AUDACCDETAUXADCSWCTRL_SEL_MASK 0x1 > +#define AUDACCDETAUXADCSWCTRL_SEL_MASK_SFT (0x1 << 9) > +#define AUDACCDETAUXADCSWCTRL_SW_ADDR \ > + MT6359_ACCDET_CON0 > +#define AUDACCDETAUXADCSWCTRL_SW_SFT 10 > +#define AUDACCDETAUXADCSWCTRL_SW_MASK 0x1 > +#define AUDACCDETAUXADCSWCTRL_SW_MASK_SFT (0x1 << 10) > +#define ACCDET_TEST_ANA_ADDR \ > + MT6359_ACCDET_CON0 > +#define ACCDET_TEST_ANA_SFT 11 > +#define ACCDET_TEST_ANA_MASK 0x1 > +#define ACCDET_TEST_ANA_MASK_SFT (0x1 << 11) > +#define RG_AUDACCDETRSV_ADDR \ > + MT6359_ACCDET_CON0 > +#define RG_AUDACCDETRSV_SFT 13 > +#define RG_AUDACCDETRSV_MASK 0x3 > +#define RG_AUDACCDETRSV_MASK_SFT (0x3 << 13) > +#define ACCDET_SW_EN_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_SW_EN_SFT 0 > +#define ACCDET_SW_EN_MASK 0x1 > +#define ACCDET_SW_EN_MASK_SFT (0x1 << 0) > +#define ACCDET_SEQ_INIT_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_SEQ_INIT_SFT 1 > +#define ACCDET_SEQ_INIT_MASK 0x1 > +#define ACCDET_SEQ_INIT_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT0_SW_EN_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT0_SW_EN_SFT 2 > +#define ACCDET_EINT0_SW_EN_MASK 0x1 > +#define ACCDET_EINT0_SW_EN_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT0_SEQ_INIT_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT0_SEQ_INIT_SFT 3 > +#define ACCDET_EINT0_SEQ_INIT_MASK 0x1 > +#define ACCDET_EINT0_SEQ_INIT_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT1_SW_EN_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT1_SW_EN_SFT 4 > +#define ACCDET_EINT1_SW_EN_MASK 0x1 > +#define ACCDET_EINT1_SW_EN_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT1_SEQ_INIT_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT1_SEQ_INIT_SFT 5 > +#define ACCDET_EINT1_SEQ_INIT_MASK 0x1 > +#define ACCDET_EINT1_SEQ_INIT_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT0_INVERTER_SW_EN_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT0_INVERTER_SW_EN_SFT 6 > +#define ACCDET_EINT0_INVERTER_SW_EN_MASK 0x1 > +#define ACCDET_EINT0_INVERTER_SW_EN_MASK_SFT (0x1 << 6) > +#define ACCDET_EINT0_INVERTER_SEQ_INIT_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT0_INVERTER_SEQ_INIT_SFT 7 > +#define ACCDET_EINT0_INVERTER_SEQ_INIT_MASK 0x1 > +#define ACCDET_EINT0_INVERTER_SEQ_INIT_MASK_SFT (0x1 << 7) > +#define ACCDET_EINT1_INVERTER_SW_EN_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT1_INVERTER_SW_EN_SFT 8 > +#define ACCDET_EINT1_INVERTER_SW_EN_MASK 0x1 > +#define ACCDET_EINT1_INVERTER_SW_EN_MASK_SFT (0x1 << 8) > +#define ACCDET_EINT1_INVERTER_SEQ_INIT_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT1_INVERTER_SEQ_INIT_SFT 9 > +#define ACCDET_EINT1_INVERTER_SEQ_INIT_MASK 0x1 > +#define ACCDET_EINT1_INVERTER_SEQ_INIT_MASK_SFT (0x1 << 9) > +#define ACCDET_EINT0_M_SW_EN_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT0_M_SW_EN_SFT 10 > +#define ACCDET_EINT0_M_SW_EN_MASK 0x1 > +#define ACCDET_EINT0_M_SW_EN_MASK_SFT (0x1 << 10) > +#define ACCDET_EINT1_M_SW_EN_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT1_M_SW_EN_SFT 11 > +#define ACCDET_EINT1_M_SW_EN_MASK 0x1 > +#define ACCDET_EINT1_M_SW_EN_MASK_SFT (0x1 << 11) > +#define ACCDET_EINT_M_DETECT_EN_ADDR \ > + MT6359_ACCDET_CON1 > +#define ACCDET_EINT_M_DETECT_EN_SFT 12 > +#define ACCDET_EINT_M_DETECT_EN_MASK 0x1 > +#define ACCDET_EINT_M_DETECT_EN_MASK_SFT (0x1 << 12) > +#define ACCDET_CMP_PWM_EN_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_CMP_PWM_EN_SFT 0 > +#define ACCDET_CMP_PWM_EN_MASK 0x1 > +#define ACCDET_CMP_PWM_EN_MASK_SFT (0x1 << 0) > +#define ACCDET_VTH_PWM_EN_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_VTH_PWM_EN_SFT 1 > +#define ACCDET_VTH_PWM_EN_MASK 0x1 > +#define ACCDET_VTH_PWM_EN_MASK_SFT (0x1 << 1) > +#define ACCDET_MBIAS_PWM_EN_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_MBIAS_PWM_EN_SFT 2 > +#define ACCDET_MBIAS_PWM_EN_MASK 0x1 > +#define ACCDET_MBIAS_PWM_EN_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT_EN_PWM_EN_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_EINT_EN_PWM_EN_SFT 3 > +#define ACCDET_EINT_EN_PWM_EN_MASK 0x1 > +#define ACCDET_EINT_EN_PWM_EN_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT_CMPEN_PWM_EN_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_EINT_CMPEN_PWM_EN_SFT 4 > +#define ACCDET_EINT_CMPEN_PWM_EN_MASK 0x1 > +#define ACCDET_EINT_CMPEN_PWM_EN_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT_CMPMEN_PWM_EN_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_EINT_CMPMEN_PWM_EN_SFT 5 > +#define ACCDET_EINT_CMPMEN_PWM_EN_MASK 0x1 > +#define ACCDET_EINT_CMPMEN_PWM_EN_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT_CTURBO_PWM_EN_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_EINT_CTURBO_PWM_EN_SFT 6 > +#define ACCDET_EINT_CTURBO_PWM_EN_MASK 0x1 > +#define ACCDET_EINT_CTURBO_PWM_EN_MASK_SFT (0x1 << 6) > +#define ACCDET_CMP_PWM_IDLE_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_CMP_PWM_IDLE_SFT 8 > +#define ACCDET_CMP_PWM_IDLE_MASK 0x1 > +#define ACCDET_CMP_PWM_IDLE_MASK_SFT (0x1 << 8) > +#define ACCDET_VTH_PWM_IDLE_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_VTH_PWM_IDLE_SFT 9 > +#define ACCDET_VTH_PWM_IDLE_MASK 0x1 > +#define ACCDET_VTH_PWM_IDLE_MASK_SFT (0x1 << 9) > +#define ACCDET_MBIAS_PWM_IDLE_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_MBIAS_PWM_IDLE_SFT 10 > +#define ACCDET_MBIAS_PWM_IDLE_MASK 0x1 > +#define ACCDET_MBIAS_PWM_IDLE_MASK_SFT (0x1 << 10) > +#define ACCDET_EINT0_CMPEN_PWM_IDLE_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_EINT0_CMPEN_PWM_IDLE_SFT 11 > +#define ACCDET_EINT0_CMPEN_PWM_IDLE_MASK 0x1 > +#define ACCDET_EINT0_CMPEN_PWM_IDLE_MASK_SFT (0x1 << 11) > +#define ACCDET_EINT1_CMPEN_PWM_IDLE_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_EINT1_CMPEN_PWM_IDLE_SFT 12 > +#define ACCDET_EINT1_CMPEN_PWM_IDLE_MASK 0x1 > +#define ACCDET_EINT1_CMPEN_PWM_IDLE_MASK_SFT (0x1 << 12) > +#define ACCDET_PWM_EN_SW_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_PWM_EN_SW_SFT 13 > +#define ACCDET_PWM_EN_SW_MASK 0x1 > +#define ACCDET_PWM_EN_SW_MASK_SFT (0x1 << 13) > +#define ACCDET_PWM_EN_SEL_ADDR \ > + MT6359_ACCDET_CON2 > +#define ACCDET_PWM_EN_SEL_SFT 14 > +#define ACCDET_PWM_EN_SEL_MASK 0x3 > +#define ACCDET_PWM_EN_SEL_MASK_SFT (0x3 << 14) > +#define ACCDET_PWM_WIDTH_ADDR \ > + MT6359_ACCDET_CON3 > +#define ACCDET_PWM_WIDTH_SFT 0 > +#define ACCDET_PWM_WIDTH_MASK 0xFFFF > +#define ACCDET_PWM_WIDTH_MASK_SFT (0xFFFF << 0) > +#define ACCDET_PWM_THRESH_ADDR \ > + MT6359_ACCDET_CON4 > +#define ACCDET_PWM_THRESH_SFT 0 > +#define ACCDET_PWM_THRESH_MASK 0xFFFF > +#define ACCDET_PWM_THRESH_MASK_SFT (0xFFFF << 0) > +#define ACCDET_RISE_DELAY_ADDR \ > + MT6359_ACCDET_CON5 > +#define ACCDET_RISE_DELAY_SFT 0 > +#define ACCDET_RISE_DELAY_MASK 0x7FFF > +#define ACCDET_RISE_DELAY_MASK_SFT (0x7FFF << 0) > +#define ACCDET_FALL_DELAY_ADDR \ > + MT6359_ACCDET_CON5 > +#define ACCDET_FALL_DELAY_SFT 15 > +#define ACCDET_FALL_DELAY_MASK 0x1 > +#define ACCDET_FALL_DELAY_MASK_SFT (0x1 << 15) > +#define ACCDET_EINT_CMPMEN_PWM_THRESH_ADDR \ > + MT6359_ACCDET_CON6 > +#define ACCDET_EINT_CMPMEN_PWM_THRESH_SFT 0 > +#define ACCDET_EINT_CMPMEN_PWM_THRESH_MASK 0x7 > +#define ACCDET_EINT_CMPMEN_PWM_THRESH_MASK_SFT (0x7 << 0) > +#define ACCDET_EINT_CMPMEN_PWM_WIDTH_ADDR \ > + MT6359_ACCDET_CON6 > +#define ACCDET_EINT_CMPMEN_PWM_WIDTH_SFT 4 > +#define ACCDET_EINT_CMPMEN_PWM_WIDTH_MASK 0x7 > +#define ACCDET_EINT_CMPMEN_PWM_WIDTH_MASK_SFT (0x7 << 4) > +#define ACCDET_EINT_EN_PWM_THRESH_ADDR \ > + MT6359_ACCDET_CON7 > +#define ACCDET_EINT_EN_PWM_THRESH_SFT 0 > +#define ACCDET_EINT_EN_PWM_THRESH_MASK 0x7 > +#define ACCDET_EINT_EN_PWM_THRESH_MASK_SFT (0x7 << 0) > +#define ACCDET_EINT_EN_PWM_WIDTH_ADDR \ > + MT6359_ACCDET_CON7 > +#define ACCDET_EINT_EN_PWM_WIDTH_SFT 4 > +#define ACCDET_EINT_EN_PWM_WIDTH_MASK 0x3 > +#define ACCDET_EINT_EN_PWM_WIDTH_MASK_SFT (0x3 << 4) > +#define ACCDET_EINT_CMPEN_PWM_THRESH_ADDR \ > + MT6359_ACCDET_CON7 > +#define ACCDET_EINT_CMPEN_PWM_THRESH_SFT 8 > +#define ACCDET_EINT_CMPEN_PWM_THRESH_MASK 0x7 > +#define ACCDET_EINT_CMPEN_PWM_THRESH_MASK_SFT (0x7 << 8) > +#define ACCDET_EINT_CMPEN_PWM_WIDTH_ADDR \ > + MT6359_ACCDET_CON7 > +#define ACCDET_EINT_CMPEN_PWM_WIDTH_SFT 12 > +#define ACCDET_EINT_CMPEN_PWM_WIDTH_MASK 0x3 > +#define ACCDET_EINT_CMPEN_PWM_WIDTH_MASK_SFT (0x3 << 12) > +#define ACCDET_DEBOUNCE0_ADDR \ > + MT6359_ACCDET_CON8 > +#define ACCDET_DEBOUNCE0_SFT 0 > +#define ACCDET_DEBOUNCE0_MASK 0xFFFF > +#define ACCDET_DEBOUNCE0_MASK_SFT (0xFFFF << 0) > +#define ACCDET_DEBOUNCE1_ADDR \ > + MT6359_ACCDET_CON9 > +#define ACCDET_DEBOUNCE1_SFT 0 > +#define ACCDET_DEBOUNCE1_MASK 0xFFFF > +#define ACCDET_DEBOUNCE1_MASK_SFT (0xFFFF << 0) > +#define ACCDET_DEBOUNCE2_ADDR \ > + MT6359_ACCDET_CON10 > +#define ACCDET_DEBOUNCE2_SFT 0 > +#define ACCDET_DEBOUNCE2_MASK 0xFFFF > +#define ACCDET_DEBOUNCE2_MASK_SFT (0xFFFF << 0) > +#define ACCDET_DEBOUNCE3_ADDR \ > + MT6359_ACCDET_CON11 > +#define ACCDET_DEBOUNCE3_SFT 0 > +#define ACCDET_DEBOUNCE3_MASK 0xFFFF > +#define ACCDET_DEBOUNCE3_MASK_SFT (0xFFFF << 0) > +#define ACCDET_CONNECT_AUXADC_TIME_DIG_ADDR \ > + MT6359_ACCDET_CON12 > +#define ACCDET_CONNECT_AUXADC_TIME_DIG_SFT 0 > +#define ACCDET_CONNECT_AUXADC_TIME_DIG_MASK 0xFFFF > +#define ACCDET_CONNECT_AUXADC_TIME_DIG_MASK_SFT (0xFFFF << 0) > +#define ACCDET_CONNECT_AUXADC_TIME_ANA_ADDR \ > + MT6359_ACCDET_CON13 > +#define ACCDET_CONNECT_AUXADC_TIME_ANA_SFT 0 > +#define ACCDET_CONNECT_AUXADC_TIME_ANA_MASK 0xFFFF > +#define ACCDET_CONNECT_AUXADC_TIME_ANA_MASK_SFT (0xFFFF << 0) > +#define ACCDET_EINT_DEBOUNCE0_ADDR \ > + MT6359_ACCDET_CON14 > +#define ACCDET_EINT_DEBOUNCE0_SFT 0 > +#define ACCDET_EINT_DEBOUNCE0_MASK 0xF > +#define ACCDET_EINT_DEBOUNCE0_MASK_SFT (0xF << 0) > +#define ACCDET_EINT_DEBOUNCE1_ADDR \ > + MT6359_ACCDET_CON14 > +#define ACCDET_EINT_DEBOUNCE1_SFT 4 > +#define ACCDET_EINT_DEBOUNCE1_MASK 0xF > +#define ACCDET_EINT_DEBOUNCE1_MASK_SFT (0xF << 4) > +#define ACCDET_EINT_DEBOUNCE2_ADDR \ > + MT6359_ACCDET_CON14 > +#define ACCDET_EINT_DEBOUNCE2_SFT 8 > +#define ACCDET_EINT_DEBOUNCE2_MASK 0xF > +#define ACCDET_EINT_DEBOUNCE2_MASK_SFT (0xF << 8) > +#define ACCDET_EINT_DEBOUNCE3_ADDR \ > + MT6359_ACCDET_CON14 > +#define ACCDET_EINT_DEBOUNCE3_SFT 12 > +#define ACCDET_EINT_DEBOUNCE3_MASK 0xF > +#define ACCDET_EINT_DEBOUNCE3_MASK_SFT (0xF << 12) > +#define ACCDET_EINT_INVERTER_DEBOUNCE_ADDR \ > + MT6359_ACCDET_CON15 > +#define ACCDET_EINT_INVERTER_DEBOUNCE_SFT 0 > +#define ACCDET_EINT_INVERTER_DEBOUNCE_MASK 0xF > +#define ACCDET_EINT_INVERTER_DEBOUNCE_MASK_SFT (0xF << 0) > +#define ACCDET_IVAL_CUR_IN_ADDR \ > + MT6359_ACCDET_CON16 > +#define ACCDET_IVAL_CUR_IN_SFT 0 > +#define ACCDET_IVAL_CUR_IN_MASK 0x3 > +#define ACCDET_IVAL_CUR_IN_MASK_SFT (0x3 << 0) > +#define ACCDET_IVAL_SAM_IN_ADDR \ > + MT6359_ACCDET_CON16 > +#define ACCDET_IVAL_SAM_IN_SFT 2 > +#define ACCDET_IVAL_SAM_IN_MASK 0x3 > +#define ACCDET_IVAL_SAM_IN_MASK_SFT (0x3 << 2) > +#define ACCDET_IVAL_MEM_IN_ADDR \ > + MT6359_ACCDET_CON16 > +#define ACCDET_IVAL_MEM_IN_SFT 4 > +#define ACCDET_IVAL_MEM_IN_MASK 0x3 > +#define ACCDET_IVAL_MEM_IN_MASK_SFT (0x3 << 4) > +#define ACCDET_EINT_IVAL_CUR_IN_ADDR \ > + MT6359_ACCDET_CON16 > +#define ACCDET_EINT_IVAL_CUR_IN_SFT 6 > +#define ACCDET_EINT_IVAL_CUR_IN_MASK 0x3 > +#define ACCDET_EINT_IVAL_CUR_IN_MASK_SFT (0x3 << 6) > +#define ACCDET_EINT_IVAL_SAM_IN_ADDR \ > + MT6359_ACCDET_CON16 > +#define ACCDET_EINT_IVAL_SAM_IN_SFT 8 > +#define ACCDET_EINT_IVAL_SAM_IN_MASK 0x3 > +#define ACCDET_EINT_IVAL_SAM_IN_MASK_SFT (0x3 << 8) > +#define ACCDET_EINT_IVAL_MEM_IN_ADDR \ > + MT6359_ACCDET_CON16 > +#define ACCDET_EINT_IVAL_MEM_IN_SFT 10 > +#define ACCDET_EINT_IVAL_MEM_IN_MASK 0x3 > +#define ACCDET_EINT_IVAL_MEM_IN_MASK_SFT (0x3 << 10) > +#define ACCDET_IVAL_SEL_ADDR \ > + MT6359_ACCDET_CON16 > +#define ACCDET_IVAL_SEL_SFT 12 > +#define ACCDET_IVAL_SEL_MASK 0x1 > +#define ACCDET_IVAL_SEL_MASK_SFT (0x1 << 12) > +#define ACCDET_EINT_IVAL_SEL_ADDR \ > + MT6359_ACCDET_CON16 > +#define ACCDET_EINT_IVAL_SEL_SFT 13 > +#define ACCDET_EINT_IVAL_SEL_MASK 0x1 > +#define ACCDET_EINT_IVAL_SEL_MASK_SFT (0x1 << 13) > +#define ACCDET_EINT_INVERTER_IVAL_CUR_IN_ADDR \ > + MT6359_ACCDET_CON17 > +#define ACCDET_EINT_INVERTER_IVAL_CUR_IN_SFT 0 > +#define ACCDET_EINT_INVERTER_IVAL_CUR_IN_MASK 0x1 > +#define ACCDET_EINT_INVERTER_IVAL_CUR_IN_MASK_SFT (0x1 << 0) > +#define ACCDET_EINT_INVERTER_IVAL_SAM_IN_ADDR \ > + MT6359_ACCDET_CON17 > +#define ACCDET_EINT_INVERTER_IVAL_SAM_IN_SFT 1 > +#define ACCDET_EINT_INVERTER_IVAL_SAM_IN_MASK 0x1 > +#define ACCDET_EINT_INVERTER_IVAL_SAM_IN_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT_INVERTER_IVAL_MEM_IN_ADDR \ > + MT6359_ACCDET_CON17 > +#define ACCDET_EINT_INVERTER_IVAL_MEM_IN_SFT 2 > +#define ACCDET_EINT_INVERTER_IVAL_MEM_IN_MASK 0x1 > +#define ACCDET_EINT_INVERTER_IVAL_MEM_IN_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT_INVERTER_IVAL_SEL_ADDR \ > + MT6359_ACCDET_CON17 > +#define ACCDET_EINT_INVERTER_IVAL_SEL_SFT 3 > +#define ACCDET_EINT_INVERTER_IVAL_SEL_MASK 0x1 > +#define ACCDET_EINT_INVERTER_IVAL_SEL_MASK_SFT (0x1 << 3) > +#define ACCDET_IRQ_ADDR \ > + MT6359_ACCDET_CON18 > +#define ACCDET_IRQ_SFT 0 > +#define ACCDET_IRQ_MASK 0x1 > +#define ACCDET_IRQ_MASK_SFT (0x1 << 0) > +#define ACCDET_EINT0_IRQ_ADDR \ > + MT6359_ACCDET_CON18 > +#define ACCDET_EINT0_IRQ_SFT 2 > +#define ACCDET_EINT0_IRQ_MASK 0x1 > +#define ACCDET_EINT0_IRQ_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT1_IRQ_ADDR \ > + MT6359_ACCDET_CON18 > +#define ACCDET_EINT1_IRQ_SFT 3 > +#define ACCDET_EINT1_IRQ_MASK 0x1 > +#define ACCDET_EINT1_IRQ_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT_IN_INVERSE_ADDR \ > + MT6359_ACCDET_CON18 > +#define ACCDET_EINT_IN_INVERSE_SFT 4 > +#define ACCDET_EINT_IN_INVERSE_MASK 0x1 > +#define ACCDET_EINT_IN_INVERSE_MASK_SFT (0x1 << 4) > +#define ACCDET_IRQ_CLR_ADDR \ > + MT6359_ACCDET_CON18 > +#define ACCDET_IRQ_CLR_SFT 8 > +#define ACCDET_IRQ_CLR_MASK 0x1 > +#define ACCDET_IRQ_CLR_MASK_SFT (0x1 << 8) > +#define ACCDET_EINT0_IRQ_CLR_ADDR \ > + MT6359_ACCDET_CON18 > +#define ACCDET_EINT0_IRQ_CLR_SFT 10 > +#define ACCDET_EINT0_IRQ_CLR_MASK 0x1 > +#define ACCDET_EINT0_IRQ_CLR_MASK_SFT (0x1 << 10) > +#define ACCDET_EINT1_IRQ_CLR_ADDR \ > + MT6359_ACCDET_CON18 > +#define ACCDET_EINT1_IRQ_CLR_SFT 11 > +#define ACCDET_EINT1_IRQ_CLR_MASK 0x1 > +#define ACCDET_EINT1_IRQ_CLR_MASK_SFT (0x1 << 11) > +#define ACCDET_EINT_M_PLUG_IN_NUM_ADDR \ > + MT6359_ACCDET_CON18 > +#define ACCDET_EINT_M_PLUG_IN_NUM_SFT 12 > +#define ACCDET_EINT_M_PLUG_IN_NUM_MASK 0x7 > +#define ACCDET_EINT_M_PLUG_IN_NUM_MASK_SFT (0x7 << 12) > +#define ACCDET_DA_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_DA_STABLE_SFT 0 > +#define ACCDET_DA_STABLE_MASK 0x1 > +#define ACCDET_DA_STABLE_MASK_SFT (0x1 << 0) > +#define ACCDET_EINT0_EN_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT0_EN_STABLE_SFT 1 > +#define ACCDET_EINT0_EN_STABLE_MASK 0x1 > +#define ACCDET_EINT0_EN_STABLE_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT0_CMPEN_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT0_CMPEN_STABLE_SFT 2 > +#define ACCDET_EINT0_CMPEN_STABLE_MASK 0x1 > +#define ACCDET_EINT0_CMPEN_STABLE_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT0_CMPMEN_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT0_CMPMEN_STABLE_SFT 3 > +#define ACCDET_EINT0_CMPMEN_STABLE_MASK 0x1 > +#define ACCDET_EINT0_CMPMEN_STABLE_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT0_CTURBO_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT0_CTURBO_STABLE_SFT 4 > +#define ACCDET_EINT0_CTURBO_STABLE_MASK 0x1 > +#define ACCDET_EINT0_CTURBO_STABLE_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT0_CEN_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT0_CEN_STABLE_SFT 5 > +#define ACCDET_EINT0_CEN_STABLE_MASK 0x1 > +#define ACCDET_EINT0_CEN_STABLE_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT1_EN_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT1_EN_STABLE_SFT 6 > +#define ACCDET_EINT1_EN_STABLE_MASK 0x1 > +#define ACCDET_EINT1_EN_STABLE_MASK_SFT (0x1 << 6) > +#define ACCDET_EINT1_CMPEN_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT1_CMPEN_STABLE_SFT 7 > +#define ACCDET_EINT1_CMPEN_STABLE_MASK 0x1 > +#define ACCDET_EINT1_CMPEN_STABLE_MASK_SFT (0x1 << 7) > +#define ACCDET_EINT1_CMPMEN_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT1_CMPMEN_STABLE_SFT 8 > +#define ACCDET_EINT1_CMPMEN_STABLE_MASK 0x1 > +#define ACCDET_EINT1_CMPMEN_STABLE_MASK_SFT (0x1 << 8) > +#define ACCDET_EINT1_CTURBO_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT1_CTURBO_STABLE_SFT 9 > +#define ACCDET_EINT1_CTURBO_STABLE_MASK 0x1 > +#define ACCDET_EINT1_CTURBO_STABLE_MASK_SFT (0x1 << 9) > +#define ACCDET_EINT1_CEN_STABLE_ADDR \ > + MT6359_ACCDET_CON19 > +#define ACCDET_EINT1_CEN_STABLE_SFT 10 > +#define ACCDET_EINT1_CEN_STABLE_MASK 0x1 > +#define ACCDET_EINT1_CEN_STABLE_MASK_SFT (0x1 << 10) > +#define ACCDET_HWMODE_EN_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_HWMODE_EN_SFT 0 > +#define ACCDET_HWMODE_EN_MASK 0x1 > +#define ACCDET_HWMODE_EN_MASK_SFT (0x1 << 0) > +#define ACCDET_HWMODE_SEL_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_HWMODE_SEL_SFT 1 > +#define ACCDET_HWMODE_SEL_MASK 0x3 > +#define ACCDET_HWMODE_SEL_MASK_SFT (0x3 << 1) > +#define ACCDET_PLUG_OUT_DETECT_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_PLUG_OUT_DETECT_SFT 3 > +#define ACCDET_PLUG_OUT_DETECT_MASK 0x1 > +#define ACCDET_PLUG_OUT_DETECT_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT0_REVERSE_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_EINT0_REVERSE_SFT 4 > +#define ACCDET_EINT0_REVERSE_MASK 0x1 > +#define ACCDET_EINT0_REVERSE_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT1_REVERSE_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_EINT1_REVERSE_SFT 5 > +#define ACCDET_EINT1_REVERSE_MASK 0x1 > +#define ACCDET_EINT1_REVERSE_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT_HWMODE_EN_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_EINT_HWMODE_EN_SFT 8 > +#define ACCDET_EINT_HWMODE_EN_MASK 0x1 > +#define ACCDET_EINT_HWMODE_EN_MASK_SFT (0x1 << 8) > +#define ACCDET_EINT_PLUG_OUT_BYPASS_DEB_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_EINT_PLUG_OUT_BYPASS_DEB_SFT 9 > +#define ACCDET_EINT_PLUG_OUT_BYPASS_DEB_MASK 0x1 > +#define ACCDET_EINT_PLUG_OUT_BYPASS_DEB_MASK_SFT (0x1 << 9) > +#define ACCDET_EINT_M_PLUG_IN_EN_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_EINT_M_PLUG_IN_EN_SFT 10 > +#define ACCDET_EINT_M_PLUG_IN_EN_MASK 0x1 > +#define ACCDET_EINT_M_PLUG_IN_EN_MASK_SFT (0x1 << 10) > +#define ACCDET_EINT_M_HWMODE_EN_ADDR \ > + MT6359_ACCDET_CON20 > +#define ACCDET_EINT_M_HWMODE_EN_SFT 11 > +#define ACCDET_EINT_M_HWMODE_EN_MASK 0x1 > +#define ACCDET_EINT_M_HWMODE_EN_MASK_SFT (0x1 << 11) > +#define ACCDET_TEST_CMPEN_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_TEST_CMPEN_SFT 0 > +#define ACCDET_TEST_CMPEN_MASK 0x1 > +#define ACCDET_TEST_CMPEN_MASK_SFT (0x1 << 0) > +#define ACCDET_TEST_VTHEN_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_TEST_VTHEN_SFT 1 > +#define ACCDET_TEST_VTHEN_MASK 0x1 > +#define ACCDET_TEST_VTHEN_MASK_SFT (0x1 << 1) > +#define ACCDET_TEST_MBIASEN_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_TEST_MBIASEN_SFT 2 > +#define ACCDET_TEST_MBIASEN_MASK 0x1 > +#define ACCDET_TEST_MBIASEN_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT_TEST_EN_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_EN_SFT 3 > +#define ACCDET_EINT_TEST_EN_MASK 0x1 > +#define ACCDET_EINT_TEST_EN_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT_TEST_INVEN_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_INVEN_SFT 4 > +#define ACCDET_EINT_TEST_INVEN_MASK 0x1 > +#define ACCDET_EINT_TEST_INVEN_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT_TEST_CMPEN_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_CMPEN_SFT 5 > +#define ACCDET_EINT_TEST_CMPEN_MASK 0x1 > +#define ACCDET_EINT_TEST_CMPEN_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT_TEST_CMPMEN_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_CMPMEN_SFT 6 > +#define ACCDET_EINT_TEST_CMPMEN_MASK 0x1 > +#define ACCDET_EINT_TEST_CMPMEN_MASK_SFT (0x1 << 6) > +#define ACCDET_EINT_TEST_CTURBO_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_CTURBO_SFT 7 > +#define ACCDET_EINT_TEST_CTURBO_MASK 0x1 > +#define ACCDET_EINT_TEST_CTURBO_MASK_SFT (0x1 << 7) > +#define ACCDET_EINT_TEST_CEN_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_CEN_SFT 8 > +#define ACCDET_EINT_TEST_CEN_MASK 0x1 > +#define ACCDET_EINT_TEST_CEN_MASK_SFT (0x1 << 8) > +#define ACCDET_TEST_B_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_TEST_B_SFT 9 > +#define ACCDET_TEST_B_MASK 0x1 > +#define ACCDET_TEST_B_MASK_SFT (0x1 << 9) > +#define ACCDET_TEST_A_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_TEST_A_SFT 10 > +#define ACCDET_TEST_A_MASK 0x1 > +#define ACCDET_TEST_A_MASK_SFT (0x1 << 10) > +#define ACCDET_EINT_TEST_CMPOUT_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_CMPOUT_SFT 11 > +#define ACCDET_EINT_TEST_CMPOUT_MASK 0x1 > +#define ACCDET_EINT_TEST_CMPOUT_MASK_SFT (0x1 << 11) > +#define ACCDET_EINT_TEST_CMPMOUT_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_CMPMOUT_SFT 12 > +#define ACCDET_EINT_TEST_CMPMOUT_MASK 0x1 > +#define ACCDET_EINT_TEST_CMPMOUT_MASK_SFT (0x1 << 12) > +#define ACCDET_EINT_TEST_INVOUT_ADDR \ > + MT6359_ACCDET_CON21 > +#define ACCDET_EINT_TEST_INVOUT_SFT 13 > +#define ACCDET_EINT_TEST_INVOUT_MASK 0x1 > +#define ACCDET_EINT_TEST_INVOUT_MASK_SFT (0x1 << 13) > +#define ACCDET_CMPEN_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_CMPEN_SEL_SFT 0 > +#define ACCDET_CMPEN_SEL_MASK 0x1 > +#define ACCDET_CMPEN_SEL_MASK_SFT (0x1 << 0) > +#define ACCDET_VTHEN_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_VTHEN_SEL_SFT 1 > +#define ACCDET_VTHEN_SEL_MASK 0x1 > +#define ACCDET_VTHEN_SEL_MASK_SFT (0x1 << 1) > +#define ACCDET_MBIASEN_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_MBIASEN_SEL_SFT 2 > +#define ACCDET_MBIASEN_SEL_MASK 0x1 > +#define ACCDET_MBIASEN_SEL_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT_EN_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_EINT_EN_SEL_SFT 3 > +#define ACCDET_EINT_EN_SEL_MASK 0x1 > +#define ACCDET_EINT_EN_SEL_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT_INVEN_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_EINT_INVEN_SEL_SFT 4 > +#define ACCDET_EINT_INVEN_SEL_MASK 0x1 > +#define ACCDET_EINT_INVEN_SEL_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT_CMPEN_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_EINT_CMPEN_SEL_SFT 5 > +#define ACCDET_EINT_CMPEN_SEL_MASK 0x1 > +#define ACCDET_EINT_CMPEN_SEL_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT_CMPMEN_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_EINT_CMPMEN_SEL_SFT 6 > +#define ACCDET_EINT_CMPMEN_SEL_MASK 0x1 > +#define ACCDET_EINT_CMPMEN_SEL_MASK_SFT (0x1 << 6) > +#define ACCDET_EINT_CTURBO_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_EINT_CTURBO_SEL_SFT 7 > +#define ACCDET_EINT_CTURBO_SEL_MASK 0x1 > +#define ACCDET_EINT_CTURBO_SEL_MASK_SFT (0x1 << 7) > +#define ACCDET_B_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_B_SEL_SFT 9 > +#define ACCDET_B_SEL_MASK 0x1 > +#define ACCDET_B_SEL_MASK_SFT (0x1 << 9) > +#define ACCDET_A_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_A_SEL_SFT 10 > +#define ACCDET_A_SEL_MASK 0x1 > +#define ACCDET_A_SEL_MASK_SFT (0x1 << 10) > +#define ACCDET_EINT_CMPOUT_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_EINT_CMPOUT_SEL_SFT 11 > +#define ACCDET_EINT_CMPOUT_SEL_MASK 0x1 > +#define ACCDET_EINT_CMPOUT_SEL_MASK_SFT (0x1 << 11) > +#define ACCDET_EINT_CMPMOUT_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_EINT_CMPMOUT_SEL_SFT 12 > +#define ACCDET_EINT_CMPMOUT_SEL_MASK 0x1 > +#define ACCDET_EINT_CMPMOUT_SEL_MASK_SFT (0x1 << 12) > +#define ACCDET_EINT_INVOUT_SEL_ADDR \ > + MT6359_ACCDET_CON22 > +#define ACCDET_EINT_INVOUT_SEL_SFT 13 > +#define ACCDET_EINT_INVOUT_SEL_MASK 0x1 > +#define ACCDET_EINT_INVOUT_SEL_MASK_SFT (0x1 << 13) > +#define ACCDET_CMPEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_CMPEN_SW_SFT 0 > +#define ACCDET_CMPEN_SW_MASK 0x1 > +#define ACCDET_CMPEN_SW_MASK_SFT (0x1 << 0) > +#define ACCDET_VTHEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_VTHEN_SW_SFT 1 > +#define ACCDET_VTHEN_SW_MASK 0x1 > +#define ACCDET_VTHEN_SW_MASK_SFT (0x1 << 1) > +#define ACCDET_MBIASEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_MBIASEN_SW_SFT 2 > +#define ACCDET_MBIASEN_SW_MASK 0x1 > +#define ACCDET_MBIASEN_SW_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT0_EN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT0_EN_SW_SFT 3 > +#define ACCDET_EINT0_EN_SW_MASK 0x1 > +#define ACCDET_EINT0_EN_SW_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT0_INVEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT0_INVEN_SW_SFT 4 > +#define ACCDET_EINT0_INVEN_SW_MASK 0x1 > +#define ACCDET_EINT0_INVEN_SW_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT0_CMPEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT0_CMPEN_SW_SFT 5 > +#define ACCDET_EINT0_CMPEN_SW_MASK 0x1 > +#define ACCDET_EINT0_CMPEN_SW_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT0_CMPMEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT0_CMPMEN_SW_SFT 6 > +#define ACCDET_EINT0_CMPMEN_SW_MASK 0x1 > +#define ACCDET_EINT0_CMPMEN_SW_MASK_SFT (0x1 << 6) > +#define ACCDET_EINT0_CTURBO_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT0_CTURBO_SW_SFT 7 > +#define ACCDET_EINT0_CTURBO_SW_MASK 0x1 > +#define ACCDET_EINT0_CTURBO_SW_MASK_SFT (0x1 << 7) > +#define ACCDET_EINT1_EN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT1_EN_SW_SFT 8 > +#define ACCDET_EINT1_EN_SW_MASK 0x1 > +#define ACCDET_EINT1_EN_SW_MASK_SFT (0x1 << 8) > +#define ACCDET_EINT1_INVEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT1_INVEN_SW_SFT 9 > +#define ACCDET_EINT1_INVEN_SW_MASK 0x1 > +#define ACCDET_EINT1_INVEN_SW_MASK_SFT (0x1 << 9) > +#define ACCDET_EINT1_CMPEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT1_CMPEN_SW_SFT 10 > +#define ACCDET_EINT1_CMPEN_SW_MASK 0x1 > +#define ACCDET_EINT1_CMPEN_SW_MASK_SFT (0x1 << 10) > +#define ACCDET_EINT1_CMPMEN_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT1_CMPMEN_SW_SFT 11 > +#define ACCDET_EINT1_CMPMEN_SW_MASK 0x1 > +#define ACCDET_EINT1_CMPMEN_SW_MASK_SFT (0x1 << 11) > +#define ACCDET_EINT1_CTURBO_SW_ADDR \ > + MT6359_ACCDET_CON23 > +#define ACCDET_EINT1_CTURBO_SW_SFT 12 > +#define ACCDET_EINT1_CTURBO_SW_MASK 0x1 > +#define ACCDET_EINT1_CTURBO_SW_MASK_SFT (0x1 << 12) > +#define ACCDET_B_SW_ADDR \ > + MT6359_ACCDET_CON24 > +#define ACCDET_B_SW_SFT 0 > +#define ACCDET_B_SW_MASK 0x1 > +#define ACCDET_B_SW_MASK_SFT (0x1 << 0) > +#define ACCDET_A_SW_ADDR \ > + MT6359_ACCDET_CON24 > +#define ACCDET_A_SW_SFT 1 > +#define ACCDET_A_SW_MASK 0x1 > +#define ACCDET_A_SW_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT0_CMPOUT_SW_ADDR \ > + MT6359_ACCDET_CON24 > +#define ACCDET_EINT0_CMPOUT_SW_SFT 2 > +#define ACCDET_EINT0_CMPOUT_SW_MASK 0x1 > +#define ACCDET_EINT0_CMPOUT_SW_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT0_CMPMOUT_SW_ADDR \ > + MT6359_ACCDET_CON24 > +#define ACCDET_EINT0_CMPMOUT_SW_SFT 3 > +#define ACCDET_EINT0_CMPMOUT_SW_MASK 0x1 > +#define ACCDET_EINT0_CMPMOUT_SW_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT0_INVOUT_SW_ADDR \ > + MT6359_ACCDET_CON24 > +#define ACCDET_EINT0_INVOUT_SW_SFT 4 > +#define ACCDET_EINT0_INVOUT_SW_MASK 0x1 > +#define ACCDET_EINT0_INVOUT_SW_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT1_CMPOUT_SW_ADDR \ > + MT6359_ACCDET_CON24 > +#define ACCDET_EINT1_CMPOUT_SW_SFT 5 > +#define ACCDET_EINT1_CMPOUT_SW_MASK 0x1 > +#define ACCDET_EINT1_CMPOUT_SW_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT1_CMPMOUT_SW_ADDR \ > + MT6359_ACCDET_CON24 > +#define ACCDET_EINT1_CMPMOUT_SW_SFT 6 > +#define ACCDET_EINT1_CMPMOUT_SW_MASK 0x1 > +#define ACCDET_EINT1_CMPMOUT_SW_MASK_SFT (0x1 << 6) > +#define ACCDET_EINT1_INVOUT_SW_ADDR \ > + MT6359_ACCDET_CON24 > +#define ACCDET_EINT1_INVOUT_SW_SFT 7 > +#define ACCDET_EINT1_INVOUT_SW_MASK 0x1 > +#define ACCDET_EINT1_INVOUT_SW_MASK_SFT (0x1 << 7) > +#define AD_AUDACCDETCMPOB_ADDR \ > + MT6359_ACCDET_CON25 > +#define AD_AUDACCDETCMPOB_SFT 0 > +#define AD_AUDACCDETCMPOB_MASK 0x1 > +#define AD_AUDACCDETCMPOB_MASK_SFT (0x1 << 0) > +#define AD_AUDACCDETCMPOA_ADDR \ > + MT6359_ACCDET_CON25 > +#define AD_AUDACCDETCMPOA_SFT 1 > +#define AD_AUDACCDETCMPOA_MASK 0x1 > +#define AD_AUDACCDETCMPOA_MASK_SFT (0x1 << 1) > +#define ACCDET_CUR_IN_ADDR \ > + MT6359_ACCDET_CON25 > +#define ACCDET_CUR_IN_SFT 2 > +#define ACCDET_CUR_IN_MASK 0x3 > +#define ACCDET_CUR_IN_MASK_SFT (0x3 << 2) > +#define ACCDET_SAM_IN_ADDR \ > + MT6359_ACCDET_CON25 > +#define ACCDET_SAM_IN_SFT 4 > +#define ACCDET_SAM_IN_MASK 0x3 > +#define ACCDET_SAM_IN_MASK_SFT (0x3 << 4) > +#define ACCDET_MEM_IN_ADDR \ > + MT6359_ACCDET_CON25 > +#define ACCDET_MEM_IN_SFT 6 > +#define ACCDET_MEM_IN_MASK 0x3 > +#define ACCDET_MEM_IN_MASK_SFT (0x3 << 6) > +#define ACCDET_STATE_ADDR \ > + MT6359_ACCDET_CON25 > +#define ACCDET_STATE_SFT 8 > +#define ACCDET_STATE_MASK 0x7 > +#define ACCDET_STATE_MASK_SFT (0x7 << 8) > +#define DA_AUDACCDETMBIASCLK_ADDR \ > + MT6359_ACCDET_CON25 > +#define DA_AUDACCDETMBIASCLK_SFT 12 > +#define DA_AUDACCDETMBIASCLK_MASK 0x1 > +#define DA_AUDACCDETMBIASCLK_MASK_SFT (0x1 << 12) > +#define DA_AUDACCDETVTHCLK_ADDR \ > + MT6359_ACCDET_CON25 > +#define DA_AUDACCDETVTHCLK_SFT 13 > +#define DA_AUDACCDETVTHCLK_MASK 0x1 > +#define DA_AUDACCDETVTHCLK_MASK_SFT (0x1 << 13) > +#define DA_AUDACCDETCMPCLK_ADDR \ > + MT6359_ACCDET_CON25 > +#define DA_AUDACCDETCMPCLK_SFT 14 > +#define DA_AUDACCDETCMPCLK_MASK 0x1 > +#define DA_AUDACCDETCMPCLK_MASK_SFT (0x1 << 14) > +#define DA_AUDACCDETAUXADCSWCTRL_ADDR \ > + MT6359_ACCDET_CON25 > +#define DA_AUDACCDETAUXADCSWCTRL_SFT 15 > +#define DA_AUDACCDETAUXADCSWCTRL_MASK 0x1 > +#define DA_AUDACCDETAUXADCSWCTRL_MASK_SFT (0x1 << 15) > +#define AD_EINT0CMPMOUT_ADDR \ > + MT6359_ACCDET_CON26 > +#define AD_EINT0CMPMOUT_SFT 0 > +#define AD_EINT0CMPMOUT_MASK 0x1 > +#define AD_EINT0CMPMOUT_MASK_SFT (0x1 << 0) > +#define AD_EINT0CMPOUT_ADDR \ > + MT6359_ACCDET_CON26 > +#define AD_EINT0CMPOUT_SFT 1 > +#define AD_EINT0CMPOUT_MASK 0x1 > +#define AD_EINT0CMPOUT_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT0_CUR_IN_ADDR \ > + MT6359_ACCDET_CON26 > +#define ACCDET_EINT0_CUR_IN_SFT 2 > +#define ACCDET_EINT0_CUR_IN_MASK 0x3 > +#define ACCDET_EINT0_CUR_IN_MASK_SFT (0x3 << 2) > +#define ACCDET_EINT0_SAM_IN_ADDR \ > + MT6359_ACCDET_CON26 > +#define ACCDET_EINT0_SAM_IN_SFT 4 > +#define ACCDET_EINT0_SAM_IN_MASK 0x3 > +#define ACCDET_EINT0_SAM_IN_MASK_SFT (0x3 << 4) > +#define ACCDET_EINT0_MEM_IN_ADDR \ > + MT6359_ACCDET_CON26 > +#define ACCDET_EINT0_MEM_IN_SFT 6 > +#define ACCDET_EINT0_MEM_IN_MASK 0x3 > +#define ACCDET_EINT0_MEM_IN_MASK_SFT (0x3 << 6) > +#define ACCDET_EINT0_STATE_ADDR \ > + MT6359_ACCDET_CON26 > +#define ACCDET_EINT0_STATE_SFT 8 > +#define ACCDET_EINT0_STATE_MASK 0x7 > +#define ACCDET_EINT0_STATE_MASK_SFT (0x7 << 8) > +#define DA_EINT0CMPEN_ADDR \ > + MT6359_ACCDET_CON26 > +#define DA_EINT0CMPEN_SFT 13 > +#define DA_EINT0CMPEN_MASK 0x1 > +#define DA_EINT0CMPEN_MASK_SFT (0x1 << 13) > +#define DA_EINT0CMPMEN_ADDR \ > + MT6359_ACCDET_CON26 > +#define DA_EINT0CMPMEN_SFT 14 > +#define DA_EINT0CMPMEN_MASK 0x1 > +#define DA_EINT0CMPMEN_MASK_SFT (0x1 << 14) > +#define DA_EINT0CTURBO_ADDR \ > + MT6359_ACCDET_CON26 > +#define DA_EINT0CTURBO_SFT 15 > +#define DA_EINT0CTURBO_MASK 0x1 > +#define DA_EINT0CTURBO_MASK_SFT (0x1 << 15) > +#define AD_EINT1CMPMOUT_ADDR \ > + MT6359_ACCDET_CON27 > +#define AD_EINT1CMPMOUT_SFT 0 > +#define AD_EINT1CMPMOUT_MASK 0x1 > +#define AD_EINT1CMPMOUT_MASK_SFT (0x1 << 0) > +#define AD_EINT1CMPOUT_ADDR \ > + MT6359_ACCDET_CON27 > +#define AD_EINT1CMPOUT_SFT 1 > +#define AD_EINT1CMPOUT_MASK 0x1 > +#define AD_EINT1CMPOUT_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT1_CUR_IN_ADDR \ > + MT6359_ACCDET_CON27 > +#define ACCDET_EINT1_CUR_IN_SFT 2 > +#define ACCDET_EINT1_CUR_IN_MASK 0x3 > +#define ACCDET_EINT1_CUR_IN_MASK_SFT (0x3 << 2) > +#define ACCDET_EINT1_SAM_IN_ADDR \ > + MT6359_ACCDET_CON27 > +#define ACCDET_EINT1_SAM_IN_SFT 4 > +#define ACCDET_EINT1_SAM_IN_MASK 0x3 > +#define ACCDET_EINT1_SAM_IN_MASK_SFT (0x3 << 4) > +#define ACCDET_EINT1_MEM_IN_ADDR \ > + MT6359_ACCDET_CON27 > +#define ACCDET_EINT1_MEM_IN_SFT 6 > +#define ACCDET_EINT1_MEM_IN_MASK 0x3 > +#define ACCDET_EINT1_MEM_IN_MASK_SFT (0x3 << 6) > +#define ACCDET_EINT1_STATE_ADDR \ > + MT6359_ACCDET_CON27 > +#define ACCDET_EINT1_STATE_SFT 8 > +#define ACCDET_EINT1_STATE_MASK 0x7 > +#define ACCDET_EINT1_STATE_MASK_SFT (0x7 << 8) > +#define DA_EINT1CMPEN_ADDR \ > + MT6359_ACCDET_CON27 > +#define DA_EINT1CMPEN_SFT 13 > +#define DA_EINT1CMPEN_MASK 0x1 > +#define DA_EINT1CMPEN_MASK_SFT (0x1 << 13) > +#define DA_EINT1CMPMEN_ADDR \ > + MT6359_ACCDET_CON27 > +#define DA_EINT1CMPMEN_SFT 14 > +#define DA_EINT1CMPMEN_MASK 0x1 > +#define DA_EINT1CMPMEN_MASK_SFT (0x1 << 14) > +#define DA_EINT1CTURBO_ADDR \ > + MT6359_ACCDET_CON27 > +#define DA_EINT1CTURBO_SFT 15 > +#define DA_EINT1CTURBO_MASK 0x1 > +#define DA_EINT1CTURBO_MASK_SFT (0x1 << 15) > +#define AD_EINT0INVOUT_ADDR \ > + MT6359_ACCDET_CON28 > +#define AD_EINT0INVOUT_SFT 0 > +#define AD_EINT0INVOUT_MASK 0x1 > +#define AD_EINT0INVOUT_MASK_SFT (0x1 << 0) > +#define ACCDET_EINT0_INVERTER_CUR_IN_ADDR \ > + MT6359_ACCDET_CON28 > +#define ACCDET_EINT0_INVERTER_CUR_IN_SFT 1 > +#define ACCDET_EINT0_INVERTER_CUR_IN_MASK 0x1 > +#define ACCDET_EINT0_INVERTER_CUR_IN_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT0_INVERTER_SAM_IN_ADDR \ > + MT6359_ACCDET_CON28 > +#define ACCDET_EINT0_INVERTER_SAM_IN_SFT 2 > +#define ACCDET_EINT0_INVERTER_SAM_IN_MASK 0x1 > +#define ACCDET_EINT0_INVERTER_SAM_IN_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT0_INVERTER_MEM_IN_ADDR \ > + MT6359_ACCDET_CON28 > +#define ACCDET_EINT0_INVERTER_MEM_IN_SFT 3 > +#define ACCDET_EINT0_INVERTER_MEM_IN_MASK 0x1 > +#define ACCDET_EINT0_INVERTER_MEM_IN_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT0_INVERTER_STATE_ADDR \ > + MT6359_ACCDET_CON28 > +#define ACCDET_EINT0_INVERTER_STATE_SFT 8 > +#define ACCDET_EINT0_INVERTER_STATE_MASK 0x7 > +#define ACCDET_EINT0_INVERTER_STATE_MASK_SFT (0x7 << 8) > +#define DA_EINT0EN_ADDR \ > + MT6359_ACCDET_CON28 > +#define DA_EINT0EN_SFT 12 > +#define DA_EINT0EN_MASK 0x1 > +#define DA_EINT0EN_MASK_SFT (0x1 << 12) > +#define DA_EINT0INVEN_ADDR \ > + MT6359_ACCDET_CON28 > +#define DA_EINT0INVEN_SFT 13 > +#define DA_EINT0INVEN_MASK 0x1 > +#define DA_EINT0INVEN_MASK_SFT (0x1 << 13) > +#define DA_EINT0CEN_ADDR \ > + MT6359_ACCDET_CON28 > +#define DA_EINT0CEN_SFT 14 > +#define DA_EINT0CEN_MASK 0x1 > +#define DA_EINT0CEN_MASK_SFT (0x1 << 14) > +#define AD_EINT1INVOUT_ADDR \ > + MT6359_ACCDET_CON29 > +#define AD_EINT1INVOUT_SFT 0 > +#define AD_EINT1INVOUT_MASK 0x1 > +#define AD_EINT1INVOUT_MASK_SFT (0x1 << 0) > +#define ACCDET_EINT1_INVERTER_CUR_IN_ADDR \ > + MT6359_ACCDET_CON29 > +#define ACCDET_EINT1_INVERTER_CUR_IN_SFT 1 > +#define ACCDET_EINT1_INVERTER_CUR_IN_MASK 0x1 > +#define ACCDET_EINT1_INVERTER_CUR_IN_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT1_INVERTER_SAM_IN_ADDR \ > + MT6359_ACCDET_CON29 > +#define ACCDET_EINT1_INVERTER_SAM_IN_SFT 2 > +#define ACCDET_EINT1_INVERTER_SAM_IN_MASK 0x1 > +#define ACCDET_EINT1_INVERTER_SAM_IN_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT1_INVERTER_MEM_IN_ADDR \ > + MT6359_ACCDET_CON29 > +#define ACCDET_EINT1_INVERTER_MEM_IN_SFT 3 > +#define ACCDET_EINT1_INVERTER_MEM_IN_MASK 0x1 > +#define ACCDET_EINT1_INVERTER_MEM_IN_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT1_INVERTER_STATE_ADDR \ > + MT6359_ACCDET_CON29 > +#define ACCDET_EINT1_INVERTER_STATE_SFT 8 > +#define ACCDET_EINT1_INVERTER_STATE_MASK 0x7 > +#define ACCDET_EINT1_INVERTER_STATE_MASK_SFT (0x7 << 8) > +#define DA_EINT1EN_ADDR \ > + MT6359_ACCDET_CON29 > +#define DA_EINT1EN_SFT 12 > +#define DA_EINT1EN_MASK 0x1 > +#define DA_EINT1EN_MASK_SFT (0x1 << 12) > +#define DA_EINT1INVEN_ADDR \ > + MT6359_ACCDET_CON29 > +#define DA_EINT1INVEN_SFT 13 > +#define DA_EINT1INVEN_MASK 0x1 > +#define DA_EINT1INVEN_MASK_SFT (0x1 << 13) > +#define DA_EINT1CEN_ADDR \ > + MT6359_ACCDET_CON29 > +#define DA_EINT1CEN_SFT 14 > +#define DA_EINT1CEN_MASK 0x1 > +#define DA_EINT1CEN_MASK_SFT (0x1 << 14) > +#define ACCDET_EN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EN_SFT 0 > +#define ACCDET_EN_MASK 0x1 > +#define ACCDET_EN_MASK_SFT (0x1 << 0) > +#define ACCDET_EINT0_EN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT0_EN_SFT 1 > +#define ACCDET_EINT0_EN_MASK 0x1 > +#define ACCDET_EINT0_EN_MASK_SFT (0x1 << 1) > +#define ACCDET_EINT1_EN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT1_EN_SFT 2 > +#define ACCDET_EINT1_EN_MASK 0x1 > +#define ACCDET_EINT1_EN_MASK_SFT (0x1 << 2) > +#define ACCDET_EINT0_M_EN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT0_M_EN_SFT 3 > +#define ACCDET_EINT0_M_EN_MASK 0x1 > +#define ACCDET_EINT0_M_EN_MASK_SFT (0x1 << 3) > +#define ACCDET_EINT0_DETECT_MOISTURE_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT0_DETECT_MOISTURE_SFT 4 > +#define ACCDET_EINT0_DETECT_MOISTURE_MASK 0x1 > +#define ACCDET_EINT0_DETECT_MOISTURE_MASK_SFT (0x1 << 4) > +#define ACCDET_EINT0_PLUG_IN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT0_PLUG_IN_SFT 5 > +#define ACCDET_EINT0_PLUG_IN_MASK 0x1 > +#define ACCDET_EINT0_PLUG_IN_MASK_SFT (0x1 << 5) > +#define ACCDET_EINT0_M_PLUG_IN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT0_M_PLUG_IN_SFT 6 > +#define ACCDET_EINT0_M_PLUG_IN_MASK 0x1 > +#define ACCDET_EINT0_M_PLUG_IN_MASK_SFT (0x1 << 6) > +#define ACCDET_EINT1_M_EN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT1_M_EN_SFT 7 > +#define ACCDET_EINT1_M_EN_MASK 0x1 > +#define ACCDET_EINT1_M_EN_MASK_SFT (0x1 << 7) > +#define ACCDET_EINT1_DETECT_MOISTURE_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT1_DETECT_MOISTURE_SFT 8 > +#define ACCDET_EINT1_DETECT_MOISTURE_MASK 0x1 > +#define ACCDET_EINT1_DETECT_MOISTURE_MASK_SFT (0x1 << 8) > +#define ACCDET_EINT1_PLUG_IN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT1_PLUG_IN_SFT 9 > +#define ACCDET_EINT1_PLUG_IN_MASK 0x1 > +#define ACCDET_EINT1_PLUG_IN_MASK_SFT (0x1 << 9) > +#define ACCDET_EINT1_M_PLUG_IN_ADDR \ > + MT6359_ACCDET_CON30 > +#define ACCDET_EINT1_M_PLUG_IN_SFT 10 > +#define ACCDET_EINT1_M_PLUG_IN_MASK 0x1 > +#define ACCDET_EINT1_M_PLUG_IN_MASK_SFT (0x1 << 10) > +#define ACCDET_CUR_DEB_ADDR \ > + MT6359_ACCDET_CON31 > +#define ACCDET_CUR_DEB_SFT 0 > +#define ACCDET_CUR_DEB_MASK 0xFFFF > +#define ACCDET_CUR_DEB_MASK_SFT (0xFFFF << 0) > +#define ACCDET_EINT0_CUR_DEB_ADDR \ > + MT6359_ACCDET_CON32 > +#define ACCDET_EINT0_CUR_DEB_SFT 0 > +#define ACCDET_EINT0_CUR_DEB_MASK 0x7FFF > +#define ACCDET_EINT0_CUR_DEB_MASK_SFT (0x7FFF << 0) > +#define ACCDET_EINT1_CUR_DEB_ADDR \ > + MT6359_ACCDET_CON33 > +#define ACCDET_EINT1_CUR_DEB_SFT 0 > +#define ACCDET_EINT1_CUR_DEB_MASK 0x7FFF > +#define ACCDET_EINT1_CUR_DEB_MASK_SFT (0x7FFF << 0) > +#define ACCDET_EINT0_INVERTER_CUR_DEB_ADDR \ > + MT6359_ACCDET_CON34 > +#define ACCDET_EINT0_INVERTER_CUR_DEB_SFT 0 > +#define ACCDET_EINT0_INVERTER_CUR_DEB_MASK 0x7FFF > +#define ACCDET_EINT0_INVERTER_CUR_DEB_MASK_SFT (0x7FFF << 0) > +#define ACCDET_EINT1_INVERTER_CUR_DEB_ADDR \ > + MT6359_ACCDET_CON35 > +#define ACCDET_EINT1_INVERTER_CUR_DEB_SFT 0 > +#define ACCDET_EINT1_INVERTER_CUR_DEB_MASK 0x7FFF > +#define ACCDET_EINT1_INVERTER_CUR_DEB_MASK_SFT (0x7FFF << 0) > +#define AD_AUDACCDETCMPOB_MON_ADDR \ > + MT6359_ACCDET_CON36 > +#define AD_AUDACCDETCMPOB_MON_SFT 0 > +#define AD_AUDACCDETCMPOB_MON_MASK 0x1 > +#define AD_AUDACCDETCMPOB_MON_MASK_SFT (0x1 << 0) > +#define AD_AUDACCDETCMPOA_MON_ADDR \ > + MT6359_ACCDET_CON36 > +#define AD_AUDACCDETCMPOA_MON_SFT 1 > +#define AD_AUDACCDETCMPOA_MON_MASK 0x1 > +#define AD_AUDACCDETCMPOA_MON_MASK_SFT (0x1 << 1) > +#define AD_EINT0CMPMOUT_MON_ADDR \ > + MT6359_ACCDET_CON36 > +#define AD_EINT0CMPMOUT_MON_SFT 2 > +#define AD_EINT0CMPMOUT_MON_MASK 0x1 > +#define AD_EINT0CMPMOUT_MON_MASK_SFT (0x1 << 2) > +#define AD_EINT0CMPOUT_MON_ADDR \ > + MT6359_ACCDET_CON36 > +#define AD_EINT0CMPOUT_MON_SFT 3 > +#define AD_EINT0CMPOUT_MON_MASK 0x1 > +#define AD_EINT0CMPOUT_MON_MASK_SFT (0x1 << 3) > +#define AD_EINT0INVOUT_MON_ADDR \ > + MT6359_ACCDET_CON36 > +#define AD_EINT0INVOUT_MON_SFT 4 > +#define AD_EINT0INVOUT_MON_MASK 0x1 > +#define AD_EINT0INVOUT_MON_MASK_SFT (0x1 << 4) > +#define AD_EINT1CMPMOUT_MON_ADDR \ > + MT6359_ACCDET_CON36 > +#define AD_EINT1CMPMOUT_MON_SFT 5 > +#define AD_EINT1CMPMOUT_MON_MASK 0x1 > +#define AD_EINT1CMPMOUT_MON_MASK_SFT (0x1 << 5) > +#define AD_EINT1CMPOUT_MON_ADDR \ > + MT6359_ACCDET_CON36 > +#define AD_EINT1CMPOUT_MON_SFT 6 > +#define AD_EINT1CMPOUT_MON_MASK 0x1 > +#define AD_EINT1CMPOUT_MON_MASK_SFT (0x1 << 6) > +#define AD_EINT1INVOUT_MON_ADDR \ > + MT6359_ACCDET_CON36 > +#define AD_EINT1INVOUT_MON_SFT 7 > +#define AD_EINT1INVOUT_MON_MASK 0x1 > +#define AD_EINT1INVOUT_MON_MASK_SFT (0x1 << 7) > +#define DA_AUDACCDETCMPCLK_MON_ADDR \ > + MT6359_ACCDET_CON37 > +#define DA_AUDACCDETCMPCLK_MON_SFT 0 > +#define DA_AUDACCDETCMPCLK_MON_MASK 0x1 > +#define DA_AUDACCDETCMPCLK_MON_MASK_SFT (0x1 << 0) > +#define DA_AUDACCDETVTHCLK_MON_ADDR \ > + MT6359_ACCDET_CON37 > +#define DA_AUDACCDETVTHCLK_MON_SFT 1 > +#define DA_AUDACCDETVTHCLK_MON_MASK 0x1 > +#define DA_AUDACCDETVTHCLK_MON_MASK_SFT (0x1 << 1) > +#define DA_AUDACCDETMBIASCLK_MON_ADDR \ > + MT6359_ACCDET_CON37 > +#define DA_AUDACCDETMBIASCLK_MON_SFT 2 > +#define DA_AUDACCDETMBIASCLK_MON_MASK 0x1 > +#define DA_AUDACCDETMBIASCLK_MON_MASK_SFT (0x1 << 2) > +#define DA_AUDACCDETAUXADCSWCTRL_MON_ADDR \ > + MT6359_ACCDET_CON37 > +#define DA_AUDACCDETAUXADCSWCTRL_MON_SFT 3 > +#define DA_AUDACCDETAUXADCSWCTRL_MON_MASK 0x1 > +#define DA_AUDACCDETAUXADCSWCTRL_MON_MASK_SFT (0x1 << 3) > +#define DA_EINT0CTURBO_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT0CTURBO_MON_SFT 0 > +#define DA_EINT0CTURBO_MON_MASK 0x1 > +#define DA_EINT0CTURBO_MON_MASK_SFT (0x1 << 0) > +#define DA_EINT0CMPMEN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT0CMPMEN_MON_SFT 1 > +#define DA_EINT0CMPMEN_MON_MASK 0x1 > +#define DA_EINT0CMPMEN_MON_MASK_SFT (0x1 << 1) > +#define DA_EINT0CMPEN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT0CMPEN_MON_SFT 2 > +#define DA_EINT0CMPEN_MON_MASK 0x1 > +#define DA_EINT0CMPEN_MON_MASK_SFT (0x1 << 2) > +#define DA_EINT0INVEN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT0INVEN_MON_SFT 3 > +#define DA_EINT0INVEN_MON_MASK 0x1 > +#define DA_EINT0INVEN_MON_MASK_SFT (0x1 << 3) > +#define DA_EINT0CEN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT0CEN_MON_SFT 4 > +#define DA_EINT0CEN_MON_MASK 0x1 > +#define DA_EINT0CEN_MON_MASK_SFT (0x1 << 4) > +#define DA_EINT0EN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT0EN_MON_SFT 5 > +#define DA_EINT0EN_MON_MASK 0x1 > +#define DA_EINT0EN_MON_MASK_SFT (0x1 << 5) > +#define DA_EINT1CTURBO_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT1CTURBO_MON_SFT 8 > +#define DA_EINT1CTURBO_MON_MASK 0x1 > +#define DA_EINT1CTURBO_MON_MASK_SFT (0x1 << 8) > +#define DA_EINT1CMPMEN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT1CMPMEN_MON_SFT 9 > +#define DA_EINT1CMPMEN_MON_MASK 0x1 > +#define DA_EINT1CMPMEN_MON_MASK_SFT (0x1 << 9) > +#define DA_EINT1CMPEN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT1CMPEN_MON_SFT 10 > +#define DA_EINT1CMPEN_MON_MASK 0x1 > +#define DA_EINT1CMPEN_MON_MASK_SFT (0x1 << 10) > +#define DA_EINT1INVEN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT1INVEN_MON_SFT 11 > +#define DA_EINT1INVEN_MON_MASK 0x1 > +#define DA_EINT1INVEN_MON_MASK_SFT (0x1 << 11) > +#define DA_EINT1CEN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT1CEN_MON_SFT 12 > +#define DA_EINT1CEN_MON_MASK 0x1 > +#define DA_EINT1CEN_MON_MASK_SFT (0x1 << 12) > +#define DA_EINT1EN_MON_ADDR \ > + MT6359_ACCDET_CON38 > +#define DA_EINT1EN_MON_SFT 13 > +#define DA_EINT1EN_MON_MASK 0x1 > +#define DA_EINT1EN_MON_MASK_SFT (0x1 << 13) > +#define ACCDET_EINT0_M_PLUG_IN_COUNT_ADDR \ > + MT6359_ACCDET_CON39 > +#define ACCDET_EINT0_M_PLUG_IN_COUNT_SFT 0 > +#define ACCDET_EINT0_M_PLUG_IN_COUNT_MASK 0x7 > +#define ACCDET_EINT0_M_PLUG_IN_COUNT_MASK_SFT (0x7 << 0) > +#define ACCDET_EINT1_M_PLUG_IN_COUNT_ADDR \ > + MT6359_ACCDET_CON39 > +#define ACCDET_EINT1_M_PLUG_IN_COUNT_SFT 4 > +#define ACCDET_EINT1_M_PLUG_IN_COUNT_MASK 0x7 > +#define ACCDET_EINT1_M_PLUG_IN_COUNT_MASK_SFT (0x7 << 4) > +#define ACCDET_MON_FLAG_EN_ADDR \ > + MT6359_ACCDET_CON40 > +#define ACCDET_MON_FLAG_EN_SFT 0 > +#define ACCDET_MON_FLAG_EN_MASK 0x1 > +#define ACCDET_MON_FLAG_EN_MASK_SFT (0x1 << 0) > +#define ACCDET_MON_FLAG_SEL_ADDR \ > + MT6359_ACCDET_CON40 > +#define ACCDET_MON_FLAG_SEL_SFT 4 > +#define ACCDET_MON_FLAG_SEL_MASK 0xF > +#define ACCDET_MON_FLAG_SEL_MASK_SFT (0xF << 4) > + > +#define RG_AUDPWDBMICBIAS0_ADDR \ > + MT6359_AUDENC_ANA_CON15 > +#define RG_AUDPWDBMICBIAS0_SFT 0 > +#define RG_AUDPWDBMICBIAS0_MASK 0x1 > +#define RG_AUDPWDBMICBIAS0_MASK_SFT (0x1 << 0) > +#define RG_AUDPREAMPLON_ADDR \ > + MT6359_AUDENC_ANA_CON0 > +#define RG_AUDPREAMPLON_SFT 0 > +#define RG_AUDPREAMPLON_MASK 0x1 > +#define RG_AUDPREAMPLON_MASK_SFT (0x1 << 0) > +#define RG_CLKSQ_EN_ADDR \ > + MT6359_AUDENC_ANA_CON23 > +#define RG_CLKSQ_EN_SFT 0 > +#define RG_CLKSQ_EN_MASK 0x1 > +#define RG_CLKSQ_EN_MASK_SFT (0x1 << 0) > +#define RG_RTC32K_CK_PDN_ADDR \ > + MT6359_TOP_CKPDN_CON0 > +#define RG_RTC32K_CK_PDN_SFT 15 > +#define RG_RTC32K_CK_PDN_MASK 0x1 > +#define RG_RTC32K_CK_PDN_MASK_SFT (0x1 << 15) > +#define RG_HPLOUTPUTSTBENH_VAUDP32_ADDR \ > + MT6359_AUDDEC_ANA_CON2 > +#define RG_HPLOUTPUTSTBENH_VAUDP32_SFT 0 > +#define RG_HPLOUTPUTSTBENH_VAUDP32_MASK 0x7 > +#define RG_HPLOUTPUTSTBENH_VAUDP32_MASK_SFT (0x7 << 0) > +#define AUXADC_RQST_CH5_ADDR \ > + MT6359_AUXADC_RQST0 > +#define AUXADC_RQST_CH5_SFT 5 > +#define AUXADC_RQST_CH5_MASK 0x1 > +#define AUXADC_RQST_CH5_MASK_SFT (0x1 << 5) > +#define RG_LDO_VUSB_HW0_OP_EN_ADDR \ > + MT6359_LDO_VUSB_OP_EN > +#define RG_LDO_VUSB_HW0_OP_EN_SFT 0 > +#define RG_LDO_VUSB_HW0_OP_EN_MASK 0x1 > +#define RG_LDO_VUSB_HW0_OP_EN_MASK_SFT (0x1 << 0) > +#define RG_HPROUTPUTSTBENH_VAUDP32_ADDR \ > + MT6359_AUDDEC_ANA_CON2 > +#define RG_HPROUTPUTSTBENH_VAUDP32_SFT 4 > +#define RG_HPROUTPUTSTBENH_VAUDP32_MASK 0x7 > +#define RG_HPROUTPUTSTBENH_VAUDP32_MASK_SFT (0x7 << 4) > +#define RG_NCP_PDDIS_EN_ADDR \ > + MT6359_AFE_NCP_CFG2 > +#define RG_NCP_PDDIS_EN_SFT 0 > +#define RG_NCP_PDDIS_EN_MASK 0x1 > +#define RG_NCP_PDDIS_EN_MASK_SFT (0x1 << 0) > +#define RG_SCK32K_CK_PDN_ADDR \ > + MT6359_TOP_CKPDN_CON0 > +#define RG_SCK32K_CK_PDN_SFT 0 > +#define RG_SCK32K_CK_PDN_MASK 0x1 > +#define RG_SCK32K_CK_PDN_MASK_SFT (0x1 << 0) > /* AUDENC_ANA_CON18: */ > -#define RG_ACCDET_MODE_ANA11_MODE1 (0x000f) > -#define RG_ACCDET_MODE_ANA11_MODE2 (0x008f) > -#define RG_ACCDET_MODE_ANA11_MODE6 (0x008f) > +#define RG_ACCDET_MODE_ANA11_MODE1 (0x000F) > +#define RG_ACCDET_MODE_ANA11_MODE2 (0x008F) > +#define RG_ACCDET_MODE_ANA11_MODE6 (0x008F) > > /* AUXADC_ADC5: Auxadc CH5 read data */ > #define AUXADC_DATA_RDY_CH5 BIT(15) > #define AUXADC_DATA_PROCEED_CH5 BIT(15) > -#define AUXADC_DATA_MASK (0x0fff) > +#define AUXADC_DATA_MASK (0x0FFF) > > /* AUXADC_RQST0_SET: Auxadc CH5 request, relevant 0x07EC */ > #define AUXADC_RQST_CH5_SET BIT(5) > /* AUXADC_RQST0_CLR: Auxadc CH5 request, relevant 0x07EC */ > #define AUXADC_RQST_CH5_CLR BIT(5) > > -#define ACCDET_CALI_MASK0 (0xff) > -#define ACCDET_CALI_MASK1 (0xff << 8) > -#define ACCDET_CALI_MASK2 (0xff) > -#define ACCDET_CALI_MASK3 (0xff << 8) > -#define ACCDET_CALI_MASK4 (0xff) > - > -#define ACCDET_EINT1_IRQ_CLR_B11 BIT(PMIC_ACCDET_EINT1_IRQ_CLR_SHIFT) > -#define ACCDET_EINT0_IRQ_CLR_B10 BIT(PMIC_ACCDET_EINT0_IRQ_CLR_SHIFT) > -#define ACCDET_EINT_IRQ_CLR_B10_11 (0x03 << \ > - PMIC_ACCDET_EINT0_IRQ_CLR_SHIFT) > -#define ACCDET_IRQ_CLR_B8 BIT(PMIC_ACCDET_IRQ_CLR_SHIFT) > +#define ACCDET_CALI_MASK0 (0xFF) > +#define ACCDET_CALI_MASK1 (0xFF << 8) > +#define ACCDET_CALI_MASK2 (0xFF) > +#define ACCDET_CALI_MASK3 (0xFF << 8) > +#define ACCDET_CALI_MASK4 (0xFF) > > -#define ACCDET_EINT1_IRQ_B3 BIT(PMIC_ACCDET_EINT1_IRQ_SHIFT) > -#define ACCDET_EINT0_IRQ_B2 BIT(PMIC_ACCDET_EINT0_IRQ_SHIFT) > -#define ACCDET_EINT_IRQ_B2_B3 (0x03 << PMIC_ACCDET_EINT0_IRQ_SHIFT) > -#define ACCDET_IRQ_B0 BIT(PMIC_ACCDET_IRQ_SHIFT) > +#define ACCDET_EINT_IRQ_B2_B3 (0x03 << ACCDET_EINT0_IRQ_SFT) > > /* ACCDET_CON25: RO, accdet FSM state,etc.*/ > -#define ACCDET_STATE_MEM_IN_OFFSET (PMIC_ACCDET_MEM_IN_SHIFT) > -#define ACCDET_STATE_AB_MASK (0x03) > -#define ACCDET_STATE_AB_00 (0x00) > -#define ACCDET_STATE_AB_01 (0x01) > -#define ACCDET_STATE_AB_10 (0x02) > -#define ACCDET_STATE_AB_11 (0x03) > +#define ACCDET_STATE_MEM_IN_OFFSET (ACCDET_MEM_IN_SFT) > +#define ACCDET_STATE_AB_MASK (0x03) > +#define ACCDET_STATE_AB_00 (0x00) > +#define ACCDET_STATE_AB_01 (0x01) > +#define ACCDET_STATE_AB_10 (0x02) > +#define ACCDET_STATE_AB_11 (0x03) > > /* ACCDET_CON19 */ > -#define ACCDET_EINT0_STABLE_VAL ((1 << PMIC_ACCDET_DA_STABLE_SHIFT) | \ > - (1 << PMIC_ACCDET_EINT0_EN_STABLE_SHIFT) | \ > - (1 << PMIC_ACCDET_EINT0_CMPEN_STABLE_SHIFT) | \ > - (1 << PMIC_ACCDET_EINT0_CEN_STABLE_SHIFT)) > - > -#define ACCDET_EINT1_STABLE_VAL ((1 << PMIC_ACCDET_DA_STABLE_SHIFT) | \ > - (1 << PMIC_ACCDET_EINT1_EN_STABLE_SHIFT) | \ > - (1 << PMIC_ACCDET_EINT1_CMPEN_STABLE_SHIFT) | \ > - (1 << PMIC_ACCDET_EINT1_CEN_STABLE_SHIFT)) > +#define ACCDET_EINT0_STABLE_VAL ((ACCDET_DA_STABLE_SFT) | \ > + (ACCDET_EINT0_EN_STABLE_SFT) | \ > + (ACCDET_EINT0_CMPEN_STABLE_SFT) | \ > + (ACCDET_EINT0_CEN_STABLE_SFT)) > + > +#define ACCDET_EINT1_STABLE_VAL ((ACCDET_DA_STABLE_SFT) | \ > + (ACCDET_EINT1_EN_STABLE_SFT) | \ > + (ACCDET_EINT1_CMPEN_STABLE_SFT) | \ > + (ACCDET_EINT1_CEN_STABLE_SFT)) > > /* The following are used for mt6359.c */ > /* MT6359_DCXO_CW12 */ > -- > 1.8.1.1.dirty >

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Re: [PATCH 1/2] dt-bindings: mediatek: mt6359: add ASoC mt6359 accdet document
by Argus Lin 03 Feb '21

03 Feb '21

Dear maintainers: Can you reply to me if any opinion? thanks B.R. Argus On Wed, 2021-01-06 at 20:19 +0800, Argus Lin wrote: > This patch adds MediaTek MT6359 accdet document. > > Signed-off-by: Argus Lin <argus.lin(a)mediatek.com> > --- > .../devicetree/bindings/sound/mt6359-accdet.yaml | 142 +++++++++++++++++++++ > 1 file changed, 142 insertions(+) > create mode 100644 Documentation/devicetree/bindings/sound/mt6359-accdet.yaml > > diff --git a/Documentation/devicetree/bindings/sound/mt6359-accdet.yaml b/Documentation/devicetree/bindings/sound/mt6359-accdet.yaml > new file mode 100644 > index 0000000..fde0d29 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/mt6359-accdet.yaml > @@ -0,0 +1,142 @@ > +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) > +%YAML 1.2 > +--- > +$id: http://devicetree.org/schemas/sound/mt6359-accdet.yaml# > +$schema: http://devicetree.org/meta-schemas/core.yaml# > + > +title: Mediatek Technologies Inc. MT6359 ASoC accdet driver bindings > + > +maintainers: > + - Chipeng Chang <chipeng.chang(a)mediatek.com> > + > +description: | > + This binding describes the Mediatek Technologies MT6359 ASoC > + accdet driver. > + > +properties: > + compatible: > + enum: > + - mediatek,mt6359-accdet > + - mediatek,pmic-accdet > + > + accdet-name: > + maxItems: 1 > + description: named as "mt6359-accdet" for snd_soc_card_jack. > + > + accdet-mic-vol: > + maxItems: 1 > + description: | > + accdet micbias1 voltage. > + > + enum: > + - 0x00 # 1.7v - micbias1 volage is 1.7v. > + - 0x01 # 1.8v - micbias1 volage is 1.8v. > + - 0x02 # 1.9v - micbias1 volage is 1.9v. > + - 0x03 # 2.0v - micbias1 volage is 2.0v. > + - 0x04 # 2.1v - micbias1 volage is 2.1v. > + - 0x05 # 2.5v - micbias1 volage is 2.5v. > + - 0x06 # 2.6v - micbias1 volage is 2.6v. > + - 0x07 # 2.7v - micbias1 volage is 2.7v. > + - 0x08 # 2.8v - micbias1 volage is 2.8v. > + - 0x09 # 2.85v - micbias1 volage is 2.85v. > + > + accdet-plugout-debounce: > + maxItems: 1 > + description: For using ap gpio eint only, it is sw debounce time. > + > + accdet-mic-mode: > + maxItems: 1 > + description: | > + value as 1, 2, 6 to indicate ACC/DCC mode. default is DCC mode 2. > + > + enum: > + - 0x01 # ACC - ACC mode. > + - 0x02 # DCC - low cost without in bias. > + - 0x06 # DCC - low cost with in bias. > + > + eint_use_ext_res: > + maxItems: 1 > + description: select HP_EINT pull up resistance, default 0 use internal resistance. > + > + headset-mode-setting: > + maxItems: 15 > + description: | > + headset-mode-setting : Indicates customized pwm, debounce setting > + accdet pwm_width, pwm_thresh, fall_delay, rise_delay > + debounce0, debounce1, debounce3, debounce4 > + eint pwm_width, eint pwm_thresh > + eint deb(debounce0, debounce1, debounce2, debounce3), inv_debounce. > + > + headset-use-ap-eint: > + maxItems: 1 > + description: indicates to use ap gpio. > + > + headset-eint-num: > + maxItems: 1 > + description: | > + incicates pmic eint usage. > + > + enum: > + - 0x0 # PMIC_EINT0 - use pmic eint0 to trigger plug interrupt. > + - 0x1 # PMIC_EINT1 - use pmic eint1 to trigger plug interrupt. > + - 0x2 # PMIC_BI_EINT - use pmic eint0/1 to trigger plug interrupt. > + > + headset-eint-trig-mode: > + maxItems: 1 > + description: | > + incicates pmic eint trigger mode. > + > + enum: > + - 0x0 # PMIC_GPIO - use pmic gpio to detect plug status by setting polarity to high or low. > + - 0x1 # PMIC_INVERTER - use pmic inverter to detect plug status. > + > + headset-key-mode: > + maxItems: 1 > + description: | > + incicates key mode type. > + > + enum: > + - 0x0 # THREE_KEY - support 3-key key mode. > + - 0x1 # FOUR_KEY - support 4-key key mode. > + - 0x1 # TRI_KEY_CDD - support 3-key google CDD key mode. > + > + headset-three-key-threshold: > + maxItems: 4 > + description: | > + 3 key device detection threshold: 0--MD_MAX--UP_MAX--DW_MAX. > + > + headset-three-key-threshold-CDD: > + maxItems: 4 > + description: | > + 3 key CDD device detection threshold: 0--MD_MAX--UP_MAX--DW_MAX. > + > + headset-four-key-threshold: > + maxItems: 4 > + description: | > + 4 key device detection threshold: 0--MD_MAX--VOICE_MAX--UP_MAX--DW_MAX. > + > +additionalProperties: false > + > +examples: > + - | > + accdet: accdet { > + compatible = "mediatek,mt6359-accdet"; > + accdet-name = "mt6359-accdet"; > + accdet-mic-vol = <8>; > + accdet-plugout-debounce = <1>; > + accdet-mic-mode = <2>; > + eint_use_ext_res = <0>; > + headset-mode-setting = <0x500 0x500 1 0x1f0 > + 0x800 0x800 0x20 0x44 > + 0x4 0x1 > + 0x5 0x3 0x3 0x5 0xe>; > + headset-use-ap-eint = <0>; > + headset-eint-num = <0>; > + headset-eint-trig-mode = <1>; > + headset-key-mode = <0>; > + headset-three-key-threshold = <0 80 220 400>; > + headset-three-key-threshold-CDD = <0 121 192 600>; > + headset-four-key-threshold = <0 58 121 192 400>; > + status = "okay"; > + }; > +... > -- > 1.8.1.1.dirty >

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Oops with "ALSA: jack: implement software jack injection via debugfs"
by Chris Wilson 03 Feb '21

03 Feb '21

commit 2d670ea2bd53 ("ALSA: jack: implement software jack injection via debugfs") is causing issues for our CI as we see a use-after-free on module unload (on all machines): https://intel-gfx-ci.01.org/tree/drm-tip/CI_DRM_9715/fi-skl-6700k2/pstore0-… <4> [241.294412] general protection fault, probably for non-canonical address 0x6b6b6b6b6b6b6c13: 0000 [#1] PREEMPT SMP PTI <4> [241.294424] CPU: 7 PID: 5820 Comm: i915_module_loa Not tainted 5.11.0-rc6-CI-CI_DRM_9715+ #1 <4> [241.294432] Hardware name: System manufacturer System Product Name/Z170 PRO GAMING, BIOS 0802 09/02/2015 <4> [241.294438] RIP: 0010:__lock_acquire+0x61a/0x25a0 <4> [241.294444] Code: 00 00 83 f8 2f 0f 87 a0 00 00 00 3b 05 97 61 07 02 c7 44 24 18 01 00 00 00 0f 86 d4 00 00 00 89 05 83 61 07 02 e9 c9 00 00 00 <48> 81 3f 80 14 d9 82 41 bc 00 00 00 00 45 0f 45 e0 83 fe 01 0f 87 <4> [241.294451] RSP: 0018:ffffc90000e77b78 EFLAGS: 00010002 <4> [241.294454] RAX: 0000000000000000 RBX: ffff888137d50040 RCX: 0000000000000000 <4> [241.294458] RDX: 0000000000000000 RSI: 0000000000000000 RDI: 6b6b6b6b6b6b6c13 <4> [241.294461] RBP: 0000000000000001 R08: 0000000000000001 R09: 0000000000000000 <4> [241.294464] R10: 0000000000000001 R11: 00000000d2337ff6 R12: 0000000000000001 <4> [241.294467] R13: 0000000000000000 R14: 0000000000000000 R15: 6b6b6b6b6b6b6c13 <4> [241.294470] FS: 00007f0e00616e40(0000) GS:ffff88824db80000(0000) knlGS:0000000000000000 <4> [241.294474] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 <4> [241.294477] CR2: 00007f375bda2cb0 CR3: 000000010ddd6001 CR4: 00000000003706e0 <4> [241.294480] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 <4> [241.294483] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 <4> [241.294486] Call Trace: <4> [241.294489] lock_acquire+0xd0/0x3b0 <4> [241.294493] ? lockref_get+0x9/0x20 <4> [241.294498] _raw_spin_lock+0x2a/0x40 <4> [241.294501] ? lockref_get+0x9/0x20 <4> [241.294505] lockref_get+0x9/0x20 <4> [241.294508] simple_recursive_removal+0x31/0x2a0 <4> [241.294511] ? debugfs_remove+0x50/0x50 <4> [241.294515] ? _raw_spin_unlock+0x24/0x40 <4> [241.294519] debugfs_remove+0x3b/0x50 <4> [241.294522] snd_card_free+0x76/0xa0 <4> [241.294527] pci_device_remove+0x36/0xb0 <4> [241.294531] device_release_driver_internal+0xf2/0x1d0 <4> [241.294820] unbind_store+0xeb/0x120 <4> [241.294824] kernfs_fop_write_iter+0x11d/0x1c0 <4> [241.294828] new_sync_write+0x11d/0x1b0 <4> [241.294832] vfs_write+0x260/0x390 <4> [241.294835] ksys_write+0x5a/0xd0 <4> [241.294838] do_syscall_64+0x33/0x80 <4> [241.294842] entry_SYSCALL_64_after_hwframe+0x44/0xa9 <4> [241.294845] RIP: 0033:0x7f0dffd80281 <4> [241.294848] Code: c3 0f 1f 84 00 00 00 00 00 48 8b 05 59 8d 20 00 c3 0f 1f 84 00 00 00 00 00 8b 05 8a d1 20 00 85 c0 75 16 b8 01 00 00 00 0f 05 <48> 3d 00 f0 ff ff 77 57 f3 c3 0f 1f 44 00 00 41 54 55 49 89 d4 53 <4> [241.294855] RSP: 002b:00007fff5a3a9c88 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 <4> [241.294860] RAX: ffffffffffffffda RBX: 0000555708dc2cf0 RCX: 00007f0dffd80281 <4> [241.294863] RDX: 000000000000000c RSI: 0000555708dc2ddb RDI: 0000000000000006 <4> [241.294866] RBP: 00007fff5a3a9d20 R08: 0000555708dc2ddb R09: 000000000000000c <4> [241.294869] R10: 00000000fffffff4 R11: 0000000000000246 R12: 00007fff5a3a9c90 <4> [241.294872] R13: 00007f0e00007b18 R14: 0000000000000006 R15: 0000555708dc2ddb <4> [241.294877] Modules linked in: vgem snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic ledtrig_audio i915 mei_hdcp x86_pkg_temp_thermal coretemp crct10dif_pclmul crc32_pclmul snd_hda_intel snd_intel_dspcfg ghash_clmulni_intel snd_hda_codec snd_hwdep snd_hda_core e1000e mei_me snd_pcm mei ptp pps_core prime_numbers [last unloaded: vgem] <4> [241.294901] ---[ end trace 98e116c0344cf275 ]--- <4> [242.399691] RIP: 0010:__lock_acquire+0x61a/0x25a0 <4> [242.399698] Code: 00 00 83 f8 2f 0f 87 a0 00 00 00 3b 05 97 61 07 02 c7 44 24 18 01 00 00 00 0f 86 d4 00 00 00 89 05 83 61 07 02 e9 c9 00 00 00 <48> 81 3f 80 14 d9 82 41 bc 00 00 00 00 45 0f 45 e0 83 fe 01 0f 87 <4> [242.399705] RSP: 0018:ffffc90000e77b78 EFLAGS: 00010002 <4> [242.399709] RAX: 0000000000000000 RBX: ffff888137d50040 RCX: 0000000000000000 <4> [242.399713] RDX: 0000000000000000 RSI: 0000000000000000 RDI: 6b6b6b6b6b6b6c13 <4> [242.399716] RBP: 0000000000000001 R08: 0000000000000001 R09: 0000000000000000 <4> [242.399719] R10: 0000000000000001 R11: 00000000d2337ff6 R12: 0000000000000001 <4> [242.399722] R13: 0000000000000000 R14: 0000000000000000 R15: 6b6b6b6b6b6b6c13 <4> [242.399725] FS: 00007f0e00616e40(0000) GS:ffff88824db80000(0000) knlGS:0000000000000000 <4> [242.399729] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 <4> [242.399732] CR2: 00007f375bda2cb0 CR3: 000000010ddd6001 CR4: 00000000003706e0 <4> [242.399735] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 <4> [242.399739] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 <6> [242.399742] note: i915_module_loa[5820] exited with preempt_count 1 <3> [242.399745] BUG: sleeping function called from invalid context at include/linux/percpu-rwsem.h:49 <3> [242.399749] in_atomic(): 0, irqs_disabled(): 1, non_block: 0, pid: 5820, name: i915_module_loa <4> [242.399753] INFO: lockdep is turned off. <4> [242.399755] irq event stamp: 125679 <4> [242.399757] hardirqs last enabled at (125679): [<ffffffff81a9e36f>] _raw_spin_unlock_irq+0x1f/0x40 <4> [242.399763] hardirqs last disabled at (125678): [<ffffffff81a9e151>] _raw_spin_lock_irq+0x41/0x50 <4> [242.399768] softirqs last enabled at (125054): [<ffffffff81e00342>] __do_softirq+0x342/0x48e <4> [242.399773] softirqs last disabled at (124021): [<ffffffff81c00f52>] asm_call_irq_on_stack+0x12/0x20 <4> [242.399779] CPU: 7 PID: 5820 Comm: i915_module_loa Tainted: G D 5.11.0-rc6-CI-CI_DRM_9715+ #1 <4> [242.399784] Hardware name: System manufacturer System Product Name/Z170 PRO GAMING, BIOS 0802 09/02/2015 <4> [242.399787] Call Trace: <4> [242.399790] dump_stack+0x77/0x97 <4> [242.399794] ___might_sleep.cold.120+0xf2/0x106 <4> [242.399799] exit_signals+0x2b/0x350 <4> [242.399803] do_exit+0xc8/0xcb0 <4> [242.399807] ? ksys_write+0x5a/0xd0 <4> [242.399811] rewind_stack_do_exit+0x17/0x17 <4> [242.399814] RIP: 0033:0x7f0dffd80281 <4> [242.399817] Code: c3 0f 1f 84 00 00 00 00 00 48 8b 05 59 8d 20 00 c3 0f 1f 84 00 00 00 00 00 8b 05 8a d1 20 00 85 c0 75 16 b8 01 00 00 00 0f 05 <48> 3d 00 f0 ff ff 77 57 f3 c3 0f 1f 44 00 00 41 54 55 49 89 d4 53 <4> [242.399824] RSP: 002b:00007fff5a3a9c88 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 <4> [242.399828] RAX: ffffffffffffffda RBX: 0000555708dc2cf0 RCX: 00007f0dffd80281 <4> [242.399831] RDX: 000000000000000c RSI: 0000555708dc2ddb RDI: 0000000000000006 <4> [242.399834] RBP: 00007fff5a3a9d20 R08: 0000555708dc2ddb R09: 000000000000000c <4> [242.399838] R10: 00000000fffffff4 R11: 0000000000000246 R12: 00007fff5a3a9c90 <4> [242.399841] R13: 00007f0e00007b18 R14: 0000000000000006 R15: 0000555708dc2ddb Created at 2021-02-02 17:16:13

3 4

[PATCH] ALSA: core: Fix the debugfs removal at snd_card_free()
by Takashi Iwai 02 Feb '21

02 Feb '21

The commit 2d670ea2bd53 ("ALSA: jack: implement software jack injection via debugfs") introduced a debugfs root for each sound card object. The debugfs entry gets removed at the card removal, too, but it turned out that debugfs_remove() is called at a wrong place; it's after the card object gets freed, hence it leads to use-after-free. Fix it by moving the debugfs_remove() at the right place, the destructor of the card device. Fixes: 2d670ea2bd53 ("ALSA: jack: implement software jack injection via debugfs") Reported-and-tested-by: Chris Wilson <chris(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/161228343605.1150.8862281636043446562@build.alpor… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index d4e78b176793..84b573e9c1f9 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -487,6 +487,10 @@ static int snd_card_do_free(struct snd_card *card) dev_warn(card->dev, "unable to free card info\n"); /* Not fatal error */ } +#ifdef CONFIG_SND_DEBUG + debugfs_remove(card->debugfs_root); + card->debugfs_root = NULL; +#endif if (card->release_completion) complete(card->release_completion); kfree(card); @@ -537,11 +541,6 @@ int snd_card_free(struct snd_card *card) /* wait, until all devices are ready for the free operation */ wait_for_completion(&released); -#ifdef CONFIG_SND_DEBUG - debugfs_remove(card->debugfs_root); - card->debugfs_root = NULL; -#endif - return 0; } EXPORT_SYMBOL(snd_card_free); -- 2.26.2

1 0

Nux Mighty Plug USB - no clock source
by Peter Allen 02 Feb '21

02 Feb '21

I have a NUX Mighty Plug guitar effects which has a USB Audio interface. Unfortunately it doesn't work in Linux, and I'd love to fix it. It gets registered with Alsa, except it ends up without any valid sample rates (by the look of it due to a buggy clock source descriptor). This is on Ubuntu-Studio 20.04 LTS _Focal Fossa with kernel 5.4.0-62-lowlatency on a HP Omen 15 with i5-7300HQ I don't know what I'm doing (but did a little module development rather a long time ago in the 2.2 days :-)) Help is greatly appreciated. When plugged in I get: [12476.413792] usb 1-1: New USB device found, idVendor=1fc9, idProduct=8260, bcdDevice= 1.00 [12476.413797] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [12476.413800] usb 1-1: Product: MIGHTY PLUG USB [12476.413804] usb 1-1: Manufacturer: NUX [12476.413807] usb 1-1: SerialNumber: 2007151403 [12476.444988] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.444992] usb 1-1: parse_audio_format_rates_v2v3(): unable to find clock source (clock -22) [12476.445413] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.445479] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.447095] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.447098] usb 1-1: parse_audio_format_rates_v2v3(): unable to find clock source (clock -22) [12476.447459] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.447684] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.450397] usbcore: registered new interface driver snd-usb-audio ===== /proc/asound/cards give ====== 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xb4428000 irq 134 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xb4000000 irq 17 2 [USB ]: USB-Audio - MIGHTY PLUG USB NUX MIGHTY PLUG USB at usb-0000:00:14.0-1, high speed ==== /proc/asound/stream======== NUX MIGHTY PLUG USB at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: Data packet interval: 1000 us Bits: 24 Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 2 IN (ASYNC) Rates: Data packet interval: 1000 us Bits: 24 ======== arecord =========== arecord -D "hw:2" -c 2 -f s24 -r 44100 -d 2 -v test.wav Recording WAVE 'xxxc' : Signed 24 bit Little Endian, Rate 44100 Hz, Stereo arecord: set_params:1314: Broken configuration for this PCM: no configuration available ========lsusb -v ========== iSerial 2 200901010001 bNumConfigurations 1 Bus 001 Device 005: ID 1fc9:8260 NXP Semiconductors MIGHTY PLUG USB Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x1fc9 NXP Semiconductors idProduct 0x8260 bcdDevice 1.00 iManufacturer 1 NUX iProduct 2 MIGHTY PLUG USB iSerial 3 2007151403 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x013e bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 2mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 0x0065 bmControls 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 41 bmAttributes 3 Internal programmable clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bClockID 40 bNrInPins 1 baCSourceID(0) 41 bmControls 0x03 Clock Selector Control (read/write) iClockSelector 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bmaControls(0) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls(1) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls(2) 0x0000000f Mute Control (read/write) Volume Control (read/write) iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 20 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 10 bCSourceID 40 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 22 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 2 bCSourceID 40 bmControls 0x0000 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 4 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 4 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 4 MIGHTY PLUG USB MIDI MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0041 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0001 Self Powered Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 5.04 iManufacturer 3 Linux 5.4.0-58-lowlatency xhci-hcd iProduct 2 xHCI Host Controller iSerial 1 0000:00:14.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0019 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 13 bDescriptorType 41 nNbrPorts 16 wHubCharacteristic 0x000a No power switching (usb 1.0) Per-port overcurrent protection TT think time 8 FS bits bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x90 0x01 0x00 PortPwrCtrlMask 0xff 0xff 0xff Hub Port Status: Port 1: 0000.0503 highspeed power enable connect Port 2: 0000.0100 power Port 3: 0000.0100 power Port 4: 0000.0503 highspeed power enable connect Port 5: 0000.0100 power Port 6: 0000.0100 power Port 7: 0000.0103 power enable connect Port 8: 0000.0100 power Port 9: 0000.0100 power Port 10: 0000.0100 power Port 11: 0000.0100 power Port 12: 0000.0100 power Port 13: 0000.0100 power Port 14: 0000.0100 power Port 15: 0000.0100 power Port 16: 0000.0100 power can't get device qualifier: Resource temporarily unavailable can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0001 Self Powered

2 3

[PATCH v2 0/2] ALSA: usb-audio: Add Pioneer DJM-450 support
by Olivia Mackintosh 02 Feb '21

02 Feb '21

Re-roll of this patchset to fix the following: 1. Missing 'Signed-of-by:' line 2. Spaces around operators 3. Test that I'm threading submissions correctly Apologies for the clumsiness on my part. Olivia Olivia Mackintosh (2): ALSA: usb-audio: Add DJM450 to Pioneer format quirk ALSA: usb-audio: Add DJM-450 to the quirks table sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 20 ++++++++++++++ 2 files changed, 77 insertions(+) -- 2.30.0

2 3

[PATCH] ASoC: fsl_spdif: Utilize the defined parameter to clear code
by Tang Bin 02 Feb '21

02 Feb '21

Utilize the defined parameter 'dev' to make the code cleaner. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/fsl/fsl_spdif.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 455f96908..b6d5563df 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1215,7 +1215,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, for (i = 0; i < STC_TXCLK_SRC_MAX; i++) { sprintf(tmp, "rxtx%d", i); - clk = devm_clk_get(&pdev->dev, tmp); + clk = devm_clk_get(dev, tmp); if (IS_ERR(clk)) { dev_err(dev, "no rxtx%d clock in devicetree\n", i); return PTR_ERR(clk); @@ -1237,14 +1237,14 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, break; } - dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n", + dev_dbg(dev, "use rxtx%d as tx clock source for %dHz sample rate\n", spdif_priv->txclk_src[index], rate[index]); - dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n", + dev_dbg(dev, "use txclk df %d for %dHz sample rate\n", spdif_priv->txclk_df[index], rate[index]); if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk)) - dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n", + dev_dbg(dev, "use sysclk df %d for %dHz sample rate\n", spdif_priv->sysclk_df[index], rate[index]); - dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n", + dev_dbg(dev, "the best rate for %dHz sample rate is %dHz\n", rate[index], spdif_priv->txrate[index]); return 0; -- 2.20.1.windows.1

3 2

[PATCH] ASoC: topology: KUnit: Convert from cpu to data format
by Amadeusz Sławiński 02 Feb '21

02 Feb '21

When creating topology templates and overriding data in specific test cases it should be done with cpu_to_le32 macro, so we operate on correct data on all architectures, as topology parser use le32_to_cpu to parse data from structures. Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology-test.c | 66 +++++++++++++++++------------------ 1 file changed, 33 insertions(+), 33 deletions(-) diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index 36e2a3486dbf..ae3968161509 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -121,19 +121,19 @@ struct tplg_tmpl_001 { static struct tplg_tmpl_001 tplg_tmpl_empty = { .header = { - .magic = SND_SOC_TPLG_MAGIC, - .abi = 5, + .magic = cpu_to_le32(SND_SOC_TPLG_MAGIC), + .abi = cpu_to_le32(5), .version = 0, - .type = SND_SOC_TPLG_TYPE_MANIFEST, - .size = sizeof(struct snd_soc_tplg_hdr), + .type = cpu_to_le32(SND_SOC_TPLG_TYPE_MANIFEST), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_hdr)), .vendor_type = 0, - .payload_size = sizeof(struct snd_soc_tplg_manifest), + .payload_size = cpu_to_le32(sizeof(struct snd_soc_tplg_manifest)), .index = 0, - .count = 1, + .count = cpu_to_le32(1), }, .manifest = { - .size = sizeof(struct snd_soc_tplg_manifest), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_manifest)), /* rest of fields is 0 */ }, }; @@ -149,60 +149,60 @@ struct tplg_tmpl_002 { static struct tplg_tmpl_002 tplg_tmpl_with_pcm = { .header = { - .magic = SND_SOC_TPLG_MAGIC, - .abi = 5, + .magic = cpu_to_le32(SND_SOC_TPLG_MAGIC), + .abi = cpu_to_le32(5), .version = 0, - .type = SND_SOC_TPLG_TYPE_MANIFEST, - .size = sizeof(struct snd_soc_tplg_hdr), + .type = cpu_to_le32(SND_SOC_TPLG_TYPE_MANIFEST), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_hdr)), .vendor_type = 0, - .payload_size = sizeof(struct snd_soc_tplg_manifest), + .payload_size = cpu_to_le32(sizeof(struct snd_soc_tplg_manifest)), .index = 0, - .count = 1, + .count = cpu_to_le32(1), }, .manifest = { - .size = sizeof(struct snd_soc_tplg_manifest), - .pcm_elems = 1, + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_manifest)), + .pcm_elems = cpu_to_le32(1), /* rest of fields is 0 */ }, .pcm_header = { - .magic = SND_SOC_TPLG_MAGIC, - .abi = 5, + .magic = cpu_to_le32(SND_SOC_TPLG_MAGIC), + .abi = cpu_to_le32(5), .version = 0, - .type = SND_SOC_TPLG_TYPE_PCM, - .size = sizeof(struct snd_soc_tplg_hdr), + .type = cpu_to_le32(SND_SOC_TPLG_TYPE_PCM), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_hdr)), .vendor_type = 0, - .payload_size = sizeof(struct snd_soc_tplg_pcm), + .payload_size = cpu_to_le32(sizeof(struct snd_soc_tplg_pcm)), .index = 0, - .count = 1, + .count = cpu_to_le32(1), }, .pcm = { - .size = sizeof(struct snd_soc_tplg_pcm), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_pcm)), .pcm_name = "KUNIT Audio", .dai_name = "kunit-audio-dai", .pcm_id = 0, .dai_id = 0, - .playback = 1, - .capture = 1, + .playback = cpu_to_le32(1), + .capture = cpu_to_le32(1), .compress = 0, .stream = { [0] = { - .channels = 2, + .channels = cpu_to_le32(2), }, [1] = { - .channels = 2, + .channels = cpu_to_le32(2), }, }, .num_streams = 0, .caps = { [0] = { .name = "kunit-audio-playback", - .channels_min = 2, - .channels_max = 2, + .channels_min = cpu_to_le32(2), + .channels_max = cpu_to_le32(2), }, [1] = { .name = "kunit-audio-capture", - .channels_min = 2, - .channels_max = 2, + .channels_min = cpu_to_le32(2), + .channels_max = cpu_to_le32(2), }, }, .flag_mask = 0, @@ -460,7 +460,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_magic(struct kunit *test) * override abi * any value != magic number is wrong */ - data->header.magic = SND_SOC_TPLG_MAGIC + 1; + data->header.magic = cpu_to_le32(SND_SOC_TPLG_MAGIC + 1); kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; @@ -516,7 +516,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_abi(struct kunit *test) * override abi * any value != accepted range is wrong */ - data->header.abi = SND_SOC_TPLG_ABI_VERSION + 1; + data->header.abi = cpu_to_le32(SND_SOC_TPLG_ABI_VERSION + 1); kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; @@ -572,7 +572,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_size(struct kunit *test) * override size * any value != struct size is wrong */ - data->header.size = sizeof(struct snd_soc_tplg_hdr) + 1; + data->header.size = cpu_to_le32(sizeof(struct snd_soc_tplg_hdr) + 1); kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; -- 2.25.1

2 1

[PATCH] ASoC: qcom: Fix typo error in HDMI regmap config callbacks
by Srinivasa Rao Mandadapu 02 Feb '21

02 Feb '21

Had a typo in lpass platform driver that resulted in crash during suspend/resume with an HDMI dongle connected. The regmap read/write/volatile regesters validation callbacks in lpass-cpu were using MI2S rdma_channels count instead of hdmi_rdma_channels. This typo error causing to read registers from the regmap beyond the length of the mapping created by ioremap(). This fix avoids the need for reducing number hdmi_rdma_channels, which is done in commit 7dfe20ee92f6 ("ASoC: qcom: Fix number of HDMI RDMA channels on sc7180"). So reverting the same. Fixes: 7cb37b7bd0d3c ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 8 ++++---- sound/soc/qcom/lpass-sc7180.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index a669202e0001..c642e5f8f28c 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -595,7 +595,7 @@ static bool lpass_hdmi_regmap_writeable(struct device *dev, unsigned int reg) return true; } - for (i = 0; i < v->rdma_channels; ++i) { + for (i = 0; i < v->hdmi_rdma_channels; ++i) { if (reg == LPAIF_HDMI_RDMACTL_REG(v, i)) return true; if (reg == LPAIF_HDMI_RDMABASE_REG(v, i)) @@ -641,7 +641,7 @@ static bool lpass_hdmi_regmap_readable(struct device *dev, unsigned int reg) if (reg == LPASS_HDMITX_APP_IRQSTAT_REG(v)) return true; - for (i = 0; i < v->rdma_channels; ++i) { + for (i = 0; i < v->hdmi_rdma_channels; ++i) { if (reg == LPAIF_HDMI_RDMACTL_REG(v, i)) return true; if (reg == LPAIF_HDMI_RDMABASE_REG(v, i)) @@ -668,7 +668,7 @@ static bool lpass_hdmi_regmap_volatile(struct device *dev, unsigned int reg) if (reg == LPASS_HDMI_TX_LEGACY_ADDR(v)) return true; - for (i = 0; i < v->rdma_channels; ++i) { + for (i = 0; i < v->hdmi_rdma_channels; ++i) { if (reg == LPAIF_HDMI_RDMACURR_REG(v, i)) return true; } @@ -812,7 +812,7 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) return PTR_ERR(drvdata->hdmiif); lpass_hdmi_regmap_config.max_register = LPAIF_HDMI_RDMAPER_REG(variant, - variant->hdmi_rdma_channels); + variant->hdmi_rdma_channels - 1); drvdata->hdmiif_map = devm_regmap_init_mmio(dev, drvdata->hdmiif, &lpass_hdmi_regmap_config); if (IS_ERR(drvdata->hdmiif_map)) { diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 735c9dac28f2..8c168d3c589e 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -171,7 +171,7 @@ static struct lpass_variant sc7180_data = { .rdma_channels = 5, .hdmi_rdma_reg_base = 0x64000, .hdmi_rdma_reg_stride = 0x1000, - .hdmi_rdma_channels = 3, + .hdmi_rdma_channels = 4, .dmactl_audif_start = 1, .wrdma_reg_base = 0x18000, .wrdma_reg_stride = 0x1000, -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[PATCH 0/6] soundwire: qcom: various improvements
by Srinivas Kandagatla 02 Feb '21

02 Feb '21

During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Srinivas Kandagatla (6): soundwire: qcom: add support to missing transport params soundwire: qcom: extract version field soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts drivers/soundwire/qcom.c | 471 ++++++++++++++++++++++++++++++--------- 1 file changed, 366 insertions(+), 105 deletions(-) -- 2.21.0

3 29

hdspmixer: Hardware output loopback toggle buttons
by GitHub pull_request - edited 02 Feb '21

02 Feb '21

alsa-project/alsa-tools pull request #4 was edited from pfandl: Hi, as promised in the [sound/pci/rme9652 patch](https://github.com/tiwai/sound/commit/da2a040ee7cfe1dd57d5bec7906cb9… here are the user land changes which provide the user the ability to toggle the hardware output loopback functionality per output channel. I tried to hide the buttons for devices which do not expose the snd controls yet, but due to probably unnecessary struggle i cancelled that. Please let me know though should you want them removed for not yet enabled devices. Request URL : https://github.com/alsa-project/alsa-tools/pull/4 Patch URL : https://github.com/alsa-project/alsa-tools/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH] ASoC: SOF: add be_hw_params_fixup() for ALH
by Kai Vehmanen 02 Feb '21

02 Feb '21

From: Rander Wang <rander.wang(a)intel.com> Fixup BE DAI links channel count to match topology settings. Normally the channel count of BE is equal to FE's so we don't have any issue. For some cases like DSM with 2-channel FE and 4-channel BE the mismatch of BE and topology will result in audio issues. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Keyon Jie <yang.jie(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 0dc39fbcd81d..61c3fe17342d 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -707,7 +707,12 @@ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_pa } break; case SOF_DAI_INTEL_ALH: - /* do nothing for ALH dai_link */ + /* + * Dai could run with different channel count compared with + * front end, so get dai channel count from topology + */ + channels->min = dai->dai_config->alh.channels; + channels->max = dai->dai_config->alh.channels; break; case SOF_DAI_IMX_ESAI: rate->min = dai->dai_config->esai.fsync_rate; base-commit: 8ac9e476b86851c94e0f33bea758e0a00e3f875e -- 2.29.2

2 1

[PATCH] ASoC: mt6359: reduce log verbosity for optional DT properties
by Tzung-Bi Shih 02 Feb '21

02 Feb '21

DT properties "dmic-mode" and "mic-type-X" are optional. Reduces the log verbosity and changes the message a bit to avoid misleading. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Inspired by https://www.spinics.net/lists/arm-kernel/msg872744.html. sound/soc/codecs/mt6359.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 6de0d744fa9e..6f4b1da52082 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -2754,7 +2754,8 @@ static int mt6359_parse_dt(struct mt6359_priv *priv) ret = of_property_read_u32(np, "mediatek,dmic-mode", &priv->dmic_one_wire_mode); if (ret) { - dev_warn(priv->dev, "%s() failed to read dmic-mode\n", + dev_info(priv->dev, + "%s() failed to read dmic-mode, use default (0)\n", __func__); priv->dmic_one_wire_mode = 0; } @@ -2762,24 +2763,27 @@ static int mt6359_parse_dt(struct mt6359_priv *priv) ret = of_property_read_u32(np, "mediatek,mic-type-0", &priv->mux_select[MUX_MIC_TYPE_0]); if (ret) { - dev_warn(priv->dev, "%s() failed to read mic-type-0\n", - __func__); + dev_info(priv->dev, + "%s() failed to read mic-type-0, use default (%d)\n", + __func__, MIC_TYPE_MUX_IDLE); priv->mux_select[MUX_MIC_TYPE_0] = MIC_TYPE_MUX_IDLE; } ret = of_property_read_u32(np, "mediatek,mic-type-1", &priv->mux_select[MUX_MIC_TYPE_1]); if (ret) { - dev_warn(priv->dev, "%s() failed to read mic-type-1\n", - __func__); + dev_info(priv->dev, + "%s() failed to read mic-type-1, use default (%d)\n", + __func__, MIC_TYPE_MUX_IDLE); priv->mux_select[MUX_MIC_TYPE_1] = MIC_TYPE_MUX_IDLE; } ret = of_property_read_u32(np, "mediatek,mic-type-2", &priv->mux_select[MUX_MIC_TYPE_2]); if (ret) { - dev_warn(priv->dev, "%s() failed to read mic-type-2\n", - __func__); + dev_info(priv->dev, + "%s() failed to read mic-type-2, use default (%d)\n", + __func__, MIC_TYPE_MUX_IDLE); priv->mux_select[MUX_MIC_TYPE_2] = MIC_TYPE_MUX_IDLE; } -- 2.30.0.365.g02bc693789-goog

2 1

[PATCH AUTOSEL 5.4 07/17] ASoC: ak4458: correct reset polarity
by Sasha Levin 02 Feb '21

02 Feb '21

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> [ Upstream commit e953daeb68b1abd8a7d44902786349fdeef5c297 ] Reset (aka power off) happens when the reset gpio is made active. Change function name to ak4458_reset to match devicetree property "reset-gpios" Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org> Link: https://lore.kernel.org/r/ce650f47-4ff6-e486-7846-cc3d033f3601@blennerhasse… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/ak4458.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 71562154c0b1e..217e8ce9a4ba4 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -523,18 +523,10 @@ static struct snd_soc_dai_driver ak4497_dai = { .ops = &ak4458_dai_ops, }; -static void ak4458_power_off(struct ak4458_priv *ak4458) +static void ak4458_reset(struct ak4458_priv *ak4458, bool active) { if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); - usleep_range(1000, 2000); - } -} - -static void ak4458_power_on(struct ak4458_priv *ak4458) -{ - if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 1); + gpiod_set_value_cansleep(ak4458->reset_gpiod, active); usleep_range(1000, 2000); } } @@ -548,7 +540,7 @@ static int ak4458_init(struct snd_soc_component *component) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_on(ak4458); + ak4458_reset(ak4458, false); ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); /* ACKS bit = 1; 10000000 */ @@ -571,7 +563,7 @@ static void ak4458_remove(struct snd_soc_component *component) { struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - ak4458_power_off(ak4458); + ak4458_reset(ak4458, true); } #ifdef CONFIG_PM @@ -581,7 +573,7 @@ static int __maybe_unused ak4458_runtime_suspend(struct device *dev) regcache_cache_only(ak4458->regmap, true); - ak4458_power_off(ak4458); + ak4458_reset(ak4458, true); if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 0); @@ -596,8 +588,8 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_off(ak4458); - ak4458_power_on(ak4458); + ak4458_reset(ak4458, true); + ak4458_reset(ak4458, false); regcache_cache_only(ak4458->regmap, false); regcache_mark_dirty(ak4458->regmap); -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 04/17] ASoC: Intel: Skylake: Zero snd_ctl_elem_value
by Sasha Levin 02 Feb '21

02 Feb '21

From: Ricardo Ribalda <ribalda(a)chromium.org> [ Upstream commit 1d8fe0648e118fd495a2cb393a34eb8d428e7808 ] Clear struct snd_ctl_elem_value before calling ->put() to avoid any data leak. Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Link: https://lore.kernel.org/r/20210121171644.131059-2-ribalda@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index aa5833001fde5..b8e393bb716fb 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3631,7 +3631,7 @@ static void skl_tplg_complete(struct snd_soc_component *component) sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { - struct snd_ctl_elem_value val; + struct snd_ctl_elem_value val = {}; if (strstr(texts[i], chan_text)) { val.value.enumerated.item[0] = i; -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 12/25] ASoC: Intel: sof_sdw: set proper flags for Dell TGL-H SKU 0A5E
by Sasha Levin 02 Feb '21

02 Feb '21

From: Libin Yang <libin.yang(a)intel.com> [ Upstream commit 9ad9bc59dde106e56dd59ce2bec7c1b08e1f0eb4 ] Add flag "SOF_RT711_JD_SRC_JD2", flag "SOF_RT715_DAI_ID_FIX" and "SOF_SDW_FOUR_SPK" to the Dell TGL-H based SKU "0A5E". Signed-off-by: Libin Yang <libin.yang(a)intel.com> Co-developed-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20210125081117.814488-1-kai.vehmanen@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b29946eb43551..a8d43c87cb5a2 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -57,6 +57,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | SOF_RT715_DAI_ID_FIX), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5E") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 11/25] ASoC: ak4458: correct reset polarity
by Sasha Levin 02 Feb '21

02 Feb '21

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> [ Upstream commit e953daeb68b1abd8a7d44902786349fdeef5c297 ] Reset (aka power off) happens when the reset gpio is made active. Change function name to ak4458_reset to match devicetree property "reset-gpios" Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org> Link: https://lore.kernel.org/r/ce650f47-4ff6-e486-7846-cc3d033f3601@blennerhasse… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/ak4458.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 1010c9ee2e836..472caad17012e 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -595,18 +595,10 @@ static struct snd_soc_dai_driver ak4497_dai = { .ops = &ak4458_dai_ops, }; -static void ak4458_power_off(struct ak4458_priv *ak4458) +static void ak4458_reset(struct ak4458_priv *ak4458, bool active) { if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); - usleep_range(1000, 2000); - } -} - -static void ak4458_power_on(struct ak4458_priv *ak4458) -{ - if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 1); + gpiod_set_value_cansleep(ak4458->reset_gpiod, active); usleep_range(1000, 2000); } } @@ -620,7 +612,7 @@ static int ak4458_init(struct snd_soc_component *component) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_on(ak4458); + ak4458_reset(ak4458, false); ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); /* ACKS bit = 1; 10000000 */ @@ -650,7 +642,7 @@ static void ak4458_remove(struct snd_soc_component *component) { struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - ak4458_power_off(ak4458); + ak4458_reset(ak4458, true); } #ifdef CONFIG_PM @@ -660,7 +652,7 @@ static int __maybe_unused ak4458_runtime_suspend(struct device *dev) regcache_cache_only(ak4458->regmap, true); - ak4458_power_off(ak4458); + ak4458_reset(ak4458, true); if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 0); @@ -685,8 +677,8 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_off(ak4458); - ak4458_power_on(ak4458); + ak4458_reset(ak4458, true); + ak4458_reset(ak4458, false); regcache_cache_only(ak4458->regmap, false); regcache_mark_dirty(ak4458->regmap); -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 10/25] ALSA: hda: intel-dsp-config: add PCI id for TGL-H
by Sasha Levin 02 Feb '21

02 Feb '21

From: Bard Liao <bard.liao(a)intel.com> [ Upstream commit c5b5ff607d6fe5f4284acabd07066f96ecf96ac4 ] Adding PCI id for TGL-H. Like for other TGL platforms, SOF is used if Soundwire codecs or PCH-DMIC is detected. Signed-off-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Xiuli Pan <xiuli.pan(a)intel.com> Reviewed-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20210125083051.828205-1-kai.vehmanen@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-dsp-config.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 1c5114dedda92..fe49e9a97f0ec 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -306,6 +306,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0xa0c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x43c8, + }, #endif /* Elkhart Lake */ -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 06/25] ASoC: Intel: Skylake: Zero snd_ctl_elem_value
by Sasha Levin 02 Feb '21

02 Feb '21

From: Ricardo Ribalda <ribalda(a)chromium.org> [ Upstream commit 1d8fe0648e118fd495a2cb393a34eb8d428e7808 ] Clear struct snd_ctl_elem_value before calling ->put() to avoid any data leak. Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Link: https://lore.kernel.org/r/20210121171644.131059-2-ribalda@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 40bee10b0c65a..1bde9ce04de33 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3631,7 +3631,7 @@ static void skl_tplg_complete(struct snd_soc_component *component) sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { - struct snd_ctl_elem_value val; + struct snd_ctl_elem_value val = {}; if (strstr(texts[i], chan_text)) { val.value.enumerated.item[0] = i; -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 03/25] ASoC: wm_adsp: Fix control name parsing for multi-fw
by Sasha Levin 02 Feb '21

02 Feb '21

From: James Schulman <james.schulman(a)cirrus.com> [ Upstream commit a8939f2e138e418c2b059056ff5b501eaf2eae54 ] When switching between firmware types, the wrong control can be selected when requesting control in kernel API. Use the currently selected DSP firwmare type to select the proper mixer control. Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210115201105.14075-1-james.schulman@cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index dec8716aa8ef5..985b2dcecf138 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2031,11 +2031,14 @@ static struct wm_coeff_ctl *wm_adsp_get_ctl(struct wm_adsp *dsp, unsigned int alg) { struct wm_coeff_ctl *pos, *rslt = NULL; + const char *fw_txt = wm_adsp_fw_text[dsp->fw]; list_for_each_entry(pos, &dsp->ctl_list, list) { if (!pos->subname) continue; if (strncmp(pos->subname, name, pos->subname_len) == 0 && + strncmp(pos->fw_name, fw_txt, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0 && pos->alg_region.alg == alg && pos->alg_region.type == type) { rslt = pos; -- 2.27.0

1 0

hdspmixer: Hardware output loopback toggle buttons
by GitHub pull_request - edited 02 Feb '21

02 Feb '21

alsa-project/alsa-tools pull request #4 was edited from pfandl: Hi, as promised in the sound/pci/rme9652 patch here are the user land changes which provide the user the ability to toggle the hardware output loopback functionality per output channel. I tried to hide the buttons for devices which do not expose the snd controls yet, but due to probably unnecessary struggle i cancelled that. Please let me know though should you want them removed for not yet enabled devices. Request URL : https://github.com/alsa-project/alsa-tools/pull/4 Patch URL : https://github.com/alsa-project/alsa-tools/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Hardware output loopback toggle buttons
by GitHub pull_request - opened 02 Feb '21

02 Feb '21

alsa-project/alsa-tools pull request #4 was opened from pfandl: Hi, as promised in the sound/pci/rme9652 patch here are the user land changes which provide the user the ability to toggle the hardware output loopback functionality per output channel. I tried to hide the buttons for devices which do not expose the snd controls yet, but due to probably unnecessary struggle i cancelled that. Please let me know though should you want them removed for not yet enabled devices. Request URL : https://github.com/alsa-project/alsa-tools/pull/4 Patch URL : https://github.com/alsa-project/alsa-tools/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH 1/2] ALSA: usb-audio: Add DJM450 to Pioneer format quirk
by Olivia Mackintosh 02 Feb '21

02 Feb '21

Like the DJM-750, ensure that the format control message is passed to the device when opening a stream. It seems as though fmt->sync_ep is not always set when this function is called hence the passing of the value at the call site. If this can be fixed, fmt->sync_up should be used as the wvalue. --- sound/usb/quirks.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e196e364cef1..08300954a50e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,6 +1470,23 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, + u16 windex) +{ + unsigned int cur_rate = subs->data_endpoint->cur_rate; + u8 sr[3]; + // Convert to little endian + sr[0] = cur_rate&0xff; + sr[1] = (cur_rate>>8)&0xff; + sr[2] = (cur_rate>>16)&0xff; + usb_set_interface(subs->dev, 0, 1); + // we should derive windex from fmt-sync_ep but it's not set + snd_usb_ctl_msg(subs->stream->chip->dev, + usb_rcvctrlpipe(subs->stream->chip->dev, 0), + 0x01, 0x22, 0x0100, windex, &sr, 0x0003); + return 0; +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1483,6 +1500,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ + pioneer_djm_set_format_quirk(subs, 0x0082); + break; } } -- 2.30.0

2 2

[PATCH v9 0/1] audio jack software injection
by Hui Wang 02 Feb '21

02 Feb '21

the changes in the v9: - move the "Phantom" jack comparison into the snd_jack_debugfs_add_inject_node(), then the comparison is only executed when SND_JACK_INJECTION_DEBUG is enabled. the changes in the v8: - change "for (i = 0; i < strlen(tname); i++)" to "for (i = 0; tname[i]; i++)" - use CONFIG_SND_DEBUG to wrap some debugfs variable and struct members. the changes in the v7: - change the format of the last part in jack-injection.rst - add dependence SND_DEBUG in the Kconfig - create debugfs_mount_dir/sound and debugfs_mount_dir/sound/cardN only SND_DEBUG is enabled - change simple_write_to_buffer(buf, count, ppos, from, count) to simple_write_to_buffer(buf, sizeof(buf) - 1, ppos, from, count) - rewrite the function parse_mask_bits() according to v6's comment the changes in the v6: - use the sizeof(buf) to replace the digits in the scnprintf - squash the document patch into the 0001-xxx.patch. the changes in the v5: - add a Kconfig to enable/disable the jack injection feature - replace all kzalloc with local char[] in the debugfs fops - replace the chars of !isalnum() to '_' for the jack folder's name - squash all .c files patches into one patch - add a document to explain jack injection, for easier review, put the document in a standalone patch. If needed, will squash this patch to the previous one. the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (1): alsa: jack: implement software jack injection via debugfs Documentation/sound/designs/index.rst | 1 + .../sound/designs/jack-injection.rst | 166 ++++++++++ include/sound/core.h | 6 + include/sound/jack.h | 1 + sound/core/Kconfig | 9 + sound/core/init.c | 16 + sound/core/jack.c | 304 +++++++++++++++++- sound/core/sound.c | 13 + 8 files changed, 512 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/designs/jack-injection.rst -- 2.25.1

2 3

[PATCH] sound: pci/rme9652 - implement and expose controls for output
by fassl 02 Feb '21

02 Feb '21

>From 980aa253c17d233966e05a43f3611693e2e02040 Mon Sep 17 00:00:00 2001 From: Jasmin Fazlic <superfassl(a)gmail.com> Date: Sun, 31 Jan 2021 22:17:22 +0100 Subject: [PATCH] sound: pci/rme9652 - implement and expose controls for output loopback - so far only tested and enabled for RME HDSP9632 Signed-off-by: Jasmin Fazlic <superfassl(a)gmail.com> --- sound/pci/rme9652/hdsp.c | 85 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 84 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cea53a878c36..7e832a502cea 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -469,6 +469,7 @@ struct hdsp { unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; + u32 io_loopback; /* output loopback channel states*/ struct snd_dma_buffer capture_dma_buf; struct snd_dma_buffer playback_dma_buf; @@ -3253,6 +3254,71 @@ static const struct snd_kcontrol_new snd_hdsp_96xx_aeb = HDSP_AnalogExtensionBoard); static struct snd_kcontrol_new snd_hdsp_adat_sync_check = HDSP_ADAT_SYNC_CHECK; + +static bool hdsp_loopback_get(struct hdsp *const hdsp, const u8 channel) +{ + return hdsp->io_loopback & (1 << channel); +} + +static int hdsp_loopback_set(struct hdsp *const hdsp, const u8 channel, const bool enable) +{ + if (hdsp_loopback_get(hdsp, channel) == enable) + return 0; + + hdsp->io_loopback ^= (1 << channel); + + hdsp_write(hdsp, HDSP_inputEnable + (4 * (hdsp->max_channels + channel)), enable); + + return 1; +} + +static int snd_hdsp_loopback_info(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_info *const uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_hdsp_loopback_get(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_value *const ucontrol) +{ + struct hdsp *const hdsp = snd_kcontrol_chip(kcontrol); + const u8 channel = kcontrol->id.index; + + if (channel >= hdsp->max_channels) + return -ENOENT; + + ucontrol->value.integer.value[0] = hdsp_loopback_get(hdsp, channel); + + return 0; +} + +static int snd_hdsp_loopback_put(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_value *const ucontrol) +{ + struct hdsp *const hdsp = snd_kcontrol_chip(kcontrol); + const u8 channel = ucontrol->id.index; + const bool enable = ucontrol->value.integer.value[0] & 1; + + if (channel >= hdsp->max_channels) + return -ENOENT; + + return hdsp_loopback_set(hdsp, channel, enable); +} + +static struct snd_kcontrol_new snd_hdsp_loopback_control = { + .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, + .name = "Output Loopback", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_hdsp_loopback_info, + .get = snd_hdsp_loopback_get, + .put = snd_hdsp_loopback_put +}; + static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) { unsigned int idx; @@ -3297,6 +3363,17 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } } + /* Output loopback controls for H9632 cards */ + if (hdsp->io_type == H9632) { + for (idx = 0; idx < hdsp->max_channels; idx++) { + snd_hdsp_loopback_control.index = idx; + kctl = snd_ctl_new1(&snd_hdsp_loopback_control, hdsp); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + } + /* AEB control for H96xx card */ if (hdsp->io_type == H9632 || hdsp->io_type == H9652) { if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_96xx_aeb, hdsp))) < 0) @@ -4956,7 +5033,7 @@ static int snd_hdsp_enable_io (struct hdsp *hdsp) static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { - int status, aebi_channels, aebo_channels; + int status, aebi_channels, aebo_channels, i; switch (hdsp->io_type) { case Digiface: @@ -4983,6 +5060,12 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_out_channels = H9632_SS_CHANNELS+aebo_channels; hdsp->ds_out_channels = H9632_DS_CHANNELS+aebo_channels; hdsp->qs_out_channels = H9632_QS_CHANNELS+aebo_channels; + /* Disable loopback of output channels, as the set function + * only sets on a change we fake all bits (channels) as enabled. + */ + hdsp->io_loopback = 0xffffffff; + for (i = 0; i < hdsp->max_channels; ++i) + hdsp_loopback_set(hdsp, i, false); break; case Multiface: -- 2.27.0

2 5

add one Advantech SSID in the ALC886
by Pshou 02 Feb '21

02 Feb '21

Hi Alsa-devel ML: Can you help me add some source code to patch_realtek.c? Add one Advantech SSID in the alc886. Best regards Pshou

2 1

[PATCH] ALSA: hda: Drop power save deny list entry for Clevo W65_67SB
by Takashi Iwai 02 Feb '21

02 Feb '21

As the runtime PM issue was addressed by the recent fix 4961167bf748 ("ALSA: hda/via: Apply the workaround generically for Clevo machines") for VIA codecs, we need no longer to keep the Clevo device off from the power saving as default. Drop the deny list entry accordingly. Depends: 4961167bf748 ("ALSA: hda/via: Apply the workaround generically for Clevo machines") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5a50d3a46445..ef1f3aabc4a3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2221,8 +2221,6 @@ static const struct snd_pci_quirk power_save_denylist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ - SND_PCI_QUIRK(0x1558, 0x6504, "Clevo W65_67SB", 0), - /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ -- 2.26.2

1 0

[bug report] potential deadlock in snd_emu10k1_voice_alloc()
by Dan Carpenter 02 Feb '21

02 Feb '21

Hello ALSA devs, The patch 1da177e4c3f4: "Linux-2.6.12-rc2" from Apr 16, 2005, leads to the following static checker warning: sound/pci/emu10k1/voice.c:112 snd_emu10k1_voice_alloc() warn: called with lock held. '&emu->voice_lock' sound/pci/emu10k1/voice.c 101 int snd_emu10k1_voice_alloc(struct snd_emu10k1 *emu, int type, int number, 102 struct snd_emu10k1_voice **rvoice) 103 { 104 unsigned long flags; 105 int result; 106 107 if (snd_BUG_ON(!rvoice)) 108 return -EINVAL; 109 if (snd_BUG_ON(!number)) 110 return -EINVAL; 111 112 spin_lock_irqsave(&emu->voice_lock, flags); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ This lock is already held in the caller. 113 for (;;) { 114 result = voice_alloc(emu, type, number, rvoice); 115 if (result == 0 || type == EMU10K1_SYNTH || type == EMU10K1_MIDI) 116 break; 117 118 /* free a voice from synth */ 119 if (emu->get_synth_voice) { The call tree is: snd_emux_note_on() takes the lock: spin_lock_irqsave(&emu->voice_lock, flags); calls vp = emu->ops.get_voice(emu, port); --> get_voice() --> snd_emu10k1_voice_alloc() regards, dan carpenter

2 1

[PATCH] Add support to expose controls of ladspa plugin
by Camel Guo 02 Feb '21

02 Feb '21

From: Camel Guo <camelg(a)axis.com> In order for external software components to adjust ladspa plugin dynamically, this commit adds an option to exposes the control array of input control ports of a ladspa plugin to a file, through which any applications with proper permission can control this plugin. Signed-off-by: Camel Guo <camelg(a)axis.com> --- src/pcm/pcm_ladspa.c | 157 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 139 insertions(+), 18 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index ad73347d..40b5d38f 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -32,6 +32,9 @@ * http://www.medianet.ag */ +#include <stdbool.h> +#include <sys/mman.h> +#include <sys/stat.h> #include <dirent.h> #include <locale.h> #include <math.h> @@ -93,7 +96,8 @@ typedef struct { unsigned int port_bindings_size; /* size of array */ unsigned int *port_bindings; /* index = channel number, value = LADSPA port */ unsigned int controls_size; /* size of array */ - unsigned char *controls_initialized; /* initialized by ALSA user */ + bool controls_new; /* if controls is new, it can be overrided by alsa config */ + bool controls_shared; /* if controls is shared memory map */ LADSPA_Data *controls; /* index = LADSPA control port index */ } snd_pcm_ladspa_plugin_io_t; @@ -101,6 +105,7 @@ typedef struct { struct list_head list; snd_pcm_ladspa_policy_t policy; char *filename; + char *controls_path; void *dl_handle; const LADSPA_Descriptor *desc; snd_pcm_ladspa_plugin_io_t input; @@ -110,6 +115,73 @@ typedef struct { #endif /* DOC_HIDDEN */ +static LADSPA_Data* mmap_ladspa_controls(const char* filename, unsigned long length, bool *is_new) +{ + LADSPA_Data *ptr = NULL; + int fd = -1; + int ret = 0; + struct stat statbuf = { 0 }; + int prot = PROT_READ; + + if (filename == NULL || is_new == NULL) return NULL; + + *is_new = false; + + fd = open(filename, O_RDONLY); + if(fd < 0) { + if (errno == ENOENT) { + fd = open(filename, O_RDWR | O_CREAT, 0666); + if(fd < 0) { + SNDERR("Failed to open controls file: %s", filename); + return NULL; + } + + prot |= PROT_WRITE; + *is_new = true; + } else { + SNDERR("Failed to open %s due to '%s'", filename, strerror(errno)); + return NULL; + } + } + + ret = fstat(fd, &statbuf); + if (ret == -1) { + SNDERR("Failed to get status of '%s' due to '%s'", filename, strerror(errno)); + goto out; + } + + if (statbuf.st_size < (off_t) length) { + /* this file is invalid, must be truncated and re-initialized later on */ + close(fd); + + fd = open(filename, O_RDWR | O_CREAT, 0666); + if(fd < 0) { + SNDERR("Failed to open controls file: %s", filename); + /* no need to close(fd), so just return */ + return NULL; + } + + ret = ftruncate(fd, length); + if (ret == -1) { + SNDERR("Failed to increase the size of '%s' due to '%s'", filename, strerror(errno)); + goto out; + } + + prot |= PROT_WRITE; + *is_new = true; + } + + ptr = (LADSPA_Data*) mmap(NULL, length, prot, MAP_SHARED, fd, 0); + if(ptr == MAP_FAILED) { + SNDERR("Failed to mmap %s due to %s", filename, strerror(errno)); + ptr = NULL; + } + +out: + close (fd); + return ptr; +} + static unsigned int snd_pcm_ladspa_count_ports(snd_pcm_ladspa_plugin_t *lplug, LADSPA_PortDescriptor pdesc) { @@ -174,8 +246,11 @@ static int snd_pcm_ladspa_find_port_idx(unsigned int *res, static void snd_pcm_ladspa_free_io(snd_pcm_ladspa_plugin_io_t *io) { - free(io->controls); - free(io->controls_initialized); + if (io->controls_shared) { + munmap(io->controls, io->controls_size * sizeof(LADSPA_Data)); + } else { + free(io->controls); + } } static void snd_pcm_ladspa_free_plugins(struct list_head *plugins) @@ -186,6 +261,8 @@ static void snd_pcm_ladspa_free_plugins(struct list_head *plugins) snd_pcm_ladspa_free_io(&plugin->output); if (plugin->dl_handle) dlclose(plugin->dl_handle); + if (plugin->controls_path) + free(plugin->controls_path); free(plugin->filename); list_del(&plugin->list); free(plugin); @@ -574,8 +651,6 @@ static int snd_pcm_ladspa_connect_controls(snd_pcm_ladspa_plugin_t *plugin, for (idx = midx = 0; idx < plugin->desc->PortCount; idx++) if ((plugin->desc->PortDescriptors[idx] & (io->pdesc | LADSPA_PORT_CONTROL)) == (io->pdesc | LADSPA_PORT_CONTROL)) { if (io->controls_size > midx) { - if (!io->controls_initialized[midx]) - snd_pcm_ladspa_get_default_cvalue(plugin->desc, idx, &io->controls[midx]); plugin->desc->connect_port(instance->handle, idx, &io->controls[midx]); } else { return -EINVAL; @@ -878,6 +953,10 @@ snd_pcm_ladspa_write_areas(snd_pcm_t *pcm, snd_pcm_ladspa_plugin_t *plugin = list_entry(pos, snd_pcm_ladspa_plugin_t, list); list_for_each(pos1, &plugin->instances) { instance = list_entry(pos1, snd_pcm_ladspa_instance_t, list); + if (plugin->input.controls_shared) { + (void) snd_pcm_ladspa_connect_controls(plugin, &plugin->input, instance); + } + /* Skip output controls since they can not be changed dynamically */ for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->input.channels.array[idx]; data = instance->input.data[idx]; @@ -939,6 +1018,10 @@ snd_pcm_ladspa_read_areas(snd_pcm_t *pcm, snd_pcm_ladspa_plugin_t *plugin = list_entry(pos, snd_pcm_ladspa_plugin_t, list); list_for_each(pos1, &plugin->instances) { instance = list_entry(pos1, snd_pcm_ladspa_instance_t, list); + if (plugin->input.controls_shared) { + (void) snd_pcm_ladspa_connect_controls(plugin, &plugin->input, instance); + } + /* Skip output controls since they can not be changed dynamically */ for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->input.channels.array[idx]; data = instance->input.data[idx]; @@ -1227,26 +1310,50 @@ static int snd_pcm_ladspa_add_default_controls(snd_pcm_ladspa_plugin_t *lplug, { unsigned int count = 0; LADSPA_Data *array; - unsigned char *initialized; unsigned long idx; + unsigned long midx; + bool controls_shared = false; + bool is_new = true; for (idx = 0; idx < lplug->desc->PortCount; idx++) if ((lplug->desc->PortDescriptors[idx] & (io->pdesc | LADSPA_PORT_CONTROL)) == (io->pdesc | LADSPA_PORT_CONTROL)) count++; - array = (LADSPA_Data *)calloc(count, sizeof(LADSPA_Data)); - if (!array) - return -ENOMEM; - initialized = (unsigned char *)calloc(count, sizeof(unsigned char)); - if (!initialized) { - free(array); - return -ENOMEM; + + /* + * Only support to expose ladspa control array for input control ports + */ + if ((io->pdesc == LADSPA_PORT_INPUT) && lplug->controls_path && count != 0) { + array = mmap_ladspa_controls(lplug->controls_path, count * sizeof(LADSPA_Data), &is_new); + if (!array) + return -ENOMEM; + controls_shared = true; + } else { + array = (LADSPA_Data *)calloc(count, sizeof(LADSPA_Data)); + if (!array) + return -ENOMEM; + } + + if (is_new) { + /* + * Initialize array of controls of this ladspa plugin with its + * default values defined in this plugin implementation. This + * array can be overrided by its alsa configuration files. + */ + for (idx = midx = 0; idx < lplug->desc->PortCount; idx++) { + if ((lplug->desc->PortDescriptors[idx] & (io->pdesc | LADSPA_PORT_CONTROL)) == (io->pdesc | LADSPA_PORT_CONTROL)) { + snd_pcm_ladspa_get_default_cvalue(lplug->desc, idx, &array[midx]); + midx++; + } + } } + io->controls_size = count; - io->controls_initialized = initialized; + io->controls_new = is_new; + io->controls_shared = controls_shared; io->controls = array; return 0; -} +} static int snd_pcm_ladspa_parse_controls(snd_pcm_ladspa_plugin_t *lplug, snd_pcm_ladspa_plugin_io_t *io, @@ -1287,8 +1394,10 @@ static int snd_pcm_ladspa_parse_controls(snd_pcm_ladspa_plugin_t *lplug, SNDERR("internal error"); return err; } - io->controls_initialized[uval] = 1; - io->controls[uval] = (LADSPA_Data)dval; + + if (io->controls_new) { + io->controls[uval] = (LADSPA_Data)dval; + } } return 0; @@ -1429,7 +1538,7 @@ static int snd_pcm_ladspa_add_plugin(struct list_head *list, int reverse) { snd_config_iterator_t i, next; - const char *label = NULL, *filename = NULL; + const char *label = NULL, *filename = NULL, *controls_path = NULL; long ladspa_id = 0; int err; snd_pcm_ladspa_plugin_t *lplug; @@ -1467,6 +1576,12 @@ static int snd_pcm_ladspa_add_plugin(struct list_head *list, output = n; continue; } + if (strcmp(id, "controls") == 0) { + err = snd_config_get_string(n, &controls_path); + if (err < 0) + return err; + continue; + } if (strcmp(id, "policy") == 0) { const char *str; err = snd_config_get_string(n, &str); @@ -1496,6 +1611,11 @@ static int snd_pcm_ladspa_add_plugin(struct list_head *list, lplug->input.pdesc = LADSPA_PORT_INPUT; lplug->output.pdesc = LADSPA_PORT_OUTPUT; INIT_LIST_HEAD(&lplug->instances); + if (controls_path) { + lplug->controls_path = strdup(controls_path); + } else { + lplug->controls_path = NULL; + } if (filename) { err = snd_pcm_ladspa_check_file(lplug, filename, label, ladspa_id); if (err < 0) { @@ -1692,6 +1812,7 @@ pcm.name { [label STR] # LADSPA plugin label (for example 'delay_5s') [filename STR] # Full filename of .so library with LADSPA plugin code [policy STR] # Policy can be 'none' or 'duplicate' + [controls STR] # Path (directory) with controls for this plugin input | output { bindings { C INT or STR # C - channel, INT - audio port index, STR - audio port name -- 2.20.1

3 2

[bug report] ALSA: rawmidi: Avoid racy info ioctl via ctl device
by Dan Carpenter 02 Feb '21

02 Feb '21

Hello Takashi Iwai, The patch c1cfd9025cc3: "ALSA: rawmidi: Avoid racy info ioctl via ctl device" from Dec 14, 2017, leads to the following static checker warning: sound/core/rawmidi.c:651 snd_rawmidi_info_select() warn: called with lock held. '&register_mutex' sound/core/seq/seq_midi.c 297 mutex_lock(&register_mutex); ^^^^^^^^^^^^^^^ Holding this lock. 298 client = synths[card->number]; 299 if (client == NULL) { 300 newclient = 1; 301 client = kzalloc(sizeof(*client), GFP_KERNEL); 302 if (client == NULL) { 303 mutex_unlock(&register_mutex); 304 kfree(info); 305 return -ENOMEM; 306 } 307 client->seq_client = 308 snd_seq_create_kernel_client( 309 card, 0, "%s", card->shortname[0] ? 310 (const char *)card->shortname : "External MIDI"); 311 if (client->seq_client < 0) { 312 kfree(client); 313 mutex_unlock(&register_mutex); 314 kfree(info); 315 return -ENOMEM; 316 } 317 } 318 319 msynth = kcalloc(ports, sizeof(struct seq_midisynth), GFP_KERNEL); 320 port = kmalloc(sizeof(*port), GFP_KERNEL); 321 if (msynth == NULL || port == NULL) 322 goto __nomem; 323 324 for (p = 0; p < ports; p++) { 325 ms = &msynth[p]; 326 327 if (snd_seq_midisynth_new(ms, card, device, p) < 0) 328 goto __nomem; 329 330 /* declare port */ 331 memset(port, 0, sizeof(*port)); 332 port->addr.client = client->seq_client; 333 port->addr.port = device * (256 / SNDRV_RAWMIDI_DEVICES) + p; 334 port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; 335 memset(info, 0, sizeof(*info)); 336 info->device = device; 337 if (p < output_count) 338 info->stream = SNDRV_RAWMIDI_STREAM_OUTPUT; 339 else 340 info->stream = SNDRV_RAWMIDI_STREAM_INPUT; 341 info->subdevice = p; 342 if (snd_rawmidi_info_select(card, info) >= 0) ^^^^^^^^^^^^^^^^^^^^^^^ We can't call this function when we're holding the lock or it leads to a deadlock. Before the patch, we used to rely on the callers to take the lock before calling snd_rawmidi_info_select() but the patch moved the lock inside the function. 343 strcpy(port->name, info->subname); 344 if (! port->name[0]) { 345 if (info->name[0]) { 346 if (ports > 1) 347 snprintf(port->name, sizeof(port->name), "%s-%u", info->name, p); 348 else 349 snprintf(port->name, sizeof(port->name), "%s", info->name); regards, dan carpenter

2 2

[PATCH] ASoC: fsl_xcvr: remove unneeded semicolon
by Yang Li 02 Feb '21

02 Feb '21

Eliminate the following coccicheck warning: ./sound/soc/fsl/fsl_xcvr.c:739:2-3: Unneeded semicolon Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/fsl/fsl_xcvr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 3d58c88..65b388a 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -736,7 +736,7 @@ static int fsl_xcvr_load_firmware(struct fsl_xcvr *xcvr) /* clean current page, including data memory */ memset_io(xcvr->ram_addr, 0, size); } - }; + } err_firmware: release_firmware(fw); -- 1.8.3.1

2 1

conf: HDA-Intel: increase the max number of HDMI/DP output to 11
by GitHub pull_request - opened 02 Feb '21

02 Feb '21

alsa-project/alsa-lib pull request #118 was opened from jason77-wang: Intel TGL HDMI/DP codec provides 9 pins (Linux kernel, 9a11ba7388f16: ALSA: hda: hdmi - add Tigerlake support), and with the DP MST enabled, the linux kernel will build 11 output devices (3, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16), and current alsa-lib only maps the first 8 output devices, if users plug the HDMI/DP monitor to the last 4 ports, the users will not see the output device from pulseaudio or gnome. We have experienced this issue on a dell TGL machine with a dock, we plugged 2 HDMP/DP monitors on the dock, but we could only see 1 HDMI/DP output device from pulseaudio or gnome, through investigation, we found one monitor is plugged in the 2nd port from last. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/118 Patch URL : https://github.com/alsa-project/alsa-lib/pull/118.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 2/2] ALSA: usb-audio: Add DJM-450 to the quirks table
by Olivia Mackintosh 02 Feb '21

02 Feb '21

As with most Pioneer devices, the device descriptor is vendor specific and as such, the number of channels, the PCM format, endpoints and sample rate need to be specified. This device has 8 inputs and 8 outputs and a sample rate of 48000 only. The PCM format is S24_3LE like other devices. There seems to be an appetite for reducing duplication amongs these Pioneer patches but again, I feel this is a step to be taken after support has been added as it's not completely clear where the commonalities are. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 93d55cd1a5a4..1165a5ac60f2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3817,6 +3817,63 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-450 + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0013), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.30.0

1 0

[PATCH 0/2] ALSA: usb-audio: Add Pioneer DJM-450 support
by Olivia Mackintosh 02 Feb '21

02 Feb '21

Here are a couple of patches to support the DJM-450. Un-suprisingly, a device quirk is required just like the other DJM models. Testing was done by "xfz" from #xwax on Freenode. Perhaps this is useful for reference or if anyone requires information about this device. They were hesitant to have their full details in the commit log. Olivia Mackintosh (2): ALSA: usb-audio: Add DJM450 to Pioneer format quirk ALSA: usb-audio: Add DJM-450 to the quirks table sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 20 ++++++++++++++ 2 files changed, 77 insertions(+) -- 2.30.0

1 0

[bug report] ASoC: cpcap: new codec
by Dan Carpenter 01 Feb '21

01 Feb '21

Hello Sebastian Reichel, The patch f6cdf2d3445d: "ASoC: cpcap: new codec" from Feb 23, 2018, leads to the following static checker warning: sound/soc/codecs/cpcap.c:1276 cpcap_voice_hw_params() warn: 'CPCAP_BIT_MIC1_RX_TIMESLOT0' is a shifter (not for '|='). sound/soc/codecs/cpcap.c:1279 cpcap_voice_hw_params() warn: 'CPCAP_BIT_MIC2_TIMESLOT0' is a shifter (not for '|='). sound/soc/codecs/cpcap.c 1253 static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, 1254 struct snd_pcm_hw_params *params, 1255 struct snd_soc_dai *dai) 1256 { 1257 struct snd_soc_component *component = dai->component; 1258 struct device *dev = component->dev; 1259 struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); 1260 static const u16 reg_cdi = CPCAP_REG_CDI; 1261 int rate = params_rate(params); 1262 int channels = params_channels(params); 1263 int direction = substream->stream; 1264 u16 val, mask; 1265 int err; 1266 1267 dev_dbg(dev, "Voice setup HW params: rate=%d, direction=%d, chan=%d", 1268 rate, direction, channels); 1269 1270 err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, rate); 1271 if (err) 1272 return err; 1273 1274 if (direction == SNDRV_PCM_STREAM_CAPTURE) { 1275 mask = 0x0000; 1276 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0; ^^^^^^^^^^^^^^^^^^^^^^^^^^^ This should probably be BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0). 1277 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1; 1278 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2; 1279 mask |= CPCAP_BIT_MIC2_TIMESLOT0; ^^^^^^^^^^^^^^^^^^^^^^^^ Same for all the others as well I guess. 1280 mask |= CPCAP_BIT_MIC2_TIMESLOT1; 1281 mask |= CPCAP_BIT_MIC2_TIMESLOT2; 1282 val = 0x0000; 1283 if (channels >= 2) 1284 val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Here it's BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0). 1285 err = regmap_update_bits(cpcap->regmap, reg_cdi, mask, val); 1286 if (err) 1287 return err; 1288 } 1289 1290 return 0; 1291 } regards, dan carpenter

4 3

[PATCH] ASoC: tegra: SND_SOC_TEGRA_AUDIO_GRAPH_CARD should depend on SND_SOC_TEGRA
by Geert Uytterhoeven 01 Feb '21

01 Feb '21

Audio Graph Card based Tegra driver is only useful on NVIDIA Tegra SoCs. Hence add a dependency on SND_SOC_TEGRA, to prevent asking the user about this driver when configuring a kernel without Tegra sound support. Wrap all Tegra sound config options inside a big if/endif block, instead of just adding the dependency to the single config option that does not have it yet, to preventing similar future mistakes. Fixes: 202e2f7745437aa5 ("ASoC: tegra: Add audio graph based card driver") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/tegra/Kconfig | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 6dc83ad8607a1de2..a4e6760944d028cf 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -9,9 +9,10 @@ config SND_SOC_TEGRA help Say Y or M here if you want support for SoC audio on Tegra. +if SND_SOC_TEGRA + config SND_SOC_TEGRA20_AC97 tristate "Tegra20 AC97 interface" - depends on SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -21,7 +22,6 @@ config SND_SOC_TEGRA20_AC97 config SND_SOC_TEGRA20_DAS tristate "Tegra20 DAS module" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to @@ -29,7 +29,6 @@ config SND_SOC_TEGRA20_DAS config SND_SOC_TEGRA20_I2S tristate "Tegra20 I2S interface" - depends on SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -38,7 +37,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support @@ -46,7 +44,6 @@ config SND_SOC_TEGRA20_SPDIF config SND_SOC_TEGRA30_AHUB tristate "Tegra30 AHUB module" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra30 AHUB module. You will also need to select the individual machine drivers to @@ -54,7 +51,6 @@ config SND_SOC_TEGRA30_AHUB config SND_SOC_TEGRA30_I2S tristate "Tegra30 I2S interface" - depends on SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -63,7 +59,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA210_AHUB tristate "Tegra210 AHUB module" - depends on SND_SOC_TEGRA help Config to enable Audio Hub (AHUB) module, which comprises of a switch called Audio Crossbar (AXBAR) used to configure or modify @@ -73,7 +68,6 @@ config SND_SOC_TEGRA210_AHUB config SND_SOC_TEGRA210_DMIC tristate "Tegra210 DMIC module" - depends on SND_SOC_TEGRA help Config to enable the Digital MIC (DMIC) controller which is used to interface with Pulse Density Modulation (PDM) input devices. @@ -84,7 +78,6 @@ config SND_SOC_TEGRA210_DMIC config SND_SOC_TEGRA210_I2S tristate "Tegra210 I2S module" - depends on SND_SOC_TEGRA help Config to enable the Inter-IC Sound (I2S) Controller which implements full-duplex and bidirectional and single direction @@ -94,7 +87,6 @@ config SND_SOC_TEGRA210_I2S config SND_SOC_TEGRA186_DSPK tristate "Tegra186 DSPK module" - depends on SND_SOC_TEGRA help Config to enable the Digital Speaker Controller (DSPK) which converts the multi-bit Pulse Code Modulation (PCM) audio input to @@ -107,7 +99,6 @@ config SND_SOC_TEGRA186_DSPK config SND_SOC_TEGRA210_ADMAIF tristate "Tegra210 ADMAIF module" - depends on SND_SOC_TEGRA help Config to enable ADMAIF which is the interface between ADMA and Audio Hub (AHUB). Each ADMA channel that sends/receives data to/ @@ -128,7 +119,7 @@ config SND_SOC_TEGRA_AUDIO_GRAPH_CARD config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -136,7 +127,7 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -144,7 +135,7 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -153,7 +144,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && GPIOLIB + depends on GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -162,7 +153,7 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" - depends on SND_SOC_TEGRA && I2C + depends on I2C select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for SoC audio on the @@ -170,7 +161,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the @@ -178,7 +169,7 @@ config SND_SOC_TEGRA_ALC5632 config SND_SOC_TEGRA_MAX98090 tristate "SoC Audio support for Tegra boards using a MAX98090 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -186,7 +177,7 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -194,9 +185,11 @@ config SND_SOC_TEGRA_RT5677 config SND_SOC_TEGRA_SGTL5000 tristate "SoC Audio support for Tegra boards using a SGTL5000 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_SGTL5000 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the SGTL5000 codec, such as Apalis T30, Apalis TK1 or Colibri T30. + +endif -- 2.25.1

3 2

[PATCH] ASoC: SOF: topology: Prevent NULL pointer dereference with no TLV
by Kai Vehmanen 01 Feb '21

01 Feb '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> When there is no TLV data in topology, extracting the TLV data could result in a NULL pointer exception. Prevent this by making sure that the TLV data exists before extracting it. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index b69f493b5faa..46089000524d 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1073,7 +1073,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, scontrol->cmd = SOF_CTRL_CMD_VOLUME; /* extract tlv data */ - if (get_tlv_data(kc->tlv.p, tlv) < 0) { + if (!kc->tlv.p || get_tlv_data(kc->tlv.p, tlv) < 0) { dev_err(scomp->dev, "error: invalid TLV data\n"); ret = -EINVAL; goto out_free; base-commit: 8ac9e476b86851c94e0f33bea758e0a00e3f875e -- 2.29.2

2 1

[PATCH] ASoC: codecs: add missing max_register in regmap config
by Srinivas Kandagatla 01 Feb '21

01 Feb '21

For some reason setting max_register was missed from regmap_config. Without this cat /sys/kernel/debug/regmap/sdw:0:217:2010:0:1/range actually throws below Warning. WARNING: CPU: 7 PID: 540 at drivers/base/regmap/regmap-debugfs.c:160 regmap_debugfs_get_dump_start.part.10+0x1e0/0x220 ... Call trace: regmap_debugfs_get_dump_start.part.10+0x1e0/0x220 regmap_reg_ranges_read_file+0xc0/0x2e0 full_proxy_read+0x64/0x98 vfs_read+0xa8/0x1e0 ksys_read+0x6c/0x100 __arm64_sys_read+0x1c/0x28 el0_svc_common.constprop.3+0x6c/0x190 do_el0_svc+0x24/0x90 el0_svc+0x14/0x20 el0_sync_handler+0x90/0xb8 el0_sync+0x158/0x180 ... Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wsa881x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 4530b74f5921..db87e07b11c9 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -640,6 +640,7 @@ static struct regmap_config wsa881x_regmap_config = { .val_bits = 8, .cache_type = REGCACHE_RBTREE, .reg_defaults = wsa881x_defaults, + .max_register = WSA881X_SPKR_STATUS3, .num_reg_defaults = ARRAY_SIZE(wsa881x_defaults), .volatile_reg = wsa881x_volatile_register, .readable_reg = wsa881x_readable_register, -- 2.21.0

2 1

[bug report] ALSA: seq: Use no intrruptible mutex_lock
by Dan Carpenter 01 Feb '21

01 Feb '21

Hello Takashi Iwai, The patch 04702e8d0092: "ALSA: seq: Use no intrruptible mutex_lock" from Aug 1, 2018, leads to the following static checker warning: sound/core/seq/seq_clientmgr.c:2194 snd_seq_create_kernel_client() warn: called with lock held. '&register_mutex' sound/core/seq/seq_midi.c 294 if (ports > (256 / SNDRV_RAWMIDI_DEVICES)) 295 ports = 256 / SNDRV_RAWMIDI_DEVICES; 296 297 mutex_lock(&register_mutex); ^^^^^^^^^^^^^^^ Holding lock 298 client = synths[card->number]; 299 if (client == NULL) { 300 newclient = 1; 301 client = kzalloc(sizeof(*client), GFP_KERNEL); 302 if (client == NULL) { 303 mutex_unlock(&register_mutex); 304 kfree(info); 305 return -ENOMEM; 306 } 307 client->seq_client = 308 snd_seq_create_kernel_client( ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Deadlock 309 card, 0, "%s", card->shortname[0] ? 310 (const char *)card->shortname : "External MIDI"); 311 if (client->seq_client < 0) { 312 kfree(client); 313 mutex_unlock(&register_mutex); 314 kfree(info); 315 return -ENOMEM; 316 } 317 } regards, dan carpenter

2 1

[PATCH v2] ASoC: qcom: lpass: Fix i2s ctl register bit map
by Jun Nie 01 Feb '21

01 Feb '21

Fix bitwidth mapping in i2s ctl register per APQ8016 document. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Jun Nie <jun.nie(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 8507ef8f6679..3efa133d1c64 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -250,7 +250,7 @@ static struct lpass_variant apq8016_data = { .micmode = REG_FIELD_ID(0x1000, 4, 7, 4, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 4, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 4, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 4, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 4, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0x8400, 12, 12, 2, 0x1000), .rdma_bursten = REG_FIELD_ID(0x8400, 11, 11, 2, 0x1000), -- 2.25.1

3 3

[PATCH] soundwire: return earlier if no slave is attached
by Bard Liao 01 Feb '21

01 Feb '21

From: Chao Song <chao.song(a)linux.intel.com> If there is no slave attached to soundwire bus, we can return earlier from sdw_bus_prep_clk_stop() and sdw_bus_exit_clk_stop(), this saves a redundant value check. Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 82df088c9333..d9deafdcf495 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -951,17 +951,17 @@ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) simple_clk_stop = false; } - if (is_slave && !simple_clk_stop) { + /* Skip remaining clock stop preparation if no Slave is attached */ + if (!is_slave) + return ret; + + if (!simple_clk_stop) { ret = sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); if (ret < 0) return ret; } - /* Don't need to inform slaves if there is no slave attached */ - if (!is_slave) - return ret; - /* Inform slaves that prep is done */ list_for_each_entry(slave, &bus->slaves, node) { if (!slave->dev_num) @@ -1075,16 +1075,13 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) "clk stop deprep failed:%d", ret); } - if (is_slave && !simple_clk_stop) + /* Skip remaining clock stop de-preparation if no Slave is attached */ + if (!is_slave) + return 0; + + if (!simple_clk_stop) sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); - /* - * Don't need to call slave callback function if there is no slave - * attached - */ - if (!is_slave) - return 0; - list_for_each_entry(slave, &bus->slaves, node) { if (!slave->dev_num) continue; -- 2.17.1

2 1

[PATCH] soundwire: bus: add better dev_dbg to track complete() calls
by Bard Liao 01 Feb '21

01 Feb '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add a dev_dbg() log for both enumeration and initialization completion to better track suspend-resume issues. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Chao Song <chao.song(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 6e1c988f3845..82df088c9333 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -784,7 +784,7 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, if (status == SDW_SLAVE_UNATTACHED) { dev_dbg(&slave->dev, - "%s: initializing completion for Slave %d\n", + "%s: initializing enumeration and init completion for Slave %d\n", __func__, slave->dev_num); init_completion(&slave->enumeration_complete); @@ -793,7 +793,7 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, } else if ((status == SDW_SLAVE_ATTACHED) && (slave->status == SDW_SLAVE_UNATTACHED)) { dev_dbg(&slave->dev, - "%s: signaling completion for Slave %d\n", + "%s: signaling enumeration completion for Slave %d\n", __func__, slave->dev_num); complete(&slave->enumeration_complete); @@ -1758,8 +1758,13 @@ int sdw_handle_slave_status(struct sdw_bus *bus, if (ret) dev_err(slave->bus->dev, "Update Slave status failed:%d\n", ret); - if (attached_initializing) + if (attached_initializing) { + dev_dbg(&slave->dev, + "%s: signaling initialization completion for Slave %d\n", + __func__, slave->dev_num); + complete(&slave->initialization_complete); + } } return ret; -- 2.17.1

2 1

jack doesn't detected plugged after suspend
by Mert Can Demir 01 Feb '21

01 Feb '21

Hi, I opened an issue on pulseaudio repository at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1125 about the problem about the detection of jack output, and I have been forwarded there, that mailing list, at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1125#note_784…. The related outputs and logs are baked into the issue, and seems that the outputs tells that alsa driver did not reported jack as plugged after suspend. Thanks for understanding, Sincerely, validatedev

2 1

[PATCH] ALSA: intel8x0: Fix missing check in snd_intel8x0m_create
by Dinghao Liu 01 Feb '21

01 Feb '21

When device_type == DEVICE_ALI, we should also check the return value of pci_iomap() to avoid potential null pointer dereference. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/pci/intel8x0m.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 1b7df0c4e57c..19872cecc9d2 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1129,13 +1129,14 @@ static int snd_intel8x0m_create(struct snd_card *card, chip->bmaddr = pci_iomap(pci, 3, 0); else chip->bmaddr = pci_iomap(pci, 1, 0); + +port_inited: if (!chip->bmaddr) { dev_err(card->dev, "Controller space ioremap problem\n"); snd_intel8x0m_free(chip); return -EIO; } - port_inited: /* initialize offsets */ chip->bdbars_count = 2; tbl = intel_regs; -- 2.17.1

2 1

[PATCH 0/2] ALSA: Constify static attribute_group structs
by Rikard Falkeborn 31 Jan '21

31 Jan '21

Constify two static attribute_group structs that are never modified. These two were the only static non-const ones in sound/. Rikard Falkeborn (2): ALSA: ac97: Constify static struct attribute_group ALSA: hda: Constify static attribute_group sound/ac97/bus.c | 2 +- sound/hda/hdac_sysfs.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.30.0

2 4

Extend card search paths
by GitHub pull_request - opened 30 Jan '21

30 Jan '21

alsa-project/alsa-ucm-conf pull request #76 was opened from lumag: On Qualcomm platforms is is typical for a single kernel module to support several sound cards. Add support for using `ucm2/module/${KernelModule}/${CardLongName}.conf` as a use case config file. See https://github.com/alsa-project/alsa-ucm-conf/pull/70 for the reference. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/76 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/76.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/8] Tegra186 and Tegra194 audio graph card
by Sameer Pujar 29 Jan '21

29 Jan '21

This series adds support for audio graph based solution on Tegra186 and Tegra194. This enables audio paths for I2S, DMIC and DSPK modules. Depending on the platform Jetson TX2 or Jetson AGX Xavier, required I/O module instances are enabled. Since the latter board has on board audio codec, DT support for the same is enabled and external audio playback and capture can be used. Sameer Pujar (8): ASoC: dt-bindings: rt5659: Update binding doc ASoC: dt-bindings: tegra: Add iommus property to Tegra graph card ASoC: audio-graph-card: Add clocks property to endpoint node ASoC: tegra: Select SND_SOC_RT5659 arm64: tegra: Add RT5658 device entry Revert "arm64: tegra: Disable the ACONNECT for Jetson TX2" arm64: tegra: Audio graph sound card for Jetson TX2 arm64: tegra: Audio graph sound card for Jetson AGX Xavier .../bindings/sound/audio-graph-port.yaml | 3 + .../sound/nvidia,tegra-audio-graph-card.yaml | 3 + Documentation/devicetree/bindings/sound/rt5659.txt | 11 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 609 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 22 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 468 ++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra194.dtsi | 20 + sound/soc/tegra/Kconfig | 1 + 8 files changed, 1137 insertions(+) -- 2.7.4

3 14

[PATCH] dt-bindings: Cleanup standard unit properties
by Rob Herring 29 Jan '21

29 Jan '21

Properties with standard unit suffixes already have a type and don't need type definitions. They also default to a single entry, so 'maxItems: 1' can be dropped. adi,ad5758 is an oddball which defined an enum of arrays. While a valid schema, it is simpler as a whole to only define scalar constraints. Cc: Jean Delvare <jdelvare(a)suse.com> Cc: Guenter Roeck <linux(a)roeck-us.net> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Ulf Hansson <ulf.hansson(a)linaro.org> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Sebastian Reichel <sre(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Serge Semin <fancer.lancer(a)gmail.com> Cc: Wolfram Sang <wolfram(a)the-dreams.de> Cc: linux-hwmon(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-input(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-serial(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-watchdog(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/arm/cpus.yaml | 1 - .../bindings/extcon/wlf,arizona.yaml | 1 - .../bindings/hwmon/adi,ltc2947.yaml | 1 - .../bindings/hwmon/baikal,bt1-pvt.yaml | 8 ++-- .../devicetree/bindings/hwmon/ti,tmp513.yaml | 1 - .../devicetree/bindings/i2c/i2c-gpio.yaml | 2 - .../bindings/i2c/snps,designware-i2c.yaml | 3 -- .../bindings/iio/adc/maxim,max9611.yaml | 1 - .../bindings/iio/adc/st,stm32-adc.yaml | 1 - .../bindings/iio/adc/ti,palmas-gpadc.yaml | 2 - .../bindings/iio/dac/adi,ad5758.yaml | 41 ++++++++++++------- .../bindings/iio/health/maxim,max30100.yaml | 1 - .../input/touchscreen/touchscreen.yaml | 2 - .../bindings/mmc/mmc-controller.yaml | 1 - .../bindings/mmc/mmc-pwrseq-simple.yaml | 2 - .../bindings/net/ethernet-controller.yaml | 2 - .../devicetree/bindings/net/snps,dwmac.yaml | 1 - .../bindings/power/supply/battery.yaml | 3 -- .../bindings/power/supply/bq2515x.yaml | 1 - .../bindings/regulator/dlg,da9121.yaml | 1 - .../bindings/regulator/fixed-regulator.yaml | 2 - .../devicetree/bindings/rtc/rtc.yaml | 2 - .../devicetree/bindings/serial/pl011.yaml | 2 - .../devicetree/bindings/sound/sgtl5000.yaml | 2 - .../bindings/watchdog/watchdog.yaml | 1 - 25 files changed, 29 insertions(+), 56 deletions(-) diff --git a/Documentation/devicetree/bindings/arm/cpus.yaml b/Documentation/devicetree/bindings/arm/cpus.yaml index 14cd727d3c4b..f02fd10de604 100644 --- a/Documentation/devicetree/bindings/arm/cpus.yaml +++ b/Documentation/devicetree/bindings/arm/cpus.yaml @@ -232,7 +232,6 @@ properties: by this cpu (see ./idle-states.yaml). capacity-dmips-mhz: - $ref: '/schemas/types.yaml#/definitions/uint32' description: u32 value representing CPU capacity (see ./cpu-capacity.txt) in DMIPS/MHz, relative to highest capacity-dmips-mhz diff --git a/Documentation/devicetree/bindings/extcon/wlf,arizona.yaml b/Documentation/devicetree/bindings/extcon/wlf,arizona.yaml index 5fe784f487c5..efdf59abb2e1 100644 --- a/Documentation/devicetree/bindings/extcon/wlf,arizona.yaml +++ b/Documentation/devicetree/bindings/extcon/wlf,arizona.yaml @@ -85,7 +85,6 @@ properties: wlf,micd-timeout-ms: description: Timeout for microphone detection, specified in milliseconds. - $ref: "/schemas/types.yaml#/definitions/uint32" wlf,micd-force-micbias: description: diff --git a/Documentation/devicetree/bindings/hwmon/adi,ltc2947.yaml b/Documentation/devicetree/bindings/hwmon/adi,ltc2947.yaml index eef614962b10..bf04151b63d2 100644 --- a/Documentation/devicetree/bindings/hwmon/adi,ltc2947.yaml +++ b/Documentation/devicetree/bindings/hwmon/adi,ltc2947.yaml @@ -49,7 +49,6 @@ properties: description: This property controls the Accumulation Dead band which allows to set the level of current below which no accumulation takes place. - $ref: /schemas/types.yaml#/definitions/uint32 maximum: 255 default: 0 diff --git a/Documentation/devicetree/bindings/hwmon/baikal,bt1-pvt.yaml b/Documentation/devicetree/bindings/hwmon/baikal,bt1-pvt.yaml index 00a6511354e6..5d3ce641fcde 100644 --- a/Documentation/devicetree/bindings/hwmon/baikal,bt1-pvt.yaml +++ b/Documentation/devicetree/bindings/hwmon/baikal,bt1-pvt.yaml @@ -73,11 +73,9 @@ properties: description: | Temperature sensor trimming factor. It can be used to manually adjust the temperature measurements within 7.130 degrees Celsius. - maxItems: 1 - items: - default: 0 - minimum: 0 - maximum: 7130 + default: 0 + minimum: 0 + maximum: 7130 additionalProperties: false diff --git a/Documentation/devicetree/bindings/hwmon/ti,tmp513.yaml b/Documentation/devicetree/bindings/hwmon/ti,tmp513.yaml index 8020d739a078..1502b22c77cc 100644 --- a/Documentation/devicetree/bindings/hwmon/ti,tmp513.yaml +++ b/Documentation/devicetree/bindings/hwmon/ti,tmp513.yaml @@ -52,7 +52,6 @@ properties: ti,bus-range-microvolt: description: | This is the operating range of the bus voltage in microvolt - $ref: /schemas/types.yaml#/definitions/uint32 enum: [16000000, 32000000] default: 32000000 diff --git a/Documentation/devicetree/bindings/i2c/i2c-gpio.yaml b/Documentation/devicetree/bindings/i2c/i2c-gpio.yaml index cc3aa2a5e70b..ff99344788ab 100644 --- a/Documentation/devicetree/bindings/i2c/i2c-gpio.yaml +++ b/Documentation/devicetree/bindings/i2c/i2c-gpio.yaml @@ -39,11 +39,9 @@ properties: i2c-gpio,delay-us: description: delay between GPIO operations (may depend on each platform) - $ref: /schemas/types.yaml#/definitions/uint32 i2c-gpio,timeout-ms: description: timeout to get data - $ref: /schemas/types.yaml#/definitions/uint32 # Deprecated properties, do not use in new device tree sources: gpios: diff --git a/Documentation/devicetree/bindings/i2c/snps,designware-i2c.yaml b/Documentation/devicetree/bindings/i2c/snps,designware-i2c.yaml index c22b66b6219e..d9293c57f573 100644 --- a/Documentation/devicetree/bindings/i2c/snps,designware-i2c.yaml +++ b/Documentation/devicetree/bindings/i2c/snps,designware-i2c.yaml @@ -66,21 +66,18 @@ properties: default: 400000 i2c-sda-hold-time-ns: - maxItems: 1 description: | The property should contain the SDA hold time in nanoseconds. This option is only supported in hardware blocks version 1.11a or newer or on Microsemi SoCs. i2c-scl-falling-time-ns: - maxItems: 1 description: | The property should contain the SCL falling time in nanoseconds. This value is used to compute the tLOW period. default: 300 i2c-sda-falling-time-ns: - maxItems: 1 description: | The property should contain the SDA falling time in nanoseconds. This value is used to compute the tHIGH period. diff --git a/Documentation/devicetree/bindings/iio/adc/maxim,max9611.yaml b/Documentation/devicetree/bindings/iio/adc/maxim,max9611.yaml index 9475a9e6e920..95774a55629d 100644 --- a/Documentation/devicetree/bindings/iio/adc/maxim,max9611.yaml +++ b/Documentation/devicetree/bindings/iio/adc/maxim,max9611.yaml @@ -23,7 +23,6 @@ properties: maxItems: 1 shunt-resistor-micro-ohms: - $ref: /schemas/types.yaml#/definitions/uint32 description: | Value in micro Ohms of the shunt resistor connected between the RS+ and RS- inputs, across which the current is measured. Value needed to compute diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml index 6364ede9bb5f..a58334c3bb76 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml @@ -248,7 +248,6 @@ patternProperties: Resolution (bits) to use for conversions: - can be 6, 8, 10 or 12 on stm32f4 - can be 8, 10, 12, 14 or 16 on stm32h7 and stm32mp1 - $ref: /schemas/types.yaml#/definitions/uint32 st,adc-channels: description: | diff --git a/Documentation/devicetree/bindings/iio/adc/ti,palmas-gpadc.yaml b/Documentation/devicetree/bindings/iio/adc/ti,palmas-gpadc.yaml index 692dacd0fee5..7b895784e008 100644 --- a/Documentation/devicetree/bindings/iio/adc/ti,palmas-gpadc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/ti,palmas-gpadc.yaml @@ -42,7 +42,6 @@ properties: const: 1 ti,channel0-current-microamp: - $ref: /schemas/types.yaml#/definitions/uint32 description: Channel 0 current in uA. enum: - 0 @@ -51,7 +50,6 @@ properties: - 20 ti,channel3-current-microamp: - $ref: /schemas/types.yaml#/definitions/uint32 description: Channel 3 current in uA. enum: - 0 diff --git a/Documentation/devicetree/bindings/iio/dac/adi,ad5758.yaml b/Documentation/devicetree/bindings/iio/dac/adi,ad5758.yaml index 626ccb6fe21e..fd4edca34a28 100644 --- a/Documentation/devicetree/bindings/iio/dac/adi,ad5758.yaml +++ b/Documentation/devicetree/bindings/iio/dac/adi,ad5758.yaml @@ -46,31 +46,42 @@ properties: two properties must be present: adi,range-microvolt: - $ref: /schemas/types.yaml#/definitions/int32-array description: | Voltage output range specified as <minimum, maximum> - enum: - - [[0, 5000000]] - - [[0, 10000000]] - - [[-5000000, 5000000]] - - [[-10000000, 10000000]] + oneOf: + - items: + - const: 0 + - enum: [5000000, 10000000] + - items: + - const: -5000000 + - const: 5000000 + - items: + - const: -10000000 + - const: 10000000 adi,range-microamp: - $ref: /schemas/types.yaml#/definitions/int32-array description: | Current output range specified as <minimum, maximum> - enum: - - [[0, 20000]] - - [[0, 24000]] - - [[4, 24000]] - - [[-20000, 20000]] - - [[-24000, 24000]] - - [[-1000, 22000]] + oneOf: + - items: + - const: 0 + - enum: [20000, 24000] + - items: + - const: 4 + - const: 24000 + - items: + - const: -20000 + - const: 20000 + - items: + - const: -24000 + - const: 24000 + - items: + - const: -1000 + - const: 22000 reset-gpios: true adi,dc-dc-ilim-microamp: - $ref: /schemas/types.yaml#/definitions/uint32 enum: [150000, 200000, 250000, 300000, 350000, 400000] description: | The dc-to-dc converter current limit. diff --git a/Documentation/devicetree/bindings/iio/health/maxim,max30100.yaml b/Documentation/devicetree/bindings/iio/health/maxim,max30100.yaml index 64b862637039..967778fb0ce8 100644 --- a/Documentation/devicetree/bindings/iio/health/maxim,max30100.yaml +++ b/Documentation/devicetree/bindings/iio/health/maxim,max30100.yaml @@ -21,7 +21,6 @@ properties: description: Connected to ADC_RDY pin. maxim,led-current-microamp: - $ref: /schemas/types.yaml#/definitions/uint32-array minItems: 2 maxItems: 2 description: | diff --git a/Documentation/devicetree/bindings/input/touchscreen/touchscreen.yaml b/Documentation/devicetree/bindings/input/touchscreen/touchscreen.yaml index a771a15f053f..046ace461cc9 100644 --- a/Documentation/devicetree/bindings/input/touchscreen/touchscreen.yaml +++ b/Documentation/devicetree/bindings/input/touchscreen/touchscreen.yaml @@ -70,11 +70,9 @@ properties: touchscreen-x-mm: description: horizontal length in mm of the touchscreen - $ref: /schemas/types.yaml#/definitions/uint32 touchscreen-y-mm: description: vertical length in mm of the touchscreen - $ref: /schemas/types.yaml#/definitions/uint32 dependencies: touchscreen-size-x: [ touchscreen-size-y ] diff --git a/Documentation/devicetree/bindings/mmc/mmc-controller.yaml b/Documentation/devicetree/bindings/mmc/mmc-controller.yaml index df4ee4c778ae..e141330c1114 100644 --- a/Documentation/devicetree/bindings/mmc/mmc-controller.yaml +++ b/Documentation/devicetree/bindings/mmc/mmc-controller.yaml @@ -261,7 +261,6 @@ properties: waiting for I/O signalling and card power supply to be stable, regardless of whether pwrseq-simple is used. Default to 10ms if no available. - $ref: /schemas/types.yaml#/definitions/uint32 default: 10 supports-cqe: diff --git a/Documentation/devicetree/bindings/mmc/mmc-pwrseq-simple.yaml b/Documentation/devicetree/bindings/mmc/mmc-pwrseq-simple.yaml index 6cd57863c1db..226fb191913d 100644 --- a/Documentation/devicetree/bindings/mmc/mmc-pwrseq-simple.yaml +++ b/Documentation/devicetree/bindings/mmc/mmc-pwrseq-simple.yaml @@ -41,13 +41,11 @@ properties: description: Delay in ms after powering the card and de-asserting the reset-gpios (if any). - $ref: /schemas/types.yaml#/definitions/uint32 power-off-delay-us: description: Delay in us after asserting the reset-gpios (if any) during power off of the card. - $ref: /schemas/types.yaml#/definitions/uint32 required: - compatible diff --git a/Documentation/devicetree/bindings/net/ethernet-controller.yaml b/Documentation/devicetree/bindings/net/ethernet-controller.yaml index 0965f6515f9e..dac4aadb6e2e 100644 --- a/Documentation/devicetree/bindings/net/ethernet-controller.yaml +++ b/Documentation/devicetree/bindings/net/ethernet-controller.yaml @@ -122,7 +122,6 @@ properties: such as flow control thresholds. rx-internal-delay-ps: - $ref: /schemas/types.yaml#/definitions/uint32 description: | RGMII Receive Clock Delay defined in pico seconds. This is used for controllers that have configurable RX internal delays. @@ -140,7 +139,6 @@ properties: is used for components that can have configurable fifo sizes. tx-internal-delay-ps: - $ref: /schemas/types.yaml#/definitions/uint32 description: | RGMII Transmit Clock Delay defined in pico seconds. This is used for controllers that have configurable TX internal delays. diff --git a/Documentation/devicetree/bindings/net/snps,dwmac.yaml b/Documentation/devicetree/bindings/net/snps,dwmac.yaml index b2f6083f556a..9ac77b8cb767 100644 --- a/Documentation/devicetree/bindings/net/snps,dwmac.yaml +++ b/Documentation/devicetree/bindings/net/snps,dwmac.yaml @@ -208,7 +208,6 @@ properties: Triplet of delays. The 1st cell is reset pre-delay in micro seconds. The 2nd cell is reset pulse in micro seconds. The 3rd cell is reset post-delay in micro seconds. - $ref: /schemas/types.yaml#/definitions/uint32-array minItems: 3 maxItems: 3 diff --git a/Documentation/devicetree/bindings/power/supply/battery.yaml b/Documentation/devicetree/bindings/power/supply/battery.yaml index 0c7e2e44793b..c3b4b7543591 100644 --- a/Documentation/devicetree/bindings/power/supply/battery.yaml +++ b/Documentation/devicetree/bindings/power/supply/battery.yaml @@ -83,21 +83,18 @@ properties: for each of the battery capacity lookup table. operating-range-celsius: - $ref: /schemas/types.yaml#/definitions/uint32-array description: operating temperature range of a battery items: - description: minimum temperature at which battery can operate - description: maximum temperature at which battery can operate ambient-celsius: - $ref: /schemas/types.yaml#/definitions/uint32-array description: safe range of ambient temperature items: - description: alert when ambient temperature is lower than this value - description: alert when ambient temperature is higher than this value alert-celsius: - $ref: /schemas/types.yaml#/definitions/uint32-array description: safe range of battery temperature items: - description: alert when battery temperature is lower than this value diff --git a/Documentation/devicetree/bindings/power/supply/bq2515x.yaml b/Documentation/devicetree/bindings/power/supply/bq2515x.yaml index 75a56773be4a..813d6afde606 100644 --- a/Documentation/devicetree/bindings/power/supply/bq2515x.yaml +++ b/Documentation/devicetree/bindings/power/supply/bq2515x.yaml @@ -50,7 +50,6 @@ properties: maxItems: 1 input-current-limit-microamp: - $ref: /schemas/types.yaml#/definitions/uint32 description: Maximum input current in micro Amps. minimum: 50000 maximum: 500000 diff --git a/Documentation/devicetree/bindings/regulator/dlg,da9121.yaml b/Documentation/devicetree/bindings/regulator/dlg,da9121.yaml index 6f2164f7bc57..228018c87bea 100644 --- a/Documentation/devicetree/bindings/regulator/dlg,da9121.yaml +++ b/Documentation/devicetree/bindings/regulator/dlg,da9121.yaml @@ -62,7 +62,6 @@ properties: description: IRQ line information. dlg,irq-polling-delay-passive-ms: - $ref: "/schemas/types.yaml#/definitions/uint32" minimum: 1000 maximum: 10000 description: | diff --git a/Documentation/devicetree/bindings/regulator/fixed-regulator.yaml b/Documentation/devicetree/bindings/regulator/fixed-regulator.yaml index d3d0dc13dd8b..8850c01bd470 100644 --- a/Documentation/devicetree/bindings/regulator/fixed-regulator.yaml +++ b/Documentation/devicetree/bindings/regulator/fixed-regulator.yaml @@ -72,11 +72,9 @@ properties: startup-delay-us: description: startup time in microseconds - $ref: /schemas/types.yaml#/definitions/uint32 off-on-delay-us: description: off delay time in microseconds - $ref: /schemas/types.yaml#/definitions/uint32 enable-active-high: description: diff --git a/Documentation/devicetree/bindings/rtc/rtc.yaml b/Documentation/devicetree/bindings/rtc/rtc.yaml index d30dc045aac6..0ec3551f12dd 100644 --- a/Documentation/devicetree/bindings/rtc/rtc.yaml +++ b/Documentation/devicetree/bindings/rtc/rtc.yaml @@ -27,7 +27,6 @@ properties: 1: chargeable quartz-load-femtofarads: - $ref: /schemas/types.yaml#/definitions/uint32 description: The capacitive load of the quartz(x-tal), expressed in femto Farad (fF). The default value shall be listed (if optional), @@ -47,7 +46,6 @@ properties: deprecated: true trickle-resistor-ohms: - $ref: /schemas/types.yaml#/definitions/uint32 description: Selected resistor for trickle charger. Should be given if trickle charger should be enabled. diff --git a/Documentation/devicetree/bindings/serial/pl011.yaml b/Documentation/devicetree/bindings/serial/pl011.yaml index c23c93b400f0..07fa6d26f2b4 100644 --- a/Documentation/devicetree/bindings/serial/pl011.yaml +++ b/Documentation/devicetree/bindings/serial/pl011.yaml @@ -88,14 +88,12 @@ properties: description: Rate at which poll occurs when auto-poll is set. default 100ms. - $ref: /schemas/types.yaml#/definitions/uint32 default: 100 poll-timeout-ms: description: Poll timeout when auto-poll is set, default 3000ms. - $ref: /schemas/types.yaml#/definitions/uint32 default: 3000 required: diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml index d116c174b545..70b4a8831073 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.yaml +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -41,14 +41,12 @@ properties: values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not mentioned or if the value is unknown, then micbias resistor is set to 4k. - $ref: "/schemas/types.yaml#/definitions/uint32" enum: [ 0, 2, 4, 8 ] micbias-voltage-m-volts: description: The bias voltage to be used in mVolts. The voltage can take values from 1.25V to 3V by 250mV steps. If this node is not mentioned or the value is unknown, then the value is set to 1.25V. - $ref: "/schemas/types.yaml#/definitions/uint32" enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ] lrclk-strength: diff --git a/Documentation/devicetree/bindings/watchdog/watchdog.yaml b/Documentation/devicetree/bindings/watchdog/watchdog.yaml index 4e2c26cd981d..e3dfb02f0ca5 100644 --- a/Documentation/devicetree/bindings/watchdog/watchdog.yaml +++ b/Documentation/devicetree/bindings/watchdog/watchdog.yaml @@ -19,7 +19,6 @@ properties: pattern: "^watchdog(@.*|-[0-9a-f])?$" timeout-sec: - $ref: /schemas/types.yaml#/definitions/uint32 description: Contains the watchdog timeout in seconds. -- 2.27.0

8 7

[PATCH AUTOSEL 5.4 11/19] ALSA: hda: Add Cometlake-R PCI ID
by Sasha Levin 29 Jan '21

29 Jan '21

From: Kai-Chuan Hsieh <kaichuan.hsieh(a)canonical.com> [ Upstream commit f84d3a1ec375e46a55cc3ba85c04272b24bd3921 ] Add HD Audio Device PCI ID for the Intel Cometlake-R platform Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Kai-Chuan Hsieh <kaichuan.hsieh(a)canonical.com> Link: https://lore.kernel.org/r/20210115031515.13100-1-kaichuan.hsieh@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5f515a29668c8..b3667a5efdc1f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2450,6 +2450,9 @@ static const struct pci_device_id azx_ids[] = { /* CometLake-S */ { PCI_DEVICE(0x8086, 0xa3f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-R */ + { PCI_DEVICE(0x8086, 0xf0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 08/19] ASoC: SOF: Intel: hda: Resume codec to do jack detection
by Sasha Levin 29 Jan '21

29 Jan '21

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit bcd7059abc19e6ec5b2260dff6a008fb99c4eef9 ] Instead of queueing jackpoll_work, runtime resume the codec to let it use different jack detection methods based on jackpoll_interval. This partially matches SOF driver's behavior with commit a6e7d0a4bdb0 ("ALSA: hda: fix jack detection with Realtek codecs when in D3"), the difference is SOF unconditionally resumes the codec. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20210112181128.1229827-1-kai.heng.feng@canonical.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 9e8233c10d860..df38616c431a6 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -68,8 +68,7 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) * has been recorded in STATESTS */ if (codec->jacktbl.used) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); + pm_request_resume(&codec->core.dev); } #else void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {} -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 19/41] ALSA: hda: Add Cometlake-R PCI ID
by Sasha Levin 29 Jan '21

29 Jan '21

From: Kai-Chuan Hsieh <kaichuan.hsieh(a)canonical.com> [ Upstream commit f84d3a1ec375e46a55cc3ba85c04272b24bd3921 ] Add HD Audio Device PCI ID for the Intel Cometlake-R platform Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Kai-Chuan Hsieh <kaichuan.hsieh(a)canonical.com> Link: https://lore.kernel.org/r/20210115031515.13100-1-kaichuan.hsieh@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 246d660164691..d393401db1ec5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2484,6 +2484,9 @@ static const struct pci_device_id azx_ids[] = { /* CometLake-S */ { PCI_DEVICE(0x8086, 0xa3f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-R */ + { PCI_DEVICE(0x8086, 0xf0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 13/41] ALSA: hda: Add AlderLake-P PCI ID and HDMI codec vid
by Sasha Levin 29 Jan '21

29 Jan '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 5e941fc033e411118fb3a7d9e0b97f8cf702cd39 ] Add HD Audio PCI ID and HDMI codec vendor ID for Intel AlderLake-P. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20210113155629.4097057-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 770ad25f1907c..246d660164691 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2507,6 +2507,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-P */ + { PCI_DEVICE(0x8086, 0x51c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d12b4799c3cb7..dc1ab4fc93a5b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4349,6 +4349,7 @@ HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 12/41] ASoC: SOF: Intel: hda: Resume codec to do jack detection
by Sasha Levin 29 Jan '21

29 Jan '21

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit bcd7059abc19e6ec5b2260dff6a008fb99c4eef9 ] Instead of queueing jackpoll_work, runtime resume the codec to let it use different jack detection methods based on jackpoll_interval. This partially matches SOF driver's behavior with commit a6e7d0a4bdb0 ("ALSA: hda: fix jack detection with Realtek codecs when in D3"), the difference is SOF unconditionally resumes the codec. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20210112181128.1229827-1-kai.heng.feng@canonical.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 8b0ddc4b8227b..8d65004c917a1 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -93,8 +93,7 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) * has been recorded in STATESTS */ if (codec->jacktbl.used) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); + pm_request_resume(&codec->core.dev); } #else void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {} -- 2.27.0

1 0

[PATCH 0/7] ASoC: codecs: add support for LPASS Codec TX and RX macros
by Srinivas Kandagatla 29 Jan '21

29 Jan '21

This patchset adds support for two Codec Macro blocks(TX and RX) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD938x Codecs to provide headphone/ear/lineout/amic/dmic etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to SM8250 MTP Development Board. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively and WCD938x TX and RX connected via Soundwire via TX and RX Macros reseptively. Thanks, srini Srinivas Kandagatla (7): ASoC: qcom: dt-bindings: add bindings for lpass rx macro codec ASoC: codec: lpass-rx-macro: add support for lpass rx macro ASoC: codec: lpass-rx-macro: add dapm widgets and route ASoC: codec: lpass-rx-macro: add iir widgets ASoC: qcom: dt-bindings: add bindings for lpass tx macro codec ASoC: codec: lpass-tx-macro: add support for lpass tx macro ASoC: codec: lpass-tx-macro: add dapm widgets and route .../bindings/sound/qcom,lpass-rx-macro.yaml | 62 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 67 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-rx-macro.c | 3623 +++++++++++++++++ sound/soc/codecs/lpass-tx-macro.c | 1881 +++++++++ 6 files changed, 5647 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml create mode 100644 sound/soc/codecs/lpass-rx-macro.c create mode 100644 sound/soc/codecs/lpass-tx-macro.c -- 2.21.0

2 11

[PATCH] ASoC: soc-component: fix undefined reference to __ffssi2
by Srinivas Kandagatla 29 Jan '21

29 Jan '21

microblaze-linux-gcc (GCC) 9.3.0 complains about missing __ffssi2 symbol while using __builtin_ffs at runtime. This is because arch/h8300 is compiled with -fno-builtin option. so fallback and use kernel ffs() instead to all the arch builds happy! Fixes: 1da0b9899abd ("ASoC: soc-component: add snd_soc_component_read/write_field()) Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-component.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 361a79d655e3..159bf88b9f8c 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -11,6 +11,7 @@ #include <linux/module.h> #include <linux/pm_runtime.h> #include <sound/soc.h> +#include <linux/bitops.h> #define soc_component_ret(dai, ret) _soc_component_ret(dai, __func__, ret) static inline int _soc_component_ret(struct snd_soc_component *component, @@ -43,7 +44,7 @@ static inline int soc_component_field_shift(struct snd_soc_component *component, return 0; } - return (__builtin_ffs(mask) - 1); + return (ffs(mask) - 1); } /* -- 2.21.0

2 1

[PATCHv2] ASoC: audio-graph-card: Add audio mixer for Motorola mdm6600
by Pavel Machek 29 Jan '21

29 Jan '21

motmdm.c handles audio configuration on "gsmtty2"; it needs to know whether we are in call or not; that part is in motmdm-state.c and listens on "gsmtty1". To configure Alsamixer for voice calls do for example: Speaker Right -> Voice Call Noise Cancellation -> Unmute Call Output -> Speakerphone Call -> 100 Mic2 -> 40 Left -> Mic 2 Voice -> 55 Mic2 -> 40 Left -> Mic 2 Tony wrote original version using custom interface to n_gsm, Pavel switched it to plain serdev and split it to two drivers to be easier to debug and understand. Credit is Tony's, bugs are probably Pavel's. Signed-off-by: Pavel Machek <pavel(a)ucw.cz> Co-authored-by: Tony Lindgren <tony(a)atomide.com> --- v2: fix compile error. diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ba4eb54aafcb..94000ab3cf3e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -932,6 +932,14 @@ config SND_SOC_MAX9860 depends on I2C select REGMAP_I2C +config SND_SOC_MOTMDM + tristate "Motorola Modem TS 27.010 Voice Call Codec" + depends on SERIAL_DEV_BUS + help + Enable support for Motorola TS 27.010 serdev voice + call codec driver for Motorola Mapphone series of + devices such as Droid 4. + config SND_SOC_MSM8916_WCD_ANALOG tristate "Qualcomm MSM8916 WCD Analog Codec" depends on SPMI || COMPILE_TEST diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d277f0366e09..f63022e0a292 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -128,6 +128,8 @@ snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-motmdm-objs := motmdm.o +snd-soc-motmdm-state-objs := motmdm-state.o snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o @@ -443,6 +445,8 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_MOTMDM) += snd-soc-motmdm.o +obj-$(CONFIG_SND_SOC_MOTMDM) += snd-soc-motmdm-state.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o diff --git a/sound/soc/codecs/motmdm-state.c b/sound/soc/codecs/motmdm-state.c new file mode 100644 index 000000000000..166de19be256 --- /dev/null +++ b/sound/soc/codecs/motmdm-state.c @@ -0,0 +1,177 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Motorola Mapphone MDM6600 voice call audio support + * Copyright 2018 - 2020 Tony Lindgren <tony(a)atomide.com> + * Copyright 2020 - 2021 Pavel Machek <pavel(a)ucw.cz> + * + * Designed to provide notifications about voice call state to the + * motmdm.c driver. This one listens on "gsmtty1". + */ + +#include <linux/init.h> +#include <linux/kfifo.h> +#include <linux/module.h> +#include <linux/of_graph.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <linux/serdev.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#define MOTMDM_HEADER_LEN 5 /* U1234 */ +#define MOTMDM_AUDIO_MAX_LEN 128 +#define MOTMDM_VOICE_RESP_LEN 7 /* U1234~+CIEV= */ + +struct motmdm_driver_data { + struct serdev_device *serdev; + unsigned char *buf; + size_t len; + spinlock_t lock; /* enable/disabled lock */ +}; + +static BLOCKING_NOTIFIER_HEAD(modem_state_chain_head); + +int register_modem_state_notifier(struct notifier_block *nb) +{ + return blocking_notifier_chain_register(&modem_state_chain_head, nb); +} +EXPORT_SYMBOL_GPL(register_modem_state_notifier); + +int unregister_modem_state_notifier(struct notifier_block *nb) +{ + return blocking_notifier_chain_unregister(&modem_state_chain_head, nb); +} +EXPORT_SYMBOL_GPL(unregister_modem_state_notifier); + +static int modem_state_notifier_call_chain(unsigned long val) +{ + int ret; + ret = __blocking_notifier_call_chain(&modem_state_chain_head, val, NULL, + -1, NULL); + return notifier_to_errno(ret); +} + +/* Parses the voice call state from unsolicited notifications on dlci1 */ +static int motmdm_voice_get_state(struct motmdm_driver_data *ddata, + const unsigned char *buf, + size_t len) +{ + struct device *dev = &ddata->serdev->dev; + bool enable; + const unsigned char *state; + + if (len < MOTMDM_HEADER_LEN + MOTMDM_VOICE_RESP_LEN + 5) + return 0; + + /* We only care about the unsolicted messages */ + if (buf[MOTMDM_HEADER_LEN] != '~') + return 0; + + if (strncmp(buf + MOTMDM_HEADER_LEN + 1, "+CIEV=", 6)) + return len; + + state = buf + MOTMDM_HEADER_LEN + MOTMDM_VOICE_RESP_LEN; + dev_info(dev, "%s: ciev=%5s\n", __func__, state); + + if (!strncmp(state, "1,1,0", 5) || /* connecting */ + !strncmp(state, "1,4,0", 5) || /* incoming call */ + !strncmp(state, "1,2,0", 5)) /* connected */ + enable = true; + else if (!strncmp(state, "1,0,0", 5) || /* disconnected */ + !strncmp(state, "1,0,2", 5)) /* call failed */ + enable = false; + else + return len; + + modem_state_notifier_call_chain(enable); + return len; +} + +static int voice_receive_data(struct serdev_device *serdev, + const unsigned char *buf, + size_t len) +{ + struct motmdm_driver_data *ddata = serdev_device_get_drvdata(serdev); + + if (len > MOTMDM_AUDIO_MAX_LEN) + len = MOTMDM_AUDIO_MAX_LEN; + + if (len <= MOTMDM_HEADER_LEN) + return 0; + + if (buf[MOTMDM_HEADER_LEN] == '~') + motmdm_voice_get_state(ddata, buf, len); + + return len; +} + +static const struct serdev_device_ops voice_serdev_ops = { + .receive_buf = voice_receive_data, + .write_wakeup = serdev_device_write_wakeup, +}; + +static void motmdm_free_voice_serdev(struct motmdm_driver_data *ddata) +{ + serdev_device_close(ddata->serdev); +} + +static int motmdm_soc_probe(struct serdev_device *serdev) +{ + struct device *dev = &serdev->dev; + struct motmdm_driver_data *ddata; + int error; + + ddata = devm_kzalloc(dev, sizeof(*ddata), GFP_KERNEL); + if (!ddata) + return -ENOMEM; + + ddata->serdev = serdev; + spin_lock_init(&ddata->lock); + ddata->len = MOTMDM_AUDIO_MAX_LEN; + + ddata->buf = devm_kzalloc(dev, ddata->len, GFP_KERNEL); + if (!ddata->buf) + return -ENOMEM; + + serdev_device_set_drvdata(ddata->serdev, ddata); + serdev_device_set_client_ops(ddata->serdev, &voice_serdev_ops); + + error = serdev_device_open(ddata->serdev); + return error; +} + +static void motmdm_state_remove(struct serdev_device *serdev) +{ + struct motmdm_driver_data *ddata = serdev_device_get_drvdata(serdev); + + motmdm_free_voice_serdev(ddata); +} + +static int motmdm_state_probe(struct serdev_device *serdev) +{ + return motmdm_soc_probe(serdev); +} + +#ifdef CONFIG_OF +static const struct of_device_id motmdm_of_match[] = { + { .compatible = "motorola,mapphone-mdm6600-modem" }, + {}, +}; +MODULE_DEVICE_TABLE(of, motmdm_of_match); +#endif + +static struct serdev_device_driver motmdm_state_driver = { + .driver = { + .name = "mot-mdm6600-modem", + .of_match_table = of_match_ptr(motmdm_of_match), + }, + .probe = motmdm_state_probe, + .remove = motmdm_state_remove, +}; +module_serdev_device_driver(motmdm_state_driver); + +MODULE_ALIAS("platform:motmdm-state"); +MODULE_DESCRIPTION("Motorola Mapphone MDM6600 modem state driver"); +MODULE_AUTHOR("Pavel Machek <pavel(a)ucw.cz>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/motmdm.c b/sound/soc/codecs/motmdm.c new file mode 100644 index 000000000000..0ee546133471 --- /dev/null +++ b/sound/soc/codecs/motmdm.c @@ -0,0 +1,688 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Motorola Mapphone MDM6600 voice call audio support + * Copyright 2018 - 2020 Tony Lindgren <tony(a)atomide.com> + * Copyright 2020 - 2021 Pavel Machek <pavel(a)ucw.cz> + * + * This one handles audio configuration on "gsmtty2"; it needs to know + * whether we are in call or not, and that part is in motmdm-state.c + * and listens on "gsmtty1". + */ + +#include <linux/init.h> +#include <linux/kfifo.h> +#include <linux/module.h> +#include <linux/of_graph.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <linux/serdev.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "motmdm-state.h" + +#define MOTMDM_HEADER_LEN 5 /* U1234 */ + +#define MOTMDM_AUDIO_RESP_LEN 6 /* U1234+XXXX= */ +#define MOTMDM_AUDIO_MAX_LEN 128 + +#define MOTMDM_VOICE_RESP_LEN 7 /* U1234~+CIEV= */ + +struct motmdm_driver_data { + struct notifier_block notifier; + struct snd_soc_component *component; + struct snd_soc_dai *master_dai; + struct device *modem; + struct serdev_device *serdev; + struct regmap *regmap; + unsigned char *buf; + size_t len; + unsigned int parsed:1; + unsigned int enabled:1; + spinlock_t lock; /* enable/disabled lock */ + struct mutex mutex; /* for sending commands */ + wait_queue_head_t read_queue; + + unsigned int dtmf_val; + unsigned int dtmf_en; +}; + +enum motmdm_cmd { + CMD_AT_EACC, + CMD_AT_CLVL, + CMD_AT_NREC, +}; + +const char * const motmd_read_fmt[] = { + [CMD_AT_EACC] = "AT+EACC?", + [CMD_AT_CLVL] = "AT+CLVL?", + [CMD_AT_NREC] = "AT+NREC?", +}; + +const char * const motmd_write_fmt[] = { + [CMD_AT_EACC] = "AT+EACC=%u,0", + [CMD_AT_CLVL] = "AT+CLVL=%u", + [CMD_AT_NREC] = "AT+NREC=%u", +}; + +/* + * Currently unconfigured additional inactive (error producing) options + * seem to be: + * "TTY Headset", "HCQ Headset", "VCQ Headset", "No-Mic Headset", + * "Handset Fluence Med", "Handset Fluence Low", "Car Dock", "Lapdock" + */ +static const char * const motmdm_out_mux_texts[] = { + "Handset", "Headset", "Speakerphone", "Bluetooth", +}; + +static SOC_ENUM_SINGLE_EXT_DECL(motmdm_out_enum, motmdm_out_mux_texts); + +static const DECLARE_TLV_DB_SCALE(motmdm_gain_tlv, 0, 100, 0); + +int motmdm_send_command(struct motmdm_driver_data *ddata, + const u8 *buf, int len) +{ + struct device *dev = ddata->component->dev; + const int timeout_ms = 1000; + unsigned char cmd[MOTMDM_AUDIO_MAX_LEN]; + int ret, cmdlen; + + cmdlen = len + 5 + 1; + if (cmdlen > MOTMDM_AUDIO_MAX_LEN) + return -EINVAL; + + mutex_lock(&ddata->mutex); + memset(ddata->buf, 0, ddata->len); + ddata->parsed = false; + snprintf(cmd, cmdlen, "U%04li%s", jiffies % 10000, buf); + + ret = serdev_device_write(ddata->serdev, cmd, cmdlen, MAX_SCHEDULE_TIMEOUT); + if (ret < 0) + goto out_unlock; + + serdev_device_wait_until_sent(ddata->serdev, 0); + + ret = wait_event_timeout(ddata->read_queue, ddata->parsed, + msecs_to_jiffies(timeout_ms)); + if (ret == 0) { + ret = -ETIMEDOUT; + goto out_unlock; + } else if (ret < 0) { + goto out_unlock; + } + + if (strstr(ddata->buf, "ERROR")) { + dev_err(dev, "command %s error %s\n", cmd, ddata->buf); + ret = -EPIPE; + } + + ret = len; + +out_unlock: + mutex_unlock(&ddata->mutex); + printk("send_command -- ret %d\n", ret); + + return ret; +} + +static int motmdm_read_reg(void *context, unsigned int reg, + unsigned int *value) +{ + struct snd_soc_component *component = context; + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + const unsigned char *cmd; + unsigned int val; + int error; + + cmd = motmd_read_fmt[reg]; + error = motmdm_send_command(ddata, cmd, strlen(cmd)); + if (error < 0) { + dev_err(component->dev, "%s: %s failed with %i\n", + __func__, cmd, error); + + return error; + } + + error = kstrtouint(ddata->buf + MOTMDM_AUDIO_RESP_LEN, 0, &val); + if (error) + return -ENODEV; + + *value = val; + + return error; +} + +static int motmdm_write_reg(void *context, unsigned int reg, + unsigned int value) +{ + struct snd_soc_component *component = context; + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + const unsigned char *fmt, *cmd; + int error; + + fmt = motmd_write_fmt[reg]; + cmd = kasprintf(GFP_KERNEL, fmt, value); + if (!cmd) { + error = -ENOMEM; + goto free; + } + + error = motmdm_send_command(ddata, cmd, strlen(cmd)); + if (error < 0) + dev_err(component->dev, "%s: %s failed with %i\n", + __func__, cmd, error); + +free: + kfree(cmd); + + return error; +} + +static const struct reg_default motmdm_reg_defaults[] = { + { CMD_AT_EACC, 0x0 }, + { CMD_AT_CLVL, 0x0 }, +}; + +static const struct regmap_config motmdm_regmap = { + .reg_bits = 32, + .reg_stride = 1, + .val_bits = 32, + .max_register = CMD_AT_NREC, + .reg_defaults = motmdm_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(motmdm_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .reg_read = motmdm_read_reg, + .reg_write = motmdm_write_reg, +}; + +static int motmdm_value_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + enum motmdm_cmd reg, + int cmd_base) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + unsigned int val; + int error; + + error = regmap_read(ddata->regmap, reg, &val); + if (error) + return error; + + if (val >= cmd_base) + val -= cmd_base; + + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +static int motmdm_value_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + enum motmdm_cmd reg, + int cmd_base) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + int error; + + error = regmap_write(ddata->regmap, reg, + ucontrol->value.enumerated.item[0] + cmd_base); + if (error) + return error; + + regcache_mark_dirty(ddata->regmap); + + return error; +} + +static int motmdm_audio_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_get(kcontrol, ucontrol, CMD_AT_EACC, 1); +} + +static int motmdm_audio_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_put(kcontrol, ucontrol, CMD_AT_EACC, 1); +} + +static int motmdm_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_get(kcontrol, ucontrol, CMD_AT_CLVL, 0); +} + +static int motmdm_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_put(kcontrol, ucontrol, CMD_AT_CLVL, 0); +} + +static int motmdm_noise_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_get(kcontrol, ucontrol, CMD_AT_NREC, 0); +} + +static int motmdm_noise_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return motmdm_value_put(kcontrol, ucontrol, CMD_AT_NREC, 0); +} + +static const char * const motmdm_tonegen_dtmf_key_txt[] = { + "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D", + "*", "#" +}; + +static SOC_ENUM_SINGLE_EXT_DECL(motmd_tonegen_dtmf_enum, + motmdm_tonegen_dtmf_key_txt); + +static int motmdm_dtmf_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = ddata->dtmf_val; + + return 0; +} + +static int motmdm_dtmf_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + + ddata->dtmf_val = ucontrol->value.enumerated.item[0]; + + return 0; +} + +static int motmdm_tonegen_dtmf_send_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = ddata->dtmf_en; + + return 0; +} + +static int motmdm_tonegen_dtmf_send_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + const unsigned char *cmd, *fmt = "AT+DTSE=%s,%i"; + const char *tone = ""; + int error; + + if (!ddata->enabled) + return 0; + + ddata->dtmf_en = ucontrol->value.enumerated.item[0]; + if (ddata->dtmf_en) + tone = motmdm_tonegen_dtmf_key_txt[ddata->dtmf_val]; + + /* Value 0 enables tone generator, 1 disables it */ + cmd = kasprintf(GFP_KERNEL, fmt, tone, !ddata->dtmf_en); + + error = motmdm_send_command(ddata, cmd, strlen(cmd)); + if (error < 0) { + dev_err(component->dev, "%s: %s failed with %i\n", + __func__, cmd, error); + goto free; + } + +free: + kfree(cmd); + + return error; +} + +static int +motmdm_enable_primary_dai(struct snd_soc_component *component) +{ + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + int error; + + if (!ddata->master_dai) + return -ENODEV; + + error = snd_soc_dai_set_sysclk(ddata->master_dai, 1, 19200000, + SND_SOC_CLOCK_OUT); + if (error) + return error; + + error = snd_soc_dai_set_fmt(ddata->master_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (error) + return error; + + error = snd_soc_dai_set_tdm_slot(ddata->master_dai, 0, 1, 1, 8); + if (error) + return error; + + return error; +} + +static int +motmdm_disable_primary_dai(struct snd_soc_component *component) +{ + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + int error; + + if (!ddata->master_dai) { + return -ENODEV; + } + + error = snd_soc_dai_set_sysclk(ddata->master_dai, 0, 26000000, + SND_SOC_CLOCK_OUT); + if (error) { + return error; + } + + error = snd_soc_dai_set_fmt(ddata->master_dai, + SND_SOC_DAIFMT_CBM_CFM); + if (error) { + return error; + } + + error = snd_soc_dai_set_tdm_slot(ddata->master_dai, 0, 0, 0, 48); + if (error) { + return error; + } + + return error; +} + +static int motmdm_find_primary_dai(struct snd_soc_component *component, + const char *name) +{ + struct motmdm_driver_data *ddata = + snd_soc_component_get_drvdata(component); + struct device_node *bitclkmaster = NULL, *framemaster = NULL; + struct device_node *ep, *master_ep, *master = NULL; + struct snd_soc_dai_link_component dlc = { 0 }; + unsigned int daifmt; + + ep = of_graph_get_next_endpoint(component->dev->of_node, NULL); + if (!ep) + return -ENODEV; + + master_ep = of_graph_get_remote_endpoint(ep); + of_node_put(ep); + if (!master_ep) + return -ENODEV; + + daifmt = snd_soc_of_parse_daifmt(master_ep, NULL, + &bitclkmaster, &framemaster); + of_node_put(master_ep); + if (bitclkmaster && framemaster) + master = of_graph_get_port_parent(bitclkmaster); + of_node_put(bitclkmaster); + of_node_put(framemaster); + if (!master) + return -ENODEV; + + dlc.of_node = master; + dlc.dai_name = name; + ddata->master_dai = snd_soc_find_dai(&dlc); + of_node_put(master); + if (!ddata->master_dai) + return -EPROBE_DEFER; + + dev_info(component->dev, "Master DAI is %s\n", + dev_name(ddata->master_dai->dev)); + + return 0; +} + +static int motmdm_parse_tdm(struct snd_soc_component *component) +{ + return motmdm_find_primary_dai(component, "cpcap-voice"); +} + +static const struct snd_kcontrol_new motmdm_snd_controls[] = { + SOC_ENUM_EXT("Call Output", motmdm_out_enum, + motmdm_audio_out_get, + motmdm_audio_out_put), + SOC_SINGLE_EXT_TLV("Call Volume", + 0, 0, 7, 0, + motmdm_gain_get, + motmdm_gain_put, + motmdm_gain_tlv), + SOC_SINGLE_BOOL_EXT("Call Noise Cancellation", 0, + motmdm_noise_get, + motmdm_noise_put), + SOC_ENUM_EXT("Call DTMF", motmd_tonegen_dtmf_enum, + motmdm_dtmf_get, + motmdm_dtmf_put), + SOC_SINGLE_BOOL_EXT("Call DTMF Send", 0, + motmdm_tonegen_dtmf_send_get, + motmdm_tonegen_dtmf_send_put), +}; + +static struct snd_soc_dai_driver motmdm_dai[] = { + { + .name = "mdm-call", + .playback = { + .stream_name = "Voice Call Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Voice Call Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, +}; + +static int +motmdm_notifier_call(struct notifier_block *bl, unsigned long state, + void *unused) +{ + struct motmdm_driver_data *ddata = + container_of(bl, struct motmdm_driver_data, notifier); + bool enable, notify = false; + unsigned long flags; + + enable = !!state; + + spin_lock_irqsave(&ddata->lock, flags); + if (ddata->enabled != enable) { + ddata->enabled = enable; + notify = true; + } + spin_unlock_irqrestore(&ddata->lock, flags); + + if (!notify) + return NOTIFY_DONE; + + if (enable) + motmdm_enable_primary_dai(ddata->component); + else + motmdm_disable_primary_dai(ddata->component); + + return NOTIFY_DONE; +} + +static int voice_receive_data(struct serdev_device *serdev, + const unsigned char *buf, + size_t len) +{ + struct motmdm_driver_data *ddata = serdev_device_get_drvdata(serdev); + struct device *dev = ddata->component->dev; + + printk("voice_receive_data: have %s %d\n", buf, len); + + if (len > MOTMDM_AUDIO_MAX_LEN) + len = MOTMDM_AUDIO_MAX_LEN; + + if (len <= MOTMDM_HEADER_LEN) + return 0; + + printk("voice_receive_data: command reply? -- %s %d\n", buf, len); + + snprintf(ddata->buf, len - MOTMDM_HEADER_LEN, buf + MOTMDM_HEADER_LEN); + dev_info(dev, "%s: received: %s\n", __func__, ddata->buf); + ddata->parsed = true; + wake_up(&ddata->read_queue); + + return len; +} + +static const struct serdev_device_ops voice_serdev_ops = { + .receive_buf = voice_receive_data, + .write_wakeup = serdev_device_write_wakeup, +}; + +static void motmdm_free_voice_serdev(struct motmdm_driver_data *ddata) +{ + serdev_device_close(ddata->serdev); +} + +static int motmdm_soc_probe(struct snd_soc_component *component) +{ + struct motmdm_driver_data *ddata; + const unsigned char *cmd = "AT+CMUT=0"; + int error; + u32 line; + + ddata = devm_kzalloc(component->dev, sizeof(*ddata), GFP_KERNEL); + if (!ddata) + return -ENOMEM; + + error = of_property_read_u32(component->dev->of_node, "reg", &line); + if (error) + return error; + + ddata->serdev = (struct serdev_device *) component->dev; + ddata->component = component; + ddata->modem = component->dev->parent; + mutex_init(&ddata->mutex); + init_waitqueue_head(&ddata->read_queue); + ddata->len = PAGE_SIZE; + spin_lock_init(&ddata->lock); + snd_soc_component_set_drvdata(component, ddata); + ddata->len = MOTMDM_AUDIO_MAX_LEN; + + ddata->buf = devm_kzalloc(component->dev, ddata->len, GFP_KERNEL); + if (!ddata->buf) + return -ENOMEM; + + ddata->regmap = devm_regmap_init(component->dev, NULL, component, + &motmdm_regmap); + if (IS_ERR(ddata->regmap)) { + error = PTR_ERR(ddata->regmap); + dev_err(component->dev, "%s: Failed to allocate regmap: %d\n", + __func__, error); + + return error; + } + + serdev_device_set_drvdata(ddata->serdev, ddata); + serdev_device_set_client_ops(ddata->serdev, &voice_serdev_ops); + + error = serdev_device_open(ddata->serdev); + if (error) + return error; + + error = motmdm_parse_tdm(component); + if (error) + goto unregister_serdev; + + regcache_sync(ddata->regmap); + + error = motmdm_send_command(ddata, cmd, strlen(cmd)); + if (error < 0) + goto unregister_serdev; + + error = motmdm_disable_primary_dai(ddata->component); + if (error) + goto unregister_serdev; + + ddata->notifier.notifier_call = motmdm_notifier_call; + register_modem_state_notifier(&ddata->notifier); + + return 0; + +unregister_serdev: + motmdm_free_voice_serdev(ddata); + serdev_device_close(ddata->serdev); + + return error; +} + +static void motmdm_soc_remove(struct snd_soc_component *component) +{ + struct motmdm_driver_data *ddata = snd_soc_component_get_drvdata(component); + + unregister_modem_state_notifier(&ddata->notifier); + + motmdm_free_voice_serdev(ddata); +} + +static struct snd_soc_component_driver soc_codec_dev_motmdm = { + .probe = motmdm_soc_probe, + .remove = motmdm_soc_remove, + .controls = motmdm_snd_controls, + .num_controls = ARRAY_SIZE(motmdm_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int motmdm_codec_probe(struct serdev_device *serdev) +{ + return devm_snd_soc_register_component(&serdev->dev, + &soc_codec_dev_motmdm, + motmdm_dai, + ARRAY_SIZE(motmdm_dai)); +} + +#ifdef CONFIG_OF +static const struct of_device_id motmdm_of_match[] = { + { .compatible = "motorola,mapphone-mdm6600-codec" }, + {}, +}; +MODULE_DEVICE_TABLE(of, motmdm_of_match); +#endif + +static struct serdev_device_driver motmdm_driver = { + .probe = motmdm_codec_probe, + .driver = { + .name = "mot-mdm6600-codec", + .of_match_table = of_match_ptr(motmdm_of_match), + }, +}; +module_serdev_device_driver(motmdm_driver); + +MODULE_ALIAS("platform:motmdm-codec"); +MODULE_DESCRIPTION("ASoC Motorola Mapphone MDM6600 codec driver"); +MODULE_AUTHOR("Tony Lindgren <tony(a)atomide.com>"); +MODULE_LICENSE("GPL v2");

2 1

[PATCH v2] ASoC: qcom: lpass-cpu: Remove bit clock state check
by Srinivasa Rao Mandadapu 28 Jan '21

28 Jan '21

No need of BCLK state maintenance from driver side as clock_enable and clk_disable API's maintaing state counter. One of the major issue was spotted when Headset jack inserted while playback continues, due to same PCM device node opens twice for playaback/capture and closes once for capture and playback continues. It can resolve the errors in such scenarios. Fixes: b1824968221c ("ASoC: qcom: Fix enabling BCLK and LRCLK in LPAIF invalid state") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v1: -- Commit message changed sound/soc/qcom/lpass-cpu.c | 22 ++++++++-------------- sound/soc/qcom/lpass-lpaif-reg.h | 3 --- sound/soc/qcom/lpass.h | 1 - 3 files changed, 8 insertions(+), 18 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index ae8efbc89af2..a669202e0001 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -286,16 +286,12 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - if (drvdata->bit_clk_state[id] == LPAIF_BIT_CLK_DISABLE) { - ret = clk_enable(drvdata->mi2s_bit_clk[id]); - if (ret) { - dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); - clk_disable(drvdata->mi2s_osr_clk[id]); - return ret; - } - drvdata->bit_clk_state[id] = LPAIF_BIT_CLK_ENABLE; + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; } - break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -310,10 +306,9 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - if (drvdata->bit_clk_state[id] == LPAIF_BIT_CLK_ENABLE) { - clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); - drvdata->bit_clk_state[id] = LPAIF_BIT_CLK_DISABLE; - } + + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + break; } @@ -861,7 +856,6 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } - drvdata->bit_clk_state[dai_id] = LPAIF_BIT_CLK_DISABLE; } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 405542832e99..c8e1d75340b2 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -60,9 +60,6 @@ #define LPAIF_I2SCTL_BITWIDTH_24 1 #define LPAIF_I2SCTL_BITWIDTH_32 2 -#define LPAIF_BIT_CLK_DISABLE 0 -#define LPAIF_BIT_CLK_ENABLE 1 - #define LPAIF_I2SCTL_RESET_STATE 0x003C0004 #define LPAIF_DMACTL_RESET_STATE 0x00200000 diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 2d68af0da34d..83b2e08ade06 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -68,7 +68,6 @@ struct lpass_data { unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; int hdmi_port_enable; - int bit_clk_state[LPASS_MAX_MI2S_PORTS]; /* low-power audio interface (LPAIF) registers */ void __iomem *lpaif; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[PATCH] ASoC: SOF: Intel: hda: cancel D0i3 work during runtime suspend
by Kai Vehmanen 28 Jan '21

28 Jan '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cancel the D0i3 work during runtime suspend as no streams are active at this point anyway. Fixes: 63e51fd33fef("ASoC: SOF: Intel: cnl: Implement feature to support DSP D0i3 in S0") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda-dsp.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 2b001151fe37..9e8b3e542548 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -799,11 +799,15 @@ int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) { + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_dsp_power_state target_state = { .state = SOF_DSP_PM_D3, }; int ret; + /* cancel any attempt for DSP D0I3 */ + cancel_delayed_work_sync(&hda->d0i3_work); + /* stop hda controller and power dsp off */ ret = hda_suspend(sdev, true); if (ret < 0) base-commit: e32df142359fb6f4d27977b7652549f0844aa62f -- 2.29.2

2 1

[PATCH 0/5] ASoC: SOF: use common code for DSP core pm
by Kai Vehmanen 28 Jan '21

28 Jan '21

Series to refactor the DSP core management: - move tracking of powered up DSP cores to common SOF code - add logic filter unnecessary power actions - modify existing implementations to use common code whenever DSP cores are powered, so the state in common code stays in sync Bard Liao (5): ASoC: SOF: Intel: hda: use snd_sof_dsp_core_power_up/down API ASoC: SOF: Intel: hda-loader: keep init cores alive ASoC: SOF: update dsp core power status in common APIs ASoC: SOF: Filter out unneeded core power up/downs ASoC: SOF: intel: hda-loader: use snd_sof_dsp_core_power_down/up APIs sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-loader.c | 9 +++++---- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/loader.c | 6 ------ sound/soc/sof/ops.h | 24 ++++++++++++++++++------ sound/soc/sof/pm.c | 1 - sound/soc/sof/topology.c | 8 -------- 7 files changed, 25 insertions(+), 27 deletions(-) base-commit: e32df142359fb6f4d27977b7652549f0844aa62f -- 2.29.2

2 6

[PATCH] ASoC: Intel: remove duplicate MODULE_LICENSE/DESCRIPTION tags
by Kai Vehmanen 28 Jan '21

28 Jan '21

The snd-soc-acpi-intel-match has duplicate module tags for all platforms separately. Remove all but one and save some storage space and cleanup modinfo output. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-cfl-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-skl-match.c | 3 --- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 3 --- 14 files changed, 42 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index 32f77e29c2ff..398cc771c835 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -85,6 +85,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index c348607b49a5..31e60df131c3 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -233,6 +233,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_baytrail_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cfl-match.c b/sound/soc/intel/common/soc-acpi-intel-cfl-match.c index 27b4b73d94d4..1733dfb23e79 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cfl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cfl-match.c @@ -18,6 +18,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cfl_sdw_machines[] = { {} }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cfl_sdw_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 2752dc955733..227424236fd5 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -196,6 +196,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cherrytrail_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index adddc91918df..2161b3b85b4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -309,6 +309,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[] = { {} }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_sdw_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index b80f032a8b76..ec77a57a07ba 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -63,6 +63,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[] = { {} }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_sdw_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c index 5c7a1214403d..6222708a98e7 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c @@ -20,6 +20,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_ehl_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 26cb3b16cdd3..6ceaab19ccb6 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -43,6 +43,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 35958553652e..fe343a95b5ff 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -53,6 +53,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { {} }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_broadwell_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index 9a529a785288..d38ff7d187c4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -185,6 +185,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_sdw_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 34f5fcad5701..52238db0bcb5 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -63,6 +63,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index a4fbe6707ca7..47dadc9d5d2a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -128,6 +128,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c index 26f9ce146523..961df8d6b5e4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-skl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c @@ -42,6 +42,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_skl_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 4933184df5f5..ee25451dfdd0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -411,6 +411,3 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_sdw_machines); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); base-commit: e32df142359fb6f4d27977b7652549f0844aa62f -- 2.29.2

2 1

[PATCH] drm/msm/dp: trigger unplug event in msm_dp_display_disable
by Judy Hsiao 28 Jan '21

28 Jan '21

1. Trigger the unplug event in msm_dp_display_disable() to shutdown audio properly. 2. Reset the completion before signal the disconnect event. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- drivers/gpu/drm/msm/dp/dp_display.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/msm/dp/dp_display.c b/drivers/gpu/drm/msm/dp/dp_display.c index 3bc7ed21de286..81f6794a25100 100644 --- a/drivers/gpu/drm/msm/dp/dp_display.c +++ b/drivers/gpu/drm/msm/dp/dp_display.c @@ -651,8 +651,8 @@ static int dp_hpd_unplug_handle(struct dp_display_private *dp, u32 data) dp_add_event(dp, EV_DISCONNECT_PENDING_TIMEOUT, 0, DP_TIMEOUT_5_SECOND); /* signal the disconnect event early to ensure proper teardown */ - dp_display_handle_plugged_change(g_dp_display, false); reinit_completion(&dp->audio_comp); + dp_display_handle_plugged_change(g_dp_display, false); dp_catalog_hpd_config_intr(dp->catalog, DP_DP_HPD_PLUG_INT_MASK | DP_DP_IRQ_HPD_INT_MASK, true); @@ -890,6 +890,9 @@ static int dp_display_disable(struct dp_display_private *dp, u32 data) /* wait only if audio was enabled */ if (dp_display->audio_enabled) { + /* signal the disconnect event */ + reinit_completion(&dp->audio_comp); + dp_display_handle_plugged_change(dp_display, false); if (!wait_for_completion_timeout(&dp->audio_comp, HZ * 5)) DRM_ERROR("audio comp timeout\n"); -- 2.29.2

2 1

[PATCH] ASoC: ti: Allocate dais dynamically for TDM and audio graph card
by Pavel Machek 28 Jan '21

28 Jan '21

From: Tony Lindgren <tony(a)atomide.com> We can have multiple connections on a single McBSP instance configured with audio graph card when using TDM (Time Division Multiplexing). Let's allow that by configuring dais dynamically. See Documentation/devicetree/bindings/sound/audio-graph-card.txt and Documentation/devicetree/bindings/graph.txt for more details for multiple endpoints. I've tested this with droid4 where cpcap pmic and modem voice are both both wired to mcbsp3. I've also tested this on droid4 both with and without the pending modem audio codec driver that is waiting for n_gsm serdev dependencies to clear. Cc: Aaro Koskinen <aaro.koskinen(a)iki.fi> Cc: Arthur D. <spinal.by(a)gmail.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Merlijn Wajer <merlijn(a)wizzup.org> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Sebastian Reichel <sre(a)kernel.org> Signed-off-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Pavel Machek <pavel(a)ucw.cz> --- sound/soc/ti/omap-mcbsp-priv.h | 2 ++ sound/soc/ti/omap-mcbsp.c | 76 +++++++++++++++++++++++++++++------------- 2 files changed, 55 insertions(+), 23 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..9464f5d35822 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -262,6 +262,8 @@ struct omap_mcbsp { struct omap_mcbsp_platform_data *pdata; struct omap_mcbsp_st_data *st_data; struct omap_mcbsp_reg_cfg cfg_regs; + struct snd_soc_dai_driver *dais; + int dai_count; struct snd_dmaengine_dai_dma_data dma_data[2]; unsigned int dma_req[2]; int dma_op_mode; diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 6025b30bbe77..189a6461b671 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -14,6 +14,7 @@ #include <linux/pm_runtime.h> #include <linux/of.h> #include <linux/of_device.h> +#include <linux/of_graph.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1299,23 +1300,53 @@ static int omap_mcbsp_remove(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_driver omap_mcbsp_dai = { - .probe = omap_mcbsp_probe, - .remove = omap_mcbsp_remove, - .playback = { - .channels_min = 1, - .channels_max = 16, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 16, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - }, - .ops = &mcbsp_dai_ops, -}; +static int omap_mcbsp_init_dais(struct omap_mcbsp *mcbsp) +{ + struct device_node *np = mcbsp->dev->of_node; + int i; + + if (np) + mcbsp->dai_count = of_graph_get_endpoint_count(np); + + if (!mcbsp->dai_count) + mcbsp->dai_count = 1; + + mcbsp->dais = devm_kcalloc(mcbsp->dev, mcbsp->dai_count, + sizeof(*mcbsp->dais), GFP_KERNEL); + if (!mcbsp->dais) + return -ENOMEM; + + for (i = 0; i < mcbsp->dai_count; i++) { + struct snd_soc_dai_driver *dai = &mcbsp->dais[i]; + + dai->name = devm_kasprintf(mcbsp->dev, GFP_KERNEL, "%s-dai%i", + dev_name(mcbsp->dev), i); + + if (i == 0) { + dai->probe = omap_mcbsp_probe; + dai->remove = omap_mcbsp_remove; + dai->ops = &mcbsp_dai_ops; + } + dai->playback.channels_min = 1; + dai->playback.channels_max = 16; + dai->playback.rates = OMAP_MCBSP_RATES; + if (mcbsp->pdata->reg_size == 2) + dai->playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + else + dai->playback.formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE; + dai->capture.channels_min = 1; + dai->capture.channels_max = 16; + dai->capture.rates = OMAP_MCBSP_RATES; + if (mcbsp->pdata->reg_size == 2) + dai->capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + else + dai->capture.formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE; + } + + return 0; +} static const struct snd_soc_component_driver omap_mcbsp_component = { .name = "omap-mcbsp", @@ -1404,18 +1435,17 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) mcbsp->dev = &pdev->dev; platform_set_drvdata(pdev, mcbsp); - ret = omap_mcbsp_init(pdev); + ret = omap_mcbsp_init_dais(mcbsp); if (ret) return ret; - if (mcbsp->pdata->reg_size == 2) { - omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; - omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; - } + ret = omap_mcbsp_init(pdev); + if (ret) + return ret; ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, - &omap_mcbsp_dai, 1); + mcbsp->dais, mcbsp->dai_count); if (ret) return ret; -- 2.11.0 -- http://www.livejournal.com/~pavelmachek

3 2

[PATCH 1/3] ASoC: SOF: allow soundwire use desc->default_fw_filename
by Kai Vehmanen 27 Jan '21

27 Jan '21

From: Libin Yang <libin.yang(a)intel.com> The old code always uses sof_fw_filename in struct snd_soc_acpi_mach as the firmware name. However, firmware name should depend on the platform instead of the machine. For example, different machines may use the same soundwire link topology, but they are using the different firmware. In this case, it's hard to determine in struct snd_soc_acpi_mach which firmware it should use. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7e703ce22fcd..d99a552eea53 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1169,7 +1169,10 @@ static int hda_sdw_machine_select(struct snd_sof_dev *sdev) mach->mach_params.links = mach->links; mach->mach_params.link_mask = mach->link_mask; mach->mach_params.platform = dev_name(sdev->dev); - pdata->fw_filename = mach->sof_fw_filename; + if (mach->sof_fw_filename) + pdata->fw_filename = mach->sof_fw_filename; + else + pdata->fw_filename = pdata->desc->default_fw_filename; pdata->tplg_filename = mach->sof_tplg_filename; } else { dev_info(sdev->dev, -- 2.29.2

3 6

[PATCH] ACPI: Test for ACPI_SUCCESS rather than !ACPI_FAILURE
by Bjorn Helgaas 27 Jan '21

27 Jan '21

From: Bjorn Helgaas <bhelgaas(a)google.com> The double negative makes it hard to read "if (!ACPI_FAILURE(status))". Replace it with "if (ACPI_SUCCESS(status))". Signed-off-by: Bjorn Helgaas <bhelgaas(a)google.com> --- This isn't really an ACPI patch, but I'm sending it to you, Rafael, since it seems easier to just apply these all at once. But I'd be happy to split them up into individual patches if you'd rather. drivers/gpu/drm/amd/amdgpu/amdgpu_bios.c | 4 ++-- drivers/gpu/drm/radeon/radeon_bios.c | 4 ++-- drivers/hwmon/acpi_power_meter.c | 4 ++-- drivers/platform/x86/asus-laptop.c | 6 +++--- drivers/spi/spi.c | 2 +- sound/pci/hda/hda_intel.c | 4 ++-- 6 files changed, 12 insertions(+), 12 deletions(-) diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu_bios.c b/drivers/gpu/drm/amd/amdgpu/amdgpu_bios.c index 6333cada1e09..055f600eeed8 100644 --- a/drivers/gpu/drm/amd/amdgpu/amdgpu_bios.c +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu_bios.c @@ -291,7 +291,7 @@ static bool amdgpu_atrm_get_bios(struct amdgpu_device *adev) continue; status = acpi_get_handle(dhandle, "ATRM", &atrm_handle); - if (!ACPI_FAILURE(status)) { + if (ACPI_SUCCESS(status)) { found = true; break; } @@ -304,7 +304,7 @@ static bool amdgpu_atrm_get_bios(struct amdgpu_device *adev) continue; status = acpi_get_handle(dhandle, "ATRM", &atrm_handle); - if (!ACPI_FAILURE(status)) { + if (ACPI_SUCCESS(status)) { found = true; break; } diff --git a/drivers/gpu/drm/radeon/radeon_bios.c b/drivers/gpu/drm/radeon/radeon_bios.c index bb29cf02974d..43bbbfd6ade8 100644 --- a/drivers/gpu/drm/radeon/radeon_bios.c +++ b/drivers/gpu/drm/radeon/radeon_bios.c @@ -205,7 +205,7 @@ static bool radeon_atrm_get_bios(struct radeon_device *rdev) continue; status = acpi_get_handle(dhandle, "ATRM", &atrm_handle); - if (!ACPI_FAILURE(status)) { + if (ACPI_SUCCESS(status)) { found = true; break; } @@ -218,7 +218,7 @@ static bool radeon_atrm_get_bios(struct radeon_device *rdev) continue; status = acpi_get_handle(dhandle, "ATRM", &atrm_handle); - if (!ACPI_FAILURE(status)) { + if (ACPI_SUCCESS(status)) { found = true; break; } diff --git a/drivers/hwmon/acpi_power_meter.c b/drivers/hwmon/acpi_power_meter.c index 848718ab7312..7d3ddcba34ce 100644 --- a/drivers/hwmon/acpi_power_meter.c +++ b/drivers/hwmon/acpi_power_meter.c @@ -161,7 +161,7 @@ static ssize_t set_avg_interval(struct device *dev, mutex_lock(&resource->lock); status = acpi_evaluate_integer(resource->acpi_dev->handle, "_PAI", &args, &data); - if (!ACPI_FAILURE(status)) + if (ACPI_SUCCESS(status)) resource->avg_interval = temp; mutex_unlock(&resource->lock); @@ -232,7 +232,7 @@ static ssize_t set_cap(struct device *dev, struct device_attribute *devattr, mutex_lock(&resource->lock); status = acpi_evaluate_integer(resource->acpi_dev->handle, "_SHL", &args, &data); - if (!ACPI_FAILURE(status)) + if (ACPI_SUCCESS(status)) resource->cap = temp; mutex_unlock(&resource->lock); diff --git a/drivers/platform/x86/asus-laptop.c b/drivers/platform/x86/asus-laptop.c index 0edafe687fa9..bfea656e910c 100644 --- a/drivers/platform/x86/asus-laptop.c +++ b/drivers/platform/x86/asus-laptop.c @@ -861,7 +861,7 @@ static ssize_t infos_show(struct device *dev, struct device_attribute *attr, * The significance of others is yet to be found. */ rv = acpi_evaluate_integer(asus->handle, "SFUN", NULL, &temp); - if (!ACPI_FAILURE(rv)) + if (ACPI_SUCCESS(rv)) len += sprintf(page + len, "SFUN value : %#x\n", (uint) temp); /* @@ -873,7 +873,7 @@ static ssize_t infos_show(struct device *dev, struct device_attribute *attr, * takes several seconds to run on some systems. */ rv = acpi_evaluate_integer(asus->handle, "HWRS", NULL, &temp); - if (!ACPI_FAILURE(rv)) + if (ACPI_SUCCESS(rv)) len += sprintf(page + len, "HWRS value : %#x\n", (uint) temp); /* @@ -884,7 +884,7 @@ static ssize_t infos_show(struct device *dev, struct device_attribute *attr, * silently ignored. */ rv = acpi_evaluate_integer(asus->handle, "ASYM", NULL, &temp); - if (!ACPI_FAILURE(rv)) + if (ACPI_SUCCESS(rv)) len += sprintf(page + len, "ASYM value : %#x\n", (uint) temp); if (asus->dsdt_info) { diff --git a/drivers/spi/spi.c b/drivers/spi/spi.c index 720ab34784c1..801d8b499788 100644 --- a/drivers/spi/spi.c +++ b/drivers/spi/spi.c @@ -2210,7 +2210,7 @@ static acpi_status acpi_register_spi_device(struct spi_controller *ctlr, return AE_OK; if (!lookup.max_speed_hz && - !ACPI_FAILURE(acpi_get_parent(adev->handle, &parent_handle)) && + ACPI_SUCCESS(acpi_get_parent(adev->handle, &parent_handle)) && ACPI_HANDLE(ctlr->dev.parent) == parent_handle) { /* Apple does not use _CRS but nested devices for SPI slaves */ acpi_spi_parse_apple_properties(adev, &lookup); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 770ad25f1907..fe8049cd2765 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1444,7 +1444,7 @@ static bool atpx_present(void) dhandle = ACPI_HANDLE(&pdev->dev); if (dhandle) { status = acpi_get_handle(dhandle, "ATPX", &atpx_handle); - if (!ACPI_FAILURE(status)) { + if (ACPI_SUCCESS(status)) { pci_dev_put(pdev); return true; } @@ -1454,7 +1454,7 @@ static bool atpx_present(void) dhandle = ACPI_HANDLE(&pdev->dev); if (dhandle) { status = acpi_get_handle(dhandle, "ATPX", &atpx_handle); - if (!ACPI_FAILURE(status)) { + if (ACPI_SUCCESS(status)) { pci_dev_put(pdev); return true; } -- 2.25.1

6 8

[PATCH] ASoC: siu: Fix build error by a wrong const prefix
by Takashi Iwai 27 Jan '21

27 Jan '21

A const prefix was put wrongly in the middle at the code refactoring commit 932eaf7c7904 ("ASoC: sh: siu_pcm: remove snd_pcm_ops"), which leads to a build error as: sound/soc/sh/siu_pcm.c:546:8: error: expected '{' before 'const' Also, another inconsistency is that the declaration of siu_component misses the const prefix. This patch corrects both failures. Fixes: 932eaf7c7904 ("ASoC: sh: siu_pcm: remove snd_pcm_ops") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Casually caught by a random build test after your latest PR. sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h index 6201840f1bc0..a675c36fc9d9 100644 --- a/sound/soc/sh/siu.h +++ b/sound/soc/sh/siu.h @@ -169,7 +169,7 @@ static inline u32 siu_read32(u32 __iomem *addr) #define SIU_BRGBSEL (0x108 / sizeof(u32)) #define SIU_BRRB (0x10c / sizeof(u32)) -extern struct snd_soc_component_driver siu_component; +extern const struct snd_soc_component_driver siu_component; extern struct siu_info *siu_i2s_data; int siu_init_port(int port, struct siu_port **port_info, struct snd_card *card); diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 45c4320976ab..4785886df4f0 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -543,7 +543,7 @@ static void siu_pcm_free(struct snd_soc_component *component, dev_dbg(pcm->card->dev, "%s\n", __func__); } -struct const snd_soc_component_driver siu_component = { +const struct snd_soc_component_driver siu_component = { .name = DRV_NAME, .open = siu_pcm_open, .close = siu_pcm_close, -- 2.26.2

2 1

[PATCH v2 1/2] ASoC: soc-component: add snd_soc_component_read/write_field()
by Srinivas Kandagatla 27 Jan '21

27 Jan '21

It's often the case that we would write or read a particular field in register. With the current soc_component apis, reading a particular field in register would involve first read the register and then perform shift operations. Ex: to read from a field mask of 0xf0 val = snd_soc_component_read(component, reg); field = ((val & 0xf0) >> 0x4); This is sometimes prone to errors and code become less readable! With this new api we could just do field = snd_soc_component_read_field(component, reg, 0xf0); this makes it bit simple, easy to write and less error prone! This also applies to writing! There are various places in kernel which provides such field interfaces however soc_component seems to be missing this. This patch is inspired by FIELD_GET/FIELD_PREP macros in include/linux/bitfield.h Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - removed open coding and call read and update_bits - made soc_component_field_shift an inline function include/sound/soc-component.h | 6 ++++ sound/soc/soc-component.c | 53 +++++++++++++++++++++++++++++++++++ 2 files changed, 59 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 0bce41fefd30..5b47768222b7 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -353,6 +353,12 @@ int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +unsigned int snd_soc_component_read_field(struct snd_soc_component *component, + unsigned int reg, unsigned int mask); +int snd_soc_component_write_field(struct snd_soc_component *component, + unsigned int reg, unsigned int mask, + unsigned int val); + /* component wide operations */ int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, int source, diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 760523382f3c..361a79d655e3 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -34,6 +34,18 @@ static inline int _soc_component_ret(struct snd_soc_component *component, return ret; } +static inline int soc_component_field_shift(struct snd_soc_component *component, + unsigned int mask) +{ + if (!mask) { + dev_err(component->dev, "ASoC: error field mask is zero for %s\n", + component->name); + return 0; + } + + return (__builtin_ffs(mask) - 1); +} + /* * We might want to check substream by using list. * In such case, we can update these macros. @@ -839,6 +851,47 @@ int snd_soc_component_update_bits_async(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_component_update_bits_async); +/** + * snd_soc_component_read_field() - Read register field value + * @component: Component to read from + * @reg: Register to read + * @mask: mask of the register field + * + * Return: read value of register field. + */ +unsigned int snd_soc_component_read_field(struct snd_soc_component *component, + unsigned int reg, unsigned int mask) +{ + unsigned int val; + + val = snd_soc_component_read(component, reg); + + val = (val & mask) >> soc_component_field_shift(component, mask); + + return val; +} +EXPORT_SYMBOL_GPL(snd_soc_component_read_field); + +/** + * snd_soc_component_write_field() - write to register field + * @component: Component to write to + * @reg: Register to write + * @mask: mask of the register field to update + * @val: value of the field to write + * + * Return: 1 for change, otherwise 0. + */ +int snd_soc_component_write_field(struct snd_soc_component *component, + unsigned int reg, unsigned int mask, + unsigned int val) +{ + + val = (val << soc_component_field_shift(component, mask)) & mask; + + return snd_soc_component_update_bits(component, reg, mask, val); +} +EXPORT_SYMBOL_GPL(snd_soc_component_write_field); + /** * snd_soc_component_async_complete() - Ensure asynchronous I/O has completed * @component: Component for which to wait -- 2.21.0

2 2

[PATCH] ASoC: q6asm: fix 'physical' typos
by Bjorn Helgaas 27 Jan '21

27 Jan '21

From: Bjorn Helgaas <bhelgaas(a)google.com> Fix misspellings of "physical". Signed-off-by: Bjorn Helgaas <bhelgaas(a)google.com> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index a6618efe22f2..36bf8bd4edd7 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -491,7 +491,7 @@ static int __q6asm_memory_map_regions(struct audio_client *ac, int dir, * * @dir: direction of audio stream * @ac: audio client instanace - * @phys: physcial address that needs mapping. + * @phys: physical address that needs mapping. * @period_sz: audio period size * @periods: number of periods * -- 2.25.1

3 2

[PATCH v2 0/2] ASoC: SOF: Intel: hda: Enable DMI L1 for power savings
by Ranjani Sridharan 27 Jan '21

27 Jan '21

This series includes 2 patches that enable DMI L1 for D0I3-compatible streams and trace DMA stream to maximise power savings on Intel HDA platforms. v2 changes: FIx compilation error when probes feature is enabled in SOF Kconfig. Ranjani Sridharan (2): ASoC: SOF: Intel: hda: enable DMI L1 for D0i3-compatible streams ASoC: SOF: Intel: hda: Enable DMI L1 for trace sound/soc/sof/intel/hda-compress.c | 2 +- sound/soc/sof/intel/hda-loader.c | 2 +- sound/soc/sof/intel/hda-pcm.c | 16 ++++++++++++- sound/soc/sof/intel/hda-stream.c | 38 +++++++++++++++++------------- sound/soc/sof/intel/hda-trace.c | 4 ++-- sound/soc/sof/intel/hda.h | 6 ++++- 6 files changed, 46 insertions(+), 22 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: SOF: add a pointer to download repo in case FW request fails
by Kai Vehmanen 27 Jan '21

27 Jan '21

The SOF firmware and topology files are not distributed via linux-firmware. To help debugging cases where correct firmware is not installed, print a pointer to the official upstream repository for Sound Open Firmware releases. BugLink: https://github.com/thesofproject/sof/issues/3665 Reported-by: Bruce Perens <bruce(a)perens.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Marc Herbert <marc.herbert(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)intel.com> --- sound/soc/sof/loader.c | 2 ++ sound/soc/sof/topology.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 08a17abb63ff..eabeaf7b260e 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -731,6 +731,8 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) if (ret < 0) { dev_err(sdev->dev, "error: request firmware %s failed err: %d\n", fw_filename, ret); + dev_err(sdev->dev, + "you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n"); goto err; } else { dev_dbg(sdev->dev, "request_firmware %s successful\n", diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index b6b32a7a91f8..59f89f3d61a2 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3740,6 +3740,8 @@ int snd_sof_load_topology(struct snd_soc_component *scomp, const char *file) if (ret < 0) { dev_err(scomp->dev, "error: tplg request firmware %s failed err: %d\n", file, ret); + dev_err(scomp->dev, + "you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n"); return ret; } -- 2.29.2

2 1

[PATCH] ASoC: max98373: Fixes a typo max98373_feedback_get
by Judy Hsiao 27 Jan '21

27 Jan '21

From: Judy Hsiao <judyhsiao(a)google.com> The snd_soc_put_volsw in max98373_feedback_get is a typo, change it to snd_soc_gut_volsw. Signed-off-by: Judy Hsiao <judyhsiao(a)google.com> --- sound/soc/codecs/max98373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 31d571d4fac1c..746c829312b87 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -190,7 +190,7 @@ static int max98373_feedback_get(struct snd_kcontrol *kcontrol, } } - return snd_soc_put_volsw(kcontrol, ucontrol); + return snd_soc_get_volsw(kcontrol, ucontrol); } static const struct snd_kcontrol_new max98373_snd_controls[] = { -- 2.29.2

2 1

[PATCH v2] ASoC: max98373: Fixes a typo in max98373_feedback_get
by Judy Hsiao 27 Jan '21

27 Jan '21

From: Judy Hsiao <judyhsiao(a)google.com> The snd_soc_put_volsw in max98373_feedback_get is a typo, change it to snd_soc_get_volsw. Fixes: 349dd23931d1 ("ASoC: max98373: don't access volatile registers in bias level off") Signed-off-by: Judy Hsiao <judyhsiao(a)google.com> --- sound/soc/codecs/max98373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 31d571d4fac1c..746c829312b87 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -190,7 +190,7 @@ static int max98373_feedback_get(struct snd_kcontrol *kcontrol, } } - return snd_soc_put_volsw(kcontrol, ucontrol); + return snd_soc_get_volsw(kcontrol, ucontrol); } static const struct snd_kcontrol_new max98373_snd_controls[] = { -- 2.29.2

2 1

Likely ALSA driver bug in snd_hda_intel per syslog
by GitHub issues - opened 27 Jan '21

27 Jan '21

alsa-project/alsa-utils issue #76 was opened from dhdurgee: [alsa-bug.log](https://github.com/alsa-project/alsa-utils/files/5881756/alsa… Issue URL : https://github.com/alsa-project/alsa-utils/issues/76 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[Feature request][alsactl] Allow -g/--ignore to be used with init command
by GitHub issues - opened 27 Jan '21

27 Jan '21

alsa-project/alsa-utils issue #75 was opened from MichaIng: It would be great if the `-g`/`--ignore` flag would have an effect on the `init` command as well, so that `alsactl -g init` does not print and exit with an error, if no sound cards have been found. Since it can be checked for attached sound cards on Linux easily via /proc or sysfs, this is of minor importance, and probably there is a reason why it is as it is. In this case, probably the `--help` output could be extended by this info, which is currently only present in the man page. Reading the `--help` only, one (or at least me) would not guess that this flag is ignored by the `init` command. Issue URL : https://github.com/alsa-project/alsa-utils/issues/75 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: qcom: lpass-cpu: Remove bit clock state check
by Srinivasa Rao Mandadapu 27 Jan '21

27 Jan '21

No need of BCLK state maintenance from driver side as clock_enable and clk_disable API's maintaing state counter. One of the major issue was spotted when Headset jack inserted while playback continues, due to same PCM device node opens twice for playaback/capture and closes once for capture and playback continues. It can resolve the errors in such scenarios. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 22 ++++++++-------------- sound/soc/qcom/lpass-lpaif-reg.h | 3 --- sound/soc/qcom/lpass.h | 1 - 3 files changed, 8 insertions(+), 18 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index ae8efbc89af2..a669202e0001 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -286,16 +286,12 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - if (drvdata->bit_clk_state[id] == LPAIF_BIT_CLK_DISABLE) { - ret = clk_enable(drvdata->mi2s_bit_clk[id]); - if (ret) { - dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); - clk_disable(drvdata->mi2s_osr_clk[id]); - return ret; - } - drvdata->bit_clk_state[id] = LPAIF_BIT_CLK_ENABLE; + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; } - break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -310,10 +306,9 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - if (drvdata->bit_clk_state[id] == LPAIF_BIT_CLK_ENABLE) { - clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); - drvdata->bit_clk_state[id] = LPAIF_BIT_CLK_DISABLE; - } + + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + break; } @@ -861,7 +856,6 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } - drvdata->bit_clk_state[dai_id] = LPAIF_BIT_CLK_DISABLE; } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 405542832e99..c8e1d75340b2 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -60,9 +60,6 @@ #define LPAIF_I2SCTL_BITWIDTH_24 1 #define LPAIF_I2SCTL_BITWIDTH_32 2 -#define LPAIF_BIT_CLK_DISABLE 0 -#define LPAIF_BIT_CLK_ENABLE 1 - #define LPAIF_I2SCTL_RESET_STATE 0x003C0004 #define LPAIF_DMACTL_RESET_STATE 0x00200000 diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 2d68af0da34d..83b2e08ade06 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -68,7 +68,6 @@ struct lpass_data { unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; int hdmi_port_enable; - int bit_clk_state[LPASS_MAX_MI2S_PORTS]; /* low-power audio interface (LPAIF) registers */ void __iomem *lpaif; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH 0/2] ASoC: SOF: Intel: hda: Enable DMI L1 for power savings
by Ranjani Sridharan 27 Jan '21

27 Jan '21

This series includes 2 patches that enable DMI L1 for D0I3-compatible streams and trace DMA stream to maximise power savings on Intel HDA platforms. Ranjani Sridharan (2): ASoC: SOF: Intel: hda: enable DMI L1 for D0i3-compatible streams ASoC: SOF: Intel: hda: Enable DMI L1 for trace sound/soc/sof/intel/hda-compress.c | 2 +- sound/soc/sof/intel/hda-loader.c | 2 +- sound/soc/sof/intel/hda-pcm.c | 16 ++++++++++++- sound/soc/sof/intel/hda-stream.c | 38 +++++++++++++++++------------- sound/soc/sof/intel/hda-trace.c | 4 ++-- sound/soc/sof/intel/hda.h | 6 ++++- 6 files changed, 46 insertions(+), 22 deletions(-) -- 2.25.1

1 3

[PATCH] ALSA: control: expand limitation on the number of user-defined control element set per card
by Takashi Sakamoto 26 Jan '21

26 Jan '21

ALSA control core allows usespace application to register control element set by call of ioctl(2) with SNDRV_CTL_IOCTL_ELEM_ADD request. The added control elements are called as 'user-defined'. Currently sound card has limitation on the number of the user-defined control element set up to 32. The limitation is inconvenient to drivers in ALSA firewire stack since the drivers expect userspace applications to implement function to control device functionalities such as mixing and routing. As the userspace application, snd-firewire-ctl-services project starts: https://github.com/alsa-project/snd-firewire-ctl-services/ The project supports many devices supported by ALSA firewire stack. The limitation is mostly good since routing and mixing controls can be represented by control element set, which includes multiple control element with the same parameters. Nevertheless, it's actually inconvenient to device which has many varied functionalities. For example, plugin effect such as channel strip and reverb has many parameters. For the case, many control elements are required to configure the parameters and control element set cannot aggregates them for the parameters. At present, the implementations for below models requires more control element sets than 32: * snd-bebob-ctl-service * Apogee Ensemble (31 sets for 34 elements) * snd-dice-ctl-service * TC Electronic Konnekt 24d (78 sets for 94 elements) * TC Electronic Studio Konnekt 48 (98 sets for 114 elements) * TC Electronic Konnekt Live (88 sets for 104 elements) * TC Electronic Impact Twin (70 sets for 86 elements) * Focusrite Liquid Saffire 56 (37 sets for 52 elements) This commit expands the limitation according to requirement from the above applications. As a result, userspace applications can add control element sets up to 150 per sound card. It results in 154,200 user-defined control elements as maximum since one control element set can include 1028 control elements. The new limitation is decided without comprehensive criteria to sound card. It could be changed according to requirement from the other type of userspace applications. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 5165741a8400..5a19bde27830 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -18,8 +18,13 @@ #include <sound/info.h> #include <sound/control.h> -/* max number of user-defined controls */ -#define MAX_USER_CONTROLS 32 +// The maximum number of control element sets per sound card added by +// userspace applications. The value is decided just to satisfy requirement +// from control service programs in userspace for devices supported by +// drivers in ALSA firewire stack. It's possible to relax the limitation +// according to requirements from the other kind of applications. +#define MAX_USER_CONTROLS 150 + #define MAX_CONTROL_COUNT 1028 struct snd_kctl_ioctl { -- 2.27.0

3 12

[PATCH] ALSA: hda/via: Apply the workaround generically for Clevo machines
by Takashi Iwai 26 Jan '21

26 Jan '21

We've got another report indicating a similar problem wrt the power-saving behavior with VIA codec on Clevo machines. Let's apply the existing workaround generically to all Clevo devices with VIA codecs to cover all in once. BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1181330 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 834367dd54e1..a5c1a2c4eae4 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1043,7 +1043,7 @@ static const struct hda_fixup via_fixups[] = { static const struct snd_pci_quirk vt2002p_fixups[] = { SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), - SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", VIA_FIXUP_POWER_SAVE), + SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", VIA_FIXUP_POWER_SAVE), {} }; -- 2.26.2

1 0

[RFC PATCH 1/2] ASoC: soc-component: add snd_soc_component_read/write_field()
by Srinivas Kandagatla 26 Jan '21

26 Jan '21

It's often the case that we would write or read a particular field in register. With the current soc_component apis, reading a particular field in register would involve first read the register and then perform shift operations. Ex: to read from a field mask of 0xf0 val = snd_soc_component_read(component, reg); field = ((val & 0xf0) >> 0x4); This is sometimes prone to errors and code become less readable! With this new api we could just do field = snd_soc_component_read_field(component, reg, 0xf0); this makes it bit simple, easy to write and less error prone! This also applies to writing! There are various places in kernel which provides such field interfaces however soc_component seems to be missing this. This patch is inspired by FIELD_GET/FIELD_PREP macros in include/linux/bitfield.h Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- include/sound/soc-component.h | 6 ++++ sound/soc/soc-component.c | 62 +++++++++++++++++++++++++++++++++++ 2 files changed, 68 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 0bce41fefd30..5b47768222b7 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -353,6 +353,12 @@ int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +unsigned int snd_soc_component_read_field(struct snd_soc_component *component, + unsigned int reg, unsigned int mask); +int snd_soc_component_write_field(struct snd_soc_component *component, + unsigned int reg, unsigned int mask, + unsigned int val); + /* component wide operations */ int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, int source, diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 760523382f3c..6bdfc6f61ed6 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -12,6 +12,7 @@ #include <linux/pm_runtime.h> #include <sound/soc.h> +#define __soc_component_field_shift(x) (__builtin_ffs(x) - 1) #define soc_component_ret(dai, ret) _soc_component_ret(dai, __func__, ret) static inline int _soc_component_ret(struct snd_soc_component *component, const char *func, int ret) @@ -839,6 +840,67 @@ int snd_soc_component_update_bits_async(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_component_update_bits_async); +/** + * snd_soc_component_read_field() - Read register field value + * @component: Component to read from + * @reg: Register to read + * @mask: mask of the register field + * + * Return: read value of register field. + */ +unsigned int snd_soc_component_read_field(struct snd_soc_component *component, + unsigned int reg, unsigned int mask) +{ + unsigned int val; + + mutex_lock(&component->io_mutex); + val = soc_component_read_no_lock(component, reg); + if (mask) + val = (val & mask) >> __soc_component_field_shift(mask); + mutex_unlock(&component->io_mutex); + + return val; +} +EXPORT_SYMBOL_GPL(snd_soc_component_read_field); + +/** + * snd_soc_component_write_field() - write to register field + * @component: Component to write to + * @reg: Register to write + * @mask: mask of the register field to update + * @val: value of the field to write + * + * Return: 1 for change, otherwise 0. + */ +int snd_soc_component_write_field(struct snd_soc_component *component, + unsigned int reg, unsigned int mask, + unsigned int val) +{ + unsigned int old, new; + int ret = 0; + bool change; + + if (!mask) + return false; + + mutex_lock(&component->io_mutex); + + old = soc_component_read_no_lock(component, reg); + + val = val << __soc_component_field_shift(mask); + + new = (old & ~mask) | (val & mask); + + change = old != new; + if (change) + ret = soc_component_write_no_lock(component, reg, new); + + mutex_unlock(&component->io_mutex); + + return change; +} +EXPORT_SYMBOL_GPL(snd_soc_component_write_field); + /** * snd_soc_component_async_complete() - Ensure asynchronous I/O has completed * @component: Component for which to wait -- 2.21.0

2 4

[GIT PULL] ASoC fixes for v5.11-rc5
by Mark Brown 26 Jan '21

26 Jan '21

The following changes since commit 7505c06dabb5e814bda610c8d83338544f15db45: MAINTAINERS: update qcom ASoC drivers list (2021-01-15 17:25:35 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.11-rc5 for you to fetch changes up to 5413dfd8ce0d5d997d442440701814a8ce7025d9: Merge series "ASoC: SOF: partial fix to Kconfig issues" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: (2021-01-25 14:15:12 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.11 More fixes for v5.11, almost all driver specific issues including new device IDs - there's one error handling fix for the topology stuff too. ---------------------------------------------------------------- Amadeusz Sławiński (1): ASoC: topology: Properly unregister DAI on removal Dan Carpenter (1): ASoC: topology: Fix memory corruption in soc_tplg_denum_create_values() Eliot Blennerhassett (1): ASoC: ak4458: correct reset polarity James Schulman (1): ASoC: wm_adsp: Fix control name parsing for multi-fw Jaroslav Kysela (1): ASoC: AMD Renoir - refine DMI entries for some Lenovo products Jonathan Neuschäfer (1): ASoC: dt-bindings: mt8192-mt6359: Fix indentation Libin Yang (1): ASoC: Intel: sof_sdw: set proper flags for Dell TGL-H SKU 0A5E Mark Brown (4): Merge series "Add KUNIT tests for ASoC topology" from Amadeusz Sławiński<amadeuszx.slawinski(a)linux.intel.com>: Merge series "ASoC: mediatek: mt8183: ignore TDM DAI link by by default" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: qcom: Fix broken lpass driver" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: SOF: partial fix to Kconfig issues" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Pierre-Louis Bossart (2): ASoC: SOF: Intel: soundwire: fix select/depend unmet dependencies ASoC: SOF: SND_INTEL_DSP_CONFIG dependency Ricardo Ribalda (2): ASoC: Intel: Skylake: skl-topology: Fix OOPs ib skl_tplg_complete ASoC: Intel: Skylake: Zero snd_ctl_elem_value Srinivas Kandagatla (3): ASoC: dt-bindings: lpass: Fix and common up lpass dai ids ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY ASoC: qcom: lpass-ipq806x: fix bitwidth regmap field Stephan Gerhold (1): ASoC: qcom: lpass: Fix out-of-bounds DAI ID lookup Stephen Boyd (1): ASoC: qcom: Fix number of HDMI RDMA channels on sc7180 Tzung-Bi Shih (3): ASoC: mediatek: mt8183-mt6358: ignore TDM DAI link by default ASoC: mediatek: mt8183-da7219: ignore TDM DAI link by default ASoC: mediatek: mt8192-mt6359: add format constraints for RT5682 .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 +- include/dt-bindings/sound/apq8016-lpass.h | 7 ++-- include/dt-bindings/sound/qcom,lpass.h | 15 +++++++ include/dt-bindings/sound/sc7180-lpass.h | 6 +-- sound/soc/amd/renoir/rn-pci-acp3x.c | 18 +++++++- sound/soc/codecs/ak4458.c | 22 ++++------ sound/soc/codecs/wm_adsp.c | 3 ++ sound/soc/intel/boards/sof_sdw.c | 10 +++++ sound/soc/intel/skylake/skl-topology.c | 15 +++---- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 5 ++- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 ++- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 49 ++++++++++++++++++++++ sound/soc/qcom/lpass-cpu.c | 22 ++++++++++ sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-lpaif-reg.h | 2 +- sound/soc/qcom/lpass-platform.c | 12 ++++++ sound/soc/qcom/lpass-sc7180.c | 11 ++--- sound/soc/qcom/lpass.h | 2 +- sound/soc/soc-topology.c | 11 ++--- sound/soc/sof/intel/Kconfig | 3 +- sound/soc/sof/sof-acpi-dev.c | 11 ++--- sound/soc/sof/sof-pci-dev.c | 10 +++-- 22 files changed, 184 insertions(+), 61 deletions(-) create mode 100644 include/dt-bindings/sound/qcom,lpass.h

2 1

[PATCH v1 1/2] dt-bindings: pinctrl: pinctrl-microchip-sgpio: Fix dt_binding_check warning
by Robert Foss 26 Jan '21

26 Jan '21

Silence indentation level warning reported by dt_binding_check in order to reduce noise during routine checks. $ make dt_binding_check mt8192-mt6359-rt1015-rt5682.yaml:10:4: [warning] wrong indentation: expected 2 but found 3 (indentation) Signed-off-by: Robert Foss <robert.foss(a)linaro.org> --- .../devicetree/bindings/pinctrl/microchip,sparx5-sgpio.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/pinctrl/microchip,sparx5-sgpio.yaml b/Documentation/devicetree/bindings/pinctrl/microchip,sparx5-sgpio.yaml index df0c83cb1c6e..4fe35e650909 100644 --- a/Documentation/devicetree/bindings/pinctrl/microchip,sparx5-sgpio.yaml +++ b/Documentation/devicetree/bindings/pinctrl/microchip,sparx5-sgpio.yaml @@ -99,8 +99,8 @@ patternProperties: '#interrupt-cells': description: - Specifies the pin (port and bit) and flags, as defined in - defined in include/dt-bindings/interrupt-controller/irq.h + Specifies the pin (port and bit) and flags, as defined in + defined in include/dt-bindings/interrupt-controller/irq.h const: 3 ngpios: -- 2.27.0

2 3

[PATCH v8 0/1] audio jack software injection
by Hui Wang 26 Jan '21

26 Jan '21

the changes in the v8: - change "for (i = 0; i < strlen(tname); i++)" to "for (i = 0; tname[i]; i++)" - use CONFIG_SND_DEBUG to wrap some debugfs variable and struct members. the changes in the v7: - change the format of the last part in jack-injection.rst - add dependence SND_DEBUG in the Kconfig - create debugfs_mount_dir/sound and debugfs_mount_dir/sound/cardN only SND_DEBUG is enabled - change simple_write_to_buffer(buf, count, ppos, from, count) to simple_write_to_buffer(buf, sizeof(buf) - 1, ppos, from, count) - rewrite the function parse_mask_bits() according to v6's comment the changes in the v6: - use the sizeof(buf) to replace the digits in the scnprintf - squash the document patch into the 0001-xxx.patch. the changes in the v5: - add a Kconfig to enable/disable the jack injection feature - replace all kzalloc with local char[] in the debugfs fops - replace the chars of !isalnum() to '_' for the jack folder's name - squash all .c files patches into one patch - add a document to explain jack injection, for easier review, put the document in a standalone patch. If needed, will squash this patch to the previous one. the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (1): alsa: jack: implement software jack injection via debugfs Documentation/sound/designs/index.rst | 1 + .../sound/designs/jack-injection.rst | 166 ++++++++++ include/sound/core.h | 6 + include/sound/jack.h | 1 + sound/core/Kconfig | 9 + sound/core/init.c | 16 + sound/core/jack.c | 301 +++++++++++++++++- sound/core/sound.c | 13 + 8 files changed, 509 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/designs/jack-injection.rst -- 2.25.1

2 3

[PATCH v7 0/1] audio jack software injection
by Hui Wang 26 Jan '21

26 Jan '21

the changes in the v7: - change the format of the last part in jack-injection.rst - add dependence SND_DEBUG in the Kconfig - create debugfs_mount_dir/sound and debugfs_mount_dir/sound/cardN only SND_DEBUG is enabled - change simple_write_to_buffer(buf, count, ppos, from, count) to simple_write_to_buffer(buf, sizeof(buf) - 1, ppos, from, count) - rewrite the function parse_mask_bits() according to v6's comment the changes in the v6: - use the sizeof(buf) to replace the digits in the scnprintf - squash the document patch into the 0001-xxx.patch. the changes in the v5: - add a Kconfig to enable/disable the jack injection feature - replace all kzalloc with local char[] in the debugfs fops - replace the chars of !isalnum() to '_' for the jack folder's name - squash all .c files patches into one patch - add a document to explain jack injection, for easier review, put the document in a standalone patch. If needed, will squash this patch to the previous one. the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (1): alsa: jack: implement software jack injection via debugfs Documentation/sound/designs/index.rst | 1 + .../sound/designs/jack-injection.rst | 166 ++++++++++ include/sound/core.h | 2 + include/sound/jack.h | 1 + sound/core/Kconfig | 9 + sound/core/init.c | 16 + sound/core/jack.c | 299 +++++++++++++++++- sound/core/sound.c | 11 + 8 files changed, 501 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/designs/jack-injection.rst -- 2.25.1

2 5

[PATCH v2 0/3] isa: Make the remove callback for isa drivers return void
by Uwe Kleine-König 26 Jan '21

26 Jan '21

Hello, changes since v1 (https://lore.kernel.org/r/20210121204812.402589-1-uwe@kleine-koenig.org): - fix build failure, found by the kernel test robot (my grep missed the struct isa_driver embedded in struct radio_isa_driver). For this I introduced patch 2 and adapted patch 3 a bit. An interdiff is below. - Added Acks/Reviews by Guenter Roeck, William Breathitt Gray, Marc Kleine-Budde, Wolfram Sang and Takashi Iwai <tiwai(a)suse.de> The base-commit I specified (by hand *sigh*) in v1 was broken, must be a cut-n-paste-error, sorry for that. Takashi suggested to take this series via sound.git given that this is the most affected tree. This is fine for me. Otherwise I can also provide an immutable branch. For both variants we still need a few acks though. Best regards Uwe Uwe Kleine-König (3): watchdog: pcwd: drop always-false if from remove callback media/radio: Make radio_isa_common_remove() return void isa: Make the remove callback for isa drivers return void drivers/base/isa.c | 2 +- drivers/i2c/busses/i2c-elektor.c | 4 +--- drivers/i2c/busses/i2c-pca-isa.c | 4 +--- drivers/input/touchscreen/htcpen.c | 4 +--- drivers/media/radio/radio-isa.c | 9 ++++----- drivers/media/radio/radio-isa.h | 2 +- drivers/media/radio/radio-sf16fmr2.c | 4 +--- drivers/net/can/sja1000/tscan1.c | 4 +--- drivers/net/ethernet/3com/3c509.c | 3 +-- drivers/scsi/advansys.c | 3 +-- drivers/scsi/aha1542.c | 3 +-- drivers/scsi/fdomain_isa.c | 3 +-- drivers/scsi/g_NCR5380.c | 5 ++--- drivers/watchdog/pcwd.c | 7 +------ include/linux/isa.h | 2 +- sound/isa/ad1848/ad1848.c | 3 +-- sound/isa/adlib.c | 3 +-- sound/isa/cmi8328.c | 3 +-- sound/isa/cmi8330.c | 3 +-- sound/isa/cs423x/cs4231.c | 3 +-- sound/isa/cs423x/cs4236.c | 3 +-- sound/isa/es1688/es1688.c | 3 +-- sound/isa/es18xx.c | 5 ++--- sound/isa/galaxy/galaxy.c | 3 +-- sound/isa/gus/gusclassic.c | 3 +-- sound/isa/gus/gusextreme.c | 3 +-- sound/isa/gus/gusmax.c | 3 +-- sound/isa/gus/interwave.c | 3 +-- sound/isa/msnd/msnd_pinnacle.c | 3 +-- sound/isa/opl3sa2.c | 3 +-- sound/isa/opti9xx/miro.c | 3 +-- sound/isa/opti9xx/opti92x-ad1848.c | 5 ++--- sound/isa/sb/jazz16.c | 3 +-- sound/isa/sb/sb16.c | 3 +-- sound/isa/sb/sb8.c | 3 +-- sound/isa/sc6000.c | 3 +-- sound/isa/sscape.c | 3 +-- sound/isa/wavefront/wavefront.c | 3 +-- 38 files changed, 44 insertions(+), 88 deletions(-) Interdiff against v1: diff --git a/drivers/media/radio/radio-isa.c b/drivers/media/radio/radio-isa.c index ad2ac16ff12d..c591c0851fa2 100644 --- a/drivers/media/radio/radio-isa.c +++ b/drivers/media/radio/radio-isa.c @@ -273,8 +273,8 @@ static int radio_isa_common_probe(struct radio_isa_card *isa, return res; } -static int radio_isa_common_remove(struct radio_isa_card *isa, - unsigned region_size) +static void radio_isa_common_remove(struct radio_isa_card *isa, + unsigned region_size) { const struct radio_isa_ops *ops = isa->drv->ops; @@ -285,7 +285,6 @@ static int radio_isa_common_remove(struct radio_isa_card *isa, release_region(isa->io, region_size); v4l2_info(&isa->v4l2_dev, "Removed radio card %s\n", isa->drv->card); kfree(isa); - return 0; } int radio_isa_probe(struct device *pdev, unsigned int dev) @@ -338,11 +337,11 @@ int radio_isa_probe(struct device *pdev, unsigned int dev) } EXPORT_SYMBOL_GPL(radio_isa_probe); -int radio_isa_remove(struct device *pdev, unsigned int dev) +void radio_isa_remove(struct device *pdev, unsigned int dev) { struct radio_isa_card *isa = dev_get_drvdata(pdev); - return radio_isa_common_remove(isa, isa->drv->region_size); + radio_isa_common_remove(isa, isa->drv->region_size); } EXPORT_SYMBOL_GPL(radio_isa_remove); diff --git a/drivers/media/radio/radio-isa.h b/drivers/media/radio/radio-isa.h index 2f0736edfda8..c9159958203e 100644 --- a/drivers/media/radio/radio-isa.h +++ b/drivers/media/radio/radio-isa.h @@ -91,7 +91,7 @@ struct radio_isa_driver { int radio_isa_match(struct device *pdev, unsigned int dev); int radio_isa_probe(struct device *pdev, unsigned int dev); -int radio_isa_remove(struct device *pdev, unsigned int dev); +void radio_isa_remove(struct device *pdev, unsigned int dev); #ifdef CONFIG_PNP int radio_isa_pnp_probe(struct pnp_dev *dev, const struct pnp_device_id *dev_id); base-commit: e609571b5ffa3528bf85292de1ceaddac342bc1c -- 2.29.2

3 3

[PATCH v3 0/6] Clock and reset improvements for Tegra ALSA drivers
by Dmitry Osipenko 26 Jan '21

26 Jan '21

This series improves the handling of clock and reset controls of NVIDA Tegra ALSA drivers. Tegra HDA and AHUB drivers aren't handling resets properly, which needs to be fixed in order to unblock other patches related to fixes of the reset controller driver since HDA/AHUB are bound to fail once reset controller driver will be corrected. In particular ALSA drivers are relying on implicit de-assertion of resets which is done by the tegra-clk driver. It's not the business of the clk driver to touch resets and we need to fix this because it breaks reset/clk programming sequences of other Tegra drivers. Changelog: v3: - Reworked "hda/tegra: Reset hardware" and "ahub: Reset hardware properly" patches, they now use usleep + reset_deassert() instead of reset_reset(). Suggested by Thierry Reding. - Added new patch "hda/tegra: Remove unnecessary null-check from hda_tegra_runtime_resume()". Suggested by Thierry Reding. - Replaced "ahub: Reset hardware properly" patch with "ahub: Add missing resets". Suggested by Thierry Reding. - Slightly improved commit messages. - Added acks from Thierry Reding. v2: - Added regcache_sync() to the "ahub: Reset hardware properly" patch, which was missed by accident in v1. - Corrected typo in the format of the error message in "ahub: Use of_reset_control_array_get_exclusive()" patch by s/%p/%pe/. Dmitry Osipenko (6): ALSA: hda/tegra: Use clk_bulk helpers ALSA: hda/tegra: Reset hardware ALSA: hda/tegra: Remove unnecessary null-check from hda_tegra_runtime_resume() ASoC: tegra: ahub: Add missing resets ASoC: tegra: ahub: Use clk_bulk helpers ASoC: tegra: ahub: Reset hardware properly sound/pci/hda/hda_tegra.c | 90 ++++++++++++---------------------- sound/soc/tegra/tegra30_ahub.c | 64 ++++++++++++++---------- sound/soc/tegra/tegra30_ahub.h | 5 +- 3 files changed, 72 insertions(+), 87 deletions(-) -- 2.29.2

3 13

[PATCH] ALSA: bebob: remove an unnecessary condition in hwdep_read()
by Takashi Sakamoto 26 Jan '21

26 Jan '21

Drivers in ALSA firewire stack supports eventing to userspace applications via ALSA hwdep interface. All of the drivers supports stream lock events. Some of them supports their unique events according to specification of target device. ALSA bebob driver supports the stream lock event only. In the case, it's enough to check condition only in loop with process blocking. However, current implementation check it again after breaking the loop. This commit removes the redundant check. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Reported-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_hwdep.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c index 8677e3ec8d14..6f9331655d43 100644 --- a/sound/firewire/bebob/bebob_hwdep.c +++ b/sound/firewire/bebob/bebob_hwdep.c @@ -37,11 +37,9 @@ hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, memset(&event, 0, sizeof(event)); count = min_t(long, count, sizeof(event.lock_status)); - if (bebob->dev_lock_changed) { - event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; - event.lock_status.status = (bebob->dev_lock_count > 0); - bebob->dev_lock_changed = false; - } + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = (bebob->dev_lock_count > 0); + bebob->dev_lock_changed = false; spin_unlock_irq(&bebob->lock); -- 2.27.0

2 1

[PATCH] ASoC: dmaengine_pcm: add peripheral configuration
by Shengjiu Wang 26 Jan '21

26 Jan '21

The commit e7bbb7acabf4 ("dmaengine: add peripheral configuration") adds peripheral configuration for dma_slave_config. This configuration is useful for some audio peripherals, for example, the peripheral supports multi fifos, we can let the DMA know which fifos are selected. So also add this configuration for snd_dmaengine_dai_dma_data. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- include/sound/dmaengine_pcm.h | 5 +++++ sound/core/pcm_dmaengine.c | 2 ++ 2 files changed, 7 insertions(+) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 8c5e38180fb0..96666efddb39 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -66,6 +66,9 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now + * @peripheral_config: peripheral configuration for programming peripheral + * for dmaengine transfer + * @peripheral_size: peripheral configuration buffer size */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -76,6 +79,8 @@ struct snd_dmaengine_dai_dma_data { const char *chan_name; unsigned int fifo_size; unsigned int flags; + void *peripheral_config; + size_t peripheral_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 4d0e8fe535a1..1fc2fa077574 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -125,6 +125,8 @@ void snd_dmaengine_pcm_set_config_from_dai_data( } slave_config->slave_id = dma_data->slave_id; + slave_config->peripheral_config = dma_data->peripheral_config; + slave_config->peripheral_size = dma_data->peripheral_size; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); -- 2.27.0

1 0

bug report: ALSA woke us up to write new data to the device, but there was actually nothing to write. Most likely this is a bug in the ALSA driver 'snd_usb_audio'
by lotus2＠tutanota.com 26 Jan '21

26 Jan '21

```$ journalctl -p3 -b --no-hostname -- Journal begins at Sun 2020-10-18 22:41:56 CEST, ends at Tue 2021-01-26 03:39:21 CET. -- Jan 26 01:29:10 pulseaudio[882]: GetManagedObjects() failed: org.freedesktop.DBus.Error.TimedOut: Failed to activate service 'org.bluez': timed out (service_start_timeout=25000ms) Jan 26 02:41:14 pulseaudio[882]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jan 26 02:41:14 pulseaudio[882]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jan 26 02:41:14 pulseaudio[882]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. $ ``` I reported the `alsa-info_alsa-info.sh` output automatically and it is attached here Operating System: Manjaro Linux KDE Plasma Version: 5.20.5 KDE Frameworks Version: 5.78.0 Qt Version: 5.15.2 Kernel Version: 5.10.10-1-MANJARO OS Type: 64-bit

1 0

[PATCH] ASoC: Intel: sof_sdw: set proper flags for Dell TGL-H SKU 0A5E
by Kai Vehmanen 25 Jan '21

25 Jan '21

From: Libin Yang <libin.yang(a)intel.com> Add flag "SOF_RT711_JD_SRC_JD2", flag "SOF_RT715_DAI_ID_FIX" and "SOF_SDW_FOUR_SPK" to the Dell TGL-H based SKU "0A5E". Signed-off-by: Libin Yang <libin.yang(a)intel.com> Co-developed-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index ca968901ac96..6d0d6ef711e0 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -67,6 +67,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | SOF_RT715_DAI_ID_FIX), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5E") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.29.2

3 2

ASoC: simple-card, fsl-sai, ak4458, imx-ak4458
by Eliot Blennerhassett 25 Jan '21

25 Jan '21

Greetings, late last year I worked on a project using $SUBJECT. That has been suspended at least for now, so I'm writing these notes to my future self, or anyone else who may want to tread the same path. I'm happy to expand on explanation, or work on patches if anyone is interested. Background: Normally, fsl-sai is used with the platform-specific imx-ak4458 (afaik only in vendor tree). However, our project wished to be able use pcm1789 which has no imx specific driver so I started trying to use simple-card with the generic ak4458 driver. I encountered a some of problems doing this: ------------------------------------------------------------------------ 1) Reset polarity of ak4458. When imx-ak4458 is used, the platform driver handles the codec reset specified in DT compatible = "fsl,imx-audio-ak4458"; ak4458,pdn-gpio = <&gpio4 20 GPIO_ACTIVE_LOW>; Used here. Afaics gpio_set_value sets the raw value given, ignoring the polarity specified by the DT? gpio_set_value_cansleep(priv->pdn_gpio, 0); usleep_range(1000, 2000); gpio_set_value_cansleep(priv->pdn_gpio, 1); usleep_range(1000, 2000); The codec driver reset code is not used. When simple-card is used, the codec driver handles reset, specified like: compatible = "asahi-kasei,ak4458"; reset-gpios = <&gpio4 20 GPIO_ACTIVE_LOW>; and used here (inverse for power on) static void ak4458_power_off(struct ak4458_priv *ak4458) { if (ak4458->reset_gpiod) { gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); usleep_range(1000, 2000); } } My understanding of gpiod functions is that value being set is the *logical* value of the GPIO. I.e. setting an active low gpio to 0 will set the hardware pin high. So it appears that the power_off and power_on functions are doing the opposite of what is intended. This is borne out by my hardware working correctly when the ------------------------------------------------------------------------ 2) Clock rate setting with simple-card When simple-card is used and DT specifies mclk fs: simple-audio-card,mclk-fs = <256>; asoc_simple_hw_params() calls snd_soc_dai_set_sysclk(..., clk_id=0, ...) The hard-coded clk_id=0 doesn't work with fsl-sai, which requires clk_id==1. For my testing purposes I changed the hard-coded value, but I think the proper solution could be to add a DT property to specify the clk_id (default=0) ? ------------------------------------------------------------------------ 3) Memory mapped stream access by aplay does not work. This precludes use of alsa plugins e.g. dmix I have found no reason or solution for this so far ------------------------------------------------------------------------ 4) Unable to get multiple source clocks working with fsl-sai With a single assigned-clock, switching between 48k and 44k1 clock rate families is accomplished by reparenting the root clock to the appropriate audio pll clock. &sai2 { assigned-clock-parents = <&clk IMX8MM_AUDIO_PLL1_OUT>; assigned-clocks = <&clk IMX8MM_CLK_SAI2>; assigned-clock-rates = <12288000>; ... }; However if two of the sai mclks could be set to 48k and 44k1 derived rates respectively, the clock reparenting would not be required, and fsl_sai_set_bclk() would search the mclks and choose the appropriate mclk for the requested rate. DT would be something like this: &sai2 { assigned-clock-parents = <&clk IMX8MM_AUDIO_PLL1_OUT>, <&clk IMX8MM_AUDIO_PLL2_OUT>; assigned-clocks = <&clk IMX8MM_CLK_SAI2>, <&clk IMX8MM_CLK_SAI1>; assigned-clock-rates = <12288000>, <11289600>; }; This setup doesn't work as I hoped it would, don't know why not. regards -- Eliot Blennerhassett

3 5

[PATCH 0/2] ASoC: SOF: partial fix to Kconfig issues
by Pierre-Louis Bossart 25 Jan '21

25 Jan '21

We've had several reports of broken dependencies. The 'right' fix is to revisit the module dependencies as suggested by Arnd Bergmann. This is WIP at https://github.com/thesofproject/linux/pull/2683. Since this is taking longer than expected, I am only sharing quick fixes for now. Pierre-Louis Bossart (2): ASoC: SOF: Intel: soundwire: fix select/depend unmet dependencies ASoC: SOF: SND_INTEL_DSP_CONFIG dependency sound/soc/sof/intel/Kconfig | 3 ++- sound/soc/sof/sof-acpi-dev.c | 11 ++++++----- sound/soc/sof/sof-pci-dev.c | 10 ++++++---- 3 files changed, 14 insertions(+), 10 deletions(-) -- 2.25.1

5 8

[PATCH v2] ASoC: mediatek: mt8192-mt6359: add format constraints for RT5682
by Tzung-Bi Shih 25 Jan '21

25 Jan '21

MT8192 determines the I2S clock rates according to the sampling rates. There is only 1 set of I2S in between MT8192 and RT5682. If playing and capturing via RT5682 in different sampling rates, the I2S data will be corrupted. Adds format constraints to the corresponding DAI links to make sure the sampling rates are symmetric. Fixes: 18b13ff23fab ("ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from previous version: - Separated from the series [1] per [2]. - Updated commit message for more details. [1]: https://patchwork.kernel.org/project/alsa-devel/list/?series=419769 [2]: https://patchwork.kernel.org/project/alsa-devel/patch/20210122100742.369912… .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 49 +++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index cc0fc72305d2..c368236d8a3a 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -410,6 +410,53 @@ static const struct snd_soc_ops mt8192_mt6359_rt1015_rt5682_capture1_ops = { .startup = mt8192_mt6359_rt1015_rt5682_cap1_startup, }; +static int +mt8192_mt6359_rt5682_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int channels[] = { + 1, 2 + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + static const unsigned int rates[] = { + 48000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list channels failed\n"); + return ret; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8192_mt6359_rt5682_ops = { + .startup = mt8192_mt6359_rt5682_startup, +}; + /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), @@ -657,6 +704,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8192_mt6359_rt5682_ops, SND_SOC_DAILINK_REG(playback3), }, { @@ -730,6 +778,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8192_mt6359_rt5682_ops, SND_SOC_DAILINK_REG(capture2), }, { -- 2.30.0.280.ga3ce27912f-goog

2 1

[PATCH v2] ASoC: qcom: lpass: Fix out-of-bounds DAI ID lookup
by Stephan Gerhold 25 Jan '21

25 Jan '21

The "dai_id" given into LPAIF_INTFDMA_REG(...) is already the real DAI ID, not an index into v->dai_driver. Looking it up again seems entirely redundant. For IPQ806x (and SC7180 since commit 09a4f6f5d21c ("ASoC: dt-bindings: lpass: Fix and common up lpass dai ids") this is now often an out-of-bounds read because the indexes in the "dai_driver" array no longer match the actual DAI ID. Cc: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Changes in v2: - Extracted from https://lore.kernel.org/alsa-devel/20210114094615.58191-2-stephan@gerhold.n… - Change commit message to clarify that this is usually not just redundant now but actually a broken out-of-bounds lookup. --- sound/soc/qcom/lpass-lpaif-reg.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 405542832e99..baf72f124ea9 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -133,7 +133,7 @@ #define LPAIF_WRDMAPERCNT_REG(v, chan) LPAIF_WRDMA_REG_ADDR(v, 0x14, (chan)) #define LPAIF_INTFDMA_REG(v, chan, reg, dai_id) \ - ((v->dai_driver[dai_id].id == LPASS_DP_RX) ? \ + ((dai_id == LPASS_DP_RX) ? \ LPAIF_HDMI_RDMA##reg##_REG(v, chan) : \ LPAIF_RDMA##reg##_REG(v, chan)) -- 2.30.0

3 2

[PATCH] ALSA: fireface: fix info leak in hwdep_read()
by Dan Carpenter 25 Jan '21

25 Jan '21

If "ff->dev_lock_changed" has not changed and "count" is too large then this will copy data beyond the end of the struct to user space. Fixes: f656edd5fb33 ("ALSA: fireface: add hwdep interface") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/fireface/ff-hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-hwdep.c b/sound/firewire/fireface/ff-hwdep.c index 4b2e0dff5ddb..b84dde609a03 100644 --- a/sound/firewire/fireface/ff-hwdep.c +++ b/sound/firewire/fireface/ff-hwdep.c @@ -35,12 +35,12 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, } memset(&event, 0, sizeof(event)); + count = min_t(long, count, sizeof(event.lock_status)); if (ff->dev_lock_changed) { event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; event.lock_status.status = (ff->dev_lock_count > 0); ff->dev_lock_changed = false; - count = min_t(long, count, sizeof(event.lock_status)); } spin_unlock_irq(&ff->lock); -- 2.29.2

4 8

[PATCH] ALSA: hda: intel-dsp-config: add PCI id for TGL-H
by Kai Vehmanen 25 Jan '21

25 Jan '21

From: Bard Liao <bard.liao(a)intel.com> Adding PCI id for TGL-H. Like for other TGL platforms, SOF is used if Soundwire codecs or PCH-DMIC is detected. Signed-off-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Xiuli Pan <xiuli.pan(a)intel.com> Reviewed-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 6a0d070c60c9..c45686172517 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -307,6 +307,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0xa0c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x43c8, + }, #endif /* Elkhart Lake */ -- 2.29.2

3 4

[PATCH] ALSA: hda: Assign boolean values to a bool variable
by Jiapeng Zhong 25 Jan '21

25 Jan '21

Fix the following coccicheck warnings: ./sound/pci/hda/hda_intel.c:2309:3-23: WARNING: Assignment of 0/1 to bool variable. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Zhong <abaci-bugfix(a)linux.alibaba.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5a50d3a..24e9762 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2304,7 +2304,7 @@ static int azx_probe_continue(struct azx *chip) /* HSW/BDW controllers need this power */ if (CONTROLLER_IN_GPU(pci)) - hda->need_i915_power = 1; + hda->need_i915_power = true; } /* Request display power well for the HDA controller or codec. For -- 1.8.3.1

2 1

[PATCH v2] ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-750
by Olivia Mackintosh 24 Jan '21

24 Jan '21

Re-add the sample-rate quirk originally implemented by Dmitry Panchenko for the DJM-900NXS2 but for the DJM-750. This commit only adds it again for the DJM-750 since I can only verify that it is absolutely required for this device. Other models may need this patch but I'm hesitant to add them as I cannot test. The 'wIndex' is passed literally in the function call as we cannot derive this from 'fmt->sync_ep' due to it not being set. We can change this if and when we revisit implicit FB. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/quirks.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e196e364cef1..b49a66f863f3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,6 +1470,23 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, + u16 windex) +{ + unsigned int cur_rate = subs->data_endpoint->cur_rate; + u8 sr[3]; + // Convert to little endian + sr[0] = cur_rate&0xff; + sr[1] = (cur_rate>>8)&0xff; + sr[2] = (cur_rate>>16)&0xff; + usb_set_interface(subs->dev, 0, 1); + // we should derive windex from fmt-sync_ep but it's not set + snd_usb_ctl_msg(subs->stream->chip->dev, + usb_rcvctrlpipe(subs->stream->chip->dev, 0), + 0x01, 0x22, 0x0100, windex, &sr, 0x0003); + return 0; +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1483,6 +1500,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x08e4, 0x017f): /* Pioneer DJM-750 */ + pioneer_djm_set_format_quirk(subs, 0x0086); + break; } } -- 2.30.0

1 1

[PATCH 0/7] ALSA: add virtio sound driver
by Anton Yakovlev 24 Jan '21

24 Jan '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. The series is based (and was actually tested) on Linus's master branch [2], on top of commit 1e2a199f6ccd ("Merge tag 'spi-fix-v5.11-rc4' of ...") As a device part was used OpenSynergy proprietary implementation. Any comments are very welcome. [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html [2] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git Anton Yakovlev (7): uapi: virtio_ids: add a sound device type ID from OASIS spec uapi: virtio_snd: add the sound device header file ALSA: virtio: add virtio sound driver ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 8 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 361 +++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 593 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 121 +++++++ sound/virtio/virtio_chmap.c | 237 +++++++++++++ sound/virtio/virtio_ctl_msg.c | 293 ++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 122 +++++++ sound/virtio/virtio_jack.c | 255 ++++++++++++++ sound/virtio/virtio_pcm.c | 582 +++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 132 +++++++ sound/virtio/virtio_pcm_msg.c | 317 +++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 524 ++++++++++++++++++++++++++++ 17 files changed, 3573 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.0

3 19

jack: use program_invocation_short_name
by GitHub pull_request - opened 24 Jan '21

24 Jan '21

alsa-project/alsa-plugins pull request #18 was opened from ArsenArsen: this results in more identifiable names in jack patchbays and control programs Request URL : https://github.com/alsa-project/alsa-plugins/pull/18 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/18.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-750
by Olivia Mackintosh 24 Jan '21

24 Jan '21

Re-add the sample-rate quirk originally implemented by Dmitry Panchenko for the DJM-900NXS2 but for the DJM-750. This commit only adds it for the DJM-750 since I can only verify that it is absolutely required for this device. Other models may need this patch but I'm hesitant to add them as I cannot test. The 'wIndex' is taken from `fmt->sync_ep` as different devices have a different sync endpoint which is specified in 'quirks-table.h'. Adding other devices should only require adding an additional case. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/quirks.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e196e364cef1..517346054fb7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,6 +1470,22 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, + const struct audioformat *fmt) +{ + unsigned int cur_rate = subs->data_endpoint->cur_rate; + u8 sr[3]; + // Convert to little endian + sr[0] = cur_rate&0xff; + sr[1] = (cur_rate>>8)&0xff; + sr[2] = (cur_rate>>16)&0xff; + usb_set_interface(subs->dev, 0, 1); + snd_usb_ctl_msg(subs->stream->chip->dev, + usb_rcvctrlpipe(subs->stream->chip->dev, 0), + 0x01, 0x22, 0x0100, fmt->sync_ep, &sr, 0x0003); + return 0; +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1483,6 +1499,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x08e4, 0x017f): /* Pioneer DJM-750 */ + pioneer_djm_set_format_quirk(subs, fmt); + break; } } -- 2.30.0

1 0

[RFC][PATCH v6 0/1] audio jack software injection
by Hui Wang 24 Jan '21

24 Jan '21

the changes in the v6: - use the sizeof(buf) to replace the digits in the scnprintf - squash the document patch into the 0001-xxx.patch. the changes in the v5: - add a Kconfig to enable/disable the jack injection feature - replace all kzalloc with local char[] in the debugfs fops - replace the chars of !isalnum() to '_' for the jack folder's name - squash all .c files patches into one patch - add a document to explain jack injection, for easier review, put the document in a standalone patch. If needed, will squash this patch to the previous one. the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (1): alsa: jack: implement software jack injection via debugfs Documentation/sound/designs/index.rst | 1 + .../sound/designs/jack-injection.rst | 124 ++++++++ include/sound/core.h | 2 + include/sound/jack.h | 1 + sound/core/Kconfig | 9 + sound/core/init.c | 10 + sound/core/jack.c | 298 +++++++++++++++++- sound/core/sound.c | 8 + 8 files changed, 449 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/designs/jack-injection.rst -- 2.25.1

2 3

[PATCH] sound/pci/hda: boolean values to a bool variable
by Jiapeng Zhong 24 Jan '21

24 Jan '21

Fix the following coccicheck warnings: ./sound/pci/hda/patch_conexant.c:570:2-20: WARNING: Assignment of 0/1 to bool variable. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Zhong <abaci-bugfix(a)linux.alibaba.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d49cc44..f2aa226 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -567,7 +567,7 @@ static void cxt_fixup_mute_led_eapd(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mute_led_eapd = 0x1b; - spec->dynamic_eapd = 1; + spec->dynamic_eapd = true; snd_hda_gen_add_mute_led_cdev(codec, cx_auto_vmaster_mute_led); } } -- 1.8.3.1

2 1

ASoC: cpcap: Implement set_tdm_slot for voice call support
by Pavel Machek 23 Jan '21

23 Jan '21

From: Tony Lindgren <tony(a)atomide.com> ASoC: cpcap: Implement set_tdm_slot for voice call support For using cpcap for voice calls, we need to route audio directly from the modem to cpcap for TDM (Time Division Multiplexing). The voice call is direct data between the modem and cpcap with no CPU involvment. In this mode, the cpcap related audio mixer controls work for the speaker selection and volume though. To do this, we need to implement standard snd_soc_dai_set_tdm_slot() for cpcap. Then the modem codec driver can use snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt(), and snd_soc_dai_set_tdm_slot() to configure a voice call. Let's add cpcap_voice_set_tdm_slot() for this, and cpcap_voice_call() helper to configure the additional registers needed for voice call. Let's also clear CPCAP_REG_VAUDIOC on init in case we have the bit for CPCAP_BIT_VAUDIO_MODE0 set on init. Signed-off-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Pavel Machek <pavel(a)ucw.cz> diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index f046987ee4cd..e266d993ab2a 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -16,6 +16,14 @@ #include <sound/soc.h> #include <sound/tlv.h> +/* Register 512 CPCAP_REG_VAUDIOC --- Audio Regulator and Bias Voltage */ +#define CPCAP_BIT_AUDIO_LOW_PWR 6 +#define CPCAP_BIT_AUD_LOWPWR_SPEED 5 +#define CPCAP_BIT_VAUDIOPRISTBY 4 +#define CPCAP_BIT_VAUDIO_MODE1 2 +#define CPCAP_BIT_VAUDIO_MODE0 1 +#define CPCAP_BIT_V_AUDIO_EN 0 + /* Register 513 CPCAP_REG_CC --- CODEC */ #define CPCAP_BIT_CDC_CLK2 15 #define CPCAP_BIT_CDC_CLK1 14 @@ -221,6 +229,7 @@ struct cpcap_reg_info { }; static const struct cpcap_reg_info cpcap_default_regs[] = { + { CPCAP_REG_VAUDIOC, 0x003F, 0x0000 }, { CPCAP_REG_CC, 0xFFFF, 0x0000 }, { CPCAP_REG_CC, 0xFFFF, 0x0000 }, { CPCAP_REG_CDI, 0xBFFF, 0x0000 }, @@ -1371,8 +1380,121 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int cpcap_voice_set_mute(struct snd_soc_dai *dai, - int mute, int direction) + +/* + * Configure codec for voice call if requested. + * + * We can configure most with snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt() + * and snd_soc_dai_set_tdm_slot(). This function configures the rest of the + * cpcap related hardware as CPU is not involved in the voice call. + */ +static int cpcap_voice_call(struct cpcap_audio *cpcap, struct snd_soc_dai *dai, + bool voice_call) +{ + int mask, err; + + /* Modem to codec VAUDIO_MODE1 */ + mask = BIT(CPCAP_BIT_VAUDIO_MODE1); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_VAUDIOC, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Clear MIC1_MUX for call */ + mask = BIT(CPCAP_BIT_MIC1_MUX); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI, + mask, voice_call ? 0 : mask); + if (err) + return err; + + /* Set MIC2_MUX for call */ + mask = BIT(CPCAP_BIT_MB_ON1L) | BIT(CPCAP_BIT_MB_ON1R) | + BIT(CPCAP_BIT_MIC2_MUX) | BIT(CPCAP_BIT_MIC2_PGA_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable LDSP for call */ + mask = BIT(CPCAP_BIT_A2_LDSP_L_EN) | BIT(CPCAP_BIT_A2_LDSP_R_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXOA, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable CPCAP_BIT_PGA_CDC_EN for call */ + mask = BIT(CPCAP_BIT_PGA_CDC_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXCOA, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Unmute voice for call */ + if (dai) { + err = snd_soc_dai_digital_mute(dai, !voice_call, + SNDRV_PCM_STREAM_PLAYBACK); + if (err) + return err; + } + + /* Set modem to codec mic CDC and HPF for call */ + mask = BIT(CPCAP_BIT_MIC2_CDC_EN) | BIT(CPCAP_BIT_CDC_EN_RX) | + BIT(CPCAP_BIT_AUDOHPF_1) | BIT(CPCAP_BIT_AUDOHPF_0) | + BIT(CPCAP_BIT_AUDIHPF_1) | BIT(CPCAP_BIT_AUDIHPF_0); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CC, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable modem to codec CDC for call*/ + mask = BIT(CPCAP_BIT_CDC_CLK_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, + mask, voice_call ? mask : 0); + + return err; +} + +static int cpcap_voice_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + int err, ts_mask, mask; + bool voice_call; + + /* + * Primitive test for voice call, probably needs more checks + * later on for 16-bit calls detected, Bluetooth headset etc. + */ + if (tx_mask == 0 && rx_mask == 1 && slot_width == 8) + voice_call = true; + else + voice_call = false; + + ts_mask = 0x7 << CPCAP_BIT_MIC2_TIMESLOT0; + ts_mask |= 0x7 << CPCAP_BIT_MIC1_RX_TIMESLOT0; + + mask = (tx_mask & 0x7) << CPCAP_BIT_MIC2_TIMESLOT0; + mask |= (rx_mask & 0x7) << CPCAP_BIT_MIC1_RX_TIMESLOT0; + + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, + ts_mask, mask); + if (err) + return err; + + err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, slot_width * 1000); + if (err) + return err; + + err = cpcap_voice_call(cpcap, dai, voice_call); + if (err) + return err; + + return 0; +} + +static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); @@ -1393,6 +1515,7 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = { .hw_params = cpcap_voice_hw_params, .set_sysclk = cpcap_voice_set_dai_sysclk, .set_fmt = cpcap_voice_set_dai_fmt, + .set_tdm_slot = cpcap_voice_set_tdm_slot, .mute_stream = cpcap_voice_set_mute, .no_capture_mute = 1, }; -- http://www.livejournal.com/~pavelmachek

2 2

[PATCH alsa-ucm-conf 1/3] chtnau8824: Move DAC Channel Source selection to Speaker/Headphones EnableSeq
by Hans de Goede 23 Jan '21

23 Jan '21

Move DAC Channel Source selection to Speaker/Headphones EnableSeq. The main reason for doing this is to make it easier to allow using variables, as the Speaker/Headphones conf files are evaluated after the main HiFi.conf had a chance to set them. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/codecs/nau8824/EnableSeq.conf | 3 --- ucm2/codecs/nau8824/HeadPhones.conf | 2 ++ ucm2/codecs/nau8824/MonoSpeaker.conf | 2 ++ ucm2/codecs/nau8824/Speaker.conf | 2 ++ 4 files changed, 6 insertions(+), 3 deletions(-) diff --git a/ucm2/codecs/nau8824/EnableSeq.conf b/ucm2/codecs/nau8824/EnableSeq.conf index 3650d06..2595e5f 100644 --- a/ucm2/codecs/nau8824/EnableSeq.conf +++ b/ucm2/codecs/nau8824/EnableSeq.conf @@ -1,7 +1,4 @@ EnableSequence [ - # Playback TDM configuration - cset "name='DACL Channel Source' 0" - cset "name='DACR Channel Source' 1" # Input Configuration cset "name='DMIC1 Enable Switch' off" cset "name='DMIC2 Enable Switch' off" diff --git a/ucm2/codecs/nau8824/HeadPhones.conf b/ucm2/codecs/nau8824/HeadPhones.conf index a807f84..46ec5d7 100644 --- a/ucm2/codecs/nau8824/HeadPhones.conf +++ b/ucm2/codecs/nau8824/HeadPhones.conf @@ -12,6 +12,8 @@ SectionDevice."Headphones" { ] EnableSequence [ + cset "name='DACL Channel Source' 0" + cset "name='DACR Channel Source' 1" cset "name='Headphone Switch' on" ] diff --git a/ucm2/codecs/nau8824/MonoSpeaker.conf b/ucm2/codecs/nau8824/MonoSpeaker.conf index 6b4ef8a..09edea5 100644 --- a/ucm2/codecs/nau8824/MonoSpeaker.conf +++ b/ucm2/codecs/nau8824/MonoSpeaker.conf @@ -11,6 +11,8 @@ SectionDevice."Speaker" { ] EnableSequence [ + cset "name='DACL Channel Source' 0" + cset "name='DACR Channel Source' 1" # nau8824 mono speaker boards have the speaker on the right chan cset "name='Speaker Left DACL Volume' 0" cset "name='Speaker Left DACR Volume' 0" diff --git a/ucm2/codecs/nau8824/Speaker.conf b/ucm2/codecs/nau8824/Speaker.conf index a995873..64d7eee 100644 --- a/ucm2/codecs/nau8824/Speaker.conf +++ b/ucm2/codecs/nau8824/Speaker.conf @@ -11,6 +11,8 @@ SectionDevice."Speaker" { ] EnableSequence [ + cset "name='DACL Channel Source' 0" + cset "name='DACR Channel Source' 1" cset "name='Speaker Left DACL Volume' 1" cset "name='Speaker Left DACR Volume' 0" cset "name='Speaker Right DACL Volume' 0" -- 2.28.0

2 6

[PATCH] ALSA: usb-audio: workaround for iface reset issue
by Takashi Iwai 23 Jan '21

23 Jan '21

The recently introduced sample rate validation code seems causing a problem on some devices; namely, after performing this, the bus gets screwed and it influences even on other USB devices. As a quick workaround, perform it only for the necessary devices; currently MOTU devices are known to need the valid altset checks, so filter out other devices. Fixes: 93db51d06b32 ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3") Reported-by: Jamie Heilman <jamie(a)audible.transient.net> BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1178203 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/format.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 9ebc5d202c87..e6ff317a6785 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -466,6 +466,17 @@ static int validate_sample_rate_table_v2v3(struct snd_usb_audio *chip, unsigned int nr_rates; int i, err; + /* performing the rate verification may lead to unexpected USB bus + * behavior afterwards by some unknown reason. Do this only for the + * known devices. + */ + switch (USB_ID_VENDOR(chip->usb_id)) { + case 0x07fd: /* MOTU */ + break; + default: + return 0; /* don't perform the validation as default */ + } + table = kcalloc(fp->nr_rates, sizeof(*table), GFP_KERNEL); if (!table) return -ENOMEM; -- 2.26.2

1 0

[PATCH] ALSA: pcm: One more dependency for hw constraints
by Takashi Iwai 23 Jan '21

23 Jan '21

The fix for a long-standing USB-audio bug required one more dependency variable to be added to the hw constraints. Unfortunately I didn't realize at debugging that the new addition may result in the overflow of the dependency array of each snd_pcm_hw_rule (up to three plus a sentinel), because USB-audio driver adds one more dependency only for a certain device and bus, hence it works as is for many devices. But in a bad case, a simple open always results in -EINVAL (with kernel WARNING if CONFIG_SND_DEBUG is set) no matter what is passed. Since the dependencies are real and unavoidable (USB-audio restricts the hw_params per looping over the format/rate/channels combos), the only good solution seems to raise the bar for one more dependency for snd_pcm_hw_rule -- so does this patch: now the hw constraint dependencies can be up to four. Fixes: 506c203cc3de ("ALSA: usb-audio: Fix hw constraints dependencies") Reported-by: Jamie Heilman <jamie(a)audible.transient.net> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 2 +- sound/core/pcm_native.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2336bf9243e1..2e1200d17d0c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -229,7 +229,7 @@ typedef int (*snd_pcm_hw_rule_func_t)(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule { unsigned int cond; int var; - int deps[4]; + int deps[5]; snd_pcm_hw_rule_func_t func; void *private; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9f3f8e953ff0..c4aac703dc22 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -382,8 +382,8 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, continue; /* - * The 'deps' array includes maximum three dependencies - * to SNDRV_PCM_HW_PARAM_XXXs for this rule. The fourth + * The 'deps' array includes maximum four dependencies + * to SNDRV_PCM_HW_PARAM_XXXs for this rule. The fifth * member of this array is a sentinel and should be * negative value. * -- 2.26.2

1 0

Question about hdac_ext_link ref_count management
by Ranjani Sridharan 22 Jan '21

22 Jan '21

Hi Takashi, While exploring some power optimizations on Intel platforms, I noticed that the hdac_ext_link ref_count is incremented during codec probe in hdac_hda_codec_probe() and the ref_count is held until the codec device is removed. I was wondering if it would be possible to call the get/put for the hdac_ext_link in the codec runtime suspend/resume callbacks so that the link is powered up only when the link is in use. Are there any downsides to doing this? Thanks, Ranjani

2 5

[PATCH v3 00/13] MFD/extcon/ASoC: Rework arizona codec jack-detect support
by Hans de Goede 22 Jan '21

22 Jan '21

Hi all, Here is v3 om my series to rework the arizona codec jack-detect support to use the snd_soc_jack helpers instead of direct extcon reporting. This is done by reworking the extcon driver into an arizona-jackdet library and then modifying the codec drivers to use that directly, replacing the old separate extcon child-devices and extcon-driver. This brings the arizona-codec jack-detect handling inline with how all other ASoC codec driver do this. This was developed and tested on a Lenovo Yoga Tablet 1051L with a WM5102 codec. There are various interdependencies between the patches in this series, so IMHO it would be best if this entire series would be merged through the MFD tree. Note this series applies on top of my "[PATCH v4 0/5] MFD/ASoC: Add support for Intel Bay Trail boards with WM5102 codec" series. Changes in v3: -Move the bugfix patches to earlier in the series so that they apply to drivers/extcon/extcon-arizona.c so that they can be cherry-picked into the stable series -Split runtime_pm_get -> runtime_pm_get_sync changes out into their own patch -Simply move drivers/extcon/extcon-arizona.c to sound/soc/codecs/arizona-jack.c instead of first adding arizona-jack.c as a copy and then later removing extcon-arizona.c -Some other small tweaks, see individual patch changelogs Regards, Hans Hans de Goede (13): mfd: arizona: Drop arizona-extcon cells extcon: arizona: Fix some issues when HPDET IRQ fires after the jack has been unplugged extcon: arizona: Fix various races on driver unbind extcon: arizona: Fix flags parameter to the gpiod_get("wlf,micd-pol") call extcon: arizona: Always use pm_runtime_get_sync() when we need the device to be awake ASoC/extcon: arizona: Move arizona jack code to sound/soc/codecs/arizona-jack.c ASoC: arizona-jack: Move jack-detect variables to struct arizona_priv ASoC: arizona-jack: Use arizona->dev for runtime-pm ASoC: arizona-jack: convert into a helper library for codec drivers ASoC: arizona-jack: Use snd_soc_jack to report jack events ASoC: arizona-jack: Cleanup logging ASoC: arizona: Make the wm5102, wm5110, wm8997 and wm8998 drivers use the new jack library ASoC: Intel: bytcr_wm5102: Add jack detect support MAINTAINERS | 1 - drivers/extcon/Kconfig | 8 - drivers/extcon/Makefile | 1 - drivers/mfd/arizona-core.c | 20 - sound/soc/codecs/Makefile | 2 +- .../soc/codecs/arizona-jack.c | 539 +++++++----------- sound/soc/codecs/arizona.h | 44 ++ sound/soc/codecs/wm5102.c | 12 +- sound/soc/codecs/wm5110.c | 12 +- sound/soc/codecs/wm8997.c | 14 +- sound/soc/codecs/wm8998.c | 9 + sound/soc/intel/boards/bytcr_wm5102.c | 28 +- 12 files changed, 307 insertions(+), 383 deletions(-) rename drivers/extcon/extcon-arizona.c => sound/soc/codecs/arizona-jack.c (77%) -- 2.28.0

2 23

[PATCH alsa-lib 0/4] pcm: Fix missing appl_ptr and hw_ptr updates in snd_pcm_status()
by Takashi Iwai 22 Jan '21

22 Jan '21

Hi, here is the fixes for a crash with dmix plugin and co, as well as the missing appl_ptr/hw_ptr updates that are found on other plugins. Takashi === Takashi Iwai (4): pcm: direct: Fix the missing appl_ptr update pcm: ioplug: Pass appl_ptr and hw_ptr in snd_pcm_status() pcm: null: Pass appl_ptr and hw_ptr in snd_pcm_status() pcm: share: Pass appl_ptr and hw_ptr in snd_pcm_status() src/pcm/pcm_dmix.c | 1 + src/pcm/pcm_dshare.c | 1 + src/pcm/pcm_dsnoop.c | 1 + src/pcm/pcm_ioplug.c | 2 ++ src/pcm/pcm_null.c | 2 ++ src/pcm/pcm_share.c | 2 ++ 6 files changed, 9 insertions(+) -- 2.26.2

2 5

[PATCH] drivers: slimbus: Fix word resposible -> responsible in slimbus.h
by Bhaskar Chowdhury 22 Jan '21

22 Jan '21

s/resposible/responsible/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- drivers/slimbus/slimbus.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..503e2a0200f5 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -368,7 +368,7 @@ struct slim_stream_runtime { * @get_laddr: It is possible that controller needs to set fixed logical * address table and get_laddr can be used in that case so that controller * can do this assignment. Use case is when the master is on the remote - * processor side, who is resposible for allocating laddr. + * processor side, who is responsible for allocating laddr. * @wakeup: This function pointer implements controller-specific procedure * to wake it up from clock-pause. Framework will call this to bring * the controller out of clock pause. -- 2.30.0

3 2

Wrong binds configuration for dmix plugin causes segmentation fault in alsa
by GitHub issues - opened 22 Jan '21

22 Jan '21

alsa-project/alsa-lib issue #117 was opened from elara-leitstellentechnik: On Debian buster I can get alsa to run into a segmentation fault with the following procedure: Configure dmix with two channels (binds) for a Mono device. (I am not sure that this is reproducible with any mono device, I tested with a Jabra LINK 230): ``` ~$ cat .asoundrc pcm."Jabra LINK 230" { type dmix ipc_key 1024 slave.pcm "hw:2,0" bindings { 0 0 # The next line is the problematic line that triggers the segfault 1 1 } } ``` Then when playing any sound file with e.g. mplayer and making it use the dmix device the following happens: ``` $ mplayer -ao alsa:device="Jabra LINK 230" /usr/share/sounds/alsa/Front_Left.wav MPlayer 1.3.0 (Debian), built with gcc-8 (C) 2000-2016 MPlayer Team do_connect: could not connect to socket connect: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /usr/share/sounds/alsa/Front_Left.wav. libavformat version 58.20.100 (external) Mismatching header version 58.12.100 Audio only file format detected. Load subtitles in /usr/share/sounds/alsa/ ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 48000 Hz, 1 ch, s16le, 768.0 kbit/100.00% (ratio: 96000->96000) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== [AO_ALSA] alsa-lib: pcm_hw.c:1711:(snd_pcm_hw_open) open '/dev/snd/pcmC2D0p' failed (-77): File descriptor in bad state AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... MPlayer interrupted by signal 11 in module: play_audio - MPlayer crashed by bad usage of CPU/FPU/RAM. Recompile MPlayer with --enable-debug and make a 'gdb' backtrace and disassembly. Details in DOCS/HTML/en/bugreports_what.html#bugreports_crash. [ This binary of MPlayer in Debian is currently compiled with '--enable-debug'; the debugging symbols are in the package 'mplayer-dbgsym'.] ``` I appreciate that configuring two bindings for a single-channel audio device is an error, but I am not sure that this justifies a segfault. Here is some more information I have been able to extract using gdb: ``` ~$ gdb --args mplayer -ao alsa:device="Jabra LINK 230" /usr/share/sounds/alsa/Front_Left.wav GNU gdb (Debian 8.2.1-2+b3) 8.2.1 Copyright (C) 2018 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. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". Type "show configuration" for configuration details. For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>. Find the GDB manual and other documentation resources online at: <http://www.gnu.org/software/gdb/documentation/>. For help, type "help". Type "apropos word" to search for commands related to "word"... Reading symbols from mplayer...Reading symbols from /usr/lib/debug/.build-id/3a/9816e9d363979a4f76e6cb809a33dffda67d05.debug...done. done. (gdb) run Starting program: /usr/bin/mplayer -ao alsa:device=Jabra\ LINK\ 230 /usr/share/sounds/alsa/Front_Left.wav [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". MPlayer 1.3.0 (Debian), built with gcc-8 (C) 2000-2016 MPlayer Team do_connect: could not connect to socket connect: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /usr/share/sounds/alsa/Front_Left.wav. libavformat version 58.20.100 (external) Mismatching header version 58.12.100 Audio only file format detected. Load subtitles in /usr/share/sounds/alsa/ ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 48000 Hz, 1 ch, s16le, 768.0 kbit/100.00% (ratio: 96000->96000) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== [AO_ALSA] alsa-lib: pcm_hw.c:1711:(snd_pcm_hw_open) open '/dev/snd/pcmC2D0p' failed (-77): File descriptor in bad state AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... Program received signal SIGSEGV, Segmentation fault. 0x00007ffff7a09b76 in mix_areas_16_smp (size=4160473088, size@entry=17424, dst=0x555555aa9510, src=0xde020109, sum=0x21fe0f3c, dst_step=2, src_step=4, sum_step=8) at pcm_dmix_x86_64.h:48 48 pcm_dmix_x86_64.h: No such file or directory. (gdb) where #0 0x00007ffff7a09b76 in mix_areas_16_smp (size=4160473088, size@entry=17424, dst=0x555555aa9510, src=0xde020109, sum=0x21fe0f3c, dst_step=2, src_step=4, sum_step=8) at pcm_dmix_x86_64.h:48 #1 0x00007ffff7a0ba1c in mix_areas (size=17424, dst_ofs=576, src_ofs=0, dst_areas=0x555555a927f0, src_areas=0x7fffe88b3000, dmix=0x555555a929e0) at pcm_dmix.c:215 #2 snd_pcm_dmix_sync_area (pcm=pcm@entry=0x555555a94850) at pcm_dmix.c:382 #3 0x00007ffff7a0bdfa in snd_pcm_dmix_start (pcm=0x555555a94850) at pcm_dmix.c:619 #4 0x00007ffff79d8b4a in __snd_pcm_start (pcm=0x555555a94850, pcm=0x555555a94850) at pcm_local.h:434 #5 snd1_pcm_write_areas (pcm=pcm@entry=0x555555a94850, areas=areas@entry=0x7fffffffcfd0, offset=offset@entry=0, size=<optimized out>, size@entry=18000, func=func@entry=0x7ffff79e46c0 <snd_pcm_mmap_write_areas>) at pcm.c:7421 #6 0x00007ffff79e4b01 in snd_pcm_mmap_writei (pcm=0x555555a94850, buffer=<optimized out>, size=18000) at pcm_mmap.c:153 #7 0x00005555556242f8 in play (data=0x555555ac4f70, len=<optimized out>, flags=<optimized out>) at libao2/ao_alsa.c:812 #8 0x000055555560bbab in fill_audio_out_buffers () at mplayer.c:2212 #9 main (argc=<optimized out>, argv=<optimized out>) at mplayer.c:3794 (gdb) ``` To me this looks like the segfault is triggered in pcm_dmix_x86_64.h:48 so I believe it is an alsa problem rather than a problem with mplayer. I hope this makes sense. If I can support with further information please let me know. Issue URL : https://github.com/alsa-project/alsa-lib/issues/117 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/4] ASoC: mediatek: mt8192-mt6359: fix I2S corrupted for RT5682
by Tzung-Bi Shih 22 Jan '21

22 Jan '21

The series fixes the I2S corrupted when mt8192-mt6359 is playing and capturing via RT5682 in different formats. The 1st and 2nd patch refactor to use asoc_substream_to_rtd(). Just realized there is such a helper for the purpose. The 3rd patch simplies ops of Capture1 DAI link. The 4th patch adds format constriants to the I2S to make sure the settings are symmetric. Tzung-Bi Shih (4): ASoC: mediatek: mt8192-mt6359: use asoc_substream_to_rtd() ASoC: mediatek: mt8192: use asoc_substream_to_rtd() ASoC: mediatek: mt8192-mt6359: simply ops for Capture1 DAI link ASoC: mediatek: mt8192-mt6359: add format constraints for RT5682 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 4 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 73 +++++++++++++++---- 2 files changed, 61 insertions(+), 16 deletions(-) -- 2.30.0.280.ga3ce27912f-goog

2 5

[PATCH v2 00/12] MFD/extcon/ASoC: Rework arizona codec jack-detect support
by Hans de Goede 22 Jan '21

22 Jan '21

Hi All, This series reworks the arizona codec jack-detect support to use the snd_soc_jack helpers instead of direct extcon reporting. This is done by reworking the extcon driver into an arizona-jackdet library and then modifying the codec drivers to use that directly, replacing the old separate extcon child-devices and extcon-driver. This brings the arizona-codec jack-detect handling inline with how all other ASoC codec driver do this. This was developed and tested on a Lenovo Yoga Tablet 1051L with a WM5102 codec. The MFD, ASoC and extcon parts can be merged independent from each-other although that could lead to a case where both the extcon driver and the new arizona-jackdet library will try to do jack-detection. If we end up with a git tree in that state then one of the 2 will fail to load because the other will already have claimed the IRQs, so this is not a problem really. Or the entire series could be merged through the MFD tree if people prefer that. Note that this series also paves the way for some further cleanups, removing some jackdetect related variables like hp_ena and hp_clamp from the arizona data struct shared between all the MFD child devices. I've deliberately not done that cleanup as part of this patch-series, since IMHO the series is big enough as is. These cleanups can be done in a follow-up series once this series has landed. Regards, Hans

6 34

[RFC][PATCH v5 0/2] audio jack software injection
by Hui Wang 22 Jan '21

22 Jan '21

the changes in the v5: - add a Kconfig to enable/disable the jack injection feature - replace all kzalloc with local char[] in the debugfs fops - replace the chars of !isalnum() to '_' for the jack folder's name - squash all .c files patches into one patch - add a document to explain jack injection, for easier review, put the document in a standalone patch. If needed, will squash this patch to the previous one. the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (2): alsa: jack: implement software jack injection via debugfs alsa: jack: add document for sound jack injection. Documentation/sound/designs/index.rst | 1 + .../sound/designs/jack-injection.rst | 124 ++++++++ include/sound/core.h | 2 + include/sound/jack.h | 1 + sound/core/Kconfig | 9 + sound/core/init.c | 10 + sound/core/jack.c | 298 +++++++++++++++++- sound/core/sound.c | 8 + 8 files changed, 449 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/designs/jack-injection.rst -- 2.25.1

2 4

[PATCH] ASoC: qcom: Fix number of HDMI RDMA channels on sc7180
by Stephen Boyd 22 Jan '21

22 Jan '21

Suspending/resuming with an HDMI dongle attached leads to crashes from an audio regmap. Unable to handle kernel paging request at virtual address ffffffc018068000 Mem abort info: ESR = 0x96000047 EC = 0x25: DABT (current EL), IL = 32 bits SET = 0, FnV = 0 EA = 0, S1PTW = 0 Data abort info: ISV = 0, ISS = 0x00000047 CM = 0, WnR = 1 swapper pgtable: 4k pages, 39-bit VAs, pgdp=0000000081b12000 [ffffffc018068000] pgd=0000000275d14003, pud=0000000275d14003, pmd=000000026365d003, pte=0000000000000000 Internal error: Oops: 96000047 [#1] PREEMPT SMP Call trace: regmap_mmio_write32le+0x2c/0x40 regmap_mmio_write+0x48/0x6c _regmap_bus_reg_write+0x34/0x44 _regmap_write+0x100/0x150 regcache_default_sync+0xc0/0x138 regcache_sync+0x188/0x26c lpass_platform_pcmops_resume+0x48/0x54 [snd_soc_lpass_platform] snd_soc_component_resume+0x28/0x40 soc_resume_deferred+0x6c/0x178 process_one_work+0x208/0x3c8 worker_thread+0x23c/0x3e8 kthread+0x144/0x178 ret_from_fork+0x10/0x18 Code: d503201f d50332bf f94002a8 8b344108 (b9000113) I can reliably reproduce this problem by running 'tail' on the registers file in debugfs for the hdmi regmap. # tail /sys/kernel/debug/regmap/62d87000.lpass-lpass_hdmi/registers [ 84.658733] Unable to handle kernel paging request at virtual address ffffffd0128e800c This crash happens because we're trying to read registers from the regmap beyond the length of the mapping created by ioremap(). The number of hdmi_rdma_channels determines the size of the regmap via this code in sound/soc/qcom/lpass-cpu.c: lpass_hdmi_regmap_config.max_register = LPAIF_HDMI_RDMAPER_REG(variant, variant->hdmi_rdma_channels); According to debugfs the size of the regmap is 0x68010 but according to the DTS file posted in [1] the size is only 0x68000 (see the first reg property of the lpass_cpu node). Let's change the number of channels to be 3 instead of 4 so the math works out to have a max register of 0x67010, nicely fitting inside of the region size of 0x68000. Note: I tried to bump up the size of the register region to the next page to include the 0x68010 register but then the tail command caused SErrors with an async abort, implying that the register region doesn't exist or it isn't clocked because the bus is telling us that the register read failed. I reduce the number of channels and played audio through the HDMI channel and it kept working so I think this is correct. Fixes: 2ad63dc8df6b ("ASoC: qcom: sc7180: Add support for audio over DP") Link: https://lore.kernel.org/r/1601448168-18396-2-git-send-email-srivasam@codeau… [1] Cc: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Cc: Srinivasa Rao <srivasam(a)codeaurora.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- sound/soc/qcom/lpass-sc7180.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 85db650c2169..3127b6022ea3 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -174,7 +174,7 @@ static struct lpass_variant sc7180_data = { .rdma_channels = 5, .hdmi_rdma_reg_base = 0x64000, .hdmi_rdma_reg_stride = 0x1000, - .hdmi_rdma_channels = 4, + .hdmi_rdma_channels = 3, .dmactl_audif_start = 1, .wrdma_reg_base = 0x18000, .wrdma_reg_stride = 0x1000, base-commit: 5c8fe583cce542aa0b84adc939ce85293de36e5e -- https://chromeos.dev

3 3

[PATCH v1 0/2] isa: Make the remove callback for isa drivers return void
by Uwe Kleine-König 22 Jan '21

22 Jan '21

Hello, as described in the commit log of the 2nd patch returning an error code from a bus' remove callback doesn't make any difference as the driver core ignores it and still considers the device removed. So change the remove callback to return void to not give driver authors an incentive to believe they could return an error. There is only a single isa driver in the tree (assuming I didn't miss any) that has a remove callback that can return a non zero return code. This is "fixed" in the first patch, to make the second patch more obviously correct. Best regards Uwe Uwe Kleine-König (2): watchdog: pcwd: drop always-false if from remove callback isa: Make the remove callback for isa drivers return void drivers/base/isa.c | 2 +- drivers/i2c/busses/i2c-elektor.c | 4 +--- drivers/i2c/busses/i2c-pca-isa.c | 4 +--- drivers/input/touchscreen/htcpen.c | 4 +--- drivers/media/radio/radio-sf16fmr2.c | 4 +--- drivers/net/can/sja1000/tscan1.c | 4 +--- drivers/net/ethernet/3com/3c509.c | 3 +-- drivers/scsi/advansys.c | 3 +-- drivers/scsi/aha1542.c | 3 +-- drivers/scsi/fdomain_isa.c | 3 +-- drivers/scsi/g_NCR5380.c | 3 +-- drivers/watchdog/pcwd.c | 7 +------ include/linux/isa.h | 2 +- sound/isa/ad1848/ad1848.c | 3 +-- sound/isa/adlib.c | 3 +-- sound/isa/cmi8328.c | 3 +-- sound/isa/cmi8330.c | 3 +-- sound/isa/cs423x/cs4231.c | 3 +-- sound/isa/cs423x/cs4236.c | 3 +-- sound/isa/es1688/es1688.c | 3 +-- sound/isa/es18xx.c | 3 +-- sound/isa/galaxy/galaxy.c | 3 +-- sound/isa/gus/gusclassic.c | 3 +-- sound/isa/gus/gusextreme.c | 3 +-- sound/isa/gus/gusmax.c | 3 +-- sound/isa/gus/interwave.c | 3 +-- sound/isa/msnd/msnd_pinnacle.c | 3 +-- sound/isa/opl3sa2.c | 3 +-- sound/isa/opti9xx/miro.c | 3 +-- sound/isa/opti9xx/opti92x-ad1848.c | 3 +-- sound/isa/sb/jazz16.c | 3 +-- sound/isa/sb/sb16.c | 3 +-- sound/isa/sb/sb8.c | 3 +-- sound/isa/sc6000.c | 3 +-- sound/isa/sscape.c | 3 +-- sound/isa/wavefront/wavefront.c | 3 +-- 36 files changed, 36 insertions(+), 79 deletions(-) base-commit: 5a158981aafa7f29709034b17bd007b15cb29983 -- 2.29.2

5 5

[PATCH] ALSA: hda/realtek: Enable headset of ASUS B1400CEPE with ALC256
by Jian-Hong Pan 22 Jan '21

22 Jan '21

ASUS B1400CEPE laptop's headset audio is not enabled until ALC256_FIXUP_ASUS_HPE quirk is applied. Here is the original pin node values: 0x12 0x40000000 0x13 0x411111f0 0x14 0x90170110 0x18 0x411111f0 0x19 0x411111f0 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x40461b45 0x1e 0x411111f0 0x21 0x04211020 Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- 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 ed5b6b894dc1..290645516313 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8006,6 +8006,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x1982, "ASUS B1400CEPE", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 2.30.0

2 1

[PATCH 1/2] ALSA: oxfw: fix info leak in hwdep_read()
by Dan Carpenter 22 Jan '21

22 Jan '21

If "count" is too large and "oxfw->dev_lock_changed" is false then this will copy beyond the end of the struct to user space. Fixes: 8985f4ac1c42 ("ALSA: oxfw: Add hwdep interface") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/oxfw/oxfw-hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw-hwdep.c b/sound/firewire/oxfw/oxfw-hwdep.c index 9e1b3e151bad..ca5477eeb663 100644 --- a/sound/firewire/oxfw/oxfw-hwdep.c +++ b/sound/firewire/oxfw/oxfw-hwdep.c @@ -35,12 +35,12 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, } memset(&event, 0, sizeof(event)); + count = min_t(long, count, sizeof(event.lock_status)); if (oxfw->dev_lock_changed) { event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; event.lock_status.status = (oxfw->dev_lock_count > 0); oxfw->dev_lock_changed = false; - count = min_t(long, count, sizeof(event.lock_status)); } spin_unlock_irq(&oxfw->lock); -- 2.29.2

1 1

duplicate line at samsung speyside driver
by Kuninori Morimoto 22 Jan '21

22 Jan '21

Hi ALSA ML I noticed samsung speyside driver has duplicate line on audio_paths. I think it is typo or something... linux/sound/soc/samsung/speyside.c static struct snd_soc_dapm_route audio_paths[] = { => { "IN1RN", NULL, "MICB1" }, { "IN1RP", NULL, "MICB1" }, { "IN1RN", NULL, "MICB2" }, { "IN1RP", NULL, "MICB2" }, { "MICB1", NULL, "Headset Mic", speyside_get_micbias }, { "MICB2", NULL, "Headset Mic", speyside_get_micbias }, { "IN1LP", NULL, "MICB2" }, => { "IN1RN", NULL, "MICB1" }, { "MICB2", NULL, "Main AMIC" }, ... Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[Regression] Game "Rise of the Tomb Raider" crashes at launch
by GitHub issues - edited 21 Jan '21

21 Jan '21

alsa-project/alsa-lib issue #116 was edited from Oschowa: The the native linux version of the game "Rise of the Tomb Raider" crashes at launch when it doesn't detect a working pulseaudio server and it falls back to alsa-lib directly. This is a regression, the last working alsa-lib version is v1.2.1.2 and bisection yields 5ee5ef31b5ff3f as the first bad commit. The crash happens when using "real" alsa hardware drivers as well as when running on the pulseaudio-alsa or pipewire-alsa software plug. Happy to provide more info if i can. Issue URL : https://github.com/alsa-project/alsa-lib/issues/116 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[Regression] Game "Rise of the Tomb Raider" crashes at launch
by GitHub issues - opened 21 Jan '21

21 Jan '21

alsa-project/alsa-lib issue #116 was opened from Oschowa: The the native linux version of the game "Rise of the Tomb Raider" crashes at launch when it doesn't detect a working pulseaudio server and it falls back alsa-lib directly. This is a regression, the last working alsa-lib version is v1.2.1.2 and bisection yields 5ee5ef31b5ff3f as the first bad commit. The crash happens when using "real" alsa hardware drivers as well as when running on the pulseaudio-alsa or pipewire-alsa software plug. Happy to provide more info if i can. Issue URL : https://github.com/alsa-project/alsa-lib/issues/116 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

SOF: Improve dmic support
by GitHub pull_request - edited 21 Jan '21

21 Jan '21

alsa-project/alsa-ucm-conf pull request #75 was edited from plbossart: DMIC support seems incomplete for HDaudio (not handling all possibilities for cfg-dmics) and missing completely for SoundWire. This is an untested draft just to get UCM3 feedback from @perexg and tests from @aigilea on an HP Spectre x360 device. Edit: adding @kv2019i @RanderWang @bardliao and @libinyang for reviews. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/75 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/75.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

SOF: Improve dmic support
by GitHub pull_request - edited 21 Jan '21

21 Jan '21

alsa-project/alsa-ucm-conf pull request #75 was edited from plbossart: DMIC support seems incomplete for HDaudio (not handling all possibilities for cfg-dmics) and missing completely for SoundWire. This is an untested draft just to get UCM3 feedback from @perexg and tests from @aigilea on an HP Spectre x360 device. Edit: adding @libinyang for reviews. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/75 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/75.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

SOF: Improve dmic support
by GitHub pull_request - opened 21 Jan '21

21 Jan '21

alsa-project/alsa-ucm-conf pull request #75 was opened from plbossart: DMIC support seems incomplete for HDaudio (not handling all possibilities for cfg-dmics) and missing completely for SoundWire. This is an untested draft just to get UCM3 feedback from @perexg and tests from @aigilea on an HP Spectre x360 device. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/75 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/75.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Support for NUX MG-300 USB interface
by Mike Oliphant 21 Jan '21

21 Jan '21

I recently got a NUX MG-300 USB guitar fx interface with high hopes that it would work in Linux. It is supposed to be class-compliant, and works in iOS and macOS without a driver. Unfortunately, while it is recognized as a USB audio device by Linux, it is not working. The core issue seems to be that "stream0" reports no playback rates. dmesg also shows errors, including "__uac_clock_find_source(): selector reported illegal value". Any help identifying the issue would be much appreciated. I have some experience troubleshooting the USB audio kernel code and am happy to try doing so if I can get a nudge in the right direction. Output from "stream0", dmesg, and lsusb follows. Thanks! Mike "stream0" --- NUX NUX MG-300 AUDIO at usb-0000:00:1a.7-1.3, high speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 2 Endpoint: 0x01 (1 OUT) (ASYNC) Rates: Data packet interval: 1000 us Bits: 24 Channel map: FL FR Sync Endpoint: 0x81 (1 IN) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: No Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 0x82 (2 IN) (ASYNC) Rates: Data packet interval: 1000 us Bits: 24 Channel map: FL FR --- dmesg log: --- [ 141.265806] usb 3-1.3: new high-speed USB device number 9 using ehci-pci [ 141.374351] usb 3-1.3: config 1 interface 3 altsetting 0 bulk endpoint 0x4 has invalid maxpacket 256 [ 141.374370] usb 3-1.3: config 1 interface 3 altsetting 0 bulk endpoint 0x83 has invalid maxpacket 256 [ 141.375373] usb 3-1.3: New USB device found, idVendor=1fc9, idProduct=8260, bcdDevice= 1.00 [ 141.375389] usb 3-1.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 141.375396] usb 3-1.3: Product: NUX MG-300 AUDIO [ 141.375401] usb 3-1.3: Manufacturer: NUX [ 141.375405] usb 3-1.3: SerialNumber: 2008101346 [ 141.468523] usb 3-1.3: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [ 141.468531] usb 3-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -22) [ 141.468771] usb 3-1.3: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [ 141.468897] usb 3-1.3: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [ 141.469274] usb 3-1.3: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [ 141.469279] usb 3-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -22) [ 141.469521] usb 3-1.3: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [ 141.469646] usb 3-1.3: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [ 141.472645] usbcore: registered new interface driver snd-usb-audio --- lsusb output: --- Bus 003 Device 009: ID 1fc9:8260 NXP Semiconductors Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x1fc9 NXP Semiconductors idProduct 0x8260 bcdDevice 1.00 iManufacturer 1 iProduct 2 iSerial 3 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x013e bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 2mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 0x0065 bmControls 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 41 bmAttributes 3 Internal programmable clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bClockID 40 bNrInPins 1 baCSourceID(0) 41 bmControls 0x03 Clock Selector Control (read/write) iClockSelector 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bmaControls(0) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls(1) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls(2) 0x0000000f Mute Control (read/write) Volume Control (read/write) iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 20 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 10 bCSourceID 40 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 22 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 2 bCSourceID 40 bmControls 0x0000 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0258 1x 600 bytes bInterval 4 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0258 1x 600 bytes bInterval 4 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 4 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0041 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 ---

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[PATCH 0/5] Add KUNIT tests for ASoC topology
by Amadeusz Sławiński 21 Jan '21

21 Jan '21

This series adds unit tests for ASoC topology. First fix problems found when developing and running test cases and then add tests implementation. Tests themselves are quite simple and just call snd_soc_tplg_component_load() with various parameters and check the result. Tests themselves are described in more detail in commits adding them. Goal is to expand the amount of test cases in following patches. Prerequisity for this patchset are 2 patches which have already been sent: https://lore.kernel.org/alsa-devel/20210114163602.911205-1-amadeuszx.slawin… Description on how typical test case itself works: In order to load topology we need to have 3 things: card, codec component & platform component. In typical test case we register card and platform component and bind to dummy codec. There are of course execeptions, when we want to test behaviour of topology API when component or card is missing. Note that this is bit different from typical scenario (in SOF and skylake drivers) where card is registered by machine driver and component by platform driver, as we register both when setting up test. If you check the test case most of them have similar architecture of: 1. /* run test */ ret = snd_soc_register_card(&kunit_comp->card); if (ret != 0 && ret != -EPROBE_DEFER) KUNIT_FAIL(test, "Failed to register card"); 2. ret = snd_soc_component_initialize(&kunit_comp->comp, &test_component, test_dev); KUNIT_EXPECT_EQ(test, 0, ret); 3. ret = snd_soc_add_component(&kunit_comp->comp, NULL, 0); KUNIT_EXPECT_EQ(test, 0, ret); Ad. 1. First we register card, which in most tests returns -EPROBE_DEFER (from snd_soc_bind_card()), as platform component is not yet created. I test for both 0 and -EPROBE_DEFER, as it makes it easier to reshuffle this code around if needed and there is one test case which does it in different order. Ad. 2. Then we initialize platform component with structure pointing at proper probe function, which calls snd_soc_tplg_component_load() with test parameters and checks expected result. Ad. 3. And then in follow up we call snd_soc_add_component() which creates platform component for us and calls snd_soc_try_rebind_card() which if everything is bound properly calls previously set probe function. Amadeusz Sławiński (5): ASoC: topology: Properly unregister DAI on removal Revert "ASoC: soc-devres: add devm_snd_soc_register_dai()" ASoC: topology: KUnit: Add KUnit tests passing various arguments to snd_soc_tplg_component_load ASoC: topology: KUnit: Add KUnit tests passing empty topology with variants to snd_soc_tplg_component_load ASoC: topology: KUnit: Add KUnit tests passing topology with PCM to snd_soc_tplg_component_load include/sound/soc.h | 4 - sound/soc/Kconfig | 17 + sound/soc/Makefile | 5 + sound/soc/soc-devres.c | 37 -- sound/soc/soc-topology-test.c | 843 ++++++++++++++++++++++++++++++++++ sound/soc/soc-topology.c | 9 +- 6 files changed, 870 insertions(+), 45 deletions(-) create mode 100644 sound/soc/soc-topology-test.c -- 2.25.1

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[PATCH] ASoC: rt5682: remove connection with LDO2 in DAPM graph
by shumingf＠realtek.com 21 Jan '21

21 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> The application circuit shall provide MICVDD power. In default, the codec driver doesn't need to enable LDO2. In case, a board wants to use VBAT for micbias, it should add a DAPM route which IN1P connects with LDO2 in the machine driver. e.g. { "IN1P", NULL, "LDO2" }, Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 42ad5bea86b5..185906d6d4e0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1854,8 +1854,6 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"CLKDET SYS", NULL, "CLKDET"}, - {"IN1P", NULL, "LDO2"}, - {"BST1 CBJ", NULL, "IN1P"}, {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, -- 2.29.0

2 1

[PATCH v2 1/2] ASoC: Intel: Skylake: skl-topology: Fix OOPs ib skl_tplg_complete
by Ricardo Ribalda 21 Jan '21

21 Jan '21

If dobj->control is not initialized we end up in an OOPs during skl_tplg_complete: [ 26.553358] BUG: kernel NULL pointer dereference, address: 0000000000000078 [ 26.561151] #PF: supervisor read access in kernel mode [ 26.566897] #PF: error_code(0x0000) - not-present page [ 26.572642] PGD 0 P4D 0 [ 26.575479] Oops: 0000 [#1] PREEMPT SMP PTI [ 26.580158] CPU: 2 PID: 2082 Comm: udevd Tainted: G C 5.4.81 #4 [ 26.588232] Hardware name: HP Soraka/Soraka, BIOS Google_Soraka.10431.106.0 12/03/2019 [ 26.597082] RIP: 0010:skl_tplg_complete+0x70/0x144 [snd_soc_skl] Fixes: 2d744ecf2b98 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHL") Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> --- sound/soc/intel/skylake/skl-topology.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ae466cd59292..1ef30ca45410 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3619,15 +3619,16 @@ static void skl_tplg_complete(struct snd_soc_component *component) list_for_each_entry(dobj, &component->dobj_list, list) { struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se = - (struct soc_enum *)kcontrol->private_value; - char **texts = dobj->control.dtexts; + struct soc_enum *se; + char **texts; char chan_text[4]; - if (dobj->type != SND_SOC_DOBJ_ENUM || - dobj->control.kcontrol->put != - skl_tplg_multi_config_set_dmic) + if (dobj->type != SND_SOC_DOBJ_ENUM || !kcontrol || + kcontrol->put != skl_tplg_multi_config_set_dmic) continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { -- 2.30.0.296.g2bfb1c46d8-goog

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[PATCH 0/2] Add sanity checks for topology API calls
by Amadeusz Sławiński 21 Jan '21

21 Jan '21

Topology API exposes just 2 function calls, to load and unload topology. This adds sanity checks to make sure that it behaves well when some of parameters are set incoorectly or not needed. This makes developer live easier by failing early instead of proceeding on and then failing in unexpected ways. As loading and unloading topology usually happens one the overhead of additional checks should be negligible. Amadeusz Sławiński (2): ASoC: topology: Ensure that needed parameters are set ASoC: topology: Check if ops is set before dereference sound/soc/soc-topology.c | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) -- 2.25.1

2 3

[PATCH 00/44] ASoC: sync parameter naming : rate / sample_bits
by Kuninori Morimoto 21 Jan '21

21 Jan '21

Hi Mark snd_pcm_runtime / snd_soc_dai / snd_soc_dai_driver / snd_soc_dai_link have related parameter which is similar but not same naming. struct snd_pcm_runtime { ... (A) unsigned int rate; ... (B) unsigned int sample_bits; ... }; struct snd_soc_dai { ... (A) unsigned int rate; (B) unsigned int sample_bits; ... }; struct snd_soc_dai_driver { ... (A) unsigned int symmetric_rates:1; (B) unsigned int symmetric_samplebits:1; ... }; struct snd_soc_dai_link { ... (A) unsigned int symmetric_rates:1; (B) unsigned int symmetric_samplebits:1; ... }; Because it is similar but not same naming rule, code can be verbose / can't share macro. This patch-set sync naming rule to struct snd_pcm_runtime base. - xxx_rates; + xxx_rate; - xxx_samplebits; + xxx_sample_bits; Kuninori Morimoto (44): ASoC: soc-pcm: revert soc_pcm_apply_symmetry() ASoC: sync parameter naming : rate / sample_bits ASoC: adi: sync parameter naming (rate/sample_bits) ASoC: atmel: sync parameter naming (rate/sample_bits) ASoC: au1x: sync parameter naming (rate/sample_bits) ASoC: bcm: sync parameter naming (rate/sample_bits) ASoC: cirrus: sync parameter naming (rate/sample_bits) ASoC: tegra: sync parameter naming (rate/sample_bits) ASoC: rockchip: sync parameter naming (rate/sample_bits) ASoC: samsung: sync parameter naming (rate/sample_bits) ASoC: sh: sync parameter naming (rate/sample_bits) ASoC: ti: sync parameter naming (rate/sample_bits) ASoC: pxa: sync parameter naming (rate/sample_bits) ASoC: mediatek: sync parameter naming (rate/sample_bits) ASoC: fsl: sync parameter naming (rate/sample_bits) ASoC: wm*: sync parameter naming (rate/sample_bits) ASoC: tlv*: sync parameter naming (rate/sample_bits) ASoC: rt*: sync parameter naming (rate/sample_bits) ASoC: nau*: sync parameter naming (rate/sample_bits) ASoC: tas*: sync parameter naming (rate/sample_bits) ASoC: da*: sync parameter naming (rate/sample_bits) ASoC: es*: sync parameter naming (rate/sample_bits) ASoC: max*: sync parameter naming (rate/sample_bits) ASoC: cs*: sync parameter naming (rate/sample_bits) ASoC: tscs*: sync parameter naming (rate/sample_bits) ASoC: ak*: sync parameter naming (rate/sample_bits) ASoC: adau*: sync parameter naming (rate/sample_bits) ASoC: jz4740: sync parameter naming (rate/sample_bits) ASoC: sunxi: sync parameter naming (rate/sample_bits) ASoC: zl38060: sync parameter naming (rate/sample_bits) ASoC: ssm2602: sync parameter naming (rate/sample_bits) ASoC: sgtl5000: sync parameter naming (rate/sample_bits) ASoC: mc13783: sync parameter naming (rate/sample_bits) ASoC: mt6660: sync parameter naming (rate/sample_bits) ASoC: lochnagar-sc: sync parameter naming (rate/sample_bits) ASoC: ab8500: sync parameter naming (rate/sample_bits) ASoC: alc5632: sync parameter naming (rate/sample_bits) ASoC: cx2072x: sync parameter naming (rate/sample_bits) ASoC: inno_rk3036: sync parameter naming (rate/sample_bits) ASoC: lm49453: sync parameter naming (rate/sample_bits) ASoC: ml26124: sync parameter naming (rate/sample_bits) ASoC: soc-dai.h: remove symmetric_rates/samplebits ASoC: soc-pcm: cleanup soc_pcm_apply_symmetry() ASoC: soc-pcm: cleanup soc_pcm_params_symmetry() include/sound/soc-dai.h | 4 +- include/sound/soc.h | 4 +- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 4 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/bcm/bcm2835-i2s.c | 4 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/ab8500-codec.c | 4 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/adau1373.c | 6 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/ak4554.c | 2 +- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/ak4641.c | 4 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4234.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/cs43130.c | 6 +- sound/soc/codecs/cs4341.c | 2 +- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs47l15.c | 12 ++-- sound/soc/codecs/cs47l24.c | 12 ++-- sound/soc/codecs/cs47l35.c | 12 ++-- sound/soc/codecs/cs47l85.c | 16 ++--- sound/soc/codecs/cs47l90.c | 16 ++--- sound/soc/codecs/cs47l92.c | 12 ++-- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/cx2072x.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da7218.c | 4 +- sound/soc/codecs/da7219.c | 4 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/jz4740.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/lochnagar-sc.c | 12 ++-- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/mt6660.c | 4 +- sound/soc/codecs/nau8810.c | 2 +- sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 4 +- sound/soc/codecs/rt298.c | 4 +- sound/soc/codecs/rt5670.c | 4 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/ssm2602.c | 4 +- sound/soc/codecs/tas2764.c | 2 +- sound/soc/codecs/tas2770.c | 2 +- sound/soc/codecs/tlv320adcx140.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 4 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tscs42xx.c | 4 +- sound/soc/codecs/tscs454.c | 12 ++-- sound/soc/codecs/wm5102.c | 12 ++-- sound/soc/codecs/wm5110.c | 12 ++-- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm8997.c | 8 +-- sound/soc/codecs/wm8998.c | 12 ++-- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/zl38060.c | 4 +- sound/soc/fsl/fsl_esai.c | 4 +- sound/soc/fsl/fsl_sai.c | 8 +-- sound/soc/fsl/fsl_ssi.c | 4 +- sound/soc/jz4740/jz4740-i2s.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 10 +-- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 8 +-- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-dai-pcm.c | 8 +-- sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 8 +-- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 4 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/sh/rcar/core.c | 4 +- sound/soc/soc-pcm.c | 80 +++++++--------------- sound/soc/soc-topology.c | 8 +-- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 12 ++-- sound/soc/tegra/tegra186_dspk.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra210_dmic.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- 116 files changed, 253 insertions(+), 283 deletions(-) -- 2.25.1

2 45

[RESEND PATCH v6 0/6] Tegra210 audio graph card
by Sameer Pujar 21 Jan '21

21 Jan '21

This series adds audio graph based sound card support for Tegra210 platforms like Jetson-TX1 an Jetson-Nano. The following preparatory audio graph enhancement series is already merged. * https://patchwork.kernel.org/project/alsa-devel/list/?series=375629&state=* Following are the summary of changes: * Add graph/audio-graph based schemas or schema updates for Tegra210 component and machine drivers. * Add Tegra audio graph machine driver. * Add required DT support for Jetson-TX1/Nano. This work is based on earlier discussion of DPCM usage for Tegra and simple card driver updates. * https://lkml.org/lkml/2020/4/30/519 * https://lkml.org/lkml/2020/6/27/4 Original v6 series was sent about 6-7 weeks back. The dependency commit, https://lore.kernel.org/alsa-devel/1610948585-16286-1-git-send-email-spujar… is now merged. Resending this now to appear in the top of the mail list. Changelog ========= v5 -> v6 -------- * Added ports or port description in YAML docs for Tegra AHUB devices and graph card in patch 1/6 and 2/6. Reference of audio-graph-port.yaml is used for AHUB devices. * Dropped redundant NULL check return for of_device_get_match_data() in patch 3/6. * Added 'Reviewed-by' tag from Jon Hunter. * No changes in remaining patches. v4 -> v5 -------- * Audio graph related changes were sent in separate v5 series as mentioned above and are dropped from current series. * Graph and audio graph doc patches are dropped from this series and are sent separately as mentioned above. * Minor change with phandle label for TX1 and Nano platform DT files. * No changes in other patches. v3 -> v4 -------- * Added new patches to convert graph.txt and audio-graph-card.txt to corresponding json-schema files. Later these references are used in Tegra audio graph schema. * AHUB component binding docs are updated to reflect the usage of ports/port/endpoint * More common stuff is moved into graph_parse_of() and this is used by both generic and Tegra audio graph. * DT binding for Tegra audio graph is updated to included "ports { }" * As per the suggestion 'void *data' member is dropped from 'asoc_simple_priv' and instead container method is used to maintain required custom data internal to Tegra audio graph. v2 -> v3 -------- * Dropped new compatible addition in generic graph driver after reviewing it with Morimoto-san. Instead added Tegra audio graph driver and new compatibles are added in the same. * Added new patches to expose new members for customization in audio graph driver. * Added new patch for Tegra audio graph driver and related documentation. * Minor change in below commit where mutex version of helper is used "ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM" * DT binding is updated to use the newly exposed compatibles * No changes in other patches v1 -> v2 -------- * Re-organized ports/endpoints description for ADMAIF and XBAR. Updated DT patches accordingly. * After above change, multiple Codec endpoint support is not required and hence dropped for now. This will be considered separately if at all required in future. * Re-ordered patches in the series. Sameer Pujar (6): ASoC: dt-bindings: tegra: Add graph bindings ASoC: dt-bindings: tegra: Add json-schema for Tegra audio graph card ASoC: tegra: Add audio graph based card driver arm64: defconfig: Enable Tegra audio graph card driver arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 .../sound/nvidia,tegra-audio-graph-card.yaml | 187 +++++++++++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 18 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 13 +- .../bindings/sound/nvidia,tegra210-ahub.yaml | 13 +- .../bindings/sound/nvidia,tegra210-dmic.yaml | 18 +- .../bindings/sound/nvidia,tegra210-i2s.yaml | 18 +- .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 153 ++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 262 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 146 ++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_audio_graph_card.c | 251 ++++++++++++++++++++ 13 files changed, 1085 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c -- 2.7.4

3 15

[PATCH v3 1/2] ASoC: Intel: Skylake: skl-topology: Fix OOPs ib skl_tplg_complete
by Ricardo Ribalda 21 Jan '21

21 Jan '21

If dobj->control is not initialized we end up in an OOPs during skl_tplg_complete: [ 26.553358] BUG: kernel NULL pointer dereference, address: 0000000000000078 [ 26.561151] #PF: supervisor read access in kernel mode [ 26.566897] #PF: error_code(0x0000) - not-present page [ 26.572642] PGD 0 P4D 0 [ 26.575479] Oops: 0000 [#1] PREEMPT SMP PTI [ 26.580158] CPU: 2 PID: 2082 Comm: udevd Tainted: G C 5.4.81 #4 [ 26.588232] Hardware name: HP Soraka/Soraka, BIOS Google_Soraka.10431.106.0 12/03/2019 [ 26.597082] RIP: 0010:skl_tplg_complete+0x70/0x144 [snd_soc_skl] Cc: <stable(a)vger.kernel.org> Fixes: 2d744ecf2b98 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHL") Tested-by: Lukasz Majczak <lma(a)semihalf.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> --- v3: order local variables by length sound/soc/intel/skylake/skl-topology.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ae466cd59292..ffd37aaecdf1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3619,15 +3619,16 @@ static void skl_tplg_complete(struct snd_soc_component *component) list_for_each_entry(dobj, &component->dobj_list, list) { struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se = - (struct soc_enum *)kcontrol->private_value; - char **texts = dobj->control.dtexts; + struct soc_enum *se; char chan_text[4]; + char **texts; - if (dobj->type != SND_SOC_DOBJ_ENUM || - dobj->control.kcontrol->put != - skl_tplg_multi_config_set_dmic) + if (dobj->type != SND_SOC_DOBJ_ENUM || !kcontrol || + kcontrol->put != skl_tplg_multi_config_set_dmic) continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { -- 2.30.0.296.g2bfb1c46d8-goog

1 1

pcm: pcm_ioplug - fix the avail_update mmap capture copy issue
by GitHub pull_request - opened 21 Jan '21

21 Jan '21

alsa-project/alsa-lib pull request #115 was opened from perexg: It seems that the commit "pcm: ioplug: Transfer all available data" introduced new regressions (wrong memory access). The second issue is that the avail_update in ioplug does not move appl_ptr nor hw_ptr, so it's possible that the transfers may be repetitive. This patch moves the transfer calls to mmap_begin callback where it should be. The pointer wraps are handled by design now. Fixes: 1714332719 ("pcm: ioplug: Transfer all available data") BugLink: https://github.com/alsa-project/alsa-lib/pull/103 Request URL : https://github.com/alsa-project/alsa-lib/pull/115 Patch URL : https://github.com/alsa-project/alsa-lib/pull/115.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RFC][PATCH v4 0/4] design a way to change audio Jack state by software
by Hui Wang 21 Jan '21

21 Jan '21

the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (4): alsa: jack: implement software jack injection via debugfs alsa: jack: adjust jack_kctl debugfs folder's name alsa: jack: add more jack_kctl debugfs nodes alsa: jack: implement save-and-restore for jack's hw status include/sound/core.h | 2 + include/sound/jack.h | 1 + sound/core/init.c | 7 + sound/core/jack.c | 352 ++++++++++++++++++++++++++++++++++++++++++- sound/core/sound.c | 8 + 5 files changed, 366 insertions(+), 4 deletions(-) -- 2.25.1

2 10

[PATCH] ALSA: control: remove limitation on the number of user-defined control element set per card
by Takashi Sakamoto 21 Jan '21

21 Jan '21

ALSA control core allows usespace application to register control element set by call of ioctl(2) with SNDRV_CTL_IOCTL_ELEM_ADD request. The added control element set is called as 'user-defined'. Currently sound card has limitation on the number of the user-defined control element set up to 32. The limitation is inconvenient to drivers in ALSA firewire stack since the drivers expect userspace applications to implement function to control device functionalities such as mixing and routing. As the userspace application, snd-firewire-ctl-services project starts: https://github.com/alsa-project/snd-firewire-ctl-services/ The project supports many devices supported by ALSA firewire stack. The limitation is mostly good since routing and mixing controls can be represented by control element set, which includes control element with the same parameters. Nevertheless, it's actually inconvenient to device which has many varied functionalities. For example, plugin effect such as channel strip and reverb has many parameters. For the case, many control elements are required to configure the parameters and control element set cannot aggregates controls for the parameters. At present, below models are implemented with the control elements and actually add control element sets over 32: * Apogee Emsemble (snd-bebob-ctl-service) * TC Electronic Konnekt 24d (snd-dice-ctl-service) * TC Electronic Studio Konnekt 48 (snd-dice-ctl-service) * TC Electronic Konnekt Live (snd-dice-ctl-service) * TC Electronic Impact Twin (snd-dice-ctl-service) It could be investigated to increase the number; e.g. quadruple to the current (=128), however it's hard to find criteria about the number for existent sound card. This commit just removes the limitation for the reason. ALSA control core uses UINT_MAX as the maximum number of control elements added to a sound card. It's limitation for both in-kernel driver and userspace application. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/core.h | 1 - sound/core/control.c | 13 ------------- 2 files changed, 14 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 0462c577d7a3..6b443ce0acf8 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -100,7 +100,6 @@ struct snd_card { struct rw_semaphore controls_rwsem; /* controls list lock */ rwlock_t ctl_files_rwlock; /* ctl_files list lock */ int controls_count; /* count of all controls */ - int user_ctl_count; /* count of all user controls */ struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ diff --git a/sound/core/control.c b/sound/core/control.c index 5165741a8400..1bcef22713f4 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -18,8 +18,6 @@ #include <sound/info.h> #include <sound/control.h> -/* max number of user-defined controls */ -#define MAX_USER_CONTROLS 32 #define MAX_CONTROL_COUNT 1028 struct snd_kctl_ioctl { @@ -539,7 +537,6 @@ static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file, ret = snd_ctl_remove(card, kctl); if (ret < 0) goto error; - card->user_ctl_count--; error: up_write(&card->controls_rwsem); return ret; @@ -1435,13 +1432,6 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, return err; } - /* - * The number of userspace controls are counted control by control, - * not element by element. - */ - if (card->user_ctl_count + 1 > MAX_USER_CONTROLS) - return -ENOMEM; - /* Check the number of elements for this userspace control. */ count = info->owner; if (count == 0) @@ -1534,9 +1524,6 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, * applications because the field originally means PID of a process * which locks the element. */ - - card->user_ctl_count++; - unlock: up_write(&card->controls_rwsem); return err; -- 2.27.0

3 3

[PATCH 1/6] ASoC: fsl_ssi: Use of_device_get_match_data()
by Fabio Estevam 21 Jan '21

21 Jan '21

The retrieval of driver data via of_device_get_match_data() can make the code simpler. Use of_device_get_match_data() to simplify the code. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 404be27c15fe..db4ba5f22b77 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1397,18 +1397,11 @@ static int fsl_ssi_probe_from_dt(struct fsl_ssi *ssi) { struct device *dev = ssi->dev; struct device_node *np = dev->of_node; - const struct of_device_id *of_id; const char *p, *sprop; const __be32 *iprop; u32 dmas[4]; int ret; - of_id = of_match_device(fsl_ssi_ids, dev); - if (!of_id || !of_id->data) - return -EINVAL; - - ssi->soc = of_id->data; - ret = of_property_match_string(np, "clock-names", "ipg"); /* Get error code if not found */ ssi->has_ipg_clk_name = ret >= 0; @@ -1492,6 +1485,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) return -ENOMEM; ssi->dev = dev; + ssi->soc = of_device_get_match_data(&pdev->dev); /* Probe from DT */ ret = fsl_ssi_probe_from_dt(ssi); -- 2.17.1

2 6

[PATCH 0/2] ASoC: qcom: Fix broken lpass driver
by Srinivas Kandagatla 21 Jan '21

21 Jan '21

LPASS driver is partially broken on DragonBoard DB410c on 5.10 and its totally broken on other Supported Qualcomm SoCs. This was due to DAI ids being over written by the SoC specific header files in the dt-bindings. Idea of having SoC specific headers is not doable when we are dealing with a common driver. So this patchset attempts to fix this properly by creating a common dt-bindings header for lpass which can be updated with new entries if required. This patchset also add an simple of_xlate function to resolve the dai names and different SoCs might not have 1:1 mapping for the dai_driver array with dai ids. Srinivas Kandagatla (2): ASoC: dt-bindings: lpass: Fix and common up lpass dai ids ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY include/dt-bindings/sound/apq8016-lpass.h | 7 +++---- include/dt-bindings/sound/qcom,lpass.h | 15 +++++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 6 ++---- sound/soc/qcom/lpass-cpu.c | 22 ++++++++++++++++++++++ sound/soc/qcom/lpass-platform.c | 12 ++++++++++++ sound/soc/qcom/lpass-sc7180.c | 6 +++--- sound/soc/qcom/lpass.h | 2 +- 7 files changed, 58 insertions(+), 12 deletions(-) create mode 100644 include/dt-bindings/sound/qcom,lpass.h -- 2.21.0

3 5

[PATCH v2 0/2] ASoC: qcom: Fix broken lpass driver
by Srinivas Kandagatla 21 Jan '21

21 Jan '21

LPASS driver is partially broken on DragonBoard DB410c on 5.10 and its totally broken on other Supported Qualcomm SoCs. This was due to DAI ids being over written by the SoC specific header files in the dt-bindings. Idea of having SoC specific headers is not doable when we are dealing with a common driver. So this patchset attempts to fix this properly by creating a common dt-bindings header for lpass which can be updated with new entries if required. This patchset also add an simple of_xlate function to resolve the dai names and different SoCs might not have 1:1 mapping for the dai_driver array with dai ids. Changes since v1: - removed array indexes as suggested by Stephan G. - rebased to sound/for-next branch - collected Srinivasa tested-by tag for sc7180 platform. Thanks, srini Srinivas Kandagatla (2): ASoC: dt-bindings: lpass: Fix and common up lpass dai ids ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY include/dt-bindings/sound/apq8016-lpass.h | 7 +++---- include/dt-bindings/sound/qcom,lpass.h | 15 +++++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 6 ++---- sound/soc/qcom/lpass-cpu.c | 22 ++++++++++++++++++++++ sound/soc/qcom/lpass-platform.c | 12 ++++++++++++ sound/soc/qcom/lpass-sc7180.c | 9 +++------ sound/soc/qcom/lpass.h | 2 +- 7 files changed, 58 insertions(+), 15 deletions(-) create mode 100644 include/dt-bindings/sound/qcom,lpass.h -- 2.21.0

3 4

[PATCH] ASoC: qcom: lpass-ipq806x: fix bitwidth regmap field
by Srinivas Kandagatla 21 Jan '21

21 Jan '21

BIT_WIDTH field in I2S_CTL register is two bits wide, however recent regmap field conversion patch trimmed it down to one bit. Fix this by correcting the bit range! Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-ipq806x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-ipq806x.c b/sound/soc/qcom/lpass-ipq806x.c index 92f98b4df47f..ef8a7984f232 100644 --- a/sound/soc/qcom/lpass-ipq806x.c +++ b/sound/soc/qcom/lpass-ipq806x.c @@ -131,7 +131,7 @@ static struct lpass_variant ipq806x_data = { .micmode = REG_FIELD_ID(0x0010, 4, 7, 5, 0x4), .micmono = REG_FIELD_ID(0x0010, 3, 3, 5, 0x4), .wssrc = REG_FIELD_ID(0x0010, 2, 2, 5, 0x4), - .bitwidth = REG_FIELD_ID(0x0010, 0, 0, 5, 0x4), + .bitwidth = REG_FIELD_ID(0x0010, 0, 1, 5, 0x4), .rdma_dyncclk = REG_FIELD_ID(0x6000, 12, 12, 4, 0x1000), .rdma_bursten = REG_FIELD_ID(0x6000, 11, 11, 4, 0x1000), -- 2.21.0

2 1

[PATCH 0/5] ASoC: mediatek: mt8192-mt6359: support DP audio
by Tzung-Bi Shih 21 Jan '21

21 Jan '21

The 1st and 2nd patches refactor the machine driver. The 3rd patch changes the platform driver to support TDM 8 channel output. The 4th patch adds an optional DT property. The 5th patch makes the machine driver support DP audio if the optional DT property is specified. Tzung-Bi Shih (5): ASoC: mediatek: mt8192-mt6359: move headset_jack to card specific data ASoC: mediatek: mt8192-mt6359: simplify mt8192_rt5682_init ASoC: mediatek: mt8192: change mclk_multiple of TDM from 128 to 512 ASoC: dt-bindings: mt8192-mt6359: add hdmi-codec property ASoC: mediatek: mt8192-mt6359: support audio over DP .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 5 ++ sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 2 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 54 ++++++++++++++++--- 3 files changed, 52 insertions(+), 9 deletions(-) -- 2.30.0.284.gd98b1dd5eaa7-goog

2 6

[PATCH 0/2] ASoC: mediatek: mt8183: ignore TDM DAI link by by default
by Tzung-Bi Shih 21 Jan '21

21 Jan '21

hdmi-codec is an optional property. The 2 patches fix DAI link binding error when the property doesn't exist in DTS. Tzung-Bi Shih (2): ASoC: mediatek: mt8183-mt6358: ignore TDM DAI link by default ASoC: mediatek: mt8183-da7219: ignore TDM DAI link by default sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 5 ++++- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 ++++- 2 files changed, 8 insertions(+), 2 deletions(-) -- 2.30.0.284.gd98b1dd5eaa7-goog

2 3

[PATCH] ASoC: topology: Fix memory corruption in soc_tplg_denum_create_values()
by Dan Carpenter 21 Jan '21

21 Jan '21

The allocation uses sizeof(u32) when it should use sizeof(unsigned long) so it leads to memory corruption later in the function when the data is initialized. Fixes: 5aebe7c7f9c2 ("ASoC: topology: fix endianness issues") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This is from static analysis, not from testing. Obviously we don't want memory corruption, so my patch is an improvement. But I feel like a better approach might be to change the type of dvalues[] to u32. I took the less risky approach because I'm not an expert and can't test it. But if someone else can take a look at it, then I'll redo the patch. sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 950c45008e24..37a5d73e643b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -902,7 +902,7 @@ static int soc_tplg_denum_create_values(struct soc_tplg *tplg, struct soc_enum * return -EINVAL; se->dobj.control.dvalues = devm_kcalloc(tplg->dev, le32_to_cpu(ec->items), - sizeof(u32), + sizeof(*se->dobj.control.dvalues), GFP_KERNEL); if (!se->dobj.control.dvalues) return -ENOMEM; -- 2.29.2

3 2

[PATCH] ASoC: AMD Renoir - refine DMI entries for some Lenovo products
by Jaroslav Kysela 21 Jan '21

21 Jan '21

Apparently, the DMI board name LNVNB161216 is also used also for products with the digital microphones connected to the AMD's audio bridge. Refine the DMI table - use product name identifiers extracted from https://bugzilla.redhat.com/show_bug.cgi?id=1892115 . The report for Lenovo Yoga Slim 7 14ARE05 (82A2) is in buglink. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211299 Cc: <stable(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index deca8c7a0e87..050a61fe9693 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -165,10 +165,24 @@ static int rn_acp_deinit(void __iomem *acp_base) static const struct dmi_system_id rn_acp_quirk_table[] = { { - /* Lenovo IdeaPad Flex 5 14ARE05, IdeaPad 5 15ARE05 */ + /* Lenovo IdeaPad S340-14API */ .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_EXACT_MATCH(DMI_BOARD_NAME, "LNVNB161216"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "81NB"), + } + }, + { + /* Lenovo IdeaPad Flex 5 14ARE05 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "81X2"), + } + }, + { + /* Lenovo IdeaPad 5 15ARE05 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "81YQ"), } }, { -- 2.29.2

2 1

[PATCH 0/3] ASoC: Simplify with dma_set_mask_and_coherent()
by Takashi Iwai 21 Jan '21

21 Jan '21

Simple conversions from two dma_set_mask*() calls into the single dma_set_mask_and_coherent(). Just a minor code refactoring, no functional changes. Takashi === Takashi Iwai (3): ASoC: intel: skl: Simplify with dma_set_mask_and_coherent() ASoC: SOF: intel: Simplify with dma_set_mask_and_coherent() ASoC: tegra: Simplify with dma_set_mask_and_coherent() sound/soc/intel/skylake/skl.c | 8 ++------ sound/soc/sof/intel/hda.c | 8 ++------ sound/soc/tegra/tegra_pcm.c | 6 +----- 3 files changed, 5 insertions(+), 17 deletions(-) -- 2.26.2

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[PATCH v7 1/2] cros_ec_commands: Add EC_CODEC_I2S_RX_RESET
by Yu-Hsuan Hsu 21 Jan '21

21 Jan '21

Add the new command EC_CODEC_I2S_RX_RESET in ec_codec_i2s_rx_subcmd, which is used for resetting the EC codec. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- include/linux/platform_data/cros_ec_commands.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 86376779ab31..95889ada83a3 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4600,6 +4600,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; -- 2.30.0.296.g2bfb1c46d8-goog

3 4

[PATCH 1/2] ALSA: usb-audio: Skip the clock selector inquiry for single connections
by Takashi Iwai 20 Jan '21

20 Jan '21

The current USB-audio driver gets an error at probing NUX MG-300 about parsing the clocks. This is because the firmware doesn't return the proper connection of the clock selector that is connected to a single clock; it's likely that the firmware was lazy^w optimized and the inquiry wasn't handled. Actually it makes little sense to inquire and set up the single connection explicitly. This patch fixes the issue by simply skipping the clock selector inquiry if it's a single connection. Reported-by: Mike Oliphant <oliphant(a)nostatic.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index dc68ed65e478..8243652d5604 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -298,6 +298,11 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (selector) { int ret, i, cur; + if (selector->bNrInPins == 1) { + ret = 1; + goto find_source; + } + /* the entity ID we are looking for is a selector. * find out what it currently selects */ ret = uac_clock_selector_get_val(chip, selector->bClockID); @@ -314,6 +319,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return -EINVAL; } + find_source: cur = ret; ret = __uac_clock_find_source(chip, fmt, selector->baCSourceID[ret - 1], -- 2.26.2

1 1

[PATCH v3 0/5] MFD/ASoC: Add support for Intel Bay Trail boards with WM5102 codec
by Hans de Goede 20 Jan '21

20 Jan '21

Hi All, Here is v3 of my series to add support for Intel Bay Trail based devices which use a WM5102 codec for audio output/input. New in v3 is that the compile error when CONFIG_ACPI is unset should be gone. Everything else is the same as in v2, see the cover-letter of v2 below: This was developed and tested on a Lenovo Yoga Tablet 1051L. The biggest difference from v1 is that, based on all the discussions about the jack-detect stuff, I've decided to split this into 2 series. One series adding the basic support for this setup, which is this series. And a second series which reworks the extcon driver into an arizona-jackdet library and then modifies the codec drivers to use that directly, replacing the old separate extcon child-device and extcon-driver. The are 2 reasons for the split: 1. With the new jack-det rework, the series really address 2 separate (but related) enhancements. 2. I expect this series to be ready for merging, while the jack-det stuff likely will need a couple more revisions. Other then the split there are some minor changes addressing various review comments, see the individual patch changelogs. The MFD and ASoC parts can be merged independent from each-other (both must be merged to get working sound on these boards, but that is only a runtime dependency and a part missing won't have any bad side-effects). Or the entire series could be merged through the MFD tree if people prefer that. Regards, Hans

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[PATCH 0/2] ASoC: remove obsolete drivers
by Arnd Bergmann 20 Jan '21

20 Jan '21

From: Arnd Bergmann <arnd(a)arndb.de> A few Arm platforms are getting removed in v5.12, this removes the corresponding sound drivers. Link: https://lore.kernel.org/linux-arm-kernel/20210120124812.2800027-1-arnd@kern… Arnd Bergmann (2): ASoC: remove sirf prima/atlas drivers ASoC: remove zte zx drivers .../bindings/sound/sirf-audio-codec.txt | 17 - .../devicetree/bindings/sound/sirf-usp.txt | 27 - .../devicetree/bindings/sound/zte,tdm.txt | 30 - .../bindings/sound/zte,zx-aud96p22.txt | 24 - .../devicetree/bindings/sound/zte,zx-i2s.txt | 45 -- .../bindings/sound/zte,zx-spdif.txt | 27 - sound/soc/Kconfig | 2 - sound/soc/Makefile | 2 - sound/soc/codecs/Makefile | 4 - sound/soc/codecs/sirf-audio-codec.c | 575 ------------------ sound/soc/codecs/zx_aud96p22.c | 401 ------------ sound/soc/sirf/Kconfig | 21 - sound/soc/sirf/Makefile | 8 - sound/soc/sirf/sirf-audio-port.c | 86 --- sound/soc/sirf/sirf-audio.c | 160 ----- sound/soc/sirf/sirf-usp.c | 435 ------------- sound/soc/sirf/sirf-usp.h | 292 --------- sound/soc/zte/Kconfig | 26 - sound/soc/zte/Makefile | 4 - sound/soc/zte/zx-i2s.c | 452 -------------- sound/soc/zte/zx-spdif.c | 363 ----------- sound/soc/zte/zx-tdm.c | 458 -------------- 22 files changed, 3459 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt delete mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,tdm.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt delete mode 100644 sound/soc/codecs/sirf-audio-codec.c delete mode 100644 sound/soc/codecs/zx_aud96p22.c delete mode 100644 sound/soc/sirf/Kconfig delete mode 100644 sound/soc/sirf/Makefile delete mode 100644 sound/soc/sirf/sirf-audio-port.c delete mode 100644 sound/soc/sirf/sirf-audio.c delete mode 100644 sound/soc/sirf/sirf-usp.c delete mode 100644 sound/soc/sirf/sirf-usp.h delete mode 100644 sound/soc/zte/Kconfig delete mode 100644 sound/soc/zte/Makefile delete mode 100644 sound/soc/zte/zx-i2s.c delete mode 100644 sound/soc/zte/zx-spdif.c delete mode 100644 sound/soc/zte/zx-tdm.c -- 2.29.2

3 4

[PATCH] ALSA: usb-audio: Fix hw constraints dependencies
by Takashi Iwai 20 Jan '21

20 Jan '21

Since the recent refactoring, it's been reported that some USB-audio devices (typically webcams) are no longer detected properly by PulseAudio. The debug session revealed that it's failing at probing by PA to try the sample rate 44.1kHz while the device has discrete sample rates other than 44.1kHz. But the puzzle was that arecord works as is, and some other devices with the discrete rates work, either. After all, this turned out to be the lack of the dependencies in a few hw constraint rules: snd_pcm_hw_rule_add() has the (variable) arguments specifying the dependent parameters, and some functions didn't set the target parameter itself as the dependencies. This resulted in an invalid parameter that could be generated only in a certain call pattern. This bug itself has been present in the code, but it didn't trigger errors just because the rules were casually avoiding such a corner case. After the recent refactoring and cleanup, however, the hw constraints work "as expected", and the problem surfaced now. For fixing the problem above, this patch adds the missing dependent parameters to each snd_pcm_hw_rule() call. Fixes: bc4e94aa8e72 ("ALSA: usb-audio: Handle discrete rates properly in hw constraints") BugLink: http://bugzilla.opensuse.org/show_bug.cgi?id=1181014 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f71965bf815f..078bb4c94033 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -981,6 +981,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, @@ -990,6 +991,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_CHANNELS, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, param_period_time_if_needed, @@ -998,6 +1000,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, -- 2.26.2

1 0

[PATCH 2/2] ASoC: rt5682: add device property to control LDO2 power
by shumingf＠realtek.com 20 Jan '21

20 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> The application circuit shall provide MICVDD power. In default, the codec driver doesn't need to enable LDO2 power. In case, the customers use VBAT for micbias, it should enable the device property "realtek,ldo2-pow-en". Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 18 +++++++++++++++++- 2 files changed, 18 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index 3900a07e3935..4b4c61b48380 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -41,6 +41,7 @@ struct rt5682_platform_data { unsigned int dmic_clk_rate; unsigned int dmic_delay; bool dmic_clk_driving_high; + bool ldo2_pow_en; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 42ad5bea86b5..68d29d71d6e4 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1343,6 +1343,19 @@ static int is_using_asrc(struct snd_soc_dapm_widget *w, } } +static int is_using_ldo2(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + if (rt5682->pdata.ldo2_pow_en) + return 1; + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, @@ -1854,7 +1867,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"CLKDET SYS", NULL, "CLKDET"}, - {"IN1P", NULL, "LDO2"}, + {"IN1P", NULL, "LDO2", is_using_ldo2}, {"BST1 CBJ", NULL, "IN1P"}, @@ -2999,6 +3012,9 @@ int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.dmic_clk_driving_high = device_property_read_bool(dev, "realtek,dmic-clk-driving-high"); + rt5682->pdata.ldo2_pow_en = device_property_read_bool(dev, + "realtek,ldo2-pow-en"); + return 0; } EXPORT_SYMBOL_GPL(rt5682_parse_dt); -- 2.29.0

2 1

[PATCH] ASoC: Intel: Skylake: Check the kcontrol against NULL
by Lukasz Majczak 20 Jan '21

20 Jan '21

There is no check for the kcontrol against NULL and in some cases it causes kernel to crash. Fixes: 2d744ecf2b984 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT") Cc: <stable(a)vger.kernel.org> # 5.4+ Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- sound/soc/intel/skylake/skl-topology.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ae466cd592922..c9abbe4ff0ba3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3618,12 +3618,18 @@ static void skl_tplg_complete(struct snd_soc_component *component) int i; list_for_each_entry(dobj, &component->dobj_list, list) { - struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se = - (struct soc_enum *)kcontrol->private_value; - char **texts = dobj->control.dtexts; + struct snd_kcontrol *kcontrol; + struct soc_enum *se; + char **texts; char chan_text[4]; + kcontrol = dobj->control.kcontrol; + if(!kcontrol) + continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; + if (dobj->type != SND_SOC_DOBJ_ENUM || dobj->control.kcontrol->put != skl_tplg_multi_config_set_dmic) base-commit: 69fe63aa100220c8fd1f451dd54dd0895df1441d -- 2.25.1

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handling different frame sizes in snd_pcm_extplug_callback_t transfer callback
by Arsen Arsenović 20 Jan '21

20 Jan '21

Hello! I am writing an external noise cancellation plugin for ALSA, based on rnnoise, but I've hit a bit of a roadblock: rnnoise processes chunks of 480 floats at a time and does no buffering, but alsa expects the transfer callback to take all the data from the input and put it onto the output, with no way to signal that any buffering is or should be happening. I have tried to return a multiple of 480 from the transfer callback and keep a buffer with the leftover data, to be prepended to the next payload, but it looks like ALSA will, when a plugin returns too little data, just send back the same non-processed data in a new call, which breaks this idea. Any way I can work around these limitations? Another solution could be for external plugins to report the amount they consumed and produced, rather than a single number for both, but I am not aware of how to do that, if it's even possible with the filter plugin SDK. Thanks in advance! EDIT: I had previously posted this question before I confirmed my mailing list membership, but seeing as it wasn't approved for a few days I cancelled the request and reposted it. Sorry about that. -- Arsen Arsenović

1 0

sof topology compilation fails
by GitHub issues - opened 20 Jan '21

20 Jan '21

alsa-project/alsa-lib issue #114 was opened from juimonen: sof topologies fail to compile because of following commit: commit 51e1d486ce4535923692bd1d7e59d5aa3a29cd50 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Thu Jan 7 17:40:42 2021 +0100 topology: parse_tuple_set() - remove dead condition code Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> I don't completely understand the change...? we are in "case SND_SOC_TPLG_TUPLE_TYPE_WORD" and then we check for SND_SOC_TPLG_TUPLE_TYPE_WORD (and not the value anymore) and error out? Issue URL : https://github.com/alsa-project/alsa-lib/issues/114 Repository URL: https://github.com/alsa-project/alsa-lib

1 0