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[alsa-devel] [PATCH 0/3 v2] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 05 Apr '16

05 Apr '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) Jose Abreu (3): drm/i2c/adv7511: Add audio support ASoC: dwc: Add I2S HDMI audio support arc: axs10x: Add support for Designware I2S on DT arch/arc/boot/dts/axs10x_mb.dtsi | 49 +- drivers/gpu/drm/i2c/Kconfig | 11 + drivers/gpu/drm/i2c/Makefile | 2 + drivers/gpu/drm/i2c/adv7511.c | 1024 ----------------------------------- drivers/gpu/drm/i2c/adv7511.h | 41 ++ drivers/gpu/drm/i2c/adv7511_audio.c | 310 +++++++++++ drivers/gpu/drm/i2c/adv7511_core.c | 1005 ++++++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 385 ++++++++++++- 10 files changed, 1788 insertions(+), 1041 deletions(-) delete mode 100644 drivers/gpu/drm/i2c/adv7511.c create mode 100644 drivers/gpu/drm/i2c/adv7511_audio.c create mode 100644 drivers/gpu/drm/i2c/adv7511_core.c -- 1.9.1

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[alsa-devel] [PATCH v2 0/7] topology: Add support for vendor tuples
by mengdong.lin＠linux.intel.com 05 Apr '16

05 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series addes support for vendor tuples to topology, to avoid importing binary data blob from other files. Backward compatibility of ABI is not impacted. A kernel patch is also submitted "ASoC: topology: ABI - Define types for vendor tuples". The 1st patch is small code cleanup. The 2nd patch is a preparation, since tuples will need the type-specific free handler. History: v2: add check on string length, use strtol() to get hex value, and fix memory leak. Mengdong Lin (7): topology: Use the generic pointer to free an element's object topology: Define a free handler for the element topology: Add doc for vendor tuples topology: ABI - Define types for vendor tuples topology: Add support for vendor tokens topology: Add support for parsing vendor tuples topology: Build data objects with tuples include/sound/asoc.h | 42 +++- include/topology.h | 79 +++++++- src/topology/data.c | 496 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/elem.c | 15 +- src/topology/parser.c | 24 +++ src/topology/tplg_local.h | 47 +++++ 6 files changed, 695 insertions(+), 8 deletions(-) -- 2.5.0

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Re: [alsa-devel] [PATCH v2- ALSA 2/3] HDA - create headset mic jack which use headphone pin sense
by Justin Monk 05 Apr '16

05 Apr '16

Team, I was wondering when this patch would be available, I figured it would already be in the down stream. If not I would love to test it. I have minimal programming skills but would love to help. Kernel Bug Tracker – Bug 107851 4.2.0-18-generic currently running 4.4.6-300.fc23.x86_64 Thanks, Justin Monk

2 1

[alsa-devel] Sound: BUG: KASAN: use-after-free in kill_fasync
by Baozeng Ding 05 Apr '16

05 Apr '16

Hi all, I've got the following report (use-after-free in kill_fasync) while running syzkaller. Unfortunately no reproducer.The kernel version is 4.5 (on Mar 16 commit 09fd671ccb2475436bd5f597f751ca4a7d177aea). ================================================================== BUG: KASAN: use-after-free in kill_fasync+0x3fb/0x420 at addr ffff880067691d88 Read of size 8 by task swapper/2/0 ============================================================================= BUG kmalloc-2048 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- Disabling lock debugging due to kernel taint INFO: Allocated in 0xbbbbbbbbbbbbbbbb age=18446678412249576073 cpu=2245704320 pid=-1 [< inline >] kmalloc /kernel/include/linux/slab.h:472 [< inline >] kzalloc /kernel/include/linux/slab.h:616 [< none >] snd_pcm_attach_substream+0x3b4/0xb10 /kernel/sound/core/pcm.c:966 [< none >] ___slab_alloc+0x4c7/0x500 /kernel/mm/slub.c:2446 [< none >] __slab_alloc+0x4c/0x90 /kernel/mm/slub.c:2475 [< inline >] slab_alloc_node /kernel/mm/slub.c:2538 [< inline >] slab_alloc /kernel/mm/slub.c:2580 [< none >] kmem_cache_alloc_trace+0x262/0x300 /kernel/mm/slub.c:2597 [< inline >] kmalloc /kernel/include/linux/slab.h:472 [< inline >] kzalloc /kernel/include/linux/slab.h:616 [< none >] snd_pcm_attach_substream+0x3b4/0xb10 /kernel/sound/core/pcm.c:966 [< none >] snd_pcm_open_substream+0x84/0x450 /kernel/sound/core/pcm_native.c:2262 [< inline >] snd_pcm_oss_open_file /kernel/sound/core/oss/pcm_oss.c:2346 [< none >] snd_pcm_oss_open.part.17+0x5a4/0x1100 /kernel/sound/core/oss/pcm_oss.c:2428 [< none >] snd_pcm_oss_open+0x35/0x50 /kernel/sound/core/oss/pcm_oss.c:2392 [< none >] soundcore_open+0x30f/0x640 /kernel/sound/sound_core.c:639 [< none >] chrdev_open+0x22a/0x4c0 /kernel/fs/char_dev.c:388 [< none >] do_dentry_open+0x6a2/0xcb0 /kernel/fs/open.c:736 [< none >] vfs_open+0x17b/0x1f0 /kernel/fs/open.c:853 [< inline >] do_last /kernel/fs/namei.c:3258 [< none >] path_openat+0x4837/0x5830 /kernel/fs/namei.c:3394 [< none >] do_filp_open+0x18e/0x250 /kernel/fs/namei.c:3429 [< none >] do_sys_open+0x201/0x420 /kernel/fs/open.c:1022 [< inline >] SYSC_open /kernel/fs/open.c:1040 [< none >] SyS_open+0x2d/0x40 /kernel/fs/open.c:1035 INFO: Freed in 0x10000b076 age=18446678416544543380 cpu=0 pid=0 [< none >] snd_pcm_detach_substream+0x134/0x280 /kernel/sound/core/pcm.c:1017 [< none >] __slab_free+0x1e8/0x300 /kernel/mm/slub.c:2657 [< inline >] slab_free /kernel/mm/slub.c:2810 [< none >] kfree+0x24e/0x2d0 /kernel/mm/slub.c:3661 [< none >] snd_pcm_detach_substream+0x134/0x280 /kernel/sound/core/pcm.c:1017 [< none >] snd_pcm_release_substream.part.38+0x219/0x2f0 /kernel/sound/core/pcm_native.c:2250 [< none >] snd_pcm_release_substream+0x59/0x70 /kernel/sound/core/pcm_native.c:2251 [< none >] snd_pcm_oss_release_file+0x45/0xb0 /kernel/sound/core/oss/pcm_oss.c:2305 [< none >] snd_pcm_oss_release+0xfa/0x250 /kernel/sound/core/oss/pcm_oss.c:2485 [< none >] __fput+0x236/0x780 /kernel/fs/file_table.c:208 [< none >] ____fput+0x15/0x20 /kernel/fs/file_table.c:244 [< none >] task_work_run+0x16b/0x200 /kernel/kernel/task_work.c:115 [< inline >] exit_task_work /kernel/include/linux/task_work.h:21 [< none >] do_exit+0x87f/0x2c90 /kernel/kernel/exit.c:748 [< none >] do_group_exit+0x108/0x330 /kernel/kernel/exit.c:878 [< inline >] SYSC_exit_group /kernel/kernel/exit.c:889 [< none >] SyS_exit_group+0x1d/0x20 /kernel/kernel/exit.c:887 [< none >] entry_SYSCALL_64_fastpath+0x23/0xc1 /kernel/arch/x86/entry/entry_64.S:207 INFO: Slab 0xffffea00019da400 objects=13 used=8 fp=0xffff880067691be0 flags=0x5fffc0000004080 INFO: Object 0xffff880067691bd8 @offset=7128 fp=0xbbbbbbbbbbbbbbbb Call Trace: <IRQ> [< inline >] __dump_stack /kernel/lib/dump_stack.c:15 <IRQ> [<ffffffff82945051>] dump_stack+0xb3/0x112 /kernel/lib/dump_stack.c:51 [<ffffffff817009ad>] print_trailer+0x10d/0x190 /kernel/mm/slub.c:668 [<ffffffff817074af>] object_err+0x2f/0x40 /kernel/mm/slub.c:675 [< inline >] print_address_description /kernel/mm/kasan/report.c:138 [<ffffffff81709ca5>] kasan_report_error+0x215/0x530 /kernel/mm/kasan/report.c:236 [< inline >] ? spin_lock /kernel/include/linux/spinlock.h:302 [<ffffffff84a1c550>] ? snd_pcm_stream_lock+0x80/0xd0 /kernel/sound/core/pcm_native.c:104 [< inline >] kasan_report /kernel/mm/kasan/report.c:259 [<ffffffff8170a0be>] __asan_report_load8_noabort+0x3e/0x40 /kernel/mm/kasan/report.c:280 [<ffffffff81792a0b>] ? kill_fasync+0x3fb/0x420 /kernel/fs/fcntl.c:729 [<ffffffff81792a0b>] kill_fasync+0x3fb/0x420 /kernel/fs/fcntl.c:729 [<ffffffff84a36e08>] snd_pcm_period_elapsed+0x1c8/0x230 /kernel/sound/core/pcm_lib.c:1890 [<ffffffff84b6e2e0>] ? get_bound_vga.isra.21.part.22+0x140/0x140 /kernel/sound/pci/hda/hda_intel.c:1327 [<ffffffff84b0637d>] stream_update+0xad/0xe0 /kernel/sound/pci/hda/hda_controller.c:923 [<ffffffff84c1583e>] snd_hdac_bus_handle_stream_irq+0x24e/0x350 /kernel/sound/hda/hdac_controller.c:449 [<ffffffff84b062d0>] ? azx_init_chip+0x100/0x100 /kernel/sound/pci/hda/hda_controller.c:890 [<ffffffff84b04d24>] ? azx_interrupt+0x14/0x4d0 /kernel/sound/pci/hda/hda_controller.c:929 [<ffffffff84b04eee>] azx_interrupt+0x1de/0x4d0 /kernel/sound/pci/hda/hda_controller.c:954 [<ffffffff84b04d10>] ? azx_stop_chip+0x20/0x20 /kernel/sound/pci/hda/hda_controller.c:908 [<ffffffff8143d9e3>] handle_irq_event_percpu+0xf3/0x790 /kernel/kernel/irq/handle.c:145 [<ffffffff8143e127>] handle_irq_event+0xa7/0x140 /kernel/kernel/irq/handle.c:192 [<ffffffff81447f81>] handle_edge_irq+0x1e1/0x8d0 /kernel/kernel/irq/chip.c:623 [< inline >] generic_handle_irq_desc /kernel/include/linux/irqdesc.h:146 [<ffffffff811ad739>] handle_irq+0x109/0x2a0 /kernel/arch/x86/kernel/irq_64.c:78 [< inline >] ? rcu_lock_release /kernel/include/linux/rcupdate.h:491 [< inline >] ? rcu_read_unlock /kernel/include/linux/rcupdate.h:926 [< inline >] ? __atomic_notifier_call_chain /kernel/kernel/notifier.c:184 [<ffffffff81369c4f>] ? atomic_notifier_call_chain+0xbf/0x140 /kernel/kernel/notifier.c:193 [<ffffffff81369b90>] ? __atomic_notifier_call_chain+0x150/0x150 /kernel/include/linux/rcupdate.h:922 [<ffffffff811ac36d>] do_IRQ+0x7d/0x1a0 /kernel/arch/x86/kernel/irq.c:240 [<ffffffff85dac40c>] common_interrupt+0x8c/0x8c /kernel/arch/x86/entry/entry_64.S:454 <EOI> [<ffffffff812245c6>] ? native_safe_halt+0x6/0x10 /kernel/./arch/x86/include/asm/irqflags.h:49 [<ffffffff81405ced>] ? trace_hardirqs_on+0xd/0x10 /kernel/kernel/locking/lockdep.c:2635 [< inline >] arch_safe_halt /kernel/./arch/x86/include/asm/paravirt.h:117 [<ffffffff811bf152>] default_idle+0x22/0x2d0 /kernel/arch/x86/kernel/process.c:307 [<ffffffff811c053a>] arch_cpu_idle+0xa/0x10 /kernel/arch/x86/kernel/process.c:298 [<ffffffff813eb618>] default_idle_call+0x48/0x70 /kernel/kernel/sched/idle.c:93 [< inline >] cpuidle_idle_call /kernel/kernel/sched/idle.c:151 [< inline >] cpu_idle_loop /kernel/kernel/sched/idle.c:242 [<ffffffff813ebaa7>] cpu_startup_entry+0x467/0x600 /kernel/kernel/sched/idle.c:291 [<ffffffff81201432>] start_secondary+0x2b2/0x380 /kernel/arch/x86/kernel/smpboot.c:259 [<ffffffff81201180>] ? set_cpu_sibling_map+0x18a0/0x18a0 /kernel/include/linux/topology.h:80 Memory state around the buggy address: ffff880067691c80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff880067691d00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >ffff880067691d80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff880067691e00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff880067691e80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== It seems that in function snd_pcm_detach_substream, when freeing runtime structure, it did not handle its fasync data pointer. Any ideas? Thanks. Best Regards, Baozeng Ding

2 1

[alsa-devel] [PATCH v2 0/2] conf: topology: Add the conf for SKL-rt286
by Vinod Koul 05 Apr '16

05 Apr '16

From: Shreyas NC <shreyas.nc(a)intel.com> The SKL RT286 I2S topology conf is provided here. The first patch adds that along with binary data for modules. The Second patch provides source of those binayy blobs and allows them to be recompiled. Shreyas NC (2): conf: topology: Add Skylake i2s conf conf: topology: Generate Private data binary blobs configure.ac | 1 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/sklrt286/Makefile.am | 4 + src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/data/Makefile.am | 4 + src/conf/topology/sklrt286/data/README | 2 + src/conf/topology/sklrt286/data/compile_local | 16 + src/conf/topology/sklrt286/data/configure.ac | 30 + src/conf/topology/sklrt286/data/pvt.c | 1815 ++++++++++++++++++++ src/conf/topology/sklrt286/data/pvt_data.c | 90 + src/conf/topology/sklrt286/data/pvt_data_local.h | 115 ++ .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/sklrt286/skl_i2s.conf | 342 ++++ 29 files changed, 2420 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/sklrt286/Makefile.am create mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin create mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin create mode 100644 src/conf/topology/sklrt286/data/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/README create mode 100755 src/conf/topology/sklrt286/data/compile_local create mode 100644 src/conf/topology/sklrt286/data/configure.ac create mode 100644 src/conf/topology/sklrt286/data/pvt.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data_local.h create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin create mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin create mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin create mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/skl_i2s.conf -- 2.4.11

1 0

[alsa-devel] Getting rid of inside_vm in intel8x0
by Andy Lutomirski 04 Apr '16

04 Apr '16

Would it be possible to revert: commit 228cf79376f13b98f2e1ac10586311312757675c Author: Konstantin Ozerkov <kozerkov(a)parallels.com> Date: Wed Oct 26 19:11:01 2011 +0400 ALSA: intel8x0: Improve performance in virtual environment Presumably one or more of the following is true: a) The inside_vm == true case is just an optimization and should apply unconditionally. b) The inside_vm == true case is incorrect and should be fixed or disabled. c) The inside_vm == true case is a special case that makes sense then IO is very very slow but doesn't make sense when IO is fast. If so, why not literally measure the time that the IO takes and switch over to the "inside VM" path when IO is slow? There are a pile of nonsensical "are we in a VM" checks of various sorts scattered throughout the kernel, they're all a mess to maintain (there are lots of kinds of VMs in the world, and Linux may not even know it's a guest), and, in most cases, it appears that the correct solution is to delete the checks. I just removed a nasty one in the x86_32 entry asm, and this one is written in C so it should be a piece of cake :) --Andy

5 11

[alsa-devel] [PATCH v3] ALSA: hda - Update chmap tlv to report sink's capability
by Subhransu S. Prusty 04 Apr '16

04 Apr '16

The existing TLV callback implementation copies all of the cea_channel_speaker_allocation map table to the TLV container irrespective of what is reported by sink. This is of little use to the userspace application. With this patch, it parses the spk_alloc block as queried from the ELD, and copies only the corresponding mapping channel allocation entries from the cea channel speaker allocation table. Thus the user can parse the TLV container to identify sink's capability and set the channel map accordingly. It shouldn't impact the behavior in AMD chipset, as this makes use of already parsed spk alloc block to calculate the channel map. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- changes in v3: Copy all possible channel map entries w.r.t. sink's capability include/sound/hda_chmap.h | 2 ++ sound/hda/hdmi_chmap.c | 44 ++++++++++++++++++++++++++++++++++++++++---- sound/pci/hda/patch_hdmi.c | 13 +++++++++++++ 3 files changed, 55 insertions(+), 4 deletions(-) diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h index e20d219..babd445 100644 --- a/include/sound/hda_chmap.h +++ b/include/sound/hda_chmap.h @@ -36,6 +36,8 @@ struct hdac_chmap_ops { int (*chmap_validate)(struct hdac_chmap *hchmap, int ca, int channels, unsigned char *chmap); + int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx); + void (*get_chmap)(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap); void (*set_chmap)(struct hdac_device *hdac, int pcm_idx, diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec862..c6c75e7 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,30 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static int spk_mask_from_spk_alloc(int spk_alloc) +{ + int i; + int spk_mask = eld_speaker_allocation_bits[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + return spk_mask; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; int chs, count = 0; + unsigned long max_chs; + int type; + int spk_alloc, spk_mask; if (size < 8) return -ENOMEM; @@ -639,40 +656,59 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + spk_mask = spk_mask_from_spk_alloc(spk_alloc); + + max_chs = hweight_long(spk_mask); + + for (chs = 2; chs <= max_chs; chs++) { int i; struct hdac_cea_channel_speaker_allocation *cap; cap = channel_allocations; for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); unsigned int tlv_chmap[8]; - if (type < 0) + if (cap->channels != chs) + continue; + + if (!(cap->spk_mask == (spk_mask & cap->spk_mask))) continue; + + type = chmap->ops.chmap_cea_alloc_validate_get_type( + chmap, cap, chs); + if (type < 0) + return -ENODEV; if (size < 8) return -ENOMEM; + if (put_user(type, dst) || put_user(chs_bytes, dst + 1)) return -EFAULT; + dst += 2; size -= 8; count += 8; + if (size < chs_bytes) return -ENOMEM; + size -= chs_bytes; count += chs_bytes; chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) return -EFAULT; dst += chs; } } + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..dd25f2c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1838,6 +1838,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2249,6 +2261,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) spec->chmap.ops.get_chmap = hdmi_get_chmap; spec->chmap.ops.set_chmap = hdmi_set_chmap; spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, codec->spec = spec; hdmi_array_init(spec, 4); -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add a sample rate quirk for Phoenix Audio TMX320
by Takashi Iwai 04 Apr '16

04 Apr '16

Phoenix Audio TMX320 gives the similar error when the sample rate is asked: usb 2-1.3: 2:1: cannot get freq at ep 0x85 usb 2-1.3: 1:1: cannot get freq at ep 0x2 .... Add the corresponding USB-device ID (1de7:0014) to snd_usb_get_sample_rate_quirk() list. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6178bb5d0731..24c7c2311b47 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1137,6 +1137,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } -- 2.7.4

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Re: [alsa-devel] Getting rid of inside_vm in intel8x0
by Takashi Iwai 04 Apr '16

04 Apr '16

On Mon, 04 Apr 2016 11:05:43 +0200, George Dunlap wrote: > > On 02/04/16 13:57, Andy Lutomirski wrote: > > On Fri, Apr 1, 2016 at 10:33 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> On Sat, 02 Apr 2016 00:28:31 +0200, > >> Luis R. Rodriguez wrote: > >>> If the former, could a we somehow detect an emulated device other than through > >>> this type of check ? Or could we *add* a capability of some sort to detect it > >>> on the driver ? This would not address the removal, but it could mean finding a > >>> way to address emulation issues. > >>> > >>> If its an IO issue -- exactly what is the causing the delays in IO ? > >> > >> Luis, there is no problem about emulation itself. It's rather an > >> optimization to lighten the host side load, as I/O access on a VM is > >> much heavier. > >> > >>>>>> This is satisfied mostly only on VM, and can't > >>>>>> be measured easily unlike the IO read speed. > >>>>> > >>>>> Interesting, note the original patch claimed it was for KVM and > >>>>> Parallels hypervisor only, but since the code uses: > >>>>> > >>>>> +#if defined(__i386__) || defined(__x86_64__) > >>>>> + inside_vm = inside_vm || boot_cpu_has(X86_FEATURE_HYPERVISOR); > >>>>> +#endif > >>>>> > >>>>> This makes it apply also to Xen as well, this makes this hack more > >>>>> broad, but does is it only applicable when an emulated device is > >>>>> used ? What about if a hypervisor is used and PCI passthrough is > >>>>> used ? > >>>> > >>>> A good question. Xen was added there at the time from positive > >>>> results by quick tests, but it might show an issue if it's running on > >>>> a very old chip with PCI passthrough. But I'm not sure whether PCI > >>>> passthrough would work on such old chipsets at all. > >>> > >>> If it did have an issue then that would have to be special cased, that > >>> is the module parameter would not need to be enabled for such type of > >>> systems, and heuristics would be needed. As you note, fortunately this > >>> may not be common though... > >> > >> Actually this *is* module parametered. If set to a boolean value, it > >> can be applied / skipped forcibly. So, if there has been a problem on > >> Xen, this should have been reported. That's why I wrote it's no > >> common case. This comes from the real experience. > >> > >>> but if this type of work around may be > >>> taken as a precedent to enable other types of hacks in other drivers > >>> I'm very fearful of more hacks later needing these considerations as > >>> well. > >>> > >>>>>>> There are a pile of nonsensical "are we in a VM" checks of various > >>>>>>> sorts scattered throughout the kernel, they're all a mess to maintain > >>>>>>> (there are lots of kinds of VMs in the world, and Linux may not even > >>>>>>> know it's a guest), and, in most cases, it appears that the correct > >>>>>>> solution is to delete the checks. I just removed a nasty one in the > >>>>>>> x86_32 entry asm, and this one is written in C so it should be a piece > >>>>>>> of cake :) > >>>>>> > >>>>>> This cake looks sweet, but a worm is hidden behind the cream. > >>>>>> The loop in the code itself is already a kludge for the buggy hardware > >>>>>> where the inconsistent read happens not so often (only at the boundary > >>>>>> and in a racy way). It would be nice if we can have a more reliably > >>>>>> way to know the hardware buggyness, but it's difficult, > >>>>>> unsurprisingly. > >>>>> > >>>>> The concern here is setting precedents for VM cases sprinkled in the kernel. > >>>>> The assumption here is such special cases are really paper'ing over another > >>>>> type of issue, so its best to ultimately try to root cause the issue in > >>>>> a more generalized fashion. > >>>> > >>>> Well, it's rather bare metal that shows the buggy behavior, thus we > >>>> need to paper over it. In that sense, it's other way round; we don't > >>>> tune for VM. The VM check we're discussing is rather for skipping the > >>>> strange workaround. > >>> > >>> What is it exactly about a VM that enables this work around to be skipped? > >>> I don't quite get it yet. > >> > >> VM -- at least the full one with the sound hardware emulation -- > >> doesn't have the hardware bug. So, the check isn't needed. > > > > Here's the issue, though: asking "am I in a VM" is not a good way to > > learn properties of hardware. Just off the top of my head, here are > > some types of VM and what they might imply about hardware: > > > > Intel Kernel Guard: your sound card is passed through from real hardware. > > > > Xen: could go either way. In dom0, it's likely passed through. In > > domU, it could be passed through or emulated, and I believe this is > > the case for all of the Xen variants. > > > > KVM: Probably emulated, but could be passed through. > > I'm not sure exactly why I was CC'd into this thread, but this is an > important point -- even if you're running in a VM, you may actually have > direct un-emulated IO access to a real (buggy) piece of hardware; in > which case it sounds like you still need the work-around. So > boot_cpu_has(X86_FEATURE_HYPERVISOR) is probably not the right check. The VM check is kept there only to show a kernel message; in case where a similar issue is seen on another VM, user may notice more easily by that. The VM check itself doesn't change any kernel behavior any longer. Takashi

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[alsa-devel] [PATCH v2] ALSA: hda - Update chmap tlv to report sink's capability
by Subhransu S. Prusty 04 Apr '16

04 Apr '16

The existing TLV callback implementation copies all of the cea_channel_speaker_allocation map table to the TLV container irrespective of what is reported by sink. This is of little use to the userspace application. With this patch, it parses the spk_alloc block as queried from the ELD, and copies only the corresponding channel map from the cea channel speaker allocation table. Thus the user can parse the TLV container to identify sink's capability and set the channel map accordingly. It shouldn't impact the behavior in AMD chipset, as this makes use of already parsed spk alloc block to calculate the channel map. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- changes in v2: Updated the commit message to make it more understandable. include/sound/hda_chmap.h | 2 ++ sound/hda/hdmi_chmap.c | 90 +++++++++++++++++++++++++++++----------------- sound/pci/hda/patch_hdmi.c | 13 +++++++ 3 files changed, 72 insertions(+), 33 deletions(-) diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h index e20d219..babd445 100644 --- a/include/sound/hda_chmap.h +++ b/include/sound/hda_chmap.h @@ -36,6 +36,8 @@ struct hdac_chmap_ops { int (*chmap_validate)(struct hdac_chmap *hchmap, int ca, int channels, unsigned char *chmap); + int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx); + void (*get_chmap)(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap); void (*set_chmap)(struct hdac_device *hdac, int pcm_idx, diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec862..9406203 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,40 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static struct hdac_cea_channel_speaker_allocation *get_cap_from_spk_alloc( + int spk_alloc) +{ + int i, spk_mask = 0; + + if (!spk_alloc) + return &channel_allocations[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* Get num channels using spk mask */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if ((spk_mask & channel_allocations[i].spk_mask) == spk_mask) + return &channel_allocations[i]; + } + + return &channel_allocations[0]; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; - int chs, count = 0; + int chs, count = 0, chs_bytes; + int type; + unsigned int tlv_chmap[8]; + int spk_alloc; + struct hdac_cea_channel_speaker_allocation *cap; if (size < 8) return -ENOMEM; @@ -639,40 +666,37 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { - int i; - struct hdac_cea_channel_speaker_allocation *cap; - - cap = channel_allocations; - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { - int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); - unsigned int tlv_chmap[8]; - - if (type < 0) - continue; - if (size < 8) - return -ENOMEM; - if (put_user(type, dst) || - put_user(chs_bytes, dst + 1)) - return -EFAULT; - dst += 2; - size -= 8; - count += 8; - if (size < chs_bytes) - return -ENOMEM; - size -= chs_bytes; - count += chs_bytes; - chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, - tlv_chmap, chs); - if (copy_to_user(dst, tlv_chmap, chs_bytes)) - return -EFAULT; - dst += chs; - } - } + + if (size < 8) + return -ENOMEM; + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + cap = get_cap_from_spk_alloc(spk_alloc); + chs = cap->channels; + + chs_bytes = chs * 4; + type = chmap->ops.chmap_cea_alloc_validate_get_type(chmap, cap, chs); + if (type < 0) + return -ENODEV; + + if (put_user(type, dst) || + put_user(chs_bytes, dst + 1)) + return -EFAULT; + + dst += 2; + size -= 8; + count += 8; + if (size < chs_bytes) + return -ENOMEM; + + count += chs_bytes; + chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) + return -EFAULT; + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..dd25f2c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1838,6 +1838,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2249,6 +2261,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) spec->chmap.ops.get_chmap = hdmi_get_chmap; spec->chmap.ops.set_chmap = hdmi_set_chmap; spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, codec->spec = spec; hdmi_array_init(spec, 4); -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: hdac_hdmi: Fix codec power state in S3
by Subhransu S. Prusty 02 Apr '16

02 Apr '16

With this series, the codec power is turned OFF when entering suspend if the playback is already in progress. And during resume the power is restored. Codec driver registers the S3 callbacks with prepare/complete to synchronize with Jack notification from display driver. Codec driver takes care of putting the AFG node to proper power state during S3 and controller driver takes care of managing the codec power. Also setting the power state for AFG node is optimized by removing the timeout loop. The driver instead uses a codec_read, which sets the power state and waits for response before the codec power is turned OFF. Subhransu S. Prusty (3): ASoC: hdac_hdmi: Fix to use dev_pm ops instead soc pm ASoC: hdac_hdmi: Fix codec power state in S3 during playback ASoC: Intel: Skylake: Fix to turn OFF codec power when entering S3 sound/soc/codecs/hdac_hdmi.c | 94 +++++++++++++++++++------------------------ sound/soc/intel/skylake/skl.c | 15 ++++++- 2 files changed, 55 insertions(+), 54 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 02 Apr '16

02 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v1: Use codec dai name to determine the capture path. --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 39 ++++++++++++++++++-- 2 files changed, 41 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..383cba9 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,17 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional properties: +- mediatek,audio-codec-dai-capture: audio codec dai on capture path + rt5645-aif1: Default setting. Connect rt5650 I2S1 for capture + rt5645-aif2: Connect rt5650 I2S2 for capture + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + mediatek,audio-codec-dai-capture = "rt5645-aif2"; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..fd2ca66 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,26 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_dai_name_capture = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, - RT5645_CLK_SEL_I2S1_ASRC); + RT5645_DA_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_dai_name_capture, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_dai_name_capture, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_err(card->dev, "Can't get the right codec dai\n"); + return -EINVAL; + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +124,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +168,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +196,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + const char *codec_dai_name_capture; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +219,17 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (device_property_present(&pdev->dev, + "mediatek,audio-codec-dai-capture")) { + device_property_read_string(&pdev->dev, + "mediatek,audio-codec-dai-capture", + &codec_dai_name_capture); + + mt8173_rt5650_codecs[1].dai_name = codec_dai_name_capture; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

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[alsa-devel] [PATCH v2 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 02 Apr '16

02 Apr '16

Driver for TI's TAS5720L/M digital audio amplifiers. The driver should be pretty standard except the optional interrupt-based fault reporting. Some background on the fault reporting since that might be a discussion of interest. The code should have that documented rather well but I wanted to bring it up right away. The TAS5720 devices can report conditions such as over-current (short), over temp, and others that can be important to know in a real-world system. However the device lacks a way to mask the generation of fault interrupts except by shutting it down, which can lead to cases where the device generates SAIF-fault host interrupts every 300us! This is clearly a lot of overhead to be thrown at the SoC (and my bench testing has demonstrated that) so after trying different things to tame this behavior I ended up implementing method to disable/enable the interrupts in the driver itself in combination with rate-limiting the processing of those events. The fact of processing interrupts in a threaded fashion (IRQF_ONESHOT) alone was not enough under certain conditions to prevent the SoC from not being able to get to all interrupts even with an empty threaded handler, and it would freeze and eventually the Kernel would disable the interrupt ("no one cared"). The scheme currently implemented seems to work reliably cycling different audio files over hours also forcing error conditions. But if anybody has a "cleaner" idea how to rate-limit IRQs on a very low level (currently done in the IRQ handler function) I'm absolutely open to suggestions. This all may not be an issue at all when using a faster/lower-latency SoC than what I'm using (BeagleBone Black) but the fact that IRQF_ONESHOT alone was not enough to prevent the system to be brought to its knees (again, even with an empty threaded handler) was a bit of a surprise to me. -- Andreas Dannenberg Texas Instruments Inc Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 37 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 734 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 873 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

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[alsa-devel] [PATCH - aconnect 1/1] Fix aconnect printing card and pid on older kernels
by Adam Goode 01 Apr '16

01 Apr '16

alsa-lib is changed to return a distinct negative value if running on an older kernel. -1 is only returned now if the kernel knows for sure that there is no card or pid. On older kernels, alsa-lib will return -ENOSYS. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 85a7770..bbc5bb3 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -177,13 +177,13 @@ static void print_port(snd_seq_t *seq, snd_seq_client_info_t *cinfo, #ifdef HAVE_SEQ_CLIENT_INFO_GET_CARD card = snd_seq_client_info_get_card(cinfo); #endif - if (card != -1) + if (card >= 0) printf(",card=%d", card); #ifdef HAVE_SEQ_CLIENT_INFO_GET_PID pid = snd_seq_client_info_get_pid(cinfo); #endif - if (pid != -1) + if (pid >= 0) printf(",pid=%d", pid); printf("]\n"); } -- 2.8.0.rc3.226.g39d4020

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[alsa-devel] [PATCH 1/2] ALSA: hda - Enable i915 binding for gen3/4 HDMI/DP
by Takashi Iwai 01 Apr '16

01 Apr '16

This patch fills the holes and now all i915 HDMI/DP codecs are managed over the audio ELD notifier, finally. The old gen3/gen4 chips have usually only a single pin/converter pair, and the digital port mapping is fixed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 11 +++++++++++ sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++------ 2 files changed, 26 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index d0da2508823e..c62a9f830b84 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -128,6 +128,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); * Pin Widget 4 - PORT B (port = 1 in i915 driver) * Pin Widget 5 - PORT C (port = 2 in i915 driver) * Pin Widget 6 - PORT D (port = 3 in i915 driver) + * + * on earlier models: + * Pin Widget 3 - PORT B */ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) { @@ -139,6 +142,14 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) case 0x80862882: /* VLV */ base_nid = 3; break; + case 0x80862801: /* Bearlake */ + case 0x80862802: /* Cantiga */ + case 0x80862803: /* Eaglelake */ + case 0x80862880: /* CedarTrail */ + case 0x808629fb: /* Crestline HDMI */ + if (WARN_ON(pin_nid != 0x03)) + return -1; + return 1; /* only port B */ default: base_nid = 4; break; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4833c7bdd1e8..2a3bf1584f3b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2286,6 +2286,15 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) case 0x80862882: /* VLV */ pin_nid = port + 0x03; break; + case 0x80862801: /* Bearlake */ + case 0x80862802: /* Cantiga */ + case 0x80862803: /* Eaglelake */ + case 0x80862880: /* CedarTrail */ + case 0x808629fb: /* Crestline HDMI */ + if (WARN_ON(port != 1)) /* assume only port B */ + return; + pin_nid = 0x03; + break; default: pin_nid = port + 0x04; break; @@ -2424,7 +2433,7 @@ static int patch_i915_byt_hdmi(struct hda_codec *codec) return 0; } -/* Intel IronLake, SandyBridge and IvyBridge; with eld notifier */ +/* Intel Gen3/Gen4, IronLake, SandyBridge and IvyBridge; with eld notifier */ static int patch_i915_cpt_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -3627,9 +3636,9 @@ HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi), -HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi), @@ -3638,10 +3647,10 @@ HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), -HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_i915_cpt_hdmi), /* special ID for generic HDMI */ HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi), {} /* terminator */ -- 2.7.4

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[alsa-devel] [PATCH] a52: Make compatible with ffmpeg 3.0
by Anatol Pomozov 01 Apr '16

01 Apr '16

audioconvert.h has been deprecated a while ago and got removed in ffmpeg 3.0 (commit 2d40968dd3ff17b12f7). Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- a52/pcm_a52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 88529eb..6fad77d 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -39,7 +39,7 @@ #endif #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53, 34, 0) -#include <libavutil/audioconvert.h> +#include <libavutil/channel_layout.h> #include <libavutil/mem.h> #define USE_AVCODEC_FRAME #endif -- 2.7.1

3 3

[alsa-devel] External Sound Card Prevents Boot, errors
by Alan Taylor 01 Apr '16

01 Apr '16

I have an external sound card--a Topping VX2. When plugged in to USB, the device prevents Ubuntu from booting properly. It hangs until I remove the USB cord from the device, then Ubuntu boots normally. Once booted, the device can be recognized by Ubuntu, but then seems to nuke PulseAudio, leaving no devices in the Sound Settings. The VX2 uses Via Technologies VT 1620A USB audio controller. I suspect that this may be the issue. I've tried this with two separate VX2 devices, with the same results. I see: -110 connection time out -71 protocol error -19 no device Attaching a Pulse Audio log here. I'm running Ubuntu 15.10. also posted: https://bugzilla.kernel.org/show_bug.cgi?id=115661

1 0

[alsa-devel] [PATCH v3 1/5] ASoC: tas571x: added missing register literals
by Petr Kulhavy 01 Apr '16

01 Apr '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to the register 0x25. Added volatile and read-only register tables into tas5711_regmap_config and tas5717_regmap_config. The chip has 256 registers in total. But from address 0x29 on (0x26 to 0x28 are reserved) the register width varies between 20, 12 and 8 bytes, which the register map is unable to represent. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> v3: added volatile and read-only register tables into regmap configs sound/soc/codecs/tas571x.c | 28 ++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 50 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..d255d65 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -241,6 +245,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -260,6 +284,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -314,6 +340,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..cf800c3 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
by Sven Eckelmann 01 Apr '16

01 Apr '16

The Lenovo Thinkpad T460s requires the alc_fixup_tpt440_dock as well in order to get working sound output on the docking stations headphone jack. Patch tested on a Thinkpad T460s (20F9CT01WW) using a ThinkPad Ultradock on kernel 4.4.6. Signed-off-by: Sven Eckelmann <sven(a)narfation.org> Tested-by: Simon Wunderlich <sw(a)simonwunderlich.de> --- sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 73978c7..5b85019 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4759,6 +4759,8 @@ enum { ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, + ALC292_FIXUP_TPT460_DOCK, + ALC292_FIXUP_TPT460, }; static const struct hda_fixup alc269_fixups[] = { @@ -5401,6 +5403,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC, }, + [ALC292_FIXUP_TPT460_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440_dock, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, + [ALC292_FIXUP_TPT460] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC292_FIXUP_TPT460_DOCK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5554,7 +5568,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -5649,6 +5663,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, + {.id = ALC292_FIXUP_TPT460_DOCK, .name = "tpt460-dock"}, + {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.8.0.rc3

1 0

[alsa-devel] [PATCH] ALSA: timer: Use mod_timer() for rearming the system timer
by Takashi Iwai 01 Apr '16

01 Apr '16

ALSA system timer backend stops the timer via del_timer() without sync and leaves del_timer_sync() at the close instead. This is because of the restriction by the design of ALSA timer: namely, the stop callback may be called from the timer handler, and calling the sync shall lead to a hangup. However, this also triggers a kernel BUG() when the timer is rearmed immediately after stopping without sync: kernel BUG at kernel/time/timer.c:966! Call Trace: <IRQ> [<ffffffff8239c94e>] snd_timer_s_start+0x13e/0x1a0 [<ffffffff8239e1f4>] snd_timer_interrupt+0x504/0xec0 [<ffffffff8122fca0>] ? debug_check_no_locks_freed+0x290/0x290 [<ffffffff8239ec64>] snd_timer_s_function+0xb4/0x120 [<ffffffff81296b72>] call_timer_fn+0x162/0x520 [<ffffffff81296add>] ? call_timer_fn+0xcd/0x520 [<ffffffff8239ebb0>] ? snd_timer_interrupt+0xec0/0xec0 .... It's the place where add_timer() checks the pending timer. It's clear that this may happen after the immediate restart without sync in our cases. So, the workaround here is just to use mod_timer() instead of add_timer(). This looks like a band-aid fix, but it's a right move, as snd_timer_interrupt() takes care of the continuous rearm of timer. Reported-by: Jiri Slaby <jslaby(a)suse.cz> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index ea4d999113ef..6469bedda2f3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1019,8 +1019,8 @@ static int snd_timer_s_start(struct snd_timer * timer) njiff += timer->sticks - priv->correction; priv->correction = 0; } - priv->last_expires = priv->tlist.expires = njiff; - add_timer(&priv->tlist); + priv->last_expires = njiff; + mod_timer(&priv->tlist, njiff); return 0; } -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Update chmap tlv to report sink's capability
by Subhransu S. Prusty 01 Apr '16

01 Apr '16

Currently channel map controls don't reflect the sink's capability to userspace. It reports the complete hdmi spec channel allocation table to the userspace which is of little use to the userspace applications. With this patch, tlv callback is modified to read the speaker allocation block from sink, and a mapping alsa channel map is presented to the userspace. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/hda_chmap.h | 2 ++ sound/hda/hdmi_chmap.c | 90 +++++++++++++++++++++++++++++----------------- sound/pci/hda/patch_hdmi.c | 13 +++++++ 3 files changed, 72 insertions(+), 33 deletions(-) diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h index e20d219..babd445 100644 --- a/include/sound/hda_chmap.h +++ b/include/sound/hda_chmap.h @@ -36,6 +36,8 @@ struct hdac_chmap_ops { int (*chmap_validate)(struct hdac_chmap *hchmap, int ca, int channels, unsigned char *chmap); + int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx); + void (*get_chmap)(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap); void (*set_chmap)(struct hdac_device *hdac, int pcm_idx, diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec862..9406203 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,40 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static struct hdac_cea_channel_speaker_allocation *get_cap_from_spk_alloc( + int spk_alloc) +{ + int i, spk_mask = 0; + + if (!spk_alloc) + return &channel_allocations[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* Get num channels using spk mask */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if ((spk_mask & channel_allocations[i].spk_mask) == spk_mask) + return &channel_allocations[i]; + } + + return &channel_allocations[0]; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; - int chs, count = 0; + int chs, count = 0, chs_bytes; + int type; + unsigned int tlv_chmap[8]; + int spk_alloc; + struct hdac_cea_channel_speaker_allocation *cap; if (size < 8) return -ENOMEM; @@ -639,40 +666,37 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { - int i; - struct hdac_cea_channel_speaker_allocation *cap; - - cap = channel_allocations; - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { - int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); - unsigned int tlv_chmap[8]; - - if (type < 0) - continue; - if (size < 8) - return -ENOMEM; - if (put_user(type, dst) || - put_user(chs_bytes, dst + 1)) - return -EFAULT; - dst += 2; - size -= 8; - count += 8; - if (size < chs_bytes) - return -ENOMEM; - size -= chs_bytes; - count += chs_bytes; - chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, - tlv_chmap, chs); - if (copy_to_user(dst, tlv_chmap, chs_bytes)) - return -EFAULT; - dst += chs; - } - } + + if (size < 8) + return -ENOMEM; + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + cap = get_cap_from_spk_alloc(spk_alloc); + chs = cap->channels; + + chs_bytes = chs * 4; + type = chmap->ops.chmap_cea_alloc_validate_get_type(chmap, cap, chs); + if (type < 0) + return -ENODEV; + + if (put_user(type, dst) || + put_user(chs_bytes, dst + 1)) + return -EFAULT; + + dst += 2; + size -= 8; + count += 8; + if (size < chs_bytes) + return -ENOMEM; + + count += chs_bytes; + chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) + return -EFAULT; + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..dd25f2c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1838,6 +1838,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2249,6 +2261,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) spec->chmap.ops.get_chmap = hdmi_get_chmap; spec->chmap.ops.set_chmap = hdmi_set_chmap; spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, codec->spec = spec; hdmi_array_init(spec, 4); -- 1.9.1

2 3

[alsa-devel] [PATCH V4 0/2] ASoC: share more product information with user space
by han.lu＠intel.com 01 Apr '16

01 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, so user space utils such as PA and UCM can distinguish different products. 1. Add a core API. 2. Apply the API to bytcr-rt5640 driver. changes on V4: 1. Replace kmalloc() and snprintf() with ksaprintf() to simplify code changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering DMI card names ASoC: bytcr-rt5640: register DMI names to card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 15 ++++++++++ sound/soc/soc-core.c | 56 +++++++++++++++++++++++++++++++++++ 3 files changed, 74 insertions(+) -- 2.5.0

3 4

[alsa-devel] [PATCH] ALSA: hda - fix front mic problem for a HP desktop
by Hui Wang 01 Apr '16

01 Apr '16

The front mic jack (pink color) can't detect any plug or unplug. After applying this fix, both detecting function and recording function work well. BugLink: https://bugs.launchpad.net/bugs/1564712 Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 73978c7..fefe83f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4759,6 +4759,7 @@ enum { ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, + ALC221_FIXUP_HP_FRONT_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -5401,6 +5402,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC, }, + [ALC221_FIXUP_HP_FRONT_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a19020 }, /* Front Mic */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5506,6 +5514,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: qcom: fix static checker warnings.
by Srinivas Kandagatla 31 Mar '16

31 Mar '16

This patchset fixes 2 static checker warnings reported by Dan Carpenter. Thanks, srini Srinivas Kandagatla (2): ASoC: qcom: Fix uninitialized symbol warning. ASoC: qcom: remove IS_ERR_VALUE usage on int. sound/soc/qcom/lpass-platform.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH 1/2] [media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
by Mauro Carvalho Chehab 31 Mar '16

31 Mar '16

Unfortunately, this patch caused several regressions at au0828 and snd-usb-audio, like this one: https://bugzilla.kernel.org/show_bug.cgi?id=115561 It also showed several troubles at the MC core that handles pretty poorly the memory protections and data lifetime management. So, better to revert it and fix the core before reapplying this change. This reverts commit aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources")' Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- sound/usb/Kconfig | 4 - sound/usb/Makefile | 2 - sound/usb/card.c | 14 --- sound/usb/card.h | 3 - sound/usb/media.c | 318 ----------------------------------------------- sound/usb/media.h | 72 ----------- sound/usb/mixer.h | 3 - sound/usb/pcm.c | 28 +---- sound/usb/quirks-table.h | 1 - sound/usb/stream.c | 2 - sound/usb/usbaudio.h | 6 - 11 files changed, 5 insertions(+), 448 deletions(-) delete mode 100644 sound/usb/media.c delete mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index d14bf411515b..a452ad7cec40 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,7 +15,6 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE - select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -23,9 +22,6 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. -config SND_USB_AUDIO_USE_MEDIA_CONTROLLER - bool - config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 8dca3c407f5a..2d2d122b069f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,8 +15,6 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o -snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o - snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 63244bbba8c7..3fc63583a537 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,7 +66,6 @@ #include "format.h" #include "power.h" #include "stream.h" -#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -612,11 +611,6 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; - if (quirk->media_device) { - /* don't want to fail when media_snd_device_create() fails */ - media_snd_device_create(chip, intf); - } - usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -673,14 +667,6 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } - /* - * Nice to check quirk && quirk->media_device - * need some special handlings. Doesn't look like - * we have access to quirk here - * Acceses mixer_list - */ - media_snd_device_delete(chip); - /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 34a0898e2238..71778ca4b26a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,8 +105,6 @@ struct snd_usb_endpoint { struct list_head list; }; -struct media_ctl; - struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -158,7 +156,6 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ - struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c deleted file mode 100644 index 93a50d01490c..000000000000 --- a/sound/usb/media.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * media.c - Media Controller specific ALSA driver code - * - * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> - * Copyright (c) 2016 Samsung Electronics Co., Ltd. - * - * This file is released under the GPLv2. - */ - -/* - * This file adds Media Controller support to ALSA driver - * to use the Media Controller API to share tuner with DVB - * and V4L2 drivers that control media device. Media device - * is created based on existing quirks framework. Using this - * approach, the media controller API usage can be added for - * a specific device. -*/ - -#include <linux/init.h> -#include <linux/list.h> -#include <linux/mutex.h> -#include <linux/slab.h> -#include <linux/usb.h> - -#include <sound/pcm.h> -#include <sound/core.h> - -#include "usbaudio.h" -#include "card.h" -#include "mixer.h" -#include "media.h" - -static int media_snd_enable_source(struct media_ctl *mctl) -{ - if (mctl && mctl->media_dev->enable_source) - return mctl->media_dev->enable_source(&mctl->media_entity, - &mctl->media_pipe); - return 0; -} - -static void media_snd_disable_source(struct media_ctl *mctl) -{ - if (mctl && mctl->media_dev->disable_source) - mctl->media_dev->disable_source(&mctl->media_entity); -} - -int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, - int stream) -{ - struct media_device *mdev; - struct media_ctl *mctl; - struct device *pcm_dev = &pcm->streams[stream].dev; - u32 intf_type; - int ret = 0; - u16 mixer_pad; - struct media_entity *entity; - - mdev = subs->stream->chip->media_dev; - if (!mdev) - return -ENODEV; - - if (subs->media_ctl) - return 0; - - /* allocate media_ctl */ - mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); - if (!mctl) - return -ENOMEM; - - mctl->media_dev = mdev; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; - mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; - mixer_pad = 1; - } else { - intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; - mctl->media_pad.flags = MEDIA_PAD_FL_SINK; - mixer_pad = 2; - } - mctl->media_entity.name = pcm->name; - media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); - ret = media_device_register_entity(mctl->media_dev, - &mctl->media_entity); - if (ret) - goto free_mctl; - - mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, - MAJOR(pcm_dev->devt), - MINOR(pcm_dev->devt)); - if (!mctl->intf_devnode) { - ret = -ENOMEM; - goto unregister_entity; - } - mctl->intf_link = media_create_intf_link(&mctl->media_entity, - &mctl->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); - if (!mctl->intf_link) { - ret = -ENOMEM; - goto devnode_remove; - } - - /* create link between mixer and audio */ - media_device_for_each_entity(entity, mdev) { - switch (entity->function) { - case MEDIA_ENT_F_AUDIO_MIXER: - ret = media_create_pad_link(entity, mixer_pad, - &mctl->media_entity, 0, - MEDIA_LNK_FL_ENABLED); - if (ret) - goto remove_intf_link; - break; - } - } - - subs->media_ctl = mctl; - return 0; - -remove_intf_link: - media_remove_intf_link(mctl->intf_link); -devnode_remove: - media_devnode_remove(mctl->intf_devnode); -unregister_entity: - media_device_unregister_entity(&mctl->media_entity); -free_mctl: - kfree(mctl); - return ret; -} - -void media_snd_stream_delete(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl && mctl->media_dev) { - struct media_device *mdev; - - mdev = subs->stream->chip->media_dev; - if (mdev && media_devnode_is_registered(&mdev->devnode)) { - media_devnode_remove(mctl->intf_devnode); - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - } - kfree(mctl); - subs->media_ctl = NULL; - } -} - -int media_snd_start_pipeline(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl) - return media_snd_enable_source(mctl); - return 0; -} - -void media_snd_stop_pipeline(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl) - media_snd_disable_source(mctl); -} - -int media_snd_mixer_init(struct snd_usb_audio *chip) -{ - struct device *ctl_dev = &chip->card->ctl_dev; - struct media_intf_devnode *ctl_intf; - struct usb_mixer_interface *mixer; - struct media_device *mdev = chip->media_dev; - struct media_mixer_ctl *mctl; - u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; - int ret; - - if (!mdev) - return -ENODEV; - - ctl_intf = chip->ctl_intf_media_devnode; - if (!ctl_intf) { - ctl_intf = media_devnode_create(mdev, intf_type, 0, - MAJOR(ctl_dev->devt), - MINOR(ctl_dev->devt)); - if (!ctl_intf) - return -ENOMEM; - chip->ctl_intf_media_devnode = ctl_intf; - } - - list_for_each_entry(mixer, &chip->mixer_list, list) { - - if (mixer->media_mixer_ctl) - continue; - - /* allocate media_mixer_ctl */ - mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); - if (!mctl) - return -ENOMEM; - - mctl->media_dev = mdev; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; - mctl->media_entity.name = chip->card->mixername; - mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; - mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; - mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; - media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, - mctl->media_pad); - ret = media_device_register_entity(mctl->media_dev, - &mctl->media_entity); - if (ret) { - kfree(mctl); - return ret; - } - - mctl->intf_link = media_create_intf_link(&mctl->media_entity, - &ctl_intf->intf, - MEDIA_LNK_FL_ENABLED); - if (!mctl->intf_link) { - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - kfree(mctl); - return -ENOMEM; - } - mctl->intf_devnode = ctl_intf; - mixer->media_mixer_ctl = mctl; - } - return 0; -} - -static void media_snd_mixer_delete(struct snd_usb_audio *chip) -{ - struct usb_mixer_interface *mixer; - struct media_device *mdev = chip->media_dev; - - if (!mdev) - return; - - list_for_each_entry(mixer, &chip->mixer_list, list) { - struct media_mixer_ctl *mctl; - - mctl = mixer->media_mixer_ctl; - if (!mixer->media_mixer_ctl) - continue; - - if (media_devnode_is_registered(&mdev->devnode)) { - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - } - kfree(mctl); - mixer->media_mixer_ctl = NULL; - } - if (media_devnode_is_registered(&mdev->devnode)) - media_devnode_remove(chip->ctl_intf_media_devnode); - chip->ctl_intf_media_devnode = NULL; -} - -int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface) -{ - struct media_device *mdev; - struct usb_device *usbdev = interface_to_usbdev(iface); - int ret; - - mdev = media_device_get_devres(&usbdev->dev); - if (!mdev) - return -ENOMEM; - if (!mdev->dev) { - /* register media device */ - mdev->dev = &usbdev->dev; - if (usbdev->product) - strlcpy(mdev->model, usbdev->product, - sizeof(mdev->model)); - if (usbdev->serial) - strlcpy(mdev->serial, usbdev->serial, - sizeof(mdev->serial)); - strcpy(mdev->bus_info, usbdev->devpath); - mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); - media_device_init(mdev); - } - if (!media_devnode_is_registered(&mdev->devnode)) { - ret = media_device_register(mdev); - if (ret) { - dev_err(&usbdev->dev, - "Couldn't register media device. Error: %d\n", - ret); - return ret; - } - } - - /* save media device - avoid lookups */ - chip->media_dev = mdev; - - /* Create media entities for mixer and control dev */ - ret = media_snd_mixer_init(chip); - if (ret) { - dev_err(&usbdev->dev, - "Couldn't create media mixer entities. Error: %d\n", - ret); - - /* clear saved media_dev */ - chip->media_dev = NULL; - - return ret; - } - return 0; -} - -void media_snd_device_delete(struct snd_usb_audio *chip) -{ - struct media_device *mdev = chip->media_dev; - - media_snd_mixer_delete(chip); - - if (mdev) { - if (media_devnode_is_registered(&mdev->devnode)) - media_device_unregister(mdev); - chip->media_dev = NULL; - } -} diff --git a/sound/usb/media.h b/sound/usb/media.h deleted file mode 100644 index 1dcdcdc5f7aa..000000000000 --- a/sound/usb/media.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * media.h - Media Controller specific ALSA driver code - * - * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> - * Copyright (c) 2016 Samsung Electronics Co., Ltd. - * - * This file is released under the GPLv2. - */ - -/* - * This file adds Media Controller support to ALSA driver - * to use the Media Controller API to share tuner with DVB - * and V4L2 drivers that control media device. Media device - * is created based on existing quirks framework. Using this - * approach, the media controller API usage can be added for - * a specific device. -*/ -#ifndef __MEDIA_H - -#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER - -#include <media/media-device.h> -#include <media/media-entity.h> -#include <sound/asound.h> - -struct media_ctl { - struct media_device *media_dev; - struct media_entity media_entity; - struct media_intf_devnode *intf_devnode; - struct media_link *intf_link; - struct media_pad media_pad; - struct media_pipeline media_pipe; -}; - -/* - * One source pad each for SNDRV_PCM_STREAM_CAPTURE and - * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link - * to AUDIO Source -*/ -#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) - -struct media_mixer_ctl { - struct media_device *media_dev; - struct media_entity media_entity; - struct media_intf_devnode *intf_devnode; - struct media_link *intf_link; - struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; - struct media_pipeline media_pipe; -}; - -int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface); -void media_snd_device_delete(struct snd_usb_audio *chip); -int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, - int stream); -void media_snd_stream_delete(struct snd_usb_substream *subs); -int media_snd_start_pipeline(struct snd_usb_substream *subs); -void media_snd_stop_pipeline(struct snd_usb_substream *subs); -#else -static inline int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface) - { return 0; } -static inline void media_snd_device_delete(struct snd_usb_audio *chip) { } -static inline int media_snd_stream_init(struct snd_usb_substream *subs, - struct snd_pcm *pcm, int stream) - { return 0; } -static inline void media_snd_stream_delete(struct snd_usb_substream *subs) { } -static inline int media_snd_start_pipeline(struct snd_usb_substream *subs) - { return 0; } -static inline void media_snd_stop_pipeline(struct snd_usb_substream *subs) { } -#endif -#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index f3789446ab9c..3417ef347e40 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,8 +3,6 @@ #include <sound/info.h> -struct media_mixer_ctl; - struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -24,7 +22,6 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; - struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0e4e0640c504..44d178ee9177 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,7 +35,6 @@ #include "pcm.h" #include "clock.h" #include "power.h" -#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -718,14 +717,10 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; - ret = media_snd_start_pipeline(subs); - if (ret) - return ret; - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - goto err_ret; + return ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -739,27 +734,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - ret = -EINVAL; - goto err_ret; + return -EINVAL; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - goto err_ret; + return ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - goto err_ret; + return ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; - -err_ret: - media_snd_stop_pipeline(subs); - return ret; } /* @@ -771,7 +761,6 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; - media_snd_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1232,7 +1221,6 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; - int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1246,12 +1234,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - ret = setup_hw_info(runtime, subs); - if (ret == 0) - ret = media_snd_stream_init(subs, as->pcm, direction); - if (ret) - snd_usb_autosuspend(subs->stream->chip); - return ret; + return setup_hw_info(runtime, subs); } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1260,7 +1243,6 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); - media_snd_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9d087b19c70c..c60a776e815d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,7 +2886,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ - .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 51258a15f653..c4dc577ab1bd 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,7 +36,6 @@ #include "format.h" #include "clock.h" #include "stream.h" -#include "media.h" /* * free a substream @@ -53,7 +52,6 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); - media_snd_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a161c7c1b126..b665d85555cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,9 +30,6 @@ * */ -struct media_device; -struct media_intf_devnode; - struct snd_usb_audio { int index; struct usb_device *dev; @@ -63,8 +60,6 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ - struct media_device *media_dev; - struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -115,7 +110,6 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; - bool media_device; const void *data; }; -- 2.5.5

1 0

[alsa-devel] [PATCH] ASoC: qcom: Fix some static checker warnings
by Srinivas Kandagatla 31 Mar '16

31 Mar '16

This patch fixes below static checker warning plus suggestions from Dan. sound/soc/qcom/lpass-platform.c:555 lpass_platform_pcm_new() error: uninitialized symbol 'ret'. sound/soc/qcom/lpass-platform.c 515 csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; 516 if (csubstream) { 517 if (v->alloc_dma_channel) 518 data->wrdma_ch = v->alloc_dma_channel(drvdata, 519 SNDRV_PCM_STREAM_CAPTURE); 520 521 if (IS_ERR_VALUE(data->wrdma_ch)) 522 goto capture_alloc_err; wrdma_ch is an int so this should just be: if (data->wrdma_ch < 0) { ret = data->wrdma_ch; goto capture_alloc_err; } Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-platform.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 6e86654..db000c6 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -474,7 +474,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) struct lpass_data *drvdata = snd_soc_platform_get_drvdata(soc_runtime->platform); struct lpass_variant *v = drvdata->variant; - int ret; + int ret = -EINVAL; struct lpass_pcm_data *data; size_t size = lpass_platform_pcm_hardware.buffer_bytes_max; @@ -491,7 +491,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->rdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_PLAYBACK); - if (IS_ERR_VALUE(data->rdma_ch)) + if (data->rdma_ch < 0) return data->rdma_ch; drvdata->substream[data->rdma_ch] = psubstream; @@ -518,8 +518,10 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->wrdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_CAPTURE); - if (IS_ERR_VALUE(data->wrdma_ch)) + if (data->wrdma_ch < 0) { + ret = data->wrdma_ch; goto capture_alloc_err; + } drvdata->substream[data->wrdma_ch] = csubstream; -- 2.5.0

2 2

[alsa-devel] [PATCH v2 1/5] Sound: SOC: TAS571x: added missing register literals
by Petr Kulhavy 31 Mar '16

31 Mar '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to register 0x25 Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> sound/soc/codecs/tas571x.c | 4 ++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..346d3da 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..cf800c3 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

4 11

[alsa-devel] [PATCH v4 1/3] ASoC: tas571x: added missing register literals
by Petr Kulhavy 31 Mar '16

31 Mar '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to the register 0x25. Added volatile and read-only register tables into tas5711_regmap_config and tas5717_regmap_config. The chip has 256 registers in total. But from address 0x29 on (0x26 to 0x28 are reserved) the register width varies between 20, 12 and 8 bytes, which the register map cannot represent. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> v3: added volatile and read-only register tables into regmap configs v4: no change, the set v3 resubmitted without the already accepted patches sound/soc/codecs/tas571x.c | 28 ++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 50 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index aafee9bbe01a..ef6c8d9b251a 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -259,6 +263,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -278,6 +302,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -332,6 +358,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471232cd..cf800c364f0f 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: nau8825: assign DAC Ch to match headset L/R
by John Hsu 31 Mar '16

31 Mar '16

The default value of DAC channel select is reverse in codec. For normal usage, switch the channel select when codec bootup. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 5 +++++ sound/soc/codecs/nau8825.h | 6 ++++++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3e7bee2..1a99a25 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -905,6 +905,11 @@ static void nau8825_init_regs(struct nau8825 *nau8825) NAU8825_RDAC_CLK_DELAY_MASK | NAU8825_RDAC_VREF_MASK, (0x2 << NAU8825_RDAC_CLK_DELAY_SFT) | (0x3 << NAU8825_RDAC_VREF_SFT)); + /* Config L/R channel */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACR_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_R); } static const struct regmap_config nau8825_regmap_config = { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 5fe009d..fcf0d0a 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -256,9 +256,15 @@ /* DACL_CTRL (0x33) */ #define NAU8825_DACL_CH_SEL_SFT 9 +#define NAU8825_DACL_CH_SEL_MASK (0x1 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_L (0x0 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_R (0x1 << NAU8825_DACL_CH_SEL_SFT) /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 +#define NAU8825_DACR_CH_SEL_MASK (0x1 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_L (0x0 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_R (0x1 << NAU8825_DACR_CH_SEL_SFT) /* CLASSG_CTRL (0x50) */ #define NAU8825_CLASSG_TIMER_SFT 8 -- 2.6.4

3 4

[alsa-devel] [PATCH] ALSA: usb-audio: Fix double-free in snd_usb_add_audio_stream()
by Vladis Dronov 31 Mar '16

31 Mar '16

Hello, Takashi, Jaroslav, all, Please, see the research and the following patch on a double-free bug in [snd-usb-audio]. 1) The upstream commits 0f886ca1, 902eb7fd and 447d6275f (many thanks to Takashi Iwai) revealed that there is a double-free bug in [snd-usb-audio] module due to alloc/free logic flaw in snd_usb_add_audio_stream() function. A double-free leads to kernel structure/list/slab corruptions and shortly to null-deref, GPF or lockup. 2.1) Let me describe what happens with the current code in usb_audio_probe(), create_fixed_stream_quirk() and snd_usb_add_audio_stream(): > [ sound/usb/card.c ] > static int usb_audio_probe(struct usb_interface *intf, > const struct usb_device_id *usb_id) > { > ... > if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { > /* need some special handlings */ > err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk); > if (err < 0) > goto __error; > } > ... > __error: > if (chip) { > if (!chip->num_interfaces) > snd_card_free(chip->card); Somewhere in the middle of usb_audio_probe() the function snd_usb_create_quirk() is called, and if it returns with an error and no interfaces were created, the sound card is destroyed with "snd_card_free(chip->card)". 2.2) create_fixed_stream_quirk() is called from snd_usb_create_quirk(): > [ sound/usb/quirks.c ] > static int create_fixed_stream_quirk(struct snd_usb_audio *chip, > struct usb_interface *iface, > struct usb_driver *driver, > const struct snd_usb_audio_quirk *quirk) > { > struct audioformat *fp; > struct usb_host_interface *alts; > struct usb_interface_descriptor *altsd; > ... > fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); > ... > (*) stream = (fp->endpoint & USB_DIR_IN) > (*) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; > (*) err = snd_usb_add_audio_stream(chip, stream, fp); > (*) if (err < 0) > (*) goto error; > > if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber || > fp->altset_idx >= iface->num_altsetting) { > err = -EINVAL; > goto error; > } > alts = &iface->altsetting[fp->altset_idx]; > altsd = get_iface_desc(alts); > if (altsd->bNumEndpoints < 1) { > err = -EINVAL; > goto error; > } > ... > error: > kfree(fp); > kfree(rate_table); > return err; > } 2.3) *fp is allocated and passed to snd_usb_add_audio_stream() where snd_usb_init_substream() is called: > [ sound/usb/stream.c ] > int snd_usb_add_audio_stream(struct snd_usb_audio *chip, > int stream, > struct audioformat *fp) > { > struct snd_usb_stream *as; > struct snd_usb_substream *subs; > struct snd_pcm *pcm; > ... > /* create a new pcm */ > as = kzalloc(sizeof(*as), GFP_KERNEL); > ... > snd_usb_init_substream(as, stream, fp); 2.4) In turn snd_usb_init_substream() adds audioformat *fp by its &fp->list to substream's fmt_list list: > [ sound/usb/stream.c ] > static void snd_usb_init_substream(struct snd_usb_stream *as, > int stream, > struct audioformat *fp) > { > struct snd_usb_substream *subs = &as->substream[stream]; > > INIT_LIST_HEAD(&subs->fmt_list); > ... > list_add_tail(&fp->list, &subs->fmt_list); > subs->num_formats++; 2.5) Things go bad from this point in case snd_usb_add_audio_stream() or the caller go the error path. The bug is that the caller frees (see "error: kfree(fp);" code) *fp on the error path, BUT the pointer to the already-freed memory remains in the substream's fmt_list list. The double-free happens after create_fixed_stream_quirk() returns with an error and usb_audio_probe() calls snd_card_free()->...->snd_usb_audio_pcm_free()->free_substream(). As subs->fmt_list is already corrupted, iterating it with list_for_each_entry_safe() leads to any and unpredictable results. > static void free_substream(struct snd_usb_substream *subs) > { > struct audioformat *fp, *n; > ... > list_for_each_entry_safe(fp, n, &subs->fmt_list, list) { > ... > kfree(fp); 3.1) The crash is reproduceable by faking the USB device with no endpoints as described in https://bugzilla.redhat.com/show_bug.cgi?id=1283355 and https://bugzilla.redhat.com/show_bug.cgi?id=1283358. Please see as a proof the following kernel log with debug printing added to the code. First, *fp is added to fmt_list in snd_usb_init_substream(): [ 322.797223] usb 1-1: new full-speed USB device number 2 using xhci_hcd [ 322.974083] usb 1-1: config 1 interface 0 altsetting 0 has 3 endpoint descriptors, different from the interface descriptor's value: 0 [ 322.987031] usb 1-1: New USB device found, idVendor=045e, idProduct=0283 [ 322.998318] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 323.083913] media: Linux media interface: v0.10 [ 323.231249] init_substream: add_tail() fp=ffff88003364ba80 fp->list.next=ffff8800b1e898b8 fp->list.prev=ffff8800b1e898b8 fmt_list=ffff8800b1e898b8 fmt_list.next=ffff88003364ba80 prev=ffff88003364ba80 num_formats=1 As you can see we have a correct list here with head at fmt_list=ffff8800b1e898b8 and single element fp=ffff88003364ba80. 3.2) The code finds out that there are too few endpoints present and goes the error path (to the "error:" label): [ 323.307927] usb 1-1: too few endpoints [ 323.312964] trace-before-free: substream-0 ffff8800b1e89818 numf 1 fmt_list ffff8800b1e898b8 next ffff88003364ba80 [ 323.353759] fp=ffff88003364ba80 next=ffff8800b1e898b8 prev=ffff8800b1e898b8 rate=(null) chmap=(null) [ 323.362118] struct sane As you can see the substream's fmt_list is sane at this point. 3.3) After "kfree(fp)" in the error path of create_fixed_stream_quirk() *fp is freed, BUT the pointer to the freed memory remains in fmt_list. After *fp is freed the list is corrupted and contains trash: [ 323.371752] KFREE(fp) ffff88003364ba80 [ 323.380383] trace-after-free: substream-0 ffff8800b1e89818 numf 1 fmt_list ffff8800b1e898b8 next ffff88003364ba80 [ 323.400003] fp=ffff88003364ba80 next=ffff88003364bf80 prev=ffff8800b1e898b8 rate=(null) chmap=(null) [ 323.406786] fp=ffff88003364bf80 next= (null) prev= (null) rate=(null) chmap=(null) [ 323.422211] next == NULL: FAIL, struct INSANE [ 323.436706] KFREE(rate_table) (null) 3.4) After error return from create_fixed_stream_quirk() the sound card is destroyed with "snd_card_free(chip->card)" in usb_audio_probe(). In the end free_substream() is called: [ 323.511256] usb 1-1: snd_usb_create_quirk() failed: -22 [ 323.565108] list_for_each_entry_safe(): fp=ffff88003364ba80 n=ffff88003364bf80 [ 323.586337] kfree fp ffff88003364ba80 <<< DOUBLE-FREE [ 323.588509] loop-end: fp=ffff88003364ba80 n=ffff88003364bf80 [ 323.599969] list_for_each_entry_safe(): fp=ffff88003364bf80 n=(null) [ 323.610460] kfree fp ffff88003364bf80 <<< FREEING SOMEONE ELSE'S MEMORY [ 323.613181] loop-end: fp=ffff88003364bf80 n=(null) <<< NULL-PTR DEREF ... [ 324.247113] BUG: unable to handle kernel NULL pointer dereference at (null) [ 324.247533] IP: [<ffffffffc02d40ef>] free_substream.part.0+0xef/0x100 [snd_usb_audio] [ 324.248088] PGD 0 [ 324.248088] Oops: 0000 [#1] SMP [ 324.248088] Modules linked in: snd_usb_audio(+) snd_usbmidi_lib snd_hwdep ... [ 324.248088] CPU: 2 PID: 767 Comm: systemd-udevd Not tainted 4.5.0-vladis #25 [ 324.248088] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.8.2-0-g33fbe13 by qemu-project.org 04/01/2014 [ 324.248088] task: ffff880034718000 ti: ffff8800b2fbc000 task.ti: ffff8800b2fbc000 [ 324.248088] RIP: 0010:[<ffffffffc02d40ef>] [<ffffffffc02d40ef>] free_substream.part.0+0xef/0x100 [snd_usb_audio] [ 324.248088] RSP: 0018:ffff8800b2fbf898 EFLAGS: 00010217 ... [ 324.248088] Call Trace: [ 324.248088] [<ffffffffc02d43fa>] snd_usb_audio_pcm_free+0x9a/0xa0 [snd_usb_audio] [ 324.248088] [<ffffffffc028d982>] snd_pcm_free+0x32/0xa0 [snd_pcm] [ 324.248088] [<ffffffffc028da02>] snd_pcm_dev_free+0x12/0x20 [snd_pcm] [ 324.248088] [<ffffffffc027d279>] __snd_device_free+0x29/0x70 [snd] [ 324.248088] [<ffffffffc027d660>] snd_device_free_all+0x30/0x60 [snd] [ 324.248088] [<ffffffffc02777a4>] release_card_device+0x34/0x90 [snd] [ 324.248088] [<ffffffff815ae2b2>] device_release+0x32/0x90 [ 324.248088] [<ffffffff81455f8a>] kobject_release+0x7a/0x190 [ 324.248088] [<ffffffff81455e47>] kobject_put+0x27/0x50 [ 324.248088] [<ffffffff815ae5f7>] put_device+0x17/0x20 [ 324.248088] [<ffffffffc0277b57>] snd_card_free+0x67/0x90 [snd] [ 324.248088] [<ffffffffc02c4f14>] usb_audio_probe+0x754/0x9d0 [snd_usb_audio] ... 4.1) The suggested patch consists of 2 changes. First, I suppose we should move the code in create_fixed_stream_quirk() marked as "(*)" (see above) after "if (altsd->bNumEndpoints < 1)" check. This way no allocations is done if we go an error path. I've checked that fp->iface and fp->altset_idx are not changed in snd_usb_add_audio_stream() and functions called from it, so it is safe to swap these 2 pieces of code. 4.2) The problem with double-free still remains. I've verified that all the callers of snd_usb_add_audio_stream() free *fp in case of error: $ git grep snd_usb_add_audio_stream sound/usb/quirks.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) > * goto error; > * error: > * kfree(fp); > * kfree(rate_table); > * return err; sound/usb/quirks.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) { > * kfree(fp); > * return err; > * } sound/usb/stream.c: err = snd_usb_add_audio_stream(chip, stream, fp); > * err = snd_usb_add_audio_stream(chip, stream, fp); > * if (err < 0) { > * kfree(fp->rate_table); > * kfree(fp->chmap); > * kfree(fp); > * return err; > * } This means that snd_usb_add_audio_stream() should remove *fp from the substream's fmt_list list on the error path, if it was already added. Such places are: > int snd_usb_add_audio_stream(struct snd_usb_audio *chip, int stream, struct audioformat *fp) > { > struct snd_usb_stream *as; > struct snd_usb_substream *subs; > struct snd_pcm *pcm; > int err; > > list_for_each_entry(as, &chip->pcm_list, list) { > subs = &as->substream[stream]; > ... > list_add_tail(&fp->list, &subs->fmt_list); <<< ADDING fp HERE > subs->num_formats++; > return 0; <<< NO ERROR PATH HERE > } > } > > /* look for an empty stream */ > list_for_each_entry(as, &chip->pcm_list, list) { > subs = &as->substream[stream]; > ... > snd_usb_init_substream(as, stream, fp); <<< ADDING fp HERE, IF add_chmap() FAILS > return add_chmap(as->pcm, stream, subs); <<< fp SHOULD BE REMOVED FROM fmt_list > } > > /* create a new pcm */ > as = kzalloc(sizeof(*as), GFP_KERNEL); > err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, > ... > if (err < 0) { <<< fp IS NOT ADDED YET HERE, SO FINE > kfree(as); > return err; > } > ... > snd_usb_init_substream(as, stream, fp); <<< ADDING fp HERE > ... <<< IF add_chmap() FAILS fp SHOULD > return add_chmap(pcm, stream, &as->substream[stream]); <<< BE REMOVED FROM fmt_list > } add_chmap() itself does not add anything to fmt_list list, so we indeed need to remove only the single list element from the list. Having all the above in mind, the patch follows. 4.3) How to handle possible error paths after successful call to snd_usb_add_audio_stream() in create_fixed_stream_quirk() is d iscussable. Properly it should be like the below, but I believe it is overcomplication here would and stick to a simple error_after_add_audio_stream: label: > error2: > snd_usb_del_audio_stream(...something...); > error: > kfree(fp); > kfree(rate_table); > return err; Best regards, Vladis Dronov | Red Hat, Inc. | Product Security Engineer

2 9

[alsa-devel] [PATCH V3 0/2] ASoC: share more product information with user space
by han.lu＠intel.com 31 Mar '16

31 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, so user space utils such as PA and UCM can distinguish different products. 1. Add a core API. 2. Apply the API to bytcr-rt5640 driver. changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering DMI card names ASoC: bytcr-rt5640: register DMI names to card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 15 ++++++++++ sound/soc/soc-core.c | 53 +++++++++++++++++++++++++++++++++++ 3 files changed, 71 insertions(+) -- 2.5.0

3 4

[alsa-devel] [PATCH v2] ALSA: hda - Bind with i915 only when Intel graphics is present
by Takashi Iwai 31 Mar '16

31 Mar '16

On Skylake and onwards, the HD-audio controller driver needs to bind with i915 for having the control of power well audio domain before actually probing the codec. This leads to the load of i915 driver from the audio driver side. But, there are systems that have no Intel graphics but Nvidia or AMD GPU, although they still use HD-audio bus for the onboard audio codecs. On these, loading the i915 driver is nothing but a useless memory and CPU consumption. A simple way to avoid it is just to look for the Intel graphics PCI entry beforehand, and try to bind with i915 only when such an entry is found. Currently, it assumes the PCI display class. If another class appears, this needs to be extended (although it's very unlikely). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Check also PCI vendor id to match only with Intel Now get out of a series, as this should be applied to 4.6, too. sound/hda/hdac_i915.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index d0da2508823e..6800e0c5a38f 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -287,6 +287,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops } EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); +/* check whether intel graphics is present */ +static bool i915_gfx_present(void) +{ + static struct pci_device_id ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), + .class = PCI_BASE_CLASS_DISPLAY << 16, + .class_mask = 0xff << 16 }, + {} + }; + return pci_dev_present(ids); +} + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -309,6 +321,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (WARN_ON(hdac_acomp)) return -EBUSY; + if (!i915_gfx_present()) + return -ENODEV; + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); if (!acomp) return -ENOMEM; -- 2.7.4

1 0

[alsa-devel] ALSA license
by Li, Steve 31 Mar '16

31 Mar '16

I am trying to use ALSA in a commercial application. I see that in the introduction page, ALSA is licensed under GPL and LGPL. Can you tell me which part of ALSA is licensed under LGPL? What is the best way to find out this information? Thanks Steve ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

3 3

[alsa-devel] [PATCH] ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
by Maruthi Srinivas Bayyavarapu 31 Mar '16

31 Mar '16

This commit fixes garbled audio on Polaris-10/11 variants Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Acked-by: Christian König <christian.koenig(a)amd.com> --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2624cfe..b680be0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2361,6 +2361,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaae8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaae0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaaf0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dpcm: prevent error for paths including static links
by Piotr Stankiewicz 31 Mar '16

31 Mar '16

According to the DPCM documentation using static codec <-> codec links along dynamic ones is a valid use case, however this causes errors of the following form to be printed to the kernel log: ASoC: can't get [playback|caputure] BE for <widget name> ASoC: no BE found for <widget name> This happens when setting up, e.g. a route starting with a dynamic DAI, which passes through a static DAI (say, "Some FE" -> "Some BE" -> "Codec to Modem link"). All DAPM widgets involved in that path will be passed to dpcm_add_paths in order to establish FE <-> BE connections, which will result in dpcm_get_be trying to locate back-ends for dynamic, as well as static links, causing a spurious error to be logged for the latter. This patch changes dpcm_get_be to look up a non front-end runtime for a stream widget, and renames it accordingly. Like this both dynamic and static links can be handled properly in dpcm_add_paths, i.e. an actual missing BE can be distinguished from a normal link. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1af4f23..ff5ea7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1220,47 +1220,47 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) } } -/* get BE for DAI widget and stream */ -static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, +/* get runtime for DAI widget and stream */ +static struct snd_soc_pcm_runtime *dpcm_get_rtd(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { - struct snd_soc_pcm_runtime *be; + struct snd_soc_pcm_runtime *rtd; int i; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - list_for_each_entry(be, &card->rtd_list, list) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (!be->dai_link->no_pcm) + if (rtd->dai_link->dynamic) continue; - if (be->cpu_dai->playback_widget == widget) - return be; + if (rtd->cpu_dai->playback_widget == widget) + return rtd; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; if (dai->playback_widget == widget) - return be; + return rtd; } } } else { - list_for_each_entry(be, &card->rtd_list, list) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (!be->dai_link->no_pcm) + if (rtd->dai_link->dynamic) continue; - if (be->cpu_dai->capture_widget == widget) - return be; + if (rtd->cpu_dai->capture_widget == widget) + return rtd; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; if (dai->capture_widget == widget) - return be; + return rtd; } } } - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", + dev_err(card->dev, "ASoC: can't get %s runtime for %s\n", stream ? "capture" : "playback", widget->name); return NULL; } @@ -1367,15 +1367,15 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; } - /* is there a valid BE rtd for this widget */ - be = dpcm_get_be(card, list->widgets[i], stream); + /* is there a valid rtd for this widget */ + be = dpcm_get_rtd(card, list->widgets[i], stream); if (!be) { - dev_err(fe->dev, "ASoC: no BE found for %s\n", + dev_err(fe->dev, "ASoC: no runtime found for %s\n", list->widgets[i]->name); continue; } - /* make sure BE is a real BE */ + /* check if the runtime is a BE */ if (!be->dai_link->no_pcm) continue; -- 1.7.10.4

3 6

[alsa-devel] [PATCH] ASoC: nau8825: Fix jack detection across suspend
by Ben Zhang 31 Mar '16

31 Mar '16

Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power. Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 126 +++++++++++++++++++++++++-------------------- 1 file changed, 71 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3e7bee2..cb08a35 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -343,9 +343,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), - /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + /* ADC for button press detection. A dapm supply widget is used to + * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON + * during suspend. + */ + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), @@ -607,6 +610,16 @@ static bool nau8825_is_jack_inserted(struct regmap *regmap) static void nau8825_restart_jack_detection(struct regmap *regmap) { + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + /* this will restart the entire jack detection process including MIC/GND * switching and create interrupts. We have to go from 0 to 1 and back * to 0 to restart. @@ -728,7 +741,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) struct regmap *regmap = nau8825->regmap; int active_irq, clear_irq = 0, event = 0, event_mask = 0; - regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) { + dev_err(nau8825->dev, "failed to read irq status\n"); + return IRQ_NONE; + } if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == NAU8825_JACK_EJECTION_DETECTED) { @@ -1198,33 +1214,74 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return ret; } } - - ret = regcache_sync(nau8825->regmap); - if (ret) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } } - break; case SND_SOC_BIAS_OFF: if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); - - regcache_mark_dirty(nau8825->regmap); break; } return 0; } +#ifdef CONFIG_PM +static int nau8825_suspend(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + disable_irq(nau8825->irq); + regcache_cache_only(nau8825->regmap, true); + regcache_mark_dirty(nau8825->regmap); + + return 0; +} + +static int nau8825_resume(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + /* The chip may lose power and reset in S3. regcache_sync restores + * register values including configurations for sysclk, irq, and + * jack/button detection. + */ + regcache_cache_only(nau8825->regmap, false); + regcache_sync(nau8825->regmap); + + /* Check the jack plug status directly. If the headset is unplugged + * during S3 when the chip has no power, there will be no jack + * detection irq even after the nau8825_restart_jack_detection below, + * because the chip just thinks no headset has ever been plugged in. + */ + if (!nau8825_is_jack_inserted(nau8825->regmap)) { + nau8825_eject_jack(nau8825); + snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); + } + + enable_irq(nau8825->irq); + + /* Run jack detection to check the type (OMTP or CTIA) of the headset + * if there is one. This handles the case where a different type of + * headset is plugged in during S3. This triggers an IRQ iff a headset + * is already plugged in. + */ + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} +#else +#define nau8825_suspend NULL +#define nau8825_resume NULL +#endif + static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, + .suspend = nau8825_suspend, + .resume = nau8825_resume, .controls = nau8825_controls, .num_controls = ARRAY_SIZE(nau8825_controls), @@ -1334,16 +1391,6 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); - ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "nau8825", nau8825); @@ -1411,36 +1458,6 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM_SLEEP -static int nau8825_suspend(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - disable_irq(client->irq); - regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); - - return 0; -} - -static int nau8825_resume(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); - - return 0; -} -#endif - -static const struct dev_pm_ops nau8825_pm = { - SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) -}; - static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1467,7 +1484,6 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), - .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, -- 2.8.0.rc3.226.g39d4020

3 2

[alsa-devel] [PATCH 0/8] ASoC: Intel: Skylake: Add Broxton DSP support
by Vinod Koul 31 Mar '16

31 Mar '16

Broxton DSP is similar to Skylake with similar DSP capabilities but differs in a few subtle ways. It doesn't have Code Loader DMA and use HDA DMA. The DSP power up/down routines are also bit different in the two platforms. Also the FW needs to be downloaded only once and not on every power sequence as it is cached by DSP in protected memory region. All these changes for new DSP ops are abstracted in a new BXTN-dsp ops file. This patch series starts with moving code around to prepare for new code and also fix whitespace issue in existing code. Since we have different loader ops we move Skylake ones to newly defined ops. Then we add device IDs and lastly the broxton ops. Jayachandran B (1): ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core Jeeja KP (4): ASoC: Intel: add dmabuffer to common sst_dsp ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops ASoC: hdac_hdmi: Add broxton device ID ASoC: Intel: Bxtn: Add Broxton DSP support Senthilnathan Veppur (1): ASoC: Intel: Bxtn: Add Broxton PCI ID Vinod Koul (2): ASoC: Intel: Skylake: Move module id defines ASoC: Intel: Skylake: Fix whitepsace issues sound/soc/codecs/hdac_hdmi.c | 1 + sound/soc/intel/Kconfig | 1 + sound/soc/intel/common/sst-dsp-priv.h | 1 + sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/bxt-sst.c | 328 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 182 +++++++++++++++++- sound/soc/intel/skylake/skl-sst-dsp.c | 14 +- sound/soc/intel/skylake/skl-sst-dsp.h | 20 +- sound/soc/intel/skylake/skl-sst.c | 3 - sound/soc/intel/skylake/skl.c | 7 + sound/soc/intel/skylake/skl.h | 12 +- 11 files changed, 549 insertions(+), 22 deletions(-) create mode 100644 sound/soc/intel/skylake/bxt-sst.c -- 1.9.1

2 16

[alsa-devel] [PATCH] ASoC: intel: add function stub when ACPI is not enabled
by Randy Dunlap 31 Mar '16

31 Mar '16

From: Randy Dunlap <rdunlap(a)infradead.org> Add function stub for "sst_acpi_find_name_from_hid()" when CONFIG_ACPI is not enabled so that the driver will build successfully. This fixes the following build errors: (loadable module) ERROR: "sst_acpi_find_name_from_hid" [sound/soc/intel/boards/snd-soc-sst-bytcr-rt5640.ko] undefined! (or built-in) bytcr_rt5640.c:(.text+0x26fc52): undefined reference to `sst_acpi_find_name_from_hid' Reported-by: Borislav Petkov <bp(a)alien8.de> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/common/sst-acpi.h | 9 +++++++++ 1 file changed, 9 insertions(+) --- linux-next-20160308.orig/sound/soc/intel/common/sst-acpi.h +++ linux-next-20160308/sound/soc/intel/common/sst-acpi.h @@ -12,10 +12,19 @@ * */ +#include <linux/kconfig.h> +#include <linux/stddef.h> #include <linux/acpi.h> /* translation fron HID to I2C name, needed for DAI codec_name */ +#if IS_ENABLED(CONFIG_ACPI) const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); +#else +inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +{ + return NULL; +} +#endif /* acpi match */ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines);

2 1

[alsa-devel] [PATCH] ASoC: wm8960: Add I2C dependency
by Fabio Estevam 30 Mar '16

30 Mar '16

Building wm8960 codec without I2C support causes build issues: sound/soc/codecs/wm8960.c:1388:1: warning: data definition has no type or storage class module_i2c_driver(wm8960_i2c_driver); ^ sound/soc/codecs/wm8960.c:1388:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/wm8960.c:1388:1: warning: parameter names (without types) in function declaration sound/soc/codecs/wm8960.c:1378:26: warning: 'wm8960_i2c_driver' defined but not used [-Wunused-variable] static struct i2c_driver wm8960_i2c_driver = { , so add the I2C dependency to fix this problem. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ba835b6..09e8c39 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -911,6 +911,7 @@ config SND_SOC_WM8955 config SND_SOC_WM8960 tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate -- 1.9.1

2 1

[alsa-devel] [RFC][PATCH] ALSA: firewire-lib: support Linux tracing to dump a part of packet data
by Takashi Sakamoto 30 Mar '16

30 Mar '16

When audio and music units have some quirks in their sequence of packet, it's really hard for non-owners to identify the quirks. To handle the quirks, developers need a dump for sequence of packets at least, however it's difficult to users who have no knowledge and equipment for this purpose. This commit adds tracepoints for this situation. When users encounter the issue, they can dump a part of packet data via Linux tracing framework as long as using drivers in ALSA firewire stack. This commit adds 'snd-firewire-lib' subsystem with 'in_packet' and 'out_packet' events. In the events, the content of CIP headers, the number of included quadlets and the index of packet managed by this module are recorded per packet. This is a sample: $ trace-cmd record -e snd_firewire_lib:out_packet -e snd_firewire_lib:in_packet /sys/kernel/tracing/events/snd_firewire_lib/out_packet/filter /sys/kernel/tracing/events/snd_firewire_lib/in_packet/filter Hit Ctrl^C to stop recording ^C $ trace-cmd report trace.dat ... <idle>-0 [002] 40398.221702: out_packet: 00070008 9001427a: 058: 31 <idle>-0 [002] 40398.221703: in_packet: 01020010 9001ffff: 002: 15 <idle>-0 [002] 40398.221703: out_packet: 00070010 9001ffff: 002: 32 <idle>-0 [002] 40398.223679: in_packet: 010b0010 900157e4: 090: 16 <idle>-0 [002] 40398.223681: out_packet: 00070010 900157e4: 058: 33 <idle>-0 [002] 40398.223681: in_packet: 010b0018 9001714f: 090: 17 ... One line represent one packet. The legend for the last four fields is: - The first quadlet of CIP header - The second quadlet of CIP header - The number of included quadlets - The index of packet inner buffer maintained by this module Currently, when detecting packet discontinuity, this module stops packet streaming. This is reasonable to packet streaming implementation. However, to identify the quirks, packet streaming should continue to dump enough sequence of packet. This commit adds a condition statement and a marker for this purpose. --- sound/firewire/Makefile | 3 ++ sound/firewire/amdtp-stream-trace.h | 68 +++++++++++++++++++++++++++++++++++++ sound/firewire/amdtp-stream.c | 17 +++++++++- 3 files changed, 87 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/amdtp-stream-trace.h diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 003c090..0ee1fb1 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,3 +1,6 @@ +# To find a header included by define_trace.h. +CFLAGS_amdtp-stream.o := -I$(src) + snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp-stream.o amdtp-am824.o snd-isight-objs := isight.o diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h new file mode 100644 index 0000000..316d495 --- /dev/null +++ b/sound/firewire/amdtp-stream-trace.h @@ -0,0 +1,68 @@ +/* + * amdtp-stream-trace.h - Linux tracing application for ALSA AMDTP engine + * + * Copyright (c) 2016 Takashi Sakamoto + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#undef TRACE_SYSTEM +#define TRACE_SYSTEM snd_firewire_lib + +#if !defined(_AMDTP_STREAM_TRACE_H) || defined(TRACE_HEADER_MULTI_READ) +#define _AMDTP_STREAM_TRACE_H + +#include <linux/tracepoint.h> + +TRACE_EVENT(in_packet, + TP_PROTO(u32 cip_header0, u32 cip_header1, unsigned int payload_quadlets, unsigned int index), + TP_ARGS(cip_header0, cip_header1, payload_quadlets, index), + TP_STRUCT__entry( + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->cip_header0 = cip_header0; + __entry->cip_header1 = cip_header1; + __entry->payload_quadlets = payload_quadlets; + __entry->index = index; + ), + TP_printk( + "%08x %08x: %03u: %02u", + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->index) +); + +TRACE_EVENT(out_packet, + TP_PROTO(u32 cip_header0, u32 cip_header1, unsigned int payload_quadlets, unsigned int index), + TP_ARGS(cip_header0, cip_header1, payload_quadlets, index), + TP_STRUCT__entry( + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->cip_header0 = cip_header0; + __entry->cip_header1 = cip_header1; + __entry->payload_quadlets = payload_quadlets; + __entry->index = index; + ), + TP_printk( + "%08x %08x: %03u: %02u", + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->index) +); + +#endif + +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#undef TRACE_INCLUDE_FILE +#define TRACE_INCLUDE_FILE amdtp-stream-trace +#include <trace/define_trace.h> diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ed29026..152375f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -15,6 +15,10 @@ #include <sound/pcm_params.h> #include "amdtp-stream.h" +/* Always support tracepoints enabled by Linux tracing subsystem. */ +#define CREATE_TRACE_POINTS +#include "amdtp-stream-trace.h" + #define TICKS_PER_CYCLE 3072 #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) @@ -426,6 +430,10 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; + + trace_out_packet(be32_to_cpu(buffer[0]), be32_to_cpu(buffer[1]), + payload_length / 4, s->packet_index); + if (queue_out_packet(s, payload_length, false) < 0) return -EIO; @@ -451,6 +459,9 @@ static int handle_in_packet(struct amdtp_stream *s, cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); + trace_in_packet(cip_header[0], cip_header[1], payload_quadlets, + s->packet_index); + /* * This module supports 'Two-quadlet CIP header with SYT field'. * For convenience, also check FMT field is AM824 or not. @@ -523,7 +534,11 @@ static int handle_in_packet(struct amdtp_stream *s, dev_err(&s->unit->device, "Detect discontinuity of CIP: %02X %02X\n", s->data_block_counter, data_block_counter); - return -EIO; + if (!trace_in_packet_enabled()) + return -EIO; + + /* To identifying this situation. */ + trace_in_packet(0xffffffff, 0xffffffff, 999, 99); } pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt); -- 2.7.3

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[alsa-devel] [PATCH] ASoC: rt5677: Avoid duplicate the same test in each switch case
by Axel Lin 30 Mar '16

30 Mar '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 24 +++++------------------- 1 file changed, 5 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 33e290b..b3f1db5 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1241,60 +1241,46 @@ static int rt5677_dmic_use_asrc(struct snd_soc_dapm_widget *source, regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO1_CLK_SEL_MASK) >> RT5677_AD_STO1_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 10: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO2_CLK_SEL_MASK) >> RT5677_AD_STO2_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 9: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO3_CLK_SEL_MASK) >> RT5677_AD_STO3_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 8: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO4_CLK_SEL_MASK) >> RT5677_AD_STO4_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 7: regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_MONOL_CLK_SEL_MASK) >> RT5677_AD_MONOL_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 6: regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_MONOR_CLK_SEL_MASK) >> RT5677_AD_MONOR_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; default: - break; + return 0; } + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + return 0; } -- 2.1.4

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[alsa-devel] Applied "ASoC: wm8960: Depends on I2C" to the asoc tree
by Mark Brown 30 Mar '16

30 Mar '16

The patch ASoC: wm8960: Depends on I2C has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3fcdfc9dad07c2a6ea19dfb98a2151cd86d06c6a Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 30 Mar 2016 11:02:49 -0700 Subject: [PATCH] ASoC: wm8960: Depends on I2C Now that this is directly user selectable it needs to care about its dependencies. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 196101b2eab5..3ebf29bc87d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -910,6 +910,7 @@ config SND_SOC_WM8955 config SND_SOC_WM8960 tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate -- 2.8.0.rc3

1 0

[alsa-devel] [PATCH 0/2] ALSA: bebob/fireworks: remove module parameters to reduce maintenance work
by Takashi Sakamoto 30 Mar '16

30 Mar '16

Hi, This patchset removes some module parameters from ALSA bebob and fireworks driver. These two modules have three parameters for users to assign preferable number and strings to a sound card as some modules in ALSA kernel land. However, they're not used so widely, and practically I suspect their advantages. To simplify and unify the shape of modules in ALSA firewire stack, I'd like to remove them. Takashi Sakamoto (2): ALSA: bebob: remove module parameters to reduce maintenance work ALSA: fireworks: remove module parameters to reduce maintenance work sound/firewire/bebob/bebob.c | 49 +++++------------------------------- sound/firewire/bebob/bebob.h | 1 - sound/firewire/fireworks/fireworks.c | 43 +++---------------------------- sound/firewire/fireworks/fireworks.h | 1 - 4 files changed, 10 insertions(+), 84 deletions(-) -- 2.7.3

2 4

[alsa-devel] [PATCH] ASoC: nau8825: reduce standby power consumption
by John Hsu 30 Mar '16

30 Mar '16

Decrease internal clock frequency for power saving when standby. But clock divider needs restore when MCLK as system clock in playback. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3e7bee2..4c8572c 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1116,6 +1116,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + /* MCLK not changed by clock tree */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0); ret = nau8825_mclk_prepare(nau8825, freq); if (ret) return ret; @@ -1126,6 +1129,13 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + /* Decrease the VCO frequency for power saving */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; -- 2.6.4

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[alsa-devel] [PATCH v8 0/6] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 30 Mar '16

30 Mar '16

The only change to v7: Remove the unused variable from tda998x_configure_audio(). Not much have changed since v6[1] patch series. I did everything addressed every comment, mostly by robots :), to previous series and there was one bug fixed in tda998x side. Unless there is any significant reaction to this series, I am going to put it to rest for a while now. I have other things to do. There is currently two other patch series[2][3] that depend on the first two (ALSA-) patches of this series. The third ("ASoC: hdmi-codec: Add audio abort() callback for video side to use") is currently not used and can be dropped if so decided. The rest depends on those the first two and adds hdmi-audio support for Beaglebone-black. Best regards, Jyri [1] "[PATCH v6 0/6] Implement generic ASoC HDMI codec and use it in tda998x" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105524.html [2] "[PATCH v6 00/10] ASoC: Add mediatek HDMI codec support" by Philipp Zabel http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105509.html [3] "[RFC v2 0/6] sti: add audio interface to the hdmi driver" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/103374.ht… and "[PATCH v4 0/6] add IEC958 channel status control helpers" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105502.html by Arnaud Pouliquen Changes since v7 * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Remove the unused "int ret" variable from tda998x_configure_audio() Changes since v6 * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Add "Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com>" and "Tested-by: Philipp Zabel <p.zabel(a)pengutronix.de>", no other changes * "ALSA: pcm: add IEC958 channel status helper for hw_param" - Added kernel-doc for snd_pcm_create_iec958_consumer_hw_params() * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Use PTR_ERR_OR_ZERO() in tda998x_audio_codec_init() - Fix possible uninitialized use of 'size' in tda998x_get_audio_ports() - Store audio configuration in tda998x_audio_hw_params() instead of tda998x_configure_audio(). Changes since v5 - Rebased on top of the latest drm-next branch - Allow 32 bit samplewidth in snd_pcm_create_iec958_consumer() and snd_pcm_create_iec958_consumer_hw_params() - Propose new simpler DT binding for tda998x audio - Squash tda998x audio DT binding together with hdmi-codec integration Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 Jyri Sarha (6): ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 + arch/arm/boot/dts/am335x-boneblack.dts | 71 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 275 ++++++++++++-- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 65 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 411 +++++++++++++++++++++ 12 files changed, 914 insertions(+), 72 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8825: fix interrupt fails and unstable after resume
by John Hsu 30 Mar '16

30 Mar '16

Modify power management function to sync behavior with set bias function. And codec needs to config interrupt setting again to recover interrupt. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 70 ++++++++++++++++++++++++++++------------------ 1 file changed, 43 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 106c391..b8af46b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1204,6 +1204,42 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static void nau8825_configure_interrupt(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable ADC needed for interrupts + * It is the same as force_enable_pin("ADC") we do later + */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); +} + +static void nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + nau8825_configure_interrupt(nau8825); + nau8825_restart_jack_detection(regmap); +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1233,6 +1269,8 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + if (nau8825->irq) + nau8825_resume_setup(nau8825); } break; @@ -1346,29 +1384,9 @@ static int nau8825_read_device_properties(struct device *dev, static int nau8825_setup_irq(struct nau8825 *nau8825) { - struct regmap *regmap = nau8825->regmap; int ret; - /* IRQ Output Enable */ - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); - - /* Enable internal VCO needed for interruptions */ - nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - - /* Enable ADC needed for interrupts */ - regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); - - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + nau8825_configure_interrupt(nau8825); ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, @@ -1440,24 +1458,22 @@ static int nau8825_i2c_remove(struct i2c_client *client) #ifdef CONFIG_PM_SLEEP static int nau8825_suspend(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); - disable_irq(client->irq); + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); return 0; } static int nau8825_resume(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); + enable_irq(nau8825->irq); return 0; } -- 2.6.4

2 2

[alsa-devel] [PATCH V2 1/1] ASoC: add a core API to share more product information with user space
by han.lu＠intel.com 30 Mar '16

30 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Add a core API to share more machine/board information, for user space apps such as PA and UCM to distinguish various products. Previously on ASoC, the card short name, driver name and long name are all the same as the machine driver name. For example, on T100TA: $ cat /proc/asound/cards 0 [bytcr-rt5640]: bytcr-rt5640 - bytcr-rt5640 bytcr-rt5640 The patch adds more board information: card driver name ---> machine driver name card short name ---> DMI_PRODUCT_NAME or DMI_BOARD_NAME card long name and card component ---> short name:driver name(:DMI_SYS_VENDOR if available)(:firmware name if available) For example, on T100TA: $ cat /proc/asound/cards 0 [T100TA ]: bytcr-rt5640 - T100TA T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.:intel/fw_ sst_0f28.bin $ amixer -c0 info Card hw:0 'T100TA'/'T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.:intel/fw_ sst_0f28.bin' Mixer name : '' Components : 'T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.:intel/fw_s st_0f28.bin' Controls : 256 Simple ctrls : 228 Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a21..4e80444 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -486,6 +486,9 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream); int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt); +int snd_soc_set_card_names(struct snd_soc_card *card, const char *board, + const char *vendor, const char *firmware); + /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 032a2e7..3162f3e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -152,6 +152,8 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { {} }; +static struct snd_soc_card byt_rt5640_card; + static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; @@ -159,6 +161,19 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_card *card = runtime->card; const struct snd_soc_dapm_route *custom_map; int num_routes; + const char *board, *vendor; + struct sst_acpi_mach *mach = byt_rt5640_card.dev->platform_data; + + /* Add machine/board information for userspace */ + board = dmi_get_system_info(DMI_PRODUCT_NAME); + if (!board) + board = dmi_get_system_info(DMI_BOARD_NAME); + vendor = dmi_get_system_info(DMI_SYS_VENDOR); + ret = snd_soc_set_card_names(card, board, vendor, mach->fw_filename); + if (ret < 0) { + dev_err(card->dev, "unable to set card names\n"); + return ret; + } card->dapm.idle_bias_off = true; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..ab8e7e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1828,6 +1828,58 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, } EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); +/** + * snd_soc_set_card_names() - Set the shortname/drivername/longname/component + * of a ASoC card. + * @card: The card to set names + * @board: The board name or product name + * @vendor: The vendor name + * @firmware: The firmware name + * + * This function adds more information for the userspace to distinguish + * different products: + * card driver name ---> machine driver name + * card short name ---> DMI_PRODUCT_NAME or DMI_BOARD_NAME + * card long name and + * card component ---> short name:driver name(:DMI_SYS_VENDOR) + * (:firmware name) + * + * Returns 0 on success, otherwise a negative error code. + */ +int snd_soc_set_card_names(struct snd_soc_card *card, const char *board, + const char *vendor, const char *firmware) +{ + int ret = 0; + char *name; + + if (!board) { + dev_err(card->dev, "ASoC: the board/product name is empty!\n"); + return -1; + } + /* card driver name */ + card->driver_name = card->name; + /* card short name */ + card->name = board; + /* card long name / card component */ + name = kstrdup(card->name, GFP_KERNEL); + if (!name) + return -ENOMEM; + strcat(name, ":"); + strcat(name, card->driver_name); + strcat(name, ":"); + if (vendor) + strcat(name, vendor); + strcat(name, ":"); + if (firmware) + strcat(name, firmware); + ret = snd_component_add(card->snd_card, name); + card->long_name = card->snd_card->components; + + kfree(name); + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_set_card_names); + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; -- 2.5.0

3 2

[alsa-devel] [PATCH 0/7] topology: Add support for vendor tuples
by mengdong.lin＠linux.intel.com 30 Mar '16

30 Mar '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series addes support for vendor tuples to topology, to avoid importing binary data blob from other files. Backward compatibility of ABI is not impacted. A kernel patch is also submitted "ASoC: topology: ABI - Define types for vendor tuples". The 1st patch is small code cleanup. The 2nd patch is a preparation, since tuples will need the type-specific free handler. Mengdong Lin (7): topology: Use the generic pointer to free an element's object topology: Define a free handler for the element topology: Add doc for vendor tuples topology: ABI - Define types for vendor tuples topology: Add support for vendor tokens topology: Add support for parsing vendor tuples topology: Build data objects with tuples include/sound/asoc.h | 42 ++++- include/topology.h | 79 +++++++- src/topology/data.c | 470 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/elem.c | 15 +- src/topology/parser.c | 24 +++ src/topology/tplg_local.h | 47 +++++ 6 files changed, 669 insertions(+), 8 deletions(-) -- 2.5.0

4 14

[alsa-devel] [PATCH] ALSA - hda: hdmi check NULL pointer in hdmi_set_chmap
by libin.yang＠linux.intel.com 30 Mar '16

30 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> Make sure per_pin is not NULL before using it. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4833c7b..c99289d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1858,6 +1858,8 @@ static void hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx, struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + if (!per_pin) + return; mutex_lock(&per_pin->lock); per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, ARRAY_SIZE(per_pin->chmap)); -- 1.9.1

3 2

[alsa-devel] bytcr_rt5640, baytrail: trying to get audio working
by Bernhard Übelacker 30 Mar '16

30 Mar '16

Hello, (This message was sent to alsa-user a week ago, but created no response there. Therefore I hope I am allowed to ask here again.) I am trying to get audio working on a small Intel Z3736F based device. Some more details I have collected in Debian wiki [1]. My tests are done with a current Debian testing. Firmware is from package firmware-intel-sound: sha1sum: 9e7c514b224d674cd3fb1301ce693c42b01f02e0 fw_sst_0f28.bin md5sum: 63765966238afad810b6a568d9b59620 fw_sst_0f28.bin Tests were done on linux v4.5-rc7-244-g03c668a. With a change adjusting the IRQ described in [2]. After a fresh boot I tried the commands below (from [3]). But did not get hearable output, except a "crack" on device open and close. Now I am not sure how to proceed. Am I using the right firmware? Is the amixer sequence ok for this device? Can all the options in alsamixer be tested or is there a risk of damaging the device? Kind regards, Bernhard [1] https://wiki.debian.org/InstallingDebianOn/PIPO/PIPO%20X8 [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094663.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/094080.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094583.html [5] https://sourceforge.net/p/alsa/mailman/alsa-user/thread/56EC0F8A.4010009%40… echo -n 'module snd_soc_sst_mfld_platform +p' > /sys/kernel/debug/dynamic_debug/control #DSP routing amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on amixer -c0 sset 'media0_out mix 0 media1_in' on amixer -c0 sset 'media1_in Gain 0' 80% amixer -c0 sset 'media1_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'media1_in Gain 0' off amixer -c0 sset 'pcm0_in Gain 0' 80% amixer -c0 sset 'pcm0_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'pcm0_in Gain 0' off amixer -c0 sset 'codec_out0 Gain 0' 80% amixer -c0 sset 'codec_out0 Gain 0 Ramp Delay' 50 amixer -c0 sset 'codec_out0 Gain 0' off #Codec routin, if you have different one please modfiy amixer -c0 sset "DAC MIXL INF1" on amixer -c0 sset "DAC MIXR INF1" on amixer -c0 sset "Stereo DAC MIXL DAC L1" on amixer -c0 sset "Stereo DAC MIXR DAC R1" on amixer -c0 sset "SPK MIXL DAC L1" on amixer -c0 sset "SPK MIXR DAC R1" on amixer -c0 sset "SPOL MIX SPKVOL L" on amixer -c0 sset "SPOR MIX SPKVOL R" on amixer -c0 sset "SPOL MIX DAC L1" on amixer -c0 sset "SPOR MIX DAC R1" on amixer -c0 sset "Speaker L" on amixer -c0 sset "Speaker R" on amixer -c0 sset "Headphone" off #amixer -c0 sset "Ext Spk" on #does not exist amixer -c0 sset "Speaker" on # LANG=C speaker-test --nloops=1 speaker-test 1.1.0 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 16 to 204800 Period size range from 8 to 102400 Using max buffer size 204800 Periods = 4 was set period_size = 51200 was set buffer_size = 204800 0 - Front Left Time per period = 2.109224 # cat /proc/interrupts | grep -E "CPU|sst" CPU0 CPU1 CPU2 CPU3 1: 1 35 0 0 IO-APIC 29-fasteoi intel_sst_driver # LANG=C dmesg -T | grep -i -E "bytcr|snd|rt56|alsa|sst|Baytrail|80860F28:00|SSP|lpe" --color [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: LPE base: 0x90a00000 size:0x200000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: IRAM base: 0x90ac0000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: DRAM base: 0x90b00000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: SHIM base: 0x90b40000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: Mailbox base: 0x90b44000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [Thu Mar 17 11:49:59 2016] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [Thu Mar 17 11:49:59 2016] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [Thu Mar 17 11:49:59 2016] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [Thu Mar 17 11:49:59 2016] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [Thu Mar 17 11:49:59 2016] bytcr_rt5640 bytcr_rt5640: rt5640-aif1 <-> ssp2-port mapping ok [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Volume 0, -1440 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Mute 1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Volume 0, 0 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Mute 1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: media1_in: Ramp Delay50 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Volume 0, -1440 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Mute 1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Volume 0, 0 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Mute 1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:56 2016] sst-mfld-platform sst-mfld-platform: pcm0_in: Ramp Delay50 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Volume 0, -1440 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Mute 1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Volume 0, 0 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Mute 1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: codec_out0: Ramp Delay50 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 2 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: rx slot - slot 3 map = 0x8 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_0 map = 0x1 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out0_1 map = 0x2 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_0 map = 0x4 [Thu Mar 17 11:50:57 2016] sst-mfld-platform sst-mfld-platform: tx channel - codec_out1_1 map = 0x8 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: enable=1, usage=0 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 01 00 00 08 00 ff ff ff ff 55 00 00 00 ............U... [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter:sst_get_ssp_mode, format=4000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter:sst_get_frame_sync_polarity, format=200 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: ssp_id: 3 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 01 00 00 1a 00 ff ff ff ff 75 00 12 00 ............u... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 03 00 03 00 98 00 03 ff 03 ff 03 00 01 00 18 00 ................ [Thu Mar 17 11:51:08 2016] writing to lpe: 00000020: 01 01 .. [Thu Mar 17 11:51:08 2016] intel_sst_acpi 80860F28:00: Alloc for str 1 pipe 0x90 [Thu Mar 17 11:51:08 2016] Baytrail Audio Port: setting buffer ptr param [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=codec_out0 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 02 01 01 00 00 1e 00 ff ff ff ff 82 00 16 00 ................ [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 82 00 12 00 03 00 01 02 04 08 10 20 40 80 01 02 ........... @... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000020: 04 08 10 20 40 80 ... @. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=codec_out0 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: control name=codec_out0 Gain 0 Volume [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 02 01 01 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 00 02 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: widget = codec_out0 mix 0 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: val = 0x2000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: reg: 0x2000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: input id: 0x8dff, nb_inputs: 1 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 01 00 00 14 00 ff ff ff ff 72 00 0c 00 ............r... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: ff 02 ff ff 03 00 01 00 ff 8d ff ff ............ [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: widget = media0_out mix 0 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: val = 0x40000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: reg: 0x40000 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: input id: 0x90ff, nb_inputs: 1 [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 03 00 00 14 00 ff ff ff ff 72 00 0c 00 ............r... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: ff 12 ff ff 03 00 01 00 ff 90 ff ff ............ [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: widget=media0_out [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: task=3, location=0x12ff [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 03 00 00 0a 00 ff 12 ff ff 77 00 02 00 ............w... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 .. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=media0_out [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: widget=pcm0_in [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: task=1, location=0x8dff [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 01 01 01 00 00 0a 00 ff 8d ff ff 77 00 02 00 ............w... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 .. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=pcm0_in [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: control name=pcm0_in Gain 0 Volume [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 02 01 01 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 00 8d 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: enter, dai-name=media-cpu-dai dir=0 [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Stream name=Headset Playback [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: send gains for widget=media1_in [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: control name=media1_in Gain 0 Volume [Thu Mar 17 11:51:08 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:08 2016] writing to lpe: 00000000: 01 02 01 03 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:08 2016] writing to lpe: 00000010: 01 00 00 90 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:08 2016] Baytrail Audio Port: sst_platform_pcm_trigger called [Thu Mar 17 11:51:08 2016] Baytrail Audio Port: sst: Trigger Start [Thu Mar 17 11:51:08 2016] SSP2-Codec: sst_platform_pcm_trigger called [Thu Mar 17 11:51:10 2016] Baytrail Audio Port: sst_platform_pcm_trigger called [Thu Mar 17 11:51:10 2016] Baytrail Audio Port: sst: in stop [Thu Mar 17 11:51:10 2016] SSP2-Codec: sst_platform_pcm_trigger called [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: enter, dai-name=media-cpu-dai dir=0 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Stream name=Headset Playback [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: send gains for widget=media1_in [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: control name=media1_in Gain 0 Volume [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 02 01 03 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 01 00 00 90 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: ssp_id: 3 [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 01 00 00 1a 00 ff ff ff ff 75 00 12 00 ............u... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 03 00 00 00 98 00 03 ff 03 ff 03 00 01 00 18 00 ................ [Thu Mar 17 11:51:10 2016] writing to lpe: 00000020: 01 01 .. [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: enable=0, usage=1 [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 01 00 00 08 00 ff ff ff ff 0e 00 00 00 ................ [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: enter, dai-name=media-cpu-dai dir=0 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Stream name=Headset Playback [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: send gains for widget=media1_in [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: control name=media1_in Gain 0 Volume [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Enter [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 02 01 03 00 00 14 00 ff ff ff ff 21 00 0c 00 ............!... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 01 00 00 90 67 00 60 fa 60 fa 32 00 ....g.`.`.2. [Thu Mar 17 11:51:10 2016] intel_sst_acpi 80860F28:00: Free for str 1 pipe 0x90 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: widget=pcm0_in [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: task=1, location=0x8dff [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 01 00 00 0a 00 ff 8d ff ff 77 00 02 00 ............w... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 00 00 .. [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: widget=media0_out [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: task=3, location=0x12ff [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 03 00 00 0a 00 ff 12 ff ff 77 00 02 00 ............w... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: 00 00 .. [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: widget = codec_out0 mix 0 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: val = 0x2000 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: reg: 0x2000 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: input id: 0x8dff, nb_inputs: 1 [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 01 00 00 14 00 ff ff ff ff 72 00 0c 00 ............r... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: ff 02 ff ff 00 00 01 00 ff 8d ff ff ............ [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: widget = media0_out mix 0 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: val = 0x40000 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: reg: 0x40000 [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: input id: 0x90ff, nb_inputs: 1 [Thu Mar 17 11:51:10 2016] writing to lpe: 00000000: 01 01 01 03 00 00 14 00 ff ff ff ff 72 00 0c 00 ............r... [Thu Mar 17 11:51:10 2016] writing to lpe: 00000010: ff 12 ff ff 00 00 01 00 ff 90 ff ff ............ [Thu Mar 17 11:51:10 2016] sst-mfld-platform sst-mfld-platform: Enter: widget=codec_out0

2 3

[alsa-devel] [PATCH 1/5] Sound: SOC: TAS571x: added missing register literals
by Petr Kulhavy 30 Mar '16

30 Mar '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to register 0x25 Signed-off-by: Petr Kulhavy <petr(a)barix.com> Tested-by: Petr Kulhavy <petr(a)barix.com> --- sound/soc/codecs/tas571x.c | 4 ++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..346d3da 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..cf800c3 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: intel: make Kconfig selects depend on ACPI
by Randy Dunlap 29 Mar '16

29 Mar '16

From: Randy Dunlap <rdunlap(a)infradead.org> Prevent SND_SST_IPC_ACPI from being selected when CONFIG_ACPI is not enabled. This fixes the kconfig warning: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/Kconfig | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) --- linux-next-20160308.orig/sound/soc/intel/Kconfig +++ linux-next-20160308/sound/soc/intel/Kconfig @@ -100,7 +100,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH depends on X86 && I2C select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR @@ -113,7 +113,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH depends on X86 && I2C select SND_SOC_RT5651 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR @@ -126,7 +126,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell @@ -139,7 +139,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell @@ -152,7 +152,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI select SND_SOC_MAX98090 select SND_SOC_TS3A227E select SND_SST_MFLD_PLATFORM - select SND_SST_IPC_ACPI + select SND_SST_IPC_ACPI if ACPI select SND_SOC_INTEL_SST_MATCH if ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell

2 2

[alsa-devel] [PATCH] ASoC: topology: ABI - Define types for vendor tuples
by mengdong.lin＠linux.intel.com 29 Mar '16

29 Mar '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Tuples, a pair of token and value, can be used to define vendor specific data, for controls and widgets. This can avoid importing binary data blob from other files. Vendor specific tuple arrays will be embeded in the private data buffer of a control or widget object. To be backward compatible, union is used to define the tuple arrays in the existing private data ABI object 'struct snd_soc_tplg_private'. Vendors need to make sure the token values defined by the topology conf file match those defined by their driver. Now supported tuple types are uuid, string, bool, byte, short and word. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index c4cc1e4..db86b447 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -116,6 +116,14 @@ #define SND_SOC_TPLG_STREAM_PLAYBACK 0 #define SND_SOC_TPLG_STREAM_CAPTURE 1 +/* vendor tuple types */ +#define SND_SOC_TPLG_TUPLE_TYPE_UUID 0 +#define SND_SOC_TPLG_TUPLE_TYPE_STRING 1 +#define SND_SOC_TPLG_TUPLE_TYPE_BOOL 2 +#define SND_SOC_TPLG_TUPLE_TYPE_BYTE 3 +#define SND_SOC_TPLG_TUPLE_TYPE_WORD 4 +#define SND_SOC_TPLG_TUPLE_TYPE_SHORT 5 + /* * Block Header. * This header precedes all object and object arrays below. @@ -132,6 +140,35 @@ struct snd_soc_tplg_hdr { __le32 count; /* number of elements in block */ } __attribute__((packed)); +/* vendor tuple for uuid */ +struct snd_soc_tplg_vendor_uuid_elem { + __le32 token; + char uuid[16]; +} __attribute__((packed)); + +/* vendor tuple for a bool/byte/short/word value */ +struct snd_soc_tplg_vendor_value_elem { + __le32 token; + __le32 value; +} __attribute__((packed)); + +/* vendor tuple for string */ +struct snd_soc_tplg_vendor_string_elem { + __le32 token; + char string[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +struct snd_soc_tplg_vendor_array { + __le32 size; /* size in bytes of the array, including all elements */ + __le32 type; /* SND_SOC_TPLG_TUPLE_TYPE_ */ + __le32 num_elems; /* number of elements in array */ + union { + struct snd_soc_tplg_vendor_uuid_elem uuid[0]; + struct snd_soc_tplg_vendor_value_elem value[0]; + struct snd_soc_tplg_vendor_string_elem string[0]; + }; +} __attribute__((packed)); + /* * Private data. * All topology objects may have private data that can be used by the driver or @@ -139,7 +176,10 @@ struct snd_soc_tplg_hdr { */ struct snd_soc_tplg_private { __le32 size; /* in bytes of private data */ - char data[0]; + union { + char data[0]; + struct snd_soc_tplg_vendor_array array[0]; + }; } __attribute__((packed)); /* -- 2.5.0

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm_adsp: Show avail in bytes to match other messages
by Charles Keepax 29 Mar '16

29 Mar '16

All other debug messages talk about data on the compressed stream in bytes except avail which is shown in words. To avoid confusion show avail in bytes as well. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d3b1cb1..3c5ef13 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2805,7 +2805,7 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) avail += wm_adsp_buffer_size(buf); adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", - buf->read_index, write_index, avail); + buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); buf->avail = avail; -- 2.1.4

2 8

[alsa-devel] [PATCH 1/2] ASoC: wm8962: Disable clock if wm8962_runtime_resume() fails
by Fabio Estevam 29 Mar '16

29 Mar '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> When regulator_bulk_enable() fails inside wm8962_runtime_resume(), we should disable the previously enabled clock. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/wm8962.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8822360..f3f71ba 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3800,7 +3800,7 @@ static int wm8962_runtime_resume(struct device *dev) if (ret != 0) { dev_err(dev, "Failed to enable supplies: %d\n", ret); - return ret; + goto disable_clock; } regcache_cache_only(wm8962->regmap, false); @@ -3838,6 +3838,10 @@ static int wm8962_runtime_resume(struct device *dev) msleep(5); return 0; + +disable_clock: + clk_disable_unprepare(wm8962->pdata.mclk); + return ret; } static int wm8962_runtime_suspend(struct device *dev) -- 1.9.1

3 5

[alsa-devel] [PATCH] ASoC: wm8962: Correct typo, when setting DSPCLK rate
by Charles Keepax 29 Mar '16

29 Mar '16

The variable dspclk holds the rate of the DSPCLK, but the variable sysclk holds an identifier for the clock. Currently if read a non-sensical value from the DSPCLK_DIV register we assign sysclk to dspclk, clearly this was intended to be sysclk_rate. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8822360..720a14e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2471,7 +2471,7 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) break; default: dev_warn(codec->dev, "Unknown DSPCLK divisor read back\n"); - dspclk = wm8962->sysclk; + dspclk = wm8962->sysclk_rate; } dev_dbg(codec->dev, "DSPCLK is %dHz, BCLK %d\n", dspclk, wm8962->bclk); -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Adjust clk definitions so that simple card can work
by Fabio Estevam 29 Mar '16

29 Mar '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> When trying to use simple card with wm8962 the following probe error happens: wm8962 0-001a: simple-card: set_sysclk error asoc-simple-card sound: ASoC: failed to init 202c000.ssi-wm8962: -22 asoc-simple-card sound: ASoC: failed to instantiate card -22 asoc-simple-card: probe of sound failed with error -22 In simple-card.c the snd_soc_dai_set_sysclk() function is called with clk_id as 0, which is an invalid clock for wm8962. Adjust the clocks source definitions in wm8962.h so that the simple card driver can work successfully. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/wm8962.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index 910aafd..e63a318 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -16,9 +16,9 @@ #include <asm/types.h> #include <sound/soc.h> -#define WM8962_SYSCLK_MCLK 1 -#define WM8962_SYSCLK_FLL 2 -#define WM8962_SYSCLK_PLL3 3 +#define WM8962_SYSCLK_MCLK 0 +#define WM8962_SYSCLK_FLL 1 +#define WM8962_SYSCLK_PLL3 2 #define WM8962_FLL 1 -- 1.9.1

3 2

[alsa-devel] ADAU1761 default register value problem
by Ricard Wanderlof 29 Mar '16

29 Mar '16

I have a problem with the ADAU1761 (although the problem I would think would manifest itself with a veriety of codecs). The register values are cached to minimize the number of writes needed, and there is a set of default values specified. However, this assumes that when the software boots the chip is always in the default state. This is not necessarily the case, because the chip has no external reset line and resets its internal state using an internal power-on-reset circuit, so if the system is rebooted without a power cycle, the regcache will contain the default values of the registers, whereas the actual registers in the chip will retain their previous values. Hence, if attempting to set a value in a register which is in fact identical to the default value, it will not be written, and the value that is currently in the register will remain. This can lead to all sorts of misconfigurations, depending on the previous state of the chip. One way to fix this is simply to avoid setting up a set of default registers (looking at regcache.c, one way seems to be to omit setting the reg_defaults and num_reg_defaults members of the struct regmap_config during initialization, and instead setting num_reg_defaults_raw to the number of registers the chip has, but not setting reg_defaults_raw; I don't know if this is how it is intended to be used however). The regcache framework will then read all the register values from the chip and use them as the new defaults. While this would seem to be technically correct, the question is if this is acceptable? A drawback is that there will be a lot of I2C or SPI accesses just to get the default values. Or should the caching behavior at startup be configured in some way, for instance via devicetree? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [PATCH] ALSA: hda - Clear the leftover component assignment at snd_hdac_i915_exit()
by Takashi Iwai 29 Mar '16

29 Mar '16

The commit [d745f5e7b8b2: ALSA: hda - Add the pin / port mapping on Intel ILK and VLV] introduced a WARN_ON() to check the pointer for avoiding the double initializations. But hdac_acomp pointer wasn't cleared at snd_hdac_i915_exit(), thus after reloading the HD-audio driver, it may result in the false positive warning. This patch makes sure to clear the leftover pointer at exit. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=94736 Reported-by: Daniela Doras-prodan <daniela.doras-prodan(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 750a4ea49fa9..ae0f305a7e41 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -372,6 +372,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; return 0; } -- 2.7.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Fix for unloading module only when it is loaded
by jeeja.kp＠intel.com 29 Mar '16

29 Mar '16

From: Jeeja KP <jeeja.kp(a)intel.com> Module needs to be unloaded only when it is loaded successfully. To fix this, first correct the module state sequence and set module state to LOADED if module is loaded successfully. When unloading the module check if module state is not in UNINIT, then unload it. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 5 ++++- sound/soc/intel/skylake/skl-topology.h | 8 ++++---- 2 files changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 545b4e7..3f393cb 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -490,6 +490,8 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) mconfig->id.module_id, mconfig->guid); if (ret < 0) return ret; + + mconfig->m_state = SKL_MODULE_LOADED; } /* update blob if blob is null for be with default value */ @@ -524,7 +526,8 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, list_for_each_entry(w_module, &pipe->w_list, node) { mconfig = w_module->w->priv; - if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod) + if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod && + mconfig->m_state > SKL_MODULE_UNINIT) return ctx->dsp->fw_ops.unload_mod(ctx->dsp, mconfig->id.module_id); } diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index de3c401..d2d9230 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -274,10 +274,10 @@ struct skl_pipe { enum skl_module_state { SKL_MODULE_UNINIT = 0, - SKL_MODULE_INIT_DONE = 1, - SKL_MODULE_LOADED = 2, - SKL_MODULE_UNLOADED = 3, - SKL_MODULE_BIND_DONE = 4 + SKL_MODULE_LOADED = 1, + SKL_MODULE_INIT_DONE = 2, + SKL_MODULE_BIND_DONE = 3, + SKL_MODULE_UNLOADED = 4, }; struct skl_module_cfg { -- 2.5.0

2 3

[alsa-devel] [PATCH] ASoC: nau8825: change output power for interrupt
by John Hsu 29 Mar '16

29 Mar '16

The interrupt clock is gated by x1[10:8], one of them needs to be enabled all the time for interrupts to happen. We change codec to enable ADC because it's helpful to reduce playback pop noise. Don't use force enable pin to enable ADC instead of ADC widget event. That won't interfere DAPM operation and let bias work normally. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 40 ++++++++++++++++++++++++++++------------ sound/soc/codecs/nau8825.h | 1 + 2 files changed, 29 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 4c8572c..106c391 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -223,6 +223,29 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) } } +static int nau8825_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + break; + case SND_SOC_DAPM_POST_PMD: + if (!nau8825->irq) + regmap_update_bits(nau8825->regmap, + NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_ADC, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + static int nau8825_pump_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -338,7 +361,9 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, + nau8825_adc_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), @@ -928,13 +953,6 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* The interrupt clock is gated by x1[10:8], - * one of them needs to be enabled all the time for - * interrupts to happen. - */ - snd_soc_dapm_force_enable_pin(dapm, "DDACR"); - snd_soc_dapm_sync(dapm); - /* Unmask interruptions. Handler uses dapm object so we can enable * interruptions only after dapm is fully initialized. */ @@ -1338,11 +1356,9 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) /* Enable internal VCO needed for interruptions */ nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - /* Enable DDACR needed for interrupts - * It is the same as force_enable_pin("DDACR") we do later - */ + /* Enable ADC needed for interrupts */ regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); /* Chip needs one FSCLK cycle in order to generate interrupts, * as we cannot guarantee one will be provided by the system. Turning diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 5fe009d..4427df9 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -99,6 +99,7 @@ #define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) #define NAU8825_ENABLE_DACL_SFT 9 #define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_ADC (1 << NAU8825_ENABLE_ADC_SFT) #define NAU8825_ENABLE_SAR_SFT 1 /* CLK_DIVIDER (0x3) */ -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: wm8960: Provide a menu selection text
by Fabio Estevam 29 Mar '16

29 Mar '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> Provide a menu selection text so that users can enable, disable or mark it as module in menuconfig. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a..196101b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -909,7 +909,7 @@ config SND_SOC_WM8955 tristate config SND_SOC_WM8960 - tristate + tristate "Wolfson Microelectronics WM8960 CODEC" config SND_SOC_WM8961 tristate -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] Asoc: bytcr-rt5640: add more machine/board information for user space
by han.lu＠intel.com 29 Mar '16

29 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Add more machine/board information for PA and UCM. Previously on ASoc, the card short name, driver name and long name are all the same as the machine driver name. On T100TA it looks like: $ cat /proc/asound/cards 0 [bytcr-rt5640]: bytcr-rt5640 - bytcr-rt5640 bytcr-rt5640 The patch made changes below: driver name ---> machine driver name short name ---> DMI_PRODUCT_NAME or DMI_BOARD_NAME long name ---> short name:driver name(:DMI_SYS_VENDOR if applicable) And add card component: long name:firmware name. On T100TA it looks like: $ cat /proc/asound/cards 0 [T100TA ]: bytcr-rt5640 - T100TA T100TA:bytcr-rt5640:ASUSTek COMPUTER INC. $ amixer -c0 info Card hw:0 'T100TA'/'T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.' Mixer name : '' Components : 'T100TA:bytcr-rt5640:ASUSTek COMPUTER INC.:intel/fw_s st_0f28.bin' Controls : 256 Simple ctrls : 228 Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 032a2e7..068cd58 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -152,6 +152,8 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { {} }; +static struct snd_soc_card byt_rt5640_card; + static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; @@ -159,6 +161,34 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_card *card = runtime->card; const struct snd_soc_dapm_route *custom_map; int num_routes; + const char *dmi_name; + char *component; + struct sst_acpi_mach *mach = byt_rt5640_card.dev->platform_data; + + /* Add machine/board information for userspace: + * card driver name ---> machine driver name + * card short name ---> DMI_PRODUCT_NAME or DMI_BOARD_NAME + * card long name ---> short name:driver name(:DMI_SYS_VENDOR) + */ + card->driver_name = card->name; + dmi_name = dmi_get_system_info(DMI_PRODUCT_NAME); + if (!dmi_name) + dmi_name = dmi_get_system_info(DMI_BOARD_NAME); + if (dmi_name) + card->name = dmi_name; + card->long_name = kstrdup(card->name, GFP_KERNEL); + strcat((char *)card->long_name, ":"); + strcat((char *)card->long_name, card->driver_name); + dmi_name = dmi_get_system_info(DMI_SYS_VENDOR); + if (dmi_name) { + strcat((char *)card->long_name, ":"); + strcat((char *)card->long_name, dmi_name); + } + /* card component ---> long name:firmware name */ + component = kstrdup(card->long_name, GFP_KERNEL); + strcat(component, ":"); + strcat(component, mach->fw_filename); + snd_component_add(card->snd_card, component); card->dapm.idle_bias_off = true; -- 2.5.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by paul.handrigan＠cirrus.com 29 Mar '16

29 Mar '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l33 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l33.h | 54 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1303 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1585 insertions(+) create mode 100644 include/sound/cs35l33.h create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h diff --git a/include/sound/cs35l33.h b/include/sound/cs35l33.h new file mode 100644 index 0000000..d18181c --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,54 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2015 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS35L33_H +#define __CS35L33_H + +struct cs35l33_hg { + bool enable_hg_algo; + unsigned int mem_depth; + unsigned int release_rate; + unsigned int hd_rm; + unsigned int ldo_thld; + unsigned int ldo_path_disable; + unsigned int ldo_entry_delay; + bool vp_hg_auto; + unsigned int vp_hg; + unsigned int vp_hg_rate; + unsigned int vp_hg_va; +}; + +struct cs35l33_pdata { + /* GPIO for IRQ */ + int gpio_irq; + + /* irq line */ + int irq; + + /* Boost Controller Voltage Setting */ + unsigned int boost_ctl; + + /* Boost Controller Peak Current */ + unsigned int boost_ipk; + + /* Amplifier Drive Select */ + unsigned int amp_drv_sel; + + /* soft volume ramp */ + unsigned int ramp_rate; + + /* IMON adc scale */ + unsigned int imon_adc_scale; + + /* H/G algo configuration */ + struct cs35l33_hg hg_config; +}; + +#endif /* __CS35L33_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ef3a0c..44bd911 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -45,6 +45,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C + select SND_SOC_CS35L33 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -372,6 +373,10 @@ config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C +config SND_SOC_CS35L33 + tristate "Cirrus Logic CS35L33 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712..40f73fa 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o +snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -243,6 +244,7 @@ obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o +obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c new file mode 100644 index 0000000..6a29047 --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1303 @@ +/* + * cs35l33.c -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/gpio.h> +#include <sound/cs35l33.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/of_gpio.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include "cs35l33.h" + +#define CS35L33_BOOT_DELAY 50 +#define CS35L33_INT_ATTEMPTS 20 + +struct cs35l33_private { + struct snd_soc_codec *codec; + struct cs35l33_pdata pdata; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + bool amp_cal; + int mclk_int; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + bool is_tdm_mode; + bool enable_soft_ramp; +}; + +static const struct reg_default cs35l33_reg[] = { + {CS35L33_PWRCTL1, 0x85}, + {CS35L33_PWRCTL2, 0xFE}, + {CS35L33_CLK_CTL, 0x0C}, + {CS35L33_BST_PEAK_CTL, 0x90}, + {CS35L33_PROTECT_CTL, 0x55}, + {CS35L33_BST_CTL1, 0x00}, + {CS35L33_BST_CTL2, 0x01}, + {CS35L33_ADSP_CTL, 0x00}, + {CS35L33_ADC_CTL, 0xC8}, + {CS35L33_DAC_CTL, 0x14}, + {CS35L33_DIG_VOL_CTL, 0x00}, + {CS35L33_CLASSD_CTL, 0x04}, + {CS35L33_AMP_CTL, 0x90}, + {CS35L33_INT_MASK_1, 0xFF}, + {CS35L33_INT_MASK_2, 0xFF}, + {CS35L33_DIAG_LOCK, 0x00}, + {CS35L33_DIAG_CTRL_1, 0x40}, + {CS35L33_DIAG_CTRL_2, 0x00}, + {CS35L33_HG_MEMLDO_CTL, 0x62}, + {CS35L33_HG_REL_RATE, 0x03}, + {CS35L33_LDO_DEL, 0x12}, + {CS35L33_HG_HEAD, 0x0A}, + {CS35L33_HG_EN, 0x05}, + {CS35L33_TX_VMON, 0x00}, + {CS35L33_TX_IMON, 0x03}, + {CS35L33_TX_VPMON, 0x02}, + {CS35L33_TX_VBSTMON, 0x05}, + {CS35L33_TX_FLAG, 0x06}, + {CS35L33_TX_EN1, 0x00}, + {CS35L33_TX_EN2, 0x00}, + {CS35L33_TX_EN3, 0x00}, + {CS35L33_TX_EN4, 0x00}, + {CS35L33_RX_AUD, 0x40}, + {CS35L33_RX_SPLY, 0x03}, + {CS35L33_RX_ALIVE, 0x04}, + {CS35L33_BST_CTL4, 0x63}, +}; + +static const struct reg_sequence cs35l33_patch[] = { + { 0x00, 0x99, 0 }, + { 0x59, 0x02, 0 }, + { 0x52, 0x30, 0 }, + { 0x39, 0x45, 0 }, + { 0x57, 0x30, 0 }, + { 0x2C, 0x68, 0 }, + { 0x00, 0x00, 0 }, +}; + +static bool cs35l33_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l33_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + /* these are read only registers */ + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return false; + default: + return true; + } +} + +static bool cs35l33_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_PWRCTL1: + case CS35L33_PWRCTL2: + case CS35L33_CLK_CTL: + case CS35L33_BST_PEAK_CTL: + case CS35L33_PROTECT_CTL: + case CS35L33_BST_CTL1: + case CS35L33_BST_CTL2: + case CS35L33_ADSP_CTL: + case CS35L33_ADC_CTL: + case CS35L33_DAC_CTL: + case CS35L33_DIG_VOL_CTL: + case CS35L33_CLASSD_CTL: + case CS35L33_AMP_CTL: + case CS35L33_INT_MASK_1: + case CS35L33_INT_MASK_2: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_DIAG_LOCK: + case CS35L33_DIAG_CTRL_1: + case CS35L33_DIAG_CTRL_2: + case CS35L33_HG_MEMLDO_CTL: + case CS35L33_HG_REL_RATE: + case CS35L33_LDO_DEL: + case CS35L33_HG_HEAD: + case CS35L33_HG_EN: + case CS35L33_TX_VMON: + case CS35L33_TX_IMON: + case CS35L33_TX_VPMON: + case CS35L33_TX_VBSTMON: + case CS35L33_TX_FLAG: + case CS35L33_TX_EN1: + case CS35L33_TX_EN2: + case CS35L33_TX_EN3: + case CS35L33_TX_EN4: + case CS35L33_RX_AUD: + case CS35L33_RX_SPLY: + case CS35L33_RX_ALIVE: + case CS35L33_BST_CTL4: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l33_snd_controls[] = { + + SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL, + 4, 0x09, 0, classd_ctl_tlv), + SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, + 0, 0x34, 0xE4, dac_tlv), +}; + +static int cs35l33_spkrdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!priv->amp_cal) { + usleep_range(8000, 9000); + priv->amp_cal = true; + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, 0); + dev_dbg(codec->dev, "Amp calibration done\n"); + } + dev_dbg(codec->dev, "Amp turned on\n"); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "Amp turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + break; + } + + return 0; +} + +static int cs35l33_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, 0); + val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM; + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, val); + dev_dbg(codec->dev, "BST turned on\n"); + break; + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "SDIN turned on\n"); + if (!priv->amp_cal) { + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, CS35L33_AMP_CAL); + dev_dbg(codec->dev, "Amp calibration started\n"); + usleep_range(10000, 11000); + } + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, CS35L33_PDN_TDM); + usleep_range(4000, 4100); + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, CS35L33_PDN_BST); + dev_dbg(codec->dev, "BST and SDIN turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + + } + + return 0; +} + +static int cs35l33_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + unsigned int mask2 = CS35L33_SDOUT_3ST_TDM; + unsigned int val, val2; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->is_tdm_mode) { + /* set sdout_3st_i2s and reset pdn_tdm */ + val = CS35L33_SDOUT_3ST_I2S; + /* reset sdout_3st_tdm */ + val2 = 0; + } else { + /* reset sdout_3st_i2s and set pdn_tdm */ + val = CS35L33_PDN_TDM; + /* set sdout_3st_tdm */ + val2 = CS35L33_SDOUT_3ST_TDM; + } + dev_dbg(codec->dev, "SDOUT turned on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + val2 = CS35L33_SDOUT_3ST_TDM; + dev_dbg(codec->dev, "SDOUT turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return 0; + } + + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + mask, val); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + mask2, val2); + + return 0; +} + +static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0, + cs35l33_spkrdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2, + 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("MON"), + + SND_SOC_DAPM_ADC("VMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1), + SND_SOC_DAPM_ADC("IMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VPMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VBSTMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1), + + SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0, + cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l33_audio_map[] = { + {"SDIN", NULL, "CS35L33 Playback"}, + {"SPKDRV", NULL, "SDIN"}, + {"SPK", NULL, "SPKDRV"}, + + {"VMON", NULL, "MON"}, + {"IMON", NULL, "MON"}, + + {"SDOUT", NULL, "VMON"}, + {"SDOUT", NULL, "IMON"}, + {"CS35L33 Capture", NULL, "SDOUT"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = { + {"SPKDRV", NULL, "VPMON"}, + {"VPMON", NULL, "CS35L33 Playback"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = { + {"SDOUT", NULL, "VPMON"}, + {"VPMON", NULL, "MON"}, + {"SDOUT", NULL, "VBSTMON"}, + {"VBSTMON", NULL, "MON"}, +}; + +static int cs35l33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int val; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, 0); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, CS35L33_PDN_ALL); + regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val); + usleep_range(1000, 1100); + if (val & CS35L33_PDN_DONE) + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, CS35L33_MCLKDIS); + break; + case SND_SOC_BIAS_OFF: + break; + default: + return -EINVAL; + } + + dapm->bias_level = level; + + return 0; +} + +struct cs35l33_mclk_div { + int mclk; + int srate; + u8 adsp_rate; + u8 int_fs_ratio; +}; + +static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = { + /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */ + {5644800, 11025, 0x4, CS35L33_INT_FS_RATE}, + {5644800, 22050, 0x8, CS35L33_INT_FS_RATE}, + {5644800, 44100, 0xC, CS35L33_INT_FS_RATE}, + + {6000000, 8000, 0x1, 0}, + {6000000, 11025, 0x2, 0}, + {6000000, 11029, 0x3, 0}, + {6000000, 12000, 0x4, 0}, + {6000000, 16000, 0x5, 0}, + {6000000, 22050, 0x6, 0}, + {6000000, 22059, 0x7, 0}, + {6000000, 24000, 0x8, 0}, + {6000000, 32000, 0x9, 0}, + {6000000, 44100, 0xA, 0}, + {6000000, 44118, 0xB, 0}, + {6000000, 48000, 0xC, 0}, + + {6144000, 8000, 0x1, CS35L33_INT_FS_RATE}, + {6144000, 12000, 0x4, CS35L33_INT_FS_RATE}, + {6144000, 16000, 0x5, CS35L33_INT_FS_RATE}, + {6144000, 24000, 0x8, CS35L33_INT_FS_RATE}, + {6144000, 32000, 0x9, CS35L33_INT_FS_RATE}, + {6144000, 48000, 0xC, CS35L33_INT_FS_RATE}, +}; + +static int cs35l33_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) { + if (cs35l33_mclk_coeffs[i].mclk == mclk && + cs35l33_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, CS35L33_MS_MASK); + dev_dbg(codec->dev, "Audio port in master mode\n"); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, 0); + dev_dbg(codec->dev, "Audio port in slave mode\n"); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + /* + * tdm mode in cs35l33 resembles dsp-a mode very + * closely, it is dsp-a with fsync shifted left by half bclk + */ + priv->is_tdm_mode = true; + dev_dbg(codec->dev, "Audio port in TDM mode\n"); + break; + case SND_SOC_DAIFMT_I2S: + priv->is_tdm_mode = false; + dev_dbg(codec->dev, "Audio port in I2S mode\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l33_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + int sample_size = params_width(params); + int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params)); + + if (coeff < 0) + return coeff; + + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_ADSP_FS | CS35L33_INT_FS_RATE, + cs35l33_mclk_coeffs[coeff].int_fs_ratio + | cs35l33_mclk_coeffs[coeff].adsp_rate); + + if (priv->is_tdm_mode) { + sample_size = (sample_size / 8) - 1; + if (sample_size > 2) + sample_size = 2; + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_AUDIN_RX_DEPTH, + sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT); + } + + dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n", + params_rate(params), params_width(params)); + + return 0; +} + +static const unsigned int cs35l33_src_rates[] = { + 8000, 11025, 11029, 12000, 16000, 22050, + 22059, 24000, 32000, 44100, 44118, 48000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { + .count = ARRAY_SIZE(cs35l33_src_rates), + .list = cs35l33_src_rates, +}; + +static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l33_constraints); + return 0; +} + +static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (tristate) { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM); + } else { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, 0); + } + + return 0; +} + +static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos, i; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) { + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "Audio starts from slots %d", slot); + } + + /* + * scan tx_mask: vmon(2 slots); imon (2 slots); + * vpmon (1 slot) vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + for (i = 0; i < 2 ; i++) { + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i, + CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE + | CS35L33_X_LOC); + } + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route, + ARRAY_SIZE(cs35l33_vp_vbst_mon_route)); + + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) { + regmap_update_bits(priv->regmap, CS35L33_TX_VMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "VMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure IMON_TX_LOC */ + if (slot_num == 3) { + regmap_update_bits(priv->regmap, CS35L33_TX_IMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "IMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure VPMON_TX_LOC */ + if (slot_num == 4) { + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[0], 2); + dev_dbg(codec->dev, "VPMON enabled in slots %d", slot); + } + + /* configure VBSTMON_TX_LOC */ + if (slot_num == 5) { + regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[2], 2); + dev_dbg(codec->dev, + "VBSTMON enabled in slots %d", slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L33_TX_EN4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + regmap_update_bits(priv->regmap, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L33_MCLK_5644: + case CS35L33_MCLK_6: + case CS35L33_MCLK_6144: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, 0); + cs35l33->mclk_int = freq; + break; + case CS35L33_MCLK_11289: + case CS35L33_MCLK_12: + case CS35L33_MCLK_12288: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, CS35L33_MCLKDIV2); + cs35l33->mclk_int = freq/2; + break; + default: + cs35l33->mclk_int = 0; + return -EINVAL; + } + + dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n", + freq, cs35l33->mclk_int); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l33_ops = { + .startup = cs35l33_pcm_startup, + .set_tristate = cs35l33_set_tristate, + .set_fmt = cs35l33_set_dai_fmt, + .hw_params = cs35l33_pcm_hw_params, + .set_tdm_slot = cs35l33_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l33_dai = { + .name = "cs35l33-dai", + .id = 0, + .playback = { + .stream_name = "CS35L33 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .capture = { + .stream_name = "CS35L33 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .ops = &cs35l33_ops, + .symmetric_rates = 1, +}; + +static int cs35l33_set_hg_data(struct snd_soc_codec *codec, + struct cs35l33_pdata *pdata) +{ + struct cs35l33_hg *hg_config = &pdata->hg_config; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (hg_config->enable_hg_algo) { + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_MEM_DEPTH_MASK, + hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT); + regmap_write(priv->regmap, CS35L33_HG_REL_RATE, + hg_config->release_rate); + regmap_update_bits(priv->regmap, CS35L33_HG_HEAD, + CS35L33_HD_RM_MASK, + hg_config->hd_rm << CS35L33_HD_RM_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_THLD_MASK, + hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_DISABLE_MASK, + hg_config->ldo_path_disable << + CS35L33_LDO_DISABLE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_LDO_ENTRY_DELAY_MASK, + hg_config->ldo_entry_delay << + CS35L33_LDO_ENTRY_DELAY_SHIFT); + if (hg_config->vp_hg_auto) { + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_AUTO_MASK, + CS35L33_VP_HG_AUTO_MASK); + snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route, + ARRAY_SIZE(cs35l33_vphg_auto_route)); + } + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_MASK, + hg_config->vp_hg << CS35L33_VP_HG_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_RATE_MASK, + hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_VA_MASK, + hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK); + } + return 0; +} + +static int cs35l33_probe(struct snd_soc_codec *codec) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + cs35l33->codec = codec; + pm_runtime_get_sync(codec->dev); + + regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL, + CS35L33_ALIVE_WD_DIS, 0x8); + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2, + CS35L33_ALIVE_WD_DIS2, + CS35L33_ALIVE_WD_DIS2); + + /* Set Platform Data */ + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1, + CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl); + regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_DRV_SEL_MASK, + cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT); + + if (cs35l33->pdata.boost_ipk) + regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL, + cs35l33->pdata.boost_ipk); + + if (cs35l33->enable_soft_ramp) { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, CS35L33_DIGSFT); + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate); + } else { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, 0); + } + + /* update IMON scaling rate if different from default of 0x8 */ + if (cs35l33->pdata.imon_adc_scale != 0x8) + snd_soc_update_bits(codec, CS35L33_ADC_CTL, + CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale); + + cs35l33_set_hg_data(codec, &(cs35l33->pdata)); + + /* + * unmask important interrupts that causes the chip to enter + * speaker safe mode and hence deserves user attention + */ + regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1, + CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT | + CS35L33_M_CAL_ERR, 0); + + pm_runtime_put_sync(codec->dev); + + return 0; +} + +static struct regmap *cs35l33_get_regmap(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + return cs35l33->regmap; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .get_regmap = cs35l33_get_regmap, + .set_bias_level = cs35l33_set_bias_level, + .set_sysclk = cs35l33_codec_set_sysclk, + + .dapm_widgets = cs35l33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets), + .dapm_routes = cs35l33_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), + .controls = cs35l33_snd_controls, + .num_controls = ARRAY_SIZE(cs35l33_snd_controls), + + .idle_bias_off = true, +}; + +static const struct regmap_config cs35l33_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L33_MAX_REGISTER, + .reg_defaults = cs35l33_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l33_reg), + .volatile_reg = cs35l33_volatile_register, + .readable_reg = cs35l33_readable_register, + .writeable_reg = cs35l33_writeable_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, +}; + +static int cs35l33_runtime_resume(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + int ret; + + dev_dbg(dev, "%s\n", __func__); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, false); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + + ret = regcache_sync(cs35l33->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + return 0; + +err: + regcache_cache_only(cs35l33->regmap, true); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return ret; +} + +static int cs35l33_runtime_suspend(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s\n", __func__); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + regcache_cache_only(cs35l33->regmap, true); + regcache_mark_dirty(cs35l33->regmap); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l33_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend, + cs35l33_runtime_resume, + NULL) +}; + +static int cs35l33_get_hg_data(const struct device_node *np, + struct cs35l33_pdata *pdata) +{ + struct device_node *hg; + struct cs35l33_hg *hg_config = &pdata->hg_config; + u32 val32; + + hg = of_get_child_by_name(np, "hg-algo"); + hg_config->enable_hg_algo = hg ? true : false; + + if (hg_config->enable_hg_algo) { + if (of_property_read_u32(hg, "mem-depth", &val32) >= 0) + hg_config->mem_depth = val32; + if (of_property_read_u32(hg, "release-rate", &val32) >= 0) + hg_config->release_rate = val32; + if (of_property_read_u32(hg, "hd-rm", &val32) >= 0) + hg_config->hd_rm = val32; + if (of_property_read_u32(hg, "ldo-thld", &val32) >= 0) + hg_config->ldo_thld = val32; + if (of_property_read_u32(hg, "ldo-path-disable", &val32) >= 0) + hg_config->ldo_path_disable = val32; + if (of_property_read_u32(hg, "ldo-entry-delay", &val32) >= 0) + hg_config->ldo_entry_delay = val32; + hg_config->vp_hg_auto = of_property_read_bool(hg, "vp-hg-auto"); + if (of_property_read_u32(hg, "vp-hg", &val32) >= 0) + hg_config->vp_hg = val32; + if (of_property_read_u32(hg, "vp-hg-rate", &val32) >= 0) + hg_config->vp_hg_rate = val32; + if (of_property_read_u32(hg, "vp-hg-va", &val32) >= 0) + hg_config->vp_hg_va = val32; + } + + of_node_put(hg); + + return 0; +} + +static irqreturn_t cs35l33_irq_thread(int irq, void *data) +{ + struct cs35l33_private *cs35l33 = data; + struct snd_soc_codec *codec = cs35l33->codec; + unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2; + int i; + + for (i = 0; i < CS35L33_INT_ATTEMPTS; i++) { + /* ack the irq by reading both status registers */ + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1); + + /* Check to see if the unmasked bits are active, + * if not then exit at the exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + break; + + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_err(codec->dev, "Amp short error\n"); + if (!(current_val & CS35L33_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, + CS35L33_AMP_SHORT_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + if (!(current_val & CS35L33_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + CS35L33_CAL_ERR_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_OTE) { + dev_err(codec->dev, "Over temperature error\n"); + if (!(current_val & CS35L33_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, + CS35L33_OTE_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + } + } + + if (sticky_val1 & CS35L33_OTW) { + dev_err(codec->dev, "Over temperature warning\n"); + if (!(current_val & CS35L33_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, + CS35L33_OTW_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + } + } + if (CS35L33_ALIVE_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n"); + + if (CS35L33_MCLK_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: MCLK Interrupt\n"); + + if (CS35L33_VMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VMON Overflow Interrupt\n"); + + if (CS35L33_IMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: IMON Overflow Interrupt\n"); + + if (CS35L33_VPMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + if (CS35L33_VBSTMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + } + + if (i == CS35L33_INT_ATTEMPTS) { + dev_err(codec->dev, + "ERROR: Interrupt handler exceeded max attpmpts\n"); + return -EINVAL; + } + + return IRQ_HANDLED; +} + +static const char * const cs35l33_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l33_of_get_pdata(struct device *dev, + struct cs35l33_private *cs35l33) +{ + struct device_node *np = dev->of_node; + struct cs35l33_pdata *pdata = &cs35l33->pdata; + u32 val32; + + if (!np) + return 0; + + if (of_property_read_u32(np, "boost-ctl", &val32) >= 0) { + pdata->boost_ctl = val32; + pdata->amp_drv_sel = 1; + } + + if (of_property_read_u32(np, "ramp-rate", &val32) >= 0) { + pdata->ramp_rate = val32; + cs35l33->enable_soft_ramp = true; + } + + if (of_property_read_u32(np, "boost-ipk", &val32) >= 0) + pdata->boost_ipk = val32; + + if (of_property_read_u32(np, "imon-adc-scale", &val32) >= 0) { + if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6)) + pdata->imon_adc_scale = val32; + else + /* use default value */ + pdata->imon_adc_scale = 0x8; + } else { + /* use default value */ + pdata->imon_adc_scale = 0x8; + } + + cs35l33_get_hg_data(np, pdata); + + return 0; +} + +static int cs35l33_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l33_private *cs35l33; + struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev); + int ret, devid, i; + unsigned int reg; + + cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private), + GFP_KERNEL); + if (!cs35l33) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l33); + cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap); + if (IS_ERR(cs35l33->regmap)) { + ret = PTR_ERR(cs35l33->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, true); + + for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++) + cs35l33->core_supplies[i].supply + = cs35l33_core_supplies[i]; + cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l33->pdata = *pdata; + } else { + cs35l33_of_get_pdata(&i2c_client->dev, cs35l33); + pdata = &cs35l33->pdata; + } + + ret = request_threaded_irq(i2c_client->irq, NULL, cs35l33_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs35l33", cs35l33); + if (ret != 0) + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + /* We could issue !RST or skip it based on AMP topology */ + cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_HIGH); + + if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { + dev_warn(&i2c_client->dev, + "%s WARNING: No reset gpio assigned\n", __func__); + } else if (IS_ERR(cs35l33->reset_gpio)) { + dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", + __func__); + return PTR_ERR(cs35l33->reset_gpio); + } + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + goto err_irq; + } + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + regcache_cache_only(cs35l33->regmap, false); + + /* initialize codec */ + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L33_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L33 Device ID (%X). Expected ID %X\n", + devid, CS35L33_CHIP_ID); + goto err_enable; + } + + ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_enable; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L33, Revision: %02X\n", ret & 0xFF); + + ret = regmap_register_patch(cs35l33->regmap, + cs35l33_patch, ARRAY_SIZE(cs35l33_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Error in applying regmap patch: %d\n", ret); + goto err_enable; + } + + /* disable mclk and tdm */ + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l33, &cs35l33_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, "%s: Register codec failed\n", + __func__); + goto err_irq; + } + + return 0; + +err_enable: + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); +err_irq: + free_irq(i2c_client->irq, cs35l33); + + return ret; +} + +static int cs35l33_i2c_remove(struct i2c_client *client) +{ + struct cs35l33_private *cs35l33 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + free_irq(client->irq, cs35l33); + + return 0; +} + +static const struct of_device_id cs35l33_of_match[] = { + { .compatible = "cirrus,cs35l33", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l33_of_match); + +static const struct i2c_device_id cs35l33_id[] = { + {"cs35l33", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l33_id); + +static struct i2c_driver cs35l33_i2c_driver = { + .driver = { + .name = "cs35l33", + .owner = THIS_MODULE, + .pm = &cs35l33_pm_ops, + .of_match_table = cs35l33_of_match, + + }, + .id_table = cs35l33_id, + .probe = cs35l33_i2c_probe, + .remove = cs35l33_i2c_remove, + +}; +module_i2c_driver(cs35l33_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L33 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h new file mode 100644 index 0000000..2f31952 --- /dev/null +++ b/sound/soc/codecs/cs35l33.h @@ -0,0 +1,221 @@ +/* + * cs35l33.h -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS35L33_H__ +#define __CS35L33_H__ + +#define CS35L33_CHIP_ID 0x00035A33 +#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */ +#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */ +#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */ +#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */ +#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */ +#define CS35L33_ADC_CTL 0x0E /* ADC Control */ +#define CS35L33_DAC_CTL 0x0F /* DAC Control */ +#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */ +#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */ +#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */ +#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */ +#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */ +#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */ +#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */ +#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */ +#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */ +#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */ +#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */ +#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */ +#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */ +#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */ +#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */ +#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */ +#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */ +#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */ +#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */ +#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */ +#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */ +#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */ +#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */ +#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */ +#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */ +#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */ +#define CS35L33_HG_STATUS 0x3F /* H/G Status */ +#define CS35L33_MAX_REGISTER 0x59 + +#define CS35L33_MCLK_5644 5644800 +#define CS35L33_MCLK_6144 6144000 +#define CS35L33_MCLK_6 6000000 +#define CS35L33_MCLK_11289 11289600 +#define CS35L33_MCLK_12 12000000 +#define CS35L33_MCLK_12288 12288000 + +/* CS35L33_PWRCTL1 */ +#define CS35L33_PDN_AMP (1 << 7) +#define CS35L33_PDN_BST (1 << 2) +#define CS35L33_PDN_ALL 1 + +/* CS35L33_PWRCTL2 */ +#define CS35L33_PDN_VMON_SHIFT 7 +#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT) +#define CS35L33_PDN_IMON_SHIFT 6 +#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT) +#define CS35L33_PDN_VPMON_SHIFT 5 +#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT) +#define CS35L33_PDN_VBSTMON_SHIFT 4 +#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT) +#define CS35L33_SDOUT_3ST_I2S_SHIFT 3 +#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT) +#define CS35L33_PDN_SDIN_SHIFT 2 +#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT) +#define CS35L33_PDN_TDM_SHIFT 1 +#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT) + +/* CS35L33_CLK_CTL */ +#define CS35L33_MCLKDIS (1 << 7) +#define CS35L33_MCLKDIV2 (1 << 6) +#define CS35L33_SDOUT_3ST_TDM (1 << 5) +#define CS35L33_INT_FS_RATE (1 << 4) +#define CS35L33_ADSP_FS 0xF + +/* CS35L33_PROTECT_CTL */ +#define CS35L33_ALIVE_WD_DIS (3 << 2) + +/* CS35L33_BST_CTL1 */ +#define CS35L33_BST_CTL_SRC (1 << 6) +#define CS35L33_BST_CTL_SHIFT (1 << 5) +#define CS35L33_BST_CTL_MASK 0x3F + +/* CS35L33_BST_CTL2 */ +#define CS35L33_TDM_WD_SEL (1 << 4) +#define CS35L33_ALIVE_WD_DIS2 (1 << 3) +#define CS35L33_VBST_SR_STEP 0x3 + +/* CS35L33_ADSP_CTL */ +#define CS35L33_ADSP_DRIVE (1 << 7) +#define CS35L33_MS_MASK (1 << 6) +#define CS35L33_SDIN_LOC (3 << 4) +#define CS35L33_ALIVE_RATE 0x3 + +/* CS35L33_ADC_CTL */ +#define CS35L33_INV_VMON (1 << 7) +#define CS35L33_INV_IMON (1 << 6) +#define CS35L33_ADC_NOTCH_DIS (1 << 5) +#define CS35L33_IMON_SCALE 0xF + +/* CS35L33_DAC_CTL */ +#define CS35L33_INV_DAC (1 << 7) +#define CS35L33_DAC_NOTCH_DIS (1 << 5) +#define CS35L33_DIGSFT (1 << 4) +#define CS35L33_DSR_RATE 0xF + +/* CS35L33_CLASSD_CTL */ +#define CS35L33_AMP_SD (1 << 6) +#define CS35L33_AMP_DRV_SEL_SRC (1 << 5) +#define CS35L33_AMP_DRV_SEL_MASK 0x10 +#define CS35L33_AMP_DRV_SEL_SHIFT 4 +#define CS35L33_AMP_CAL (1 << 3) +#define CS35L33_GAIN_CHG_ZC_MASK 0x04 +#define CS35L33_GAIN_CHG_ZC_SHIFT 2 +#define CS35L33_CLASS_D_CTL_MASK 0x3F + +/* CS35L33_AMP_CTL */ +#define CS35L33_AMP_GAIN 0xF0 +#define CS35L33_CAL_ERR_RLS (1 << 3) +#define CS35L33_AMP_SHORT_RLS (1 << 2) +#define CS35L33_OTW_RLS (1 << 1) +#define CS35L33_OTE_RLS 1 + +/* CS35L33_INT_MASK_1 */ +#define CS35L33_M_CAL_ERR_SHIFT 6 +#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT) +#define CS35L33_M_ALIVE_ERR_SHIFT 5 +#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT) +#define CS35L33_M_AMP_SHORT_SHIFT 2 +#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT) +#define CS35L33_M_OTW_SHIFT 1 +#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT) +#define CS35L33_M_OTE_SHIFT 0 +#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT) + +/* CS35L33_INT_STATUS_1 */ +#define CS35L33_CAL_ERR (1 << 6) +#define CS35L33_ALIVE_ERR (1 << 5) +#define CS35L33_ADSPCLK_ERR (1 << 4) +#define CS35L33_MCLK_ERR (1 << 3) +#define CS35L33_AMP_SHORT (1 << 2) +#define CS35L33_OTW (1 << 1) +#define CS35L33_OTE (1 << 0) + +/* CS35L33_INT_STATUS_2 */ +#define CS35L33_VMON_OVFL (1 << 7) +#define CS35L33_IMON_OVFL (1 << 6) +#define CS35L33_VPMON_OVFL (1 << 5) +#define CS35L33_VBSTMON_OVFL (1 << 4) +#define CS35L33_PDN_DONE 1 + +/* CS35L33_BST_CTL4 */ +#define CS35L33_BST_RGS 0x70 +#define CS35L33_BST_COEFF3 0xF + +/* CS35L33_HG_MEMLDO_CTL */ +#define CS35L33_MEM_DEPTH_SHIFT 5 +#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT) +#define CS35L33_LDO_THLD_SHIFT 1 +#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT) +#define CS35L33_LDO_DISABLE_SHIFT 0 +#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT) + +/* CS35L33_LDO_DEL */ +#define CS35L33_VP_HG_VA_SHIFT 5 +#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT) +#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2 +#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT) +#define CS35L33_VP_HG_RATE_SHIFT 0 +#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT) + +/* CS35L33_HG_HEAD */ +#define CS35L33_HD_RM_SHIFT 0 +#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT) + +/* CS35L33_HG_EN */ +#define CS35L33_CLASS_HG_ENA_SHIFT 7 +#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT) +#define CS35L33_VP_HG_AUTO_SHIFT 6 +#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6) +#define CS35L33_VP_HG_SHIFT 0 +#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT) + +#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000) +#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* CS35L33_{RX,TX}_X */ +#define CS35L33_X_STATE_SHIFT 7 +#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT) +#define CS35L33_X_LOC_SHIFT 0 +#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT) + +/* CS35L33_RX_AUD */ +#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5 +#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT) + +#endif -- 1.9.1

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Re: [alsa-devel] writing driver for MOTU PCI 324, 2408 MK-II
by Jef 28 Mar '16

28 Mar '16

Forest Bond <f_a_b <at> softhome.net> writes: > > Hello all, > > I'm new to the list. Joined up because I've decided I want to write a > driver for the Mark of the Unicorn PCI-324 PCI card. This card allows a > variety of MOTU external interfaces to be used. I own a MOTU 2408 MK > II, an 8 channel interface. This (and similar interfaces from MOTU) are > common in the pro audio world. > > I have emailed MOTU tech support, but have not gotten a response. I > have a feeling this will be the case in the future, as well. MOTU has > strong ties with Apple. > > I've searched the list archive, and, from what I can see, noone has > begun work on a driver for these interfaces. It may not be feasible, > given the lack of technical information available (yeah, yeah, I know, > welcome to the wonderful world of linux driver development). If, > however, anyone is aware of an attempt to write a driver, please let me > know. Also, if anyone is interested in pitching in, send me an email. > > I have limited experience writing drivers (well, next to none, > actually), but I'm gonna give it a shot anyway. > > Couple of questions: > > Can released ASIO drivers be used to gather technical information on the > hardware? > > Are there any good techniques for probing the hardware and discovering > necessary technical info? > > Any help would be appreciated. > > Thanks! > Forest > Hello! I know this is an old post, but I was hoping that the sentiment was still alive and strong. I have recently made the transition to Linux as my primary OS, and I, too, am looking to move my audio gear along as well. Has there been any progress on the PCI-324 linux driver development front? Is there an active project in the works, or has this idea been abandoned? Cheers! And thank you. Jef

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[alsa-devel] [PATCH 0/4] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 28 Mar '16

28 Mar '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Jose Abreu (4): drm/i2c/adv7511: Add audio support ARC: axs10x: Update defconfigs so that audio is enabled ASoC: dwc: Add I2S HDMI audio support using custom platform driver ARC: axs10x: Update defconfigs so that I2S is enabled arch/arc/boot/dts/axs10x_mb.dtsi | 20 +- arch/arc/configs/axs101_defconfig | 4 + arch/arc/configs/axs103_defconfig | 6 + arch/arc/configs/axs103_smp_defconfig | 6 + drivers/gpu/drm/i2c/Kconfig | 8 + drivers/gpu/drm/i2c/Makefile | 2 + drivers/gpu/drm/i2c/adv7511.c | 1024 --------------------------------- drivers/gpu/drm/i2c/adv7511.h | 41 ++ drivers/gpu/drm/i2c/adv7511_audio.c | 310 ++++++++++ drivers/gpu/drm/i2c/adv7511_core.c | 1005 ++++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 384 ++++++++++++- 13 files changed, 1771 insertions(+), 1041 deletions(-) delete mode 100644 drivers/gpu/drm/i2c/adv7511.c create mode 100644 drivers/gpu/drm/i2c/adv7511_audio.c create mode 100644 drivers/gpu/drm/i2c/adv7511_core.c -- 1.9.1

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[alsa-devel] [PATCH 0/2] ALSA: dice: fix a fatal bug when no sound card is registered
by Takashi Sakamoto 28 Mar '16

28 Mar '16

Hi, This patchset is for Linux 4.6-rc2. ALSA dice driver causes kernel NULL pointer dereference in a case that no sound card instance is registered and unit is going to be removed. This is observed when dice-based units are connected to Linux system and disconnected within 2 seconds, thus it's rare. Additionally, even if the bug is fixed, the driver dumps warnings for most dice-based models which supports a pair of isochronous streams. This can occur regardless of the former bug. Users can see this for unsupported dice-based models. This patchset fixes these bugs. Takashi Sakamoto (2): ALSA: dice: fix NULL pointer dereference at remove units when sound card is not registered ALSA: firewire-lib: skip releasing stream data when it's not initialized sound/firewire/amdtp-stream.c | 4 ++++ sound/firewire/dice/dice-stream.c | 14 ++++---------- 2 files changed, 8 insertions(+), 10 deletions(-) -- 2.7.3

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[alsa-devel] [PATCH 0/4] ALSA: firewire: some trivial fixes
by Takashi Sakamoto 28 Mar '16

28 Mar '16

Hi, This patchset includes some trivial fixes for ALSA firewire stack. I don't mind to postpone them to 4.7 instead of 4.6-rc2. Regards Takashi Sakamoto (4): ALSA: bebob: remove needless argument from local function ALSA: oxfw: remove needless member from private structure ALSA: fireworks: move model quirk detection code to information parser ALSA: firewire-tascam: add Kconfig entry for TASCAM FW-1804 sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 4 ++-- sound/firewire/fireworks/fireworks.c | 21 +++++++++++---------- sound/firewire/oxfw/oxfw.h | 1 - 4 files changed, 14 insertions(+), 13 deletions(-) -- 2.7.3

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[alsa-devel] [PATCH v2] ALSA: dice: fix memory leak when unplugging
by Takashi Sakamoto 28 Mar '16

28 Mar '16

When sound card is going to be released, dice private data is also released. Then all of data should be released. However, stream data is not released. This causes memory leak when unplugging dice unit. This commit fixes the bug. Fixes: 4bdc495c87b3('ALSA: dice: handle several PCM substreams when any isochronous streams are available') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 845d5e5..ec4db3a 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -446,18 +446,12 @@ end: void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { - struct reg_params tx_params, rx_params; - - snd_dice_transaction_clear_enable(dice); + unsigned int i; - if (get_register_params(dice, &tx_params, &rx_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, &tx_params); - stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); + for (i = 0; i < MAX_STREAMS; i++) { + destroy_stream(dice, AMDTP_IN_STREAM, i); + destroy_stream(dice, AMDTP_OUT_STREAM, i); } - - release_resources(dice); - - dice->substreams_counter = 0; } void snd_dice_stream_update_duplex(struct snd_dice *dice) -- 2.7.3

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[alsa-devel] [PATCH] ALSA: dice: fix memory leak when unplugging
by Takashi Sakamoto 28 Mar '16

28 Mar '16

When sound card is going to be released, dice private data is also released. Then all of data should be released. However, stream data is not released. This causes memory leak when unplugging dice unit. This commit fixes the bug. Fixes: 4bdc495c87b3('ALSA: dice: handle several PCM substreams when any isochronous streams are available') --- sound/firewire/dice/dice-stream.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 845d5e5..ec4db3a 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -446,18 +446,12 @@ end: void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { - struct reg_params tx_params, rx_params; - - snd_dice_transaction_clear_enable(dice); + unsigned int i; - if (get_register_params(dice, &tx_params, &rx_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, &tx_params); - stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); + for (i = 0; i < MAX_STREAMS; i++) { + destroy_stream(dice, AMDTP_IN_STREAM, i); + destroy_stream(dice, AMDTP_OUT_STREAM, i); } - - release_resources(dice); - - dice->substreams_counter = 0; } void snd_dice_stream_update_duplex(struct snd_dice *dice) -- 2.7.3

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Re: [alsa-devel] ASoC: qcom: add mic support
by Dan Carpenter 25 Mar '16

25 Mar '16

Hello Srinivas Kandagatla, The patch fb5d11524eda: "ASoC: qcom: add mic support" from Feb 11, 2016, leads to the following static checker warning: sound/soc/qcom/lpass-platform.c:555 lpass_platform_pcm_new() error: uninitialized symbol 'ret'. sound/soc/qcom/lpass-platform.c 515 csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; 516 if (csubstream) { 517 if (v->alloc_dma_channel) 518 data->wrdma_ch = v->alloc_dma_channel(drvdata, 519 SNDRV_PCM_STREAM_CAPTURE); 520 521 if (IS_ERR_VALUE(data->wrdma_ch)) 522 goto capture_alloc_err; ->wrdma_ch is an int so this should just be: if (data->wrdma_ch < 0) { ret = data->wrdma_ch; goto capture_alloc_err; } The other ->alloc_dma_channel uses the same IS_ERR_VALUE() check as well so we should probably clean that up as well. The IS_ERR_VALUE() function is for places which return unsigned long basically. No need to be fancy for normal "negative returns are error" code. 523 524 drvdata->substream[data->wrdma_ch] = csubstream; 525 526 ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, 527 soc_runtime->platform->dev, 528 size, &csubstream->dma_buffer); 529 if (ret) 530 goto capture_alloc_err; 531 532 ret = regmap_write(drvdata->lpaif_map, 533 LPAIF_WRDMACTL_REG(v, data->wrdma_ch), 0); 534 if (ret) { 535 dev_err(soc_runtime->dev, 536 "%s() error writing to wrdmactl reg: %d\n", 537 __func__, ret); 538 goto capture_reg_err; 539 } 540 } 541 542 return 0; 543 544 capture_reg_err: 545 if (csubstream) 546 snd_dma_free_pages(&csubstream->dma_buffer); 547 548 capture_alloc_err: 549 if (psubstream) 550 snd_dma_free_pages(&psubstream->dma_buffer); 551 552 playback_alloc_err: 553 dev_err(soc_runtime->dev, "Cannot allocate buffer(s)\n"); 554 555 return ret; 556 } regards, dan carpenter

1 0

[alsa-devel] [RFC PATCH v2 0/3] start to support MST
by libin.yang＠linux.intel.com 25 Mar '16

25 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> These patches are draft patches to support MST. The patches doesn't work because of lacking support in gfx driver. Libin Yang (3): ALSA: hda - add DP mst verb support ALSA - hda: add devid support in acom ALSA - hda: add DP MST support drivers/gpu/drm/i915/intel_audio.c | 2 +- include/drm/i915_component.h | 2 +- include/sound/hda_i915.h | 6 +- sound/hda/hdac_i915.c | 7 +- sound/pci/hda/hda_codec.c | 86 ++++++++++++++++- sound/pci/hda/hda_codec.h | 3 + sound/pci/hda/patch_hdmi.c | 184 +++++++++++++++++++++++++++---------- 7 files changed, 227 insertions(+), 63 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: adau1701: Reset codec based on sample rate changes
by pascal.huerst＠gmail.com 24 Mar '16

24 Mar '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> Instead of checking if mclk/lrclk ratio has changed, check if sample rate has changed. In certain cases, the mclk might be changed in the machine driver, which can lead to the same mclk/lrclk ration, eventhow the sample rate has changed. Since the codec has to be programmed differently for every sample rate, its better to check for samplerate changes instead of mclk/lrclk ration changes. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/adau1701.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 202dea1..1dbbcda 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -111,6 +111,7 @@ struct adau1701 { unsigned int dai_fmt; unsigned int pll_clkdiv; unsigned int sysclk; + unsigned int current_rate; struct regmap *regmap; struct i2c_client *client; u8 pin_config[12]; @@ -436,22 +437,24 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); - unsigned int clkdiv = adau1701->sysclk / params_rate(params); + unsigned int rate = params_rate(params); + unsigned int clkdiv = adau1701->sysclk / rate; unsigned int val; int ret; /* - * If the mclk/lrclk ratio changes, the chip needs updated PLL + * If the sample rate changes, the chip needs updated PLL * mode GPIO settings, and a full reset cycle, including a new * firmware upload. */ - if (clkdiv != adau1701->pll_clkdiv) { - ret = adau1701_reset(codec, clkdiv, params_rate(params)); + if (adau1701->current_rate != rate) { + adau1701->current_rate = rate; + ret = adau1701_reset(codec, clkdiv, rate); if (ret < 0) return ret; } - switch (params_rate(params)) { + switch (rate) { case 192000: val = ADAU1701_DSPCTRL_SR_192; break; -- 2.5.0

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[alsa-devel] [PATCH - tinycompress - brace issue 1/1] tinycompress: fix compilation error: missing brace issue
by francois.gaffie＠intel.com 24 Mar '16

24 Mar '16

From: François Gaffie <francois.gaffie(a)intel.com> Signed-off-by: François Gaffie <francois.gaffie(a)intel.com> diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 68626a4..d35801b 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -273,4 +273,7 @@ int is_compress_ready(struct compress *compress); /* Returns a human readable reason for the last error */ const char *compress_get_error(struct compress *compress); +#if defined(__cplusplus) +} +#endif #endif diff --git a/include/tinycompress/tinymp3.h b/include/tinycompress/tinymp3.h index a709c39..7015320 100644 --- a/include/tinycompress/tinymp3.h +++ b/include/tinycompress/tinymp3.h @@ -102,5 +102,7 @@ enum mp3_stereo_mode { DUAL = 0x02, MONO = 0x03 }; - +#if defined(__cplusplus) +} +#endif #endif -- 2.5.0 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

1 0

[alsa-devel] [PATCH 4/4] [meida] media-device: dynamically allocate struct media_devnode
by Mauro Carvalho Chehab 24 Mar '16

24 Mar '16

struct media_devnode is currently embedded at struct media_device. While this works fine during normal usage, it leads to a race condition during devnode unregister. the problem is that drivers assume that, after calling media_device_unregister(), the struct that contains media_device can be freed. This is not true, as it can't be freed until userspace closes all opened /dev/media devnodes. In other words, if the media devnode is still open, and media_device gets freed, any call to an ioctl will make the core to try to access struct media_device, with will cause an use-after-free and even GPF. Fix this by dynamically allocating the struct media_devnode and only freeing it when it is safe. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- drivers/media/media-device.c | 38 ++++++++++++++++++++++------------ drivers/media/media-devnode.c | 7 ++++++- drivers/media/usb/au0828/au0828-core.c | 4 ++-- drivers/media/usb/uvc/uvc_driver.c | 2 +- include/media/media-device.h | 5 +---- include/media/media-devnode.h | 15 ++++++++++++-- sound/usb/media.c | 8 +++---- 7 files changed, 52 insertions(+), 27 deletions(-) diff --git a/drivers/media/media-device.c b/drivers/media/media-device.c index 10cc4766de10..d10dc615e7a8 100644 --- a/drivers/media/media-device.c +++ b/drivers/media/media-device.c @@ -428,7 +428,7 @@ static long media_device_ioctl(struct file *filp, unsigned int cmd, unsigned long arg) { struct media_devnode *devnode = media_devnode_data(filp); - struct media_device *dev = to_media_device(devnode); + struct media_device *dev = devnode->media_dev; long ret; switch (cmd) { @@ -504,7 +504,7 @@ static long media_device_compat_ioctl(struct file *filp, unsigned int cmd, unsigned long arg) { struct media_devnode *devnode = media_devnode_data(filp); - struct media_device *dev = to_media_device(devnode); + struct media_device *dev = devnode->media_dev; long ret; switch (cmd) { @@ -540,7 +540,8 @@ static const struct media_file_operations media_device_fops = { static ssize_t show_model(struct device *cd, struct device_attribute *attr, char *buf) { - struct media_device *mdev = to_media_device(to_media_devnode(cd)); + struct media_devnode *devnode = to_media_devnode(cd); + struct media_device *mdev = devnode->media_dev; return sprintf(buf, "%.*s\n", (int)sizeof(mdev->model), mdev->model); } @@ -718,25 +719,36 @@ EXPORT_SYMBOL_GPL(media_device_cleanup); int __must_check __media_device_register(struct media_device *mdev, struct module *owner) { + struct media_devnode *devnode; int ret; mutex_lock(&mdev->graph_mutex); + devnode = kzalloc(sizeof(*devnode), GFP_KERNEL); + if (!devnode) + return -ENOMEM; + /* Register the device node. */ - mdev->devnode.fops = &media_device_fops; - mdev->devnode.parent = mdev->dev; - mdev->devnode.release = media_device_release; + mdev->devnode = devnode; + devnode->fops = &media_device_fops; + devnode->parent = mdev->dev; + devnode->release = media_device_release; /* Set version 0 to indicate user-space that the graph is static */ mdev->topology_version = 0; - ret = media_devnode_register(&mdev->devnode, owner); - if (ret < 0) + ret = media_devnode_register(mdev, devnode, owner); + if (ret < 0) { + mdev->devnode = NULL; + kfree(devnode); goto err; + } - ret = device_create_file(&mdev->devnode.dev, &dev_attr_model); + ret = device_create_file(&devnode->dev, &dev_attr_model); if (ret < 0) { - media_devnode_unregister(&mdev->devnode); + mdev->devnode = NULL; + media_devnode_unregister(devnode); + kfree(devnode); goto err; } @@ -800,9 +812,9 @@ static void __media_device_unregister(struct media_device *mdev) } /* Check if mdev devnode was registered */ - if (media_devnode_is_registered(&mdev->devnode)) { - device_remove_file(&mdev->devnode.dev, &dev_attr_model); - media_devnode_unregister(&mdev->devnode); + if (media_devnode_is_registered(mdev->devnode)) { + device_remove_file(&mdev->devnode->dev, &dev_attr_model); + media_devnode_unregister(mdev->devnode); } dev_dbg(mdev->dev, "Media device unregistered\n"); diff --git a/drivers/media/media-devnode.c b/drivers/media/media-devnode.c index ae2bc0b7a368..db47063d8801 100644 --- a/drivers/media/media-devnode.c +++ b/drivers/media/media-devnode.c @@ -44,6 +44,7 @@ #include <linux/uaccess.h> #include <media/media-devnode.h> +#include <media/media-device.h> #define MEDIA_NUM_DEVICES 256 #define MEDIA_NAME "media" @@ -74,6 +75,8 @@ static void media_devnode_release(struct device *cd) /* Release media_devnode and perform other cleanups as needed. */ if (devnode->release) devnode->release(devnode); + + kfree(devnode); } static struct bus_type media_bus_type = { @@ -218,7 +221,8 @@ static const struct file_operations media_devnode_fops = { .llseek = no_llseek, }; -int __must_check media_devnode_register(struct media_devnode *devnode, +int __must_check media_devnode_register(struct media_device *mdev, + struct media_devnode *devnode, struct module *owner) { int minor; @@ -237,6 +241,7 @@ int __must_check media_devnode_register(struct media_devnode *devnode, mutex_unlock(&media_devnode_lock); devnode->minor = minor; + devnode->media_dev = mdev; /* Part 2: Initialize and register the character device */ cdev_init(&devnode->cdev, &media_devnode_fops); diff --git a/drivers/media/usb/au0828/au0828-core.c b/drivers/media/usb/au0828/au0828-core.c index 85c13ca5178f..598a85388d77 100644 --- a/drivers/media/usb/au0828/au0828-core.c +++ b/drivers/media/usb/au0828/au0828-core.c @@ -142,7 +142,7 @@ static void au0828_unregister_media_device(struct au0828_dev *dev) struct media_device *mdev = dev->media_dev; struct media_entity_notify *notify, *nextp; - if (!mdev || !media_devnode_is_registered(&mdev->devnode)) + if (!mdev || !media_devnode_is_registered(mdev->devnode)) return; /* Remove au0828 entity_notify callbacks */ @@ -480,7 +480,7 @@ static int au0828_media_device_register(struct au0828_dev *dev, if (!dev->media_dev) return 0; - if (!media_devnode_is_registered(&dev->media_dev->devnode)) { + if (!media_devnode_is_registered(dev->media_dev->devnode)) { /* register media device */ ret = media_device_register(dev->media_dev); diff --git a/drivers/media/usb/uvc/uvc_driver.c b/drivers/media/usb/uvc/uvc_driver.c index 451e84e962e2..302e284a95eb 100644 --- a/drivers/media/usb/uvc/uvc_driver.c +++ b/drivers/media/usb/uvc/uvc_driver.c @@ -1674,7 +1674,7 @@ static void uvc_delete(struct uvc_device *dev) if (dev->vdev.dev) v4l2_device_unregister(&dev->vdev); #ifdef CONFIG_MEDIA_CONTROLLER - if (media_devnode_is_registered(&dev->mdev.devnode)) + if (media_devnode_is_registered(dev->mdev.devnode)) media_device_unregister(&dev->mdev); media_device_cleanup(&dev->mdev); #endif diff --git a/include/media/media-device.h b/include/media/media-device.h index e59772ed8494..b04cfa907350 100644 --- a/include/media/media-device.h +++ b/include/media/media-device.h @@ -347,7 +347,7 @@ struct media_entity_notify { struct media_device { /* dev->driver_data points to this struct. */ struct device *dev; - struct media_devnode devnode; + struct media_devnode *devnode; char model[32]; char driver_name[32]; @@ -403,9 +403,6 @@ struct usb_device; #define MEDIA_DEV_NOTIFY_PRE_LINK_CH 0 #define MEDIA_DEV_NOTIFY_POST_LINK_CH 1 -/* media_devnode to media_device */ -#define to_media_device(node) container_of(node, struct media_device, devnode) - /** * media_entity_enum_init - Initialise an entity enumeration * diff --git a/include/media/media-devnode.h b/include/media/media-devnode.h index e1d5af077adb..cc2b3155593c 100644 --- a/include/media/media-devnode.h +++ b/include/media/media-devnode.h @@ -33,6 +33,8 @@ #include <linux/device.h> #include <linux/cdev.h> +struct media_device; + /* * Flag to mark the media_devnode struct as registered. Drivers must not touch * this flag directly, it will be set and cleared by media_devnode_register and @@ -81,6 +83,8 @@ struct media_file_operations { * before registering the node. */ struct media_devnode { + struct media_device *media_dev; + /* device ops */ const struct media_file_operations *fops; @@ -103,7 +107,8 @@ struct media_devnode { /** * media_devnode_register - register a media device node * - * @devnode: media device node structure we want to register + * @media_dev: struct media_device we want to register a device node + * @devnode: the device node to unregister * @owner: should be filled with %THIS_MODULE * * The registration code assigns minor numbers and registers the new device node @@ -116,7 +121,8 @@ struct media_devnode { * the media_devnode structure is *not* called, so the caller is responsible for * freeing any data. */ -int __must_check media_devnode_register(struct media_devnode *devnode, +int __must_check media_devnode_register(struct media_device *mdev, + struct media_devnode *devnode, struct module *owner); /** @@ -146,9 +152,14 @@ static inline struct media_devnode *media_devnode_data(struct file *filp) * false otherwise. * * @devnode: pointer to struct &media_devnode. + * + * Note: If mdev is NULL, it also returns false. */ static inline int media_devnode_is_registered(struct media_devnode *devnode) { + if (!devnode) + return false; + return test_bit(MEDIA_FLAG_REGISTERED, &devnode->flags); } diff --git a/sound/usb/media.c b/sound/usb/media.c index 6db4878045e5..8fed0adec9e1 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -136,7 +136,7 @@ void media_snd_stream_delete(struct snd_usb_substream *subs) struct media_device *mdev; mdev = mctl->media_dev; - if (mdev && media_devnode_is_registered(&mdev->devnode)) { + if (mdev && media_devnode_is_registered(mdev->devnode)) { media_devnode_remove(mctl->intf_devnode); media_device_unregister_entity(&mctl->media_entity); media_entity_cleanup(&mctl->media_entity); @@ -241,14 +241,14 @@ static void media_snd_mixer_delete(struct snd_usb_audio *chip) if (!mixer->media_mixer_ctl) continue; - if (media_devnode_is_registered(&mdev->devnode)) { + if (media_devnode_is_registered(mdev->devnode)) { media_device_unregister_entity(&mctl->media_entity); media_entity_cleanup(&mctl->media_entity); } kfree(mctl); mixer->media_mixer_ctl = NULL; } - if (media_devnode_is_registered(&mdev->devnode)) + if (media_devnode_is_registered(mdev->devnode)) media_devnode_remove(chip->ctl_intf_media_devnode); chip->ctl_intf_media_devnode = NULL; } @@ -268,7 +268,7 @@ int media_snd_device_create(struct snd_usb_audio *chip, if (!mdev->dev) media_device_usb_init(mdev, usbdev, NULL); - if (!media_devnode_is_registered(&mdev->devnode)) { + if (!media_devnode_is_registered(mdev->devnode)) { ret = media_device_register(mdev); if (ret) { dev_err(&usbdev->dev, -- 2.5.5

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[alsa-devel] [PATCH] ASoC: Add max98371 codec driver
by anish kumar 24 Mar '16

24 Mar '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/max98371.c | 428 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 63 +++ 3 files changed, 508 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 0000000..6c28523 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 0000000..1aba140 --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,428 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const voltage_level_text[] = { + "9.5dB", "10.5dB", "11.5dB", "11.5dB", "12.5dB", + "13.5dB", "14.5dB", "15.5dB", "16.5dB", "17.5dB", + "18.5dB", "19.5dB", "20.5dB" +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_voltage, MAX98371_GAIN, 0, + voltage_level_text); + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const unsigned int max98371_gain_tlv[] = { + 0, 8, TLV_DB_SCALE_ITEM(0, 50, 0), + 9, 10, TLV_DB_SCALE_ITEM(500, 100, 0), + 11, 15, TLV_DB_SCALE_ITEM(0, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Gain", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Voltage Level", max98371_voltage), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + pr_info("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + pr_info("%s: ratio unsupported %d", __func__, blr_clk_ratio); + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + pr_info("%s: rate unsupported %d", __func__, rate); + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", + "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Enable", + "HiFi Playback", MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", + MAX98371_GLOBAL_ENABLE, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC Enable", NULL, "DAI_OUT"}, + {"SPK_OUT", NULL, "DAC Enable"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static int max98371_probe(struct snd_soc_codec *codec) +{ + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + + max98371->codec = codec; + return 0; +} + +static struct snd_soc_codec_driver max98371_codec = { + .probe = max98371_probe, + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 0000000..644316a --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,63 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- 1.9.3

4 11

[alsa-devel] ASoC: TLV320AIC3x how to reduce alsa playback/capture latency?
by Timur Karaldin 24 Mar '16

24 Mar '16

Hello, I have an issue to reduce playback/capture audio latency for AIC3106. Now I'm using following strategy for initial of alsa backend driver in user level application: snd_pcm_hw_params_set_period_size(...,frame_size,...); snd_pcm_hw_params_set_periods_min(...,&periodsp,...); if(periodsp< user_nperiods) periodsp = user_nperiods snd_pcm_hw_params_set_periods_near(....,&periodsp,...); snd_pcm_hw_set_buffer_size(...,frame_size*periodsp,....); [...] snd_pcm_sw_params_set_start_threshold(..., 0U); snd_pcm_sw_params_set_stop_threshold(..., frame_size*periodsp); if(playback) { availmin=frame_size*(periodsp-user_nperiods+1); }else { availmin=frame_size; } snd_pcm_sw_params_set_avail_min(...,availmin) [...] On far end I hear an echo with delay about 1 second, I definitely know that it is a playback/capture (acoustic echo), and delay is too much for any AEC because of (as I think) low latency in alsa driver or misunderstanding how it should be initialized. I already exclude other delays (jitter delay, network delay) My initial parameters: frame_size = 640, rate = 16000, user_nperiods = 2, but after calling snd_pcm_hw_params_set_periods_min I got minimum periodsp = 16. For these parameter buffer (16 buffers 40ms each) total length of buffer is 0.64 sec. My question is how could I reduce this latency? May be some improvements which allow me to set up less then 16 buffers, or I did wrong initialization or is there any another internal buffer which make such long delay for echo? Cheers, Tim Karaldin

1 0

[alsa-devel] question about cross compilation with alsa
by Alvarado, Claudia 23 Mar '16

23 Mar '16

Hello! I have a general question. We are cross compiling on a Linux machine for an ARM processor on an embedded device. When compiling, we specify paths for the include files and our cross compiled Alsa library. We then try to port this to our embedded device. When trying to run the executable on the embedded device we get two problems: 1. it doesn’t recognize the PCM device called ‘default’— I think we just need to call this out with hw:0,0 but we need to play with this. 2. We get this error: “ALSA lib conf.c(snd_config_update_r)Cannot access file /path/to/alsa.conf" * The path that it lists is the path to our cross compiled alsa lib on our linux box… so it seems that in the compilation process, this path is preserved and transferred to the executable somehow. So my question is, is there a way to specify two different library paths for the two devices (linux and embedded) while compiling so that when we port to the embedded device, the executable knows where to look for the alsa library? Normally, this wouldn’t be an issue because the default on a linux machine is usr/include, which I believe gets searched by default. However, on our embedded device, the path is different. Thanks in advance!

1 0

[alsa-devel] [PATCH v2] [media] media-device: use kref for media_device instance
by Mauro Carvalho Chehab 23 Mar '16

23 Mar '16

Now that the media_device can be used by multiple drivers, via devres, we need to be sure that it will be dropped only when all drivers stop using it. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- v2: The kref is now used only when media_device is allocated via the media_device*_devress. This warrants that other drivers won't be affected, and that we can keep media_device_cleanup() balanced with media_device_init(). drivers/media/media-device.c | 117 +++++++++++++++++++++++++-------- drivers/media/usb/au0828/au0828-core.c | 3 +- include/media/media-device.h | 28 ++++++++ sound/usb/media.c | 3 +- 4 files changed, 118 insertions(+), 33 deletions(-) diff --git a/drivers/media/media-device.c b/drivers/media/media-device.c index c32fa15cc76e..4a97d92a7e7d 100644 --- a/drivers/media/media-device.c +++ b/drivers/media/media-device.c @@ -707,11 +707,16 @@ void media_device_init(struct media_device *mdev) } EXPORT_SYMBOL_GPL(media_device_init); -void media_device_cleanup(struct media_device *mdev) +static void __media_device_cleanup(struct media_device *mdev) { ida_destroy(&mdev->entity_internal_idx); mdev->entity_internal_idx_max = 0; media_entity_graph_walk_cleanup(&mdev->pm_count_walk); +} + +void media_device_cleanup(struct media_device *mdev) +{ + __media_device_cleanup(mdev); mutex_destroy(&mdev->graph_mutex); } EXPORT_SYMBOL_GPL(media_device_cleanup); @@ -721,6 +726,9 @@ int __must_check __media_device_register(struct media_device *mdev, { int ret; + /* Check if mdev was ever registered at all */ + mutex_lock(&mdev->graph_mutex); + /* Register the device node. */ mdev->devnode.fops = &media_device_fops; mdev->devnode.parent = mdev->dev; @@ -731,17 +739,19 @@ int __must_check __media_device_register(struct media_device *mdev, ret = media_devnode_register(&mdev->devnode, owner); if (ret < 0) - return ret; + goto err; ret = device_create_file(&mdev->devnode.dev, &dev_attr_model); if (ret < 0) { media_devnode_unregister(&mdev->devnode); - return ret; + goto err; } dev_dbg(mdev->dev, "Media device registered\n"); - return 0; +err: + mutex_unlock(&mdev->graph_mutex); + return ret; } EXPORT_SYMBOL_GPL(__media_device_register); @@ -773,24 +783,13 @@ void media_device_unregister_entity_notify(struct media_device *mdev, } EXPORT_SYMBOL_GPL(media_device_unregister_entity_notify); -void media_device_unregister(struct media_device *mdev) +static void __media_device_unregister(struct media_device *mdev) { struct media_entity *entity; struct media_entity *next; struct media_interface *intf, *tmp_intf; struct media_entity_notify *notify, *nextp; - if (mdev == NULL) - return; - - mutex_lock(&mdev->graph_mutex); - - /* Check if mdev was ever registered at all */ - if (!media_devnode_is_registered(&mdev->devnode)) { - mutex_unlock(&mdev->graph_mutex); - return; - } - /* Remove all entities from the media device */ list_for_each_entry_safe(entity, next, &mdev->entities, graph_obj.list) __media_device_unregister_entity(entity); @@ -807,38 +806,98 @@ void media_device_unregister(struct media_device *mdev) kfree(intf); } - mutex_unlock(&mdev->graph_mutex); - - device_remove_file(&mdev->devnode.dev, &dev_attr_model); - media_devnode_unregister(&mdev->devnode); + /* Check if mdev devnode was registered */ + if (media_devnode_is_registered(&mdev->devnode)) { + device_remove_file(&mdev->devnode.dev, &dev_attr_model); + media_devnode_unregister(&mdev->devnode); + } dev_dbg(mdev->dev, "Media device unregistered\n"); } + +void media_device_unregister(struct media_device *mdev) +{ + if (mdev == NULL) + return; + + mutex_lock(&mdev->graph_mutex); + __media_device_unregister(mdev); + mutex_unlock(&mdev->graph_mutex); +} EXPORT_SYMBOL_GPL(media_device_unregister); static void media_device_release_devres(struct device *dev, void *res) { } -struct media_device *media_device_get_devres(struct device *dev) +static void do_media_device_unregister_devres(struct kref *kref) { + struct media_device_devres *mdev_devres; struct media_device *mdev; + int ret; - mdev = devres_find(dev, media_device_release_devres, NULL, NULL); - if (mdev) - return mdev; + mdev_devres = container_of(kref, struct media_device_devres, kref); - mdev = devres_alloc(media_device_release_devres, - sizeof(struct media_device), GFP_KERNEL); - if (!mdev) + if (!mdev_devres) + return; + + mdev = &mdev_devres->mdev; + + mutex_lock(&mdev->graph_mutex); + __media_device_unregister(mdev); + __media_device_cleanup(mdev); + mutex_unlock(&mdev->graph_mutex); + mutex_destroy(&mdev->graph_mutex); + + ret = devres_destroy(mdev->dev, media_device_release_devres, + NULL, NULL); + pr_debug("%s: devres_destroy() returned %d\n", __func__, ret); +} + +void media_device_unregister_devres(struct media_device *mdev) +{ + struct media_device_devres *mdev_devres; + + mdev_devres = container_of(mdev, struct media_device_devres, mdev); + kref_put(&mdev_devres->kref, do_media_device_unregister_devres); +} +EXPORT_SYMBOL_GPL(media_device_unregister_devres); + +struct media_device *media_device_get_devres(struct device *dev) +{ + struct media_device_devres *mdev_devres, *ptr; + + mdev_devres = devres_find(dev, media_device_release_devres, NULL, NULL); + if (mdev_devres) { + kref_get(&mdev_devres->kref); + return &mdev_devres->mdev; + } + + mdev_devres = devres_alloc(media_device_release_devres, + sizeof(struct media_device_devres), + GFP_KERNEL); + if (!mdev_devres) return NULL; - return devres_get(dev, mdev, NULL, NULL); + + ptr = devres_get(dev, mdev_devres, NULL, NULL); + if (ptr) + kref_init(&ptr->kref); + else + devres_free(mdev_devres); + + return &ptr->mdev; } EXPORT_SYMBOL_GPL(media_device_get_devres); struct media_device *media_device_find_devres(struct device *dev) { - return devres_find(dev, media_device_release_devres, NULL, NULL); + struct media_device_devres *mdev_devres; + + mdev_devres = devres_find(dev, media_device_release_devres, NULL, NULL); + if (!mdev_devres) + return NULL; + + return &mdev_devres->mdev; } EXPORT_SYMBOL_GPL(media_device_find_devres); diff --git a/drivers/media/usb/au0828/au0828-core.c b/drivers/media/usb/au0828/au0828-core.c index 06da73f1ff22..060904ed8f20 100644 --- a/drivers/media/usb/au0828/au0828-core.c +++ b/drivers/media/usb/au0828/au0828-core.c @@ -157,8 +157,7 @@ static void au0828_unregister_media_device(struct au0828_dev *dev) dev->media_dev->enable_source = NULL; dev->media_dev->disable_source = NULL; - media_device_unregister(dev->media_dev); - media_device_cleanup(dev->media_dev); + media_device_unregister_devres(dev->media_dev); dev->media_dev = NULL; #endif } diff --git a/include/media/media-device.h b/include/media/media-device.h index b21ef244ad3e..e59772ed8494 100644 --- a/include/media/media-device.h +++ b/include/media/media-device.h @@ -23,6 +23,7 @@ #ifndef _MEDIA_DEVICE_H #define _MEDIA_DEVICE_H +#include <linux/kref.h> #include <linux/list.h> #include <linux/mutex.h> @@ -382,6 +383,16 @@ struct media_device { unsigned int notification); }; +/** + * struct media_device_devres - Media device device resource + * @mdev: pointer to struct media_device + * @kref: Object refcount + */ +struct media_device_devres { + struct media_device mdev; + struct kref kref; +}; + /* We don't need to include pci.h or usb.h here */ struct pci_dev; struct usb_device; @@ -604,6 +615,19 @@ struct media_device *media_device_get_devres(struct device *dev); */ struct media_device *media_device_find_devres(struct device *dev); +/** + * media_device_unregister_devres) - Unregister media device allocated as + * as device resource + * + * @dev: pointer to struct &device. + * + * Devices allocated via media_device_get_devres should be de-alocalted + * and freed via this function. Callers should not call + * media_device_unregister() nor media_device_cleanup() on devices + * allocated via media_device_get_devres(). + */ +void media_device_unregister_devres(struct media_device *mdev); + /* Iterate over all entities. */ #define media_device_for_each_entity(entity, mdev) \ list_for_each_entry(entity, &(mdev)->entities, graph_obj.list) @@ -688,6 +712,10 @@ static inline struct media_device *media_device_find_devres(struct device *dev) return NULL; } +static inline void media_device_unregister_devres(struct media_device *mdev) +{ +} + static inline void media_device_pci_init(struct media_device *mdev, struct pci_dev *pci_dev, char *name) diff --git a/sound/usb/media.c b/sound/usb/media.c index 93a50d01490c..f78955fd0d6e 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -311,8 +311,7 @@ void media_snd_device_delete(struct snd_usb_audio *chip) media_snd_mixer_delete(chip); if (mdev) { - if (media_devnode_is_registered(&mdev->devnode)) - media_device_unregister(mdev); + media_device_unregister_devres(mdev); chip->media_dev = NULL; } } -- 2.5.0

3 9

[alsa-devel] [PATCH] ucm: add cset-tlv command
by Hsin-Yu Chao 23 Mar '16

23 Mar '16

This patch enables UCM to set a file in TLV format to kcontrol by: cset-tlv "name='<kcontrol-name>' <path-to-file>" This new 'cset-tlv' command will be used to write audio DSP to specific alsa control, where the driver expectes a file in TLV format. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/ucm/main.c | 80 ++++++++++++++++++++++++++++++++++++++++++++--------- src/ucm/parser.c | 10 +++++++ src/ucm/ucm_local.h | 1 + 3 files changed, 78 insertions(+), 13 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7e44603..a4ccb65 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -161,6 +161,47 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int tlv_parse(char **res, + snd_ctl_elem_info_t *info, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + + if (!snd_ctl_elem_info_is_tlv_writable(info)) { + err = -EINVAL; + return err; + } + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + *res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, *res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -errno; + free(*res); + *res = NULL; + } + + __fail: + close(fd); + return err; +} + static int binary_file_parse(snd_ctl_elem_value_t *dst, snd_ctl_elem_info_t *info, const char *filepath) @@ -226,6 +267,7 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *value; snd_ctl_elem_info_t *info; + char *res = NULL; snd_ctl_elem_id_malloc(&id); snd_ctl_elem_value_malloc(&value); @@ -241,23 +283,32 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) err = -EINVAL; goto __fail; } - snd_ctl_elem_value_set_id(value, id); snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_read(ctl, value); - if (err < 0) - goto __fail; err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) - err = binary_file_parse(value, info, pos); - else - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); - if (err < 0) - goto __fail; - err = snd_ctl_elem_write(ctl, value); - if (err < 0) - goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_TLV) { + err = tlv_parse(&res, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_tlv_write(ctl, id, (unsigned int *)res); + if (err < 0) + goto __fail; + } else { + snd_ctl_elem_value_set_id(value, id); + err = snd_ctl_elem_read(ctl, value); + if (err < 0) + goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_write(ctl, value); + if (err < 0) + goto __fail; + } err = 0; __fail: if (id != NULL) @@ -266,6 +317,8 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) free(value); if (info != NULL) free(info); + if (res != NULL) + free(res); return err; } @@ -298,6 +351,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, break; case SEQUENCE_ELEMENT_TYPE_CSET: case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: + case SEQUENCE_ELEMENT_TYPE_CSET_TLV: if (cdev == NULL) { char *playback_ctl = NULL; char *capture_ctl = NULL; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 9e1cb41..d781e1b 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -316,6 +316,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-tlv") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_TLV; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-tlv requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 3a5d2c2..b89de2a 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -48,6 +48,7 @@ #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 #define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 +#define SEQUENCE_ELEMENT_TYPE_CSET_TLV 6 struct ucm_value { struct list_head list; -- 2.1.2

2 5

[alsa-devel] [PATCH V3 0/6] alsabat: clean structure and tinyalsa support
by han.lu＠intel.com 23 Mar '16

23 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> 1. Refactoring alsa process to make structure simpler and cleaner, use general functions and unified return value for convenience of maintaining; 2. Add tinyalsa support, user can choose tinyalsa with "configure --enable-alsabat-backend-tiny" option. By default ALSA lib is used. The intention is for alsabat to run on tinyalsa platforms such as Android and some IoT devices. changes on v3: 1. use option "--enable-alsabat-backend-tiny" instead of "--enable-tinyalsa" to avoid ambiguous. 2. fix inconsistent in man page and configure.ac. changes on v2: 1. add more description to explain the purpose of each patch. 2. use configure option to select tinyalsa, rather than auto detect environment and decide. Lu, Han (6): alsabat: refactoring alsa capture thread alsabat: use general function for wav header update alsabat: clean return value for playback and capture threads alsabat: use general data generator function alsabat: move alsa process to a single block alsabat: add tinyalsa support bat/Makefile.am | 12 +- bat/alsa.c | 217 +++++++++++--------------- bat/alsabat.1 | 3 + bat/bat.c | 58 +++---- bat/common.c | 68 +++++++++ bat/common.h | 17 ++- bat/signal.c | 13 +- bat/tinyalsa.c | 460 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/tinyalsa.h | 20 +++ configure.ac | 17 +++ 10 files changed, 711 insertions(+), 174 deletions(-) create mode 100644 bat/tinyalsa.c create mode 100644 bat/tinyalsa.h -- 2.5.0

2 7

[alsa-devel] [PATCH 1/3] ALSA: hda - Asus N750JV external subwoofer fixup
by Takashi Iwai 23 Mar '16

23 Mar '16

From: Bobi Mihalca <bobbymihalca(a)touchtech.ro> ASUS N750JV needs the same fixup as N550 for enabling its subwoofer. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=115181 Signed-off-by: Bobi Mihalca <bobbymihalca(a)touchtech.ro> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f5ca0b9ce27..44c52c2b7be7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6670,6 +6670,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), + SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), -- 2.7.4

1 2

[alsa-devel] [PATCH] ASoC: rt5645: Add dmi_system_id "Google Setzer"
by John Lin 23 Mar '16

23 Mar '16

Add platform specific data for Setzer project. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 93e8c90..75e5a4a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3552,6 +3552,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, + { + .ident = "Google Setzer", + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ucm: add cset-tlv
by Hsin-Yu Chao 23 Mar '16

23 Mar '16

This patch enables UCM to set a file in TLV format to kcontrol by: cset-tlv "name='<kcontrol-name>' <path-to-file>" This new 'cset-tlv' command will be used to write audio DSP to specific alsa control, where the driver expectes a file in TLV format. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/ucm/main.c | 91 +++++++++++++++++++++++++++++++++++++++++++++-------- src/ucm/parser.c | 10 ++++++ src/ucm/ucm_local.h | 1 + 3 files changed, 89 insertions(+), 13 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7e44603..c6bbf27 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -161,6 +161,54 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int read_tlv_file(char **res, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + unsigned int *tlv; + + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + if (sz > 1024 * 1024 || sz < 8) { + uc_error("unsupported file size"); + goto __fail; + } + *res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, *res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -errno; + free(*res); + *res = NULL; + } + /* Check if the tlv file specifies valid size. */ + tlv = (unsigned int *)(*res); + if (tlv[1] + 2 * sizeof(unsigned int) != sz) { + uc_error("Invalid tlv size"); + free(*res); + *res = NULL; + } + + __fail: + close(fd); + return err; +} + static int binary_file_parse(snd_ctl_elem_value_t *dst, snd_ctl_elem_info_t *info, const char *filepath) @@ -226,6 +274,7 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *value; snd_ctl_elem_info_t *info; + char *res = NULL; snd_ctl_elem_id_malloc(&id); snd_ctl_elem_value_malloc(&value); @@ -241,23 +290,36 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) err = -EINVAL; goto __fail; } - snd_ctl_elem_value_set_id(value, id); snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_read(ctl, value); - if (err < 0) - goto __fail; err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) - err = binary_file_parse(value, info, pos); - else - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); - if (err < 0) - goto __fail; - err = snd_ctl_elem_write(ctl, value); - if (err < 0) - goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_TLV) { + if (!snd_ctl_elem_info_is_tlv_writable(info)) { + err = -EINVAL; + goto __fail; + } + err = read_tlv_file(&res, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_tlv_write(ctl, id, (unsigned int *)res); + if (err < 0) + goto __fail; + } else { + snd_ctl_elem_value_set_id(value, id); + err = snd_ctl_elem_read(ctl, value); + if (err < 0) + goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_write(ctl, value); + if (err < 0) + goto __fail; + } err = 0; __fail: if (id != NULL) @@ -266,6 +328,8 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) free(value); if (info != NULL) free(info); + if (res != NULL) + free(res); return err; } @@ -298,6 +362,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, break; case SEQUENCE_ELEMENT_TYPE_CSET: case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: + case SEQUENCE_ELEMENT_TYPE_CSET_TLV: if (cdev == NULL) { char *playback_ctl = NULL; char *capture_ctl = NULL; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 9e1cb41..d781e1b 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -316,6 +316,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-tlv") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_TLV; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-tlv requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 3a5d2c2..b89de2a 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -48,6 +48,7 @@ #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 #define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 +#define SEQUENCE_ELEMENT_TYPE_CSET_TLV 6 struct ucm_value { struct list_head list; -- 2.1.2

1 0

[alsa-devel] [PATCH v3] ALSA: timer: fix gparams ioctl compatibility for different architectures
by Takashi Sakamoto 23 Mar '16

23 Mar '16

'struct snd_timer_gparams' includes some members with 'unsigned long', therefore its size differs depending on data models of architecture. As a result, x86/x32 applications fail to execute ioctl(2) with SNDRV_TIMER_GPARAMS command on x86_64 machine. This commit fixes this bug by adding a pair of structure and ioctl command for the compatibility. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/timer.c | 20 +++++++++++++------- sound/core/timer_compat.c | 30 +++++++++++++++++++++++++++++- 2 files changed, 42 insertions(+), 8 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index aa1b15c..ea4d999 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1502,17 +1502,13 @@ static int snd_timer_user_ginfo(struct file *file, return err; } -static int snd_timer_user_gparams(struct file *file, - struct snd_timer_gparams __user *_gparams) +static int timer_set_gparams(struct snd_timer_gparams *gparams) { - struct snd_timer_gparams gparams; struct snd_timer *t; int err; - if (copy_from_user(&gparams, _gparams, sizeof(gparams))) - return -EFAULT; mutex_lock(&register_mutex); - t = snd_timer_find(&gparams.tid); + t = snd_timer_find(&gparams->tid); if (!t) { err = -ENODEV; goto _error; @@ -1525,12 +1521,22 @@ static int snd_timer_user_gparams(struct file *file, err = -ENOSYS; goto _error; } - err = t->hw.set_period(t, gparams.period_num, gparams.period_den); + err = t->hw.set_period(t, gparams->period_num, gparams->period_den); _error: mutex_unlock(&register_mutex); return err; } +static int snd_timer_user_gparams(struct file *file, + struct snd_timer_gparams __user *_gparams) +{ + struct snd_timer_gparams gparams; + + if (copy_from_user(&gparams, _gparams, sizeof(gparams))) + return -EFAULT; + return timer_set_gparams(&gparams); +} + static int snd_timer_user_gstatus(struct file *file, struct snd_timer_gstatus __user *_gstatus) { diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 2e90822..6a437eb 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -22,6 +22,19 @@ #include <linux/compat.h> +/* + * ILP32/LP64 has different size for 'long' type. Additionally, the size + * of storage alignment differs depending on architectures. Here, '__packed' + * qualifier is used so that the size of this structure is multiple of 4 and + * it fits to any architectures with 32 bit storage alignment. + */ +struct snd_timer_gparams32 { + struct snd_timer_id tid; + u32 period_num; + u32 period_den; + unsigned char reserved[32]; +} __packed; + struct snd_timer_info32 { u32 flags; s32 card; @@ -32,6 +45,19 @@ struct snd_timer_info32 { unsigned char reserved[64]; }; +static int snd_timer_user_gparams_compat(struct file *file, + struct snd_timer_gparams32 __user *user) +{ + struct snd_timer_gparams gparams; + + if (copy_from_user(&gparams.tid, &user->tid, sizeof(gparams.tid)) || + get_user(gparams.period_num, &user->period_num) || + get_user(gparams.period_den, &user->period_den)) + return -EFAULT; + + return timer_set_gparams(&gparams); +} + static int snd_timer_user_info_compat(struct file *file, struct snd_timer_info32 __user *_info) { @@ -99,6 +125,7 @@ static int snd_timer_user_status_compat(struct file *file, */ enum { + SNDRV_TIMER_IOCTL_GPARAMS32 = _IOW('T', 0x04, struct snd_timer_gparams32), SNDRV_TIMER_IOCTL_INFO32 = _IOR('T', 0x11, struct snd_timer_info32), SNDRV_TIMER_IOCTL_STATUS32 = _IOW('T', 0x14, struct snd_timer_status32), #ifdef CONFIG_X86_X32 @@ -114,7 +141,6 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PVERSION: case SNDRV_TIMER_IOCTL_TREAD: case SNDRV_TIMER_IOCTL_GINFO: - case SNDRV_TIMER_IOCTL_GPARAMS: case SNDRV_TIMER_IOCTL_GSTATUS: case SNDRV_TIMER_IOCTL_SELECT: case SNDRV_TIMER_IOCTL_PARAMS: @@ -128,6 +154,8 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PAUSE_OLD: case SNDRV_TIMER_IOCTL_NEXT_DEVICE: return snd_timer_user_ioctl(file, cmd, (unsigned long)argp); + case SNDRV_TIMER_IOCTL_GPARAMS32: + return snd_timer_user_gparams_compat(file, argp); case SNDRV_TIMER_IOCTL_INFO32: return snd_timer_user_info_compat(file, argp); case SNDRV_TIMER_IOCTL_STATUS32: -- 2.7.3

2 1

[alsa-devel] [PATCH] ALSA: timer: fix gparams ioctl compatibility for different architectures
by Takashi Sakamoto 23 Mar '16

23 Mar '16

'struct snd_timer_gparams' includes some members with 'unsigned long', therefore its size differs depending on data models of architecture. As a result, x86/x32 applications fail to execute ioctl(2) with SNDRV_TIMER_GPARAMS command on x86_64 machine. This commit fixes this bug by adding a pair of structure and ioctl command for the compatibility. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/timer.c | 20 +++++++++++++------- sound/core/timer_compat.c | 34 +++++++++++++++++++++++++++++++++- 2 files changed, 46 insertions(+), 8 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index aa1b15c..ea4d999 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1502,17 +1502,13 @@ static int snd_timer_user_ginfo(struct file *file, return err; } -static int snd_timer_user_gparams(struct file *file, - struct snd_timer_gparams __user *_gparams) +static int timer_set_gparams(struct snd_timer_gparams *gparams) { - struct snd_timer_gparams gparams; struct snd_timer *t; int err; - if (copy_from_user(&gparams, _gparams, sizeof(gparams))) - return -EFAULT; mutex_lock(&register_mutex); - t = snd_timer_find(&gparams.tid); + t = snd_timer_find(&gparams->tid); if (!t) { err = -ENODEV; goto _error; @@ -1525,12 +1521,22 @@ static int snd_timer_user_gparams(struct file *file, err = -ENOSYS; goto _error; } - err = t->hw.set_period(t, gparams.period_num, gparams.period_den); + err = t->hw.set_period(t, gparams->period_num, gparams->period_den); _error: mutex_unlock(&register_mutex); return err; } +static int snd_timer_user_gparams(struct file *file, + struct snd_timer_gparams __user *_gparams) +{ + struct snd_timer_gparams gparams; + + if (copy_from_user(&gparams, _gparams, sizeof(gparams))) + return -EFAULT; + return timer_set_gparams(&gparams); +} + static int snd_timer_user_gstatus(struct file *file, struct snd_timer_gstatus __user *_gstatus) { diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 2e90822..d10c792 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -22,6 +22,19 @@ #include <linux/compat.h> +/* + * ILP32/LP64 has different size for 'long' type. Additionally, the size + * of storage alignment differs depending on architectures. Here, '__packed' + * qualifier is used so that the size of this structure is multiple of 4 and + * it fits to any architectures with 32 bit storage alignment. + */ +struct snd_timer_gparams32 { + struct snd_timer_id tid; + u32 period_num; + u32 period_den; + unsigned char reserved[32]; +} __packed; + struct snd_timer_info32 { u32 flags; s32 card; @@ -32,6 +45,23 @@ struct snd_timer_info32 { unsigned char reserved[64]; }; +static int snd_timer_user_gparams_compat(struct file *file, + struct snd_timer_gparams32 __user *user) +{ + struct snd_timer_gparams gparams; + + if (get_user(gparams.tid.dev_class, &user->tid.dev_class) || + get_user(gparams.tid.dev_sclass, &user->tid.dev_sclass) || + get_user(gparams.tid.card, &user->tid.card) || + get_user(gparams.tid.device, &user->tid.device) || + get_user(gparams.tid.subdevice, &user->tid.subdevice) || + get_user(gparams.period_num, &user->period_num) || + get_user(gparams.period_den, &user->period_den)) + return -EFAULT; + + return timer_set_gparams(&gparams); +} + static int snd_timer_user_info_compat(struct file *file, struct snd_timer_info32 __user *_info) { @@ -99,6 +129,7 @@ static int snd_timer_user_status_compat(struct file *file, */ enum { + SNDRV_TIMER_IOCTL_GPARAMS32 = _IOW('T', 0x04, struct snd_timer_gparams32), SNDRV_TIMER_IOCTL_INFO32 = _IOR('T', 0x11, struct snd_timer_info32), SNDRV_TIMER_IOCTL_STATUS32 = _IOW('T', 0x14, struct snd_timer_status32), #ifdef CONFIG_X86_X32 @@ -114,7 +145,6 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PVERSION: case SNDRV_TIMER_IOCTL_TREAD: case SNDRV_TIMER_IOCTL_GINFO: - case SNDRV_TIMER_IOCTL_GPARAMS: case SNDRV_TIMER_IOCTL_GSTATUS: case SNDRV_TIMER_IOCTL_SELECT: case SNDRV_TIMER_IOCTL_PARAMS: @@ -128,6 +158,8 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PAUSE_OLD: case SNDRV_TIMER_IOCTL_NEXT_DEVICE: return snd_timer_user_ioctl(file, cmd, (unsigned long)argp); + case SNDRV_TIMER_IOCTL_GPARAMS32: + return snd_timer_user_gparams_compat(file, argp); case SNDRV_TIMER_IOCTL_INFO32: return snd_timer_user_info_compat(file, argp); case SNDRV_TIMER_IOCTL_STATUS32: -- 2.7.3

2 2

[alsa-devel] [PATCH 0/3] ARM: OMAP3: Fix McBSP2/3 hwmod setup for sidetone
by Peter Ujfalusi 22 Mar '16

22 Mar '16

Hi, The series addresses a long standing issue with McBSP2/3 regarding to hwmod setup. When booting with DT a warning is printed that mcbsp2/3 is using two hwmod. The root of the issue is the way how the hwmod data was constructed in the first place for OMAP3 McBSP2/3. After re-reading the TRM it is clear that the sidetone should not have it's own hwmod data as it is not a separate IP, it is part of the McBSP module. It can not affect PRCM either since it's SYSCONFIG register's AUTOIDLE bit is only sets the autoidle from the internal McBSP_iclk clock to the sidetone block of the same McBSP. The first two patch will remove the omap2/3_sidetone hwmod and it's use from DT. The ASoC patch is clear the autoidle bit in McBSP sidetone as it is adviced in the TRM. Mark: The ASoC patch can be safely merged via ASoC tree. Regards, Peter --- Peter Ujfalusi (3): ARM: DTS: omap3: Remove mcbsp2/3_sidetone hwmod reference for McBSP2/3 ARM: OMAP3: hwmod data: Merge and remove the McBSP sidetone related data ASoC: omap-mcbsp: Enable/disable sidetone block auto clock gating for omap3 arch/arm/boot/dts/omap3.dtsi | 4 +- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 120 ++++------------------------- sound/soc/omap/mcbsp.c | 8 ++ 3 files changed, 24 insertions(+), 108 deletions(-) -- 2.7.3

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[alsa-devel] [PATCH] sound/usb: fix to release stream resources from media_snd_device_delete()
by Shuah Khan 22 Mar '16

22 Mar '16

Fix to release stream resources from media_snd_device_delete() before media device is unregistered. Without this change, stream resource free is attempted after the media device is unregistered which would result in use-after-free errors. Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- - Ran bind/unbind loop (1000 iteration) test on snd-usb-audio while running mc_nextgen_test loop (1000 iterations) in parallel. - Ran bind/unbind and rmmod/modprobe tests on both drivers. Also generated graphs when after bind/unbind, rmmod/modprobe. Graphs look good. - Note: Please apply the following patch to fix memory leak: sound/usb: Fix memory leak in media_snd_stream_delete() during unbind https://lkml.org/lkml/2016/3/16/1050 sound/usb/media.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/media.c b/sound/usb/media.c index de4a815..e35af88 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -301,6 +301,13 @@ int media_snd_device_create(struct snd_usb_audio *chip, void media_snd_device_delete(struct snd_usb_audio *chip) { struct media_device *mdev = chip->media_dev; + struct snd_usb_stream *stream; + + /* release resources */ + list_for_each_entry(stream, &chip->pcm_list, list) { + media_snd_stream_delete(&stream->substream[0]); + media_snd_stream_delete(&stream->substream[1]); + } media_snd_mixer_delete(chip); -- 2.5.0

2 9

[alsa-devel] [PATCH V2 0/6] alsabat: clean structure and tinyalsa support
by han.lu＠intel.com 22 Mar '16

22 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> 1. refactoring alsa process to make structure simpler and cleaner, use general functions and unified return value for convenience of maintaining; 2. add tinyalsa support, user can choose tinyalsa with "configure --enable-tinyalsa" option. By default ALSA lib is used. The purpose is for alsabat to run on tinyalsa platforms such as Android and some IoT devices. Lu, Han (6): alsabat: refactoring alsa capture thread alsabat: use general function for wav header update alsabat: clean return value for playback and capture threads alsabat: use general data generator function alsabat: move alsa process to a single block alsabat: add tinyalsa support bat/Makefile.am | 12 +- bat/alsa.c | 217 +++++++++++--------------- bat/alsabat.1 | 3 + bat/bat.c | 58 +++---- bat/common.c | 68 +++++++++ bat/common.h | 17 ++- bat/signal.c | 13 +- bat/tinyalsa.c | 460 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/tinyalsa.h | 20 +++ configure.ac | 17 +++ 10 files changed, 711 insertions(+), 174 deletions(-) create mode 100644 bat/tinyalsa.c create mode 100644 bat/tinyalsa.h -- 2.5.0

2 7

[alsa-devel] [PATCH 0/6] *** alsabat: clean structure and tinyalsa support ***
by han.lu＠intel.com 22 Mar '16

22 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> 1. Clean structure, use general function to replace local processes. 2. Add tinyalsa support, configure will read environment and decide to link to ALSA lib or tinyalsa lib. Lu, Han (6): alsabat: clean file process on capture thread loop alsabat: use common wav process function in playback loop alsabat: clean return value of playback and capture loops alsabat: use common data generator function alsabat: move alsa process to alsa.c alsabat: add tinyalsa support bat/Makefile.am | 12 +- bat/alsa.c | 217 +++++++++++--------------- bat/alsabat.1 | 3 + bat/bat.c | 58 +++---- bat/common.c | 68 +++++++++ bat/common.h | 17 ++- bat/signal.c | 13 +- bat/tinyalsa.c | 460 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/tinyalsa.h | 20 +++ configure.ac | 4 + 10 files changed, 698 insertions(+), 174 deletions(-) create mode 100644 bat/tinyalsa.c create mode 100644 bat/tinyalsa.h -- 2.5.0

3 10

[alsa-devel] [PATCH] ALSA: timer: fix another ioctl compatibility bug
by Takashi Sakamoto 22 Mar '16

22 Mar '16

Hi, In ALSA timer interface, I found another compatibility bug to different ABIs. This patch is to fix it. I note that the bug is in really, really-minor functionality and no drivers use it, therefore it has little impact to this subsystem. This patch is just for manners between kernel/userspace interfaces. To check the compatibility, I wrote a series of easy and small programs for each ALSA interface. It's available in my repository on github. https://github.com/takaswie/alsa-ioctl-test As long as I tested with these programs, this patch is the last fix for the compatibility issues which comes from differences of data model. On the other hand, I confirmed that each compatibility layer is not fully compatible to core implementation. For example, ioctl() with SNDRV_PCM_IOCTL_WRITEI_FRAMES and argument filled by zero for PCM capture character device (not for playback device) returns different error codes for each of x86/x86_64 binary. I think this should be fixed, but don't start to work for it yet. Regards Takashi Sakamoto (1): ALSA: timer: fix ioctl compatibility for different data models sound/core/timer_compat.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) -- 2.7.3

2 3

[alsa-devel] [PATCH 0/7] Intel HD-audio HDMI codec driver rewrites
by Takashi Iwai 22 Mar '16

22 Mar '16

Hi, here is a patch series for rewrite / cleanup of Intel HD-audio codec driver. The main purpose of this patchset is to re-add the support of i915 ELD notification for old chipsets. For that, it starts from the cleanup of the HDMI codec driver code to get rid of excessive usages of is_*() macros. Then the fixes for supporting the old chips in HDA core side and HDMI codec driver side. Although this could be included for 4.6, my current plan is to postpone this for 4.7. Takashi === Takashi Iwai (7): ALSA: hda - Split out Intel-specific codes from patch_generic_hdmi() ALSA: hda - Apply AMP fix in hdmi_setup_audio_infoframe() generically ALSA: hda - Override HDMI setup_stream ops for Intel HSW+ ALSA: hda - Introduce pin_cvt_fixup() ops to hdmi parser ALSA: hda - Use eld notifier for Intel SandyBridge and IvyBridge HDMI/DP ALSA: hda - Add the pin / port mapping on Intel ILK and VLV ALSA: hda - Enable i915 ELD notifier for Intel IronLake and Baytrail include/sound/hda_i915.h | 10 +- sound/hda/hdac_i915.c | 45 ++++-- sound/pci/hda/patch_hdmi.c | 373 ++++++++++++++++++++++++++++++--------------- 3 files changed, 293 insertions(+), 135 deletions(-) -- 2.7.4

1 7

[alsa-devel] [PATCH 0/3] *** Some trival fixes on alsabat ***
by han.lu＠intel.com 21 Mar '16

21 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Fix misusing of errno, fopen, prints and return value Lu, Han (3): alsabat: fix misusing of errno alsabat: fix fopen and messages alsabat: use variable for thread return value bat/alsa.c | 50 ++++++++++++++++++++++++++------------------------ bat/analyze.c | 6 +++--- bat/bat.c | 39 ++++++++++++++++++++++----------------- bat/common.c | 1 - 4 files changed, 51 insertions(+), 45 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 21 Mar '16

21 Mar '16

Driver for TI's TAS5720L/M digital audio amplifiers. The driver should be pretty standard except the optional interrupt-based fault reporting. Some background on the fault reporting since that might be a discussion of interest. The code should have that documented rather well but I wanted to bring it up right away. The TAS5720 devices can report conditions such as over-current (short), over temp, and others that can be important to know in a real-world system. However the device lacks a way to mask the generation of fault interrupts except by shutting it down, which can lead to cases where the device generates SAIF-fault host interrupts every 300us! This is clearly a lot of overhead to be thrown at the SoC (and my bench testing has demonstrated that) so after trying different things to tame this behavior I ended up implementing method to disable/enable the interrupts in the driver itself in combination with rate-limiting the processing of those events. The fact of processing interrupts in a threaded fashion (IRQF_ONESHOT) alone was not enough under certain conditions to prevent the SoC from not being able to get to all interrupts even with an empty threaded handler, and it would freeze and eventually the Kernel would disable the interrupt ("no one cared"). The scheme currently implemented seems to work reliably cycling different audio files over hours also forcing error conditions. But if anybody has a "cleaner" idea how to rate-limit IRQs on a very low level (currently done in the IRQ handler function) I'm absolutely open to suggestions. This all may not be an issue at all when using a faster/lower-latency SoC than what I'm using (BeagleBone Black) but the fact that IRQF_ONESHOT alone was not enough to prevent the system to be brought to its knees (again, even with an empty threaded handler) was a bit of a surprise to me. -- Andreas Dannenberg Texas Instruments Inc Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 37 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 733 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 872 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

2 4

[alsa-devel] New snd-hda warning spew
by Ville Syrjälä 21 Mar '16

21 Mar '16

We have a few new WARN spews from snd-hda causing some grief in i915 CI. This one happens on ILK and BYT. Looks like it happens 100% of the time on driver load: [ 18.809850] ------------[ cut here ]------------ [ 18.809866] WARNING: CPU: 0 PID: 39 at sound/hda/hdac_i915.c:129 pin2port+0x25/0x30 [snd_hda_core]() [ 18.809869] Modules linked in: ax88179_178a usbnet mii intel_powerclamp coretemp crct10dif_pclmul i915 snd_hda_codec_hdmi snd_hda_codec_generic crc32_pclmul ghash_clmulni_intel snd_hda_intel snd_hda_codec snd_hwdep snd_hda_core snd_pcm mei_me mei lpc_ich sdhci_pci sdhci mmc_core e1000e ptp pps_core [ 18.809903] CPU: 0 PID: 39 Comm: kworker/0:1 Not tainted 4.5.0-gfxbench+ #1 [ 18.809905] Hardware name: Hewlett-Packard HP EliteBook 8440p/172A, BIOS 68CCU Ver. F.24 09/13/2013 [ 18.809912] Workqueue: events azx_probe_work [snd_hda_intel] [ 18.809915] 0000000000000000 ffff8800b646b9e8 ffffffff813fef15 0000000000000000 [ 18.809920] ffffffffa00f277c ffff8800b646ba20 ffffffff81078a21 ffff8800b2436000 [ 18.809924] ffff8800b24755e8 ffff8800b24755f0 0000000000000100 ffff8800aed08330 [ 18.809929] Call Trace: [ 18.809936] [<ffffffff813fef15>] dump_stack+0x67/0x92 [ 18.809942] [<ffffffff81078a21>] warn_slowpath_common+0x81/0xc0 [ 18.809945] [<ffffffff81078b15>] warn_slowpath_null+0x15/0x20 [ 18.809953] [<ffffffffa00f12d5>] pin2port+0x25/0x30 [snd_hda_core] [ 18.809960] [<ffffffffa00f1378>] snd_hdac_acomp_get_eld+0x38/0x70 [snd_hda_core] [ 18.809966] [<ffffffffa017da52>] hdmi_present_sense+0x92/0x370 [snd_hda_codec_hdmi] [ 18.809971] [<ffffffffa017e0ff>] generic_hdmi_build_controls+0x11f/0x200 [snd_hda_codec_hdmi] [ 18.809985] [<ffffffffa01105b1>] snd_hda_codec_build_controls+0x191/0x1d0 [snd_hda_codec] [ 18.809993] [<ffffffffa0110881>] ? snd_hda_codec_build_pcms+0xe1/0x1a0 [snd_hda_codec] [ 18.809999] [<ffffffffa010b442>] hda_codec_driver_probe+0x82/0x100 [snd_hda_codec] [ 18.810005] [<ffffffff8153d729>] driver_probe_device+0x229/0x450 [ 18.810008] [<ffffffff8153da7d>] __device_attach_driver+0x6d/0x80 [ 18.810011] [<ffffffff8153da10>] ? driver_allows_async_probing+0x30/0x30 [ 18.810015] [<ffffffff8153b4b8>] bus_for_each_drv+0x58/0x90 [ 18.810018] [<ffffffff8153d3c8>] __device_attach+0xb8/0x130 [ 18.810022] [<ffffffff8153dc0e>] device_initial_probe+0xe/0x10 [ 18.810025] [<ffffffff8153c76e>] bus_probe_device+0x9e/0xb0 [ 18.810028] [<ffffffff8153a5a1>] device_add+0x421/0x5f0 [ 18.810035] [<ffffffff817c2599>] ? mutex_unlock+0x9/0x10 [ 18.810041] [<ffffffffa00eba7f>] snd_hdac_device_register+0xf/0x40 [snd_hda_core] [ 18.810048] [<ffffffffa010b1f6>] snd_hda_codec_configure+0x36/0x180 [snd_hda_codec] [ 18.810057] [<ffffffffa01150c3>] azx_codec_configure+0x23/0x40 [snd_hda_codec] [ 18.810062] [<ffffffffa0134707>] azx_probe_work+0x567/0x870 [snd_hda_intel] [ 18.810067] [<ffffffff81093c8b>] process_one_work+0x1cb/0x680 [ 18.810070] [<ffffffff81093c06>] ? process_one_work+0x146/0x680 [ 18.810073] [<ffffffff81094189>] worker_thread+0x49/0x490 [ 18.810076] [<ffffffff81094140>] ? process_one_work+0x680/0x680 [ 18.810081] [<ffffffff8109a87a>] kthread+0xea/0x100 [ 18.810085] [<ffffffff817c5407>] ? _raw_spin_unlock_irq+0x27/0x50 [ 18.810090] [<ffffffff8109a790>] ? kthread_create_on_node+0x1f0/0x1f0 [ 18.810093] [<ffffffff817c60bf>] ret_from_fork+0x3f/0x70 [ 18.810097] [<ffffffff8109a790>] ? kthread_create_on_node+0x1f0/0x1f0 [ 18.810277] ---[ end trace 2d4983bcfed2cd4b ]--- This other one was seen at least on on SKL: [ 124.808525] ------------[ cut here ]------------ [ 124.808545] WARNING: CPU: 3 PID: 173 at sound/hda/hdac_i915.c:91 snd_hdac_display_power+0xf1/0x110 [snd_hda_core]() [ 124.808551] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic x86_pkg_temp_thermal intel_powerclamp coretemp i915 crct10dif_pclmul snd_hda_intel crc32_pclmul snd_hda_codec ghash_clmulni_intel snd_hwdep snd_hda_core snd_pcm mei_me mei i2c_hid sdhci_pci e1000e ptp sdhci pps_core mmc_core [ 124.808603] CPU: 3 PID: 173 Comm: kworker/3:2 Tainted: G W 4.5.0-gfxbench+ #1 [ 124.808608] Hardware name: /NUC6i5SYB, BIOS SYSKLi35.86A.0028.2015.1112.1822 11/12/2015 [ 124.808616] Workqueue: pm pm_runtime_work [ 124.808639] 0000000000000000 ffff88026423fbb8 ffffffff813fef15 0000000000000000 [ 124.808644] ffffffffa00f877c ffff88026423fbf0 ffffffff81078a21 0000000000000000 [ 124.808648] ffff880085a11148 ffff880085586ab8 ffff8800861b8000 0000000000000000 [ 124.808652] Call Trace: [ 124.808656] [<ffffffff813fef15>] dump_stack+0x67/0x92 [ 124.808660] [<ffffffff81078a21>] warn_slowpath_common+0x81/0xc0 [ 124.808662] [<ffffffff81078b15>] warn_slowpath_null+0x15/0x20 [ 124.808668] [<ffffffffa00f77e1>] snd_hdac_display_power+0xf1/0x110 [snd_hda_core] [ 124.808673] [<ffffffffa015039d>] azx_intel_link_power+0xd/0x10 [snd_hda_intel] [ 124.808680] [<ffffffffa011e32a>] azx_link_power+0x1a/0x30 [snd_hda_codec] [ 124.808684] [<ffffffffa00f21f9>] snd_hdac_link_power+0x29/0x40 [snd_hda_core] [ 124.808690] [<ffffffffa01192a6>] hda_codec_runtime_suspend+0x76/0xa0 [snd_hda_codec] [ 124.808697] [<ffffffffa0119230>] ? hda_codec_runtime_resume+0x50/0x50 [snd_hda_codec] [ 124.808701] [<ffffffff8154719d>] __rpm_callback+0x2d/0x70 [ 124.808704] [<ffffffff815471ff>] rpm_callback+0x1f/0x80 [ 124.808710] [<ffffffffa0119230>] ? hda_codec_runtime_resume+0x50/0x50 [snd_hda_codec] [ 124.808713] [<ffffffff81547764>] rpm_suspend+0x134/0x7f0 [ 124.808716] [<ffffffff810c80a9>] ? __lock_is_held+0x49/0x70 [ 124.808719] [<ffffffff81548129>] rpm_idle+0x209/0x530 [ 124.808722] [<ffffffff815494f9>] pm_runtime_work+0xa9/0xc0 [ 124.808725] [<ffffffff81093c8b>] process_one_work+0x1cb/0x680 [ 124.808728] [<ffffffff81093c06>] ? process_one_work+0x146/0x680 [ 124.808731] [<ffffffff81094189>] worker_thread+0x49/0x490 [ 124.808733] [<ffffffff81094140>] ? process_one_work+0x680/0x680 [ 124.808735] [<ffffffff81094140>] ? process_one_work+0x680/0x680 [ 124.808738] [<ffffffff8109a87a>] kthread+0xea/0x100 [ 124.808742] [<ffffffff817c53d7>] ? _raw_spin_unlock_irq+0x27/0x50 [ 124.808745] [<ffffffff8109a790>] ? kthread_create_on_node+0x1f0/0x1f0 [ 124.808749] [<ffffffff817c607f>] ret_from_fork+0x3f/0x70 [ 124.808752] [<ffffffff8109a790>] ? kthread_create_on_node+0x1f0/0x1f0 [ 124.808755] ---[ end trace 2b97e417eb08d213 ]--- -- Ville Syrjälä Intel OTC

4 23

[alsa-devel] amidi.c.patch review request
by Martin Tarenskeen 21 Mar '16

21 Mar '16

Hi, I improved my own patch to amidi.c As a result I can now succesfully fetch a SysEx dump from my Yamaha Reface DX, and it exits succesfully after the timeout that I specify using the -t option. Problem: My synth sends continuously not only 0xFE (Active Sensing) but also 0xF8 (Timing Clock). 24 times per quarter note to be exactly. With a fast tempo like 250 BPM this gives a 0xF8 byte every 10 ms. Until now when I tried to fetch a SysEx dump from my synth using amidi the data were polluted with lots of 0xF8 bytes and the -t option would not work. A bug I would say. The attached patch seems to solve this issue, but please comment and improve where needed. Please try with other synths also. Can someone submit my patch to the official sources? I don't know how to. See attached. -- Martin

2 1

Re: [alsa-devel] [Alsa-devel] [PATCH] amidi active sensing option
by Clemens Ladisch 21 Mar '16

21 Mar '16

Martin Tarenskeen wrote: > I searched on Google and found in the old alsa-devel archives the > anouncement of this option to enable/disable detecting Active Sensing > (0xfe) bytes with the amidi utility. > But not only 0xfe (Active Sensing) should be blocked by default, also > other Midi System Realtime messages. In other SysEx dump utilities > that I know there is a simialr option called something like > "rt-filter" or "realtime-filter". > [...] > I have attached my patch. Thank you for this improvement. However, the name of the option ("--active-sensing") is no longer correct, and the man page must be updated, too. Also, patches need a Signed-off-by line to indicate that the submitter accepts the DCO; see <http://elinux.org/Developer_Certificate_Of_Origin>. Regards, Clemens

1 0

[alsa-devel] [PATCH] sound: usb: add Microsoft HD-5001 to quirks
by Victor Clément 20 Mar '16

20 Mar '16

The Microsoft HD-5001 webcam microphone does not support sample rate reading as the HD-5000 one. This results in dmesg errors and sound hanging with pulseaudio. Signed-off-by: Victor Clément <victor.clement(a)openmailbox.org> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4f6ce1c..f27016f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1121,6 +1121,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) switch (chip->usb_id) { case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ + case USB_ID(0x045E, 0x076E): /* MS Lifecam HD-5001 */ case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ -- 2.7.2

2 1

[alsa-devel] [GIT PULL]: Update sst audio firmware for Skylake
by Vinod Koul 19 Mar '16

19 Mar '16

Hello Linux firmware maintainers, Please PULL to receive the updated version of Skylake audio firmware version 896. This contains some bug fixes and wov module fixes. The following changes since commit 9be9ff282635c25694819c39aae0f6cc6c251c12: iwlwifi: add -21.ucode to WHENCE (2016-03-10 11:28:28 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to 780f4eed8ff971bd4d6c41ecded4c99135cd36f9: linux-firmware: intel: Update Skylake audio firmware (2016-03-14 11:38:19 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v896.bin | Bin 0 -> 233472 bytes 2 files changed, 1 insertion(+), 1 deletion(-) create mode 100644 intel/dsp_fw_release_v896.bin Thanks -- ~Vinod

2 1

[alsa-devel] USB Sound oops in Linus's latest tree
by Greg KH 19 Mar '16

19 Mar '16

Hi, I'm getting the following oops with my USB sound device using Linus's latest tree of the moment, which has the sound tree pull in it. Anyone seen this before? thanks, greg k-h [ +0.002298] input: Schiit Audio USB Modi Device as /devices/pci0000:00/0000:00:14.0/usb2/2-10/2-10.1/2-10.1:1.2/0003:0D8C:1319.0001/input/input17 [ +0.000056] hid-generic 0003:0D8C:1319.0001: input,hidraw0: USB HID v1.00 Device [Schiit Audio USB Modi Device] on usb-0000:00:14.0-10.1/input2 [ +0.008749] BUG: unable to handle kernel NULL pointer dereference at 0000000000000014 [ +0.001151] IP: [<ffffffffc0e79921>] usb_audio_probe+0x1e1/0x8f0 [snd_usb_audio] [ +0.001142] PGD 0 [ +0.001114] Oops: 0000 [#1] PREEMPT SMP [ +0.001115] Modules linked in: snd_usb_audio(+) hid_generic snd_usbmidi_lib usbhid snd_rawmidi snd_seq_device hid media arc4 fuse btusb btrtl btbcm btintel bluetooth rt2800usb rt2x00usb rt2800lib rt2x00lib mac80211 crc_ccitt cfg80211 rfkill nls_cp437 vfat fat iTCO_wdt iTCO_vendor_support mxm_wmi x86_pkg_temp_thermal intel_powerclamp coretemp kvm_intel kvm irqbypass crct10dif_pclmul crct10dif_common crc32_pclmul crc32c_intel ghash_clmulni_intel aesni_intel aes_x86_64 glue_helper lrw gf128mul ablk_helper cryptd psmouse serio_raw pcspkr firewire_ohci firewire_core crc_itu_t xhci_pci ehci_pci xhci_hcd ehci_hcd mei_me usbcore mei lpc_ich mfd_core usb_common wmi evdev ip_tables x_tables [ +0.004079] CPU: 4 PID: 615 Comm: systemd-udevd Not tainted 4.6.0-rc1+ #103 [ +0.001382] Hardware name: / , BIOS KLZ8711D.86A.0457.2015.0713.1319 07/13/2015 [ +0.001387] task: ffff880098ec0e00 ti: ffff880251920000 task.ti: ffff880251920000 [ +0.001417] RIP: 0010:[<ffffffffc0e79921>] [<ffffffffc0e79921>] usb_audio_probe+0x1e1/0x8f0 [snd_usb_audio] [ +0.001414] RSP: 0018:ffff880251923af0 EFLAGS: 00010246 [ +0.001409] RAX: 0000000000000000 RBX: ffff88024d602000 RCX: 0000000000000000 [ +0.001404] RDX: ffff880254e0b340 RSI: ffff8802559b99f8 RDI: ffffffff82181030 [ +0.001413] RBP: ffff880251923b50 R08: 0000000000019d00 R09: ffffffff81385837 [ +0.001431] R10: ffffea0009461200 R11: ffff880253b3fe72 R12: 0000000000000000 [ +0.001411] R13: 0000000000000000 R14: 0000000000000001 R15: ffff88025338c212 [ +0.001436] FS: 00007f7a8db567c0(0000) GS:ffff88025fb00000(0000) knlGS:0000000000000000 [ +0.001431] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ +0.001445] CR2: 0000000000000014 CR3: 0000000251854000 CR4: 00000000001406e0 [ +0.001451] Stack: [ +0.001433] ffff880253302e88 ffff88024d602054 ffff88009879d000 ffff880253302e88 [ +0.001480] ffff880255f7e000 386430425355d030 ff00393133313a63 ffff8800984f4098 [ +0.001466] ffff8800984f4000 ffffffffc0e95728 ffff88009879d030 ffffffffc0e955a0 [ +0.001497] Call Trace: [ +0.001295] [<ffffffffc0067cfd>] usb_probe_interface+0x1bd/0x300 [usbcore] [ +0.001427] [<ffffffff815987e9>] driver_probe_device+0x249/0x450 [ +0.001400] [<ffffffff81598a74>] __driver_attach+0x84/0x90 [ +0.001398] [<ffffffff815989f0>] ? driver_probe_device+0x450/0x450 [ +0.001417] [<ffffffff815963e4>] bus_for_each_dev+0x64/0xa0 [ +0.001369] [<ffffffff81597ede>] driver_attach+0x1e/0x20 [ +0.001396] [<ffffffff815979ab>] bus_add_driver+0x1eb/0x280 [ +0.001365] [<ffffffff81599390>] driver_register+0x60/0xe0 [ +0.001369] [<ffffffffc0066834>] usb_register_driver+0x84/0x140 [usbcore] [ +0.001422] [<ffffffffc0e9d000>] ? 0xffffffffc0e9d000 [ +0.001370] [<ffffffffc0e9d01e>] usb_audio_driver_init+0x1e/0x20 [snd_usb_audio] [ +0.001472] [<ffffffff8100211b>] do_one_initcall+0xab/0x1d0 [ +0.001547] [<ffffffff8115e732>] do_init_module+0x5f/0x1cd [ +0.001469] [<ffffffff810f9ec8>] load_module+0x1368/0x1510 [ +0.001542] [<ffffffff810f75d0>] ? symbol_put_addr+0x50/0x50 [ +0.001468] [<ffffffff810fa2a8>] SyS_finit_module+0xc8/0xf0 [ +0.001451] [<ffffffff81003c6e>] do_syscall_64+0x5e/0xc0 [ +0.001538] [<ffffffff817fb425>] entry_SYSCALL64_slow_path+0x25/0x25 [ +0.001397] Code: f8 01 00 44 89 e6 48 89 df e8 dc 73 00 00 85 c0 89 c1 0f 88 99 00 00 00 48 8b 7b 10 e8 c9 0b 80 c0 85 c0 89 c1 0f 88 86 00 00 00 <41> 80 7d 14 00 0f 85 90 04 00 00 48 63 03 48 89 1c c5 40 90 e9 [ +0.001513] RIP [<ffffffffc0e79921>] usb_audio_probe+0x1e1/0x8f0 [snd_usb_audio] [ +0.001468] RSP <ffff880251923af0> [ +0.001576] CR2: 0000000000000014 [ +0.010610] ---[ end trace c9cdd61401b8f1d9 ]---

3 4

Re: [alsa-devel] snd_hda_core oops with Linux v4.5-6229-gf7813ad5cbfd
by Takashi Iwai 18 Mar '16

18 Mar '16

On Fri, 18 Mar 2016 21:59:51 +0100, Josh Boyer wrote: > > Hi, > > I have a celeron based NUC that is giving me the following backtraces > after Linus merged the sound tree today. This is with the top of the > tree at f7813ad5cbfd as mentioned in the subject. > > The machine has an HDMI port, but it is not connected to any thing. It should have been fixed now on my sound.git tree. Try to pull for-linus branch. I'll send another pull request in early next week. thanks, Takashi

1 0

[alsa-devel] [RFC PATCH 0/3] ASoC: pcm1792a: Add support for pcm1795 and pcm1796
by Michael Trimarchi 18 Mar '16

18 Mar '16

Those patches add support for pcm1795 and pcm1796. Both are compatible with the pcm1792a with a minor change for the pcm1795 Michael Trimarchi (3): ASoC: pcm1792a: Rename internal data and function to pcm179x ASoC: pcm1792a: Rename pcm1792a to pcm179x ASoC: pcm179x: Add support for pcm1795 and pcm1796 .../devicetree/bindings/sound/pcm1792a.txt | 18 -- .../devicetree/bindings/sound/pcm179x.txt | 18 ++ sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/pcm1792a.c | 271 ------------------ sound/soc/codecs/pcm1792a.h | 27 -- sound/soc/codecs/pcm179x.c | 314 +++++++++++++++++++++ sound/soc/codecs/pcm179x.h | 31 ++ 8 files changed, 368 insertions(+), 321 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm179x.txt delete mode 100644 sound/soc/codecs/pcm1792a.c delete mode 100644 sound/soc/codecs/pcm1792a.h create mode 100644 sound/soc/codecs/pcm179x.c create mode 100644 sound/soc/codecs/pcm179x.h -- 2.7.0

4 21

[alsa-devel] Playing mono files on a stereo only device
by Champ, Andy 18 Mar '16

18 Mar '16

Hi All, I'm new to this area, so please forgive any too-obvious questions. I'm working on a device driver for an SOC. The audio device is connected via an I2S bus, which is inherently stereo. The driver code that I've inherited (from a similar, but not the same, chip) claimed to support mono - but didn't. I've changed the configuration in the device's snd_soc_dai_driver structure to stereo only (channels_min=channels_max=2) and everything works fine when playing a stereo file. AIUI an Alsa plug should be able to take a mono file, convert it on the fly to stereo, and play that out. I've spent a couple of days playing with it, and not really got anywhere. Does anyone have an example where this is being done on an SOC? Currently the error I am having indicates that it's a problem with access. An IOCTL is being sent by the ALSA code to the kernel, of type SNDRV_PCM_IOCTL_HW_REFINE, requesting an access. This fails. I've not even been able to decide if an access is meant to be a mask or a value. The type is typedef int __bitwise snd_pcm_access_t; which would tend to say it's mask (bitwise?) though I am guessing here, but the values are 0 1 2 3 4 - which says it isn't. Just to really confuse me the mask is logged as 7. Is there anything which explains what this snd_pcm_access_t is? Thanks

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix kbuild dependency
by Vinod Koul 18 Mar '16

18 Mar '16

The Skylake driver selects SND_HDA_I915 which causes kbuild to spew warning: warning: (SND_SOC_INTEL_SKYLAKE) selects SND_HDA_I915 which has unmet direct dependencies (SOUND && !M68K && !UML && SND && DRM_I915 && SND_HDA_CORE) The SND_HDA_I915 should not be selected so drop that. Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b3e6c2300457..1120f4f4d011 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -163,7 +163,6 @@ config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE select SND_SOC_TOPOLOGY - select SND_HDA_I915 select SND_SOC_INTEL_SST config SND_SOC_INTEL_SKL_RT286_MACH -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix forgotten HDMI monitor_present update
by Takashi Iwai 18 Mar '16

18 Mar '16

We forgot to copy monitor_present value when updating the ELD information. This won't change the ELD retrieval and the jack notification behavior, but appears only in the proc output. In that sense, it's no fatal error, but a bug is a bug is a bug. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8cdb804aa9cf..e7d9453ecd10 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1358,6 +1358,7 @@ static void update_eld(struct hda_codec *codec, eld->eld_size) != 0) eld_changed = true; + pin_eld->monitor_present = eld->monitor_present; pin_eld->eld_valid = eld->eld_valid; pin_eld->eld_size = eld->eld_size; if (eld->eld_valid) -- 2.7.4

1 0

[alsa-devel] [PATCH 5/5] dt/bindings: sound: new chip added into TAS571x binding
by Petr Kulhavy 18 Mar '16

18 Mar '16

This adds the TAS5721 into the TAS571x binding. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- Documentation/devicetree/bindings/sound/tas571x.txt | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt index 0ac31d8..f1aade3 100644 --- a/Documentation/devicetree/bindings/sound/tas571x.txt +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -1,4 +1,4 @@ -Texas Instruments TAS5711/TAS5717/TAS5719 stereo power amplifiers +Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 stereo power amplifiers The codec is controlled through an I2C interface. It also has two other signals that can be wired up to GPIOs: reset (strongly recommended), and @@ -6,7 +6,7 @@ powerdown (optional). Required properties: -- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719" +- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719", "ti,tas5721" - reg: The I2C address of the device - #sound-dai-cells: must be equal to 0 @@ -25,6 +25,8 @@ Optional properties: - PVDD_B-supply: regulator phandle for the PVDD_B supply (5711) - PVDD_C-supply: regulator phandle for the PVDD_C supply (5711) - PVDD_D-supply: regulator phandle for the PVDD_D supply (5711) +- DRVDD-supply: regulator phandle for the DRVDD supply (5721) +- PVDD-supply: regulator phandle for the PVDD supply (5721) Example: -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] sst: select DW_DMAC_CORE since it's mandatory
by Andy Shevchenko 18 Mar '16

18 Mar '16

While here group selectors under SND_SOC_INTEL_HASWELL and SND_SOC_INTEL_BAYTRAIL. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/Kconfig | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 05fde5e6e..bc5281e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -35,15 +35,17 @@ config SND_SOC_INTEL_SST_ACPI config SND_SOC_INTEL_HASWELL tristate + select DW_DMAC_CORE + select SND_SOC_INTEL_SST config SND_SOC_INTEL_BAYTRAIL tristate + select DW_DMAC_CORE + select SND_SOC_INTEL_SST config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -55,8 +57,6 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help @@ -66,8 +66,6 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help @@ -76,10 +74,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on X86_INTEL_LPSS && I2C && DW_DMAC && \ - I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST + depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help -- 2.5.1

2 2

[alsa-devel] [PATCH] ASoC: dapm: Make sure we have a card when displaying component widgets
by Mark Brown 18 Mar '16

18 Mar '16

The dummy component is reused for all cards so we special case and don't bind it to any of them. This means that code like that displaying the component widgets that tries to look at the card will crash. In the future we will fix this by ensuring that the dummy component looks like other components but that is invasive and so not suitable for a fix. Instead add a special case check here. Reported-by: Harry Pan <harry.pan(a)intel.com> Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a81dfd..c4464858bf01 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2188,6 +2188,13 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, int count = 0; char *state = "not set"; + /* card won't be set for the dummy component, as a spot fix + * we're checking for that case specifically here but in future + * we will ensure that the dummy component looks like others. + */ + if (!cmpnt->card) + return 0; + list_for_each_entry(w, &cmpnt->card->widgets, list) { if (w->dapm != dapm) continue; -- 2.7.0

1 0

Re: [alsa-devel] [Intel-gfx] ✗ Fi.CI.BAT: failure for drm/i915/bxt: Fix off-by-one error in Broxton PLL IDs (rev3)
by Daniel Vetter 18 Mar '16

18 Mar '16

On Wed, Mar 16, 2016 at 02:37:24PM +0200, Tomi Sarvela wrote: > On Wednesday 16 March 2016 10:48:43 Imre Deak wrote: > > Tomi, noticed two things that maybe infrastructure related, see below: > > > > > Test drv_module_reload_basic: > > > skip -> PASS (bdw-nuci7) > > > Test gem_ringfill: > > > Subgroup basic-default-s3: > > > pass -> DMESG-WARN (skl-nuci5) > > > Test gem_storedw_loop: > > > Subgroup basic-bsd: > > > pass -> DMESG-WARN (skl-nuci5) > > > Subgroup basic-bsd1: > > > pass -> DMESG-WARN (skl-nuci5) > > > > Unrelated platform. All of the above are SND resume failures. Ville > > said he contacted Takashi about them. > > SND is recurring problem. I'd like to see drm-intel-nightly not breaking when > pulling sound updates. We need to escalate this again, snd team seriously sucks at quality control :( Adding Libin&Takashi. If this doesn't improve I need to throw sound trees out of -nightly for real. -Daniel -- Daniel Vetter Software Engineer, Intel Corporation http://blog.ffwll.ch

4 9

[alsa-devel] Bad rever-select of CONFIG_SND_HDA_I915
by Takashi Iwai 18 Mar '16

18 Mar '16

Hi, while build testing, I noticed the following warning: warning: (SND_SOC_INTEL_SKYLAKE) selects SND_HDA_I915 which has unmet direct dependencies (SOUND && !M68K && !UML && SND && DRM_I915 && SND_HDA_CORE) And it's indeed broken: you can't select CONFIG_SND_HDA_I915 from the driver side. This is the condition depending on DRM i915 driver. Fixing this doesn't look too trivial, though. Since SND_SOC_INTEL_SKYLAKE is the item to be selected, it can't have "depends on". So, the top-level configs that select SND_SOC_INTEL_SKYLAKE need to have "depends on DRM_I915" lines, or modify the driver to be built without i915 binding. Takashi

2 7

[alsa-devel] [PATCH] ASoC: rt5640: Correct the digital interface data select
by Sugar Zhang 18 Mar '16

18 Mar '16

this patch corrects the interface adc/dac control register definition according to datasheet. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5640.h | 36 ++++++++++++++++++------------------ 2 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 11d032c..48dbb2f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -359,7 +359,7 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, /* Interface data select */ static const char * const rt5640_data_select[] = { - "Normal", "left copy to right", "right copy to left", "Swap"}; + "Normal", "Swap", "left copy to right", "right copy to left"}; static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 83a7150..f84231e 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -442,39 +442,39 @@ #define RT5640_IF1_DAC_SEL_MASK (0x3 << 14) #define RT5640_IF1_DAC_SEL_SFT 14 #define RT5640_IF1_DAC_SEL_NOR (0x0 << 14) -#define RT5640_IF1_DAC_SEL_L2R (0x1 << 14) -#define RT5640_IF1_DAC_SEL_R2L (0x2 << 14) -#define RT5640_IF1_DAC_SEL_SWAP (0x3 << 14) +#define RT5640_IF1_DAC_SEL_SWAP (0x1 << 14) +#define RT5640_IF1_DAC_SEL_L2R (0x2 << 14) +#define RT5640_IF1_DAC_SEL_R2L (0x3 << 14) #define RT5640_IF1_ADC_SEL_MASK (0x3 << 12) #define RT5640_IF1_ADC_SEL_SFT 12 #define RT5640_IF1_ADC_SEL_NOR (0x0 << 12) -#define RT5640_IF1_ADC_SEL_L2R (0x1 << 12) -#define RT5640_IF1_ADC_SEL_R2L (0x2 << 12) -#define RT5640_IF1_ADC_SEL_SWAP (0x3 << 12) +#define RT5640_IF1_ADC_SEL_SWAP (0x1 << 12) +#define RT5640_IF1_ADC_SEL_L2R (0x2 << 12) +#define RT5640_IF1_ADC_SEL_R2L (0x3 << 12) #define RT5640_IF2_DAC_SEL_MASK (0x3 << 10) #define RT5640_IF2_DAC_SEL_SFT 10 #define RT5640_IF2_DAC_SEL_NOR (0x0 << 10) -#define RT5640_IF2_DAC_SEL_L2R (0x1 << 10) -#define RT5640_IF2_DAC_SEL_R2L (0x2 << 10) -#define RT5640_IF2_DAC_SEL_SWAP (0x3 << 10) +#define RT5640_IF2_DAC_SEL_SWAP (0x1 << 10) +#define RT5640_IF2_DAC_SEL_L2R (0x2 << 10) +#define RT5640_IF2_DAC_SEL_R2L (0x3 << 10) #define RT5640_IF2_ADC_SEL_MASK (0x3 << 8) #define RT5640_IF2_ADC_SEL_SFT 8 #define RT5640_IF2_ADC_SEL_NOR (0x0 << 8) -#define RT5640_IF2_ADC_SEL_L2R (0x1 << 8) -#define RT5640_IF2_ADC_SEL_R2L (0x2 << 8) -#define RT5640_IF2_ADC_SEL_SWAP (0x3 << 8) +#define RT5640_IF2_ADC_SEL_SWAP (0x1 << 8) +#define RT5640_IF2_ADC_SEL_L2R (0x2 << 8) +#define RT5640_IF2_ADC_SEL_R2L (0x3 << 8) #define RT5640_IF3_DAC_SEL_MASK (0x3 << 6) #define RT5640_IF3_DAC_SEL_SFT 6 #define RT5640_IF3_DAC_SEL_NOR (0x0 << 6) -#define RT5640_IF3_DAC_SEL_L2R (0x1 << 6) -#define RT5640_IF3_DAC_SEL_R2L (0x2 << 6) -#define RT5640_IF3_DAC_SEL_SWAP (0x3 << 6) +#define RT5640_IF3_DAC_SEL_SWAP (0x1 << 6) +#define RT5640_IF3_DAC_SEL_L2R (0x2 << 6) +#define RT5640_IF3_DAC_SEL_R2L (0x3 << 6) #define RT5640_IF3_ADC_SEL_MASK (0x3 << 4) #define RT5640_IF3_ADC_SEL_SFT 4 #define RT5640_IF3_ADC_SEL_NOR (0x0 << 4) -#define RT5640_IF3_ADC_SEL_L2R (0x1 << 4) -#define RT5640_IF3_ADC_SEL_R2L (0x2 << 4) -#define RT5640_IF3_ADC_SEL_SWAP (0x3 << 4) +#define RT5640_IF3_ADC_SEL_SWAP (0x1 << 4) +#define RT5640_IF3_ADC_SEL_L2R (0x2 << 4) +#define RT5640_IF3_ADC_SEL_R2L (0x3 << 4) /* REC Left Mixer Control 1 (0x3b) */ #define RT5640_G_HP_L_RM_L_MASK (0x7 << 13) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dapm: Do not traverse widget hooks to snd-soc-dummy
by Harry Pan 18 Mar '16

18 Mar '16

All components are initially given an empty card when registering platform, and since the commit 6e78108bda78 ("ASoC: core: Don't probe the component which is dummy")', snd-soc-dummy will not be probed so that it remains an empty card assigned. This patch ignores to iterate widget hooks to the 'snd-soc-dummy' component, else it will trigger memory fault because of invalid dereference address of card->widgets. Test by grep -rsI "hello" /sys/devices/platform/skl_nau88l25_ssm4567_i2s/ Conflicts: sound/soc/soc-dapm.c Signed-off-by: Harry Pan <harry.pan(a)intel.com> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 581175a..0bc15c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2188,6 +2188,9 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, int count = 0; char *state = "not set"; + if (!strcmp(cmpnt->name, "snd-soc-dummy")) + return 0; + list_for_each_entry(w, &cmpnt->card->widgets, list) { if (w->dapm != dapm) continue; -- 2.1.2

5 7

Re: [alsa-devel] [pulseaudio-discuss] ASoC and pulseaudio
by Liam Girdwood 17 Mar '16

17 Mar '16

+ alsa, Takashi and Mark since it's a naming convention being discussed as part of this work. On Mon, 2016-03-14 at 08:17 +0000, Lu, Han wrote: > Hi, > > > -----Original Message----- > > From: pulseaudio-discuss [mailto:pulseaudio-discuss- > > bounces(a)lists.freedesktop.org] On Behalf Of Liam Girdwood > > Sent: Tuesday, February 23, 2016 7:13 PM > > To: General PulseAudio Discussion <pulseaudio- > > discuss(a)lists.freedesktop.org> > > Cc: Linux Upstreaming Team <linux(a)endlessm.com>; han.lu > > <han.lu(a)linux.intel.com> > > Subject: Re: [pulseaudio-discuss] ASoC and pulseaudio > > > > On Sat, 2016-02-20 at 12:22 -0600, Pierre-Louis Bossart wrote: > > > r > > > On 02/17/2016 09:27 AM, Carlo Caione wrote: > > > > Hi, > > > > In our daily work we are seeing more and more laptops coming in with > > > > SoC audio (ASoC). In the most recent case, we are working with a new > > > > consumer laptop based on Intel Cherry Trail. As you probably know to > > > > have audio working on these laptops / codecs a fairly lengthy hw > > > > specific mixer configuration is needed. > > > > The problem we are facing is how to ship a single PA / ALSA package > > > > containing a working configuration for several of those laptops and > > > > we were wondering what is the most correct / upstream-friendly way > > > > to do that. > > > > > > > > I'm already excluding here some trivial workarounds like: shipping a > > > > different PA package for each hw or using an hw-tailored script to > > > > load the correct ALSA configuration. > > > > > > > > The first solution we came up with is just to write a profile-set > > > > configuration file loaded by udev for each cards / laptop with the > > > > correct mixer configuration already embedded in the path > > > > configuration files. This looks a bit like an overkill when you have > > > > to change tens of controls most of which just one time (once you > > > > have setup the codec selecting the main path is usually just matter > > > > of changing a few controls to change the output / input path). > > > > > > > > Another possibility is to use UCM and let PA create automatically > > > > from that profile and path configuration files. This looks promising > > > > but we were a bit worried about the UCM support in PA and how > > > > stable/adopted is UCM itself. > > > > Fwiw, I've not had any problems using PA and UCM together on my BDW + > > RT286 based development platforms. The problem we have atm is matching > > and quirks as Pierre describes. > > > > > > > > > > So, what is the suggested way to accomplish this and in general how > > > > PA is trying to address this problem? I expect in the future to see > > > > many more ASoC coming into the laptops world, how will the community > > > > make it so that you install a distro and then sound "just works"? > > > > > > Not an easy problem. I generated a set of UCM files for baytrail-based > > platforms and I was planning on doing the same for CHT now that there are a > > set of commercially available devices. > > > There are however a set of limitations that are known already, e.g. > > > the UCM file matches the machine driver name, so if there are any quirks > > inside the driver we'd need to add a DMI-based info to the card name and > > also build UCM files based on an include mechanism, otherwise it's going to > > be copy/paste hell Adding Liam to make sure he sees this thread. > > > > > > > Fwiw, we have been looking at providing some fixes for naming and quirks. > > We should probably schedule the work now. > > > > For naming we have the problem of duplication in the driver name e.g. > > > > cat /proc/asound/cards > > 0 [bytrt5640 ]: byt-rt5640 - byt-rt5640 > > byt-rt5640 > > > > The driver name actually has three strings iirc and we could differentiate here > > (by adding DMI name, plus any others?? ) to help UCM and PA deal with any > > quirks. > > > > UCM also does not currently support #include. The intention is to provide a > > method to define mixer settings on a codec per codec basis and the these > > could be #included into a machine UCM file. The machine UCM file would > > also define settings for any quirks and would #include any codec UCM files > > that were needed. PA would load the machine UCM file (based on > > improvements to the driver naming above) intead of the generic UCM file > > that is loaded atm. > > > > Liam > > > > We can fix this from driver level and user space level: > (1) to reporting more information from driver. > ALSA sound cards have 3 names. longname, shortname and drivername. e.g. > from soc-core.c :- > > snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), > "%s", card->name); > snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), > "%s", card->long_name ? card->long_name : card->name); > snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), > "%s", card->driver_name ? card->driver_name : card->name); > > We can additionally set the long_name and driver_name in ASoC if they were not set. > > The naming scheme could be :- > > a) shortname is current machine driver name i.e. "byt-rt5640" > b) longname is DMI boardname + 1 i.e. "Minnowboard Max: byt-rt5640" > c) driver name is platform driver name i.e. "baytrail-pcm-audio" > > (2) to enable UCM to support "include" parsing, so each codec will have its own UCM file to define all the use cases for the codec only, and the codec UCM file can be included and the codec use cases can be referenced by different machine driver UCM files. > > I'll try to work out RFC patches to implement the features above. > > BR, > Han > > > > > > > > _______________________________________________ > > > pulseaudio-discuss mailing list > > > pulseaudio-discuss(a)lists.freedesktop.org > > > https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss > > > > > > _______________________________________________ > > pulseaudio-discuss mailing list > > pulseaudio-discuss(a)lists.freedesktop.org > > https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss > _______________________________________________ > pulseaudio-discuss mailing list > pulseaudio-discuss(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss

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[alsa-devel] [PATCH] ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
by Takashi Sakamoto 17 Mar '16

17 Mar '16

In control compatibility layer, when no elements are found by ELEM_READ/ELEM_WRITE ioctl commands, ENXIO is returned. On the other hand, in core implementation, ENOENT is returned. This is not good for ALSA ctl applications. This commit changes the return value from the compatibility layer so that the same value is returned. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control_compat.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 0608f21..1fa7076 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -196,7 +196,7 @@ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, kctl = snd_ctl_find_id(card, id); if (! kctl) { up_read(&card->controls_rwsem); - return -ENXIO; + return -ENOENT; } info = kzalloc(sizeof(*info), GFP_KERNEL); if (info == NULL) { -- 2.7.0

2 1

[alsa-devel] adau1701: Different sample rates need different firmwares
by Pascal Huerst 17 Mar '16

17 Mar '16

Hey Lars, I'm having an issue with the adau1701 codec. There is this line which checks, if the mclk/lrclk ratio has changed and whether it is necessary to reset the codec and upload a new firmware. http://lxr.free-electrons.com/source/sound/soc/codecs/adau1701.c#L444 But if we change mclk in the machine driver (for example to playback 44.1), this ration might not change, but still we have to upload a new firmware, so shouldn't we probably be checking for sample rate changes instead: if (adau1701->current_rate != params_rate(params) { adau1701->current_rate = params_rate(params); ret = adau1701_reset(codec, clkdiv, params_rate(params)); if (ret < 0) return ret; } Or did I miss something? regards pascal

2 1

Re: [alsa-devel] different return value from some ioctl commands for x86 binary executed on x86_64
by Takashi Iwai 17 Mar '16

17 Mar '16

On Thu, 17 Mar 2016 04:52:52 +0100, Takashi Sakamoto wrote: > > Hi, > > When testing ioctl compatibility layer for ALSA ctl interface, I found > that error code of some ioctl commands are different depending on binary > architecture. > > The commands are SNDRV_CTL_IOCTL_ELEM_READ/SNDRV_CTL_IOCTL_ELEM_WRITE. > When passing data for non-existent element, ENOENT is returned by the > core implementation. On the other hand, ENXIO is returned by the > compatibility layer. > > This code is for reproduction of this bug. > > #include <stdio.h> > #include <stdlib.h> > #include <stdbool.h> > #include <sys/types.h> > #include <sys/stat.h> > #include <fcntl.h> > #include <sys/ioctl.h> > #include <errno.h> > #include <string.h> > > #include <sound/asound.h> > > static void check_elem_read(int fd) > { > struct snd_ctl_elem_value data = {0}; > if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_READ, &data) < 0) { > printf("ioctl(SNDRV_CTL_IOCTL_ELEM_READ): %s\n", > strerror(errno)); > } > } > > static void check_elem_write(int fd) > { > struct snd_ctl_elem_value data = {0}; > if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &data) < 0) { > printf("ioctl(SNDRV_CTL_IOCTL_ELEM_WRITE): %s\n", > strerror(errno)); > } > } > > int main(void) > { > int fd; > > fd = open("/dev/snd/controlC0", O_RDONLY); > if (fd < 0) { > printf("open(2): %s\n", strerror(errno)); > return EXIT_FAILURE; > } > > check_elem_read(fd); > check_elem_write(fd); > > return EXIT_SUCCESS; > } > > I tested on x86_64 machine with enough support for multi-arch. > > $ gcc -Wall -m64 -o ./test ./test.c ; ./test > ioctl(SNDRV_CTL_IOCTL_ELEM_READ): No such file or directory > ioctl(SNDRV_CTL_IOCTL_ELEM_WRITE): No such file or directory > > $ gcc -Wall -m32 -o ./test ./test.c ; ./test > ioctl(SNDRV_CTL_IOCTL_ELEM_READ): No such device or address > ioctl(SNDRV_CTL_IOCTL_ELEM_WRITE): No such device or address > > > This bug is due to return value of condition statement for > 'snd_ctl_find()'. I wrote a patch to fix it. > > > >From d7a8fd14a1bc100dbaf9bf28af47094d5ccb882b Mon Sep 17 00:00:00 2001 > From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> > Date: Wed, 16 Mar 2016 13:40:31 +0900 > Subject: [PATCH] ALSA: ctl: change error code in compatibility layer so that > it's the same code as core implementation > > In control compatibility layer, when no elements are found by > ELEM_READ/ELEM_WRITE ioctl commands, ENXIO is returned. On the other > hand, in core implementation, ENOENT is returned. This is not good for > applications. > > This commit changes the return value from the compatibility layer so > that the same error code is returned. > > diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c > index 0608f21..1fa7076 100644 > --- a/sound/core/control_compat.c > +++ b/sound/core/control_compat.c > @@ -196,7 +196,7 @@ static int get_ctl_type(struct snd_card *card, > struct snd_ctl_elem_id *id, > kctl = snd_ctl_find_id(card, id); > if (! kctl) { > up_read(&card->controls_rwsem); > - return -ENXIO; > + return -ENOENT; > } > info = kzalloc(sizeof(*info), GFP_KERNEL); > if (info == NULL) { > > Well, it's already in merge window for Linux 4.6. I don't mind to > postpone this patch to next developing cycle, while I also think it > better to fix the bug now. So I leave the decision to maintainers. Looks good to me, and trivial enough to be merged for 4.6. Please resubmit with a proper sign off. thanks, Takashi

2 1

[alsa-devel] [PATCH v7 0/6] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 17 Mar '16

17 Mar '16

Not much have changed since v6[1] patch series. I did everything addressed every comment, mostly by robots :), to previous series and there was one bug fixed in tda998x side. Unless there is any significant reaction to this series, I am going to put it to rest for a while now. I have other things to do. There is currently two other patch series[2][3] that depend on the first two (ALSA-) patches of this series. The third ("ASoC: hdmi-codec: Add audio abort() callback for video side to use") is currently not used and can be dropped if so decided. The rest depends on those the first two and adds hdmi-audio support for Beaglebone-black. Best regards, Jyri [1] "[PATCH v6 0/6] Implement generic ASoC HDMI codec and use it in tda998x" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105524.html [2] "[PATCH v6 00/10] ASoC: Add mediatek HDMI codec support" by Philipp Zabel http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105509.html [3] "[RFC v2 0/6] sti: add audio interface to the hdmi driver" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/103374.ht… and "[PATCH v4 0/6] add IEC958 channel status control helpers" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105502.html by Arnaud Pouliquen Changes since v6 * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Add "Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com>" and "Tested-by: Philipp Zabel <p.zabel(a)pengutronix.de>", no other changes * "ALSA: pcm: add IEC958 channel status helper for hw_param" - Added kernel-doc for snd_pcm_create_iec958_consumer_hw_params() * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Use PTR_ERR_OR_ZERO() in tda998x_audio_codec_init() - Fix possible uninitialized use of 'size' in tda998x_get_audio_ports() - Store audio configuration in tda998x_audio_hw_params() instead of tda998x_configure_audio(). Changes since v5 - Rebased on top of the latest drm-next branch - Allow 32 bit samplewidth in snd_pcm_create_iec958_consumer() and snd_pcm_create_iec958_consumer_hw_params() - Propose new simpler DT binding for tda998x audio - Squash tda998x audio DT binding together with hdmi-codec integration Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 Jyri Sarha (6): ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 + arch/arm/boot/dts/am335x-boneblack.dts | 71 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 276 ++++++++++++-- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 65 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 411 +++++++++++++++++++++ 12 files changed, 915 insertions(+), 72 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

2 7

[alsa-devel] [PATCH v3 00/22] Sharing media resources across ALSA and au0828 drivers
by Shuah Khan 17 Mar '16

17 Mar '16

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API and Media Controller API to share media resource (tuner). This Patch v3 series is based on linux_media master. This work addresses Mauro and Takashi's comments on Patch v2 series. Changes since Patch v2 series: - Added documentation for uapi and new MC interfaces 1. Ran make for documentation and Mauro's doc script. Thanks for sharing the script. - Updated commit logs as needed for clarity. - media: v4l-core add enable/disable source common interfaces Addressed comments - added static inlines in header file - media: dvb-frontend invoke enable/disable_source handlers Moved the patch after enable/disable handlers and left pipe in dvb_frontend_private - media: au0828 video remove au0828_enable_analog_tuner() and media: au0828 video change to use v4l_enable_media_source() 1. Collapsed these two patches - it doesn't make sense to keep them separate. Also added FIXME as per Mauro's suggestion. - media: au0828 change to use Managed Media Controller API 1. Added comment explaining the need for using udev->product instead of dev->board.name - media: au0828 add enable, disable source handlers 1. Changed to handle S-Video and Composite inputs. 2. Fixed a bug that lets Video in when ALSA holds the source. 3. Added comments to one FIXME for s_input changes. Fix is in progress and will send it in a day or two combined with other issues found in au0828 video in the way it handles input. Doesn't impact this routine, changes are all in au0828-video. - sound/usb: Use Media Controller API to share media resources 1. Addressed comments from Mauro and Takashi about void pointer use. 2. Fixed kbuildbot error when MEDIA_SUPPORT=m - Dropped media: Ensure media device unregister is done only once Device removal worked without any changes to media unregister path. This work isn't necessary. Shuah Khan (22): [media] Docbook: media-types.xml: Add ALSA Media Controller Intf types uapi/media.h: Declare interface types for ALSA [media] Docbook: media-types.xml: Add Audio Function Entities media: Add ALSA Media Controller function entities media: Media Controller register/unregister entity_notify API media: Media Controller enable/disable source handler API media: Media Controller export non locking __media_entity_setup_link() edia: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add enable/disable source common interfaces media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: Change v4l-core to check if source is free media: au0828 change to use Managed Media Controller API media: au0828 handle media_init and media_register window media: au0828 create tuner to decoder link in disabled state media: au0828 disable tuner to demod link media: au0828-core register entity_notify hook media: au0828 add enable, disable source handlers media: dvb-frontend invoke enable/disable_source handlers media: au0828 video change to use v4l_enable_media_source() sound/usb: Use Media Controller API to share media resources Documentation/DocBook/media/v4l/media-types.xml | 52 ++++ drivers/media/dvb-core/dvb_frontend.c | 135 ++-------- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 42 +++ drivers/media/media-entity.c | 51 +++- drivers/media/usb/au0828/au0828-core.c | 325 ++++++++++++++++++++++-- drivers/media/usb/au0828/au0828-video.c | 102 +++----- drivers/media/usb/au0828/au0828.h | 6 + drivers/media/v4l2-core/Makefile | 1 + drivers/media/v4l2-core/v4l2-fh.c | 2 + drivers/media/v4l2-core/v4l2-ioctl.c | 30 +++ drivers/media/v4l2-core/v4l2-mc.c | 52 ++++ drivers/media/v4l2-core/videobuf2-core.c | 4 + include/media/media-device.h | 86 +++++++ include/media/media-entity.h | 19 ++ include/media/v4l2-dev.h | 1 + include/media/v4l2-mc.h | 65 +++++ include/uapi/linux/media.h | 17 ++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 + sound/usb/card.h | 3 + sound/usb/media.c | 318 +++++++++++++++++++++++ sound/usb/media.h | 72 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 +- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 31 files changed, 1240 insertions(+), 220 deletions(-) create mode 100644 drivers/media/v4l2-core/v4l2-mc.c create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.5.0

5 46

[alsa-devel] [PATCH] sound/usb: Fix memory leak in media_snd_stream_delete() during unbind
by Shuah Khan 17 Mar '16

17 Mar '16

media_snd_stream_delete() fails to release resources during unbind. This leads to use-after-free in media_gobj_create() on a subsequent bind. [ 1445.086410] BUG: KASAN: use-after-free in media_gobj_create+0x3a1/0x470 [media] at addr ffff8801ead49998 [ 1445.086771] Call Trace: [ 1445.086779] [<ffffffff81ade373>] dump_stack+0x67/0x94 [ 1445.086785] [<ffffffff81523c29>] print_trailer+0xf9/0x150 [ 1445.086790] [<ffffffff81529644>] object_err+0x34/0x40 [ 1445.086796] [<ffffffff8152b9e1>] kasan_report_error+0x221/0x530 [ 1445.086803] [<ffffffff8152bfb3>] __asan_report_store8_noabort+0x43/0x50 [ 1445.086813] [<ffffffffa0a79341>] ? media_gobj_create+0x3a1/0x470 [media] [ 1445.086822] [<ffffffffa0a79341>] media_gobj_create+0x3a1/0x470 [media] [ 1445.086831] [<ffffffffa0a705a9>] media_device_register_entity+0x259/0x6f0 [media] [ 1445.086841] [<ffffffffa0a70350>] ? media_device_unregister_entity_notify+0x100/0x100 [media] [ 1445.086846] [<ffffffff81526232>] ? ___slab_alloc+0x172/0x500 [ 1445.086854] [<ffffffff81203548>] ? mark_held_locks+0xc8/0x120 [ 1445.086859] [<ffffffff81526610>] ? __slab_alloc+0x50/0x70 [ 1445.086878] [<ffffffffa0fe711c>] ? media_snd_mixer_init+0x16c/0x500 [snd_usb_audio] [ 1445.086884] [<ffffffff8152b086>] ? kasan_unpoison_shadow+0x36/0x50 [ 1445.086890] [<ffffffff8152b086>] ? kasan_unpoison_shadow+0x36/0x50 [ 1445.086895] [<ffffffff8152b0fe>] ? kasan_kmalloc+0x5e/0x70 Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- sound/usb/media.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index 44a5de9..0d03773 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -135,7 +135,7 @@ void media_snd_stream_delete(struct snd_usb_substream *subs) if (mctl && mctl->media_dev) { struct media_device *mdev; - mdev = subs->stream->chip->media_dev; + mdev = mctl->media_dev; if (mdev && media_devnode_is_registered(&mdev->devnode)) { media_devnode_remove(mctl->intf_devnode); media_device_unregister_entity(&mctl->media_entity); -- 2.5.0

2 2

[alsa-devel] [PATCH] ASoC: nau8825: support different clock source for FLL function
by John Hsu 17 Mar '16

17 Mar '16

Extend FLL clock source selection. The source can be from MCLK, BCLK or FS. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 82 +++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 8 +++++ 2 files changed, 67 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 1c87299..ac264ec 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1053,6 +1053,37 @@ static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, return 0; } +static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) +{ + int ret = 0; + + nau8825->mclk = devm_clk_get(nau8825->dev, "mclk"); + if (IS_ERR(nau8825->mclk)) { + dev_info(nau8825->dev, "No 'mclk' clock found, assume MCLK is managed externally"); + return 0; + } + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + if (nau8825->mclk_freq != freq) { + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + return ret; + } + nau8825->mclk_freq = freq; + } + + return 0; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -1064,29 +1095,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); - - /* We selected MCLK source but the clock itself managed externally */ - if (!nau8825->mclk) - break; - - if (!nau8825->mclk_freq) { - ret = clk_prepare_enable(nau8825->mclk); - if (ret) { - dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); - return ret; - } - } - - if (nau8825->mclk_freq != freq) { - nau8825->mclk_freq = freq; - - freq = clk_round_rate(nau8825->mclk, freq); - ret = clk_set_rate(nau8825->mclk, freq); - if (ret) { - dev_err(nau8825->dev, "Unable to set mclk rate\n"); - return ret; - } - } + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; break; case NAU8825_CLK_INTERNAL: @@ -1094,7 +1105,32 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + break; + case NAU8825_CLK_FLL_MCLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_BLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + case NAU8825_CLK_FLL_FS: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8ceb5f3..ed0d8f3 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -113,6 +113,11 @@ /* FLL3 (0x06) */ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8825_FLL_CLK_SRC_SFT 10 +#define NAU8825_FLL_CLK_SRC_MASK (0x3 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_MCLK (0 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_BLK (0x2 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) @@ -320,6 +325,9 @@ enum { NAU8825_CLK_MCLK = 0, NAU8825_CLK_INTERNAL, + NAU8825_CLK_FLL_MCLK, + NAU8825_CLK_FLL_BLK, + NAU8825_CLK_FLL_FS, }; struct nau8825 { -- 2.6.4

3 4

[alsa-devel] [PATCH] ALSA: hda - codec add DP MST support for connection list
by libin.yang＠linux.intel.com 17 Mar '16

17 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patches adds the support of connection list for DP MST. With this, hdmi driver in DP MST mode can easily reuse the connection list mechanism. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 52 +++++++++++++++++++++++++++--------------- sound/pci/hda/hda_codec.h | 15 ++++++------ sound/pci/hda/hda_generic.c | 18 +++++++-------- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/patch_analog.c | 18 ++++++++------- sound/pci/hda/patch_hdmi.c | 8 ++++--- sound/pci/hda/patch_realtek.c | 20 ++++++++-------- sound/pci/hda/patch_sigmatel.c | 4 ++-- sound/pci/hda/patch_via.c | 10 ++++---- 9 files changed, 84 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8374188..d17e38e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -110,23 +110,24 @@ struct hda_conn_list { struct list_head list; int len; hda_nid_t nid; + int dev_id; hda_nid_t conns[0]; }; /* look up the cached results */ static struct hda_conn_list * -lookup_conn_list(struct hda_codec *codec, hda_nid_t nid) +lookup_conn_list(struct hda_codec *codec, hda_nid_t nid, int dev_id) { struct hda_conn_list *p; list_for_each_entry(p, &codec->conn_list, list) { - if (p->nid == nid) + if ((p->nid == nid) && (p->dev_id == dev_id)) return p; } return NULL; } -static int add_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, - const hda_nid_t *list) +static int add_conn_list(struct hda_codec *codec, hda_nid_t nid, + int dev_id, int len, const hda_nid_t *list) { struct hda_conn_list *p; @@ -135,6 +136,7 @@ static int add_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, return -ENOMEM; p->len = len; p->nid = nid; + p->dev_id = dev_id; memcpy(p->conns, list, len * sizeof(hda_nid_t)); list_add(&p->list, &codec->conn_list); return 0; @@ -150,8 +152,13 @@ static void remove_conn_list(struct hda_codec *codec) } } -/* read the connection and add to the cache */ -static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) +/* + * read the connection and add to the cache + * the caller should select the device entry by sending the + * corresponding verb if necessary before calling this function + */ +static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid, + int dev_id) { hda_nid_t list[32]; hda_nid_t *result = list; @@ -166,7 +173,8 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) len = snd_hda_get_raw_connections(codec, nid, result, len); } if (len >= 0) - len = snd_hda_override_conn_list(codec, nid, len, result); + len = snd_hda_override_conn_list(codec, nid, dev_id, + len, result); if (result != list) kfree(result); return len; @@ -176,6 +184,7 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) * snd_hda_get_conn_list - get connection list * @codec: the HDA codec * @nid: NID to parse + * @dev_id: device entry id * @listp: the pointer to store NID list * * Parses the connection list of the given widget and stores the pointer @@ -188,7 +197,7 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) * concurrently, protect with a mutex appropriately. */ int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, - const hda_nid_t **listp) + int dev_id, const hda_nid_t **listp) { bool added = false; @@ -197,7 +206,7 @@ int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, const struct hda_conn_list *p; /* if the connection-list is already cached, read it */ - p = lookup_conn_list(codec, nid); + p = lookup_conn_list(codec, nid, dev_id); if (p) { if (listp) *listp = p->conns; @@ -206,7 +215,7 @@ int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, if (snd_BUG_ON(added)) return -EINVAL; - err = read_and_add_raw_conns(codec, nid); + err = read_and_add_raw_conns(codec, nid, dev_id); if (err < 0) return err; added = true; @@ -218,6 +227,7 @@ EXPORT_SYMBOL_GPL(snd_hda_get_conn_list); * snd_hda_get_connections - copy connection list * @codec: the HDA codec * @nid: NID to parse + * @dev_id: device entry id * @conn_list: connection list array; when NULL, checks only the size * @max_conns: max. number of connections to store * @@ -227,10 +237,11 @@ EXPORT_SYMBOL_GPL(snd_hda_get_conn_list); * Returns the number of connections, or a negative error code. */ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns) + int dev_id, hda_nid_t *conn_list, + int max_conns) { const hda_nid_t *list; - int len = snd_hda_get_conn_list(codec, nid, &list); + int len = snd_hda_get_conn_list(codec, nid, dev_id, &list); if (len > 0 && conn_list) { if (len > max_conns) { @@ -249,6 +260,7 @@ EXPORT_SYMBOL_GPL(snd_hda_get_connections); * snd_hda_override_conn_list - add/modify the connection-list to cache * @codec: the HDA codec * @nid: NID to parse + * @dev_id: device entry id * @len: number of connection list entries * @list: the list of connection entries * @@ -257,18 +269,18 @@ EXPORT_SYMBOL_GPL(snd_hda_get_connections); * * Returns zero or a negative error code. */ -int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, - const hda_nid_t *list) +int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, + int dev_id, int len, const hda_nid_t *list) { struct hda_conn_list *p; - p = lookup_conn_list(codec, nid); + p = lookup_conn_list(codec, nid, dev_id); if (p) { list_del(&p->list); kfree(p); } - return add_conn_list(codec, nid, len, list); + return add_conn_list(codec, nid, dev_id, len, list); } EXPORT_SYMBOL_GPL(snd_hda_override_conn_list); @@ -277,6 +289,7 @@ EXPORT_SYMBOL_GPL(snd_hda_override_conn_list); * @codec: the HDA codec * @mux: NID containing the list * @nid: NID to select + * @dev_id: device entry id * @recursive: 1 when searching NID recursively, otherwise 0 * * Parses the connection list of the widget @mux and checks whether the @@ -284,12 +297,12 @@ EXPORT_SYMBOL_GPL(snd_hda_override_conn_list); * Otherwise it returns -1. */ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, - hda_nid_t nid, int recursive) + hda_nid_t nid, int dev_id, int recursive) { const hda_nid_t *conn; int i, nums; - nums = snd_hda_get_conn_list(codec, mux, &conn); + nums = snd_hda_get_conn_list(codec, mux, dev_id, &conn); for (i = 0; i < nums; i++) if (conn[i] == nid) return i; @@ -304,7 +317,8 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i])); if (type == AC_WID_PIN || type == AC_WID_AUD_OUT) continue; - if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0) + if (snd_hda_get_conn_index(codec, conn[i], nid, dev_id, + recursive) >= 0) return i; } return -1; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 373fcad..4852c1a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -329,11 +329,12 @@ snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, #define snd_hda_get_sub_nodes(codec, nid, start_nid) \ snd_hdac_get_sub_nodes(&(codec)->core, nid, start_nid) int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns); + int dev_id, hda_nid_t *conn_list, + int max_conns); static inline int -snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) +snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid, int dev_id) { - return snd_hda_get_connections(codec, nid, NULL, 0); + return snd_hda_get_connections(codec, nid, dev_id, NULL, 0); } #define snd_hda_get_raw_connections(codec, nid, list, max_conns) \ @@ -341,12 +342,12 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) #define snd_hda_get_num_raw_conns(codec, nid) \ snd_hdac_get_connections(&(codec)->core, nid, NULL, 0); -int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, +int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, int dev_id, const hda_nid_t **listp); -int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, - const hda_nid_t *list); +int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, + int dev_id, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, - hda_nid_t nid, int recursive); + hda_nid_t nid, int dev_id, int recursive); int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, u8 *dev_list, int max_devices); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7ca5b89..f8300b4 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -338,7 +338,7 @@ static bool is_reachable_path(struct hda_codec *codec, { if (!from_nid || !to_nid) return false; - return snd_hda_get_conn_index(codec, to_nid, from_nid, true) >= 0; + return snd_hda_get_conn_index(codec, to_nid, from_nid, 0, true) >= 0; } /* nid, dir and idx */ @@ -397,7 +397,7 @@ static bool __parse_nid_path(struct hda_codec *codec, else if (to_nid == (hda_nid_t)(-anchor_nid)) return false; /* hit the exclusive nid */ - nums = snd_hda_get_conn_list(codec, to_nid, &conn); + nums = snd_hda_get_conn_list(codec, to_nid, 0, &conn); for (i = 0; i < nums; i++) { if (conn[i] != from_nid) { /* special case: when from_nid is 0, @@ -696,9 +696,9 @@ static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid, int dir) return true; if (dir != HDA_INPUT || get_wcaps_type(wcaps) != AC_WID_AUD_MIX) return false; - if (snd_hda_get_num_conns(codec, nid) != 1) + if (snd_hda_get_num_conns(codec, nid, 0) != 1) return false; - if (snd_hda_get_connections(codec, nid, &conn, 1) < 0) + if (snd_hda_get_connections(codec, nid, 0, &conn, 1) < 0) return false; return !!(get_wcaps(codec, conn) & AC_WCAP_STEREO); } @@ -791,7 +791,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, int type; hda_nid_t nid = path->path[i]; - nums = snd_hda_get_conn_list(codec, nid, &conn); + nums = snd_hda_get_conn_list(codec, nid, 0, &conn); type = get_wcaps_type(get_wcaps(codec, nid)); if (type == AC_WID_PIN || (type == AC_WID_AUD_IN && codec->single_adc_amp)) { @@ -1058,7 +1058,7 @@ static int add_sw_ctl(struct hda_codec *codec, const char *pfx, int cidx, val = amp_val_replace_channels(val, chs); if (get_amp_direction_(val) == HDA_INPUT) { hda_nid_t nid = get_amp_nid_(val); - int nums = snd_hda_get_num_conns(codec, nid); + int nums = snd_hda_get_num_conns(codec, nid, 0); if (nums > 1) { type = HDA_CTL_BIND_MUTE; val |= nums << 19; @@ -3004,7 +3004,7 @@ static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, const hda_nid_t *list; hda_nid_t nid; - idx = snd_hda_get_conn_index(codec, mix_nid, pin, true); + idx = snd_hda_get_conn_index(codec, mix_nid, pin, 0, true); if (idx < 0) return false; @@ -3017,7 +3017,7 @@ static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, return true; /* check leaf node */ - num_conns = snd_hda_get_conn_list(codec, mix_nid, &list); + num_conns = snd_hda_get_conn_list(codec, mix_nid, 0, &list); if (num_conns < idx) return false; nid = list[idx]; @@ -4760,7 +4760,7 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) const hda_nid_t *conn; bool has_amp; - nums = snd_hda_get_conn_list(codec, mix, &conn); + nums = snd_hda_get_conn_list(codec, mix, 0, &conn); has_amp = nid_has_mute(codec, mix, HDA_INPUT); for (i = 0; i < nums; i++) { if (has_amp) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 033aa84..c747183 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -636,7 +636,7 @@ static void print_conn_list(struct snd_info_buffer *buffer, } /* Get Cache connections info */ - cache_len = snd_hda_get_conn_list(codec, nid, &list); + cache_len = snd_hda_get_conn_list(codec, nid, 0, &list); if (cache_len >= 0 && (cache_len != conn_len || memcmp(list, conn, conn_len) != 0)) { snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e0fb8c6..6f007ff 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -466,7 +466,7 @@ static int ad1983_auto_smux_enum_info(struct snd_kcontrol *kcontrol, static const char * const texts2[] = { "PCM", "ADC" }; static const char * const texts3[] = { "PCM", "ADC1", "ADC2" }; hda_nid_t dig_out = spec->gen.multiout.dig_out_nid; - int num_conns = snd_hda_get_num_conns(codec, dig_out); + int num_conns = snd_hda_get_num_conns(codec, dig_out, 0); if (num_conns == 2) return snd_hda_enum_helper_info(kcontrol, uinfo, 2, texts2); @@ -493,7 +493,7 @@ static int ad1983_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; hda_nid_t dig_out = spec->gen.multiout.dig_out_nid; - int num_conns = snd_hda_get_num_conns(codec, dig_out); + int num_conns = snd_hda_get_num_conns(codec, dig_out, 0); if (val >= num_conns) return -EINVAL; @@ -521,7 +521,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) if (!dig_out) return 0; - num_conns = snd_hda_get_num_conns(codec, dig_out); + num_conns = snd_hda_get_num_conns(codec, dig_out, 0); if (num_conns != 2 && num_conns != 3) return 0; if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &ad1983_auto_smux_mixer)) @@ -546,8 +546,10 @@ static int patch_ad1983(struct hda_codec *codec) set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); /* limit the loopback routes not to confuse the parser */ - snd_hda_override_conn_list(codec, 0x0c, ARRAY_SIZE(conn_0c), conn_0c); - snd_hda_override_conn_list(codec, 0x0d, ARRAY_SIZE(conn_0d), conn_0d); + snd_hda_override_conn_list(codec, 0x0c, 0, ARRAY_SIZE(conn_0c), + conn_0c); + snd_hda_override_conn_list(codec, 0x0d, 0, ARRAY_SIZE(conn_0d), + conn_0d); err = ad198x_parse_auto_config(codec, false); if (err < 0) @@ -745,7 +747,7 @@ static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, static const char * const texts[] = { "PCM", "ADC1", "ADC2", "ADC3", }; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = snd_hda_get_num_conns(codec, 0x0b, 0) + 1; if (num_conns > 4) num_conns = 4; return snd_hda_enum_helper_info(kcontrol, uinfo, num_conns, texts); @@ -768,7 +770,7 @@ static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; unsigned int val = ucontrol->value.enumerated.item[0]; struct nid_path *path; - int num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + int num_conns = snd_hda_get_num_conns(codec, 0x0b, 0) + 1; if (val >= num_conns) return -EINVAL; @@ -856,7 +858,7 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec) get_wcaps_type(get_wcaps(codec, 0x1d)) != AC_WID_AUD_MIX) return 0; - num_conns = snd_hda_get_num_conns(codec, 0x0b) + 1; + num_conns = snd_hda_get_num_conns(codec, 0x0b, 0) + 1; if (num_conns != 3 && num_conns != 4) return 0; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index fe4141c..7e03200 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1719,7 +1719,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return -EINVAL; } - per_pin->num_mux_nids = snd_hda_get_connections(codec, pin_nid, + per_pin->num_mux_nids = snd_hda_get_connections(codec, pin_nid, 0, per_pin->mux_nids, HDA_MAX_CONNECTIONS); @@ -2818,14 +2818,16 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, hda_nid_t conns[4]; int nconns; - nconns = snd_hda_get_connections(codec, nid, conns, ARRAY_SIZE(conns)); + nconns = snd_hda_get_connections(codec, nid, 0, conns, + ARRAY_SIZE(conns)); if (nconns == spec->num_cvts && !memcmp(conns, spec->cvt_nids, spec->num_cvts * sizeof(hda_nid_t))) return; /* override pins connection list */ codec_dbg(codec, "hdmi: haswell: override pin connection 0x%x\n", nid); - snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); + snd_hda_override_conn_list(codec, nid, 0, spec->num_cvts, + spec->cvt_nids); } #define INTEL_VENDOR_NID 0x08 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 93d2156..fb70f18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1851,17 +1851,17 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, /* fake the connections during parsing the tree */ hda_nid_t conn1[2] = { 0x0c, 0x0d }; hda_nid_t conn2[2] = { 0x0e, 0x0f }; - snd_hda_override_conn_list(codec, 0x14, 2, conn1); - snd_hda_override_conn_list(codec, 0x15, 2, conn1); - snd_hda_override_conn_list(codec, 0x18, 2, conn2); - snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + snd_hda_override_conn_list(codec, 0x14, 0, 2, conn1); + snd_hda_override_conn_list(codec, 0x15, 0, 2, conn1); + snd_hda_override_conn_list(codec, 0x18, 0, 2, conn2); + snd_hda_override_conn_list(codec, 0x1a, 0, 2, conn2); } else if (action == HDA_FIXUP_ACT_PROBE) { /* restore the connections */ hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; - snd_hda_override_conn_list(codec, 0x14, 5, conn); - snd_hda_override_conn_list(codec, 0x15, 5, conn); - snd_hda_override_conn_list(codec, 0x18, 5, conn); - snd_hda_override_conn_list(codec, 0x1a, 5, conn); + snd_hda_override_conn_list(codec, 0x14, 0, 5, conn); + snd_hda_override_conn_list(codec, 0x15, 0, 5, conn); + snd_hda_override_conn_list(codec, 0x18, 0, 5, conn); + snd_hda_override_conn_list(codec, 0x1a, 0, 5, conn); } } @@ -4607,8 +4607,8 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, make sure 0x14 (front speaker) and 0x15 (headphones) use the stereo DAC, while leaving 0x17 (bass speaker) for node 0x03. */ hda_nid_t conn1[2] = { 0x0c }; - snd_hda_override_conn_list(codec, 0x14, 1, conn1); - snd_hda_override_conn_list(codec, 0x15, 1, conn1); + snd_hda_override_conn_list(codec, 0x14, 0, 1, conn1); + snd_hda_override_conn_list(codec, 0x15, 0, 1, conn1); } } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 37b70f8..5bff793 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -984,7 +984,7 @@ static int stac_create_spdif_mux_ctls(struct hda_codec *codec) if (cfg->dig_outs < 1) return 0; - num_cons = snd_hda_get_num_conns(codec, cfg->dig_out_pins[0]); + num_cons = snd_hda_get_num_conns(codec, cfg->dig_out_pins[0], 0); if (num_cons <= 1) return 0; @@ -4543,7 +4543,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; - num_dacs = snd_hda_get_num_conns(codec, 0x0a) - 1; + num_dacs = snd_hda_get_num_conns(codec, 0x0a, 0) - 1; if (num_dacs < 3 || num_dacs > 5) { codec_warn(codec, "Could not determine number of channels defaulting to DAC count\n"); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fc30d1e..14d9010 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -867,7 +867,7 @@ static int add_secret_dac_path(struct hda_codec *codec) if (!spec->gen.mixer_nid) return 0; - nums = snd_hda_get_connections(codec, spec->gen.mixer_nid, conn, + nums = snd_hda_get_connections(codec, spec->gen.mixer_nid, 0, conn, ARRAY_SIZE(conn) - 1); for (i = 0; i < nums; i++) { if (get_wcaps_type(get_wcaps(codec, conn[i])) == AC_WID_AUD_OUT) @@ -882,7 +882,7 @@ static int add_secret_dac_path(struct hda_codec *codec) conn[nums++] = nid; return snd_hda_override_conn_list(codec, spec->gen.mixer_nid, - nums, conn); + 0, nums, conn); } } return 0; @@ -1082,8 +1082,10 @@ static void fix_vt1802_connections(struct hda_codec *codec) static hda_nid_t conn_24[] = { 0x14, 0x1c }; static hda_nid_t conn_33[] = { 0x1c }; - snd_hda_override_conn_list(codec, 0x24, ARRAY_SIZE(conn_24), conn_24); - snd_hda_override_conn_list(codec, 0x33, ARRAY_SIZE(conn_33), conn_33); + snd_hda_override_conn_list(codec, 0x24, 0, ARRAY_SIZE(conn_24), + conn_24); + snd_hda_override_conn_list(codec, 0x33, 0, ARRAY_SIZE(conn_33), + conn_33); } /* patch for vt2002P */ -- 1.9.1

3 12

[alsa-devel] [PATCH] ASoC: nau8825: add crosstalk suppression feature
by John Hsu 17 Mar '16

17 Mar '16

Add crosstalk signal suppression function. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 403 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 77 ++++++++- 2 files changed, 479 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index d270f63..81554be 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -82,6 +82,380 @@ static const struct nau8825_fll_attr fll_pre_scalar[] = { { 8, 0x3 }, }; +/* Crosstalk detection */ +#define LOG10_MAGIC 646456993 +#define GAIN_AUGMENT 22500 +#define SIDETONE_BASE 207000 + +static struct reg_default nau8825_xtalk_baktab[] = { + { NAU8825_REG_FLL6, 0 }, + { NAU8825_REG_CLK_DIVIDER, 0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0 }, + { NAU8825_REG_HSVOL_CTRL, 0 }, + { NAU8825_REG_DACL_CTRL, 0 }, + { NAU8825_REG_DACR_CTRL, 0 }, + { NAU8825_REG_CLASSG_CTRL, 0 }, +}; + +static const struct reg_default nau8825_biq_reg[] = { + { NAU8825_REG_BIQ_CTRL, 0x0000 }, + { NAU8825_REG_BIQ_COF1, 0x009B }, + { NAU8825_REG_BIQ_COF2, 0x0006 }, + { NAU8825_REG_BIQ_COF3, 0xff66 }, + { NAU8825_REG_BIQ_COF4, 0x0000 }, + { NAU8825_REG_BIQ_COF5, 0xffb3 }, + { NAU8825_REG_BIQ_COF6, 0x0000 }, + { NAU8825_REG_BIQ_COF7, 0x009A }, + { NAU8825_REG_BIQ_COF8, 0x0006 }, + { NAU8825_REG_BIQ_COF9, 0xffb3 }, + { NAU8825_REG_BIQ_COF10, 0x8000 }, + { NAU8825_REG_BIQ_CTRL, 0x0010 }, +}; + +static const unsigned short logtable[256] = { + 0x0000, 0x0171, 0x02e0, 0x044e, 0x05ba, 0x0725, 0x088e, 0x09f7, + 0x0b5d, 0x0cc3, 0x0e27, 0x0f8a, 0x10eb, 0x124b, 0x13aa, 0x1508, + 0x1664, 0x17bf, 0x1919, 0x1a71, 0x1bc8, 0x1d1e, 0x1e73, 0x1fc6, + 0x2119, 0x226a, 0x23ba, 0x2508, 0x2656, 0x27a2, 0x28ed, 0x2a37, + 0x2b80, 0x2cc8, 0x2e0f, 0x2f54, 0x3098, 0x31dc, 0x331e, 0x345f, + 0x359f, 0x36de, 0x381b, 0x3958, 0x3a94, 0x3bce, 0x3d08, 0x3e41, + 0x3f78, 0x40af, 0x41e4, 0x4319, 0x444c, 0x457f, 0x46b0, 0x47e1, + 0x4910, 0x4a3f, 0x4b6c, 0x4c99, 0x4dc5, 0x4eef, 0x5019, 0x5142, + 0x526a, 0x5391, 0x54b7, 0x55dc, 0x5700, 0x5824, 0x5946, 0x5a68, + 0x5b89, 0x5ca8, 0x5dc7, 0x5ee5, 0x6003, 0x611f, 0x623a, 0x6355, + 0x646f, 0x6588, 0x66a0, 0x67b7, 0x68ce, 0x69e4, 0x6af8, 0x6c0c, + 0x6d20, 0x6e32, 0x6f44, 0x7055, 0x7165, 0x7274, 0x7383, 0x7490, + 0x759d, 0x76aa, 0x77b5, 0x78c0, 0x79ca, 0x7ad3, 0x7bdb, 0x7ce3, + 0x7dea, 0x7ef0, 0x7ff6, 0x80fb, 0x81ff, 0x8302, 0x8405, 0x8507, + 0x8608, 0x8709, 0x8809, 0x8908, 0x8a06, 0x8b04, 0x8c01, 0x8cfe, + 0x8dfa, 0x8ef5, 0x8fef, 0x90e9, 0x91e2, 0x92db, 0x93d2, 0x94ca, + 0x95c0, 0x96b6, 0x97ab, 0x98a0, 0x9994, 0x9a87, 0x9b7a, 0x9c6c, + 0x9d5e, 0x9e4f, 0x9f3f, 0xa02e, 0xa11e, 0xa20c, 0xa2fa, 0xa3e7, + 0xa4d4, 0xa5c0, 0xa6ab, 0xa796, 0xa881, 0xa96a, 0xaa53, 0xab3c, + 0xac24, 0xad0c, 0xadf2, 0xaed9, 0xafbe, 0xb0a4, 0xb188, 0xb26c, + 0xb350, 0xb433, 0xb515, 0xb5f7, 0xb6d9, 0xb7ba, 0xb89a, 0xb97a, + 0xba59, 0xbb38, 0xbc16, 0xbcf4, 0xbdd1, 0xbead, 0xbf8a, 0xc065, + 0xc140, 0xc21b, 0xc2f5, 0xc3cf, 0xc4a8, 0xc580, 0xc658, 0xc730, + 0xc807, 0xc8de, 0xc9b4, 0xca8a, 0xcb5f, 0xcc34, 0xcd08, 0xcddc, + 0xceaf, 0xcf82, 0xd054, 0xd126, 0xd1f7, 0xd2c8, 0xd399, 0xd469, + 0xd538, 0xd607, 0xd6d6, 0xd7a4, 0xd872, 0xd93f, 0xda0c, 0xdad9, + 0xdba5, 0xdc70, 0xdd3b, 0xde06, 0xded0, 0xdf9a, 0xe063, 0xe12c, + 0xe1f5, 0xe2bd, 0xe385, 0xe44c, 0xe513, 0xe5d9, 0xe69f, 0xe765, + 0xe82a, 0xe8ef, 0xe9b3, 0xea77, 0xeb3b, 0xebfe, 0xecc1, 0xed83, + 0xee45, 0xef06, 0xefc8, 0xf088, 0xf149, 0xf209, 0xf2c8, 0xf387, + 0xf446, 0xf505, 0xf5c3, 0xf680, 0xf73e, 0xf7fb, 0xf8b7, 0xf973, + 0xfa2f, 0xfaea, 0xfba5, 0xfc60, 0xfd1a, 0xfdd4, 0xfe8e, 0xff47 +}; + +/** + * computes log10 of a value; the result is round off to 3 decimal. + * this function takes reference to dvb-math. + * + * @return log10(value) * 1000 + */ +static u32 nau8825_intlog10_dec3(u32 value) +{ + u32 msb, logentry, significand, interpolation, log10val; + u64 log2val; + + msb = fls(value) - 1; + significand = value << (31 - msb); + logentry = (significand >> 23) & 0xff; + interpolation = ((significand & 0x7fffff) * + ((logtable[(logentry + 1) & 0xff] - + logtable[logentry]) & 0xffff)) >> 15; + + log2val = ((msb << 24) + (logtable[logentry] << 8) + interpolation); + log10val = (log2val * LOG10_MAGIC) >> 31; + return log10val / ((1 << 24) / 1000); +} + +/** + * computes cross-talk sidetone with orignal and cross-talk signal + * + * @sig_org: orignal signal + * @sig_cros: cross-talk signal + * + * return cross-talk sidetone + */ +static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) +{ + u32 gain, sidetone; + + if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { + WARN_ON(1); + return 0; + } + + sig_org = nau8825_intlog10_dec3(sig_org); + sig_cros = nau8825_intlog10_dec3(sig_cros); + gain = (sig_org - sig_cros) * 20 + GAIN_AUGMENT; + sidetone = SIDETONE_BASE - gain * 2; + sidetone /= 1000; + + return sidetone; +} + +static void nau8825_xtalk_backup(struct nau8825 *nau8825) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(nau8825_xtalk_baktab); i++) + regmap_read(nau8825->regmap, nau8825_xtalk_baktab[i].reg, + &nau8825_xtalk_baktab[i].def); +} + +static void nau8825_xtalk_restore(struct nau8825 *nau8825) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(nau8825_xtalk_baktab); i++) + regmap_write(nau8825->regmap, nau8825_xtalk_baktab[i].reg, + nau8825_xtalk_baktab[i].def); +} + +static void nau8825_xtalk_biq_apply(struct nau8825 *nau8825) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(nau8825_biq_reg); i++) + regmap_write(nau8825->regmap, nau8825_biq_reg[i].reg, + nau8825_biq_reg[i].def); +} + +static void nau8825_xtalk_biq_cancel(struct nau8825 *nau8825) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(nau8825_biq_reg); i++) + regmap_write(nau8825->regmap, nau8825_biq_reg[i].reg, 0); +} + +static void nau8825_xtalk_prepare_dac(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDACL_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_HP_BOOST_DIS, NAU8825_HP_BOOST_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSD_CTRL, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR | NAU8825_ENABLE_DACL | + NAU8825_ENABLE_ADC | NAU8825_ENABLE_ADC_CLK | + NAU8825_ENABLE_DAC_CLK, NAU8825_ENABLE_DACR | + NAU8825_ENABLE_DACL | NAU8825_ENABLE_ADC | + NAU8825_ENABLE_ADC_CLK | NAU8825_ENABLE_DAC_CLK); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLASSG_CTRL, + NAU8825_CLASSG_LDAC_EN | NAU8825_CLASSG_RDAC_EN, + NAU8825_CLASSG_LDAC_EN | NAU8825_CLASSG_RDAC_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW | NAU8825_CHANRGE_PUMP_EN, + NAU8825_JAMNODCLOW | NAU8825_CHANRGE_PUMP_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L | + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L | + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSD_CTRL, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, 0); +} + +static void nau8825_xtalk_prepare_adc(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ANALOG_ADC_2, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_MASK, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB); +} + +static void nau8825_xtalk_clock(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, NAU8825_DCO_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_VCO | 0x3); +} + +static void nau8825_xtalk_prepare(struct nau8825 *nau8825) +{ + /* backup value of specific register used by cross-talk */ + nau8825_xtalk_backup(nau8825); + nau8825_xtalk_clock(nau8825); + nau8825_xtalk_prepare_dac(nau8825); + nau8825_xtalk_prepare_adc(nau8825); + /* Config channel */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_MASK, NAU8825_DACR_CH_SEL_R); + /* Config headphone volume */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HPL_VOL_MASK | NAU8825_HPR_VOL_MASK, 0x0492); + /* Config IIS parameter */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_DRV_MASK | NAU8825_I2S_BLK_DIV_MASK, + (0x2 << NAU8825_I2S_DRV_SFT) | 0x1); + /* Config crosstalk */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, + NAU8825_IMM_THD_MASK | NAU8825_IMM_CYC_MASK | + NAU8825_IMM_DAC_SRC_MASK, (0x2 << NAU8825_IMM_THD_SFT) | + NAU8825_IMM_CYC_8192 | NAU8825_IMM_DAC_SRC_SIN); + regmap_update_bits(nau8825->regmap, + NAU8825_REG_INTERRUPT_MASK, NAU8825_IRQ_RMS_EN, 0); + nau8825_xtalk_biq_apply(nau8825); +} + +static void nau8825_xtalk_clean_dac(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_HP_BOOST_DIS, NAU8825_HP_BOOST_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL | + NAU8825_JAMNODCLOW); + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSD_CTRL, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLASSG_CTRL, + NAU8825_CLASSG_LDAC_EN | NAU8825_CLASSG_RDAC_EN, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_CHANRGE_PUMP_EN, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP | + NAU8825_BIAS_TESTDAC_EN, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_HP_BOOST_DIS, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL | NAU8825_ENABLE_ADC | + NAU8825_ENABLE_ADC_CLK | NAU8825_ENABLE_DAC_CLK, 0); + if (!nau8825->irq) + regmap_update_bits(nau8825->regmap, + NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN, 0); +} + +static void nau8825_xtalk_clean_adc(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ANALOG_ADC_2, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_MASK, 0); +} + +static void nau8825_xtalk_clean(struct nau8825 *nau8825) +{ + /* restore value of specific register for cross-talk */ + nau8825_xtalk_restore(nau8825); + nau8825_xtalk_clean_dac(nau8825); + nau8825_xtalk_clean_adc(nau8825); + regmap_write(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_RMS_EN, NAU8825_IRQ_RMS_EN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK | + NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + nau8825_xtalk_biq_cancel(nau8825); +} + +static void nau8825_xtalk_imm_start(struct nau8825 *nau8825, int vol) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ADC_DGAIN_CTRL, + NAU8825_ADC_DIG_VOL_MASK, vol); + switch (nau8825->xtalk_state) { + case NAU8825_XTALK_HPR_R2L: + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP, + NAU8825_BIAS_HPR_IMP); + break; + case NAU8825_XTALK_HPL_R2L: + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP, + NAU8825_BIAS_HPL_IMP); + break; + default: + break; + } + msleep(100); + regmap_update_bits(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, + NAU8825_IMM_EN, NAU8825_IMM_EN); +} + +static void nau8825_xtalk_imm_stop(struct nau8825 *nau8825) +{ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_IMM_MODE_CTRL, NAU8825_IMM_EN, 0); +} + +static void nau8825_xtalk_measure(struct nau8825 *nau8825) +{ + u32 sidetone; + + switch (nau8825->xtalk_state) { + case NAU8825_XTALK_DONE: + nau8825_xtalk_prepare(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_HPR_R2L; + msleep(280); + nau8825_xtalk_imm_start(nau8825, 0x00d2); + break; + case NAU8825_XTALK_HPR_R2L: + regmap_read(nau8825->regmap, NAU8825_REG_IMM_RMS_L, + &nau8825->imp_rms[NAU8825_XTALK_HPR_R2L]); + dev_dbg(nau8825->dev, "HPR_R2L imm: %x\n", + nau8825->imp_rms[NAU8825_XTALK_HPR_R2L]); + nau8825_xtalk_imm_stop(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_HPL_R2L; + nau8825_xtalk_imm_start(nau8825, 0x00ff); + break; + case NAU8825_XTALK_HPL_R2L: + regmap_read(nau8825->regmap, NAU8825_REG_IMM_RMS_L, + &nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + dev_dbg(nau8825->dev, "HPL_R2L imm: %x\n", + nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + nau8825_xtalk_imm_stop(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_IMM; + break; + case NAU8825_XTALK_IMM: + sidetone = nau8825_xtalk_sidetone( + nau8825->imp_rms[NAU8825_XTALK_HPR_R2L], + nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + dev_dbg(nau8825->dev, "cross-talk sidetone: %x\n", sidetone); + regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, + (sidetone << 8) | sidetone); + nau8825_xtalk_clean(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_DONE; + break; + default: + break; + } +} + +static void nau8825_xtalk_work(struct work_struct *work) +{ + struct nau8825 *nau8825 = container_of( + work, struct nau8825, xtalk_work); + + nau8825_xtalk_measure(nau8825); + if (nau8825->xtalk_state == NAU8825_XTALK_IMM) + nau8825_xtalk_measure(nau8825); +} + + static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_ENA_CTRL, 0x00ff }, { NAU8825_REG_IIC_ADDR_SET, 0x0 }, @@ -211,6 +585,7 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) case NAU8825_REG_RESET: case NAU8825_REG_IRQ_STATUS: case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_BIQ_CTRL ... NAU8825_REG_BIQ_COF10: case NAU8825_REG_IMM_RMS_L: case NAU8825_REG_IMM_RMS_R: case NAU8825_REG_I2C_DEVICE_ID: @@ -741,6 +1116,10 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); mic_detected = (jack_status_reg >> 10) & 3; + if (mic_detected == 0x3) + nau8825->high_imped = 1; + else + nau8825->high_imped = 0; switch (mic_detected) { case 0: @@ -836,6 +1215,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); + if (!nau8825->high_imped) { + nau8825->xtalk_state = NAU8825_XTALK_DONE; + schedule_work(&nau8825->xtalk_work); + } } else { dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); nau8825_eject_jack(nau8825); @@ -843,6 +1226,9 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } else if (active_irq & NAU8825_IMPEDANCE_MEAS_IRQ) { + schedule_work(&nau8825->xtalk_work); + clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; } if (!clear_irq) @@ -901,6 +1287,11 @@ static void nau8825_init_regs(struct nau8825 *nau8825) /* Latch IIC LSB value */ regmap_write(regmap, NAU8825_REG_IIC_ADDR_SET, 0x0001); + /* Config channel */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_MASK, NAU8825_DACL_CH_SEL_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_MASK, NAU8825_DACR_CH_SEL_R); /* Enable Bias/Vmid */ regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); @@ -1007,6 +1398,17 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); nau8825_restart_jack_detection(nau8825->regmap); + INIT_WORK(&nau8825->xtalk_work, nau8825_xtalk_work); + + return 0; +} + +static int nau8825_codec_remove(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + cancel_work_sync(&nau8825->xtalk_work); + return 0; } @@ -1326,6 +1728,7 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, + .remove = nau8825_codec_remove, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 029134a..808474d 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -98,7 +98,13 @@ #define NAU8825_ENABLE_DACR_SFT 10 #define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) #define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_DACL (1 << NAU8825_ENABLE_DACL_SFT) #define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_ADC (1 << NAU8825_ENABLE_ADC_SFT) +#define NAU8825_ENABLE_ADC_CLK_SFT 7 +#define NAU8825_ENABLE_ADC_CLK (1 << NAU8825_ENABLE_ADC_CLK_SFT) +#define NAU8825_ENABLE_DAC_CLK_SFT 6 +#define NAU8825_ENABLE_DAC_CLK (1 << NAU8825_ENABLE_DAC_CLK_SFT) #define NAU8825_ENABLE_SAR_SFT 1 /* CLK_DIVIDER (0x3) */ @@ -153,6 +159,7 @@ /* INTERRUPT_MASK (0xf) */ #define NAU8825_IRQ_OUTPUT_EN (1 << 11) #define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_RMS_EN (1 << 8) #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) @@ -225,10 +232,13 @@ /* I2S_PCM_CTRL2 (0x1d) */ #define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */ +#define NAU8825_I2S_DRV_SFT 12 +#define NAU8825_I2S_DRV_MASK (0x3 << NAU8825_I2S_DRV_SFT) #define NAU8825_I2S_MS_SFT 3 #define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_BLK_DIV_MASK 0x7 /* ADC_RATE (0x2b) */ #define NAU8825_ADC_SYNC_DOWN_SFT 0 @@ -247,22 +257,66 @@ #define NAU8825_DAC_OVERSAMPLE_128 2 #define NAU8825_DAC_OVERSAMPLE_32 4 +/* ADC_DGAIN_CTRL (0x30) */ +#define NAU8825_ADC_DAC_ST1_SFT 12 +#define NAU8825_ADC_DAC_ST1_MASK 0xf +#define NAU8825_ADC_DAC_ST2_SFT 8 +#define NAU8825_ADC_DAC_ST2_MASK 0xf +#define NAU8825_ADC_DIG_VOL_MASK 0xff + /* MUTE_CTRL (0x31) */ #define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) #define NAU8825_DAC_SOFT_MUTE (1 << 9) /* HSVOL_CTRL (0x32) */ #define NAU8825_HP_MUTE (1 << 15) +#define NAU8825_HP_MUTE_AUTO (1 << 14) +#define NAU8825_HPL_MUTE (1 << 13) +#define NAU8825_HPR_MUTE (1 << 12) +#define NAU8825_HPL_VOL_SFT 6 +#define NAU8825_HPL_VOL_MASK (0x3f << NAU8825_HPL_VOL_SFT) +#define NAU8825_HPR_VOL_SFT 0 +#define NAU8825_HPR_VOL_MASK (0x3f << NAU8825_HPR_VOL_SFT) /* DACL_CTRL (0x33) */ #define NAU8825_DACL_CH_SEL_SFT 9 +#define NAU8825_DACL_CH_SEL_MASK (0x1 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_L (0x0 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_SEL_R (0x1 << NAU8825_DACL_CH_SEL_SFT) /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 +#define NAU8825_DACR_CH_SEL_MASK (0x1 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_L (0x0 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_SEL_R (0x1 << NAU8825_DACR_CH_SEL_SFT) + +/* IMM_MODE_CTRL (0x4C) */ +#define NAU8825_IMM_THD_SFT 8 +#define NAU8825_IMM_THD_MASK (0x3f << NAU8825_IMM_THD_SFT) +#define NAU8825_IMM_CYC_SFT 4 +#define NAU8825_IMM_CYC_MASK (0x3 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_1024 (0x0 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_2048 (0x1 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_4096 (0x2 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_8192 (0x3 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_EN (1 << 3) +#define NAU8825_IMM_DAC_SRC_MASK 0x7 +#define NAU8825_IMM_DAC_SRC_BIQ 0x0 +#define NAU8825_IMM_DAC_SRC_DRC 0x1 +#define NAU8825_IMM_DAC_SRC_MIX 0x2 +#define NAU8825_IMM_DAC_SRC_SIN 0x3 /* CLASSG_CTRL (0x50) */ #define NAU8825_CLASSG_TIMER_SFT 8 #define NAU8825_CLASSG_TIMER_MASK (0x3f << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_1ms (0x1 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_2ms (0x2 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_8ms (0x4 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_16ms (0x8 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_32ms (0x10 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_64ms (0x20 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_LDAC_EN (0x1 << 2) +#define NAU8825_CLASSG_RDAC_EN (0x1 << 1) #define NAU8825_CLASSG_EN (1 << 0) /* I2C_DEVICE_ID (0x58) */ @@ -271,7 +325,12 @@ #define NAU8825_SOFTWARE_ID_NAU8825 0x0 /* BIAS_ADJ (0x66) */ -#define NAU8825_BIAS_TESTDAC_EN (0x3 << 8) +#define NAU8825_BIAS_HPR_IMP (1 << 15) +#define NAU8825_BIAS_HPL_IMP (1 << 14) +#define NAU8825_BIAS_TESTDAC_SFT 8 +#define NAU8825_BIAS_TESTDAC_EN (0x3 << NAU8825_BIAS_TESTDAC_SFT) +#define NAU8825_BIAS_TESTDACR_EN (0x2 << NAU8825_BIAS_TESTDAC_SFT) +#define NAU8825_BIAS_TESTDACL_EN (0x1 << NAU8825_BIAS_TESTDAC_SFT) #define NAU8825_BIAS_VMID (1 << 6) #define NAU8825_BIAS_VMID_SEL_SFT 4 #define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) @@ -290,6 +349,10 @@ #define NAU8825_POWERUP_ADCL (1 << 6) /* RDAC (0x73) */ +#define NAU8825_RDAC_EN_SFT 12 +#define NAU8825_RDAC_EN (0x3 << NAU8825_RDAC_EN_SFT) +#define NAU8825_RDAC_CLK_EN_SFT 8 +#define NAU8825_RDAC_CLK_EN (0x3 << NAU8825_RDAC_CLK_EN_SFT) #define NAU8825_RDAC_CLK_DELAY_SFT 4 #define NAU8825_RDAC_CLK_DELAY_MASK (0x7 << NAU8825_RDAC_CLK_DELAY_SFT) #define NAU8825_RDAC_VREF_SFT 2 @@ -333,12 +396,21 @@ enum { NAU8825_CLK_FLL_FS, }; +/* Crosstalk detection state */ +enum { + NAU8825_XTALK_HPR_R2L = 0, + NAU8825_XTALK_HPL_R2L, + NAU8825_XTALK_IMM, + NAU8825_XTALK_DONE, +}; + struct nau8825 { struct device *dev; struct regmap *regmap; struct snd_soc_dapm_context *dapm; struct snd_soc_jack *jack; struct clk *mclk; + struct work_struct xtalk_work; int irq; int mclk_freq; /* 0 - mclk is disabled */ int button_pressed; @@ -357,6 +429,9 @@ struct nau8825 { int key_debounce; int jack_insert_debounce; int jack_eject_debounce; + int xtalk_state; + int imp_rms[NAU8825_XTALK_IMM]; + int high_imped; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

2 4

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Skylake: Fix module removal
by Vinod Koul 16 Mar '16

16 Mar '16

Skylake driver were crashing on module removal when we run "alsa reload". The first issues was caused by component framework regression which Russell fixed and in in 4.5. In driver there were still some missing order which needs to be fixed. This series first update the list parsing in core, then ensure proper freeup of code objects, and then ensuring driver does proper cleanup and i915 dependency Changes in v2: - Add acked by Takashi - Remove codec destructor wrapper Vinod Koul (6): ALSA: hda: use list macro for parsing on cleanup ASoC: Intel: Skylake: free codec objects on removal ASoC: Intel: Skylake: Freeup properly on skl_dsp_free ASoC: Intel: Skylake: Unmap the address last ASoC: Intel: Skylake: Call i915 exit last ASoC: Intel: Skylake: remove call to pci_dev_put sound/hda/ext/hdac_ext_stream.c | 5 ++--- sound/soc/intel/skylake/skl-sst-dsp.c | 5 +++++ sound/soc/intel/skylake/skl.c | 17 ++++++++++------- 3 files changed, 17 insertions(+), 10 deletions(-) -- 1.9.1

3 15

[alsa-devel] [PATCH v4] ASoC: Intel: Skylake: remove call to pci_dev_put
by Vinod Koul 16 Mar '16

16 Mar '16

The PCI bus takes pci_dev_get() and pci_dev_put() is also there. So no need for drivers to invoke these. In SKL driver we were calling pci_dev_put() only which is not right, so remove this Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 72971dc55c52..07d9bc1cf3cb 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -724,7 +724,6 @@ static void skl_remove(struct pci_dev *pci) if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); - pci_dev_put(pci); /* codec removal, invoke bus_device_remove */ snd_hdac_ext_bus_device_remove(ebus); -- 1.9.1

1 0

[alsa-devel] [PATCH v3] ASoC: Intel: Skylake: remove call to pci_dev_put
by Vinod Koul 16 Mar '16

16 Mar '16

The PCI bus takes pci_dev_get() and pci_dev_put() is also there. So no need for drivers to invoke these. In SKL driver we were calling pci_dev_put() only which is not right, so remove this Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index b67f3bd0fccd..c337d06793ec 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -732,7 +732,6 @@ static void skl_remove(struct pci_dev *pci) if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); - pci_dev_put(pci); skl_codecs_remove(ebus); skl_platform_unregister(&pci->dev); skl_free_dsp(skl); -- 1.9.1

2 3

[alsa-devel] [patch] ALSA: mixart: silence an uninitialized variable warning
by Dan Carpenter 16 Mar '16

16 Mar '16

We could print the uninitialized value of "stat" in the error message. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index bc81b9f..25c0ddd 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -132,7 +132,7 @@ static int mixart_set_pipe_state(struct mixart_mgr *mgr, } if(start) { - u32 stat; + u32 stat = 0; group_state.pipe_count = 0; /* in case of start same command once again with pipe_count=0 */

3 2

Re: [alsa-devel] [media] sound/usb: Use Media Controller API to share media resources
by Dan Carpenter 16 Mar '16

16 Mar '16

Hello Shuah Khan, This is a semi-automatic email about new static checker warnings. The patch aebb2b89bff0: "[media] sound/usb: Use Media Controller API to share media resources" from Mar 2, 2016, leads to the following Smatch complaint: sound/usb/card.c:615 usb_audio_probe() error: we previously assumed 'quirk' could be null (see line 534) sound/usb/card.c 583 /* 584 * For devices with more than one control interface, we assume the 585 * first contains the audio controls. We might need a more specific 586 * check here in the future. 587 */ 588 if (!chip->ctrl_intf) 589 chip->ctrl_intf = alts; 590 591 chip->txfr_quirk = 0; 592 err = 1; /* continue */ 593 if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { ^^^^^ Old code assumes "quirk" can be NULL. 594 /* need some special handlings */ 595 err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk); 596 if (err < 0) 597 goto __error; 598 } 599 600 if (err > 0) { 601 /* create normal USB audio interfaces */ 602 err = snd_usb_create_streams(chip, ifnum); 603 if (err < 0) 604 goto __error; 605 err = snd_usb_create_mixer(chip, ifnum, ignore_ctl_error); 606 if (err < 0) 607 goto __error; 608 } 609 610 /* we are allowed to call snd_card_register() many times */ 611 err = snd_card_register(chip->card); 612 if (err < 0) 613 goto __error; 614 615 if (quirk->media_device) { ^^^^^^^^^^^^^^^^^^^ Patch adds an unchecked dereference. 616 /* don't want to fail when media_snd_device_create() fails */ 617 media_snd_device_create(chip, intf); regards, dan carpenter

2 1

[alsa-devel] [PATCH 1/2] sound/usb/media: use core routine to initialize media_device
by Mauro Carvalho Chehab 16 Mar '16

16 Mar '16

The media core has now a routine to initialize media_device for USB devices. Use it, instead of doing its own logic, as it warrants that all USB drivers will behave the same. It also warrants that the device will get the same data, no matter if it was initialized initially via snd-usb-audio or via some other driver, like au0828. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- sound/usb/media.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index 93a50d01490c..44a5de91f748 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -263,19 +263,11 @@ int media_snd_device_create(struct snd_usb_audio *chip, mdev = media_device_get_devres(&usbdev->dev); if (!mdev) return -ENOMEM; - if (!mdev->dev) { - /* register media device */ - mdev->dev = &usbdev->dev; - if (usbdev->product) - strlcpy(mdev->model, usbdev->product, - sizeof(mdev->model)); - if (usbdev->serial) - strlcpy(mdev->serial, usbdev->serial, - sizeof(mdev->serial)); - strcpy(mdev->bus_info, usbdev->devpath); - mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); - media_device_init(mdev); - } + + /* Initialize media device */ + if (!mdev->dev) + media_device_usb_init(mdev, usbdev, NULL); + if (!media_devnode_is_registered(&mdev->devnode)) { ret = media_device_register(mdev); if (ret) { -- 2.5.0

1 0

[alsa-devel] [PATCH] media: add GFP flag to media_*() that could get called in atomic context
by Shuah Khan 16 Mar '16

16 Mar '16

Add GFP flags to media_create_pad_link(), media_create_intf_link(), media_devnode_create(), and media_add_link() that could get called in atomic context to allow callers to pass in the right flags for memory allocation. tree-wide driver changes for media_*() GFP flags change: Change drivers to add gfpflags to interffaces, media_create_pad_link(), media_create_intf_link() and media_devnode_create(). Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> Suggested-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- Ran through kbuild-all compile testing. Tested the changes in Win-TV HVR-950Q device drivers/media/dvb-core/dvbdev.c | 26 +++++++----- drivers/media/i2c/s5c73m3/s5c73m3-core.c | 6 ++- drivers/media/i2c/s5k5baf.c | 3 +- drivers/media/i2c/smiapp/smiapp-core.c | 3 +- drivers/media/i2c/tvp5150.c | 3 +- drivers/media/media-entity.c | 30 ++++++++------ drivers/media/platform/exynos4-is/media-dev.c | 19 +++++---- drivers/media/platform/omap3isp/isp.c | 47 ++++++++++++++-------- drivers/media/platform/s3c-camif/camif-core.c | 4 +- drivers/media/platform/vsp1/vsp1_drm.c | 6 +-- drivers/media/platform/vsp1/vsp1_drv.c | 9 +++-- drivers/media/platform/xilinx/xilinx-vipp.c | 4 +- drivers/media/usb/au0828/au0828-core.c | 3 +- drivers/media/usb/uvc/uvc_entity.c | 2 +- drivers/media/v4l2-core/v4l2-dev.c | 5 ++- drivers/media/v4l2-core/v4l2-device.c | 3 +- drivers/media/v4l2-core/v4l2-mc.c | 25 +++++++----- drivers/staging/media/davinci_vpfe/dm365_ipipeif.c | 3 +- drivers/staging/media/davinci_vpfe/dm365_isif.c | 2 +- drivers/staging/media/davinci_vpfe/dm365_resizer.c | 10 +++-- .../staging/media/davinci_vpfe/vpfe_mc_capture.c | 10 ++--- drivers/staging/media/omap4iss/iss.c | 17 +++++--- drivers/staging/media/omap4iss/iss_csi2.c | 6 ++- drivers/staging/media/omap4iss/iss_ipipeif.c | 3 +- drivers/staging/media/omap4iss/iss_resizer.c | 3 +- include/media/media-entity.h | 9 +++-- sound/usb/media.c | 15 ++++--- 27 files changed, 170 insertions(+), 106 deletions(-) diff --git a/drivers/media/dvb-core/dvbdev.c b/drivers/media/dvb-core/dvbdev.c index e1684c5..57f3e1e 100644 --- a/drivers/media/dvb-core/dvbdev.c +++ b/drivers/media/dvb-core/dvbdev.c @@ -399,7 +399,8 @@ static int dvb_register_media_device(struct dvb_device *dvbdev, dvbdev->intf_devnode = media_devnode_create(dvbdev->adapter->mdev, intf_type, 0, - DVB_MAJOR, minor); + DVB_MAJOR, minor, + GFP_KERNEL); if (!dvbdev->intf_devnode) return -ENOMEM; @@ -416,7 +417,7 @@ static int dvb_register_media_device(struct dvb_device *dvbdev, return 0; link = media_create_intf_link(dvbdev->entity, &dvbdev->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (!link) return -ENOMEM; #endif @@ -558,7 +559,8 @@ static int dvb_create_io_intf_links(struct dvb_adapter *adap, if (strncmp(entity->name, name, strlen(name))) continue; link = media_create_intf_link(entity, intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) return -ENOMEM; } @@ -680,7 +682,8 @@ int dvb_create_media_graph(struct dvb_adapter *adap, } if (demux && ca) { ret = media_create_pad_link(demux, 1, ca, - 0, MEDIA_LNK_FL_ENABLED); + 0, MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return -ENOMEM; } @@ -693,7 +696,8 @@ int dvb_create_media_graph(struct dvb_adapter *adap, strlen(DVR_TSOUT))) { ret = media_create_pad_link(demux, ++dvr_pad, - entity, 0, 0); + entity, 0, 0, + GFP_KERNEL); if (ret) return ret; } @@ -701,7 +705,8 @@ int dvb_create_media_graph(struct dvb_adapter *adap, strlen(DEMUX_TSOUT))) { ret = media_create_pad_link(demux, ++demux_pad, - entity, 0, 0); + entity, 0, 0, + GFP_KERNEL); if (ret) return ret; } @@ -713,14 +718,16 @@ int dvb_create_media_graph(struct dvb_adapter *adap, media_device_for_each_intf(intf, mdev) { if (intf->type == MEDIA_INTF_T_DVB_CA && ca) { link = media_create_intf_link(ca, intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) return -ENOMEM; } if (intf->type == MEDIA_INTF_T_DVB_FE && tuner) { link = media_create_intf_link(tuner, intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) return -ENOMEM; } @@ -732,7 +739,8 @@ int dvb_create_media_graph(struct dvb_adapter *adap, */ if (intf->type == MEDIA_INTF_T_DVB_DVR && demux) { link = media_create_intf_link(demux, intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) return -ENOMEM; } diff --git a/drivers/media/i2c/s5c73m3/s5c73m3-core.c b/drivers/media/i2c/s5c73m3/s5c73m3-core.c index 08af58f..dd994f1 100644 --- a/drivers/media/i2c/s5c73m3/s5c73m3-core.c +++ b/drivers/media/i2c/s5c73m3/s5c73m3-core.c @@ -1484,11 +1484,13 @@ static int s5c73m3_oif_registered(struct v4l2_subdev *sd) ret = media_create_pad_link(&state->sensor_sd.entity, S5C73M3_ISP_PAD, &state->oif_sd.entity, OIF_ISP_PAD, - MEDIA_LNK_FL_IMMUTABLE | MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_IMMUTABLE | MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); ret = media_create_pad_link(&state->sensor_sd.entity, S5C73M3_JPEG_PAD, &state->oif_sd.entity, OIF_JPEG_PAD, - MEDIA_LNK_FL_IMMUTABLE | MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_IMMUTABLE | MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); return ret; } diff --git a/drivers/media/i2c/s5k5baf.c b/drivers/media/i2c/s5k5baf.c index db82ed0..3692bec 100644 --- a/drivers/media/i2c/s5k5baf.c +++ b/drivers/media/i2c/s5k5baf.c @@ -1759,7 +1759,8 @@ static int s5k5baf_registered(struct v4l2_subdev *sd) ret = media_create_pad_link(&state->cis_sd.entity, PAD_CIS, &state->sd.entity, PAD_CIS, MEDIA_LNK_FL_IMMUTABLE | - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); return ret; } diff --git a/drivers/media/i2c/smiapp/smiapp-core.c b/drivers/media/i2c/smiapp/smiapp-core.c index a215efe..8c2e9ad 100644 --- a/drivers/media/i2c/smiapp/smiapp-core.c +++ b/drivers/media/i2c/smiapp/smiapp-core.c @@ -2508,7 +2508,8 @@ static int smiapp_register_subdevs(struct smiapp_sensor *sensor) &last->sd.entity, last->sink_pad, MEDIA_LNK_FL_ENABLED | - MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_IMMUTABLE, + GFP_KERNEL); if (rval) { dev_err(&client->dev, "media_create_pad_link failed\n"); diff --git a/drivers/media/i2c/tvp5150.c b/drivers/media/i2c/tvp5150.c index ff18444..fd99683 100644 --- a/drivers/media/i2c/tvp5150.c +++ b/drivers/media/i2c/tvp5150.c @@ -1197,7 +1197,8 @@ static int tvp5150_registered_async(struct v4l2_subdev *sd) return ret; ret = media_create_pad_link(input, 0, &sd->entity, - DEMOD_PAD_IF_INPUT, 0); + DEMOD_PAD_IF_INPUT, 0, + GFP_KERNEL); if (ret < 0) { media_device_unregister_entity(input); return ret; diff --git a/drivers/media/media-entity.c b/drivers/media/media-entity.c index 66a5392..115435b 100644 --- a/drivers/media/media-entity.c +++ b/drivers/media/media-entity.c @@ -566,14 +566,14 @@ EXPORT_SYMBOL_GPL(media_entity_put); * Links management */ -static struct media_link *media_add_link(struct list_head *head) +static struct media_link *media_add_link(struct list_head *head, gfp_t gfpflags) { struct media_link *link; if (in_atomic()) dump_stack(); - link = kzalloc(sizeof(*link), GFP_KERNEL); + link = kzalloc(sizeof(*link), gfpflags); if (link == NULL) return NULL; @@ -615,7 +615,8 @@ static void __media_entity_remove_link(struct media_entity *entity, int media_create_pad_link(struct media_entity *source, u16 source_pad, - struct media_entity *sink, u16 sink_pad, u32 flags) + struct media_entity *sink, u16 sink_pad, u32 flags, + gfp_t gfpflags) { struct media_link *link; struct media_link *backlink; @@ -624,7 +625,7 @@ media_create_pad_link(struct media_entity *source, u16 source_pad, BUG_ON(source_pad >= source->num_pads); BUG_ON(sink_pad >= sink->num_pads); - link = media_add_link(&source->links); + link = media_add_link(&source->links, gfpflags); if (link == NULL) return -ENOMEM; @@ -639,7 +640,7 @@ media_create_pad_link(struct media_entity *source, u16 source_pad, /* Create the backlink. Backlinks are used to help graph traversal and * are not reported to userspace. */ - backlink = media_add_link(&sink->links); + backlink = media_add_link(&sink->links, gfpflags); if (backlink == NULL) { __media_entity_remove_link(source, link); return -ENOMEM; @@ -682,7 +683,7 @@ int media_create_pad_links(const struct media_device *mdev, /* Trivial case: 1:1 relation */ if (source && sink) return media_create_pad_link(source, source_pad, - sink, sink_pad, flags); + sink, sink_pad, flags, GFP_KERNEL); /* Worse case scenario: n:n relation */ if (!source && !sink) { @@ -696,7 +697,7 @@ int media_create_pad_links(const struct media_device *mdev, continue; ret = media_create_pad_link(source, source_pad, sink, sink_pad, - flags); + flags, GFP_KERNEL); if (ret) return ret; flags &= ~(MEDIA_LNK_FL_ENABLED | @@ -718,10 +719,12 @@ int media_create_pad_links(const struct media_device *mdev, if (source) ret = media_create_pad_link(source, source_pad, - entity, sink_pad, flags); + entity, sink_pad, flags, + GFP_KERNEL); else ret = media_create_pad_link(entity, source_pad, - sink, sink_pad, flags); + sink, sink_pad, flags, + GFP_KERNEL); if (ret) return ret; flags &= ~(MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE); @@ -890,14 +893,15 @@ static void media_interface_init(struct media_device *mdev, struct media_intf_devnode *media_devnode_create(struct media_device *mdev, u32 type, u32 flags, - u32 major, u32 minor) + u32 major, u32 minor, + gfp_t gfpflags) { struct media_intf_devnode *devnode; if (in_atomic()) dump_stack(); - devnode = kzalloc(sizeof(*devnode), GFP_KERNEL); + devnode = kzalloc(sizeof(*devnode), gfpflags); if (!devnode) return NULL; @@ -921,11 +925,11 @@ EXPORT_SYMBOL_GPL(media_devnode_remove); struct media_link *media_create_intf_link(struct media_entity *entity, struct media_interface *intf, - u32 flags) + u32 flags, gfp_t gfpflags) { struct media_link *link; - link = media_add_link(&intf->links); + link = media_add_link(&intf->links, gfpflags); if (link == NULL) return NULL; diff --git a/drivers/media/platform/exynos4-is/media-dev.c b/drivers/media/platform/exynos4-is/media-dev.c index feb521f..145d90e 100644 --- a/drivers/media/platform/exynos4-is/media-dev.c +++ b/drivers/media/platform/exynos4-is/media-dev.c @@ -798,7 +798,8 @@ static int __fimc_md_create_fimc_sink_links(struct fimc_md *fmd, sink = &fmd->fimc[i]->vid_cap.subdev.entity; ret = media_create_pad_link(source, pad, sink, - FIMC_SD_PAD_SINK_CAM, flags); + FIMC_SD_PAD_SINK_CAM, flags, + GFP_KERNEL); if (ret) return ret; @@ -818,7 +819,8 @@ static int __fimc_md_create_fimc_sink_links(struct fimc_md *fmd, sink = &fmd->fimc_lite[i]->subdev.entity; ret = media_create_pad_link(source, pad, sink, - FLITE_SD_PAD_SINK, 0); + FLITE_SD_PAD_SINK, 0, + GFP_KERNEL); if (ret) return ret; @@ -850,13 +852,13 @@ static int __fimc_md_create_flite_source_links(struct fimc_md *fmd) sink = &fimc->ve.vdev.entity; /* FIMC-LITE's subdev and video node */ ret = media_create_pad_link(source, FLITE_SD_PAD_SOURCE_DMA, - sink, 0, 0); + sink, 0, 0, GFP_KERNEL); if (ret) break; /* Link from FIMC-LITE to IS-ISP subdev */ sink = &fmd->fimc_is->isp.subdev.entity; ret = media_create_pad_link(source, FLITE_SD_PAD_SOURCE_ISP, - sink, 0, 0); + sink, 0, 0, GFP_KERNEL); if (ret) break; } @@ -880,7 +882,8 @@ static int __fimc_md_create_fimc_is_links(struct fimc_md *fmd) /* Link from FIMC-IS-ISP subdev to FIMC */ sink = &fmd->fimc[i]->vid_cap.subdev.entity; ret = media_create_pad_link(source, FIMC_ISP_SD_PAD_SRC_FIFO, - sink, FIMC_SD_PAD_SINK_FIFO, 0); + sink, FIMC_SD_PAD_SINK_FIFO, 0, + GFP_KERNEL); if (ret) return ret; } @@ -893,7 +896,7 @@ static int __fimc_md_create_fimc_is_links(struct fimc_md *fmd) return 0; return media_create_pad_link(source, FIMC_ISP_SD_PAD_SRC_DMA, - sink, 0, 0); + sink, 0, 0, GFP_KERNEL); } /** @@ -944,7 +947,7 @@ static int fimc_md_create_links(struct fimc_md *fmd) ret = media_create_pad_link(&sensor->entity, pad, &csis->entity, CSIS_PAD_SINK, MEDIA_LNK_FL_IMMUTABLE | - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; @@ -996,7 +999,7 @@ static int fimc_md_create_links(struct fimc_md *fmd) sink = &fmd->fimc[i]->vid_cap.ve.vdev.entity; ret = media_create_pad_link(source, FIMC_SD_PAD_SOURCE, - sink, 0, flags); + sink, 0, flags, GFP_KERNEL); if (ret) break; } diff --git a/drivers/media/platform/omap3isp/isp.c b/drivers/media/platform/omap3isp/isp.c index 5d54e2c..f8d9c84 100644 --- a/drivers/media/platform/omap3isp/isp.c +++ b/drivers/media/platform/omap3isp/isp.c @@ -1669,7 +1669,7 @@ static int isp_link_entity( return -EINVAL; } - return media_create_pad_link(entity, i, input, pad, flags); + return media_create_pad_link(entity, i, input, pad, flags, GFP_KERNEL); } static int isp_register_entities(struct isp_device *isp) @@ -1748,43 +1748,50 @@ static int isp_create_links(struct isp_device *isp) /* Create links between entities and video nodes. */ ret = media_create_pad_link( &isp->isp_csi2a.subdev.entity, CSI2_PAD_SOURCE, - &isp->isp_csi2a.video_out.video.entity, 0, 0); + &isp->isp_csi2a.video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccp2.video_in.video.entity, 0, - &isp->isp_ccp2.subdev.entity, CCP2_PAD_SINK, 0); + &isp->isp_ccp2.subdev.entity, CCP2_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_OF, - &isp->isp_ccdc.video_out.video.entity, 0, 0); + &isp->isp_ccdc.video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_prev.video_in.video.entity, 0, - &isp->isp_prev.subdev.entity, PREV_PAD_SINK, 0); + &isp->isp_prev.subdev.entity, PREV_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_prev.subdev.entity, PREV_PAD_SOURCE, - &isp->isp_prev.video_out.video.entity, 0, 0); + &isp->isp_prev.video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_res.video_in.video.entity, 0, - &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0); + &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_res.subdev.entity, RESZ_PAD_SOURCE, - &isp->isp_res.video_out.video.entity, 0, 0); + &isp->isp_res.video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; @@ -1792,52 +1799,60 @@ static int isp_create_links(struct isp_device *isp) /* Create links between entities. */ ret = media_create_pad_link( &isp->isp_csi2a.subdev.entity, CSI2_PAD_SOURCE, - &isp->isp_ccdc.subdev.entity, CCDC_PAD_SINK, 0); + &isp->isp_ccdc.subdev.entity, CCDC_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccp2.subdev.entity, CCP2_PAD_SOURCE, - &isp->isp_ccdc.subdev.entity, CCDC_PAD_SINK, 0); + &isp->isp_ccdc.subdev.entity, CCDC_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_VP, - &isp->isp_prev.subdev.entity, PREV_PAD_SINK, 0); + &isp->isp_prev.subdev.entity, PREV_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_OF, - &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0); + &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_prev.subdev.entity, PREV_PAD_SOURCE, - &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0); + &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_VP, &isp->isp_aewb.subdev.entity, 0, - MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_VP, &isp->isp_af.subdev.entity, 0, - MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_VP, &isp->isp_hist.subdev.entity, 0, - MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE, + GFP_KERNEL); if (ret < 0) return ret; diff --git a/drivers/media/platform/s3c-camif/camif-core.c b/drivers/media/platform/s3c-camif/camif-core.c index 0b44b9a..a49b4c3 100644 --- a/drivers/media/platform/s3c-camif/camif-core.c +++ b/drivers/media/platform/s3c-camif/camif-core.c @@ -266,7 +266,7 @@ static int camif_create_media_links(struct camif_dev *camif) ret = media_create_pad_link(&camif->sensor.sd->entity, 0, &camif->subdev.entity, CAMIF_SD_PAD_SINK, MEDIA_LNK_FL_IMMUTABLE | - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; @@ -274,7 +274,7 @@ static int camif_create_media_links(struct camif_dev *camif) ret = media_create_pad_link(&camif->subdev.entity, i, &camif->vp[i - 1].vdev.entity, 0, MEDIA_LNK_FL_IMMUTABLE | - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); } return ret; diff --git a/drivers/media/platform/vsp1/vsp1_drm.c b/drivers/media/platform/vsp1/vsp1_drm.c index 021fe57..f868dec 100644 --- a/drivers/media/platform/vsp1/vsp1_drm.c +++ b/drivers/media/platform/vsp1/vsp1_drm.c @@ -525,7 +525,7 @@ int vsp1_drm_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&rpf->entity.subdev.entity, RWPF_PAD_SOURCE, &vsp1->bru->entity.subdev.entity, - i, flags); + i, flags, GFP_KERNEL); if (ret < 0) return ret; @@ -536,7 +536,7 @@ int vsp1_drm_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&vsp1->bru->entity.subdev.entity, vsp1->bru->entity.source_pad, &vsp1->wpf[0]->entity.subdev.entity, - RWPF_PAD_SINK, flags); + RWPF_PAD_SINK, flags, GFP_KERNEL); if (ret < 0) return ret; @@ -546,7 +546,7 @@ int vsp1_drm_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&vsp1->wpf[0]->entity.subdev.entity, RWPF_PAD_SOURCE, &vsp1->lif->entity.subdev.entity, - LIF_PAD_SINK, flags); + LIF_PAD_SINK, flags, GFP_KERNEL); if (ret < 0) return ret; diff --git a/drivers/media/platform/vsp1/vsp1_drv.c b/drivers/media/platform/vsp1/vsp1_drv.c index 25750a0..7295587 100644 --- a/drivers/media/platform/vsp1/vsp1_drv.c +++ b/drivers/media/platform/vsp1/vsp1_drv.c @@ -125,7 +125,8 @@ static int vsp1_create_sink_links(struct vsp1_device *vsp1, ret = media_create_pad_link(&source->subdev.entity, source->source_pad, - entity, pad, flags); + entity, pad, flags, + GFP_KERNEL); if (ret < 0) return ret; @@ -157,7 +158,7 @@ static int vsp1_uapi_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&vsp1->wpf[0]->entity.subdev.entity, RWPF_PAD_SOURCE, &vsp1->lif->entity.subdev.entity, - LIF_PAD_SINK, 0); + LIF_PAD_SINK, 0, GFP_KERNEL); if (ret < 0) return ret; } @@ -169,7 +170,7 @@ static int vsp1_uapi_create_links(struct vsp1_device *vsp1) &rpf->entity.subdev.entity, RWPF_PAD_SINK, MEDIA_LNK_FL_ENABLED | - MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_IMMUTABLE, GFP_KERNEL); if (ret < 0) return ret; } @@ -188,7 +189,7 @@ static int vsp1_uapi_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&wpf->entity.subdev.entity, RWPF_PAD_SOURCE, &wpf->video->video.entity, 0, - flags); + flags, GFP_KERNEL); if (ret < 0) return ret; } diff --git a/drivers/media/platform/xilinx/xilinx-vipp.c b/drivers/media/platform/xilinx/xilinx-vipp.c index e795a45..b444db4 100644 --- a/drivers/media/platform/xilinx/xilinx-vipp.c +++ b/drivers/media/platform/xilinx/xilinx-vipp.c @@ -158,7 +158,7 @@ static int xvip_graph_build_one(struct xvip_composite_device *xdev, ret = media_create_pad_link(local, local_pad->index, remote, remote_pad->index, - link_flags); + link_flags, GFP_KERNEL); if (ret < 0) { dev_err(xdev->dev, "failed to create %s:%u -> %s:%u link\n", @@ -272,7 +272,7 @@ static int xvip_graph_build_dma(struct xvip_composite_device *xdev) ret = media_create_pad_link(source, source_pad->index, sink, sink_pad->index, - link_flags); + link_flags, GFP_KERNEL); if (ret < 0) { dev_err(xdev->dev, "failed to create %s:%u -> %s:%u link\n", diff --git a/drivers/media/usb/au0828/au0828-core.c b/drivers/media/usb/au0828/au0828-core.c index 552ac58..de9ab11 100644 --- a/drivers/media/usb/au0828/au0828-core.c +++ b/drivers/media/usb/au0828/au0828-core.c @@ -251,7 +251,8 @@ create_link: ret = media_create_pad_link(decoder, DEMOD_PAD_AUDIO_OUT, mixer, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) dev_err(&dev->usbdev->dev, "Mixer Pad Link Create Error: %d\n", ret); diff --git a/drivers/media/usb/uvc/uvc_entity.c b/drivers/media/usb/uvc/uvc_entity.c index ac386bb..d9555dc 100644 --- a/drivers/media/usb/uvc/uvc_entity.c +++ b/drivers/media/usb/uvc/uvc_entity.c @@ -53,7 +53,7 @@ static int uvc_mc_create_links(struct uvc_video_chain *chain, remote_pad = remote->num_pads - 1; ret = media_create_pad_link(source, remote_pad, - sink, i, flags); + sink, i, flags, GFP_KERNEL); if (ret < 0) return ret; } diff --git a/drivers/media/v4l2-core/v4l2-dev.c b/drivers/media/v4l2-core/v4l2-dev.c index d8e5994..b0f53a2 100644 --- a/drivers/media/v4l2-core/v4l2-dev.c +++ b/drivers/media/v4l2-core/v4l2-dev.c @@ -786,7 +786,7 @@ static int video_register_media_controller(struct video_device *vdev, int type) vdev->intf_devnode = media_devnode_create(vdev->v4l2_dev->mdev, intf_type, 0, VIDEO_MAJOR, - vdev->minor); + vdev->minor, GFP_KERNEL); if (!vdev->intf_devnode) { media_device_unregister_entity(&vdev->entity); return -ENOMEM; @@ -797,7 +797,8 @@ static int video_register_media_controller(struct video_device *vdev, int type) link = media_create_intf_link(&vdev->entity, &vdev->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) { media_devnode_remove(vdev->intf_devnode); media_device_unregister_entity(&vdev->entity); diff --git a/drivers/media/v4l2-core/v4l2-device.c b/drivers/media/v4l2-core/v4l2-device.c index 06fa5f1..c7514e7 100644 --- a/drivers/media/v4l2-core/v4l2-device.c +++ b/drivers/media/v4l2-core/v4l2-device.c @@ -266,7 +266,8 @@ int v4l2_device_register_subdev_nodes(struct v4l2_device *v4l2_dev) link = media_create_intf_link(&sd->entity, &vdev->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) { err = -ENOMEM; goto clean_up; diff --git a/drivers/media/v4l2-core/v4l2-mc.c b/drivers/media/v4l2-core/v4l2-mc.c index 40ee864..b1652f9 100644 --- a/drivers/media/v4l2-core/v4l2-mc.c +++ b/drivers/media/v4l2-core/v4l2-mc.c @@ -92,7 +92,8 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) continue; ret = media_create_pad_link(entity, 0, io_v4l, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; } @@ -110,18 +111,21 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) ret = media_create_pad_link(tuner, TUNER_PAD_OUTPUT, if_vid, IF_VID_DEC_PAD_IF_INPUT, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; ret = media_create_pad_link(if_vid, IF_VID_DEC_PAD_OUT, decoder, DEMOD_PAD_IF_INPUT, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; } else { ret = media_create_pad_link(tuner, TUNER_PAD_OUTPUT, decoder, DEMOD_PAD_IF_INPUT, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; } @@ -130,7 +134,8 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) ret = media_create_pad_link(tuner, TUNER_PAD_AUD_OUT, if_aud, IF_AUD_DEC_PAD_IF_INPUT, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; } else { @@ -143,7 +148,7 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) if (io_v4l) { ret = media_create_pad_link(decoder, DEMOD_PAD_VID_OUT, io_v4l, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; } @@ -151,7 +156,7 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) if (io_swradio) { ret = media_create_pad_link(decoder, DEMOD_PAD_VID_OUT, io_swradio, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; } @@ -159,7 +164,7 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) if (io_vbi) { ret = media_create_pad_link(decoder, DEMOD_PAD_VBI_OUT, io_vbi, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; } @@ -174,13 +179,13 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) ret = media_create_pad_link(entity, 0, tuner, TUNER_PAD_RF_INPUT, - flags); + flags, GFP_KERNEL); break; case MEDIA_ENT_F_CONN_SVIDEO: case MEDIA_ENT_F_CONN_COMPOSITE: ret = media_create_pad_link(entity, 0, decoder, DEMOD_PAD_IF_INPUT, - flags); + flags, GFP_KERNEL); break; default: continue; diff --git a/drivers/staging/media/davinci_vpfe/dm365_ipipeif.c b/drivers/staging/media/davinci_vpfe/dm365_ipipeif.c index 633d645..d1442b9 100644 --- a/drivers/staging/media/davinci_vpfe/dm365_ipipeif.c +++ b/drivers/staging/media/davinci_vpfe/dm365_ipipeif.c @@ -977,7 +977,8 @@ vpfe_ipipeif_register_entities(struct vpfe_ipipeif_device *ipipeif, flags = 0; ret = media_create_pad_link(&ipipeif->video_in.video_dev.entity, 0, - &ipipeif->subdev.entity, 0, flags); + &ipipeif->subdev.entity, 0, flags, + GFP_KERNEL); if (ret < 0) goto fail; diff --git a/drivers/staging/media/davinci_vpfe/dm365_isif.c b/drivers/staging/media/davinci_vpfe/dm365_isif.c index 9905789..2470655 100644 --- a/drivers/staging/media/davinci_vpfe/dm365_isif.c +++ b/drivers/staging/media/davinci_vpfe/dm365_isif.c @@ -1824,7 +1824,7 @@ int vpfe_isif_register_entities(struct vpfe_isif_device *isif, /* connect isif to video node */ ret = media_create_pad_link(&isif->subdev.entity, 1, &isif->video_out.video_dev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_create_link; return 0; diff --git a/drivers/staging/media/davinci_vpfe/dm365_resizer.c b/drivers/staging/media/davinci_vpfe/dm365_resizer.c index a91395c..067f699 100644 --- a/drivers/staging/media/davinci_vpfe/dm365_resizer.c +++ b/drivers/staging/media/davinci_vpfe/dm365_resizer.c @@ -1833,26 +1833,28 @@ int vpfe_resizer_register_entities(struct vpfe_resizer_device *resizer, /* create link between Resizer Crop----> Resizer A*/ ret = media_create_pad_link(&resizer->crop_resizer.subdev.entity, 1, &resizer->resizer_a.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_create_link; /* create link between Resizer Crop----> Resizer B*/ ret = media_create_pad_link(&resizer->crop_resizer.subdev.entity, 2, &resizer->resizer_b.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_create_link; /* create link between Resizer A ----> video out */ ret = media_create_pad_link(&resizer->resizer_a.subdev.entity, 1, - &resizer->resizer_a.video_out.video_dev.entity, 0, flags); + &resizer->resizer_a.video_out.video_dev.entity, 0, flags, + GFP_KERNEL); if (ret < 0) goto out_create_link; /* create link between Resizer B ----> video out */ ret = media_create_pad_link(&resizer->resizer_b.subdev.entity, 1, - &resizer->resizer_b.video_out.video_dev.entity, 0, flags); + &resizer->resizer_b.video_out.video_dev.entity, 0, flags, + GFP_KERNEL); if (ret < 0) goto out_create_link; diff --git a/drivers/staging/media/davinci_vpfe/vpfe_mc_capture.c b/drivers/staging/media/davinci_vpfe/vpfe_mc_capture.c index ec46f36..bee9f5c 100644 --- a/drivers/staging/media/davinci_vpfe/vpfe_mc_capture.c +++ b/drivers/staging/media/davinci_vpfe/vpfe_mc_capture.c @@ -447,32 +447,32 @@ static int vpfe_register_entities(struct vpfe_device *vpfe_dev) if (vpfe_dev->sd[i]->entity.num_pads) { ret = media_create_pad_link(&vpfe_dev->sd[i]->entity, 0, &vpfe_dev->vpfe_isif.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; } ret = media_create_pad_link(&vpfe_dev->vpfe_isif.subdev.entity, 1, &vpfe_dev->vpfe_ipipeif.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; ret = media_create_pad_link(&vpfe_dev->vpfe_ipipeif.subdev.entity, 1, &vpfe_dev->vpfe_ipipe.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; ret = media_create_pad_link(&vpfe_dev->vpfe_ipipe.subdev.entity, 1, &vpfe_dev->vpfe_resizer.crop_resizer.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; ret = media_create_pad_link(&vpfe_dev->vpfe_ipipeif.subdev.entity, 1, &vpfe_dev->vpfe_resizer.crop_resizer.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; diff --git a/drivers/staging/media/omap4iss/iss.c b/drivers/staging/media/omap4iss/iss.c index fb80d2b..933c8a7 100644 --- a/drivers/staging/media/omap4iss/iss.c +++ b/drivers/staging/media/omap4iss/iss.c @@ -1065,7 +1065,7 @@ static int iss_register_entities(struct iss_device *iss) } ret = media_create_pad_link(&sensor->entity, 0, input, pad, - flags); + flags, GFP_KERNEL); if (ret < 0) goto done; } @@ -1110,31 +1110,36 @@ static int iss_create_links(struct iss_device *iss) /* Connect the submodules. */ ret = media_create_pad_link( &iss->csi2a.subdev.entity, CSI2_PAD_SOURCE, - &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SINK, 0); + &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &iss->csi2b.subdev.entity, CSI2_PAD_SOURCE, - &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SINK, 0); + &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SOURCE_VP, - &iss->resizer.subdev.entity, RESIZER_PAD_SINK, 0); + &iss->resizer.subdev.entity, RESIZER_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SOURCE_VP, - &iss->ipipe.subdev.entity, IPIPE_PAD_SINK, 0); + &iss->ipipe.subdev.entity, IPIPE_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &iss->ipipe.subdev.entity, IPIPE_PAD_SOURCE_VP, - &iss->resizer.subdev.entity, RESIZER_PAD_SINK, 0); + &iss->resizer.subdev.entity, RESIZER_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; diff --git a/drivers/staging/media/omap4iss/iss_csi2.c b/drivers/staging/media/omap4iss/iss_csi2.c index aaca39d..4b40132 100644 --- a/drivers/staging/media/omap4iss/iss_csi2.c +++ b/drivers/staging/media/omap4iss/iss_csi2.c @@ -1352,13 +1352,15 @@ int omap4iss_csi2_create_links(struct iss_device *iss) /* Connect the CSI2a subdev to the video node. */ ret = media_create_pad_link(&csi2a->subdev.entity, CSI2_PAD_SOURCE, - &csi2a->video_out.video.entity, 0, 0); + &csi2a->video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; /* Connect the CSI2b subdev to the video node. */ ret = media_create_pad_link(&csi2b->subdev.entity, CSI2_PAD_SOURCE, - &csi2b->video_out.video.entity, 0, 0); + &csi2b->video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; diff --git a/drivers/staging/media/omap4iss/iss_ipipeif.c b/drivers/staging/media/omap4iss/iss_ipipeif.c index 23de833..2f124b0 100644 --- a/drivers/staging/media/omap4iss/iss_ipipeif.c +++ b/drivers/staging/media/omap4iss/iss_ipipeif.c @@ -827,7 +827,8 @@ int omap4iss_ipipeif_create_links(struct iss_device *iss) /* Connect the IPIPEIF subdev to the video node. */ return media_create_pad_link(&ipipeif->subdev.entity, IPIPEIF_PAD_SOURCE_ISIF_SF, - &ipipeif->video_out.video.entity, 0, 0); + &ipipeif->video_out.video.entity, 0, 0, + GFP_KERNEL); } /* diff --git a/drivers/staging/media/omap4iss/iss_resizer.c b/drivers/staging/media/omap4iss/iss_resizer.c index f1d352c..8d5766a 100644 --- a/drivers/staging/media/omap4iss/iss_resizer.c +++ b/drivers/staging/media/omap4iss/iss_resizer.c @@ -869,7 +869,8 @@ int omap4iss_resizer_create_links(struct iss_device *iss) /* Connect the RESIZER subdev to the video node. */ return media_create_pad_link(&resizer->subdev.entity, RESIZER_PAD_SOURCE_MEM, - &resizer->video_out.video.entity, 0, 0); + &resizer->video_out.video.entity, 0, 0, + GFP_KERNEL); } /* diff --git a/include/media/media-entity.h b/include/media/media-entity.h index 6dc9e4e..0665c2d 100644 --- a/include/media/media-entity.h +++ b/include/media/media-entity.h @@ -595,6 +595,7 @@ static inline void media_entity_cleanup(struct media_entity *entity) {}; * @sink: pointer to &media_entity of the sink pad. * @sink_pad: number of the sink pad in the pads array. * @flags: Link flags, as defined in include/uapi/linux/media.h. + * @gfpflags: Get Free Page (GFP) flags, as defined in include/linux/gfp.h * * Valid values for flags: * A %MEDIA_LNK_FL_ENABLED flag indicates that the link is enabled and can be @@ -613,7 +614,7 @@ static inline void media_entity_cleanup(struct media_entity *entity) {}; */ __must_check int media_create_pad_link(struct media_entity *source, u16 source_pad, struct media_entity *sink, - u16 sink_pad, u32 flags); + u16 sink_pad, u32 flags, gfp_t gfpflags); /** * media_create_pad_links() - creates a link between two entities. @@ -883,11 +884,12 @@ void __media_entity_pipeline_stop(struct media_entity *entity); struct media_intf_devnode * __must_check media_devnode_create(struct media_device *mdev, u32 type, u32 flags, - u32 major, u32 minor); + u32 major, u32 minor, gfp_t gfpflags); /** * media_devnode_remove() - removes a device node interface * * @devnode: pointer to &media_intf_devnode to be freed. + * @gfpflags: Get Free Page (GFP) flags, as defined in include/linux/gfp.h * * When a device node interface is removed, all links to it are automatically * removed. @@ -901,6 +903,7 @@ struct media_link * * @entity: pointer to %media_entity * @intf: pointer to %media_interface * @flags: Link flags, as defined in include/uapi/linux/media.h. + * @gfpflags: Get Free Page (GFP) flags, as defined in include/linux/gfp.h * * * Valid values for flags: @@ -921,7 +924,7 @@ struct media_link * */ __must_check media_create_intf_link(struct media_entity *entity, struct media_interface *intf, - u32 flags); + u32 flags, gfp_t gfpflags); /** * __media_remove_intf_link() - remove a single interface link * diff --git a/sound/usb/media.c b/sound/usb/media.c index 93a50d01..f5ac038 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -88,14 +88,16 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, MAJOR(pcm_dev->devt), - MINOR(pcm_dev->devt)); + MINOR(pcm_dev->devt), + GFP_KERNEL); if (!mctl->intf_devnode) { ret = -ENOMEM; goto unregister_entity; } mctl->intf_link = media_create_intf_link(&mctl->media_entity, &mctl->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!mctl->intf_link) { ret = -ENOMEM; goto devnode_remove; @@ -107,7 +109,8 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, case MEDIA_ENT_F_AUDIO_MIXER: ret = media_create_pad_link(entity, mixer_pad, &mctl->media_entity, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) goto remove_intf_link; break; @@ -180,7 +183,8 @@ int media_snd_mixer_init(struct snd_usb_audio *chip) if (!ctl_intf) { ctl_intf = media_devnode_create(mdev, intf_type, 0, MAJOR(ctl_dev->devt), - MINOR(ctl_dev->devt)); + MINOR(ctl_dev->devt), + GFP_KERNEL); if (!ctl_intf) return -ENOMEM; chip->ctl_intf_media_devnode = ctl_intf; @@ -213,7 +217,8 @@ int media_snd_mixer_init(struct snd_usb_audio *chip) mctl->intf_link = media_create_intf_link(&mctl->media_entity, &ctl_intf->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!mctl->intf_link) { media_device_unregister_entity(&mctl->media_entity); media_entity_cleanup(&mctl->media_entity); -- 2.5.0

5 11

[alsa-devel] [PATCH 0/3] ALSA: usb-audio: Fix NULL dereferences via invalid EP accesses
by Takashi Iwai 16 Mar '16

16 Mar '16

Hi, this is a small patchset to address the possible Oops by accessing the invalid EP that was borked by a malformed firmware. One patch is a clean up while two are fixes. Takashi === Takashi Iwai (3): ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk() ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk() ALSA: usb-audio: Add sanity checks for endpoint accesses sound/usb/clock.c | 2 ++ sound/usb/endpoint.c | 3 +++ sound/usb/mixer_quirks.c | 4 ++++ sound/usb/pcm.c | 2 ++ sound/usb/quirks.c | 22 ++++++++++++++-------- 5 files changed, 25 insertions(+), 8 deletions(-) -- 2.7.3

1 3

[alsa-devel] pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large
by Klement Sekera 16 Mar '16

16 Mar '16

Hi devs, I'm experiencing random hick-ups on my Carbon X1 2nd version with a Xonar U3 USB sound card plugged into the onelink dock. When this happens, the whole desktop freezes, whatever was last playing (second or two) loops and after a while (~30seconds) everything recovers.. I see this in my logs: Mar 16 11:37:07 ksekera-arch rtkit-daemon[1207]: The canary thread is apparently starving. Taking action. Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 4194088 bytes (23776 ms). Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA develop Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_dump(): Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Hardware PCM card 2 'USB Advanced Audio Device' device 0 subdevice 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Its setup is: Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: stream : PLAYBACK Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: access : MMAP_INTERLEAVED Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: format : S16_LE Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: subformat : STD Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: channels : 2 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: rate : 44100 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: exact rate : 44100 (44100/1) Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: msbits : 16 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: buffer_size : 88200 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_size : 44100 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_time : 1000000 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: tstamp_mode : ENABLE Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: tstamp_type : MONOTONIC Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_step : 1 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: avail_min : 87319 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_event : 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: start_threshold : -1 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: stop_threshold : 6206523236469964800 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: silence_threshold: 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: silence_size : 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: boundary : 6206523236469964800 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: appl_ptr : 20607492 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: hw_ptr : 21567814 Mar 16 11:37:07 ksekera-arch rtkit-daemon[1207]: Demoting known real-time threads. Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -3674960 bytes (-20833 ms). Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA develop Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_dump(): Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Hardware PCM card 2 'USB Advanced Audio Device' device 0 subdevice 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Its setup is: Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: stream : PLAYBACK Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: access : MMAP_INTERLEAVED Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: format : S16_LE Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: subformat : STD Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: channels : 2 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: rate : 44100 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: exact rate : 44100 (44100/1) Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: msbits : 16 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: buffer_size : 88200 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_size : 44100 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_time : 1000000 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: tstamp_mode : ENABLE Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: tstamp_type : MONOTONIC Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_step : 1 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: avail_min : 87319 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_event : 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: start_threshold : -1 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: stop_threshold : 6206523236469964800 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: silence_threshold: 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: silence_size : 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: boundary : 6206523236469964800 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: appl_ptr : 20653792 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: hw_ptr : 21572797 Mar 16 11:37:07 ksekera-arch rtkit-daemon[1207]: Successfully demoted thread 1203 of process 1203 (/usr/bin/pulseaudio). Mar 16 11:37:07 ksekera-arch rtkit-daemon[1207]: Demoted 1 threads. I'm on arch linux: ksekera@ksekera-arch ~> cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.3.6-1-ck. ksekera@ksekera-arch ~> uname -a Linux ksekera-arch 4.3.6-1-ck #1 SMP PREEMPT Fri Feb 19 20:24:27 EST 2016 x86_64 GNU/Linux Let me know if you need any more info, Regards, Klement

1 0

[alsa-devel] [PATCH] ASoC: dapm: Do not traverse widget hooks to snd-soc-dummy
by Harry Pan 16 Mar '16

16 Mar '16

All components are initially given an empty card when registering platform, and since the commit 6e78108bda78 ("ASoC: core: Don't probe the component which is dummy")', snd-soc-dummy will not be probed so that it remains an empty card assigned. This patch ignores to iterate widget hooks to the 'snd-soc-dummy' component, else it will trigger memory fault because of invalid dereference address of card->widgets. Test by grep -rsI "hello" /sys/devices/platform/skl_nau88l25_ssm4567_i2s/ Conflicts: sound/soc/soc-dapm.c Signed-off-by: Harry Pan <harry.pan(a)intel.com> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 581175a..0bc15c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2188,6 +2188,9 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, int count = 0; char *state = "not set"; + if (!strcmp(cmpnt->name, "snd-soc-dummy")) + return 0; + list_for_each_entry(w, &cmpnt->card->widgets, list) { if (w->dapm != dapm) continue; -- 2.1.2

1 0

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Fix module removal
by Vinod Koul 16 Mar '16

16 Mar '16

Skyalke driver were crashing on module removal when we run "alsa reload". The first issues was caused by component framwork regression which Russell fixed and in in 4.5. In driver there were still some missing order which needs to be fixed. This series first update the list parsing in core, then ensure proper freeup of code objects, and then esnureing driver does proper cleanup and i915 dependency Vinod Koul (6): ALSA: hda: use list macro for parsing on cleanup ASoC: Intel: Skylake: free codec objects on removal ASoC: Intel: Skylake: Freeup properly on skl_dsp_free ASoC: Intel: Skylake: Unmap the address last ASoC: Intel: Skylake: Call i915 exit last ASoC: Intel: Skylake: remove call to pci_dev_put sound/hda/ext/hdac_ext_stream.c | 5 ++--- sound/soc/intel/skylake/skl-sst-dsp.c | 5 +++++ sound/soc/intel/skylake/skl.c | 22 +++++++++++++++------- 3 files changed, 22 insertions(+), 10 deletions(-) Thanks -- 1.9.1

3 14

[alsa-devel] [PATCH] ASoC: rt5616: add I2C dependency
by Arnd Bergmann 16 Mar '16

16 Mar '16

The rt5616 codec registers itself as an i2c driver, but can be enabled even when i2c is turned off, which leads to a build error: codecs/rt5616.c:1419:1: error: data definition has no type or storage class [-Werror] module_i2c_driver(rt5616_i2c_driver); This adds an explicit Kconfig dependency, like the other codec drivers. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 288bc356a881 ("ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616") --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..7ef3a0c16478 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -629,6 +629,7 @@ config SND_SOC_RT5514 config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" + depends on I2C config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" -- 2.7.0

2 1

[alsa-devel] [PATCH v6] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 16 Mar '16

16 Mar '16

This is an initial imx-wm8958 device-tree-only machine driver. The sound card driver support three dai link, AIF1-DAI, AIF2-DAI and AIF3-DAI. We can configure the cpu dai from device tree, if present it will create corresponding sound card subdevice. At least one cpu dai phandle should be set from device tree. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- v5->v6: some header files should be moved to the next patch. So remove it. .../devicetree/bindings/sound/imx-audio-wm8958.txt | 69 +++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 325 +++++++++++++++++++++ 4 files changed, 410 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..7978c62 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,69 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + +Optional properties: + + - cpu-dai1 : The phandle of CPU DAI1 controller, which is for + aif1 of wm8958. At least one cpu dai is required. + + - cpu-dai2 : The phandle of CPU DAI2 controller, which is for + aif2 of wm8958. At least one cpu dai is required. + + - cpu-dai3 : The phandle of CPU DAI3 controller, which is for + aif3 of wm8958. At least one cpu dai is required. + + - dai-link1-name : If present, set the dai link1 name and stream name + to the specific strings. Otherwise, set the dai + link1 name and stream name to "AIF1-DAI". + + - dai-link2-name : If present, set the dai link2 name and stream name + to the specific strings. Otherwise, set the dai + link2 name and stream name to "AIF2-DAI". + + - dai-link3-name : If present, set the dai link3 name and stream name + to the specific strings. Otherwise, set the dai + link3 name and stream name to "AIF3-DAI". + + - fsl,cpu-dai1-master : If present, cpu dai1 works as master, and will + provide bit clock and frame clock. Otherwise, + cpu dai1 works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-ddr3-arm2-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai1 = <&sai1>; + dai-link1-name = "Hifi"; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..5ac200e --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,325 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define AIF1_DAI (0) +#define AIF2_DAI (1) +#define AIF3_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == AIF1_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == AIF1_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == AIF1_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link dai_link[DAI_LINK_NUM]; + struct snd_soc_card card; + unsigned long mclk_freq[WM8958_MCLK_MAX]; + bool is_cpu_dai1_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == AIF1_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == AIF1_DAI) { + if (data->is_cpu_dai1_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (data->is_cpu_dai1_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == AIF1_DAI && data->is_cpu_dai1_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_wm8958_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [AIF1_DAI] = { + .name = "AIF1-DAI", + .stream_name = "AIF1-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [AIF2_DAI] = { + .name = "AIF2-DAI", + .stream_name = "AIF2-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [AIF3_DAI] = { + .name = "AIF3-DAI", + .stream_name = "AIF3-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np[DAI_LINK_NUM], *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev[DAI_LINK_NUM]; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + struct clk *mclk; + char tmp[8], *name; + int num_links = 0; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "cpu-dai%d", i + 1); + cpu_np[i] = of_parse_phandle(np, tmp, 0); + if (!cpu_np[i]) + continue; + + cpu_pdev[i] = of_find_device_by_node(cpu_np[i]); + if (!cpu_pdev[i]) { + dev_err(&pdev->dev, + "failed to get cpu dai%d platform device\n", i); + ret = -EINVAL; + goto fail; + } + data->dai_link[num_links] = imx_wm8958_dai_link[i]; + data->dai_link[num_links].cpu_dai_name = dev_name(&cpu_pdev[i]->dev); + data->dai_link[num_links].platform_of_node = cpu_np[i]; + + sprintf(tmp, "dai-link%d-name", i + 1); + if(!of_property_read_string(np, tmp, (const char **)&name)) { + data->dai_link[num_links].name = name; + data->dai_link[num_links].stream_name = name; + } + num_links++; + } + + if (!num_links) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + /* cpu dai1 link support cpu dai master and slave mode */ + if (of_property_read_bool(np, "fsl,cpu-dai1-master")) { + data->is_cpu_dai1_master = true; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else { + data->is_cpu_dai1_master = false; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + mclk = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(mclk)) + data->mclk_freq[i] = clk_get_rate(mclk); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = num_links; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np[AIF1_DAI]); + of_node_put(cpu_np[AIF2_DAI]); + of_node_put(cpu_np[AIF3_DAI]); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: cs35l32: prevent unexpected parse result of device tree
by Takashi Sakamoto 16 Mar '16

16 Mar '16

Hi, When compiling recent kernel, I got a warning caused by a codec in ASoC. sound/soc/codecs/cs35l32.c: In function ‘cs35l32_i2c_probe’: sound/soc/codecs/cs35l32.c:278:2: warning: ‘val’ may be used uninitialized in this function [-Wmaybe-uninitialized] switch (val) { ^ sound/soc/codecs/cs35l32.c:272:15: note: ‘val’ was declared here unsigned int val; ^ I'm not an ASoC developer, furthermore, have no knowledge about this Cirrus codec. Although, as a quick glance it may include a logical bug to assign wrong information to structure member as a reult of parsing device tree. This patch is my proposal to fix it. I'm happy to receive your comments. Regards Takashi Sakamoto (1): ASoC: cs35l32: fix unexpected parse result of device tree blob sound/soc/codecs/cs35l32.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.7.0

2 3

[alsa-devel] [PATCH] ASoC: nau8825: fix interrupt fails and unstable after resume
by John Hsu 16 Mar '16

16 Mar '16

This patch is to fix interrupt fails when resume back. In solution, we add IRQ re-initiation, and reconstruct suspend and resume with set_bias_level function. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 42 ++++++++++++++++++++++++++++++++++++------ 1 file changed, 36 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 504c969..2e2e11a 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1092,6 +1092,36 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable DDACR needed for interrupts */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + nau8825_restart_jack_detection(regmap); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1121,6 +1151,8 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + if (nau8825->irq) + nau8825_resume_setup(nau8825); } break; @@ -1330,24 +1362,22 @@ static int nau8825_i2c_remove(struct i2c_client *client) #ifdef CONFIG_PM_SLEEP static int nau8825_suspend(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); - disable_irq(client->irq); + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); return 0; } static int nau8825_resume(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); + enable_irq(nau8825->irq); return 0; } -- 2.6.4

2 8

[alsa-devel] [PATCH v5] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 15 Mar '16

15 Mar '16

This is an initial imx-wm8958 device-tree-only machine driver. The sound card driver support three dai link, HIFI, VOICE and BT. We can configure the cpu dai from device tree, if present it will create corresponding sound card subdevice. So at least one cpu dai phandle should be set from device tree. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v4->v5: support set dai link name and stream name from device tree. .../devicetree/bindings/sound/imx-audio-wm8958.txt | 69 +++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 328 +++++++++++++++++++++ 4 files changed, 413 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..7978c62 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,69 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + +Optional properties: + + - cpu-dai1 : The phandle of CPU DAI1 controller, which is for + aif1 of wm8958. At least one cpu dai is required. + + - cpu-dai2 : The phandle of CPU DAI2 controller, which is for + aif2 of wm8958. At least one cpu dai is required. + + - cpu-dai3 : The phandle of CPU DAI3 controller, which is for + aif3 of wm8958. At least one cpu dai is required. + + - dai-link1-name : If present, set the dai link1 name and stream name + to the specific strings. Otherwise, set the dai + link1 name and stream name to "AIF1-DAI". + + - dai-link2-name : If present, set the dai link2 name and stream name + to the specific strings. Otherwise, set the dai + link2 name and stream name to "AIF2-DAI". + + - dai-link3-name : If present, set the dai link3 name and stream name + to the specific strings. Otherwise, set the dai + link3 name and stream name to "AIF3-DAI". + + - fsl,cpu-dai1-master : If present, cpu dai1 works as master, and will + provide bit clock and frame clock. Otherwise, + cpu dai1 works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-ddr3-arm2-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai1 = <&sai1>; + dai-link1-name = "Hifi"; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..650b0c8 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,328 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define AIF1_DAI (0) +#define AIF2_DAI (1) +#define AIF3_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == AIF1_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == AIF1_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == AIF1_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link dai_link[DAI_LINK_NUM]; + struct snd_soc_card card; + unsigned long mclk_freq[WM8958_MCLK_MAX]; + bool is_cpu_dai1_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == AIF1_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == AIF1_DAI) { + if (data->is_cpu_dai1_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (data->is_cpu_dai1_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == AIF1_DAI && data->is_cpu_dai1_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_wm8958_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [AIF1_DAI] = { + .name = "AIF1-DAI", + .stream_name = "AIF1-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [AIF2_DAI] = { + .name = "AIF2-DAI", + .stream_name = "AIF2-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [AIF3_DAI] = { + .name = "AIF3-DAI", + .stream_name = "AIF3-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np[DAI_LINK_NUM], *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev[DAI_LINK_NUM]; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + struct clk *mclk; + char tmp[8], *name; + int num_links = 0; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "cpu-dai%d", i + 1); + cpu_np[i] = of_parse_phandle(np, tmp, 0); + if (!cpu_np[i]) + continue; + + cpu_pdev[i] = of_find_device_by_node(cpu_np[i]); + if (!cpu_pdev[i]) { + dev_err(&pdev->dev, + "failed to get cpu dai%d platform device\n", i); + ret = -EINVAL; + goto fail; + } + data->dai_link[num_links] = imx_wm8958_dai_link[i]; + data->dai_link[num_links].cpu_dai_name = dev_name(&cpu_pdev[i]->dev); + data->dai_link[num_links].platform_of_node = cpu_np[i]; + + sprintf(tmp, "dai-link%d-name", i + 1); + if(!of_property_read_string(np, tmp, (const char **)&name)) { + data->dai_link[num_links].name = name; + data->dai_link[num_links].stream_name = name; + } + num_links++; + } + + if (!num_links) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + /* cpu dai1 link support cpu dai master and slave mode */ + if (of_property_read_bool(np, "fsl,cpu-dai1-master")) { + data->is_cpu_dai1_master = true; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else { + data->is_cpu_dai1_master = false; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + mclk = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(mclk)) + data->mclk_freq[i] = clk_get_rate(mclk); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = num_links; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np[AIF1_DAI]); + of_node_put(cpu_np[AIF2_DAI]); + of_node_put(cpu_np[AIF3_DAI]); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix unconditional GPIO toggle via automute
by Takashi Iwai 15 Mar '16

15 Mar '16

Cirrus HD-audio driver may adjust GPIO pins for EAPD dynamically depending on the jack plug state. This works fine for the auto-mute mode where the speaker gets muted upon the HP jack plug. OTOH, when the auto-mute mode is off, this turns off the EAPD unexpectedly depending on the jack state, which results in the silent speaker output. This patch fixes the silent speaker output issue by setting GPIO bits constantly when the auto-mute mode is off. Reported-and-tested-by: moosotc(a)gmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c1c855a6c0af..a47e8ae0eb30 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -174,8 +174,12 @@ static void cs_automute(struct hda_codec *codec) snd_hda_gen_update_outputs(codec); if (spec->gpio_eapd_hp || spec->gpio_eapd_speaker) { - spec->gpio_data = spec->gen.hp_jack_present ? - spec->gpio_eapd_hp : spec->gpio_eapd_speaker; + if (spec->gen.automute_speaker) + spec->gpio_data = spec->gen.hp_jack_present ? + spec->gpio_eapd_hp : spec->gpio_eapd_speaker; + else + spec->gpio_data = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_data); } -- 2.7.3

1 0

[alsa-devel] No audio output via built-in speaker on a MacMini while headphone jack is plugged in
by moosotc＠gmail.com 15 Mar '16

15 Mar '16

`amixer set Speaker unmute' makes no difference physically unplugging the headphone jack from the socket does, i.e. `amixer set Speaker (un)mute' does the right thing. -- mailto:moosotc@gmail.com

2 8

[alsa-devel] [PATCH] [media] sound/usb: fix NULL dereference in usb_audio_probe()
by Nicolai Stange 15 Mar '16

15 Mar '16

With commit aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources") an access to quirk->media_device without checking for quirk != NULL has been introduced in usb_audio_probe(). With a Plantronics USB headset (device ID 0x047f:0xc010) attached, this results in the following splat at boot time: BUG: unable to handle kernel NULL pointer dereference at 0000000000000014 IP: [<ffffffffa089aa6c>] usb_audio_probe+0x2cc/0x9a0 [snd_usb_audio] Oops: 0000 [#1] SMP [...] CPU: 2 PID: 696 Comm: systemd-udevd Not tainted 4.5.0-next-20160315 #13 Hardware name: Dell Inc. Latitude E6540/0725FP, BIOS A10 06/26/2014 task: ffff88021c88d7c0 ti: ffff88003d5b0000 task.ti: ffff88003d5b0000 RIP: 0010:[<ffffffffa089aa6c>] [<ffffffffa089aa6c>] usb_audio_probe+0x2cc/0x9a0 [snd_usb_audio] [...] Call Trace: [<ffffffff815a8e16>] usb_probe_interface+0x136/0x2d0 [<ffffffff81509edc>] driver_probe_device+0x22c/0x440 [<ffffffff8150a1c1>] __driver_attach+0xd1/0xf0 [<ffffffff8150a0f0>] ? driver_probe_device+0x440/0x440 [<ffffffff815077ec>] bus_for_each_dev+0x6c/0xc0 [<ffffffff815095ce>] driver_attach+0x1e/0x20 [<ffffffff81509013>] bus_add_driver+0x1c3/0x280 [<ffffffff8150ab10>] driver_register+0x60/0xe0 [<ffffffff815a7711>] usb_register_driver+0x81/0x140 [<ffffffffa08c7000>] ? 0xffffffffa08c7000 [<ffffffffa08c701e>] usb_audio_driver_init+0x1e/0x1000 [snd_usb_audio] [<ffffffff81002123>] do_one_initcall+0xb3/0x1f0 [<ffffffff811fb091>] ? __vunmap+0x81/0xd0 [<ffffffff8121b8d2>] ? kmem_cache_alloc_trace+0x182/0x1d0 [<ffffffff811b0267>] ? do_init_module+0x27/0x1d8 [<ffffffff811b029f>] do_init_module+0x5f/0x1d8 [<ffffffff8112ce35>] load_module+0x1fe5/0x27a0 [<ffffffff81129870>] ? __symbol_put+0x60/0x60 [<ffffffff81241690>] ? vfs_read+0x110/0x130 [<ffffffff8112d866>] SYSC_finit_module+0xe6/0x120 [<ffffffff8112d8be>] SyS_finit_module+0xe/0x10 [<ffffffff81003d94>] do_syscall_64+0x64/0x110 [<ffffffff817c0b61>] entry_SYSCALL64_slow_path+0x25/0x25 After encountering this, the system-udevd process seems to be blocked until it is killed when hitting its timeout of 3min. In analogy to the other accesses to members of quirk in usb_audio_probe(), check for quirk != NULL before accessing its ->media_device. Fixes: aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources") Signed-off-by: Nicolai Stange <nicstange(a)gmail.com> --- Applicable to linux-next-20160315. sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 63244bb..479621e 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -612,7 +612,7 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; - if (quirk->media_device) { + if (quirk && quirk->media_device) { /* don't want to fail when media_snd_device_create() fails */ media_snd_device_create(chip, intf); } -- 2.7.2

4 4

Re: [alsa-devel] ASoC: TLV320AIC3x: Adding additional functionality for 3106 with [Patch] for discuss
by Peter Ujfalusi 15 Mar '16

15 Mar '16

On 03/14/2016 04:57 PM, Timur Karaldin wrote: > Hi, Peter! > What's about placing capture controls in the Playback section of Alsamixer > view? Yeah, it is annoying. They can be fixed by: "Line1L Switch" -> "Line1L Capture Switch" and so on. On the other hand we had these unfortunate control names I think since the beginning and it is kind of an ABI for existing products. I don't have issue changing them, but existing setups relying on the control names might break... > Does resetting of some registers during start/stop of recording > connect with it? Not sure what you mean. -- Péter > > 14.03.2016 17:47, Peter Ujfalusi пишет: >> Hi Timur, >> >> On 03/14/16 13:06, Timur Karaldin wrote: >>> Hi, colleagues! >>> >>> I'm newbie with ALSA devel. I need to get extra functionality from codec >>> through ALSA mixer. >>> Task: 1. making possibility to set-up gain for each input (XXX_2_LADCCTRL and >>> XXX_2_RADCCTRL registers) >> This part is a bit tricky as in the registers: >> value 0..8 is gain from 0dB to -12dB in steps of -1.5dB and value 0xf means >> basically mute/disconnect. >> >>> 2. making possibility to set-up debounce for button and headset detection ( >>> HEADSET_DETECT_CTRL_A register) >>> 3. making possibility to enable high-power AC-coupled output >>> (HEADSET_DETECT_CTRL_A register) >>> 4. making possibility to enable stereo pseudo differential output >>> (HEADSET_DETECT_CTRL_B register) >>> 5. enable headset detection (HEADSET_DETECT_CTRL_A) >>> 6. getting headset detect status (HEADSET_DETECT_CTRL_A) >>> >>> Well, my patch works in generally, but I don't like it very much, because: >>> 1. All controls for input amplifiers' gain are shown in Alsamixer-Playback >>> section. But not only mine, capture controls from base codec's version are >>> shown in playback section too, for example (Right or Left) PGA Mixer Line1(L >>> or R)/2(L or R)/MIC3(L or R), which is definitely capture settings. Should I >>> worried about it? because, as I understand, it could not be setted correctly >>> during changing playback/capture mode for power-saving reason. >>> 2. "Right PGA Mixer Line1R" control (and similar) using the same bits in >>> LINE1R_2_RADCCTRL register as my "Line1R RADC Gain" control (and similar) and >>> put there b"1111" when user switch disconnect it from ADC. So my control is >>> not shown when user disconnect it from ADC. >>> 3. When I start/stop arecord some of my options were reset (debounces, headset >>> enable, high-power AC-coupled, pseudo differential output). When I set up it >>> in shell and then start arecord, it's reset again. Is it the reason because of >>> displaying controls in playback mixer section? >>> >>> I will be very pleased for any advise, thank you! >>> >>> >>> here is my patch for tlv320aic3x.c >>> --- >>> sound/soc/codecs/tlv320aic3x.c | 55 >>> +++++++++++++++++++++++++++++++++++++++--- >>> 1 file changed, 52 insertions(+), 3 deletions(-) >>> >>> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c >>> index d7349bc..27d031e 100644 >>> --- a/sound/soc/codecs/tlv320aic3x.c >>> +++ b/sound/soc/codecs/tlv320aic3x.c >>> @@ -122,11 +122,24 @@ static const struct reg_default aic3x_reg[] = { >>> { 108, 0x00 }, { 109, 0x00 }, >>> }; >>> >>> +static bool aic3106_volatile(struct device *dev, unsigned int reg) >>> +{ >>> + switch(reg) >>> + { >>> + case AIC3X_HEADSET_DETECT_CTRL_A: >>> + case AIC3X_HEADSET_DETECT_CTRL_B: >>> + return true; >>> + default: >>> + return false; >>> + } >>> +} >>> + >>> static const struct regmap_config aic3x_regmap = { >>> .reg_bits = 8, >>> .val_bits = 8, >>> >>> .max_register = DAC_ICC_ADJ, >>> + .volatile_reg = aic3106_volatile, >>> .reg_defaults = aic3x_reg, >>> .num_reg_defaults = ARRAY_SIZE(aic3x_reg), >>> .cache_type = REGCACHE_RBTREE, >>> @@ -196,7 +209,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, >>> struct snd_kcontrol *kcontrol, int event) >>> { >>> struct snd_soc_codec *codec = w->codec; >>> - struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); >>> + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); >> You have couple of these nop changes in the patch... >> >>> switch (event) { >>> case SND_SOC_DAPM_POST_PMU: >>> @@ -270,6 +283,26 @@ static const struct soc_enum aic3x_agc_decay_enum[] = { >>> SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), >>> }; >>> >>> +static const char *aic3x_adc_gain[] = { "0dB", "-1.5dB", "-3dB", "-4.5dB", >>> "-6dB", "-7.5dB", "-9dB", "-10.5dB", "-12dB" }; >>> +static const struct soc_enum aic3x_adc_gain_enum[] = { >>> + SOC_ENUM_SINGLE(MIC3LR_2_LADC_CTRL, 4, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(MIC3LR_2_RADC_CTRL, 0, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), >>> +}; >> So the issue is that we have the DAPM switches controlling exactly the same >> registers. I believe if you would set the gain in a way that bit0 is 1, then >> DAPM will think that the path is disconnected. >> Also if you would set the gain and then mute and unmute the path you would >> have lost the gain you wanted to have.. >> >> The only way I can think of implementing these mixers is to have two sets of >> custom callbacks. one set is to set/get the gain and the other is to set/get >> the mute/disconnect on these. >> When the path is disconnected you should not write the gain change to the >> chip, but cache it and if the path is unmuted, you write the cached gain. When >> you mute the path you should take the set gain first, cache it, then >> disconnect the path. >> >> For these gains you should have DECLARE_TLV_DB_SCALE() and use >> SOC_SINGLE_TLV(). Make sure that the control name matches with the >> corresponding DAPM widget's name so ALSA can match them correctly. >> >>> +static const char *aic3x_headset_debounce[] = { "16ms", "32ms", "64ms", >>> "128ms", "256ms", "512ms" }; >>> +static const struct soc_enum aic3x_headset_debounce_enum[] = { >>> + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 2, 6, >>> aic3x_headset_debounce), >>> +}; >>> +static const char *aic3x_button_debounce[] = { "0ms", "8ms", "16ms", >>> "32ms" }; >>> +static const struct soc_enum aic3x_button_debounce_enum[] = { >>> + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 0, 4, >>> aic3x_button_debounce), >>> +}; >>> + >>> + >>> + >>> /* >>> * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps >>> */ >>> @@ -399,6 +432,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] >>> = { >>> SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), >>> >>> SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), >>> + /* Additional controls */ >>> + SOC_ENUM("Mic3L LADC Gain", aic3x_adc_gain_enum[0]), >>> + SOC_ENUM("Mic3R RADC Gain", aic3x_adc_gain_enum[1]), >>> + SOC_ENUM("Line1L LADC Gain", aic3x_adc_gain_enum[2]), >>> + SOC_ENUM("Line1R RADC Gain", aic3x_adc_gain_enum[3]), >>> + SOC_ENUM("Line2L LADC Gain", aic3x_adc_gain_enum[4]), >>> + SOC_ENUM("Line2R RADC Gain", aic3x_adc_gain_enum[5]), >> Please avoid using arrays as the driver no longer use these anymore. >> >>> + SOC_ENUM("Headset Debounce", aic3x_headset_debounce_enum), >>> + SOC_ENUM("Button Debounce", aic3x_button_debounce_enum), >>> + SOC_SINGLE("Headset Detected", AIC3X_HEADSET_DETECT_CTRL_A, >>> AIC3X_HEADSET_DETECT_SHIFT, AIC3X_HEADSET_DETECT_MASK, 0), >>> + SOC_SINGLE("Headset Detect Enable", AIC3X_HEADSET_DETECT_CTRL_A, 7, 1, >>> 0), >>> + SOC_SINGLE("High power output Ac-coupled", AIC3X_HEADSET_DETECT_CTRL_B, >>> 7, 1, 0), >>> + SOC_SINGLE("Stereo pseudodifferential output", >>> AIC3X_HEADSET_DETECT_CTRL_B, 3, 1, 0), >>> }; >>> >>> static const struct snd_kcontrol_new aic3x_mono_controls[] = { >>> @@ -614,7 +660,7 @@ static const struct snd_soc_dapm_widget >>> aic3x_dapm_widgets[] = { >>> SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", >>> AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), >>> >>> - /* Mic Bias */ >>> + /* Mic Bias */ >> nop change >> >>> SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, >>> mic_bias_event, >>> SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), >>> @@ -692,7 +738,7 @@ static const struct snd_soc_dapm_route intercon[] = { >>> {"Left Line2L Mux", "single-ended", "LINE2L"}, >>> {"Left Line2L Mux", "differential", "LINE2L"}, >>> >>> - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, >>> + {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, >> nop change >> >>> {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, >>> {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, >>> {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, >>> @@ -814,6 +860,7 @@ static const struct snd_soc_dapm_route intercon[] = { >>> {"Right HPCOM Mux", "external feedback", "Right HPCOM Mixer"}, >>> {"Right HP Com", NULL, "Right HPCOM Mux"}, >>> {"HPRCOM", NULL, "Right HP Com"}, >>> + >> why? >> >>> }; >>> >>> static const struct snd_soc_dapm_route intercon_mono[] = { >>> @@ -918,7 +965,9 @@ static int aic3x_hw_params(struct snd_pcm_substream >>> *substream, >>> data = (LDAC2LCH | RDAC2RCH); >>> data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; >>> if (params_rate(params) >= 64000) >>> + { >>> data |= DUAL_RATE_MODE; >>> + } >> why? >> >>> snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data); >>> >>> /* codec sample rate select */ >>> -- >>> and tlv320aic3x.h >>> --- >>> sound/soc/codecs/tlv320aic3x.h | 6 ++++++ >>> 1 file changed, 6 insertions(+) >>> >>> diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h >>> index e521ac3..8ff42d2 100644 >>> --- a/sound/soc/codecs/tlv320aic3x.h >>> +++ b/sound/soc/codecs/tlv320aic3x.h >>> @@ -271,6 +271,12 @@ enum { >>> AIC3X_BUTTON_DEBOUNCE_32MS = 3 >>> }; >>> >>> +typedef struct { >>> + struct platform_device* pdev; >>> + struct proc_dir_entry *proc_value; >>> + struct snd_soc_codec *codec; >>> +} aic3106_detect_t; >>> + >>> #define AIC3X_HEADSET_DETECT_ENABLED 0x80 >>> #define AIC3X_HEADSET_DETECT_SHIFT 5 >>> #define AIC3X_HEADSET_DETECT_MASK 3 >>> -- >> >> When submitting please separate the patches by functionality. It is easier to >> review the changes. >> > >

2 2

[alsa-devel] [PATCH v3] ASoC: cs35l32: avoid uninitialized variable access
by Arnd Bergmann 15 Mar '16

15 Mar '16

gcc warns about the possibilty of accessing a property read from devicetree in cs35l32_i2c_probe() when it has not been initialized because CONFIG_OF is disabled: sound/soc/codecs/cs35l32.c: In function 'cs35l32_i2c_probe': sound/soc/codecs/cs35l32.c:278:2: warning: 'val' may be used uninitialized in this function [-Wmaybe-uninitialized] The code is actually correct because it checks the dev->of_node variable first and we know this is NULL here when CONFIG_OF is disabled, but Russell King noticed that it's broken when we probe the device using DT, and the properties are absent. The code already has some checking for incorrect values, and I keep that checking unchanged here, but add an additional check for an error returned by the property accessor functions that now gets handled the same way as incorrect data in the properties. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v3: Restore a line that was accidentally removed, as pointed out by Brian Austin. v2: fix bug in case of CONFIG_OF and missing properties v1: only address warning for the !CONFIG_OF case sound/soc/codecs/cs35l32.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..287d13740be4 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -274,7 +274,9 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) pdata->sdout_share = val; - of_property_read_u32(np, "cirrus,boost-manager", &val); + if (of_property_read_u32(np, "cirrus,boost-manager", &val)) + val = -1u; + switch (val) { case CS35L32_BOOST_MGR_AUTO: case CS35L32_BOOST_MGR_AUTO_AUDIO: @@ -282,13 +284,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BOOST_MGR_FIXED: pdata->boost_mng = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,boost-manager DT value %d\n", val); pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; } - of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + if (of_property_read_u32(np, "cirrus,sdout-datacfg", &val)) + val = -1u; switch (val) { case CS35L32_DATA_CFG_LR_VP: case CS35L32_DATA_CFG_LR_STAT: @@ -296,13 +300,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_DATA_CFG_LR_VPSTAT: pdata->sdout_datacfg = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,sdout-datacfg DT value %d\n", val); pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; } - of_property_read_u32(np, "cirrus,battery-threshold", &val); + if (of_property_read_u32(np, "cirrus,battery-threshold", &val)) + val = -1u; switch (val) { case CS35L32_BATT_THRESH_3_1V: case CS35L32_BATT_THRESH_3_2V: @@ -310,13 +316,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_THRESH_3_4V: pdata->batt_thresh = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-threshold DT value %d\n", val); pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; } - of_property_read_u32(np, "cirrus,battery-recovery", &val); + if (of_property_read_u32(np, "cirrus,battery-recovery", &val)) + val = -1u; switch (val) { case CS35L32_BATT_RECOV_3_1V: case CS35L32_BATT_RECOV_3_2V: @@ -326,6 +334,7 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_RECOV_3_6V: pdata->batt_recov = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-recovery DT value %d\n", val); -- 2.7.0

3 2

[alsa-devel] [PATCH] ASoC: nau8825: improve FLL function for better performance
by John Hsu 15 Mar '16

15 Mar '16

In FLL calculation, increase VCO/DCO frequency for better performance. Besides, have different register configuration according to fraction or not when apply FLL parameters. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 47 +++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 13 ++++++++----- 2 files changed, 42 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 1c87299..9023dbd 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -31,7 +31,7 @@ #include "nau8825.h" #define NAU_FREF_MAX 13500000 -#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 struct nau8825_fll { @@ -957,8 +957,8 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, struct nau8825_fll *fll_param) { - u64 fvco; - unsigned int fref, i; + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. @@ -983,18 +983,23 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fll_param->ratio = fll_ratio[i].val; /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. - * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be * guaranteed across the full range of operation. * FDCO = freq_out * 2 * mclk_src_scaling */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { fvco = 256 * fs * 2 * mclk_src_scaling[i].param; - if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) - break; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } } - if (i == ARRAY_SIZE(mclk_src_scaling)) + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) return -EINVAL; - fll_param->mclk_src = mclk_src_scaling[i].val; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional * input based on FDCO, FREF and FLL ratio. @@ -1009,7 +1014,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -1022,10 +1028,25 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ - regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_FILTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, NAU8825_FLL_FTR_SW_ACCU); + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_SDM_EN, 0); + } } /* freq_out must be 256*Fs in order to achieve the best performance */ diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8ceb5f3..01bd386 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -118,15 +118,18 @@ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8825_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8825_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8825_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8825_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8825_FLL_FTR_SW_FILTER (0x0 << 12) /* FLL6 (0x9) */ -#define NAU8825_DCO_EN_MASK (0x1 << 15) #define NAU8825_DCO_EN (0x1 << 15) -#define NAU8825_DCO_DIS (0x0 << 15) -#define NAU8825_SDM_EN_MASK (0x1 << 14) #define NAU8825_SDM_EN (0x1 << 14) -#define NAU8825_SDM_DIS (0x0 << 14) /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) -- 2.6.4

2 1

[alsa-devel] [PATCH 0/3] *** new features for alsabat ***
by han.lu＠intel.com 15 Mar '16

15 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Add 3 features for alsabat: 1. add default device name for playback and capture 2. add standalone mode, to bypass data analysis 3. add bash test script to cover basic alsabat features Lu, Han (3): alsabat: add default device name for playback and capture alsabat: add standalone mode alsabat: add bash test script bat/Makefile.am | 13 +++--- bat/alsa.c | 12 ------ bat/alsabat-test.sh | 112 ++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsabat.1 | 9 +++++ bat/bat.c | 18 ++++++++- bat/common.h | 3 ++ configure.ac | 5 ++- 7 files changed, 153 insertions(+), 19 deletions(-) create mode 100755 bat/alsabat-test.sh -- 2.5.0

2 4

[alsa-devel] [patch] ALSA: mixart: silence unitialized variable warnings
by Dan Carpenter 15 Mar '16

15 Mar '16

We print can print the uninitialized memory on error. Which is an info leak, I suppose but it's basically harmless. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 24a1955..58fd79eb 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -726,7 +726,7 @@ int mixart_update_playback_stream_level(struct snd_mixart* chip, int is_aes, int int volume[2]; struct mixart_msg request; struct mixart_set_out_stream_level_req set_level; - u32 status; + u32 status = 0; struct mixart_pipe *pipe; memset(&set_level, 0, sizeof(set_level)); @@ -778,7 +778,7 @@ int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes) struct mixart_pipe *pipe; struct mixart_msg request; struct mixart_set_in_audio_level_req set_level; - u32 status; + u32 status = 0; if(is_aes) { idx = 1;

2 1

[alsa-devel] Max9768 driver in kernel 4.1
by Micka 15 Mar '16

15 Mar '16

Hi, I'm trying to use the max9768, I have defined it correctly in the dts file : max9768: max9768@49 { compatible = "max,max9768"; reg = <0x49>; }; I can see that in the driver, i go in the function max9768_i2c_probe <http://lxr.free-electrons.com/ident?v=4.1;i=max9768_i2c_probe> without problem, and I don't see errors when executing this line : err <http://lxr.free-electrons.com/ident?v=4.1;i=err> = snd_soc_register_codec <http://lxr.free-electrons.com/ident?v=4.1;i=snd_soc_register_codec>(&client <http://lxr.free-electrons.com/ident?v=4.1;i=client>->dev <http://lxr.free-electrons.com/ident?v=4.1;i=dev>, &max9768_codec_driver <http://lxr.free-electrons.com/ident?v=4.1;i=max9768_codec_driver>, NULL <http://lxr.free-electrons.com/ident?v=4.1;i=NULL>, 0); I have put some log : [ 9.213797] ASoC: dai register 2-0049 #0 [ 9.213806] ASoC: Registered codec 'max9768.2-0049' my problem is that i don't know how to configure the volume ? Any idea ? where is it ? I don't see where is it in the amixer controls list : root@beaglebone:~# amixer controls numid=1,iface=MIXER,name='PCM Playback Volume' numid=15,iface=MIXER,name='Line DAC Playback Volume' numid=14,iface=MIXER,name='Line PGA Bypass Volume' numid=20,iface=MIXER,name='Line Playback Switch' numid=33,iface=MIXER,name='ADC HPF Cut-off' numid=23,iface=MIXER,name='AGC Switch' numid=30,iface=MIXER,name='De-emphasis Switch' numid=17,iface=MIXER,name='HP DAC Playback Volume' numid=16,iface=MIXER,name='HP PGA Bypass Volume' numid=21,iface=MIXER,name='HP Playback Switch' numid=19,iface=MIXER,name='HPCOM DAC Playback Volume' numid=18,iface=MIXER,name='HPCOM PGA Bypass Volume' numid=22,iface=MIXER,name='HPCOM Playback Switch' numid=26,iface=MIXER,name='Left AGC Attack time' numid=28,iface=MIXER,name='Left AGC Decay time' numid=24,iface=MIXER,name='Left AGC Target level' numid=75,iface=MIXER,name='Left DAC Mux' numid=49,iface=MIXER,name='Left HP Mixer DACL1 Switch' numid=7,iface=MIXER,name='Left HP Mixer DACR1 Playback Volume' numid=51,iface=MIXER,name='Left HP Mixer DACR1 Switch' numid=48,iface=MIXER,name='Left HP Mixer PGAL Bypass Switch' numid=50,iface=MIXER,name='Left HP Mixer PGAR Bypass Switch' numid=6,iface=MIXER,name='Left HP Mixer PGAR Bypass Volume' numid=41,iface=MIXER,name='Left HPCOM Mixer DACL1 Switch' numid=11,iface=MIXER,name='Left HPCOM Mixer DACR1 Playback Volume' numid=43,iface=MIXER,name='Left HPCOM Mixer DACR1 Switch' numid=40,iface=MIXER,name='Left HPCOM Mixer PGAL Bypass Switch' numid=42,iface=MIXER,name='Left HPCOM Mixer PGAR Bypass Switch' numid=10,iface=MIXER,name='Left HPCOM Mixer PGAR Bypass Volume' numid=74,iface=MIXER,name='Left HPCOM Mux' numid=57,iface=MIXER,name='Left Line Mixer DACL1 Switch' numid=3,iface=MIXER,name='Left Line Mixer DACR1 Playback Volume' numid=59,iface=MIXER,name='Left Line Mixer DACR1 Switch' numid=56,iface=MIXER,name='Left Line Mixer PGAL Bypass Switch' numid=58,iface=MIXER,name='Left Line Mixer PGAR Bypass Switch' numid=2,iface=MIXER,name='Left Line Mixer PGAR Bypass Volume' numid=71,iface=MIXER,name='Left Line1L Mux' numid=70,iface=MIXER,name='Left Line1R Mux' numid=64,iface=MIXER,name='Left PGA Mixer Line1L Switch' numid=65,iface=MIXER,name='Left PGA Mixer Line1R Switch' numid=66,iface=MIXER,name='Left PGA Mixer Mic2L Switch' numid=67,iface=MIXER,name='Left PGA Mixer Mic2R Switch' numid=34,iface=MIXER,name='Output Driver Power-On time' numid=35,iface=MIXER,name='Output Driver Ramp-up step' numid=32,iface=MIXER,name='PGA Capture Switch' numid=31,iface=MIXER,name='PGA Capture Volume' numid=27,iface=MIXER,name='Right AGC Attack time' numid=29,iface=MIXER,name='Right AGC Decay time' numid=25,iface=MIXER,name='Right AGC Target level' numid=73,iface=MIXER,name='Right DAC Mux' numid=9,iface=MIXER,name='Right HP Mixer DACL1 Playback Volume' numid=45,iface=MIXER,name='Right HP Mixer DACL1 Switch' numid=47,iface=MIXER,name='Right HP Mixer DACR1 Switch' numid=44,iface=MIXER,name='Right HP Mixer PGAL Bypass Switch' numid=8,iface=MIXER,name='Right HP Mixer PGAL Bypass Volume' numid=46,iface=MIXER,name='Right HP Mixer PGAR Bypass Switch' numid=13,iface=MIXER,name='Right HPCOM Mixer DACL1 Playback Volume' numid=37,iface=MIXER,name='Right HPCOM Mixer DACL1 Switch' numid=39,iface=MIXER,name='Right HPCOM Mixer DACR1 Switch' numid=36,iface=MIXER,name='Right HPCOM Mixer PGAL Bypass Switch' numid=12,iface=MIXER,name='Right HPCOM Mixer PGAL Bypass Volume' numid=38,iface=MIXER,name='Right HPCOM Mixer PGAR Bypass Switch' numid=72,iface=MIXER,name='Right HPCOM Mux' numid=5,iface=MIXER,name='Right Line Mixer DACL1 Playback Volume' numid=53,iface=MIXER,name='Right Line Mixer DACL1 Switch' numid=55,iface=MIXER,name='Right Line Mixer DACR1 Switch' numid=52,iface=MIXER,name='Right Line Mixer PGAL Bypass Switch' numid=4,iface=MIXER,name='Right Line Mixer PGAL Bypass Volume' numid=54,iface=MIXER,name='Right Line Mixer PGAR Bypass Switch' numid=69,iface=MIXER,name='Right Line1L Mux' numid=68,iface=MIXER,name='Right Line1R Mux' numid=61,iface=MIXER,name='Right PGA Mixer Line1L Switch' numid=60,iface=MIXER,name='Right PGA Mixer Line1R Switch' numid=62,iface=MIXER,name='Right PGA Mixer Mic2L Switch' numid=63,iface=MIXER,name='Right PGA Mixer Mic2R Switch' Thank you for your work and help !

1 0

[alsa-devel] [PATCH 00/10] *** alsabat features and fixes ***
by han.lu＠intel.com 15 Mar '16

15 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> add alsabat features and some fixes. 1. add default device name 2. add standalone mode 3. add tinyalsa support 4. add bash script for basic features test on alsa and tinyalsa 5. refactor alsa thread process to make clearer structure, and allow more common functions shared by alsa and tinyalsa 6. some tiny fixes on code, text and comments Lu, Han (10): alsabat: add default device name for playback and capture alsabat: add standalone mode alsabat: add tinyalsa support alsabat: clean the thread loopback of alsa capture alsabat: refactor wav file process alsabat: use common data generator function alsabat: add interrupt handler for shutdown alsabat: fix an incorrect print alsabat: use variable for thread return value alsabat: add bash test script bat/Makefile.am | 14 +- bat/alsa.c | 215 +++++++++-------------------- bat/alsa.h | 3 + bat/alsabat.1 | 12 ++ bat/bat.c | 80 ++++++++++- bat/common.c | 89 ++++++++++++ bat/common.h | 14 ++ bat/testbat.sh | 160 +++++++++++++++++++++ bat/tinyalsa.c | 422 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/tinyalsa.h | 23 +++ configure.ac | 9 +- 11 files changed, 883 insertions(+), 158 deletions(-) create mode 100644 bat/testbat.sh create mode 100644 bat/tinyalsa.c create mode 100644 bat/tinyalsa.h -- 2.5.0

3 24

[alsa-devel] [PATCH v5 0/9] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 15 Mar '16

15 Mar '16

Hi, these patches add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. Since there haven't been many comments after the last submission, the patches are mostly unchanged. A new patch adds HDMI dai-links to the recently merged mt8173-rt5650 machine driver and I have updated the HDMI notification patch (and renamed patch 3 to mention the mt8173-rt5650-rt5676 machine driver). Changes since v4: - Update hdmi notifier to latest version - Added HDMI dai-links to new mt8173-rt5650 machine driver The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/8337071/ ("drm/mediatek: Add HDMI support"), and need this patch: [2] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") The asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver PC Liao (1): ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Philipp Zabel (7): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver ASoC: mediatek: address dai link array entries by enum video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: hdmi-codec: Add ELD control drm/mediatek: hdmi: use helper function for N and CTS calculation .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 148 +++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 89 +++++----- drivers/gpu/drm/mediatek/mtk_hdmi.h | 10 +- drivers/gpu/drm/mediatek/mtk_hdmi_hw.c | 179 ++++----------------- drivers/gpu/drm/mediatek/mtk_hdmi_hw.h | 4 +- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 +++++++ include/linux/hdmi-notifier.h | 44 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/hdmi-codec.c | 123 +++++++++++++- sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 66 +++++++- sound/soc/mediatek/mt8173-rt5650.c | 47 ++++++ 18 files changed, 587 insertions(+), 217 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.7.0

4 13

Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Mark Brown 14 Mar '16

14 Mar '16

On Fri, Mar 11, 2016 at 02:27:03AM +0530, Jeeja KP wrote: > On Wed, Mar 02, 2016 at 11:02:15AM +0900, Mark Brown wrote: > > On Tue, Feb 16, 2016 at 09:47:47PM +0530, Jeeja KP wrote: > > Like I keep saying if you're thinking about this in terms of DPCM you're > > solving the wrong problem - we already support supplying parameters for > > device<->device links, we already have systems that need to configure > > things for device<->device links so we can already see that if we're > > doing something that only works for things that are on SoC and can use > > DPCM then we're not solving the problem. > No this is not exactly about DPCM. It is about the device<->device links that > assume you can have only codecs as hostless and thus change the direction > for the link. Since we are rendering from SoC we don't need direction change. You're routing them via DSP though? > > This is clearly confused, if we can't figure out what the two devices we > > are connecting are based on just looking at the devices then that's > > really not a good sign that our interfaces are sensible and easy to work > > with. > Right now dai_link assumes device<->device links based on params so we > cannot specify params for anything where we dont want direction change and would > like to treat as normal direction dai_link. So we propose this to move to > codec_link flag and use params only for creating controls based on params > provided. What I'm telling you is that you shouldn't be writing hacks like this to bodge things but coming up with scalable and maintainable solutions. I already discussed this with Vinod several times, we've discussed the idea that if you're having trouble with the asymmetries in the definitions of DAI links you should make the DAI links symmetric so things like this just work. We need clear, comprehensible abstractions which work robustly not some mess of special case handling.

2 1

[alsa-devel] [PATCH] topology: fix debug output to print correct "max" value.
by Liam Girdwood 14 Mar '16

14 Mar '16

Debug log is printing num_regs instead of max in the max section. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/ctl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/topology/ctl.c b/src/topology/ctl.c index 1c073f7..c250227 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -350,7 +350,7 @@ int tplg_parse_control_bytes(snd_tplg_t *tplg, return -EINVAL; be->max = atoi(val); - tplg_dbg("\t%s: %d\n", id, be->num_regs); + tplg_dbg("\t%s: %d\n", id, be->max); continue; } -- 2.5.0

2 9

[alsa-devel] [GIT PULL] ASoC updates for v4.6
by Mark Brown 14 Mar '16

14 Mar '16

The following changes since commit f6cede5b49e822ebc41a099fe41ab4989f64e2cb: Linux 4.5-rc7 (2016-03-06 14:48:03 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.6 for you to fetch changes up to d4a6360f19c1c551afcba42be98df04651fab31b: Merge remote-tracking branches 'asoc/topic/sunxi', 'asoc/topic/topology' and 'asoc/topic/wm8974' into asoc-next (2016-03-13 15:17:56 +0700) ---------------------------------------------------------------- ASoC: Updates for v4.6 The main thing in terms of the core this time around has been some additional framework work for dynamic topologies (though we *still* don't appear to have a stable ABI for the topology code, it's probably worth considering if this will ever happen...). Otherwise the work has almost all been in the drivers: - HDMI support for Sky Lake, along with other fixes and enhancements for the Intel drivers. - Lots of improvements to the Renesas drivers. - Capture support for Qualcomm drivers. - Support for TI DaVinci DRA7xxx devices. - New machine drivers for Freescale systems with Cirrus CODECs, Mediatek systems with RT5650 CODECs. - New CPU drivers for Allwinner S/PDIF controllers - New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514. ---------------------------------------------------------------- Alan Cox (1): ASoC: Intel: Skylake: remove bogus comparison of an array with NULL Andreas Irestål (4): ASoC: adau17x1: Cache writes when core clock is disabled ASoC: adau17x1: Correct typos in file headers ASoC: adau17x1: Add adau17x1 bindings document ASoC: adau17x1: Add basic DT support for adau17x1 Andrzej Hajda (1): ASoC: max9867: fix type of variable containing error codes Antonio Ospite (1): ASoC: rk3036: fix missing dependency on REGMAP_MMIO Arnd Bergmann (3): ASoC: s3c24xx: use const snd_soc_component_driver pointer ASoC: pxa: remove unused variable ASoC: rockchip: use __maybe_unused to hide st_irq_syscfg_resume Axel Lin (7): ASoC: rt5659: Staticise rt5659_i2c_shutdown ASoC: max98926: Fix blr_clk_ratio calculation ASoC: max98926: Constify max98926_reg and max98926_regmap ASoC: max9867: Add CONFIG_SND_SOC_MAX9867 to build ASoC: max9867: Make max9867_i2c_id NULL terminated ASoC: max9867: Constify max9867_reg and max9867_regmap ASoC: max9867: Use params_width instead of open-coded Bard Liao (4): ASoC: rt298: enable IRQ for jack detection ASoC: rt298: Don't enable IRQ in i2c_probe ASoC: rt5640: Set PLL src according to source ASoC: rt5640: add supplys for dac power Caesar Wang (4): ASoC: rt5616: add mclk property for rt5616 document ASoC: rt5616: trivial: fix the typo ASoC: rt5616: add the mclk for the codec driver ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616 Carlo Caione (1): ASoC: cht_bsw_rt5645: Enable jack detection Charles Keepax (14): ASoC: wm5110: Fix up snd_soc_register_platform error path ASoC: wm_adsp: Use more generic naming for regions and caps definitions ASoC: wm_adsp: Add debug audio trace firmware ASoC: wm_adsp: Move setting of DSP speed into CODEC specific code ASoC: wm5110: Add support for audio trace firmware ASoC: wm5102: Add support for the audio trace firmware ASoC: wm_adsp: Add missing const from region definitions ASoC: wm_adsp: Return sample rate in wm_adsp_compr_pointer ASoC: arizona: Add support for SNDRV_PCM_RATE_KNOT ASoC: wm_adsp: Avoid narrow race condition on compr pointer ASoC: wm_adsp: Explicitly set the control access flags in all cases ASoC: wm_adsp: Fixup some odd line wrapping ASoC: wm_adsp: wm_coeff_{read|write}_control should use passed length ASoC: wm_adsp: Use correct local length in error message Damien.Horsley (1): ASoC: pcm3168a: Call clk_set_rate in pcm3168a_set_dai_sysclk Dan Carpenter (1): ASoC: max9867: silence and array overflow warning Dharageswari.R (3): ASoC: Intel: Skylake: Get node id using helper ASoC: Intel: Skylake: Add skl_tplg_be_get_cpr_module() helper ASoC: Intel: Skylake: Add MCLK configuration Fabio Estevam (1): ASoC: hdac_hdmi: Remove 'edev' NULL check Felipe F. Tonello (1): ASoC: fsl-asoc-card: add cs4271 and cs4272 support Florian Vaussard (2): ASoC: ads117x: Add bindings documentation for TI ADS117x ADC ASoC: ads117x: Add device tree compatible string Jacob Siverskog (3): ASoC: pcm179x: Split into core and SPI parts ASoC: pcm179x: Add I2C interface driver ASoC: pcm179x: Support continuous rates Jayachandran B (1): ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core Jeeja KP (16): ASoC: Intel: Skylake: Allow module parameter set after bind ASoC: Intel: Skylake: Add shutdown callback ASoC: Intel: Skylake: Fill BE blob with default params ASoC: hdac_hdmi: Add jack reporting ASoC: hdac_hdmi: Add PM support ASoC: Intel: boards: Enable HDMI and DP on skl_rt286 machine ASoC: Intel: boards: Enable HDMI and DP on nau88l2 machine ASoC: Intel: boards: Remove WoV sink in skl_rt286 ASoC: Intel: boards: Remove WoV sink in nau88125_ssm4567 machine ASoC: Intel: boards: Remove WoV sink in nau88125_max98357a machine ASoC: Intel: Skylake: correct direction in blob query ASoC: Intel: Skylake: Add IVI feedback path in nau88125_ssm4567 machine ASoC: Intel: Skylake: remove IVI codec loop in skl_nau88l25_ssm4567 machine ASoC: Intel: add dmabuffer to common sst_dsp ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops ASoC: hdac_hdmi: Add broxton device ID Jianqun Xu (1): ASoC: rockchip: add bindings for rk3399 i2s John Hsu (1): ASoC: nau8825: fix issue that pop noise when start playback John Lin (3): ASoC: rt5645: improve headphone pop when system resumes from S3 ASoC: rt5645: extend delay time for headphone pop noise ASoC: rt5616: add missing mute control for HPVOL Koro Chen (3): ASoC: mediatek: Add machine driver for ALC5650 codec ASoC: mediatek: Add I03/I04 widgets and corresponding routes ASoC: mediatek: Add machine driver for rt5650 rt5514 codec Kuninori Morimoto (59): ASoC: rsnd: remove unsed *parent ASoC: rsnd: use rsnd_mod_init() for ADG ASoC: rsnd: add debug message for rsnd_mod_call() ASoC: rsnd: don't update status if rsnd_mod_call() doesn't match ASoC: rsnd: try to connect connected mod is not error ASoC: rsnd: always call probe/remove for MUX ASoC: rsnd: don't overwrite io on rsnd_cmd_init() ASoC: rsnd: select each SRC correctly for CMD data path ASoC: rsnd: each mod has status again for CTU/MUX support ASoC: rsnd: attach Audio-DMAC-periperi correctly ASoC: rsnd: ctu: add rsnd_mix_activation() ASoC: rsnd: ctu: add rsnd_ctu_halt() ASoC: rsnd: ctu: settings matches to datasheet ASoC: rsnd: tidyup Playback/Capture sequence ASoC: rsnd: disable SRC.out/in in same time ASoC: rsnd: don't auto-recover when under/over run error ASoC: rsnd: add .irq callback ASoC: soc-pcm: copy dpcm->hw_params and call be_hw_params_fixup every time ASoC: rsnd: rsnd_ssi_is_multi_slave() macro uses rsnd_ssi_multi_slaves() ASoC: rsnd: rename RSND_DVC_CHANNELS to RSND_MAX_CHANNELS ASoC: rsnd: setup SRC_ROUTE_MODE0 before SRC_SRCIR ASoC: rsnd: indicates status failed SSI ASoC: rsnd: SSI function parameter uses "mod" instead of "ssi" ASoC: rsnd: fixup forever loop bug on SSI ASoC: rsnd: add missing .irq callback for DMA ASoC: rsnd: tidyup SSI init/start sequence ASoC: rsnd: Parent SSI attach is not needed if not clock master ASoC: rsnd: move rsnd_ssi_irq() position ASoC: rsnd: judge multi SSI in runtime ASoC: rsnd: judge work SSI in runtime regmap: add regmap_update_bits_base() regmap: merge regmap_update_bits() into macro regmap: merge regmap_update_bits_async() into macro regmap: merge regmap_update_bits_check() into macro regmap: merge regmap_update_bits_check_async() into macro regmap: add regmap_field_update_bits_base() regmap: merge regmap_field_write() into macro regmap: merge regmap_field_update_bits() into macro regmap: add regmap_fields_update_bits_base() regmap: merge regmap_fields_write() into macro regmap: merge regmap_fields_update_bits() into macro regmap: add regmap_field_force_xxx() macros regmap: add regmap_fields_force_xxx() macros ASoC: rsnd: rsnd_write() / rsnd_bset() uses regmap _force_ function ASoC: rsnd: don't call update callback if it was NULL ASoC: rsrc-card: add convert channels support ASoC: rsnd: tidyup rsnd_ssiu_init_gen2() ASoC: rsnd: add CTU basic support ASoC: rsnd: add rsnd_runtime_channel_xxx() ASoC: rsnd: add CTU support ASoC: rsrc-card: tidyup convert-channels explain ASoC: rsnd: don't enable SRC Sync mode when Capture with CMD ASoC: rsnd: SRC sync mode is not related to clock master mode ASoC: rsnd: SRC function parameter uses "mod" instead of "src" ASoC: rsnd: rename rsnd_enable_sync_convert() to rsnd_src_sync_is_enabled() ASoC: rsnd: add rsnd_src_get_in/out_rate() ASoC: rsnd: SRC TIMSEL support for Capture ASoC: rsnd: CMD TIMSEL support for Capture ASoC: rsnd: add simplified module explanation Lars-Peter Clausen (1): ASoC: ssm4567: Reset device before regcache_sync() Lukas Wunner (1): ASoC: Intel: Use acpi_dev_present() Maciej S. Szmigiero (1): ASoC: fsl_ssi: remove explicit register defaults Mans Rullgard (2): ASoC: wm8974: configure pll and mclk divider automatically ASoC: mxs-saif: check BUSY bit in hw_params() only if not mclk_in_use Marcus Cooper (2): ASoC: sunxi: add sun4i SPDIF transceiver ASoC: sunxi: Add support for the SPDIF block Mark Brown (31): regmap: Add explict native endian flag to DT bindings MIPS: dt: Explicitly specify native endian behaviour for syscon regmap: mmio: Convert to regmap_bus and fix accessor usage regmap: Return an error if a caller attempts to do an unsupported raw read regmap: cache: Fall back to register by register read for cache defaults Merge branch 'fix/mmio' of git://git.kernel.org/.../broonie/regmap into regmap-mmio Merge branch 'fix/intel' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/drm' of git://git.kernel.org/.../broonie/sound into asoc-hdac Merge tag 'asoc-fix-v4.5-rc4' into asoc-qcom Merge branch 'topic/mmio' of git://git.kernel.org/.../broonie/regmap into asoc-fsl-ssl Merge tag 'v4.5-rc5' into asoc-mtk Merge branch 'topic/hdac' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/update-bits' of git://git.kernel.org/.../broonie/regmap into asoc-rcar Merge tag 'asoc-fix-v4.5-rc6' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/core-pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm179x' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3168a' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ab8500', 'asoc/topic/adau17x1', 'asoc/topic/ads117x', 'asoc/topic/adsp' and 'asoc/topic/arizona' into asoc-next Merge remote-tracking branches 'asoc/topic/atmel', 'asoc/topic/bcm2835' and 'asoc/topic/cs42xx8' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/fsl-card' and 'asoc/topic/fsl-mpc5200' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-sai', 'asoc/topic/fsl-ssl', 'asoc/topic/hdac' and 'asoc/topic/max9867' into asoc-next Merge remote-tracking branches 'asoc/topic/max98926', 'asoc/topic/mtk', 'asoc/topic/mxs-saif', 'asoc/topic/nau8825' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa' and 'asoc/topic/qcom' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/rockchip', 'asoc/topic/rt298', 'asoc/topic/rt5514' and 'asoc/topic/rt5616' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5640', 'asoc/topic/rt5659', 'asoc/topic/samsung' and 'asoc/topic/ssm4567' into asoc-next Merge remote-tracking branches 'asoc/topic/sunxi', 'asoc/topic/topology' and 'asoc/topic/wm8974' into asoc-next Martin Sperl (1): ASoC: bcm2835: move to use the clock framework Mengdong Lin (4): ALSA: pcm: Add snd_pcm_rate_range_to_bits() ASoC: topology: Add FE DAIs dynamically ASoC: topology: Add FE DAI links dynamically ASoC: topology: Fix setting of stream rates, rate_min and rate_max Nicolin Chen (1): ASoC: fsl-asoc-card: Don't add DAPM routes for ASRC if it doesn't exist Oder Chiou (1): ASoC: rt5514: add rt5514 codec driver PC Liao (2): ASoC: mediatek: Enable 33bit memory address to support 4GB DRAM ASoC: mediatek: remove soft reset and add second I2S clock Peter Ujfalusi (4): ASoC: davinci-mcasp: Discourage use of fck_parent for clock reparenting ASoC: davinci-mcasp: dai format runtime reconfiguration ASoC: davinci: Kconfig: Update the edma-pcm section's dependency and help ASoC: omap-hdmi-audio: Support for DRA7xx family Philipp Zabel (1): ASoC: mediatek: address dai link array entries by enum Pierre-Louis Bossart (5): ASoC: Intel: common: filter ACPI devices with _STA return value ASoC: Intel: common: add translation from HID to codec-name ASoC: Intel: bytcr_rt5640: change quirk position ASoC: Intel: bytcr_rt5640: use HID translation util ASoC: Intel: Atom: add support for CHT w/ RT5640 Rajan Vaja (1): ASoC: dapm: Remove unneeded variable ret returning 0 Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Richard Fitzgerald (2): ASoC: cs47l24: Add voice control compressed stream ASoC: arizona: Make logging of FLL calculations clearer Sathya Prakash M R (1): ASoC: Intel: boards: Add IV feedback route for nau88l25_ssm4567 machine Sathyanarayana Nujella (1): ASoC: Intel: boards: Enable HDMI and DP on SKL nau88l25_max98357 machine Senthilnathan Veppur (1): ASoC: Intel: Bxtn: Add Broxton PCI ID Songjun Wu (1): ASoC: atmel_ssc_dai: distinguish the different SSC Srinivas Kandagatla (14): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma register definitions ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: add mic support ASoC: qcom: apq8016-sbc: add mic support ASoC: qcom: apq8016: add wrdma support Stephen Boyd (1): ASoC: qcom: Don't specify LE device endianness Subhransu S. Prusty (15): ASoC: hdac_hdmi: Add hotplug notification and read ELD ASoC: hdac_hdmi: Apply constraints based on ELD drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi: create dais based on number of cvts ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Enable playback on all enumerated ports ASoC: hdac_hdmi: Add infoframe support for dp audio ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume ASoC: Intel: Skylake: Fix possible memory corruption in codec dai dma params ASoC: Intel: Skylake: Add HDMI FE and BE cpu dais ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec ASoC: hdac_hdmi: Fix infoframe programming Sudip Mukherjee (6): ASoC: topology: fix build warning ASoC: fix memory leak ASoC: intel: remove unused variable ASoC: pcm3168a: remove unused variable ASoC: rt5640: remove unused variable ASoC: qcom: fix build error Sugar Zhang (3): ASoC: rockchip: i2s: add default values for registers ASoC: rt5640: add master clock handling for rt5640 ASoC: rockchip: add bindings for spdif controller Vinod Koul (5): ASoC: Intel: Skylake: Update the delay check ASoC: Intel: Skylake: Generate topology name for NHLT table header ASoC: Intel: Skylake: Add i915 enabling in skl probe ASoC: Intel: Skylake: Move module id defines ASoC: Intel: Skylake: Fix whitepsace issues Wolfram Sang (2): ASoC: sh: rcar: core: don't open code of_device_get_match_data() ASoC: sh: rcar: rsrc-card: don't open code of_device_get_match_data() Xing Zheng (1): ASoC: rt5616: Add support sample rate to 192KHz Xiubo Li (3): ASoC: ab8500: remove the useless 'break' after 'return' ASoC: fsl_sai: Use usleep_range() instead of msleep() ASoC: fsl-mpc5200: Use usleep_range() instead of msleep() Zidan Wang (1): ASoC: cs42xx8: fix the noise in the right dac channel with mono playback anish kumar (3): ASoC: Add max98926 codec driver ASoC: Add max9867 codec driver ASoC: max9867: Changed the if-else case to switch case .../devicetree/bindings/regmap/regmap.txt | 11 +- .../devicetree/bindings/sound/adi,adau17x1.txt | 24 + .../devicetree/bindings/sound/fsl-asoc-card.txt | 9 + .../devicetree/bindings/sound/max9867.txt | 17 + .../devicetree/bindings/sound/max98926.txt | 32 + .../bindings/sound/mt8173-rt5650-rt5514.txt | 15 + .../devicetree/bindings/sound/mt8173-rt5650.txt | 15 + .../devicetree/bindings/sound/pcm179x.txt | 11 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 340 ++++++ .../bindings/sound/renesas,rsrc-card.txt | 1 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + .../devicetree/bindings/sound/rockchip-spdif.txt | 8 +- Documentation/devicetree/bindings/sound/rt5514.txt | 25 + Documentation/devicetree/bindings/sound/rt5616.txt | 6 + Documentation/devicetree/bindings/sound/rt5640.txt | 3 + .../bindings/sound/sunxi,sun4i-spdif.txt | 39 + .../devicetree/bindings/sound/ti,ads117x.txt | 11 + arch/mips/boot/dts/brcm/bcm6328.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm6368.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7125.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7346.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7358.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7360.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7362.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7420.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7425.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7435.dtsi | 2 +- drivers/base/regmap/regcache.c | 41 +- drivers/base/regmap/regmap-mmio.c | 259 +++-- drivers/base/regmap/regmap.c | 211 +--- drivers/misc/atmel-ssc.c | 1 + include/drm/drm_edid.h | 12 + include/linux/platform_data/adau17x1.h | 2 +- include/linux/regmap.h | 85 +- include/sound/pcm.h | 2 + include/sound/soc-topology.h | 21 +- include/sound/soc.h | 2 +- include/trace/events/asoc.h | 8 +- sound/core/pcm_misc.c | 30 + sound/soc/atmel/atmel_ssc_dai.c | 27 +- sound/soc/bcm/bcm2835-i2s.c | 284 +---- sound/soc/codecs/Kconfig | 38 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 9 +- sound/soc/codecs/adau1761-i2c.c | 14 +- sound/soc/codecs/adau1761-spi.c | 14 +- sound/soc/codecs/adau1761.c | 10 +- sound/soc/codecs/adau1781-i2c.c | 10 + sound/soc/codecs/adau1781-spi.c | 10 + sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.h | 6 +- sound/soc/codecs/ads117x.c | 12 + sound/soc/codecs/arizona.c | 71 +- sound/soc/codecs/arizona.h | 4 +- sound/soc/codecs/cs42l51.c | 8 +- sound/soc/codecs/cs42xx8.c | 10 +- sound/soc/codecs/cs47l24.c | 123 +- sound/soc/codecs/da732x.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 1219 ++++++++++++++++++-- sound/soc/codecs/hdac_hdmi.h | 6 + sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 +- sound/soc/codecs/max9867.c | 546 +++++++++ sound/soc/codecs/max9867.h | 83 ++ sound/soc/codecs/max98926.c | 606 ++++++++++ sound/soc/codecs/max98926.h | 848 ++++++++++++++ sound/soc/codecs/nau8825.c | 169 ++- sound/soc/codecs/nau8825.h | 16 +- sound/soc/codecs/pcm179x-i2c.c | 73 ++ sound/soc/codecs/pcm179x-spi.c | 72 ++ sound/soc/codecs/pcm179x.c | 56 +- sound/soc/codecs/pcm179x.h | 9 +- sound/soc/codecs/pcm3168a.c | 8 +- sound/soc/codecs/rt298.c | 7 +- sound/soc/codecs/rt298.h | 8 + sound/soc/codecs/rt5514.c | 982 ++++++++++++++++ sound/soc/codecs/rt5514.h | 252 ++++ sound/soc/codecs/rt5616.c | 414 ++++--- sound/soc/codecs/rt5640.c | 91 +- sound/soc/codecs/rt5640.h | 2 + sound/soc/codecs/rt5645.c | 15 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/ssm4567.c | 5 + sound/soc/codecs/tlv320dac33.c | 9 +- sound/soc/codecs/wl1273.c | 13 +- sound/soc/codecs/wm5102.c | 93 +- sound/soc/codecs/wm5110.c | 75 +- sound/soc/codecs/wm8753.c | 6 +- sound/soc/codecs/wm8904.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 8 +- sound/soc/codecs/wm8974.c | 93 ++ sound/soc/codecs/wm8983.c | 14 +- sound/soc/codecs/wm8985.c | 14 +- sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/codecs/wm9081.c | 8 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_adsp.c | 139 ++- sound/soc/codecs/wm_adsp.h | 8 +- sound/soc/davinci/Kconfig | 3 +- sound/soc/davinci/davinci-mcasp.c | 12 + sound/soc/fsl/Kconfig | 4 +- sound/soc/fsl/fsl-asoc-card.c | 26 +- sound/soc/fsl/fsl_sai.c | 3 +- sound/soc/fsl/mpc5200_psc_ac97.c | 3 +- sound/soc/intel/Kconfig | 4 + sound/soc/intel/atom/sst/sst_acpi.c | 4 + sound/soc/intel/atom/sst/sst_ipc.c | 2 - sound/soc/intel/boards/bytcr_rt5640.c | 50 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 17 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 34 +- sound/soc/intel/boards/mfld_machine.c | 16 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 122 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 130 ++- sound/soc/intel/boards/skl_rt286.c | 113 +- sound/soc/intel/common/sst-acpi.h | 3 + sound/soc/intel/common/sst-dsp-priv.h | 1 + sound/soc/intel/common/sst-match-acpi.c | 45 +- sound/soc/intel/skylake/skl-messages.c | 124 +- sound/soc/intel/skylake/skl-nhlt.c | 34 + sound/soc/intel/skylake/skl-pcm.c | 106 +- sound/soc/intel/skylake/skl-sst-dsp.c | 14 +- sound/soc/intel/skylake/skl-sst-dsp.h | 7 +- sound/soc/intel/skylake/skl-sst.c | 3 - sound/soc/intel/skylake/skl-topology.c | 210 +++- sound/soc/intel/skylake/skl-topology.h | 27 + sound/soc/intel/skylake/skl-tplg-interface.h | 3 +- sound/soc/intel/skylake/skl.c | 88 ++ sound/soc/intel/skylake/skl.h | 15 +- sound/soc/mediatek/Kconfig | 22 +- sound/soc/mediatek/Makefile | 2 + sound/soc/mediatek/mt8173-rt5650-rt5514.c | 258 +++++ sound/soc/mediatek/mt8173-rt5650-rt5676.c | 18 +- sound/soc/mediatek/mt8173-rt5650.c | 236 ++++ sound/soc/mediatek/mtk-afe-common.h | 1 + sound/soc/mediatek/mtk-afe-pcm.c | 47 +- sound/soc/mxs/mxs-saif.c | 2 +- sound/soc/omap/n810.c | 18 +- sound/soc/omap/omap-hdmi-audio.c | 1 + sound/soc/omap/rx51.c | 18 +- sound/soc/pxa/brownstone.c | 2 - sound/soc/pxa/corgi.c | 12 +- sound/soc/pxa/magician.c | 6 +- sound/soc/pxa/poodle.c | 12 +- sound/soc/pxa/spitz.c | 12 +- sound/soc/pxa/tosa.c | 12 +- sound/soc/qcom/Kconfig | 7 +- sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +- sound/soc/qcom/lpass-cpu.c | 147 ++- sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 116 +- sound/soc/qcom/lpass-platform.c | 244 ++-- sound/soc/qcom/lpass.h | 10 +- sound/soc/rockchip/rockchip_i2s.c | 13 + sound/soc/rockchip/rockchip_spdif.c | 17 +- sound/soc/samsung/i2s.c | 21 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c-i2s-v2.h | 2 +- sound/soc/sh/rcar/adg.c | 218 ++-- sound/soc/sh/rcar/cmd.c | 18 +- sound/soc/sh/rcar/core.c | 160 ++- sound/soc/sh/rcar/ctu.c | 281 ++++- sound/soc/sh/rcar/dma.c | 56 +- sound/soc/sh/rcar/dvc.c | 33 +- sound/soc/sh/rcar/gen.c | 56 +- sound/soc/sh/rcar/mix.c | 4 +- sound/soc/sh/rcar/rsnd.h | 115 +- sound/soc/sh/rcar/rsrc-card.c | 27 +- sound/soc/sh/rcar/src.c | 171 ++- sound/soc/sh/rcar/ssi.c | 334 +++--- sound/soc/sh/rcar/ssiu.c | 20 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dapm.c | 12 +- sound/soc/soc-pcm.c | 24 +- sound/soc/soc-topology.c | 240 ++-- sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 550 +++++++++ 181 files changed, 10507 insertions(+), 2285 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau17x1.txt create mode 100644 Documentation/devicetree/bindings/sound/max9867.txt create mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5514.txt create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,ads117x.txt create mode 100644 sound/soc/codecs/hdac_hdmi.h create mode 100755 sound/soc/codecs/max9867.c create mode 100755 sound/soc/codecs/max9867.h create mode 100644 sound/soc/codecs/max98926.c create mode 100644 sound/soc/codecs/max98926.h create mode 100644 sound/soc/codecs/pcm179x-i2c.c create mode 100644 sound/soc/codecs/pcm179x-spi.c create mode 100644 sound/soc/codecs/rt5514.c create mode 100644 sound/soc/codecs/rt5514.h create mode 100644 sound/soc/mediatek/mt8173-rt5650-rt5514.c create mode 100644 sound/soc/mediatek/mt8173-rt5650.c create mode 100644 sound/soc/sunxi/sun4i-spdif.c

2 1

Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Mark Brown 14 Mar '16

14 Mar '16

On Mon, Mar 14, 2016 at 11:27:42PM +0530, Jeeja KP wrote: > We discussed this and yes we are looking at ways to solve this longer > term with DAPM but today DAPM doesn't help us in managing DMAs for our > controller. I believe Vinod did discuss this limitation during ELC. > We are starting this discussion internally and will try to solve > this part. We discussed using CODEC-CODEC links for this. If you want to add DMA operations to them then that's a simple matter of programming (which IIRC we did discuss). > Now for us to mange the internal sinks and source how should we go about > modelling them in DAPM and provide hw_params for the configuration? In the same way as anyone not doing it on a SoC does; at the minute that means working out any settings that need configuration by looking at what happens when the signal goes out onto an audio interface. Ideally we'd have a mechanism for propagating digital configuration along audio paths.

1 0

[alsa-devel] [PATCH - tinycompress - brace issue 1/1] tinycompress: fix compilation error: missing brace issue
by francois.gaffie＠intel.com 14 Mar '16

14 Mar '16

From: François Gaffie <francois.gaffie(a)intel.com> Signed-off-by: François Gaffie <francois.gaffie(a)intel.com> diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 68626a4..d35801b 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -273,4 +273,7 @@ int is_compress_ready(struct compress *compress); /* Returns a human readable reason for the last error */ const char *compress_get_error(struct compress *compress); +#if defined(__cplusplus) +} +#endif #endif diff --git a/include/tinycompress/tinymp3.h b/include/tinycompress/tinymp3.h index a709c39..7015320 100644 --- a/include/tinycompress/tinymp3.h +++ b/include/tinycompress/tinymp3.h @@ -102,5 +102,7 @@ enum mp3_stereo_mode { DUAL = 0x02, MONO = 0x03 }; - +#if defined(__cplusplus) +} +#endif #endif -- 2.5.0 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

1 0

[alsa-devel] [PATCH] ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
by Subhransu S. Prusty 14 Mar '16

14 Mar '16

nvhdmi_chmap_cea_alloc_validate_get_type calls itself recursively using chmap ops causing the double fault. Fixed by adding the default validate_get_type handling inside nvdia validate_get_type handler. Link: https://bugzilla.kernel.org/show_bug.cgi?id=114311 Fixes: 67b90cb ("ALSA: hda - Create common chmap object") Reported-by: Andreas Reis <andreas.reis(a)gmail.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Tested-by: Andreas Reis <andreas.reis(a)gmail.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f59e87b..a386a65 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2827,8 +2827,12 @@ static int nvhdmi_chmap_cea_alloc_validate_get_type(struct hdac_chmap *chmap, if (cap->ca_index == 0x00 && channels == 2) return SNDRV_CTL_TLVT_CHMAP_FIXED; - return chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, channels); + /* If the speaker allocation matches the channel count, it is OK. */ + if (cap->channels != channels) + return -1; + + /* all channels are remappable freely */ + return SNDRV_CTL_TLVT_CHMAP_VAR; } static int nvhdmi_chmap_validate(struct hdac_chmap *chmap, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
by Vittorio Gambaletta (VittGam) 14 Mar '16

14 Mar '16

The clock measurement on the AC'97 audio card found in the IBM ThinkPad X41 will often fail, so add a quirk entry to fix it. Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=441087 Cc: <stable(a)vger.kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Vittorio Gambaletta <linuxbugs(a)vittgam.net> --- When the clock measurement doesn't fail, the `pos` variable after line 2861 will be around 48367, but when it does fail it will be somewhere between 45000 and 47500. So the bus clock gets set between 48500 and 52000 (!), resulting in faster audio for rates with final clocks below 48000, and distorted audio for rates with final clocks above 48000. The problem is always reproducible on kernel 4.4.5, but very rarely on 3.13. This patch fixes the problem for my computer, and will also speed up module loading a bit as said on Bugzilla, but I've got some questions regarding other hardware... Do we want to protect against overclock for all hardware, by never setting the measured bus clock higher than 48000? Do we actually risk burning or otherwise permanently ruining something here, or not? Does hardware clocked above 48000 exist? Also, is line 2874 actually correct? As `pos` gets higher, the bus clock gets lower, and vice versa. Note that in this function the bus clock is always 48000 before it gets updated here; so the code first computes 48000 * 48000, and then it divides the result by `pos`. It doesn't seem right to me... By the way, the history of that line goes beyond Git (so before kernel 2.6.12-rc2). --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2879,6 +2879,7 @@ static struct snd_pci_quirk intel8x0_clock_list[] = { SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000), + SND_PCI_QUIRK(0x1014, 0x0581, "AD1981B", 48000), SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100), SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000), SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),

2 1

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
by Aaron Plattner 14 Mar '16

14 Mar '16

Vendor ID 0x10de0082 is used by a yet-to-be-named GPU chip. This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is appropriate here. Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- Another day, another new chip with the same stupid bug. The hardware folks still claim they're going to fix it someday. sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bcbc4ee10130..651a9ee494db 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3659,6 +3659,7 @@ HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0082, "GPU 82 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0083, "GPU 83 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), -- 2.7.3

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: dai format runtime reconfiguration
by Peter Ujfalusi 13 Mar '16

13 Mar '16

In case when the dai format is set via the dai_link the format configuration happens once when the links are probed. If the McASP lose context after this, the information will be lost and McASP will not going to work correctly. To overcome this issue, we save the fmt and set it within hw_params as well. Reported-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1bd86c8e7bc8..b0d1b9f93fa1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -78,6 +78,7 @@ struct davinci_mcasp { u32 fifo_base; struct device *dev; struct snd_pcm_substream *substreams[2]; + unsigned int dai_fmt; /* McASP specific data */ int tdm_slots; @@ -399,6 +400,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, bool fs_pol_rising; bool inv_fs = false; + if (!fmt) + return 0; + pm_runtime_get_sync(mcasp->dev); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: @@ -530,6 +534,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); } + + mcasp->dai_fmt = fmt; out: pm_runtime_put(mcasp->dev); return ret; @@ -1048,6 +1054,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; + ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt); + if (ret) + return ret; + /* * If mcasp is BCLK master, and a BCLK divider was not provided by * the machine driver, we need to calculate the ratio. -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: davinci: Kconfig: Update the edma-pcm section's dependency and help
by Peter Ujfalusi 13 Mar '16

13 Mar '16

Instead of depending on individual SoCs make the edma-pcm depend on the eDMA dmaengine driver (TI_EDMA). Update the help text and add DRA7xx family since they have eDMA integrated as well along with sDMA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index e0f6581494bf..a834558fd01e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -5,7 +5,7 @@ config SND_DAVINCI_SOC config SND_EDMA_SOC tristate "SoC Audio for Texas Instruments chips using eDMA" - depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI + depends on TI_EDMA select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. @@ -13,6 +13,7 @@ config SND_EDMA_SOC - daVinci devices - AM335x - AM437x/AM438x + - DRA7xx family config SND_DAVINCI_SOC_I2S tristate -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: qcom: fix build error
by Sudip Mukherjee 13 Mar '16

13 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> While building m32r allmodconfig the build failed with: ERROR: "bad_dma_ops" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! ERROR: "dma_common_mmap" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! To satisfy the dependency CONFIG_SND_SOC_LPASS_PLATFORM should depend on HAS_DMA. Some other configs also needs the dependency on HAS_DMA as they are directly or indirectly selecting SND_SOC_LPASS_PLATFORM. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- m32r build log at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/115245240 sound/soc/qcom/Kconfig | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 3cc252e..8ec9a07 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -11,21 +11,24 @@ config SND_SOC_LPASS_CPU config SND_SOC_LPASS_PLATFORM tristate + depends on HAS_DMA select REGMAP_MMIO config SND_SOC_LPASS_IPQ806X tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_APQ8016 tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" - depends on SND_SOC_QCOM + depends on SND_SOC_QCOM && HAS_DMA select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -34,7 +37,7 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" - depends on SND_SOC_QCOM + depends on SND_SOC_QCOM && HAS_DMA select SND_SOC_LPASS_APQ8016 help Support for Qualcomm Technologies LPASS audio block in -- 1.9.1

2 1

[alsa-devel] [PATCH v4] ASoC: nau8825: fix issue that pop noise when start playback
by Ben Zhang 13 Mar '16

13 Mar '16

From: John Hsu <KCHSU0(a)nuvoton.com> Reduce pop noise in power up and down sequence when playback. The DAPM widgets graph is reconstructed to ensure the register write sequence at playback matches exactly to the v5 clickless sequence provided by Nuvoton. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 169 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/nau8825.h | 16 ++++- 2 files changed, 141 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c1b87c5..1c87299 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -84,6 +84,7 @@ static const struct nau8825_fll_attr fll_pre_scalar[] = { static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_IIC_ADDR_SET, 0x0 }, { NAU8825_REG_CLK_DIVIDER, 0x0050 }, { NAU8825_REG_FLL1, 0x0 }, { NAU8825_REG_FLL2, 0x3126 }, @@ -158,8 +159,7 @@ static const struct reg_default nau8825_reg_defaults[] = { static bool nau8825_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { - case NAU8825_REG_ENA_CTRL: - case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_ENA_CTRL ... NAU8825_REG_FLL_VCO_RSV: case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: @@ -184,8 +184,7 @@ static bool nau8825_readable_reg(struct device *dev, unsigned int reg) static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) { switch (reg) { - case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: - case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_RESET ... NAU8825_REG_FLL_VCO_RSV: case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: case NAU8825_REG_INTERRUPT_MASK: case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: @@ -227,10 +226,42 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) static int nau8825_pump_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: /* Prevent startup click by letting charge pump to ramp up */ msleep(10); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, NAU8825_JAMNODCLOW); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8825_output_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Disables the TESTDAC to let DAC signal pass through. */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDAC_EN); break; default: return -EINVAL; @@ -316,10 +347,10 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, NAU8825_SAR_ADC_EN_SFT, 0), - SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), - SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), - SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL Clock", 3, NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR Clock", 3, NAU8825_REG_RDAC, 9, 0, NULL, 0), SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR_SFT, 0), @@ -330,29 +361,48 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), - SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, - 0), + SND_SOC_DAPM_PGA_S("HP amp L", 0, + NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP amp R", 0, + NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, - nau8825_pump_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("Charge Pump", 1, NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_pump_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA("Output Driver R Stage 1", + SND_SOC_DAPM_PGA_S("Output Driver R Stage 1", 4, NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver L Stage 1", + SND_SOC_DAPM_PGA_S("Output Driver L Stage 1", 4, NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver R Stage 2", + SND_SOC_DAPM_PGA_S("Output Driver R Stage 2", 5, NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver L Stage 2", + SND_SOC_DAPM_PGA_S("Output Driver L Stage 2", 5, NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1, + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 6, NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1, + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 6, NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0), - SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACL", 7, + NAU8825_REG_CHARGE_PUMP, 8, 1, nau8825_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_S("Output DACR", 7, + NAU8825_REG_CHARGE_PUMP, 9, 1, nau8825_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* HPOL/R are ungrounded by disabling 16 Ohm pull-downs on playback */ + SND_SOC_DAPM_PGA_S("HPOL Pulldown", 8, + NAU8825_REG_HSD_CTRL, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("HPOR Pulldown", 8, + NAU8825_REG_HSD_CTRL, 1, 1, NULL, 0), + + /* High current HPOL/R boost driver */ + SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, + NAU8825_REG_BOOST, 9, 1, NULL, 0), + + /* Class G operation control*/ + SND_SOC_DAPM_PGA_S("Class G", 10, + NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), @@ -375,24 +425,27 @@ static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { {"DACR Mux", "DACR", "DDACR"}, {"HP amp L", NULL, "DACL Mux"}, {"HP amp R", NULL, "DACR Mux"}, - {"HP amp L", NULL, "HP amp power"}, - {"HP amp R", NULL, "HP amp power"}, - {"ADACL", NULL, "HP amp L"}, - {"ADACR", NULL, "HP amp R"}, - {"ADACL", NULL, "ADACL Clock"}, - {"ADACR", NULL, "ADACR Clock"}, - {"Output Driver L Stage 1", NULL, "ADACL"}, - {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Charge Pump", NULL, "HP amp L"}, + {"Charge Pump", NULL, "HP amp R"}, + {"ADACL", NULL, "Charge Pump"}, + {"ADACR", NULL, "Charge Pump"}, + {"ADACL Clock", NULL, "ADACL"}, + {"ADACR Clock", NULL, "ADACR"}, + {"Output Driver L Stage 1", NULL, "ADACL Clock"}, + {"Output Driver R Stage 1", NULL, "ADACR Clock"}, {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, {"Output DACL", NULL, "Output Driver L Stage 3"}, {"Output DACR", NULL, "Output Driver R Stage 3"}, - {"HPOL", NULL, "Output DACL"}, - {"HPOR", NULL, "Output DACR"}, - {"HPOL", NULL, "Charge Pump"}, - {"HPOR", NULL, "Charge Pump"}, + {"HPOL Pulldown", NULL, "Output DACL"}, + {"HPOR Pulldown", NULL, "Output DACR"}, + {"HP Boost Driver", NULL, "HPOL Pulldown"}, + {"HP Boost Driver", NULL, "HPOR Pulldown"}, + {"Class G", NULL, "HP Boost Driver"}, + {"HPOL", NULL, "Class G"}, + {"HPOR", NULL, "Class G"}, }; static int nau8825_hw_params(struct snd_pcm_substream *substream, @@ -659,11 +712,10 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) break; } - if (type & SND_JACK_HEADPHONE) { - /* Unground HPL/R */ - regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); - } - + /* Leaving HPOL/R grounded after jack insert by default. They will be + * ungrounded as part of the widget power up sequence at the beginning + * of playback to reduce pop. + */ return type; } @@ -768,6 +820,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) { struct regmap *regmap = nau8825->regmap; + /* Latch IIC LSB value */ + regmap_write(regmap, NAU8825_REG_IIC_ADDR_SET, 0x0001); /* Enable Bias/Vmid */ regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); @@ -780,10 +834,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825) nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT); /* Disable Boost Driver, Automatic Short circuit protection enable */ regmap_update_bits(regmap, NAU8825_REG_BOOST, - NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | - NAU8825_SHORT_SHUTDOWN_EN, - NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | - NAU8825_SHORT_SHUTDOWN_EN); + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_DIS | + NAU8825_HP_BOOST_G_DIS | NAU8825_SHORT_SHUTDOWN_EN, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_DIS | + NAU8825_HP_BOOST_G_DIS | NAU8825_SHORT_SHUTDOWN_EN); regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, NAU8825_JKDET_OUTPUT_EN, @@ -822,6 +876,35 @@ static void nau8825_init_regs(struct nau8825 *nau8825) NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128); regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128); + /* Disable DACR/L power */ + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL); + /* Enable TESTDAC. This sets the analog DAC inputs to a '0' input + * signal to avoid any glitches due to power up transients in both + * the analog and digital DAC circuit. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDAC_EN); + /* CICCLP off */ + regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, + NAU8825_DAC_CLIP_OFF, NAU8825_DAC_CLIP_OFF); + + /* Class AB bias current to 2x, DAC Capacitor enable MSB/LSB */ + regmap_update_bits(regmap, NAU8825_REG_ANALOG_CONTROL_2, + NAU8825_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8825_DAC_CAPACITOR_MSB | NAU8825_DAC_CAPACITOR_LSB, + NAU8825_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8825_DAC_CAPACITOR_MSB | NAU8825_DAC_CAPACITOR_LSB); + /* Class G timer 64ms */ + regmap_update_bits(regmap, NAU8825_REG_CLASSG_CTRL, + NAU8825_CLASSG_TIMER_MASK, + 0x20 << NAU8825_CLASSG_TIMER_SFT); + /* DAC clock delay 2ns, VREF */ + regmap_update_bits(regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_CLK_DELAY_MASK | NAU8825_RDAC_VREF_MASK, + (0x2 << NAU8825_RDAC_CLK_DELAY_SFT) | + (0x3 << NAU8825_RDAC_VREF_SFT)); } static const struct regmap_config nau8825_regmap_config = { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..8ceb5f3 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -14,6 +14,7 @@ #define NAU8825_REG_RESET 0x00 #define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_IIC_ADDR_SET 0x02 #define NAU8825_REG_CLK_DIVIDER 0x03 #define NAU8825_REG_FLL1 0x04 #define NAU8825_REG_FLL2 0x05 @@ -129,7 +130,7 @@ /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) -/* 0 - short to GND, 1 - open */ +/* 0 - open, 1 - short to GND */ #define NAU8825_SPKR_DWN1R (1 << 1) #define NAU8825_SPKR_DWN1L (1 << 0) @@ -251,12 +252,18 @@ /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 +/* CLASSG_CTRL (0x50) */ +#define NAU8825_CLASSG_TIMER_SFT 8 +#define NAU8825_CLASSG_TIMER_MASK (0x3f << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_EN (1 << 0) + /* I2C_DEVICE_ID (0x58) */ #define NAU8825_GPIO2JD1 (1 << 7) #define NAU8825_SOFTWARE_ID_MASK 0x3 #define NAU8825_SOFTWARE_ID_NAU8825 0x0 /* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_TESTDAC_EN (0x3 << 8) #define NAU8825_BIAS_VMID (1 << 6) #define NAU8825_BIAS_VMID_SEL_SFT 4 #define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) @@ -274,6 +281,12 @@ #define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) #define NAU8825_POWERUP_ADCL (1 << 6) +/* RDAC (0x73) */ +#define NAU8825_RDAC_CLK_DELAY_SFT 4 +#define NAU8825_RDAC_CLK_DELAY_MASK (0x7 << NAU8825_RDAC_CLK_DELAY_SFT) +#define NAU8825_RDAC_VREF_SFT 2 +#define NAU8825_RDAC_VREF_MASK (0x3 << NAU8825_RDAC_VREF_SFT) + /* MIC_BIAS (0x74) */ #define NAU8825_MICBIAS_JKSLV (1 << 14) #define NAU8825_MICBIAS_JKR2 (1 << 12) @@ -284,6 +297,7 @@ /* BOOST (0x76) */ #define NAU8825_PRECHARGE_DIS (1 << 13) #define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 9) #define NAU8825_HP_BOOST_G_DIS (1 << 8) #define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) -- 2.7.0.rc3.207.g0ac5344

2 1

[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: Support for DRA7xx family
by Peter Ujfalusi 13 Mar '16

13 Mar '16

DRA7xx family is compatible with the OMAP5 HDMI. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-hdmi-audio.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index f83cc2bc0fc4..64425d352962 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -345,6 +345,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) dai_drv = &omap4_hdmi_dai; break; case OMAPDSS_VER_OMAP5: + case OMAPDSS_VER_DRA7xx: dai_drv = &omap5_hdmi_dai; break; default: -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: rsnd: add simplified module explanation
by Kuninori Morimoto 13 Mar '16

13 Mar '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound driver user needs to read its datasheet when create DT. But it is difficult to understand, because it has many modules (SRC/CTU/MIX/DVC/SSIU/SSI/AudioDMAC/AudioDMACperiperi), and many features (Asynchronous/Synchronous mode on SRC, CTU matrix, DVC volume settings feature, Multi-SSI/TDM-SSI, etc). This patch adds simplified explanation to help setting/understanding. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 340 +++++++++++++++++++++ sound/soc/sh/rcar/dvc.c | 23 ++ 2 files changed, 363 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8ee0fa9..c7b29df 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -1,6 +1,337 @@ Renesas R-Car sound +============================================= +* Modules +============================================= + +Renesas R-Car sound is constructed from below modules +(for Gen2 or later) + + SCU : Sampling Rate Converter Unit + - SRC : Sampling Rate Converter + - CMD + - CTU : Channel Transfer Unit + - MIX : Mixer + - DVC : Digital Volume and Mute Function + SSIU : Serial Sound Interface Unit + SSI : Serial Sound Interface + +See detail of each module's channels, connection, limitation on datasheet + +============================================= +* Multi channel +============================================= + +Multi channel is supported by Multi-SSI, or TDM-SSI. + + Multi-SSI : 6ch case, you can use stereo x 3 SSI + TDM-SSI : 6ch case, you can use TDM + +============================================= +* Enable/Disable each modules +============================================= + +See datasheet to check SRC/CTU/MIX/DVC connect-limitation. +DT controls enabling/disabling module. +${LINUX}/arch/arm/boot/dts/r8a7790-lager.dts can be good example. +This is example of + +Playback: [MEM] -> [SRC2] -> [DVC0] -> [SSIU0/SSI0] -> [codec] +Capture: [MEM] <- [DVC1] <- [SRC3] <- [SSIU1/SSI1] <- [codec] + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&ssi0 &src2 &dvc0>; + capture = <&ssi1 &src3 &dvc1>; + }; + }; + }; + +You can use below. +${LINUX}/arch/arm/boot/dts/r8a7790.dts can be good example. + + &src0 &ctu00 &mix0 &dvc0 &ssi0 + &src1 &ctu01 &mix1 &dvc1 &ssi1 + &src2 &ctu02 &ssi2 + &src3 &ctu03 &ssi3 + &src4 &ssi4 + &src5 &ctu10 &ssi5 + &src6 &ctu11 &ssi6 + &src7 &ctu12 &ssi7 + &src8 &ctu13 &ssi8 + &src9 &ssi9 + +============================================= +* SRC (Sampling Rate Converter) +============================================= + + [xx]Hz [yy]Hz + ------> [SRC] ------> + +SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes + + Asynchronous mode: input data / output data are based on different clocks. + you can use this mode on Playback / Capture + Synchronous mode: input data / output data are based on same clocks. + This mode will be used if system doesn't have its input clock, + for example digital TV case. + you can use this mode on Playback + +------------------ +** Asynchronous mode +------------------ + +You need to use "renesas,rsrc-card" sound card for it. +example) + + sound { + compatible = "renesas,rsrc-card"; + ... + /* + * SRC Asynchronous mode setting + * Playback: + * All input data will be converted to 48kHz + * Capture: + * Inputed 48kHz data will be converted to + * system specified Hz + */ + convert-rate = <48000>; + ... + cpu { + sound-dai = <&rcar_sound>; + }; + codec { + ... + }; + }; + +------------------ +** Synchronous mode +------------------ + + > amixer set "SRC Out Rate" on + > aplay xxxx.wav + > amixer set "SRC Out Rate" 48000 + > amixer set "SRC Out Rate" 44100 + +============================================= +* CTU (Channel Transfer Unit) +============================================= + + [xx]ch [yy]ch + ------> [CTU] --------> + +CTU can convert [xx]ch to [yy]ch, or exchange outputed channel. +CTU conversion needs matrix settings. +For more detail information, see below + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - SCU Operation + - CMD Block + - Functional Blocks in CMD + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - Register Description + - CTUn Scale Value exx Register (CTUn_SVxxR) + + ${LINUX}/sound/soc/sh/rcar/ctu.c + - comment of header + +You need to use "renesas,rsrc-card" sound card for it. +example) + + sound { + compatible = "renesas,rsrc-card"; + ... + /* + * CTU setting + * All input data will be converted to 2ch + * as output data + */ + convert-channels = <2>; + ... + cpu { + sound-dai = <&rcar_sound>; + }; + codec { + ... + }; + }; + +Ex) Exchange output channel + Input -> Output + 1ch -> 0ch + 0ch -> 1ch + + example of using matrix + output 0ch = (input 0ch x 0) + (input 1ch x 1) + output 1ch = (input 0ch x 1) + (input 1ch x 0) + + amixer set "CTU Reset" on + amixer set "CTU Pass" 9,10 + amixer set "CTU SV0" 0,4194304 + amixer set "CTU SV1" 4194304,0 + + example of changing connection + amixer set "CTU Reset" on + amixer set "CTU Pass" 2,1 + +============================================= +* MIX (Mixer) +============================================= + +MIX merges 2 sounds path. You can see 2 sound interface on system, +and these sounds will be merged by MIX. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 yyyy.wav + +You need to use "renesas,rsrc-card" sound card for it. +Ex) + [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] + | + [MEM] -> [SRC2] -> [CTU03] -+ + + sound { + compatible = "renesas,rsrc-card"; + ... + cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + codec { + ... + }; + }; + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; + }; + dai1 { + playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; + }; + }; + }; + +============================================= +* DVC (Digital Volume and Mute Function) +============================================= + +DVC controls Playback/Capture volume. + +Playback Volume + amixer set "DVC Out" 100% + +Capture Volume + amixer set "DVC In" 100% + +Playback Mute + amixer set "DVC Out Mute" on + +Capture Mute + amixer set "DVC In Mute" on + +Volume Ramp + amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + amixer set "DVC Out Ramp" on + aplay xxx.wav & + amixer set "DVC Out" 80% // Volume Down + amixer set "DVC Out" 100% // Volume Up + +============================================= +* SSIU (Serial Sound Interface Unit) +============================================= + +There is no DT settings for SSIU, because SSIU will be automatically +selected via SSI. +SSIU can avoid some under/over run error, because it has some buffer. +But you can't use it if SSI was PIO mode. +In DMA mode, you can select not to use SSIU by using "no-busif" on DT. + + &ssi0 { + no-busif; + }; + +============================================= +* SSI (Serial Sound Interface) +============================================= + +** PIO mode + +You can use PIO mode which is for connection check by using. +Note: The system will drop non-SSI modules in PIO mode +even though if DT is selecting other modules. + + &ssi0 { + pio-transfer + }; + +** DMA mode without SSIU + +You can use DMA without SSIU. +Note: under/over run, or noise are likely to occur + + &ssi0 { + no-busif; + }; + +** PIN sharing + +Each SSI can share WS pin. It is based on platform. +This is example if SSI1 want to share WS pin with SSI0 + + &ssi1 { + shared-pin; + }; + +** Multi-SSI + +You can use Multi-SSI. +This is example of SSI0/SSI1/SSI2 (= for 6ch) + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; + }; + }; + }; + +** TDM-SSI + +You can use TDM with SSI. +This is example of TDM 6ch. +Driver can automatically switches TDM <-> stereo mode in this case. + + rsnd_tdm: sound { + compatible = "simple-audio-card"; + ... + simple-audio-card,cpu { + /* system can use TDM 6ch */ + dai-tdm-slot-num = <6>; + sound-dai = <&rcar_sound>; + }; + simple-audio-card,codec { + ... + }; + }; + + +============================================= Required properties: +============================================= + - compatible : "renesas,rcar_sound-<soctype>", fallbacks "renesas,rcar_sound-gen1" if generation1, and "renesas,rcar_sound-gen2" if generation2 @@ -64,7 +395,10 @@ DAI subnode properties: - playback : list of playback modules - capture : list of capture modules + +============================================= Example: +============================================= rcar_sound: sound@ec500000 { #sound-dai-cells = <1>; @@ -250,7 +584,9 @@ rcar_sound: sound@ec500000 { }; }; +============================================= Example: simple sound card +============================================= rsnd_ak4643: sound { compatible = "simple-audio-card"; @@ -290,7 +626,9 @@ Example: simple sound card shared-pin; }; +============================================= Example: simple sound card for TDM +============================================= rsnd_tdm: sound { compatible = "simple-audio-card"; @@ -309,7 +647,9 @@ Example: simple sound card for TDM }; }; +============================================= Example: simple sound card for Multi channel +============================================= &rcar_sound { pinctrl-0 = <&sound_pins &sound_clk_pins>; diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 93b11e1..02d971f 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -8,6 +8,29 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ + +/* + * Playback Volume + * amixer set "DVC Out" 100% + * + * Capture Volume + * amixer set "DVC In" 100% + * + * Playback Mute + * amixer set "DVC Out Mute" on + * + * Capture Mute + * amixer set "DVC In Mute" on + * + * Volume Ramp + * amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + * amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + * amixer set "DVC Out Ramp" on + * aplay xxx.wav & + * amixer set "DVC Out" 80% // Volume Down + * amixer set "DVC Out" 100% // Volume Up + */ + #include "rsnd.h" #define RSND_DVC_NAME_SIZE 16 -- 1.9.1

2 1

[alsa-devel] [PATCH v4 0/6] add IEC958 channel status control helpers
by Arnaud Pouliquen 12 Mar '16

12 Mar '16

V4 - Link pcm control, defined in dai, to PCM device: Discard solution based on "pcm_new" ops for DAI initialisations replace it by an helper function that registers pcm control and links it to pcm device. - Define helpers function to handle 'IEC958 Playback Default' control . add snd_pcm_iec958_ctl_new helper for compatibility with snd_soc_add_dai_pcm_controls implementation . add substream parameter for snd_pcm_create_iec958_ctl function . fixes based on Takashi remarks. V3: - Define helpers function to handle 'IEC958 Playback Default' control - add "pcm_new" ops for DAI initialisations that need pcm runtime context. This patchset is needed to be able to associate control to the PCM device. The iec control is linked to the DAI but also to the PCM device. Furthermore, some platforms like sti support both HDMI and SPDIF outputs in parallel. Logical way to differentiate them is to link them to pcm device index. This patchset is extracted from following RFC http://permalink.gmane.org/gmane.linux.alsa.devel/149876 (sti: add audio interface to the hdmi driver) Notice that "ASoC: hdmi-codec: add IEC control" patch depends on integration of the hdmi-codec driver Patches update: - ALSA: pcm: add IEC958 channel status control helper - make mutex usage mandatory. - Set control index to pcm device id ( needed to be supported by iecset) - ASoC: core: add code to complete dai init after pcm creation - fix dai param of pcm_new ops for cpu dai. - ASoC: sti: use iec channel status control helper new patch: implementation of the helper for sti cpu-dai V2: - patch: ALSA: pcm: add IEC958 channel status control helper - Return 1 instead of 0 in snd_pcm_iec958_put - Add .access field in control structure - I have kept condition on mutex for flexibility (but could be cleaned to force user to use a mutex) V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Arnaud Pouliquen (5): ASoC: core: add snd_soc_add_dai_pcm_controls helper ASoC: sti: use snd_soc_add_dai_pcm_controls helper ALSA: pcm: add IEC958 channel status control helper ASoC: core: allow private data for snd_soc_add_dai_pcm_controls ASoC: sti: use iec channel status control helper include/sound/hdmi-codec.h | 1 + include/sound/pcm_iec958.h | 21 +++++ include/sound/soc-dai.h | 1 + include/sound/soc.h | 3 + sound/core/pcm_iec958.c | 125 ++++++++++++++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 54 ++++++++++++++++++++++++++++++++----------- sound/soc/soc-core.c | 168 +++++++++++++++++++++++++++++++++++----- sound/soc/sti/Kconfig | 1 + sound/soc/sti/sti_uniperif.c | 57 +++++++------- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 80 +++++-------------- 11 files changed, 386 insertions(+), 126 deletions(-) -- 1.9.1

4 11

[alsa-devel] [PATCH 1/1] aplay/arecord: handle parsing errors of parameter values
by nebelbank＠posteo.de 11 Mar '16

11 Mar '16

From: erwin <nebelbank(a)posteo.de> Signed-off-by: erwin <nebelbank(a)posteo.de> --- It took me a while to find out why the signal from "-d hw:1,0" sounded so unexpected: was from another pair of microphones... diff --git a/aplay/aplay.c b/aplay/aplay.c index 7eacee3..7acaa83 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -427,6 +427,22 @@ enum { OPT_FATAL_ERRORS, }; +static long parse_long(const char *str, int *err) +{ + long val; + char *endptr; + + errno = 0; + val = strtol(str, &endptr, 0); + + if (errno != 0 || *endptr != '\0') + *err = -1; + else + *err = 0; + + return val; +} + int main(int argc, char *argv[]) { int option_index; @@ -558,7 +574,11 @@ int main(int argc, char *argv[]) } break; case 'c': - rhwparams.channels = strtol(optarg, NULL, 0); + rhwparams.channels = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid channels argument '%s'"), optarg); + return 1; + } if (rhwparams.channels < 1 || rhwparams.channels > 256) { error(_("value %i for channels is invalid"), rhwparams.channels); return 1; @@ -585,7 +605,11 @@ int main(int argc, char *argv[]) } break; case 'r': - tmp = strtol(optarg, NULL, 0); + tmp = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid rate argument '%s'"), optarg); + return 1; + } if (tmp < 300) tmp *= 1000; rhwparams.rate = tmp; @@ -595,32 +619,64 @@ int main(int argc, char *argv[]) } break; case 'd': - timelimit = strtol(optarg, NULL, 0); + timelimit = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid duration argument '%s'"), optarg); + return 1; + } break; case 'N': nonblock = 1; open_mode |= SND_PCM_NONBLOCK; break; case 'F': - period_time = strtol(optarg, NULL, 0); + period_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid period time argument '%s'"), optarg); + return 1; + } break; case 'B': - buffer_time = strtol(optarg, NULL, 0); + buffer_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid buffer time argument '%s'"), optarg); + return 1; + } break; case OPT_PERIOD_SIZE: - period_frames = strtol(optarg, NULL, 0); + period_frames = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid period size argument '%s'"), optarg); + return 1; + } break; case OPT_BUFFER_SIZE: - buffer_frames = strtol(optarg, NULL, 0); + buffer_frames = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid buffer size argument '%s'"), optarg); + return 1; + } break; case 'A': - avail_min = strtol(optarg, NULL, 0); + avail_min = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid min available space argument '%s'"), optarg); + return 1; + } break; case 'R': - start_delay = strtol(optarg, NULL, 0); + start_delay = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid start delay argument '%s'"), optarg); + return 1; + } break; case 'T': - stop_delay = strtol(optarg, NULL, 0); + stop_delay = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid stop delay argument '%s'"), optarg); + return 1; + } break; case 'v': verbose++; @@ -671,7 +727,11 @@ int main(int argc, char *argv[]) test_position = 1; break; case OPT_TEST_COEF: - test_coef = strtol(optarg, NULL, 0); + test_coef = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid test coef argument '%s'"), optarg); + return 1; + } if (test_coef < 1) test_coef = 1; break; @@ -679,7 +739,11 @@ int main(int argc, char *argv[]) test_nowait = 1; break; case OPT_MAX_FILE_TIME: - max_file_time = strtol(optarg, NULL, 0); + max_file_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid max file time argument '%s'"), optarg); + return 1; + } break; case OPT_PROCESS_ID_FILE: pidfile_name = optarg; -- 2.5.0

2 3

[alsa-devel] [PATCH] ASoC: wm_adsp: Use correct local length in error message
by Charles Keepax 11 Mar '16

11 Mar '16

Fixes: 44029e9e1290 ("ASoC: wm_adsp: wm_coeff_{read|write}_control should use passed length") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3ee1c72..3c15481 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -807,7 +807,7 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, ret = regmap_raw_read(dsp->regmap, reg, scratch, len); if (ret) { adsp_err(dsp, "Failed to read %zu bytes from %x: %d\n", - ctl->len, reg, ret); + len, reg, ret); kfree(scratch); return ret; } -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - fix the mic mute button and led problem for a Lenovo AIO
by Hui Wang 11 Mar '16

11 Mar '16

This Lenovo ThinkCentre AIO also uses Line2 as mic mute button and uses GPIO2 to control the mic mute led, so applying this quirk can make both the button and led work. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1555912 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 93d2156..4f5ca0b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5556,6 +5556,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), + SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: dice: use structure to represent register parameters instead of array with basic type element
by Takashi Sakamoto 11 Mar '16

11 Mar '16

In dice interface, two blocks of register are accessible via IEEE 1394 asynchronous transaction to represent the number of supported isochronous streams and the number of quadlets for stream information. Current ALSA dice driver uses array with 'unsigned int' element for temporary cache of these information. But using structure is preferable for begin easily comprehensible. This commit applies a local structure for this aim. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 110 +++++++++++++++++++------------------- 1 file changed, 54 insertions(+), 56 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 2077f18..845d5e5 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -12,6 +12,11 @@ #define CALLBACK_TIMEOUT 200 #define NOTIFICATION_TIMEOUT_MS (2 * MSEC_PER_SEC) +struct reg_params { + unsigned int count; + unsigned int size; +}; + const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { /* mode 0 */ [0] = 32000, @@ -65,7 +70,9 @@ static int ensure_phase_lock(struct snd_dice *dice) return 0; } -static int get_register_params(struct snd_dice *dice, unsigned int params[4]) +static int get_register_params(struct snd_dice *dice, + struct reg_params *tx_params, + struct reg_params *rx_params) { __be32 reg[2]; int err; @@ -73,14 +80,16 @@ static int get_register_params(struct snd_dice *dice, unsigned int params[4]) err = snd_dice_transaction_read_tx(dice, TX_NUMBER, reg, sizeof(reg)); if (err < 0) return err; - params[0] = min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); - params[1] = be32_to_cpu(reg[1]) * 4; + tx_params->count = + min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); + tx_params->size = be32_to_cpu(reg[1]) * 4; err = snd_dice_transaction_read_rx(dice, RX_NUMBER, reg, sizeof(reg)); if (err < 0) return err; - params[2] = min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); - params[3] = be32_to_cpu(reg[1]) * 4; + rx_params->count = + min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); + rx_params->size = be32_to_cpu(reg[1]) * 4; return 0; } @@ -105,21 +114,21 @@ static void release_resources(struct snd_dice *dice) } static void stop_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, - unsigned int count, unsigned int size) + struct reg_params *params) { __be32 reg; unsigned int i; - for (i = 0; i < count; i++) { + for (i = 0; i < params->count; i++) { reg = cpu_to_be32((u32)-1); if (dir == AMDTP_IN_STREAM) { snd_dice_transaction_write_tx(dice, - size * i + TX_ISOCHRONOUS, - &reg, sizeof(reg)); + params->size * i + TX_ISOCHRONOUS, + &reg, sizeof(reg)); } else { snd_dice_transaction_write_rx(dice, - size * i + RX_ISOCHRONOUS, - &reg, sizeof(reg)); + params->size * i + RX_ISOCHRONOUS, + &reg, sizeof(reg)); } } } @@ -180,8 +189,7 @@ static int keep_resources(struct snd_dice *dice, } static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, - unsigned int rate, unsigned int count, - unsigned int size) + unsigned int rate, struct reg_params *params) { __be32 reg[2]; unsigned int i, pcm_chs, midi_ports; @@ -197,15 +205,15 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, resources = dice->rx_resources; } - for (i = 0; i < count; i++) { + for (i = 0; i < params->count; i++) { if (dir == AMDTP_IN_STREAM) { err = snd_dice_transaction_read_tx(dice, - size * i + TX_NUMBER_AUDIO, - reg, sizeof(reg)); + params->size * i + TX_NUMBER_AUDIO, + reg, sizeof(reg)); } else { err = snd_dice_transaction_read_rx(dice, - size * i + RX_NUMBER_AUDIO, - reg, sizeof(reg)); + params->size * i + RX_NUMBER_AUDIO, + reg, sizeof(reg)); } if (err < 0) return err; @@ -219,12 +227,12 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, reg[0] = cpu_to_be32(resources[i].channel); if (dir == AMDTP_IN_STREAM) { err = snd_dice_transaction_write_tx(dice, - size * i + TX_ISOCHRONOUS, - reg, sizeof(reg[0])); + params->size * i + TX_ISOCHRONOUS, + reg, sizeof(reg[0])); } else { err = snd_dice_transaction_write_rx(dice, - size * i + RX_ISOCHRONOUS, - reg, sizeof(reg[0])); + params->size * i + RX_ISOCHRONOUS, + reg, sizeof(reg[0])); } if (err < 0) return err; @@ -247,14 +255,14 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) { unsigned int curr_rate; unsigned int i; - unsigned int reg_params[4]; + struct reg_params tx_params, rx_params; bool need_to_start; int err; if (dice->substreams_counter == 0) return -EIO; - err = get_register_params(dice, reg_params); + err = get_register_params(dice, &tx_params, &rx_params); if (err < 0) return err; @@ -271,12 +279,12 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) /* Judge to need to restart streams. */ for (i = 0; i < MAX_STREAMS; i++) { - if (i < reg_params[0]) { + if (i < tx_params.count) { if (amdtp_streaming_error(&dice->tx_stream[i]) || !amdtp_stream_running(&dice->tx_stream[i])) break; } - if (i < reg_params[2]) { + if (i < rx_params.count) { if (amdtp_streaming_error(&dice->rx_stream[i]) || !amdtp_stream_running(&dice->rx_stream[i])) break; @@ -287,10 +295,8 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) if (need_to_start) { /* Stop transmission. */ snd_dice_transaction_clear_enable(dice); - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], - reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], - reg_params[3]); + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); release_resources(dice); err = ensure_phase_lock(dice); @@ -301,12 +307,10 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) } /* Start both streams. */ - err = start_streams(dice, AMDTP_IN_STREAM, rate, reg_params[0], - reg_params[1]); + err = start_streams(dice, AMDTP_IN_STREAM, rate, &tx_params); if (err < 0) goto error; - err = start_streams(dice, AMDTP_OUT_STREAM, rate, reg_params[2], - reg_params[3]); + err = start_streams(dice, AMDTP_OUT_STREAM, rate, &rx_params); if (err < 0) goto error; @@ -318,10 +322,10 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) } for (i = 0; i < MAX_STREAMS; i++) { - if ((i < reg_params[0] && + if ((i < tx_params.count && !amdtp_stream_wait_callback(&dice->tx_stream[i], CALLBACK_TIMEOUT)) || - (i < reg_params[2] && + (i < rx_params.count && !amdtp_stream_wait_callback(&dice->rx_stream[i], CALLBACK_TIMEOUT))) { err = -ETIMEDOUT; @@ -333,8 +337,8 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) return err; error: snd_dice_transaction_clear_enable(dice); - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], reg_params[3]); + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); release_resources(dice); return err; } @@ -346,18 +350,16 @@ error: */ void snd_dice_stream_stop_duplex(struct snd_dice *dice) { - unsigned int reg_params[4]; + struct reg_params tx_params, rx_params; if (dice->substreams_counter > 0) return; snd_dice_transaction_clear_enable(dice); - if (get_register_params(dice, reg_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], - reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], - reg_params[3]); + if (get_register_params(dice, &tx_params, &rx_params) == 0) { + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); } release_resources(dice); @@ -444,15 +446,13 @@ end: void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { - unsigned int reg_params[4]; + struct reg_params tx_params, rx_params; snd_dice_transaction_clear_enable(dice); - if (get_register_params(dice, reg_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], - reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], - reg_params[3]); + if (get_register_params(dice, &tx_params, &rx_params) == 0) { + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); } release_resources(dice); @@ -462,7 +462,7 @@ void snd_dice_stream_destroy_duplex(struct snd_dice *dice) void snd_dice_stream_update_duplex(struct snd_dice *dice) { - unsigned int reg_params[4]; + struct reg_params tx_params, rx_params; /* * On a bus reset, the DICE firmware disables streaming and then goes @@ -474,11 +474,9 @@ void snd_dice_stream_update_duplex(struct snd_dice *dice) */ dice->global_enabled = false; - if (get_register_params(dice, reg_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], - reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], - reg_params[3]); + if (get_register_params(dice, &tx_params, &rx_params) == 0) { + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); } } -- 2.7.0

2 2

Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Mark Brown 10 Mar '16

10 Mar '16

On Tue, Feb 16, 2016 at 09:47:47PM +0530, Jeeja KP wrote: > We have no mechanism to configure hw_params for DAIs which are hostless. > Like a loop or DAIs connected to internal sinks/sources. Please propose patches, don't send enormous design documents. They're big, take a long time to read and don't really get us anywhere unless they're extremely clear. > Approach: > 1. Allow params to be defined for other BE dai link also. this > currently applicable only for codec_codec link. Like I keep saying if you're thinking about this in terms of DPCM you're solving the wrong problem - we already support supplying parameters for device<->device links, we already have systems that need to configure things for device<->device links so we can already see that if we're doing something that only works for things that are on SoC and can use DPCM then we're not solving the problem. DPCM works pretty well for small, simple DSPs that are mostly in line in the middle of a standard DAI but it does have limitations which you're now running into. Your systems don't fit well with what DPCM supports naturally. > struct snd_soc_dai_link { > const struct snd_soc_pcm_stream *params; > unsigned int num_params; > > + /* flag to create codec codec based on the flag*/ > + unsigned int codec_link:1; > + This is clearly confused, if we can't figure out what the two devices we are connecting are based on just looking at the devices then that's really not a good sign that our interfaces are sensible and easy to work with.

2 1

[alsa-devel] [PATCH] ALSA: pcm: Avoid "BUG:" string for warnings again
by Takashi Iwai 10 Mar '16

10 Mar '16

The commit [d507941beb1e: ALSA: pcm: Correct PCM BUG error message] made the warning prefix back to "BUG:" due to its previous wrong prefix. But a kernel message containing "BUG:" seems taken as an Oops message wrongly by some brain-dead daemons, and it annoys users in the end. Instead of teaching daemons, change the string again to a more reasonable one. Fixes: 507941beb1e ('ALSA: pcm: Correct PCM BUG error message') Cc: <stable(a)vger.kernel.org> # v3.19+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6b5a811e01a5..3a9b66c6e09c 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -322,7 +322,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, char name[16]; snd_pcm_debug_name(substream, name, sizeof(name)); pcm_err(substream->pcm, - "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", + "invalid position: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, runtime->period_size); } -- 2.7.2

1 0

[alsa-devel] [PATCH - pcm 1/1] pcm: fix 'unable to create IPC shm instance' caused by fork from a thread
by IceBsi 10 Mar '16

10 Mar '16

From: Qing Cai <bsiice(a)msn.com> As stated in manpage SHMCTL(2), shm_nattch is "No. of current attaches" (i.e., number of processes attached to the shared memeory). If an application uses alsa-lib and invokes fork() from a thread of the application, there may be the following execution sequence: 1. execute the following statement: pcm_direct.c:110: dmix->shmptr = shmat(dmix->shmid, 0, 0) (shm_nattch becomes 1) 2. invoke fork() in some thread. (shm_nattch becomes 2) 3. execute the following statement: pcm_direct.c:122: if (buf.shm_nattch == 1) 4. execute the following statement: pcm_direct.c:131: if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) (As stated in manpage SHMGET(2), "When a new shared memory segment is created, its contents are initialized to zero values", so dmix->shmptr->magic is 0) 5. execute the following statements: pcm_direct.c:132: snd_pcm_direct_shm_discard(dmix) pcm_direct.c:133: return -EINVAL The above execution sequence will cause the following error: unable to create IPC shm instance This error causes multimedia application has no sound. This error rarely occurs, probability is about 1%. Signed-off-by: Qing Cai <bsiice(a)msn.com> Signed-off-by: Qing Cai <caiqing(a)neusoft.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index fd3877c..14de734 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -91,13 +91,19 @@ int snd_pcm_direct_semaphore_create_or_connect(snd_pcm_direct_t *dmix) int snd_pcm_direct_shm_create_or_connect(snd_pcm_direct_t *dmix) { struct shmid_ds buf; - int tmpid, err; + int tmpid, err, first_instance = 0; retryget: dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), - IPC_CREAT | dmix->ipc_perm); + dmix->ipc_perm); + if (dmix->shmid < 0) { + if (errno == ENOENT) + if ((dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), + IPC_CREAT | IPC_EXCL | dmix->ipc_perm)) != -1) + first_instance = 1; + } err = -errno; - if (dmix->shmid < 0){ + if (dmix->shmid < 0) { if (errno == EINVAL) if ((tmpid = shmget(dmix->ipc_key, 0, dmix->ipc_perm)) != -1) if (!shmctl(tmpid, IPC_STAT, &buf)) @@ -119,7 +125,7 @@ retryget: snd_pcm_direct_shm_discard(dmix); return err; } - if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ + if (first_instance) { /* we're the first user, clear the segment */ memset(dmix->shmptr, 0, sizeof(snd_pcm_direct_share_t)); if (dmix->ipc_gid >= 0) { buf.shm_perm.gid = dmix->ipc_gid; -- 2.7.2

3 3

[alsa-devel] [PATCH] ALSA: hda - Add a sanity check of pin / port mapping on i915 HDMI/DP
by Takashi Iwai 10 Mar '16

10 Mar '16

There is an implicit rule to map between pin NID and port number on Intel HDMI/DP codec: the mapping is fixed only for NID 0x05, 0x06 and 0x07. For avoiding the possible memory corruption, add a sanity check for the NID value and splat WARN_ON() for invalid accesses. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index f6854dbd7d8d..fb96aead8257 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -126,6 +126,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); */ static int pin2port(hda_nid_t pin_nid) { + if (WARN_ON(pin_nid < 5 || pin_nid > 7)) + return -1; return pin_nid - 4; } @@ -144,10 +146,14 @@ static int pin2port(hda_nid_t pin_nid) int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) { struct i915_audio_component *acomp = bus->audio_component; + int port; if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) return -ENODEV; - return acomp->ops->sync_audio_rate(acomp->dev, pin2port(nid), rate); + port = pin2port(nid); + if (port < 0) + return -EINVAL; + return acomp->ops->sync_audio_rate(acomp->dev, port, rate); } EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); @@ -175,11 +181,15 @@ int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes) { struct i915_audio_component *acomp = bus->audio_component; + int port; if (!acomp || !acomp->ops || !acomp->ops->get_eld) return -ENODEV; - return acomp->ops->get_eld(acomp->dev, pin2port(nid), audio_enabled, + port = pin2port(nid); + if (port < 0) + return -EINVAL; + return acomp->ops->get_eld(acomp->dev, port, audio_enabled, buffer, max_bytes); } EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); -- 2.7.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Don't handle ELD notify from invalid port
by Takashi Iwai 10 Mar '16

10 Mar '16

The current Intel HDMI codec driver supports only three fixed ports from port B to port D. However, i915 driver may assign a DP on other ports, e.g. port A, when no eDP is used. This incompatibility is caught later at pin_nid_to_pin_index() and results in a warning message like "HDMI: pin nid 4 not registered" at each time. This patch filters out such invalid events beforehand, so that the kernel won't be too grumbling. Reported-by: Stefan Assmann <sassmann(a)kpanic.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bcbc4ee10130..b90e4f3967de 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2432,6 +2432,10 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) struct hda_codec *codec = audio_ptr; int pin_nid = port + 0x04; + /* we assume only from port-B to port-D */ + if (port < 1 || port > 3) + return; + /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ -- 2.7.2

1 0

Re: [alsa-devel] [PATCH v6 5/6] drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding
by Jyri Sarha 10 Mar '16

10 Mar '16

On 03/09/16 15:52, kbuild test robot wrote: > Hi Jyri, > > [auto build test WARNING on drm/drm-next] > [also build test WARNING on v4.5-rc7 next-20160309] > [if your patch is applied to the wrong git tree, please drop us a note to help improving the system] > > url: https://github.com/0day-ci/linux/commits/Jyri-Sarha/Implement-generic-ASoC-… > base: git://people.freedesktop.org/~airlied/linux.git drm-next > config: x86_64-rhel (attached as .config) > reproduce: > # save the attached .config to linux build tree > make ARCH=x86_64 > > All warnings (new ones prefixed by >>): > > drivers/gpu/drm/i2c/tda998x_drv.c: In function 'tda998x_get_audio_ports': >>> drivers/gpu/drm/i2c/tda998x_drv.c:1345:7: warning: 'size' is used uninitialized in this function [-Wuninitialized] > size /= sizeof(u32); > ^ > > vim +/size +1345 drivers/gpu/drm/i2c/tda998x_drv.c > > 1329 if (IS_ERR(priv->audio_pdev)) > 1330 return PTR_ERR(priv->audio_pdev); > 1331 > 1332 return 0; > 1333 } > 1334 > 1335 /* I2C driver functions */ > 1336 > 1337 static int tda998x_get_audio_ports(struct tda998x_priv *priv, > 1338 struct device_node *np) > 1339 { > 1340 const u32 *port_data; > 1341 u32 size; > 1342 int i; > 1343 > 1344 port_data = of_get_property(np, "audio-ports", &size); >> 1345 size /= sizeof(u32); Oops, that is a real bug. I should test if port_data is not null before assuming that the size has been set (/me thinks that he had that line there at some point but then it vanished). Actually initializing size to 0 would be enough, but I think I'll do the both just to be more explicit. And I'll start using PTR_ERR_OR_ZERO() while I am at it. > 1346 > 1347 if (size > 2 * ARRAY_SIZE(priv->audio_port) || size % 2 != 0) { > 1348 dev_err(&priv->hdmi->dev, > 1349 "Bad number of elements in audio-ports dt-property\n"); > 1350 return -EINVAL; > 1351 } > 1352 > 1353 size /= 2; > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation >

1 0

[alsa-devel] [PATCH] ALSA: hda - Apply reboot D3 fix for CX20724 codec, too
by Takashi Iwai 10 Mar '16

10 Mar '16

Just like CX20722, CX7024 codec also requires the power down at reboot in order to reduce the noise at reboot/shutdown. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=113511 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6122b8ca872f..56fefbd85782 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -204,8 +204,13 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - if (codec->core.vendor_id != 0x14f150f2) + switch (codec->core.vendor_id) { + case 0x14f150f2: /* CX20722 */ + case 0x14f150f4: /* CX20724 */ + break; + default: return; + } /* Turn the CX20722 codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ -- 2.7.2

1 0

[alsa-devel] [PATCH] tinycompress: INSTALL: Update the instructions
by Vinod Koul 10 Mar '16

10 Mar '16

Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- INSTALL | 33 +++++++++++++++++++++++++++++++-- 1 file changed, 31 insertions(+), 2 deletions(-) diff --git a/INSTALL b/INSTALL index 2099840756e6..5ac7670b0a7c 100644 --- a/INSTALL +++ b/INSTALL @@ -1,5 +1,34 @@ -Installation Instructions -************************* + + Tinycompress Installation Instructions + ************************************** + +Install from tarball +==================== + +For installation you can use these commands: + + ./configure + make install + + +Compilation from Git +==================== + +gitcompile script allows to build and then install + + ./gitcompile + make install + + +Compilation for Android +======================= + +Android makefile Android.mk is inculded in this project and should be +packaged into Android build system + + +Configure help +============== Copyright (C) 1994-1996, 1999-2002, 2004-2013 Free Software Foundation, Inc. -- 1.9.1

1 0

[alsa-devel] [PATCH 0/6] Support for CherryTrail devices
by Pierre-Louis Bossart 10 Mar '16

10 Mar '16

Fix dependency on DW_DMAC_CORE Fix BIOS issues with better ACPI filtering for active devices Change codec name based on filtered HID Re-use bytcr-rt5640 driver Tested on Tronsmart Ara x5 (CHT, headset only) and Asus T100TA (BYT) to make sure nothing was broken Thanks to Mika Westerberg for his support on ACPI Pierre-Louis Bossart (6): ASoC: Intel: allow for DW_DMAC_CORE as module ASoC: Intel: common: filter ACPI devices with _STA return value ASoC: Intel: common: add translation from HID to codec-name ASoC: Intel: bytcr_rt5640: change quirk position ASoC: Intel: bytcr_rt5640: use HID translation util ASoC: Intel: Atom: add support for CHT w/ RT5640 sound/soc/intel/Kconfig | 4 +-- sound/soc/intel/atom/sst/sst_acpi.c | 4 +++ sound/soc/intel/boards/bytcr_rt5640.c | 50 +++++++++++++++++++++---------- sound/soc/intel/common/sst-acpi.h | 3 ++ sound/soc/intel/common/sst-match-acpi.c | 53 +++++++++++++++++++++++++++++++-- 5 files changed, 94 insertions(+), 20 deletions(-) -- 1.9.1

3 18

[alsa-devel] [PATCH] ASoC: nau8825: extend FLL function
by John Hsu 10 Mar '16

10 Mar '16

Extend FLL clock source from MCLK/BCLK/FS. Modify FLL calculation and parameter setting. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 128 ++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 21 ++++++-- 2 files changed, 108 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c1b87c5..504c969 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -31,7 +31,7 @@ #include "nau8825.h" #define NAU_FREF_MAX 13500000 -#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 struct nau8825_fll { @@ -874,8 +874,8 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, struct nau8825_fll *fll_param) { - u64 fvco; - unsigned int fref, i; + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. @@ -900,18 +900,23 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fll_param->ratio = fll_ratio[i].val; /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. - * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be * guaranteed across the full range of operation. * FDCO = freq_out * 2 * mclk_src_scaling */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { fvco = 256 * fs * 2 * mclk_src_scaling[i].param; - if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) - break; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } } - if (i == ARRAY_SIZE(mclk_src_scaling)) + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) return -EINVAL; - fll_param->mclk_src = mclk_src_scaling[i].val; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional * input based on FDCO, FREF and FLL ratio. @@ -926,7 +931,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -939,10 +945,25 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ - regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_FILTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, NAU8825_FLL_FTR_SW_ACCU); + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_SDM_EN, 0); + } } /* freq_out must be 256*Fs in order to achieve the best performance */ @@ -970,6 +991,36 @@ static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, return 0; } +static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) +{ + int ret = 0; + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + return 0; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + return ret; + } + } + + return 0; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -981,29 +1032,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); - - /* We selected MCLK source but the clock itself managed externally */ - if (!nau8825->mclk) - break; - - if (!nau8825->mclk_freq) { - ret = clk_prepare_enable(nau8825->mclk); - if (ret) { - dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); - return ret; - } - } - - if (nau8825->mclk_freq != freq) { - nau8825->mclk_freq = freq; - - freq = clk_round_rate(nau8825->mclk, freq); - ret = clk_set_rate(nau8825->mclk, freq); - if (ret) { - dev_err(nau8825->dev, "Unable to set mclk rate\n"); - return ret; - } - } + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; break; case NAU8825_CLK_INTERNAL: @@ -1011,7 +1042,32 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + case NAU8825_CLK_FLL_MCLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_BLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + break; + case NAU8825_CLK_FLL_FS: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..8237693 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -112,20 +112,28 @@ /* FLL3 (0x06) */ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8825_FLL_CLK_SRC_SFT 10 +#define NAU8825_FLL_CLK_SRC_MASK (0x3 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_MCLK (0 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_BLK (0x2 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8825_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8825_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8825_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8825_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8825_FLL_FTR_SW_FILTER (0x0 << 12) /* FLL6 (0x9) */ -#define NAU8825_DCO_EN_MASK (0x1 << 15) #define NAU8825_DCO_EN (0x1 << 15) -#define NAU8825_DCO_DIS (0x0 << 15) -#define NAU8825_SDM_EN_MASK (0x1 << 14) #define NAU8825_SDM_EN (0x1 << 14) -#define NAU8825_SDM_DIS (0x0 << 14) /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) @@ -306,6 +314,9 @@ enum { NAU8825_CLK_MCLK = 0, NAU8825_CLK_INTERNAL, + NAU8825_CLK_FLL_MCLK, + NAU8825_CLK_FLL_BLK, + NAU8825_CLK_FLL_FS, }; struct nau8825 { -- 2.6.4

2 5

[alsa-devel] [PATCH v6 0/6] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 10 Mar '16

10 Mar '16

There is currently two other patch series[1][2] that depend on the three first (ALSA-) patches of this series. It would be nice to get at least those three applied sooner rather than later, so please tell me if I can do anything to get those in. BTW, I applied the relevant patches from the two series to on top of this one. Obviously the two series conflicted with each other a bit and also the IEC958 helper patch from [2] conflicted with the IEC958 patch from my series (which is weird since Arnaud's series depends on my series...). After resolving the conflicts everything compiled fine, but for some reason neither the 'ELD' or IEC958 Playback Default' appeared on my BBB. I'll try to find time to debug this further at some point. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105509.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105502.html Changes since v5 - Rebased on top of the latest drm-next branch - Allow 32 bit samplewidth in snd_pcm_create_iec958_consumer() and snd_pcm_create_iec958_consumer_hw_params() - Propose new simpler DT binding for tda998x audio - Squash tda998x audio DT binding together with hdmi-codec integration Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 Jyri Sarha (6): ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 + arch/arm/boot/dts/am335x-boneblack.dts | 71 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 271 ++++++++++++-- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 53 ++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 411 +++++++++++++++++++++ 12 files changed, 898 insertions(+), 72 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

5 16

[alsa-devel] [PATCH] ASoC: hdac_hdmi: Fix infoframe programming
by Subhransu S. Prusty 10 Mar '16

10 Mar '16

Audio infoframe used incorrect buffer, so fix it. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7851dd2..4f7bd1f 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -303,7 +303,6 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, if (ret < 0) return ret; - dip = (u8 *)&frame; break; case DRM_ELD_CONN_TYPE_DP: @@ -332,9 +331,9 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, /* Fill infoframe. Index auto-incremented */ hdac_hdmi_set_dip_index(hdac, pin_nid, 0x0, 0x0); if (conn_type == DRM_ELD_CONN_TYPE_HDMI) { - for (i = 0; i < sizeof(frame); i++) + for (i = 0; i < sizeof(buffer); i++) snd_hdac_codec_write(&hdac->hdac, pin_nid, 0, - AC_VERB_SET_HDMI_DIP_DATA, dip[i]); + AC_VERB_SET_HDMI_DIP_DATA, buffer[i]); } else { for (i = 0; i < sizeof(dp_ai); i++) snd_hdac_codec_write(&hdac->hdac, pin_nid, 0, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] ALSA: dice: enable to handle several streams
by Takashi Sakamoto 09 Mar '16

09 Mar '16

Hi, This patchset updates my previous RFT, and goes for merging. [alsa-devel] [RFT][PATCH 0/4] ALSA: dice: enabled to handle several streams http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105387.html For the previous one, Stefan Richter worked as a tester with his devices (Focusrite Saffire Pro 24 and 40), while current patchset includes changes related to userspace interface. Therefore, I don't add Tested-by. In previous one, ALSA dice driver uses subdevice for userspace to access any PCM substreams. On the other hand, the design of PCM subdevice leaves a bit cumbersome for userspace to handle several subdevices simultaneously in one process. These PCM substreams on a batch of AMDTP streams and it's better for ALSA dice driver to enable one process to handle them with simpler way. Based on this concept, this patchset adds PCM character devices, instead of PCM subdevice. Thus, userspace applications can utilize PCM substreams via individual PCM character devices. In configuration space of alsa-lib, via 'hw:0,0' and 'hw:0,1' if available, like: $ arecord -l ... card 1: Pro26000f5a [Pro26-000f5a], device 0: DICE [Pro26-000f5a] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Pro26000f5a [Pro26-000f5a], device 1: DICE [Pro26-000f5a] Subdevices: 1/1 Subdevice #0: subdevice #0 For the other parts, there's no code changes. Therefore, Stefan's result still has meanings. Changes: * Add additional PCM character device instead of PCM subdevice * Correct string entries to force-pcm * Improve comments Takashi Sakamoto (4): ALSA: dice: have two sets of isochronous resources/streams ALSA: dice: handle whole available isochronous streams ALSA: dice: handle several PCM substreams when any isochronous streams are available ALSA: dice: force to add two pcm devices for listed models sound/firewire/dice/dice-midi.c | 8 +- sound/firewire/dice/dice-pcm.c | 147 +++++++++----- sound/firewire/dice/dice-stream.c | 398 ++++++++++++++++++++++++-------------- sound/firewire/dice/dice.c | 41 ++++ sound/firewire/dice/dice.h | 33 +++- 5 files changed, 424 insertions(+), 203 deletions(-) -- 2.7.0

3 8

[alsa-devel] Scheduled alsa-lib release?
by Takashi Iwai 09 Mar '16

09 Mar '16

Hi, in the previous meeting, there have been requests about the regular "predictable" release of alsa-lib. We thought of aligning with kernel release cycle, judging from the nature of alsa-lib. alsa-lib 1.1.0 was released in the beginning of November, which was just after 4.3 kernel release. Now 4.5 kernel release will be likely in the next week or one week after that. Jaroslav, how about releasing alsa-lib 1.1.1 around that? There are a few pending patches, but they should be available soon. thanks, Takashi

4 6

[alsa-devel] alsa-plugins: pcm_a52.c uses recently removed avcodec_{alloc, free}_frame
by Daniel Kirchner 09 Mar '16

09 Mar '16

pcm_a52.c uses avcodec_alloc_frame and avcodec_free_frame that have been replaced by av_frame_alloc and av_frame_free in recent versions of libavcodec. The following patch should fix this while remaining backwards compatible. --- a/a52/pcm_a52.c.orig 2016-03-04 18:21:12.606805770 +0100 +++ b/a52/pcm_a52.c 2016-03-04 18:34:22.653240501 +0100 @@ -62,6 +62,11 @@ #define AV_CODEC_ID_AC3 CODEC_ID_AC3 #endif +#if LIBAVCODEC_VERSION_INT < 0x371c01 +#define av_frame_alloc avcodec_alloc_frame +#define av_frame_free avcodec_free_frame +#endif + struct a52_ctx { snd_pcm_ioplug_t io; snd_pcm_t *slave; @@ -513,7 +518,7 @@ rec->inbuf = NULL; } #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) - avcodec_free_frame(&rec->frame); + av_frame_free(&rec->frame); #else av_freep(&rec->frame); #endif @@ -557,7 +562,7 @@ { struct a52_ctx *rec = io->private_data; #ifdef USE_AVCODEC_FRAME - rec->frame = avcodec_alloc_frame(); + rec->frame = av_frame_alloc(); if (!rec->frame) return -ENOMEM; if (av_samples_alloc(rec->frame->data, rec->frame->linesize,

2 1

[alsa-devel] [PATCH] Show sequencer sound card numer/PID via aconnect
by Martin Koegler 09 Mar '16

09 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This patch adds support for displaying the sound card number/PID to aconnect. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- configure.ac | 9 +++++++++ seq/aconnect/aconnect.c | 17 ++++++++++++++++- 2 files changed, 25 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index bdb133c..9e48190 100644 --- a/configure.ac +++ b/configure.ac @@ -41,6 +41,15 @@ AC_CHECK_HEADERS([alsa/topology.h], [have_topology="yes"], [have_topology="no"], AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) +AC_CHECK_LIB([asound], [snd_seq_client_info_get_card], [HAVE_SEQ_CLIENT_INFO_GET_CARD="yes"]) +if test "$HAVE_SEQ_CLIENT_INFO_GET_CARD" = "yes" ; then + AC_DEFINE([HAVE_SEQ_CLIENT_INFO_GET_CARD], 1, [alsa-lib supports snd_seq_client_info_get_card]) +fi +AC_CHECK_LIB([asound], [snd_seq_client_info_get_pid], [HAVE_SEQ_CLIENT_INFO_GET_PID="yes"]) +if test "$HAVE_SEQ_CLIENT_INFO_GET_PID" = "yes" ; then + AC_DEFINE([HAVE_SEQ_CLIENT_INFO_GET_PID], 1, [alsa-lib supports snd_seq_client_info_get_pid]) +fi + AM_CONDITIONAL(HAVE_PCM, test "$have_pcm" = "yes") AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 8d6cebb..85a7770 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -166,11 +166,26 @@ static void print_port(snd_seq_t *seq, snd_seq_client_info_t *cinfo, snd_seq_port_info_t *pinfo, int count) { if (! count) { - printf(_("client %d: '%s' [type=%s]\n"), + int card = -1, pid = -1; + + printf(_("client %d: '%s' [type=%s"), snd_seq_client_info_get_client(cinfo), snd_seq_client_info_get_name(cinfo), (snd_seq_client_info_get_type(cinfo) == SND_SEQ_USER_CLIENT ? _("user") : _("kernel"))); + +#ifdef HAVE_SEQ_CLIENT_INFO_GET_CARD + card = snd_seq_client_info_get_card(cinfo); +#endif + if (card != -1) + printf(",card=%d", card); + +#ifdef HAVE_SEQ_CLIENT_INFO_GET_PID + pid = snd_seq_client_info_get_pid(cinfo); +#endif + if (pid != -1) + printf(",pid=%d", pid); + printf("]\n"); } printf(" %3d '%-16s'\n", snd_seq_port_info_get_port(pinfo), -- 2.1.4

2 1

[alsa-devel] alsa-plugins: pcm_a52.c uses recently removed avcodec_{alloc, free}_frame
by Daniel Kirchner 09 Mar '16

09 Mar '16

pcm_a52.c uses avcodec_alloc_frame and avcodec_free_frame that have been replaced by av_frame_alloc and av_frame_free in recent versions of libavcodec. The following patch should fix this while remaining backwards compatible. --- a/a52/pcm_a52.c.orig 2016-03-04 18:21:12.606805770 +0100 +++ b/a52/pcm_a52.c 2016-03-04 18:34:22.653240501 +0100 @@ -62,6 +62,11 @@ #define AV_CODEC_ID_AC3 CODEC_ID_AC3 #endif +#if LIBAVCODEC_VERSION_INT < 0x371c01 +#define av_frame_alloc avcodec_alloc_frame +#define av_frame_free avcodec_free_frame +#endif + struct a52_ctx { snd_pcm_ioplug_t io; snd_pcm_t *slave; @@ -513,7 +518,7 @@ rec->inbuf = NULL; } #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) - avcodec_free_frame(&rec->frame); + av_frame_free(&rec->frame); #else av_freep(&rec->frame); #endif @@ -557,7 +562,7 @@ { struct a52_ctx *rec = io->private_data; #ifdef USE_AVCODEC_FRAME - rec->frame = avcodec_alloc_frame(); + rec->frame = av_frame_alloc(); if (!rec->frame) return -ENOMEM; if (av_samples_alloc(rec->frame->data, rec->frame->linesize,

1 0

[alsa-devel] [PATCH] ASoC: rt5640: remove unused variable
by Sudip Mukherjee 09 Mar '16

09 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> We are getting build warning about: sound/soc/codecs/rt5640.c:1892:11: warning: unused variable 'dai_sel' The use of the variable was removed but the variable itself was not removed. Fixes: c467fc0e010b ("ASoC: rt5640: Set PLL src according to source") Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index af9b5f1..e8b5ba0 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1889,7 +1889,7 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); struct rl6231_pll_code pll_code; - int ret, dai_sel; + int ret; if (source == rt5640->pll_src && freq_in == rt5640->pll_in && freq_out == rt5640->pll_out) -- 1.9.1

2 1

[alsa-devel] [PATCH v4] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 08 Mar '16

08 Mar '16

This is an initial imx-wm8958 device-tree-only machine driver. The sound card driver support three dai link, HIFI, VOICE and BT. We can configure the cpu dai from device tree, if present it will create corresponding sound card subdevice. So at least one cpu dai phandle should be set from device tree. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v3->v4: remove the dummy cpu dai, support set cpu dai1/2/3 from device tree. When cpu-dai1/2/3 present it will create croresponding sound card subdevice. At least one cpu dai should be configured. .../devicetree/bindings/sound/imx-audio-wm8958.txt | 56 ++++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 327 +++++++++++++++++++++ 5 files changed, 400 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..c013287 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,56 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + +Optional properties: + + - cpu-dai1 : The phandle of CPU DAI1 controller. At least one + cpu dai is required. + + - cpu-dai2 : The phandle of CPU DAI2 controller. At least one + cpu dai is required. + + - cpu-dai3 : The phandle of CPU DAI3 controller. At least one + cpu dai is required. + + - fsl,cpu-dai1-master : If present, cpu dai1 works as master, and will + provide bit clock and frame clock. Otherwise, + cpu dai1 works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai1 = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..0e8a9b2 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,327 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link dai_link[DAI_LINK_NUM]; + int num_links; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_cpu_dai1_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + if (data->is_cpu_dai1_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (data->is_cpu_dai1_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && data->is_cpu_dai1_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np[DAI_LINK_NUM], *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev[DAI_LINK_NUM]; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "cpu-dai%d", i + 1); + cpu_np[i] = of_parse_phandle(np, tmp, 0); + if (cpu_np[i]) { + cpu_pdev[i] = of_find_device_by_node(cpu_np[i]); + if (!cpu_pdev[i]) { + dev_err(&pdev->dev, + "failed to get cpu dai%d platform device\n", i); + ret = -EINVAL; + goto fail; + } + + data->dai_link[data->num_links] = imx_wm8958_dai_link[i]; + data->dai_link[data->num_links].cpu_dai_name = dev_name(&cpu_pdev[i]->dev); + data->dai_link[data->num_links++].platform_of_node = cpu_np[i]; + } + } + + if (!data->num_links) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + /* Hifi dai link support codec master and slave mode */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,cpu-dai1-master")) { + data->is_cpu_dai1_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else { + data->is_cpu_dai1_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(data->mclk[i])) + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = data->num_links; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np[HIFI_DAI]); + of_node_put(cpu_np[VOICE_DAI]); + of_node_put(cpu_np[BT_DAI]); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

3 3

[alsa-devel] [PATCH RFC v5 0/8] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 08 Mar '16

08 Mar '16

There is very little change to this series since the RFC v4, but because of the recent interest on the hdmi-codec patch I decided to send another version of the series. Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI There was also some comments about the DT ports binding for tda998x by Jean-Francois Moine. However I did not have time to come up with alternative approach. All in all my tda998x patches should be considered more as a proof of concept for the hdmi-codec part, rather than a serious attempt to get those patches in. However, all comments to those patch are more than welcome as they will help me to come up with something that could finally get merged. Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (7): ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 ++++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 300 ++++++++++++--- include/drm/i2c/tda998x.h | 24 +- include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 52 ++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 411 +++++++++++++++++++++ 11 files changed, 964 insertions(+), 79 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

7 27

Re: [alsa-devel] Testing SAI with mainline
by Fabio Estevam 08 Mar '16

08 Mar '16

On Tue, Mar 8, 2016 at 12:03 AM, Zidan Wang <zidan.wang(a)nxp.com> wrote: > IMX_DMATYPE_SAI is 24. Thanks for the clarification. I have submitted the patches adding SAI support for mx6sx. Now I am trying to get SAI to work on mx7d-sdb. Please see the consolidated changes below. After running audio playback the system hangs. Any ideas? Thanks diff --git a/arch/arm/boot/dts/imx7d-sdb.dts b/arch/arm/boot/dts/imx7d-sdb.dts index b267f79..17320a1 100644 --- a/arch/arm/boot/dts/imx7d-sdb.dts +++ b/arch/arm/boot/dts/imx7d-sdb.dts @@ -95,6 +95,30 @@ regulator-max-microvolt = <1800000>; }; }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "mx7d-sdb-sound-card"; + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&dailink_master>; + simple-audio-card,frame-master = <&dailink_master>; + + simple-audio-card,cpu { + sound-dai = <&sai1>; + }; + + dailink_master: simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; + }; + }; +}; + +&clks { + assigned-clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_SRC>, + <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; + assigned-clock-parents = <&clks IMX7D_PLL_AUDIO_POST_DIV>; + assigned-clock-rates = <0>, <12288000>; }; &adc1 { @@ -264,14 +288,23 @@ status = "okay"; codec: wm8960@1a { + #sound-dai-cells = <0>; compatible = "wlf,wm8960"; reg = <0x1a>; - clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; - clock-names = "mclk"; wlf,shared-lrclk; }; }; +&sai1 { + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai1>; + assigned-clocks = <&clks IMX7D_SAI1_ROOT_SRC>, + <&clks IMX7D_SAI1_ROOT_CLK>; + assigned-clock-parents = <&clks IMX7D_PLL_AUDIO_POST_DIV>; + assigned-clock-rates = <0>, <36864000>; + status = "okay"; +}; + &uart1 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_uart1>; @@ -390,6 +423,16 @@ >; }; + pinctrl_sai1: sai1grp { + fsl,pins = < + MX7D_PAD_SAI1_MCLK__SAI1_MCLK 0x1f + MX7D_PAD_ENET1_RX_CLK__SAI1_TX_BCLK 0x1f + MX7D_PAD_ENET1_CRS__SAI1_TX_SYNC 0x1f + MX7D_PAD_ENET1_COL__SAI1_TX_DATA0 0x30 + MX7D_PAD_ENET1_TX_CLK__SAI1_RX_DATA0 0x1f + >; + }; + pinctrl_uart1: uart1grp { fsl,pins = < MX7D_PAD_UART1_TX_DATA__UART1_DCE_TX 0x79 diff --git a/arch/arm/boot/dts/imx7d.dtsi b/arch/arm/boot/dts/imx7d.dtsi index b5a50e0..f21b010 100644 --- a/arch/arm/boot/dts/imx7d.dtsi +++ b/arch/arm/boot/dts/imx7d.dtsi @@ -693,6 +693,20 @@ status = "disabled"; }; + sai1: sai@308a0000 { + #sound-dai-cells = <0>; + compatible = "fsl,imx7d-sai"; + reg = <0x308a0000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX7D_SAI1_IPG_CLK>, + <&clks IMX7D_SAI1_ROOT_CLK>, + <&clks IMX7D_CLK_DUMMY>, <&clks IMX7D_CLK_DUMMY>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "rx", "tx"; + dmas = <&sdma 8 24 0>, <&sdma 9 24 0>; + status = "disabled"; + }; + i2c1: i2c@30a20000 { #address-cells = <1>; #size-cells = <0>; @@ -884,6 +898,17 @@ status = "disabled"; }; + sdma: sdma@30bd0000 { + compatible = "fsl,imx7d-sdma"; + reg = <0x30bd0000 0x10000>; + interrupts = <GIC_SPI 2 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX7D_SDMA_CORE_CLK>, + <&clks IMX7D_AHB_CHANNEL_ROOT_CLK>; + clock-names = "ipg", "ahb"; + #dma-cells = <3>; + fsl,sdma-ram-script-name = "imx/sdma/sdma-imx7d.bin"; + }; + fec1: ethernet@30be0000 { compatible = "fsl,imx7d-fec", "fsl,imx6sx-fec"; reg = <0x30be0000 0x10000>; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..6e69bf55 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -258,6 +258,7 @@ CONFIG_SND_SOC_IMX_MC13783=y CONFIG_SND_SOC_FSL_ASOC_CARD=y CONFIG_SND_SOC_CS42XX8_I2C=y CONFIG_SND_SOC_TLV320AIC3X=y +CONFIG_SND_SOC_WM8960=y CONFIG_SND_SIMPLE_CARD=y CONFIG_USB=y CONFIG_USB_EHCI_HCD=y diff --git a/drivers/clk/imx/clk-imx7d.c b/drivers/clk/imx/clk-imx7d.c index fbb6a8c..c2f0c85 100644 --- a/drivers/clk/imx/clk-imx7d.c +++ b/drivers/clk/imx/clk-imx7d.c @@ -22,6 +22,10 @@ #include "clk.h" +static u32 share_count_sai1; +static u32 share_count_sai2; +static u32 share_count_sai3; + static struct clk *clks[IMX7D_CLK_END]; static const char *arm_a7_sel[] = { "osc", "pll_arm_main_clk", "pll_enet_500m_clk", "pll_dram_main_clk", @@ -777,9 +781,12 @@ static void __init imx7d_clocks_init(struct device_node *ccm_node) clks[IMX7D_MIPI_DSI_ROOT_CLK] = imx_clk_gate2("mipi_dsi_root_clk", "mipi_dsi_post_div", base + 0x4650, 0); clks[IMX7D_MIPI_CSI_ROOT_CLK] = imx_clk_gate2("mipi_csi_root_clk", "mipi_csi_post_div", base + 0x4640, 0); clks[IMX7D_MIPI_DPHY_ROOT_CLK] = imx_clk_gate2("mipi_dphy_root_clk", "mipi_dphy_post_div", base + 0x4660, 0); - clks[IMX7D_SAI1_ROOT_CLK] = imx_clk_gate2("sai1_root_clk", "sai1_post_div", base + 0x48c0, 0); - clks[IMX7D_SAI2_ROOT_CLK] = imx_clk_gate2("sai2_root_clk", "sai2_post_div", base + 0x48d0, 0); - clks[IMX7D_SAI3_ROOT_CLK] = imx_clk_gate2("sai3_root_clk", "sai3_post_div", base + 0x48e0, 0); + clks[IMX7D_SAI1_ROOT_CLK] = imx_clk_gate2_shared("sai1_root_clk", "sai1_post_div", base + 0x48c0, 0, &share_count_sai1); + clks[IMX7D_SAI1_IPG_CLK] = imx_clk_gate2_shared("sai1_ipg_clk", "ipg_root_clk", base + 0x48c0, 0, &share_count_sai1); + clks[IMX7D_SAI2_ROOT_CLK] = imx_clk_gate2_shared("sai2_root_clk", "sai2_post_div", base + 0x48d0, 0, &share_count_sai2); + clks[IMX7D_SAI2_IPG_CLK] = imx_clk_gate2_shared("sai2_ipg_clk", "ipg_root_clk", base + 0x48d0, 0, &share_count_sai2); + clks[IMX7D_SAI3_ROOT_CLK] = imx_clk_gate2_shared("sai3_root_clk", "sai3_post_div", base + 0x48e0, 0, &share_count_sai3); + clks[IMX7D_SAI3_IPG_CLK] = imx_clk_gate2_shared("sai3_ipg_clk", "ipg_root_clk", base + 0x48e0, 0, &share_count_sai3); clks[IMX7D_SPDIF_ROOT_CLK] = imx_clk_gate2("spdif_root_clk", "spdif_post_div", base + 0x44d0, 0); clks[IMX7D_ENET1_REF_ROOT_CLK] = imx_clk_gate2("enet1_ref_root_clk", "enet1_ref_post_div", base + 0x44e0, 0); clks[IMX7D_ENET1_TIME_ROOT_CLK] = imx_clk_gate2("enet1_time_root_clk", "enet1_time_post_div", base + 0x44f0, 0); diff --git a/drivers/dma/imx-sdma.c b/drivers/dma/imx-sdma.c index 0f6fd42..620849f 100644 --- a/drivers/dma/imx-sdma.c +++ b/drivers/dma/imx-sdma.c @@ -477,6 +477,24 @@ static struct sdma_driver_data sdma_imx6q = { .script_addrs = &sdma_script_imx6q, }; +static struct sdma_script_start_addrs sdma_script_imx7d = { + .ap_2_ap_addr = 644, + .uart_2_mcu_addr = 819, + .mcu_2_app_addr = 749, + .uartsh_2_mcu_addr = 1034, + .mcu_2_shp_addr = 962, + .app_2_mcu_addr = 685, + .shp_2_mcu_addr = 893, + .spdif_2_mcu_addr = 1102, + .mcu_2_spdif_addr = 1136, +}; + +static struct sdma_driver_data sdma_imx7d = { + .chnenbl0 = SDMA_CHNENBL0_IMX35, + .num_events = 48, + .script_addrs = &sdma_script_imx7d, +}; + static const struct platform_device_id sdma_devtypes[] = { { .name = "imx25-sdma", @@ -497,6 +515,9 @@ static const struct platform_device_id sdma_devtypes[] = { .name = "imx6q-sdma", .driver_data = (unsigned long)&sdma_imx6q, }, { + .name = "imx7d-sdma", + .driver_data = (unsigned long)&sdma_imx7d, + }, { /* sentinel */ } }; @@ -509,6 +530,7 @@ static const struct of_device_id sdma_dt_ids[] = { { .compatible = "fsl,imx35-sdma", .data = &sdma_imx35, }, { .compatible = "fsl,imx31-sdma", .data = &sdma_imx31, }, { .compatible = "fsl,imx25-sdma", .data = &sdma_imx25, }, + { .compatible = "fsl,imx7d-sdma", .data = &sdma_imx7d, }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, sdma_dt_ids); @@ -1384,6 +1406,7 @@ static void sdma_issue_pending(struct dma_chan *chan) #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V1 34 #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V2 38 #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V3 41 +#define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V4 42 static void sdma_add_scripts(struct sdma_engine *sdma, const struct sdma_script_start_addrs *addr) @@ -1433,6 +1456,9 @@ static void sdma_load_firmware(const struct firmware *fw, void *context) case 3: sdma->script_number = SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V3; break; + case 4: + sdma->script_number = SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V4; + break; default: dev_err(sdma->dev, "unknown firmware version\n"); goto err_firmware; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a..62d041c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -909,7 +909,8 @@ config SND_SOC_WM8955 tristate config SND_SOC_WM8960 - tristate +tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..b60497c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx7d-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx7d-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids);

1 0

[alsa-devel] [PATCH v2 2/3] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 08 Mar '16

08 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This changes adds support for exposing the card number for kernel clients as well as the PID for user client. It supports kernels with and without the required support. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 2 ++ include/sound/asequencer.h | 6 ++++-- src/seq/seq.c | 26 ++++++++++++++++++++++++++ src/seq/seq_hw.c | 12 +++++++++++- 4 files changed, 43 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..d05940e 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,8 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..3ac70fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,9 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card; /* RO: card number[kernel] */ + int pid; /* RO: pid[user] */ + char reserved[56]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..4405e68 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,32 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number. + * \param info client_info container + * \return card number or -1 if value is not available. + * + * Only available for SND_SEQ_KERNEL_CLIENT clients. + */ +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) +{ + assert(info); + return info->card; +} + +/** + * \brief Get the owning PID. + * \param info client_info container + * \return pid or -1 if value is not available. + * + * Only available for SND_SEQ_USER_CLIENT clients. + */ +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info) +{ + assert(info); + return info->pid; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..24350b6 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int micro_version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,10 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -368,6 +373,10 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -480,6 +489,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->micro_version = SNDRV_PROTOCOL_MICRO(ver); if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.1.4

3 8

[alsa-devel] [PATCH 0/9] ASoC: wm*: Fix enum ctl accesses
by Takashi Iwai 08 Mar '16

08 Mar '16

Hi, here is a patchset for correcting the wrong access types to the enum ctl values in various wm* codec drivers. I didn't put Cc to stable. If they might be used for 64 bit architectures, it's a real problem. In that case, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (9): ASoC: wm8753: Fix enum ctl accesses in a wrong type ASoC: wm8904: Fix enum ctl accesses in a wrong type ASoC: wm8958: Fix enum ctl accesses in a wrong type ASoC: wm8983: Fix enum ctl accesses in a wrong type ASoC: wm8985: Fix enum ctl accesses in a wrong type ASoC: wm8994: Fix enum ctl accesses in a wrong type ASoC: wm8996: Fix enum ctl accesses in a wrong type ASoC: wm9081: Fix enum ctl accesses in a wrong type ASoC: wm_adsp: Fix enum ctl accesses in a wrong type sound/soc/codecs/wm8753.c | 6 +++--- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8958-dsp2.c | 8 ++++---- sound/soc/codecs/wm8983.c | 14 +++++++------- sound/soc/codecs/wm8985.c | 14 +++++++------- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 8 ++++---- sound/soc/codecs/wm_adsp.c | 8 ++++---- 9 files changed, 34 insertions(+), 34 deletions(-) -- 2.7.2

3 19

[alsa-devel] [RFC PATCH 0/4] MST DP audio support
by libin.yang＠linux.intel.com 08 Mar '16

08 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> The patchset are tmp patches. The API between audio and gfx is still in discussion. Libin Yang (4): ALSA: hda - codec add DP MST support for connection list ALSA: hda - add DP mst verb support ALSA - hda: add devid support in acom ALSA - hda: add DP MST support drivers/gpu/drm/i915/intel_audio.c | 2 +- include/drm/i915_component.h | 2 +- include/sound/hda_i915.h | 6 +- sound/hda/hdac_i915.c | 7 +- sound/pci/hda/hda_codec.c | 217 ++++++++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.h | 8 ++ sound/pci/hda/patch_hdmi.c | 172 ++++++++++++++++++++--------- 7 files changed, 340 insertions(+), 74 deletions(-) -- 1.9.1

3 12

Re: [alsa-devel] Testing SAI with mainline
by Fabio Estevam 08 Mar '16

08 Mar '16

Hi Zidan, On Mon, Mar 7, 2016 at 11:49 PM, Zidan Wang <zidan.wang(a)nxp.com> wrote: > You should change the sdma peripheral types of SAI to 24. Thanks! That did the trick. I have also tried with 25, which is what NXP BSP uses, but it did not work. Care to explain why 24 is needed here? I will prepare some patches and submit them tomorrow so that we have SAI working on this board. Thanks for your help

1 0

[alsa-devel] [PATCH v5 22/22] sound/usb: Use Media Controller API to share media resources
by Shuah Khan 07 Mar '16

07 Mar '16

Change ALSA driver to use Media Controller API to share media resources with DVB and V4L2 drivers on a AU0828 media device. Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> --- Changes since v4: 1. Renamed media_* interfaces in sound/usb to media_snd* to avoid potential conflicts with the MC interfaces. This addresses Laurent Pinchart's comment. 2. Added Acked-by from Takashi Iwai sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 +++ sound/usb/card.h | 3 + sound/usb/media.c | 318 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 72 +++++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 11 files changed, 448 insertions(+), 5 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index a452ad7..d14bf41 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,6 +15,7 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE + select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -22,6 +23,9 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_AUDIO_USE_MEDIA_CONTROLLER + bool + config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2d2d122..8dca3c4 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,6 +15,8 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o +snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o + snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 1f09d95..258cf70 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,6 +66,7 @@ #include "format.h" #include "power.h" #include "stream.h" +#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -561,6 +562,11 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; + if (quirk->media_device) { + /* don't want to fail when media_snd_device_create() fails */ + media_snd_device_create(chip, intf); + } + usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -617,6 +623,14 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } + /* + * Nice to check quirk && quirk->media_device + * need some special handlings. Doesn't look like + * we have access to quirk here + * Acceses mixer_list + */ + media_snd_device_delete(chip); + /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 71778ca..34a0898 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,6 +105,8 @@ struct snd_usb_endpoint { struct list_head list; }; +struct media_ctl; + struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -156,6 +158,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c new file mode 100644 index 0000000..6c0dfd8 --- /dev/null +++ b/sound/usb/media.c @@ -0,0 +1,318 @@ +/* + * media.c - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ + +#include <linux/init.h> +#include <linux/list.h> +#include <linux/mutex.h> +#include <linux/slab.h> +#include <linux/usb.h> + +#include <sound/pcm.h> +#include <sound/core.h> + +#include "usbaudio.h" +#include "card.h" +#include "mixer.h" +#include "media.h" + +static int media_snd_enable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->enable_source) + return mctl->media_dev->enable_source(&mctl->media_entity, + &mctl->media_pipe); + return 0; +} + +static void media_snd_disable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->disable_source) + mctl->media_dev->disable_source(&mctl->media_entity); +} + +int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream) +{ + struct media_device *mdev; + struct media_ctl *mctl; + struct device *pcm_dev = &pcm->streams[stream].dev; + u32 intf_type; + int ret = 0; + u16 mixer_pad; + struct media_entity *entity; + + mdev = subs->stream->chip->media_dev; + if (!mdev) + return -ENODEV; + + if (subs->media_ctl) + return 0; + + /* allocate media_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; + mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; + mixer_pad = 1; + } else { + intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; + mctl->media_pad.flags = MEDIA_PAD_FL_SINK; + mixer_pad = 2; + } + mctl->media_entity.name = pcm->name; + media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) + goto err1; + + mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, + MAJOR(pcm_dev->devt), + MINOR(pcm_dev->devt)); + if (!mctl->intf_devnode) { + ret = -ENOMEM; + goto err2; + } + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &mctl->intf_devnode->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + ret = -ENOMEM; + goto err3; + } + + /* create link between mixer and audio */ + media_device_for_each_entity(entity, mdev) { + switch (entity->function) { + case MEDIA_ENT_F_AUDIO_MIXER: + ret = media_create_pad_link(entity, mixer_pad, + &mctl->media_entity, 0, + MEDIA_LNK_FL_ENABLED); + if (ret) + goto err4; + break; + } + } + + subs->media_ctl = mctl; + return 0; + +err4: + media_remove_intf_link(mctl->intf_link); +err3: + media_devnode_remove(mctl->intf_devnode); +err2: + media_device_unregister_entity(&mctl->media_entity); +err1: + kfree(mctl); + return ret; +} + +void media_snd_stream_delete(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl && mctl->media_dev) { + struct media_device *mdev; + + mdev = subs->stream->chip->media_dev; + if (mdev && media_devnode_is_registered(&mdev->devnode)) { + media_devnode_remove(mctl->intf_devnode); + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + subs->media_ctl = NULL; + } +} + +int media_snd_start_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + return media_snd_enable_source(mctl); + return 0; +} + +void media_snd_stop_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + media_snd_disable_source(mctl); +} + +int media_snd_mixer_init(struct snd_usb_audio *chip) +{ + struct device *ctl_dev = &chip->card->ctl_dev; + struct media_intf_devnode *ctl_intf; + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + struct media_mixer_ctl *mctl; + u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; + int ret; + + if (!mdev) + return -ENODEV; + + ctl_intf = chip->ctl_intf_media_devnode; + if (!ctl_intf) { + ctl_intf = media_devnode_create(mdev, intf_type, 0, + MAJOR(ctl_dev->devt), + MINOR(ctl_dev->devt)); + if (!ctl_intf) + return -ENOMEM; + chip->ctl_intf_media_devnode = ctl_intf; + } + + list_for_each_entry(mixer, &chip->mixer_list, list) { + + if (mixer->media_mixer_ctl) + continue; + + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; + mctl->media_entity.name = chip->card->mixername; + mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; + mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; + mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; + media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, + mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) { + kfree(mctl); + return ret; + } + + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &ctl_intf->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + kfree(mctl); + return -ENOMEM; + } + mctl->intf_devnode = ctl_intf; + mixer->media_mixer_ctl = mctl; + } + return 0; +} + +static void media_snd_mixer_delete(struct snd_usb_audio *chip) +{ + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + + if (!mdev) + return; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + struct media_mixer_ctl *mctl; + + mctl = mixer->media_mixer_ctl; + if (!mixer->media_mixer_ctl) + continue; + + if (media_devnode_is_registered(&mdev->devnode)) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + mixer->media_mixer_ctl = NULL; + } + if (media_devnode_is_registered(&mdev->devnode)) + media_devnode_remove(chip->ctl_intf_media_devnode); + chip->ctl_intf_media_devnode = NULL; +} + +int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) +{ + struct media_device *mdev; + struct usb_device *usbdev = interface_to_usbdev(iface); + int ret; + + mdev = media_device_get_devres(&usbdev->dev); + if (!mdev) + return -ENOMEM; + if (!mdev->dev) { + /* register media device */ + mdev->dev = &usbdev->dev; + if (usbdev->product) + strlcpy(mdev->model, usbdev->product, + sizeof(mdev->model)); + if (usbdev->serial) + strlcpy(mdev->serial, usbdev->serial, + sizeof(mdev->serial)); + strcpy(mdev->bus_info, usbdev->devpath); + mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); + media_device_init(mdev); + } + if (!media_devnode_is_registered(&mdev->devnode)) { + ret = media_device_register(mdev); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't register media device. Error: %d\n", + ret); + return ret; + } + } + + /* save media device - avoid lookups */ + chip->media_dev = mdev; + + /* Create media entities for mixer and control dev */ + ret = media_snd_mixer_init(chip); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't create media mixer entities. Error: %d\n", + ret); + + /* clear saved media_dev */ + chip->media_dev = NULL; + + return ret; + } + return 0; +} + +void media_snd_device_delete(struct snd_usb_audio *chip) +{ + struct media_device *mdev = chip->media_dev; + + media_snd_mixer_delete(chip); + + if (mdev) { + if (media_devnode_is_registered(&mdev->devnode)) + media_device_unregister(mdev); + chip->media_dev = NULL; + } +} diff --git a/sound/usb/media.h b/sound/usb/media.h new file mode 100644 index 0000000..1dcdcdc --- /dev/null +++ b/sound/usb/media.h @@ -0,0 +1,72 @@ +/* + * media.h - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ +#ifndef __MEDIA_H + +#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER + +#include <media/media-device.h> +#include <media/media-entity.h> +#include <sound/asound.h> + +struct media_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad; + struct media_pipeline media_pipe; +}; + +/* + * One source pad each for SNDRV_PCM_STREAM_CAPTURE and + * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link + * to AUDIO Source +*/ +#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) + +struct media_mixer_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; + struct media_pipeline media_pipe; +}; + +int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface); +void media_snd_device_delete(struct snd_usb_audio *chip); +int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream); +void media_snd_stream_delete(struct snd_usb_substream *subs); +int media_snd_start_pipeline(struct snd_usb_substream *subs); +void media_snd_stop_pipeline(struct snd_usb_substream *subs); +#else +static inline int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) + { return 0; } +static inline void media_snd_device_delete(struct snd_usb_audio *chip) { } +static inline int media_snd_stream_init(struct snd_usb_substream *subs, + struct snd_pcm *pcm, int stream) + { return 0; } +static inline void media_snd_stream_delete(struct snd_usb_substream *subs) { } +static inline int media_snd_start_pipeline(struct snd_usb_substream *subs) + { return 0; } +static inline void media_snd_stop_pipeline(struct snd_usb_substream *subs) { } +#endif +#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3417ef3..f378944 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,6 +3,8 @@ #include <sound/info.h> +struct media_mixer_ctl; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -22,6 +24,7 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; + struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9245f52..b0370d5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,6 +35,7 @@ #include "pcm.h" #include "clock.h" #include "power.h" +#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -715,10 +716,14 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; + ret = media_snd_start_pipeline(subs); + if (ret) + return ret; + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto err_ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -732,22 +737,27 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - return -EINVAL; + ret = -EINVAL; + goto err_ret; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; + +err_ret: + media_snd_stop_pipeline(subs); + return ret; } /* @@ -759,6 +769,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; + media_snd_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1219,6 +1230,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; + int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1232,7 +1244,12 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - return setup_hw_info(runtime, subs); + ret = setup_hw_info(runtime, subs); + if (ret == 0) + ret = media_snd_stream_init(subs, as->pcm, direction); + if (ret) + snd_usb_autosuspend(subs->stream->chip); + return ret; } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1241,6 +1258,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); + media_snd_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c60a776..9d087b1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,6 +2886,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c4dc577..51258a1 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,6 +36,7 @@ #include "format.h" #include "clock.h" #include "stream.h" +#include "media.h" /* * free a substream @@ -52,6 +53,7 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); + media_snd_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85..a161c7c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,6 +30,9 @@ * */ +struct media_device; +struct media_intf_devnode; + struct snd_usb_audio { int index; struct usb_device *dev; @@ -60,6 +63,8 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ + struct media_device *media_dev; + struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -110,6 +115,7 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; + bool media_device; const void *data; }; -- 2.5.0

3 8

[alsa-devel] Testing SAI with mainline
by Fabio Estevam 07 Mar '16

07 Mar '16

Hi, I am trying to get SAI to work on a mx6sx sdb board. Here are the changes I made: http://pastebin.com/aq32B24v I can see the driver getting probed: [ 2.198276] fsl-asoc-card sound: wm8962 <-> 21d4000.sai mapping ok ,but when attempting a playback it fails: $ aplay clarinet.wav [ 110.843619] fsl-sai 21d4000.sai: isr: Transmit underrun detected [ 110.849841] fsl-sai 21d4000.sai: isr: Transmit underrun detected [ 110.855995] fsl-sai 21d4000.sai: isr: Transmit underrun detected (repeats forever). Any ideas as to how to get SAI working on this platform? Zidan, How have you tested SAI when you usptream the driver? Thanks

2 2

[alsa-devel] [PATCH] ALSA: hda - Fix unexpected resume through regmap code path
by Takashi Iwai 07 Mar '16

07 Mar '16

HD-audio driver has a mechanism to trigger the runtime resume automatically at accessing the verbs. This auto-resume, however, causes the mutex deadlock when invoked from the regmap handler since the regmap keeps the mutex while auto-resuming. For avoiding that, there is some tricky check in the HDA regmap handler to return -EAGAIN error to back-off when the codec is powered down. Then the caller of regmap r/w will retry after properly turning on the codec power. This works in most cases, but there seems a slight race between the codec power check and the actual on-demand auto-resume trigger. This resulted in the lockdep splat, eventually leading to a real deadlock. This patch tries to address the race window by getting the runtime PM refcount at the check time using pm_runtime_get_if_in_use(). With this call, we can keep the power on only when the codec has been already turned on, and back off if not. For keeping the code consistency, the code touching the runtime PM is stored in hdac_device.c although it's used only locally in hdac_regmap.c. Reported-by: Jiri Slaby <jslaby(a)suse.cz> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 2 ++ sound/hda/hdac_device.c | 16 ++++++++++++ sound/hda/hdac_regmap.c | 69 ++++++++++++++++++++++++++++++++----------------- 3 files changed, 64 insertions(+), 23 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index c21c38ce7450..93e63c56f48f 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -168,11 +168,13 @@ int snd_hdac_power_up(struct hdac_device *codec); int snd_hdac_power_down(struct hdac_device *codec); int snd_hdac_power_up_pm(struct hdac_device *codec); int snd_hdac_power_down_pm(struct hdac_device *codec); +int snd_hdac_keep_power_up(struct hdac_device *codec); #else static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; } #endif /* diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index e361024eabb6..d1a4d6973330 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -611,6 +611,22 @@ int snd_hdac_power_up_pm(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); +/* like snd_hdac_power_up_pm(), but only increment the pm count when + * already powered up. Returns -1 if not powered up, 1 if incremented + * or 0 if unchanged. Only used in hdac_regmap.c + */ +int snd_hdac_keep_power_up(struct hdac_device *codec) +{ + if (!atomic_inc_not_zero(&codec->in_pm)) { + int ret = pm_runtime_get_if_in_use(&codec->dev); + if (!ret) + return -1; + if (ret < 0) + return 0; + } + return 1; +} + /** * snd_hdac_power_down_pm - power down the codec * @codec: the codec object diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index eb8f7c30cb09..bdbcd6b75ff6 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -21,13 +21,16 @@ #include <sound/hdaudio.h> #include <sound/hda_regmap.h> -#ifdef CONFIG_PM -#define codec_is_running(codec) \ - (atomic_read(&(codec)->in_pm) || \ - !pm_runtime_suspended(&(codec)->dev)) -#else -#define codec_is_running(codec) true -#endif +static int codec_pm_lock(struct hdac_device *codec) +{ + return snd_hdac_keep_power_up(codec); +} + +static void codec_pm_unlock(struct hdac_device *codec, int lock) +{ + if (lock == 1) + snd_hdac_power_down_pm(codec); +} #define get_verb(reg) (((reg) >> 8) & 0xfff) @@ -238,20 +241,28 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) struct hdac_device *codec = context; int verb = get_verb(reg); int err; + int pm_lock = 0; - if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE) - return -EAGAIN; + if (verb != AC_VERB_GET_POWER_STATE) { + pm_lock = codec_pm_lock(codec); + if (pm_lock < 0) + return -EAGAIN; + } reg |= (codec->addr << 28); - if (is_stereo_amp_verb(reg)) - return hda_reg_read_stereo_amp(codec, reg, val); - if (verb == AC_VERB_GET_PROC_COEF) - return hda_reg_read_coef(codec, reg, val); + if (is_stereo_amp_verb(reg)) { + err = hda_reg_read_stereo_amp(codec, reg, val); + goto out; + } + if (verb == AC_VERB_GET_PROC_COEF) { + err = hda_reg_read_coef(codec, reg, val); + goto out; + } if ((verb & 0x700) == AC_VERB_SET_AMP_GAIN_MUTE) reg &= ~AC_AMP_FAKE_MUTE; err = snd_hdac_exec_verb(codec, reg, 0, val); if (err < 0) - return err; + goto out; /* special handling for asymmetric reads */ if (verb == AC_VERB_GET_POWER_STATE) { if (*val & AC_PWRST_ERROR) @@ -259,7 +270,9 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) else /* take only the actual state */ *val = (*val >> 4) & 0x0f; } - return 0; + out: + codec_pm_unlock(codec, pm_lock); + return err; } static int hda_reg_write(void *context, unsigned int reg, unsigned int val) @@ -267,6 +280,7 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) struct hdac_device *codec = context; unsigned int verb; int i, bytes, err; + int pm_lock = 0; if (codec->caps_overwriting) return 0; @@ -275,14 +289,21 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) reg |= (codec->addr << 28); verb = get_verb(reg); - if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE) - return codec->lazy_cache ? 0 : -EAGAIN; + if (verb != AC_VERB_SET_POWER_STATE) { + pm_lock = codec_pm_lock(codec); + if (pm_lock < 0) + return codec->lazy_cache ? 0 : -EAGAIN; + } - if (is_stereo_amp_verb(reg)) - return hda_reg_write_stereo_amp(codec, reg, val); + if (is_stereo_amp_verb(reg)) { + err = hda_reg_write_stereo_amp(codec, reg, val); + goto out; + } - if (verb == AC_VERB_SET_PROC_COEF) - return hda_reg_write_coef(codec, reg, val); + if (verb == AC_VERB_SET_PROC_COEF) { + err = hda_reg_write_coef(codec, reg, val); + goto out; + } switch (verb & 0xf00) { case AC_VERB_SET_AMP_GAIN_MUTE: @@ -319,10 +340,12 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) reg |= (verb + i) << 8 | ((val >> (8 * i)) & 0xff); err = snd_hdac_exec_verb(codec, reg, 0, NULL); if (err < 0) - return err; + goto out; } - return 0; + out: + codec_pm_unlock(codec, pm_lock); + return err; } static const struct regmap_config hda_regmap_cfg = { -- 2.7.2

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[alsa-devel] [PATCH v3 0/7] ALSA: hda: Move chmap helpers to core
by Subhransu S. Prusty 07 Mar '16

07 Mar '16

The HDMI channel map helper APIs and control registration can be reused by ASoC skylake HDMI drivers. A common chmap object is created, channel map control callback handlers, helper APIs and data structures are moved to the core. Driver object store an instance of this chmap object to register chmap controls and program channel allocation for HDMI infoframe. Driver registers ops for specific operations. changes in v3: - Namespace managed as per hda frmwrk requirement - Address review comments in v2 Subhransu S. Prusty (7): ALSA: hda - Create common chmap object ALSA: hda - Register chmap obj as priv data instead of codec ALSA: hda - Use hdac name space for CEA spk alloc structure ALSA: hda - Add hdmi chmap verb programming ops to chmap object ALSA: hda - chmap helper args modified to use generic hdac objs. ALSA: hda - Move chmap support helpers/ops to core ALSA: hda - Use snd_hdac namespace prefix for chmap exported APIs include/sound/hda_chmap.h | 76 +++++ sound/hda/Makefile | 2 +- sound/hda/hdmi_chmap.c | 791 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_eld.c | 31 +- sound/pci/hda/patch_hdmi.c | 824 +++++---------------------------------------- 5 files changed, 950 insertions(+), 774 deletions(-) create mode 100644 include/sound/hda_chmap.h create mode 100644 sound/hda/hdmi_chmap.c -- 1.9.1

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Re: [alsa-devel] [PATCH 4/4] ALSA: dice: force to add two pcm devices for listed models
by Takashi Sakamoto 07 Mar '16

07 Mar '16

On Mar 7 2016 22:43, Stefan Richter wrote: > On Mar 07 Takashi Sakamoto wrote: >> Thanks for your report. Totally: >> * The patched snd-dice still works fine with Saffire Pro 24. >> * The pached snd-dice successfully manages several isochronous streams >> without code bugs (i.e. kernel NULL pointer dereference or >> continuous locking) with Saffire Pro 24/40. >> * The pached snd-dice still has mute issue with Saffire Pro 40. > > I concur. While I was only able to perform just a few tests yet, there > doesn't appear to be any regression, and this patchset adds support for > the previously inaccessible channels at the second transmitter and > receiver. > >> Anyway, the result cannot block applying this patchset for 4.6. I'll >> post it in this night for merging. > > Thanks. I posted. Thanks for your test ;) >> Well, about the mute issue, I have no idea to fix it at all. At least, >> for my test units, the patched snd-dice works fine without such issues. >> Therefore, it's nearly model-dependent issue and I have no way to do for >> it. FYI, I tested with: >> * OHCI 1394 host controllers >> * VIA VT6315 >> * TI XIO2213 >> * Dice II units >> * TC Electronic ImpactTwin >> * TC Electronic Konnekt24D >> * TC Electronic Konnekt8 >> * Dice Jr. units >> * TC Electronic DesktopKonnect6 >> * Dice Mini units >> * Focusrite Saffire Pro 26 > > FWIW, Saffire PRO 24 and 40 are based on Dice Junior according to FFADO's > device database. If desired I could open up my 24 and 40 to check. If you don't mind it, please. (Because once opening your unit, usually, you cannot get repair service from sellers.) And I wrote wrong for Dice Jr./Mini units. Correctly: * Dice Jr. units * TC Electronic DesktopKonnect6 * Focusrite Saffire Pro 26 * I have no units with Dice Mini (Oh...). Anyway, all of my units with Dice Jr. works fine. So we can judge that Pro 40 has its own quirk which we don't know. > Furthermore FYI, the FFADO userspace streaming driver has got the following > two known issues with the PRO 40: > http://subversion.ffado.org/ticket/326 > playback at 96 kHz is choppy (but fine at 44.1, 48, 88.2 kHz; > reproduced by multiple users, 100% reproducible by myself too) > http://subversion.ffado.org/ticket/372 > MIDI playback problem (perhaps dependent on firmware revision; > cannot reproduce this myself due to lack of MIDI equipment) Hm. With a lack of something like packet dump or detailed case-investigation, I can do nothing for the unit. I have no fashion to lost my time for meaningless talks, sorry. For our information, I put your result of hinawa-dice-common-cui from Saffire Pro 40. (meaningless lines are omitted.) $ ./hinawa-dice-common-cui 00130e040140dd28 current-status owner address: ffc1000100000000 global enabled: False clock status: locked: 1 nominal: 48000 clock source detection: locked: slip: measured rate: 48000 dice version: 1.0.12.0 latest notification: 0x00000010 clock source: Internal sampling rate: 48000 nickname: Pro40-00dd28 Transmitted streams: Stream 0: iso-channel: 4294967295 speed: 2 pcm: 10 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: IP 1 1: IP 2 2: IP 3 3: IP 4 4: IP 5 5: IP 6 6: IP 7 7: IP 8 8: SPDIF L 9: SPDIF R Stream 1: iso-channel: 4294967295 speed: 2 pcm: 10 midi: 0 iec60958: caps: 0 enable: 0 formation: 0: ADAT 1 1: ADAT 2 2: ADAT 3 3: ADAT 4 4: ADAT 5 5: ADAT 6 6: ADAT 7 7: ADAT 8 8: Loop 1 9: Loop 2 Received streams: Stream 0: iso-channel: 4294967295 start: 0 pcm: 12 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: Mon 1 1: Mon 2 2: Line 3 3: Line 4 4: Line 5 5: Line 6 6: Line 7 7: Line 8 8: Line 9 9: Line 10 10: SPDIF L 11: SPDIF R Stream 1: iso-channel: 4294967295 start: 0 pcm: 8 midi: 0 iec60958: caps: 0 enable: 0 formation: 0: ADAT 1 1: ADAT 2 2: ADAT 3 3: ADAT 4 4: ADAT 5 5: ADAT 6 6: ADAT 7 7: ADAT 8 Synchronization infomation: clock-source: Internal locked: 1 sampling-rate: 48000 ADAT: data-bits: 0 no-data: 1 $ ./hinawa-dice-common-cui 00130e040140dd28 sampling-rate set 96000 $ ./hinawa-dice-common-cui 00130e040140dd28 current-status owner address: ffc1000100000000 global enabled: False clock status: locked: 1 nominal: 96000 clock source detection: locked: slip: measured rate: 96000 dice version: 1.0.12.0 latest notification: 0x00000010 clock source: Internal sampling rate: 96000 nickname: Pro40-00dd28 Transmitted streams: Stream 0: iso-channel: 4294967295 speed: 2 pcm: 16 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: IP 1 1: IP 2 2: IP 3 3: IP 4 4: IP 5 5: IP 6 6: IP 7 7: IP 8 8: SPDIF L 9: SPDIF R 10: ADAT 1 11: ADAT 2 12: ADAT 3 13: ADAT 4 14: Loop 1 15: Loop 2 Received streams: Stream 0: iso-channel: 4294967295 start: 0 pcm: 16 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: Mon 1 1: Mon 2 2: Line 3 3: Line 4 4: Line 5 5: Line 6 6: Line 7 7: Line 8 8: Line 9 9: Line 10 10: SPDIF L 11: SPDIF R 12: ADAT 1 13: ADAT 2 14: ADAT 3 15: ADAT 4 Synchronization infomation: clock-source: Internal locked: 1 sampling-rate: 96000 ADAT: data-bits: 0 no-data: 1 Regards Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: cs35l32: avoid uninitialized variable access
by Arnd Bergmann 07 Mar '16

07 Mar '16

gcc warns about the possibilty of accessing a property read from devicetree in cs35l32_i2c_probe() when it has not been initialized because CONFIG_OF is disabled: sound/soc/codecs/cs35l32.c: In function 'cs35l32_i2c_probe': sound/soc/codecs/cs35l32.c:278:2: warning: 'val' may be used uninitialized in this function [-Wmaybe-uninitialized] The code is actually correct because it checks the dev->of_node variable first and we know this is NULL here when CONFIG_OF is disabled, but Russell King noticed that it's broken when we probe the device using DT, and the properties are absent. The code already has some checking for incorrect values, and I keep that checking unchanged here, but add an additional check for an error returned by the property accessor functions that now gets handled the same way as incorrect data in the properties. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cs35l32.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..c490dc74121b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -274,21 +274,24 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) pdata->sdout_share = val; - of_property_read_u32(np, "cirrus,boost-manager", &val); + if (of_property_read_u32(np, "cirrus,boost-manager", &val)) + val = -1u; + switch (val) { case CS35L32_BOOST_MGR_AUTO: case CS35L32_BOOST_MGR_AUTO_AUDIO: case CS35L32_BOOST_MGR_BYPASS: case CS35L32_BOOST_MGR_FIXED: - pdata->boost_mng = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,boost-manager DT value %d\n", val); pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; } - of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + if (of_property_read_u32(np, "cirrus,sdout-datacfg", &val)) + val = -1u; switch (val) { case CS35L32_DATA_CFG_LR_VP: case CS35L32_DATA_CFG_LR_STAT: @@ -296,13 +299,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_DATA_CFG_LR_VPSTAT: pdata->sdout_datacfg = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,sdout-datacfg DT value %d\n", val); pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; } - of_property_read_u32(np, "cirrus,battery-threshold", &val); + if (of_property_read_u32(np, "cirrus,battery-threshold", &val)) + val = -1u; switch (val) { case CS35L32_BATT_THRESH_3_1V: case CS35L32_BATT_THRESH_3_2V: @@ -310,13 +315,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_THRESH_3_4V: pdata->batt_thresh = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-threshold DT value %d\n", val); pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; } - of_property_read_u32(np, "cirrus,battery-recovery", &val); + if (of_property_read_u32(np, "cirrus,battery-recovery", &val)) + val = -1u; switch (val) { case CS35L32_BATT_RECOV_3_1V: case CS35L32_BATT_RECOV_3_2V: -- 2.7.0

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[alsa-devel] [GIT PULL] ASoC fixes for v4.5
by Mark Brown 07 Mar '16

07 Mar '16

The following changes since commit fc77dbd34c5c99bce46d40a2491937c3bcbd10af: Linux 4.5-rc6 (2016-02-28 08:41:20 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.5-rc6 for you to fetch changes up to 02db17ff376893f65f45974b0897e70e6695034e: Merge remote-tracking branches 'asoc/fix/wm8985', 'asoc/fix/wm8994', 'asoc/fix/wm8996', 'asoc/fix/wm9081' and 'asoc/fix/wm9713' into asoc-linus (2016-03-05 21:27:13 +0900) ---------------------------------------------------------------- ASoC: Fixes for v4.5 This is far too big a set of fixes for this late in the release cycle but the overwhelming bulk is essentially the same simple fix from Takashi for a cut'n'pasted 64 bit cleanliness issue in the userspace interface where drivers were accessing things using the wrong element in a union which worked OK on 32 bit platforms as the correct element happened to be aligned the same way but with 64 bit platforms ABIs are different and the two members of the union are laid out in different places. They aren't all tagged to stable since some of these chips have vanishingly little chance of being used in 64 bit systems. The other changes are: - A fix for Qualcomm devices to work on big endian systems. The original change is actually correct but triggered a bug in regmap which is too invasive to fix for this cycle and can be worked around by just letting regmap pick the default. - A fix for the Samsung I2S driver locking which wasn't using IRQ safe spinlocks when it needed to. - A fix for the new Intel Sky Lake driver forgetting that C pointer arithmetic takes the type of the pointer into consideration. - A revert of a change to the FSL SSI driver that broke some systems. - A fix for the cleanup path of the wm9713 driver. - A fix for some incorrect register definitions in the ADAU17x1 driver that caused misclocking in some configurations. - A fix for the tracepoints for jack detection to avoid using an internal field of the core jack structure which is no longer present in all configurations. - A fix for another of the new Intel drivers which tried to write to a string literal. ---------------------------------------------------------------- Alan (1): ASoC: Intel: Skylake: fix pointer scaling Andreas Irestål (1): ASoC: adau17x1: Fix incorrect BCLK ratio definitions Arnd Bergmann (1): ASoC: trace: fix printing jack name Carlo Caione (1): ASoC: cht_bsw_rt5645: Fix writing to string literal Charles Keepax (1): ASoC: samsung: Use IRQ safe spin lock calls Fabio Estevam (1): ASoC: fsl_ssi: Go back to explicit register defaults Mark Brown (6): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/ab8500', 'asoc/fix/adau17x1', 'asoc/fix/adsp', 'asoc/fix/cs42l51', 'asoc/fix/da732x' and 'asoc/fix/fsl-ssi' into asoc-linus Merge remote-tracking branches 'asoc/fix/jack', 'asoc/fix/max98088', 'asoc/fix/max98095', 'asoc/fix/omap', 'asoc/fix/pxa' and 'asoc/fix/qcom-be' into asoc-linus Merge remote-tracking branches 'asoc/fix/samsung', 'asoc/fix/tlv320dac33', 'asoc/fix/wl1273', 'asoc/fix/wm8753', 'asoc/fix/wm8904', 'asoc/fix/wm8958' and 'asoc/fix/wm8983' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8985', 'asoc/fix/wm8994', 'asoc/fix/wm8996', 'asoc/fix/wm9081' and 'asoc/fix/wm9713' into asoc-linus Robert Jarzmik (1): ASoC: wm9713: fix regmap free path Stephen Boyd (1): ASoC: qcom: Don't specify LE device endianness Takashi Iwai (25): ASoC: dapm: Fix ctl value accesses in a wrong type ASoC: pxa: corgi: Fix enum ctl accesses in a wrong type ASoC: pxa: magician: Fix enum ctl accesses in a wrong type ASoC: pxa: poodle: Fix enum ctl accesses in a wrong type ASoC: pxa: spitz: Fix enum ctl accesses in a wrong type ASoC: pxa: tosa: Fix enum ctl accesses in a wrong type ASoC: omap: n810: Fix enum ctl accesses in a wrong type ASoC: omap: rx51: Fix enum ctl accesses in a wrong type ASoC: intel: mfld: Fix enum ctl accesses in a wrong type ASoC: cs42l51: Fix enum ctl accesses in a wrong type ASoC: da732x: Fix enum ctl accesses in a wrong type ASoC: ab8500: Fix enum ctl accesses in a wrong type ASoC: max98088: Fix enum ctl accesses in a wrong type ASoC: max98095: Fix enum ctl accesses in a wrong type ASoC: tlv320dac33: Fix enum ctl accesses in a wrong type ASoC: wl1273: Fix enum ctl accesses in a wrong type ASoC: wm8753: Fix enum ctl accesses in a wrong type ASoC: wm8904: Fix enum ctl accesses in a wrong type ASoC: wm8958: Fix enum ctl accesses in a wrong type ASoC: wm8983: Fix enum ctl accesses in a wrong type ASoC: wm8985: Fix enum ctl accesses in a wrong type ASoC: wm8994: Fix enum ctl accesses in a wrong type ASoC: wm8996: Fix enum ctl accesses in a wrong type ASoC: wm9081: Fix enum ctl accesses in a wrong type ASoC: wm_adsp: Fix enum ctl accesses in a wrong type include/trace/events/asoc.h | 8 +++---- sound/soc/codecs/ab8500-codec.c | 8 +++---- sound/soc/codecs/adau17x1.h | 6 ++--- sound/soc/codecs/cs42l51.c | 8 +++---- sound/soc/codecs/da732x.c | 8 +++---- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/codecs/tlv320dac33.c | 9 ++++--- sound/soc/codecs/wl1273.c | 13 +++++----- sound/soc/codecs/wm8753.c | 6 ++--- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8958-dsp2.c | 8 +++---- sound/soc/codecs/wm8983.c | 14 +++++------ sound/soc/codecs/wm8985.c | 14 +++++------ sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 8 +++---- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_adsp.c | 8 +++---- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++----------------- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 ++++- sound/soc/intel/boards/mfld_machine.c | 16 ++++++------- sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/omap/n810.c | 18 +++++++------- sound/soc/omap/rx51.c | 18 +++++++------- sound/soc/pxa/corgi.c | 12 +++++----- sound/soc/pxa/magician.c | 6 ++--- sound/soc/pxa/poodle.c | 12 +++++----- sound/soc/pxa/spitz.c | 12 +++++----- sound/soc/pxa/tosa.c | 12 +++++----- sound/soc/qcom/lpass-cpu.c | 1 - sound/soc/samsung/i2s.c | 21 ++++++++++------- sound/soc/soc-dapm.c | 8 +++---- 33 files changed, 162 insertions(+), 160 deletions(-)

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