April 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2 v6] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 21 Apr '16

21 Apr '16

Because applied patch (commit 0beeb8d0b434b8b007ac8058fb2cac5997fff6df "ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") is dropped, I upload new version which include dependent patch. Changes since v5: - add "ASoC: core: export snd_soc_get_dai_name" patch. Because snd_soc_of_get_dai_name() fix the property name "sound-dai". Our patch needs to change name to use, so export snd_soc_get_dai_name() which is included in snd_soc_of_get_dai_name(). Changes since v4: Change name from "sound-dai" to "capture-dai" to determine the capture path Changes since v2,v3: Fix unclear code and use sound-dai to determine the capture path Changes since v1: Use codec dai name to determine the capture path. Jean-Francois Moine (1): ASoC: core: export snd_soc_get_dai_name PC Liao (1): ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ include/sound/soc.h | 2 + sound/soc/mediatek/mt8173-rt5650.c | 44 ++++++++++++++++++-- sound/soc/soc-core.c | 5 ++- 4 files changed, 52 insertions(+), 5 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH v8 0/2] ASoc: sun4i-codec: Add FM, Line and Mic inputs
by Danny Milosavljevic 21 Apr '16

21 Apr '16

Hi, this is the eighth version of the patch set that adds inputs to sun4i-codec. Changes compared to v7 are: - fix the routes for line and mic capturing. The patches are on top of the branch "sunxi-next" in <git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git>. Regards, Danny --- Danny Milosavljevic (2): sun4i-codec: Add FM, Line and Mic inputs sun4i-codec: make it possible to use different codec_widgets for A10 and A20. b/sound/soc/sunxi/sun4i-codec.c | 230 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 215 insertions(+), 15 deletions(-)

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[alsa-devel] sound: use-after-free in snd_timer_interrupt
by Dmitry Vyukov 21 Apr '16

21 Apr '16

Hello, I am hitting the following use-after-free while running syzkaller fuzzer on commit 8e0f93cda48ed054e1216bab5c60017e1a5fc1e8 ================================================================== BUG: KASAN: use-after-free in __list_del_entry+0x1d3/0x1e0 at addr ffff88002ebf6e20 Read of size 8 by task syz-executor/7684 ============================================================================= BUG kmalloc-256 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- INFO: Allocated in snd_timer_instance_new+0x52/0x3a0 age=5 cpu=0 pid=7693 [< none >] ___slab_alloc+0x578/0x5d0 mm/slub.c:2464 [< none >] __slab_alloc+0x66/0xc0 mm/slub.c:2493 [< inline >] slab_alloc_node mm/slub.c:2556 [< inline >] slab_alloc mm/slub.c:2598 [< none >] kmem_cache_alloc_trace+0x242/0x3b0 mm/slub.c:2615 [< inline >] kmalloc include/linux/slab.h:463 [< inline >] kzalloc include/linux/slab.h:607 [< none >] snd_timer_instance_new+0x52/0x3a0 sound/core/timer.c:106 [< none >] snd_timer_open+0x522/0xd20 sound/core/timer.c:289 [< inline >] snd_timer_user_tselect sound/core/timer.c:1612 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1888 [< none >] snd_timer_user_ioctl+0x8f4/0x2490 sound/core/timer.c:1918 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 INFO: Freed in snd_timer_close+0x3ee/0x750 age=9 cpu=0 pid=7693 [< none >] __slab_free+0x1fc/0x320 mm/slub.c:2674 [< inline >] slab_free mm/slub.c:2829 [< none >] kfree+0x2f5/0x370 mm/slub.c:3660 [< none >] snd_timer_close+0x3ee/0x750 sound/core/timer.c:375 [< inline >] snd_timer_user_tselect sound/core/timer.c:1602 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1888 [< none >] snd_timer_user_ioctl+0x7cd/0x2490 sound/core/timer.c:1918 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 INFO: Slab 0xffffea0000bafd00 objects=22 used=11 fp=0xffff88002ebf6d80 flags=0x1fffc0000004080 INFO: Object 0xffff88002ebf6d80 @offset=11648 fp=0xffff88002ebf5110 CPU: 3 PID: 7684 Comm: syz-executor Tainted: G B 4.5.0-rc7+ #337 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87b7d480 ffff88006d707bb0 ffffffff82c2125f ffffffff00bafd00 fffffbfff0f6fa90 ffff88003e807000 ffff88002ebf6d80 ffff88002ebf4000 ffffea0000bafd00 ffff88002ebf6e18 ffff88006d707be0 ffffffff8176dcc4 Call Trace: [<ffffffff81777e4e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:295 [<ffffffff82c88f63>] __list_del_entry+0x1d3/0x1e0 lib/list_debug.c:48 [< inline >] list_del_init include/linux/list.h:145 [<ffffffff852942f9>] snd_timer_interrupt+0x5b9/0xc80 sound/core/timer.c:791 [<ffffffff8529acc9>] snd_hrtimer_callback+0x169/0x230 sound/core/hrtimer.c:54 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1248 [<ffffffff814c3911>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1312 [<ffffffff814c62e2>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1346 [<ffffffff81255ec2>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff812593e9>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff866d256c>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:520 [< inline >] spin_unlock_irqrestore include/linux/spinlock.h:362 [<ffffffff8177369e>] __slab_free+0x1ae/0x320 mm/slub.c:2681 [< inline >] slab_free mm/slub.c:2829 [<ffffffff81773ed8>] kmem_cache_free+0x318/0x440 mm/slub.c:2838 [<ffffffff81704cb8>] ptlock_free+0x38/0x50 mm/memory.c:3912 [< inline >] pgtable_pmd_page_dtor include/linux/mm.h:1702 [<ffffffff812951ca>] ___pmd_free_tlb+0xaa/0x110 arch/x86/mm/pgtable.c:74 [< inline >] __pmd_free_tlb ./arch/x86/include/asm/pgalloc.h:106 [< inline >] free_pmd_range mm/memory.c:432 [< inline >] free_pud_range mm/memory.c:450 [<ffffffff816f4dc3>] free_pgd_range+0x973/0xbe0 mm/memory.c:526 [<ffffffff816f52f5>] free_pgtables+0x2c5/0x3b0 mm/memory.c:558 [<ffffffff817137c3>] exit_mmap+0x233/0x410 mm/mmap.c:2868 [<ffffffff813543f5>] mmput+0x95/0x230 kernel/fork.c:706 [< inline >] exit_mm kernel/exit.c:436 [<ffffffff81366eb2>] do_exit+0x7b2/0x2d00 kernel/exit.c:735 [<ffffffff81369578>] do_group_exit+0x108/0x330 kernel/exit.c:878 [<ffffffff8138cf14>] get_signal+0x634/0x15e0 kernel/signal.c:2307 [<ffffffff811a7db3>] do_signal+0x83/0x1c90 arch/x86/kernel/signal.c:712 [<ffffffff81006685>] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:247 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:282 [<ffffffff8100866a>] syscall_return_slowpath+0x2ba/0x340 arch/x86/entry/common.c:344 [<ffffffff866d18e2>] int_ret_from_sys_call+0x25/0x9f arch/x86/entry/entry_64.S:281 ================================================================== kasan: GPF could be caused by NULL-ptr deref or user memory accessgeneral protection fault: 0000 [#1] SMP DEBUG_PAGEALLOC KASAN Modules linked in: CPU: 3 PID: 7684 Comm: syz-executor Tainted: G B 4.5.0-rc7+ #337 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff880061062f80 ti: ffff88005ee10000 task.ti: ffff88005ee10000 RIP: 0010:[<ffffffff82c88e16>] [<ffffffff82c88e16>] __list_del_entry+0x86/0x1e0 RSP: 0018:ffff88006d707cd0 EFLAGS: 00010046 RAX: dffffc0000000000 RBX: 0000000000000000 RCX: ffff88002ebf6e18 RDX: 0000000000000000 RSI: 0000000000000001 RDI: 0000000000000000 RBP: ffff88006d707cf0 R08: ffffffff89cbcf00 R09: 0000000000000000 R10: ffffed000dae0f8c R11: 0000000000000000 R12: ffff88005ee0d120 R13: ffff8800670a2298 R14: ffff88005ee0d110 R15: ffff88002ebf6e18 FS: 0000000000000000(0000) GS:ffff88006d700000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 00000000006e0000 CR3: 0000000007ae9000 CR4: 00000000000006e0 Stack: ffff88006d707cf0 ffff88002ebf6e18 dffffc0000000000 ffff88005ee0d120 ffff88006d707db0 ffffffff852942f9 ffff8800670a2368 0000000000000082 dffffc0000000000 ffff8800670a234c ffffffff87b7d480 ffffed000ce1446d Call Trace: <IRQ> [< inline >] list_del_init include/linux/list.h:145 [<ffffffff852942f9>] snd_timer_interrupt+0x5b9/0xc80 sound/core/timer.c:791 [<ffffffff8529acc9>] snd_hrtimer_callback+0x169/0x230 sound/core/hrtimer.c:54 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1248 [<ffffffff814c3911>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1312 [<ffffffff814c62e2>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1346 [<ffffffff81255ec2>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff812593e9>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff866d256c>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:520 <EOI> [< inline >] spin_unlock_irqrestore include/linux/spinlock.h:362 [<ffffffff8177369e>] __slab_free+0x1ae/0x320 mm/slub.c:2681 [< inline >] slab_free mm/slub.c:2829 [<ffffffff81773ed8>] kmem_cache_free+0x318/0x440 mm/slub.c:2838 [<ffffffff81704cb8>] ptlock_free+0x38/0x50 mm/memory.c:3912 [< inline >] pgtable_pmd_page_dtor include/linux/mm.h:1702 [<ffffffff812951ca>] ___pmd_free_tlb+0xaa/0x110 arch/x86/mm/pgtable.c:74 [< inline >] __pmd_free_tlb ./arch/x86/include/asm/pgalloc.h:106 [< inline >] free_pmd_range mm/memory.c:432 [< inline >] free_pud_range mm/memory.c:450 [<ffffffff816f4dc3>] free_pgd_range+0x973/0xbe0 mm/memory.c:526 [<ffffffff816f52f5>] free_pgtables+0x2c5/0x3b0 mm/memory.c:558 [<ffffffff817137c3>] exit_mmap+0x233/0x410 mm/mmap.c:2868 [<ffffffff813543f5>] mmput+0x95/0x230 kernel/fork.c:706 [< inline >] exit_mm kernel/exit.c:436 [<ffffffff81366eb2>] do_exit+0x7b2/0x2d00 kernel/exit.c:735 [<ffffffff81369578>] do_group_exit+0x108/0x330 kernel/exit.c:878 [<ffffffff8138cf14>] get_signal+0x634/0x15e0 kernel/signal.c:2307 [<ffffffff811a7db3>] do_signal+0x83/0x1c90 arch/x86/kernel/signal.c:712 [<ffffffff81006685>] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:247 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:282 [<ffffffff8100866a>] syscall_return_slowpath+0x2ba/0x340 arch/x86/entry/common.c:344 [<ffffffff866d18e2>] int_ret_from_sys_call+0x25/0x9f arch/x86/entry/entry_64.S:281 Code: c0 0f 84 91 00 00 00 48 b8 00 02 00 00 00 00 ad de 48 39 c3 0f 84 9f 00 00 00 48 89 da 48 b8 00 00 00 00 00 fc ff df 48 c1 ea 03 <80> 3c 02 00 0f 85 f6 00 00 00 4c 8b 23 4c 39 e1 0f 85 95 00 00 RIP [<ffffffff82c88e16>] __list_del_entry+0x86/0x1e0 lib/list_debug.c:57 RSP <ffff88006d707cd0> ---[ end trace fd16e1eaa1720656 ]--- Kernel panic - not syncing: Fatal exception in interrupt Shutting down cpus with NMI Kernel Offset: disabled ---[ end Kernel panic - not syncing: Fatal exception in interrupt It is not easily reproducible. I've hit several times while running fuzzer for a week. Here is one of the logs for the record: https://gist.githubusercontent.com/dvyukov/c84798ee55721563ecb537c4d51dc9f5…

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[alsa-devel] linux-4.6-rc4/sound/pci/ens1370.c:1551: possible bad expression ?
by David Binderman 21 Apr '16

21 Apr '16

Hello there, [linux-4.6-rc4/sound/pci/ens1370.c:1551]: (style) Expression '(X & 0xf0000)>= 0x4' is always false. Source code is if ((ensoniq->ctrl & ES_1371_GPIO_OUTM)>= 4) val = 1; Regards David Binderman

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[alsa-devel] [PATCH 0/4] ASoC: topology: configure more on DAI & DAI links
by mengdong.lin＠linux.intel.com 21 Apr '16

21 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. Two patches are small changes in ASoC core, to allow topology to further configure DAI and DAI links. In the other two patches, topology will configure more on FE DAI links. Guneshwor Singh (1): ASoC: topology: Set CPU DAI name and enable DPCM by default for FE link Mengdong Lin (3): ASoC: Export snd_soc_find_dai() ASoC: Change DAI link's be_id to a generic id ASoC: topology: Set the link ID when creating a FE DAI link include/sound/soc.h | 5 ++++- include/uapi/sound/asoc.h | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 12 ++++++------ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 12 ++++++------ sound/soc/intel/boards/skl_rt286.c | 10 +++++----- sound/soc/soc-core.c | 3 ++- sound/soc/soc-topology.c | 11 +++++++++++ 14 files changed, 42 insertions(+), 27 deletions(-) -- 2.5.0

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Re: [alsa-devel] Sound: BUG: KASAN: use-after-free in kill_fasync
by Baozeng Ding 21 Apr '16

21 Apr '16

On 2016/4/6 19:37, Baozeng Ding wrote: > > > On 2016/4/5 22:18, Takashi Iwai wrote: >> On Tue, 05 Apr 2016 15:51:30 +0200, >> Baozeng Ding wrote: >>> 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. >> Well, fasync doesn't need a cleanup usually. Do you have a proper >> reproducer code? > Sorry, I cannot reproduce it. I will apply your patch and test it. I > will tell you the result whether it is okay a few days later. > Thanks. > >> In anyway, try the patch below. I'm not sure whether this would help, >> but if the path is via kill_fasync(), it'd be good to put in the >> protected context. >> During the last few weeks, I have not seen the bug any more after applying your patch. Thanks. >> >> Takashi >> >> --- >> diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c >> index 3a9b66c6e09c..0aca39762ed0 100644 >> --- a/sound/core/pcm_lib.c >> +++ b/sound/core/pcm_lib.c >> @@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct >> snd_pcm_substream *substream) >> snd_timer_interrupt(substream->timer, 1); >> #endif >> _end: >> - snd_pcm_stream_unlock_irqrestore(substream, flags); >> kill_fasync(&runtime->fasync, SIGIO, POLL_IN); >> + snd_pcm_stream_unlock_irqrestore(substream, flags); >> } >> EXPORT_SYMBOL(snd_pcm_period_elapsed); >

2 1

[alsa-devel] [RFC PATCH v2 0/5] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 21 Apr '16

21 Apr '16

Refactor and go v2 because of the 60KB limitation This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. The interface to connect with BT chip is not finished yet, but as an upstream rookie I really need some comments. Any comments will be greatly appreciated. Garlic Tseng (5): ASoC: mediatek: Refine mt8173 driver ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add clock and irq control for 2701 platform driver ASoC: mediatek: add mt2701 platform driver implementation. ASoC: mediatek: Add mt2701-cs42448 driver and config option. .../devicetree/bindings/sound/mt2701-cs42448.txt | 39 + .../devicetree/bindings/sound/mtk2701-afe-pcm.txt | 150 ++ sound/soc/mediatek/Kconfig | 21 + sound/soc/mediatek/Makefile | 15 +- sound/soc/mediatek/mt2701/Makefile | 20 + sound/soc/mediatek/mt2701/mt2701-cs42448.c | 376 +++++ sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.c | 259 +++ sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.h | 28 + sound/soc/mediatek/mt2701/mtk2701-afe-common.h | 217 +++ sound/soc/mediatek/mt2701/mtk2701-afe-pcm.c | 1752 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mtk2701-irq.c | 109 ++ sound/soc/mediatek/mt2701/mtk2701-irq.h | 30 + sound/soc/mediatek/mt2701/mtk2701-reg.h | 165 ++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/{ => mt8173}/mtk-afe-common.h | 0 sound/soc/mediatek/{ => mt8173}/mtk-afe-pcm.c | 0 20 files changed, 3186 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 Documentation/devicetree/bindings/sound/mtk2701-afe-pcm.txt create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-irq.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-irq.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) rename sound/soc/mediatek/{ => mt8173}/mtk-afe-common.h (100%) rename sound/soc/mediatek/{ => mt8173}/mtk-afe-pcm.c (100%) -- 1.9.1

4 11

[alsa-devel] [PATCH v7] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 20 Apr '16

20 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 v6: Use subnode to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 ++++ sound/soc/mediatek/mt8173-rt5650.c | 49 ++++++++++++++++++-- 2 files changed, 56 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..5bfa6b6 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,21 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + codec-capture { + sound-dai = <&rt5650 1>; + }; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..b8ed965 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,28 @@ 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_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "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 +126,11 @@ 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 +170,7 @@ 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 +198,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +222,26 @@ 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 (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ASoC: ak4642: Remove CLK_IS_ROOT
by Stephen Boyd 20 Apr '16

20 Apr '16

This flag is a no-op now (see commit 47b0eeb3dc8a "clk: Deprecate CLK_IS_ROOT", 2016-02-02) so remove it. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> --- sound/soc/codecs/ak4642.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index cda27c22812a..1ee8506c06c7 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -608,9 +608,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev) of_property_read_string(np, "clock-output-names", &clk_name); - clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, - (parent_clk_name) ? 0 : CLK_IS_ROOT, - rate); + clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate); if (!IS_ERR(clk)) of_clk_add_provider(np, of_clk_src_simple_get, clk); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Remove CLK_IS_ROOT
by Stephen Boyd 20 Apr '16

20 Apr '16

This flag is a no-op now (see commit 47b0eeb3dc8a "clk: Deprecate CLK_IS_ROOT", 2016-02-02) so remove it. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> --- sound/soc/sh/rcar/adg.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 606399de684d..49354d17ea55 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, req_rate); + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); @@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, else { for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->onecell.clks = adg->clkout; -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

3 2

[alsa-devel] [PATCH 0/4] ASoC: add graph base connection on simple-card
by Kuninori Morimoto 20 Apr '16

20 Apr '16

Hi Mark Current simple-card is using "sound-dai" base connection on DT, but V4L2 is using graph base connection. For example HDMI case, we would like to use both connection. To above confusable connection method, and to reuse current resource, this patch adds new function, and detect both "sound-dai" and "remote-endpoint" on simple-card. like this sound { compatible = "simple-audio-card"; ... sndcpu: simple-audio-card,cpu { sound-dai = <&xxx>; }; sndcodec: simple-audio-card,codec { => remote-endpoint = <&out_hdmi>; }; }; xxx { ... ports { ... port@0 { reg = <0>; out_rgb: endpoint { }; }; port@1 { reg = <1>; => out_hdmi: endpoint { }; }; ... }; }; Kuninori Morimoto (4): of_graph: add of_graph_get_endpoint_count() ASoC: add of_parse_snd_connection_with_args() for sound-dai/graph connection ASoC: snd_soc_of_get_dai_name() uses of_parse_snd_soc_connection_with_args() ASoC: simple-card: probe both sound-dai and remote-endpoint Documentation/devicetree/bindings/sound/simple-card.txt | 19 +++++++++++++++++++ include/linux/of_graph.h | 11 +++++++++++ include/sound/soc.h | 2 ++ sound/soc/generic/simple-card.c | 9 ++++++--- sound/soc/soc-core.c | 45 +++++++++++++++++++++++++++++++++++++++++++-- 5 files changed, 81 insertions(+), 5 deletions(-)

3 13

Re: [alsa-devel] Issues with ALSA:ice1724 sound cards on powerpc machines
by Julian Margetson 20 Apr '16

20 Apr '16

On 2/17/2016 2:29 PM, Julian Margetson wrote: > Hi > > Having an issue issue with ice1724 sound cards on powerpc motherboards. > Motherboards in question Acube Sam460ex . > The card appears as a valid device but there is no sound output on the > majority of boots of the computer. > Logs attached. > On the first set boot audio playback produced sound ok. > On the second set boot no sound was produce with audio playback > > > Regards > Julian Margetson > > > Anyone have any ideas ? Resubmitting again with links to logs instead of included as attachments. The issue also seen on A-Eon Cyrus P5020 motherboard . https://www.dropbox.com/s/72labcje9z1r1b7/1alsa-info.txt.7CLHMG0Gap?dl=0 https://www.dropbox.com/s/v1pn5kjqrvwu16f/boot_1.log?dl=0 https://www.dropbox.com/s/uuaa14rkpmmz1o0/2alsa-info.txt.VKiEVuytiJ?dl=0 https://www.dropbox.com/s/wwu2f5t93l8g26d/boot_2.log?dl=0 Thanks

3 7

[alsa-devel] GPIO control from ALSA
by Petr Kulhavy 20 Apr '16

20 Apr '16

Hello, I'm writing a driver for ASoC "sound card" hardware which uses CPU GPIO to mute/unmute an analogue amplifier and another GPIO to control an analogue switch for line/mic switching. Is there an abstraction in ALSA to model this kind of hardware? Thanks Petr

2 5

[alsa-devel] [PATCH 1/1] ALSA: hda - add PCI ID for Intel Broxton-T
by han.lu＠intel.com 20 Apr '16

20 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Add HD Audio Device PCI ID for the Intel Broxton-T platform. It is an HDA Intel PCH controller. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b680be0..637b8a0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2232,6 +2232,9 @@ static const struct pci_device_id azx_ids[] = { /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, + /* Broxton-T */ + { PCI_DEVICE(0x8086, 0x1a98), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, -- 2.5.0

2 1

[alsa-devel] [asoc:topic/mtk 1/1] sound/soc/mediatek/mt8173-rt5650.c:236:3: error: implicit declaration of function 'snd_soc_get_dai_name'
by kbuild test robot 20 Apr '16

20 Apr '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/mtk head: 0beeb8d0b434b8b007ac8058fb2cac5997fff6df commit: 0beeb8d0b434b8b007ac8058fb2cac5997fff6df [1/1] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver config: arm-allmodconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 0beeb8d0b434b8b007ac8058fb2cac5997fff6df # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc/mediatek/mt8173-rt5650.c: In function 'mt8173_rt5650_dev_probe': >> sound/soc/mediatek/mt8173-rt5650.c:236:3: error: implicit declaration of function 'snd_soc_get_dai_name' [-Werror=implicit-function-declaration] ret = snd_soc_get_dai_name(&args, &codec_dai_capture); ^ cc1: some warnings being treated as errors vim +/snd_soc_get_dai_name +236 sound/soc/mediatek/mt8173-rt5650.c 230 dev_err(&pdev->dev, 231 "%s capture-dai name fail %d\n", 232 __func__, ret); 233 return ret; 234 } 235 > 236 ret = snd_soc_get_dai_name(&args, &codec_dai_capture); 237 mt8173_rt5650_codecs[1].dai_name = codec_dai_capture; 238 } 239 --- 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 - Keep powering up ADCs on Cirrus codecs
by Takashi Iwai 19 Apr '16

19 Apr '16

Although one weird behavior about the input path (inconsistent D0/D3 switch) on Cirrus CS420x codecs was fixed in the previous commit, there is still an issue on some Mac machines: the capture stream stalls when switching the ADCs on the fly. More badly, this keeps stuck until the next reboot. The dynamic ADC switching is already a bit fragile and assuming optimistically that the chip accepts the frequent power changes. On Cirrus codecs, this doesn't seem applicable. As a quick workaround, we pin down the ADCs to keep up in D0 when spec->dyn_adc_switch is set. In this way, the ADCs are kept up only for the system that were confirmed to be broken. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116171 Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a47e8ae0eb30..80bbadc83721 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -361,6 +361,7 @@ static int cs_parse_auto_config(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; int err; + int i; err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); if (err < 0) @@ -370,6 +371,19 @@ static int cs_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + /* keep the ADCs powered up when it's dynamically switchable */ + if (spec->gen.dyn_adc_switch) { + unsigned int done = 0; + for (i = 0; i < spec->gen.input_mux.num_items; i++) { + int idx = spec->gen.dyn_adc_idx[i]; + if (done & (1 << idx)) + continue; + snd_hda_gen_fix_pin_power(codec, + spec->gen.adc_nids[idx]); + done |= 1 << idx; + } + } + return 0; } -- 2.8.1

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Add 96k sample rate
by Jeremy McDermond 19 Apr '16

19 Apr '16

The TLV320AIC32x4 series supports 96ksps rates in hardware. This patch adds the necessary PLL divider values and clock settings to the table to make 96ksps work. Signed-off-by: Jeremy McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c6b6d55..2f8480c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -159,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 48k rate */ {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, - {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}, + + /* 96k rate */ + {AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1}, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -564,7 +567,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.5.0

2 1

[alsa-devel] [PATCH v2] ASoC: sti: update dt-bindings for tdm mode
by Moise Gergaud 19 Apr '16

19 Apr '16

- Add "TDM" in the st,mode property list - st,mode property is also mandatory for reader - add tdm playback dai-link example Signed-off-by: Moise Gergaud <moise.gergaud(a)st.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- I forgot to include this patch into "[PATCH v2] ASoC: sti: unip tdm mode" serie that has been applied. history: - V2: rename patch with a subject line reflecting the style for the subsystem .../devicetree/bindings/sound/st,sti-asoc-card.txt | 44 +++++++++++++++++++--- 1 file changed, 39 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index f546dd9..4d9a83d 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -37,17 +37,18 @@ Required properties: - dai-name: DAI name that describes the IP. + - IP mode: IP working mode depending on associated codec. + "HDMI" connected to HDMI codec and support IEC HDMI formats (player only). + "SPDIF" connected to SPDIF codec and support SPDIF formats (player only). + "PCM" PCM standard mode for I2S or TDM bus. + "TDM" TDM mode for TDM bus. + Required properties ("st,sti-uni-player" compatibility only): - clocks: CPU_DAI IP clock source, listed in the same order than the CPU_DAI properties. - uniperiph-id: internal SOC IP instance ID. - - IP mode: IP working mode depending on associated codec. - "HDMI" connected to HDMI codec IP and IEC HDMI formats. - "SPDIF"connected to SPDIF codec and support SPDIF formats. - "PCM" PCM standard mode for I2S or TDM bus. - Optional properties: - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for external codecs connection. @@ -56,6 +57,22 @@ Optional properties: Example: + sti_uni_player1: sti-uni-player@1 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + st,dai-name = "Uni Player #1 (I2S)"; + dma-names = "tx"; + st,uniperiph-id = <1>; + st,version = <5>; + st,mode = "TDM"; + }; + sti_uni_player2: sti-uni-player@2 { compatible = "st,sti-uni-player"; status = "okay"; @@ -99,6 +116,7 @@ Example: dma-names = "rx"; dai-name = "Uni Reader #1 (HDMI RX)"; version = <3>; + st,mode = "PCM"; }; 2) sti-sas-codec: internal audio codec IPs driver @@ -152,4 +170,20 @@ Example of audio card declaration: sound-dai = <&sti_sasg_codec 0>; }; }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; }; -- 1.9.1

2 1

[alsa-devel] [PATCH v4 0/4] Propagate errors out from compressed streams
by Charles Keepax 19 Apr '16

19 Apr '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v3: - Add missing comment for new field in snd_compr_stream - Return -EPIPE on XRUN to match what the PCM framework does It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 76 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 ++++-- sound/soc/soc-compress.c | 5 +-- 4 files changed, 88 insertions(+), 9 deletions(-) -- 2.1.4

2 10

[alsa-devel] [PATCH] ASoC: core: export snd_soc_get_dai_name
by PC Liao 19 Apr '16

19 Apr '16

From: Jean-Francois Moine <moinejf(a)free.fr> snd_soc_get_dai_name() may be used to define a sound card with a different syntax from the one of the simple-card. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- [1] is applied and needs this patch. Otherwise, [1] will build fail. [1] https://patchwork.kernel.org/patch/8878491/ ("ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 5 +++-- 2 files changed, 5 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a21..03b2d19 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1669,6 +1669,8 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, struct device_node **framemaster); +int snd_soc_get_dai_name(struct of_phandle_args *args, + const char **dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name); int snd_soc_of_get_dai_link_codecs(struct device *dev, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..69cbbb0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3786,8 +3786,8 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); -static int snd_soc_get_dai_name(struct of_phandle_args *args, - const char **dai_name) +int snd_soc_get_dai_name(struct of_phandle_args *args, + const char **dai_name) { struct snd_soc_component *pos; struct device_node *component_of_node; @@ -3838,6 +3838,7 @@ static int snd_soc_get_dai_name(struct of_phandle_args *args, mutex_unlock(&client_mutex); return ret; } +EXPORT_SYMBOL_GPL(snd_soc_get_dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name) -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/2] ASoC: da7219: Small clocking related updates for PLL and sysclk
by Adam Thomson 19 Apr '16

19 Apr '16

The patch set contains 2 small fixes/updates relating to clocking in the codec. The first improves PLL locking for a corner scenario with a low frequency MCLK provided, and the second is to remove 32KHz support from the codec set_dai_sysclk() function, to align with updates already made to the set_dai_pll() function. Patches are based on v4.6-rc4 kernel version. Adam Thomson (2): ASoC: da7219: Update PLL ranges and dividers to improve locking ASoC: da7219: Disallow unsupported 32KHz clock setting in set_dai_sysclk() sound/soc/codecs/da7219.c | 30 +++++++++++++++--------------- sound/soc/codecs/da7219.h | 20 ++++++++++---------- 2 files changed, 25 insertions(+), 25 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH v9 0/7] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 19 Apr '16

19 Apr '16

Couple of minor changes since v8 to ALSA/ASoC side patches. The series is based on top of drm-next. However, it does not look like there is enough attention to the DRM side so that the tda998x patches would get in any time soon. So I am happy to rebase the first ALSA-side patches on top of Mark Brown's for-next tree, for instance, if they would have a change to go in on their own. The first two IEC958 helper patches should apply as such. There is currently two other patch series[2][3] that depend on the first three (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 patches 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 v8 * "ALSA: pcm: add IEC958 channel status helper for hw_params" - Split out the 32 bit support - add: "ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper" * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Set hw_params->msbits to 24 in hw_params() callback if sample width is greater than 24 bits 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 (7): ALSA: pcm: add IEC958 channel status helper for hw_params ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper 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 | 414 +++++++++++++++++++++ 12 files changed, 917 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

6 21

[alsa-devel] [PATCH v5] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 19 Apr '16

19 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. This patch depends on [1], because snd_soc_of_get_dai_name() fix the property name "sound-dai". This patch needs to change name to use, so export snd_soc_get_dai_name() which is included in snd_soc_of_get_dai_name(). [1] https://patchwork.kernel.org/patch/5671961/ ("ASoC: core: export snd_soc_get_dai_name") Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v4: Change name from "sound-dai" to "capture-dai" to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 44 ++++++++++++++++++-- 2 files changed, 47 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..802dd1b 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: +- capture-dai: audio codec dai on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + capture-dai = <&rt5650 1>; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..45dea22 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_capture = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_dai_capture, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_dai_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 +196,8 @@ 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_capture; + struct of_phandle_args args; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +220,23 @@ 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, "capture-dai")) { + ret = of_parse_phandle_with_args(pdev->dev.of_node, + "capture-dai", + "#sound-dai-cells", 0, &args); + if (ret < 0) { + dev_err(&pdev->dev, + "%s capture-dai name fail %d\n", + __func__, ret); + return ret; + } + + ret = snd_soc_get_dai_name(&args, &codec_dai_capture); + mt8173_rt5650_codecs[1].dai_name = codec_dai_capture; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

2 4

[alsa-devel] [PATCH v3] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 19 Apr '16

19 Apr '16

Provide *our* view of what the rules are for the different DAI formats, so that we do not have to trust external interpretations for this crucial bit of interoperability. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- Documentation/sound/alsa/soc/clocking.txt | 146 +++++++++++++++++++++++++++++- 1 file changed, 144 insertions(+), 2 deletions(-) Changed PCM mode A to start left word one BCLK after positive LRC edge. Added link to I2S spec as indicated by Mark Brown. Cheers, Peter diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index b1300162e01c..7eb02f0758e7 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -47,5 +47,147 @@ rate, number of channels and word size) to save on power. It is also desirable to use the codec (if possible) to drive (or master) the audio clocks as it usually gives more accurate sample rates than the CPU. - - +The below diagrams all have BCLK as the first signal, LRC as the second signal +and DATA as the third. Below that is an indication about which DATA bits belong +in what channel. + +A "..." marking as DATA indicates that there may be more bits that are not +shown. Also, all DATA bits marked X may or may not be present. DAI +transmitters must add them should LRC not match the word size exactly and DAI +receivers must be prepared to ignore them. DAI transmitters must insert zeros +for I2S and Left Justified modes, and preferably not drive DATA for the DSP +modes for these extra X bits. + + +I2S + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word starts one BCLK cycle after a negative flank of LRC, and +the right channel word starts one BCLK cycle after a positive flank of LRC. The +words start with the MSB. Receivers must truncate words if more bits per word +are transmitted than they can use, and transmitters must pad words with zeros +if LRC is not matching the word size. Also, see +https://www.sparkfun.com/datasheets/BreakoutBoards/I2SBUS.pdf + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified (aka MSB) + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word starts directly after a positive flank of LRC, and the +right channel word starts directly after a negative flank of LRC. The words +start with the MSB. Receivers must truncate words if more bits per word are +transmitted than they can use, and transmitters must pad words with zeros if +LRC is not matching the word size. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Right Justified (aka LSB) + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word end right before a negative flank of LRC, and the right +channel word ends right before a positive flank of LRC. The words end with LSB. +Transmitters must pad words if LRC is not matching the word size and the +transmitter and the receiver have to agree on the word leghth. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | X |...| X |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +DSP mode A + +LRC should have positive flanks synchronized with a negative flank of BCLK. +The left channel word starts one BCLK after a positive flank of LRC, and the +right channel word starts directly after the left channel word. The words +start with MSB. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | | + +LRC examples for DSP mode A + .---. .---. +---' '-------------------------------------------------------' '----- + .-. .-. +---' '---------------------------------------------------------' '------- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +DSP mode B + +LRC should have positive flanks synchronized with a negative flank of BCLK. +The left channel word starts directly after a positive flank of LRC, and the +right channel word starts directly after the left channel word. The words +start with MSB. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | | + +LRC examples for DSP mode B + .---. .---. +---' '-------------------------------------------------------' '----- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +The above diagrams show normal BCLK and LRC clocking where DAI transmitters +change DATA on the falling edge of BCLK and DAI receivers read DATA on the +rising edge. For inverted BCLK it is naturally the other way around. Inverted +LRC is just that. Note that DSP mode A is compatible with DSP mode B with +inverted LRC, and vice versa. + + +Left Justified, inverted BCLK +-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .- + '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified, inverted LRC + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | -- 2.1.4

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[alsa-devel] [PATCH] Documentation: sti-asoc-card: update tdm mode
by Moise Gergaud 19 Apr '16

19 Apr '16

- Add "TDM" in the st,mode property list - st,mode property is also mandatory for reader - add tdm playback dai-link example Signed-off-by: Moise Gergaud <moise.gergaud(a)st.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- I forgot to include this patch into "[PATCH v2] ASoC: sti: unip tdm mode" serie that has been applied. This patch is the same than the one previously sent in http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106392.html .../devicetree/bindings/sound/st,sti-asoc-card.txt | 44 +++++++++++++++++++--- 1 file changed, 39 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index f546dd9..4d9a83d 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -37,17 +37,18 @@ Required properties: - dai-name: DAI name that describes the IP. + - IP mode: IP working mode depending on associated codec. + "HDMI" connected to HDMI codec and support IEC HDMI formats (player only). + "SPDIF" connected to SPDIF codec and support SPDIF formats (player only). + "PCM" PCM standard mode for I2S or TDM bus. + "TDM" TDM mode for TDM bus. + Required properties ("st,sti-uni-player" compatibility only): - clocks: CPU_DAI IP clock source, listed in the same order than the CPU_DAI properties. - uniperiph-id: internal SOC IP instance ID. - - IP mode: IP working mode depending on associated codec. - "HDMI" connected to HDMI codec IP and IEC HDMI formats. - "SPDIF"connected to SPDIF codec and support SPDIF formats. - "PCM" PCM standard mode for I2S or TDM bus. - Optional properties: - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for external codecs connection. @@ -56,6 +57,22 @@ Optional properties: Example: + sti_uni_player1: sti-uni-player@1 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + st,dai-name = "Uni Player #1 (I2S)"; + dma-names = "tx"; + st,uniperiph-id = <1>; + st,version = <5>; + st,mode = "TDM"; + }; + sti_uni_player2: sti-uni-player@2 { compatible = "st,sti-uni-player"; status = "okay"; @@ -99,6 +116,7 @@ Example: dma-names = "rx"; dai-name = "Uni Reader #1 (HDMI RX)"; version = <3>; + st,mode = "PCM"; }; 2) sti-sas-codec: internal audio codec IPs driver @@ -152,4 +170,20 @@ Example of audio card declaration: sound-dai = <&sti_sasg_codec 0>; }; }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; }; -- 1.9.1

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