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[alsa-devel] [PATCH] ALSA: hda - fix the Headset Mic Jack name for Realtek
by Jaroslav Kysela 06 Nov '19

06 Nov '19

The HDA auto parser creates kcontrol 'Headset Mic Jack'. Use this name also for the alc_fixup_headset_jack() function. It is necessary to have common name for the user space like the UCM configs: https://github.com/alsa-project/alsa-ucm-conf/pull/5 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 80f66ba85f87..2633e45144d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5679,7 +5679,7 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, case HDA_FIXUP_ACT_PRE_PROBE: snd_hda_jack_detect_enable_callback(codec, 0x55, alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + snd_hda_jack_add_kctl(codec, 0x55, "Headset Mic Jack", false, SND_JACK_HEADSET, alc_headset_btn_keymap); break; case HDA_FIXUP_ACT_INIT: -- 2.20.1

2 2

[alsa-devel] INFO: task hung in snd_timer_close
by syzbot 06 Nov '19

06 Nov '19

Hello, syzbot found the following crash on: HEAD commit: a99d8080 Linux 5.4-rc6 git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=10afae68e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=8c5e2eca3f31f9bf dashboard link: https://syzkaller.appspot.com/bug?extid=f1048ebddb93befb085f compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=126bc658e00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f1048ebddb93befb085f(a)syzkaller.appspotmail.com INFO: task syz-executor.0:12251 blocked for more than 143 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.0 D28160 12251 8868 0x00000004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_close+0x7e/0x100 sound/core/timer.c:422 snd_timer_user_release+0x98/0x130 sound/core/timer.c:1478 __fput+0x2ff/0x890 fs/file_table.c:280 ____fput+0x16/0x20 fs/file_table.c:313 task_work_run+0x145/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_usermode_loop+0x316/0x380 arch/x86/entry/common.c:163 prepare_exit_to_usermode arch/x86/entry/common.c:194 [inline] syscall_return_slowpath arch/x86/entry/common.c:274 [inline] do_syscall_64+0x65f/0x760 arch/x86/entry/common.c:300 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x413db1 Code: c0 c3 0f 1f 80 00 00 00 00 c7 07 00 00 00 00 31 c0 c3 0f 1f 80 00 00 00 00 31 c0 c3 66 66 66 66 2e 0f 1f 84 00 00 00 00 00 b8 <2d> f8 ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 31 c0 c3 66 66 66 66 RSP: 002b:00007ffdda4a34b0 EFLAGS: 00000293 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000004 RCX: 0000000000413db1 RDX: 0000001b2dd20000 RSI: 0000000000000000 RDI: 0000000000000003 RBP: 0000000000000001 R08: ffffffffffffffff R09: ffffffffffffffff R10: 00007ffdda4a3590 R11: 0000000000000293 R12: 000000000075c9a0 R13: 000000000075c9a0 R14: 0000000000760a90 R15: 000000000075bf2c INFO: task syz-executor.2:12256 blocked for more than 143 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.2 D28160 12256 8864 0x00000004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_close+0x7e/0x100 sound/core/timer.c:422 snd_timer_user_release+0x98/0x130 sound/core/timer.c:1478 __fput+0x2ff/0x890 fs/file_table.c:280 ____fput+0x16/0x20 fs/file_table.c:313 task_work_run+0x145/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_usermode_loop+0x316/0x380 arch/x86/entry/common.c:163 prepare_exit_to_usermode arch/x86/entry/common.c:194 [inline] syscall_return_slowpath arch/x86/entry/common.c:274 [inline] do_syscall_64+0x65f/0x760 arch/x86/entry/common.c:300 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x413db1 Code: c0 c3 0f 1f 80 00 00 00 00 c7 07 00 00 00 00 31 c0 c3 0f 1f 80 00 00 00 00 31 c0 c3 66 66 66 66 2e 0f 1f 84 00 00 00 00 00 b8 <2d> f8 ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 31 c0 c3 66 66 66 66 RSP: 002b:00007ffe078ed7d0 EFLAGS: 00000293 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000004 RCX: 0000000000413db1 RDX: 0000001b2da20000 RSI: 0000000000000000 RDI: 0000000000000003 RBP: 0000000000000001 R08: ffffffffffffffff R09: ffffffffffffffff R10: 00007ffe078ed8b0 R11: 0000000000000293 R12: 000000000075c9a0 R13: 000000000075c9a0 R14: 0000000000760a90 R15: 000000000075bf2c INFO: task syz-executor.4:12266 blocked for more than 144 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.4 D28480 12266 8871 0x00000004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_open+0x98/0x1860 sound/core/timer.c:245 snd_timer_user_tselect sound/core/timer.c:1728 [inline] __snd_timer_user_ioctl.isra.0+0x68b/0x1fd0 sound/core/timer.c:2003 snd_timer_user_ioctl+0x7a/0xa7 sound/core/timer.c:2033 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0xdb6/0x13e0 fs/ioctl.c:696 ksys_ioctl+0xab/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0xfa/0x760 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x45a219 Code: Bad RIP value. RSP: 002b:00007f2806975c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 000000000045a219 RDX: 0000000020029fcc RSI: 0000000040345410 RDI: 0000000000000003 RBP: 000000000075bf20 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f28069766d4 R13: 00000000004cf428 R14: 00000000004d9760 R15: 00000000ffffffff INFO: task syz-executor.4:12269 blocked for more than 144 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.4 D29712 12269 8871 0x00000004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0xdb6/0x13e0 fs/ioctl.c:696 ksys_ioctl+0xab/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0xfa/0x760 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x45a219 Code: Bad RIP value. RSP: 002b:00007f2806954c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000002 RCX: 000000000045a219 RDX: 0000000000000000 RSI: 00000000000054a0 RDI: 0000000000000003 RBP: 000000000075bfc8 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f28069556d4 R13: 00000000004c4ae8 R14: 00000000004d9778 R15: 00000000ffffffff INFO: task syz-executor.4:12271 blocked for more than 144 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.4 D28000 12271 8871 0x00000004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_open+0x98/0x1860 sound/core/timer.c:245 snd_seq_timer_open+0x240/0x590 sound/core/seq/seq_timer.c:275 queue_use+0xf1/0x270 sound/core/seq/seq_queue.c:489 snd_seq_queue_alloc+0x2c5/0x4d0 sound/core/seq/seq_queue.c:176 snd_seq_ioctl_create_queue+0xb0/0x330 sound/core/seq/seq_clientmgr.c:1548 snd_seq_kernel_client_ctl+0xf8/0x140 sound/core/seq/seq_clientmgr.c:2353 alloc_seq_queue.isra.0+0xdc/0x180 sound/core/seq/oss/seq_oss_init.c:357 snd_seq_oss_open+0x2ff/0x960 sound/core/seq/oss/seq_oss_init.c:215 odev_open+0x70/0x90 sound/core/seq/oss/seq_oss.c:125 soundcore_open+0x453/0x610 sound/sound_core.c:593 chrdev_open+0x245/0x6b0 fs/char_dev.c:414 do_dentry_open+0x4e6/0x1380 fs/open.c:797 vfs_open+0xa0/0xd0 fs/open.c:914 do_last fs/namei.c:3408 [inline] path_openat+0x10e9/0x46d0 fs/namei.c:3525 do_filp_open+0x1a1/0x280 fs/namei.c:3555 do_sys_open+0x3fe/0x5d0 fs/open.c:1097 __do_sys_openat fs/open.c:1124 [inline] __se_sys_openat fs/open.c:1118 [inline] __x64_sys_openat+0x9d/0x100 fs/open.c:1118 do_syscall_64+0xfa/0x760 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x45a219 Code: Bad RIP value. RSP: 002b:00007f2806933c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 RAX: ffffffffffffffda RBX: 0000000000000004 RCX: 000000000045a219 RDX: 0000000000000001 RSI: 0000000020000000 RDI: ffffffffffffff9c RBP: 000000000075c070 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f28069346d4 R13: 00000000004c734a R14: 00000000004dcf88 R15: 00000000ffffffff INFO: task syz-executor.4:12273 blocked for more than 145 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.4 D28552 12273 8871 0x00000004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0xdb6/0x13e0 fs/ioctl.c:696 ksys_ioctl+0xab/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0xfa/0x760 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x45a219 Code: Bad RIP value. RSP: 002b:00007f2806912c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000002 RCX: 000000000045a219 RDX: 0000000000000000 RSI: 00000000000054a0 RDI: 0000000000000003 RBP: 000000000075c118 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f28069136d4 R13: 00000000004c4ae8 R14: 00000000004d9778 R15: 00000000ffffffff INFO: task syz-executor.4:12275 blocked for more than 145 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.4 D29312 12275 8871 0x00004004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_open+0x98/0x1860 sound/core/timer.c:245 snd_timer_user_tselect sound/core/timer.c:1728 [inline] __snd_timer_user_ioctl.isra.0+0x68b/0x1fd0 sound/core/timer.c:2003 snd_timer_user_ioctl+0x7a/0xa7 sound/core/timer.c:2033 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0xdb6/0x13e0 fs/ioctl.c:696 ksys_ioctl+0xab/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0xfa/0x760 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x45a219 Code: Bad RIP value. RSP: 002b:00007f28068f1c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 000000000045a219 RDX: 0000000020029fcc RSI: 0000000040345410 RDI: 0000000000000003 RBP: 000000000075c1c0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f28068f26d4 R13: 00000000004cf428 R14: 00000000004d9760 R15: 00000000ffffffff INFO: task syz-executor.4:12276 blocked for more than 146 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.4 D28536 12276 8871 0x00004004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 odev_open+0x5f/0x90 sound/core/seq/oss/seq_oss.c:124 soundcore_open+0x453/0x610 sound/sound_core.c:593 chrdev_open+0x245/0x6b0 fs/char_dev.c:414 do_dentry_open+0x4e6/0x1380 fs/open.c:797 vfs_open+0xa0/0xd0 fs/open.c:914 do_last fs/namei.c:3408 [inline] path_openat+0x10e9/0x46d0 fs/namei.c:3525 do_filp_open+0x1a1/0x280 fs/namei.c:3555 do_sys_open+0x3fe/0x5d0 fs/open.c:1097 __do_sys_openat fs/open.c:1124 [inline] __se_sys_openat fs/open.c:1118 [inline] __x64_sys_openat+0x9d/0x100 fs/open.c:1118 do_syscall_64+0xfa/0x760 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x45a219 Code: Bad RIP value. RSP: 002b:00007f28068d0c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 RAX: ffffffffffffffda RBX: 0000000000000004 RCX: 000000000045a219 RDX: 0000000000000001 RSI: 0000000020000000 RDI: ffffffffffffff9c RBP: 000000000075c268 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f28068d16d4 R13: 00000000004c734a R14: 00000000004dcf88 R15: 00000000ffffffff INFO: task syz-executor.1:12261 blocked for more than 146 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.1 D28160 12261 8866 0x00000004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_close+0x7e/0x100 sound/core/timer.c:422 snd_timer_user_release+0x98/0x130 sound/core/timer.c:1478 __fput+0x2ff/0x890 fs/file_table.c:280 ____fput+0x16/0x20 fs/file_table.c:313 task_work_run+0x145/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_usermode_loop+0x316/0x380 arch/x86/entry/common.c:163 prepare_exit_to_usermode arch/x86/entry/common.c:194 [inline] syscall_return_slowpath arch/x86/entry/common.c:274 [inline] do_syscall_64+0x65f/0x760 arch/x86/entry/common.c:300 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x413db1 Code: c0 c3 0f 1f 80 00 00 00 00 c7 07 00 00 00 00 31 c0 c3 0f 1f 80 00 00 00 00 31 c0 c3 66 66 66 66 2e 0f 1f 84 00 00 00 00 00 b8 <2d> f8 ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 31 c0 c3 66 66 66 66 RSP: 002b:00007ffe3503b380 EFLAGS: 00000293 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000004 RCX: 0000000000413db1 RDX: 0000001b2db20000 RSI: 0000000000000000 RDI: 0000000000000003 RBP: 0000000000000001 R08: ffffffffffffffff R09: ffffffffffffffff R10: 00007ffe3503b460 R11: 0000000000000293 R12: 000000000075c9a0 R13: 000000000075c9a0 R14: 0000000000760a90 R15: 000000000075bf2c INFO: task syz-executor.5:12267 blocked for more than 147 seconds. Not tainted 5.4.0-rc6 #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor.5 D28552 12267 8870 0x00000004 Call Trace: context_switch kernel/sched/core.c:3384 [inline] __schedule+0x94f/0x1e70 kernel/sched/core.c:4069 schedule+0xd9/0x260 kernel/sched/core.c:4136 schedule_preempt_disabled+0x13/0x20 kernel/sched/core.c:4195 __mutex_lock_common kernel/locking/mutex.c:1033 [inline] __mutex_lock+0x7b0/0x13c0 kernel/locking/mutex.c:1103 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1118 snd_timer_open+0x98/0x1860 sound/core/timer.c:245 snd_timer_user_tselect sound/core/timer.c:1728 [inline] __snd_timer_user_ioctl.isra.0+0x68b/0x1fd0 sound/core/timer.c:2003 snd_timer_user_ioctl+0x7a/0xa7 sound/core/timer.c:2033 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:509 [inline] do_vfs_ioctl+0xdb6/0x13e0 fs/ioctl.c:696 ksys_ioctl+0xab/0xd0 fs/ioctl.c:713 __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:718 do_syscall_64+0xfa/0x760 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x45a219 Code: Bad RIP value. RSP: 002b:00007ff76c742c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 000000000045a219 RDX: 0000000020029fcc RSI: 0000000040345410 RDI: 0000000000000003 RBP: 000000000075bf20 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007ff76c7436d4 R13: 00000000004cf428 R14: 00000000004d9760 R15: 00000000ffffffff Showing all locks held in the system: 1 lock held by khungtaskd/1070: #0: ffffffff88fab340 (rcu_read_lock){....}, at: debug_show_all_locks+0x5f/0x27e kernel/locking/lockdep.c:5337 1 lock held by rsyslogd/8724: #0: ffff88809ae4bd60 (&f->f_pos_lock){+.+.}, at: __fdget_pos+0xee/0x110 fs/file.c:801 2 locks held by getty/8813: #0: ffff8880a459b090 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x33/0x40 drivers/tty/tty_ldsem.c:340 #1: ffffc90005f152e0 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x232/0x1c10 drivers/tty/n_tty.c:2156 2 locks held by getty/8814: #0: ffff8880a9021090 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x33/0x40 drivers/tty/tty_ldsem.c:340 #1: ffffc90005f212e0 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x232/0x1c10 drivers/tty/n_tty.c:2156 2 locks held by getty/8815: #0: ffff8880a818a090 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x33/0x40 drivers/tty/tty_ldsem.c:340 #1: ffffc90005f192e0 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x232/0x1c10 drivers/tty/n_tty.c:2156 2 locks held by getty/8816: #0: ffff8880a843a090 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x33/0x40 drivers/tty/tty_ldsem.c:340 #1: ffffc90005f1d2e0 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x232/0x1c10 drivers/tty/n_tty.c:2156 2 locks held by getty/8817: #0: ffff8880a2834090 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x33/0x40 drivers/tty/tty_ldsem.c:340 #1: ffffc90005f252e0 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x232/0x1c10 drivers/tty/n_tty.c:2156 2 locks held by getty/8818: #0: ffff88809aafe090 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x33/0x40 drivers/tty/tty_ldsem.c:340 #1: ffffc90005f292e0 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x232/0x1c10 drivers/tty/n_tty.c:2156 2 locks held by getty/8819: #0: ffff8880a514a090 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x33/0x40 drivers/tty/tty_ldsem.c:340 #1: ffffc90005ef12e0 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x232/0x1c10 drivers/tty/n_tty.c:2156 2 locks held by syz-executor.3/12245: 2 locks held by syz-executor.0/12251: #0: ffff888082e84550 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_release+0x67/0x130 sound/core/timer.c:1476 #1: ffffffff89914b00 (register_mutex){+.+.}, at: snd_timer_close+0x7e/0x100 sound/core/timer.c:422 2 locks held by syz-executor.2/12256: #0: ffff88809c005150 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_release+0x67/0x130 sound/core/timer.c:1476 #1: ffffffff89914b00 (register_mutex){+.+.}, at: snd_timer_close+0x7e/0x100 sound/core/timer.c:422 2 locks held by syz-executor.4/12266: #0: ffff8880a9b93950 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 #1: ffffffff89914b00 (register_mutex){+.+.}, at: snd_timer_open+0x98/0x1860 sound/core/timer.c:245 1 lock held by syz-executor.4/12269: #0: ffff8880a9b93950 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 2 locks held by syz-executor.4/12271: #0: ffffffff89924320 (register_mutex#4){+.+.}, at: odev_open+0x5f/0x90 sound/core/seq/oss/seq_oss.c:124 #1: ffffffff89914b00 (register_mutex){+.+.}, at: snd_timer_open+0x98/0x1860 sound/core/timer.c:245 1 lock held by syz-executor.4/12273: #0: ffff888096c8d550 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 2 locks held by syz-executor.4/12275: #0: ffff888096c8d550 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 #1: ffffffff89914b00 (register_mutex){+.+.}, at: snd_timer_open+0x98/0x1860 sound/core/timer.c:245 1 lock held by syz-executor.4/12276: #0: ffffffff89924320 (register_mutex#4){+.+.}, at: odev_open+0x5f/0x90 sound/core/seq/oss/seq_oss.c:124 2 locks held by syz-executor.1/12261: #0: ffff88809a826950 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_release+0x67/0x130 sound/core/timer.c:1476 #1: ffffffff89914b00 (register_mutex){+.+.}, at: snd_timer_close+0x7e/0x100 sound/core/timer.c:422 2 locks held by syz-executor.5/12267: #0: ffff8880970f0950 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 #1: ffffffff89914b00 (register_mutex){+.+.}, at: snd_timer_open+0x98/0x1860 sound/core/timer.c:245 1 lock held by syz-executor.5/12270: #0: ffff8880970f0950 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 1 lock held by syz-executor.5/12272: #0: ffffffff89924320 (register_mutex#4){+.+.}, at: odev_open+0x5f/0x90 sound/core/seq/oss/seq_oss.c:124 1 lock held by syz-executor.5/12274: #0: ffff8880a0113d50 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 2 locks held by syz-executor.5/12277: #0: ffff8880a0113d50 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_ioctl+0x51/0xa7 sound/core/timer.c:2032 #1: ffffffff89914b00 (register_mutex){+.+.}, at: snd_timer_open+0x98/0x1860 sound/core/timer.c:245 1 lock held by syz-executor.5/12279: #0: ffffffff89924320 (register_mutex#4){+.+.}, at: odev_open+0x5f/0x90 sound/core/seq/oss/seq_oss.c:124 ============================================= NMI backtrace for cpu 1 CPU: 1 PID: 1070 Comm: khungtaskd Not tainted 5.4.0-rc6 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x172/0x1f0 lib/dump_stack.c:113 nmi_cpu_backtrace.cold+0x70/0xb2 lib/nmi_backtrace.c:101 nmi_trigger_cpumask_backtrace+0x23b/0x28b lib/nmi_backtrace.c:62 arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 trigger_all_cpu_backtrace include/linux/nmi.h:146 [inline] check_hung_uninterruptible_tasks kernel/hung_task.c:205 [inline] watchdog+0x9d0/0xef0 kernel/hung_task.c:289 kthread+0x361/0x430 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Sending NMI from CPU 1 to CPUs 0: NMI backtrace for cpu 0 CPU: 0 PID: 12245 Comm: syz-executor.3 Not tainted 5.4.0-rc6 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:rdtsc_ordered arch/x86/include/asm/msr.h:236 [inline] RIP: 0010:delay_tsc+0x4a/0xc0 arch/x86/lib/delay.c:61 Code: 0f 01 f9 66 90 48 c1 e2 20 48 09 c2 49 89 d4 eb 16 f3 90 bf 01 00 00 00 e8 03 c2 01 fa e8 de cd f9 fb 44 39 e8 75 36 0f 01 f9 <66> 90 48 c1 e2 20 48 89 d3 48 09 c3 48 89 d8 4c 29 e0 4c 39 f0 73 RSP: 0018:ffff88808034fc70 EFLAGS: 00000246 RAX: 000000004e0d2263 RBX: 000002d24e0d2178 RCX: 0000000000000000 RDX: 00000000000002d2 RSI: ffffffff8347345c RDI: 0000000000000005 RBP: ffff88808034fc90 R08: ffff8880998302c0 R09: fffffbfff14efd42 R10: fffffbfff14efd41 R11: ffffffff8a77ea0f R12: 000002d24e0d173d R13: 0000000000000000 R14: 00000000000059d9 R15: ffff88809ccb0108 FS: 0000000000fa4940(0000) GS:ffff8880ae800000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: ffffffffff600400 CR3: 0000000095a99000 CR4: 00000000001406f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: __delay arch/x86/lib/delay.c:161 [inline] __const_udelay+0x59/0x80 arch/x86/lib/delay.c:175 snd_timer_close_locked+0x22a/0xda0 sound/core/timer.c:369 snd_timer_close+0x8a/0x100 sound/core/timer.c:423 snd_timer_user_release+0x98/0x130 sound/core/timer.c:1478 __fput+0x2ff/0x890 fs/file_table.c:280 ____fput+0x16/0x20 fs/file_table.c:313 task_work_run+0x145/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_usermode_loop+0x316/0x380 arch/x86/entry/common.c:163 prepare_exit_to_usermode arch/x86/entry/common.c:194 [inline] syscall_return_slowpath arch/x86/entry/common.c:274 [inline] do_syscall_64+0x65f/0x760 arch/x86/entry/common.c:300 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x413db1 Code: 75 14 b8 03 00 00 00 0f 05 48 3d 01 f0 ff ff 0f 83 04 1b 00 00 c3 48 83 ec 08 e8 0a fc ff ff 48 89 04 24 b8 03 00 00 00 0f 05 <48> 8b 3c 24 48 89 c2 e8 53 fc ff ff 48 89 d0 48 83 c4 08 48 3d 01 RSP: 002b:00007fff58da8ec0 EFLAGS: 00000293 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000004 RCX: 0000000000413db1 RDX: 0000000000000000 RSI: 0000000000000081 RDI: 0000000000000003 RBP: 0000000000000000 R08: 0000000000760a88 R09: ffffffffffffffff R10: 00007fff58da8f90 R11: 0000000000000293 R12: 000000000075bf20 R13: 0000000000000003 R14: 0000000000760a90 R15: 000000000075bf2c --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

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Re: [alsa-devel] [PATCH V2 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm
by S.j. Wang 06 Nov '19

06 Nov '19

Hi Rob > > Hi > > > > On Wed, Oct 30, 2019 at 07:41:26PM +0800, Shengjiu Wang wrote: > > > In order to support the two asrc modules in imx8qm, we need to add > > > compatible string "fsl,imx8qm-asrc0" and "fsl,imx8qm-asrc1" > > > > Are the blocks different in some way? > > > > If not, why do you need to distinguish them? > > > The internal clock mapping is different for each module. > Or we can use one compatible string, but need add another property "fsl,asrc-clk-map" to distinguish the different clock map. The change is in below. Which one do you think is better? Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Contains "fsl,imx35-asrc", "fsl,imx53-asrc", + "fsl,imx8qm-asrc". - reg : Offset and length of the register set for the device. @@ -35,6 +36,11 @@ Required properties: - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. + - fsl,asrc-clk-map : Defines clock map used in driver. which is required + by imx8qm + <0> - select the map for asrc0 + <1> - select the map for asrc1 + Optional properties: Best regards Wang shengjiu

1 0

[alsa-devel] broonie-sound/for-next boot bisection: v5.4-rc6-292-gc1efaea10be0 on sun8i-h3-libretech-all-h3-cc
by kernelci.org bot 06 Nov '19

06 Nov '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * * * If you do send a fix, please include this trailer: * * Reported-by: "kernelci.org bot" <bot(a)kernelci.org> * * * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * broonie-sound/for-next boot bisection: v5.4-rc6-292-gc1efaea10be0 on sun8i-h3-libretech-all-h3-cc Summary: Start: c1efaea10be0 Merge branch 'asoc-5.5' into asoc-next Details: https://kernelci.org/boot/id/5dc218e159b5142346138dff Plain log: https://storage.kernelci.org//broonie-sound/for-next/v5.4-rc6-292-gc1efaea1… HTML log: https://storage.kernelci.org//broonie-sound/for-next/v5.4-rc6-292-gc1efaea1… Result: ac6a4dd3e9f0 ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() Checks: revert: PASS verify: PASS Parameters: Tree: broonie-sound URL: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Branch: for-next Target: sun8i-h3-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Compiler: gcc-8 Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit ac6a4dd3e9f09697ab6a1774d7ab6a34e7ab36fa Author: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue Nov 5 15:46:51 2019 +0900 ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() snd_soc_unregister_component() is now finding component manually, but we already have snd_soc_lookup_component() to find component; Let's use existing function. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/87zhha252c.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bb0592159414..0ce333669138 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2876,29 +2876,19 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); * * @dev: The device to unregister */ -static int __snd_soc_unregister_component(struct device *dev) +void snd_soc_unregister_component(struct device *dev) { struct snd_soc_component *component; - int found = 0; mutex_lock(&client_mutex); - for_each_component(component) { - if (dev != component->dev) - continue; + while (1) { + component = snd_soc_lookup_component(dev, NULL); + if (!component) + break; snd_soc_del_component_unlocked(component); - found = 1; - break; } mutex_unlock(&client_mutex); - - return found; -} - -void snd_soc_unregister_component(struct device *dev) -{ - while (__snd_soc_unregister_component(dev)) - ; } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [a99d8080aaf358d5d23581244e5da23b35e340b9] Linux 5.4-rc6 git bisect good a99d8080aaf358d5d23581244e5da23b35e340b9 # bad: [c1efaea10be0d91b6986af5c14cd3482ab160981] Merge branch 'asoc-5.5' into asoc-next git bisect bad c1efaea10be0d91b6986af5c14cd3482ab160981 # good: [1092b09708882e3c216f0b9c02e606b3c0942c5b] ASoC: tlv320aic32x4: add a check for devm_clk_get git bisect good 1092b09708882e3c216f0b9c02e606b3c0942c5b # good: [f03412b78a947857bbd20899e1423482fba55761] ASoC: rockchip-max98090: Support usage with and without HDMI git bisect good f03412b78a947857bbd20899e1423482fba55761 # good: [acbf27746ecfa96b290b54cc7f05273482ea128a] ASoC: pcm: update FE/BE trigger order based on the command git bisect good acbf27746ecfa96b290b54cc7f05273482ea128a # bad: [e443c20593de9f8efd9b2935ed40eb0bbacce30b] ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() git bisect bad e443c20593de9f8efd9b2935ed40eb0bbacce30b # good: [b8132657990b5a09ad8e1c9e2c8efc20b5f9372a] ASoC: soc-core: move snd_soc_lookup_component() git bisect good b8132657990b5a09ad8e1c9e2c8efc20b5f9372a # bad: [ac6a4dd3e9f09697ab6a1774d7ab6a34e7ab36fa] ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() git bisect bad ac6a4dd3e9f09697ab6a1774d7ab6a34e7ab36fa # good: [486c7978ff665eb763f70cc9477e0de6326e1c41] ASoC: soc-core: add snd_soc_del_component_unlocked() git bisect good 486c7978ff665eb763f70cc9477e0de6326e1c41 # good: [b18768f56162964f70bbb9119dba59a947d7d577] ASoC: soc-core: remove snd_soc_component_add/del() git bisect good b18768f56162964f70bbb9119dba59a947d7d577 # first bad commit: [ac6a4dd3e9f09697ab6a1774d7ab6a34e7ab36fa] ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() -------------------------------------------------------------------------------

2 1

[alsa-devel] [PATCH V2 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm
by Shengjiu Wang 06 Nov '19

06 Nov '19

In order to support the two asrc modules in imx8qm, we need to add compatible string "fsl,imx8qm-asrc0" and "fsl,imx8qm-asrc1" Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,asrc.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..cd2bd3daa7e1 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,8 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Contains "fsl,imx35-asrc", "fsl,imx53-asrc", + "fsl,imx8qm-asrc0" or "fsl,imx8qm-asrc1". - reg : Offset and length of the register set for the device. -- 2.21.0

3 3

[alsa-devel] [PATCH v3 00/11] Add Samus Hotwording for RT5677
by Curtis Malainey 06 Nov '19

06 Nov '19

This patch series adds the hotwording implementation used in the Pixelbook on the RT5677 driver. Known Issues: There is a known issue where the system will fail to detect a hotword if suspended while the stream is open. This is due to the fact that the haswell-dsp suspends its I2S MCLK before the RT5677 suspends which causes the writes and reads to become corrupted as a result. Any recommendations to correct this behaviour would be appreciated. Changes since v1: * Addressed Mark's and Kuninori's comments * Adjusted DAI Links * Squashed related patches * Removed all kcontrol writes * Addressed race condition with dapm * Added firmware boot check * Moved all clocking to driver API * Moved DAPM routes that spanned drivers to machine driver Changes since v2: * Updated and removed necessary dapm forced power on * Addressed Cexzary's comments Cc: Dylan Reid <dgreid(a)chromium.org> Cc: Ben Zhang <benzh(a)chromium.org> Ben Zhang (7): ASoC: rt5677: Load firmware via SPI using delayed work ASoC: bdw-rt5677: Add a DAI link for rt5677 SPI PCM device ASoC: rt5677: Enable jack detect while DSP is running ASoC: rt5677: Add DAPM audio path for hotword stream ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile ASoC: rt5677: Stop and restart DSP over suspend/resume ASoC: rt5677: Disable irq at suspend Curtis Malainey (4): ASoC: rt5677: Allow VAD to be shut on/off at all times ASoC: bdw-rt5677: Turn on MCLK1 for DSP via DAPM ASoC: rt5677: Set ADC clock to use PLL and enable ASRC ASoC: rt5677: Wait for DSP to boot before loading firmware sound/soc/codecs/rt5677-spi.c | 13 + sound/soc/codecs/rt5677.c | 445 +++++++++++++++++++++++----- sound/soc/codecs/rt5677.h | 11 +- sound/soc/intel/boards/bdw-rt5677.c | 51 ++++ 4 files changed, 447 insertions(+), 73 deletions(-) -- 2.24.0.rc1.363.gb1bccd3e3d-goog

2 12

[alsa-devel] [PATCH] ALSA: hda: hdmi - add Tigerlake support
by Kai Vehmanen 05 Nov '19

05 Nov '19

Add Tigerlake HDMI codec support. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205379 BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=112171 Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b72553710ffb..3c720703ebb8 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2851,6 +2851,18 @@ static int patch_i915_icl_hdmi(struct hda_codec *codec) return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); } +static int patch_i915_tgl_hdmi(struct hda_codec *codec) +{ + /* + * pin to port mapping table where the value indicate the pin number and + * the index indicate the port number with 1 base. + */ + static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; + + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); +} + + /* Intel Baytrail and Braswell; with eld notifier */ static int patch_i915_byt_hdmi(struct hda_codec *codec) { @@ -4153,6 +4165,7 @@ HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), +HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.17.1

3 3

[alsa-devel] [PATCH 0/2] ALSA: pcm: minor enhancements
by Takashi Iwai 05 Nov '19

05 Nov '19

Hi, here are a couple of more small enhnacements on PCM memory allocation stuff. Takashi === Takashi Iwai (2): ALSA: pcm: Warn if doubly preallocated ALSA: pcm: Create proc files only for non-empty preallocations sound/core/pcm_memory.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) -- 2.16.4

1 2

[alsa-devel] [PATCH] ASoC: stm32: sai: add restriction on mmap support
by Olivier Moysan 05 Nov '19

05 Nov '19

Do not support mmap in S/PDIF mode. In S/PDIF mode the buffer has to be copied, to allow the channel status bits insertion. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index a4060813bc74..48e629ac2d88 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1218,6 +1218,16 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream, return 0; } +/* No support of mmap in S/PDIF mode */ +static const struct snd_pcm_hardware stm32_sai_pcm_hw_spdif = { + .info = SNDRV_PCM_INFO_INTERLEAVED, + .buffer_bytes_max = 8 * PAGE_SIZE, + .period_bytes_min = 1024, + .period_bytes_max = PAGE_SIZE, + .periods_min = 2, + .periods_max = 8, +}; + static const struct snd_pcm_hardware stm32_sai_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP, .buffer_bytes_max = 8 * PAGE_SIZE, @@ -1270,7 +1280,7 @@ static const struct snd_dmaengine_pcm_config stm32_sai_pcm_config = { }; static const struct snd_dmaengine_pcm_config stm32_sai_pcm_config_spdif = { - .pcm_hardware = &stm32_sai_pcm_hw, + .pcm_hardware = &stm32_sai_pcm_hw_spdif, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .process = stm32_sai_pcm_process_spdif, }; -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ASoC: max98373: replace gpio_request with devm_gpio_request
by Yong Zhi 05 Nov '19

05 Nov '19

Use devm_gpio_request() to automatic unroll when fails and avoid resource leaks at error paths. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- v2 change: - use devm_gpio_request to avoid the goto statement based on review comments. sound/soc/codecs/max98373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index eb709d528259..cae1def8902d 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -960,11 +960,11 @@ static int max98373_i2c_probe(struct i2c_client *i2c, /* Power on device */ if (gpio_is_valid(max98373->reset_gpio)) { - ret = gpio_request(max98373->reset_gpio, "MAX98373_RESET"); + ret = devm_gpio_request(&i2c->dev, max98373->reset_gpio, + "MAX98373_RESET"); if (ret) { dev_err(&i2c->dev, "%s: Failed to request gpio %d\n", __func__, max98373->reset_gpio); - gpio_free(max98373->reset_gpio); return -EINVAL; } gpio_direction_output(max98373->reset_gpio, 0); -- 2.7.4

3 2

[alsa-devel] Incorrect buffer handling in dw-hdmi bridge audio
by Takashi Iwai 05 Nov '19

05 Nov '19

Hi, while recently working on the ALSA memory allocator API cleanup, I noticed that dw-hdmi bridge driver seems doing weird about the buffer management. It pre-allocates the usual device buffers fully at the probe time, while each stream allocates the buffer via the vmalloc helpers and replaces with it at each open. I guess it's no expected behavior? It's basically a full waste of resources, and the vmalloc buffer isn't guaranteed to work well for mmap on every architecture. Below is the patch to address it. Can anyone check whether this still works? Since I have a cleanup series and this is involved, I'd like to take the fix through my tree once after it's confirmed (and get ACK if possible). Thanks! Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] drm/bridge: dw-hdmi: Fix the incorrect buffer allocations The driver sets up the buffer preallocation with SNDRV_DMA_TYPE_DEV, while it re-allocates and releases vmalloc pages. This is not only a lot of waste resources but also causes the mmap malfunction. Change / drop the vmalloc-related code and use the standard buffer allocation / release code instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c index 2b7539701b42..8fe7a6e8ff94 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c @@ -384,15 +384,14 @@ static int dw_hdmi_close(struct snd_pcm_substream *substream) static int dw_hdmi_hw_free(struct snd_pcm_substream *substream) { - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int dw_hdmi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { /* Allocate the PCM runtime buffer, which is exposed to userspace. */ - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(params)); + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } static int dw_hdmi_prepare(struct snd_pcm_substream *substream) @@ -511,7 +510,6 @@ static const struct snd_pcm_ops snd_dw_hdmi_ops = { .prepare = dw_hdmi_prepare, .trigger = dw_hdmi_trigger, .pointer = dw_hdmi_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static int snd_dw_hdmi_probe(struct platform_device *pdev) -- 2.16.4

3 6

[alsa-devel] [PATCH v2] dt-bindings: sound: adau7118: Fix example warning
by Maxime Ripard 05 Nov '19

05 Nov '19

The ADAU7118 has an example where the codec has an i2c address of 14, and the unit address set to 14 as well. However, while the address is expressed in decimal, the unit-address is supposed to be in hexadecimal, which ends up with two different addresses that trigger a DTC warning. Fix this by setting the address to 0x14. Cc: Nuno Sá <nuno.sa(a)analog.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Fixes: 969d49b2cdc8 ("dt-bindings: asoc: Add ADAU7118 documentation") Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Changes from v1: - Fix the address instead of the unit-address. --- Documentation/devicetree/bindings/sound/adi,adau7118.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml index c3f10afbdd6f..75e0cbe6be70 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -67,7 +67,7 @@ examples: #size-cells = <0>; adau7118_codec: audio-codec@14 { compatible = "adi,adau7118"; - reg = <14>; + reg = <0x14>; #sound-dai-cells = <0>; iovdd-supply = <&supply>; dvdd-supply = <&supply>; -- 2.23.0

2 1

[alsa-devel] [PATCH 00/24] Clean up memalloc callers
by Takashi Iwai 05 Nov '19

05 Nov '19

Hi, this is a trivial cleanup series corresponding to the recent ALSA memalloc API enhancement[1]. Most of patches do simple and systematic changes: either convert to the new way of vmalloc buffer handling, replacing the superfluous snd_dma_continous_data() macro with NULL, or replacing the snd_dma_pci_data() with the direct device pointer assignment. Some have a slightly more changes but they are straightforward enough. Also, a few document updates are provided. There are still a few reminders (noticeably ASoC), and I'll post the patches for them later. Once after all conversions done, we can drop the superfluous API functions and macros. thanks, Takashi [1] https://lore.kernel.org/r/20191105080138.1260-1-tiwai@suse.de === Takashi Iwai (24): ALSA: drivers: Remove superfluous snd_dma_continuous_data() ALSA: mips: Remove superfluous snd_dma_continuous_data() ALSA: rme32: Remove superfluous snd_dma_continuous_data() ALSA: sh: Remove superfluous snd_dma_continuous_data() ALSA: sparc: Remove superfluous snd_dma_continuous_data() ALSA: usb: Remove superfluous snd_dma_continuous_data() ALSA: 6fire: Drop the dead code ALSA: usb-audio: Convert to the common vmalloc memalloc ALSA: aloop: Convert to the common vmalloc memalloc ALSA: firewire: Convert to the common vmalloc memalloc ALSA: 6fire: Convert to the common vmalloc memalloc ALSA: caiaq: Convert to the common vmalloc memalloc ALSA: hiface: Convert to the common vmalloc memalloc ALSA: ua101: Convert to the common vmalloc memalloc ALSA: vx: Convert to the common vmalloc memalloc ALSA: pdaudiocf: Convert to the common vmalloc memalloc ALSA: mips: Convert to the common vmalloc memalloc ALSA: pci: Drop superfluous snd_pcm_sgbuf_ops_page ALSA: aoa: Avoid non-standard macro usage ALSA: echoaudio: Avoid non-standard macro usage ALSA: es1968: Avoid non-standard macro usage ALSA: rme: Avoid non-standard macro usage ALSA: pci: Avoid non-standard macro usage ALSA: docs: Remove snd_dma_pci_data() usage in documentation .../sound/kernel-api/writing-an-alsa-driver.rst | 8 ++-- sound/aoa/soundbus/i2sbus/pcm.c | 2 +- sound/drivers/aloop.c | 8 ++-- sound/drivers/dummy.c | 2 +- sound/drivers/ml403-ac97cr.c | 2 +- sound/drivers/pcsp/pcsp_lib.c | 4 +- sound/drivers/vx/vx_pcm.c | 10 ++--- sound/firewire/bebob/bebob_pcm.c | 9 ++-- sound/firewire/dice/dice-pcm.c | 11 ++--- sound/firewire/digi00x/digi00x-pcm.c | 9 ++-- sound/firewire/fireface/ff-pcm.c | 9 ++-- sound/firewire/fireworks/fireworks_pcm.c | 9 ++-- sound/firewire/isight.c | 8 ++-- sound/firewire/motu/motu-pcm.c | 9 ++-- sound/firewire/oxfw/oxfw-pcm.c | 14 +++--- sound/firewire/tascam/tascam-pcm.c | 9 ++-- sound/mips/hal2.c | 3 +- sound/mips/sgio2audio.c | 12 +++--- sound/pci/ad1889.c | 6 +-- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 3 +- sound/pci/als4000.c | 3 +- sound/pci/asihpi/asihpi.c | 4 +- sound/pci/atiixp.c | 6 +-- sound/pci/atiixp_modem.c | 4 +- sound/pci/au88x0/au88x0_pcm.c | 3 +- sound/pci/aw2/aw2-alsa.c | 6 +-- sound/pci/azt3328.c | 8 ++-- sound/pci/bt87x.c | 5 +-- sound/pci/ca0106/ca0106_main.c | 6 +-- sound/pci/cmipci.c | 6 +-- sound/pci/cs4281.c | 3 +- sound/pci/cs46xx/cs46xx_lib.c | 16 ++++--- sound/pci/cs5535audio/cs5535audio_pcm.c | 6 +-- sound/pci/ctxfi/ctpcm.c | 5 +-- sound/pci/ctxfi/ctvmem.c | 2 +- sound/pci/echoaudio/echoaudio.c | 24 +++-------- sound/pci/emu10k1/emu10k1.c | 5 ++- sound/pci/emu10k1/emu10k1x.c | 6 +-- sound/pci/emu10k1/emufx.c | 2 +- sound/pci/emu10k1/emupcm.c | 12 +++--- sound/pci/emu10k1/memory.c | 4 +- sound/pci/emu10k1/p16v.c | 4 +- sound/pci/ens1370.c | 8 ++-- sound/pci/es1938.c | 3 +- sound/pci/es1968.c | 4 +- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_controller.c | 1 - sound/pci/ice1712/ice1712.c | 9 ++-- sound/pci/ice1712/ice1724.c | 6 +-- sound/pci/intel8x0.c | 4 +- sound/pci/intel8x0m.c | 4 +- sound/pci/korg1212/korg1212.c | 8 ++-- sound/pci/lola/lola.c | 2 +- sound/pci/lola/lola_pcm.c | 5 +-- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 3 +- sound/pci/mixart/mixart.c | 7 +-- sound/pci/oxygen/oxygen_pcm.c | 10 ++--- sound/pci/pcxhr/pcxhr.c | 4 +- sound/pci/riptide/riptide.c | 6 +-- sound/pci/rme32.c | 4 +- sound/pci/rme9652/hdsp.c | 7 +-- sound/pci/rme9652/hdspm.c | 3 +- sound/pci/rme9652/rme9652.c | 7 +-- sound/pci/sis7019.c | 3 +- sound/pci/sonicvibes.c | 3 +- sound/pci/trident/trident_main.c | 24 ++++++----- sound/pci/via82xx.c | 17 +++----- sound/pci/via82xx_modem.c | 6 +-- sound/pci/ymfpci/ymfpci_main.c | 16 ++++--- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 9 ++-- sound/sh/aica.c | 2 +- sound/sh/sh_dac_audio.c | 2 +- sound/sparc/amd7930.c | 2 +- sound/sparc/dbri.c | 2 +- sound/usb/6fire/pcm.c | 16 +++---- sound/usb/caiaq/audio.c | 8 ++-- sound/usb/hiface/pcm.c | 9 ++-- sound/usb/line6/pcm.c | 4 +- sound/usb/misc/ua101.c | 14 +++--- sound/usb/pcm.c | 50 ++++------------------ sound/usb/usx2y/usbusx2yaudio.c | 4 +- sound/usb/usx2y/usx2yhwdeppcm.c | 4 +- 84 files changed, 266 insertions(+), 327 deletions(-) -- 2.16.4

1 24

[alsa-devel] [PATCH 00/62] Add definition for GPIO direction
by Matti Vaittinen 05 Nov '19

05 Nov '19

The patch series adds definitions for GPIO line directions. For occasional GPIO contributor like me it is always a pain to remember whether 1 or 0 was used for GPIO direction INPUT/OUTPUT. Judging the fact that I removed few comments like: /* Return 0 if output, 1 if input */ /* This means "out" */ return 1; /* input */ return 0; /* output */ it seems at least some others may find it hard to remember too. Adding defines for these values helps us who really have good - but short duration - memory :] Please also see the last patch. It adds warning prints to be emitted from gpiolib if other than defined values are used. This patch can be dropped out if there is a reason for that to still be allowed. This idea comes from RFC series for ROHM BD71828 PMIC and was initially discussed with Linus Walleij here: https://lore.kernel.org/lkml/c06725c3dd34118a324907137758d8b85b3d4043.camel… but as this has no dependencies to BD71828 work (which probably takes a while) I decided to make it independent series. Patches are compile-tested only. I have no HW to really test them. Thus I'd appreciate carefull review. This work is mainly about converting zeros and ones to the new defines but it wouldn't be first time I get it wrong in one of the patches :) Patch 1: - adds the defines Patches 2 - 61: - convert drivers to use new defines Patch 62: - Add warning print if values other than the defines is used for direction. Patches are created on top of Linux v5.4-rc6. -- Matti Vaittinen (62): gpio: Add definition for GPIO direction gpio: gpio-104-dio-48e: Use new GPIO_LINE_DIRECTION gpio: add gpio-104-idi-48: Use new GPIO_LINE_DIRECTION gpio: gpio-104-idio-16: Use new GPIO_LINE_DIRECTION gpio: gpio-74xx-mmio: Use new GPIO_LINE_DIRECTION gpio: gpio-amd-fch: Use new GPIO_LINE_DIRECTION gpio: gpio-aspeed: Use new GPIO_LINE_DIRECTION gpio: gpio-bcm-kona: Use new GPIO_LINE_DIRECTION gpio: gpio-bd70528: Use new GPIO_LINE_DIRECTION gpio: gpio-bd9571mwv: Use new GPIO_LINE_DIRECTION gpio: gpio-dln2: Use new GPIO_LINE_DIRECTION gpio: gpio-exar: Use new GPIO_LINE_DIRECTION gpio: gpio-f7188x: Use new GPIO_LINE_DIRECTION gpio: gpio-gpio-mm: Use new GPIO_LINE_DIRECTION gpio: gpio-htc-egpio: Use new GPIO_LINE_DIRECTION gpio: gpio-ich: Use new GPIO_LINE_DIRECTION gpio: gpio-kempld: Use new GPIO_LINE_DIRECTION gpio: gpio-lp873x: Use new GPIO_LINE_DIRECTION gpio: gpio-lp87565: Use new GPIO_LINE_DIRECTION gpio: gpio-madera: Use new GPIO_LINE_DIRECTION gpio: gpio-max3191x: Use new GPIO_LINE_DIRECTION gpio: gpio-mmio: Use new GPIO_LINE_DIRECTION gpio: gpio-merrifield: Use new GPIO_LINE_DIRECTION gpio: gpio-mockup: Use new GPIO_LINE_DIRECTION gpio: gpio-moxtet: Use new GPIO_LINE_DIRECTION gpio: gpio-mvebu: Use new GPIO_LINE_DIRECTION gpio: gpio-mxs: Use new GPIO_LINE_DIRECTION gpio: gpio-omap: Use new GPIO_LINE_DIRECTION gpio: gpio-pca953x: Use new GPIO_LINE_DIRECTION gpio: gpio-pcie-idio-24: Use new GPIO_LINE_DIRECTION gpio: gpio-pci-idio-16: Use new GPIO_LINE_DIRECTION gpio: gpio-pisosr: Use new GPIO_LINE_DIRECTION gpio: gpio-pl061: Use new GPIO_LINE_DIRECTION gpio: gpio-raspberrypi-exp: Use new GPIO_LINE_DIRECTION gpio: gpio-rcar: Use new GPIO_LINE_DIRECTION gpio: gpio-reg: Use new GPIO_LINE_DIRECTION gpio: gpio-sama5d2-piobu: Use new GPIO_LINE_DIRECTION gpio: gpio-sch311x: Use new GPIO_LINE_DIRECTION gpio: gpio-sch: Use new GPIO_LINE_DIRECTION gpio: gpio-siox: Use new GPIO_LINE_DIRECTION gpio: gpio-stmpe: Use new GPIO_LINE_DIRECTION gpio: gpio-tc3589x: Use new GPIO_LINE_DIRECTION gpio: gpio-tegra186: Use new GPIO_LINE_DIRECTION gpio: gpio-tegra: Use new GPIO_LINE_DIRECTION gpio: gpio-thunderx: Use new GPIO_LINE_DIRECTION gpio: gpio-tpic2810: Use new GPIO_LINE_DIRECTION gpio: gpio-tps65086: Use new GPIO_LINE_DIRECTION gpio: gpio-tps65912: Use new GPIO_LINE_DIRECTION gpio: gpio-tps68470: Use new GPIO_LINE_DIRECTION gpio: gpio-tqmx86: Use new GPIO_LINE_DIRECTION gpio: gpio-ts4900: Use new GPIO_LINE_DIRECTION gpio: gpio-twl4030: Use new GPIO_LINE_DIRECTION gpio: gpio-twl6040: Use new GPIO_LINE_DIRECTION gpio: gpio-uniphier: Use new GPIO_LINE_DIRECTION gpio: gpio-wcove: Use new GPIO_LINE_DIRECTION gpio: gpio-ws16c48: Use new GPIO_LINE_DIRECTION gpio: gpio-xgene: Use new GPIO_LINE_DIRECTION gpio: gpio-xra1403: Use new GPIO_LINE_DIRECTION gpio: gpio-xtensa: Use new GPIO_LINE_DIRECTION gpio: gpio-zynq: Use new GPIO_LINE_DIRECTION gpio: gpio-sa1100: Use new GPIO_LINE_DIRECTION gpiolib: Nag for INPUT direction values other than GPIO_LINE_DIRECTION_IN drivers/gpio/gpio-104-dio-48e.c | 5 ++++- drivers/gpio/gpio-104-idi-48.c | 2 +- drivers/gpio/gpio-104-idio-16.c | 4 ++-- drivers/gpio/gpio-74xx-mmio.c | 5 ++++- drivers/gpio/gpio-amd-fch.c | 2 +- drivers/gpio/gpio-aspeed.c | 7 +++---- drivers/gpio/gpio-bcm-kona.c | 6 +++--- drivers/gpio/gpio-bd70528.c | 8 +++++--- drivers/gpio/gpio-bd9571mwv.c | 4 +++- drivers/gpio/gpio-dln2.c | 6 +++--- drivers/gpio/gpio-exar.c | 5 ++++- drivers/gpio/gpio-f7188x.c | 5 ++++- drivers/gpio/gpio-gpio-mm.c | 5 ++++- drivers/gpio/gpio-htc-egpio.c | 5 ++++- drivers/gpio/gpio-ich.c | 5 ++++- drivers/gpio/gpio-kempld.c | 5 ++++- drivers/gpio/gpio-lp873x.c | 2 +- drivers/gpio/gpio-lp87565.c | 5 ++++- drivers/gpio/gpio-madera.c | 5 ++++- drivers/gpio/gpio-max3191x.c | 2 +- drivers/gpio/gpio-merrifield.c | 5 ++++- drivers/gpio/gpio-mmio.c | 21 +++++++++++++++------ drivers/gpio/gpio-mockup.c | 11 +++-------- drivers/gpio/gpio-moxtet.c | 4 ++-- drivers/gpio/gpio-mvebu.c | 5 ++++- drivers/gpio/gpio-mxs.c | 5 ++++- drivers/gpio/gpio-omap.c | 6 ++++-- drivers/gpio/gpio-pca953x.c | 5 ++++- drivers/gpio/gpio-pci-idio-16.c | 4 ++-- drivers/gpio/gpio-pcie-idio-24.c | 9 ++++++--- drivers/gpio/gpio-pisosr.c | 2 +- drivers/gpio/gpio-pl061.c | 5 ++++- drivers/gpio/gpio-raspberrypi-exp.c | 5 ++++- drivers/gpio/gpio-rcar.c | 5 ++++- drivers/gpio/gpio-reg.c | 3 ++- drivers/gpio/gpio-sa1100.c | 5 ++++- drivers/gpio/gpio-sama5d2-piobu.c | 7 ++++--- drivers/gpio/gpio-sch.c | 5 ++++- drivers/gpio/gpio-sch311x.c | 5 ++++- drivers/gpio/gpio-siox.c | 4 ++-- drivers/gpio/gpio-stmpe.c | 5 ++++- drivers/gpio/gpio-tc3589x.c | 5 ++++- drivers/gpio/gpio-tegra.c | 5 ++++- drivers/gpio/gpio-tegra186.c | 4 ++-- drivers/gpio/gpio-thunderx.c | 5 ++++- drivers/gpio/gpio-tpic2810.c | 2 +- drivers/gpio/gpio-tps65086.c | 2 +- drivers/gpio/gpio-tps65912.c | 4 ++-- drivers/gpio/gpio-tps68470.c | 6 +++--- drivers/gpio/gpio-tqmx86.c | 5 ++++- drivers/gpio/gpio-ts4900.c | 5 ++++- drivers/gpio/gpio-twl4030.c | 10 +++++----- drivers/gpio/gpio-twl6040.c | 3 +-- drivers/gpio/gpio-uniphier.c | 5 ++++- drivers/gpio/gpio-wcove.c | 7 +++++-- drivers/gpio/gpio-ws16c48.c | 5 ++++- drivers/gpio/gpio-xgene.c | 5 ++++- drivers/gpio/gpio-xra1403.c | 5 ++++- drivers/gpio/gpio-xtensa.c | 4 ++-- drivers/gpio/gpio-zynq.c | 7 +++++-- drivers/gpio/gpiolib.c | 16 ++++++++++++++-- include/linux/gpio/driver.h | 3 +++ 62 files changed, 228 insertions(+), 104 deletions(-) -- 2.21.0 -- Matti Vaittinen, Linux device drivers ROHM Semiconductors, Finland SWDC Kiviharjunlenkki 1E 90220 OULU FINLAND ~~~ "I don't think so," said Rene Descartes. Just then he vanished ~~~ Simon says - in Latin please. ~~~ "non cogito me" dixit Rene Descarte, deinde evanescavit ~~~ Thanks to Simon Glass for the translation =]

5 18

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Fix possible workqueue stall
by Takashi Iwai 05 Nov '19

05 Nov '19

The unsolicited event handler for the headphone jack on CA0132 codec driver tries to reschedule the another delayed work with cancel_delayed_work_sync(). It's no good idea, unfortunately, especially after we changed the work queue to the standard global one; this may lead to a stall because both works are using the same global queue. Fix it by dropping the _sync but does call cancel_delayed_work() instead. Fixes: 993884f6a26c ("ALSA: hda/ca0132 - Delay HP amp turnon.") BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1155836 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 6d1fb7c11f17..b7a1abb3e231 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -7604,7 +7604,7 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) /* Delay enabling the HP amp, to let the mic-detection * state machine run. */ - cancel_delayed_work_sync(&spec->unsol_hp_work); + cancel_delayed_work(&spec->unsol_hp_work); schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); tbl = snd_hda_jack_tbl_get(codec, cb->nid); if (tbl) -- 2.16.4

1 0

[alsa-devel] [PATCH] ASoC: bdw-rt5677: enable runtime channel merge
by Brent Lu 05 Nov '19

05 Nov '19

In the DAI link "Capture PCM", the FE DAI "Capture Pin" supports 4-channel capture but the BE DAI supports only 2-channel capture. To fix the channel mismatch, we need to enable the runtime channel merge for this DAI link. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 4a4d335..8d07038 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -273,6 +273,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { }, .dpcm_capture = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, SND_SOC_DAILINK_REG(fe, dummy, platform), }, -- 2.7.4

3 4

[alsa-devel] amixer: "5%-" or "5%+" doesn't work with cset
by GitHub issues - opened 05 Nov '19

05 Nov '19

alsa-project/alsa-utils issue #32 was opened from MarioMey: I want to raise and low volume with keys. This works, this rise the volume 20%: `amixer set Master 20%+` And every time I run this command, it rise 20% more. But, when I want to use the card (because I need to do this): `amixer -c 0 cset name='Master Playback Volume' 20%+` It doesn't work. It seems that amixer takes only "20%" value and doesn't care about "+" or "-". I have Debian Buster, Mate desktop. ``` mario@debian:~/$ apt-cache showpkg alsa-tools Package: alsa-tools Versions: 1.1.7-1 (/var/lib/apt/lists/ftp.us.debian.org_debian_dists_buster_main_binary-amd64_Packages) Description Language: File: /var/lib/apt/lists/ftp.us.debian.org_debian_dists_buster_main_binary-amd64_Packages MD5: 4d64eace7e51fb1fc1a18ed780ee5e62 Description Language: en File: /var/lib/apt/lists/ftp.us.debian.org_debian_dists_buster_main_i18n_Translation-en MD5: 4d64eace7e51fb1fc1a18ed780ee5e62 Reverse Depends: ld10k1,alsa-tools multimedia-jack,alsa-tools Dependencies: 1.1.7-1 - libasound2 (2 1.0.16) libc6 (2 2.7) libgcc1 (2 1:3.0) libglib2.0-0 (2 2.12.0) libstdc++6 (2 5) as10k1 (0 (null)) as10k1 (0 (null)) Provides: 1.1.7-1 - as10k1 (= ) Reverse Provides: ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/32 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] tegra30 tdm support
by Ben Dooks 05 Nov '19

05 Nov '19

New series for TDM audio support on tegra30 v2: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock v3: - cleanup fsync patch - fix rebase issue with tdm patch v4: - fix comment style issues - change tdm-a to data-offset 1

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[alsa-devel] [PATCH 20/62] gpio: gpio-madera: Use new GPIO_LINE_DIRECTION
by Matti Vaittinen 05 Nov '19

05 Nov '19

It's hard for occasional GPIO code reader/writer to know if values 0/1 equal to IN or OUT. Use defined GPIO_LINE_DIRECTION_IN and GPIO_LINE_DIRECTION_OUT to help them out. Signed-off-by: Matti Vaittinen <matti.vaittinen(a)fi.rohmeurope.com> --- drivers/gpio/gpio-madera.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/gpio/gpio-madera.c b/drivers/gpio/gpio-madera.c index 7086f8b5388f..8f38303fcbc4 100644 --- a/drivers/gpio/gpio-madera.c +++ b/drivers/gpio/gpio-madera.c @@ -34,7 +34,10 @@ static int madera_gpio_get_direction(struct gpio_chip *chip, if (ret < 0) return ret; - return !!(val & MADERA_GP1_DIR_MASK); + if (val & MADERA_GP1_DIR_MASK) + return GPIO_LINE_DIRECTION_IN; + + return GPIO_LINE_DIRECTION_OUT; } static int madera_gpio_direction_in(struct gpio_chip *chip, unsigned int offset) -- 2.21.0 -- Matti Vaittinen, Linux device drivers ROHM Semiconductors, Finland SWDC Kiviharjunlenkki 1E 90220 OULU FINLAND ~~~ "I don't think so," said Rene Descartes. Just then he vanished ~~~ Simon says - in Latin please. ~~~ "non cogito me" dixit Rene Descarte, deinde evanescavit ~~~ Thanks to Simon Glass for the translation =]

1 0

[alsa-devel] [PATCH] ALSA: hda: Disable regmap internal locking
by Takashi Iwai 05 Nov '19

05 Nov '19

Since we apply the own mutex (bus->cmd_mutex) in HDA core side, the internal locking in regmap is superfluous. This patch adds the flag to indicate that. Also, an infamous side-effect by this change is that it disables the regmap debugfs, too, and this is seen rather good; the regmap debugfs isn't quite useful for HD-audio as it provides the very sparse registers and its debugfs access tends to lead to the way too high resource usages or sometimes hang up. So it'd be rather safe to disable it altogether. Link: https://lore.kernel.org/r/2029139028.10333037.1572874551626.JavaMail.zimbra… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_regmap.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 286361ecd640..906b1e20bae0 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -363,6 +363,7 @@ static const struct regmap_config hda_regmap_cfg = { .reg_write = hda_reg_write, .use_single_read = true, .use_single_write = true, + .disable_locking = true, }; /** -- 2.16.4

1 0

[alsa-devel] [PATCH v2 resend 00/18] ASoC: soc-core cleanup - step 4
by Kuninori Morimoto 05 Nov '19

05 Nov '19

Hi Mark These are resend of soc-core cleanup step4. These are based on mark/for-5.5 branch. I already got report from Intel that these had no issue on Intel CI machine. Kuninori Morimoto (18): ASoC: soc-core: remove unneeded snd_soc_tplg_component_remove() ASoC: soc-core: move soc_init_dai_link() ASoC: soc-core: tidyup soc_init_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: call soc_bind_dai_link() under snd_soc_add_dai_link() ASoC: soc-core: add soc_unbind_dai_link() ASoC: soc-core: move snd_soc_lookup_component() ASoC: soc-core: add snd_soc_del_component_unlocked() ASoC: soc-core: remove snd_soc_component_add/del() ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() ASoC: soc-core: move snd_soc_register_dai() ASoC: soc-core: move snd_soc_unregister_dais() ASoC: soc-core: add snd_soc_unregister_dai() ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() ASoC: soc-core: call snd_soc_register_dai() from snd_soc_register_dais() ASoC: soc-core: remove topology specific operation ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY include/sound/soc.h | 15 +- sound/soc/soc-core.c | 536 +++++++++++++++++++++-------------------------- sound/soc/soc-topology.c | 17 +- 3 files changed, 263 insertions(+), 305 deletions(-) -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: wm8524: Add support S32_LE
by S.j. Wang 05 Nov '19

05 Nov '19

Hi > On Fri, Nov 01, 2019 at 06:34:54AM +0000, S.j. Wang wrote: > > Allow 32bit sample with this codec. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/codecs/wm8524.c | 4 +++- > > 1 file changed, 3 insertions(+), 1 deletion(-) > > > > diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c > > index 91e3d1570c45..4e9ab542f648 100644 > > --- a/sound/soc/codecs/wm8524.c > > +++ b/sound/soc/codecs/wm8524.c > > @@ -159,7 +159,9 @@ static int wm8524_mute_stream(struct > snd_soc_dai > > *dai, int mute, int stream) > > > > #define WM8524_RATES SNDRV_PCM_RATE_8000_192000 > > > > -#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | > > SNDRV_PCM_FMTBIT_S24_LE) > > +#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ > > + SNDRV_PCM_FMTBIT_S24_LE |\ > > + SNDRV_PCM_FMTBIT_S32_LE) > > > > The device doesn't actually support 32bit though, I guess it will ignore the > extra LSBs so it should work. But is that really supporting 32 bit? > I also think it may ignore the extra LSBs just as you said, but don't really Support 32bit. This change is to make driver accept S32_LE format. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH v4 0/2] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 05 Nov '19

05 Nov '19

Thanks for reviewing the v3 patchset. Here is new patchset addressing all the comments from v3 This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is tested on on DragonBoard DB845c based of SDM845 SoC and Lenovo YOGA C630 Laptop based on SDM850. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since v3: - Updated bindings as suggested by Rob. - Fixed sdw_disable/unprepare order. - removed setting stream_name as suggested by Pierre - removed v1.5.0 controller support for now, will be added after testing. Srinivas Kandagatla (2): dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller .../bindings/soundwire/qcom,sdw.txt | 179 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 904 ++++++++++++++++++ 4 files changed, 1096 insertions(+) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

3 8

[alsa-devel] [PATCH] slimbus: qcom: add missed clk_disable_unprepare in remove
by Chuhong Yuan 04 Nov '19

04 Nov '19

The remove misses to disable and unprepare rclk and hclk. Add calls to clk_disable_unprepare to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- drivers/slimbus/qcom-ctrl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index a444badd8df5..4aad2566f52d 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -641,6 +641,8 @@ static int qcom_slim_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); slim_unregister_controller(&ctrl->ctrl); + clk_disable_unprepare(ctrl->rclk); + clk_disable_unprepare(ctrl->hclk); destroy_workqueue(ctrl->rxwq); return 0; } -- 2.23.0

1 0

[alsa-devel] [PATCH] ASoC: max98373: Fix calls to free_gpio()
by Yong Zhi 04 Nov '19

04 Nov '19

Don't Call free_gpio() when request_gpio() fails, call it at error paths to avoid resource leak. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/codecs/max98373.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index eb709d528259..83984b5244c5 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -964,7 +964,6 @@ static int max98373_i2c_probe(struct i2c_client *i2c, if (ret) { dev_err(&i2c->dev, "%s: Failed to request gpio %d\n", __func__, max98373->reset_gpio); - gpio_free(max98373->reset_gpio); return -EINVAL; } gpio_direction_output(max98373->reset_gpio, 0); @@ -979,16 +978,23 @@ static int max98373_i2c_probe(struct i2c_client *i2c, if (ret < 0) { dev_err(&i2c->dev, "Failed to read: 0x%02X\n", MAX98373_R21FF_REV_ID); - return ret; + goto err_gpio_free; } dev_info(&i2c->dev, "MAX98373 revisionID: 0x%02X\n", reg); /* codec registeration */ ret = devm_snd_soc_register_component(&i2c->dev, &soc_codec_dev_max98373, max98373_dai, ARRAY_SIZE(max98373_dai)); - if (ret < 0) + if (ret < 0) { dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + goto err_gpio_free; + } + + return 0; +err_gpio_free: + if (gpio_is_valid(max98373->reset_gpio)) + gpio_free(max98373->reset_gpio); return ret; } -- 2.7.4

5 4

[alsa-devel] [PATCH] ASoC: hdac_hda: fix race in device removal
by Pierre-Louis Bossart 04 Nov '19

04 Nov '19

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> When ASoC card instance is removed containing a HDA codec, hdac_hda_codec_remove() may run in parallel with codec resume. This will cause problems if the HDA link is freed with snd_hdac_ext_bus_link_put() while the codec is still in middle of its resume process. To fix this, change the order such that pm_runtime_disable() is called before the link is freed. This will ensure any pending runtime PM action is completed before proceeding to free the link. This issue can be easily hit with e.g. SOF driver by loading and unloading the drivers. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 298761a26180..6803d39e09a5 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -500,8 +500,8 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component) return; } - snd_hdac_ext_bus_link_put(hdev->bus, hlink); pm_runtime_disable(&hdev->dev); + snd_hdac_ext_bus_link_put(hdev->bus, hlink); } static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = { -- 2.20.1

2 1

[alsa-devel] Applied "ASoC: ux500: Remove redundant variable "status"" to the asoc tree
by Mark Brown 04 Nov '19

04 Nov '19

The patch ASoC: ux500: Remove redundant variable "status" has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 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 e0859710516c98b189879966b48ea1c77e0cd979 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Thu, 31 Oct 2019 21:34:15 +0800 Subject: [PATCH] ASoC: ux500: Remove redundant variable "status" local variable "status" is not used. hence it is safe to remove and just return 0. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Link: https://lore.kernel.org/r/1572528855-25990-1-git-send-email-zhongjiang@huaw… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/ux500/ux500_msp_i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index a90e0d7f0b73..394d8b2a4a16 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -533,7 +533,6 @@ static void disable_msp_tx(struct ux500_msp *msp) static int disable_msp(struct ux500_msp *msp, unsigned int dir) { u32 reg_val_GCR; - int status = 0; unsigned int disable_tx, disable_rx; reg_val_GCR = readl(msp->registers + MSP_GCR); @@ -566,7 +565,7 @@ static int disable_msp(struct ux500_msp *msp, unsigned int dir) else if (disable_rx) disable_msp_rx(msp); - return status; + return 0; } int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) -- 2.20.1

1 0

[alsa-devel] [PATCH v5 0/2] ASoC: Intel: boards: Add CML M/C for RT5682 and RT1011
by Pierre-Louis Bossart 04 Nov '19

04 Nov '19

The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Changes since v4 (from Kai Vehmanen) added prefix for series, reformatted changes fixed bad filename in Makefile Support for hdmi common codec Changes since v3 Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> repost with more recipients and edits to commit messages Changes since v2: re ordered match table to ensure non i2c device matches are at the end Changes since V1: Addressed comments on ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 Naveen Manohar (2): ASoC: Intel: Add acpi match for rt1011 based m/c driver ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 487 ++++++++++++++++++ .../intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 510 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- 2.20.1

2 4

[alsa-devel] [PATCH] dt-bindings: sound: adau7118: Fix example warning
by Maxime Ripard 04 Nov '19

04 Nov '19

The ADAU7118 has an example where the codec has an i2c address of 14, and the unit address set to 14 as well. However, while the address is expressed in decimal, the unit-address is supposed to be in hexadecimal, which ends up with two different addresses that trigger a DTC warning. Fix this by setting the unit address to (0x)e. Cc: Nuno Sá <nuno.sa(a)analog.com> Fixes: 969d49b2cdc8 ("dt-bindings: asoc: Add ADAU7118 documentation") Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Documentation/devicetree/bindings/sound/adi,adau7118.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml index c3f10afbdd6f..65f6844a0c6d 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -65,7 +65,7 @@ examples: /* example with i2c support */ #address-cells = <1>; #size-cells = <0>; - adau7118_codec: audio-codec@14 { + adau7118_codec: audio-codec@e { compatible = "adi,adau7118"; reg = <14>; #sound-dai-cells = <0>; -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: wm8524: Add support S32_LE
by S.j. Wang 04 Nov '19

04 Nov '19

Allow 32bit sample with this codec. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8524.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 91e3d1570c45..4e9ab542f648 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -159,7 +159,9 @@ static int wm8524_mute_stream(struct snd_soc_dai *dai, int mute, int stream) #define WM8524_RATES SNDRV_PCM_RATE_8000_192000 -#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8524_dai_ops = { .startup = wm8524_startup, -- 2.21.0

2 1

[alsa-devel] [PATCH] ALSA: pcm: Fix missing check of the new non-cached buffer type
by Takashi Iwai 04 Nov '19

04 Nov '19

The check for the mmap support via hw_support_mmap() function misses the case where the device is with SNDRV_DMA_TYPE_DEV_UC, which should have been treated equally as SNDRV_DMA_TYPE_DEV. Let's fix it. Note that this bug doesn't hit any practical problem, because SNDRV_DMA_TYPE_DEV_UC is used only for x86-specific drivers (snd-hda-intel and snd-intel8x0) for the specific platforms that need the non-cached buffers. And, on such platforms, hw_support_mmap() already returns true in anyway. That's the reason I didn't put Cc-to-stable mark here. This is only for any theoretical future extension. Fixes: 425da159707b ("ALSA: pcm: use dma_can_mmap() to check if a device supports dma_mmap_*") Fixes: 42e748a0b325 ("ALSA: memalloc: Add non-cached buffer type") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 91c6ad58729f..c3a139436ac2 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -222,7 +222,8 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) return false; if (substream->ops->mmap || - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV) + (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && + substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) return true; return dma_can_mmap(substream->dma_buffer.dev.dev); -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: bebob: fix to detect configured source of sampling clock for Focusrite Saffire Pro i/o series
by Takashi Sakamoto 04 Nov '19

04 Nov '19

For Focusrite Saffire Pro i/o, the lowest 8 bits of register represents configured source of sampling clock. The next lowest 8 bits represents whether the configured source is actually detected or not just after the register is changed for the source. Current implementation evaluates whole the register to detect configured source. This results in failure due to the next lowest 8 bits when the source is connected in advance. This commit fixes the bug. Fixes: 25784ec2d034 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series") Cc: <stable(a)vger.kernel.org> # v3.16+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_focusrite.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index 32b864bee25f..06d6a37cd853 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -27,6 +27,8 @@ #define SAFFIRE_CLOCK_SOURCE_SPDIF 1 /* clock sources as returned from register of Saffire Pro 10 and 26 */ +#define SAFFIREPRO_CLOCK_SOURCE_SELECT_MASK 0x000000ff +#define SAFFIREPRO_CLOCK_SOURCE_DETECT_MASK 0x0000ff00 #define SAFFIREPRO_CLOCK_SOURCE_INTERNAL 0 #define SAFFIREPRO_CLOCK_SOURCE_SKIP 1 /* never used on hardware */ #define SAFFIREPRO_CLOCK_SOURCE_SPDIF 2 @@ -189,6 +191,7 @@ saffirepro_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id) map = saffirepro_clk_maps[1]; /* In a case that this driver cannot handle the value of register. */ + value &= SAFFIREPRO_CLOCK_SOURCE_SELECT_MASK; if (value >= SAFFIREPRO_CLOCK_SOURCE_COUNT || map[value] < 0) { err = -EIO; goto end; -- 2.20.1

2 1

[alsa-devel] [GIT PULL] soundwire fixes for v5.4-rc6
by Vinod Koul 02 Nov '19

02 Nov '19

Hi Greg, Please pull to receive couple of fixes for soundwire. All these changes have been sitting in linux-next as well The following changes since commit 54ecb8f7028c5eb3d740bb82b0f1d90f2df63c5c: Linux 5.4-rc1 (2019-09-30 10:35:40 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.4-rc6 for you to fetch changes up to 7b47ad3322d114736e08cc244a84fe7af9b861f6: soundwire: slave: fix scanf format (2019-10-24 16:55:45 +0530) ---------------------------------------------------------------- soundwire fixes for v5.4-rc6 - Kconfig fixes to ensure soundwire is built only for ACPI and DT platform - fix for intel PDI offsets and numbers - slave scanf format fix ---------------------------------------------------------------- Bard Liao (1): soundwire: intel: fix intel_register_dai PDI offsets and numbers Michal Suchanek (2): soundwire: depend on ACPI soundwire: depend on ACPI || OF Pierre-Louis Bossart (1): soundwire: slave: fix scanf format drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 4 ++-- drivers/soundwire/slave.c | 3 ++- 3 files changed, 5 insertions(+), 3 deletions(-) Thanks -- ~Vinod

2 1

[alsa-devel] Question about FE & BE DPCM
by Daniel Baluta 01 Nov '19

01 Nov '19

Hi Morimoto-san, I'm really confused on what exactly is Front End and Back End. Is FE a DAI or a DAI link or part of a DAI link? In simple-card.c there is simple_dai_link_of_dpcm function: struct snd_soc_dai_link_component *codecs = dai_link->codecs; » /* BE is dummy */ » » codecs->of_node»» = NULL; » » codecs->dai_name» = "snd-soc-dummy-dai"; » » codecs->name» » = "snd-soc-dummy"; » » /* FE settings */ » » dai_link->dynamic» » = 1; » » dai_link->dpcm_merged_format» = 1; So, from here it looks like BE/FE are components being part of the same link. Another thing is: asoc_simple_canonicalize_platform function where it does say: » * DPCM BE can be no platform. » * Alloced memory will be waste, but not leak. » */ » if (!dai_link->platforms->of_node) » » dai_link->num_platforms = 0; But I do want to create a BE with a platform. So, should I try to revert the following patch: commit 794fcee8da3c0c8a01b08ecad1c181cb0a622868 Author: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed Jul 10 17:01:12 2019 +0900 ASoC: simple-card-utils: care no Platform for DPCM commit 34614739988ad ("ASoC: soc-core: support dai_link with platforms_num != 1") supports multi Platform, and commit 9f3eb91753451 ("ASoC: simple-card-utils: consider CPU-Platform possibility") removed no Platform from simple-card. Multi Platform is now checking both Platform name/of_node are NULL case. But in normal case, DPCM be doesn't have Platform. Basically, what I want to do is: after the DAI link is created to call: snd_soc_fixup_dai_links_platform_name function to set a proper name for BE platform. The problem is that above mentioned patch sets dai_link->num_platforms = 0; so even the name is set the platform component is not probed. thanks, Daniel.

3 4

[alsa-devel] [PATCH 2/5] ASoC: rt1011: improve the rt1011_set_dai_fmt() function
by shumingf＠realtek.com 31 Oct '19

31 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> If there is a wrong format setting, the driver will goto the end of the function directly. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 8a74f374d779..57a3a34b5f18 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1631,6 +1631,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1641,6 +1642,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1657,6 +1659,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (dai->id) { @@ -1674,6 +1677,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = -EINVAL; } +_set_fmt_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } -- 2.23.0

2 1

[alsa-devel] [PATCH 3/5] ASoC: rt1011: remove unnecessary tabs using spaces instead
by shumingf＠realtek.com 31 Oct '19

31 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> There are unnecessary tabs inside some statements. It uses sapces instead. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 45 +++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 57a3a34b5f18..2f1de933f3ac 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -989,7 +989,7 @@ static SOC_ENUM_SINGLE_DECL(rt1011_din_source_enum, RT1011_CROSS_BQ_SET_1, 5, static const char * const rt1011_tdm_data_out_select[] = { "TDM_O_LR", "BQ1", "DVOL", "BQ10", "ALC", "DMIX", "ADC_SRC_LR", - "ADC_O_LR", "ADC_MONO", "RSPK_BPF_LR", "DMIX_ADD", "ENVELOPE_FS", + "ADC_O_LR", "ADC_MONO", "RSPK_BPF_LR", "DMIX_ADD", "ENVELOPE_FS", "SEP_O_GAIN", "ALC_BK_GAIN", "STP_V_C", "DMIX_ABST" }; @@ -1002,7 +1002,7 @@ static SOC_ENUM_SINGLE_DECL(rt1011_tdm2_l_dac1_enum, RT1011_TDM2_SET_4, 12, rt1011_tdm_l_ch_data_select); static SOC_ENUM_SINGLE_DECL(rt1011_tdm1_adc1_dat_enum, - RT1011_ADCDAT_OUT_SOURCE, 0, rt1011_tdm_data_out_select); + RT1011_ADCDAT_OUT_SOURCE, 0, rt1011_tdm_data_out_select); static SOC_ENUM_SINGLE_DECL(rt1011_tdm1_adc1_loc_enum, RT1011_TDM1_SET_2, 0, rt1011_tdm_l_ch_data_select); @@ -1024,9 +1024,9 @@ static const char * const rt1011_tdm_adc_swap_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, +static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, rt1011_tdm_adc_swap_select); -static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, +static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, rt1011_tdm_adc_swap_select); static void rt1011_reset(struct regmap *regmap) @@ -1092,9 +1092,9 @@ static bool rt1011_validate_bq_drc_coeff(unsigned short reg) { if ((reg == RT1011_DAC_SET_1) | (reg >= RT1011_ADC_SET && reg <= RT1011_ADC_SET_1) | - (reg == RT1011_ADC_SET_4) | (reg == RT1011_ADC_SET_5) | + (reg == RT1011_ADC_SET_4) | (reg == RT1011_ADC_SET_5) | (reg == RT1011_MIXER_1) | - (reg == RT1011_A_TIMING_1) | (reg >= RT1011_POWER_7 && + (reg == RT1011_A_TIMING_1) | (reg >= RT1011_POWER_7 && reg <= RT1011_POWER_8) | (reg == RT1011_CLASS_D_POS) | (reg == RT1011_ANALOG_CTRL) | (reg >= RT1011_SPK_TEMP_PROTECT_0 && @@ -1289,7 +1289,7 @@ static int rt1011_r0_load_mode_put(struct snd_kcontrol *kcontrol, r0_integer = format / rt1011->r0_reg / 128; r0_factor = ((format / rt1011->r0_reg * 100) / 128) - (r0_integer * 100); - dev_info(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, rt1011->r0_reg); if (rt1011->r0_reg) @@ -1977,14 +1977,14 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | - RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); + RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); snd_soc_component_update_bits(component, RT1011_TDM2_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | - RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); + RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, - RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); + RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, - RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); + RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); if (tx_slotnum) snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, @@ -2109,7 +2109,7 @@ static const struct snd_soc_component_driver soc_component_dev_rt1011 = { .remove = rt1011_remove, .suspend = rt1011_suspend, .resume = rt1011_resume, - .set_bias_level = rt1011_set_bias_level, + .set_bias_level = rt1011_set_bias_level, .controls = rt1011_snd_controls, .num_controls = ARRAY_SIZE(rt1011_snd_controls), .dapm_widgets = rt1011_dapm_widgets, @@ -2118,9 +2118,9 @@ static const struct snd_soc_component_driver soc_component_dev_rt1011 = { .num_dapm_routes = ARRAY_SIZE(rt1011_dapm_routes), .set_sysclk = rt1011_set_component_sysclk, .set_pll = rt1011_set_component_pll, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1011_regmap = { @@ -2222,17 +2222,17 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_ADC_OFFSET_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "ADC offset=0x%x\n", dc_offset); + dev_info(dev, "ADC offset=0x%x\n", dc_offset); regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G0_20_16, &value); dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G0_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "Gain0 offset=0x%x\n", dc_offset); + dev_info(dev, "Gain0 offset=0x%x\n", dc_offset); regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G1_20_16, &value); dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G1_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); + dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); if (cali_flag) { @@ -2252,7 +2252,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) while (count < chk_cnt) { msleep(100); regmap_read(rt1011->regmap, - RT1011_INIT_RECIPROCAL_SYN_24_16, &value); + RT1011_INIT_RECIPROCAL_SYN_24_16, &value); r0[count%3] = value << 16; regmap_read(rt1011->regmap, RT1011_INIT_RECIPROCAL_SYN_15_0, &value); @@ -2267,7 +2267,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) break; } if (count > chk_cnt) { - dev_err(dev, "Calibrate R0 Failure\n"); + dev_err(dev, "Calibrate R0 Failure\n"); ret = -EAGAIN; } else { format = 2147483648U; /* 2^24 * 128 */ @@ -2276,7 +2276,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) - (r0_integer * 100); rt1011->r0_reg = r0[0]; rt1011->cali_done = 1; - dev_info(dev, "r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(dev, "r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, r0[0]); } } @@ -2354,7 +2354,7 @@ static void rt1011_calibration_work(struct work_struct *work) r0_integer = format / rt1011->r0_reg / 128; r0_factor = ((format / rt1011->r0_reg * 100) / 128) - (r0_integer * 100); - dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, rt1011->r0_reg); rt1011_r0_load(rt1011); @@ -2420,7 +2420,6 @@ static void rt1011_i2c_shutdown(struct i2c_client *client) rt1011_reset(rt1011->regmap); } - static struct i2c_driver rt1011_i2c_driver = { .driver = { .name = "rt1011", -- 2.23.0

2 1

[alsa-devel] [PATCH 4/5] ASoC: rt1011: add the range check for temperature_calib from device property
by shumingf＠realtek.com 31 Oct '19

31 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> The driver will check the range for temperature_calib. It should be from 1 to 255. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 2f1de933f3ac..f2c581b66d32 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2341,7 +2341,8 @@ static void rt1011_calibration_work(struct work_struct *work) rt1011_reg_init(component); /* Apply temperature and calibration data from device property */ - if (rt1011->temperature_calib) { + if (rt1011->temperature_calib <= 0xff && + rt1011->temperature_calib > 0) { snd_soc_component_update_bits(component, RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, (rt1011->temperature_calib << 2)); -- 2.23.0

2 1

[alsa-devel] [PATCH 1/5] ASoC: rt1011: remove redundant code in kcontrol
by shumingf＠realtek.com 31 Oct '19

31 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> The !component->card->instantiated statement should remove in kcontrol. It is no need to check the card->instantiated in kcontrol. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index dcd397a83cb4..8a74f374d779 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1163,9 +1163,6 @@ static int rt1011_bq_drc_coeff_put(struct snd_kcontrol *kcontrol, (struct rt1011_bq_drc_params *)ucontrol->value.integer.value; unsigned int i, mode_idx = 0; - if (!component->card->instantiated) - return 0; - if (strstr(ucontrol->id.name, "AdvanceMode Initial Set")) mode_idx = RT1011_ADVMODE_INITIAL_SET; else if (strstr(ucontrol->id.name, "AdvanceMode SEP BQ Coeff")) @@ -1236,9 +1233,6 @@ static int rt1011_r0_cali_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - if (!component->card->instantiated) - return 0; - rt1011->cali_done = 0; if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF && ucontrol->value.integer.value[0]) @@ -1284,9 +1278,6 @@ static int rt1011_r0_load_mode_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0] == rt1011->r0_reg) return 0; - if (!component->card->instantiated) - return 0; - if (ucontrol->value.integer.value[0] == 0) return -EINVAL; -- 2.23.0

2 1

[alsa-devel] [PATCH 5/5] ASoC: rt1011: some minor changes to improve readability
by shumingf＠realtek.com 31 Oct '19

31 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> There is no other code use the RT1011_INIT_REG_LEN definition, except rt1011_reg_init(). Hence, we remove it and fix the typo. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index f2c581b66d32..2552073e54ce 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -61,7 +61,6 @@ static const struct reg_sequence init_list[] = { { RT1011_DAC_SET_1, 0xe702 }, { RT1011_DAC_SET_3, 0x2004 }, }; -#define RT1011_INIT_REG_LEN ARRAY_SIZE(init_list) static const struct reg_default rt1011_reg[] = { {0x0000, 0x0000}, @@ -684,7 +683,8 @@ static int rt1011_reg_init(struct snd_soc_component *component) { struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - regmap_multi_reg_write(rt1011->regmap, init_list, RT1011_INIT_REG_LEN); + regmap_multi_reg_write(rt1011->regmap, + init_list, ARRAY_SIZE(init_list)); return 0; } @@ -1773,7 +1773,8 @@ static int rt1011_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); return ret; } @@ -2384,7 +2385,7 @@ static int rt1011_i2c_probe(struct i2c_client *i2c, rt1011 = devm_kzalloc(&i2c->dev, sizeof(struct rt1011_priv), GFP_KERNEL); - if (rt1011 == NULL) + if (!rt1011) return -ENOMEM; i2c_set_clientdata(i2c, rt1011); -- 2.23.0

2 1

[alsa-devel] [PATCH v9 0/6] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 31 Oct '19

31 Oct '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c commit 6fa5963c37a2 ("ASoC: hdmi-codec: Add an op to set callback function for plug event") has been merged to upstream. Changes from v8 to v9: 1. rockchip_max98090: Use the presence of rockchip,audio-codec to determine the presense of max98090 in sound card. Use the presence of rockchip,hdmi-codec to determine the presence of HDMI in sound card. Remove the compatible strings added in v8. 2. #include <sound/hdmi-codec.h> should be in the patch of adding HDMI jack support. Cheng-Yi Chiang (6): drm: bridge: dw-hdmi: Report connector status using callback ASoC: rockchip-max98090: Support usage with and without HDMI ASoC: rockchip_max98090: Optionally support HDMI use case ASoC: rockchip_max98090: Add HDMI jack support ARM: dts: rockchip: Add HDMI support to rk3288-veyron-analog-audio ARM: dts: rockchip: Add HDMI audio support to rk3288-veyron-mickey.dts .../bindings/sound/rockchip-max98090.txt | 27 +- .../boot/dts/rk3288-veyron-analog-audio.dtsi | 1 + arch/arm/boot/dts/rk3288-veyron-mickey.dts | 7 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 11 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rockchip_max98090.c | 313 ++++++++++++++---- 8 files changed, 338 insertions(+), 69 deletions(-) -- 2.24.0.rc0.303.g954a862665-goog

4 14

[alsa-devel] sound/core/timer: Deadlock on register_mutex
by Kirill A. Shutemov 30 Oct '19

30 Oct '19

Hi, I've stepped on this after pulling USB sound card: ============================================ WARNING: possible recursive locking detected 5.4.0-rc4-00090-g95b5dc072cc3-dirty #48 Not tainted -------------------------------------------- xdg-screensaver/1321 is trying to acquire lock: ffffffffbaf6b3a0 (register_mutex){+.+.}, at: snd_timer_free.part.0 (/include/linux/compiler.h:199 /include/linux/list.h:268 /sound/core/timer.c:944) but task is already holding lock: ffffffffbaf6b3a0 (register_mutex){+.+.}, at: snd_timer_close (/sound/core/timer.c:416) other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(register_mutex); lock(register_mutex); *** DEADLOCK *** May be due to missing lock nesting notation 2 locks held by xdg-screensaver/1321: #0: ffff9f74bbf5ef50 (&tu->ioctl_lock){+.+.}, at: snd_timer_user_release (/sound/core/timer.c:1467) #1: ffffffffbaf6b3a0 (register_mutex){+.+.}, at: snd_timer_close (/sound/core/timer.c:416) stack backtrace: CPU: 27 PID: 1321 Comm: xdg-screensaver Not tainted 5.4.0-rc4-00090-g95b5dc072cc3-dirty #48 Hardware name: Gigabyte Technology Co., Ltd. X299 AORUS Gaming 3 Pro/X299 AORUS Gaming 3 Pro-CF, BIOS F3 12/28/2017 Call Trace: dump_stack (/lib/dump_stack.c:115) __lock_acquire.cold (/kernel/locking/lockdep.c:2371 /kernel/locking/lockdep.c:2412 /kernel/locking/lockdep.c:2955 /kernel/locking/lockdep.c:3955) ? __lock_acquire (/kernel/locking/lockdep.c:3962) lock_acquire (/arch/x86/include/asm/current.h:15 /kernel/locking/lockdep.c:4489) ? snd_timer_free.part.0 (/include/linux/compiler.h:199 /include/linux/list.h:268 /sound/core/timer.c:944) __mutex_lock (/include/linux/compiler.h:199 /arch/x86/include/asm/atomic64_64.h:22 /include/asm-generic/atomic-instrumented.h:837 /include/asm-generic/atomic-long.h:28 /kernel/locking/mutex.c:111 /kernel/locking/mutex.c:152 /kernel/locking/mutex.c:958 /kernel/locking/mutex.c:1103) ? snd_timer_free.part.0 (/include/linux/compiler.h:199 /include/linux/list.h:268 /sound/core/timer.c:944) ? __mutex_lock (/include/linux/compiler.h:199 /arch/x86/include/asm/atomic64_64.h:22 /include/asm-generic/atomic-instrumented.h:837 /include/asm-generic/atomic-long.h:28 /kernel/locking/mutex.c:111 /kernel/locking/mutex.c:152 /kernel/locking/mutex.c:958 /kernel/locking/mutex.c:1103) ? __mutex_lock (/arch/x86/include/asm/preempt.h:102 /kernel/locking/mutex.c:964 /kernel/locking/mutex.c:1103) ? snd_timer_free.part.0 (/include/linux/compiler.h:199 /include/linux/list.h:268 /sound/core/timer.c:944) ? snd_timer_free.part.0 (/include/linux/compiler.h:199 /include/linux/list.h:268 /sound/core/timer.c:944) ? lockdep_hardirqs_on (/kernel/locking/lockdep.c:3394 /kernel/locking/lockdep.c:3434) snd_timer_free.part.0 (/include/linux/compiler.h:199 /include/linux/list.h:268 /sound/core/timer.c:944) snd_timer_dev_free (/sound/core/timer.c:967) __snd_device_free (/sound/core/device.c:76) snd_device_free_all (/sound/core/device.c:228) release_card_device (/sound/core/init.c:471 /sound/core/init.c:140) device_release (/drivers/base/core.c:1105) kobject_put (/lib/kobject.c:697 /lib/kobject.c:722 /include/linux/kref.h:65 /lib/kobject.c:739) snd_timer_close_locked (/sound/core/timer.c:398) snd_timer_close (/sound/core/timer.c:417) snd_timer_user_release (/sound/core/timer.c:1469) __fput (/fs/file_table.c:281) task_work_run (/kernel/task_work.c:115 (discriminator 1)) exit_to_usermode_loop (/include/linux/tracehook.h:188 /arch/x86/entry/common.c:163) do_syscall_64 (/arch/x86/entry/common.c:194 /arch/x86/entry/common.c:274 /arch/x86/entry/common.c:300) entry_SYSCALL_64_after_hwframe (/arch/x86/entry/entry_64.S:177) -- Kirill A. Shutemov

2 4

[alsa-devel] wm9713 regression on pxa platforms
by Robert Jarzmik 30 Oct '19

30 Oct '19

Hi Wolfson maintainers, hi Mark, Recently, the wm9713 support broke on my platforms. I'm pretty sure that a previous harmless warning became an error preventing the probe of my driver (sound/soc/pxa/mioa701_wm9713.c) : the error is shown in [1] ("Control not supported"). The warning "ASoC: no dapm match for Mic B Source" has be there as far as I'm aware for ages, and it never prevented the probe before. Now, if I apply the ugly patch in [2], everything works as expected ... but that's definitely not the right fix. Would any of you be able to figure out a fix please ? Cheers. -- Robert [1] Kernel log [ 129.888985] pxa2xx-ac97 pxa2xx-ac97: ASoC: dai register pxa2xx-ac97 #3 [ 129.889052] pxa2xx-ac97 pxa2xx-ac97: ASoC: dynamically register DAI pxa2xx-ac97 [ 129.889102] pxa2xx-ac97 pxa2xx-ac97: ASoC: Registered DAI 'pxa2xx-ac97' [ 129.889135] pxa2xx-ac97 pxa2xx-ac97: ASoC: dynamically register DAI pxa2xx-ac97 [ 129.889176] pxa2xx-ac97 pxa2xx-ac97: ASoC: Registered DAI 'pxa2xx-ac97-aux' [ 129.889207] pxa2xx-ac97 pxa2xx-ac97: ASoC: dynamically register DAI pxa2xx-ac97 [ 129.889249] pxa2xx-ac97 pxa2xx-ac97: ASoC: Registered DAI 'pxa2xx-ac97-mic' [ 135.341942] wm9713-codec wm9713-codec: ASoC: dai register wm9713-codec #3 [ 135.342019] wm9713-codec wm9713-codec: ASoC: dynamically register DAI wm9713-codec [ 135.342077] wm9713-codec wm9713-codec: ASoC: Registered DAI 'wm9713-hifi' [ 135.342109] wm9713-codec wm9713-codec: ASoC: dynamically register DAI wm9713-codec [ 135.342162] wm9713-codec wm9713-codec: ASoC: Registered DAI 'wm9713-aux' [ 135.342193] wm9713-codec wm9713-codec: ASoC: dynamically register DAI wm9713-codec [ 135.342234] wm9713-codec wm9713-codec: ASoC: Registered DAI 'wm9713-voice' [ 139.362113] mioa701-wm9713 mioa701-wm9713: ASoC: binding AC97 [ 139.362302] mioa701-wm9713 mioa701-wm9713: ASoC: binding AC97 Aux [ 139.367170] wm9713-codec wm9713-codec: Control not supported for path Mic B Source -> [MPB] -> Mic B Pre Amp [ 139.375450] wm9713-codec wm9713-codec: ASoC: no dapm match for Mic B Source --> MPB --> Mic B Pre Amp [ 139.383448] wm9713-codec wm9713-codec: ASoC: Failed to add route Mic B Source -> MPB -> Mic B Pre Amp [ 139.393422] mioa701-wm9713 mioa701-wm9713: ASoC: failed to instantiate card -22 [ 139.403089] mioa701-wm9713: probe of mioa701-wm9713 failed with error -22 [2] Ugly test diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 6497c1ea6228..ec140ccbab0b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -630,8 +630,8 @@ static const struct snd_soc_dapm_route wm9713_audio_map[] = { {"Mic A Source", "Mic 1", "MIC1"}, {"Mic A Source", "Mic 2 A", "MIC2A"}, {"Mic A Source", "Mic 2 B", "Mic B Source"}, - {"Mic B Pre Amp", "MPB", "Mic B Source"}, - {"Mic B Source", NULL, "MIC2B"}, + /* {"Mic B Pre Amp", "MPB", "Mic B Source"}, */ + /* {"Mic B Source", NULL, "MIC2B"}, */ /* headphone capture */ {"Capture Headphone Mux", "Stereo", "Capture Mixer"},

2 2

[alsa-devel] [PATCH v3 0/2] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 30 Oct '19

30 Oct '19

Thanks for reviewing the v2 patchset. Here is new patchset addressing all the comments from v2 This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is tested on on DragonBoard DB845c based of SDM845 SoC and Lenovo YOGA C630 Laptop based on SDM850. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since v2: - Added support to set_sdw_stream Srinivas Kandagatla (2): dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller .../bindings/soundwire/qcom,sdw.txt | 167 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 935 ++++++++++++++++++ 4 files changed, 1115 insertions(+) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

4 14

[alsa-devel] pcm_lock deadlock
by Ville Syrjälä 30 Oct '19

30 Oct '19

Hi Takashi, I just got this deadlock when I tried to modprobe i915 on an ELK: [ 203.716416] ============================================ [ 203.716417] WARNING: possible recursive locking detected [ 203.716418] 5.4.0-rc5-elk+ #206 Not tainted [ 203.716419] -------------------------------------------- [ 203.716420] kworker/0:1/12 is trying to acquire lock: [ 203.716421] efb1c138 (&spec->pcm_lock){+.+.}, at: generic_hdmi_init+0x21/0x140 [snd_hda_codec_hdmi] [ 203.716426] but task is already holding lock: [ 203.716427] efb1c138 (&spec->pcm_lock){+.+.}, at: check_presence_and_report+0x67/0xb0 [snd_hda_codec_hdmi] [ 203.716430] other info that might help us debug this: [ 203.716431] Possible unsafe locking scenario: [ 203.716431] CPU0 [ 203.716432] ---- [ 203.716432] lock(&spec->pcm_lock); [ 203.716433] lock(&spec->pcm_lock); [ 203.716434] *** DEADLOCK *** [ 203.716435] May be due to missing lock nesting notation [ 203.716436] 3 locks held by kworker/0:1/12: [ 203.716436] #0: f14096a0 ((wq_completion)events){+.+.}, at: process_one_work+0x1b8/0x530 [ 203.716442] #1: f14dbf4c ((work_completion)(&bus->unsol_work)){+.+.}, at: process_one_work+0x1b8/0x530 [ 203.716444] #2: efb1c138 (&spec->pcm_lock){+.+.}, at: check_presence_and_report+0x67/0xb0 [snd_hda_codec_hdmi] [ 203.716448] stack backtrace: [ 203.716449] CPU: 0 PID: 12 Comm: kworker/0:1 Not tainted 5.4.0-rc5-elk+ #206 [ 203.716450] Hardware name: System manufacturer P5Q-EM/P5Q-EM, BIOS 2203 07/08/2009 [ 203.716457] Workqueue: events snd_hdac_bus_process_unsol_events [snd_hda_core] [ 203.716459] Call Trace: [ 203.716463] dump_stack+0x66/0x8e [ 203.716466] __lock_acquire.cold.62+0x3bf/0x3c7 [ 203.716468] ? mark_held_locks+0x3f/0x60 [ 203.716470] ? _raw_spin_unlock_irq+0x22/0x30 [ 203.716478] ? azx_rirb_get_response+0xd7/0x220 [snd_hda_codec] [ 203.716479] ? lockdep_hardirqs_on+0xec/0x1a0 [ 203.716480] ? _raw_spin_unlock_irq+0x22/0x30 [ 203.716483] ? trace_hardirqs_on+0x4a/0xf0 [ 203.716484] ? find_held_lock+0x26/0xb0 [ 203.716486] lock_acquire+0x74/0x150 [ 203.716488] ? generic_hdmi_init+0x21/0x140 [snd_hda_codec_hdmi] [ 203.716490] __mutex_lock+0x60/0x810 [ 203.716492] ? generic_hdmi_init+0x21/0x140 [snd_hda_codec_hdmi] [ 203.716496] ? snd_hdac_exec_verb+0x16/0x40 [snd_hda_core] [ 203.716499] ? codec_read+0x29/0x40 [snd_hda_core] [ 203.716501] mutex_lock_nested+0x14/0x20 [ 203.716503] ? generic_hdmi_init+0x21/0x140 [snd_hda_codec_hdmi] [ 203.716505] generic_hdmi_init+0x21/0x140 [snd_hda_codec_hdmi] [ 203.716507] generic_hdmi_resume+0x18/0x60 [snd_hda_codec_hdmi] [ 203.716512] hda_call_codec_resume+0xc2/0x130 [snd_hda_codec] [ 203.716517] hda_codec_runtime_resume+0x2a/0x60 [snd_hda_codec] [ 203.716520] __rpm_callback+0x7a/0x140 [ 203.716524] ? snd_hda_codec_device_new+0x2a0/0x2a0 [snd_hda_codec] [ 203.716529] ? snd_hda_codec_device_new+0x2a0/0x2a0 [snd_hda_codec] [ 203.716531] rpm_callback+0x1a/0x70 [ 203.716535] ? snd_hda_codec_device_new+0x2a0/0x2a0 [snd_hda_codec] [ 203.716537] rpm_resume+0x52c/0x700 [ 203.716538] ? _raw_spin_lock_irqsave+0x32/0x40 [ 203.716540] __pm_runtime_resume+0x43/0x90 [ 203.716543] snd_hdac_power_up_pm+0x4d/0x50 [snd_hda_core] [ 203.716546] hdmi_present_sense+0x34/0x340 [snd_hda_codec_hdmi] [ 203.716548] ? finish_task_switch+0x89/0x210 [ 203.716550] check_presence_and_report+0x7a/0xb0 [snd_hda_codec_hdmi] [ 203.716553] hdmi_unsol_event+0x57/0x60 [snd_hda_codec_hdmi] [ 203.716557] ? hda_codec_match+0x70/0x70 [snd_hda_codec] [ 203.716561] hda_codec_unsol_event+0x12/0x20 [snd_hda_codec] [ 203.716564] snd_hdac_bus_process_unsol_events+0x51/0x60 [snd_hda_core] [ 203.716566] process_one_work+0x230/0x530 [ 203.716567] worker_thread+0x37/0x410 [ 203.716569] kthread+0xf5/0x110 [ 203.716570] ? process_one_work+0x530/0x530 [ 203.716572] ? kthread_create_worker_on_cpu+0x20/0x20 [ 203.716574] ret_from_fork+0x2e/0x38 Looks like commit ade49db337a9 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI") introduced pcm_lock to generic_hdmi_init(). -- Ville Syrjälä Intel

2 3

[alsa-devel] [PATCH 1/3] ASoC: rt5682: improve the sensitivity of push button
by shumingf＠realtek.com 30 Oct '19

30 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> The sensitivity could improve by decreasing the HW debounce time and reduce the delay time of workequeue. This patch added a device property for HW debounce time control. We could change this value to tune the sensitivity of push button. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index bf2ee75aabb1..bc2c31734df1 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -31,6 +31,7 @@ struct rt5682_platform_data { enum rt5682_dmic1_data_pin dmic1_data_pin; enum rt5682_dmic1_clk_pin dmic1_clk_pin; enum rt5682_jd_src jd_src; + unsigned int btndet_delay; }; #endif diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index c50b75ce82e0..35dcec135c8a 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -44,6 +44,7 @@ static const struct rt5682_platform_data i2s_default_platform_data = { .dmic1_data_pin = RT5682_DMIC1_DATA_GPIO2, .dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3, .jd_src = RT5682_JD1, + .btndet_delay = 16, }; struct rt5682_priv { @@ -1026,6 +1027,18 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, RT5682_JD1_EN | RT5682_JD1_POL_NOR); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); mod_delayed_work(system_power_efficient_wq, &rt5682->jack_detect_work, msecs_to_jiffies(250)); break; @@ -2467,6 +2480,8 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) &rt5682->pdata.dmic1_clk_pin); device_property_read_u32(dev, "realtek,jd-src", &rt5682->pdata.jd_src); + device_property_read_u32(dev, "realtek,btndet-delay", + &rt5682->pdata.btndet_delay); rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); -- 2.23.0

3 3

[alsa-devel] [PATCH 2/3] ASoC: dt-bindings: rt5682: add button delay device property
by shumingf＠realtek.com 30 Oct '19

30 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> The btndet-delay device property could control the HW debounce time. It is easy to adjust the sensitivity of push button. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5682.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 312e9a129530..30e927a28369 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -27,6 +27,11 @@ Optional properties: - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. +- realtek,btndet-delay + The debounce delay for push button. + The delay time is realtek,btndet-delay value multiple of 8.192 ms. + If absent, the default is 16. + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 @@ -47,4 +52,5 @@ rt5682 { realtek,dmic1-data-pin = <1>; realtek,dmic1-clk-pin = <1>; realtek,jd-src = <1>; + realtek,btndet-delay = <16>; }; -- 2.23.0

2 2

[alsa-devel] [PATCH 3/3] ASoC: rt1011: improve readability and remove redundant code
by shumingf＠realtek.com 30 Oct '19

30 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> 1. remove !component->card->instantiated statement in kcontrol 2. goto the end of fucntion if "ret = -EINVAL" in rt1011_set_dai_fmt() 3. fix typo 4. add the range check for temperature_calib from device property 5. remove unnecessary tabs using spaces instead Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 72 ++++++++++++++++++--------------------- 1 file changed, 34 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index dcd397a83cb4..10bbdb37d689 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -61,7 +61,6 @@ static const struct reg_sequence init_list[] = { { RT1011_DAC_SET_1, 0xe702 }, { RT1011_DAC_SET_3, 0x2004 }, }; -#define RT1011_INIT_REG_LEN ARRAY_SIZE(init_list) static const struct reg_default rt1011_reg[] = { {0x0000, 0x0000}, @@ -684,7 +683,8 @@ static int rt1011_reg_init(struct snd_soc_component *component) { struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - regmap_multi_reg_write(rt1011->regmap, init_list, RT1011_INIT_REG_LEN); + regmap_multi_reg_write(rt1011->regmap, + init_list, ARRAY_SIZE(init_list)); return 0; } @@ -989,7 +989,7 @@ static SOC_ENUM_SINGLE_DECL(rt1011_din_source_enum, RT1011_CROSS_BQ_SET_1, 5, static const char * const rt1011_tdm_data_out_select[] = { "TDM_O_LR", "BQ1", "DVOL", "BQ10", "ALC", "DMIX", "ADC_SRC_LR", - "ADC_O_LR", "ADC_MONO", "RSPK_BPF_LR", "DMIX_ADD", "ENVELOPE_FS", + "ADC_O_LR", "ADC_MONO", "RSPK_BPF_LR", "DMIX_ADD", "ENVELOPE_FS", "SEP_O_GAIN", "ALC_BK_GAIN", "STP_V_C", "DMIX_ABST" }; @@ -1002,7 +1002,7 @@ static SOC_ENUM_SINGLE_DECL(rt1011_tdm2_l_dac1_enum, RT1011_TDM2_SET_4, 12, rt1011_tdm_l_ch_data_select); static SOC_ENUM_SINGLE_DECL(rt1011_tdm1_adc1_dat_enum, - RT1011_ADCDAT_OUT_SOURCE, 0, rt1011_tdm_data_out_select); + RT1011_ADCDAT_OUT_SOURCE, 0, rt1011_tdm_data_out_select); static SOC_ENUM_SINGLE_DECL(rt1011_tdm1_adc1_loc_enum, RT1011_TDM1_SET_2, 0, rt1011_tdm_l_ch_data_select); @@ -1024,9 +1024,9 @@ static const char * const rt1011_tdm_adc_swap_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, +static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, rt1011_tdm_adc_swap_select); -static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, +static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, rt1011_tdm_adc_swap_select); static void rt1011_reset(struct regmap *regmap) @@ -1092,9 +1092,9 @@ static bool rt1011_validate_bq_drc_coeff(unsigned short reg) { if ((reg == RT1011_DAC_SET_1) | (reg >= RT1011_ADC_SET && reg <= RT1011_ADC_SET_1) | - (reg == RT1011_ADC_SET_4) | (reg == RT1011_ADC_SET_5) | + (reg == RT1011_ADC_SET_4) | (reg == RT1011_ADC_SET_5) | (reg == RT1011_MIXER_1) | - (reg == RT1011_A_TIMING_1) | (reg >= RT1011_POWER_7 && + (reg == RT1011_A_TIMING_1) | (reg >= RT1011_POWER_7 && reg <= RT1011_POWER_8) | (reg == RT1011_CLASS_D_POS) | (reg == RT1011_ANALOG_CTRL) | (reg >= RT1011_SPK_TEMP_PROTECT_0 && @@ -1163,9 +1163,6 @@ static int rt1011_bq_drc_coeff_put(struct snd_kcontrol *kcontrol, (struct rt1011_bq_drc_params *)ucontrol->value.integer.value; unsigned int i, mode_idx = 0; - if (!component->card->instantiated) - return 0; - if (strstr(ucontrol->id.name, "AdvanceMode Initial Set")) mode_idx = RT1011_ADVMODE_INITIAL_SET; else if (strstr(ucontrol->id.name, "AdvanceMode SEP BQ Coeff")) @@ -1236,9 +1233,6 @@ static int rt1011_r0_cali_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - if (!component->card->instantiated) - return 0; - rt1011->cali_done = 0; if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF && ucontrol->value.integer.value[0]) @@ -1284,9 +1278,6 @@ static int rt1011_r0_load_mode_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0] == rt1011->r0_reg) return 0; - if (!component->card->instantiated) - return 0; - if (ucontrol->value.integer.value[0] == 0) return -EINVAL; @@ -1298,7 +1289,7 @@ static int rt1011_r0_load_mode_put(struct snd_kcontrol *kcontrol, r0_integer = format / rt1011->r0_reg / 128; r0_factor = ((format / rt1011->r0_reg * 100) / 128) - (r0_integer * 100); - dev_info(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, rt1011->r0_reg); if (rt1011->r0_reg) @@ -1640,6 +1631,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1650,6 +1642,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1666,6 +1659,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (dai->id) { @@ -1683,6 +1677,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = -EINVAL; } +_set_fmt_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } @@ -1778,7 +1773,8 @@ static int rt1011_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); return ret; } @@ -1982,14 +1978,14 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | - RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); + RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); snd_soc_component_update_bits(component, RT1011_TDM2_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | - RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); + RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, - RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); + RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, - RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); + RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); if (tx_slotnum) snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, @@ -2114,7 +2110,7 @@ static const struct snd_soc_component_driver soc_component_dev_rt1011 = { .remove = rt1011_remove, .suspend = rt1011_suspend, .resume = rt1011_resume, - .set_bias_level = rt1011_set_bias_level, + .set_bias_level = rt1011_set_bias_level, .controls = rt1011_snd_controls, .num_controls = ARRAY_SIZE(rt1011_snd_controls), .dapm_widgets = rt1011_dapm_widgets, @@ -2123,9 +2119,9 @@ static const struct snd_soc_component_driver soc_component_dev_rt1011 = { .num_dapm_routes = ARRAY_SIZE(rt1011_dapm_routes), .set_sysclk = rt1011_set_component_sysclk, .set_pll = rt1011_set_component_pll, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1011_regmap = { @@ -2227,17 +2223,17 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_ADC_OFFSET_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "ADC offset=0x%x\n", dc_offset); + dev_info(dev, "ADC offset=0x%x\n", dc_offset); regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G0_20_16, &value); dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G0_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "Gain0 offset=0x%x\n", dc_offset); + dev_info(dev, "Gain0 offset=0x%x\n", dc_offset); regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G1_20_16, &value); dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G1_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); + dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); if (cali_flag) { @@ -2257,7 +2253,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) while (count < chk_cnt) { msleep(100); regmap_read(rt1011->regmap, - RT1011_INIT_RECIPROCAL_SYN_24_16, &value); + RT1011_INIT_RECIPROCAL_SYN_24_16, &value); r0[count%3] = value << 16; regmap_read(rt1011->regmap, RT1011_INIT_RECIPROCAL_SYN_15_0, &value); @@ -2272,7 +2268,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) break; } if (count > chk_cnt) { - dev_err(dev, "Calibrate R0 Failure\n"); + dev_err(dev, "Calibrate R0 Failure\n"); ret = -EAGAIN; } else { format = 2147483648U; /* 2^24 * 128 */ @@ -2281,7 +2277,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) - (r0_integer * 100); rt1011->r0_reg = r0[0]; rt1011->cali_done = 1; - dev_info(dev, "r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(dev, "r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, r0[0]); } } @@ -2346,7 +2342,8 @@ static void rt1011_calibration_work(struct work_struct *work) rt1011_reg_init(component); /* Apply temperature and calibration data from device property */ - if (rt1011->temperature_calib) { + if (rt1011->temperature_calib <= 0xff && + rt1011->temperature_calib > 0) { snd_soc_component_update_bits(component, RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, (rt1011->temperature_calib << 2)); @@ -2359,7 +2356,7 @@ static void rt1011_calibration_work(struct work_struct *work) r0_integer = format / rt1011->r0_reg / 128; r0_factor = ((format / rt1011->r0_reg * 100) / 128) - (r0_integer * 100); - dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, rt1011->r0_reg); rt1011_r0_load(rt1011); @@ -2380,7 +2377,7 @@ static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) } static int rt1011_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *id) { struct rt1011_priv *rt1011; int ret; @@ -2388,7 +2385,7 @@ static int rt1011_i2c_probe(struct i2c_client *i2c, rt1011 = devm_kzalloc(&i2c->dev, sizeof(struct rt1011_priv), GFP_KERNEL); - if (rt1011 == NULL) + if (!rt1011) return -ENOMEM; i2c_set_clientdata(i2c, rt1011); @@ -2425,7 +2422,6 @@ static void rt1011_i2c_shutdown(struct i2c_client *client) rt1011_reset(rt1011->regmap); } - static struct i2c_driver rt1011_i2c_driver = { .driver = { .name = "rt1011", -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: rt5677-spi: fixup compile warning
by Kuninori Morimoto 30 Oct '19

30 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch fixup this warning LINUX/sound/soc/codecs/rt5677-spi.c: In function ‘rt5677_spi_pcm_close’: LINUX/sound/soc/codecs/rt5677-spi.c:114:30: warning: unused variable ‘rtd’ [-Wunused-variable] struct snd_soc_pcm_runtime *rtd = substream->private_data; ^~~ Fixes: a0e0d135427c ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/rt5677-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index dac1097..3aa3ea7 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -111,7 +111,6 @@ static int rt5677_spi_pcm_close( struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct rt5677_dsp *rt5677_dsp = snd_soc_component_get_drvdata(component); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: firewire-motu: Correct a typo in the clock proc string
by Takashi Iwai 30 Oct '19

30 Oct '19

Just fix a typo of "S/PDIF" in the clock name string. Fixes: 4638ec6ede08 ("ALSA: firewire-motu: add proc node to show current statuc of clock and packet formats") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/firewire/motu/motu-proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c index b47ba818343f..187f6abd878c 100644 --- a/sound/firewire/motu/motu-proc.c +++ b/sound/firewire/motu/motu-proc.c @@ -16,7 +16,7 @@ static const char *const clock_names[] = { [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT] = "S/PDIF on optical interface", [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_A] = "S/PDIF on optical interface A", [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_B] = "S/PDIF on optical interface B", - [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PCIF on coaxial interface", + [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PDIF on coaxial interface", [SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR] = "AESEBU on XLR interface", [SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC] = "Word clock on BNC interface", [SND_MOTU_CLOCK_SOURCE_SPH] = "Source packet header", -- 2.16.4

2 1

[alsa-devel] [PATCH 0/6] ALSA: firewire-motu: add support for MOTU UltraLite and source packet header clock mode
by Takashi Sakamoto 30 Oct '19

30 Oct '19

Hi, Devices in MOTU FireWire series uses source packet header (SPH) in IEC 61883-1 for each data block in CIP. The devices have a mode to use the series of SPH for generating sampling clock. This patchset supports the above mechanism. Additionally, this patchset also supports MOTU UltraLite which adopts protocol version 2. Takashi Sakamoto (6): ALSA: firewire-motu: fix wrong spelling for macro ALSA: firewire-motu: print for unknown source of sampling clock ALSA: firewire-motu: detect SPH source of sampling clock ALSA: firewire-motu: minor code refactoring for protocol version 2 ALSA: firewire-motu: code refactoring to handle model specific switch for protocol v2 ALSA: firewire-motu: add support for MOTU UltraLite sound/firewire/motu/motu-pcm.c | 3 +- sound/firewire/motu/motu-proc.c | 2 + sound/firewire/motu/motu-protocol-v2.c | 142 ++++++++++++++----------- sound/firewire/motu/motu-protocol-v3.c | 4 +- sound/firewire/motu/motu.c | 34 ++++-- sound/firewire/motu/motu.h | 6 +- 6 files changed, 113 insertions(+), 78 deletions(-) -- 2.20.1

2 7

[alsa-devel] snd-bebob : from kernel 4.13 to 5.3.19 and .20
by Jean-Paul Argudo 30 Oct '19

30 Oct '19

Hi there, I wrote 1st this here: https://github.com/takaswie/snd-firewire-improve/issues/25 But I think it was the bad place, sorry for that. I think this ML is better? My problem in one line: I upgraded my Ubuntu Disco (kernel 4.13) to Ubuntu Eoan (kernel 5.3.19) and I can report my SaffireLE / Firewire doesnt work anymore, snd-bebob reports "discontinuity of CIP" on dmesg... At startup it lights green ok, but no sound is playable, then the lights turn orange (like it is when it's not working), I hear a "relay sound" (a electric clic of a relay), then, the Saffire LE disapears from the sound menu in Ubuntu sound menu. I can see this in dmesg: [ 19.083583] snd-bebob fw1.0: Detect discontinuity of CIP: 10 50 [ 19.746665] snd-bebob fw1.0: Detect discontinuity of CIP: A0 A8 [ 19.846775] snd-bebob fw1.0: Detect discontinuity of CIP: 70 B0 [ 20.335861] snd-bebob fw1.0: Detect discontinuity of CIP: 90 D0 [ 20.856301] snd-bebob fw1.0: Detect discontinuity of CIP: 78 B8 [ 21.312770] snd-bebob fw1.0: Detect discontinuity of CIP: 30 70 [ 21.796716] snd-bebob fw1.0: Detect discontinuity of CIP: 00 40 [ 22.347664] snd-bebob fw1.0: Detect discontinuity of CIP: 00 40 [ 22.818247] snd-bebob fw1.0: Detect discontinuity of CIP: A8 E8 [ 23.306079] snd-bebob fw1.0: Detect discontinuity of CIP: 98 D8 [ 29.544652] snd-bebob fw1.0: Detect discontinuity of CIP: 10 50 [ 30.272090] snd-bebob fw1.0: Detect discontinuity of CIP: A8 E8 [ 30.776168] snd-bebob fw1.0: Detect discontinuity of CIP: 00 40 [ 31.268123] snd-bebob fw1.0: Detect discontinuity of CIP: E0 20 [ 31.766085] snd-bebob fw1.0: Detect discontinuity of CIP: E0 20 [ 32.267903] snd-bebob fw1.0: Detect discontinuity of CIP: 88 C8 [ 32.767989] snd-bebob fw1.0: Detect discontinuity of CIP: F8 38 [ 33.263808] snd-bebob fw1.0: Detect discontinuity of CIP: 78 B8 [ 33.757156] snd-bebob fw1.0: Detect discontinuity of CIP: 08 48 [ 34.258860] snd-bebob fw1.0: Detect discontinuity of CIP: 80 C0 [ 34.755571] snd-bebob fw1.0: Detect discontinuity of CIP: 40 80 [ 35.261252] snd-bebob fw1.0: Detect discontinuity of CIP: 60 A0 [ 35.753980] snd-bebob fw1.0: Detect discontinuity of CIP: 78 B8 [ 36.253547] snd-bebob fw1.0: Detect discontinuity of CIP: C0 00 [ 36.752500] snd-bebob fw1.0: Detect discontinuity of CIP: 08 48 [ 37.255203] snd-bebob fw1.0: Detect discontinuity of CIP: B0 F0 [ 277.760550] snd-bebob fw1.0: Detect discontinuity of CIP: 40 80 [ 278.488972] snd-bebob fw1.0: Detect discontinuity of CIP: C0 00 [ 278.986290] snd-bebob fw1.0: Detect discontinuity of CIP: E0 20 [ 279.485998] snd-bebob fw1.0: Detect discontinuity of CIP: 18 58 [ 279.983838] snd-bebob fw1.0: Detect discontinuity of CIP: D0 10 [ 280.486776] snd-bebob fw1.0: Detect discontinuity of CIP: C0 00 [ 280.979487] snd-bebob fw1.0: Detect discontinuity of CIP: 30 70 [ 281.480952] snd-bebob fw1.0: Detect discontinuity of CIP: 78 B8 [ 281.975413] snd-bebob fw1.0: Detect discontinuity of CIP: 38 78 [ 282.475487] snd-bebob fw1.0: Detect discontinuity of CIP: 60 A0 [ 282.979446] snd-bebob fw1.0: Detect discontinuity of CIP: 88 C8 [ 283.472403] snd-bebob fw1.0: Detect discontinuity of CIP: 88 C8 [ 283.968484] snd-bebob fw1.0: Detect discontinuity of CIP: 68 A8 [ 284.471585] snd-bebob fw1.0: Detect discontinuity of CIP: 30 70 [ 284.562886] usb 3-4.1: reset high-speed USB device number 5 using xhci_hcd [ 284.965508] snd-bebob fw1.0: Detect discontinuity of CIP: D0 10 [ 285.469348] snd-bebob fw1.0: Detect discontinuity of CIP: 68 A8 [ 285.965174] snd-bebob fw1.0: Detect discontinuity of CIP: 20 60 [ 285.981618] firewire_core 0000:05:01.0: phy config: new root=ffc1, gap_count=5 [ 290.103982] firewire_core 0000:05:01.0: phy config: new root=ffc1, gap_count=5 I tried update the kernel with eoan-proposed I know run 5.3.20 and I have the same problem (the trace above is made with 5.3.20. First, I hope I report this the best way, really not sure here. If not, please tell me what to bring (files? logs?) and where (on this list? elsewhere?). I have spare time to investigate anything that could help here. Thanks! -- Jean-Paul Argudo

2 2

Re: [alsa-devel] [PATCH 2/2] ASoC: fsl_asrc: Add support for imx8qm
by S.j. Wang 30 Oct '19

30 Oct '19

Hi > > On Tue, Oct 29, 2019 at 05:17:09PM +0800, Shengjiu Wang wrote: > > There are two asrc module in imx8qm, each module has different clock > > configuration, and the DMA type is EDMA. > > > > So in this patch, we define the new clocks, refine the clock map, and > > include struct fsl_asrc_soc_data for different soc usage. > > > > The EDMA channel is fixed with each dma request, one dma request > > corresponding to one dma channel. So we need to request dma channel > > with dma request of asrc module. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 91 +++++++++++++++++++++++++++++------- > > sound/soc/fsl/fsl_asrc.h | 65 +++++++++++++++++++++++++- > > sound/soc/fsl/fsl_asrc_dma.c | 39 ++++++++++++---- > > 3 files changed, 167 insertions(+), 28 deletions(-) > > > diff --git a/sound/soc/fsl/fsl_asrc_dma.c > > b/sound/soc/fsl/fsl_asrc_dma.c index d6146de9acd2..dbb07a486504 > 100644 > > --- a/sound/soc/fsl/fsl_asrc_dma.c > > +++ b/sound/soc/fsl/fsl_asrc_dma.c > > @@ -199,19 +199,40 @@ static int fsl_asrc_dma_hw_params(struct > > snd_soc_component *component, > > > > /* Get DMA request of Back-End */ > > tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); > > - tmp_data = tmp_chan->private; > > - pair->dma_data.dma_request = tmp_data->dma_request; > > - dma_release_channel(tmp_chan); > > + /* tmp_chan may be NULL for it is already allocated by Back-End */ > > + if (tmp_chan) { > > + tmp_data = tmp_chan->private; > > + if (tmp_data) > > + pair->dma_data.dma_request = > > + tmp_data->dma_request; > > If this patch is supposed to add a !tmp_chan case for EDMA, we probably > shouldn't mute the !tmp_data case because dma_request will be NULL, > although the code previously didn't have a check either. I mean we might > need to error-out for !tmp_chan. Or... > is this intentional? > Yes, intentional. May be we can change to if (!asrc_priv->soc->use_edma) { /* Get DMA request of Back-End */ tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); tmp_data = tmp_chan->private; pair->dma_data.dma_request = tmp_data->dma_request; dma_release_channel(tmp_chan); /* Get DMA request of Front-End */ tmp_chan = fsl_asrc_get_dma_channel(pair, dir); tmp_data = tmp_chan->private; pair->dma_data.dma_request2 = tmp_data->dma_request; pair->dma_data.peripheral_type = tmp_data->peripheral_type; pair->dma_data.priority = tmp_data->priority; dma_release_channel(tmp_chan); } Best regards Wang shengjiu

2 1

[alsa-devel] [PATCH v1] ALSA: usb-audio: sound: usb: usb true/false for bool return type
by Saurav Girepunje 30 Oct '19

30 Oct '19

Use true/false for bool type return in uac_clock_source_is_valid(). Signed-off-by: Saurav Girepunje <saurav.girepunje(a)gmail.com> --- Changes in v1: - Update the patch subject suggested by Takashi Iwai <tiwai(a)suse.de> - Drop "!!" but keep the ternary operator in last return statement so that function will return always true/false. sound/usb/clock.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 72e9bdf76115..8d5538063598 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -167,21 +167,21 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = le32_to_cpu(cs_desc->bmControls); } else { /* UAC_VERSION_1/2 */ struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = cs_desc->bmControls; } /* If a clock source can't tell us whether it's valid, we assume it is */ if (!uac_v2v3_control_is_readable(bmControls, UAC2_CS_CONTROL_CLOCK_VALID)) - return 1; + return true; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, @@ -193,10 +193,10 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, dev_warn(&dev->dev, "%s(): cannot get clock validity for id %d\n", __func__, source_id); - return 0; + return false; } - return !!data; + return data ? true : false; } static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, -- 2.20.1

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[alsa-devel] axfer: break transmission of audio data frame when transmission backend queues no PCM event within timeout
by GitHub pull_request - opened 30 Oct '19

30 Oct '19

alsa-project/alsa-utils pull request #31 was opened from takaswie: Axfer internally has the waiter abstraction for polling system call. This is used without explicit timeout for IRQ-based scheduling model. As a result, axfer process is corrupted to be blocked due to a call of polling system call when transmission backend has a bug to queue PCM event in I/O thread or hardware IRQ context. This patchset implements error reporting for such buggy backend. When call of polling system call returns with timeout, axfer reports it as a bug. The timeout is equivalent to the size of buffer. Precisely it should be the size of period, however the larger size is used to stabilize runtime for existent PCM plugins. ``` Takashi Sakamoto (6): axfer: return the number of file descriptors for I/O events from select(2) waiter axfer: fix to return error code when a call of select(2) fails axfer: return ETIMEDOUT when no event occurs after waiter expiration axfer: handle -ETIMEDOUT before handle mmap I/O operation axfer: code refactoring for a helper function to wait for avail buffer space axfer: handle -ETIMEDOUT before non-block I/O operation axfer/waiter-select.c | 4 +-- axfer/xfer-libasound-irq-mmap.c | 19 +++++++++- axfer/xfer-libasound-irq-rw.c | 62 +++++++++++++++++++-------------- axfer/xfer-libasound.c | 21 +++++++---- 4 files changed, 70 insertions(+), 36 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/31 Patch URL : https://github.com/alsa-project/alsa-utils/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm
by Shengjiu Wang 30 Oct '19

30 Oct '19

In order to support the two asrc modules in imx8qm, we need to add compatible string "fsl,imx8qm-asrc0" and "fsl,imx8qm-asrc1" Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,asrc.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..cd2bd3daa7e1 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,8 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Contains "fsl,imx35-asrc", "fsl,imx53-asrc", + "fsl,imx8qm-asrc0" or "fsl,imx8qm-asrc1". - reg : Offset and length of the register set for the device. -- 2.21.0

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[alsa-devel] Adding support for Android's Wired Audio Headset Specification v1.1
by Mads Lønsethagen 29 Oct '19

29 Oct '19

Posting this to alsa-devel without being actually sure it belongs here, so please do tell if it belongs elsewhere. But here goes: If you have e.g. a Bose QC25 headset (wired, not bluetooth) in Linux, you can't get the volume buttons nor the play/pause button on the headphone cable itself to work. As far as I know, these buttons works in both Android and Windows 10, so it would be nice if they would work with plain Linux too. This StackExchange thread discusses how it works: https://electronics.stackexchange.com/questions/242883/how-do-media-buttons… As one poster on StackExchange mentions, Android has a defined standard on these kind of buttons: https://source.android.com/devices/accessories/headset/plug-headset-spec Since these kind of button presses must be decoded from impedance changes in the 3.5mm jack connector, does it belong in alsa somewhere? Bose QC25 buttons works on Intel HDA cards on new Dell XPS laptops, so I'm guessing it should be possible within snd-hda-intel... - Mads

1 0

[alsa-devel] [PATCH v8 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 29 Oct '19

29 Oct '19

Hey, here's the 8th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. Intended to go in via Mark's ASoC tree. v8 changes: - rebased on top of Mark's for-next (50484b6a523a) - addressed Takashi's review comment to patch 5 - addressed two sparse warnings in patch 3 reported by kbuild test robot <lkp(a)intel.com> - fixed itnull.cocci warning in patch 3 reported by Julia Lawall <julia.lawall(a)lip6.fr> and kbuild test robot <lkp(a)intel.com> Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 114 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 +- sound/soc/intel/boards/Makefile | 10 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.c | 85 +++++++++++++ sound/soc/intel/boards/hda_dsp_common.h | 32 +++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 6 + sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 19 files changed, 367 insertions(+), 31 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.c create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

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[alsa-devel] [PATCH 00/26] ASoC: SOF: enable S0ix support for Intel platforms
by Pierre-Louis Bossart 29 Oct '19

29 Oct '19

In S0ix (aka S0 Idle), the audio DSP can enter the D0i3 substate and keep some of the pipelines and processing active for e.g. Hotwording and other acoustic event detection while optimizing power consumption. This patchset mainly adds flags and state machines to track which streams may remain active (tagged as such in the topology file), and keep them active when the ALSA core tries to suspend everything. The Intel specific parts are limited to the changes in the IPC, where instead of using an SRAM window, the IPC commands needed to enter and exit D0i3 are sent in a compressed form using IPC registers themselves. This patchset is made of 26 patches, which is rather large, but each change is rather small and contained. The code was initially reviewed on GitHub, with 165+ comments provided and no regressions found [1]. The S0ix mode was tested on a CometLake platform. Future improvements will consist in an additional idle detection so that the D0i3 mode can be entered even on devices where S0ix is not enabled. [1] https://github.com/thesofproject/linux/pull/1337 Keyon Jie (26): ASoC: SOF: add a field to store the current D0 substate of DSP ASoC: SOF: reset default d0_substate at probe() and resume() ASoC: SOF: add set_power_state() to dsp_ops for power state update ASoC: SOF: Intel: hda-dsp: Add helper for setting DSP D0ix substate ASoC: SOF: Intel: CNL: add set_power_state() ops ASoC: SOF: Intel: APL: add set_power_state() ops ASoC: SOF: add flag to snd_sof_pcm_stream for D0i3 compatible stream ASoC: SOF: token: add tokens for PCM compatible with D0i3 substate ASoC: SOF: topology: parse and store d0i3_compatible flag ASoC: SOF: Intel: hda-dsp: align the comments for D0I3C update ASoC: SOF: Intel: HDA: use macro for register polling retry count ASoC: SOF: PM: rename sof_send_pm_ipc to sof_send_pm_ctx_ipc ASoC: SOF: ipc: introduce message for DSP power gating ASoC: SOF: Intel: hda-ipc: Don't read mailbox for PM_GATE reply ASoC: SOF: Intel: HDA: add cAVS specific compact IPC header file ASoC: SOF: configure D0ix IPC flags in set_power_state ASoC: SOF: PM: add helpers for setting D0 substate for ADSP ASoC: SOF: Intel: CNL: add support for sending compact IPC ASoC: SOF: add a flag to indicate the system suspend target ASoC: SOF: add a flag suspend_ignored for sof stream ASoC: SOF: PM: implement prepare/complete callbacks ASoC: SOF: ignore suspend/resume for D0ix compatible streams ASoC: SOF: Intel: hda-dsp: implement suspend/resume for S0ix<->S0 transition ASoC: SOF: return -ENOTSUPP if D0I3 is not supported ASoC: SOF: PM: Add support for DSP D0i3 state when entering S0ix ASoC: SOF: pci: Add prepare/complete PM callbacks include/sound/sof/header.h | 1 + include/sound/sof/pm.h | 8 +++ include/uapi/sound/sof/tokens.h | 4 ++ sound/soc/sof/core.c | 3 ++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 43 +++++++++++++-- sound/soc/sof/intel/hda-dsp.c | 91 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 6 ++- sound/soc/sof/intel/hda-ipc.h | 51 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 12 +++++ sound/soc/sof/ops.h | 10 ++++ sound/soc/sof/pcm.c | 31 +++++++++++ sound/soc/sof/pm.c | 95 +++++++++++++++++++++++++++++++-- sound/soc/sof/sof-pci-dev.c | 2 + sound/soc/sof/sof-priv.h | 23 ++++++++ sound/soc/sof/topology.c | 43 ++++++++++++++- 16 files changed, 413 insertions(+), 11 deletions(-) create mode 100644 sound/soc/sof/intel/hda-ipc.h -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: bxt_da7219_max98357a: update dai_link platform name
by mac.chiang＠intel.com 29 Oct '19

29 Oct '19

From: Mac Chiang <mac.chiang(a)intel.com> redefine soc platform name "0000:00:1f.3" if cml core Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index ac1dea5..38b7dbe 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -40,6 +40,13 @@ struct bxt_card_private { struct list_head hdmi_pcm_list; }; +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + enum { BXT_DPCM_AUDIO_PB = 0, BXT_DPCM_AUDIO_CP, @@ -708,6 +715,7 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_dais[i].name = "SSP0-Codec"; broxton_dais[i].cpus->dai_name = "SSP0 Pin"; } + broxton_dais[i].platforms = platform_component; } } -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: intel - fix the card names
by Jaroslav Kysela 29 Oct '19

29 Oct '19

Those strings are exposed to the user space as the card name thus used in the GUIs. The common standard is to avoid '_' here. The worst case is 'sof-skl_hda_card' string. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a..d14ca327c684 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -623,7 +623,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) * kabylake audio machine driver for MAX98927 + RT5514 + RT5663 */ static struct snd_soc_card kabylake_audio_card = { - .name = "kbl_r5514_5663_max", + .name = "kbl-r5514-5663-max", .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 1778acdc367c..e8d676c192f6 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -90,7 +90,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) } static struct snd_soc_card hda_soc_card = { - .name = "skl_hda_card", + .name = "hda-dsp", .owner = THIS_MODULE, .dai_link = skl_hda_be_dai_links, .dapm_widgets = skl_hda_widgets, diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a..320a9b9273db 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -370,7 +370,7 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { - .name = "sof_rt5682", + .name = "rt5682", /* the sof- prefix is added by the core */ .owner = THIS_MODULE, .controls = sof_controls, .num_controls = ARRAY_SIZE(sof_controls), -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: SOF - remove the dead code (skylake/kabylake)
by Jaroslav Kysela 29 Oct '19

29 Oct '19

Appearently the CONFIG_SND_SOC_SOF_KABYLAKE and CONFIG_SND_SOC_SOF_SKYLAKE options are not present in Kconfig and 'struct snd_sof_dsp_ops sof_skl_ops' is not declared in the code, too. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.h | 1 - sound/soc/sof/sof-pci-dev.c | 44 ------------------------------------- 2 files changed, 45 deletions(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 23e430d3e056..f521c48ec2eb 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -596,7 +596,6 @@ extern struct snd_soc_dai_driver skl_dai[]; */ extern const struct snd_sof_dsp_ops sof_apl_ops; extern const struct snd_sof_dsp_ops sof_cnl_ops; -extern const struct snd_sof_dsp_ops sof_skl_ops; extern const struct sof_intel_dsp_desc apl_chip_info; extern const struct sof_intel_dsp_desc cnl_chip_info; diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 3a9e0e2a150d..bca3fb791a6b 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -168,42 +168,6 @@ static const struct sof_dev_desc icl_desc = { }; #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) -static const struct sof_dev_desc skl_desc = { - .machines = snd_soc_acpi_intel_skl_machines, - .resindex_lpe_base = 0, - .resindex_pcicfg_base = -1, - .resindex_imr_base = -1, - .irqindex_host_ipc = -1, - .resindex_dma_base = -1, - .chip_info = &skl_chip_info, - .default_fw_path = "intel/sof", - .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-skl.ri", - .nocodec_tplg_filename = "sof-skl-nocodec.tplg", - .ops = &sof_skl_ops, - .arch_ops = &sof_xtensa_arch_ops -}; -#endif - -#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) -static const struct sof_dev_desc kbl_desc = { - .machines = snd_soc_acpi_intel_kbl_machines, - .resindex_lpe_base = 0, - .resindex_pcicfg_base = -1, - .resindex_imr_base = -1, - .irqindex_host_ipc = -1, - .resindex_dma_base = -1, - .chip_info = &skl_chip_info, - .default_fw_path = "intel/sof", - .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-kbl.ri", - .nocodec_tplg_filename = "sof-kbl-nocodec.tplg", - .ops = &sof_skl_ops, - .arch_ops = &sof_xtensa_arch_ops -}; -#endif - #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) static const struct sof_dev_desc tgl_desc = { .machines = snd_soc_acpi_intel_tgl_machines, @@ -407,14 +371,6 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0xa348), .driver_data = (unsigned long)&cfl_desc}, #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) - { PCI_DEVICE(0x8086, 0x9d71), - .driver_data = (unsigned long)&kbl_desc}, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) - { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = (unsigned long)&skl_desc}, -#endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) { PCI_DEVICE(0x8086, 0x34C8), .driver_data = (unsigned long)&icl_desc}, -- 2.20.1

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[alsa-devel] [ANNOUNCE] Draft schedule for 2019 Linux Audio miniconference
by Mark Brown 29 Oct '19

29 Oct '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue) and will be free of charge to attend, generously sponsored by Intel. The agenda is looking pretty full now, what we've got so far is: - Use case management enhancements (Curtis) - DSP framework integration (Liam) - SOF integration with ACPI and DT (Daniel) - SOF support for non-modular drivers (Daniel) - Gapless compressed playback (Patrick) - Soundwire status (Liam?) - Multi-CPU support - Mixing with HDA/I2S in a single system - PCM issues (Patrick) - PCM device volume control - Attaching timestamps to buffers - Virtualization (Liam) - User experiences (Jerome) - Unified graph (Patrick) - Componentisation status/plans (Me) If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

2 2

[alsa-devel] [PATCH] usb: clock.c : usb true/false for bool return type
by Saurav Girepunje 29 Oct '19

29 Oct '19

Use true/false for bool type return in uac_clock_source_is_valid(). Signed-off-by: Saurav Girepunje <saurav.girepunje(a)gmail.com> --- sound/usb/clock.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 6b8c14f9b5d4..8b8ab83fac0d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -165,21 +165,21 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = le32_to_cpu(cs_desc->bmControls); } else { /* UAC_VERSION_1/2 */ struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = cs_desc->bmControls; } /* If a clock source can't tell us whether it's valid, we assume it is */ if (!uac_v2v3_control_is_readable(bmControls, UAC2_CS_CONTROL_CLOCK_VALID)) - return 1; + return true; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, @@ -191,10 +191,10 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, dev_warn(&dev->dev, "%s(): cannot get clock validity for id %d\n", __func__, source_id); - return 0; + return false; } - return !!data; + return !!data ? true : false; } static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, -- 2.20.1

2 1

[alsa-devel] [PATCH 00/21] Enable BACO for power savings (v2)
by Alex Deucher 29 Oct '19

29 Oct '19

This patch set enables BACO (Bus Active Chip Off) for power savings on VI+ asics. Similar to PowerXpress and Hybrid Graphics (PX/HG) laptops, we can disable GPUs at runtime when they are not in use is they support BACO. The runtime pm code in amdgpu was originally developed for PX/HG laptops, so it was pretty entangled with the vga_switcheroo and ACPI code. Since the GPU contains an audio codec for HDMI/DP, there is some interaction with the hda driver. I am by no means an expert on alsa, so any advice on those patches is much appreciated. I had to enable runtime pm on the hda device to allow the GPU to enter runtime pm because they are linked. These patches depend on the clean patches and CI/VI BACO patches I sent out earlier today. The full tree can be found here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=baco TODO: - Turn off runtime pm when KFD is active - Make sure audio still works - Make sure PX/HG still works v2: - Split whitespace changes from baco callback addition - Fix pci ref counting in px/hg handling in hda driver - Handle CONFIG_ACPI properly - Split adding new hda pci ids from adding runpm flag Alex Deucher (21): drm/amdgpu: add asic callback for BACO support drm/amdgpu/soc15: move struct definition around to align with other soc15 asics drm/amdgpu: add supports_baco callback for soc15 asics. (v2) drm/amdgpu: add supports_baco callback for SI asics. drm/amdgpu: add supports_baco callback for CIK asics. drm/amdgpu: add supports_baco callback for VI asics. drm/amdgpu: add supports_baco callback for NV asics. drm/amdgpu: add a amdgpu_device_supports_baco helper drm/amdgpu: rename amdgpu_device_is_px to amdgpu_device_supports_boco (v2) drm/amdgpu: add additional boco checks to runtime suspend/resume (v2) drm/amdgpu: split swSMU baco_reset into enter and exit drm/amdgpu: add helpers for baco entry and exit drm/amdgpu: add baco support to runtime suspend/resume drm/amdgpu: start to disentangle boco from runtime pm drm/amdgpu: disentangle runtime pm and vga_switcheroo drm/amdgpu: enable runtime pm on BACO capable boards if runpm=1 drm/amdgpu/runpm: enable runpm on baco capable VI+ asics ALSA: hda/hdmi - fix vgaswitcheroo detection for AMD ALSA: hda/hdmi - Add new pci ids for AMD GPU display audio ALSA: hda/hdmi - enable runtime pm for newer AMD display audio ALSA: hda/hdmi - enable automatic runtime pm for AMD HDMI codecs by default drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 +- drivers/gpu/drm/amd/amdgpu/amdgpu_device.c | 106 ++++++++++++++++-- drivers/gpu/drm/amd/amdgpu/amdgpu_drv.c | 63 ++++++----- drivers/gpu/drm/amd/amdgpu/amdgpu_kms.c | 20 +++- drivers/gpu/drm/amd/amdgpu/cik.c | 18 +++ drivers/gpu/drm/amd/amdgpu/nv.c | 18 ++- drivers/gpu/drm/amd/amdgpu/si.c | 6 + drivers/gpu/drm/amd/amdgpu/soc15.c | 36 +++++- drivers/gpu/drm/amd/amdgpu/vi.c | 22 ++++ drivers/gpu/drm/amd/powerplay/amdgpu_smu.c | 20 +++- drivers/gpu/drm/amd/powerplay/arcturus_ppt.c | 3 +- .../gpu/drm/amd/powerplay/inc/amdgpu_smu.h | 6 +- drivers/gpu/drm/amd/powerplay/inc/smu_v11_0.h | 3 +- drivers/gpu/drm/amd/powerplay/navi10_ppt.c | 3 +- drivers/gpu/drm/amd/powerplay/smu_v11_0.c | 9 +- drivers/gpu/drm/amd/powerplay/vega20_ppt.c | 3 +- sound/pci/hda/hda_intel.c | 84 +++++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 18 files changed, 371 insertions(+), 62 deletions(-) -- 2.23.0

2 25

[alsa-devel] axfer: call of polling system call never return when driver fails to queue no period-wakeup event due to the lack of hw IRQ context
by GitHub issues - edited 29 Oct '19

29 Oct '19

alsa-project/alsa-utils issue #30 was edited from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-utils/issues/30 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] axfer: call of polling system call never return when driver fails to queue any events due to the lack of hw IRQ context
by GitHub issues - opened 29 Oct '19

29 Oct '19

alsa-project/alsa-utils issue #30 was opened from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-utils/issues/30 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH v8 0/6] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 29 Oct '19

29 Oct '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c 6fa5963c37a2 ASoC: hdmi-codec: Add an op to set callback function for plug event has been merged to upstream. Changes from v7 to v8: 1. rockchip_max98090: Allow three different use cases: max98090-only: For backward compatibility where DTS does not specify HDMI node. HDMI-only: For HDMI-only board like veyron-mickey. max98090 + HDMI: For other veyron boards. Pass different compatible string to specify the use case. 2. Add more maintainers to cc-list for new device property reviewing. Cheng-Yi Chiang (6): drm: bridge: dw-hdmi: Report connector status using callback ASoC: rockchip-max98090: Support usage with and without HDMI ASoC: rockchip_max98090: Optionally support HDMI use case ASoC: rockchip_max98090: Add HDMI jack support ARM: dts: rockchip: Add HDMI support to rk3288-veyron-analog-audio ARM: dts: rockchip: Add HDMI audio support to rk3288-veyron-mickey.dts .../bindings/sound/rockchip-max98090.txt | 38 +- .../boot/dts/rk3288-veyron-analog-audio.dtsi | 3 +- arch/arm/boot/dts/rk3288-veyron-mickey.dts | 7 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 11 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 +- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rockchip_max98090.c | 392 +++++++++++++++--- 8 files changed, 425 insertions(+), 74 deletions(-) -- 2.24.0.rc0.303.g954a862665-goog

3 12

[alsa-devel] [PATCH 0/2] Update adau7118 dt-bindings
by Nuno Sá 29 Oct '19

29 Oct '19

This is a follow-up patch as requested by Rob. The dt-bindings patch for this device was applied before Rob's review. Here is the link to the review: https://lore.kernel.org/alsa-devel/20191010210850.GA20184@bogus/ Nuno Sá (2): dt-bindings: asoc: adau7118: Cleanup ASOC: adau7118: Change regulators id .../bindings/sound/adi,adau7118.yaml | 25 ++++++++----------- sound/soc/codecs/adau7118.c | 4 +-- 2 files changed, 12 insertions(+), 17 deletions(-) -- 2.23.0

4 5

[alsa-devel] [PATCH] Revert "ALSA: hda: Flush interrupts on disabling"
by Takashi Iwai 28 Oct '19

28 Oct '19

This reverts commit caa8422d01e983782548648e125fd617cadcec3f. It turned out that this commit caused a regression at shutdown / reboot, as the synchronize_irq() calls seems blocking the whole shutdown. Also another part of the change about shuffling the call order looks suspicious; the azx_stop_chip() call disables the CORB / RIRB while the others may still need the CORB/RIRB update. Since the original commit itself was a cargo-fix, let's revert the whole patch. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205333 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_controller.c | 2 -- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index d3999e7b0705..7e7be8e4dcf9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -447,8 +447,6 @@ static void azx_int_disable(struct hdac_bus *bus) list_for_each_entry(azx_dev, &bus->stream_list, list) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0); - synchronize_irq(bus->irq); - /* disable SIE for all streams */ snd_hdac_chip_writeb(bus, INTCTL, 0); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a815bc811799..cf53fbd872ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1348,9 +1348,9 @@ static int azx_free(struct azx *chip) } if (bus->chip_init) { - azx_stop_chip(chip); azx_clear_irq_pending(chip); azx_stop_all_streams(chip); + azx_stop_chip(chip); } if (bus->irq >= 0) -- 2.16.4

3 4

[alsa-devel] [PATCH v7 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 28 Oct '19

28 Oct '19

Hey, here's the 7th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. v7 changes: - Only change since v6 is moving inlined logic from hda_dsp_common.h to hda_dsp_common.c and related changes to machine driver make rules. This is to address Takashi's feedback. - Rebased on top of Mark's tree (agreed with Takashi and Pierre to submit via Mark's ASoC tree). Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 114 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 +- sound/soc/intel/boards/Makefile | 10 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.c | 85 +++++++++++++ sound/soc/intel/boards/hda_dsp_common.h | 32 +++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 19 files changed, 370 insertions(+), 32 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.c create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

3 15

[alsa-devel] [PATCH] ASoC: intel - fix the card names
by Jaroslav Kysela 28 Oct '19

28 Oct '19

Those strings are exposed to the user space as the card name thus used in the GUIs. The common standard is to avoid '_' here. The worst case is 'sof-skl_hda_card' string. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a..d14ca327c684 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -623,7 +623,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) * kabylake audio machine driver for MAX98927 + RT5514 + RT5663 */ static struct snd_soc_card kabylake_audio_card = { - .name = "kbl_r5514_5663_max", + .name = "kbl-r5514-5663-max", .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 1778acdc367c..5261f7bf0b3b 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -90,7 +90,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) } static struct snd_soc_card hda_soc_card = { - .name = "skl_hda_card", + .name = "skl-hda", .owner = THIS_MODULE, .dai_link = skl_hda_be_dai_links, .dapm_widgets = skl_hda_widgets, diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a..fc4638b8e03b 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -370,7 +370,7 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { - .name = "sof_rt5682", + .name = "sof-rt5682", .owner = THIS_MODULE, .controls = sof_controls, .num_controls = ARRAY_SIZE(sof_controls), -- 2.20.1

2 2

[alsa-devel] [PATCH] ALSA: usb-audio: Fix memory leak in __snd_usbmidi_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of __snd_usbmidi_create() there is a memory leak caused by incorrect goto destination. Go to free_midi if snd_usbmidi_create_endpoints_midiman() or snd_usbmidi_create_endpoints() fail. Fixes: 731209cc0417 ("ALSA: usb-midi: Use common error handling code in __snd_usbmidi_create()") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b737f0ec77d0..22db37fbfbbd 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2476,7 +2476,7 @@ int __snd_usbmidi_create(struct snd_card *card, else err = snd_usbmidi_create_endpoints(umidi, endpoints); if (err < 0) - goto exit; + goto free_midi; usb_autopm_get_interface_no_resume(umidi->iface); -- 2.17.1

3 5

[alsa-devel] [PATCH] ASoC: SOF: ipc: Fix memory leak in sof_set_get_large_ctrl_data
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of sof_set_get_large_ctrl_data() there is a memory leak in case an error. Release partdata if sof_get_ctrl_copy_params() fails. Fixes: 54d198d5019d ("ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/soc/sof/ipc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index b2f359d2f7e5..086eeeab8679 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -572,8 +572,10 @@ static int sof_set_get_large_ctrl_data(struct snd_sof_dev *sdev, else err = sof_get_ctrl_copy_params(cdata->type, partdata, cdata, sparams); - if (err < 0) + if (err < 0) { + kfree(partdata); return err; + } msg_bytes = sparams->msg_bytes; pl_size = sparams->pl_size; -- 2.17.1

3 3

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda-stream: fix the CONFIG_ prefix missing
by Pierre-Louis Bossart 28 Oct '19

28 Oct '19

From: Keyon Jie <yang.jie(a)linux.intel.com> We are missing the 'CONFIG_' prefix when using the kernel configure item SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1, here correct them. Fixes: 43b2ab9009b13b ('ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture') Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 450f9c55785f..29ab43281670 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -190,7 +190,7 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction) * Workaround to address a known issue with host DMA that results * in xruns during pause/release in capture scenarios. */ - if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) if (stream && direction == SNDRV_PCM_STREAM_CAPTURE) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, @@ -228,7 +228,7 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) spin_unlock_irq(&bus->reg_lock); /* Enable DMI L1 entry if there are no capture streams open */ - if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) if (!active_capture_stream) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: SOF: Fix memory leak in sof_dfsentry_write
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of sof_dfsentry_write() memory allocated for string is leaked in case of an error. Go to error handling path if the d_name.name is not valid. Fixes: 091c12e1f50c ("ASoC: SOF: debug: add new debugfs entries for IPC flood test") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/soc/sof/debug.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 54cd431faab7..5529e8eeca46 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -152,8 +152,10 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, */ dentry = file->f_path.dentry; if (strcmp(dentry->d_name.name, "ipc_flood_count") && - strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) - return -EINVAL; + strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) { + ret = -EINVAL; + goto out; + } if (!strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) flood_duration_test = true; -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: eve: implement set_bias_level function for rt5514
by Brent Lu 28 Oct '19

28 Oct '19

The first DMIC capture always fail (zero sequence data from PCM port) after using DSP hotwording function (i.e. Google assistant). This rt5514 codec requires to control mclk directly in the set_bias_level function. Implement this function in machine driver to control the ssp1_mclk clock explicitly could fix this issue. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 50 ++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index dc09a85..b546de8 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -653,6 +653,55 @@ static struct snd_soc_dai_link kabylake_dais[] = { }, }; +static int kabylake_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + if (!component || strcmp(component->name, RT5514_DEV_NAME)) + return 0; + + if (IS_ERR(priv->mclk)) + return 0; + + /* + * It's required to control mclk directly in the set_bias_level + * function for rt5514 codec or the recording function could + * break. + */ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_ON) { + dev_dbg(card->dev, "Disable mclk"); + clk_disable_unprepare(priv->mclk); + } else { + dev_dbg(card->dev, "Enable mclk"); + ret = clk_set_rate(priv->mclk, 24000000); + if (ret) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", + ret); + + /* mclk is already enabled in FW */ + ret = 0; + } + } + break; + default: + break; + } + + return ret; +} + static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); @@ -692,6 +741,7 @@ static struct snd_soc_card kabylake_audio_card = { .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), + .set_bias_level = kabylake_set_bias_level, .controls = kabylake_controls, .num_controls = ARRAY_SIZE(kabylake_controls), .dapm_widgets = kabylake_widgets, -- 2.7.4

4 6

[alsa-devel] [PATCH -next] ASoC: sunxi: sun4i-codec: remove unneeded semicolon
by YueHaibing 28 Oct '19

28 Oct '19

remove unneeded semicolon. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sunxi/sun4i-codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index ee448d5..34f3e0b 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1442,7 +1442,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) @@ -1480,7 +1480,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) @@ -1518,7 +1518,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: fsl: fsl_dma: fix build failure
by Michael Ellerman 28 Oct '19

28 Oct '19

Commit 4ac85de9977e ("ASoC: fsl: fsl_dma: remove snd_pcm_ops") removed fsl_dma_ops but left a usage, leading to a build error for some configs, eg. mpc85xx_defconfig: sound/soc/fsl/fsl_dma.c: In function ‘fsl_soc_dma_probe’: sound/soc/fsl/fsl_dma.c:905:18: error: ‘fsl_dma_ops’ undeclared (first use in this function) dma->dai.ops = &fsl_dma_ops; ^~~~~~~~~~~ Remove the usage to fix the build. Fixes: 4ac85de9977e ("ASoC: fsl: fsl_dma: remove snd_pcm_ops") Signed-off-by: Michael Ellerman <mpe(a)ellerman.id.au> --- sound/soc/fsl/fsl_dma.c | 1 - 1 file changed, 1 deletion(-) This breakage is only in linux-next. diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index a092726510d4..2868c4f97cb2 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -901,7 +901,6 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) } dma->dai.name = DRV_NAME; - dma->dai.ops = &fsl_dma_ops; dma->dai.open = fsl_dma_open; dma->dai.close = fsl_dma_close; dma->dai.ioctl = snd_soc_pcm_lib_ioctl; -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: rockchip: rockchip_max98090: Enable SHDN to fix headset detection
by Cheng-Yi Chiang 28 Oct '19

28 Oct '19

max98090 spec states that chip needs to be in turned-on state to supply mic bias. Enable SHDN dapm widget along with MICBIAS widget to actually turn on mic bias for proper headset button detection. This is similar to cht_ti_jack_event in sound/soc/intel/boards/cht_bsw_max98090_ti.c. Note that due to ts3a227e reports the jack event right away before the notifier is registered, if headset is plugged on boot, headset button will not get detected until headset is unplugged and plugged. This is still an issue to be fixed. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/rockchip/rockchip_max98090.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 0097df1fae66..e80b09143b63 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -66,10 +66,13 @@ static int rk_jack_event(struct notifier_block *nb, unsigned long event, struct snd_soc_jack *jack = (struct snd_soc_jack *)data; struct snd_soc_dapm_context *dapm = &jack->card->dapm; - if (event & SND_JACK_MICROPHONE) + if (event & SND_JACK_MICROPHONE) { snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); - else + snd_soc_dapm_force_enable_pin(dapm, "SHDN"); + } else { snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SHDN"); + } snd_soc_dapm_sync(dapm); -- 2.24.0.rc0.303.g954a862665-goog

2 1

[alsa-devel] [PATCH] ASoC: ti: sdma-pcm: Add back the flags parameter for non standard dma names
by Peter Ujfalusi 28 Oct '19

28 Oct '19

When non standard names are used it is possible that one of the directions are not provided, thus the flags needs to be present to tell the core that we have half duplex setup. Fixes: 642aafea8889 ("ASoC: ti: remove compat dma probing") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/sdma-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index a236350beb10..2b0bc234e1b6 100644 --- a/sound/soc/ti/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c @@ -62,7 +62,7 @@ int sdma_pcm_platform_register(struct device *dev, config->chan_names[0] = txdmachan; config->chan_names[1] = rxdmachan; - return devm_snd_dmaengine_pcm_register(dev, config, 0); + return devm_snd_dmaengine_pcm_register(dev, config, flags); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: rsnd: core.c: add WARN_ON() on rsnd_channel_normalization()
by Kuninori Morimoto 28 Oct '19

28 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> chan > 8 or chan < 0 shouldn't happen. This patch uses WARN_ON() for such case. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e9242a0..295faad 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -302,7 +302,7 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, int rsnd_channel_normalization(int chan) { - if ((chan > 8) || (chan < 0)) + if (WARN_ON((chan > 8) || (chan < 0))) return 0; /* TDM Extend Mode needs 8ch */ -- 2.7.4

2 1

[alsa-devel] [PATCH V3] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 28 Oct '19

28 Oct '19

The output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitation in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v3 - refine comments Change in v2 - remove p2p/m2m word - use use_ideal_rate sound/soc/fsl/fsl_asrc.c | 45 ++++++++++++++++++++++++++++++---------- 1 file changed, 34 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..a3cfceea7d2f 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -259,8 +259,15 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * It configures those ASRC registers according to a configuration instance * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. + * + * Note: + * The ideal ratio configuration can work with a flexible clock rate setting. + * Using IDEAL_RATIO_RATE gives a faster converting speed but overloads ASRC. + * For a regular audio playback, the clock rate should not be slower than an + * clock rate aligning with the output sample rate; For a use case requiring + * faster conversion, set use_ideal_rate to have the faster speed. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +275,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,27 +359,42 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; - if (div[IN] == 0) { + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; + + /* + * The divider range is [1, 1024], defined by the hardware. For non- + * ideal ratio configuration, clock rate has to be strictly aligned + * with the sample rate. For ideal ratio configuration, clock rates + * only result in different converting speeds. So remainder does not + * matter, as long as we keep the divider within its valid range. + */ + if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; } + div[IN] = min_t(u32, 1024, div[IN]); + clk = asrc_priv->asrck_clk[clk_index[OUT]]; - - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + clk_rate = clk_get_rate(clk); + if (ideal && use_ideal_rate) + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); else - div[OUT] = clk_get_rate(clk) / outrate; + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; - if (div[OUT] == 0) { + /* Output divider has the same limitation as the input one */ + if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; } + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +583,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH V3] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Shengjiu Wang 28 Oct '19

28 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v3 -refine comments Change in v2 -add lock for fsl_esai_trigger_start. sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..453392a9cd68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock between hw_reset() and trigger() * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect hw_reset and trigger */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH] ALSA: pci: Fix memory leak in snd_korg1212_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_korg1212_create() the allocated memory for korg1212 is leaked in cases of error. Release korg1212 via snd_korg1212_free() if either of these calls fail: snd_korg1212_downloadDSPCode(), snd_pcm_new(), or snd_ctl_add(). Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/korg1212/korg1212.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 0d81eac0a478..e976e857d915 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2367,8 +2367,10 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, mdelay(CARD_BOOT_DELAY_IN_MS); - if (snd_korg1212_downloadDSPCode(korg1212)) + if (snd_korg1212_downloadDSPCode(korg1212)) { + snd_korg1212_free(korg1212); return -EBUSY; + } K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], " "PlayDataPhy = %08x L[%08x]\n" @@ -2383,8 +2385,11 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy), korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy)); - if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0) + err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm); + if (err < 0) { + snd_korg1212_free(korg1212); return err; + } korg1212->pcm->private_data = korg1212; korg1212->pcm->private_free = snd_korg1212_free_pcm; @@ -2398,8 +2403,10 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, for (i = 0; i < ARRAY_SIZE(snd_korg1212_controls); i++) { err = snd_ctl_add(korg1212->card, snd_ctl_new1(&snd_korg1212_controls[i], korg1212)); - if (err < 0) + if (err < 0) { + snd_korg1212_free(korg1212); return err; + } } snd_korg1212_proc_init(korg1212); -- 2.17.1

3 7

[alsa-devel] KASAN: slab-out-of-bounds Read in build_audio_procunit
by syzbot 28 Oct '19

28 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=103893cf600000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=0620f79a1978b1133fd7 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=16fd6a40e00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1033215f600000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+0620f79a1978b1133fd7(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: slab-out-of-bounds in uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] BUG: KASAN: slab-out-of-bounds in build_audio_procunit+0x1306/0x13f0 sound/usb/mixer.c:2436 Read of size 1 at addr ffff8881d537c7b7 by task kworker/0:2/101 CPU: 0 PID: 101 Comm: kworker/0:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] build_audio_procunit+0x1306/0x13f0 sound/usb/mixer.c:2436 parse_audio_processing_unit sound/usb/mixer.c:2468 [inline] parse_audio_unit+0x17e9/0x36f0 sound/usb/mixer.c:2757 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 101: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x314/0x3050 drivers/usb/core/config.c:857 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 17: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 urb_destroy drivers/usb/core/urb.c:26 [inline] kref_put include/linux/kref.h:65 [inline] usb_free_urb.part.0+0x7a/0xc0 drivers/usb/core/urb.c:95 usb_free_urb+0x1b/0x30 drivers/usb/core/urb.c:94 usb_start_wait_urb+0x1e5/0x2b0 drivers/usb/core/message.c:79 usb_internal_control_msg drivers/usb/core/message.c:101 [inline] usb_control_msg+0x31c/0x4a0 drivers/usb/core/message.c:152 get_port_status drivers/usb/core/hub.c:569 [inline] hub_ext_port_status+0x125/0x460 drivers/usb/core/hub.c:586 hub_port_status drivers/usb/core/hub.c:608 [inline] hub_activate+0x497/0x1570 drivers/usb/core/hub.c:1111 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 The buggy address belongs to the object at ffff8881d537c700 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 183 bytes inside of 192-byte region [ffff8881d537c700, ffff8881d537c7c0) The buggy address belongs to the page: page:ffffea000754df00 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea000754d680 0000000600000006 ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d537c680: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d537c700: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > ffff8881d537c780: 00 00 00 02 fc fc fc fc fc fc fc fc fc fc fc fc ^ ffff8881d537c800: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8881d537c880: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

3 3

[alsa-devel] [PATCH] ASoC: change 'HDMI/DP, pcm=' to 'HDMI/DP, pcm=' Jack control names
by Jaroslav Kysela 28 Oct '19

28 Oct '19

There is an inconsistency in the names for the HDMI/DP Jack control names between some ASoC drivers and the HDA HDMI driver which introduced this naming in 2011. There might be an impact for the user space (UCM). I will fix the UCM configurations when this patch is applied. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Documentation/sound/designs/control-names.rst | 5 +++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 13 files changed, 15 insertions(+), 14 deletions(-) diff --git a/Documentation/sound/designs/control-names.rst b/Documentation/sound/designs/control-names.rst index 7fedd0f33cd9..a5f6e3842df8 100644 --- a/Documentation/sound/designs/control-names.rst +++ b/Documentation/sound/designs/control-names.rst @@ -50,7 +50,7 @@ Internal internal SOURCE ~~~~~~ -=================== ================================================= +=================== =================================================== Master Master Mono Hardware Master @@ -92,7 +92,8 @@ SPDIF output only SPDIF In Digital In HDMI/DP either HDMI or DisplayPort -=================== ================================================= +HDMI/DP,pcm=<num> either HDMI or DisplayPort with the PCM device link +=================== =================================================== Exceptions (deprecated) ----------------------- diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index ac1dea5f9d11..b49028c6fd10 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -618,7 +618,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &broxton_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index adf416a49b48..caaf2f2af287 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -530,7 +530,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &broxton_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index bd2d371f2acd..5474f28e0c31 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -548,7 +548,7 @@ static int glk_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &geminilake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb6..367415efed9b 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -548,7 +548,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 829f95fc4179..b30c2148d1f4 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -977,7 +977,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &kabylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 74fe1f3a5479..0965d1f02b2c 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -470,7 +470,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 7cefda341fbf..9f420e978459 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -888,7 +888,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index 58409b6e476e..f717b1d179bd 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -139,7 +139,7 @@ int skl_hda_hdmi_jack_init(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &pcm->hdmi_jack, NULL, 0); diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 3ce8efbeed12..b009e2df25d3 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -607,7 +607,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 1a7ac8bdf543..0a6e650dc698 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -648,7 +648,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 231349a47cc9..56ee4b55dce9 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -489,7 +489,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a..862a7e35cb5c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -277,7 +277,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &sof_hdmi[i], NULL, 0); -- 2.20.1

6 20

[alsa-devel] [PATCH] ALSA: hda: Allow non-Intel device probe gracefully
by Takashi Iwai 28 Oct '19

28 Oct '19

The recent addition of snd_intel_dsp_driver_probe() check caused a spurious kernel warning when the driver is loaded for a non-Intel hardware due to snd_BUG_ON(). Moreover, for such a hardware, we should always return SND_INTEL_DSP_DRIVER_ANY, not check the dsp_driver option at all. This patch fixes these issues for non-Intel devices. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/intel-dsp-config.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 0b2a7201a46a..be1df80ed013 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -307,13 +307,13 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) { const struct config_entry *cfg; - if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) - return dsp_driver; - /* Intel vendor only */ - if (snd_BUG_ON(pci->vendor != 0x8086)) + if (pci->vendor != 0x8086) return SND_INTEL_DSP_DRIVER_ANY; + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + /* * detect DSP by checking class/subclass/prog-id information * class=04 subclass 03 prog-if 00: no DSP, use legacy driver -- 2.16.4

1 0

[alsa-devel] Questions about max98090 codec driver
by Tzung-Bi Shih 28 Oct '19

28 Oct '19

Hi, I am studying an odd issue of max98090 codec on Baytrail-based chromebook. The issue is: when user playback and capture simultaneously, it seems the PLL never get locked if msleep(10) between the SHDN off and on (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…). The playback stream becomes silent and the console keeps printing "PLL unlocked". But, if comment out the msleep(10) between the SHDN off and on, the issue fixed. I am trying to find the reason but facing further more questions and may need your inputs. 1. The commit b8a3ee820f7b ("ASoC: max98090: Add recovery for PLL lock failure") enables ULK interrupts (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…) when PCM stream starting. If max98090 claims its PLL is unlocked, max98090_pll_work() get scheduled to workaround it by SHDN off and on (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…). I feel it is weird to sleep in max98090_pll_work(). Especially, at this line https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.… (it makes less sense to "wait" in another thread). Note that, the threaded IRQF_ONESHOT handler and max98090_pll_work() are in 2 different threads. I guess the original intention is: - disable ULK interrupt in IRQ handler (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…) - schedule max98090_pll_work() to workaround it - wait 10ms to give PLL chance to lock - enable ULK interrupt again If max98090 claims its PLL is unlocked again, repeat the above by receiving another ULK interrupt. Unfortunately, the odd issue seems not be fixed by my rough implementation of these. 2. According to the datasheet page 164 table 90 (https://datasheets.maximintegrated.com/en/ds/MAX98090.pdf), there are some registers should only be adjusted when SHDN==0. But I fail to find max98090.c tries to set SHDN to 0 and restore it afterwards when writing to these registers. For example, https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…. I am wondering if it would bring any side effects because the datasheet states "Changing these settings during normal operation (SHDN=1) can compromise device stability and performance specifications." 3. By searching some history data, I found a previous version did not have the msleep(10) between the SHDN off and on (https://crrev.com/c/191740, click the file name in the middle of the window to see the diff. Pardon me, I do not find another public repository for this). I am curious if anyone of you still remember why the upstream version contains the msleep(10). I am also curious if anyone of your environment works well with the upstream version max98090.c.

4 5

Re: [alsa-devel] [PATCH V2] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by S.j. Wang 28 Oct '19

28 Oct '19

Hi > > On Fri, Oct 25, 2019 at 03:13:53PM +0800, Shengjiu Wang wrote: > > xrun may happen at the end of stream, the > > trigger->fsl_esai_trigger_stop maybe called in the middle of > > fsl_esai_hw_reset, this may cause esai in wrong state after stop, and > > there may be endless xrun interrupt. > > > > This issue may also happen with trigger->fsl_esai_trigger_start. > > > > So Add spin lock to lock those functions. > > > > Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after > > xrun") > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > Some small comments inline. Once they are addressed, please add: > > Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> > > Thanks > > > --- > > Change in v2 > > -add lock for fsl_esai_trigger_start. > > > > sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ > > 1 file changed, 12 insertions(+) > > > > diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index > > 37b14c48b537..9b28e2af26e4 100644 > > --- a/sound/soc/fsl/fsl_esai.c > > +++ b/sound/soc/fsl/fsl_esai.c > > @@ -33,6 +33,7 @@ > > * @fsysclk: system clock source to derive HCK, SCK and FS > > * @spbaclk: SPBA clock (optional, depending on SoC design) > > * @task: tasklet to handle the reset operation > > + * @lock: spin lock to handle reset and stop behavior > > Should be "between hw_reset() and trigger()" now. > > > * @fifo_depth: depth of tx/rx FIFO > > * @slot_width: width of each DAI slot > > * @slots: number of slots > > @@ -56,6 +57,7 @@ struct fsl_esai { > > struct clk *fsysclk; > > struct clk *spbaclk; > > struct tasklet_struct task; > > + spinlock_t lock; /* Protect reset and stop */ > > We can drop the comments here since you add it to the top. Here is required by the checkpatch.pl, so still is needed. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ALSA: pci: Fix memory leak in snd_ali_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_ali_create() the memory allocated for codec is leaked in case of an error. Release codec if snd_ali_chip_init() fails. Fixes: f9ab2b1c3ab5 ("[ALSA] ali5451 - Code clean up, irq handler fix") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 6e28e381c21a..3ed07a18be4d 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2179,6 +2179,7 @@ static int snd_ali_create(struct snd_card *card, err = snd_ali_chip_init(codec); if (err < 0) { dev_err(card->dev, "ali create: chip init error.\n"); + snd_ali_free(codec); return err; } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: lx6464es: Fix memory leaks in snd_lx6464es_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_lx6464es_create() it there are memory leaks when error happens. Go to error path if any of these calls fail: lx_init_dsp(), lx_pcm_create(), lx_proc_create(), snd_ctl_add(). Fixes: 02bec4904508 ("ALSA: lx6464es - driver for the digigram lx6464es interface") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/lx6464es/lx6464es.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index fe10714380f2..7c6e8f4ef826 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1020,25 +1020,26 @@ static int snd_lx6464es_create(struct snd_card *card, err = lx_init_dsp(chip); if (err < 0) { dev_err(card->dev, "error during DSP initialization\n"); - return err; + goto cleanup; } err = lx_pcm_create(chip); if (err < 0) - return err; + goto cleanup; err = lx_proc_create(card, chip); if (err < 0) - return err; + goto cleanup; err = snd_ctl_add(card, snd_ctl_new1(&lx_control_playback_switch, chip)); if (err < 0) - return err; + goto cleanup; *rchip = chip; return 0; +cleanup: device_new_failed: free_irq(pci->irq, chip); -- 2.17.1

3 2

[alsa-devel] [PATCH -next] ASoC: SOF: select SND_INTEL_DSP_CONFIG in SND_SOC_SOF_PCI
by Mao Wenan 28 Oct '19

28 Oct '19

When SND_SOC_SOF_PCI=y, and SND_INTEL_DSP_CONFIG=m, below errors can be seen: sound/soc/sof/sof-pci-dev.o: In function `sof_pci_probe': sof-pci-dev.c:(.text+0xb9): undefined reference to `snd_intel_dsp_driver_probe' After commit 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code"), sof_pci_probe() will call snd_intel_dsp_driver_probe(), so it should select SND_INTEL_DSP_CONFIG in Kconfig SND_SOC_SOF_PCI. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Mao Wenan <maowenan(a)huawei.com> --- sound/soc/sof/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 56a3ab6..a9b2be2 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_SOF_PCI select SND_SOC_ACPI if ACPI select SND_SOC_SOF_OPTIONS select SND_SOC_SOF_INTEL_PCI if SND_SOC_SOF_INTEL_TOPLEVEL + select SND_INTEL_DSP_CONFIG help This adds support for PCI enumeration. This option is required to enable Intel Skylake+ devices -- 2.7.4

2 1

[alsa-devel] [PATCH] Fix JACK plugin buffer sizes
by John McKay 27 Oct '19

27 Oct '19

I was having a conversation on the Archlinux forums[1] about using the alsa-plugins JACK plugin. While one of the bugs that we found has already been fixed in the git alsa-lib, there is another bug relating to buffer sizes. Currently snd_pcm_hw_params_set_buffer_size_near() doesn't really work when you are using this plugin unless you happen to request a buffer size that almost exactly matches a whole multiple of periods that the JACK is asking for. After some debugging, I discovered the reason is that some plugins, notably the 'plug' plugin that most everybody is going to use to change frequency/format directly attempt to manipulate the buffer size. This causes the code in pcm_ioplug.c to fail because it cannot find a number of periods/period time that can fit the buffer. The included patch stops this by specifying the list of acceptable buffer sizes as well, so that when other plugins try to change the buffer it always picks one that will work. It does have some problems in the JACK plugin now offers a number of periods/sizes that don't exist but in practice asking for huge buffers just seems to give you a more sane value. [1] https://bbs.archlinux.org/viewtopic.php?id=250116 --- jack/pcm_jack.c 2019-05-10 06:57:24.000000000 +0000 +++ jack/pcm_jack.c 2019-10-27 03:43:36.053734268 +0000 @@ -428,6 +428,7 @@ unsigned int format = SND_PCM_FORMAT_FLOAT; unsigned int rate = jack_get_sample_rate(jack->client); unsigned int psize_list[MAX_PERIODS_MULTIPLE]; + unsigned int bsize_list[MAX_PERIODS_MULTIPLE]; unsigned int nframes = jack_get_buffer_size(jack->client); unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * jack->num_ports); @@ -441,6 +442,9 @@ for (i = 1; i <= ARRAY_SIZE(psize_list); i++) psize_list[i-1] = jack_buffer_bytes * i; + for (i = 1; i <= ARRAY_SIZE(bsize_list); i++) + bsize_list[i-1] = jack_buffer_bytes * (i + 1); + jack->sample_bits = snd_pcm_format_physical_width(format); if ((err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list)) < 0 || @@ -452,6 +456,8 @@ rate, rate)) < 0 || (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, ARRAY_SIZE(psize_list), psize_list)) < 0 || + (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, + ARRAY_SIZE(bsize_list), bsize_list)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_PERIODS, 2, 64)) < 0) return err;

1 0

[alsa-devel] [PATCH] ALSA: bebob: Fix prototype of helper function to return negative value
by Takashi Sakamoto 26 Oct '19

26 Oct '19

A helper function of ALSA bebob driver returns negative value in a function which has a prototype to return unsigned value. This commit fixes it by changing the prototype. Fixes: eb7b3a056cd8 ("ALSA: bebob: Add commands and connections/streams management" Cc: <stable(a)vger.kernel.org> # v3.16+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 7ac0d9f495c4..f7f0db5aa811 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -252,8 +252,7 @@ int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob, return err; } -static unsigned int -map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) +static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) { unsigned int sec, sections, ch, channels; unsigned int pcm, midi, location; -- 2.20.1

2 1

[alsa-devel] [PATCH v2 00/12] Add Samus Hotwording for RT5677
by Curtis Malainey 26 Oct '19

26 Oct '19

This patch series adds the hotwording implementation used in the Pixelbook on the RT5677 driver. Known Issues: There is a known issue where the system will fail to detect a hotword if suspended while the stream is open. This is due to the fact that the haswell-dsp suspends its I2S MCLK before the RT5677 suspends which causes the writes and reads to become corrupted as a result. Any recommendations to correct this behaviour would be appreciated. Changes since v1: * Addressed Mark's and Kuninori's comments * Adjusted DAI Links * Squashed related patches * Removed all kcontrol writes * Addressed race condition with dapm * Added firmware boot check * Moved all clocking to driver API * Moved DAPM routes that spanned drivers to machine driver Cc: Dylan Reid <dgreid(a)chromium.org> Cc: Ben Zhang <benzh(a)chromium.org> Ben Zhang (8): ASoC: rt5677: Add a PCM device for streaming hotword via SPI ASoC: rt5677: Load firmware via SPI using delayed work ASoC: bdw-rt5677: Add a DAI link for rt5677 SPI PCM device ASoC: rt5677: Enable jack detect while DSP is running ASoC: rt5677: Add DAPM audio path for hotword stream ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile ASoC: rt5677: Stop and restart DSP over suspend/resume ASoC: rt5677: Disable irq at suspend Curtis Malainey (4): ASoC: rt5677: Allow VAD to be shut on/off at all times ASoC: bdw-rt5677: Turn on MCLK1 for DSP via DAPM ASoC: rt5677: Set ADC clock to use PLL and enable ASRC ASoC: rt5677: Wait for DSP to boot before loading firmware sound/soc/codecs/rt5677-spi.c | 397 ++++++++++++++++++++++++ sound/soc/codecs/rt5677-spi.h | 1 + sound/soc/codecs/rt5677.c | 447 +++++++++++++++++++++++----- sound/soc/codecs/rt5677.h | 11 +- sound/soc/intel/boards/bdw-rt5677.c | 51 ++++ 5 files changed, 834 insertions(+), 73 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

4 19

[alsa-devel] [PATCH V2] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 25 Oct '19

25 Oct '19

The output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitaion in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Change in v2 - remove p2p/m2m word - use use_ideal_rate sound/soc/fsl/fsl_asrc.c | 37 +++++++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..89cf333154c7 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -259,8 +259,11 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * It configures those ASRC registers according to a configuration instance * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. + * + * Note: + * use_ideal_rate = true is need by some case which need higher performance. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +271,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,8 +355,10 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; - if (div[IN] == 0) { + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; + if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; @@ -360,18 +366,29 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) clk = asrc_priv->asrck_clk[clk_index[OUT]]; - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + /* + * Output rate should be align with the out samplerate. If set too + * high output rate, there will be lots of Overload. + * But some case need higher performance, then we can use + * IDEAL_RATIO_RATE specifically for such case. + */ + clk_rate = clk_get_rate(clk); + if (ideal && use_ideal_rate) + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); else - div[OUT] = clk_get_rate(clk) / outrate; + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; - if (div[OUT] == 0) { + if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; } + /* Divider range is [1, 1024] */ + div[IN] = min_t(u32, 1024, div[IN]); + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +577,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH V2] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Shengjiu Wang 25 Oct '19

25 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Change in v2 -add lock for fsl_esai_trigger_start. sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..9b28e2af26e4 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock to handle reset and stop behavior * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect reset and stop */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH v2] ASoC: ti: omap-mcpdm: Add support for pdmclk clock handling
by Peter Ujfalusi 25 Oct '19

25 Oct '19

McPDM module receives it's functional clock from external source. This clock is the pdmclk provided by the twl6040 audio IC. If the clock is not available all register accesses to McPDM fails and the module is not operational. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Hi, this has been lurking in my next-wip branch for some time... It might not needed anymore as I have not heard about issues with McPDM for a while. It was dropped back then because some core parts were not picked in time and this caused some issues, but can not find the exact reason Regards, Peter sound/soc/ti/omap-mcpdm.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index b8c8290265c7..2c536947b69f 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -17,6 +17,7 @@ #include <linux/err.h> #include <linux/io.h> #include <linux/irq.h> +#include <linux/clk.h> #include <linux/slab.h> #include <linux/pm_runtime.h> #include <linux/of_device.h> @@ -42,6 +43,7 @@ struct omap_mcpdm { int irq; struct pm_qos_request pm_qos_req; int latency[2]; + struct clk *pdmclk; struct mutex mutex; @@ -416,6 +418,10 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int ret; + ret = clk_prepare_enable(mcpdm->pdmclk); + if (ret) + return ret; + pm_runtime_enable(mcpdm->dev); /* Disable lines while request is ongoing */ @@ -454,6 +460,7 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) if (pm_qos_request_active(&mcpdm->pm_qos_req)) pm_qos_remove_request(&mcpdm->pm_qos_req); + clk_disable_unprepare(mcpdm->pdmclk); return 0; } @@ -473,12 +480,19 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai) mcpdm->pm_active_count++; } + clk_disable_unprepare(mcpdm->pdmclk); + return 0; } static int omap_mcpdm_resume(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(mcpdm->pdmclk); + if (ret) + return ret; if (mcpdm->pm_active_count) { while (mcpdm->pm_active_count--) @@ -573,6 +587,15 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; + mcpdm->pdmclk = devm_clk_get(&pdev->dev, "pdmclk"); + if (IS_ERR(mcpdm->pdmclk)) { + if (PTR_ERR(mcpdm->pdmclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + dev_warn(&pdev->dev, "Error getting pdmclk (%ld)!\n", + PTR_ERR(mcpdm->pdmclk)); + mcpdm->pdmclk = NULL; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, &omap_mcpdm_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 2

[alsa-devel] sof-skl_hda: move hdmi as separate use case
by GitHub pull_request - opened 25 Oct '19

25 Oct '19

alsa-project/alsa-ucm-conf pull request #6 was opened from juimonen: Move hdmi definitions as separate use case and conf file. This will enable Pulseaudio to create separate profile for hdmi and make routing much more fluent than with single profile. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/6 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/6.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] sof-skl_hda: fix headphone jack name
by GitHub pull_request - opened 25 Oct '19

25 Oct '19

alsa-project/alsa-ucm-conf pull request #5 was opened from juimonen: Change "Headphone Jack" to "Headphone Mic Jack". Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/5 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/5.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [ANNOUNCE] 2019 Linux Audio miniconference
by Mark Brown 25 Oct '19

25 Oct '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue) and will be free of charge to attend, generously sponsored by Intel. Thus far for the agenda we have: - Use case management enhancements (Curtis) - DSP framework integration (Liam?) - Soundwire status (Pierre?) - Componentisation status/plans (Me) As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

8 19

[alsa-devel] [PATCH v5 1/2] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 25 Oct '19

25 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems. The examples are taken from the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/intel-dsp-config.h | 34 +++++ sound/hda/Kconfig | 10 +- sound/hda/Makefile | 5 +- sound/hda/intel-dsp-config.c | 249 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +- sound/pci/hda/hda_intel.c | 49 ++---- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 +-- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 + 11 files changed, 322 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..601e617918b8 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,6 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +snd-intel-dspcfg-$(CONFIG_SND_INTEL_NHLT) += intel-nhlt.o +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..d9f6d9e872b4 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,249 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <linux/bits.h> +#include <linux/dmi.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/intel-dsp-config.h> +#include <sound/intel-nhlt.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); + +#define FLAG_SST BIT(0) +#define FLAG_SOF BIT(1) +#define FLAG_SOF_ONLY_IF_DMIC BIT(16) + +struct config_entry { + u32 flags; + u16 device; + const struct dmi_system_id *dmi_table; +}; + +/* + * configuration table + * - the order of similar PCI ID entries is important! + * - the first successful match will win + */ +static const struct config_entry config_table[] = { +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) + { + /* prefer SST */ + .flags = FLAG_SST, + .device = 0x02c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) + { + .flags = FLAG_SST, + .device = 0x06c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF, + .device = 0x06c8, + }, +#endif +/* Merrifield */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { + .flags = FLAG_SOF, + .device = 0x119a, + }, +#endif +/* Broxton-T */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x1a98, + }, +#endif +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) + { + .flags = FLAG_SST, + .device = 0x3198, + }, +#endif +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif +/* Appololake (Broxton-P) */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF, + .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Up Squared", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + { + .flags = FLAG_SST, + .device = 0x5a98, + }, +#endif +/* Cannonlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9dc8, + }, +#endif +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d70, + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d71, + }, +#endif +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif +/* Coffelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa348, + }, +#endif +}; + +static const struct config_entry *snd_intel_dsp_find_config + (struct pci_dev *pci, const struct config_entry *table, u32 len) +{ + u16 device; + + device = pci->device; + for (; len > 0; len--, table++) { + if (table->device != device) + continue; + if (table->dmi_table && !dmi_check_system(table->dmi_table)) + continue; + return table; + } + return NULL; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + const struct config_entry *cfg; + + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* find the configuration for the specific device */ + cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); + if (!cfg) + return SND_INTEL_DSP_DRIVER_ANY; + + if (cfg->flags & FLAG_SOF) { + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + } else { + return SND_INTEL_DSP_DRIVER_SOF; + } + } + + if (cfg->flags & FLAG_SST) + return SND_INTEL_DSP_DRIVER_SST; + + return SND_INTEL_DSP_DRIVER_LEGACY; +} +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..a916963822e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool dsp_driver = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,9 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(dsp_driver, bool, 0444); +MODULE_PARM_DESC(dsp_driver, "Allow DSP driver selection (bypass this driver) " + "(0=off, 1=on) (default=1)"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2021,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2038,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2061,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

3 6

[alsa-devel] [PATCH -next] ALSA: intel_hdmi: Remove dev_err() on platform_get_irq() failure
by YueHaibing 25 Oct '19

25 Oct '19

platform_get_irq() will call dev_err() itself on failure, so there is no need for the driver to also do this. This is detected by coccinelle. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/x86/intel_hdmi_audio.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 5fd4e32..cd389d2 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1708,10 +1708,8 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* get resources */ irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "Could not get irq resource: %d\n", irq); + if (irq < 0) return irq; - } res_mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res_mmio) { -- 2.7.4

2 1

[alsa-devel] [PATCH][next] ASoC: rt5677: Add missing null check for failed allocation of rt5677_dsp
by Colin King 25 Oct '19

25 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> The allocation of rt5677_dsp can potentially fail and return null, so add a null check and return -ENOMEM on a memory allocation failure. Addresses-Coverity: ("Dereference null return") Fixes: a0e0d135427c ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5677-spi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 36c02d200cfc..3a17643fcd9f 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -376,6 +376,8 @@ static int rt5677_spi_pcm_probe(struct snd_soc_component *component) rt5677_dsp = devm_kzalloc(component->dev, sizeof(*rt5677_dsp), GFP_KERNEL); + if (!rt5677_dsp) + return -ENOMEM; rt5677_dsp->dev = &g_spi->dev; mutex_init(&rt5677_dsp->dma_lock); INIT_DELAYED_WORK(&rt5677_dsp->copy_work, rt5677_spi_copy_work); -- 2.20.1

3 2

[alsa-devel] [PATCH trivial] ASoC: Spelling s/configr/configur/
by Geert Uytterhoeven 25 Oct '19

25 Oct '19

Fix misspellings of "configuration" and "configure". Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- include/sound/wm8904.h | 2 +- sound/soc/codecs/cx2072x.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 14074405f5012f9a..88ac1870510ec063 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -120,7 +120,7 @@ * DRC configurations are specified with a label and a set of register * values to write (the enable bits will be ignored). At runtime an * enumerated control will be presented for each DRC block allowing - * the user to choose the configration to use. + * the user to choose the configuration to use. * * Configurations may be generated by hand or by using the DRC control * panel provided by the WISCE - see http://www.wolfsonmicro.com/wisce/ diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index 1c1ba7bea4d81969..2ad00ed21bec6c59 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -1507,7 +1507,7 @@ static int cx2072x_probe(struct snd_soc_component *codec) regmap_multi_reg_write(cx2072x->regmap, cx2072x_reg_init, ARRAY_SIZE(cx2072x_reg_init)); - /* configre PortC as input device */ + /* configure PortC as input device */ regmap_update_bits(cx2072x->regmap, CX2072X_PORTC_PIN_CTRL, 0x20, 0x20); -- 2.17.1

2 1

[alsa-devel] [PATCH trivial] ASoC: pxa: poodle: Spelling s/enpoints/endpoints/, s/connetion/connection/
by Geert Uytterhoeven 25 Oct '19

25 Oct '19

Fix misspelling of "endpoints" and "connection". Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/pxa/poodle.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 48d5c2252b10081d..59ef04d0467a622d 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -56,7 +56,7 @@ static void poodle_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); } - /* set the enpoints to their new connetion states */ + /* set the endpoints to their new connection states */ if (poodle_spk_func == POODLE_SPK_ON) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: rt5677-spi: fix sparse warnings
by Curtis Malainey 25 Oct '19

25 Oct '19

Fix bugs reported by kbuild test robot Fixes: a0e0d135427c ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI") Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 36c02d200cfc..958871b342ea 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -145,7 +145,7 @@ static int rt5677_spi_hw_free( snd_soc_component_get_drvdata(component); mutex_lock(&rt5677_dsp->dma_lock); - rt5677_dsp->substream = 0; + rt5677_dsp->substream = NULL; mutex_unlock(&rt5677_dsp->dma_lock); return snd_pcm_lib_free_vmalloc_buffer(substream); @@ -362,7 +362,7 @@ static void rt5677_spi_copy_work(struct work_struct *work) mutex_unlock(&rt5677_dsp->dma_lock); } -struct page *rt5677_spi_pcm_page( +static struct page *rt5677_spi_pcm_page( struct snd_soc_component *component, struct snd_pcm_substream *substream, unsigned long offset) -- 2.24.0.rc0.303.g954a862665-goog

2 1

[alsa-devel] [PATCH v2] ASoC: soc-dpcm: tidyup for_each_dpcm_xx() macro
by Kuninori Morimoto 25 Oct '19

25 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> for_each_dpcm_xx() macro is using "dpcm" as parameter (1), but, it is also struct member (2). #define for_each_dpcm_fe(be, stream, dpcm) \ list_for_each_entry(dpcm, &(be)->dpcm[stream]...) ^^^^(1) ^^^^(2) Thus, it will be compile error if user not used "dpcm" as parameter for_each_dpcm_fe(be, stream, dp) ^^ This patch fixup it. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - based on mark/for-5.5 branch include/sound/soc-dpcm.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index e55aeb0..b654ebf 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,15 +103,15 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; -#define for_each_dpcm_fe(be, stream, dpcm) \ - list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe) - -#define for_each_dpcm_be(fe, stream, dpcm) \ - list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be) -#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm) \ - list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be) -#define for_each_dpcm_be_rollback(fe, stream, dpcm) \ - list_for_each_entry_continue_reverse(dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_fe(be, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(be)->dpcm[stream].fe_clients, list_fe) + +#define for_each_dpcm_be(fe, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_safe(fe, stream, _dpcm, __dpcm) \ + list_for_each_entry_safe(_dpcm, __dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_rollback(fe, stream, _dpcm) \ + list_for_each_entry_continue_reverse(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) /* can this BE stop and free */ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, -- 2.7.4

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: SOF: make hw_info platform specific
by Pierre-Louis Bossart 25 Oct '19

25 Oct '19

The SOF core blindly sets the hw_info, when it should really be platform-specific. This move enable us to temporarily fix PulseAudio issues resulting from inaccuracies in the hw_ptr position provided by the firmware on Baytrail-Broadwell. This is the first time we experiment with the GitHub low-level API to automagically keep track of 'approvers' by inserting a Reviewed-by tag when merging a pull-request, thanks to Ranjani for her initial work and bearing with my habit of breaking things :-) v2: make the code apply on Mark's tree. Sorry about that bad patch. Pierre-Louis Bossart (2): ASoC: SOF: define INFO_ flags in dsp_ops ASoC: SOF: Intel: only support INFO_BATCH for legacy platforms sound/soc/sof/imx/imx8.c | 7 +++++++ sound/soc/sof/intel/apl.c | 7 +++++++ sound/soc/sof/intel/bdw.c | 9 ++++++++- sound/soc/sof/intel/byt.c | 21 +++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 7 +++++++ sound/soc/sof/pcm.c | 8 +++----- sound/soc/sof/sof-priv.h | 3 +++ 7 files changed, 56 insertions(+), 6 deletions(-) -- 2.20.1

2 4

[alsa-devel] [PATCH] ASoC: Intel: eve: Enable mclk and ssp sclk early for rt5514
by Brent Lu 25 Oct '19

25 Oct '19

The first DMIC capture always fail (zero sequence data from PCM port) after using DSP hotwording function (i.e. Google assistant). The DMIC is attached to rt5514 which also require eaily mclk/sclk like rt5663. Therefore we add a dapm route to provide ssp mclk/sclk early to rt5514. The rt5514 is attached to SSP0 and the rt5663 is attached to SSP1 so we create another supply widget to provide SSP0 mclk/sclk to rt5514. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 136 ++++++++++++++------- 1 file changed, 92 insertions(+), 44 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index dc09a85..1755efa 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -53,8 +53,10 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; - struct clk *mclk; - struct clk *sclk; + struct clk *ssp0_mclk; + struct clk *ssp0_sclk; + struct clk *ssp1_mclk; + struct clk *ssp1_sclk; }; enum { @@ -77,13 +79,31 @@ static const struct snd_kcontrol_new kabylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event, int ssp_num) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + struct clk *mclk, *sclk; + unsigned long sclk_rate; int ret = 0; + switch (ssp_num) { + case 0: + mclk = priv->ssp0_mclk; + sclk = priv->ssp0_sclk; + sclk_rate = 6144000; + break; + case 1: + mclk = priv->ssp1_mclk; + sclk = priv->ssp1_sclk; + sclk_rate = 3072000; + break; + default: + dev_err(card->dev, "Invalid ssp_num %d\n", ssp_num); + return -EINVAL; + } + /* * MCLK/SCLK need to be ON early for a successful synchronization of * codec internal clock. And the clocks are turned off during @@ -92,37 +112,39 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: /* Enable MCLK */ - ret = clk_set_rate(priv->mclk, 24000000); + ret = clk_set_rate(mclk, 24000000); if (ret < 0) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); + dev_err(card->dev, "Can't set rate for ssp%d_mclk, err: %d\n", + ssp_num, ret); return ret; } - ret = clk_prepare_enable(priv->mclk); + ret = clk_prepare_enable(mclk); if (ret < 0) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + dev_err(card->dev, "Can't enable ssp%d_mclk, err: %d\n", + ssp_num, ret); return ret; } /* Enable SCLK */ - ret = clk_set_rate(priv->sclk, 3072000); + ret = clk_set_rate(sclk, sclk_rate); if (ret < 0) { - dev_err(card->dev, "Can't set rate for sclk, err: %d\n", - ret); - clk_disable_unprepare(priv->mclk); + dev_err(card->dev, "Can't set rate for ssp%d_sclk, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(mclk); return ret; } - ret = clk_prepare_enable(priv->sclk); + ret = clk_prepare_enable(sclk); if (ret < 0) { - dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); - clk_disable_unprepare(priv->mclk); + dev_err(card->dev, "Can't enable ssp%d_sclk, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(mclk); } break; case SND_SOC_DAPM_POST_PMD: - clk_disable_unprepare(priv->mclk); - clk_disable_unprepare(priv->sclk); + clk_disable_unprepare(mclk); + clk_disable_unprepare(sclk); break; default: return 0; @@ -131,6 +153,18 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int platform_clock_control_ssp0(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 0); +} + +static int platform_clock_control_ssp1(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 1); +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -139,15 +173,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_SUPPLY("Platform Clock SSP0", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp0, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Platform Clock SSP1", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp1, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ - { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "Platform Clock SSP1" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -156,7 +192,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock SSP1" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -180,6 +216,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Rx", NULL, "Right HiFi Capture" }, /* DMIC */ + { "DMIC", NULL, "Platform Clock SSP0" }, { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, { "DMIC2L", NULL, "DMIC" }, @@ -704,6 +741,29 @@ static struct snd_soc_card kabylake_audio_card = { .late_probe = kabylake_card_late_probe, }; +static int kabylake_audio_clk_get(struct device *dev, const char *id, + struct clk **clk) +{ + int ret = 0; + + if (!clk) + return -EINVAL; + + *clk = devm_clk_get(dev, id); + if (IS_ERR(*clk)) { + ret = PTR_ERR(*clk); + if (ret == -ENOENT) { + dev_info(dev, "Failed to get %s, defer probe\n", id); + return -EPROBE_DEFER; + } + + dev_err(dev, "Failed to get %s with err:%d\n", id, ret); + return ret; + } + + return ret; +} + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; @@ -724,33 +784,21 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; - ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_mclk", &ctx->ssp0_mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_sclk", &ctx->ssp0_sclk); + if (ret != 0) return ret; - } - ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_mclk", &ctx->ssp1_mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_sclk", &ctx->ssp1_sclk); + if (ret != 0) return ret; - } return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

3 2

Re: [alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by S.j. Wang 25 Oct '19

25 Oct '19

Hi > > On Wed, Oct 23, 2019 at 06:25:20AM +0000, S.j. Wang wrote: > > > On Thu, Oct 17, 2019 at 02:21:08PM +0800, Shengjiu Wang wrote: > > > > For P2P output, the output divider should align with the output > > > > sample > > > > > > I think we should avoid "P2P" (or "M2M") keyword in the mainline > > > code as we know M2M will never get merged while somebody working > > > with the mainline and caring about new feature might be confused. > > > > Ok. But we still curious that is there a way to upstream m2m? > > Hmm..I would love to see that happening. Here is an old discussion that > you may want to take a look: > https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fmail > man.alsa-project.org%2Fpipermail%2Falsa-devel%2F2014- > May%2F076797.html&data=02%7C01%7Cshengjiu.wang%40nxp.com%7 > Ce902d2bac4254d2faa0f08d757ecac0e%7C686ea1d3bc2b4c6fa92cd99c5c301 > 635%7C0%7C0%7C637074546320396681&sdata=bg%2BLwRQnUPhW8f > mE972O%2F53MyVftJkK140PSnmC%2FDKQ%3D&reserved=0 > > > > It makes sense to me, yet I feel that the delay at the beginning of > > > the audio playback might be longer as a compromise. I am okay with > > > this decision though... > > > > > > > The maximum divider of asrc clock is 1024, but there is no > > > > judgement for this limitaion in driver, which may cause the > > > > divider setting not correct. > > > > > > > > For non-ideal ratio mode, the clock rate should divide the sample > > > > rate with no remainder, and the quotient should be less than 1024. > > > > > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > > > > @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct > > > > fsl_asrc_pair > > > *pair) > > > > /* We only have output clock for ideal ratio mode */ > > > > clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; > > > > > > > > - div[IN] = clk_get_rate(clk) / inrate; > > > > + clk_rate = clk_get_rate(clk); > > > > > > The fsl_asrc.c file has config.inclk being set to INCLK_NONE and > > > this sets the "ideal" in this function to true. So, although we tend > > > to not use ideal ratio setting for p2p cases, yet the input clock is > > > still not physically connected, so we still use output clock for div[IN] > calculation? > > > > For p2p case, it can be ideal or non-ideal. For non-ideal, we still > > use Output clock for div calculation. > > > > > > > > I am thinking something simplier: if we decided not to use ideal > > > ratio for "P2P", instead of adding "bool p2p" with the confusing > > > "ideal" in this function, could we just set config.inclk to the same > > > clock as the output one for "P2P"? By doing so, "P2P" won't go > > > through ideal ratio mode while still having a clock rate from the output > clock for div[IN] calculation here. > > > > Bool p2p is to force output rate to be sample rate, no impact to ideal > > Ratio mode. > > I just realized that the function has a bottom part for ideal mode > exclusively -- if we treat p2p as !ideal, those configurations will be missing. > So you're right, should have an extra boolean variable. > > > > > > > > + rem[IN] = do_div(clk_rate, inrate); > > > > + div[IN] = (u32)clk_rate; > > > > if (div[IN] == 0) { > > > > > > Could we check div[IN] and rem[IN] here? Like: > > > if (div[IN] == 0 || div[IN] > 1024) { > > > pair_err(); > > > goto out; > > > } > > > > > > if (!ideal && rem[IN]) { > > > pair_err(); > > > goto out; > > > } > > > > > > According to your commit log, I think the max-1024 limitation should > > > be applied to all cases, not confined to "!ideal" cases right? And > > > we should add some comments also, indicating it is limited by hardware. > > > > For ideal mode, my test result is the divider not impact the output result. > > Which means it is ok for ideal mode even divider is not correct... > > OK. > > > > > > > > pair_err("failed to support input sample rate %dHz > > > > by > > > asrck_%x\n", > > > > inrate, clk_index[ideal ? OUT : > > > > IN]); @@ > > > > -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct > > > > fsl_asrc_pair *pair) > > > > > > > > clk = asrc_priv->asrck_clk[clk_index[OUT]]; > > > > > > > > - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ > > > > - if (ideal) > > > > - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; > > > > - else > > > > - div[OUT] = clk_get_rate(clk) / outrate; > > > > + /* > > > > + * When P2P mode, output rate should align with the out > samplerate. > > > > + * if set too high output rate, there will be lots of Overload. > > > > + * When M2M mode, output rate should also need to align with > > > > + the out > > > > > > For this "should", do you actually mean "M2M could also"? Sorry, I'm > > > just trying to understand everyting here, not intentionally being picky at > words. > > > My understanding is that we still keep the ideal ratio setting > > > because "M2M" still uses it. > > > > We use IDEAL_RATIO_RATE as output rate for m2m mode, it likes a Tricky > > operation, in order to improve the performance. I think The correct > > operation is to use the real output rate, but the performance Is bad. > > So it is a compromise. > > I see. > > > > > > > > + * samplerate, but M2M must use less time to achieve good > > > performance. > > > > + */ > > > > + clk_rate = clk_get_rate(clk); > > > > + if (p2p || !ideal) { > > > > + rem[OUT] = do_div(clk_rate, outrate); > > > > + div[OUT] = clk_rate; > > > > + } else { > > > > + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); > > > > + div[OUT] = clk_rate; > > > > + } > > > > > > > > if (div[OUT] == 0) { > > > > pair_err("failed to support output sample rate %dHz > > > > by asrck_%x\n", @@ -372,6 +384,16 @@ static int > > > > fsl_asrc_config_pair(struct > > > fsl_asrc_pair *pair) > > > > return -EINVAL; > > > > } > > > > > > > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || > > > > + rem[IN] != 0 || rem[OUT] != 0)) { > > > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || rem[IN] > > > > + || rem[OUT] != 0)) { > > > > > > So for ideal == true, these limitaions are not applied any more? > > > Remember that the "ideal" is true for "p2p == true" cases here. > > > > No, not applied. for ideal, the div don't impact the output result > > Even it is not accurate. > > I see. > > > > > > > > + pair_err("The divider can't be used for non ideal mode\n"); > > > > + return -EINVAL; > > > > + } > > > > + > > > > + /* Divider range is [1, 1024] */ > > > > + div[IN] = min_t(u32, 1024, div[IN]); > > > > + div[OUT] = min_t(u32, 1024, div[OUT]); > > > > > > Hmm, this looks like we want to allow ideal ratio cases and p2p > > > cases to operate any way, even if the divider wasn't within the > > > range to get the in/out rates from the output clock? > > > > Yes. We still allow the p2p = true, ideal = false. Note that p2p is > > not Equal to ideal. > > Got it. > > Overall, I feel it's better to have a naming to state the purpose of using > ideal configurations without the IDEAL_RATIO_RATE setup. > bool use_ideal_rate; > And we can put into the asrc_config structure if there's no major problem. > Asrc_config may exposed to user, I don't think user need to care about The using of ideal rate or not. > So the condition check for the calculation would be: > + if (ideal && config->use_ideal_rate) > + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); > + else > + rem[OUT] = do_div(clk_rate, outrate); > + div[OUT] = clk_rate; > > And for that if (!ideal && div[IN]....rem[OUT]), I feel it would be clear to > have them separately, as the existing "div[IN] == 0" > and "div[OUT] == 0" checks, so that we can tell users which side of the > divider is out of range and what the sample rate and clock rate are. > Do you mean need to combine this judgement with "div[IN] == 0" Or "div[OUT] == 0"? Best regards Wang shengjiu

2 1

Re: [alsa-devel] [PATCH] ASoC: fsl_esai: Add spin lock to protect reset and stop
by S.j. Wang 25 Oct '19

25 Oct '19

Hi > > On Wed, Oct 23, 2019 at 03:29:49PM +0800, Shengjiu Wang wrote: > > xrun may happen at the end of stream, the > > trigger->fsl_esai_trigger_stop maybe called in the middle of > > fsl_esai_hw_reset, this may cause esai in wrong state after stop, and > > there may be endless xrun interrupt. > > What about fsl_esai_trigger_start? It touches ESAI_xFCR_xFEN bit that is > being checked in the beginning of fsl_esai_hw_reset. > > Could the scenario below be possible also? > > 1) ESAI TX starts > 2) Xrun happens to TX > 3) Starting fsl_esai_hw_reset (enabled[TX] = true; enabled[RX] = false) > 4) ESAI RX starts > 5) Finishing fsl_esai_hw_reset (enabled[RX] is still false) > > Good catch, this may possible. Will update in v2. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH v2 00/22] ASoC: soc-core cleanup - step 4
by Kuninori Morimoto 25 Oct '19

25 Oct '19

Hi Mark These are v2 of soc-core cleanup. These are related to topology, but according to Intel CI, there is no damage to it. Some of them need more review. I added some mark to such patch. I merged posted-but-not-yet-accepted patch on top of it. Kuninori Morimoto (22): ASoC: soc-pcm: fixup dpcm_prune_paths() loop continue ASoC: soc-core: remove for_each_rtdcom_safe() ASoC: soc-core: add for_each_rtd_components() and replace ASoC: soc-core: snd_soc_unbind_card() cleanup ASoC: soc-core: remove unneeded snd_soc_tplg_component_remove() ASoC: soc-core: move soc_init_dai_link() ASoC: soc-core: tidyup soc_init_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: call soc_bind_dai_link() under snd_soc_add_dai_link() ASoC: soc-core: add soc_unbind_dai_link() ASoC: soc-core: move snd_soc_lookup_component() ASoC: soc-core: add snd_soc_del_component_unlocked() ASoC: soc-core: remove snd_soc_component_add/del() ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() ASoC: soc-core: move snd_soc_register_dai() ASoC: soc-core: move snd_soc_unregister_dais() ASoC: soc-core: add snd_soc_unregister_dai() ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() ASoC: soc-core: call snd_soc_register_dai() from snd_soc_register_dais() ASoC: soc-core: remove topology specific operation ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY include/sound/soc.h | 25 ++- sound/soc/soc-component.c | 43 +--- sound/soc/soc-compress.c | 52 ++--- sound/soc/soc-core.c | 561 ++++++++++++++++++++-------------------------- sound/soc/soc-pcm.c | 55 ++--- sound/soc/soc-topology.c | 17 +- 6 files changed, 322 insertions(+), 431 deletions(-) -- 2.7.4

2 28

[alsa-devel] [PATCH] ASoC: soc-dpcm: tidyup for_each_dpcm_xx() macro
by Kuninori Morimoto 25 Oct '19

25 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> for_each_dpcm_xx() macro is using "dpcm" as parameter (1), but, it is also struct member (2). #define for_each_dpcm_fe(be, stream, dpcm) \ list_for_each_entry(dpcm, &(be)->dpcm[stream]...) ^^^^(1) ^^^^(2) Thus, it will be compile error if user not used "dpcm" as parameter for_each_dpcm_fe(be, stream, dp) ^^ This patch fixup it. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dpcm.h | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ff53037..fbe9e3d 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,13 +103,13 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; -#define for_each_dpcm_fe(be, stream, dpcm) \ - list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe) +#define for_each_dpcm_fe(be, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(be)->dpcm[stream].fe_clients, list_fe) -#define for_each_dpcm_be(fe, stream, dpcm) \ - list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be) -#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm) \ - list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be(fe, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_safe(fe, stream, _dpcm, __dpcm) \ + list_for_each_entry_safe(_dpcm, __dpcm, &(fe)->dpcm[stream].be_clients, list_be) /* can this BE stop and free */ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, -- 2.7.4

2 2

[alsa-devel] [PATCH v4 0/2] Add CML M/C for RT5682 and RT1011
by sathya.prakash.m.r＠intel.com 24 Oct '19

24 Oct '19

From: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Naveen Manohar (2): ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 ASoC: Intel: Add acpi match for rt1011 based m/c driver sound/soc/intel/boards/Kconfig | 18 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 473 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 500 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- V3 Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Changes from V3: repost with more recipients and edits to commit messages Changes from V2: re ordered match table to ensure non i2c device matches are at the end Changes from V1: Addressed comments on ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 1.9.1

3 5

[alsa-devel] KASAN: use-after-free Read in build_audio_procunit
by syzbot 24 Oct '19

24 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=17cdccc4e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=fd965c77e4711eb13b82 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=13fe752f600000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17989fd8e00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+fd965c77e4711eb13b82(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: use-after-free in uac_extension_unit_iExtension include/uapi/linux/usb/audio.h:483 [inline] BUG: KASAN: use-after-free in build_audio_procunit+0xeab/0x13f0 sound/usb/mixer.c:2434 Read of size 1 at addr ffff8881d4c0140d by task kworker/0:1/12 CPU: 0 PID: 12 Comm: kworker/0:1 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_extension_unit_iExtension include/uapi/linux/usb/audio.h:483 [inline] build_audio_procunit+0xeab/0x13f0 sound/usb/mixer.c:2434 parse_audio_extension_unit sound/usb/mixer.c:2483 [inline] parse_audio_unit+0x1812/0x36f0 sound/usb/mixer.c:2761 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 12: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_alloc_urb+0x65/0xb0 drivers/usb/core/urb.c:73 usb_internal_control_msg drivers/usb/core/message.c:94 [inline] usb_control_msg+0x1c9/0x4a0 drivers/usb/core/message.c:152 usb_get_descriptor+0xc1/0x1b0 drivers/usb/core/message.c:647 usb_get_configuration+0x28e/0x3050 drivers/usb/core/config.c:836 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 12: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 urb_destroy drivers/usb/core/urb.c:26 [inline] kref_put include/linux/kref.h:65 [inline] usb_free_urb.part.0+0x7a/0xc0 drivers/usb/core/urb.c:95 usb_free_urb+0x1b/0x30 drivers/usb/core/urb.c:94 usb_start_wait_urb+0x1e5/0x2b0 drivers/usb/core/message.c:79 usb_internal_control_msg drivers/usb/core/message.c:101 [inline] usb_control_msg+0x31c/0x4a0 drivers/usb/core/message.c:152 usb_get_descriptor+0xc1/0x1b0 drivers/usb/core/message.c:647 usb_get_configuration+0x28e/0x3050 drivers/usb/core/config.c:836 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 The buggy address belongs to the object at ffff8881d4c01400 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 13 bytes inside of 192-byte region [ffff8881d4c01400, ffff8881d4c014c0) The buggy address belongs to the page: page:ffffea0007530040 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea000754fec0 0000000200000002 ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d4c01300: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8881d4c01380: 00 00 00 07 fc fc fc fc fc fc fc fc fc fc fc fc > ffff8881d4c01400: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8881d4c01480: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d4c01500: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] KASAN: use-after-free Read in parse_term_proc_unit
by syzbot 24 Oct '19

24 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=13a38628e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=97e24236f1e2918ad968 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=12184764e00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1432ae28e00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+97e24236f1e2918ad968(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: use-after-free in uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] BUG: KASAN: use-after-free in parse_term_proc_unit+0x57a/0x5e0 sound/usb/mixer.c:896 Read of size 1 at addr ffff8881d5346d0e by task kworker/1:2/83 CPU: 1 PID: 83 Comm: kworker/1:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] parse_term_proc_unit+0x57a/0x5e0 sound/usb/mixer.c:896 __check_input_term+0xc32/0x13f0 sound/usb/mixer.c:984 parse_audio_feature_unit sound/usb/mixer.c:1875 [inline] parse_audio_unit+0x101d/0x36f0 sound/usb/mixer.c:2753 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 83: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_alloc_urb+0x65/0xb0 drivers/usb/core/urb.c:73 usb_internal_control_msg drivers/usb/core/message.c:94 [inline] usb_control_msg+0x1c9/0x4a0 drivers/usb/core/message.c:152 usb_get_descriptor+0xc1/0x1b0 drivers/usb/core/message.c:647 usb_get_configuration+0x28e/0x3050 drivers/usb/core/config.c:836 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 83: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 urb_destroy drivers/usb/core/urb.c:26 [inline] kref_put include/linux/kref.h:65 [inline] usb_free_urb.part.0+0x7a/0xc0 drivers/usb/core/urb.c:95 usb_free_urb+0x1b/0x30 drivers/usb/core/urb.c:94 usb_start_wait_urb+0x1e5/0x2b0 drivers/usb/core/message.c:79 usb_internal_control_msg drivers/usb/core/message.c:101 [inline] usb_control_msg+0x31c/0x4a0 drivers/usb/core/message.c:152 usb_get_descriptor+0xc1/0x1b0 drivers/usb/core/message.c:647 usb_get_configuration+0x28e/0x3050 drivers/usb/core/config.c:836 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 The buggy address belongs to the object at ffff8881d5346d00 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 14 bytes inside of 192-byte region [ffff8881d5346d00, ffff8881d5346dc0) The buggy address belongs to the page: page:ffffea000754d180 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea0007548ac0 0000000900000009 ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d5346c00: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8881d5346c80: 00 00 00 00 05 fc fc fc fc fc fc fc fc fc fc fc > ffff8881d5346d00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8881d5346d80: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d5346e00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix 2 front mics of codec 0x623
by Aaron Ma 24 Oct '19

24 Oct '19

These 2 ThinkCentres installed a new realtek codec ID 0x623, it has 2 front mics with the same location on pin 0x18 and 0x19. Apply fixup ALC283_FIXUP_HEADSET_MIC to change 1 front mic location to right, then pulseaudio can handle them. One "Front Mic" and one "Mic" will be shown, and audio output works fine. Signed-off-by: Aaron Ma <aaron.ma(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..c34d8b435f58 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7186,6 +7186,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), 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, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -9187,6 +9189,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269), HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0623, "ALC623", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), -- 2.17.1

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[alsa-devel] [PATCH 0/2] ASoC: hdmi-codec: fix locking issue
by Jerome Brunet 24 Oct '19

24 Oct '19

This patchset fixes the locking issue reported by Russell. As explained a mutex was used as flag and held while returning to userspace. Patch 2 is entirely optional and switches from bit atomic operation to mutex again. I tend to prefer bit atomic operation in this particular case but either way should be fine. Jerome Brunet (2): Revert "ASoC: hdmi-codec: re-introduce mutex locking" ASoC: hdmi-codec: re-introduce mutex locking again sound/soc/codecs/hdmi-codec.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH] ASoC: rt5650: Add Kahlee platform specfic changes
by Curtis Malainey 24 Oct '19

24 Oct '19

From: Akshu Agrawal <akshu.agrawal(a)amd.com> Add platform specific data for Kahlee project. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Shirish S <shirish.s(a)amd.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5645.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1c06b3b9218c5..a15e4ecd2a241 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3636,6 +3636,12 @@ static const struct rt5645_platform_data lattepanda_board_platform_data = { .inv_jd1_1 = true }; +static const struct rt5645_platform_data kahlee_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + static const struct dmi_system_id dmi_platform_data[] = { { .ident = "Chrome Buddy", @@ -3742,6 +3748,13 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&lattepanda_board_platform_data, }, + { + .ident = "Chrome Kahlee", + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Kahlee"), + }, + .driver_data = (void *)&kahlee_platform_data, + }, { } }; -- 2.24.0.rc0.303.g954a862665-goog

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[alsa-devel] [PATCH v3 0/5] soundwire: intel/cadence: better initialization
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

Changes since v2: addressed feedback from Vinod Koul on patch 2&4 Add kernel taint when using debugfs hw_reset (similar to regmap) Remove useless goto label Changes since v1: addressed feedback from Vinod Koul clarified init changes impact Intel and Cadence sides remove unnecessary intermediate variable disable interrupts when exit_reset fails, updated error handling returned -EINVAL on debugfs invalid parameter Pierre-Louis Bossart (5): soundwire: intel/cadence: fix startup sequence soundwire: cadence_master: add hw_reset capability in debugfs soundwire: intel: add helper for initialization soundwire: intel/cadence: add flag for interrupt enable soundwire: cadence_master: make clock stop exit configurable on init drivers/soundwire/cadence_master.c | 134 +++++++++++++++++++++-------- drivers/soundwire/cadence_master.h | 5 +- drivers/soundwire/intel.c | 39 ++++++--- 3 files changed, 129 insertions(+), 49 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: SOF: make hw_info platform specific
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

The SOF core blindly sets the hw_info, when it should really be platform-specific. This move enable us to temporarily fix PulseAudio issues resulting from inaccuracies in the hw_ptr position provided by the firmware on Baytrail-Broadwell. This is the first time we experiment with the GitHub low-level API to automagically keep track of 'approvers' by inserting a Reviewed-by tag when merging a pull-request, thanks to Ranjani for her initial work and bearing with my habit of breaking things :-) Pierre-Louis Bossart (2): ASoC: SOF: define INFO_ flags in dsp_ops ASoC: SOF: Intel: only support INFO_BATCH for legacy platforms sound/soc/sof/imx/imx8.c | 7 +++++++ sound/soc/sof/intel/apl.c | 7 +++++++ sound/soc/sof/intel/bdw.c | 9 ++++++++- sound/soc/sof/intel/byt.c | 21 +++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 7 +++++++ sound/soc/sof/pcm.c | 8 +++----- sound/soc/sof/sof-priv.h | 3 +++ 7 files changed, 56 insertions(+), 6 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] soundwire: slave: fix scanf format
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

fix cppcheck warning: [drivers/soundwire/slave.c:145]: (warning) %x in format string (no. 1) requires 'unsigned int *' but the argument type is 'signed int *'. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/slave.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 48a63ca130d2..6473fa602f82 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -128,7 +128,8 @@ int sdw_of_find_slaves(struct sdw_bus *bus) struct device_node *node; for_each_child_of_node(bus->dev->of_node, node) { - int link_id, sdw_version, ret, len; + int link_id, ret, len; + unsigned int sdw_version; const char *compat = NULL; struct sdw_slave_id id; const __be32 *addr; -- 2.20.1

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[alsa-devel] AC97 audio-routing [was: Recording on WM9712]
by Matej Kupljen 24 Oct '19

24 Oct '19

Hi all, I am not sure how to specify audio-routing for AC97 codec. If I use the similar setting as the UDOO board (which was the only board that uses AC97 codec I could find) I get warnings during booting: [ 3.261369] fsl-asoc-card sound: ac97-hifi <-> 202c000.ssi mapping ok [ 3.267917] fsl-asoc-card sound: ASoC: no sink widget found for Playback [ 3.274642] fsl-asoc-card sound: ASoC: Failed to add route AC97 Playback -> direct -> Playback [ 3.283309] fsl-asoc-card sound: ASoC: no source widget found for Capture [ 3.290141] fsl-asoc-card sound: ASoC: Failed to add route Capture -> direct -> AC97 Capture This was tested on kernel Linux version 4.14.134. However, the card is detected: [ 4.041646] ALSA device list: [ 4.044628] #0: ac97-audio and I can use it for playback normally, but the capture is "slow" as described in my previous e-mail. Then I tried to build and boot the latest kernel from GIT (Linux version 5.4.0-rc4) using the same DTS, I get additional warnings from the kernel: [ 2.639878] fsl-asoc-card sound: ac97-hifi <-> 202c000.ssi mapping ok [ 2.651321] fsl-asoc-card sound: ASoC: source widget AC97 Playback overwritten [ 2.658595] fsl-asoc-card sound: ASoC: no sink widget found for Playback [ 2.665317] fsl-asoc-card sound: ASoC: Failed to add route AC97 Playback -> direct -> Playback [ 2.673974] fsl-asoc-card sound: ASoC: sink widget AC97 Capture overwritten [ 2.680986] fsl-asoc-card sound: ASoC: no source widget found for Capture [ 2.687822] fsl-asoc-card sound: ASoC: Failed to add route Capture -> direct -> AC97 Capture [ 2.718636] fsl-asoc-card sound: snd_soc_register_card failed (-19) and the card is not detected: [ 3.285543] ALSA device list: [ 3.288518] No soundcards found. As you can see, there are additional warnings about widget overwrite, because this check was added to the latest kernel, as seen from the kernel source code. This is the setting from my DTS file: sound { compatible = "fsl,imx-audio-ac97"; model = "fsl,imx6q-ac97"; audio-cpu = <&ssi2>; audio-routing = "RX", "Mic Jack", "Headphone Jack", "TX"; mux-int-port = <2>; mux-ext-port = <5>; }; As you can see, I do not even specify any routing about AC97 Playback/Capture and I still get those warnings. For the test, I commented out the code in sound/soc/soc-core.c in function snd_soc_instantiate_card, to ignore the return values from the call to snd_soc_dapm_add_routes and then I can use the card for playback, but the recording has the issues again. What seems to be wrong with this AC97 audio routing? Any suggestions? I am willing to test this on the latest kernel, or from some other GIT repo if needed. Thanks and BR, Matej

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add Gustard U16/X26 device family to quirks for native DSD support
by Fly Pig 24 Oct '19

24 Oct '19

Hi There This patch add native DSD support for Gustard U16/X26 USB Interface. Tested on my unit, working properly. Hence submit the patch. Cheers. Thank you very much. Pig --- a/sound/usb/quirks.c 2019-09-22 21:28:09.757897000 +0000 +++ b/sound/usb/quirks.c 2019-09-22 21:33:36.892274800 +0000 @@ -1369,6 +1369,7 @@ case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ + case USB_ID(0x292b, 0xc4b3): /* Gustard U16 */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break;

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[alsa-devel] [asoc:for-5.5 160/172] sound/soc/codecs/rt5677-spi.c:148:33: sparse: sparse: Using plain integer as NULL pointer
by kbuild test robot 24 Oct '19

24 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 175fc928198236037174e5c5c066fe3c4691903e commit: a0e0d135427cf699fe2dee77da0924e0b47f3170 [160/172] ASoC: rt5677: Add a PCM device for streaming hotword via SPI reproduce: # apt-get install sparse # sparse version: v0.6.1-dirty git checkout a0e0d135427cf699fe2dee77da0924e0b47f3170 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/rt5677-spi.c:148:33: sparse: sparse: Using plain integer as NULL pointer >> sound/soc/codecs/rt5677-spi.c:365:13: sparse: sparse: symbol 'rt5677_spi_pcm_page' was not declared. Should it be static? vim +148 sound/soc/codecs/rt5677-spi.c 139 140 static int rt5677_spi_hw_free( 141 struct snd_soc_component *component, 142 struct snd_pcm_substream *substream) 143 { 144 struct rt5677_dsp *rt5677_dsp = 145 snd_soc_component_get_drvdata(component); 146 147 mutex_lock(&rt5677_dsp->dma_lock); > 148 rt5677_dsp->substream = 0; 149 mutex_unlock(&rt5677_dsp->dma_lock); 150 151 return snd_pcm_lib_free_vmalloc_buffer(substream); 152 } 153 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 00/18] soundwire: code hardening and suspend-resume support
by Pierre-Louis Bossart 23 Oct '19

23 Oct '19

this patchset applies on top of "[PATCH 00/14] soundwire: intel: implement new ASoC interfaces". It implements a series of improvements for: a) interrupt handling on Intel platforms in MSI mode b) race conditions on codec probe and enumeration c) suspend-resume issues (clock-stop mode not supported for now) d) underflow handling e) updates to the stream state machine which did not support valid ALSA transitions. These patches were tested extensively on 4 different platforms and are viewed as required for any sort of SoundWire-based product. Bard Liao (3): soundwire: intel/cadence: fix timeouts in MSI mode soundwire: stream: only prepare stream when it is configured. soundwire: intel: reinitialize IP+DSP in .prepare() Pierre-Louis Bossart (15): soundwire: fix race between driver probe and update_status callback soundwire: bus: add PM/no-PM versions of read/write functions soundwire: bus: write Slave Device Number without runtime_pm soundwire: intel: add helpers for link power down and shim wake soundwire: intel: Add basic power management support soundwire: intel: add pm_runtime support soundwire: intel: reset pm_runtime status during system resume soundwire: bus: add helper to reset Slave status to UNATTACHED soundwire: intel: call helper to reset Slave states on resume soundwire: bus: check first if Slaves become UNATTACHED soundwire: add enumeration_complete signaling soundwire: intel: disable pm_runtime when removing a master soundwire: bus: disable pm_runtime in sdw_slave_delete soundwire: stream: update state machine and add state checks soundwire: stream: do not update parameters during DISABLED-PREPARED transition Documentation/driver-api/soundwire/stream.rst | 63 +++- drivers/soundwire/bus.c | 157 +++++++-- drivers/soundwire/bus.h | 3 + drivers/soundwire/bus_type.c | 5 + drivers/soundwire/cadence_master.c | 17 +- drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 328 ++++++++++++++++-- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 45 ++- drivers/soundwire/slave.c | 3 + drivers/soundwire/stream.c | 64 +++- include/linux/soundwire/sdw.h | 1 + include/linux/soundwire/sdw_intel.h | 2 + 13 files changed, 623 insertions(+), 71 deletions(-) -- 2.20.1

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Re: [alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by S.j. Wang 23 Oct '19

23 Oct '19

Hi > > On Thu, Oct 17, 2019 at 02:21:08PM +0800, Shengjiu Wang wrote: > > For P2P output, the output divider should align with the output sample > > I think we should avoid "P2P" (or "M2M") keyword in the mainline code as > we know M2M will never get merged while somebody working with the > mainline and caring about new feature might be confused. Ok. But we still curious that is there a way to upstream m2m? > > > rate, if use ideal sample rate, there will be a lot of overload, which > > would cause underrun. > > If I understand it correctly, setting to ideal ratio provides a faster converting > speed but increases the load of the processor of ASRC. So we choose a > slower converting speed here since real- time playback mode doesn't really > need a faster conversion? Yes. Slower speed is enough for real-time playback > > It makes sense to me, yet I feel that the delay at the beginning of the audio > playback might be longer as a compromise. I am okay with this decision > though... > > > The maximum divider of asrc clock is 1024, but there is no judgement > > for this limitaion in driver, which may cause the divider setting not > > correct. > > > > For non-ideal ratio mode, the clock rate should divide the sample rate > > with no remainder, and the quotient should be less than 1024. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 40 > > +++++++++++++++++++++++++++++++--------- > > 1 file changed, 31 insertions(+), 9 deletions(-) > > > > diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index > > 0bf91a6f54b9..44d05ec28bd3 100644 > > --- a/sound/soc/fsl/fsl_asrc.c > > +++ b/sound/soc/fsl/fsl_asrc.c > > @@ -260,7 +260,7 @@ static int fsl_asrc_set_ideal_ratio(struct > fsl_asrc_pair *pair, > > * of struct asrc_config which includes in/output sample rate, width, > channel > > * and clock settings. > > */ > > -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) > > +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p) > > { > > struct asrc_config *config = pair->config; > > struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +268,8 > > @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) > > enum asrc_word_width input_word_width; > > enum asrc_word_width output_word_width; > > u32 inrate, outrate, indiv, outdiv; > > - u32 clk_index[2], div[2]; > > + u32 clk_index[2], div[2], rem[2]; > > + u64 clk_rate; > > int in, out, channels; > > int pre_proc, post_proc; > > struct clk *clk; > > @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair > *pair) > > /* We only have output clock for ideal ratio mode */ > > clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; > > > > - div[IN] = clk_get_rate(clk) / inrate; > > + clk_rate = clk_get_rate(clk); > > The fsl_asrc.c file has config.inclk being set to INCLK_NONE and this sets the > "ideal" in this function to true. So, although we tend to not use ideal ratio > setting for p2p cases, yet the input clock is still not physically connected, so > we still use output clock for div[IN] calculation? For p2p case, it can be ideal or non-ideal. For non-ideal, we still use Output clock for div calculation. > > I am thinking something simplier: if we decided not to use ideal ratio for > "P2P", instead of adding "bool p2p" with the confusing "ideal" in this > function, could we just set config.inclk to the same clock as the output one > for "P2P"? By doing so, "P2P" won't go through ideal ratio mode while still > having a clock rate from the output clock for div[IN] calculation here. Bool p2p is to force output rate to be sample rate, no impact to ideal Ratio mode. > > > + rem[IN] = do_div(clk_rate, inrate); > > + div[IN] = (u32)clk_rate; > > if (div[IN] == 0) { > > Could we check div[IN] and rem[IN] here? Like: > if (div[IN] == 0 || div[IN] > 1024) { > pair_err(); > goto out; > } > > if (!ideal && rem[IN]) { > pair_err(); > goto out; > } > > According to your commit log, I think the max-1024 limitation should be > applied to all cases, not confined to "!ideal" cases right? And we should > add some comments also, indicating it is limited by hardware. For ideal mode, my test result is the divider not impact the output result. Which means it is ok for ideal mode even divider is not correct... > > > pair_err("failed to support input sample rate %dHz by > asrck_%x\n", > > inrate, clk_index[ideal ? OUT : IN]); @@ > > -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct > > fsl_asrc_pair *pair) > > > > clk = asrc_priv->asrck_clk[clk_index[OUT]]; > > > > - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ > > - if (ideal) > > - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; > > - else > > - div[OUT] = clk_get_rate(clk) / outrate; > > + /* > > + * When P2P mode, output rate should align with the out samplerate. > > + * if set too high output rate, there will be lots of Overload. > > + * When M2M mode, output rate should also need to align with the > > + out > > For this "should", do you actually mean "M2M could also"? Sorry, I'm just > trying to understand everyting here, not intentionally being picky at words. > My understanding is that we still keep the ideal ratio setting because > "M2M" still uses it. We use IDEAL_RATIO_RATE as output rate for m2m mode, it likes a Tricky operation, in order to improve the performance. I think The correct operation is to use the real output rate, but the performance Is bad. So it is a compromise. > > > + * samplerate, but M2M must use less time to achieve good > performance. > > + */ > > + clk_rate = clk_get_rate(clk); > > + if (p2p || !ideal) { > > + rem[OUT] = do_div(clk_rate, outrate); > > + div[OUT] = clk_rate; > > + } else { > > + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); > > + div[OUT] = clk_rate; > > + } > > > > if (div[OUT] == 0) { > > pair_err("failed to support output sample rate %dHz by > > asrck_%x\n", @@ -372,6 +384,16 @@ static int fsl_asrc_config_pair(struct > fsl_asrc_pair *pair) > > return -EINVAL; > > } > > > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || > > + rem[IN] != 0 || rem[OUT] != 0)) { > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || rem[IN] || > > + rem[OUT] != 0)) { > > So for ideal == true, these limitaions are not applied any more? > Remember that the "ideal" is true for "p2p == true" cases here. No, not applied. for ideal, the div don't impact the output result Even it is not accurate. > > > + pair_err("The divider can't be used for non ideal mode\n"); > > + return -EINVAL; > > + } > > + > > + /* Divider range is [1, 1024] */ > > + div[IN] = min_t(u32, 1024, div[IN]); > > + div[OUT] = min_t(u32, 1024, div[OUT]); > > Hmm, this looks like we want to allow ideal ratio cases and p2p cases to > operate any way, even if the divider wasn't within the range to get the > in/out rates from the output clock? Yes. We still allow the p2p = true, ideal = false. Note that p2p is not Equal to ideal. Best regards Wang shengjiu

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[alsa-devel] [PATCH] ASoC: rsnd: dma: set bus width to data width for monaural data
by Eugeniu Rosca 23 Oct '19

23 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> According to R-Car3 HW manual 40.3.3 (Data Format on Audio Local Bus), in case of monaural data writing or reading through Audio-DMAC, it's always in Left Justified format, so both src and dst DMA Bus width should be equal to physical data width. Therefore set src and dst's DMA bus width to: - [monaural case] data width - [non-monaural case] 32bits (as prior applying the patch) Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Cc: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- sound/soc/sh/rcar/dma.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 28f65eba2bb4..95aa26d62e4f 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -165,14 +165,40 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; + enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; int is_play = rsnd_io_is_play(io); int ret; + /* + * in case of monaural data writing or reading through Audio-DMAC + * data is always in Left Justified format, so both src and dst + * DMA Bus width need to be set equal to physical data width. + */ + if (rsnd_runtime_channel_original(io) == 1) { + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + int bits = snd_pcm_format_physical_width(runtime->format); + + switch (bits) { + case 8: + buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; + break; + case 16: + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(dev, "invalid format width %d\n", bits); + return -EINVAL; + } + } + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; cfg.src_addr = dma->src_addr; cfg.dst_addr = dma->dst_addr; - cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.src_addr_width = buswidth; + cfg.dst_addr_width = buswidth; dev_dbg(dev, "%s %pad -> %pad\n", rsnd_mod_name(mod), -- 2.23.0

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[alsa-devel] [PATCH] ASoC: rsnd: add missing of_node_put()
by Kuninori Morimoto 23 Oct '19

23 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch adds missing of_node_put() for rsnd_parse_tdm_split_mode() rsnd_parse_connect_graph() Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index fe681c5..14e86e4 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1075,7 +1075,10 @@ static void rsnd_parse_tdm_split_mode(struct rsnd_priv *priv, j++; } + of_node_put(node); } + + of_node_put(ssiu_np); } static void rsnd_parse_connect_simple(struct rsnd_priv *priv, @@ -1093,11 +1096,13 @@ static void rsnd_parse_connect_graph(struct rsnd_priv *priv, struct device_node *endpoint) { struct device *dev = rsnd_priv_to_dev(priv); - struct device_node *remote_node = of_graph_get_remote_port_parent(endpoint); + struct device_node *remote_node; if (!rsnd_io_to_mod_ssi(io)) return; + remote_node = of_graph_get_remote_port_parent(endpoint); + /* HDMI0 */ if (strstr(remote_node->full_name, "hdmi@fead0000")) { rsnd_flags_set(io, RSND_STREAM_HDMI0); @@ -1111,6 +1116,8 @@ static void rsnd_parse_connect_graph(struct rsnd_priv *priv, } rsnd_parse_tdm_split_mode(priv, io, endpoint); + + of_node_put(remote_node); } void rsnd_parse_connect_common(struct rsnd_dai *rdai, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: mediatek: Check SND_SOC_CROS_EC_CODEC dependency
by Mao Wenan 23 Oct '19

23 Oct '19

If SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A=y, below errors can be seen: sound/soc/codecs/cros_ec_codec.o: In function `send_ec_host_command': cros_ec_codec.c:(.text+0x534): undefined reference to `cros_ec_cmd_xfer_status' cros_ec_codec.c:(.text+0x101c): undefined reference to `cros_ec_get_host_event' This is because it will select SND_SOC_CROS_EC_CODEC after commit 2cc3cd5fdc8b ("ASoC: mediatek: mt8183: support WoV"), but SND_SOC_CROS_EC_CODEC depends on CROS_EC. Fixes: 2cc3cd5fdc8b ("ASoC: mediatek: mt8183: support WoV") Signed-off-by: Mao Wenan <maowenan(a)huawei.com> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 8b29f39..a656d20 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -125,7 +125,7 @@ config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A select SND_SOC_MAX98357A select SND_SOC_BT_SCO select SND_SOC_TS3A227E - select SND_SOC_CROS_EC_CODEC + select SND_SOC_CROS_EC_CODEC if CROS_EC help This adds ASoC driver for Mediatek MT8183 boards with the MT6358 TS3A227E MAX98357A audio codec. -- 2.7.4

4 6

[alsa-devel] [PATCH] ASoC: kirkwood: fix external clock probe defer
by Russell King 23 Oct '19

23 Oct '19

When our call to get the external clock fails, we forget to clean up the enabled internal clock correctly. Enable the clock after we have obtained all our resources. Fixes: 84aac6c79bfd ("ASoC: kirkwood: fix loss of external clock at probe time") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/kirkwood/kirkwood-i2s.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3446a113f482..c323ae314b55 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -559,10 +559,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return PTR_ERR(priv->clk); } - err = clk_prepare_enable(priv->clk); - if (err < 0) - return err; - priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (IS_ERR(priv->extclk)) { if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) @@ -578,6 +574,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } } + err = clk_prepare_enable(priv->clk); + if (err < 0) + return err; + /* Some sensible defaults - this reflects the powerup values */ priv->ctl_play = KIRKWOOD_PLAYCTL_SIZE_24; priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; -- 2.20.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: SOF: Intel: improve loader error handling
by Pierre-Louis Bossart 23 Oct '19

23 Oct '19

Pierre-Louis Bossart (2): ASoC: SOF: Intel: hda-loader: improve error handling ASoC: SOF: Intel: hda: add dev_err() traces for snd_sof_dsp_read_poll_timeout() sound/soc/sof/intel/hda-dsp.c | 30 +++++++++++++++++++++++++++--- sound/soc/sof/intel/hda-loader.c | 25 ++++++++++++++++++++++--- sound/soc/sof/intel/hda-stream.c | 24 ++++++++++++++++++++---- 3 files changed, 69 insertions(+), 10 deletions(-) -- 2.20.1

2 4

[alsa-devel] [PATCH 0/2] ASoC: SOF: initial support for JasperLake
by Pierre-Louis Bossart 23 Oct '19

23 Oct '19

Add PCI IDs and dummy tables for ACPI matching (to be populated when machine drivers are provided, used for nocodec mode for now). Pan Xiuli (2): ASoC: Intel: common: add ACPI matching tables for JSL ASoC: SOF: Intel: initial support to JasperLake. include/sound/soc-acpi-intel-match.h | 1 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-jsl-match.c | 18 +++++++++++++++ sound/soc/sof/intel/Kconfig | 16 ++++++++++++++ sound/soc/sof/intel/cnl.c | 17 ++++++++++++++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/sof-pci-dev.c | 22 +++++++++++++++++++ 7 files changed, 76 insertions(+) create mode 100644 sound/soc/intel/common/soc-acpi-intel-jsl-match.c -- 2.20.1

2 4

[alsa-devel] [PATCH] ASoC: kirkwood: fix IRQ error handling
by Russell King 23 Oct '19

23 Oct '19

Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 6f69f314f2c2..d2d5c25bf550 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.20.1

2 1

[alsa-devel] [RESEND PATCH] ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address
by Eugeniu Rosca 23 Oct '19

23 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit's busif dma address is calculated by following calculation formula: 0xec540000 + 0x1000 * id + busif / 4 * 0xA000 + busif % 4 * 0x400 But according to R-Car3 HW manual 41.1.4 Register Configuration, ssi9 4/5/6/7 busif data register address (SSI9_4_BUSIF/SSI9_5_BUSIF/SSI9_6_BUSIF/SSI9_7_BUSIF) are out of this rule. This patch updates the calculation formula to correct ssi9 4/5/6/7 busif data register address. Fixes: 5e45a6fab3b9 ("ASoc: rsnd: dma: Calculate dma address with consider of BUSIF") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> [erosca: minor improvements in commit description] Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Cc: stable(a)vger.kernel.org # v4.20+ Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- Originally submitted as https://patchwork.kernel.org/patch/10825513/ ("ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address") --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0324a5c39619..28f65eba2bb4 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -508,10 +508,10 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) -- 2.23.0

3 3

[alsa-devel] [PATCH] ASoC: kirkwood: fix device remove ordering
by Russell King 23 Oct '19

23 Oct '19

The devm conversion of kirkwood was incorrect; on removal, devm takes effect after the "remove" function has returned. So, the effect of the conversion was to change the order during remove from: - snd_soc_unregister_component() (unpublishes interfaces) - clk_disable_unprepare() - cleanup resources After the conversion, this became: - clk_disable_unprepare() - while the device may still be active - snd_soc_unregister_component() - cleanup resources Hence, it introduces a bug, where the internal clock for the device may be shut down before the device itself has been shut down. It is known that Marvell SoCs, including Dove, locks up if registers for a peripheral that has its clocks disabled are accessed. Fixes: f98fc0f8154e ("ASoC: kirkwood: replace platform to component") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/kirkwood/kirkwood-i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index c323ae314b55..eb38cdb37f0e 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -591,7 +591,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_128; } - err = devm_snd_soc_register_component(&pdev->dev, &kirkwood_soc_component, + err = snd_soc_register_component(&pdev->dev, &kirkwood_soc_component, soc_dai, 2); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); @@ -614,6 +614,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); if (!IS_ERR(priv->extclk)) clk_disable_unprepare(priv->extclk); clk_disable_unprepare(priv->clk); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_esai: Add spin lock to protect reset and stop
by Shengjiu Wang 23 Oct '19

23 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. So Add spin lock to lock these two function. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..6a797648b66d 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock to handle reset and stop behavior * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect reset and stop */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -734,7 +741,9 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1011,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

2 1

Re: [alsa-devel] Question about "957ce0c6b8a1 ASoC: soc-pcm: check symmetry after hw_params"
by S.j. Wang 23 Oct '19

23 Oct '19

Hi Mark > > On Wed, Oct 16, 2019 at 03:05:16AM +0000, S.j. Wang wrote: > > > The soc_pcm_params_symmetry can't take effect, because its > > position is after the change of cpu_dai->rate. > > > Is this an issue? > > I think so, yes. Thanks, but how to fix it? because I don't really understand this commit want to do, it is hard for me to send a fixed patch. Or shall we revert it? Best regards Wang shengjiu

2 1

[alsa-devel] alsa-info: lots of mostly syntactic changes
by GitHub pull_request - opened 23 Oct '19

23 Oct '19

alsa-project/alsa-utils pull request #29 was opened from E5ten: - remove all bashisms and change shebang to #!/bin/sh - replace use of which with command -v for portability - remove awk dependency - general cosmetic consistency: - indentation - always put then or done on the same line as if, for, or while - quote much more, especially when the variable refers to a file - replace many uses of >> $FILE with command grouping and single redirects of the command groups - simplify text processing pipelines, mostly by turning them into a single sed command Request URL : https://github.com/alsa-project/alsa-utils/pull/29 Patch URL : https://github.com/alsa-project/alsa-utils/pull/29.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [RFC] sof-skl_hda: change to use common HDMI codec driver
by GitHub pull_request - opened 23 Oct '19

23 Oct '19

alsa-project/alsa-ucm-conf pull request #4 was opened from kv2019i: Here's a companion patch to the hda-hdmi kernel patchset on alsa-devel: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/157201.h… Modify UCM sequences to use user-space interface of the common HDMI codec driver. Use of hdac-hdmi driver is no longer supported by this UCM file. To use this UCM file, kernel must be compiled with SND_SOC_SOF_HDA_COMMON_HDMI_CODEC Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/4 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

2 1

[alsa-devel] alsaloop change avail_min question
by ed nwave 23 Oct '19

23 Oct '19

Hi, Over time I see the message below and the value slowly creeps up. Is this a simple information output or more of a warning? If run alsaloop for a long time - many hours of music, will alsaloop be prone to failure because of this value increasing? playback plughw:Siso: change avail_min=7299 Thanks.

2 2

[alsa-devel] [PATCH] ASoC: Intel: Reduce Resource notification trace to once per minute.
by vani.ganji＠intel.com 23 Oct '19

23 Oct '19

From: Vani Ganji <vani.ganji(a)intel.com> Resource notification trace "MCPS Budget violation" observed very frequently which might cause losing critical debug information. Added a counter to print trace on first instance and once per minute. It is required to be notified when module resource is used more than a threshold so enabled once per minute. Suggested-by: Grant Grundler <grundler(a)google.com> Signed-off-by: Vani Ganji <vani.ganji(a)intel.com> --- sound/soc/intel/skylake/skl-sst-ipc.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 667cdddc289f..03ff0d8cea3e 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -346,6 +346,7 @@ int skl_ipc_process_notification(struct sst_generic_ipc *ipc, struct skl_ipc_header header) { struct skl_dev *skl = container_of(ipc, struct skl_dev, ipc); + static unsigned long j; if (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { switch (IPC_GLB_NOTIFY_TYPE(header.primary)) { @@ -355,12 +356,17 @@ int skl_ipc_process_notification(struct sst_generic_ipc *ipc, break; case IPC_GLB_NOTIFY_RESOURCE_EVENT: - dev_err(ipc->dev, "MCPS Budget Violation: %x\n", + /*Print trace on first instance and for every minute*/ + if (printk_timed_ratelimit(&j, 60000)) { + dev_err(ipc->dev, "MCPS Budget Violation: %x\n", header.primary); + } break; case IPC_GLB_NOTIFY_FW_READY: skl->boot_complete = true; + /* reset timer on every FW load */ + j = 0; wake_up(&skl->boot_wait); break; -- 2.17.1

1 0

[alsa-devel] [PATCH 00/36] ARM: samsung platform cleanup
by Arnd Bergmann 23 Oct '19

23 Oct '19

I've spent some time looking at the remaining ARMv4/ARMv5 platforms that are not part of ARCH_MULTIPLATFORM, and tried to get them closer to that. Here is what came out of that for the samsung platforms: * Exynos and s5pv210 are made independent of plat-samsung * device drivers stop using mach/*.h headers for s3c24xx (and other platforms not in this series) * s3c24xx and s3c64xx get merged into mach-s3c, removing the need for plat-samsung (I have other patches for the remaining plat-* directories) * mach/io.h gets cleaned up to only be needed for BAST PC104 mode (looking for ideas to proceed) * mach/irqs.h remains for now, this still needs to be converted to sparse IRQs. Some bits are a little ugly, but overall I think this a big improvement. The contents are available for testing in git://kernel.org:/pub/scm/linux/kernel/git/arnd/playground.git s3c-multiplatform Given the overall size of the series I expect that there will be bugs, so at this points I'm looking for reviews, acks and testing from anyone who still has the hardware, in particular s3c24xx, but also the other samsung platforms I'm touching. Arnd Arnd Bergmann (36): ARM: samsung: make S3C24XX_MISCCR access indirect ARM: s3c: simplify mach/io.h usb: gadget: s3c: use platform resources usb: gadget: s3c-hsudc: remove platform header dependency ARM: samsung: make pm-debug platform independent ARM: samsung: move CONFIG_DEBUG_S3C_UART to Kconfig.debug ARM: exynos: use private samsung_cpu_id copy ARM: exynos: stop selecting PLAT_SAMSUNG ARM: samsung: move pm check code to drivers/soc ARM: s5pv210: use private pm save/restore ARM: s5pv210: split from plat-samsung ARM: s3c64xx: remove mach/hardware.h ARM: s3c: move regs-spi.h into spi driver ARM: s3c: move irqchip driver back into platform ARM: s3c: adc: move header to linux/soc/samsung ARM: s3c: move spi fiq handler into platform ARM: s3c: h1940-audio: turn into platform driver ARM: s3c: gta02-audio: turn into platform driver ARM: s3c: rx1950: turn into platform driver ASoC: samsung: s3c2412-i2s: avoid hardcoded S3C2410_PA_IIS ARM: s3c: move iis pinctrl config into boards ARM: s3c: leds: move setpull() calls into board files ARM: s3c: move s3cmci pinctrl handling into board files ARM: s3c: include mach/irqs.h where needed ARM: s3c: spi: avoid hardcoding fiq number in driver ARM: s3c: bast: avoid irq_desc array usage ARM: s3c: fix mmc gpio lookup tables fbdev: s3c2410fb: remove mach header dependency ARM: s3c: cpufreq: split out registers ARM: s3c: remove cpufreq header dependencies ARM: s3c: cpufreq: use global s3c2412_cpufreq_setrefresh ARM: s3c: iotiming: make functions static ARM: s3c: move low-level clk reg access into platform code ARM: s3c: stop including mach/hardware.h from mach/io.h ARM: s3c: move into a common directory ARM: s3c: make headers local if possible MAINTAINERS | 16 +- arch/arm/Kconfig | 5 +- arch/arm/Kconfig.debug | 18 +- arch/arm/Makefile | 7 +- arch/arm/mach-exynos/Makefile | 4 - arch/arm/mach-exynos/common.h | 6 +- arch/arm/mach-exynos/exynos.c | 19 +- arch/arm/mach-exynos/include/mach/map.h | 18 -- arch/arm/mach-exynos/platsmp.c | 4 +- arch/arm/mach-exynos/pm.c | 8 +- arch/arm/{plat-samsung => mach-s3c}/Kconfig | 62 +---- .../Kconfig => mach-s3c/Kconfig.s3c24xx} | 0 .../Kconfig => mach-s3c/Kconfig.s3c64xx} | 0 arch/arm/{plat-samsung => mach-s3c}/Makefile | 15 +- .../{mach-s3c24xx => mach-s3c}/Makefile.boot | 0 .../Makefile => mach-s3c/Makefile.s3c24xx} | 15 +- .../Makefile => mach-s3c/Makefile.s3c64xx} | 14 +- .../include/plat => mach-s3c}/adc-core.h | 0 arch/arm/{plat-samsung => mach-s3c}/adc.c | 4 +- arch/arm/{mach-s3c24xx => mach-s3c}/anubis.h | 0 .../arm/{mach-s3c64xx => mach-s3c}/ata-core.h | 0 .../{mach-s3c64xx => mach-s3c}/backlight.h | 0 .../arm/{mach-s3c24xx => mach-s3c}/bast-ide.c | 3 +- .../arm/{mach-s3c24xx => mach-s3c}/bast-irq.c | 8 +- arch/arm/{mach-s3c24xx => mach-s3c}/bast.h | 0 .../{mach-s3c24xx => mach-s3c}/common-smdk.c | 17 +- .../{mach-s3c24xx => mach-s3c}/common-smdk.h | 0 arch/arm/{plat-samsung => mach-s3c}/cpu.c | 21 +- .../include/plat => mach-s3c}/cpu.h | 11 - .../cpufreq-utils.c | 38 +++- arch/arm/{mach-s3c64xx => mach-s3c}/cpuidle.c | 4 +- .../arm/{mach-s3c64xx => mach-s3c}/crag6410.h | 2 +- .../{mach-s3c64xx => mach-s3c}/dev-audio.c | 9 +- .../dev-backlight.c | 4 +- .../dev-uart-s3c64xx.c} | 5 +- .../arm/{plat-samsung => mach-s3c}/dev-uart.c | 2 +- arch/arm/{plat-samsung => mach-s3c}/devs.c | 56 +++-- .../include/plat => mach-s3c}/devs.h | 0 .../mach/dma.h => mach-s3c/dma-s3c24xx.h} | 0 .../mach/dma.h => mach-s3c/dma-s3c64xx.h} | 0 arch/arm/mach-s3c/dma.h | 7 + arch/arm/{mach-s3c24xx => mach-s3c}/fb-core.h | 0 .../include/plat => mach-s3c}/fb.h | 0 .../plat => mach-s3c}/gpio-cfg-helpers.h | 0 .../include/plat => mach-s3c}/gpio-cfg.h | 0 .../include/plat => mach-s3c}/gpio-core.h | 2 +- .../gpio-samsung-s3c24xx.h} | 2 + .../gpio-samsung-s3c64xx.h} | 0 .../{plat-samsung => mach-s3c}/gpio-samsung.c | 18 +- arch/arm/mach-s3c/gpio-samsung.h | 7 + arch/arm/{mach-s3c24xx => mach-s3c}/gta02.h | 2 +- .../h1940-bluetooth.c | 7 +- arch/arm/{mach-s3c24xx => mach-s3c}/h1940.h | 0 .../include/mach => mach-s3c}/hardware.h | 7 - .../include/plat => mach-s3c}/iic-core.h | 0 arch/arm/mach-s3c/include/mach/io.h | 50 +++++ .../include/mach/irqs-s3c24xx.h} | 0 .../include/mach/irqs-s3c64xx.h} | 0 arch/arm/mach-s3c/include/mach/irqs.h | 7 + .../plat => mach-s3c/include/mach}/map-base.h | 0 arch/arm/{plat-samsung => mach-s3c}/init.c | 4 +- .../iotiming-s3c2410.c | 12 +- .../iotiming-s3c2412.c | 6 +- .../irq-pm.c => mach-s3c/irq-pm-s3c24xx.c} | 16 +- .../irq-pm.c => mach-s3c/irq-pm-s3c64xx.c} | 8 +- arch/arm/mach-s3c/irq-s3c24xx-fiq-exports.c | 9 + .../arm/mach-s3c/irq-s3c24xx-fiq.S | 10 +- .../arm/mach-s3c}/irq-s3c24xx.c | 23 +- .../arm/{mach-s3c64xx => mach-s3c}/irq-uart.h | 0 .../include/plat => mach-s3c}/keypad.h | 0 .../mach-amlm5900.c | 37 ++- .../{mach-s3c24xx => mach-s3c}/mach-anubis.c | 14 +- .../{mach-s3c64xx => mach-s3c}/mach-anw6410.c | 17 +- .../mach-at2440evb.c | 28 ++- .../{mach-s3c24xx => mach-s3c}/mach-bast.c | 22 +- .../mach-crag6410-module.c | 2 +- .../mach-crag6410.c | 27 ++- .../{mach-s3c24xx => mach-s3c}/mach-gta02.c | 57 ++++- .../{mach-s3c24xx => mach-s3c}/mach-h1940.c | 65 ++++-- .../arm/{mach-s3c64xx => mach-s3c}/mach-hmt.c | 15 +- .../{mach-s3c24xx => mach-s3c}/mach-jive.c | 27 ++- .../mach-mini2440.c | 48 ++-- .../mach-mini6410.c | 22 +- .../arm/{mach-s3c24xx => mach-s3c}/mach-n30.c | 40 +++- .../arm/{mach-s3c64xx => mach-s3c}/mach-ncp.c | 13 +- .../mach-nexcoder.c | 20 +- .../mach-osiris-dvs.c | 4 +- .../{mach-s3c24xx => mach-s3c}/mach-osiris.c | 22 +- .../{mach-s3c24xx => mach-s3c}/mach-otom.c | 17 +- .../{mach-s3c24xx => mach-s3c}/mach-qt2410.c | 37 ++- .../mach-real6410.c | 18 +- .../{mach-s3c24xx => mach-s3c}/mach-rx1950.c | 57 +++-- .../{mach-s3c24xx => mach-s3c}/mach-rx3715.c | 26 ++- .../mach-s3c2416-dt.c | 8 +- .../mach-s3c64xx-dt.c | 6 +- .../{mach-s3c64xx => mach-s3c}/mach-smartq.c | 18 +- .../{mach-s3c64xx => mach-s3c}/mach-smartq.h | 0 .../{mach-s3c64xx => mach-s3c}/mach-smartq5.c | 18 +- .../{mach-s3c64xx => mach-s3c}/mach-smartq7.c | 18 +- .../mach-smdk2410.c | 14 +- .../mach-smdk2413.c | 21 +- .../mach-smdk2416.c | 23 +- .../mach-smdk2440.c | 25 ++- .../mach-smdk2443.c | 14 +- .../mach-smdk6400.c | 13 +- .../mach-smdk6410.c | 23 +- .../mach-tct_hammer.c | 24 +- .../{mach-s3c24xx => mach-s3c}/mach-vr1000.c | 20 +- .../{mach-s3c24xx => mach-s3c}/mach-vstms.c | 20 +- .../include/plat => mach-s3c}/map-s3c.h | 12 +- .../mach/map.h => mach-s3c/map-s3c24xx.h} | 6 +- .../mach/map.h => mach-s3c/map-s3c64xx.h} | 4 +- .../include/plat => mach-s3c}/map-s5p.h | 4 +- arch/arm/mach-s3c/map.h | 7 + .../{mach-s3c24xx => mach-s3c}/nand-core.h | 0 .../{mach-s3c64xx => mach-s3c}/onenand-core.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/osiris.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/otom.h | 0 arch/arm/{mach-s3c64xx => mach-s3c}/pl080.c | 4 +- .../{plat-samsung => mach-s3c}/platformdata.c | 4 +- .../{mach-s3c24xx => mach-s3c}/pll-s3c2410.c | 4 +- .../pll-s3c2440-12000000.c | 4 +- .../pll-s3c2440-16934400.c | 4 +- .../{plat-samsung => mach-s3c}/pm-common.c | 2 +- arch/arm/mach-s3c/pm-common.h | 40 ++++ .../pm-core.h => mach-s3c/pm-core-s3c24xx.h} | 8 +- .../pm-core.h => mach-s3c/pm-core-s3c64xx.h} | 50 +---- arch/arm/mach-s3c/pm-core.h | 7 + arch/arm/{plat-samsung => mach-s3c}/pm-gpio.c | 6 +- .../arm/{mach-s3c24xx => mach-s3c}/pm-h1940.S | 5 +- .../{mach-s3c24xx => mach-s3c}/pm-s3c2410.c | 11 +- .../{mach-s3c24xx => mach-s3c}/pm-s3c2412.c | 11 +- .../{mach-s3c24xx => mach-s3c}/pm-s3c2416.c | 6 +- .../pm.c => mach-s3c/pm-s3c24xx.c} | 12 +- .../pm.c => mach-s3c/pm-s3c64xx.c} | 66 +++++- arch/arm/{plat-samsung => mach-s3c}/pm.c | 20 +- .../include/plat => mach-s3c}/pm.h | 2 +- .../include/plat => mach-s3c}/pwm-core.h | 0 .../include/plat => mach-s3c}/regs-adc.h | 0 .../regs-clock-s3c24xx.h} | 2 + .../regs-clock-s3c64xx.h} | 0 arch/arm/mach-s3c/regs-clock.h | 7 + .../arm/{mach-s3c24xx => mach-s3c}/regs-dsc.h | 0 .../regs-gpio-memport.h | 0 .../regs-gpio-s3c24xx.h} | 2 + .../regs-gpio-s3c64xx.h} | 0 arch/arm/mach-s3c/regs-gpio.h | 7 + .../regs-irq-s3c24xx.h} | 2 + .../regs-irq-s3c64xx.h} | 0 arch/arm/mach-s3c/regs-irq.h | 7 + .../include/plat => mach-s3c}/regs-irqtype.h | 0 .../arm/{mach-s3c24xx => mach-s3c}/regs-mem.h | 2 + .../{mach-s3c64xx => mach-s3c}/regs-modem.h | 0 .../mach => mach-s3c}/regs-s3c2443-clock.h | 50 +++++ .../{mach-s3c64xx => mach-s3c}/regs-srom.h | 0 .../arm/{mach-s3c64xx => mach-s3c}/regs-sys.h | 0 .../regs-syscon-power.h | 0 .../regs-usb-hsotg-phy.h | 0 .../include/mach => mach-s3c}/rtc-core.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/s3c2410.c | 21 +- .../s3c2412-power.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/s3c2412.c | 16 +- .../include/mach => mach-s3c}/s3c2412.h | 2 + arch/arm/{mach-s3c24xx => mach-s3c}/s3c2416.c | 28 +-- arch/arm/{mach-s3c24xx => mach-s3c}/s3c2440.c | 17 +- arch/arm/{mach-s3c24xx => mach-s3c}/s3c2442.c | 17 +- arch/arm/{mach-s3c24xx => mach-s3c}/s3c2443.c | 21 +- arch/arm/{mach-s3c24xx => mach-s3c}/s3c244x.c | 16 +- .../common.c => mach-s3c/s3c24xx.c} | 21 +- .../common.h => mach-s3c/s3c24xx.h} | 1 + arch/arm/{mach-s3c64xx => mach-s3c}/s3c6400.c | 14 +- arch/arm/{mach-s3c64xx => mach-s3c}/s3c6410.c | 17 +- .../common.c => mach-s3c/s3c64xx.c} | 24 +- .../common.h => mach-s3c/s3c64xx.h} | 0 .../include/plat => mach-s3c}/samsung-time.h | 0 .../include/plat => mach-s3c}/sdhci.h | 2 +- .../{mach-s3c24xx => mach-s3c}/setup-camif.c | 4 +- .../setup-fb-24bpp.c | 6 +- .../{mach-s3c24xx => mach-s3c}/setup-i2c.c | 7 +- .../{mach-s3c64xx => mach-s3c}/setup-i2c0.c | 4 +- .../{mach-s3c64xx => mach-s3c}/setup-i2c1.c | 4 +- .../{mach-s3c64xx => mach-s3c}/setup-ide.c | 8 +- .../{mach-s3c64xx => mach-s3c}/setup-keypad.c | 6 +- .../setup-sdhci-gpio-s3c24xx.c} | 6 +- .../setup-sdhci-gpio-s3c64xx.c} | 6 +- .../setup-spi-s3c24xx.c} | 6 +- .../setup-spi-s3c64xx.c} | 4 +- .../arm/{mach-s3c24xx => mach-s3c}/setup-ts.c | 5 +- .../setup-usb-phy.c | 6 +- .../{mach-s3c24xx => mach-s3c}/simtec-audio.c | 11 +- .../{mach-s3c24xx => mach-s3c}/simtec-nor.c | 2 +- .../{mach-s3c24xx => mach-s3c}/simtec-pm.c | 8 +- .../{mach-s3c24xx => mach-s3c}/simtec-usb.c | 6 +- arch/arm/{mach-s3c24xx => mach-s3c}/simtec.h | 0 .../sleep-s3c2410.S | 7 +- .../sleep-s3c2412.S | 5 +- .../sleep.S => mach-s3c/sleep-s3c24xx.S} | 7 +- .../sleep.S => mach-s3c/sleep-s3c64xx.S} | 4 +- .../arm/{mach-s3c24xx => mach-s3c}/spi-core.h | 0 .../include/plat => mach-s3c}/usb-phy.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/vr1000.h | 0 .../{plat-samsung => mach-s3c}/wakeup-mask.c | 4 +- .../include/plat => mach-s3c}/wakeup-mask.h | 0 .../watchdog-reset.c | 0 .../watchdog-reset.h | 0 arch/arm/mach-s3c24xx/include/mach/fb.h | 2 - arch/arm/mach-s3c24xx/include/mach/io.h | 212 ------------------ arch/arm/mach-s3c64xx/include/mach/hardware.h | 17 -- arch/arm/mach-s5pv210/Kconfig | 1 + arch/arm/mach-s5pv210/Makefile | 7 - arch/arm/mach-s5pv210/pm.c | 51 ++++- arch/arm/mach-s5pv210/regs-clock.h | 2 +- arch/arm/mach-s5pv210/s5pv210.c | 2 - drivers/clk/samsung/clk-s3c2410-dclk.c | 10 +- drivers/cpufreq/s3c2410-cpufreq.c | 10 +- drivers/cpufreq/s3c2412-cpufreq.c | 44 ++-- drivers/cpufreq/s3c2440-cpufreq.c | 29 ++- drivers/cpufreq/s3c24xx-cpufreq-debugfs.c | 2 +- drivers/cpufreq/s3c24xx-cpufreq.c | 14 +- drivers/hwmon/s3c-hwmon.c | 2 +- drivers/input/touchscreen/s3c2410_ts.c | 37 ++- drivers/irqchip/Makefile | 1 - drivers/leds/leds-s3c24xx.c | 7 - drivers/mmc/host/Kconfig | 2 +- drivers/mmc/host/s3cmci.c | 72 ++---- drivers/power/supply/s3c_adc_battery.c | 2 +- drivers/pwm/Kconfig | 2 +- drivers/soc/samsung/Kconfig | 48 +++- drivers/soc/samsung/Makefile | 3 + .../soc/samsung/s3c-pm-check.c | 2 +- .../soc/samsung/s3c-pm-debug.c | 26 +-- drivers/spi/Kconfig | 2 +- drivers/spi/Makefile | 1 - .../spi/spi-s3c24xx-regs.h | 3 +- drivers/spi/spi-s3c24xx.c | 28 +-- drivers/tty/serial/Kconfig | 2 +- drivers/usb/gadget/udc/s3c-hsudc.c | 55 +---- drivers/usb/gadget/udc/s3c2410_udc.c | 31 +-- drivers/usb/gadget/udc/s3c2410_udc.h | 1 + .../usb/gadget/udc/s3c2410_udc_regs.h | 2 + .../video/fbdev/s3c2410fb-regs-lcd.h | 28 +-- drivers/video/fbdev/s3c2410fb.c | 16 +- .../linux/platform_data}/fb-s3c2410.h | 33 ++- include/linux/platform_data/mmc-s3cmci.h | 2 + include/linux/platform_data/s3c-hsudc.h | 2 + .../linux/soc/samsung/s3c-adc.h | 0 .../linux/soc/samsung/s3c-cpu-freq.h | 0 .../linux/soc/samsung/s3c-cpufreq-core.h | 10 +- .../linux/soc/samsung/s3c-pm.h | 69 +++--- .../linux/spi/s3c24xx-fiq.h | 5 + include/linux/spi/s3c24xx.h | 2 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/h1940_uda1380.c | 71 ++---- sound/soc/samsung/neo1973_wm8753.c | 85 +++---- sound/soc/samsung/rx1950_uda1380.c | 72 ++---- sound/soc/samsung/s3c-i2s-v2.c | 3 +- sound/soc/samsung/s3c-i2s-v2.h | 3 +- sound/soc/samsung/s3c2412-i2s.c | 9 +- sound/soc/samsung/s3c24xx-i2s.c | 7 - 259 files changed, 1716 insertions(+), 1648 deletions(-) delete mode 100644 arch/arm/mach-exynos/include/mach/map.h rename arch/arm/{plat-samsung => mach-s3c}/Kconfig (73%) rename arch/arm/{mach-s3c24xx/Kconfig => mach-s3c/Kconfig.s3c24xx} (100%) rename arch/arm/{mach-s3c64xx/Kconfig => mach-s3c/Kconfig.s3c64xx} (100%) rename arch/arm/{plat-samsung => mach-s3c}/Makefile (67%) rename arch/arm/{mach-s3c24xx => mach-s3c}/Makefile.boot (100%) rename arch/arm/{mach-s3c24xx/Makefile => mach-s3c/Makefile.s3c24xx} (86%) rename arch/arm/{mach-s3c64xx/Makefile => mach-s3c/Makefile.s3c64xx} (83%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/adc-core.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/adc.c (99%) rename arch/arm/{mach-s3c24xx => mach-s3c}/anubis.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/ata-core.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/backlight.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/bast-ide.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/bast-irq.c (95%) rename arch/arm/{mach-s3c24xx => mach-s3c}/bast.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/common-smdk.c (93%) rename arch/arm/{mach-s3c24xx => mach-s3c}/common-smdk.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/cpu.c (62%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/cpu.h (91%) rename arch/arm/{mach-s3c24xx => mach-s3c}/cpufreq-utils.c (70%) rename arch/arm/{mach-s3c64xx => mach-s3c}/cpuidle.c (96%) rename arch/arm/{mach-s3c64xx => mach-s3c}/crag6410.h (95%) rename arch/arm/{mach-s3c64xx => mach-s3c}/dev-audio.c (97%) rename arch/arm/{mach-s3c64xx => mach-s3c}/dev-backlight.c (98%) rename arch/arm/{mach-s3c64xx/dev-uart.c => mach-s3c/dev-uart-s3c64xx.c} (95%) rename arch/arm/{plat-samsung => mach-s3c}/dev-uart.c (97%) rename arch/arm/{plat-samsung => mach-s3c}/devs.c (96%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/devs.h (100%) rename arch/arm/{mach-s3c24xx/include/mach/dma.h => mach-s3c/dma-s3c24xx.h} (100%) rename arch/arm/{mach-s3c64xx/include/mach/dma.h => mach-s3c/dma-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/dma.h rename arch/arm/{mach-s3c24xx => mach-s3c}/fb-core.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/fb.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/gpio-cfg-helpers.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/gpio-cfg.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/gpio-core.h (99%) rename arch/arm/{mach-s3c24xx/include/mach/gpio-samsung.h => mach-s3c/gpio-samsung-s3c24xx.h} (99%) rename arch/arm/{mach-s3c64xx/include/mach/gpio-samsung.h => mach-s3c/gpio-samsung-s3c64xx.h} (100%) rename arch/arm/{plat-samsung => mach-s3c}/gpio-samsung.c (99%) create mode 100644 arch/arm/mach-s3c/gpio-samsung.h rename arch/arm/{mach-s3c24xx => mach-s3c}/gta02.h (94%) rename arch/arm/{mach-s3c24xx => mach-s3c}/h1940-bluetooth.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/h1940.h (100%) rename arch/arm/{mach-s3c24xx/include/mach => mach-s3c}/hardware.h (76%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/iic-core.h (100%) create mode 100644 arch/arm/mach-s3c/include/mach/io.h rename arch/arm/{mach-s3c24xx/include/mach/irqs.h => mach-s3c/include/mach/irqs-s3c24xx.h} (100%) rename arch/arm/{mach-s3c64xx/include/mach/irqs.h => mach-s3c/include/mach/irqs-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/include/mach/irqs.h rename arch/arm/{plat-samsung/include/plat => mach-s3c/include/mach}/map-base.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/init.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/iotiming-s3c2410.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/iotiming-s3c2412.c (98%) rename arch/arm/{mach-s3c24xx/irq-pm.c => mach-s3c/irq-pm-s3c24xx.c} (93%) rename arch/arm/{mach-s3c64xx/irq-pm.c => mach-s3c/irq-pm-s3c64xx.c} (97%) create mode 100644 arch/arm/mach-s3c/irq-s3c24xx-fiq-exports.c rename drivers/spi/spi-s3c24xx-fiq.S => arch/arm/mach-s3c/irq-s3c24xx-fiq.S (94%) rename {drivers/irqchip => arch/arm/mach-s3c}/irq-s3c24xx.c (99%) rename arch/arm/{mach-s3c64xx => mach-s3c}/irq-uart.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/keypad.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-amlm5900.c (84%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-anubis.c (97%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-anw6410.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-at2440evb.c (88%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-bast.c (98%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-crag6410-module.c (99%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-crag6410.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-gta02.c (90%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-h1940.c (92%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-hmt.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-jive.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-mini2440.c (93%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-mini6410.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-n30.c (93%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-ncp.c (92%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-nexcoder.c (92%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-osiris-dvs.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-osiris.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-otom.c (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-qt2410.c (88%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-real6410.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-rx1950.c (93%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-rx3715.c (90%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-s3c2416-dt.c (92%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-s3c64xx-dt.c (95%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smartq.c (97%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smartq.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smartq5.c (93%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smartq7.c (94%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2410.c (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2413.c (90%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2416.c (95%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2440.c (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2443.c (93%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smdk6400.c (92%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smdk6410.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-tct_hammer.c (82%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-vr1000.c (95%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-vstms.c (90%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/map-s3c.h (87%) rename arch/arm/{mach-s3c24xx/include/mach/map.h => mach-s3c/map-s3c24xx.h} (97%) rename arch/arm/{mach-s3c64xx/include/mach/map.h => mach-s3c/map-s3c64xx.h} (98%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/map-s5p.h (85%) create mode 100644 arch/arm/mach-s3c/map.h rename arch/arm/{mach-s3c24xx => mach-s3c}/nand-core.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/onenand-core.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/osiris.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/otom.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/pl080.c (99%) rename arch/arm/{plat-samsung => mach-s3c}/platformdata.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pll-s3c2410.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pll-s3c2440-12000000.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pll-s3c2440-16934400.c (98%) rename arch/arm/{plat-samsung => mach-s3c}/pm-common.c (98%) create mode 100644 arch/arm/mach-s3c/pm-common.h rename arch/arm/{mach-s3c24xx/include/mach/pm-core.h => mach-s3c/pm-core-s3c24xx.h} (95%) rename arch/arm/{mach-s3c64xx/include/mach/pm-core.h => mach-s3c/pm-core-s3c64xx.h} (58%) create mode 100644 arch/arm/mach-s3c/pm-core.h rename arch/arm/{plat-samsung => mach-s3c}/pm-gpio.c (99%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pm-h1940.S (80%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pm-s3c2410.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pm-s3c2412.c (95%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pm-s3c2416.c (95%) rename arch/arm/{mach-s3c24xx/pm.c => mach-s3c/pm-s3c24xx.c} (94%) rename arch/arm/{mach-s3c64xx/pm.c => mach-s3c/pm-s3c64xx.c} (83%) rename arch/arm/{plat-samsung => mach-s3c}/pm.c (94%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/pm.h (98%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/pwm-core.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/regs-adc.h (100%) rename arch/arm/{mach-s3c24xx/include/mach/regs-clock.h => mach-s3c/regs-clock-s3c24xx.h} (99%) rename arch/arm/{mach-s3c64xx/include/mach/regs-clock.h => mach-s3c/regs-clock-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/regs-clock.h rename arch/arm/{mach-s3c24xx => mach-s3c}/regs-dsc.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-gpio-memport.h (100%) rename arch/arm/{mach-s3c24xx/include/mach/regs-gpio.h => mach-s3c/regs-gpio-s3c24xx.h} (99%) rename arch/arm/{mach-s3c64xx/include/mach/regs-gpio.h => mach-s3c/regs-gpio-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/regs-gpio.h rename arch/arm/{mach-s3c24xx/include/mach/regs-irq.h => mach-s3c/regs-irq-s3c24xx.h} (98%) rename arch/arm/{mach-s3c64xx/include/mach/regs-irq.h => mach-s3c/regs-irq-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/regs-irq.h rename arch/arm/{plat-samsung/include/plat => mach-s3c}/regs-irqtype.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/regs-mem.h (98%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-modem.h (100%) rename arch/arm/{mach-s3c24xx/include/mach => mach-s3c}/regs-s3c2443-clock.h (81%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-srom.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-sys.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-syscon-power.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-usb-hsotg-phy.h (100%) rename arch/arm/{mach-s3c24xx/include/mach => mach-s3c}/rtc-core.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2410.c (90%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2412-power.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2412.c (94%) rename arch/arm/{mach-s3c24xx/include/mach => mach-s3c}/s3c2412.h (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2416.c (88%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2440.c (85%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2442.c (82%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2443.c (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c244x.c (93%) rename arch/arm/{mach-s3c24xx/common.c => mach-s3c/s3c24xx.c} (98%) rename arch/arm/{mach-s3c24xx/common.h => mach-s3c/s3c24xx.h} (99%) rename arch/arm/{mach-s3c64xx => mach-s3c}/s3c6400.c (89%) rename arch/arm/{mach-s3c64xx => mach-s3c}/s3c6410.c (89%) rename arch/arm/{mach-s3c64xx/common.c => mach-s3c/s3c64xx.c} (97%) rename arch/arm/{mach-s3c64xx/common.h => mach-s3c/s3c64xx.h} (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/samsung-time.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/sdhci.h (99%) rename arch/arm/{mach-s3c24xx => mach-s3c}/setup-camif.c (95%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-fb-24bpp.c (86%) rename arch/arm/{mach-s3c24xx => mach-s3c}/setup-i2c.c (79%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-i2c0.c (90%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-i2c1.c (90%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-ide.c (89%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-keypad.c (86%) rename arch/arm/{mach-s3c24xx/setup-sdhci-gpio.c => mach-s3c/setup-sdhci-gpio-s3c24xx.c} (90%) rename arch/arm/{mach-s3c64xx/setup-sdhci-gpio.c => mach-s3c/setup-sdhci-gpio-s3c64xx.c} (95%) rename arch/arm/{mach-s3c24xx/setup-spi.c => mach-s3c/setup-spi-s3c24xx.c} (85%) rename arch/arm/{mach-s3c64xx/setup-spi.c => mach-s3c/setup-spi-s3c64xx.c} (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/setup-ts.c (88%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-usb-phy.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec-audio.c (86%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec-nor.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec-pm.c (92%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec-usb.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/sleep-s3c2410.S (92%) rename arch/arm/{mach-s3c24xx => mach-s3c}/sleep-s3c2412.S (93%) rename arch/arm/{mach-s3c24xx/sleep.S => mach-s3c/sleep-s3c24xx.S} (92%) rename arch/arm/{mach-s3c64xx/sleep.S => mach-s3c/sleep-s3c64xx.S} (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/spi-core.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/usb-phy.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/vr1000.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/wakeup-mask.c (94%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/wakeup-mask.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/watchdog-reset.c (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/watchdog-reset.h (100%) delete mode 100644 arch/arm/mach-s3c24xx/include/mach/fb.h delete mode 100644 arch/arm/mach-s3c24xx/include/mach/io.h delete mode 100644 arch/arm/mach-s3c64xx/include/mach/hardware.h rename arch/arm/plat-samsung/pm-check.c => drivers/soc/samsung/s3c-pm-check.c (99%) rename arch/arm/plat-samsung/pm-debug.c => drivers/soc/samsung/s3c-pm-debug.c (78%) rename arch/arm/plat-samsung/include/plat/regs-spi.h => drivers/spi/spi-s3c24xx-regs.h (95%) rename arch/arm/plat-samsung/include/plat/regs-udc.h => drivers/usb/gadget/udc/s3c2410_udc_regs.h (99%) rename arch/arm/mach-s3c24xx/include/mach/regs-lcd.h => drivers/video/fbdev/s3c2410fb-regs-lcd.h (84%) rename {arch/arm/plat-samsung/include/plat => include/linux/platform_data}/fb-s3c2410.h (57%) rename arch/arm/plat-samsung/include/plat/adc.h => include/linux/soc/samsung/s3c-adc.h (100%) rename arch/arm/plat-samsung/include/plat/cpu-freq.h => include/linux/soc/samsung/s3c-cpu-freq.h (100%) rename arch/arm/plat-samsung/include/plat/cpu-freq-core.h => include/linux/soc/samsung/s3c-cpufreq-core.h (96%) rename arch/arm/plat-samsung/include/plat/pm-common.h => include/linux/soc/samsung/s3c-pm.h (55%) rename drivers/spi/spi-s3c24xx-fiq.h => include/linux/spi/s3c24xx-fiq.h (78%) Cc: linux-kernel(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: patches(a)opensource.cirrus.com Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-clk(a)vger.kernel.org Cc: linux-hwmon(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-pwm(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-serial(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-fbdev(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org -- 2.20.0

3 3

[alsa-devel] [PATCH 11/36] ARM: s5pv210: split from plat-samsung
by Arnd Bergmann 23 Oct '19

23 Oct '19

These can be build completely independently, so split the two Kconfig symbols. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/Kconfig.debug | 6 +++--- arch/arm/Makefile | 1 - arch/arm/plat-samsung/Kconfig | 2 +- drivers/mmc/host/Kconfig | 2 +- drivers/pwm/Kconfig | 2 +- drivers/spi/Kconfig | 2 +- drivers/tty/serial/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 8 files changed, 9 insertions(+), 10 deletions(-) diff --git a/arch/arm/Kconfig.debug b/arch/arm/Kconfig.debug index 9c4f2d6deb06..4c4e97ae4fcb 100644 --- a/arch/arm/Kconfig.debug +++ b/arch/arm/Kconfig.debug @@ -998,7 +998,7 @@ choice via SCIFA4 on Renesas SH-Mobile AG5 (SH73A0). config DEBUG_S3C_UART0 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX @@ -1010,7 +1010,7 @@ choice by the boot-loader before use. config DEBUG_S3C_UART1 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX @@ -1022,7 +1022,7 @@ choice by the boot-loader before use. config DEBUG_S3C_UART2 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX diff --git a/arch/arm/Makefile b/arch/arm/Makefile index f492d7c338fe..a1bc15cda751 100644 --- a/arch/arm/Makefile +++ b/arch/arm/Makefile @@ -235,7 +235,6 @@ machine-$(CONFIG_PLAT_SPEAR) += spear # by CONFIG_* macro name. plat-$(CONFIG_ARCH_OMAP) += omap plat-$(CONFIG_ARCH_S3C64XX) += samsung -plat-$(CONFIG_ARCH_S5PV210) += samsung plat-$(CONFIG_PLAT_ORION) += orion plat-$(CONFIG_PLAT_PXA) += pxa plat-$(CONFIG_PLAT_S3C24XX) += samsung diff --git a/arch/arm/plat-samsung/Kconfig b/arch/arm/plat-samsung/Kconfig index 740bdb23f38a..1530946cc672 100644 --- a/arch/arm/plat-samsung/Kconfig +++ b/arch/arm/plat-samsung/Kconfig @@ -4,7 +4,7 @@ config PLAT_SAMSUNG bool - depends on PLAT_S3C24XX || ARCH_S3C64XX || ARCH_S5PV210 + depends on PLAT_S3C24XX || ARCH_S3C64XX default y select GENERIC_IRQ_CHIP select NO_IOPORT_MAP diff --git a/drivers/mmc/host/Kconfig b/drivers/mmc/host/Kconfig index 400a581c918c..16a0e5430b44 100644 --- a/drivers/mmc/host/Kconfig +++ b/drivers/mmc/host/Kconfig @@ -275,7 +275,7 @@ config MMC_SDHCI_TEGRA config MMC_SDHCI_S3C tristate "SDHCI support on Samsung S3C SoC" - depends on MMC_SDHCI && (PLAT_SAMSUNG || ARCH_EXYNOS) + depends on MMC_SDHCI && (PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS) help This selects the Secure Digital Host Controller Interface (SDHCI) often referrered to as the HSMMC block in some of the Samsung S3C diff --git a/drivers/pwm/Kconfig b/drivers/pwm/Kconfig index e3a2518503ed..8eb738cac0c7 100644 --- a/drivers/pwm/Kconfig +++ b/drivers/pwm/Kconfig @@ -394,7 +394,7 @@ config PWM_ROCKCHIP config PWM_SAMSUNG tristate "Samsung PWM support" - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS help Generic PWM framework driver for Samsung. diff --git a/drivers/spi/Kconfig b/drivers/spi/Kconfig index 6f7fdcbb9151..355391ee643d 100644 --- a/drivers/spi/Kconfig +++ b/drivers/spi/Kconfig @@ -625,7 +625,7 @@ config SPI_S3C24XX_FIQ config SPI_S3C64XX tristate "Samsung S3C64XX series type SPI" - depends on (PLAT_SAMSUNG || ARCH_EXYNOS || COMPILE_TEST) + depends on (PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS || COMPILE_TEST) help SPI driver for Samsung S3C64XX and newer SoCs. diff --git a/drivers/tty/serial/Kconfig b/drivers/tty/serial/Kconfig index 4789b5d62f63..17f01cf3009c 100644 --- a/drivers/tty/serial/Kconfig +++ b/drivers/tty/serial/Kconfig @@ -237,7 +237,7 @@ config SERIAL_CLPS711X_CONSOLE config SERIAL_SAMSUNG tristate "Samsung SoC serial support" - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select SERIAL_CORE help Support for the on-chip UARTs on the Samsung S3C24XX series CPUs, diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 638983123d8f..7a0035dd9995 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on PLAT_SAMSUNG || ARCH_EXYNOS || COMPILE_TEST + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS || COMPILE_TEST depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM ---help--- -- 2.20.0

5 9

[alsa-devel] [PATCH 21/36] ARM: s3c: move iis pinctrl config into boards
by Arnd Bergmann 23 Oct '19

23 Oct '19

The s3c_gpio_cfgall_range() function is an internal interface of the samsung gpio driver and should not be called directly by drivers, so move the iis pin initialization into the boards. Note that the s3c2412-i2s driver has no boards using it in mainline linux, the driver gets selected for the jive machine but is never instantiated. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-gta02.c | 4 ++++ arch/arm/mach-s3c24xx/mach-h1940.c | 3 +++ arch/arm/mach-s3c24xx/mach-mini2440.c | 5 +++++ arch/arm/mach-s3c24xx/mach-n30.c | 5 +++++ arch/arm/mach-s3c24xx/mach-nexcoder.c | 5 +++++ arch/arm/mach-s3c24xx/mach-otom.c | 6 ++++++ arch/arm/mach-s3c24xx/mach-qt2410.c | 3 +++ arch/arm/mach-s3c24xx/mach-rx1950.c | 3 +++ arch/arm/mach-s3c24xx/mach-rx3715.c | 4 ++++ arch/arm/mach-s3c24xx/mach-smdk2410.c | 5 +++++ arch/arm/mach-s3c24xx/mach-smdk2413.c | 4 ++++ arch/arm/mach-s3c24xx/mach-smdk2440.c | 6 +++++- arch/arm/mach-s3c24xx/mach-vstms.c | 6 +++++- arch/arm/mach-s3c24xx/simtec-audio.c | 6 ++++++ sound/soc/samsung/s3c2412-i2s.c | 7 ------- sound/soc/samsung/s3c24xx-i2s.c | 7 ------- 16 files changed, 63 insertions(+), 16 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-gta02.c b/arch/arm/mach-s3c24xx/mach-gta02.c index 526fd0933289..1ca0460d82f4 100644 --- a/arch/arm/mach-s3c24xx/mach-gta02.c +++ b/arch/arm/mach-s3c24xx/mach-gta02.c @@ -540,6 +540,10 @@ static void __init gta02_machine_init(void) i2c_register_board_info(0, gta02_i2c_devs, ARRAY_SIZE(gta02_i2c_devs)); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + gpiod_add_lookup_table(&gta02_audio_gpio_table); platform_add_devices(gta02_devices, ARRAY_SIZE(gta02_devices)); pm_power_off = gta02_poweroff; diff --git a/arch/arm/mach-s3c24xx/mach-h1940.c b/arch/arm/mach-s3c24xx/mach-h1940.c index 922d53bacbdf..11f705fb5e32 100644 --- a/arch/arm/mach-s3c24xx/mach-h1940.c +++ b/arch/arm/mach-s3c24xx/mach-h1940.c @@ -709,6 +709,9 @@ static void __init h1940_init(void) s3c24xx_fb_set_platdata(&h1940_fb_info); gpiod_add_lookup_table(&h1940_mmc_gpio_table); gpiod_add_lookup_table(&h1940_audio_gpio_table); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); s3c24xx_mci_set_platdata(&h1940_mmc_cfg); s3c24xx_udc_set_platdata(&h1940_udc_cfg); s3c24xx_ts_set_platdata(&h1940_ts_cfg); diff --git a/arch/arm/mach-s3c24xx/mach-mini2440.c b/arch/arm/mach-s3c24xx/mach-mini2440.c index 9035f868fb34..bb44b8704d64 100644 --- a/arch/arm/mach-s3c24xx/mach-mini2440.c +++ b/arch/arm/mach-s3c24xx/mach-mini2440.c @@ -685,6 +685,11 @@ static void __init mini2440_init(void) s3c_gpio_setpull(mini2440_buttons[i].gpio, S3C_GPIO_PULL_UP); s3c_gpio_cfgpin(mini2440_buttons[i].gpio, S3C2410_GPIO_INPUT); } + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + if (features.lcd_index != -1) { int li; diff --git a/arch/arm/mach-s3c24xx/mach-n30.c b/arch/arm/mach-s3c24xx/mach-n30.c index d856f23939af..1836f1cc0992 100644 --- a/arch/arm/mach-s3c24xx/mach-n30.c +++ b/arch/arm/mach-s3c24xx/mach-n30.c @@ -36,6 +36,7 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> #include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <asm/mach/arch.h> #include <asm/mach/irq.h> @@ -570,6 +571,10 @@ static void __init n30_init(void) S3C2410_MISCCR_USBSUSPND0 | S3C2410_MISCCR_USBSUSPND1, 0x0); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + if (machine_is_n30()) { /* Turn off suspend on both USB ports, and switch the * selectable USB port to USB device mode. */ diff --git a/arch/arm/mach-s3c24xx/mach-nexcoder.c b/arch/arm/mach-s3c24xx/mach-nexcoder.c index c2f34758ccb6..a05c92b6de2e 100644 --- a/arch/arm/mach-s3c24xx/mach-nexcoder.c +++ b/arch/arm/mach-s3c24xx/mach-nexcoder.c @@ -145,6 +145,11 @@ static void __init nexcoder_init_time(void) static void __init nexcoder_init(void) { s3c_i2c0_set_platdata(NULL); + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + platform_add_devices(nexcoder_devices, ARRAY_SIZE(nexcoder_devices)); }; diff --git a/arch/arm/mach-s3c24xx/mach-otom.c b/arch/arm/mach-s3c24xx/mach-otom.c index 4e24d89e870b..c834f16e5ba0 100644 --- a/arch/arm/mach-s3c24xx/mach-otom.c +++ b/arch/arm/mach-s3c24xx/mach-otom.c @@ -24,6 +24,8 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <plat/cpu.h> #include <plat/devs.h> @@ -107,6 +109,10 @@ static void __init otom11_init_time(void) static void __init otom11_init(void) { s3c_i2c0_set_platdata(NULL); + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(otom11_devices, ARRAY_SIZE(otom11_devices)); } diff --git a/arch/arm/mach-s3c24xx/mach-qt2410.c b/arch/arm/mach-s3c24xx/mach-qt2410.c index 5d48e5b6e738..3cf8423c76c6 100644 --- a/arch/arm/mach-s3c24xx/mach-qt2410.c +++ b/arch/arm/mach-s3c24xx/mach-qt2410.c @@ -337,6 +337,9 @@ static void __init qt2410_machine_init(void) s3c24xx_udc_set_platdata(&qt2410_udc_cfg); s3c_i2c0_set_platdata(NULL); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); gpiod_add_lookup_table(&qt2410_spi_gpiod_table); platform_add_devices(qt2410_devices, ARRAY_SIZE(qt2410_devices)); s3c_pm_init(); diff --git a/arch/arm/mach-s3c24xx/mach-rx1950.c b/arch/arm/mach-s3c24xx/mach-rx1950.c index 9f3b0f18cd36..4eeede398775 100644 --- a/arch/arm/mach-s3c24xx/mach-rx1950.c +++ b/arch/arm/mach-s3c24xx/mach-rx1950.c @@ -825,6 +825,9 @@ static void __init rx1950_init_machine(void) pwm_add_table(rx1950_pwm_lookup, ARRAY_SIZE(rx1950_pwm_lookup)); gpiod_add_lookup_table(&rx1950_audio_gpio_table); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(rx1950_devices, ARRAY_SIZE(rx1950_devices)); i2c_register_board_info(0, rx1950_i2c_devices, diff --git a/arch/arm/mach-s3c24xx/mach-rx3715.c b/arch/arm/mach-s3c24xx/mach-rx3715.c index 529c6faf862f..2542c523ad46 100644 --- a/arch/arm/mach-s3c24xx/mach-rx3715.c +++ b/arch/arm/mach-s3c24xx/mach-rx3715.c @@ -39,6 +39,7 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> #include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <plat/cpu.h> #include <plat/devs.h> @@ -199,6 +200,9 @@ static void __init rx3715_init_machine(void) s3c_nand_set_platdata(&rx3715_nand_info); s3c24xx_fb_set_platdata(&rx3715_fb_info); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(rx3715_devices, ARRAY_SIZE(rx3715_devices)); } diff --git a/arch/arm/mach-s3c24xx/mach-smdk2410.c b/arch/arm/mach-s3c24xx/mach-smdk2410.c index 18dfef52c8bf..02adaac4d6df 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2410.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2410.c @@ -19,6 +19,8 @@ #include <linux/serial_s3c.h> #include <linux/platform_device.h> #include <linux/io.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <asm/mach/arch.h> #include <asm/mach/map.h> @@ -94,6 +96,9 @@ static void __init smdk2410_init(void) { s3c_i2c0_set_platdata(NULL); platform_add_devices(smdk2410_devices, ARRAY_SIZE(smdk2410_devices)); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); smdk_machine_init(); } diff --git a/arch/arm/mach-s3c24xx/mach-smdk2413.c b/arch/arm/mach-s3c24xx/mach-smdk2413.c index ca80167f268d..2bba4f1fa7ee 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2413.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2413.c @@ -37,6 +37,7 @@ #include <linux/platform_data/i2c-s3c2410.h> #include <mach/gpio-samsung.h> #include <mach/fb.h> +#include <plat/gpio-cfg.h> #include <plat/devs.h> #include <plat/cpu.h> @@ -119,6 +120,9 @@ static void __init smdk2413_machine_init(void) s3c24xx_udc_set_platdata(&smdk2413_udc_cfg); s3c_i2c0_set_platdata(NULL); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(smdk2413_devices, ARRAY_SIZE(smdk2413_devices)); smdk_machine_init(); diff --git a/arch/arm/mach-s3c24xx/mach-smdk2440.c b/arch/arm/mach-s3c24xx/mach-smdk2440.c index 7bafcd8ea104..f0fceda3e8e6 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2440.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2440.c @@ -29,6 +29,8 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <mach/fb.h> #include <linux/platform_data/i2c-s3c2410.h> @@ -167,7 +169,9 @@ static void __init smdk2440_machine_init(void) { s3c24xx_fb_set_platdata(&smdk2440_fb_info); s3c_i2c0_set_platdata(NULL); - + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(smdk2440_devices, ARRAY_SIZE(smdk2440_devices)); smdk_machine_init(); } diff --git a/arch/arm/mach-s3c24xx/mach-vstms.c b/arch/arm/mach-s3c24xx/mach-vstms.c index d76b28b65e65..f333ca05de68 100644 --- a/arch/arm/mach-s3c24xx/mach-vstms.c +++ b/arch/arm/mach-s3c24xx/mach-vstms.c @@ -31,6 +31,8 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <mach/fb.h> @@ -149,7 +151,9 @@ static void __init vstms_init(void) { s3c_i2c0_set_platdata(NULL); s3c_nand_set_platdata(&vstms_nand_info); - + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(vstms_devices, ARRAY_SIZE(vstms_devices)); } diff --git a/arch/arm/mach-s3c24xx/simtec-audio.c b/arch/arm/mach-s3c24xx/simtec-audio.c index 12e17f82dae3..0c12ba4a4704 100644 --- a/arch/arm/mach-s3c24xx/simtec-audio.c +++ b/arch/arm/mach-s3c24xx/simtec-audio.c @@ -14,6 +14,8 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <linux/platform_data/asoc-s3c24xx_simtec.h> #include <plat/devs.h> @@ -65,6 +67,10 @@ int __init simtec_audio_add(const char *name, bool has_lr_routing, if (has_lr_routing) simtec_audio_platdata.startup = simtec_audio_startup_lrroute; + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + platform_device_register(&s3c_device_iis); platform_device_register(&simtec_audio_dev); return 0; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 5bf571762e89..fd17765670f1 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -19,9 +19,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/gpio-samsung.h> -#include <plat/gpio-cfg.h> - #include "dma.h" #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" @@ -70,10 +67,6 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) if (ret) goto err; - /* Configure the I2S pins (GPE0...GPE4) in correct mode */ - s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), - S3C_GPIO_PULL_NONE); - return 0; err: diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 92bdaf0878f8..5ca263f6a39e 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -18,10 +18,7 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/gpio-samsung.h> -#include <plat/gpio-cfg.h> #include "regs-iis.h" - #include "dma.h" #include "s3c24xx-i2s.h" @@ -348,10 +345,6 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) if (ret) return ret; - /* Configure the I2S pins (GPE0...GPE4) in correct mode */ - s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), - S3C_GPIO_PULL_NONE); - writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); s3c24xx_snd_txctrl(0); -- 2.20.0

2 2

[alsa-devel] [RFC PATCH] ALSA: pcm: fix blocking while loop in snd_pcm_update_hw_ptr0()
by Srinivas Kandagatla 23 Oct '19

23 Oct '19

From: Preetam Singh Ranawat <apranawat(a)codeaurora.org> When period interrupts are disabled, while loop in snd_pcm_update_hw_ptr0() results in the machine locking up if runtime->hw_ptr_buffer_jiffies is 0. Validate runtime->hw_ptr_buffer_jiffies value before while loop to avoid delta check. Signed-off-by: Preetam Singh Ranawat <apranawat(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- This issue was noticed in Android setup from fastcapture path. sound/core/pcm_lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d80041ea4e01..f407f6895f8a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -341,7 +341,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, * the elapsed time to detect xruns. */ jdelta = curr_jiffies - runtime->hw_ptr_jiffies; - if (jdelta < runtime->hw_ptr_buffer_jiffies / 2) + if ((jdelta < runtime->hw_ptr_buffer_jiffies / 2) || + !runtime->hw_ptr_buffer_jiffies) goto no_delta_check; hdelta = jdelta - delta * HZ / runtime->rate; xrun_threshold = runtime->hw_ptr_buffer_jiffies / 2 + 1; -- 2.21.0

2 1

[alsa-devel] alsa conformance test fails to run for playback (and works for capture)
by Hin-Tak Leung 23 Oct '19

23 Oct '19

Hi Yu-Hsuan, I have encountered another problem with the alsa conformance test. It works for capture, but not for playback. The device is - http://wiki.seeedstudio.com/ReSpeaker_6-Mic_Circular_Array_kit_for_Raspberr… and running on a pi / raspbian buster. The actual message is: Fail - ALSA lib pcm_hw.c:1822:(_snd_pcm_hw_open) Invalid value for card snd_pcm_open hw:CARD=seeed8micvoi: No such device I think the truncation is harmless . The actual device name according to aplay -L is "hw:CARD=seeed8micvoicec,DEV=0" and as I noted earlier, arecord -L shows it as "hw:CARD=seeed8micvoicec,DEV=0" also, and the comformance test running in CAPTURE mode works, just not in PLAYBACK mode. Is there anything I can do to look further? Since it says "invalid value", I tried deleting the parameters around line 190 of src/alsa_conformance_helper.c, but perhaps I should do something else? src/alsa_conformance_helper.c-191- conformance_timer_start(timer, SND_PCM_OPEN); src/alsa_conformance_helper.c:192: rc = snd_pcm_open(handle, src/alsa_conformance_helper.c-193- dev_name, src/alsa_conformance_helper.c-194- stream, src/alsa_conformance_helper.c-195- SND_PCM_NONBLOCK | src/alsa_conformance_helper.c-196- SND_PCM_NO_AUTO_RESAMPLE | src/alsa_conformance_helper.c-197- SND_PCM_NO_AUTO_CHANNELS | src/alsa_conformance_helper.c-198- SND_PCM_NO_AUTO_FORMAT); src/alsa_conformance_helper.c-199- conformance_timer_stop(timer, SND_PCM_OPEN); Regards, Hin-Tak

3 8

[alsa-devel] KASAN: slab-out-of-bounds Read in parse_term_proc_unit
by syzbot 23 Oct '19

23 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=141c4097600000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=f586a582f79122007598 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=169fe2df600000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=13e2f3ef600000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f586a582f79122007598(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: slab-out-of-bounds in uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] BUG: KASAN: slab-out-of-bounds in parse_term_proc_unit+0x57a/0x5e0 sound/usb/mixer.c:896 Read of size 1 at addr ffff8881d48eeca9 by task kworker/1:2/83 CPU: 1 PID: 83 Comm: kworker/1:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] parse_term_proc_unit+0x57a/0x5e0 sound/usb/mixer.c:896 __check_input_term+0xc32/0x13f0 sound/usb/mixer.c:984 parse_audio_feature_unit sound/usb/mixer.c:1875 [inline] parse_audio_unit+0x101d/0x36f0 sound/usb/mixer.c:2753 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 83: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x314/0x3050 drivers/usb/core/config.c:857 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 0: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 free_task+0xe3/0x110 kernel/fork.c:472 __put_task_struct+0x1e2/0x4c0 kernel/fork.c:748 put_task_struct include/linux/sched/task.h:119 [inline] delayed_put_task_struct+0x1b4/0x2c0 kernel/exit.c:182 __rcu_reclaim kernel/rcu/rcu.h:222 [inline] rcu_do_batch kernel/rcu/tree.c:2157 [inline] rcu_core+0x630/0x1ca0 kernel/rcu/tree.c:2377 __do_softirq+0x221/0x912 kernel/softirq.c:292 The buggy address belongs to the object at ffff8881d48eec00 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 169 bytes inside of 192-byte region [ffff8881d48eec00, ffff8881d48eecc0) The buggy address belongs to the page: page:ffffea0007523b80 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea000753ce40 0000000e0000000e ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d48eeb80: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d48eec00: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > ffff8881d48eec80: 00 00 00 00 03 fc fc fc fc fc fc fc fc fc fc fc ^ ffff8881d48eed00: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8881d48eed80: 00 00 00 00 00 00 00 fc fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] KMSAN: uninit-value in build_audio_procunit
by syzbot 23 Oct '19

23 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 3c8ca708 test_kmsan.c: fix SPDX comment git tree: https://github.com/google/kmsan.git master console output: https://syzkaller.appspot.com/x/log.txt?x=133fb3cf600000 kernel config: https://syzkaller.appspot.com/x/.config?x=c07a3d4f8a59e198 dashboard link: https://syzkaller.appspot.com/bug?extid=85f45d8bd343473faf15 compiler: clang version 9.0.0 (/home/glider/llvm/clang 80fee25776c2fb61e74c1ecb1a523375c2500b69) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=12201cef600000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1281b8ff600000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+85f45d8bd343473faf15(a)syzkaller.appspotmail.com usb 1-1: New USB device found, idVendor=1d6b, idProduct=0101, bcdDevice= 0.40 usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ===================================================== BUG: KMSAN: uninit-value in build_audio_procunit+0x1be8/0x2490 sound/usb/mixer.c:2438 CPU: 1 PID: 3381 Comm: kworker/1:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x191/0x1f0 lib/dump_stack.c:113 kmsan_report+0x14a/0x2f0 mm/kmsan/kmsan_report.c:110 __msan_warning+0x73/0xf0 mm/kmsan/kmsan_instr.c:245 build_audio_procunit+0x1be8/0x2490 sound/usb/mixer.c:2438 parse_audio_unit+0x115c/0x7490 sound/usb/mixer.c:2483 snd_usb_mixer_controls sound/usb/mixer.c:3095 [inline] snd_usb_create_mixer+0x1d7c/0x4070 sound/usb/mixer.c:3445 usb_audio_probe+0x286b/0x3eb0 sound/usb/card.c:653 usb_probe_interface+0xd19/0x1310 drivers/usb/core/driver.c:361 really_probe+0xd91/0x1f90 drivers/base/dd.c:552 driver_probe_device+0x1ba/0x510 drivers/base/dd.c:721 __device_attach_driver+0x5b8/0x790 drivers/base/dd.c:828 bus_for_each_drv+0x28e/0x3b0 drivers/base/bus.c:430 __device_attach+0x489/0x750 drivers/base/dd.c:894 device_initial_probe+0x4a/0x60 drivers/base/dd.c:941 bus_probe_device+0x131/0x390 drivers/base/bus.c:490 device_add+0x25b5/0x2df0 drivers/base/core.c:2201 usb_set_configuration+0x309f/0x3710 drivers/usb/core/message.c:2027 generic_probe+0xe7/0x280 drivers/usb/core/generic.c:210 usb_probe_device+0x146/0x200 drivers/usb/core/driver.c:266 really_probe+0xd91/0x1f90 drivers/base/dd.c:552 driver_probe_device+0x1ba/0x510 drivers/base/dd.c:721 __device_attach_driver+0x5b8/0x790 drivers/base/dd.c:828 bus_for_each_drv+0x28e/0x3b0 drivers/base/bus.c:430 __device_attach+0x489/0x750 drivers/base/dd.c:894 device_initial_probe+0x4a/0x60 drivers/base/dd.c:941 bus_probe_device+0x131/0x390 drivers/base/bus.c:490 device_add+0x25b5/0x2df0 drivers/base/core.c:2201 usb_new_device+0x23e5/0x2fb0 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5098 [inline] hub_port_connect_change drivers/usb/core/hub.c:5213 [inline] port_event drivers/usb/core/hub.c:5359 [inline] hub_event+0x581d/0x72f0 drivers/usb/core/hub.c:5441 process_one_work+0x1572/0x1ef0 kernel/workqueue.c:2269 worker_thread+0x111b/0x2460 kernel/workqueue.c:2415 kthread+0x4b5/0x4f0 kernel/kthread.c:256 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:355 Uninit was created at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:151 [inline] kmsan_internal_poison_shadow+0x60/0x110 mm/kmsan/kmsan.c:134 kmsan_slab_alloc+0xaa/0x130 mm/kmsan/kmsan_hooks.c:88 slab_alloc_node mm/slub.c:2792 [inline] slab_alloc mm/slub.c:2801 [inline] __kmalloc+0x28e/0x430 mm/slub.c:3832 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x50d/0x76a0 drivers/usb/core/config.c:857 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0x224/0x2fb0 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5098 [inline] hub_port_connect_change drivers/usb/core/hub.c:5213 [inline] port_event drivers/usb/core/hub.c:5359 [inline] hub_event+0x581d/0x72f0 drivers/usb/core/hub.c:5441 process_one_work+0x1572/0x1ef0 kernel/workqueue.c:2269 worker_thread+0x111b/0x2460 kernel/workqueue.c:2415 kthread+0x4b5/0x4f0 kernel/kthread.c:256 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:355 ===================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] [PATCH] ALSA: hda: Add Tigerlake/Jasperlake PCI ID
by Pierre-Louis Bossart 23 Oct '19

23 Oct '19

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> Add HD Audio Device PCI ID for the Intel Tigerlake and Jasperlake platform. Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..a815bc811799 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2399,6 +2399,12 @@ static const struct pci_device_id azx_ids[] = { /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Jasperlake */ + { PCI_DEVICE(0x8086, 0x38c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Tigerlake */ + { PCI_DEVICE(0x8086, 0xa0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.20.1

2 1

[alsa-devel] [PATCH v2 0/5] soundwire: intel/cadence: better initialization
by Pierre-Louis Bossart 22 Oct '19

22 Oct '19

V2 of the original series 'soundwire: inits and PM additions for 5.4', with PM additions removed since more tests on hardware are required. Changes since v1: addressed feedback from Vinod Koul clarified init changes impact Intel and Cadence sides remove unnecessary intermediate variable disable interrupts when exit_reset fails, updated error handling returned -EINVAL on debugfs invalid parameter Pierre-Louis Bossart (5): soundwire: intel/cadence: fix startup sequence soundwire: cadence_master: add hw_reset capability in debugfs soundwire: intel: add helper for initialization soundwire: intel/cadence: add flag for interrupt enable soundwire: cadence_master: make clock stop exit configurable on init drivers/soundwire/cadence_master.c | 131 +++++++++++++++++++++-------- drivers/soundwire/cadence_master.h | 5 +- drivers/soundwire/intel.c | 38 ++++++--- 3 files changed, 126 insertions(+), 48 deletions(-) -- 2.20.1

2 13

[alsa-devel] [PATCH] ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address
by twischer＠de.adit-jv.com 22 Oct '19

22 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif dma address is calculated by following calculation formulation: 0xec540000 + 0x1000 * id + busif / 4 * 0xA000 + busif % 4 * 0x400 But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif data register address (SSI9_4_BUSIF/SSI9_5_BUSIF/SSI9_6_BUSIF/SSI9_7_BUSIF) are out of this rule. This patch updates the calculation formulation to correct ssi9 4/5/6/7 busif data register address Fixes: commit 5e45a6fab3b9 ("ASoc: rsnd: dma: Calculate dma address with consider of BUSIF") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0324a5c..28f65eb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -508,10 +508,10 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) -- 2.7.4

4 7

[alsa-devel] Recording on WM9712
by Matej Kupljen 22 Oct '19

22 Oct '19

Hi all, I have an iMX6q board with external AC97 WM9712 codec connected to SSI on iMX6. The card is detected correctly and it is using generic AC97 codec driver, not the specific one for WM9712. I was able to hard code the values into fsl-asoc-card.c and fsl_ssi.c files, but my problem still remains. I can playback WAV file normally, also speaker-test works as expected. The problem is when I try to record something from Line In. It does record the file, but when I playback it, it is "too slow". I tried different sample rates, but the problem is only worse (strange beeping in the recording). The command for recording is: # arecord -f dat -vv /tmp/record.wav (I tired also with -M option, but there is no difference). If I copy the recorded file to PC and use audacity to play it back using 1.5x speed, the recording seems O.K. Checking codec register values trough /proc/asound/ac97audio/codec97#0/ac97#0-0+regs it seems that the sample rates are set O.K. to 48kHz. Any idea what might be wrong? Is there anything else I can check? Thanks and BR, Matej

1 0

[alsa-devel] [REGRESSION 5.3] WARNING: pulseaudio/1297 still has locks held!
by Russell King - ARM Linux admin 22 Oct '19

22 Oct '19

When upgrading from 5.2 to 5.3, I now encounter this error after boot: ==================================== WARNING: pulseaudio/1297 still has locks held! 5.3.0+ #1826 Not tainted ------------------------------------ 1 lock held by pulseaudio/1297: #0: ee815308 (&hcp->lock){....}, at: hdmi_codec_startup+0x20/0x130 stack backtrace: CPU: 0 PID: 1297 Comm: pulseaudio Not tainted 5.3.0+ #1826 Hardware name: Marvell Dove (Cubox) [<c0017b4c>] (unwind_backtrace) from [<c0014d10>] (show_stack+0x10/0x14) [<c0014d10>] (show_stack) from [<c00a2d18>] (futex_wait_queue_me+0x13c/0x19c) [<c00a2d18>] (futex_wait_queue_me) from [<c00a3630>] (futex_wait+0x184/0x24c) [<c00a3630>] (futex_wait) from [<c00a5e1c>] (do_futex+0x334/0x598) [<c00a5e1c>] (do_futex) from [<c00a62e8>] (sys_futex_time32+0x118/0x180) [<c00a62e8>] (sys_futex_time32) from [<c0009000>] (ret_fast_syscall+0x0/0x54) Exception stack(0xebd31fa8 to 0xebd31ff0) 1fa0: 00000000 ffffffff 000c8748 00000189 00000001 00000000 1fc0: 00000000 ffffffff 00000000 000000f0 00000000 00000000 00000000 00056200 1fe0: 000000f0 beac03a8 b6d6c835 b6d6f456 This seems to be caused by: eb1ecadb7f67 ("ASoC: hdmi-codec: re-introduce mutex locking") which holds a mutex and returns to userspace. Meanwhile, userspace issues a futex operation, which ends up in futex_wait_queue_me(). futex_wait_queue_me() can wait, and if the system will be hibernating, is a potential hibernation point. One of the requirements there is that no locks shall be held, and the freezer code verifies that by calling debug_check_no_locks_held(). With lockdep enabled, it reveals that hdmi-codec is indeed holding a lock. Reverting this commit solves the problem. -- RMK's Patch system: https://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line in suburbia: sync at 12.1Mbps down 622kbps up According to speedtest.net: 11.9Mbps down 500kbps up

3 2

[alsa-devel] [PATCH v6 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 22 Oct '19

22 Oct '19

Hey, here's the 6th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. v6 changes: - Since v5, changes in patches 002, 003 and 005. - Fixed a bug with S3 suspend/resume during HDMI playback on Intel ICL platforms. As part of the bug fix, cleaned up the new display power control code added in v5. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 114 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.h | 97 +++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 345 insertions(+), 27 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

3 17

[alsa-devel] sof-skl_hda: add device definition to speaker disable sequence
by GitHub pull_request - opened 22 Oct '19

22 Oct '19

alsa-project/alsa-ucm-conf pull request #3 was opened from juimonen: For example Pulseaudio is not able to run the disable sequence if the device is not defined. So add the definition. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/3 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/3.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [asoc:for-5.5 122/122] sound/soc/codecs/tas2562.c:195:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 22 Oct '19

22 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: c173dba44c2d2ec2adaa4e607e9729b74236bd57 commit: c173dba44c2d2ec2adaa4e607e9729b74236bd57 [122/122] ASoC: tas2562: Introduce the TAS2562 amplifier reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-42-g38eda53-dirty git checkout c173dba44c2d2ec2adaa4e607e9729b74236bd57 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2562.c:195:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2562.c:202:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2562.c:209:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer >> sound/soc/codecs/tas2562.c:246:58: sparse: sparse: incorrect type in argument 2 (different base types) @@ expected int bitwidth @@ got restricted snd_pcm_formint bitwidth @@ >> sound/soc/codecs/tas2562.c:246:58: sparse: expected int bitwidth >> sound/soc/codecs/tas2562.c:246:58: sparse: got restricted snd_pcm_format_t sound/soc/codecs/tas2562.c:472:39: sparse: sparse: restricted snd_pcm_format_t degrades to integer vim +195 sound/soc/codecs/tas2562.c 189 190 static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) 191 { 192 int ret; 193 194 switch (bitwidth) { > 195 case SNDRV_PCM_FORMAT_S16_LE: 196 snd_soc_component_update_bits(tas2562->component, 197 TAS2562_TDM_CFG2, 198 TAS2562_TDM_CFG2_RXWLEN_MASK, 199 TAS2562_TDM_CFG2_RXWLEN_16B); 200 tas2562->v_sense_slot = tas2562->i_sense_slot + 2; 201 break; 202 case SNDRV_PCM_FORMAT_S24_LE: 203 snd_soc_component_update_bits(tas2562->component, 204 TAS2562_TDM_CFG2, 205 TAS2562_TDM_CFG2_RXWLEN_MASK, 206 TAS2562_TDM_CFG2_RXWLEN_24B); 207 tas2562->v_sense_slot = tas2562->i_sense_slot + 4; 208 break; 209 case SNDRV_PCM_FORMAT_S32_LE: 210 snd_soc_component_update_bits(tas2562->component, 211 TAS2562_TDM_CFG2, 212 TAS2562_TDM_CFG2_RXWLEN_MASK, 213 TAS2562_TDM_CFG2_RXWLEN_32B); 214 tas2562->v_sense_slot = tas2562->i_sense_slot + 4; 215 break; 216 217 default: 218 dev_info(tas2562->dev, "Not supported params format\n"); 219 } 220 221 ret = snd_soc_component_update_bits(tas2562->component, 222 TAS2562_TDM_CFG5, 223 TAS2562_TDM_CFG5_VSNS_EN | TAS2562_TDM_CFG5_VSNS_SLOT_MASK, 224 TAS2562_TDM_CFG5_VSNS_EN | tas2562->v_sense_slot); 225 if (ret < 0) 226 return ret; 227 228 ret = snd_soc_component_update_bits(tas2562->component, 229 TAS2562_TDM_CFG6, 230 TAS2562_TDM_CFG6_ISNS_EN | TAS2562_TDM_CFG6_ISNS_SLOT_MASK, 231 TAS2562_TDM_CFG6_ISNS_EN | tas2562->i_sense_slot); 232 if (ret < 0) 233 return ret; 234 235 return 0; 236 } 237 238 static int tas2562_hw_params(struct snd_pcm_substream *substream, 239 struct snd_pcm_hw_params *params, 240 struct snd_soc_dai *dai) 241 { 242 struct snd_soc_component *component = dai->component; 243 struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); 244 int ret; 245 > 246 ret = tas2562_set_bitwidth(tas2562, params_format(params)); 247 if (ret) { 248 dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); 249 return ret; 250 } 251 252 ret = tas2562_set_samplerate(tas2562, params_rate(params)); 253 if (ret) 254 dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); 255 256 return ret; 257 } 258 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v2] ASoC: samsung: i2s: Document clocks macros
by Marek Szyprowski 21 Oct '19

21 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Document clocks macros with their description from 'Documentation/devicetree/bindings/sound/samsung-i2s.txt' Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - Added proper commit message and description - Moved comments to the right side Best regards, Maciej Falkowski --- include/dt-bindings/sound/samsung-i2s.h | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/include/dt-bindings/sound/samsung-i2s.h b/include/dt-bindings/sound/samsung-i2s.h index 77545f14c379..250de0d6c734 100644 --- a/include/dt-bindings/sound/samsung-i2s.h +++ b/include/dt-bindings/sound/samsung-i2s.h @@ -2,8 +2,14 @@ #ifndef _DT_BINDINGS_SAMSUNG_I2S_H #define _DT_BINDINGS_SAMSUNG_I2S_H -#define CLK_I2S_CDCLK 0 -#define CLK_I2S_RCLK_SRC 1 -#define CLK_I2S_RCLK_PSR 2 +#define CLK_I2S_CDCLK 0 /* the CDCLK (CODECLKO) gate clock */ + +#define CLK_I2S_RCLK_SRC 1 /* the RCLKSRC mux clock (corresponding to + * RCLKSRC bit in IISMOD register) + */ + +#define CLK_I2S_RCLK_PSR 2 /* the RCLK prescaler divider clock + * (corresponding to the IISPSR register) + */ #endif /* _DT_BINDINGS_SAMSUNG_I2S_H */ -- 2.17.1

4 3

[alsa-devel] [PATCH 1/2] ASoC: msm8916-wcd-analog: Fix RX1 selection in RDAC2 MUX
by Stephan Gerhold 21 Oct '19

21 Oct '19

According to the PM8916 Hardware Register Description, CDC_D_CDC_CONN_HPHR_DAC_CTL has only a single bit (RX_SEL) to switch between RX1 (0) and RX2 (1). It is not possible to disable it entirely to achieve the "ZERO" state. However, at the moment the "RDAC2 MUX" mixer defines three possible values ("ZERO", "RX2" and "RX1"). Setting the mixer to "ZERO" actually configures it to RX1. Setting the mixer to "RX1" has (seemingly) no effect. Remove "ZERO" and replace it with "RX1" to fix this. Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/msm8916-wcd-analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 667e9f73aba3..e3d311fb510e 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -306,7 +306,7 @@ struct pm8916_wcd_analog_priv { }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; -static const char *const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" }; +static const char *const rdac2_mux_text[] = { "RX1", "RX2" }; static const char *const hph_text[] = { "ZERO", "Switch", }; static const struct soc_enum hph_enum = SOC_ENUM_SINGLE_VIRT( @@ -321,7 +321,7 @@ static const struct soc_enum adc2_enum = SOC_ENUM_SINGLE_VIRT( /* RDAC2 MUX */ static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE( - CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text); + CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 2, rdac2_mux_text); static const struct snd_kcontrol_new spkr_switch[] = { SOC_DAPM_SINGLE("Switch", CDC_A_SPKR_DAC_CTL, 7, 1, 0) -- 2.23.0

3 5

[alsa-devel] [PATCH] ASoC: compress: fix unsigned integer overflow check
by Srinivas Kandagatla 21 Oct '19

21 Oct '19

From: Xiaojun Sang <xsang(a)codeaurora.org> Parameter fragments and fragment_size are type of u32. U32_MAX is the correct check. Signed-off-by: Xiaojun Sang <xsang(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/core/compress_offload.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 41905afada63..f34ce564d92c 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -528,7 +528,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments > U32_MAX / params->buffer.fragment_size || params->buffer.fragments == 0) return -EINVAL; -- 2.21.0

3 2

[alsa-devel] [PATCH v5 0/4] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 21 Oct '19

21 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). ASoC: cros_ec_codec: support WoV - makes cros_ec_codec support WoV. ASoC: mediatek: mt6358: support WoV - sets necessary registers on mt6358 to support WoV. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV - make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. Changes from v4: - remove the first 6 commits which have merged - fix a compile error by adding Kconfig dependency https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156992.h… Changes from v3: - fix a compile error by adding ifdef https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156750.h… - fix a coccinelle warning https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156754.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (4): ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 1 + .../linux/platform_data/cros_ec_commands.h | 69 ++ sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/cros_ec_codec.c | 700 +++++++++++++++++- sound/soc/codecs/mt6358.c | 105 +++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 946 insertions(+), 5 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

3 12

[alsa-devel] [PATCH 0/6] ALSA: firewire: handle several IR/IT context in callback of the IT context
by Takashi Sakamoto 21 Oct '19

21 Oct '19

Hi, This patchset is a part of my continuous work to improve ALSA IEC 61883-1/6 packet streaming engine for clock-recovery, addressed in my previous message: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-August/153388.ht… For the clock-recovery, information in the sequence of tx packet from device should be used to generate the sequence of rx packet to the device. In current implementation of the engine, the packets are processed in different tasklet contexts for each IR/IT context. This is inconvenient to bypass information between these IR/IT contexts. In this patchset, the engine is improved to process all of IR/IT contexts in the same domain by a tasklet context for one of IT context. For convenience, the IT context is called as 'IRQ target'. As a result, 1394 OHCI controller is managed to generate hardware IRQ just for the IRQ target. All of rx/tx packets are processed in tasklet for the hardware IRQ. Takashi Sakamoto (6): ALSA: firewire-lib: add irq_target member into amdtp_domain struct ALSA: firewire-lib: replace pointer callback to flush isoc contexts in AMDTP domain ALSA: firewire-lib: replace ack callback to flush isoc contexts in AMDTP domain ALSA: firewire-lib: cancel flushing isoc context in the laste step to process context callback ALSA: firewire-lib: handle several AMDTP streams in callback handler of IRQ target ALSA: firewire-lib: postpone to start IR context sound/firewire/amdtp-stream.c | 329 +++++++++++++++----- sound/firewire/amdtp-stream.h | 17 +- sound/firewire/bebob/bebob_pcm.c | 18 +- sound/firewire/bebob/bebob_stream.c | 10 +- sound/firewire/dice/dice-pcm.c | 8 +- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 8 +- sound/firewire/digi00x/digi00x-stream.c | 2 +- sound/firewire/fireface/ff-pcm.c | 8 +- sound/firewire/fireface/ff-stream.c | 10 +- sound/firewire/fireworks/fireworks_pcm.c | 10 +- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/motu/motu-pcm.c | 8 +- sound/firewire/motu/motu-stream.c | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 8 +- sound/firewire/oxfw/oxfw-stream.c | 2 +- sound/firewire/tascam/tascam-pcm.c | 8 +- sound/firewire/tascam/tascam-stream.c | 2 +- 18 files changed, 325 insertions(+), 129 deletions(-) -- 2.20.1

2 9

[alsa-devel] [sound:for-next 37/38] include/linux/pci.h:2317:13: error: 'pci_pr3_present' defined but not used
by kbuild test robot 21 Oct '19

21 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: bacd861452d2be86a4df341b12e32db7dac8021e commit: 52525b7a3cf82adec5c6cf0ecbd23ff228badc94 [37/38] PCI: Add a helper to check Power Resource Requirements _PR3 existence config: sparc64-allmodconfig (attached as .config) compiler: sparc64-linux-gcc (GCC) 7.4.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 52525b7a3cf82adec5c6cf0ecbd23ff228badc94 # save the attached .config to linux build tree GCC_VERSION=7.4.0 make.cross ARCH=sparc64 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from arch/sparc/kernel/iommu.c:20:0: >> include/linux/pci.h:2317:13: error: 'pci_pr3_present' defined but not used [-Werror=unused-function] static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ cc1: all warnings being treated as errors vim +/pci_pr3_present +2317 include/linux/pci.h 2310 2311 void 2312 pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); 2313 bool pci_pr3_present(struct pci_dev *pdev); 2314 #else 2315 static inline struct irq_domain * 2316 pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } > 2317 static bool pci_pr3_present(struct pci_dev *pdev) { return false; } 2318 #endif 2319 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [GIT PULL] ASoC fixes for 5.4
by Mark Brown 21 Oct '19

21 Oct '19

The following changes since commit 147162f575152db80000fb3de26358264768ee9f: ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies (2019-09-20 12:06:48 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.4-rc4 for you to fetch changes up to 95a32c98055f664f9b3f34c41e153d4dcedd0eff: ASoC: SOF: control: return true when kcontrol values change (2019-10-18 18:09:22 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.4 A collection of fixes that have arrived since the merge window. There are a small number of core fixes here but they are smaller ones around error handling. ---------------------------------------------------------------- Bard Liao (1): ASoC: intel: sof_rt5682: use separate route map for dmic Chuhong Yuan (1): ASoC: Intel: sof-rt5682: add a check for devm_clk_get Dan Carpenter (2): ASoC: soc-component: fix a couple missing error assignments ASoC: topology: Fix a signedness bug in soc_tplg_dapm_widget_create() Daniel Baluta (1): ASoC: simple_card_utils.h: Fix potential multiple redefinition error Dragos Tarcatu (1): ASoC: SOF: control: return true when kcontrol values change Hans de Goede (1): ASoC: core: Fix pcm code debugfs error Jaska Uimonen (3): ASoC: rt5682: add NULL handler to set_jack function ASoC: intel: sof_rt5682: add remove function to disable jack ASoC: intel: bytcr_rt5651: add null check to support_button_press Junya Monden (1): ASoC: rsnd: Reinitialize bit clock inversion flag for every format setting Kai Vehmanen (2): ASoC: SOF: pcm: fix resource leak in hw_free ASoC: SOF: Intel: hda: fix warnings during FW load Keyon Jie (1): ASoC: SOF: topology: fix parse fail issue for byte/bool tuple types Li Xu (1): ASoC: wm_adsp: Fix theoretical NULL pointer for alg_region Liam Girdwood (1): ASoC: SOF: Intel: initialise and verify FW crash dump data. Olivier Moysan (1): ASoC: stm32: sai: fix sysclk management on shutdown Pan Xiuli (1): ASoC: SOF: pcm: harden PCM STOP sequence Pierre-Louis Bossart (1): ASoC: SOF: loader: fix kernel oops on firmware boot failure Ranjani Sridharan (1): ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture Robin Murphy (1): ASoc: rockchip: i2s: Fix RPM imbalance Sathyanarayana Nujella (1): ASoC: max98373: check for device node before parsing Srinivas Kandagatla (1): ASoC: msm8916-wcd-digital: add missing MIX2 path for RX1/2 Stuart Henderson (1): ASoC: wm_adsp: Don't generate kcontrols without READ flags Sylwester Nawrocki (2): ASoC: samsung: arndale: Add missing OF node dereferencing ASoC: wm8994: Do not register inapplicable controls for WM1811 include/sound/simple_card_utils.h | 8 ++--- sound/soc/codecs/max98373.c | 20 +++++++----- sound/soc/codecs/msm8916-wcd-digital.c | 22 +++++++++++++ sound/soc/codecs/rt5651.c | 3 ++ sound/soc/codecs/rt5682.c | 12 +++++-- sound/soc/codecs/wm8994.c | 43 ++++++++++++++---------- sound/soc/codecs/wm_adsp.c | 10 ++++-- sound/soc/intel/boards/sof_rt5682.c | 60 +++++++++++++++++++++++++++++++--- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/samsung/arndale_rt5631.c | 34 ++++++++++++++++--- sound/soc/sh/rcar/core.c | 1 + sound/soc/soc-pcm.c | 17 +++++++--- sound/soc/soc-topology.c | 2 +- sound/soc/sof/control.c | 26 ++++++++++----- sound/soc/sof/intel/Kconfig | 10 ++++++ sound/soc/sof/intel/bdw.c | 7 ++++ sound/soc/sof/intel/byt.c | 6 ++++ sound/soc/sof/intel/hda-ctrl.c | 12 +++---- sound/soc/sof/intel/hda-loader.c | 1 + sound/soc/sof/intel/hda-stream.c | 45 +++++++++++++++++++++---- sound/soc/sof/intel/hda.c | 7 ++++ sound/soc/sof/intel/hda.h | 5 ++- sound/soc/sof/loader.c | 4 +-- sound/soc/sof/pcm.c | 35 +++++++++++++------- sound/soc/sof/topology.c | 4 ++- sound/soc/stm/stm32_sai_sub.c | 21 ++++++------ 26 files changed, 320 insertions(+), 97 deletions(-)

2 1

[alsa-devel] ALC711 support
by Kailang 21 Oct '19

21 Oct '19

Hi Takashi, Attach patch was support ALC711 HDA mode. BR, Kailang

2 1

[alsa-devel] Question about "957ce0c6b8a1 ASoC: soc-pcm: check symmetry after hw_params"
by S.j. Wang 21 Oct '19

21 Oct '19

Hi Jiada, Mark After this commit "957ce0c6b8a1 ASoC: soc-pcm: check symmetry after hw_params" The soc_pcm_params_symmetry can't take effect, because its position is after the change of cpu_dai->rate. The code of soc_pcm_params_symmetry is: static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { ... if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", cpu_dai->rate, rate); return -EINVAL; } ... } This code soc_pcm_hw_params of is: static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { ... /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); cpu_dai->channels = params_channels(params); cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); snd_soc_dapm_update_dai(substream, params, cpu_dai); ... ret = soc_pcm_params_symmetry(substream, params); if (ret) goto component_err; ... } Is this an issue? Best regards Wang shengjiu

2 1

[alsa-devel] [PATCH v2] dt-bindings: sound: Convert Samsung Exynos Odroid XU3/XU4 audio complex to dt-schema
by Marek Szyprowski 21 Oct '19

21 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec to newer dt-schema format. 'clocks' property is unneeded in the bindings and is left undefined in 'properties'. 'samsung,audio-widgets' and 'samsung,audio-routing' are optional from driver perspective and they are set as unrequired. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: reordered non-standard properties] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,odroid.txt | 54 ----------- .../bindings/sound/samsung,odroid.yaml | 91 +++++++++++++++++++ 2 files changed, 91 insertions(+), 54 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt deleted file mode 100644 index e9da2200e173..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ /dev/null @@ -1,54 +0,0 @@ -Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec - -Required properties: - - - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated), - "samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated), - "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - - model - the user-visible name of this sound complex - - clocks - should contain entries matching clock names in the clock-names - property - - samsung,audio-widgets - this property specifies off-codec audio elements - like headphones or speakers, for details see widgets.txt - - samsung,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 for sources and sinks are the MAX98090's pins (as - documented in its binding), and the jacks on the board - - For Odroid X2: - "Headphone Jack", "Mic Jack", "DMIC" - - For Odroid U3, XU3: - "Headphone Jack", "Speakers" - - For Odroid XU4: - no entries - -Required sub-nodes: - - - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S - controller - - 'codec' subnode with a 'sound-dai' property containing list of phandles - to the CODEC nodes, first entry must be corresponding to the MAX98090 - CODEC and the second entry must be the phandle of the HDMI IP block node - -Example: - -sound { - compatible = "hardkernel,odroid-xu3-audio"; - model = "Odroid-XU3"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; - - cpu { - sound-dai = <&i2s0 0>; - }; - codec { - sound-dai = <&hdmi>, <&max98090>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml new file mode 100644 index 000000000000..c6b244352d05 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,odroid.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + oneOf: + - const: hardkernel,odroid-xu3-audio + + - const: hardkernel,odroid-xu4-audio + deprecated: true + + - const: samsung,odroid-xu3-audio + deprecated: true + + - const: samsung,odroid-xu4-audio + deprecated: true + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandles to the I2S controllers + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles to the CODEC nodes, + first entry must be corresponding to the MAX98090 CODEC and + the second entry must be the phandle of the HDMI IP block node. + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 for sources and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board. + For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" + For Odroid U3, XU3: "Headphone Jack", "Speakers" + For Odroid XU4: no entries + + samsung,audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + This property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + +required: + - compatible + - model + - cpu + - codec + +examples: + - | + sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&hdmi>, <&max98090>; + }; + }; + -- 2.17.1

4 3

[alsa-devel] [PATCH 0/6] soundwire: intel/cadence: simplify PDI handling
by Pierre-Louis Bossart 21 Oct '19

21 Oct '19

These patches were originally submitted as '[RFC PATCH 00/11] soundwire: intel: simplify DAI/PDI handling'. There were no comments received. This series only provides the PDI changes, which makes it simpler to review. The DAI changes will be provided with the complete series for ASoC/SOF integration, which is a larger change. Bard Liao (3): soundwire: intel: fix intel_register_dai PDI offsets and numbers soundwire: intel: remove playback/capture stream_name soundwire: cadence_master: improve PDI allocation Pierre-Louis Bossart (3): soundwire: remove DAI_ID_RANGE definitions soundwire: cadence/intel: simplify PDI/port mapping soundwire: intel: don't filter out PDI0/1 drivers/soundwire/cadence_master.c | 158 +++++++---------------------- drivers/soundwire/cadence_master.h | 34 ++----- drivers/soundwire/intel.c | 155 ++++++---------------------- include/linux/soundwire/sdw.h | 3 - 4 files changed, 73 insertions(+), 277 deletions(-) -- 2.20.1

2 8

[alsa-devel] Fwd: [PATCH 29/46] Input: touchscreen: use wrapper for pxa2xx ac97 registers
by Arnd Bergmann 18 Oct '19

18 Oct '19

On Fri, Oct 18, 2019 at 8:48 PM Dmitry Torokhov <dmitry.torokhov(a)gmail.com> wrote: > > On Fri, Oct 18, 2019 at 05:41:44PM +0200, Arnd Bergmann wrote: > > To avoid a dependency on the pxa platform header files with > > hardcoded registers, change the driver to call a wrapper > > in the pxa2xx-ac97-lib that encapsulates all the other > > ac97 stuff. > > Not supper happy about adding module dependencies. Can we include > mach/regs-ac97.h from include/sound/pxa2xx-lib.h and use static inlines? > Someone needs to include mach/regs-ac97.h in the end... > > Or there is something later in the series that needs it? One of the goals of the series is to completely remove all mach/*.h headers and place them somewhere else, ideally inaccessible to device drivers. In case of mach/regs-ac97.h, the later patch "ASoC: pxa: ac97: use normal MMIO accessors" passes the physical register base address as a platform device resource that gets ioremapped in the ac97 driver, rather than hardcoding the virtual address in a global header. I agree that the exported function is still ugly, but I hope it's enough of an improvement over the previous state that we can do it anyway. If you have any better ideas though, I can give that a try, too. One possibility would be a higher-level interface exported on top of 'struct snd_ac97', but I could not figure out how to do this. Arnd

2 1

[alsa-devel] [PATCH v4 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 18 Oct '19

18 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v3: - fix a compile error by adding ifdef https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156750.h… - fix a coccinelle warning https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156754.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1128 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1296 insertions(+), 325 deletions(-) -- 2.23.0.700.g56cf767bdb-goog

2 16

[alsa-devel] [PATCH] ASoC: rsnd: Reinitialize bit clock inversion flag for every format setting
by Eugeniu Rosca 18 Oct '19

18 Oct '19

From: Junya Monden <jmonden(a)jp.adit-jv.com> Unlike other format-related DAI parameters, rdai->bit_clk_inv flag is not properly re-initialized when setting format for new stream processing. The inversion, if requested, is then applied not to default, but to a previous value, which leads to SCKP bit in SSICR register being set incorrectly. Fix this by re-setting the flag to its initial value, determined by format. Fixes: 1a7889ca8aba3 ("ASoC: rsnd: fixup SND_SOC_DAIFMT_xB_xF behavior") Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Cc: Jiada Wang <jiada_wang(a)mentor.com> Cc: Timo Wischer <twischer(a)de.adit-jv.com> Cc: stable(a)vger.kernel.org # v3.17+ Signed-off-by: Junya Monden <jmonden(a)jp.adit-jv.com> Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index bda5b958d0dc..e9596c2096cd 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -761,6 +761,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set format */ + rdai->bit_clk_inv = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; -- 2.23.0

3 2

[alsa-devel] [PATCH] ASoC: Intel: sof-rt5682: add a check for devm_clk_get
by Chuhong Yuan 18 Oct '19

18 Oct '19

sof_audio_probe misses a check for devm_clk_get and may cause problems. Add a check for it to fix the bug. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index a437567b8cee..6d15c7ff66bf 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -576,6 +576,15 @@ static int sof_audio_probe(struct platform_device *pdev) /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret); + return ret; + } + ret = clk_prepare_enable(ctx->mclk); if (ret < 0) { dev_err(&pdev->dev, -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: stm32: sai: fix sysclk management on shutdown
by Olivier Moysan 18 Oct '19

18 Oct '19

The commit below, adds a call to sysclk callback on shutdown. This introduces a regression in stm32 SAI driver, as some clock services are called twice, leading to unbalanced calls. Move processing related to mclk from shutdown to sysclk callback. When requested frequency is 0, assume shutdown and release mclk. Fixes: 2458adb8f92a ("SoC: simple-card-utils: set 0Hz to sysclk when shutdown") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index d7501f88aaa6..a4060813bc74 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -505,10 +505,20 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - (unsigned int)~SAI_XCR1_NODIV); + freq ? 0 : SAI_XCR1_NODIV); if (ret < 0) return ret; + /* Assume shutdown if requested frequency is 0Hz */ + if (!freq) { + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } + return 0; + } + /* If master clock is used, set parent clock now */ ret = stm32_sai_set_parent_clock(sai, freq); if (ret) @@ -1093,15 +1103,6 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0); - regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - SAI_XCR1_NODIV); - - /* Release mclk rate only if rate was actually set */ - if (sai->mclk_rate) { - clk_rate_exclusive_put(sai->sai_mclk); - sai->mclk_rate = 0; - } - clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); -- 2.17.1

2 1

[alsa-devel] [PATCH 1/2] ASoc: Add Texas Instruments TAS2562 amplifier binding
by Dan Murphy 18 Oct '19

18 Oct '19

Add the DT binding for the TAS2562 amplifier. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 +++++++++++++++++++ 1 file changed, 34 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt new file mode 100644 index 000000000000..658e1fb18a99 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -0,0 +1,34 @@ +Texas Instruments TAS2562 Smart PA + +The TAS2562 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - compatible: - Should contain "ti,tas2562". + - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. + - ti,imon-slot-no:- TDM TX current sense time slot. + +Optional properties: +- interrupt-parent: phandle to the interrupt controller which provides + the interrupt. +- interrupts: (GPIO) interrupt to which the chip is connected. +- shut-down: GPIO used to control the state of the device. + +Examples: +tas2562@4c { + #address-cells = <1>; + #size-cells = <0>; + compatible = "ti,tas2562"; + reg = <0x4c>; + + interrupt-parent = <&gpio1>; + interrupts = <14>; + + shut-down = <&gpio1 15 0>; + ti,imon-slot-no = <0>; +}; + -- 2.22.0.214.g8dca754b1e

3 4

[alsa-devel] [PATCH v3 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 18 Oct '19

18 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Tzung-Bi Shih (10): WIP: platform/chrome: cros_ec: remove unused EC feature WIP: ASoC: cros_ec_codec: refactor I2S RX WIP: ASoC: cros_ec_codec: extract DMIC EC command from I2S RX WIP: platform/chrome: cros_ec: add common commands for EC codec WIP: ASoC: cros_ec_codec: read max DMIC gain from EC codec WIP: ASoC: dt-bindings: cros_ec_codec: add SHM bindings WIP: ASoC: cros_ec_codec: support WoV WIP: ASoC: mediatek: mt6358: support WoV WIP: ASoC: dt-bindings: mt8183: add ec-codec WIP: ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1128 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1296 insertions(+), 325 deletions(-) -- 2.23.0.700.g56cf767bdb-goog

5 28

[alsa-devel] [PATCH] ASoC: SOF: control: return true when kcontrol values change
by Pierre-Louis Bossart 18 Oct '19

18 Oct '19

From: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> All the kcontrol put() functions are currently returning 0 when successful. This does not go well with alsamixer as it does not seem to get notified on SND_CTL_EVENT_MASK_VALUE callbacks when values change for (some of) the sof kcontrols. This patch fixes that by returning true for volume, switch and enum type kcontrols when values do change in put(). Signed-off-by: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/control.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 2c4abd406c4f..7baf7f1507c3 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -91,13 +91,16 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = - mixer_to_ipc(ucontrol->value.integer.value[i], + value = mixer_to_ipc(ucontrol->value.integer.value[i], scontrol->volume_table, sm->max + 1); + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of mixer updates */ @@ -107,8 +110,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, SOF_CTRL_TYPE_VALUE_CHAN_GET, SOF_CTRL_CMD_VOLUME, true); - - return 0; + return change; } int snd_sof_switch_get(struct snd_kcontrol *kcontrol, @@ -136,11 +138,15 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.integer.value[i]; + value = ucontrol->value.integer.value[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } if (scontrol->led_ctl.use_led) @@ -154,7 +160,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_SWITCH, true); - return 0; + return change; } int snd_sof_enum_get(struct snd_kcontrol *kcontrol, @@ -182,11 +188,15 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.enumerated.item[i]; + value = ucontrol->value.enumerated.item[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of enum updates */ @@ -197,7 +207,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_ENUM, true); - return 0; + return change; } int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, -- 2.20.1

2 1

[alsa-devel] [PATCH v3] ASoC: dt-bindings: Convert Allwinner A10 codec to a schema
by Maxime Ripard 18 Oct '19

18 Oct '19

The Allwinner SoCs have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by: Maxime Ripard <mripard(a)kernel.org> --- Changes from v2: - Change the audio-routing values to an enum, and enforce boundaries on the size - Add restrictions to the possible values of audio-routing Changes from v1: - Fix subject prefix --- .../sound/allwinner,sun4i-a10-codec.yaml | 267 ++++++++++++++++++ .../devicetree/bindings/sound/sun4i-codec.txt | 94 ------ 2 files changed, 267 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..b8f89c7258eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,267 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + 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. + allOf: + - $ref: /schemas/types.yaml#definitions/non-unique-string-array + - minItems: 2 + maxItems: 18 + items: + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "allwinner,sun7i-a20-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- dmas: DMA channels for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- resets: phandle to the reset control for this device -- allwinner,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 include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -Required properties for the following compatibles: - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- allwinner,codec-analog-controls: A phandle to the codec analog controls - block in the PRCM. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; -- 2.23.0

3 2

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: add a check for devm_clk_get
by Chuhong Yuan 18 Oct '19

18 Oct '19

aic32x4_set_dai_sysclk misses a check for devm_clk_get and may miss the failure. Add a check to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 68165de1c8de..b4e9a6c73f90 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -573,6 +573,9 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct clk *pll; pll = devm_clk_get(component->dev, "pll"); + if (IS_ERR(pll)) + return PTR_ERR(pll); + mclk = clk_get_parent(pll); return clk_set_rate(mclk, freq); -- 2.20.1

2 1

[alsa-devel] [PATCH][next] ASoC: rt1011: fix spelling mistake "temperture" -> "temperature"
by Colin King 18 Oct '19

18 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_dbg message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1011.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index ad049cfddcb0..dcd397a83cb4 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2373,7 +2373,7 @@ static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) device_property_read_u32(dev, "realtek,r0_calib", &rt1011->r0_calib); - dev_dbg(dev, "%s: r0_calib: 0x%x, temperture_calib: 0x%x", + dev_dbg(dev, "%s: r0_calib: 0x%x, temperature_calib: 0x%x", __func__, rt1011->r0_calib, rt1011->temperature_calib); return 0; -- 2.20.1

2 1

[alsa-devel] [PATCH v6] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 18 Oct '19

18 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: integrated fix for minor spelling issues] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v6: - integrated fix for minor spelling issues - collected tags --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 138 ++++++++++++++++++ 2 files changed, 138 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..53e3bad4178c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,138 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel (5.1) I2S with + secondary FIFO, s/w reset control and internal mux for root clock + source. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel (5.1) I2S for + playback, stereo channel capture, secondary FIFO using internal + or external DMA, s/w reset control, internal mux for root clock + source and 7.1 channel TDM support for playback; TDM (Time division + multiplexing) is to allow transfer of multiple channel audio data on + single data line. + + samsung,exynos7-i2s: with all the available features of Exynos5 I2S. + Exynos7 I2S has 7.1 channel TDM support for capture, secondary FIFO + with only external DMA and more number of root clock sampling + frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for I2S0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for I2S1 and I2S2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the I2S bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clock. I2S0 has internal mux to select the source + of root clock and I2S1 and I2S2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for I2S0 + - const: i2s_cdclk0 + - items: # for I2S1 + - const: i2s_cdclk1 + - items: # for I2S2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + subsystem (used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; -- 2.17.1

2 1

[alsa-devel] [PATCH 00/12] ALSA: firewire: configure interval of hw IRQ via ALSA PCM interface
by Takashi Sakamoto 18 Oct '19

18 Oct '19

Hi, ALSA IEC 61883-1/6 packet streaming engine controls OHCI 1394 controller to generate hardware IRQ for fixed number (=16) of isoc packets. Additionally, the size of packet buffer is fixed for triple of interval of hardware IRQ (=48). This commit allows ALSA PCM applications to configure these parameters by ALSA PCM interface. As a result, the size of PCM period is available in the range between 125 usec and 64 msec (or 32 msec). The hardware IRQ is scheduled for isoc packet which includes event across PCM period boundary. Takashi Sakamoto (12): ALSA: firewire-lib: use variable size of queue for isoc packets instead of fixed size ALSA: bebob: share PCM buffer size for both direction ALSA: fireworks: share PCM buffer size for both direction ALSA: oxfw: share PCM buffer size for both direction ALSA: dice: share PCM buffer size for both direction ALSA: firewire-digi00x: share PCM buffer size for both direction ALSA: firewire-tascam: share PCM buffer size for both direction ALSA: firewire-motu: share PCM buffer size for both direction ALSA: fireface: share PCM buffer size for both direction ALSA: firewire-lib: schedule hardware IRQ according to the size of PCM period ALSA: firewire-lib: tune the maximum available size of PCM period ALSA: firewire-lib: tune the minimum available size of PCM period sound/firewire/amdtp-stream.c | 154 ++++++++++++++------ sound/firewire/amdtp-stream.h | 9 +- sound/firewire/bebob/bebob.h | 3 +- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 5 +- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 20 ++- sound/firewire/dice/dice-stream.c | 5 +- sound/firewire/dice/dice.h | 3 +- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 12 +- sound/firewire/digi00x/digi00x-stream.c | 5 +- sound/firewire/digi00x/digi00x.h | 3 +- sound/firewire/fireface/ff-pcm.c | 13 +- sound/firewire/fireface/ff-stream.c | 5 +- sound/firewire/fireface/ff.h | 3 +- sound/firewire/fireworks/fireworks.h | 3 +- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 12 +- sound/firewire/fireworks/fireworks_stream.c | 5 +- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 12 +- sound/firewire/motu/motu-stream.c | 5 +- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-midi.c | 4 +- sound/firewire/oxfw/oxfw-pcm.c | 17 ++- sound/firewire/oxfw/oxfw-stream.c | 5 +- sound/firewire/oxfw/oxfw.h | 3 +- sound/firewire/tascam/tascam-pcm.c | 12 +- sound/firewire/tascam/tascam-stream.c | 5 +- sound/firewire/tascam/tascam.h | 3 +- 32 files changed, 266 insertions(+), 85 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 17 Oct '19

17 Oct '19

For P2P output, the output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitaion in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 40 +++++++++++++++++++++++++++++++--------- 1 file changed, 31 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..44d05ec28bd3 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -260,7 +260,7 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +268,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; if (div[IN] == 0) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); @@ -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) clk = asrc_priv->asrck_clk[clk_index[OUT]]; - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; - else - div[OUT] = clk_get_rate(clk) / outrate; + /* + * When P2P mode, output rate should align with the out samplerate. + * if set too high output rate, there will be lots of Overload. + * When M2M mode, output rate should also need to align with the out + * samplerate, but M2M must use less time to achieve good performance. + */ + clk_rate = clk_get_rate(clk); + if (p2p || !ideal) { + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; + } else { + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); + div[OUT] = clk_rate; + } if (div[OUT] == 0) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", @@ -372,6 +384,16 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || + rem[IN] != 0 || rem[OUT] != 0)) { + pair_err("The divider can't be used for non ideal mode\n"); + return -EINVAL; + } + + /* Divider range is [1, 1024] */ + div[IN] = min_t(u32, 1024, div[IN]); + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +582,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, true); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH v3] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 17 Oct '19

17 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: reordered non-standard properties, added list of values for widgets, minor other fixes] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,tm2-audio.txt | 42 -------- .../bindings/sound/samsung,tm2-audio.yaml | 99 +++++++++++++++++++ 2 files changed, 99 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,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 for sources and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml new file mode 100644 index 000000000000..c9178d928ad4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos SoC Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-codec: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: | + phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt; + - description: phandle of the HDMI transmitter node. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + 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 for sources and sinks are: + the WM5110's pins: + "HPOUT1L", + "HPOUT1R", + "HPOUT2L", + "HPOUT2R", + "HPOUT3L", + "HPOUT3R", + MAX98504's pins: + "SPKOUT" + and the jacks on the board: + "HP", + "SPK", + "RCV", + "Main Mic", + "Sub Mic", + "Third Mic", + "Headset Mic". + +required: + - compatible + - model + - i2s-controller + - audio-codec + - audio-amplifier + - mic-bias-gpios + - samsung,audio-routing + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-codec = <&wm5110>, <&hdmi>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3>; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; + }; + -- 2.17.1

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Re: [alsa-devel] linux-next: Tree for Oct 17 (soundwire)
by Randy Dunlap 17 Oct '19

17 Oct '19

On 10/16/19 9:26 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20191016: > on x86_64: # CONFIG_BLOCK is not set CC [M] drivers/soundwire/intel_init.o In file included from ../drivers/soundwire/intel_init.c:12:0: ../include/linux/iomap.h: In function ‘iomap_sector’: ../include/linux/iomap.h:76:48: error: ‘SECTOR_SHIFT’ undeclared (first use in this function); did you mean ‘SECTIONS_SHIFT’? return (iomap->addr + pos - iomap->offset) >> SECTOR_SHIFT; ^~~~~~~~~~~~ SECTIONS_SHIFT -- ~Randy

2 1

[alsa-devel] [PATCH] ALSA: hda - Force runtime PM on Nvidia HDMI codecs
by Lukas Wunner 17 Oct '19

17 Oct '19

Przemysław Kopa reports that since commit b516ea586d71 ("PCI: Enable NVIDIA HDA controllers"), the discrete GPU Nvidia GeForce GT 540M on his 2011 Samsung laptop refuses to runtime suspend, resulting in a power regression and excessive heat. Rivera Valdez witnesses the same issue with a GeForce GT 525M (GF108M) of the same era, as does another Arch Linux user named "R0AR" with a more recent GeForce GTX 1050 Ti (GP107M). The commit exposes the discrete GPU's HDA controller and all four codecs on the controller do not set the CLKSTOP and EPSS bits in the Supported Power States Response. They also do not set the PS-ClkStopOk bit in the Get Power State Response. hda_codec_runtime_suspend() therefore does not call snd_hdac_codec_link_down(), which prevents each codec and the PCI device from runtime suspending. The same issue is present on some AMD discrete GPUs and we addressed it by forcing runtime PM despite the bits not being set, see commit 57cb54e53bdd ("ALSA: hda - Force to link down at runtime suspend on ATI/AMD HDMI"). Do the same for Nvidia HDMI codecs. Fixes: b516ea586d71 ("PCI: Enable NVIDIA HDA controllers") Link: https://bbs.archlinux.org/viewtopic.php?pid=1865512 Link: https://bugs.freedesktop.org/show_bug.cgi?id=75985#c81 Reported-by: Przemysław Kopa <prymoo(a)gmail.com> Reported-by: Rivera Valdez <riveravaldez(a)ysinembargo.com> Signed-off-by: Lukas Wunner <lukas(a)wunner.de> Cc: Daniel Drake <dan(a)reactivated.net> Cc: stable(a)vger.kernel.org # v5.3+ --- 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 bca5de7..795cbda 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3474,6 +3474,8 @@ static int patch_nvhdmi(struct hda_codec *codec) nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + codec->link_down_at_suspend = 1; + generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin); return 0; -- 2.20.1

2 1

[alsa-devel] [PATCH 1/7] ASoC: amd: No need PCI-MSI interrupts
by Ravulapati Vishnu vardhan rao 17 Oct '19

17 Oct '19

ACP-PCI controller driver does not depends msi interrupts. So removed msi related pci functions which have no use and does not impact on existing functionality. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..8f6bf00 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -46,14 +46,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, goto release_regions; } - /* check for msi interrupt support */ - ret = pci_enable_msi(pci); - if (ret) - /* msi is not enabled */ - irqflags = IRQF_SHARED; - else - /* msi is enabled */ - irqflags = 0; + irqflags = 0; addr = pci_resource_start(pci, 0); adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); @@ -112,7 +105,6 @@ static int snd_acp3x_probe(struct pci_dev *pci, return 0; unmap_mmio: - pci_disable_msi(pci); iounmap(adata->acp3x_base); release_regions: pci_release_regions(pci); @@ -129,7 +121,6 @@ static void snd_acp3x_remove(struct pci_dev *pci) platform_device_unregister(adata->pdev); iounmap(adata->acp3x_base); - pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); } -- 2.7.4

6 23

[alsa-devel] [PATCH -next] ASoC: atmel: Fix build error
by YueHaibing 17 Oct '19

17 Oct '19

when do randbuilding, I got this error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x12f6): undefined reference to `atmel_pcm_pdc_platform_register' This is because SND_ATMEL_SOC_SSC_DMA=y, SND_ATMEL_SOC_SSC=y, but SND_ATMEL_SOC_SSC_PDC=m. Fix it bt reintroducing the default Kconfig. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/atmel/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c22..79e45f2 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -12,10 +12,14 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC tristate depends on HAS_DMA + default m if SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=m + default y if SND_ATMEL_SOC_SSC_PDC=y || (SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=y) config SND_ATMEL_SOC_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM + default m if SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=m + default y if SND_ATMEL_SOC_SSC_DMA=y || (SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=y) config SND_ATMEL_SOC_SSC tristate -- 2.7.4

4 5

[alsa-devel] [sound:for-next 2/20] drivers/pci/pci.c:5859:6: error: redefinition of 'pci_pr3_present'
by kbuild test robot 17 Oct '19

17 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: fa5a68b2abcefe2204075c80662e02cf59567a9f commit: 2e1a529c10e18e62da55eac42c278f11216dddaa [2/20] PCI: Add pci_pr3_present() helper to check Power Resource for D3hot config: parisc-c3000_defconfig (attached as .config) compiler: hppa-linux-gcc (GCC) 7.4.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 2e1a529c10e18e62da55eac42c278f11216dddaa # save the attached .config to linux build tree GCC_VERSION=7.4.0 make.cross ARCH=parisc If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): >> drivers/pci/pci.c:5859:6: error: redefinition of 'pci_pr3_present' bool pci_pr3_present(struct pci_dev *pdev) ^~~~~~~~~~~~~~~ In file included from include/linux/of_pci.h:5:0, from drivers/pci/pci.c:17: include/linux/pci.h:2317:13: note: previous definition of 'pci_pr3_present' was here static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ drivers/pci/pci.c: In function 'pci_pr3_present': >> drivers/pci/pci.c:5870:13: error: dereferencing pointer to incomplete type 'struct acpi_device' return adev->power.flags.power_resources && ^~ >> drivers/pci/pci.c:5871:3: error: implicit declaration of function 'acpi_has_method'; did you mean 'acpi_has_watchdog'? [-Werror=implicit-function-declaration] acpi_has_method(adev->handle, "_PR3"); ^~~~~~~~~~~~~~~ acpi_has_watchdog In file included from include/linux/of_pci.h:5:0, from drivers/pci/pci.c:17: At top level: include/linux/pci.h:2317:13: warning: 'pci_pr3_present' defined but not used [-Wunused-function] static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/pci_pr3_present +5859 drivers/pci/pci.c 5858 > 5859 bool pci_pr3_present(struct pci_dev *pdev) 5860 { 5861 struct acpi_device *adev; 5862 5863 if (acpi_disabled) 5864 return false; 5865 5866 adev = ACPI_COMPANION(&pdev->dev); 5867 if (!adev) 5868 return false; 5869 > 5870 return adev->power.flags.power_resources && > 5871 acpi_has_method(adev->handle, "_PR3"); 5872 } 5873 EXPORT_SYMBOL_GPL(pci_pr3_present); 5874 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Enable headset mic on Asus MJ401TA
by Daniel Drake 17 Oct '19

17 Oct '19

On Asus MJ401TA (with Realtek ALC256), the headset mic is connected to pin 0x19, with default configuration value 0x411111f0 (indicating no physical connection). Enable this by quirking the pin. Mic jack detection was also tested and found to be working. This enables use of the headset mic on this product. Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..e9fc5d8ff497 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5869,6 +5869,7 @@ enum { ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, + ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, @@ -6912,6 +6913,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE }, + [ALC256_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, [ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7108,6 +7118,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 2.20.1

2 1

[alsa-devel] [PATCH v5 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 17 Oct '19

17 Oct '19

Hey, here's the 5th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. v5 changes: - Fixed a bug with codec power management at probe. To imitate snd_hda_intel driver, SOF needs to enable display power when doing codec probe for HDMI codecs. Affects patches 2 and 5. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 110 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 ++- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.h | 93 +++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 337 insertions(+), 27 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

2 11

[alsa-devel] [PATCH 00/17] firewire: share the size of period for both playback and capture PCM substreams
by Takashi Sakamoto 17 Oct '19

17 Oct '19

Hi, This patchset is a preparation for enhancement of AMDTP domain that I addressed to my previous patchset: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/152430.html When queued packets for several IT/IR contexts in the same domain are handled in hardware IRQ context for an IT context, the number of events in the packets are mostly the same for each of contexts. This means that the size of period for PCM substreams associated to the contexts is also the same. At present, PCM substreams for the contexts can have own size of period. This commit adds a member into AMDTP domain to share the size of period for PCM substreams on AMDTP streams in the same domain. This patchset includes optimization to detect whether isochronous resources for AMDTP streams are reserved or not in pcm.open callback. Takashi Sakamoto (17): firewire-lib: add a member into AMDTP domain for events per period bebob: register the size of PCM period to AMDTP domain fireworks: register the size of PCM period to AMDTP domain oxfw: register the size of PCM period to AMDTP domain dice: register the size of PCM period to AMDTP domain firewire-digi00x: register the size of PCM period to AMDTP domain firewire-tascam: register the size of PCM period to AMDTP domain firewire-motu: register the size of PCM period to AMDTP domain fireface: register the size of PCM period to AMDTP domain bebob: use the same size of period for PCM substream in AMDTP streams fireworks: use the same size of period for PCM substream in AMDTP streams oxfw: use the same size of period for PCM substream in AMDTP streams dice: use the same size of period for PCM substream in AMDTP streams firewire-digi00x: use the same size of period for PCM substream in AMDTP streams firewire-tascam: use the same size of period for PCM substream in AMDTP streams firewire-motu: use the same size of period for PCM substream in AMDTP streams fireface: use the same size of period for PCM substreams in AMDTP streams sound/firewire/amdtp-stream.c | 4 ++ sound/firewire/amdtp-stream.h | 10 +++++ sound/firewire/bebob/bebob.h | 3 +- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 43 ++++++++++++------ sound/firewire/bebob/bebob_stream.c | 11 ++++- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 50 +++++++++++++++------ sound/firewire/dice/dice-stream.c | 8 +++- sound/firewire/dice/dice.h | 3 +- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 39 ++++++++++++---- sound/firewire/digi00x/digi00x-stream.c | 11 ++++- sound/firewire/digi00x/digi00x.h | 3 +- sound/firewire/fireface/ff-pcm.c | 32 ++++++++++--- sound/firewire/fireface/ff-stream.c | 11 ++++- sound/firewire/fireface/ff.h | 3 +- sound/firewire/fireworks/fireworks.h | 3 +- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 43 +++++++++++++----- sound/firewire/fireworks/fireworks_stream.c | 11 ++++- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 33 ++++++++++---- sound/firewire/motu/motu-stream.c | 11 ++++- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-midi.c | 4 +- sound/firewire/oxfw/oxfw-pcm.c | 45 +++++++++++++------ sound/firewire/oxfw/oxfw-stream.c | 12 ++++- sound/firewire/oxfw/oxfw.h | 3 +- sound/firewire/tascam/tascam-pcm.c | 38 ++++++++++++---- sound/firewire/tascam/tascam-stream.c | 11 ++++- sound/firewire/tascam/tascam.h | 3 +- 32 files changed, 354 insertions(+), 107 deletions(-) -- 2.20.1

2 18

[alsa-devel] [PATCH v4 1/2] PCI: Add a helper to check Power Resource Requirements _PR3 existence
by Kai-Heng Feng 17 Oct '19

17 Oct '19

A driver may want to know the existence of _PR3, to choose different runtime suspend behavior. A user will be add in next patch. This is mostly the same as nouveau_pr3_present(). Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v4: - Let caller to find its upstream port device. drivers/pci/pci.c | 16 ++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 18 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index e7982af9a5d8..d03f624d8928 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,22 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct acpi_device *adev; + + if (acpi_disabled) + return false; + + adev = ACPI_COMPANION(&pdev->dev); + if (!adev) + return false; + + return adev->power.flags.power_resources && + acpi_has_method(adev->handle, "_PR3"); +} +EXPORT_SYMBOL_GPL(pci_pr3_present); + /** * pci_add_dma_alias - Add a DMA devfn alias for a device * @dev: the PCI device for which alias is added diff --git a/include/linux/pci.h b/include/linux/pci.h index f9088c89a534..1d15c5d49cdd 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2310,9 +2310,11 @@ struct irq_domain *pci_host_bridge_acpi_msi_domain(struct pci_bus *bus); void pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); +bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } +static bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH -- 2.17.1

3 5

[alsa-devel] [PATCH v5.3] ASoC: sgtl5000: add ADC mute control
by Richard Leitner 16 Oct '19

16 Oct '19

From: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> Upstream commit 631bc8f0134a ("ASoC: sgtl5000: Fix of unmute outputs on probe"), which is e9f621efaebd in v5.3 replaced snd_soc_component_write with snd_soc_component_update_bits and therefore no longer cleared the MUTE_ADC flag. This caused the ADC to stay muted and recording doesn't work any longer. This patch fixes this problem by adding a Switch control for MUTE_ADC. commit 694b14554d75 ("ASoC: sgtl5000: add ADC mute control") upstream This control mute/unmute the ADC input of SGTL5000 using its CHIP_ANA_CTRL register. Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> Reviewed-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Reviewed-by: Igor Opaniuk <igor.opaniuk(a)toradex.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20190719100524.23300-5-oleksandr.suvorov@toradex.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> Fixes: e9f621efaebd ("ASoC: sgtl5000: Fix of unmute outputs on probe") --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3f28e7862b5b..b65232521ea8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -720,6 +720,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SGTL5000_CHIP_ANA_ADC_CTRL, 8, 1, 0, capture_6db_attenuate), SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + SOC_SINGLE("Capture Switch", SGTL5000_CHIP_ANA_CTRL, 0, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", SGTL5000_CHIP_ANA_HP_CTRL, -- 2.21.0

2 1

[alsa-devel] [PATCH v6 0/7] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 16 Oct '19

16 Oct '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, To be able to add support for the Allwinner H6 I've changed some of the original reg fields into function calls as this made it easier to setup for multi-channel audio especially across different SoCs. I've also stripped out all the other patches unrelated to this which I will deliver after support for the H6 has gone in. These other patches are required for HDMI audio which is driving this patchset and they can be found here https://github.com/codekipper/linux-sunxi/commits/upstream-i2s BR, CK --- v6 changes compared to v5 are: - rebased onto the recent tdm delivery - stripped out patches not required for the H6 delivery v5 changes compared to v4 are: - removed delivered patches. - Added more details to commit messages. - replaced some reg fields with function calls. - Added DSP_A and DSP_B support for H3 and later SoCs. - Added support for the Allwinner H6. v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Jernej Skrabec (2): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S Marcus Cooper (5): ASoC: sun4i-i2s: Move channel select offset ASoC: sun4i-i2s: Add functions for RX and TX channel offsets ASoC: sun4i-i2s: Add functions for RX and TX channel enables ASoC: sun4i-i2s: Add functions for RX and TX channel selects ASoC: sun4i-i2s: Add functions for RX and TX channel mapping .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + sound/soc/sunxi/sun4i-i2s.c | 337 ++++++++++++++++-- 2 files changed, 305 insertions(+), 34 deletions(-) -- 2.23.0

3 13

[alsa-devel] [PATCH v2] ASoC: rt1011: set tx/rx slots from tx/rx_mask in TDM case
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> The TX/RX slot configuration use tx/rx_mask which requested by the machine driver. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 142 ++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt1011.h | 23 ++++++ 2 files changed, 160 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index be1e276e3631..b927e61b32df 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1805,8 +1805,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, struct snd_soc_component *component = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - unsigned int val = 0, tdm_en = 0; - int ret = 0; + unsigned int val = 0, tdm_en = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit, last_bit; snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) @@ -1829,6 +1829,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, break; default: ret = -EINVAL; + goto _set_tdm_err_; } switch (slot_width) { @@ -1848,6 +1849,135 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, break; default: ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + first_bit = find_next_bit((unsigned long *)&rx_mask, 32, 0); + if (rx_slotnum > 1 || rx_slotnum == 0) { + ret = -EINVAL; + dev_dbg(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + first_bit = find_next_bit((unsigned long *)&tx_mask, 32, 0); + last_bit = find_last_bit((unsigned long *)&tx_mask, 32); + if (tx_slotnum > 2 || (last_bit-first_bit) > 1) { + ret = -EINVAL; + dev_dbg(component->dev, "too many tx slots or tx slot location error\n"); + goto _set_tdm_err_; + } + + if (tx_slotnum == 1) { + snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, first_bit); + switch (first_bit) { + case 1: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, + RT1011_TDM_I2S_RX_ADC1_1_LL); + break; + case 3: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, + RT1011_TDM_I2S_RX_ADC2_1_LL); + break; + case 5: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, + RT1011_TDM_I2S_RX_ADC3_1_LL); + break; + case 7: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, + RT1011_TDM_I2S_RX_ADC4_1_LL); + break; + case 0: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, 0); + break; + case 2: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, 0); + break; + case 4: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, 0); + break; + case 6: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, 0); + break; + default: + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location error\n"); + goto _set_tdm_err_; + } + } else if (tx_slotnum == 2) { + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, + RT1011_TDM_I2S_DOCK_ADCDAT_2CH | first_bit); + break; + default: + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location should be paired and start from slot0/2/4/6\n"); + goto _set_tdm_err_; + } } snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, @@ -1860,10 +1990,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); - snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, - RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, - RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); + if (tx_slotnum) + snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, + RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, + RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); +_set_tdm_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..3f7dea1191ad 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -460,6 +460,23 @@ #define RT1011_TDM_I2S_DOCK_EN_1_MASK (0x1 << 3) #define RT1011_TDM_I2S_DOCK_EN_1_SFT 3 #define RT1011_TDM_I2S_DOCK_EN_1 (0x1 << 3) +#define RT1011_TDM_ADCDAT1_DATA_LOCATION (0x7 << 0) + +/* TDM1 Setting-3 (0x0118) */ +#define RT1011_TDM_I2S_RX_ADC1_1_MASK (0x3 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_MASK (0x3 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_MASK (0x3 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_MASK (0x3 << 0) +#define RT1011_TDM_I2S_RX_ADC1_1_LL (0x2 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_LL (0x2 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_LL (0x2 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_LL (0x2 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1011_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1011_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1011_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1011_TDM_I2S_TX_R_DAC1_1_SFT 8 /* TDM2 Setting-2 (0x0120) */ #define RT1011_TDM_I2S_DOCK_ADCDAT_LEN_2_MASK (0x7 << 13) @@ -585,6 +602,12 @@ #define RT1011_STP_T0_EN_BIT 6 #define RT1011_STP_T0_EN (0x1 << 6) +/* Cross Biquad Setting-1 (0x0702) */ +#define RT1011_MONO_LR_SEL_MASK (0x3 << 5) +#define RT1011_MONO_L_CHANNEL (0x0 << 5) +#define RT1011_MONO_R_CHANNEL (0x1 << 5) +#define RT1011_MONO_LR_MIX_CHANNEL (0x2 << 5) + /* ClassD Internal Setting-1 (0x1300) */ #define RT1011_DRIVER_READY_SPK (0x1 << 12) #define RT1011_DRIVER_READY_SPK_BIT 12 -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: dt-bindings: rt1011: add r0 and temperature device property
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> Typically, the r0 (calibration data) and temperature were measured in the factory. This information is written into the non-volatile area where keeps data whether factory reset or OS update. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt1011.txt | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt1011.txt b/Documentation/devicetree/bindings/sound/rt1011.txt index 35a23e60d679..02d53b9aa247 100644 --- a/Documentation/devicetree/bindings/sound/rt1011.txt +++ b/Documentation/devicetree/bindings/sound/rt1011.txt @@ -20,6 +20,14 @@ Required properties: | 1 | 1 | 0x3b | ------------------------------------- +Optional properties: + +- realtek,temperature_calib + u32. The temperature was measured while doing the calibration. Units: Celsius degree + +- realtek,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + Pins on the device (for linking into audio routes) for RT1011: * SPO @@ -29,4 +37,6 @@ Example: rt1011: codec@38 { compatible = "realtek,rt1011"; reg = <0x38>; + realtek,temperature_calib = <25>; + realtek,r0_calib = <0x224050>; }; -- 2.23.0

2 3

[alsa-devel] [PATCH] ASoC: wm8958: use <asm/unaligned.h> to simplify code
by Ben Dooks (Codethink) 16 Oct '19

16 Oct '19

Simplify the memcpy/be32_to_cpu() code by simply using get_unaligned_be32() throughout and makes the code nicer to look at. This fixes the following warnings from sparse: sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: patches(a)opensource.cirrus.com Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/wm8958-dsp2.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 18535b326680..ca42445b649d 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -25,6 +25,8 @@ #include <linux/mfd/wm8994/pdata.h> #include <linux/mfd/wm8994/gpio.h> +#include <asm/unaligned.h> + #include "wm8994.h" #define WM_FW_BLOCK_INFO 0xff @@ -58,18 +60,15 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, } if (memcmp(fw->data, "WMFW", 4) != 0) { - memcpy(&data32, fw->data, sizeof(data32)); - data32 = be32_to_cpu(data32); + data32 = get_unaligned_be32(fw->data); dev_err(component->dev, "%s: firmware has bad file magic %08x\n", name, data32); goto err; } - memcpy(&data32, fw->data + 4, sizeof(data32)); - len = be32_to_cpu(data32); + len = get_unaligned_be32(fw->data + 4); + data32 = get_unaligned_be32(fw->data + 8); - memcpy(&data32, fw->data + 8, sizeof(data32)); - data32 = be32_to_cpu(data32); if ((data32 >> 24) & 0xff) { dev_err(component->dev, "%s: unsupported firmware version %d\n", name, (data32 >> 24) & 0xff); @@ -87,9 +86,8 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, } if (check) { - memcpy(&data64, fw->data + 24, sizeof(u64)); - dev_info(component->dev, "%s timestamp %llx\n", - name, be64_to_cpu(data64)); + data64 = get_unaligned_be64(fw->data + 24); + dev_info(component->dev, "%s timestamp %llx\n", name, data64); } else { snd_soc_component_write(component, 0x102, 0x2); snd_soc_component_write(component, 0x900, 0x2); @@ -104,8 +102,7 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, goto err; } - memcpy(&data32, data + 4, sizeof(data32)); - block_len = be32_to_cpu(data32); + block_len = get_unaligned_be32(data + 4); if (block_len + 8 > len) { dev_err(component->dev, "%zd byte block longer than file\n", block_len); @@ -116,8 +113,7 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, goto err; } - memcpy(&data32, data, sizeof(data32)); - data32 = be32_to_cpu(data32); + data32 = get_unaligned_be32(data); switch ((data32 >> 24) & 0xff) { case WM_FW_BLOCK_INFO: -- 2.23.0

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[alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> Typically, the r0 (calibration data) and temperature were measured in the factory. This information is written into the non-volatile area where keeps data whether factory reset or OS update. In Chromium OS case, the coreboot will read the info from VPD and create the device property for each rt1011. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 41 ++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt1011.h | 1 + 2 files changed, 41 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index b927e61b32df..ad049cfddcb0 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2328,8 +2328,12 @@ static void rt1011_calibration_work(struct work_struct *work) struct rt1011_priv *rt1011 = container_of(work, struct rt1011_priv, cali_work); struct snd_soc_component *component = rt1011->component; + unsigned int r0_integer, r0_factor, format; - rt1011_calibrate(rt1011, 1); + if (rt1011->r0_calib) + rt1011_calibrate(rt1011, 0); + else + rt1011_calibrate(rt1011, 1); /* * This flag should reset after booting. @@ -2340,6 +2344,39 @@ static void rt1011_calibration_work(struct work_struct *work) /* initial */ rt1011_reg_init(component); + + /* Apply temperature and calibration data from device property */ + if (rt1011->temperature_calib) { + snd_soc_component_update_bits(component, + RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, + (rt1011->temperature_calib << 2)); + } + + if (rt1011->r0_calib) { + rt1011->r0_reg = rt1011->r0_calib; + + format = 2147483648U; /* 2^24 * 128 */ + r0_integer = format / rt1011->r0_reg / 128; + r0_factor = ((format / rt1011->r0_reg * 100) / 128) + - (r0_integer * 100); + dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", + r0_integer, r0_factor, rt1011->r0_reg); + + rt1011_r0_load(rt1011); + } +} + +static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) +{ + device_property_read_u32(dev, "realtek,temperature_calib", + &rt1011->temperature_calib); + device_property_read_u32(dev, "realtek,r0_calib", + &rt1011->r0_calib); + + dev_dbg(dev, "%s: r0_calib: 0x%x, temperture_calib: 0x%x", + __func__, rt1011->r0_calib, rt1011->temperature_calib); + + return 0; } static int rt1011_i2c_probe(struct i2c_client *i2c, @@ -2356,6 +2393,8 @@ static int rt1011_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1011); + rt1011_parse_dp(rt1011, &i2c->dev); + rt1011->regmap = devm_regmap_init_i2c(i2c, &rt1011_regmap); if (IS_ERR(rt1011->regmap)) { ret = PTR_ERR(rt1011->regmap); diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 3f7dea1191ad..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -690,6 +690,7 @@ struct rt1011_priv { int bq_drc_set; unsigned int r0_reg, cali_done; + unsigned int r0_calib, temperature_calib; int recv_spk_mode; }; -- 2.23.0

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[alsa-devel] [PATCH] ASoC: sof: include types.h at header.h
by Kuninori Morimoto 16 Oct '19

16 Oct '19

Content-Transfer-Encoding: 8bit Without <types.h> we will get these error linux/include/sound/sof/header.h:125:2: error: unknown type name ‘uint32_t’uint32_t size; linux/include/sound/sof/header.h:136:2: error: unknown type name ‘uint32_t’uint32_t size; linux/include/sound/sof/header.h:137:2: error: unknown type name ‘uint32_t’uint32_t cmd; ... linux/include/sound/sof/dai-imx.h:18:2: error: unknown type name ‘uint16_t’uint16_t reserved1; linux/include/sound/sof/dai-imx.h:30:2: error: unknown type name ‘uint16_t’uint16_t tdm_slot_width; linux/include/sound/sof/dai-imx.h:31:2: error: unknown type name ‘uint16_t’uint16_t reserved2; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/sof/header.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 10f00c0..332143f 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -9,6 +9,7 @@ #ifndef __INCLUDE_SOUND_SOF_HEADER_H__ #define __INCLUDE_SOUND_SOF_HEADER_H__ +#include <linux/types.h> #include <uapi/sound/sof/abi.h> /** \addtogroup sof_uapi uAPI -- 2.7.4

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