March 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Silence PM ops build warning
by Lukas Wunner 13 Apr '18

13 Apr '18

The system sleep PM ops azx_suspend() and azx_resume() were previously called by vga_switcheroo, but commit 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") removed their invocation. Unfortunately the commit neglected to update the #ifdef surrounding the two functions, so if CONFIG_PM_SLEEP is *not* enabled but all three of CONFIG_PM, CONFIG_VGA_SWITCHEROO and CONFIG_SND_HDA_CODEC_HDMI *are* enabled, the compiler now emits the following warning: sound/pci/hda/hda_intel.c:1024:12: warning: 'azx_resume' defined but not used [-Wunused-function] static int azx_resume(struct device *dev) ^~~~~~~~~~ sound/pci/hda/hda_intel.c:989:12: warning: 'azx_suspend' defined but not used [-Wunused-function] static int azx_suspend(struct device *dev) ^~~~~~~~~~~ Silence by updating the #ifdef. Because the #ifdef block now uses the same condition as the one immediately succeeding it, the two blocks can be collapsed together, shaving off another two lines. Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Cc: Takashi Iwai <tiwai(a)suse.com> Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Lukas Wunner <lukas(a)wunner.de> --- sound/pci/hda/hda_intel.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec4e6b829ee2..d653c56e9141 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -982,7 +982,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -#if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1063,9 +1063,7 @@ static int azx_resume(struct device *dev) trace_azx_resume(chip); return 0; } -#endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ -#ifdef CONFIG_PM_SLEEP /* put codec down to D3 at hibernation for Intel SKL+; * otherwise BIOS may still access the codec and screw up the driver */ -- 2.16.2

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[alsa-devel] [PATCH] ASoC: rsnd: mark PM functions __maybe_unused
by Arnd Bergmann 12 Apr '18

12 Apr '18

The suspend/resume callbacks are now optional, leading to a warning when they are unused: sound/soc/sh/rcar/core.c:1548:12: error: 'rsnd_resume' defined but not used [-Werror=unused-function] static int rsnd_resume(struct device *dev) ^~~~~~~~~~~ sound/soc/sh/rcar/core.c:1539:12: error: 'rsnd_suspend' defined but not used [-Werror=unused-function] static int rsnd_suspend(struct device *dev) This marks the as __maybe_unused to avoid the warning. Fixes: f8a9a29c4fe9 ("ASoC: rsnd: set pm_ops in hibernate-compatible way") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a76688a8ba9..94f081b93258 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1536,7 +1536,7 @@ static int rsnd_remove(struct platform_device *pdev) return ret; } -static int rsnd_suspend(struct device *dev) +static int __maybe_unused rsnd_suspend(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); @@ -1545,7 +1545,7 @@ static int rsnd_suspend(struct device *dev) return 0; } -static int rsnd_resume(struct device *dev) +static int __maybe_unused rsnd_resume(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); -- 2.9.0

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[alsa-devel] [PATCH] ASoC: Intel: atom: fix ACPI/PCI Kconfig
by Pierre-Louis Bossart 12 Apr '18

12 Apr '18

The split between ACPI and PCI platforms generated issues with randconfig: with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! To keep things simple, let's expose two configs for SND_SST_ATOM_HIFI2_PLATFORM_PCI and SND_SST_ATOM_HIFI2_PLATFORM_ACPI, which select a common SND_SST_ATOM_HIFI2_PLATFORM option. To avoid breaking existing solutions with the semantics change, SND_SST_ATOM_HIFI2_PLATFORM_ACPI uses "default ACPI" so that "make oldnoconfig" and "make olddefconfig" still work as expected. Also remove mentions of Medfield while we are at it since it was removed recently. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f2c9e8c5970a..d7ac77b0197b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. +config SND_SST_ATOM_HIFI2_PLATFORM + tristate + select SND_SOC_COMPRESS + config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI select SND_SST_IPC_PCI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM help - If you have a Intel Medfield or Merrifield/Edison platform, then + If you have a Intel Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree + enable this option: while Merrifield/Edison can run a mainline + kernel with limited functionality it will require a firmware file + which is not in the standard firmware tree -config SND_SST_ATOM_HIFI2_PLATFORM +config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + default ACPI depends on X86 && ACPI select SND_SST_IPC_ACPI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI help -- 2.14.1

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[alsa-devel] [Xen-devel][PATCH v3 0/5] sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 12 Apr '18

12 Apr '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Hello, all! In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - bump protocol version to 2 - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events - add XENSND_OP_HW_PARAM_QUERY request to read/update stream configuration space: request passes desired intervals/formats for the stream parameters and the response returns allowed intervals and formats mask that can be used. Changes since v2: 1. Konrad's r-b tag for version patch 2. MAJOR: changed req/resp/evt packet sizes from 32 to 64 octets 3. Reworked XENSND_OP_HW_PARAM_QUERY so it now sends all parameters at once, allowing to check all the configuration space. 4. Minor documentation cleanup (added missed "reserved" fields) Changes since v1: 1. Changed protocol version definition from string to integer, so it can easily be used in comparisons. Konrad, I have removed your r-b tag for the reason of this change. 2. In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameter given: request passes desired parameter interval (mask) and the response to this request returns min/max interval (mask) for the parameter to be used. Parameters supported by this request/response: - format mask - sample rate interval - number of channels interval - buffer size, interval, frames - period size, interval, frames Oleksandr Andrushchenko (5): sndif: Introduce protocol version sndif: Fix missed "reserved" fields in comments sndif: Make requests and responses 64 octets long sndif: Add explicit back and front synchronization sndif: Add explicit back and front parameter negotiation xen/include/public/io/sndif.h | 322 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 306 insertions(+), 16 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: AMD: Support headset button on Stoney DA7219
by Akshu Agrawal 12 Apr '18

12 Apr '18

Adds headset button support. TEST=Tested Volume UP/Down functionality Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..d9491e1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -33,6 +33,7 @@ #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/input.h> #include <linux/acpi.h> #include "../codecs/da7219.h" @@ -51,6 +52,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack *jack; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -80,6 +82,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + jack = &cz_jack; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &cz_jack); return 0; -- 1.9.1

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[alsa-devel] [PATCH v2 1/3] dell-led: Change dell-led.h to dell-common.h
by Kai-Heng Feng 07 Apr '18

07 Apr '18

This header will be used for more than just led. Change it to a more generic name. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: Mario suggested to squash the HDA part into the same series. drivers/platform/x86/dell-laptop.c | 2 +- include/linux/{dell-led.h => dell-common.h} | 4 ++-- sound/pci/hda/dell_wmi_helper.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) rename include/linux/{dell-led.h => dell-common.h} (61%) diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index c52c6723374b..8ba820e6c3d0 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -29,7 +29,7 @@ #include <linux/mm.h> #include <linux/i8042.h> #include <linux/debugfs.h> -#include <linux/dell-led.h> +#include <linux/dell-common.h> #include <linux/seq_file.h> #include <acpi/video.h> #include "dell-rbtn.h" diff --git a/include/linux/dell-led.h b/include/linux/dell-common.h similarity index 61% rename from include/linux/dell-led.h rename to include/linux/dell-common.h index 92521471517f..37e4b614dd74 100644 --- a/include/linux/dell-led.h +++ b/include/linux/dell-common.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ -#ifndef __DELL_LED_H__ -#define __DELL_LED_H__ +#ifndef __DELL_COMMON_H__ +#define __DELL_COMMON_H__ int dell_micmute_led_set(int on); diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..56050cc3c0ee 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -4,7 +4,7 @@ */ #if IS_ENABLED(CONFIG_DELL_LAPTOP) -#include <linux/dell-led.h> +#include <linux/dell-common.h> enum { MICMUTE_LED_ON, -- 2.15.1

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[alsa-devel] KASAN: use-after-free Read in snd_pcm_timer_resolution
by syzbot 05 Apr '18

05 Apr '18

Hello, syzbot hit the following crash on upstream commit 0b412605ef5f5c64b31f19e2910b1d5eba9929c3 (Thu Mar 29 01:07:23 2018 +0000) Merge tag 'drm-fixes-for-v4.16-rc8' of git://people.freedesktop.org/~airlied/linux syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=8e62ff4e07aa2ce87826 So far this crash happened 3 times on upstream. C reproducer: https://syzkaller.appspot.com/x/repro.c?id=6219387178582016 syzkaller reproducer: https://syzkaller.appspot.com/x/repro.syz?id=5176598244360192 Raw console output: https://syzkaller.appspot.com/x/log.txt?id=5815694612496384 Kernel config: https://syzkaller.appspot.com/x/.config?id=-8440362230543204781 compiler: gcc (GCC) 7.1.1 20170620 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+8e62ff4e07aa2ce87826(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. ================================================================== BUG: KASAN: use-after-free in snd_pcm_timer_resolution+0xa3/0xc0 sound/core/pcm_timer.c:72 Read of size 4 at addr ffff8801bdd58490 by task syzkaller702213/13238 CPU: 0 PID: 13238 Comm: syzkaller702213 Not tainted 4.16.0-rc7+ #370 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x24d lib/dump_stack.c:53 print_address_description+0x73/0x250 mm/kasan/report.c:256 kasan_report_error mm/kasan/report.c:354 [inline] kasan_report+0x23c/0x360 mm/kasan/report.c:412 __asan_report_load4_noabort+0x14/0x20 mm/kasan/report.c:432 snd_pcm_timer_resolution+0xa3/0xc0 sound/core/pcm_timer.c:72 snd_timer_resolution sound/core/timer.c:439 [inline] snd_timer_notify1+0x10f/0x400 sound/core/timer.c:462 snd_timer_start1+0x586/0x8f0 sound/core/timer.c:522 snd_timer_start+0x5d/0xa0 sound/core/timer.c:634 snd_timer_user_start.isra.12+0xde/0x130 sound/core/timer.c:1878 snd_timer_user_continue sound/core/timer.c:1902 [inline] __snd_timer_user_ioctl+0x1771/0x2c10 sound/core/timer.c:1977 snd_timer_user_ioctl+0x5f/0x7d sound/core/timer.c:1992 vfs_ioctl fs/ioctl.c:46 [inline] do_vfs_ioctl+0x1b1/0x1520 fs/ioctl.c:686 SYSC_ioctl fs/ioctl.c:701 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:692 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x4453c9 RSP: 002b:00007ffd46abffe8 EFLAGS: 00000217 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 00000000004453c9 RDX: 00000000004453c9 RSI: 00000000000054a2 RDI: 0000000000000003 RBP: 0000000000000000 R08: 0000000046ac0168 R09: 0000000046ac0168 R10: 0000000046ac0168 R11: 0000000000000217 R12: 00000000000142ea R13: 0000000000402c90 R14: 00000000006d1440 R15: 0000000000000000 Allocated by task 13234: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:552 kmem_cache_alloc_trace+0x136/0x740 mm/slab.c:3608 kmalloc include/linux/slab.h:512 [inline] kzalloc include/linux/slab.h:701 [inline] snd_pcm_attach_substream+0x3c5/0xb60 sound/core/pcm.c:1006 snd_pcm_open_substream+0x95/0x3b0 sound/core/pcm_native.c:2372 snd_pcm_oss_open_file sound/core/oss/pcm_oss.c:2309 [inline] snd_pcm_oss_open+0x7cf/0x1540 sound/core/oss/pcm_oss.c:2391 soundcore_open+0x321/0x630 sound/sound_core.c:597 chrdev_open+0x257/0x730 fs/char_dev.c:417 do_dentry_open+0x667/0xd40 fs/open.c:752 vfs_open+0x107/0x220 fs/open.c:866 do_last fs/namei.c:3379 [inline] path_openat+0x1151/0x3530 fs/namei.c:3519 do_filp_open+0x25b/0x3b0 fs/namei.c:3554 do_sys_open+0x502/0x6d0 fs/open.c:1059 SYSC_openat fs/open.c:1086 [inline] SyS_openat+0x30/0x40 fs/open.c:1080 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 Freed by task 13234: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] __kasan_slab_free+0x11a/0x170 mm/kasan/kasan.c:520 kasan_slab_free+0xe/0x10 mm/kasan/kasan.c:527 __cache_free mm/slab.c:3486 [inline] kfree+0xd9/0x260 mm/slab.c:3801 snd_pcm_detach_substream+0x134/0x290 sound/core/pcm.c:1057 snd_pcm_release_substream.part.44+0x23d/0x300 sound/core/pcm_native.c:2361 snd_pcm_release_substream+0x59/0x70 sound/core/pcm_native.c:2362 snd_pcm_oss_release_file.part.18+0x71/0xa0 sound/core/oss/pcm_oss.c:2268 snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2263 [inline] snd_pcm_oss_release+0x104/0x280 sound/core/oss/pcm_oss.c:2448 __fput+0x327/0x7e0 fs/file_table.c:209 ____fput+0x15/0x20 fs/file_table.c:243 task_work_run+0x199/0x270 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x9bb/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 SYSC_exit_group kernel/exit.c:979 [inline] SyS_exit_group+0x1d/0x20 kernel/exit.c:977 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 The buggy address belongs to the object at ffff8801bdd58180 which belongs to the cache kmalloc-2048 of size 2048 The buggy address is located 784 bytes inside of 2048-byte region [ffff8801bdd58180, ffff8801bdd58980) The buggy address belongs to the page: page:ffffea0006f75600 count:1 mapcount:0 mapping:ffff8801bdd58180 index:0x0 compound_mapcount: 0 flags: 0x2fffc0000008100(slab|head) raw: 02fffc0000008100 ffff8801bdd58180 0000000000000000 0000000100000003 raw: ffffea0007330d20 ffffea0007268aa0 ffff8801dac00c40 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8801bdd58380: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8801bdd58400: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb > ffff8801bdd58480: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8801bdd58500: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8801bdd58580: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] [PATCH 00/13] soundwire: Add stream support
by Vinod Koul 05 Apr '18

05 Apr '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Sanyog Kale (5): soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (6): soundwire: Add support for port management soundwire: Add stream configuration APIs soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 ++ Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 +++ Documentation/driver-api/soundwire/stream.rst | 369 +++++++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 3 +- drivers/soundwire/bus.c | 44 + drivers/soundwire/bus.h | 81 ++ drivers/soundwire/cadence_master.c | 449 +++++++++- drivers/soundwire/cadence_master.h | 151 ++++ drivers/soundwire/intel.c | 589 ++++++++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/runtime.c | 907 +++++++++++++++++++++ drivers/soundwire/stream.c | 678 +++++++++++++++ include/linux/soundwire/sdw.h | 337 +++++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 5 + sound/soc/soc-core.c | 20 + 19 files changed, 3816 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/runtime.c create mode 100644 drivers/soundwire/stream.c -- 2.7.4

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[alsa-devel] [PATCHv5 0/5] Motorola Droid 4 Audio Support
by Sebastian Reichel 04 Apr '18

04 Apr '18

Hi, This adds audio support to Motorola Droid 4. I dropped the regulator from the DT binding as requested by Mark. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (this needs something more sophisticated than the audio-graph-card driver. I will have a look once the basic support has been merged) Changes since PATCHv4: * Replace verbose GPL text with SPDX header * Use "GPL v2" as MODULE_LICENSE * Drop "default MFD_CPCAP" for the codec driver * Use SOC_SINGLE instead of SOC_ENUM for Phase invert switches * Find DT node by node name instead of by compatible property - Drop Acked-by from Rob Herring * Add new patch for MFD driver to implement the mentioned DT change Changes since PATCHv3: * Drop regulator from DT binding, request VAUDIO from global regulator namespace instead Changes since PATCHv2: * Fix a whitespace issue * Fix const notes Takashi provided * Fix a DAPM route issue I accidently introduced in v2 Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (5): dt-bindings: mfd: motorola-cpcap: document audio-codec ASoC: codec: cpcap: new codec mfd: motorola-cpcap: Add audio-codec support ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: omap4-droid4: add soundcard .../devicetree/bindings/mfd/motorola-cpcap.txt | 42 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 13 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 78 + drivers/mfd/motorola-cpcap.c | 51 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1568 ++++++++++++++++++++ 7 files changed, 1757 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/cpcap.c -- 2.16.1

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[alsa-devel] [PATCH 1/3 v3] ASoC: TSCS42xx: Shorten lines and other cleanup
by Steven Eckhoff 03 Apr '18

03 Apr '18

Shorten lines greater than 80 chars Add const to struct snd_soc_component_driver Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 87 ++++++++++++++++++++++++++++----------------- 1 file changed, 55 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index cfc71b5411ee..37636d6efe60 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -204,7 +204,8 @@ static int power_up_audio_plls(struct snd_soc_component *component) break; default: ret = -EINVAL; - dev_err(component->dev, "Unrecognized PLL output freq (%d)\n", ret); + dev_err(component->dev, + "Unrecognized PLL output freq (%d)\n", ret); return ret; } @@ -261,7 +262,8 @@ static int power_down_audio_plls(struct snd_soc_component *component) static int coeff_ram_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -280,7 +282,8 @@ static int coeff_ram_get(struct snd_kcontrol *kcontrol, static int coeff_ram_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); struct coeff_ram_ctl *ctl = (struct coeff_ram_ctl *)kcontrol->private_value; @@ -363,7 +366,8 @@ static int dapm_micb_event(struct snd_soc_dapm_widget *w, static int pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); int ret; if (SND_SOC_DAPM_EVENT_ON(event)) @@ -377,7 +381,8 @@ static int pll_event(struct snd_soc_dapm_widget *w, static int dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); int ret; @@ -819,16 +824,19 @@ static int setup_sample_format(struct snd_soc_component *component, dev_err(component->dev, "Unsupported format width (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_WL, width); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_WL, width); if (ret < 0) { - dev_err(component->dev, "Failed to set sample width (%d)\n", ret); + dev_err(component->dev, + "Failed to set sample width (%d)\n", ret); return ret; } return 0; } -static int setup_sample_rate(struct snd_soc_component *component, unsigned int rate) +static int setup_sample_rate(struct snd_soc_component *component, + unsigned int rate) { struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); unsigned int br, bm; @@ -881,24 +889,32 @@ static int setup_sample_rate(struct snd_soc_component *component, unsigned int r } /* DAC and ADC share bit and frame clock */ - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBR, br); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBR, br); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_DACSR_DBM, bm); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_DACSR_DBM, bm); if (ret < 0) { - dev_err(component->dev, "Failed to update register (%d)\n", ret); + dev_err(component->dev, + "Failed to update register (%d)\n", ret); return ret; } @@ -1076,7 +1092,8 @@ static int tscs42xx_hw_params(struct snd_pcm_substream *substream, ret = setup_sample_rate(component, params_rate(params)); if (ret < 0) { - dev_err(component->dev, "Failed to setup sample rate (%d)\n", ret); + dev_err(component->dev, + "Failed to setup sample rate (%d)\n", ret); return ret; } @@ -1087,7 +1104,8 @@ static inline int dac_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute DAC (%d)\n", @@ -1102,7 +1120,8 @@ static inline int dac_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR1, RM_CNVRTR1_DACMU, + ret = snd_soc_component_update_bits(component, + R_CNVRTR1, RM_CNVRTR1_DACMU, RV_CNVRTR1_DACMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute DAC (%d)\n", @@ -1117,8 +1136,8 @@ static inline int adc_mute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_ENABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_ENABLE); if (ret < 0) { dev_err(component->dev, "Failed to mute ADC (%d)\n", ret); @@ -1132,8 +1151,8 @@ static inline int adc_unmute(struct snd_soc_component *component) { int ret; - ret = snd_soc_component_update_bits(component, R_CNVRTR0, RM_CNVRTR0_ADCMU, - RV_CNVRTR0_ADCMU_DISABLE); + ret = snd_soc_component_update_bits(component, + R_CNVRTR0, RM_CNVRTR0_ADCMU, RV_CNVRTR0_ADCMU_DISABLE); if (ret < 0) { dev_err(component->dev, "Failed to unmute ADC (%d)\n", ret); @@ -1171,8 +1190,8 @@ static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, /* Slave mode not supported since it needs always-on frame clock */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_MS, - RV_AIC1_MS_MASTER); + ret = snd_soc_component_update_bits(component, + R_AIC1, RM_AIC1_MS, RV_AIC1_MS_MASTER); if (ret < 0) { dev_err(component->dev, "Failed to set codec DAI master (%d)\n", ret); @@ -1211,14 +1230,18 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_component_update_bits(component, R_DACSR, RM_DACSR_DBCM, value); + ret = snd_soc_component_update_bits(component, + R_DACSR, RM_DACSR_DBCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set DAC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set DAC BCLK ratio (%d)\n", ret); return ret; } - ret = snd_soc_component_update_bits(component, R_ADCSR, RM_ADCSR_ABCM, value); + ret = snd_soc_component_update_bits(component, + R_ADCSR, RM_ADCSR_ABCM, value); if (ret < 0) { - dev_err(component->dev, "Failed to set ADC BCLK ratio (%d)\n", ret); + dev_err(component->dev, + "Failed to set ADC BCLK ratio (%d)\n", ret); return ret; } @@ -1309,7 +1332,7 @@ static int part_is_valid(struct tscs42xx *tscs42xx) }; } -static struct snd_soc_component_driver soc_codec_dev_tscs42xx = { +static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1416,8 +1439,8 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, mutex_init(&tscs42xx->coeff_ram_lock); mutex_init(&tscs42xx->pll_lock); - ret = devm_snd_soc_register_component(tscs42xx->dev, &soc_codec_dev_tscs42xx, - &tscs42xx_dai, 1); + ret = devm_snd_soc_register_component(tscs42xx->dev, + &soc_codec_dev_tscs42xx, &tscs42xx_dai, 1); if (ret) { dev_err(tscs42xx->dev, "Failed to register codec (%d)\n", ret); return ret; -- 2.15.1

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