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[alsa-devel] Driver-alsa for Icon Producer Works-192
by Андрей Король 20 Oct '17

20 Oct '17

Hello ! Is it possible to support the sound card Icon Producer Works-192 in the alsa? More about the map: VIA ICEnsemble ICE1724HT (Envy24HT/PT) http://soundcard.hw-specs.com/ICON_Producer_192/ https://nson.org/komplektuyushhie/zvukovye-karty/icon-producer-works-192x.h…

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Re: [alsa-devel] Add support for ALC236
by Takashi Iwai 20 Oct '17

20 Oct '17

On Fri, 20 Oct 2017 09:13:44 +0200, Kailang wrote: > > Hi Takashi, > > Attach patch was support ALC236. Applied, thanks. Takashi

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[alsa-devel] [PATCH] ASoC: rt5616: fix 0x91 default value
by Bard Liao 20 Oct '17

20 Oct '17

The default value of register 0x91 is 0x0c00 instead of 0x0000. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5616.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index c94e94f..0e5f54a 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -98,7 +98,7 @@ static const struct reg_default rt5616_reg[] = { { 0x8e, 0x0004 }, { 0x8f, 0x1100 }, { 0x90, 0x0000 }, - { 0x91, 0x0000 }, + { 0x91, 0x0c00 }, { 0x92, 0x0000 }, { 0x93, 0x2000 }, { 0x94, 0x0200 }, -- 2.7.4

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[alsa-devel] [PATCH] tinycompress: clarify dual-license wording
by Pierre-Louis Bossart 20 Oct '17

20 Oct '17

There were reports of confusion in legal circles introduced by the wording "tinycompress is provided under LGPL and BSD dual license" in the README. Clarify with the addition of the wording traditionally used for dual licenses: "This file is provided under a dual BSD/LGPLv2.1 license. When using or redistributing this file, you may do so under either license" Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- COPYING | 3 +++ README | 3 ++- include/tinycompress/tinycompress.h | 3 +++ include/tinycompress/tinymp3.h | 3 +++ include/tinycompress/version.h | 3 +++ src/lib/compress.c | 3 +++ src/utils/cplay.c | 3 +++ src/utils/crecord.c | 3 +++ 8 files changed, 23 insertions(+), 1 deletion(-) diff --git a/COPYING b/COPYING index f963926..dfc7426 100644 --- a/COPYING +++ b/COPYING @@ -1,3 +1,6 @@ +tinycompress is provided under a dual BSD/LGPLv2.1 license. When using or + redistributing tinycompress, you may do so under either license. + BSD LICENSE tinycompress library for compress audio offload in alsa diff --git a/README b/README index 73c19a6..689ca75 100644 --- a/README +++ b/README @@ -22,7 +22,8 @@ Http: http://git.alsa-project.org/?p=tinycompress.git alsa-devel(a)alsa-project.org. 4. LICENSE - tinycompress is provided under LGPL and BSD dual license + tinycompress is provided under a dual BSD/LGPLv2.1 license. When using or + redistributing tinycompress, you may do so under either license. 5. CREDITS - Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> for library design diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 68626a4..780dce6 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * Copyright (c) 2011-2012, Intel Corporation diff --git a/include/tinycompress/tinymp3.h b/include/tinycompress/tinymp3.h index a709c39..b0d429d 100644 --- a/include/tinycompress/tinymp3.h +++ b/include/tinycompress/tinymp3.h @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * mp3 header and prasing diff --git a/include/tinycompress/version.h b/include/tinycompress/version.h index 000f6a2..82a3e6c 100644 --- a/include/tinycompress/version.h +++ b/include/tinycompress/version.h @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * Copyright (c) 2011-2012, Intel Corporation diff --git a/src/lib/compress.c b/src/lib/compress.c index 84738d2..8ae6bbd 100644 --- a/src/lib/compress.c +++ b/src/lib/compress.c @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * tinycompress library for compress audio offload in alsa diff --git a/src/utils/cplay.c b/src/utils/cplay.c index 032a971..e0603fd 100644 --- a/src/utils/cplay.c +++ b/src/utils/cplay.c @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * tinyplay command line player for compress audio offload in alsa diff --git a/src/utils/crecord.c b/src/utils/crecord.c index 1fd3698..e051c2d 100644 --- a/src/utils/crecord.c +++ b/src/utils/crecord.c @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * crecord command line recorder for compress audio record in alsa -- 2.11.0

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[alsa-devel] [PATCH 0/2] rt5651: Enable jack detection
by Carlo Caione 20 Oct '17

20 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> Small set of patches to enable jack detection for the rt5651 codec. Please note that: - The jack detection is restricted to the JD1_1 pin only (the only hardware I have uses this pin for jack detection) - AFAICT (reading the rt5651 datasheet) there is no way to detect runtime the type of the jack inserted as done for the others rt56* codecs, so in the driver I always assume that we are dealing with an headset. Carlo Caione (2): ASoC: rt5651: Enable jack detection on JD1_1 ASoC: intel: byt: Add headset jack include/sound/rt5651.h | 7 +++ sound/soc/codecs/rt5651.c | 91 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5651.h | 3 ++ sound/soc/intel/boards/bytcr_rt5651.c | 24 +++++++++ 4 files changed, 123 insertions(+), 2 deletions(-) -- 2.14.2

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[alsa-devel] [PATCH] ALSA: hda: Avoid racy recreation of widget kobjects
by Takashi Iwai 19 Oct '17

19 Oct '17

The refresh of HD-audio widget sysfs kobjects via snd_hdac_refresh_widget_sysfs() is slightly racy. The driver recreates the whole tree from scratch after deleting the whole. When CONFIG_DEBUG_KOBJECT_RELEASE option is used, kobject release doesn't happen immediately but delayed, while the re-creation of the same named kobject happens soon after invoking kobject_put(). This may end up with the conflicts of duplicated kobjects, as found in the bug report below. In this patch, we take another approach to refresh the tree: instead of recreating the whole tree, just add the new nodes and delete the non-existing nodes. Since the refresh happens only once at initialization, no longer race would happen. Along with the code change, merge snd_hdac_refresh_widget_sysfs() with the existing snd_hdac_refresh_widgets() with an additional bool flag for simplifying the code. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=197307 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 3 +-- sound/hda/hdac_device.c | 43 ++++++++++--------------------------------- sound/hda/hdac_sysfs.c | 47 +++++++++++++++++++++++++++++++++++++++++++++++ sound/hda/local.h | 2 ++ sound/pci/hda/hda_codec.c | 2 +- 5 files changed, 61 insertions(+), 36 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 96546b30e900..6f1118545bb8 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -111,8 +111,7 @@ void snd_hdac_device_unregister(struct hdac_device *codec); int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size); -int snd_hdac_refresh_widgets(struct hdac_device *codec); -int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec); +int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs); unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, unsigned int verb, unsigned int parm); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 19deb306facb..06f845e293cb 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -87,7 +87,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, fg = codec->afg ? codec->afg : codec->mfg; - err = snd_hdac_refresh_widgets(codec); + err = snd_hdac_refresh_widgets(codec, false); if (err < 0) goto error; @@ -388,11 +388,12 @@ static void setup_fg_nodes(struct hdac_device *codec) /** * snd_hdac_refresh_widgets - Reset the widget start/end nodes * @codec: the codec object + * @sysfs: re-initialize sysfs tree, too */ -int snd_hdac_refresh_widgets(struct hdac_device *codec) +int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs) { hda_nid_t start_nid; - int nums; + int nums, err; nums = snd_hdac_get_sub_nodes(codec, codec->afg, &start_nid); if (!start_nid || nums <= 0 || nums >= 0xff) { @@ -401,6 +402,12 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec) return -EINVAL; } + if (sysfs) { + err = hda_widget_sysfs_reinit(codec, start_nid, nums); + if (err < 0) + return err; + } + codec->num_nodes = nums; codec->start_nid = start_nid; codec->end_nid = start_nid + nums; @@ -408,36 +415,6 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_refresh_widgets); -/** - * snd_hdac_refresh_widget_sysfs - Reset the codec widgets and reinit the - * codec sysfs - * @codec: the codec object - * - * first we need to remove sysfs, then refresh widgets and lastly - * recreate it - */ -int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec) -{ - int ret; - - if (device_is_registered(&codec->dev)) - hda_widget_sysfs_exit(codec); - ret = snd_hdac_refresh_widgets(codec); - if (ret) { - dev_err(&codec->dev, "failed to refresh widget: %d\n", ret); - return ret; - } - if (device_is_registered(&codec->dev)) { - ret = hda_widget_sysfs_init(codec); - if (ret) { - dev_err(&codec->dev, "failed to init sysfs: %d\n", ret); - return ret; - } - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_hdac_refresh_widget_sysfs); - /* return CONNLIST_LEN parameter of the given widget */ static unsigned int get_num_conns(struct hdac_device *codec, hda_nid_t nid) { diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 42d61bf41969..e123ba6b2e81 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -414,3 +414,50 @@ void hda_widget_sysfs_exit(struct hdac_device *codec) { widget_tree_free(codec); } + +int hda_widget_sysfs_reinit(struct hdac_device *codec, + hda_nid_t start_nid, int num_nodes) +{ + struct hdac_widget_tree *tree; + hda_nid_t end_nid = start_nid + num_nodes; + hda_nid_t nid; + int i; + + if (!codec->widgets) + return hda_widget_sysfs_init(codec); + + tree = kmemdup(codec->widgets, sizeof(*tree), GFP_KERNEL); + if (!tree) + return -ENOMEM; + + tree->nodes = kcalloc(num_nodes + 1, sizeof(*tree->nodes), GFP_KERNEL); + if (!tree->nodes) { + kfree(tree); + return -ENOMEM; + } + + /* prune non-existing nodes */ + for (i = 0, nid = codec->start_nid; i < codec->num_nodes; i++, nid++) { + if (nid < start_nid || nid >= end_nid) + free_widget_node(codec->widgets->nodes[i], + &widget_node_group); + } + + /* add new nodes */ + for (i = 0, nid = start_nid; i < num_nodes; i++, nid++) { + if (nid < codec->start_nid || nid >= codec->end_nid) + add_widget_node(tree->root, nid, &widget_node_group, + &tree->nodes[i]); + else + tree->nodes[i] = + codec->widgets->nodes[nid - codec->start_nid]; + } + + /* replace with the new tree */ + kfree(codec->widgets->nodes); + kfree(codec->widgets); + codec->widgets = tree; + + kobject_uevent(tree->root, KOBJ_CHANGE); + return 0; +} diff --git a/sound/hda/local.h b/sound/hda/local.h index 0d5bb159d538..c586ea62d49e 100644 --- a/sound/hda/local.h +++ b/sound/hda/local.h @@ -28,6 +28,8 @@ static inline unsigned int get_wcaps_channels(u32 wcaps) extern const struct attribute_group *hdac_dev_attr_groups[]; int hda_widget_sysfs_init(struct hdac_device *codec); +int hda_widget_sysfs_reinit(struct hdac_device *codec, hda_nid_t start_nid, + int num_nodes); void hda_widget_sysfs_exit(struct hdac_device *codec); #endif /* __HDAC_LOCAL_H */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3db26c451837..bb086665c3e0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -977,7 +977,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) hda_nid_t fg; int err; - err = snd_hdac_refresh_widget_sysfs(&codec->core); + err = snd_hdac_refresh_widgets(&codec->core, true); if (err < 0) return err; -- 2.14.2

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[alsa-devel] [PATCH v2] ALSA: hda: Abort capability probe on invalid capability
by Ughreja, Rakesh A 19 Oct '17

19 Oct '17

From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Due to bugs in BIOS it's possible that the HDA capability link list is not constructed properly. This may lead to driver going into unknown state. So whenever driver discovers unknown HDA capability, log it as error and stop traversing the link list further. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc18..8f7d0d9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -314,7 +314,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) break; default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + dev_err(bus->dev, "Unknown capability %d\n", cur_cap); + cur_cap = 0; break; } -- 2.7.4

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[alsa-devel] [BUG] aplay doesn't honor `-t raw` when WAVE header is detected on input
by Nutchanon Wetchasit 19 Oct '17

19 Oct '17

Hello, I ran into a problem on how alsa-utils's aplay command handles `-t` option when I was trying to play a PCM WAVE file in loop, using following command: ( while true; do cat audio.wav; done ) | aplay -t raw -f S16_LE -r 44100 -c 2 An "audio.wav" can be any 44100 Hz 16-bit stereo PCM WAVE file, and `-t raw` was specially specified to make aplay reject length information in the header. However, once the command is run, first line of the output would read: Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Which shows that aplay still read the input as WAVE format, despite the `-t raw` option being explicitly specified. And you would notice that the audio *does not loop*. Instead, it will play just once, followed by silence. However, if WAVE header was stripped out (or corrupted), aplay would now treat the input as raw audio stream, and play a continuously repeated sound as expected: ( while true; do dd if=audio.wav of=/dev/stdout ibs=44 skip=1; done ) | \ aplay -t raw -f S16_LE -r 44100 -c 2 The first line of output would also show the stream as "raw data" too: Playing raw data 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo This problem exists on the latest alsa-utils 1.1.4, as well as the older 1.0.25 which came with my GNU/Linux distribution; though I'm not certain whether this is also applicable to other file format (like Sun .AU) or not. Please investigate. Regards, Nutchanon Wetchasit alsa-utils: 1.1.4 (source distribution) libasound: 1.1.4 (source distribution) System: Debian GNU/Linux 7.0 "Wheezy" i386 P.S. The "audio.wav" I used in my testing is a siren sound generated by Sox: sox -b 16 -c 2 -e signed-integer -r 44100 -n audio.wav \ synth 0.5 sine 800 gain -6 : synth 0.5 sine 600 gain -6

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[alsa-devel] [PATCH v2 0/2] ALSA: hda - Fix incorrect TLV callback check
by Takashi Iwai 19 Oct '17

19 Oct '17

Hi, this is a revised patchset for addressing the incorrect TLV callback check introduced by a set_fs() removal commit. A fix for a superfluous '-' in the code is split as another fix so that it can be applied to older kernels without the rest. Other than that, only the changelog text updates since the older patch (that was embedded in the mail thread "4.13 regression: get_kctl_0dB_offset doesn't handle all possible callbacks"). thanks, Takashi === Takashi Iwai (2): ALSA: hda: Remove superfluous '-' added by printk conversion ALSA: hda - Fix incorrect TLV callback check introduced during set_fs() removal include/sound/control.h | 3 ++ sound/core/vmaster.c | 31 +++++++++++++++ sound/pci/hda/hda_codec.c | 97 +++++++++++++++++++++++++++-------------------- 3 files changed, 89 insertions(+), 42 deletions(-) -- 2.14.2

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[alsa-devel] [PATCH] ALSA: hda: Abort capability probe on invalid capability
by Ughreja, Rakesh A 18 Oct '17

18 Oct '17

From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> When an invalid capability is discovered, stop traversing the capability link list further. print the invalid capability error and set it to zero Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc18..8f7d0d9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -314,7 +314,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) break; default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + dev_err(bus->dev, "Unknown capability %d\n", cur_cap); + cur_cap = 0; break; } -- 2.7.4

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[alsa-devel] [PATCH 0/2] rt5651: Enable platforms with int mic on IN2
by Carlo Caione 18 Oct '17

18 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> While working on enabling a cherry-trail laptop shipping the rt5651 codec I realized that the machine driver needed some fixup. In particular the laptop I'm working on (KIANO SlimNote 14.2) has the internal mic connected to the IN2 port. All the laptop-specific work with the related quirks will follow. Carlo Caione (2): ASoC: intel: byt: Enable quirk logic for custom maps ASoC: intel: byt: Introduce new custom IN2 map sound/soc/intel/boards/bytcr_rt5651.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) -- 2.14.2

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Re: [alsa-devel] linux-next: manual merge of the sound-asoc tree with the drm-misc tree
by Mark Brown 18 Oct '17

18 Oct '17

On Wed, Oct 18, 2017 at 10:57:33AM +0100, Mark Brown wrote: > I fixed it up (see below) and can carry the fix as necessary. This > is now fixed as far as linux-next is concerned, but any non trivial > conflicts should be mentioned to your upstream maintainer when your tree > is submitted for merging. You may also want to consider cooperating > with the maintainer of the conflicting tree to minimise any particularly > complex conflicts. Actually I'm just going to discard the AMD drivers from the ASoC tree because the build produces reams of errors like those below, the changes to move the chip type definitions around weren't fully baked. Please resend both the pull request and the patches with this fixed. Note also that if you're basing something on Linus' tree you should use a tagged release rather than just a random commit. In file included from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/amdgpu.h:51:0, from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/amdgpu_connectors.c:31: /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/../include/amd_shared.h:33:6: error: nested redefinition of 'enum amd_asic_type' enum amd_asic_type { ^ /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/../include/amd_shared.h:33:6: error: redeclaration of 'enum amd_asic_type' In file included from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/../include/amd_shared.h:26:0, from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/amdgpu.h:51, from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/amdgpu_connectors.c:31: /home/broonie/tmpfs/next/include/drm/amd_asic_type.h:28:6: note: originally defined here enum amd_asic_type { ^

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[alsa-devel] [PATCH 0/4 resend] Add ASoC support for AMD Stoney APUs
by Alex Deucher 18 Oct '17

18 Oct '17

This patch set updates the AMD GPU and Audio CoProcessor (ACP) audio drivers and the designware i2s driver for Stoney (ST). ST is an APU similar to Carrizo (CZ) which already has ACP audio support. The i2s controller and ACP audio DMA engine are part of the GPU and both need updating so I would like to upstream the whole patch set via one tree if possible. I just sent a pull request for the pre-requisite patches which are available here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=linus-4.14-rc4-acp-prereq This is a resend of just the non pre-requisite audio patches at Mark's request. Thanks! Alex v2: - Patch 1 is already applied to the audio tree, just including it for completeness since it's required for this patch set and it's not yet in the drm tree. - New patch to share asic types between gpu and audio drivers - ACPI ID changed for rt5650 machine driver - Integrate feedback on other patches v3: - resend remaining patches that have not been applied yet v4: - Integrate feedback from v3 - Add Mark's acks from v3 - Integrate a machine driver fix from Dylan Reid Akshu Agrawal (1): ASoC: AMD: Add machine driver for cz rt5650 Vijendar Mukunda (3): ASoC: AMD: disabling memory gating in stoney platform ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Audio buffer related changes for Stoney sound/soc/amd/Kconfig | 7 ++ sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-pcm-dma.c | 249 +++++++++++++++++++++++++++++++++++--------- sound/soc/amd/acp-rt5645.c | 199 +++++++++++++++++++++++++++++++++++ sound/soc/amd/acp.h | 2 + 5 files changed, 408 insertions(+), 51 deletions(-) create mode 100644 sound/soc/amd/acp-rt5645.c -- 2.5.5

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[alsa-devel] [pull v2] amdgpu pre-reqs for stoney ACP audio support
by Alex Deucher 18 Oct '17

18 Oct '17

Hi Dave, Mark, This branch against Linus' 4.14-rc4 contains the 3 pre-requisite patches for Stoney ACP audio. Mark needs these patches to apply the rest of the Stoney ACP patch set on the audio side. I'll resend the stony audio patches shortly, but there are no differences from the last version I send out. Note that depending on when these changes end up in drm-next, there may be a conflict due to f674bd28146045 (drm/amdgpu Moving amdgpu asic types to a separate file). If you end up with a conflict, please take the changes in f674bd28146045, otherwise, you'll end up with the enum defined twice. Please pull. Thanks! The following changes since commit 8a5776a5f49812d29fe4b2d0a2d71675c3facf3f: Linux 4.14-rc4 (2017-10-08 20:53:29 -0700) are available in the git repository at: git://people.freedesktop.org/~agd5f/linux linus-4.14-rc4-acp-prereq for you to fetch changes up to f674bd281460451e8784f3bc917a650e8937a8ed: drm/amdgpu Moving amdgpu asic types to a separate file (2017-10-18 10:56:27 -0400) ---------------------------------------------------------------- Akshu Agrawal (1): drm/amdgpu Moving amdgpu asic types to a separate file Vijendar Mukunda (2): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: AMD: Added asic_type as ACP DMA driver platform data drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 ++ drivers/gpu/drm/amd/include/amd_shared.h | 29 ++---------------- include/drm/amd_asic_type.h | 52 ++++++++++++++++++++++++++++++++ sound/soc/amd/acp-pcm-dma.c | 8 ++--- sound/soc/amd/acp.h | 7 +++++ 5 files changed, 65 insertions(+), 33 deletions(-) create mode 100644 include/drm/amd_asic_type.h

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[alsa-devel] [PATCH] ALSA: hda - Add model string for Intel reference board quirk
by Takashi Iwai 18 Oct '17

18 Oct '17

For allowing user to apply the existing quirk on a machine with a different SSID, add a new model string entry, alc700-ref. The quirk itself was introduced in the commit b84e843644f2: "ALSA: hda/realtek - Enable jack detection function for Intel ALC700") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/hd-audio/models.rst | 2 ++ sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 3 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 773d2bfacc6c..1fee5a4f6660 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -82,6 +82,8 @@ tpt460 Lenovo Thinkpad T460/560 setup dual-codecs Lenovo laptops with dual codecs +alc700-ref + Intel reference board with ALC700 codec ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ce71111b4e3..1a0163b81f5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6357,6 +6357,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, + {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.14.2

1 0

[alsa-devel] [PATCH v2 0/9] Baytrail-Cherrytrail updates
by Pierre-Louis Bossart 18 Oct '17

18 Oct '17

As discussed on the alsa-devel mailing list, here is a set of patches to fix 1. previous feedback on bytcht_rt5651 - no new functionality beyond what was already reviewed 2. cosmetic issues (variables in reverse xmas tree order, etc) 3. unnecessary clock tests 4. addition of a helper to remove duplicate code across machine drivers The last two patches were not included in the first pass, they correct an off-by-one issue which has no functional impact and remove the hard-codec compressed dailinks which don't currently work with the mainline code+publicly available firmware. Pierre-Louis Bossart (9): ASoC: Intel: bytcr_rt5651: add MCLK, quirks and cleanups ASoC: Intel: bytcr_rt5640: cosmetic fixes ASoC: Intel: cht_bsw_rt5645: cosmetic fixes ASoC: Intel: bytcht_da7213: cosmetic fixes ASoC: Intel: bytcht_es8316: remove useless code ASoC: Add helper to find codec_dai from dai_name ASoC: Intel: boards: use helper to get codec_dai ASoC: Intel: boards: fix off-by-one dailink id ASoC: Intel: boards: remove hard-coded compressed dailinks include/sound/soc.h | 14 ++ sound/soc/intel/boards/bxt_da7219_max98357a.c | 16 +- sound/soc/intel/boards/bytcht_da7213.c | 17 +- sound/soc/intel/boards/bytcht_es8316.c | 25 +-- sound/soc/intel/boards/bytcht_nocodec.c | 10 +- sound/soc/intel/boards/bytcr_rt5640.c | 105 ++++------- sound/soc/intel/boards/bytcr_rt5651.c | 236 +++++++++++++++++++++--- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 24 +-- sound/soc/intel/boards/cht_bsw_rt5645.c | 103 ++++------- sound/soc/intel/boards/cht_bsw_rt5672.c | 24 +-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 16 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +- 12 files changed, 318 insertions(+), 288 deletions(-) -- 2.11.0

4 17

[alsa-devel] [PATCH 0/6] Baytrail-Cherrytrail updates
by Pierre-Louis Bossart 18 Oct '17

18 Oct '17

As discussed on the alsa-devel mailing list, here is a set of patches to fix 1. previous feedback on bytcht_rt5651 - no new functionality beyond what was already reviewed 2. cosmetic issues (variables in reverse xmas tree order, etc) 3. unnecessary clock tests Since the changes are very small, I grouped them by machine driver instead of generating more patches. let me know if a different split is needed. The last patch addresses Andy's comment that we should not use strncmp(a,b,strlen(b)) when strcmp(a,b) is fine. This change is applied across all machine drivers. Pierre-Louis Bossart (6): ASoC: Intel: bytcr_rt5651: add MCLK support and fix quirks ASoC: Intel: bytcr_rt5651: filter codec name and fix variable declarations ASoC: Intel: bytcr_rt5640: cosmetic fixes ASoC: Intel: cht_bsw_rt5645: cosmetic fixes ASoC: Intel: bytcht_da7213: cosmetic fixes ASoC: Intel: boards: use strcmp with constant literals sound/soc/intel/boards/bxt_da7219_max98357a.c | 3 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 3 +- sound/soc/intel/boards/bytcr_rt5640.c | 79 ++++---- sound/soc/intel/boards/bytcr_rt5651.c | 257 ++++++++++++++++++++++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 79 ++++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 3 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 3 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 3 +- 10 files changed, 324 insertions(+), 115 deletions(-) -- 2.11.0

5 15

[alsa-devel] Applied "ASoC: Intel: boards: remove hard-coded compressed dailinks" to the asoc tree
by Mark Brown 18 Oct '17

18 Oct '17

The patch ASoC: Intel: boards: remove hard-coded compressed dailinks has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2193eb9631d272e5a8a7e7534830f555305aa556 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 12 Oct 2017 18:38:05 -0500 Subject: [PATCH] ASoC: Intel: boards: remove hard-coded compressed dailinks The hard-coded compressed dailinks are not supported using publicly-available firmwares, which creates unnecessary user confusion [1]. Even if the firmware was available, the mainline code does not have the required .dynamic=1 and .dpcm_playback=1 fields so probably never worked as is, and last and they conflict with topology-defined streams. Remove them and move on. This can be re-enabled with SOF later in a more flexible manner. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124868.html Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcht_da7213.c | 9 --------- sound/soc/intel/boards/bytcht_es8316.c | 9 --------- sound/soc/intel/boards/bytcht_nocodec.c | 8 -------- sound/soc/intel/boards/bytcr_rt5640.c | 11 +---------- sound/soc/intel/boards/bytcr_rt5651.c | 11 +---------- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 8 -------- sound/soc/intel/boards/cht_bsw_rt5645.c | 8 -------- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 -------- 8 files changed, 2 insertions(+), 70 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 28a493e5318d..296b50640f46 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -185,14 +185,6 @@ static struct snd_soc_dai_link dailink[] = { .dpcm_playback = 1, .ops = &aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* CODEC<->CODEC link */ /* back ends */ { @@ -243,7 +235,6 @@ static int bytcht_da7213_probe(struct platform_device *pdev) card->dev = &pdev->dev; /* fix index of codec dai */ - dai_index = MERR_DPCM_COMPR + 1; for (i = 0; i < ARRAY_SIZE(dailink); i++) { if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 808e825c3b05..6acacc0aeedb 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -194,15 +194,6 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { .ops = &byt_cht_es8316_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, - /* back ends */ { /* Only SSP2 has been tested here, so BYT-CR platforms that diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c index 511e87601104..b80ec027a0e8 100644 --- a/sound/soc/intel/boards/bytcht_nocodec.c +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -133,14 +133,6 @@ static struct snd_soc_dai_link dais[] = { .dpcm_playback = 1, .ops = &aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* CODEC<->CODEC link */ /* back ends */ { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6c7b71d7fed7..3eeb02036aa1 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -678,14 +678,6 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .dynamic = 1, .dpcm_playback = 1, .ops = &byt_rt5640_aif1_ops, - }, - [MERR_DPCM_COMPR] = { - .name = "Baytrail Compressed Port", - .stream_name = "Baytrail Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", }, /* back ends */ { @@ -748,7 +740,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) struct sst_acpi_mach *mach; const char *i2c_name = NULL; int ret_val = 0; - int dai_index; + int dai_index = 0; int i; is_bytcr = false; @@ -762,7 +754,6 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&byt_rt5640_card, priv); /* fix index of codec dai */ - dai_index = MERR_DPCM_COMPR + 1; for (i = 0; i < ARRAY_SIZE(byt_rt5640_dais); i++) { if (!strcmp(byt_rt5640_dais[i].codec_name, "i2c-10EC5640:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index e0caee26fb0a..3076bfc0db5e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -389,14 +389,6 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { .dpcm_playback = 1, .ops = &byt_rt5651_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* CODEC<->CODEC link */ /* back ends */ { @@ -440,7 +432,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) struct sst_acpi_mach *mach; const char *i2c_name = NULL; int ret_val = 0; - int dai_index; + int dai_index = 0; int i; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); @@ -454,7 +446,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&byt_rt5651_card, priv); /* fix index of codec dai */ - dai_index = MERR_DPCM_COMPR + 1; for (i = 0; i < ARRAY_SIZE(byt_rt5651_dais); i++) { if (!strcmp(byt_rt5651_dais[i].codec_name, "i2c-10EC5651:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index a2194d993615..14dd68edfce4 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -350,14 +350,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .dpcm_playback = 1, .ops = &cht_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* back ends */ { .name = "SSP2-Codec", diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 1901b3b8f4d2..83c923c39393 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -445,14 +445,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .dpcm_playback = 1, .ops = &cht_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* CODEC<->CODEC link */ /* back ends */ { diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 3464cf41c7d1..d135640702e9 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -302,14 +302,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .dpcm_playback = 1, .ops = &cht_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* Back End DAI links */ { -- 2.14.1

1 0

[alsa-devel] [PATCH] ASoC: wm97xx: fix compilation corner case
by Robert Jarzmik 18 Oct '17

18 Oct '17

When the old AC97 is not used, CONFIG_SND_SOC_AC97_BUS is not defined. As a consequence, in the error path, snd_soc_free_ac97_codec() is not defined and triggers a compilation error. Fix it for wm9705 and wm9712, as wm9713 is correctly written. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/codecs/wm9705.c | 8 ++------ sound/soc/codecs/wm9712.c | 7 ++----- 2 files changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 68c204e3599f..2c09f71fe433 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -321,7 +321,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { struct wm9705_priv *wm9705 = snd_soc_codec_get_drvdata(codec); struct regmap *regmap; - int ret; if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; @@ -337,8 +336,8 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) regmap = regmap_init_ac97(wm9705->ac97, &wm9705_regmap_config); if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - goto err_free_ac97_codec; + snd_soc_free_ac97_codec(wm9705->ac97); + return PTR_ERR(regmap); } #endif } @@ -347,9 +346,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) snd_soc_codec_init_regmap(codec, regmap); return 0; -err_free_ac97_codec: - snd_soc_free_ac97_codec(wm9705->ac97); - return ret; } static int wm9705_soc_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9db5c7783de5..4f6d1a442bc4 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -656,8 +656,8 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) regmap = regmap_init_ac97(wm9712->ac97, &wm9712_regmap_config); if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - goto err_free_ac97_codec; + snd_soc_free_ac97_codec(wm9712->ac97); + return PTR_ERR(regmap); } #endif } @@ -668,9 +668,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, AC97_VIDEO, 0x3000, 0x3000); return 0; -err_free_ac97_codec: - snd_soc_free_ac97_codec(wm9712->ac97); - return ret; } static int wm9712_soc_remove(struct snd_soc_codec *codec) -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: samsung: i2s: disable secondary DAI until it gets fixed
by Marek Szyprowski 18 Oct '17

18 Oct '17

Secondary DAI in Exynos I2S driver is not used by any of the currently supported boards and it causes problems due to some limitations in the ASoC code. Disable it until it gets proper support both by board-specific and ASoC core code. Also disable IDMA support, which relies on secondary DAI presence. This patch fixes following kernel warning: samsung-i2s 3830000.i2s: ASoC: Failed to create component debugfs directory samsung-i2s 3830000.i2s: ASoC: Failed to create component debugfs directory ------------[ cut here ]------------ WARNING: CPU: 3 PID: 82 at fs/proc/generic.c:330 proc_register+0xec/0x10c proc_dir_entry 'sub0/prealloc' already registered Modules linked in: CPU: 3 PID: 82 Comm: kworker/3:1 Not tainted 4.14.0-rc5-next-20171017 #3089 Hardware name: SAMSUNG EXYNOS (Flattened Device Tree) Workqueue: events deferred_probe_work_func [<c0110114>] (unwind_backtrace) from [<c010c900>] (show_stack+0x10/0x14) [<c010c900>] (show_stack) from [<c083e664>] (dump_stack+0x90/0xc8) [<c083e664>] (dump_stack) from [<c011d2b8>] (__warn+0xd4/0x100) [<c011d2b8>] (__warn) from [<c011d384>] (warn_slowpath_fmt+0x38/0x48) [<c011d384>] (warn_slowpath_fmt) from [<c0271268>] (proc_register+0xec/0x10c) [<c0271268>] (proc_register) from [<c027130c>] (proc_create_data+0x84/0xc8) [<c027130c>] (proc_create_data) from [<c061afbc>] (snd_info_register+0x64/0xcc) [<c061afbc>] (snd_info_register) from [<c062a6e0>] (snd_pcm_lib_preallocate_pages1+0x78/0x1a0) [<c062a6e0>] (snd_pcm_lib_preallocate_pages1) from [<c063eef4>] (dmaengine_pcm_new+0xa0/0x1ec) [<c063eef4>] (dmaengine_pcm_new) from [<c062b9f8>] (snd_soc_platform_drv_pcm_new+0x1c/0x28) [<c062b9f8>] (snd_soc_platform_drv_pcm_new) from [<c063d54c>] (soc_new_pcm+0x2f4/0x4f4) [<c063d54c>] (soc_new_pcm) from [<c063107c>] (snd_soc_register_card+0xc4c/0xdc4) [<c063107c>] (snd_soc_register_card) from [<c063db30>] (devm_snd_soc_register_card+0x34/0x70) [<c063db30>] (devm_snd_soc_register_card) from [<c064af60>] (asoc_simple_card_probe+0x230/0x47c) [<c064af60>] (asoc_simple_card_probe) from [<c047f8fc>] (platform_drv_probe+0x50/0xb0) [<c047f8fc>] (platform_drv_probe) from [<c047dee0>] (driver_probe_device+0x2a0/0x46c) [<c047dee0>] (driver_probe_device) from [<c047c0bc>] (bus_for_each_drv+0x44/0x8c) [<c047c0bc>] (bus_for_each_drv) from [<c047db50>] (__device_attach+0xa0/0x134) [<c047db50>] (__device_attach) from [<c047cf7c>] (bus_probe_device+0x88/0x90) [<c047cf7c>] (bus_probe_device) from [<c047d484>] (deferred_probe_work_func+0x3c/0x168) [<c047d484>] (deferred_probe_work_func) from [<c01371f8>] (process_one_work+0x188/0x41c) [<c01371f8>] (process_one_work) from [<c01374b4>] (process_scheduled_works+0x28/0x38) [<c01374b4>] (process_scheduled_works) from [<c01376d4>] (worker_thread+0x210/0x4dc) [<c01376d4>] (worker_thread) from [<c013d9cc>] (kthread+0x128/0x164) [<c013d9cc>] (kthread) from [<c0108848>] (ret_from_fork+0x14/0x2c) ---[ end trace bad8db6ee771d094 ]-- Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- sound/soc/samsung/i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 8d5e1861abb1..233f1c9a4b6c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1292,6 +1292,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } } } + quirks &= ~(QUIRK_SEC_DAI | QUIRK_SUPPORTS_IDMA); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); pri_dai->addr = devm_ioremap_resource(&pdev->dev, res); -- 2.14.2

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5659: register power bit of LOUT Amp
by Bard Liao 18 Oct '17

18 Oct '17

There is a power bit for LOUT Amp. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5659.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 71216db..999432e 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2744,7 +2744,8 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5659_hp_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_PGA("LOUT Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT Amp", 1, RT5659_PWR_ANLG_1, RT5659_PWR_LM_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Charge Pump", SND_SOC_NOPM, 0, 0, rt5659_charge_pump_event, SND_SOC_DAPM_PRE_PMU | -- 2.7.4

2 7

[alsa-devel] [PATCH] ALSA: hda: Abort capability probe on invalid capability
by Ughreja, Rakesh A 18 Oct '17

18 Oct '17

From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> When an invalid capability is discovered, stop traversing the capability link list further. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc18..4fa0cc0 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -314,7 +314,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) break; default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + cur_cap = 0; + dev_err(bus->dev, "Unknown capability %d\n", cur_cap); break; } -- 2.7.4

2 3

[alsa-devel] [PATCH] ALSA: hda: Abort capability probe at invalid register read
by Takashi Iwai 18 Oct '17

18 Oct '17

The loop in snd_hdac_bus_parse_capabilities() may go to nirvana when it hits an invalid register value read: BUG: unable to handle kernel paging request at ffffad5dc41f3fff IP: pci_azx_readl+0x5/0x10 [snd_hda_intel] Call Trace: snd_hdac_bus_parse_capabilities+0x3c/0x1f0 [snd_hda_core] azx_probe_continue+0x7d5/0x940 [snd_hda_intel] ..... This happened on a new Intel machine, and we need to check the value and abort the loop accordingly. [Note: the fixes tag below indicates only the commit where this patch can be applied; the original problem was introduced even before that commit] Fixes: 6720b38420a0 ("ALSA: hda - move bus_parse_capabilities to core") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_controller.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc1801b3a..f6d2985b2520 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -284,6 +284,11 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) dev_dbg(bus->dev, "HDA capability ID: 0x%x\n", (cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF); + if (cur_cap == -1) { + dev_dbg(bus->dev, "Invalid capability reg read\n"); + break; + } + switch ((cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF) { case AZX_ML_CAP_ID: dev_dbg(bus->dev, "Found ML capability\n"); -- 2.14.2

3 8

[alsa-devel] [PATCH v2 0/2] Add snd_card_disconnect_sync() helper
by Takashi Iwai 18 Oct '17

18 Oct '17

Hi, here is a refreshed series containing ALSA core changes Morimoto-san posted. Basically the code is same but it's split, and more comments and an error message are added. I tested lightly with the existing hot-pluggable device and no regression was seen. The commits are found in topic/card-disconnect branch of sound git tree, too. Mark, please pull it to yours when applying the further patches relying on this new helper function. thanks, Takashi === Takashi Iwai (2): ALSA: add snd_card_disconnect_sync() ALSA: pcm: Forcibly stop at disconnect callback include/sound/core.h | 2 ++ sound/core/init.c | 32 ++++++++++++++++++++++++++++++++ sound/core/pcm.c | 4 ++++ 3 files changed, 38 insertions(+) -- 2.14.2

4 22

[alsa-devel] [PATCH] ALSA: seq: Enable 'use' locking in all configurations
by Ben Hutchings 18 Oct '17

18 Oct '17

The 'use' locking macros are no-ops if neither SMP or SND_DEBUG is enabled. This might once have been OK in non-preemptible configurations, but even in that case snd_seq_read() may sleep while relying on a 'use' lock. So always use the proper implementations. Cc: stable(a)vger.kernel.org Signed-off-by: Ben Hutchings <ben.hutchings(a)codethink.co.uk> --- sound/core/seq/seq_lock.c | 4 ---- sound/core/seq/seq_lock.h | 12 ------------ 2 files changed, 16 deletions(-) diff --git a/sound/core/seq/seq_lock.c b/sound/core/seq/seq_lock.c index 0ff7926a5a69..cda64b489e42 100644 --- a/sound/core/seq/seq_lock.c +++ b/sound/core/seq/seq_lock.c @@ -23,8 +23,6 @@ #include <sound/core.h> #include "seq_lock.h" -#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG) - /* wait until all locks are released */ void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line) { @@ -41,5 +39,3 @@ void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line) } } EXPORT_SYMBOL(snd_use_lock_sync_helper); - -#endif diff --git a/sound/core/seq/seq_lock.h b/sound/core/seq/seq_lock.h index 54044bc2c9ef..ac38031c370e 100644 --- a/sound/core/seq/seq_lock.h +++ b/sound/core/seq/seq_lock.h @@ -3,8 +3,6 @@ #include <linux/sched.h> -#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG) - typedef atomic_t snd_use_lock_t; /* initialize lock */ @@ -20,14 +18,4 @@ typedef atomic_t snd_use_lock_t; void snd_use_lock_sync_helper(snd_use_lock_t *lock, const char *file, int line); #define snd_use_lock_sync(lockp) snd_use_lock_sync_helper(lockp, __BASE_FILE__, __LINE__) -#else /* SMP || CONFIG_SND_DEBUG */ - -typedef spinlock_t snd_use_lock_t; /* dummy */ -#define snd_use_lock_init(lockp) /**/ -#define snd_use_lock_use(lockp) /**/ -#define snd_use_lock_free(lockp) /**/ -#define snd_use_lock_sync(lockp) /**/ - -#endif /* SMP || CONFIG_SND_DEBUG */ - #endif /* __SND_SEQ_LOCK_H */ -- 2.15.0.rc0

2 1

[alsa-devel] Applied "drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data" to the asoc tree
by Mark Brown 18 Oct '17

18 Oct '17

The patch drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d4ba4ef2e894c10f2c16497fb0659e5ded502f97 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Date: Mon, 9 Oct 2017 16:35:12 -0400 Subject: [PATCH] drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data asic_type information is passed to ACP DMA Driver as platform data. Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c b/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c index a52795d9b458..ebca22302ebb 100644 --- a/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c @@ -371,6 +371,8 @@ static int acp_hw_init(void *handle) adev->acp.acp_cell[0].name = "acp_audio_dma"; adev->acp.acp_cell[0].num_resources = 4; adev->acp.acp_cell[0].resources = &adev->acp.acp_res[0]; + adev->acp.acp_cell[0].platform_data = &adev->asic_type; + adev->acp.acp_cell[0].pdata_size = sizeof(adev->asic_type); adev->acp.acp_cell[1].name = "designware-i2s"; adev->acp.acp_cell[1].num_resources = 1; -- 2.14.1

1 0

[alsa-devel] Applied "ASoC: AMD: Added asic_type as ACP DMA driver platform data" to the asoc tree
by Mark Brown 18 Oct '17

18 Oct '17

The patch ASoC: AMD: Added asic_type as ACP DMA driver platform data has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 01a6b1634d50dba65133a89fe19f182173bb8e8f Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Date: Mon, 9 Oct 2017 16:36:08 -0400 Subject: [PATCH] ASoC: AMD: Added asic_type as ACP DMA driver platform data asic_type information is passed to ACP DMA Driver as platform data. Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 8 ++------ sound/soc/amd/acp.h | 7 +++++++ 2 files changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 08b1399d1da2..dcbf9973884d 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -73,12 +73,6 @@ static const struct snd_pcm_hardware acp_pcm_hardware_capture = { .periods_max = CAPTURE_MAX_NUM_PERIODS, }; -struct audio_drv_data { - struct snd_pcm_substream *play_stream; - struct snd_pcm_substream *capture_stream; - void __iomem *acp_mmio; -}; - static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -916,6 +910,7 @@ static int acp_audio_probe(struct platform_device *pdev) int status; struct audio_drv_data *audio_drv_data; struct resource *res; + const u32 *pdata = pdev->dev.platform_data; audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); @@ -932,6 +927,7 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_stream = NULL; audio_drv_data->capture_stream = NULL; + audio_drv_data->asic_type = *pdata; res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); if (!res) { diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 330832ef4e5e..28cf9140f49c 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -84,6 +84,13 @@ struct audio_substream_data { void __iomem *acp_mmio; }; +struct audio_drv_data { + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + void __iomem *acp_mmio; + u32 asic_type; +}; + enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- 2.14.1

1 0

[alsa-devel] [PATCH 0/9 v2] Add ASoC support for AMD Stoney APUs
by Alex Deucher 18 Oct '17

18 Oct '17

This patch set updates the AMD GPU and Audio CoProcessor (ACP) audio drivers and the designware i2s driver for Stoney (ST). ST is an APU similar to Carrizo (CZ) which already has ACP audio support. The i2s controller and ACP audio DMA engine are part of the GPU and both need updating so I would like to upstream the whole patch set via one tree if possible. The current code is based on drm-next, but I'm happy to rebase on whatever tree this ends up going through if there are any problems applying. The entire patch set can be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=stoney_acp Thanks! Alex v2: - Patch 1 is already applied to the audio tree, just including it for completeness since it's required for this patch set and it's not yet in the drm tree. - New patch to share asic types between gpu and audio drivers - ACPI ID changed for rt5650 machine driver - Integrate feedback on other patches Akshu Agrawal (2): drm/amdgpu Moving amdgpu asic types to a separate file ASoC: AMD: Add machine driver for cz rt5650 Vijendar Mukunda (7): ASoC: dwc: Added a quirk DW_I2S_QUIRK_16BIT_IDX_OVERRIDE to dwc driver drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data drm/amd/amdgpu: Added a quirk for Stoney platform ASoC: AMD: disabling memory gating in stoney platform ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Audio buffer related changes for Stoney drm/amd/amdgpu: Disabling Power Gating for Stoney platform drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 74 ++++++---- drivers/gpu/drm/amd/include/amd_shared.h | 29 +--- include/drm/amd_asic_type.h | 52 +++++++ include/sound/designware_i2s.h | 1 + sound/soc/amd/Kconfig | 7 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-pcm-dma.c | 243 ++++++++++++++++++++++++------- sound/soc/amd/acp-rt5645.c | 210 ++++++++++++++++++++++++++ sound/soc/amd/acp.h | 9 ++ sound/soc/dwc/dwc-i2s.c | 6 + 10 files changed, 525 insertions(+), 108 deletions(-) create mode 100644 include/drm/amd_asic_type.h create mode 100644 sound/soc/amd/acp-rt5645.c -- 2.5.5

2 14

[alsa-devel] [PATCH 0/6 v4] Add ASoC support for AMD Stoney APUs
by Alex Deucher 18 Oct '17

18 Oct '17

This patch set updates the AMD GPU and Audio CoProcessor (ACP) audio drivers and the designware i2s driver for Stoney (ST). ST is an APU similar to Carrizo (CZ) which already has ACP audio support. The i2s controller and ACP audio DMA engine are part of the GPU and both need updating so I would like to upstream the whole patch set via one tree if possible. The entire patch set can be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=stoney_acp3 I just need acks from audio on the remaining patches that touch audio (1, 5, 6), and then I can send a pull request. Thanks! Alex v2: - Patch 1 is already applied to the audio tree, just including it for completeness since it's required for this patch set and it's not yet in the drm tree. - New patch to share asic types between gpu and audio drivers - ACPI ID changed for rt5650 machine driver - Integrate feedback on other patches v3: - resend remaining patches that have not been applied yet v4: - Integrate feedback from v3 - Add Mark's acks from v3 - Integrate a machine driver fix from Dylan Reid Akshu Agrawal (2): drm/amdgpu Moving amdgpu asic types to a separate file ASoC: AMD: Add machine driver for cz rt5650 Vijendar Mukunda (4): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: AMD: disabling memory gating in stoney platform ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Audio buffer related changes for Stoney drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 + drivers/gpu/drm/amd/include/amd_shared.h | 29 +--- include/drm/amd_asic_type.h | 52 +++++++ sound/soc/amd/Kconfig | 7 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-pcm-dma.c | 253 ++++++++++++++++++++++++------- sound/soc/amd/acp-rt5645.c | 199 ++++++++++++++++++++++++ sound/soc/amd/acp.h | 9 ++ 8 files changed, 471 insertions(+), 82 deletions(-) create mode 100644 include/drm/amd_asic_type.h create mode 100644 sound/soc/amd/acp-rt5645.c -- 2.5.5

3 14

[alsa-devel] [PATCH] ALSA: emu10k1: remove redundant assignment to tmp
by Colin King 17 Oct '17

17 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The assignment to variable tmp is redundant as the value is never read and a new value is assigned to tmp in the following for-loop, so remove the assignment. Cleans up clang warning: Value stored to 'tmp' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/emu10k1/emuproc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index cf05229b569b..bde0d1954f56 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -280,7 +280,6 @@ static void snd_emu10k1_proc_rates_read(struct snd_info_entry *entry, struct snd_emu10k1 *emu = entry->private_data; unsigned int val, tmp, n; val = snd_emu10k1_ptr20_read(emu, CAPTURE_RATE_STATUS, 0); - tmp = (val >> 16) & 0x8; for (n = 0; n < 4; n++) { tmp = val >> (16 + (n*4)); if (tmp & 0x8) snd_iprintf(buffer, "Channel %d: Rate=%d\n", n, samplerate[tmp & 0x7]); -- 2.14.1

2 1

Re: [alsa-devel] stable-rc build: 9 warnings 0 failures (stable-rc/v3.18.75-20-g4ffe554)
by Arnd Bergmann 17 Oct '17

17 Oct '17

On Tue, Oct 17, 2017 at 12:22 PM, Olof's autobuilder <build(a)lixom.net> wrote: > Warnings: > > 9 sound/usb/mixer.c:2161:27: warning: passing argument 1 of 'snd_usb_mixer_disconnect' from incompatible pointer type [-Wincompatible-pointer-types] This broke after backporting 124751d5e63c ("ALSA: usb-audio: Kill stray URB at exiting") into linux-3.18. The interface was changed in commit a6cece9d8199 ("ALSA: usb-audio: Pass direct struct pointer instead of list_head"), which was merged in linux-3.19. I don't immediately see how this can be fixed. Backporting a6cece9d8199 would probably work. Arnd

3 2

[alsa-devel] [PATCH] ALSA: au88x0: remove redundant assignment of variable i
by Colin King 17 Oct '17

17 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The assignment to variable i is redundant as it is never read because it is updated a little later on. Remove this assignment. Cleans up clang warning: Value stored to 'i' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/au88x0/au88x0_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index c308a4f70550..4083c8b01619 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2628,7 +2628,7 @@ static void vortex_spdif_init(vortex_t * vortex, int spdif_sr, int spdif_mode) else edi = 0x1ffff; } else { - i = edi = 0x800; + edi = 0x800; } /* this_04 and this_08 are the CASp4Src's (samplerate converters) */ vortex_src_setupchannel(vortex, this_04, edi, 0, 1, -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: tpa61xx: add back chip name to the controls
by Lucas Stach 17 Oct '17

17 Oct '17

Commit cb7e62256e99( ASoC: tpa6130a2: Register component) removed the chip name from the controls, which causes issues as the TPA61xx is not a standalone codec, but a headphone amplifier, which may be connected to the line out of a codec which already exposes a "Headphone Playback Volume" control. In that case we end up with duplicate controls, causing the the probe of the soundcard to be aborted. Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- sound/soc/codecs/tpa6130a2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 2e014c80d113..abca7eedb6be 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -136,7 +136,7 @@ static const DECLARE_TLV_DB_RANGE(tpa6130_tlv, ); static const struct snd_kcontrol_new tpa6130a2_controls[] = { - SOC_SINGLE_TLV("Headphone Playback Volume", + SOC_SINGLE_TLV("TPA6130A2 Headphone Playback Volume", TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0, tpa6130_tlv), }; @@ -148,7 +148,7 @@ static const DECLARE_TLV_DB_RANGE(tpa6140_tlv, ); static const struct snd_kcontrol_new tpa6140a2_controls[] = { - SOC_SINGLE_TLV("Headphone Playback Volume", + SOC_SINGLE_TLV("TPA6140A2 Headphone Playback Volume", TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0, tpa6140_tlv), }; -- 2.11.0

2 2

Re: [alsa-devel] [PATCH] ASoC: sgtl5000: add support for a fixed sysclk frequency
by Fabio Estevam 17 Oct '17

17 Oct '17

[Sorry for the top-posting. I did not see your message in the mailing list] Hi Jakob, arch/arm/boot/dts/imx51-babbage.dts has SYS_MCLK generated by an external fixed oscillator and it works fine. Do you really need this new property? ________________________________ From: Jakob Unterwurzacher <jakob.unterwurzacher(a)theobroma-systems.com> Sent: Friday, October 6, 2017 5:34:44 AM To: linux-kernel(a)vger.kernel.org Cc: Jakob Unterwurzacher; Klaus Goger; Liam Girdwood; Mark Brown; Rob Herring; Mark Rutland; Jaroslav Kysela; Takashi Iwai; Richard Leitner; Fabio Estevam; Bhumika Goyal; alsa-devel(a)alsa-project.org; devicetree(a)vger.kernel.org Subject: [PATCH] ASoC: sgtl5000: add support for a fixed sysclk frequency The system clock, also called SYS_MCLK in the SGTL5000 datasheet, is usually generated by the I2S master with a frequency proportional to the sample rate. However, this is not always the case. Compute modules following the Qseven specification, for example, do not even have a pin for I2S SYS_MCLK on their board-to-board connector. Only I2S_WS, I2S_RST# I2S_CLK, I2S_SDI and I2S_SDO are specified. Instead, SYS_MCLK is usually generated by an external fixed oscillator. Add support for this scenario, enabled by the new "sysclk-fixed" devicetree property. Tested on a "Puma" RK3399-Q7 compute module with a "Haikou" baseboard that has a fixed 24.576 MHz oscillator connected to the codec's SYS_MCLK. Signed-off-by: Jakob Unterwurzacher <jakob.unterwurzacher(a)theobroma-systems.com> Cc: Klaus Goger <klaus.goger(a)theobroma-systems.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 5 +++++ sound/soc/codecs/sgtl5000.c | 12 ++++++++++++ 2 files changed, 17 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 7a73a9d62015..0d0dd58b2d4a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -35,6 +35,11 @@ VDDIO 1.8V 2.5V 3.3V 2 = 3.33 mA 5.74 mA 8.03 mA 3 = 4.99 mA 8.61 mA 12.05 mA +- sysclk-fixed : If this property exists, the sysclk is considered to be + generated by a fixed-frequency oscillator. The frequency value is read + from the clock defined in the "clocks" property. set_sysclk requests + are ignored. + Example: codec: sgtl5000@0a { diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f2bb4feba3b6..be3aea89371c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -133,6 +133,7 @@ struct sgtl5000_priv { u8 micbias_resistor; u8 micbias_voltage; u8 lrclk_strength; + u8 sysclk_fixed; /* sysclk rate cannot be changed */ }; /* @@ -613,6 +614,10 @@ static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case SGTL5000_SYSCLK: + if (sgtl5000->sysclk_fixed) { + dev_dbg(codec->dev, "sysclk-fixed: ignoring set_sysclk request\n"); + return 0; + } sgtl5000->sysclk = freq; break; default: @@ -1444,6 +1449,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, } else { sgtl5000->micbias_voltage = 0; } + if (of_property_read_bool(np, "sysclk-fixed")) { + sgtl5000->sysclk = clk_get_rate(sgtl5000->mclk); + sgtl5000->sysclk_fixed = 1; + dev_dbg(&client->dev, + "sysclk-fixed: setting sysclk to fixed value %d\n", + sgtl5000->sysclk); + } } sgtl5000->lrclk_strength = I2S_LRCLK_STRENGTH_LOW; -- 2.11.0

3 6

[alsa-devel] [PATCH] ALSA: usb-audio: Convert timers to use timer_setup()
by Kees Cook 17 Oct '17

17 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Andrej Krutak <dev(a)andree.sk> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: "Hans P. Möller Ebner" <hmoller(a)uc.cl> Cc: Bhumika Goyal <bhumirks(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/usb/line6/driver.c | 4 ++-- sound/usb/line6/driver.h | 3 +-- sound/usb/line6/pod.c | 11 +++++------ sound/usb/line6/podhd.c | 10 +++++----- sound/usb/line6/toneport.c | 7 +++---- sound/usb/line6/variax.c | 32 ++++++++++++++++---------------- sound/usb/midi.c | 7 +++---- 7 files changed, 35 insertions(+), 39 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 8d5a454842f4..4f9613e5fc9e 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -199,9 +199,9 @@ static int line6_send_raw_message_async_part(struct message *msg, Setup and start timer. */ void line6_start_timer(struct timer_list *timer, unsigned long msecs, - void (*function)(unsigned long), unsigned long data) + void (*function)(struct timer_list *t)) { - setup_timer(timer, function, data); + timer->function = (TIMER_FUNC_TYPE)function; mod_timer(timer, jiffies + msecs_to_jiffies(msecs)); } EXPORT_SYMBOL_GPL(line6_start_timer); diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index dc97895547be..61425597eb61 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -198,8 +198,7 @@ extern int line6_send_sysex_message(struct usb_line6 *line6, extern ssize_t line6_set_raw(struct device *dev, struct device_attribute *attr, const char *buf, size_t count); extern void line6_start_timer(struct timer_list *timer, unsigned long msecs, - void (*function)(unsigned long), - unsigned long data); + void (*function)(struct timer_list *t)); extern int line6_version_request_async(struct usb_line6 *line6); extern int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, unsigned datalen); diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index 358224cc5638..020c81818951 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -174,7 +174,7 @@ static const char pod_version_header[] = { }; /* forward declarations: */ -static void pod_startup2(unsigned long data); +static void pod_startup2(struct timer_list *t); static void pod_startup3(struct usb_line6_pod *pod); static char *pod_alloc_sysex_buffer(struct usb_line6_pod *pod, int code, @@ -286,13 +286,12 @@ static void pod_startup1(struct usb_line6_pod *pod) CHECK_STARTUP_PROGRESS(pod->startup_progress, POD_STARTUP_INIT); /* delay startup procedure: */ - line6_start_timer(&pod->startup_timer, POD_STARTUP_DELAY, pod_startup2, - (unsigned long)pod); + line6_start_timer(&pod->startup_timer, POD_STARTUP_DELAY, pod_startup2); } -static void pod_startup2(unsigned long data) +static void pod_startup2(struct timer_list *t) { - struct usb_line6_pod *pod = (struct usb_line6_pod *)data; + struct usb_line6_pod *pod = from_timer(pod, t, startup_timer); struct usb_line6 *line6 = &pod->line6; CHECK_STARTUP_PROGRESS(pod->startup_progress, POD_STARTUP_VERSIONREQ); @@ -413,7 +412,7 @@ static int pod_init(struct usb_line6 *line6, line6->process_message = line6_pod_process_message; line6->disconnect = line6_pod_disconnect; - init_timer(&pod->startup_timer); + timer_setup(&pod->startup_timer, NULL, 0); INIT_WORK(&pod->startup_work, pod_startup4); /* create sysfs entries: */ diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 46fa8c046cb5..36ed9c85c0eb 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -158,7 +158,7 @@ static struct line6_pcm_properties podx3_pcm_properties = { }; static struct usb_driver podhd_driver; -static void podhd_startup_start_workqueue(unsigned long data); +static void podhd_startup_start_workqueue(struct timer_list *t); static void podhd_startup_workqueue(struct work_struct *work); static int podhd_startup_finalize(struct usb_line6_podhd *pod); @@ -208,12 +208,12 @@ static void podhd_startup(struct usb_line6_podhd *pod) /* delay startup procedure: */ line6_start_timer(&pod->startup_timer, PODHD_STARTUP_DELAY, - podhd_startup_start_workqueue, (unsigned long)pod); + podhd_startup_start_workqueue); } -static void podhd_startup_start_workqueue(unsigned long data) +static void podhd_startup_start_workqueue(struct timer_list *t) { - struct usb_line6_podhd *pod = (struct usb_line6_podhd *)data; + struct usb_line6_podhd *pod = from_timer(pod, t, startup_timer); CHECK_STARTUP_PROGRESS(pod->startup_progress, PODHD_STARTUP_SCHEDULE_WORKQUEUE); @@ -319,7 +319,7 @@ static int podhd_init(struct usb_line6 *line6, line6->disconnect = podhd_disconnect; - init_timer(&pod->startup_timer); + timer_setup(&pod->startup_timer, NULL, 0); INIT_WORK(&pod->startup_work, podhd_startup_workqueue); if (pod->line6.properties->capabilities & LINE6_CAP_CONTROL) { diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index ba7975c0d03d..750467fb95db 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -241,9 +241,9 @@ static int snd_toneport_source_put(struct snd_kcontrol *kcontrol, return 1; } -static void toneport_start_pcm(unsigned long arg) +static void toneport_start_pcm(struct timer_list *t) { - struct usb_line6_toneport *toneport = (struct usb_line6_toneport *)arg; + struct usb_line6_toneport *toneport = from_timer(toneport, t, timer); struct usb_line6 *line6 = &toneport->line6; line6_pcm_acquire(line6->line6pcm, LINE6_STREAM_MONITOR, true); @@ -415,8 +415,7 @@ static int toneport_init(struct usb_line6 *line6, struct usb_line6_toneport *toneport = (struct usb_line6_toneport *) line6; toneport->type = id->driver_info; - setup_timer(&toneport->timer, toneport_start_pcm, - (unsigned long)toneport); + timer_setup(&toneport->timer, toneport_start_pcm, 0); line6->disconnect = line6_toneport_disconnect; diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c index 0c4512d0382e..e8c852b2ce35 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -82,9 +82,9 @@ static const char variax_activate[] = { }; /* forward declarations: */ -static void variax_startup2(unsigned long data); -static void variax_startup4(unsigned long data); -static void variax_startup5(unsigned long data); +static void variax_startup2(struct timer_list *t); +static void variax_startup4(struct timer_list *t); +static void variax_startup5(struct timer_list *t); static void variax_activate_async(struct usb_line6_variax *variax, int a) { @@ -106,12 +106,12 @@ static void variax_startup1(struct usb_line6_variax *variax) /* delay startup procedure: */ line6_start_timer(&variax->startup_timer1, VARIAX_STARTUP_DELAY1, - variax_startup2, (unsigned long)variax); + variax_startup2); } -static void variax_startup2(unsigned long data) +static void variax_startup2(struct timer_list *t) { - struct usb_line6_variax *variax = (struct usb_line6_variax *)data; + struct usb_line6_variax *variax = from_timer(variax, t, startup_timer1); struct usb_line6 *line6 = &variax->line6; /* schedule another startup procedure until startup is complete: */ @@ -120,7 +120,7 @@ static void variax_startup2(unsigned long data) variax->startup_progress = VARIAX_STARTUP_VERSIONREQ; line6_start_timer(&variax->startup_timer1, VARIAX_STARTUP_DELAY1, - variax_startup2, (unsigned long)variax); + variax_startup2); /* request firmware version: */ line6_version_request_async(line6); @@ -132,12 +132,12 @@ static void variax_startup3(struct usb_line6_variax *variax) /* delay startup procedure: */ line6_start_timer(&variax->startup_timer2, VARIAX_STARTUP_DELAY3, - variax_startup4, (unsigned long)variax); + variax_startup4); } -static void variax_startup4(unsigned long data) +static void variax_startup4(struct timer_list *t) { - struct usb_line6_variax *variax = (struct usb_line6_variax *)data; + struct usb_line6_variax *variax = from_timer(variax, t, startup_timer2); CHECK_STARTUP_PROGRESS(variax->startup_progress, VARIAX_STARTUP_ACTIVATE); @@ -145,12 +145,12 @@ static void variax_startup4(unsigned long data) /* activate device: */ variax_activate_async(variax, 1); line6_start_timer(&variax->startup_timer2, VARIAX_STARTUP_DELAY4, - variax_startup5, (unsigned long)variax); + variax_startup5); } -static void variax_startup5(unsigned long data) +static void variax_startup5(struct timer_list *t) { - struct usb_line6_variax *variax = (struct usb_line6_variax *)data; + struct usb_line6_variax *variax = from_timer(variax, t, startup_timer2); CHECK_STARTUP_PROGRESS(variax->startup_progress, VARIAX_STARTUP_WORKQUEUE); @@ -190,7 +190,7 @@ static void line6_variax_process_message(struct usb_line6 *line6) } else if (memcmp(buf + 1, variax_init_done + 1, sizeof(variax_init_done) - 1) == 0) { /* notify of complete initialization: */ - variax_startup4((unsigned long)variax); + variax_startup4(&variax->startup_timer2); } break; } @@ -222,8 +222,8 @@ static int variax_init(struct usb_line6 *line6, line6->process_message = line6_variax_process_message; line6->disconnect = line6_variax_disconnect; - init_timer(&variax->startup_timer1); - init_timer(&variax->startup_timer2); + timer_setup(&variax->startup_timer1, NULL, 0); + timer_setup(&variax->startup_timer2, NULL, 0); INIT_WORK(&variax->startup_work, variax_startup6); /* initialize USB buffers: */ diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 7ab25de5ca0a..2c1aaa3292bf 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -352,9 +352,9 @@ static void snd_usbmidi_out_tasklet(unsigned long data) } /* called after transfers had been interrupted due to some USB error */ -static void snd_usbmidi_error_timer(unsigned long data) +static void snd_usbmidi_error_timer(struct timer_list *t) { - struct snd_usb_midi *umidi = (struct snd_usb_midi *)data; + struct snd_usb_midi *umidi = from_timer(umidi, t, error_timer); unsigned int i, j; spin_lock(&umidi->disc_lock); @@ -2369,8 +2369,7 @@ int __snd_usbmidi_create(struct snd_card *card, usb_id = USB_ID(le16_to_cpu(umidi->dev->descriptor.idVendor), le16_to_cpu(umidi->dev->descriptor.idProduct)); umidi->usb_id = usb_id; - setup_timer(&umidi->error_timer, snd_usbmidi_error_timer, - (unsigned long)umidi); + timer_setup(&umidi->error_timer, snd_usbmidi_error_timer, 0); /* detect the endpoint(s) to use */ memset(endpoints, 0, sizeof(endpoints)); -- 2.7.4 -- Kees Cook Pixel Security

2 1

[alsa-devel] [PATCH] ASoC: cht-bsw-rt5645: Also enable/disable the clock on Cherry Trail
by Hans de Goede 16 Oct '17

16 Oct '17

Just like on Bay Trail the codec mclk is also controlled by the common clk framework as pmc_plt_clk_3 on Cherry Trail and the cht-bsw-rt5645 needs to control it so that it has the right frequency and gets en/disabled as appropriate. This fixes the sound crackling (like a bonfire) when the clock was not set up properly before Linux boots and turns the clock off when not in use saving power. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=196351 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 5bcde01d15e6..d553e2b67c92 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -682,14 +682,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) cht_rt5645_cpu_dai_name; } - if (is_valleyview()) { - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); - if (IS_ERR(drv->mclk)) { - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); - return PTR_ERR(drv->mclk); - } + drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(drv->mclk)) { + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %ld\n", + PTR_ERR(drv->mclk)); + return PTR_ERR(drv->mclk); } snd_soc_card_set_drvdata(card, drv); -- 2.14.2

2 1

[alsa-devel] [PATCH] ALSA: 6fire: remove unused variable card
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The pointer card is being assigned a value but it is never used. Remove this redundant variable. Cleans up clang warning: Value stored to 'card' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/6fire/chip.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index c7641cb50616..17d5e3ee6d73 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -174,11 +174,9 @@ static int usb6fire_chip_probe(struct usb_interface *intf, static void usb6fire_chip_disconnect(struct usb_interface *intf) { struct sfire_chip *chip; - struct snd_card *card; chip = usb_get_intfdata(intf); if (chip) { /* if !chip, fw upload has been performed */ - card = chip->card; chip->intf_count--; if (!chip->intf_count) { mutex_lock(&register_mutex); -- 2.14.1

2 1

[alsa-devel] [PATCH] asihpi: clean up a couple of build warnings
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> Variable dpcm is never used and hence can be removed. Variable runtime is being assigned but is never read, so the assignment is redundant and can be removed. Cleans up two clang warnings: Value stored to 'dpcm' is never read Value stored to 'runtime' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/asihpi/asihpi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 614322171c79..720361455c60 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -573,10 +573,8 @@ static void snd_card_asihpi_pcm_int_start(struct snd_pcm_substream *substream) static void snd_card_asihpi_pcm_int_stop(struct snd_pcm_substream *substream) { - struct snd_card_asihpi_pcm *dpcm; struct snd_card_asihpi *card; - dpcm = (struct snd_card_asihpi_pcm *)substream->runtime->private_data; card = snd_pcm_substream_chip(substream); hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index, @@ -863,7 +861,6 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) snd_pcm_group_for_each_entry(s, substream) { struct snd_card_asihpi_pcm *ds = s->runtime->private_data; - runtime = s->runtime; /* don't link Cap and Play */ if (substream->stream != s->stream) -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: emux: remove unused redundant variable p2
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The variable p2 is being assigned but never used, it is redundant and can be safely removed. Cleans up clang warning: Value stored to 'p2' is never read. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/synth/emux/emux_oss.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index de19e108974a..764ff4bc2089 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -430,7 +430,6 @@ gusspec_control(struct snd_emux *emu, struct snd_emux_port *port, int cmd, { int voice; unsigned short p1; - short p2; int plong; struct snd_midi_channel *chan; @@ -445,7 +444,6 @@ gusspec_control(struct snd_emux *emu, struct snd_emux_port *port, int cmd, chan = &port->chset.channels[voice]; p1 = *(unsigned short *) &event[4]; - p2 = *(short *) &event[6]; plong = *(int*) &event[4]; switch (cmd) { -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: ens137x: remove redundant variable result
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> Variable result is being assigned a value from a calculation however the variable is never read, so this redundant variable can be removed. Cleans up clang warning: Value stored to 'result' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/ens1370.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index d4cd6451fdca..39f79a6b5283 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -732,7 +732,7 @@ static void snd_es1371_codec_wait(struct snd_ac97 *ac97) static void snd_es1371_adc_rate(struct ensoniq * ensoniq, unsigned int rate) { - unsigned int n, truncm, freq, result; + unsigned int n, truncm, freq; mutex_lock(&ensoniq->src_mutex); n = rate / 3000; @@ -740,7 +740,6 @@ static void snd_es1371_adc_rate(struct ensoniq * ensoniq, unsigned int rate) n--; truncm = (21 * n - 1) | 1; freq = ((48000UL << 15) / rate) * n; - result = (48000UL << 15) / (freq / n); if (rate >= 24000) { if (truncm > 239) truncm = 239; -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: pcm: remove redundant variable runtime
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> An earlier commit removed the access to variable runtime and we are now left with unused variable that is redundant, so remove it. Cleans up the clang warning: Value stored to 'runtime' is never read Fixes: e11f0f90a626 ("ALSA: pcm: remove SNDRV_PCM_IOCTL1_INFO internal command") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/pcm_native.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 2fec2feac387..a4d92e46c459 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -195,7 +195,6 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore); int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) { - struct snd_pcm_runtime *runtime; struct snd_pcm *pcm = substream->pcm; struct snd_pcm_str *pstr = substream->pstr; @@ -211,7 +210,6 @@ int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) info->subdevices_count = pstr->substream_count; info->subdevices_avail = pstr->substream_count - pstr->substream_opened; strlcpy(info->subname, substream->name, sizeof(info->subname)); - runtime = substream->runtime; return 0; } -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add native DSD support for Pro-Ject Pre Box S2 Digital
by Jussi Laako 16 Oct '17

16 Oct '17

Add native DSD support quirk for Pro-Ject Pre Box S2 Digital USB id 2772:0230. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9aeb05f..5a040e3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1348,6 +1348,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */ + case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.7.4

2 1

[alsa-devel] 4.13 regression: get_kctl_0dB_offset doesn't handle all possible callbacks
by PaX Team 16 Oct '17

16 Oct '17

hello devs, upstream commit 99b5c5bb9a5435a5ae5d46445ac0f2bf6aa5ee52 removed the use of set_fs in get_kctl_0dB_offset under the assumption that the only runtime value of kctl->tlv.c was snd_hda_mixer_amp_tlv. alas, recently, the KERNEXEC and UDEREF features in PaX reported a violation of this assumption as it turns out that _snd_ctl_add_slave sets this callback to slave_tlv_cmd instead. it seems to me that there're several other possible values for this callback so it's not clear why it was assumed that get_kctl_0dB_offset would only be called with only one of them in practice, but nevertheless, it happens here so clearly something is wrong. the backtrace that shows how things go wrong in this case: PAX: suspicious general protection fault: 0000 [#1] PREEMPT SMP Modules linked in: CPU: 4 PID: 31 Comm: kworker/4:0 Not tainted 4.13.5-i386-pax #17 Workqueue: events azx_probe_work task: eb61c880 task.stack: eb62a000 EIP: 0060:[<009bbd2d>] init_slave_0dB+0x2d/0xa0 EFLAGS: 00210202 CPU: 4 EAX: 00991fd0 EBX: e9883a80 ECX: e9883a80 EDX: eb62bd74 ESI: eb62bd74 EDI: e9098800 EBP: eb62bd08 ESP: eb62bcec DS: 0068 ES: 0068 FS: 00d8 GS: 0068 SS: 0068 CR0: 80050033 CR2: 00000000 CR3: 03b2d100 CR4: 000406f0 shadow CR4: 000406f0 Call Trace: [<009bb469>] map_slaves+0xb9/0xe0 [<009bce0e>] __snd_hda_add_vmaster+0xde/0x110 [<009c82f0>] snd_hda_gen_build_controls+0x1a0/0x1b0 [<009d0a4d>] cx_auto_build_controls+0xd/0x70 [<009bdf76>] snd_hda_codec_build_controls+0x186/0x1d0 [<009b92ad>] hda_codec_driver_probe+0x6d/0xf0 [<006a3be9>] driver_probe_device+0x289/0x420 [<006a3ed6>] __device_attach_driver+0x76/0x100 [<006a1edf>] bus_for_each_drv+0x3f/0x70 [<006a3813>] __device_attach+0xa3/0x110 [<006a3f9d>] device_initial_probe+0xd/0x10 [<006a2d6f>] bus_probe_device+0x6f/0x80 [<006a100f>] device_add+0x2cf/0x590 [<009d8d8c>] snd_hdac_device_register+0xc/0x40 [<009b90b4>] snd_hda_codec_configure+0x34/0x140 [<009c2875>] azx_codec_configure+0x25/0x50 [<009d8081>] azx_probe_continue+0x621/0x9e0 [<009d84bd>] azx_probe_work+0xd/0x10 [<0006fff2>] process_one_work+0x122/0x2a0 [<000701a9>] worker_thread+0x39/0x390 [<00074df2>] kthread+0xe2/0x110 [<00cee619>] ret_from_fork+0x19/0x30 Code: e5 57 89 c7 56 89 d6 53 89 cb 83 ec 10 8b 41 6c a9 00 00 00 10 74 09 81 79 58 90 bc 9b 00 74 4e a8 10 74 38 8b 43 58 85 c0 74 31 <83> 38 01 75 2c 8b 48 0c 81 e1 ff ff fe ff 74 21 8b 16 39 d1 74 EIP: [<009bbd2d>] init_slave_0dB+0x2d/0xa0 SS:ESP: 0068:eb62bcec what KERNEXEC on i386 does is that it executes kernel code in its own 0-based code segment hence the 'low' code addresses. due to the current logic that checks for SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK in get_kctl_0dB_offset, this callback address is instead treated as a data pointer (as apparently SNDRV_CTL_ELEM_ACCESS_TLV_READ is also set) and since it's not a valid kernel address, it causes a GPF under the UDEREF PaX feature (without UDEREF it'd have been an oops since such low addresses are not mapped for kernel threads). on vanilla kernels all this is a silent read of kernel code bytes that are then interpreted as the tlv[] array content, which is probably not what you want either. as for fixing this, first the above mentioned assumption should be re-evaluated. if it's considered correct then there is some logic bug in my case (i can help debug it if you tell me what to do) otherwise the current pattern of if (SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK && callback == snd_hda_mixer_amp_tlv) { call wrapped function underneath snd_hda_mixer_amp_tlv } else if (SNDRV_CTL_ELEM_ACCESS_TLV_READ) { treat unrecognized callback address as data ptr } should be changed to if (SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK( { if (callback == snd_hda_mixer_amp_tlv) { call wrapped function underneath snd_hda_mixer_amp_tlv } else if (callback == others) { handle others, WARN/BUG/etc } } else if (SNDRV_CTL_ELEM_ACCESS_TLV_READ) { no longer treat unrecognized callback address as data ptr } and all other callbacks with userland access should be refactored the same way as snd_hda_mixer_amp_tlv was. i'd also suggest that you at least do the above suggested if/else pattern change in order to prevent the misuse of unexpected callbacks in the future. cheers, PaX Team

3 2

[alsa-devel] [PATCH 0/2] sound: oss: make the function argument and structure const
by Bhumika Goyal 16 Oct '17

16 Oct '17

Make the function argument of function sound_install_audiodrv const. After this change, make the structures audio_driver const as they are only passed to the const argument (made const in Patch[1/2]) of the function sound_install_audiodrv . Bhumika Goyal (2): sound: oss: make the function argument as const sound: oss: make audio_driver structure const sound/oss/dev_table.c | 2 +- sound/oss/dev_table.h | 2 +- sound/oss/pas2_pcm.c | 2 +- sound/oss/waveartist.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) -- 1.9.1

2 3

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Decrease loglevel for topology loading
by Chintan Patel 16 Oct '17

16 Oct '17

In case of failure in loading customize topology firmware, dfw_sst.bin gets loaded. However, current log provides this message as error even after successfully falling back to default topology "dfw_sst.bin". Hence to convey proper message, changing log level and message. Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- Changes in v2: Addressing review comments from Takashi Iwai by changing loglevel as info sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22f768ca3c73..12f383fd7c39 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3404,7 +3404,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_err(bus->dev, "tplg fw %s load failed with %d\n", + dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", skl->tplg_name, ret); ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { -- 1.9.1

2 1

[alsa-devel] Surface 3 audio broken between 4.11 and 4.13 (rt5645)
by Bastien Nocera 15 Oct '17

15 Oct '17

Hey, As I was trying to test PulseAudio HDMI LPE audio support fixes, I realised that the internal sound card of my Surface 3 wasn't working anymore, using the same user-space configuration. Looks like detection of the device regressed between 2 versions. Does this look familiar to anyone, or should I start bisecting? (which will take me absolutely ages, as the CPU is anemic on that machine) Working: $ dmesg | grep rt5 [ 13.707825] rt5640 i2c-10EC5640:00: Device with ID register 0x6308 is not rt5640/39 [ 15.010548] cht-bsw-rt5645 cht-bsw-rt5645: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 15.011132] cht-bsw-rt5645 cht-bsw-rt5645: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [ 15.012756] cht-bsw-rt5645 cht-bsw-rt5645: rt5645-aif1 <-> ssp2-port mapping ok [ 15.100152] input: chtrt5645 Headset as /devices/pci0000:00/808622A8:00/cht-bsw-rt5645/sound/card0/input27 $ uname -r 4.11.0-0.rc0.git4.21.surface3.fc26.x86_64 Not working: $ journalctl -b -2 | grep rt5 Oct 13 20:10:19 surface kernel: rt5640 i2c-10EC5640:00: Device with ID register 0x6308 is not rt5640/39 Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: ASoC: CODEC DAI rt5640-aif1 not registered Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: devm_snd_soc_register_card failed -517 Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: ASoC: CODEC DAI rt5640-aif1 not registered Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: devm_snd_soc_register_card failed -517 Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: ASoC: CODEC DAI rt5640-aif1 not registered Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: devm_snd_soc_register_card failed -517 Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: ASoC: CODEC DAI rt5640-aif1 not registered Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: devm_snd_soc_register_card failed -517 $ journalctl -b -2 | head -n3 | tail -n1 Oct 13 22:10:10 surface kernel: Linux version 4.13.0-0.rc2.git0.26.surface3.fc27.x86_64 (hadess@classic) (gcc version 7.1.1 20170526 (Red Hat 7.1.1-2) (GCC)) #1 SMP Tue Jul 25 18:47:03 CEST 2017 Cheers

2 2

[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Add support for loading FE dais from topology
by Guneshwor Singh 13 Oct '17

13 Oct '17

This series adds support for loading FE dais using topology. A flag is introduced to decide whether to register FE dais defined in the platform driver or use from topology. By default, it is set to register the FE dais defined in the platform driver. While at it, fix a missing sentinel too. Guneshwor Singh (3): ASoC: Intel: Skylake: Add flag to check to register FE dais from topology ASoC: Intel: Skylake: Add dai load ops for dais from topology ASoC: Intel: Skylake: Fix missing sentinel in sst_acpi_mach sound/soc/intel/skylake/skl-pcm.c | 46 +++++++++++++++++++++++++++++++--- sound/soc/intel/skylake/skl-topology.c | 1 + sound/soc/intel/skylake/skl-topology.h | 3 +++ sound/soc/intel/skylake/skl.c | 10 +++++++- sound/soc/intel/skylake/skl.h | 4 ++- 5 files changed, 58 insertions(+), 6 deletions(-) -- 2.14.2

3 7

[alsa-devel] [PATCH] ASoC: rsnd: more clear ADG clock debug info
by Kuninori Morimoto 13 Oct '17

13 Oct '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ADG inputs clock from CLK{A,B,C,I} and outputs clock from CLKOUT{0,1,2,3} which is selected by BRG{A,B}. Now, ADG is assuming BRGA is for 44100Hz related clocks, BRGB is for 48000Hz related clocks. Clock related debug is very difficult/confusable. This patch cleanups clock related debug info. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 59 ++++++++++++++++++++++++++++++++++--------------- 1 file changed, 41 insertions(+), 18 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 19213077..8ddb087 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -57,6 +57,13 @@ struct rsnd_adg { i++) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) +static const char * const clk_name[] = { + [CLKA] = "clk_a", + [CLKB] = "clk_b", + [CLKC] = "clk_c", + [CLKI] = "clk_i", +}; + static u32 rsnd_adg_calculate_rbgx(unsigned long div) { int i, ratio; @@ -279,6 +286,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod); struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + struct device *dev = rsnd_priv_to_dev(priv); int id = rsnd_mod_id(ssi_mod); int shift = (id % 4) * 8; u32 mask = 0xFF << shift; @@ -305,12 +313,13 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val); break; } + + dev_dbg(dev, "AUDIO_CLK_SEL is 0x%x\n", val); } int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) { struct rsnd_adg *adg = rsnd_priv_to_adg(priv); - struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; int i; int sel_table[] = { @@ -320,8 +329,6 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) [CLKI] = 0x0, }; - dev_dbg(dev, "request clock = %d\n", rate); - /* * find suitable clock from * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. @@ -377,9 +384,10 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) rsnd_mod_write(adg_mod, BRRA, adg->rbga); rsnd_mod_write(adg_mod, BRRB, adg->rbgb); - dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n", - rsnd_mod_name(ssi_mod), rsnd_mod_id(ssi_mod), - data, rate); + dev_dbg(dev, "CLKOUT is based on BRG%c (= %dHz)\n", + (ckr) ? 'B' : 'A', + (ckr) ? adg->rbgb_rate_for_48khz : + adg->rbga_rate_for_441khz); return 0; } @@ -408,21 +416,12 @@ static void rsnd_adg_get_clkin(struct rsnd_priv *priv, { struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; - static const char * const clk_name[] = { - [CLKA] = "clk_a", - [CLKB] = "clk_b", - [CLKC] = "clk_c", - [CLKI] = "clk_i", - }; int i; for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); adg->clk[i] = IS_ERR(clk) ? NULL : clk; } - - for_each_rsnd_clk(clk, adg, i) - dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk)); } static void rsnd_adg_get_clkout(struct rsnd_priv *priv, @@ -571,12 +570,35 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->ckr = ckr; adg->rbga = rbga; adg->rbgb = rbgb; +} + +#ifdef DEBUG +static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + int i; + + for_each_rsnd_clk(clk, adg, i) + dev_dbg(dev, "%s : %p : %ld\n", + clk_name[i], clk, clk_get_rate(clk)); - for_each_rsnd_clkout(clk, adg, i) - dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk)); dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", - ckr, rbga, rbgb); + adg->ckr, adg->rbga, adg->rbgb); + dev_dbg(dev, "BRGA (for 44100 base) = %d\n", adg->rbga_rate_for_441khz); + dev_dbg(dev, "BRGB (for 48000 base) = %d\n", adg->rbgb_rate_for_48khz); + + /* + * Actual CLKOUT will be exchanged in rsnd_adg_ssi_clk_try_start() + * by BRGCKR::BRGCKR_31 + */ + for_each_rsnd_clkout(clk, adg, i) + dev_dbg(dev, "clkout %d : %p : %ld\n", i, + clk, clk_get_rate(clk)); } +#else +#define rsnd_adg_clk_dbg_info(priv, adg) +#endif int rsnd_adg_probe(struct rsnd_priv *priv) { @@ -595,6 +617,7 @@ int rsnd_adg_probe(struct rsnd_priv *priv) rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); + rsnd_adg_clk_dbg_info(priv, adg); priv->adg = adg; -- 1.9.1

2 1

[alsa-devel] No sound on samsung galaxy book 12 (HDA Intel ALC298)
by Nathanael inkson 13 Oct '17

13 Oct '17

Hi, I have just installed ubuntu 16.04.3 on a samsung galaxy book 12. I have no sound from either speakers or headphones. the intel driver seems to be loaded . I have tried all the tricks of unmuting sound in the alsa mixers to no avail. aplay -l gives: card 0: PCH [HDA Intel PCH], device 0: ALC298 Analog [ALC298 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 Can anyone help? The output of the alsa script is upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.64 !!################################ !!Script ran on: Fri Oct 13 13:41:05 UTC 2017 !!Linux Distribution !!------------------ Ubuntu 16.04.3 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 16.04.3 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 16.04.3 LTS" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" UBUNTU_CODENAME=xenial !!DMI Information !!--------------- Manufacturer: SAMSUNG ELECTRONICS CO., LTD. Product Name: Galaxy Book 12 Product Version: P02HAC Firmware Version: P02HAC.004.170626.WY.1442 Board Vendor: SAMSUNG ELECTRONICS CO., LTD. Board Name: SM-W720NZKBBTU !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/ACPI0003:00/status 31 /sys/bus/acpi/devices/ACPI000C:00/status 15 /sys/bus/acpi/devices/INT33A1:00/status 15 /sys/bus/acpi/devices/INT33D2:00/status 11 /sys/bus/acpi/devices/INT33D3:00/status 15 /sys/bus/acpi/devices/INT33D4:00/status 15 /sys/bus/acpi/devices/INT33D6:00/status 15 /sys/bus/acpi/devices/INT3400:00/status 15 /sys/bus/acpi/devices/INT3403:01/status 15 /sys/bus/acpi/devices/INT3403:02/status 15 /sys/bus/acpi/devices/INT3406:00/status 15 /sys/bus/acpi/devices/INT3446:00/status 15 /sys/bus/acpi/devices/INT3447:00/status 15 /sys/bus/acpi/devices/INT344B:00/status 15 /sys/bus/acpi/devices/INT3472:01/status 15 /sys/bus/acpi/devices/INT3472:02/status 15 /sys/bus/acpi/devices/INT347F:00/status 15 /sys/bus/acpi/devices/INT3F0D:00/status 15 /sys/bus/acpi/devices/MSF0001:00/status 15 /sys/bus/acpi/devices/MSFT0101:00/status 15 /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0C02:03/status 3 /sys/bus/acpi/devices/PNP0C02:05/status 3 /sys/bus/acpi/devices/PNP0C0A:00/status 31 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/SAM0101:00/status 15 /sys/bus/acpi/devices/SAM0201:00/status 15 /sys/bus/acpi/devices/SAM0203:01/status 15 /sys/bus/acpi/devices/SAM0204:00/status 15 /sys/bus/acpi/devices/SAM0603:00/status 15 /sys/bus/acpi/devices/SAM0701:00/status 15 /sys/bus/acpi/devices/SONY258A:00/status 15 /sys/bus/acpi/devices/SSE4752:00/status 15 /sys/bus/acpi/devices/SYNP1234:00/status 15 /sys/bus/acpi/devices/WCOM0027:00/status 15 /sys/bus/acpi/devices/device:10/status 15 /sys/bus/acpi/devices/device:11/status 15 /sys/bus/acpi/devices/device:4f/status 15 /sys/bus/acpi/devices/device:50/status 15 /sys/bus/acpi/devices/device:51/status 15 /sys/bus/acpi/devices/device:52/status 15 /sys/bus/acpi/devices/device:53/status 15 /sys/bus/acpi/devices/device:54/status 15 /sys/bus/acpi/devices/device:55/status 15 /sys/bus/acpi/devices/device:56/status 15 /sys/bus/acpi/devices/device:5b/status 15 /sys/bus/acpi/devices/device:5c/status 15 /sys/bus/acpi/devices/device:5d/status 15 /sys/bus/acpi/devices/device:5e/status 15 /sys/bus/acpi/devices/device:5f/status 15 /sys/bus/acpi/devices/device:60/status 15 /sys/bus/acpi/devices/device:61/status 15 !!Kernel Information !!------------------ Kernel release: 4.10.0-37-generic Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k4.10.0-37-generic Library version: 1.1.0 Utilities version: 1.1.0 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xdf638000 irq 280 !!PCI Soundcards installed in the system !!-------------------------------------- 00:05.0 Multimedia controller: Intel Corporation Sky Lake Imaging Unit (rev 01) 00:14.3 Multimedia controller: Intel Corporation Device 9d32 (rev 01) 00:1f.3 Multimedia audio controller: Intel Corporation Device 9d71 (rev 21) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1f.3 0401: 8086:9d71 (rev 21) Subsystem: 144d:c14f !!Modprobe options (Sound related) !!-------------------------------- snd_pcsp: index=-2 snd_usb_audio: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : -1 !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC298 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0298 Subsystem Id: 0x144dc14f Revision Id: 0x100103 No Modem Function Group found Default PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=8, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC298 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=0 Amp-Out vals: [0x73 0x73] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=0 Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x40]: 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0xc3 0xc3] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC298 Analog", type="Audio", device=0 Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0x6e 0x6e] Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100511: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x40]: 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x25 Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x9d 0x9d] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 4 0x18 0x19 0x1a 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] Connection: 1 0x03 Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0xc3 0xc3] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x24 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x90a601c0: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xc, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x14 [Pin Complex] wcaps 0x40050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010010: OUT EAPD EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 2 0x0c* 0x0d Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x0c* 0x0d 0x06 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x04a11020: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001373c: IN OUT HP EAPD Detect Vref caps: HIZ 50 GRD 80 100 EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 2 0x0c* 0x0d Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40420105: [N/A] SPDIF Out at Ext N/A Conn = 1/4, Color = Unknown DefAssociation = 0x0, Sequence = 0x5 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x1e [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=150 Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000001c: OUT HP Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 2 0x0c* 0x0d Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18 0x19 0x1a 0x1d 0x0b 0x12* Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18 0x19 0x1a* 0x1d 0x0b 0x13 Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x12* 0x13 Node 0x25 [Audio Selector] wcaps 0x300101: Stereo Connection: 1 0x1f Codec: Intel Kabylake HDMI Address: 2 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x8086280b Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1a]: 16 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1a]: 16 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1a]: 16 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 2 Oct 13 14:39 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 8 Oct 13 14:39 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 9 Oct 13 14:39 /dev/snd/hwC0D2 crw-rw----+ 1 root audio 116, 4 Oct 13 14:39 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 3 Oct 13 14:39 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 5 Oct 13 14:39 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 6 Oct 13 14:39 /dev/snd/pcmC0D7p crw-rw----+ 1 root audio 116, 7 Oct 13 14:39 /dev/snd/pcmC0D8p crw-rw----+ 1 root audio 116, 1 Oct 13 14:39 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Oct 13 14:39 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Oct 13 14:39 . drwxr-xr-x 3 root root 260 Oct 13 14:39 .. lrwxrwxrwx 1 root root 12 Oct 13 14:39 pci-0000:00:1f.3 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC298 Analog [ALC298 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC298 Analog [ALC298 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [PCH] Card hw:0 'PCH'/'HDA Intel PCH at 0xdf638000 irq 280' Mixer name : 'Realtek ALC298' Components : 'HDA:10ec0298,144dc14f,00100103 HDA:8086280b,80860101,00100000' Controls : 36 Simple ctrls : 10 Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 115 [91%] [-6.00dB] [on] Front Right: Playback 115 [91%] [-6.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 29 [94%] [9.00dB] [off] Front Right: Playback 29 [94%] [9.00dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 110 [87%] [21.50dB] [on] Front Right: Capture 110 [87%] [21.50dB] [on] Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Master Playback Volume' value.0 115 value.1 115 comment { access 'read write' type INTEGER count 2 range '0 - 127' dbmin -6350 dbmax 0 dbvalue.0 -600 dbvalue.1 -600 } } control.2 { iface MIXER name 'Master Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Loopback Mixing' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.4 { iface MIXER name 'Mic Playback Volume' value.0 29 value.1 29 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 900 dbvalue.1 900 } } control.5 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Capture Volume' value.0 110 value.1 110 comment { access 'read write' type INTEGER count 2 range '0 - 127' dbmin -3350 dbmax 3000 dbvalue.0 2150 dbvalue.1 2150 } } control.7 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.9 { iface MIXER name 'Internal Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.10 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.11 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.12 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.13 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.14 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.15 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.16 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.18 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.19 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.20 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.21 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.22 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.23 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.24 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.25 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.26 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.27 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.28 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.29 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.30 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.31 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.32 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.33 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.34 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.35 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.36 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module ccm rfcomm arc4 bnep snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic binfmt_misc nls_iso8859_1 joydev 8250_dw i2c_designware_platform i2c_designware_core snd_soc_skl snd_soc_skl_ipc snd_soc_sst_ipc snd_soc_sst_dsp snd_hda_ext_core snd_soc_sst_match snd_soc_core snd_compress ac97_bus snd_pcm_dmaengine snd_hda_intel snd_hda_codec snd_hda_core snd_hwdep snd_pcm snd_seq_midi snd_seq_midi_event snd_rawmidi intel_rapl x86_pkg_temp_thermal intel_powerclamp coretemp kvm_intel kvm irqbypass crct10dif_pclmul crc32_pclmul ghash_clmulni_intel pcbc snd_seq snd_seq_device snd_timer ath10k_pci hid_multitouch ath10k_core snd ath aesni_intel mac80211 aes_x86_64 crypto_simd glue_helper cryptd cfg80211 soundcore input_leds serio_raw btusb btrtl btbcm btintel shpchp bluetooth idma64 mei_me virt_dma mei processor_thermal_device intel_pch_thermal intel_soc_dts_iosf intel_lpss_pci intel_vbtn int3403_thermal soc_button_array int340x_thermal_zone mac_hid intel_lpss_acpi intel_lpss int3406_thermal tpm_crb dptf_power int3400_thermal acpi_thermal_rel intel_hid sparse_keymap acpi_pad parport_pc ppdev lp parport autofs4 mmc_block hid_generic usbhid i915 i2c_algo_bit drm_kms_helper syscopyarea sysfillrect sysimgblt fb_sys_fops sdhci_pci drm sdhci ahci libahci video i2c_hid hid wmi pinctrl_sunrisepoint pinctrl_intel fjes !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x12 0x90a601c0 0x13 0x411111f0 0x14 0x411111f0 0x17 0x90170110 0x18 0x04a11020 0x19 0x411111f0 0x1a 0x411111f0 0x1d 0x40420105 0x1e 0x411111f0 0x1f 0x411111f0 0x21 0x411111f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC0D2/init_pin_configs: 0x05 0x18560010 0x06 0x18560010 0x07 0x18560010 /sys/class/sound/hwC0D2/driver_pin_configs: /sys/class/sound/hwC0D2/user_pin_configs: /sys/class/sound/hwC0D2/init_verbs: /sys/class/sound/hwC0D2/hints: !!ALSA/HDA dmesg !!-------------- [ 2.431375] intel_rapl: Found RAPL domain dram [ 2.462483] snd_hda_intel 0000:00:1f.3: enabling device (0000 -> 0002) [ 2.570682] input: SYNP1234:00 06CB:5810 as /devices/pci0000:00/INT3446:00/i2c_designware.0/i2c-6/i2c-SYNP1234:00/0018:06CB:5810.0004/input/input12 -- [ 2.855095] hid-generic 0018:2D1F:000B.0005: input,hidraw4: I2C HID v1.00 Mouse [WCOM0027:00 2D1F:000B] on i2c-WCOM0027:00 [ 3.588848] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 3.613596] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC298: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 3.613598] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 3.613599] snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 3.613600] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 3.613600] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 3.613601] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 3.613602] snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 [ 3.626328] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input18 [ 3.626383] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input19 [ 3.626430] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 [ 3.626479] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 [ 3.749827] audit: type=1400 audit(1507901955.077:2): apparmor="STATUS" operation="profile_load" profile="unconfined" name="/sbin/dhclient" pid=737 comm="apparmor_parser"

1 0

[alsa-devel] [PATCH] ALSA: hda - silence uninitialized variable warning in activate_amp_in()
by Dan Carpenter 13 Oct '17

13 Oct '17

If snd_hda_get_conn_list() fails then "conn" isn't initialized. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 28e265a88383..5cc65093d941 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -795,6 +795,8 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, hda_nid_t nid = path->path[i]; nums = snd_hda_get_conn_list(codec, nid, &conn); + if (nums < 0) + return; type = get_wcaps_type(get_wcaps(codec, nid)); if (type == AC_WID_PIN || (type == AC_WID_AUD_IN && codec->single_adc_amp)) {

2 1

[alsa-devel] [PATCH v2] ALSA: hda/ca0132 - use ARRAY_SIZE
by Jérémy Lefaure 13 Oct '17

13 Oct '17

Using the ARRAY_SIZE macro improves the readability of the code. Found with Coccinelle with the following semantic patch: @r depends on (org || report)@ type T; T[] E; position p; @@ ( (sizeof(E)@p /sizeof(*E)) | (sizeof(E)@p /sizeof(E[...])) | (sizeof(E)@p /sizeof(T)) ) Signed-off-by: Jérémy Lefaure <jeremy.lefaure(a)lse.epita.fr> --- Changes since v1: - keep sound/oss untouched sound/pci/hda/patch_ca0132.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 3e73d5c6ccfc..768ea8651993 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -27,6 +27,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> +#include <linux/kernel.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -3605,8 +3606,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, static int ca0132_voicefx_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - unsigned int items = sizeof(ca0132_voicefx_presets) - / sizeof(struct ct_voicefx_preset); + unsigned int items = ARRAY_SIZE(ca0132_voicefx_presets); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -3635,10 +3635,8 @@ static int ca0132_voicefx_put(struct snd_kcontrol *kcontrol, struct ca0132_spec *spec = codec->spec; int i, err = 0; int sel = ucontrol->value.enumerated.item[0]; - unsigned int items = sizeof(ca0132_voicefx_presets) - / sizeof(struct ct_voicefx_preset); - if (sel >= items) + if (sel >= ARRAY_SIZE(ca0132_voicefx_presets)) return 0; codec_dbg(codec, "ca0132_voicefx_put: sel=%d, preset=%s\n", -- 2.14.2

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: add support for CX2072X CODEC
by simon.ho.cnxt＠gmail.com 13 Oct '17

13 Oct '17

From: Simon Ho <simon.ho(a)conexant.com> Hi Takashi, Mark I corrected the driver based on your input as below. Please help review the code. Mark: - Corrected the coding style. - EQ/DRC settings can be configured from userspace via mixer controls. - Wrongly cast void pointers. - Print I2C error code. - Use memcmp() to check if the new EQ value is different. - Removed unused TDM code. - Removed noisy "out of memory" message. - Corrected ACPI and I2C match table. - Removed unnessary CONFIG_ACPI define. Takashi: - Fixed printf int argument. - Fixed unexpected error code at no MCLK fallback. Thanks, Simon Simon Ho (2): ASoC: cx2072x: Add DT bingings documentation for CX2072X CODEC ASoC: add support for Conexant CX2072X CODEC .../devicetree/bindings/sound/cx2072x.txt | 36 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2253 ++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 319 +++ 5 files changed, 2615 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cx2072x.txt create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h -- 2.7.4

4 10

[alsa-devel] [PATCH] ASoC: max98925: Return proper error if revision mismatch
by Axel Lin 13 Oct '17

13 Oct '17

Return proper error instead of 0 if the revision does not match. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98925.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 327eaa25c9bd..921f95fc396d 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -579,7 +579,7 @@ static int max98925_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(max98925->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); - goto err_out; + return ret; } if (!of_property_read_u32(i2c->dev.of_node, "vmon-slot-no", &value)) { @@ -596,16 +596,20 @@ static int max98925_i2c_probe(struct i2c_client *i2c, } max98925->i_slot = value; } - ret = regmap_read(max98925->regmap, - MAX98925_REV_VERSION, &reg); - if ((ret < 0) || - ((reg != MAX98925_VERSION) && - (reg != MAX98925_VERSION1))) { - dev_err(&i2c->dev, - "device initialization error (%d 0x%02X)\n", + + ret = regmap_read(max98925->regmap, MAX98925_REV_VERSION, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Read revision failed\n"); + return ret; + } + + if ((reg != MAX98925_VERSION) && (reg != MAX98925_VERSION1)) { + ret = -ENODEV; + dev_err(&i2c->dev, "Invalid revision (%d 0x%02X)\n", ret, reg); - goto err_out; + return ret; } + dev_info(&i2c->dev, "device version 0x%02X\n", reg); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98925, @@ -613,7 +617,6 @@ static int max98925_i2c_probe(struct i2c_client *i2c, if (ret < 0) dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); -err_out: return ret; } -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: cht_bsw_max98090_ti: Fix NULL pointer dereference while accessing jack
by Guenter Roeck 12 Oct '17

12 Oct '17

From: Guenter Roeck <groeck(a)chromium.org> Commit f2ed6b07645e ("ASoC: Make aux_dev more like a generic component") caused a regression on this driver, since now a kernel oops is seen when the driver is loaded, or more specifically when ts3a227e_enable_jack_detect() is called. That commit changed the probing of aux_devs before checking new DAI links, so cht_max98090_headset_init is now called before cht_codec_init. The kernel crashes due a NULL pointer dereference in cht_max98090_headset_init since there is a call that tries to access the jack pointer which has not been allocated yet. This patch moves the new jack object creation from cht_codec_init to cht_max98090_headset_init, making sure the jack is created before is accessed. Also see upstream commit 5f22449344d9 ("ASoC: rockchip-max98090: Fix NULL pointer dereference while accessing to jack.") for a similar fix in the rockchip-max98090 driver. Fixes: f2ed6b07645e ("ASoC: Make aux_dev more like a generic component") Cc: Mengdong Lin <mengdong.lin(a)linux.intel.com> Signed-off-by: Guenter Roeck <groeck(a)chromium.org> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 55 ++++++++++++++-------------- 1 file changed, 27 insertions(+), 28 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 20755ecc7f9e..c1518db7bb61 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -111,37 +111,12 @@ static struct notifier_block cht_jack_nb = { static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { - int ret; - int jack_type; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); - struct snd_soc_jack *jack = &ctx->jack; - - /** - * TI supports 4 butons headset detection - * KEY_MEDIA - * KEY_VOICECOMMAND - * KEY_VOLUMEUP - * KEY_VOLUMEDOWN - */ - if (ctx->ts3a227e_present) - jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3; - else - jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; - - ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", - jack_type, jack, NULL, 0); - - if (ret) { - dev_err(runtime->dev, "Headset Jack creation failed %d\n", ret); - return ret; - } if (ctx->ts3a227e_present) - snd_soc_jack_notifier_register(jack, &cht_jack_nb); + snd_soc_jack_notifier_register(&ctx->jack, &cht_jack_nb); - return ret; + return 0; } static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, @@ -188,8 +163,32 @@ static int cht_max98090_headset_init(struct snd_soc_component *component) { struct snd_soc_card *card = component->card; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_jack *jack = &ctx->jack; + int jack_type; + int ret; + + /* + * TI supports 4 butons headset detection + * KEY_MEDIA + * KEY_VOICECOMMAND + * KEY_VOLUMEUP + * KEY_VOLUMEDOWN + */ + if (ctx->ts3a227e_present) + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + else + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; + + ret = snd_soc_card_jack_new(card, "Headset Jack", jack_type, jack, + NULL, 0); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed %d\n", ret); + return ret; + } - return ts3a227e_enable_jack_detect(component, &ctx->jack); + return ts3a227e_enable_jack_detect(component, jack); } static const struct snd_soc_ops cht_aif1_ops = { -- 2.7.4

2 1

[alsa-devel] regression introduced by commit 70d435ba1cd6 ("ASoC: imx-pcm-dma: simplify pcm_config")
by Lothar Waßmann 12 Oct '17

12 Oct '17

Hi, the commit 70d435ba1cd6 ("ASoC: imx-pcm-dma: simplify pcm_config") introduced a regression on the Ka-Ro TX series modules (and possibly other hardware as well). The problem is, that this commit restricts the supported sound formats to S24_LE only rather than S16_LE, S24_LE, S20_3LE without the patch. Therefore aplay fails with the following error messages, when trying to play a sound file with S16_LE format: |ALSA lib alsa-lib-1.1.2/src/pcm/pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable |aplay: set_params:1204: Broken configuration for this PCM: no configurations available This is with the simple-sound-card driver as well as the imx-sgtl5000 driver. Can someone tell me on what hardware and with what software has this patch been tested? Lothar Waßmann

3 3

[alsa-devel] [PATCH] ASoC: rsnd: add rsnd_kctrl_xxx() macro
by Kuninori Morimoto 12 Oct '17

12 Oct '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current CTU/MIX/DVC are directly using rsnd_kctrl_cfg_m/s to control val etc, but it is difficult to read/understand. And there was no uniformity in access method. This patch adds new rsnd_kctrl_xxx() and implements uniformed access method. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ctu.c | 80 ++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/dvc.c | 19 ++++++------ sound/soc/sh/rcar/mix.c | 10 +++--- sound/soc/sh/rcar/rsnd.h | 4 +++ 4 files changed, 59 insertions(+), 54 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index ad07ba8..d201d55 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -133,7 +133,7 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, int i; for (i = 0; i < RSND_MAX_CHANNELS; i++) { - u32 val = ctu->pass.val[i]; + u32 val = rsnd_kctrl_valm(ctu->pass, i); cpmdr |= val << (28 - (i * 4)); @@ -150,44 +150,44 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, rsnd_mod_write(mod, CTU_SCMDR, scmdr); if (scmdr > 0) { - rsnd_mod_write(mod, CTU_SV00R, ctu->sv0.val[0]); - rsnd_mod_write(mod, CTU_SV01R, ctu->sv0.val[1]); - rsnd_mod_write(mod, CTU_SV02R, ctu->sv0.val[2]); - rsnd_mod_write(mod, CTU_SV03R, ctu->sv0.val[3]); - rsnd_mod_write(mod, CTU_SV04R, ctu->sv0.val[4]); - rsnd_mod_write(mod, CTU_SV05R, ctu->sv0.val[5]); - rsnd_mod_write(mod, CTU_SV06R, ctu->sv0.val[6]); - rsnd_mod_write(mod, CTU_SV07R, ctu->sv0.val[7]); + rsnd_mod_write(mod, CTU_SV00R, rsnd_kctrl_valm(ctu->sv0, 0)); + rsnd_mod_write(mod, CTU_SV01R, rsnd_kctrl_valm(ctu->sv0, 1)); + rsnd_mod_write(mod, CTU_SV02R, rsnd_kctrl_valm(ctu->sv0, 2)); + rsnd_mod_write(mod, CTU_SV03R, rsnd_kctrl_valm(ctu->sv0, 3)); + rsnd_mod_write(mod, CTU_SV04R, rsnd_kctrl_valm(ctu->sv0, 4)); + rsnd_mod_write(mod, CTU_SV05R, rsnd_kctrl_valm(ctu->sv0, 5)); + rsnd_mod_write(mod, CTU_SV06R, rsnd_kctrl_valm(ctu->sv0, 6)); + rsnd_mod_write(mod, CTU_SV07R, rsnd_kctrl_valm(ctu->sv0, 7)); } if (scmdr > 1) { - rsnd_mod_write(mod, CTU_SV10R, ctu->sv1.val[0]); - rsnd_mod_write(mod, CTU_SV11R, ctu->sv1.val[1]); - rsnd_mod_write(mod, CTU_SV12R, ctu->sv1.val[2]); - rsnd_mod_write(mod, CTU_SV13R, ctu->sv1.val[3]); - rsnd_mod_write(mod, CTU_SV14R, ctu->sv1.val[4]); - rsnd_mod_write(mod, CTU_SV15R, ctu->sv1.val[5]); - rsnd_mod_write(mod, CTU_SV16R, ctu->sv1.val[6]); - rsnd_mod_write(mod, CTU_SV17R, ctu->sv1.val[7]); + rsnd_mod_write(mod, CTU_SV10R, rsnd_kctrl_valm(ctu->sv1, 0)); + rsnd_mod_write(mod, CTU_SV11R, rsnd_kctrl_valm(ctu->sv1, 1)); + rsnd_mod_write(mod, CTU_SV12R, rsnd_kctrl_valm(ctu->sv1, 2)); + rsnd_mod_write(mod, CTU_SV13R, rsnd_kctrl_valm(ctu->sv1, 3)); + rsnd_mod_write(mod, CTU_SV14R, rsnd_kctrl_valm(ctu->sv1, 4)); + rsnd_mod_write(mod, CTU_SV15R, rsnd_kctrl_valm(ctu->sv1, 5)); + rsnd_mod_write(mod, CTU_SV16R, rsnd_kctrl_valm(ctu->sv1, 6)); + rsnd_mod_write(mod, CTU_SV17R, rsnd_kctrl_valm(ctu->sv1, 7)); } if (scmdr > 2) { - rsnd_mod_write(mod, CTU_SV20R, ctu->sv2.val[0]); - rsnd_mod_write(mod, CTU_SV21R, ctu->sv2.val[1]); - rsnd_mod_write(mod, CTU_SV22R, ctu->sv2.val[2]); - rsnd_mod_write(mod, CTU_SV23R, ctu->sv2.val[3]); - rsnd_mod_write(mod, CTU_SV24R, ctu->sv2.val[4]); - rsnd_mod_write(mod, CTU_SV25R, ctu->sv2.val[5]); - rsnd_mod_write(mod, CTU_SV26R, ctu->sv2.val[6]); - rsnd_mod_write(mod, CTU_SV27R, ctu->sv2.val[7]); + rsnd_mod_write(mod, CTU_SV20R, rsnd_kctrl_valm(ctu->sv2, 0)); + rsnd_mod_write(mod, CTU_SV21R, rsnd_kctrl_valm(ctu->sv2, 1)); + rsnd_mod_write(mod, CTU_SV22R, rsnd_kctrl_valm(ctu->sv2, 2)); + rsnd_mod_write(mod, CTU_SV23R, rsnd_kctrl_valm(ctu->sv2, 3)); + rsnd_mod_write(mod, CTU_SV24R, rsnd_kctrl_valm(ctu->sv2, 4)); + rsnd_mod_write(mod, CTU_SV25R, rsnd_kctrl_valm(ctu->sv2, 5)); + rsnd_mod_write(mod, CTU_SV26R, rsnd_kctrl_valm(ctu->sv2, 6)); + rsnd_mod_write(mod, CTU_SV27R, rsnd_kctrl_valm(ctu->sv2, 7)); } if (scmdr > 3) { - rsnd_mod_write(mod, CTU_SV30R, ctu->sv3.val[0]); - rsnd_mod_write(mod, CTU_SV31R, ctu->sv3.val[1]); - rsnd_mod_write(mod, CTU_SV32R, ctu->sv3.val[2]); - rsnd_mod_write(mod, CTU_SV33R, ctu->sv3.val[3]); - rsnd_mod_write(mod, CTU_SV34R, ctu->sv3.val[4]); - rsnd_mod_write(mod, CTU_SV35R, ctu->sv3.val[5]); - rsnd_mod_write(mod, CTU_SV36R, ctu->sv3.val[6]); - rsnd_mod_write(mod, CTU_SV37R, ctu->sv3.val[7]); + rsnd_mod_write(mod, CTU_SV30R, rsnd_kctrl_valm(ctu->sv3, 0)); + rsnd_mod_write(mod, CTU_SV31R, rsnd_kctrl_valm(ctu->sv3, 1)); + rsnd_mod_write(mod, CTU_SV32R, rsnd_kctrl_valm(ctu->sv3, 2)); + rsnd_mod_write(mod, CTU_SV33R, rsnd_kctrl_valm(ctu->sv3, 3)); + rsnd_mod_write(mod, CTU_SV34R, rsnd_kctrl_valm(ctu->sv3, 4)); + rsnd_mod_write(mod, CTU_SV35R, rsnd_kctrl_valm(ctu->sv3, 5)); + rsnd_mod_write(mod, CTU_SV36R, rsnd_kctrl_valm(ctu->sv3, 6)); + rsnd_mod_write(mod, CTU_SV37R, rsnd_kctrl_valm(ctu->sv3, 7)); } rsnd_mod_write(mod, CTU_CTUIR, 0); @@ -199,17 +199,17 @@ static void rsnd_ctu_value_reset(struct rsnd_dai_stream *io, struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); int i; - if (!ctu->reset.val) + if (!rsnd_kctrl_vals(ctu->reset)) return; for (i = 0; i < RSND_MAX_CHANNELS; i++) { - ctu->pass.val[i] = 0; - ctu->sv0.val[i] = 0; - ctu->sv1.val[i] = 0; - ctu->sv2.val[i] = 0; - ctu->sv3.val[i] = 0; + rsnd_kctrl_valm(ctu->pass, i) = 0; + rsnd_kctrl_valm(ctu->sv0, i) = 0; + rsnd_kctrl_valm(ctu->sv1, i) = 0; + rsnd_kctrl_valm(ctu->sv2, i) = 0; + rsnd_kctrl_valm(ctu->sv3, i) = 0; } - ctu->reset.val = 0; + rsnd_kctrl_vals(ctu->reset) = 0; } static int rsnd_ctu_init(struct rsnd_mod *mod, diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 4ef318a..dbe54f0 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -73,8 +73,9 @@ static void rsnd_dvc_halt(struct rsnd_mod *mod) rsnd_mod_write(mod, DVC_SWRSR, 0); } -#define rsnd_dvc_get_vrpdr(dvc) (dvc->rup.val << 8 | dvc->rdown.val) -#define rsnd_dvc_get_vrdbr(dvc) (0x3ff - (dvc->volume.val[0] >> 13)) +#define rsnd_dvc_get_vrpdr(dvc) (rsnd_kctrl_vals(dvc->rup) << 8 | \ + rsnd_kctrl_vals(dvc->rdown)) +#define rsnd_dvc_get_vrdbr(dvc) (0x3ff - (rsnd_kctrl_valm(dvc->volume, 0) >> 13)) static void rsnd_dvc_volume_parameter(struct rsnd_dai_stream *io, struct rsnd_mod *mod) @@ -84,12 +85,12 @@ static void rsnd_dvc_volume_parameter(struct rsnd_dai_stream *io, int i; /* Enable Ramp */ - if (dvc->ren.val) + if (rsnd_kctrl_vals(dvc->ren)) for (i = 0; i < RSND_MAX_CHANNELS; i++) - val[i] = dvc->volume.cfg.max; + val[i] = rsnd_kctrl_max(dvc->volume); else for (i = 0; i < RSND_MAX_CHANNELS; i++) - val[i] = dvc->volume.val[i]; + val[i] = rsnd_kctrl_valm(dvc->volume, i); /* Enable Digital Volume */ rsnd_mod_write(mod, DVC_VOL0R, val[0]); @@ -119,7 +120,7 @@ static void rsnd_dvc_volume_init(struct rsnd_dai_stream *io, dvucr |= 0x101; /* Enable Ramp */ - if (dvc->ren.val) { + if (rsnd_kctrl_vals(dvc->ren)) { dvucr |= 0x10; /* @@ -161,10 +162,10 @@ static void rsnd_dvc_volume_update(struct rsnd_dai_stream *io, u32 vrdbr = 0; int i; - for (i = 0; i < dvc->mute.cfg.size; i++) - zcmcr |= (!!dvc->mute.cfg.val[i]) << i; + for (i = 0; i < rsnd_kctrl_size(dvc->mute); i++) + zcmcr |= (!!rsnd_kctrl_valm(dvc->mute, i)) << i; - if (dvc->ren.val) { + if (rsnd_kctrl_vals(dvc->ren)) { vrpdr = rsnd_dvc_get_vrpdr(dvc); vrdbr = rsnd_dvc_get_vrdbr(dvc); } diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 912cfec..7998380 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -84,7 +84,7 @@ static void rsnd_mix_halt(struct rsnd_mod *mod) #define rsnd_mix_get_vol(mix, X) \ rsnd_flags_has(mix, HAS_VOL##X) ? \ - (VOL_MAX - mix->volume##X.cfg.val[0]) : 0 + (VOL_MAX - rsnd_kctrl_vals(mix->volume##X)) : 0 static void rsnd_mix_volume_parameter(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { @@ -116,9 +116,9 @@ static void rsnd_mix_volume_init(struct rsnd_dai_stream *io, rsnd_mod_write(mod, MIX_ADINR, rsnd_runtime_channel_after_ctu(io)); /* volume step */ - rsnd_mod_write(mod, MIX_MIXMR, mix->ren.cfg.val[0]); - rsnd_mod_write(mod, MIX_MVPDR, mix->rup.cfg.val[0] << 8 | - mix->rdw.cfg.val[0]); + rsnd_mod_write(mod, MIX_MIXMR, rsnd_kctrl_vals(mix->ren)); + rsnd_mod_write(mod, MIX_MVPDR, rsnd_kctrl_vals(mix->rup) << 8 | + rsnd_kctrl_vals(mix->rdw)); /* common volume parameter */ rsnd_mix_volume_parameter(io, mod); @@ -217,7 +217,7 @@ static int rsnd_mix_pcm_new(struct rsnd_mod *mod, volume, VOL_MAX); if (ret < 0) return ret; - volume->cfg.val[0] = VOL_MAX; + rsnd_kctrl_vals(*volume) = VOL_MAX; if (rsnd_flags_has(mix, ONCE_KCTRL_INITIALIZED)) return ret; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0d9f9c3..2a224fa 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -631,6 +631,10 @@ struct rsnd_kctrl_cfg_s { struct rsnd_kctrl_cfg cfg; u32 val; }; +#define rsnd_kctrl_size(x) ((x).cfg.size) +#define rsnd_kctrl_max(x) ((x).cfg.max) +#define rsnd_kctrl_valm(x, i) ((x).val[i]) /* = (x).cfg.val[i] */ +#define rsnd_kctrl_vals(x) ((x).val) /* = (x).cfg.val[0] */ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); -- 1.9.1

2 1

[alsa-devel] Pulseaudio say me to report a issue.
by Diniz Bortolotto 12 Oct '17

12 Oct '17

Hi buddies! My name is Diniz and I'm new here in alsa-devel list. I just entered in the list because pulseaudio reported a error in my syslog. I'm running a Slackware 64bits current machine and receive the messages below: pulseaudio[1938]: [alsa-sink-ALC887-VD Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. pulseaudio[1938]: [alsa-sink-ALC887-VD Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver "snd_hda_intel". Please report this issue to the ALSA developers. pulseaudio[1938]: [alsa-sink-ALC887-VD Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. I don't know if that is a big deal. Sorry if it is not. I only wanna help. Sorry my poor english. :-) Best Regards, Diniz Bortolotto

1 0

[alsa-devel] Click sound when playing "silence" buffer
by Владимир Мартьянов 11 Oct '17

11 Oct '17

Hello! I'm completely new to ALSA and I tried to do a simply buzzer using it. During the debug I noticed a strange behaviour: I used SND_PCM_FORMAT_S8 format, tried to play the buffer filled with zero bytes and I heard click sound when my program starts and end playing the buffer. When I filled the buffer with 0x80 bytes I didn't hear any clicks. The closer the bytes value to 0x80, the quieter the click is. I tried snd_pcm_format_set_silence() instead of memset() and got clicks too. Why I got them? Does it mean what when I use some formats, the values in buffer for zero output voltage are not 0x00? My source code is in attachment, my sound card is Intel HDA.

1 0

[alsa-devel] [PATCH] ALSA: seq: Fix use-after-free at creating a port
by Takashi Iwai 11 Oct '17

11 Oct '17

There is a potential race window opened at creating and deleting a port via ioctl, as spotted by fuzzing. snd_seq_create_port() creates a port object and returns its pointer, but it doesn't take the refcount, thus it can be deleted immediately by another thread. Meanwhile, snd_seq_ioctl_create_port() still calls the function snd_seq_system_client_ev_port_start() with the created port object that is being deleted, and this triggers use-after-free like: BUG: KASAN: use-after-free in snd_seq_ioctl_create_port+0x504/0x630 [snd_seq] at addr ffff8801f2241cb1 ============================================================================= BUG kmalloc-512 (Tainted: G B ): kasan: bad access detected ----------------------------------------------------------------------------- INFO: Allocated in snd_seq_create_port+0x94/0x9b0 [snd_seq] age=1 cpu=3 pid=4511 ___slab_alloc+0x425/0x460 __slab_alloc+0x20/0x40 kmem_cache_alloc_trace+0x150/0x190 snd_seq_create_port+0x94/0x9b0 [snd_seq] snd_seq_ioctl_create_port+0xd1/0x630 [snd_seq] snd_seq_do_ioctl+0x11c/0x190 [snd_seq] snd_seq_ioctl+0x40/0x80 [snd_seq] do_vfs_ioctl+0x54b/0xda0 SyS_ioctl+0x79/0x90 entry_SYSCALL_64_fastpath+0x16/0x75 INFO: Freed in port_delete+0x136/0x1a0 [snd_seq] age=1 cpu=2 pid=4717 __slab_free+0x204/0x310 kfree+0x15f/0x180 port_delete+0x136/0x1a0 [snd_seq] snd_seq_delete_port+0x235/0x350 [snd_seq] snd_seq_ioctl_delete_port+0xc8/0x180 [snd_seq] snd_seq_do_ioctl+0x11c/0x190 [snd_seq] snd_seq_ioctl+0x40/0x80 [snd_seq] do_vfs_ioctl+0x54b/0xda0 SyS_ioctl+0x79/0x90 entry_SYSCALL_64_fastpath+0x16/0x75 Call Trace: [<ffffffff81b03781>] dump_stack+0x63/0x82 [<ffffffff81531b3b>] print_trailer+0xfb/0x160 [<ffffffff81536db4>] object_err+0x34/0x40 [<ffffffff815392d3>] kasan_report.part.2+0x223/0x520 [<ffffffffa07aadf4>] ? snd_seq_ioctl_create_port+0x504/0x630 [snd_seq] [<ffffffff815395fe>] __asan_report_load1_noabort+0x2e/0x30 [<ffffffffa07aadf4>] snd_seq_ioctl_create_port+0x504/0x630 [snd_seq] [<ffffffffa07aa8f0>] ? snd_seq_ioctl_delete_port+0x180/0x180 [snd_seq] [<ffffffff8136be50>] ? taskstats_exit+0xbc0/0xbc0 [<ffffffffa07abc5c>] snd_seq_do_ioctl+0x11c/0x190 [snd_seq] [<ffffffffa07abd10>] snd_seq_ioctl+0x40/0x80 [snd_seq] [<ffffffff8136d433>] ? acct_account_cputime+0x63/0x80 [<ffffffff815b515b>] do_vfs_ioctl+0x54b/0xda0 ..... We may fix this in a few different ways, and in this patch, it's fixed simply by taking the refcount properly at snd_seq_create_port() and letting the caller unref the object after use. Also, there is another potential use-after-free by sprintf() call in snd_seq_create_port(), and this is moved inside the lock. This fix covers CVE-2017-15265. Reported-and-tested-by: Michael23 Yu <ycqzsy(a)gmail.com> Suggested-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 6 +++++- sound/core/seq/seq_ports.c | 7 +++++-- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index ea2d0ae85bd3..6c9cba2166d9 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1259,6 +1259,7 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg) struct snd_seq_port_info *info = arg; struct snd_seq_client_port *port; struct snd_seq_port_callback *callback; + int port_idx; /* it is not allowed to create the port for an another client */ if (info->addr.client != client->number) @@ -1269,7 +1270,9 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg) return -ENOMEM; if (client->type == USER_CLIENT && info->kernel) { - snd_seq_delete_port(client, port->addr.port); + port_idx = port->addr.port; + snd_seq_port_unlock(port); + snd_seq_delete_port(client, port_idx); return -EINVAL; } if (client->type == KERNEL_CLIENT) { @@ -1290,6 +1293,7 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg) snd_seq_set_port_info(port, info); snd_seq_system_client_ev_port_start(port->addr.client, port->addr.port); + snd_seq_port_unlock(port); return 0; } diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 0a7020c82bfc..d21ece9f8d73 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -122,7 +122,9 @@ static void port_subs_info_init(struct snd_seq_port_subs_info *grp) } -/* create a port, port number is returned (-1 on failure) */ +/* create a port, port number is returned (-1 on failure); + * the caller needs to unref the port via snd_seq_port_unlock() appropriately + */ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, int port) { @@ -151,6 +153,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, snd_use_lock_init(&new_port->use_lock); port_subs_info_init(&new_port->c_src); port_subs_info_init(&new_port->c_dest); + snd_use_lock_use(&new_port->use_lock); num = port >= 0 ? port : 0; mutex_lock(&client->ports_mutex); @@ -165,9 +168,9 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, list_add_tail(&new_port->list, &p->list); client->num_ports++; new_port->addr.port = num; /* store the port number in the port */ + sprintf(new_port->name, "port-%d", num); write_unlock_irqrestore(&client->ports_lock, flags); mutex_unlock(&client->ports_mutex); - sprintf(new_port->name, "port-%d", num); return new_port; } -- 2.14.2

2 2

[alsa-devel] PCM2902: click/pop sound when it should be silent.
by Theodor van Nahl 11 Oct '17

11 Oct '17

Hello, I do have a Behringer U-PHORIA UMC22 USB soundcard which uses the PCM2902 soundchip. I am using that device with pulseaudio on top and the trouble occurs when I do not use the device. Problem ======= In various intervals, but only when I don't play audio, I do hear a pop or click sound comparable to the sound a power amp does when connecting or disconnecting the power. My Analysis =========== I did watch the usb port using wireshark and the following frame has been the last frame before that sound: Frame 3470: 64 bytes on wire (512 bits), 64 bytes captured (512 bits) on interface 0 USB URB [Source: host] [Destination: 2.4.0] URB id: 0xffff9522ee93f9c0 URB type: URB_SUBMIT ('S') URB transfer type: URB_CONTROL (0x02) Endpoint: 0x00, Direction: OUT Device: 4 URB bus id: 2 Device setup request: relevant (0) Data: present (0) URB sec: 1507602131 URB usec: 34054 URB status: Operation now in progress (-EINPROGRESS) (-115) URB length [bytes]: 0 Data length [bytes]: 0 [Response in: 3471] Interval: 0 Start frame: 0 Copy of Transfer Flags: 0x00000000 Number of ISO descriptors: 0 URB setup bmRequestType: 0x01 0... .... = Direction: Host-to-device .00. .... = Type: Standard (0x0) ...0 0001 = Recipient: Interface (0x01) bRequest: SET INTERFACE (11) bAlternateSetting: 0 wInterface: 1 wLength: 0 From what I understand from the PCM2902 Manual [0] on page 22 is that the streaming interface gets deactivated (zero bandwith setting) by this frame. If I read the documentation of the PCM2902 correctly that step is not nesessary, the chip suspends itself when no data comes in (p. 26). Reading *linux/sound/usb/endpoint.c* I am convinced that the frame above and its counter part are send by this code and send everytime a first endpoint (pulseaudio, aplay, …) connects and the last endpoint disconnected. Due to that I do believe that this is a bug in the driver used for the pcm2902 chip. References ========== That bug is not unheard of as you can see in [1] and [2] but the workarounds provided in those links I cannot confirm to work. I hope that my analysis is as conclusive for you as they sound to me. From this point I am unsure where to head next and was hoping for help here. So my followup questions are: * What do you believe where and "why there" this should be fixed? * Why is the frame above beeing send to the device? The soundcard does seem to have power after that frame. * Isn't it possible to never set the pcm2902 into "zero bandwith mode"? Greetings, Theodor [0]: http://www.ti.com/product/pcm2902 [1]: https://ubuntuforums.org/showthread.php?t=2292143&p=13345100#post13345100 [2]: https://www.reddit.com/r/linux/comments/3idb1h/clicky_noise_before_and_afte…

1 0

[alsa-devel] [PATCH v2 0/9] sound: Add sanity checks for invalid EPs
by Takashi Iwai 11 Oct '17

11 Oct '17

Hi, this is a revised patch set to cover the codes that may submit URBs containing invalid EPs without validation, which result in the kernel warning from the USB core. The first patch adds a new helper for simplifying the EP check, and the rest applies it at needed places. The original issues were spotted by syzkaller, and I put a few others for covering more similar cases. The only updated code in v2 is about the first patch to add usb_urb_ep_type_check() helper (in addition to tested-by tag from Andrey). Typos were fixed and it's called also from usb_submit_urb() as Greg suggested, too. USB devs: does this look OK now? If yes, and if I get acks, I can take the patches to sound tree. Or I don't mind if you take all to usb tree, too. Or maybe better, I'll prepare an immutable branch based on 4.4.14-rc (rc4 for now) in case you want to pull into both trees. Let me know your wish. thanks, Takashi === Takashi Iwai (9): usb: core: Add a helper function to check the validity of EP type in URB ALSA: bcd2000: Add a sanity check for invalid EPs ALSA: caiaq: Add a sanity check for invalid EPs ALSA: line6: Add a sanity check for invalid EPs ALSA: usb-audio: Add sanity checks for invalid EPs ALSA: usx2y: Add sanity checks for invalid EPs ALSA: hiface: Add sanity checks for invalid EPs ALSA: caiaq: Add yet more sanity checks for invalid EPs ALSA: line6: Add yet more sanity checks for invalid EPs drivers/usb/core/urb.c | 30 ++++++++++++++++++++++++++---- include/linux/usb.h | 2 ++ sound/usb/bcd2000/bcd2000.c | 7 +++++++ sound/usb/caiaq/device.c | 7 +++++++ sound/usb/caiaq/input.c | 9 +++++++++ sound/usb/hiface/pcm.c | 9 +++++++-- sound/usb/line6/driver.c | 30 ++++++++++++++++++++++-------- sound/usb/line6/midi.c | 17 +++++++++++------ sound/usb/midi.c | 38 ++++++++++++++++++++++++++++++-------- sound/usb/usx2y/usbusx2y.c | 8 ++++++++ sound/usb/usx2y/usbusx2yaudio.c | 3 +++ 11 files changed, 132 insertions(+), 28 deletions(-) -- 2.14.2

3 26

[alsa-devel] [PATCH 0/3] ALSA: add snd_card_disconnect_sync()
by Kuninori Morimoto 11 Oct '17

11 Oct '17

Hi Takashi-san, Mark Current ALSA (SoC) might be kernel panic if user unbinds sound driver during playback. > aplay xxx.wav & > echo xxxx > /sys/bus/platform/drivers/xxxx/unbind Main issue is we *can't* skip unbind with return -Exxx, because unbind operation doesn't checks return value from each driver. Thus, we *must* stop driver immediately when unbind. 1) is for ALSA 2) is for ALSA SoC 3) is for Renesas sound driver. I think almost all driver want to have 3) type patch on .remove but this patch-set doesn't care at this point. Please let me know if they should have it. Takashi Iwai (1): 1) ALSA: add snd_card_disconnect_sync() Kuninori Morimoto (2): 2) ASoC: add snd_soc_card_disconnect_sync() 3) ASoC: rsnd: call snd_soc_card_disconnect_sync() when remove include/sound/core.h | 2 ++ include/sound/soc.h | 2 ++ sound/core/init.c | 18 ++++++++++++++++++ sound/core/pcm.c | 4 ++++ sound/soc/sh/rcar/core.c | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ 6 files changed, 38 insertions(+) -- 1.9.1 Best regards --- Kuninori Morimoto

3 8

[alsa-devel] [PATCH] ASoC: rsnd: stop all working stream when .remove
by Kuninori Morimoto 11 Oct '17

11 Oct '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Driver should stop all working stream when .remove timing. Current Renesas sound driver is assuming that all stream was stopped when .remove but it was wrong. This patch stops all working stream when .remove, otherwise kernel will get damage for example in below case. Special thanks to Truong, Hiep > cd /sys/bus/platform/drivers/rcar_sound > aplay xxx.wav & > echo ec500000.sound > unbind Reported-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 68 ++++++++++++++++++++++++++++++++++-------------- 1 file changed, 48 insertions(+), 20 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1071332..1bf472f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -197,12 +197,6 @@ void rsnd_mod_interrupt(struct rsnd_mod *mod, } } -int rsnd_io_is_working(struct rsnd_dai_stream *io) -{ - /* see rsnd_dai_stream_init/quit() */ - return !!io->substream; -} - int rsnd_runtime_channel_original(struct rsnd_dai_stream *io) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); @@ -610,20 +604,24 @@ struct rsnd_dai_stream *rsnd_rdai_to_io(struct rsnd_dai *rdai, return &rdai->capture; } -static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) +int rsnd_io_is_working(struct rsnd_dai_stream *io) +{ + /* see rsnd_dai_stream_init/quit() */ + return !!io->substream; +} + +#define rsnd_io_start(priv, io, sub) rsnd_io_operation(priv, io, sub) +#define rsnd_io_stop(priv, io) rsnd_io_operation(priv, io, NULL) +static int rsnd_io_operation(struct rsnd_priv *priv, + struct rsnd_dai_stream *io, + struct snd_pcm_substream *substream) { - struct rsnd_priv *priv = rsnd_dai_to_priv(dai); - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); int ret; unsigned long flags; spin_lock_irqsave(&priv->lock, flags); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: + if (substream) { rsnd_dai_stream_init(io, substream); ret = rsnd_dai_call(init, io, priv); @@ -638,9 +636,7 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, if (ret < 0) goto dai_trigger_end; - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: + } else { ret = rsnd_dai_call(irq, io, priv, 0); ret |= rsnd_dai_call(stop, io, priv); @@ -648,9 +644,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, ret |= rsnd_dai_call(quit, io, priv); rsnd_dai_stream_quit(io); - break; - default: - ret = -EINVAL; } dai_trigger_end: @@ -659,6 +652,30 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct rsnd_priv *priv = rsnd_dai_to_priv(dai); + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + ret = rsnd_io_start(priv, io, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ret = rsnd_io_stop(priv, io); + break; + default: + ret = -EINVAL; + } + + return ret; +} + static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); @@ -1477,6 +1494,17 @@ static int rsnd_remove(struct platform_device *pdev) }; int ret = 0, i; + /* + * Stop all working io + */ + for_each_rsnd_dai(rdai, priv, i) { + if (rsnd_io_is_working(&rdai->playback)) + rsnd_io_stop(priv, &rdai->playback); + + if (rsnd_io_is_working(&rdai->capture)) + rsnd_io_stop(priv, &rdai->capture); + } + pm_runtime_disable(&pdev->dev); for_each_rsnd_dai(rdai, priv, i) { -- 1.9.1

3 21

[alsa-devel] [PATCH 0/3] ALSA: add snd_card_disconnect_sync()
by Kuninori Morimoto 11 Oct '17

11 Oct '17

Hi Takashi-san, Mark Current ALSA (SoC) might be kernel panic if user unbinds sound driver during playback. > aplay xxx.wav & > echo xxxx > /sys/bus/platform/drivers/xxxx/unbind Main issue is we *can't* skip unbind with return -Exxx, because unbind operation doesn't checks return value from each driver. Thus, we *must* stop driver immediately when unbind. 1) is for ALSA 2) is for ALSA SoC 3) is for Renesas sound driver. I think almost all driver want to have 3) type patch on .remove but this patch-set doesn't care at this point. Please let me know if they should have it. Takashi Iwai (1): 1) ALSA: add snd_card_disconnect_sync() Kuninori Morimoto (2): 2) ASoC: add snd_soc_card_disconnect_sync() 3) ASoC: rsnd: call snd_soc_card_disconnect_sync() when remove include/sound/core.h | 2 ++ include/sound/soc.h | 2 ++ sound/core/init.c | 18 ++++++++++++++++++ sound/core/pcm.c | 4 ++++ sound/soc/sh/rcar/core.c | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ 6 files changed, 38 insertions(+) -- 1.9.1 Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH 0/3] ALSA: add snd_card_disconnect_sync()
by Kuninori Morimoto 11 Oct '17

11 Oct '17

Hi Takashi-san, Mark Current ALSA (SoC) might be kernel panic if user unbinds sound driver during playback. > aplay xxx.wav & > echo xxxx > /sys/bus/platform/drivers/xxxx/unbind Main issue is we *can't* skip unbind with return -Exxx, because unbind operation doesn't checks return value from each driver. Thus, we *must* stop driver immediately when unbind. 1) is for ALSA 2) is for ALSA SoC 3) is for Renesas sound driver. I think almost all driver want to have 3) type patch on .remove but this patch-set doesn't care at this point. Please let me know if they should have it. Takashi Iwai (1): 1) ALSA: add snd_card_disconnect_sync() Kuninori Morimoto (2): 2) ASoC: add snd_soc_card_disconnect_sync() 3) ASoC: rsnd: call snd_soc_card_disconnect_sync() when remove include/sound/core.h | 2 ++ include/sound/soc.h | 2 ++ sound/core/init.c | 18 ++++++++++++++++++ sound/core/pcm.c | 4 ++++ sound/soc/sh/rcar/core.c | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ 6 files changed, 38 insertions(+) -- 1.9.1 Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH 0/9] sound: Add sanity checks for invalid EPs
by Takashi Iwai 10 Oct '17

10 Oct '17

Hi, this is a patchset to cover the codes that may submit URBs containing invalid EPs without validation, which result in the kernel warning from the USB core. The first patch adds a new helper for simplifying the EP check, and the rest applies it at needed places. USB devs: could you check the first patch? I tried usb_find_common_endpoints() and friends at first, but it made the code more complex in the end, because we're dealing with the fixed endpoints and the validation is required for them. The original issues were spotted by syzkaller, and I put a few others for covering more similar cases. thanks, Takashi === Takashi Iwai (9): usb: core: Add a helper function to check the validity of EP type in URB ALSA: bcd2000: Add a sanity check for invalid EPs ALSA: caiaq: Add a sanity check for invalid EPs ALSA: line6: Add a sanity check for invalid EPs ALSA: usb-audio: Add sanity checks for invalid EPs ALSA: usx2y: Add sanity checks for invalid EPs ALSA: hiface: Add sanity checks for invalid EPs ALSA: caiaq: Add yet more sanity checks for invalid EPs ALSA: line6: Add yet more sanity checks for invalid EPs drivers/usb/core/urb.c | 28 +++++++++++++++++++++++++--- include/linux/usb.h | 2 ++ sound/usb/bcd2000/bcd2000.c | 7 +++++++ sound/usb/caiaq/device.c | 7 +++++++ sound/usb/caiaq/input.c | 9 +++++++++ sound/usb/hiface/pcm.c | 9 +++++++-- sound/usb/line6/driver.c | 30 ++++++++++++++++++++++-------- sound/usb/line6/midi.c | 17 +++++++++++------ sound/usb/midi.c | 38 ++++++++++++++++++++++++++++++-------- sound/usb/usx2y/usbusx2y.c | 8 ++++++++ sound/usb/usx2y/usbusx2yaudio.c | 3 +++ 11 files changed, 131 insertions(+), 27 deletions(-) -- 2.14.2

3 14

[alsa-devel] [PATCH 6/8] Documentation: fix sound related doc refs
by Tom Saeger 10 Oct '17

10 Oct '17

Make sound doc refs valid. Signed-off-by: Tom Saeger <tom.saeger(a)oracle.com> --- Documentation/sound/hd-audio/notes.rst | 2 +- Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index f59c3cdbfaf4..9f7347830ba4 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -192,7 +192,7 @@ preset model instead of PCI (and codec-) SSID look-up. What ``model`` option values are available depends on the codec chip. Check your codec chip from the codec proc file (see "Codec Proc-File" section below). It will show the vendor/product name of your codec -chip. Then, see Documentation/sound/HD-Audio-Models.rst file, +chip. Then, see Documentation/sound/hd-audio/models.rst file, the section of HD-audio driver. You can find a list of codecs and ``model`` options belonging to each codec. For example, for Realtek ALC262 codec chip, pass ``model=ultra`` for devices that are compatible diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 58ffa3f5bda7..a0b268466cb1 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -2498,7 +2498,7 @@ Mic boost Mic-boost switch is set as “Mic Boost” or “Mic Boost (6dB)”. More precise information can be found in -``Documentation/sound/alsa/ControlNames.txt``. +``Documentation/sound/designs/control-names.rst``. Access Flags ------------ -- 2.14.2

2 1

[alsa-devel] [PATCH 3/8] Documentation: fix input related doc refs
by Tom Saeger 10 Oct '17

10 Oct '17

Make `input` document refs valid including: - joystick - joystick-parport Signed-off-by: Tom Saeger <tom.saeger(a)oracle.com> --- Documentation/admin-guide/kernel-parameters.txt | 10 +++++----- Documentation/hid/hiddev.txt | 2 +- Documentation/input/devices/xpad.rst | 3 ++- Documentation/sound/cards/joystick.rst | 2 +- 4 files changed, 9 insertions(+), 8 deletions(-) diff --git a/Documentation/admin-guide/kernel-parameters.txt b/Documentation/admin-guide/kernel-parameters.txt index e857bbbc8575..411b41127eee 100644 --- a/Documentation/admin-guide/kernel-parameters.txt +++ b/Documentation/admin-guide/kernel-parameters.txt @@ -314,7 +314,7 @@ amijoy.map= [HW,JOY] Amiga joystick support Map of devices attached to JOY0DAT and JOY1DAT Format: <a>,<b> - See also Documentation/input/joystick.txt + See also Documentation/input/joydev/joystick.rst analog.map= [HW,JOY] Analog joystick and gamepad support Specifies type or capabilities of an analog joystick @@ -724,7 +724,7 @@ db9.dev[2|3]= [HW,JOY] Multisystem joystick support via parallel port (one device per port) Format: <port#>,<type> - See also Documentation/input/joystick-parport.txt + See also Documentation/input/devices/joystick-parport.rst ddebug_query= [KNL,DYNAMIC_DEBUG] Enable debug messages at early boot time. See @@ -1220,7 +1220,7 @@ [HW,JOY] Multisystem joystick and NES/SNES/PSX pad support via parallel port (up to 5 devices per port) Format: <port#>,<pad1>,<pad2>,<pad3>,<pad4>,<pad5> - See also Documentation/input/joystick-parport.txt + See also Documentation/input/devices/joystick-parport.rst gamma= [HW,DRM] @@ -1766,7 +1766,7 @@ ivrs_acpihid[00:14.5]=AMD0020:0 js= [HW,JOY] Analog joystick - See Documentation/input/joystick.txt. + See Documentation/input/joydev/joystick.rst. nokaslr [KNL] When CONFIG_RANDOMIZE_BASE is set, this disables @@ -4199,7 +4199,7 @@ TurboGraFX parallel port interface Format: <port#>,<js1>,<js2>,<js3>,<js4>,<js5>,<js6>,<js7> - See also Documentation/input/joystick-parport.txt + See also Documentation/input/devices/joystick-parport.rst udbg-immortal [PPC] When debugging early kernel crashes that happen after console_init() and before a proper diff --git a/Documentation/hid/hiddev.txt b/Documentation/hid/hiddev.txt index 6e8c9f1d2f22..638448707aa2 100644 --- a/Documentation/hid/hiddev.txt +++ b/Documentation/hid/hiddev.txt @@ -12,7 +12,7 @@ To support these disparate requirements, the Linux USB system provides HID events to two separate interfaces: * the input subsystem, which converts HID events into normal input device interfaces (such as keyboard, mouse and joystick) and a -normalised event interface - see Documentation/input/input.txt +normalised event interface - see Documentation/input/input.rst * the hiddev interface, which provides fairly raw HID events The data flow for a HID event produced by a device is something like diff --git a/Documentation/input/devices/xpad.rst b/Documentation/input/devices/xpad.rst index 5a709ab77c8d..b8bd65962dd8 100644 --- a/Documentation/input/devices/xpad.rst +++ b/Documentation/input/devices/xpad.rst @@ -230,4 +230,5 @@ Historic Edits 2005-03-19 - Dominic Cerquetti <binary1230(a)yahoo.com> - added stuff for dance pads, new d-pad->axes mappings -Later changes may be viewed with 'git log Documentation/input/xpad.txt' +Later changes may be viewed with +'git log --follow Documentation/input/devices/xpad.rst' diff --git a/Documentation/sound/cards/joystick.rst b/Documentation/sound/cards/joystick.rst index a6e468c81d02..488946fc1079 100644 --- a/Documentation/sound/cards/joystick.rst +++ b/Documentation/sound/cards/joystick.rst @@ -11,7 +11,7 @@ General First of all, you need to enable GAMEPORT support on Linux kernel for using a joystick with the ALSA driver. For the details of gameport -support, refer to Documentation/input/joystick.txt. +support, refer to Documentation/input/joydev/joystick.rst. The joystick support of ALSA drivers is different between ISA and PCI cards. In the case of ISA (PnP) cards, it's usually handled by the -- 2.14.2

2 1

[alsa-devel] [RESEND PATCH] ASoC: codecs: msm8916-wcd-digital: fix RX2 MIX1 and RX3 MIX1
by Jean-François Têtu 10 Oct '17

10 Oct '17

The kcontrol for the third input (rxN_mix1_inp3) of both RX2 and RX3 mixers are not using the correct control register. This simple patch fixes this. Signed-off-by: Jean-François Têtu <jean-francois.tetu(a)savoirfairelinux.com> --- sound/soc/codecs/msm8916-wcd-digital.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 66df8f810f0d..a10a724eb448 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -238,7 +238,7 @@ static const struct soc_enum rx_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( static const struct soc_enum rx2_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 0, 6, rx_mix1_text), SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 3, 6, rx_mix1_text), - SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 0, 6, rx_mix1_text), + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B2_CTL, 0, 6, rx_mix1_text), }; /* RX2 MIX2 */ @@ -249,7 +249,7 @@ static const struct soc_enum rx2_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( static const struct soc_enum rx3_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 0, 6, rx_mix1_text), SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 3, 6, rx_mix1_text), - SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 0, 6, rx_mix1_text), + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text), }; /* DEC */ -- 2.14.2

2 2

[alsa-devel] usb/sound: use-after-free in snd_usb_mixer_interrupt
by Andrey Konovalov 10 Oct '17

10 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit 8a5776a5f49812d29fe4b2d0a2d71675c3facf3f (4.14-rc4). gadgetfs: bound to dummy_udc driver usb 1-1: new full-speed USB device number 2 using dummy_hcd gadgetfs: connected gadgetfs: disconnected gadgetfs: connected usb 1-1: config 1 has an invalid interface number: 1 but max is 0 usb 1-1: config 1 has 2 interfaces, different from the descriptor's value: 1 usb 1-1: config 1 interface 0 altsetting 0 endpoint 0x81 has an invalid bInterval 0, changing to 10 usb 1-1: too many endpoints for config 1 interface 1 altsetting 174: 131, using maximum allowed: 30 usb 1-1: config 1 interface 1 altsetting 174 has 0 endpoint descriptors, different from the interface de scriptor's value: 131 usb 1-1: config 1 interface 1 has no altsetting 0 usb 1-1: New USB device found, idVendor=0dba, idProduct=1000 usb 1-1: New USB device strings: Mfr=0, Product=0, SerialNumber=252 usb 1-1: SerialNumber: a gadgetfs: configuration #1 hub 1-1:1.0: USB hub found hub 1-1:1.0: config failed, can't read hub descriptor (err -22) snd-usb-audio: probe of 1-1:1.0 failed with error -22 ================================================================== BUG: KASAN: use-after-free in snd_usb_mixer_interrupt+0x604/0x6f0 Read of size 8 at addr ffff8800636f8900 by task swapper/1/0 CPU: 1 PID: 0 Comm: swapper/1 Not tainted 4.14.0-rc4-43418-g43a3f84d2109-dirty #391 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:16 dump_stack+0x292/0x395 lib/dump_stack.c:52 print_address_description+0x78/0x280 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 kasan_report+0x23d/0x350 mm/kasan/report.c:409 __asan_report_load8_noabort+0x19/0x20 mm/kasan/report.c:430 snd_usb_mixer_interrupt+0x604/0x6f0 sound/usb/mixer.c:2490 __usb_hcd_giveback_urb+0x2e0/0x650 drivers/usb/core/hcd.c:1779 usb_hcd_giveback_urb+0x337/0x420 drivers/usb/core/hcd.c:1845 dummy_timer+0xba9/0x39f0 drivers/usb/gadget/udc/dummy_hcd.c:1965 call_timer_fn+0x2a2/0x940 kernel/time/timer.c:1281 expire_timers kernel/time/timer.c:1320 __run_timers+0x87f/0xd40 kernel/time/timer.c:1620 run_timer_softirq+0x83/0x140 kernel/time/timer.c:1646 __do_softirq+0x2ee/0xc0f kernel/softirq.c:284 invoke_softirq kernel/softirq.c:364 irq_exit+0x171/0x1a0 kernel/softirq.c:405 exiting_irq ./arch/x86/include/asm/apic.h:638 smp_apic_timer_interrupt+0x2b9/0x8d0 arch/x86/kernel/apic/apic.c:1048 apic_timer_interrupt+0x9d/0xb0 </IRQ> RIP: 0010:native_safe_halt+0x6/0x10 ./arch/x86/include/asm/irqflags.h:53 RSP: 0018:ffff88006bac7b90 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff10 RAX: dffffc0000000020 RBX: 1ffff1000d758f76 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffff88006baa3a64 RBP: ffff88006bac7b90 R08: ffffffff813d5301 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 1ffff1000d758f82 R13: ffff88006bac7cd0 R14: ffffffff8881ecc8 R15: 0000000000000000 arch_safe_halt ./arch/x86/include/asm/paravirt.h:93 default_idle+0x127/0x690 arch/x86/kernel/process.c:341 arch_cpu_idle+0xf/0x20 arch/x86/kernel/process.c:332 default_idle_call+0x3b/0x60 kernel/sched/idle.c:98 cpuidle_idle_call kernel/sched/idle.c:156 do_idle+0x33a/0x410 kernel/sched/idle.c:246 cpu_startup_entry+0x1d/0x20 kernel/sched/idle.c:351 start_secondary+0x3de/0x500 arch/x86/kernel/smpboot.c:278 secondary_startup_64+0xa5/0xa5 arch/x86/kernel/head_64.S:235 Allocated by task 1484: save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59 save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551 kmem_cache_alloc_trace+0x11e/0x2d0 mm/slub.c:2772 kmalloc ./include/linux/slab.h:493 kzalloc ./include/linux/slab.h:666 snd_usb_create_mixer+0x145/0x1010 sound/usb/mixer.c:2540 create_standard_mixer_quirk+0x58/0x80 sound/usb/quirks.c:516 snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:560 create_composite_quirk+0x1c4/0x3e0 sound/usb/quirks.c:59 snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:560 usb_audio_probe+0x1040/0x2c10 sound/usb/card.c:618 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Freed by task 1484: save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59 save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_slab_free+0x72/0xc0 mm/kasan/kasan.c:524 slab_free_hook mm/slub.c:1390 slab_free_freelist_hook mm/slub.c:1412 slab_free mm/slub.c:2988 kfree+0xf6/0x2f0 mm/slub.c:3919 snd_usb_mixer_free+0x11a/0x160 sound/usb/mixer.c:2244 snd_usb_mixer_dev_free+0x36/0x50 sound/usb/mixer.c:2250 __snd_device_free+0x1ff/0x380 sound/core/device.c:91 snd_device_free_all+0x8f/0xe0 sound/core/device.c:244 snd_card_do_free sound/core/init.c:461 release_card_device+0x47/0x170 sound/core/init.c:181 device_release+0x13f/0x210 drivers/base/core.c:814 kobject_cleanup lib/kobject.c:648 kobject_release lib/kobject.c:677 kref_put ./include/linux/kref.h:70 kobject_put+0x145/0x240 lib/kobject.c:694 put_device+0x25/0x30 drivers/base/core.c:1931 snd_card_free_when_closed sound/core/init.c:489 snd_card_free+0x156/0x220 sound/core/init.c:514 usb_audio_probe+0x1e36/0x2c10 sound/usb/card.c:648 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 The buggy address belongs to the object at ffff8800636f8900 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 0 bytes inside of 192-byte region [ffff8800636f8900, ffff8800636f89c0) The buggy address belongs to the page: page:ffffea00018dbe00 count:1 mapcount:0 mapping: (null) index:0x0 flags: 0x100000000000100(slab) raw: 0100000000000100 0000000000000000 0000000000000000 0000000180100010 raw: ffffea00018dbc00 0000000400000004 ffff88006c003380 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8800636f8800: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8800636f8880: 00 00 00 00 00 00 00 00 fc fc fc fc fc fc fc fc >ffff8800636f8900: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8800636f8980: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8800636f8a00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ==================================================================

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[alsa-devel] [PATCH] ALSA: seq: Add sanity check for user-space pointer delivery
by Takashi Iwai 10 Oct '17

10 Oct '17

The sequencer event may contain a user-space pointer with its SNDRV_SEQ_EXT_USRPTR bit, and we assure that its delivery is limited with non-atomic mode. Otherwise the copy_from_user() may hit the fault and cause a problem. Although the core code doesn't set such a flag (only set at snd_seq_write()), any wild driver may set it mistakenly and lead to an unexpected crash. This patch adds a sanity check of such events at the delivery core code to filter out the invalid invocation in the atomic mode. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 6c9cba2166d9..84759d9c54bf 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -802,6 +802,10 @@ static int snd_seq_deliver_event(struct snd_seq_client *client, struct snd_seq_e return -EMLINK; } + if (snd_seq_ev_is_variable(event) && + snd_BUG_ON(atomic && (event->data.ext.len & SNDRV_SEQ_EXT_USRPTR))) + return -EINVAL; + if (event->queue == SNDRV_SEQ_ADDRESS_SUBSCRIBERS || event->dest.client == SNDRV_SEQ_ADDRESS_SUBSCRIBERS) result = deliver_to_subscribers(client, event, atomic, hop); -- 2.14.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: dapm: add initialization for w_param_text pointer
by Arnd Bergmann 10 Oct '17

10 Oct '17

We now allocate the array conditionally, but we always pass the pointer to the new snd_soc_dapm_free_kcontrol() function, which introduces a warning for the case that it is not initialized: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3940:2: error: 'w_param_text' may be used uninitialized in this function [-Werror=maybe-uninitialized] As snd_soc_dapm_free_kcontrol() is global, it doesn't get inlined and gcc fails to notice that we don't actually access the array in that case, so the code is actually safe. Adding an initialization for the array pointer shuts up the warning. Fixes: 19ad683abc85 ("ASoC: dapm: Avoid creating kcontrol for params") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ceec185c871f..19babdae1a06 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3905,6 +3905,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, ret = -ENOMEM; goto param_fail; } + } else { + w_param_text = NULL; } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); -- 2.9.0

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[alsa-devel] [PATCH 0/2 v2] ASoC: soc-core: add missing component functions
by Kuninori Morimoto 10 Oct '17

10 Oct '17

Hi Mark These are v2 of one of "replace to component" patch set. These add missing functions which exist in "platforms" but not exist in "component". Kuninori Morimoto (3): ASoC: soc-core: add component lookup functions ASoC: soc-core: add snd_soc_add_component() ASoC: soc-core: remove unnecessary message from snd_soc_register_component() include/sound/soc.h | 7 +++++++ sound/soc/soc-core.c | 58 ++++++++++++++++++++++++++++++++++++++++++---------- 2 files changed, 54 insertions(+), 11 deletions(-) -- 1.9.1

2 4

[alsa-devel] [bug report] ASoC: Intel: Refactor common IPC/mailbox code into generic APIs
by Dan Carpenter 10 Oct '17

10 Oct '17

Hello Jin Yao, The patch a33c1ec5cf82: "ASoC: Intel: Refactor common IPC/mailbox code into generic APIs" from Apr 7, 2015, leads to the following static checker warning: sound/soc/intel/common/sst-ipc.c:242 sst_ipc_reply_find_msg() error: uninitialized symbol 'mask'. sound/soc/intel/common/sst-ipc.c 226 struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, 227 u64 header) 228 { 229 struct ipc_message *msg; 230 u64 mask; 231 232 if (ipc->ops.reply_msg_match != NULL) 233 header = ipc->ops.reply_msg_match(header, &mask); What should "mask" be initialized to if ops.reply_msg_match is NULL? 234 235 if (list_empty(&ipc->rx_list)) { 236 dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n", 237 header); 238 return NULL; 239 } 240 241 list_for_each_entry(msg, &ipc->rx_list, list) { 242 if ((msg->header & mask) == header) ^^^^^^^^^^^^^^^^^^ Used uninitialized here. 243 return msg; 244 } 245 246 return NULL; 247 } regards, dan carpenter

1 0

[alsa-devel] [pull] amdgpu pre-reqs for stoney ACP audio support
by Alex Deucher 10 Oct '17

10 Oct '17

Hi Dave, Mark, This branch against Linus' master contains the 3 pre-requisite patches for Stoney ACP audio. Mark needs these patches to apply the rest of the Stoney ACP patch set on the audio side. Please pull. Thanks! The following changes since commit 68ebe3cbe786a954481d6ded0dc83f7d551640bf: Merge tag 'nfs-for-4.14-3' of git://git.linux-nfs.org/projects/trondmy/linux-nfs (2017-10-09 10:55:37 -0700) are available in the git repository at: git://people.freedesktop.org/~agd5f/linux linus-acp-prereq for you to fetch changes up to 1e4448648333a8fb7830a7a4b76c86e160a79b1e: drm/amdgpu Moving amdgpu asic types to a separate file (2017-10-09 17:54:24 -0400) ---------------------------------------------------------------- Akshu Agrawal (1): drm/amdgpu Moving amdgpu asic types to a separate file Vijendar Mukunda (2): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: AMD: Added asic_type as ACP DMA driver platform data drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 ++ drivers/gpu/drm/amd/include/amd_shared.h | 29 ++---------------- include/drm/amd_asic_type.h | 52 ++++++++++++++++++++++++++++++++ sound/soc/amd/acp-pcm-dma.c | 8 ++--- sound/soc/amd/acp.h | 7 +++++ 5 files changed, 65 insertions(+), 33 deletions(-) create mode 100644 include/drm/amd_asic_type.h

1 0

[alsa-devel] [PATCH v2] ALSA: seq: resize buffer for overflow
by Mark Salyzyn 09 Oct '17

09 Oct '17

Can not replicate, issue discovered in fuzzing. Stack trace below. No functional or performance testing done regarding the fix. Trap at (reformatted): snd_seq_oss_readq_puts(struct seq_oss_readq *q, int dev, unsigned char *data, int len) { union evrec rec; int result; memset(&rec, 0, sizeof(rec)); rec.c[0] = SEQ_MIDIPUTC; rec.c[2] = dev; while (len-- > 0) { rec.c[1] = *data++; // data is RBX HERE 'data' pointer just passed a page boundary, so the buffer supplied was short. Caller must have been handed an ev->type equal to SNDDRV_SEQ_EVENT_SYSEX, which resulted in handing off ev->data.ext.ptr[ev->data.ext.len] buffer. Intuited that the source of the event and buffer was referenced in snd_midi_event_encode_byte() passing a larger length than the allocated buffer. BUG: unable to handle kernel paging request at ffffc900008ab000 IP: [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 PGD 1da091067 PUD 1da092067 Oops: 0000 [#1] PREEMPT SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 1 PID: 3264 Comm: XXXXXXXX Hardware name: XXXXXXXXXX task: ffff8801cdd9e000 task.stack: ffff8801ce648000 RIP: 0010:[<ffffffff82e2fc05>] [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP: 0018:ffff8801ce64f1c0 EFLAGS: 00010246 RAX: 0000000000000000 RBX: ffffc900008ab000 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffffffff858e5780 RBP: ffff8801ce64f260 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 1ffff10039cc9df2 R12: 000000003fffffa4 R13: dffffc0000000000 R14: ffff8801ce64f238 R15: ffffc900008ab001 FS: 00007fe3d3d9e700(0000) GS:ffff8801db300000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: ffffc900008ab000 CR3: 00000001d19b7000 CR4: 00000000001406e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Stack: 1ffff10039cc9e3b ffff8801ce64f1f8 ffff8801d0b9aa00 0000000041b58ab3 ffffffff841daf3c ffffffff82e2fb30 0000000000000286 0000000000000005 ffffffff838aa5d5 ffffffff861962c0 dffffc0000000000 ffff8801ce64f260 Call Trace: [<ffffffff82e2ebbe>] send_midi_event sound/core/seq/oss/seq_oss_midi.c:616 [inline] [<ffffffff82e2ebbe>] snd_seq_oss_midi_input+0x8ce/0xa70 sound/core/seq/oss/seq_oss_midi.c:535 [<ffffffff82e2758d>] snd_seq_oss_event_input+0x15d/0x220 sound/core/seq/oss/seq_oss_event.c:439 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0be16>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] [<ffffffff82e0be16>] snd_seq_deliver_event+0x316/0x740 sound/core/seq/seq_clientmgr.c:807 [<ffffffff82e0cf9e>] snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2314 [<ffffffff82e31775>] dummy_input+0x235/0x320 sound/core/seq/seq_dummy.c:104 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0bc2d>] snd_seq_deliver_event+0x12d/0x740 sound/core/seq/seq_clientmgr.c:818 [<ffffffff82e0d0ed>] snd_seq_dispatch_event+0x11d/0x520 sound/core/seq/seq_clientmgr.c:892 [<ffffffff82e1117e>] snd_seq_check_queue.part.3+0x38e/0x510 sound/core/seq/seq_queue.c:285 [<ffffffff82e11d8d>] snd_seq_check_queue sound/core/seq/seq_queue.c:357 [inline] [<ffffffff82e11d8d>] snd_seq_enqueue_event+0x32d/0x3d0 sound/core/seq/seq_queue.c:363 [<ffffffff82e0c444>] snd_seq_client_enqueue_event+0x204/0x3e0 sound/core/seq/seq_clientmgr.c:951 [<ffffffff82e0cc55>] kernel_client_enqueue.part.10+0xb5/0xd0 sound/core/seq/seq_clientmgr.c:2251 [<ffffffff82e0cd3f>] kernel_client_enqueue sound/core/seq/seq_clientmgr.c:2241 [inline] [<ffffffff82e0cd3f>] snd_seq_kernel_client_enqueue_blocking+0xcf/0x110 sound/core/seq/seq_clientmgr.c:2279 [<ffffffff82e27f68>] insert_queue sound/core/seq/oss/seq_oss_rw.c:189 [inline] [<ffffffff82e27f68>] snd_seq_oss_write+0x538/0x850 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff82e1fab4>] odev_write+0x64/0x90 sound/core/seq/oss/seq_oss.c:177 [<ffffffff8156ab43>] __vfs_write+0x103/0x680 fs/read_write.c:510 [<ffffffff8156ec70>] vfs_write+0x170/0x4e0 fs/read_write.c:560 [<ffffffff81572669>] SYSC_write fs/read_write.c:607 [inline] [<ffffffff81572669>] SyS_write+0xd9/0x1b0 fs/read_write.c:599 [<ffffffff838aad85>] entry_SYSCALL_64_fastpath+0x23/0xc6 Code: 4d 9a eb 4e e8 2d aa 53 fe 4c 8d 7b 01 48 89 d8 48 89 d9 48 c1 e8 03 83 e1 07 42 0f b6 04 28 38 c8 7f 08 84 c0 0f 85 80 00 00 00 <41> 0f b6 47 ff 41 83 ec 01 48 8b b5 68 ff ff ff 48 8b bd 70 ff RIP [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP <ffff8801ce64f1c0> CR2: ffffc900008ab000 Signed-off-by: Mark Salyzyn <salyzyn(a)android.com> v2: removed nested locks in snd_midi_event_resize_buffer --- sound/core/seq/seq_midi_event.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 90bbbdbeba03..ed3206ef0cd4 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -192,8 +192,7 @@ EXPORT_SYMBOL(snd_midi_event_no_status); /* * resize buffer */ -#if 0 -int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) +static int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) { unsigned char *new_buf, *old_buf; unsigned long flags; @@ -203,16 +202,13 @@ int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) new_buf = kmalloc(bufsize, GFP_KERNEL); if (new_buf == NULL) return -ENOMEM; - spin_lock_irqsave(&dev->lock, flags); old_buf = dev->buf; dev->buf = new_buf; dev->bufsize = bufsize; reset_encode(dev); - spin_unlock_irqrestore(&dev->lock, flags); kfree(old_buf); return 0; } -#endif /* 0 */ /* * read bytes and encode to sequencer event if finished @@ -297,6 +293,8 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || dev->read >= dev->bufsize) { + if (dev->read > dev->bufsize) + snd_midi_event_resize_buffer(dev, dev->read); ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK; ev->flags |= SNDRV_SEQ_EVENT_LENGTH_VARIABLE; ev->type = SNDRV_SEQ_EVENT_SYSEX; -- 2.14.2.920.gcf0c67979c-goog

2 2

[alsa-devel] usb/sound/line6: trying to register non-static key in podhd_disconnect
by Andrey Konovalov 09 Oct '17

09 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e19b205be43d11bff638cad4487008c48d21c103 (4.14-rc2). INFO: trying to register non-static key. the code is fine but needs lockdep annotation. turning off the locking correctness validator. CPU: 1 PID: 1845 Comm: kworker/1:2 Not tainted 4.14.0-rc2-42613-g1488251d1a98 #238 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:16 dump_stack+0x292/0x395 lib/dump_stack.c:52 register_lock_class+0x6c4/0x1a00 kernel/locking/lockdep.c:769 __lock_acquire+0x27e/0x4550 kernel/locking/lockdep.c:3385 lock_acquire+0x259/0x620 kernel/locking/lockdep.c:4002 del_timer_sync+0x12c/0x280 kernel/time/timer.c:1237 podhd_disconnect+0x8c/0x160 sound/usb/line6/podhd.c:299 line6_probe+0x844/0x1310 sound/usb/line6/driver.c:783 podhd_probe+0x64/0x70 sound/usb/line6/podhd.c:474 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431

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[alsa-devel] [PATCH] ASoC: Intel: cht_bsw_max98090: remove useless code, align with ChromeOS driver
by Pierre-Louis Bossart 09 Oct '17

09 Oct '17

Remove dead code which was missed in previous changes. This is not visible with git diff but there is a test+return on the same condition just above. Also reuse local variable instead of fetching the jack information twice. Tested on Acer R11 (cyan) Fixes: 3bbda5a38601 ('ASoC: Intel: cht_bsw_max98090_ti: Fix jack initialization') Cc: Thierry Escande <thierry.escande(a)collabora.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 74c047a4b05c..119da5503fee 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -219,10 +219,6 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - - if (ctx->ts3a227e_present) - snd_soc_jack_notifier_register(jack, &cht_jack_nb); - ret = snd_soc_jack_add_gpiods(runtime->card->dev->parent, jack, ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios); @@ -323,7 +319,7 @@ static int cht_max98090_headset_init(struct snd_soc_component *component) return ret; } - return ts3a227e_enable_jack_detect(component, &ctx->jack); + return ts3a227e_enable_jack_detect(component, jack); } static const struct snd_soc_ops cht_aif1_ops = { -- 2.11.0

3 2

[alsa-devel] [BUG] sound/core/seq: A possible sleep-in-atomic bug in snd_virmidi_dev_receive_event
by Jia-Ju Bai 09 Oct '17

09 Oct '17

According to seq_virmidi.c, the driver may sleep under a read spinlock. The function call path is: snd_virmidi_dev_receive_event (acquire the spinlock) snd_seq_dump_var_event copy_from_user --> may sleep This bug is found by my static analysis tool and my code review. Thanks, Jia-Ju Bai

2 1

[alsa-devel] Applied "ASoC: cygnus: Remove unnecessary active_slots check" to the asoc tree
by Mark Brown 09 Oct '17

09 Oct '17

The patch ASoC: cygnus: Remove unnecessary active_slots check has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4c75968a1bbfb3f190e2e624c83929451e4730ac Mon Sep 17 00:00:00 2001 From: Christos Gkekas <chris.gekas(a)gmail.com> Date: Sun, 8 Oct 2017 19:20:30 +0100 Subject: [PATCH] ASoC: cygnus: Remove unnecessary active_slots check Variable active_slots is unsigned so checking whether it is less than zero is not necessary. Signed-off-by: Christos Gkekas <chris.gekas(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/bcm/cygnus-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index cd8aef8ed8a5..abafadc0b534 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -986,7 +986,7 @@ static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, active_slots = hweight32(tx_mask); - if ((active_slots < 0) || (active_slots > 16)) + if (active_slots > 16) return -EINVAL; /* Slot value must be even */ -- 2.14.1

1 0

[alsa-devel] [PATCH] ASoC: img-parallel-out: Check pm_runtime_get_sync return code
by Ed Blake 09 Oct '17

09 Oct '17

Check the the pm_runtime_get_sync return code and return if an error. Signed-off-by: Ed Blake <ed.blake(a)sondrel.com> --- sound/soc/img/img-parallel-out.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 2fc8a6372206..acc005217be0 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -153,6 +153,7 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct img_prl_out *prl = snd_soc_dai_get_drvdata(dai); u32 reg, control_set = 0; + int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -164,7 +165,10 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - pm_runtime_get_sync(prl->dev); + ret = pm_runtime_get_sync(prl->dev); + if (ret < 0) + return ret; + reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL); reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set; img_prl_out_writel(prl, reg, IMG_PRL_OUT_CTL); -- 2.11.0

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: img-i2s-in: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-i2s-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Changes in v2: - Changed from ASoC specific pm callbacks to device ones - Errors returned from pm_runtime_get_sync are now handled Ed Blake (2): ASoC: img-i2s-in: Add suspend / resume handling ASoC: img-i2s-in: Add runtime PM sound/soc/img/img-i2s-in.c | 130 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 118 insertions(+), 12 deletions(-) -- 2.11.0

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: img-i2s-out: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-i2s-out driver. The first patch adds system PM suspend / resume handling and the second patch adds control of the sys clock to runtime PM. Changes in v2: - Changed from ASoC specific pm callbacks to device ones - Errors returned from pm_runtime_get_sync are now handled Ed Blake (2): ASoC: img-i2s-out: Add suspend / resume handling ASoC: img-i2s-out: Add control of sys clock to runtime PM sound/soc/img/img-i2s-out.c | 108 ++++++++++++++++++++++++++++++++++---------- 1 file changed, 85 insertions(+), 23 deletions(-) -- 2.11.0

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: img-spdif-in: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-spdif-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Changes in v2: - Changed from ASoC specific pm callbacks to device ones Ed Blake (2): ASoC: img-spdif-in: Add suspend / resume handling ASoC: img-spdif-in: Add runtime PM sound/soc/img/img-spdif-in.c | 110 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 99 insertions(+), 11 deletions(-) -- 2.11.0

2 4

[alsa-devel] [PATCH v2 0/2] dma: img-spdif-out: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-spdif-out driver. The first patch adds system PM suspend / resume handling and the second patch adds control of the sys clock to runtime PM. Changes in v2: - Changed from ASoC specific pm callbacks to device ones Ed Blake (2): ASoC: img-spdif-out: Add suspend / resume handling ASoC: img-spdif-out: Add control of sys clock to runtime PM sound/soc/img/img-spdif-out.c | 73 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 60 insertions(+), 13 deletions(-) -- 2.11.0

2 3

[alsa-devel] [PATCH 0/3] dma: img-spdif-out: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-spdif-out driver. The first patch renames existing 'suspend' and 'resume' functions to 'runtime_suspend' and 'runtime_resume', which is what they actually are, the second patch adds system PM suspend / resume handling and the third patch adds control of the sys clock to runtime PM. Ed Blake (3): ASoC: img-spdif-out: Rename suspend / resume funcs ASoC: img-spdif-out: Add suspend / resume handling ASoC: img-spdif-out: Add control of sys clock to runtime PM sound/soc/img/img-spdif-out.c | 89 +++++++++++++++++++++++++++++++++---------- 1 file changed, 68 insertions(+), 21 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCH] sound/soc: fix typos in Kconfig (trivial)
by Nik Nyby 09 Oct '17

09 Oct '17

This fixes a few typos in the snd_designware description. Signed-off-by: Nik Nyby <nikolas(a)gnu.org> --- sound/soc/dwc/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index c6fd95fa5ca6..aa0c6ec4d93c 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -4,8 +4,8 @@ config SND_DESIGNWARE_I2S select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for I2S driver for - Synopsys desigwnware I2S device. The device supports upto - maximum of 8 channels each for play and record. + Synopsys designware I2S device. The device supports up to + a maximum of 8 channels each for play and record. config SND_DESIGNWARE_PCM bool "PCM PIO extension for I2S driver" -- 2.14.2

2 2

[alsa-devel] [PATCH v3] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 09 Oct '17

09 Oct '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in this version: - Fixed the coding guideline comment from charles. Changes in v2: - Review comments from charles to move common function of error handling in seperate function. Changes in v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 140 +++++++++++++++++++++++++++++++-------------------- 1 file changed, 86 insertions(+), 54 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..dbf3838 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,27 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +void +snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, + unsigned long *private_value, + int num_params, + const char **w_param_text) +{ + int count; + + devm_kfree(card->dev, (void *)*private_value); + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); +} + +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params, const char **w_param_text, + unsigned long *private_value) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3807,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; - - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); - if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + int count; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,47 +3826,86 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ - private_value = + *private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), sizeof(struct soc_enum), GFP_KERNEL); - if (!private_value) { + if (!*private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } - kcontrol_dai_link[0].private_value = private_value; + kcontrol_dai_link[0].private_value = *private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], + kcontrol_news = devm_kmemdup(card->dev, &kcontrol_dai_link[0], sizeof(struct snd_kcontrol_new), GFP_KERNEL); - if (!template.kcontrol_news) { + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_private_value; + goto outfree_w_param; } + return kcontrol_news; +outfree_w_param: + snd_soc_dapm_free_kcontrol(card, private_value, num_params, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + const char **w_param_text; + unsigned long private_value; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params, + w_param_text, &private_value); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto param_fail; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3937,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); -outfree_link_name: + snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); +param_fail: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

4 5

[alsa-devel] [PATCH] sound/pci/riptide or drivers/base/firmware: Fix a possible sleep-in-atomic bug
by Jia-Ju Bai 09 Oct '17

09 Oct '17

The riptide driver may sleep under a spinlock, and the function call path is: snd_riptide_prepare (acquire the spinlock) setsampleformat sendcmd riptide_reset try_to_load_firmware request_firmware _request_firmware (drivers/base/firmware_class.c) _request_firmware_prepare kzalloc(GFP_KERNEL) --> may sleep To fix it, GFP_KERNEL is replaced with GFP_ATOMIC in _request_firmware_prepare. This bug is found by my static analysis tool and my code review. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)163.com> --- drivers/base/firmware_class.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/base/firmware_class.c b/drivers/base/firmware_class.c index 4b57cf5..d89f676 100644 --- a/drivers/base/firmware_class.c +++ b/drivers/base/firmware_class.c @@ -1138,7 +1138,7 @@ static inline void kill_pending_fw_fallback_reqs(bool only_kill_custom) { } struct firmware_buf *buf; int ret; - *firmware_p = firmware = kzalloc(sizeof(*firmware), GFP_KERNEL); + *firmware_p = firmware = kzalloc(sizeof(*firmware), GFP_ATOMIC); if (!firmware) { dev_err(device, "%s: kmalloc(struct firmware) failed\n", __func__); -- 1.7.9.5

2 2

[alsa-devel] [PATCH] opl3: Fix a possible sleep-in-atomic bug in snd_opl3_find_patch
by Jia-Ju Bai 09 Oct '17

09 Oct '17

The driver may sleep under a spinlock, and the function call path is: snd_opl3_note_on (acquire the spinlock) snd_opl3_find_patch kzalloc(GFP_KERNEL) --> may sleep To fix it, GFP_KERNEL is replaced with GFP_ATOMIC. This bug is found by my static analysis tool and my code review. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)163.com> --- sound/drivers/opl3/opl3_synth.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index ddcc1a3..2e1cb2b 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -325,7 +325,7 @@ struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, if (!create_patch) return NULL; - patch = kzalloc(sizeof(*patch), GFP_KERNEL); + patch = kzalloc(sizeof(*patch), GFP_ATOMIC); if (!patch) return NULL; patch->prog = prog; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: cygnus: Remove unnecessary active_slots check
by Christos Gkekas 08 Oct '17

08 Oct '17

Variable active_slots is unsigned so checking whether it is less than zero is not necessary. Signed-off-by: Christos Gkekas <chris.gekas(a)gmail.com> --- sound/soc/bcm/cygnus-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 15c438f..cb46128 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -998,7 +998,7 @@ static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, active_slots = hweight32(tx_mask); - if ((active_slots < 0) || (active_slots > 16)) + if (active_slots > 16) return -EINVAL; /* Slot value must be even */ -- 2.7.4

1 0

[alsa-devel] [BUG] haswell: A possible sleep-in-atomic bug in hsw_irq_thread
by Jia-Ju Bai 08 Oct '17

08 Oct '17

According to sst-haswell-ipc.c, the driver may sleep under a spinlock, and the function call path is: hsw_irq_thread (acquire the spinlock) hsw_process_notification hsw_log_message mutex_lock --> may sleep This bug is found by my static analysis tool and my code review. Thanks, Jia-Ju Bai

1 0

[alsa-devel] [PATCH] ALSA: seq: resize buffer for overflow
by Mark Salyzyn 06 Oct '17

06 Oct '17

Can not replicate, issue discovered in fuzzing. Stack trace below. No functional or performance testing done regarding the fix. Trap at (reformatted): snd_seq_oss_readq_puts(struct seq_oss_readq *q, int dev, unsigned char *data, int len) { union evrec rec; int result; memset(&rec, 0, sizeof(rec)); rec.c[0] = SEQ_MIDIPUTC; rec.c[2] = dev; while (len-- > 0) { rec.c[1] = *data++; // data is RBX HERE 'data' pointer just passed a page boundary, so the buffer supplied was short. Caller must have been handed an ev->type equal to SNDDRV_SEQ_EVENT_SYSEX, which resulted in handing off ev->data.ext.ptr[ev->data.ext.len] buffer. Intuited that the source of the event and buffer was referenced in snd_midi_event_encode_byte() passing a larger length than the allocated buffer. BUG: unable to handle kernel paging request at ffffc900008ab000 IP: [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 PGD 1da091067 PUD 1da092067 Oops: 0000 [#1] PREEMPT SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 1 PID: 3264 Comm: XXXXXXXX Hardware name: XXXXXXXXXX task: ffff8801cdd9e000 task.stack: ffff8801ce648000 RIP: 0010:[<ffffffff82e2fc05>] [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP: 0018:ffff8801ce64f1c0 EFLAGS: 00010246 RAX: 0000000000000000 RBX: ffffc900008ab000 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffffffff858e5780 RBP: ffff8801ce64f260 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 1ffff10039cc9df2 R12: 000000003fffffa4 R13: dffffc0000000000 R14: ffff8801ce64f238 R15: ffffc900008ab001 FS: 00007fe3d3d9e700(0000) GS:ffff8801db300000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: ffffc900008ab000 CR3: 00000001d19b7000 CR4: 00000000001406e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Stack: 1ffff10039cc9e3b ffff8801ce64f1f8 ffff8801d0b9aa00 0000000041b58ab3 ffffffff841daf3c ffffffff82e2fb30 0000000000000286 0000000000000005 ffffffff838aa5d5 ffffffff861962c0 dffffc0000000000 ffff8801ce64f260 Call Trace: [<ffffffff82e2ebbe>] send_midi_event sound/core/seq/oss/seq_oss_midi.c:616 [inline] [<ffffffff82e2ebbe>] snd_seq_oss_midi_input+0x8ce/0xa70 sound/core/seq/oss/seq_oss_midi.c:535 [<ffffffff82e2758d>] snd_seq_oss_event_input+0x15d/0x220 sound/core/seq/oss/seq_oss_event.c:439 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0be16>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] [<ffffffff82e0be16>] snd_seq_deliver_event+0x316/0x740 sound/core/seq/seq_clientmgr.c:807 [<ffffffff82e0cf9e>] snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2314 [<ffffffff82e31775>] dummy_input+0x235/0x320 sound/core/seq/seq_dummy.c:104 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0bc2d>] snd_seq_deliver_event+0x12d/0x740 sound/core/seq/seq_clientmgr.c:818 [<ffffffff82e0d0ed>] snd_seq_dispatch_event+0x11d/0x520 sound/core/seq/seq_clientmgr.c:892 [<ffffffff82e1117e>] snd_seq_check_queue.part.3+0x38e/0x510 sound/core/seq/seq_queue.c:285 [<ffffffff82e11d8d>] snd_seq_check_queue sound/core/seq/seq_queue.c:357 [inline] [<ffffffff82e11d8d>] snd_seq_enqueue_event+0x32d/0x3d0 sound/core/seq/seq_queue.c:363 [<ffffffff82e0c444>] snd_seq_client_enqueue_event+0x204/0x3e0 sound/core/seq/seq_clientmgr.c:951 [<ffffffff82e0cc55>] kernel_client_enqueue.part.10+0xb5/0xd0 sound/core/seq/seq_clientmgr.c:2251 [<ffffffff82e0cd3f>] kernel_client_enqueue sound/core/seq/seq_clientmgr.c:2241 [inline] [<ffffffff82e0cd3f>] snd_seq_kernel_client_enqueue_blocking+0xcf/0x110 sound/core/seq/seq_clientmgr.c:2279 [<ffffffff82e27f68>] insert_queue sound/core/seq/oss/seq_oss_rw.c:189 [inline] [<ffffffff82e27f68>] snd_seq_oss_write+0x538/0x850 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff82e1fab4>] odev_write+0x64/0x90 sound/core/seq/oss/seq_oss.c:177 [<ffffffff8156ab43>] __vfs_write+0x103/0x680 fs/read_write.c:510 [<ffffffff8156ec70>] vfs_write+0x170/0x4e0 fs/read_write.c:560 [<ffffffff81572669>] SYSC_write fs/read_write.c:607 [inline] [<ffffffff81572669>] SyS_write+0xd9/0x1b0 fs/read_write.c:599 [<ffffffff838aad85>] entry_SYSCALL_64_fastpath+0x23/0xc6 Code: 4d 9a eb 4e e8 2d aa 53 fe 4c 8d 7b 01 48 89 d8 48 89 d9 48 c1 e8 03 83 e1 07 42 0f b6 04 28 38 c8 7f 08 84 c0 0f 85 80 00 00 00 <41> 0f b6 47 ff 41 83 ec 01 48 8b b5 68 ff ff ff 48 8b bd 70 ff RIP [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP <ffff8801ce64f1c0> CR2: ffffc900008ab000 Signed-off-by: Mark Salyzyn <salyzyn(a)android.com> --- sound/core/seq/seq_midi_event.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 90bbbdbeba03..cbc501fd47c2 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -192,8 +192,7 @@ EXPORT_SYMBOL(snd_midi_event_no_status); /* * resize buffer */ -#if 0 -int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) +static int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) { unsigned char *new_buf, *old_buf; unsigned long flags; @@ -212,7 +211,6 @@ int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) kfree(old_buf); return 0; } -#endif /* 0 */ /* * read bytes and encode to sequencer event if finished @@ -297,6 +295,8 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || dev->read >= dev->bufsize) { + if (dev->read > dev->bufsize) + snd_midi_event_resize_buffer(dev, dev->read); ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK; ev->flags |= SNDRV_SEQ_EVENT_LENGTH_VARIABLE; ev->type = SNDRV_SEQ_EVENT_SYSEX; -- 2.14.2.920.gcf0c67979c-goog

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Re: [alsa-devel] [PATCH] alsactl: Move systemd unit start-up from basic.target to sound.target
by Takashi Iwai 06 Oct '17

06 Oct '17

On Tue, 26 Sep 2017 20:36:12 +0200, Chris Mayo wrote: > > Ensures soundcard is ready before restoring state. > > sound.target added to systemd in v18: > https://cgit.freedesktop.org/systemd/systemd/commit/?id=88dfa2938af > > Simplify dependencies: > - After=alsa-state.service is not needed because both units test for > @daemonswitch@ with opposite outcomes. > > - After=sysinit.target is automatically added by systemd. > > First proposed by Tom Yan. > > Signed-off-by: Chris Mayo <aklhfex(a)gmail.com> > --- > > I need this because my Sound Blaster Titanium is not ready by the time > basic.target is reached. > > It was proposed last year by Tom Yan but looks like it didn't get in > because requested config checks and documentation didn't materialise. > https://www.spinics.net/lists/alsa-devel/msg45537.html > > I've added those (PKG_CHECK_MODULES outputs a message at configure time). > > I've left ConditionPathExistsGlob=/dev/snd/control* in > alsa-restore.service because although it would normally be started by the > udev rule and sound.target there's no guarantee that is how it was started. Thanks, this looks like a sensitive change. Applied now. Takashi

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[alsa-devel] [PATCH] ALSA: usb-audio: Add sample rate quirk for Plantronics P610
by Kai-Heng Feng 06 Oct '17

06 Oct '17

Like other Plantronics devices, P610 does not support sample rate reading. Apply sample rate quirk to it. BugLink: https://bugs.launchpad.net/bugs/1719853 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b8cb57aeec77..9ddaae3784f5 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1138,6 +1138,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x047F, 0xC022): /* Plantronics C310 */ + case USB_ID(0x047F, 0xC02F): /* Plantronics P610 */ case USB_ID(0x047F, 0xC036): /* Plantronics C520-M */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ -- 2.14.1

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[alsa-devel] [asoc:topic/dapm 4/4] sound//soc/soc-dapm.c:3940:2: warning: 'w_param_text' may be used uninitialized in this function
by kbuild test robot 05 Oct '17

05 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/dapm head: 19ad683abc8534cd755403381c73c0ddce7b9f6d commit: 19ad683abc8534cd755403381c73c0ddce7b9f6d [4/4] ASoC: dapm: Avoid creating kcontrol for params config: i386-randconfig-x079-201740 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 19ad683abc8534cd755403381c73c0ddce7b9f6d # save the attached .config to linux build tree make ARCH=i386 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound//soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': >> sound//soc/soc-dapm.c:3940:2: warning: 'w_param_text' may be used uninitialized in this function [-Wmaybe-uninitialized] snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +/w_param_text +3940 sound//soc/soc-dapm.c 3862 3863 int snd_soc_dapm_new_pcm(struct snd_soc_card *card, 3864 const struct snd_soc_pcm_stream *params, 3865 unsigned int num_params, 3866 struct snd_soc_dapm_widget *source, 3867 struct snd_soc_dapm_widget *sink) 3868 { 3869 struct snd_soc_dapm_widget template; 3870 struct snd_soc_dapm_widget *w; 3871 const char **w_param_text; 3872 unsigned long private_value; 3873 char *link_name; 3874 int ret; 3875 3876 link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", 3877 source->name, sink->name); 3878 if (!link_name) 3879 return -ENOMEM; 3880 3881 memset(&template, 0, sizeof(template)); 3882 template.reg = SND_SOC_NOPM; 3883 template.id = snd_soc_dapm_dai_link; 3884 template.name = link_name; 3885 template.event = snd_soc_dai_link_event; 3886 template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | 3887 SND_SOC_DAPM_PRE_PMD; 3888 template.kcontrol_news = NULL; 3889 3890 /* allocate memory for control, only in case of multiple configs */ 3891 if (num_params > 1) { 3892 w_param_text = devm_kcalloc(card->dev, num_params, 3893 sizeof(char *), GFP_KERNEL); 3894 if (!w_param_text) { 3895 ret = -ENOMEM; 3896 goto param_fail; 3897 } 3898 3899 template.num_kcontrols = 1; 3900 template.kcontrol_news = 3901 snd_soc_dapm_alloc_kcontrol(card, 3902 link_name, params, num_params, 3903 w_param_text, &private_value); 3904 if (!template.kcontrol_news) { 3905 ret = -ENOMEM; 3906 goto param_fail; 3907 } 3908 } 3909 dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); 3910 3911 w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); 3912 if (IS_ERR(w)) { 3913 ret = PTR_ERR(w); 3914 /* Do not nag about probe deferrals */ 3915 if (ret != -EPROBE_DEFER) 3916 dev_err(card->dev, 3917 "ASoC: Failed to create %s widget (%d)\n", 3918 link_name, ret); 3919 goto outfree_kcontrol_news; 3920 } 3921 if (!w) { 3922 dev_err(card->dev, "ASoC: Failed to create %s widget\n", 3923 link_name); 3924 ret = -ENOMEM; 3925 goto outfree_kcontrol_news; 3926 } 3927 3928 w->params = params; 3929 w->num_params = num_params; 3930 3931 ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); 3932 if (ret) 3933 goto outfree_w; 3934 return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); 3935 3936 outfree_w: 3937 devm_kfree(card->dev, w); 3938 outfree_kcontrol_news: 3939 devm_kfree(card->dev, (void *)template.kcontrol_news); > 3940 snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); 3941 param_fail: 3942 devm_kfree(card->dev, link_name); 3943 return ret; 3944 } 3945 --- 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 00/18] use ARRAY_SIZE macro
by Jérémy Lefaure 05 Oct '17

05 Oct '17

Hi everyone, Using ARRAY_SIZE improves the code readability. I used coccinelle (I made a change to the array_size.cocci file [1]) to find several places where ARRAY_SIZE could be used instead of other macros or sizeof division. I tried to divide the changes into a patch per subsystem (excepted for staging). If one of the patch should be split into several patches, let me know. In order to reduce the size of the To: and Cc: lines, each patch of the series is sent only to the maintainers and lists concerned by the patch. This cover letter is sent to every list concerned by this series. This series is based on linux-next next-20170929. Each patch has been tested by building the relevant files with W=1. This series contains the following patches: [PATCH 01/18] sound: use ARRAY_SIZE [PATCH 02/18] tracing/filter: use ARRAY_SIZE [PATCH 03/18] media: use ARRAY_SIZE [PATCH 04/18] IB/mlx5: Use ARRAY_SIZE [PATCH 05/18] net: use ARRAY_SIZE [PATCH 06/18] drm: use ARRAY_SIZE [PATCH 07/18] scsi: bfa: use ARRAY_SIZE [PATCH 08/18] ecryptfs: use ARRAY_SIZE [PATCH 09/18] nfsd: use ARRAY_SIZE [PATCH 10/18] orangefs: use ARRAY_SIZE [PATCH 11/18] dm space map metadata: use ARRAY_SIZE [PATCH 12/18] x86: use ARRAY_SIZE [PATCH 13/18] tpm: use ARRAY_SIZE [PATCH 14/18] ipmi: use ARRAY_SIZE [PATCH 15/18] acpi: use ARRAY_SIZE [PATCH 16/18] media: staging: atomisp: use ARRAY_SIZE [PATCH 17/18] staging: rtl8723bs: use ARRAY_SIZE [PATCH 18/18] staging: rtlwifi: use ARRAY_SIZE [1]: https://lkml.org/lkml/2017/9/13/689

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[alsa-devel] [PATCH] ALSA: sh: aica: Convert timers to use timer_setup()
by Kees Cook 05 Oct '17

05 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. This requires adding a pointer to hold the timer's target substream, as there won't be a way to pass this in the future. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Bhumika Goyal <bhumirks(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- This requires commit 686fef928bba ("timer: Prepare to change timer callback argument type") in v4.14-rc3, but should be otherwise stand-alone. --- sound/sh/aica.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index fdc680ae8aa0..2b26311405a4 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -299,14 +299,14 @@ static void run_spu_dma(struct work_struct *work) } } -static void aica_period_elapsed(unsigned long timer_var) +static void aica_period_elapsed(struct timer_list *t) { + struct snd_card_aica *dreamcastcard = from_timer(dreamcastcard, + t, timer); + struct snd_pcm_substream *substream = dreamcastcard->timer_substream; /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; - struct snd_pcm_substream *substream; - struct snd_card_aica *dreamcastcard; - substream = (struct snd_pcm_substream *) timer_var; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; /* Have we played out an additional period? */ @@ -336,12 +336,12 @@ static void spu_begin_dma(struct snd_pcm_substream *substream) /*get the queue to do the work */ schedule_work(&(dreamcastcard->spu_dma_work)); /* Timer may already be running */ - if (unlikely(dreamcastcard->timer.data)) { + if (unlikely(dreamcastcard->timer_substream)) { mod_timer(&dreamcastcard->timer, jiffies + 4); return; } - setup_timer(&dreamcastcard->timer, aica_period_elapsed, - (unsigned long) substream); + timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); + dreamcastcard->timer_substream = substream; mod_timer(&dreamcastcard->timer, jiffies + 4); } @@ -379,7 +379,7 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; flush_work(&(dreamcastcard->spu_dma_work)); - if (dreamcastcard->timer.data) + if (dreamcastcard->timer_substream) del_timer(&dreamcastcard->timer); kfree(dreamcastcard->channel); spu_disable(); @@ -600,7 +600,7 @@ static int snd_aica_probe(struct platform_device *devptr) { int err; struct snd_card_aica *dreamcastcard; - dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + dreamcastcard = kzalloc(sizeof(struct snd_card_aica), GFP_KERNEL); if (unlikely(!dreamcastcard)) return -ENOMEM; err = snd_card_new(&devptr->dev, index, SND_AICA_DRIVER, @@ -619,8 +619,6 @@ static int snd_aica_probe(struct platform_device *devptr) err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) goto freedreamcast; - dreamcastcard->timer.data = 0; - dreamcastcard->channel = NULL; /* Add basic controls */ err = add_aicamixer_controls(dreamcastcard); if (unlikely(err < 0)) -- 2.7.4 -- Kees Cook Pixel Security

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[alsa-devel] [PATCH 0/2] ASoC: img-i2s-in: Add power management
by Ed Blake 05 Oct '17

05 Oct '17

This patch set adds power managment to the img-i2s-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Ed Blake (2): ASoC: img-i2s-in: Add suspend / resume handling ASoC: img-i2s-in: Add runtime PM sound/soc/img/img-i2s-in.c | 126 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 114 insertions(+), 12 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] conf: HdmiLpeAudio: remove the "front" pcm definition
by Tanu Kaskinen 05 Oct '17

05 Oct '17

PulseAudio assumes that the "front" pcm device always refers to an analog device, not HDMI. While that assumption is not really valid, the reality is that without that assumption PulseAudio can't know whether "front" and "hdmi" refer to a different or the same device. The HDMI LPE driver doesn't allow audio streaming while the HDMI cable is unplugged, so PulseAudio has to know when it's plugged in and when it's not. If both "front" and "hdmi" devices exist, PulseAudio will notice that HDMI is unplugged, but it doesn't know that "front" refers to the same device, and PulseAudio will try to use the "front" device with bad consequences. The kernel driver's refusal to stream any audio makes PulseAudio enter an infinite loop and then the kernel kills PulseAudio, because it consumes too much CPU time in a realtime thread. While the looping in PulseAudio could probably be fixed, that wouldn't change the fact that PulseAudio thinks that there is an analog device. I believe it's best to avoid having the same device as both "front" and "hdmi" in alsa-lib. I removed also the surround configuration includes. I don't think they had any effect anyway, so I wonder why they were there in the first place. BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=100488 --- src/conf/cards/HdmiLpeAudio.conf | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/src/conf/cards/HdmiLpeAudio.conf b/src/conf/cards/HdmiLpeAudio.conf index 9fa30da0..a1e493da 100644 --- a/src/conf/cards/HdmiLpeAudio.conf +++ b/src/conf/cards/HdmiLpeAudio.conf @@ -2,30 +2,6 @@ # Configuration for the Intel HDMI/DP LPE audio # -<confdir:pcm/front.conf> - -HdmiLpeAudio.pcm.front.0 { - @args [ CARD ] - @args.CARD { - type string - } - type softvol - slave.pcm { - type hw - card $CARD - } - control { - name "PCM Playback Volume" - card $CARD - } -} - -<confdir:pcm/surround40.conf> -<confdir:pcm/surround21.conf> -<confdir:pcm/surround41.conf> -<confdir:pcm/surround50.conf> -<confdir:pcm/surround51.conf> - <confdir:pcm/hdmi.conf> HdmiLpeAudio.pcm.hdmi.0 { -- 2.14.2

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[alsa-devel] SEGFault when optininal snd_ctl_ext_callback::read_event() not set
by Wischer, Timo (ADITG/ESB) 05 Oct '17

05 Oct '17

Hi all, snd_ctl_ext_callback::read_event() callback is mentioned as optional (see http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=include/control_extern…) but there is no NULL check and the NULL pointer will be called if the read_event function callback pointer is not set. (see http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/control/control_ex…) I think a default function has to be provided which will be called when the callback is not set or the read_event() callback should not be marked as optional. What is your opinion? Best regards Timo Wischer Advanced Driver Information Technology GmbH Engineering Software Base (ADITG/ESB) Robert-Bosch-Str. 200 31139 Hildesheim Germany Tel. +49 5121 49 6938 Fax +49 5121 49 6999 twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschäftsführung: Wilhelm Grabow, Ken Yaguchi

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[alsa-devel] notifying userspace of hwparams change
by Tim Harvey 05 Oct '17

05 Oct '17

Greetings, I've got an HDMI receiver chip I'm working on a driver for which provides the audio signal from the HDMI source. An HDMI source can be a variety of sample-rates, channels, widths, and formats. If I understand correctly I need to register the device as supporting all the possible rates/channels/formats then constrain these to what is actually available at the time in the startup callback with snd_pcm_hw_contstraint_minmax and snd_pcm_hw_constraint_mask. How do I notify userspace when the chip tells me that the HDMI audio source rate/channels/format has changed? I know on the Video side the driver sends a V4L2_EVENT_SOURCE_CHANGE event to userspace when input format/resolution has changed so that userspace can stop streaming, call VIDIOC_QUERY_DV_TIMINGS and then VIDIOC_S_DV_TIMINGS with new timings. Regards, Tim

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[alsa-devel] [PATCH] ALSA: core: Add new ACCESS_TLV flag for coefficient TLV controls
by Charles Keepax 05 Oct '17

05 Oct '17

ASoC can use the TLV mechanism to pass coefficient data for controls that exceed the normal 512 byte limit. However, there is no mechanism to inform user-space the control expects data through the TLV mechanism. Currently amixer uses a combination of the absense of non-TLV read/write permissions and the type being BYTES. This is not ideal as it is not a clear signal and prevents controls that might support both access mechanisms. Add a specific flag SNDRV_CTL_ELEM_ACCESS_TLV_COEFF to let user-space know when it is dealing with a TLV control that is being used to pass actual coefficient data. Then add this flag to the controls generated by the current users of these types of controls. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Hi, This builds on work done by Sakamoto-san [1] and the changes mentioned to amixer in the commit message are found in [2]. I think something like this should probably be added, what I am less clear on is where we go from here? Sakamoto-san's original patches also added a tag type SNDRV_CTL_TVLT_COEFF, to keep the data matching the normal format found in TLV controls. The trouble is neither of the two users really follow this convention at the moment. The ADSP code just takes the data as is with no additional headers and the Skylake code does have a tag and a length as part of the data passed through the TLV however the tag part looks like it is used for some internal purpose, so I am far from clear it would support a fixed value to identify the data without some changes as well. Changing the ADSP code is probably acceptable since we have had very few users in the field, and since this work has been done improvements in the firmware tool chains mean most firmwares are created with much smaller dedicated purpose controls rather than the large block ones that were initially the reason for using these controls on our side. Finally, since [3] tinymix has been updated to add some TLV information into control writes/reads which will break the ADSP code once it trickles down to users. So I would be keen to discuss this at this years conference and decide how we are going to handle these controls? As Mark says [4] maybe we should see if there is a better approach we should all migrate across to? Thanks, Charles [1] https://patchwork.kernel.org/patch/9304649/ [2] https://patchwork.kernel.org/patch/8147001/ [3] https://github.com/tinyalsa/tinyalsa/commit/3f813e47784674a3909fe1277bc10b7… [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-September/125743.… include/sound/soc.h | 3 ++- include/uapi/sound/asound.h | 1 + sound/core/control.c | 1 + sound/soc/codecs/wm_adsp.c | 3 ++- sound/soc/intel/skylake/skl-topology.c | 1 + sound/soc/soc-ops.c | 3 +++ 6 files changed, 10 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d776cdee30d7..d82297a95caf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -323,7 +323,8 @@ #define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK | \ + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF, \ .tlv.c = (snd_soc_bytes_tlv_callback), \ .info = snd_soc_bytes_info_ext, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1949923a40bf..4bf0b2f835e9 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -853,6 +853,7 @@ typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_ACCESS_TLV_WRITE (1<<5) /* TLV write is possible */ #define SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE (SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) #define SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND (1<<6) /* TLV command is possible */ +#define SNDRV_CTL_ELEM_ACCESS_TLV_COEFF (1<<7) /* Coefficient data is accepted */ #define SNDRV_CTL_ELEM_ACCESS_INACTIVE (1<<8) /* control does actually nothing, but may be updated */ #define SNDRV_CTL_ELEM_ACCESS_LOCK (1<<9) /* write lock */ #define SNDRV_CTL_ELEM_ACCESS_OWNER (1<<10) /* write lock owner */ diff --git a/sound/core/control.c b/sound/core/control.c index 56b3e2d49c82..e7a74d48633b 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -266,6 +266,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, SNDRV_CTL_ELEM_ACCESS_INACTIVE | SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND | + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); err = snd_ctl_new(&kctl, count, access, NULL); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 65c059b5ffd7..1d7fe0d8a18a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1156,7 +1156,8 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; - out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK | + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF; } else { rd = SNDRV_CTL_ELEM_ACCESS_READ; wr = SNDRV_CTL_ELEM_ACCESS_WRITE; diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 27bcb62568fb..01b01c6a2bf3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2870,6 +2870,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, tplg_bc = container_of(hdr, struct snd_soc_tplg_bytes_control, hdr); if (kctl->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + kctl->access |= SNDRV_CTL_ELEM_ACCESS_TLV_COEFF; sb = (struct soc_bytes_ext *)kctl->private_value; if (tplg_bc->priv.size) return skl_init_algo_data( diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 500f98c730b9..5ac2afe12b5d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -776,6 +776,9 @@ int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, unsigned int count = size < params->max ? size : params->max; int ret = -ENXIO; + if (!(kcontrol->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_COEFF)) + return -ENXIO; + switch (op_flag) { case SNDRV_CTL_TLV_OP_READ: if (params->get) -- 2.11.0

2 2

[alsa-devel] [PATCH] sound/pci/asihpi: Convert timers to use timer_setup()
by Kees Cook 05 Oct '17

05 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- This requires commit 686fef928bba ("timer: Prepare to change timer callback argument type") in v4.14-rc3, but should be otherwise stand-alone. --- sound/pci/asihpi/asihpi.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 70d023a85bf5..614322171c79 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -749,9 +749,9 @@ static inline unsigned int modulo_min(unsigned int a, unsigned int b, /** Timer function, equivalent to interrupt service routine for cards */ -static void snd_card_asihpi_timer_function(unsigned long data) +static void snd_card_asihpi_timer_function(struct timer_list *t) { - struct snd_card_asihpi_pcm *dpcm = (struct snd_card_asihpi_pcm *)data; + struct snd_card_asihpi_pcm *dpcm = from_timer(dpcm, t, timer); struct snd_pcm_substream *substream = dpcm->substream; struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime; @@ -948,7 +948,7 @@ static void snd_card_asihpi_int_task(unsigned long data) asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; if (asihpi->llmode_streampriv) snd_card_asihpi_timer_function( - (unsigned long)asihpi->llmode_streampriv); + &asihpi->llmode_streampriv->timer); } static void snd_card_asihpi_isr(struct hpi_adapter *a) @@ -1059,8 +1059,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) If internal and other stream playing, can't switch */ - setup_timer(&dpcm->timer, snd_card_asihpi_timer_function, - (unsigned long) dpcm); + timer_setup(&dpcm->timer, snd_card_asihpi_timer_function, 0); dpcm->substream = substream; runtime->private_data = dpcm; runtime->private_free = snd_card_asihpi_runtime_free; @@ -1240,8 +1239,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) if (err) return -EIO; - setup_timer(&dpcm->timer, snd_card_asihpi_timer_function, - (unsigned long) dpcm); + timer_setup(&dpcm->timer, snd_card_asihpi_timer_function, 0); dpcm->substream = substream; runtime->private_data = dpcm; runtime->private_free = snd_card_asihpi_runtime_free; -- 2.7.4 -- Kees Cook Pixel Security

1 0

[alsa-devel] [PATCH] sound/core: Convert timers to use timer_setup()
by Kees Cook 05 Oct '17

05 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. This adds a pointer back to struct snd_timer. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Ingo Molnar <mingo(a)kernel.org> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: alsa-devel(a)alsa-project.org Cc: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- This requires commit 686fef928bba ("timer: Prepare to change timer callback argument type") in v4.14-rc3, but should be otherwise stand-alone. --- sound/core/timer.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 6cdd04a45962..09acaf2b2e57 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1028,15 +1028,17 @@ EXPORT_SYMBOL(snd_timer_global_register); struct snd_timer_system_private { struct timer_list tlist; + struct snd_timer *snd_timer; unsigned long last_expires; unsigned long last_jiffies; unsigned long correction; }; -static void snd_timer_s_function(unsigned long data) +static void snd_timer_s_function(struct timer_list *t) { - struct snd_timer *timer = (struct snd_timer *)data; - struct snd_timer_system_private *priv = timer->private_data; + struct snd_timer_system_private *priv = from_timer(priv, t, + tlist); + struct snd_timer *timer = priv->snd_timer; unsigned long jiff = jiffies; if (time_after(jiff, priv->last_expires)) priv->correction += (long)jiff - (long)priv->last_expires; @@ -1118,7 +1120,8 @@ static int snd_timer_register_system(void) snd_timer_free(timer); return -ENOMEM; } - setup_timer(&priv->tlist, snd_timer_s_function, (unsigned long) timer); + priv->snd_timer = timer; + timer_setup(&priv->tlist, snd_timer_s_function, 0); timer->private_data = priv; timer->private_free = snd_timer_free_system; return snd_timer_global_register(timer); -- 2.7.4 -- Kees Cook Pixel Security

1 0

[alsa-devel] [PATCH] ASoC: fsi: Use of_device_get_match_data() helper
by Geert Uytterhoeven 04 Oct '17

04 Oct '17

Use the of_device_get_match_data() helper instead of open coding. Note that when used with DT, there's always a valid match. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/fsi.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 6d3c7706d93fbc10..c3aaf4788557c4e1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1932,14 +1932,9 @@ static int fsi_probe(struct platform_device *pdev) core = NULL; if (np) { - const struct of_device_id *of_id; - - of_id = of_match_device(fsi_of_match, &pdev->dev); - if (of_id) { - core = of_id->data; - fsi_of_parse("fsia", np, &info.port_a, &pdev->dev); - fsi_of_parse("fsib", np, &info.port_b, &pdev->dev); - } + core = of_device_get_match_data(&pdev->dev); + fsi_of_parse("fsia", np, &info.port_a, &pdev->dev); + fsi_of_parse("fsib", np, &info.port_b, &pdev->dev); } else { const struct platform_device_id *id_entry = pdev->id_entry; if (id_entry) -- 2.7.4

2 1

[alsa-devel] usb/sound/bcd2000: warning in bcd2000_init_device
by Andrey Konovalov 04 Oct '17

04 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e19b205be43d11bff638cad4487008c48d21c103 (4.14-rc2). It seems that there's no check of the endpoint type. usb 1-1: BOGUS urb xfer, pipe 1 != type 3 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1846 at drivers/usb/core/urb.c:449 usb_submit_urb+0xf8a/0x11d0 Modules linked in: CPU: 0 PID: 1846 Comm: kworker/0:2 Not tainted 4.14.0-rc2-42613-g1488251d1a98 #238 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff880064296300 task.stack: ffff8800643b0000 RIP: 0010:usb_submit_urb+0xf8a/0x11d0 drivers/usb/core/urb.c:448 RSP: 0018:ffff8800643b6140 EFLAGS: 00010286 RAX: 0000000000000029 RBX: ffff880063842400 RCX: 0000000000000000 RDX: 0000000000000029 RSI: ffffffff85a58800 RDI: ffffed000c876c1a RBP: ffff8800643b6240 R08: 1ffff1000c876ac0 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 1ffff1000c876c2f R13: 0000000000000003 R14: 0000000000000001 R15: ffff88006325c768 FS: 0000000000000000(0000) GS:ffff88006c800000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fa2a51c0000 CR3: 000000006acc8000 CR4: 00000000000006f0 Call Trace: bcd2000_init_device sound/usb/bcd2000/bcd2000.c:289 bcd2000_init_midi sound/usb/bcd2000/bcd2000.c:345 bcd2000_probe+0xe64/0x19e0 sound/usb/bcd2000/bcd2000.c:406 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Code: 48 8b 85 30 ff ff ff 48 8d b8 98 00 00 00 e8 6e 91 8b ff 45 89 e8 44 89 f1 4c 89 fa 48 89 c6 48 c7 c7 c0 5a c8 85 e8 10 50 dd fd <0f> ff e9 9b f7 ff ff e8 ba cf 26 fe e9 80 f7 ff ff e8 a0 a5 f4 ---[ end trace bad127706d5fe2d6 ]---

5 15

[alsa-devel] [PATCH v2] ASoC: codecs: msm8916-wcd-analog: configure micbias in mbhc setup
by Damien Riegel 04 Oct '17

04 Oct '17

The very first time a headset is plugged in, detection is unreliable because bias hasn't been configured yet, it's done once a mechanical insertion interrupt has been triggered, so following insertions (and thus detections) are not affected. To fix the very first detection, the bias must also be configured in the function that setup the MBHC. Move pm8916_wcd_setup_mbhc after pm8916_mbhc_configure_bias to avoid a forward declaration. Signed-off-by: Damien Riegel <damien.riegel(a)savoirfairelinux.com> --- Changes in v2: - squash the two patches in one patch only sound/soc/codecs/msm8916-wcd-analog.c | 94 +++++++++++++++++++---------------- 1 file changed, 50 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index f562f2d86907..71494e9dbdcb 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -443,50 +443,6 @@ static int pm8916_wcd_analog_enable_micbias_int1(struct wcd->micbias1_cap_mode); } -static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) -{ - struct snd_soc_codec *codec = wcd->codec; - u32 plug_type = 0; - u32 int_en_mask; - - snd_soc_write(codec, CDC_A_MBHC_DET_CTL_1, - CDC_A_MBHC_DET_CTL_L_DET_EN | - CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION | - CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO | - CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN); - - if (wcd->hphl_jack_type_normally_open) - plug_type |= CDC_A_HPHL_PLUG_TYPE_NO; - - if (wcd->gnd_jack_type_normally_open) - plug_type |= CDC_A_GND_PLUG_TYPE_NO; - - snd_soc_write(codec, CDC_A_MBHC_DET_CTL_2, - CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 | - CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD | - plug_type | - CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN); - - - snd_soc_write(codec, CDC_A_MBHC_DBNC_TIMER, - CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS | - CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS); - - /* enable MBHC clock */ - snd_soc_update_bits(codec, CDC_D_CDC_DIG_CLK_CTL, - DIG_CLK_CTL_D_MBHC_CLK_EN_MASK, - DIG_CLK_CTL_D_MBHC_CLK_EN); - - int_en_mask = MBHC_SWITCH_INT; - if (wcd->mbhc_btn_enabled) - int_en_mask |= MBHC_BUTTON_PRESS_DET | MBHC_BUTTON_RELEASE_DET; - - snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); - snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); - wcd->mbhc_btn0_released = false; - wcd->detect_accessory_type = true; -} - static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv, bool micbias2_enabled) { @@ -534,6 +490,56 @@ static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv, return 0; } +static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) +{ + struct snd_soc_codec *codec = wcd->codec; + bool micbias_enabled = false; + u32 plug_type = 0; + u32 int_en_mask; + + snd_soc_write(codec, CDC_A_MBHC_DET_CTL_1, + CDC_A_MBHC_DET_CTL_L_DET_EN | + CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION | + CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO | + CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN); + + if (wcd->hphl_jack_type_normally_open) + plug_type |= CDC_A_HPHL_PLUG_TYPE_NO; + + if (wcd->gnd_jack_type_normally_open) + plug_type |= CDC_A_GND_PLUG_TYPE_NO; + + snd_soc_write(codec, CDC_A_MBHC_DET_CTL_2, + CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 | + CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD | + plug_type | + CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN); + + + snd_soc_write(codec, CDC_A_MBHC_DBNC_TIMER, + CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS | + CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS); + + /* enable MBHC clock */ + snd_soc_update_bits(codec, CDC_D_CDC_DIG_CLK_CTL, + DIG_CLK_CTL_D_MBHC_CLK_EN_MASK, + DIG_CLK_CTL_D_MBHC_CLK_EN); + + if (snd_soc_read(codec, CDC_A_MICB_2_EN) & CDC_A_MICB_2_EN_ENABLE) + micbias_enabled = true; + + pm8916_mbhc_configure_bias(wcd, micbias_enabled); + + int_en_mask = MBHC_SWITCH_INT; + if (wcd->mbhc_btn_enabled) + int_en_mask |= MBHC_BUTTON_PRESS_DET | MBHC_BUTTON_RELEASE_DET; + + snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); + snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); + wcd->mbhc_btn0_released = false; + wcd->detect_accessory_type = true; +} + static int pm8916_wcd_analog_enable_micbias_int2(struct snd_soc_dapm_widget *w, struct snd_kcontrol -- 2.14.1

3 4

[alsa-devel] [PATCH] ASoC: rockchip: Allocate enough memory so we don't overflow routes
by Douglas Anderson 04 Oct '17

04 Oct '17

In the recent commit d9f9c167edae ("ASoC: rockchip: Init dapm routes dynamically") we improperly allocated memory for the card->dapm_routes causing us to overflow the allocation on every boot. Oops. Let's allocate the correct amount of memory. We'll also add a check to make sure that we don't overrun memory even if we encounter some sort of weird device tree. Fixes: d9f9c167edae ("ASoC: rockchip: Init dapm routes dynamically") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 30eed83e8a13..d64fbbd50544 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -494,13 +494,17 @@ static int rockchip_sound_of_parse_dais(struct device *dev, struct snd_soc_dai_link *dai; struct snd_soc_dapm_route *routes; int i, index; + int num_routes; card->dai_link = devm_kzalloc(dev, sizeof(rockchip_dais), GFP_KERNEL); if (!card->dai_link) return -ENOMEM; - routes = devm_kzalloc(dev, sizeof(rockchip_routes), + num_routes = 0; + for (i = 0; i < ARRAY_SIZE(rockchip_routes); i++) + num_routes += rockchip_routes[i].num_routes; + routes = devm_kzalloc(dev, num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) return -ENOMEM; @@ -538,6 +542,12 @@ static int rockchip_sound_of_parse_dais(struct device *dev, dai->platform_of_node = np_cpu; dai->cpu_of_node = np_cpu; + if (card->num_dapm_routes + rockchip_routes[index].num_routes > + num_routes) { + dev_err(dev, "Too many routes\n"); + return -EINVAL; + } + memcpy(routes + card->num_dapm_routes, rockchip_routes[index].routes, rockchip_routes[index].num_routes * sizeof(*routes)); -- 2.14.2.822.g60be5d43e6-goog

2 1

[alsa-devel] [PATCH 0/3] ASoC: img-i2s-out: Add power management
by Ed Blake 04 Oct '17

04 Oct '17

This patch set adds power managment to the img-i2s-out driver. The first patch renames existing 'suspend' and 'resume' functions to 'runtime_suspend' and 'runtime_resume', which is what they actually are, the second patch adds system PM suspend / resume handling and the third patch adds control of the sys clock to runtime PM. Ed Blake (3): ASoC: img-i2s-out: Rename suspend / resume funcs ASoC: img-i2s-out: Add suspend / resume handling ASoC: img-i2s-out: Add control of sys clock to runtime PM sound/soc/img/img-i2s-out.c | 116 +++++++++++++++++++++++++++++++++----------- 1 file changed, 87 insertions(+), 29 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: img-parallel-out: Add pm_runtime_get/put to set_fmt callback
by Ed Blake 04 Oct '17

04 Oct '17

Add pm_runtime_get_sync and pm_runtime_put calls to set_fmt callback function. This fixes a bus error during boot when CONFIG_SUSPEND is defined when this function gets called while the device is runtime disabled and device registers are accessed while the clock is disabled. Signed-off-by: Ed Blake <ed.blake(a)sondrel.com> --- sound/soc/img/img-parallel-out.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 23b0f0f..2fc8a63 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -164,9 +164,11 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + pm_runtime_get_sync(prl->dev); reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL); reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set; img_prl_out_writel(prl, reg, IMG_PRL_OUT_CTL); + pm_runtime_put(prl->dev); return 0; } -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: codecs: msm8916-wcd-analog: fix micbias level
by Jean-François Têtu 04 Oct '17

04 Oct '17

The macro used to set the microphone bias level causes the snd_soc_write() call to overwrite other fields in the CDC_A_MICB_1_VAL register. The macro also does not return the proper level value to use. This fixes this by preserving all bits from the register that are not the level while setting the level. Signed-off-by: Jean-François Têtu <jean-francois.tetu(a)savoirfairelinux.com> --- sound/soc/codecs/msm8916-wcd-analog.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index f562f2d..e1534fe 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -104,7 +104,7 @@ #define CDC_A_MICB_1_VAL (0xf141) #define MICB_MIN_VAL 1600 #define MICB_STEP_SIZE 50 -#define MICB_VOLTAGE_REGVAL(v) ((v - MICB_MIN_VAL)/MICB_STEP_SIZE) +#define MICB_VOLTAGE_REGVAL(v) (((v - MICB_MIN_VAL)/MICB_STEP_SIZE) << 3) #define MICB_1_VAL_MICB_OUT_VAL_MASK GENMASK(7, 3) #define MICB_1_VAL_MICB_OUT_VAL_V2P70V ((0x16) << 3) #define MICB_1_VAL_MICB_OUT_VAL_V1P80V ((0x4) << 3) @@ -349,8 +349,9 @@ static void pm8916_wcd_analog_micbias_enable(struct snd_soc_codec *codec) | MICB_1_CTL_EXT_PRECHARG_EN_ENABLE); if (wcd->micbias_mv) { - snd_soc_write(codec, CDC_A_MICB_1_VAL, - MICB_VOLTAGE_REGVAL(wcd->micbias_mv)); + snd_soc_update_bits(codec, CDC_A_MICB_1_VAL, + MICB_1_VAL_MICB_OUT_VAL_MASK, + MICB_VOLTAGE_REGVAL(wcd->micbias_mv)); /* * Special headset needs MICBIAS as 2.7V so wait for * 50 msec for the MICBIAS to reach 2.7 volts. -- 2.7.4

3 2

[alsa-devel] [PATCH 0/6 v3] ASoC: rsnd: add MIX Volume Ramp support
by Kuninori Morimoto 04 Oct '17

04 Oct '17

Hi Mark These are v3 of MIX Volume Ramp support on Renesas R-Car sound. 0001 - 0005 are prepare/cleanup patch for it. v2 -> v3 - tidyup typo: INITIALIZEED -> INITIALIZED - flags macro Kuninori Morimoto (6): ASoC: rsnd: add generic rsnd_flags_xxx() macro ASoC: rsnd: use generic rsnd_flags_xxx() macro on ADG ASoC: rsnd: DVC kctrl sets once ASoC: rsnd: CTU kctrl sets once ASoC: rsnd: makes volume ramp rate list generic ASoC: rsnd: add MIX Volume Ramp support sound/soc/sh/rcar/adg.c | 13 ++-- sound/soc/sh/rcar/core.c | 27 ++++++++ sound/soc/sh/rcar/ctu.c | 8 +++ sound/soc/sh/rcar/dvc.c | 41 ++++-------- sound/soc/sh/rcar/mix.c | 158 +++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/rsnd.h | 12 +++- sound/soc/sh/rcar/ssi.c | 27 ++++---- 7 files changed, 227 insertions(+), 59 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH 0/6] ASoC: rsnd: add MIX Volume Ramp support
by Kuninori Morimoto 04 Oct '17

04 Oct '17

Hi Mark These adds MIX Volume Ramp support on Renesas R-Car sound. 0001 - 0005 are prepare/cleanup patch for it. Kuninori Morimoto (6): ASoC: rsnd: add generic rsnd_flags_xxx() macro ASoC: rsnd: use generic rsnd_flags_xxx() macro on ADG ASoC: rsnd: DVC kctrl sets once ASoC: rsnd: CTU kctrl sets once ASoC: rsnd: makes volume ramp rate list generic ASoC: rsnd: add MIX Volume Ramp support sound/soc/sh/rcar/adg.c | 13 ++-- sound/soc/sh/rcar/core.c | 27 ++++++++ sound/soc/sh/rcar/ctu.c | 8 +++ sound/soc/sh/rcar/dvc.c | 41 ++++-------- sound/soc/sh/rcar/mix.c | 158 +++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/rsnd.h | 12 +++- sound/soc/sh/rcar/ssi.c | 27 ++++---- 7 files changed, 227 insertions(+), 59 deletions(-) -- 1.9.1

3 12

[alsa-devel] [PATCH 0/6 v2] ASoC: rsnd: add MIX Volume Ramp support
by Kuninori Morimoto 04 Oct '17

04 Oct '17

Hi Mark These are v2 of MIX Volume Ramp support on Renesas R-Car sound. 0001 - 0005 are prepare/cleanup patch for it. Kuninori Morimoto (6): ASoC: rsnd: add generic rsnd_flags_xxx() macro ASoC: rsnd: use generic rsnd_flags_xxx() macro on ADG ASoC: rsnd: DVC kctrl sets once ASoC: rsnd: CTU kctrl sets once ASoC: rsnd: makes volume ramp rate list generic ASoC: rsnd: add MIX Volume Ramp support sound/soc/sh/rcar/adg.c | 13 ++-- sound/soc/sh/rcar/core.c | 27 ++++++++ sound/soc/sh/rcar/ctu.c | 8 +++ sound/soc/sh/rcar/dvc.c | 41 ++++-------- sound/soc/sh/rcar/mix.c | 158 +++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/rsnd.h | 12 +++- sound/soc/sh/rcar/ssi.c | 27 ++++---- 7 files changed, 227 insertions(+), 59 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCH 0/9] ASoC: cygnus: Various improvements and fixes
by Lori Hikichi 04 Oct '17

04 Oct '17

This patch series contains an number of improvements and refinements to the driver. There is also a fix for a problem when transferring four or more channels in TDM mode. Lori Hikichi (9): ASoC: cygnus: Add support for 384kHz frame rates ASoC: cygnus: Update bindings for audio clock changes ASoC: cygnus: Allow each port to select its clock source ASoC: cygnus: Only enable MCLK pins when in use ASoC: cygnus: Remove support for 8 bit audio and for mono ASoc: cygnus: Fix problems with multichannel transfers ASoC: cygnus: Remove set_fmt from SPDIF dai ops ASoC: cygnus: Add EXPORT_SYMBOL for helper function ASoC: cygnus: Tidy up of structure access .../bindings/sound/brcm,cygnus-audio.txt | 70 +- sound/soc/bcm/cygnus-pcm.c | 50 +- sound/soc/bcm/cygnus-ssp.c | 1336 ++++++++++---------- sound/soc/bcm/cygnus-ssp.h | 51 +- 4 files changed, 739 insertions(+), 768 deletions(-) -- 1.9.1

2 18

[alsa-devel] [PATCH v1 0/3] Minor updates to driver
by Lori Hikichi 04 Oct '17

04 Oct '17

This patchset contains a few small updates to the DAI driver. Lori Hikichi (3): ASoC: cygnus: Add EXPORT_SYMBOL for helper function ASoC: cygnus: Remove set_fmt from SPDIF dai ops ASoC: cygnus: Remove support for 8 bit audio and for mono sound/soc/bcm/cygnus-ssp.c | 32 +++++++++++++------------------- 1 file changed, 13 insertions(+), 19 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCH 0/5] ASoC: soc-core: component functions cleanup
by Kuninori Morimoto 04 Oct '17

04 Oct '17

Hi Mark These are component functions customize. And is preparation for platform replace to component. I'm posting platform replace patch-set [RFC] to ML. These are preparation but independent from it. Kuninori Morimoto (5): ASoC: soc-core: rename "cmpnt" to "component" ASoC: soc-core: add component lookup functions ASoC: soc-core: add component /unregister_exp/remove functions ASoC: soc-core: add snd_soc_add_component() ASoC: soc-devres: use expanded unregister on devm register component include/sound/soc.h | 16 ++++++-- sound/soc/soc-core.c | 102 ++++++++++++++++++++++++++++++++++--------------- sound/soc/soc-devres.c | 6 +-- 3 files changed, 87 insertions(+), 37 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCH] ASoC: codecs: msm8916-wcd-analog: fix module autoload
by Nicolas Dechesne 03 Oct '17

03 Oct '17

If the driver is built as a module, autoload won't work because the module alias information is not filled. So user-space can't match the registered device with the corresponding module. Export the module alias information using the MODULE_DEVICE_TABLE() macro. Before this patch: $ modinfo snd_soc_msm8916_analog | grep alias $ After this patch: $ modinfo snd_soc_msm8916_analog | grep alias alias: of:N*T*Cqcom,pm8916-wcd-analog-codecC* alias: of:N*T*Cqcom,pm8916-wcd-analog-codec Signed-off-by: Nicolas Dechesne <nicolas.dechesne(a)linaro.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 549c269acc7d..d79e5b35e003 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1241,6 +1241,8 @@ static const struct of_device_id pm8916_wcd_analog_spmi_match_table[] = { { } }; +MODULE_DEVICE_TABLE(of, pm8916_wcd_analog_spmi_match_table); + static struct platform_driver pm8916_wcd_analog_spmi_driver = { .driver = { .name = "qcom,pm8916-wcd-spmi-codec", -- 2.11.0

2 1

[alsa-devel] "alsactl store" fails to save Echo Gina3G mixer
by S. Christian Collins 03 Oct '17

03 Oct '17

Greetings, I have two PCI sound cards in my PC: 1. Sound Blaster Audigy 2 ZS Gold (primary card) 2. Echo Gina3G If I save the mixer for the Audigy 2 via: alsactl -f ~/asound0.state store 0 ...the Audigy 2's mixer is properly saved. However, when I try to save the Gina3G mixer: alsactl -f ~/asound1.state store 1 ...I get the following error: alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. Aborted I have tried to find any info on this error online, but to no avail. Does anybody know what is going on here? ** My System ** OS: KDE Neon 5.10.5 64-bit (Plasma Desktop 5.10.5, KDE Frameworks 5.37.0, Qt 5.9.1) Linux Kernel: 4.10.0.32-generic Motherboard: ASRock X58 Extreme3 (Intel X58 chipset) CPU: Intel Core i7 930 (2.8 GHz quad-core) RAM: 12GB DDR3 Video: NVIDIA GeForce GTX 970 w/ 4GB RAM (PCI Express) Sound Card #1: Sound Blaster Audigy 2 ZS Gold Sound Card #2: Echo Gina3G NVIDIA video driver: 375.66

3 11

[alsa-devel] Applied "ASoC: codecs: msm8916-wcd-analog: fix module autoload" to the asoc tree
by Mark Brown 03 Oct '17

03 Oct '17

The patch ASoC: codecs: msm8916-wcd-analog: fix module autoload has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 46d69e141d479585c105a4d5b2337cd2ce6967e5 Mon Sep 17 00:00:00 2001 From: Nicolas Dechesne <nicolas.dechesne(a)linaro.org> Date: Tue, 3 Oct 2017 11:49:51 +0200 Subject: [PATCH] ASoC: codecs: msm8916-wcd-analog: fix module autoload If the driver is built as a module, autoload won't work because the module alias information is not filled. So user-space can't match the registered device with the corresponding module. Export the module alias information using the MODULE_DEVICE_TABLE() macro. Before this patch: $ modinfo snd_soc_msm8916_analog | grep alias $ After this patch: $ modinfo snd_soc_msm8916_analog | grep alias alias: of:N*T*Cqcom,pm8916-wcd-analog-codecC* alias: of:N*T*Cqcom,pm8916-wcd-analog-codec Signed-off-by: Nicolas Dechesne <nicolas.dechesne(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index f562f2d86907..3593c578e3e7 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1239,6 +1239,8 @@ static const struct of_device_id pm8916_wcd_analog_spmi_match_table[] = { { } }; +MODULE_DEVICE_TABLE(of, pm8916_wcd_analog_spmi_match_table); + static struct platform_driver pm8916_wcd_analog_spmi_driver = { .driver = { .name = "qcom,pm8916-wcd-spmi-codec", -- 2.14.1

1 0

[alsa-devel] [PATCH] ASoC: rcar: skip disabled-SSI nodes
by Kuninori Morimoto 03 Oct '17

03 Oct '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The current device tree representation of the R-Car SSI assumes that they are numbered consecutively, starting from 0. Alas, this is not the case with the R8A77995 (D3) SoC which SSI1/SSI2 aren't present. In order to keep the existing device trees working, I'm suggesting to use a disabled node for SSI0/SSI1. Teach the SSI probe to just skip disabled nodes. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fffc07e..03c2a1f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1112,6 +1112,9 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) i = 0; for_each_child_of_node(node, np) { + if (!of_device_is_available(np)) + goto skip; + ssi = rsnd_ssi_get(priv, i); snprintf(name, RSND_SSI_NAME_SIZE, "%s.%d", @@ -1148,7 +1151,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) of_node_put(np); goto rsnd_ssi_probe_done; } - +skip: i++; } -- 1.9.1

2 1

[alsa-devel] [alsa-lib][PATCH] test/seq-decoder: enable timestamping for external subscribers
by Antonio Ospite 03 Oct '17

03 Oct '17

Events sent by external clients subscribed to the input port are not timestamped. This inconsistent behavior may surprise newbies who look at seq-decoder as a reference example. See the example below using "vkeybd --addr 128:0" to connect to seq-decoder, the events sent by vkeybd are on a different queue with no timestamps: ... EVENT>>> Type = 66, flags = 0x0, time = 0 ticks Source = 0.1, dest = 128.0, queue = 253 Event = Port Subscribed; 129:0 -> 128:0 EVENT>>> Type = 66, flags = 0x1, time = 4.829712627 Source = 0.1, dest = 128.0, queue = 0 Event = Port Subscribed; 129:0 -> 128:0 EVENT>>> Type = 10, flags = 0x0, time = 0 ticks Source = 129.0, dest = 128.0, queue = 253 Event = Controller; ch=0, param=0, value=0 EVENT>>> Type = 11, flags = 0x0, time = 0 ticks Source = 129.0, dest = 128.0, queue = 253 Event = Program Change; ch=0, program=0 ... After the change events are on the main queue and are timestamped: ... EVENT>>> Type = 66, flags = 0x1, time = 4.280907223 Source = 0.1, dest = 128.0, queue = 0 Event = Port Subscribed; 129:0 -> 128:0 EVENT>>> Type = 66, flags = 0x1, time = 4.280912063 Source = 0.1, dest = 128.0, queue = 0 Event = Port Subscribed; 129:0 -> 128:0 EVENT>>> Type = 10, flags = 0x1, time = 4.280990702 Source = 129.0, dest = 128.0, queue = 0 Event = Controller; ch=0, param=0, value=0 EVENT>>> Type = 11, flags = 0x1, time = 4.280994862 Source = 129.0, dest = 128.0, queue = 0 Event = Program Change; ch=0, program=0 ... Signed-off-by: Antonio Ospite <ao2(a)ao2.it> --- Hi, while looking into improving the alsamidisrc[1,2] GStreamer element, I took a better look at what the sequencer API could do and started using test/seq-decoder.c as a reference example. I noticed the issue discussed above and I tried to fix it in order to have a more consistent behavior of event timestamping. I think the new behavior is less surprising for newbies but I may still be missing something, please let me know if I am. Thanks, Antonio [1] https://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/alsa/gstal… [2] https://bugzilla.gnome.org/show_bug.cgi?id=787683 test/seq-decoder.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/test/seq-decoder.c b/test/seq-decoder.c index b110e98d..38755553 100644 --- a/test/seq-decoder.c +++ b/test/seq-decoder.c @@ -283,6 +283,12 @@ void event_decoder(snd_seq_t *handle, int argc, char *argv[]) snd_seq_port_info_set_name(pinfo, "Input"); snd_seq_port_info_set_type(pinfo, SND_SEQ_PORT_TYPE_MIDI_GENERIC); snd_seq_port_info_set_capability(pinfo, SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_SUBS_WRITE); + + /* Enable timestamping for events sent by external subscribers. */ + snd_seq_port_info_set_timestamping(pinfo, 1); + snd_seq_port_info_set_timestamp_real(pinfo, 1); + snd_seq_port_info_set_timestamp_queue(pinfo, queue); + if ((err = snd_seq_create_port(handle, pinfo)) < 0) { fprintf(stderr, "Cannot create input port: %s\n", snd_strerror(err)); return; -- 2.14.1

2 1

[alsa-devel] [alsa-lib][PATCH] seq: fix snd_seq_set_queue_tempo() usage example in the documentation
by Antonio Ospite 03 Oct '17

03 Oct '17

snd_seq_set_queue_tempo() requires a queue id as the second argument, fix the example in documentation to reflect that. Also add the queue id as an argument of the set_tempo() function, just to keep the whole example compilable. Signed-off-by: Antonio Ospite <ao2(a)ao2.it> --- src/seq/seq.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/seq/seq.c b/src/seq/seq.c index d5ed1c6a..808c7915 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -452,13 +452,13 @@ For setting these tempo parameters, use #snd_seq_queue_tempo_t record. For example, to set the tempo of the queue <code>q</code> to 48 PPQ, 60 BPM, \code -void set_tempo(snd_seq_t *handle) +void set_tempo(snd_seq_t *handle, int queue) { snd_seq_queue_tempo_t *tempo; snd_seq_queue_tempo_alloca(&tempo); snd_seq_queue_tempo_set_tempo(tempo, 1000000); // 60 BPM snd_seq_queue_tempo_set_ppq(tempo, 48); // 48 PPQ - snd_seq_set_queue_tempo(handle, tempo); + snd_seq_set_queue_tempo(handle, queue, tempo); } \endcode -- 2.14.2

3 3

[alsa-devel] ASoC Sound Driver I2S over LPE, Intel Baytrail SoC. MinnowBoard Max
by Michael Pogrebinsky 02 Oct '17

02 Oct '17

Hi all, I have the MinnowBoard Max with Baytrail Atom (ValleyView) SoC. I'm attempting to write an Alsa (ASoC) driver for my sound card to work with this board. PCM should work through I2S for this sound card I have. >From my understanding Baytrail's I2S goes over LPE. I read this thread:https://www.mail-archive.com/elinux-minnowboard@lists.elinux.org/msg… So I understand that I need the 3 components for my ASoC driver. However I am having trouble at understanding how to communicate with the LPE to get the I2S working. Here is what I have done so far: I am running Ubuntu 14.04 LTS, kernel : 4.3.0-040300rc4 First of all I downloaded the firmware fw_sst_0f28.bin-48kHz_i2s_master fromhttps://kernel.googlesource.com/pub/scm/linux/kernel/git/firmware/linux… And put it /lib/firmware/intel/. Later I went to the Bios and enabled LPE in PCI mode. I do see the LPE controller now, but it shows as UNCLAIMED. Later I looked at the sound/soc/inte/baytrail/ and from my understanding in order for me to get the I2S working I need to load this module so I did modprobe snd-soc-sst-baytrail-pcm. Later I wrote a simple codec and machine driver (looking at the example of sound/soc/intel/boards/byt_max98090) however everytime I attempt to register the sound card with snd_soc_register_card, I get the following error message in dmesg : "ASoC: CPU DAI baytrail-pcm-audio not registered". >From this I deduct that although I loaded the snd-soc-sst-baytrail-pcm module, the driver has not actually been loaded. I'm clueless on how to proceed from here. I would highly appreciate your help.

3 7

[alsa-devel] usb/sound/usx2y: warning in usb_stream_new/__alloc_pages_slowpath
by Andrey Konovalov 02 Oct '17

02 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit ebb2c2437d8008d46796902ff390653822af6cc4 (Sep 18). ------------[ cut here ]------------ WARNING: CPU: 1 PID: 1846 at mm/page_alloc.c:3883 __alloc_pages_slowpath+0x1ef2/0x2d70 Modules linked in: CPU: 1 PID: 1846 Comm: kworker/1:2 Not tainted 4.14.0-rc1-42251-gebb2c2437d80 #215 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff8800643c18c0 task.stack: ffff88006b658000 RIP: 0010:trace_reclaim_retry_zone ./include/trace/events/oom.h:31 RIP: 0010:should_reclaim_retry mm/page_alloc.c:3783 RIP: 0010:__alloc_pages_slowpath+0x1ef2/0x2d70 mm/page_alloc.c:4039 RSP: 0018:ffff88006b65d938 EFLAGS: 00010246 RAX: 00000000ffffa666 RBX: 00000000014000c0 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000034 RDI: 000000000140c0c0 RBP: ffff88006b65e088 R08: 0000000000000000 R09: fffffffffff00f88 R10: 0000000000000000 R11: 0000000000000085 R12: ffff88006b65e130 R13: ffff88006b65e270 R14: ffff88006b65e0f0 R15: 000000000140c0c0 FS: 0000000000000000(0000) GS:ffff88006c900000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000020829000 CR3: 0000000063cce000 CR4: 00000000000006e0 Call Trace: __alloc_pages_nodemask+0x921/0xf70 mm/page_alloc.c:4217 alloc_pages_current+0xbb/0x1f0 mm/mempolicy.c:2035 alloc_pages ./include/linux/gfp.h:505 __get_free_pages+0x14/0x50 mm/page_alloc.c:4248 usb_stream_new+0x50f/0x9f0 sound/usb/usx2y/usb_stream.c:214 us122l_start+0xb0/0x250 sound/usb/usx2y/us122l.c:355 us122l_create_card sound/usb/usx2y/us122l.c:502 us122l_usb_probe sound/usb/usx2y/us122l.c:588 snd_us122l_probe+0xa62/0x15d0 sound/usb/usx2y/us122l.c:623 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Code: be 47 00 00 00 48 c7 c7 20 42 8b 85 c6 05 5e 35 99 05 01 e8 91 1f cb ff 48 8b 8d d8 f8 ff ff 44 8b 9d 50 f9 ff ff e9 57 fc ff ff <0f> ff e9 79 e3 ff ff 0f ff c7 85 e8 f8 ff ff 00 00 00 00 89 d8 ---[ end trace 78800971cbd5f94f ]---

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[alsa-devel] [PATCH] ASoC: rockchip: Fix wrong allocation size of dapm routes
by Jeffy Chen 02 Oct '17

02 Oct '17

The allocation size of dapm routes is wrong, correct it. Fixes: d9f9c167edae ("ASoC: rockchip: Init dapm routes dynamically") Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 30eed83e8a13..008452e55ef8 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -493,14 +493,18 @@ static int rockchip_sound_of_parse_dais(struct device *dev, struct device_node *np_codec; struct snd_soc_dai_link *dai; struct snd_soc_dapm_route *routes; - int i, index; + int i, index, max_num_routes; card->dai_link = devm_kzalloc(dev, sizeof(rockchip_dais), GFP_KERNEL); if (!card->dai_link) return -ENOMEM; - routes = devm_kzalloc(dev, sizeof(rockchip_routes), + max_num_routes = 0; + for (i = 0; i < ARRAY_SIZE(rockchip_dais); i++) + max_num_routes += rockchip_routes[i].num_routes; + + routes = devm_kzalloc(dev, max_num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) return -ENOMEM; -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: line6: make snd_pcm_ops const
by Bhumika Goyal 02 Oct '17

02 Oct '17

Make these const as they are only passed to a const argument of the function snd_pcm_set_ops in the file referencing them. Also, add const to the declaration in the headers. Structures found using Coccinelle and changes done by hand. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/usb/line6/capture.c | 2 +- sound/usb/line6/capture.h | 2 +- sound/usb/line6/playback.c | 2 +- sound/usb/line6/playback.h | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c index 7c81256..947d616 100644 --- a/sound/usb/line6/capture.c +++ b/sound/usb/line6/capture.c @@ -248,7 +248,7 @@ static int snd_line6_capture_close(struct snd_pcm_substream *substream) } /* capture operators */ -struct snd_pcm_ops snd_line6_capture_ops = { +const struct snd_pcm_ops snd_line6_capture_ops = { .open = snd_line6_capture_open, .close = snd_line6_capture_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/line6/capture.h b/sound/usb/line6/capture.h index 890b21b..b67ccc3 100644 --- a/sound/usb/line6/capture.h +++ b/sound/usb/line6/capture.h @@ -17,7 +17,7 @@ #include "driver.h" #include "pcm.h" -extern struct snd_pcm_ops snd_line6_capture_ops; +extern const struct snd_pcm_ops snd_line6_capture_ops; extern void line6_capture_copy(struct snd_line6_pcm *line6pcm, char *fbuf, int fsize); diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 812d181..819e9b2 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -393,7 +393,7 @@ static int snd_line6_playback_close(struct snd_pcm_substream *substream) } /* playback operators */ -struct snd_pcm_ops snd_line6_playback_ops = { +const struct snd_pcm_ops snd_line6_playback_ops = { .open = snd_line6_playback_open, .close = snd_line6_playback_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/line6/playback.h b/sound/usb/line6/playback.h index 51fce29..d8d3b8a 100644 --- a/sound/usb/line6/playback.h +++ b/sound/usb/line6/playback.h @@ -27,7 +27,7 @@ */ #define USE_CLEAR_BUFFER_WORKAROUND 1 -extern struct snd_pcm_ops snd_line6_playback_ops; +extern const struct snd_pcm_ops snd_line6_playback_ops; extern int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm); extern int line6_submit_audio_out_all_urbs(struct snd_line6_pcm *line6pcm); -- 1.9.1

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[alsa-devel] [PATCH] Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"
by Takashi Sakamoto 02 Oct '17

02 Oct '17

This reverts commit 275353bb684e to fix a regression which can abort 'alsactl' program in alsa-utils due to assertion in alsa-lib. alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. This commit is a band-aid. In a point of usage of ALSA control interface, the drivers still bring an issue that they prevent userspace applications to have a consistent way to parse each levels of the dimension information via ALSA control interface. Let me investigate this issue. Current implementation of the drivers have three control element sets with dimension information: * 'Monitor Mixer Volume' (type: integer) * 'VMixer Volume' (type: integer) * 'VU-meters' (type: boolean) Although the number of elements named as 'Monitor Mixer Volume' differs depending on drivers in this group, it can be calculated by macros defined by each driver (= (BX_NUM - BX_ANALOG_IN) * BX_ANALOG_IN). Each of the elements has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * (BX_NUM - BX_ANALOG_IN)). For these elements, userspace applications are expected to handle the dimension information so that all of the elements construct a matrix where the number of rows and columns are represented by the dimension information. The same way is applied to elements named as 'VMixer Volume'. The number of these elements can also be calculated by macros defined by each drivers (= PX_ANALOG_IN * BX_ANALOG_IN). Each of the element has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * PX_ANALOG_IN). All of the elements construct a matrix with the dimension information. An element named as 'VU-meters' gets a different way in a point of dimension information. The element includes 96 members for value. The element has dimension information with 3 levels (= 3 or 2 * 16 * 2). For this element, userspace applications are expected to handle the dimension information so that all of the members for value construct a matrix where the number of rows and columns are represented by the dimension information. This is different from the way for the former. As a summary, the drivers were not designed to produce a consistent way to parse the dimension information. This makes it hard for general userspace applications such as amixer to parse the information by a consistent way, and actually no userspace applications except for 'echomixer' utilize the dimension information. Additionally, no drivers excluding this group use the information. The reverted commit was written based on the latter way. A commit 860c1994a70a ('ALSA: control: add dimension validator for userspace elements') is written based on the latter way, too. The patch should be reconsider too in the same time to re-define a consistent way to parse the dimension information. Reported-by: Mark Hills <mark(a)xwax.org> Reported-by: S. Christian Collins <s.chriscollins(a)gmail.com> Fixes: 275353bb684e ('ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members') Cc: <stable(a)vger.kernel.org> # v4.8+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/pci/echoaudio/echoaudio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 7326695bca33..d68f99e076a8 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1272,11 +1272,11 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_busses_in(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1344,11 +1344,11 @@ static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_pipes_out(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1728,6 +1728,7 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 96; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = 0; #ifdef ECHOCARD_HAS_VMIXER @@ -1737,7 +1738,6 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, #endif uinfo->dimen.d[1] = 16; /* 16 channels */ uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */ - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1] * uinfo->dimen.d[2]; return 0; } -- 2.11.0

2 1

[alsa-devel] [PATCH 0/2] ASoC: img-spdif-in: Add power management
by Ed Blake 02 Oct '17

02 Oct '17

This patch set adds power managment to the img-spdif-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Ed Blake (2): ASoC: img-spdif-in: Add suspend / resume handling ASoC: img-spdif-in: Add runtime PM sound/soc/img/img-spdif-in.c | 111 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 99 insertions(+), 12 deletions(-) -- 1.9.1

1 2

[alsa-devel] [PATCH v1 0/2] Fix first detection issue with headset
by Damien Riegel 02 Oct '17

02 Oct '17

This series fix an issue with headset being misidentified on the very first time a device is plugged in by calling the function that configures the bias when the MBHC is setup. I've split this series into two patches because pm8916_wcd_setup_mbhc needs to be moved a few lines below to avoid forward declaring it. It is overkill but that was not part of the code snippet that Srinivas acked, and I wanted to be extra sure not to misattribute an Acked-by. Damien Riegel (2): ASoC: codecs: msm8916-wcd-analog: move pm8916_wcd_setup_mbhc ASoC: codecs: msm8916-wcd-analog: configure micbias in mbhc setup sound/soc/codecs/msm8916-wcd-analog.c | 94 +++++++++++++++++++---------------- 1 file changed, 50 insertions(+), 44 deletions(-) -- 2.14.1

2 3

[alsa-devel] [PATCH v2] ALSA: hda - program ICT bits to support HBR audio
by Pierre-Louis Bossart 02 Oct '17

02 Oct '17

From: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> On recent Intel platforms (Haswell, Broadwell, Skylake, ApolloLake, KabyLake, ...), the IEC Coding Type (ICT) bitfield in the Digital Converter Control #3 needs to be set explicitly for HDMI/DisplayPort High Bit Rate (HBR) audio playback to work. This was not required in earlier platforms when HBR was first introduced. The ICT bits are defined in Section 7.3.3.9 of the HDaudio 1.0a specification. Since the ICT bitfield was not specified for HDAudio 1.0 devices (before 2009), we only program it on machines more recent than Haswell. We tested that this fix is not needed on Baytrail-I (MinnowBoard Turbot) and believe by extension it also does not apply to Braswell. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=98797 Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v2: move digi3 verb in Set section, only set ICT for haswell and newer include/sound/hda_verbs.h | 1 + sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index d0509db6d0ec..d2004d7feb2b 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -54,6 +54,7 @@ enum { #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d #define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ +#define AC_VERB_SET_DIGI_CONVERT_3 0x73e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 53f9311370de..8566de298ea3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -906,6 +906,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; + unsigned int param; int err; err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); @@ -915,6 +916,26 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return err; } + if (is_haswell_plus(codec)) { + + /* + * on recent platforms IEC Coding Type is required for HBR + * support, read current Digital Converter settings and set + * ICT bitfield if needed. + */ + param = snd_hda_codec_read(codec, cvt_nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + + param = (param >> 16) & ~(AC_DIG3_ICT); + + /* on recent platforms ICT mode is required for HBR support */ + if (is_hbr_format(format)) + param |= 0x1; + + snd_hda_codec_write(codec, cvt_nid, 0, + AC_VERB_SET_DIGI_CONVERT_3, param); + } + snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); return 0; } -- 2.11.0

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[alsa-devel] [PATCH 0/2] ASoC: soc-core: add missing component functions
by Kuninori Morimoto 02 Oct '17

02 Oct '17

Hi Mark These are resend of one of "replace to component" patch set. These add missing functions which exist in "platforms" but not exist in "component". Kuninori Morimoto (2): ASoC: soc-core: add component lookup functions ASoC: soc-core: add snd_soc_add_component() include/sound/soc.h | 7 ++++++ sound/soc/soc-core.c | 60 ++++++++++++++++++++++++++++++++++++++++++---------- 2 files changed, 56 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] 4.14-rc2: audio stopped working?
by Pavel Machek 29 Sep '17

29 Sep '17

Hi! I've had sound problems in chromium for long long time. It produces choppy auduo just after boot and then it works, probably after chromium restart. May have something to do with pulseaudio. Not sure. Anyway... sound in flightgear usually worked, and now it does not. mplayer works, chromium works (sometimes choppy, as usual). Any ideas? Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

1 1

[alsa-devel] [RFC] ASoC: codecs: msm8916-wcd-analog: use btn0 released detection
by Damien Riegel 28 Sep '17

28 Sep '17

msm8916-wcd-analog uses button0 to differentiate between headphone and headset. Under some circumstances, button pressed and released interrupts are not fired as the driver expects it. For instance, with some connectors, there are spurious button-pressed interrupts when unplugging a headphone, without the corresponding button-released interrupt. But the codec always alternates between button pressed and released interrupts, it cannot fire two interrupts of the same kind in a row. That means that when the headphone is plugged back, only a button-released interrupt will be fired instead of pressed then released. This causes the driver to report headphone as headset. By changing the logic and relying on button 0 release interrupt, the driver could be made more robust for connectors that differ from the one used on the Dragonboard's audio mezzanine. Signed-off-by: Damien Riegel <damien.riegel(a)savoirfairelinux.com> --- sound/soc/codecs/msm8916-wcd-analog.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 549c269acc7d..f562f2d86907 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -285,7 +285,7 @@ struct pm8916_wcd_analog_priv { u16 codec_version; bool mbhc_btn_enabled; /* special event to detect accessory type */ - bool mbhc_btn0_pressed; + int mbhc_btn0_released; bool detect_accessory_type; struct clk *mclk; struct snd_soc_codec *codec; @@ -483,7 +483,7 @@ static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); - wcd->mbhc_btn0_pressed = false; + wcd->mbhc_btn0_released = false; wcd->detect_accessory_type = true; } @@ -950,7 +950,7 @@ static irqreturn_t mbhc_btn_release_irq_handler(int irq, void *arg) /* check if its BTN0 thats released */ if ((val != -1) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK)) - priv->mbhc_btn0_pressed = false; + priv->mbhc_btn0_released = true; } else { snd_soc_jack_report(priv->jack, 0, btn_mask); @@ -983,9 +983,7 @@ static irqreturn_t mbhc_btn_press_irq_handler(int irq, void *arg) break; case 0x0: /* handle BTN_0 specially for type detection */ - if (priv->detect_accessory_type) - priv->mbhc_btn0_pressed = true; - else + if (!priv->detect_accessory_type) snd_soc_jack_report(priv->jack, SND_JACK_BTN_0, btn_mask); break; @@ -1029,19 +1027,19 @@ static irqreturn_t pm8916_mbhc_switch_irq_handler(int irq, void *arg) * both press and release event received then its * a headset. */ - if (priv->mbhc_btn0_pressed) + if (priv->mbhc_btn0_released) snd_soc_jack_report(priv->jack, - SND_JACK_HEADPHONE, hs_jack_mask); + SND_JACK_HEADSET, hs_jack_mask); else snd_soc_jack_report(priv->jack, - SND_JACK_HEADSET, hs_jack_mask); + SND_JACK_HEADPHONE, hs_jack_mask); priv->detect_accessory_type = false; } else { /* removal */ snd_soc_jack_report(priv->jack, 0, hs_jack_mask); priv->detect_accessory_type = true; - priv->mbhc_btn0_pressed = false; + priv->mbhc_btn0_released = false; } return IRQ_HANDLED; -- 2.14.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix jack name format substitution
by Chintan Patel 28 Sep '17

28 Sep '17

Jack name is not getting formatted correctly hence resulting in invalid name for HDMI/DP input devices. This was recently exposed due changes brought by MST: commit 3a13347f05fd ("ASoC: Intel: kbl: Add jack port initialize in kbl machine drivers") Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 4689991bcc86..c127b7b0ee3f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -611,6 +611,8 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { codec = pcm->codec_dai->codec; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &ctx->kabylake_hdmi[i], NULL, 0); -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: rt5670: add set_bclk_ratio in dai ops
by Bard Liao 28 Sep '17

28 Sep '17

We need to set a specific bit for 50 bclk rate. So add set_bclk_ratio function to set the bit. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2: * add define for tdm data format select bit. v3: * change if (ratio == 50) to if ((ratio % 50) == 0) since the bit needs to be set if bclk ratio is divisible by 50. --- sound/soc/codecs/rt5670.c | 19 +++++++++++++++++++ sound/soc/codecs/rt5670.h | 4 ++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9545764..78517d8 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2582,6 +2582,24 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5670_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (dai->id != RT5670_AIF1) + return 0; + + if ((ratio % 50) == 0) + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_50FS); + else + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_NOR); + + return 0; +} + static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2712,6 +2730,7 @@ static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_fmt = rt5670_set_dai_fmt, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, + .set_bclk_ratio = rt5670_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5670_dai[] = { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 5ba485c..265df80 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1816,6 +1816,10 @@ #define RT5670_ZCD_HP_DIS (0x0 << 15) #define RT5670_ZCD_HP_EN (0x1 << 15) +/* General Control 3 (0xfc) */ +#define RT5670_TDM_DATA_MODE_SEL (0x1 << 11) +#define RT5670_TDM_DATA_MODE_NOR (0x0 << 11) +#define RT5670_TDM_DATA_MODE_50FS (0x1 << 11) /* Codec Private Register definition */ /* 3D Speaker Control (0x63) */ -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 28 Sep '17

28 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in this version: - Review comments from charles to move common function of error handling in seperate function. Changes in v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 145 +++++++++++++++++++++++++++++++-------------------- 1 file changed, 89 insertions(+), 56 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..5dae9d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,27 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +void +snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, + unsigned long *private_value, + int num_params, + const char **w_param_text) +{ + int count; + + devm_kfree(card->dev, (void *)*private_value); + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); +} + +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params, const char **w_param_text, + unsigned long *private_value) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3807,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; - - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); - if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + int count; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,47 +3826,87 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ - private_value = + *private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), sizeof(struct soc_enum), GFP_KERNEL); - if (!private_value) { + if (!*private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } - kcontrol_dai_link[0].private_value = private_value; + kcontrol_dai_link[0].private_value = *private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_private_value; + goto outfree_w_param; } + return kcontrol_news; +outfree_w_param: + snd_soc_dapm_free_kcontrol(card, private_value, num_params, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + const char **w_param_text; + unsigned long private_value; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params, + w_param_text, &private_value); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto param_fail; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3938,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); -outfree_link_name: + snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); +param_fail: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

3 3

[alsa-devel] [PATCH] ASoC: kirkwood: make kirkwood_soc_platform const
by Bhumika Goyal 27 Sep '17

27 Sep '17

Make kirkwood_soc_platform const as it only passed to a const argument of the function snd_soc_register_platform in the file referencing it. Make the declaration const too. Done using Coccinelle. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/kirkwood/kirkwood.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cf23af1..505b0ff 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -318,7 +318,7 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) } } -struct snd_soc_platform_driver kirkwood_soc_platform = { +const struct snd_soc_platform_driver kirkwood_soc_platform = { .ops = &kirkwood_dma_ops, .pcm_new = kirkwood_dma_new, .pcm_free = kirkwood_dma_free_dma_buffers, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 90e32a7..783cb1a4 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -143,6 +143,6 @@ struct kirkwood_dma_data { int burst; }; -extern struct snd_soc_platform_driver kirkwood_soc_platform; +extern const struct snd_soc_platform_driver kirkwood_soc_platform; #endif -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Kabylake machine driver changes for a FE DAI
by Harsha Priya 27 Sep '17

27 Sep '17

'Kbl Audio Headset Playback' FE DAI also needs SSP hw parama fix. Adding this dai also to be handled in kabylake_ssp_fixup() call. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 88ff542..e767283 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -302,6 +302,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || + !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; channels->min = channels->max = 2; -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: pcm: Sync delayed work before releasing resources
by Takashi Iwai 27 Sep '17

27 Sep '17

When ASoC driver is unbound dynamically during its operation (i.e. a kind of hot-unplug), we may hit Oops due to the resource access after the release by a delayed work, something like: Unable to handle kernel paging request at virtual address dead000000000220 .... PC is at soc_dapm_dai_stream_event.isra.14+0x20/0xd0 LR is at snd_soc_dapm_stream_event+0x74/0xa8 .... [<ffff000008715610>] soc_dapm_dai_stream_event.isra.14+0x20/0xd0 [<ffff00000871989c>] snd_soc_dapm_stream_event+0x74/0xa8 [<ffff00000871b23c>] close_delayed_work+0x3c/0x50 [<ffff0000080bbd6c>] process_one_work+0x1ac/0x318 [<ffff0000080bbf20>] worker_thread+0x48/0x420 [<ffff0000080c201c>] kthread+0xfc/0x128 [<ffff0000080842f0>] ret_from_fork+0x10/0x18 For fixing the race, this patch adds a sync-point in pcm private_free callback to finish the delayed work before actually releasing the resources. Reported-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-pcm.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 94b88b897c3b..c0f0b09cb433 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2632,6 +2632,17 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } +static void soc_pcm_private_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_soc_platform *platform = rtd->platform; + + /* need to sync the delayed work before releasing resources */ + flush_delayed_work(&rtd->delayed_work); + if (platform->driver->pcm_free) + platform->driver->pcm_free(pcm); +} + /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -2757,7 +2768,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) } } - pcm->private_free = platform->driver->pcm_free; + pcm->private_free = soc_pcm_private_free; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, -- 2.14.0

3 4

[alsa-devel] [PATCH] ASoC: dapm: Fix typos in comment
by Peng Donglin 27 Sep '17

27 Sep '17

Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..0867eaa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4167,7 +4167,7 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, * @dapm: DAPM context * @pin: pin name * - * Enables input/output pin and its parents or children widgets iff there is + * Enables input/output pin and its parents or children widgets if there is * a valid audio route and active audio stream. * * Requires external locking. @@ -4187,7 +4187,7 @@ int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, * @dapm: DAPM context * @pin: pin name * - * Enables input/output pin and its parents or children widgets iff there is + * Enables input/output pin and its parents or children widgets if there is * a valid audio route and active audio stream. * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to -- 1.9.1

3 2

[alsa-devel] usb/sound/caiaq: warning in init_card/usb_submit_urb
by Andrey Konovalov 27 Sep '17

27 Sep '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e19b205be43d11bff638cad4487008c48d21c103 (4.14-rc2). It seems that there's no check on the actual type of the endpoint. usb 1-1: BOGUS urb xfer, pipe 3 != type 1 ------------[ cut here ]------------ WARNING: CPU: 1 PID: 1150 at drivers/usb/core/urb.c:449 usb_submit_urb+0xf8a/0x11d0 Modules linked in: CPU: 1 PID: 1150 Comm: kworker/1:1 Not tainted 4.14.0-rc2-42660-g24b7bd59eec0 #277 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff880069428000 task.stack: ffff880069440000 RIP: 0010:usb_submit_urb+0xf8a/0x11d0 drivers/usb/core/urb.c:448 RSP: 0018:ffff8800694462e8 EFLAGS: 00010282 RAX: 0000000000000029 RBX: ffff88006a570ef0 RCX: 0000000000000000 RDX: 0000000000000029 RSI: ffffffff85876dc0 RDI: ffffed000d288c4f RBP: ffff8800694463e8 R08: 1ffff1000d288af5 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 1ffff1000d288c64 R13: 0000000000000001 R14: 0000000000000003 R15: ffff8800697c4288 FS: 0000000000000000(0000) GS:ffff88006c900000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007facf21b4000 CR3: 000000006a6d4000 CR4: 00000000000006e0 Call Trace: init_card sound/usb/caiaq/device.c:467 snd_probe+0x81c/0x1150 sound/usb/caiaq/device.c:525 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Code: 48 8b 85 30 ff ff ff 48 8d b8 98 00 00 00 e8 4e 92 8b ff 45 89 e8 44 89 f1 4c 89 fa 48 89 c6 48 c7 c7 c0 5a c8 85 e8 f0 4a dd fd <0f> ff e9 9b f7 ff ff e8 da cb 26 fe e9 80 f7 ff ff e8 70 a1 f4 ---[ end trace 1d323f93391df1ca ]---

1 0

[alsa-devel] [PATCH 0/3] ALSA: BeBoB: Fine-tuning for six function implementations
by SF Markus Elfring 27 Sep '17

27 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 6 Sep 2017 12:16:54 +0200 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Use common error handling code in snd_bebob_stream_start_duplex() Adjust six checks for null pointers Improve a size determination in parse_stream_formation() sound/firewire/bebob/bebob_stream.c | 35 +++++++++++++++++------------------ 1 file changed, 17 insertions(+), 18 deletions(-) -- 2.14.1

2 8

[alsa-devel] [PATCH v2] ASoC: Intel: Kbl: Add Playback DAI for fixup
by Harsha Priya 27 Sep '17

27 Sep '17

'Kbl Audio Headset Playback' FE DAI also needs SSP hw param fix. Add this dai also to be handled in kabylake_ssp_fixup() call. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- v2: change patch title and log sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 88ff542..e767283 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -302,6 +302,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || + !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; channels->min = channels->max = 2; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: add Component level items
by Kuninori Morimoto 27 Sep '17

27 Sep '17

Hi Mark These are resend of one of "replace component" patch series. I posted it on below, but few weeks past, and its base branch was updated now. I need/want to resend these. Subject: Prepare to covert Codec/Platform into Component Date: Fri, 01 Sep 2017 14:18:11 +0900 Because it is big patch-set, to avoid sending patch bomb, I will repost it as small amount of patch set. To review from big picture, please check above patch-set. I will continue to resend remainings of it if this small patch-set were accepted Kuninori Morimoto (2): ASoC: add Component level pcm_new/pcm_free v2 ASoC: add Component level set_bias_level include/sound/soc.h | 14 +++++++++++++ sound/soc/soc-core.c | 55 +++++++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/soc-pcm.c | 29 +++++++++++++++++++-------- 3 files changed, 89 insertions(+), 9 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

2 4

[alsa-devel] [RFC PATCH 0/7] Fix year 2038 issue for sound subsystem
by Baolin Wang 27 Sep '17

27 Sep '17

Since many structures will use timespec type variables to record time stamp in uapi/asound.h, which are not year 2038 safe on 32bit system. This patchset tries to introduce new structures removing timespec type to compatible native mode and compat mode. Moreover this patchset also converts the internal structrures to use timespec64 type and related APIs. Baolin Wang (7): sound: Replace timespec with timespec64 sound: core: Avoid using timespec for struct snd_pcm_status sound: core: Avoid using timespec for struct snd_pcm_sync_ptr sound: core: Avoid using timespec for struct snd_rawmidi_status sound: core: Avoid using timespec for struct snd_timer_status uapi: sound: Avoid using timespec for struct snd_ctl_elem_value sound: core: Avoid using timespec for struct snd_timer_tread include/sound/pcm.h | 113 ++++++++- include/sound/timer.h | 4 +- include/uapi/sound/asound.h | 15 +- sound/core/pcm.c | 14 +- sound/core/pcm_compat.c | 466 +++++++++++++++++++++++++++++-------- sound/core/pcm_lib.c | 33 +-- sound/core/pcm_native.c | 227 ++++++++++++++---- sound/core/rawmidi.c | 74 +++++- sound/core/rawmidi_compat.c | 90 +++++-- sound/core/timer.c | 247 ++++++++++++++++---- sound/core/timer_compat.c | 25 +- sound/pci/hda/hda_controller.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 4 +- 13 files changed, 1046 insertions(+), 276 deletions(-) -- 1.7.9.5

5 31

[alsa-devel] [PATCH v2] ASoC: add null_snd_soc_ops and reduce NULL ops check
by Kuninori Morimoto 26 Sep '17

26 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Double NULL pointer check for ops and ops->func is difficult to read and might be forget to check it if new func was add. This patch adds new null_snd_soc_ops and use it if rtd->dai_link didn't have it to avoid NULL ops, and reduces ops NULL check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased on mark/topic/core sound/soc/soc-core.c | 5 +++++ sound/soc/soc-pcm.c | 16 ++++++++-------- 2 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e162206..1b5b519 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -614,6 +614,8 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); +static const struct snd_soc_ops null_snd_soc_ops; + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -626,6 +628,9 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( INIT_LIST_HEAD(&rtd->component_list); rtd->card = card; rtd->dai_link = dai_link; + if (!rtd->dai_link->ops) + rtd->dai_link->ops = &null_snd_soc_ops; + rtd->codec_dais = kzalloc(sizeof(struct snd_soc_dai *) * dai_link->num_codecs, GFP_KERNEL); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 975965d..09fed701 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -511,7 +511,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->rx_mask = 0; } - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + if (rtd->dai_link->ops->startup) { ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { pr_err("ASoC: %s startup failed: %d\n", @@ -585,7 +585,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); machine_err: @@ -692,7 +692,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->driver->ops->shutdown(substream, codec_dai); } - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); if (platform->driver->ops && platform->driver->ops->close) @@ -751,7 +751,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + if (rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: machine prepare error:" @@ -855,7 +855,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: machine hw_params" @@ -948,7 +948,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_dai->rate = 0; } - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); mutex_unlock(&rtd->pcm_mutex); @@ -994,7 +994,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); /* free any DMA resources */ @@ -1045,7 +1045,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (rtd->dai_link->ops && rtd->dai_link->ops->trigger) { + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"
by Takashi Sakamoto 26 Sep '17

26 Sep '17

This reverts commit 275353bb684e to fix a regression which can abort 'alsactl' program in alsa-utils due to assertion in alsa-lib. alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. This commit is a band-aid. In a point of usage of ALSA control interface, the drivers still bring an issue that they prevent userspace applications to have a consistent way to parse each levels of the dimension information via ALSA control interface. Let me investigate this issue. Current implementation of the drivers have three control element sets with dimension information: * 'Monitor Mixer Volume' (type: integer) * 'VMixer Volume' (type: integer) * 'VU-meters' (type: boolean) Although the number of elements named as 'Monitor Mixer Volume' differs depending on drivers in this group, it can be calculated by macros defined by each driver (= (BX_NUM - BX_ANALOG_IN) * BX_ANALOG_IN). Each of the elements has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * (BX_NUM - BX_ANALOG_IN)). For these elements, userspace applications are expected to handle the dimension information so that all of the elements construct a matrix where the number of rows and columns are represented by the dimension information. The same way is applied to elements named as 'VMixer Volume'. The number of these elements can also be calculated by macros defined by each drivers (= PX_ANALOG_IN * BX_ANALOG_IN). Each of the element has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * PX_ANALOG_IN). All of the elements construct a matrix with the dimension information. An element named as 'VU-meters' gets a different way in a point of dimension information. The element includes 96 members for value. The element has dimension information with 3 levels (= 3 or 2 * 16 * 2). For this element, userspace applications are expected to handle the dimension information so that all of the members for value construct a matrix where the number of rows and columns are represented by the dimension information. This is different from the way for the former. As a summary, the drivers were not designed to produce a consistent way to parse the dimension information. This makes it hard for general userspace applications such as amixer to parse the information by a consistent way, and actually no userspace applications except for 'echomixer' utilize the dimension information. Additionally, no drivers excluding this group use the information. The reverted commit was written based on the latter way. A commit 860c1994a70a ('ALSA: control: add dimension validator for userspace elements') is written based on the latter way, too. The patch should be reconsider too in the same time to re-define a consistent way to parse the dimension information. Reported-by: Mark Hills <mark(a)xwax.org> Reported-by: S. Christian Collins <s.chriscollins(a)gmail.com> Fixes: 275353bb684e ('ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members') Cc: <stable(a)vger.kernel.org> # v4.8+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/pci/echoaudio/echoaudio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 7326695bca33..d68f99e076a8 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1272,11 +1272,11 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_busses_in(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1344,11 +1344,11 @@ static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_pipes_out(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1728,6 +1728,7 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 96; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = 0; #ifdef ECHOCARD_HAS_VMIXER @@ -1737,7 +1738,6 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, #endif uinfo->dimen.d[1] = 16; /* 16 channels */ uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */ - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1] * uinfo->dimen.d[2]; return 0; } -- 2.11.0

1 0

[alsa-devel] [Resend] [PATCH v1 3/4] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 25 Sep '17

25 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 130 ++++++++++++++++++++++++++++++--------------------- 1 file changed, 78 insertions(+), 52 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..e659c74 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,13 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params, const char **w_param_text, + unsigned long *private_value) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3793,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; - - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); - if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + int count; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,47 +3812,90 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ - private_value = + *private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), sizeof(struct soc_enum), GFP_KERNEL); - if (!private_value) { + if (!*private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } - kcontrol_dai_link[0].private_value = private_value; + kcontrol_dai_link[0].private_value = *private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; goto outfree_private_value; } + return kcontrol_news; + +outfree_private_value: + devm_kfree(card->dev, (void *)*private_value); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + return NULL; +} +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + const char **w_param_text; + unsigned long private_value; + char *link_name; + int ret, count; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params, + w_param_text, &private_value); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto outfree_link_name; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3927,13 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: devm_kfree(card->dev, (void *)private_value); -outfree_link_name: - devm_kfree(card->dev, link_name); for (count = 0 ; count < num_params; count++) devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: +outfree_link_name: devm_kfree(card->dev, w_param_text); - +param_fail: + devm_kfree(card->dev, link_name); return ret; } -- 2.5.4 (Apple Git-61)

4 4

[alsa-devel] Applied "ASoC: max98927: Added support for DSP_A and DSP_B format" to the asoc tree
by Mark Brown 25 Sep '17

25 Sep '17

The patch ASoC: max98927: Added support for DSP_A and DSP_B format has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4eee20246c0fa77a7a7b189ac806d5f78d675546 Mon Sep 17 00:00:00 2001 From: Ryan Lee <ryans.lee(a)maximintegrated.com> Date: Thu, 14 Sep 2017 17:30:38 -0700 Subject: [PATCH] ASoC: max98927: Added support for DSP_A and DSP_B format Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98927.c | 35 ++++++++++++++++++++--------------- sound/soc/codecs/max98927.h | 6 ++++-- 2 files changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe72f8ad..6f7b3ef48e25 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); - - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +235,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + } return 0; } diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a608cbe1..bf7a6f92562a 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) -- 2.14.1

1 0

[alsa-devel] Applied "ASoC: max98927: Added max98927_dai_tdm_slot function" to the asoc tree
by Mark Brown 25 Sep '17

25 Sep '17

The patch ASoC: max98927: Added max98927_dai_tdm_slot function has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d4a8bce81cbd53420988f5db0d096ad04960f189 Mon Sep 17 00:00:00 2001 From: Ryan Lee <ryans.lee(a)maximintegrated.com> Date: Thu, 14 Sep 2017 17:30:39 -0700 Subject: [PATCH] ASoC: max98927: Added max98927_dai_tdm_slot function Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98927.c | 120 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/max98927.h | 1 + 2 files changed, 105 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index 6f7b3ef48e25..a1d39353719d 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -250,6 +250,21 @@ static const int rate_table[] = { 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -275,23 +290,20 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: - return -EINVAL; + if (!max98927->tdm_mode) { + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + value); } - regmap_update_bits(max98927->regmap, - MAX98927_R0022_PCM_CLK_SETUP, - MAX98927_PCM_CLK_SETUP_BSEL_MASK, - value); return 0; } @@ -391,6 +403,78 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + max98927->tdm_mode = true; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -410,6 +494,7 @@ static const struct snd_soc_dai_ops max98927_dai_ops = { .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, @@ -419,6 +504,9 @@ static int max98927_dac_event(struct snd_soc_dapm_widget *w, struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + max98927->tdm_mode = 0; + break; case SND_SOC_DAPM_POST_PMU: regmap_update_bits(max98927->regmap, MAX98927_R003A_AMP_EN, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index bf7a6f92562a..9ea839735433 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -270,5 +270,6 @@ struct max98927_priv { unsigned int iface; unsigned int master; unsigned int digital_gain; + bool tdm_mode; }; #endif -- 2.14.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 25 Sep '17

25 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 35 ++++++++++++++++++++--------------- sound/soc/codecs/max98927.h | 6 ++++-- 2 files changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe7..6f7b3ef 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); - - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +235,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + } return 0; } diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..bf7a6f92 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: dapm: Make snd_soc_dapm_add/del_routes use runtime mutex subclass
by Stuart Henderson 25 Sep '17

25 Sep '17

Make snd_soc_dapm_add_routes and snd_soc_dapm_del_routes use the SND_SOC_DAPM_CLASS_RUNTIME dapm subclass to allow dynamic dapm route updates. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..e6b6011 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2884,7 +2884,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, { int i, r, ret = 0; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); for (i = 0; i < num; i++) { r = snd_soc_dapm_add_route(dapm, route); if (r < 0) { @@ -2915,7 +2915,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, { int i; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); for (i = 0; i < num; i++) { snd_soc_dapm_del_route(dapm, route); route++; -- 2.1.4

2 1

[alsa-devel] [PATCH 0/4][RESEND] ASoC: rtd NULL check and unneeded ops check
by Kuninori Morimoto 25 Sep '17

25 Sep '17

Hi Mark These are posted few weeks ago, and nothting happen. Thus, I resend these. I'm posting "Convert Codec/Platform to Component" patch-set too, but I know it is very big patch-set, need more review. OTOH, These are very light patches. Thus, I will repost these 1st. Then, I will repost "Convert ..." patch-set again on top of these if these are accepted. Kuninori Morimoto (4): ASoC: soc-core: remove rtd NULL check on soc_free_pcm_runtime() ASoC: remove unneeded dai->driver check ASoC: remove unneeded dai->driver->ops check ASoC: add null_snd_soc_ops and reduce NULL ops check sound/soc/soc-core.c | 24 ++++++++++++++---------- sound/soc/soc-dapm.c | 8 ++++---- sound/soc/soc-pcm.c | 47 +++++++++++++++++++++++------------------------ 3 files changed, 41 insertions(+), 38 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

2 8

[alsa-devel] [PATCH] ASoC: rt5663: Change the dev getting function in rt5663_irq
by Oder Chiou 25 Sep '17

25 Sep '17

In the irq handler "rt5663_irq", while the codec is not initialized, rt5663->codec will be null, and it will cause the kernel panic in the debug print enabled. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 767f219..fcd1643 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1827,7 +1827,8 @@ static irqreturn_t rt5663_irq(int irq, void *data) { struct rt5663_priv *rt5663 = data; - dev_dbg(rt5663->codec->dev, "%s IRQ queue work\n", __func__); + dev_dbg(regmap_get_device(rt5663->regmap), "%s IRQ queue work\n", + __func__); queue_delayed_work(system_wq, &rt5663->jack_detect_work, msecs_to_jiffies(250)); -- 2.7.4

2 1

[alsa-devel] [PATCH 0/3] Intel-rt5670 updates
by Pierre-Louis Bossart 25 Sep '17

25 Sep '17

Instead of submitting all my patches at ones, I broke them by platforms/codecs. This series is only for Intel-rt5670 platforms. It fixes a mistake on the Dell Venue 5585 - corrected with the help from Realtek and Dell who kindly provided schematics and data sheet information needed to fix non-working DMIC capture. Since there are still some reports of non-functional audio on some platforms, I added quirks that can be changed from the kernel command line (and later added as DMI-based quirks). The last two patches are minor corrections to fix ACPI HID naming issues and use a more meaningful card name. Pierre-Louis Bossart (3): ASoC: rt5670: refactor DMI quirks and fix Dell Venue settings ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name sound/soc/codecs/rt5670.c | 124 +++++++++++++++++++++++++++----- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++- 2 files changed, 112 insertions(+), 22 deletions(-) -- 2.9.3

4 7

[alsa-devel] [PATCH 00/14] BYT/CHT codec/machine drivers update
by Pierre-Louis Bossart 25 Sep '17

25 Sep '17

Finally took the time to cleanup known fixes for codec and machine drivers. I tested on Baytrail/Cherrytrail and w/ RT5640, 45 and 72. This should be the last batch of fixes before we move on to the integration of SOF (Sound Open Firmware) support in the mainline. This batch contains two patches already submitted by Thierry Escande on alsa-devel but never merged. Fang, Yang A (1): ASoC: ts3a227e: add acpi table Pierre-Louis Bossart (12): ASoC: Intel: boards: use devm_clk_get() unconditionally ASoC: intel: cht_bsw_max98090: add support for clock framework ASoC: Intel: cht_bsw_max98090: Fix I2S config + unused code ASoC: Intel: atom: add support for Baytrail Chromebooks ASoC: Intel: bytcr-rt5651: fix capture routes ASoC: Intel: bytcr_rt5651: add MCLK support ASoC: Intel: bytcr_rt5651: filter codec name ASoC: codecs: rt5645: add quirks for Asus T100HA ASoC: Intel: bytcr_rt5640: simplify MCLK quirk tests ASoC: codecs: rt5670: add jack detection quirk for Dell Venue 5585 ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name Thierry Escande (1): ASoc: cht_bsw_max98090_ti: Fix jack initialization sound/soc/codecs/rt5645.c | 20 ++ sound/soc/codecs/rt5670.c | 16 ++ sound/soc/codecs/ts3a227e.c | 10 + sound/soc/intel/atom/sst/sst_acpi.c | 8 + sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 278 ++++++++++++++++++++++++--- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 157 ++++++++++++--- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 36 ++-- 9 files changed, 460 insertions(+), 87 deletions(-) -- 2.9.3

3 27

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix swapped order of function arguments dir and pin_index
by Colin King 25 Sep '17

25 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> The call to slk_tplg_fill_res_tkn is passing dir and pin_index in the wrong order, they need to be in pin_index, dir order to match the function slk_tplg_fill_res_tkn correctly. Detected by CoveritScan, CID#1454992 ("Arguments in wrong order") Fixes: f6fa56e22559 ("ASoC: Intel: Skylake: Parse and update module config structure") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22f768ca3c73..27bcb62568fb 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2382,7 +2382,7 @@ static int skl_tplg_get_token(struct device *dev, case SKL_TKN_U32_MAX_MCPS: case SKL_TKN_U32_OBS: case SKL_TKN_U32_IBS: - ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, dir, pin_index); + ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, pin_index, dir); if (ret < 0) return ret; -- 2.14.1

3 2

[alsa-devel] [PATCH] ASoC: wm8998: Correct handling of input muxes
by Charles Keepax 25 Sep '17

25 Sep '17

Currently, wm8998 has two input mux controls on IN1 and attempts to switch these together when the A position is configured to be in digital mode. This is because the digital mode requires pins from both the L and R channels. However, this doesn't work as intended because whilst the registers on the chip are changed the corresponding DAPM representation is only updated for the mux actually being changed by the user. The DAPM graph being out of sync with the hardware can cause some odd issues with incorrect things being powered etc. To avoid this issue and simplify the code somewhat, simply let the user set the muxes as they desire. If they set an invalid configuration they might not get audio from the DMIC but most of the chip requires you to set a valid audio route to get audio. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm8998.c | 84 +++++++++++------------------------------------ 1 file changed, 19 insertions(+), 65 deletions(-) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index c59caaa75ba0..2d211dbe7422 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -101,7 +101,7 @@ static int wm8998_asrc_ev(struct snd_soc_dapm_widget *w, return 0; } -static int wm8998_in1mux_put(struct snd_kcontrol *kcontrol, +static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); @@ -109,84 +109,38 @@ static int wm8998_in1mux_put(struct snd_kcontrol *kcontrol, struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = wm8998->core.arizona; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux, inmode; - unsigned int mode_val, src_val; + unsigned int mode_reg, mode_index; + unsigned int mux, inmode, src_val, mode_val; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) return -EINVAL; - /* L and R registers have same shift and mask */ - inmode = arizona->pdata.inmode[2 * mux]; - src_val = mux << ARIZONA_IN1L_SRC_SHIFT; - if (inmode & ARIZONA_INMODE_SE) - src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT; - - switch (arizona->pdata.inmode[0]) { - case ARIZONA_INMODE_DMIC: - if (mux) - mode_val = 0; /* B always analogue */ - else - mode_val = 1 << ARIZONA_IN1_MODE_SHIFT; - - snd_soc_update_bits(codec, ARIZONA_IN1L_CONTROL, - ARIZONA_IN1_MODE_MASK, mode_val); - - /* IN1A is digital so L and R must change together */ - /* src_val setting same for both registers */ - snd_soc_update_bits(codec, - ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, src_val); - snd_soc_update_bits(codec, - ARIZONA_ADC_DIGITAL_VOLUME_1R, - ARIZONA_IN1R_SRC_MASK | - ARIZONA_IN1R_SRC_SE_MASK, src_val); + switch (e->reg) { + case ARIZONA_ADC_DIGITAL_VOLUME_2L: + mode_reg = ARIZONA_IN2L_CONTROL; + mode_index = 1 + (2 * mux); break; default: - /* both analogue */ - snd_soc_update_bits(codec, - e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + mode_reg = ARIZONA_IN1L_CONTROL; + mode_index = (2 * mux); break; } - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); -} - -static int wm8998_in2mux_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); - struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec); - struct arizona *arizona = wm8998->core.arizona; - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux, inmode, src_val, mode_val; - - mux = ucontrol->value.enumerated.item[0]; - if (mux > 1) - return -EINVAL; - - inmode = arizona->pdata.inmode[1 + (2 * mux)]; + inmode = arizona->pdata.inmode[mode_index]; if (inmode & ARIZONA_INMODE_DMIC) - mode_val = 1 << ARIZONA_IN2_MODE_SHIFT; + mode_val = 1 << ARIZONA_IN1_MODE_SHIFT; else mode_val = 0; - src_val = mux << ARIZONA_IN2L_SRC_SHIFT; + src_val = mux << ARIZONA_IN1L_SRC_SHIFT; if (inmode & ARIZONA_INMODE_SE) - src_val |= 1 << ARIZONA_IN2L_SRC_SE_SHIFT; + src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT; - snd_soc_update_bits(codec, ARIZONA_IN2L_CONTROL, - ARIZONA_IN2_MODE_MASK, mode_val); + snd_soc_update_bits(codec, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_update_bits(codec, ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_IN2L_SRC_MASK | ARIZONA_IN2L_SRC_SE_MASK, + snd_soc_update_bits(codec, e->reg, + ARIZONA_IN1L_SRC_MASK | ARIZONA_IN1L_SRC_SE_MASK, src_val); return snd_soc_dapm_mux_update_power(dapm, kcontrol, @@ -216,14 +170,14 @@ static SOC_ENUM_SINGLE_DECL(wm8998_in2mux_enum, static const struct snd_kcontrol_new wm8998_in1mux[2] = { SOC_DAPM_ENUM_EXT("IN1L Mux", wm8998_in1muxl_enum, - snd_soc_dapm_get_enum_double, wm8998_in1mux_put), + snd_soc_dapm_get_enum_double, wm8998_inmux_put), SOC_DAPM_ENUM_EXT("IN1R Mux", wm8998_in1muxr_enum, - snd_soc_dapm_get_enum_double, wm8998_in1mux_put), + snd_soc_dapm_get_enum_double, wm8998_inmux_put), }; static const struct snd_kcontrol_new wm8998_in2mux = SOC_DAPM_ENUM_EXT("IN2 Mux", wm8998_in2mux_enum, - snd_soc_dapm_get_enum_double, wm8998_in2mux_put); + snd_soc_dapm_get_enum_double, wm8998_inmux_put); static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); -- 2.11.0

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[alsa-devel] [PATCH] Documentation: DT bindings: Add chip compatible string for audio codec.
by Shreeya Patel 25 Sep '17

25 Sep '17

This patch adds chip compatible string to remove the warnings generated by checkpatch that -DT compatible string "toshiba,apb-dummy-codec" appears un-documented. Signed-off-by: Shreeya Patel <shreeya.patel23498(a)gmail.com> --- Documentation/devicetree/bindings/sound/sirf-audio-codec.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt index 062f5ec..a61560b 100644 --- a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt +++ b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt @@ -3,6 +3,7 @@ SiRF internal audio CODEC Required properties: - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec" + "toshiba,apb-dummy-codec" - reg : the register address of the device. @@ -11,7 +12,7 @@ Required properties: Example: audiocodec: audiocodec@b0040000 { - compatible = "sirf,atlas6-audio-codec"; + compatible = "sirf,atlas6-audio-codec", "toshiba,apb-dummy-codec"; reg = <0xb0040000 0x10000>; clocks = <&clks 27>; }; -- 2.7.4

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[alsa-devel] ASoC: pll/clkdiv issue with simple-card driver & WM8510/Atmel SSC
by jukka.hynninen＠vaisala.com 25 Sep '17

25 Sep '17

Hi, > I think simple-card doesn't have .set_pll / .set_clkdiv feature today. > You can add such feature if needed. Implementing such a feature into simple-card driver alone might not be possible, as the PLL output frequency and resulting MCLK/BCLK divisors are codec dependent. I wonder if the right place would be within the codec driver, after all? The machine code would just request a certain bitclock frequency and frame length in case the codec is selected to be the clock master. Is this in line with ASoC architecture? -Jukka

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[alsa-devel] usb/sound/line6: warning in line6_start_listen/usb_submit_urb
by Andrey Konovalov 25 Sep '17

25 Sep '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e19b205be43d11bff638cad4487008c48d21c103 (4.14-rc2). It seems there's no check on the actual endpoint type. usb 1-1: BOGUS urb xfer, pipe 3 != type 1 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 24 at drivers/usb/core/urb.c:449 usb_submit_urb+0xf8a/0x11d0 Modules linked in: CPU: 0 PID: 24 Comm: kworker/0:1 Not tainted 4.14.0-rc2-42613-g1488251d1a98 #238 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff88006befe300 task.stack: ffff88006bf78000 RIP: 0010:usb_submit_urb+0xf8a/0x11d0 drivers/usb/core/urb.c:448 RSP: 0018:ffff88006bf7e218 EFLAGS: 00010286 RAX: 0000000000000029 RBX: ffff88006a878200 RCX: 0000000000000000 RDX: 0000000000000029 RSI: ffffffff85876dc0 RDI: ffffed000d7efc35 RBP: ffff88006bf7e318 R08: 1ffff1000d7efadb R09: 0000000000000000 R10: 0000000000000002 R11: 0000000000000000 R12: 1ffff1000d7efc4a R13: 0000000000000001 R14: 0000000000000003 R15: ffff88006902d7f8 FS: 0000000000000000(0000) GS:ffff88006c800000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000001904fa0 CR3: 0000000062019000 CR4: 00000000000006f0 Call Trace: line6_start_listen+0x55f/0x9e0 sound/usb/line6/driver.c:82 line6_init_cap_control sound/usb/line6/driver.c:690 line6_probe+0x7c9/0x1310 sound/usb/line6/driver.c:764 podhd_probe+0x64/0x70 sound/usb/line6/podhd.c:474 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Code: 48 8b 85 30 ff ff ff 48 8d b8 98 00 00 00 e8 6e 91 8b ff 45 89 e8 44 89 f1 4c 89 fa 48 89 c6 48 c7 c7 c0 5a c8 85 e8 10 50 dd fd <0f> ff e9 9b f7 ff ff e8 ba cf 26 fe e9 80 f7 ff ff e8 a0 a5 f4 ---[ end trace 29c2f0b769cf6f62 ]---

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[alsa-devel] ASoC: pll/clkdiv issue with simple-card driver & WM8510/Atmel SSC
by jukka.hynninen＠vaisala.com 25 Sep '17

25 Sep '17

Hi all, We're looking into using the generic simple-card.c driver to glue together WM8510 codec driver and Atmel SSC platform driver without a specific machine code. It turned out that the DTS configuration below fails to set the codec PLL and clkdiv values. In fact, wm8510_set_dai_pll() and wm8510_set_dai_clkdiv() get never called. After adding following lines (omitting determination of myboard_* values) into wm8510_pcm_hw_params(), the audio starts working properly: wm8510_set_dai_clkdiv(dai, WM8510_BCLKDIV, myboard_bclk_div); wm8510_set_dai_pll(dai, 0, 0, 12000000, myboard_pll_out); wm8510_set_dai_clkdiv(dai, WM8510_MCLKDIV, myboard_mclk_div); What is actually the missing part? Did we forget to configure something in DTS? Or is this a limitation of simple-card driver? If so, what would be the proper way to implement the missing features? We're using Linux v4.14-rc1. The codec clock comes from a constantly running 12 MHz oscillator, and the codec is the clock and frame master. DTS configuration is as follows: ahb { apb { ssc0: ssc@fff9c000 { #sound-dai-cells = <0>; status = "okay"; dmas = <&dma 1 AT91_DMA_CFG_PER_ID(5)>, <&dma 1 AT91_DMA_CFG_PER_ID(6)>; dma-names = "tx", "rx"; pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; }; spi0: spi@fffa4000{ status = "okay"; cs-gpios = <&pioB 3 0>, <&pioA 28 0>, <0>, <0>; audio_codec: wm8510@1 { #sound-dai-cells = <0>; compatible = "wlf,wm8510"; spi-max-frequency = <2000000>; reg = <1>; }; }; }; sound { compatible = "simple-audio-card"; simple-audio-card,name = "wm8510_WID"; simple-audio-card,format = "i2s"; simple-audio-card,routing = "Mono Out", "MONOOUT"; simple-audio-card,codec { sound-dai = <&audio_codec>; system-clock-frequency = <12000000>; bitclock-master; frame-master; }; simple-audio-card,cpu { sound-dai = <&ssc0>; }; }; -Jukka

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[alsa-devel] Roland Boutique USB Implicit Feedback Weirdness
by Nathan Fraser 24 Sep '17

24 Sep '17

Hi, I've been trying to get some Roland Boutique devices to work with ALSA, and have had some initial success, but I've hit a wall and was wondering if anyone on the list might be able to help point me in the right direction to get it done. Some background: The Roland Boutique series is a collection of small drum machines and synthesizers from Roland that include a compposite USB interface which carries both PCM and MIDI. The PCM on most units is 2 channels in and out, on some there are additional channels. The devices present themselves to ALSA as many Roland devices seem to, and all the interfaces seem to be correctly auto-detected with current ALSA code. For example, the JP-08 (0582:01b5): $ amidi -l Dir Device Name IO hw:2,0,0 Boutiq MIDI 1 $ ./alsacap -C 2 *** Scanning for playback devices on card 2 *** Card 2, ID `Boutiq', name `Boutiq' Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available) 2 channels, sampling rate 44100..44100 Hz Sample formats: S32_LE Subdevice 0, name `subdevice #0' $ ./alsacap -R -C 2 *** Scanning for recording devices on card 2 *** Card 2, ID `Boutiq', name `Boutiq' Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available) 2 channels, sampling rate 44100..44100 Hz Sample formats: S32_LE Subdevice 0, name `subdevice #0' The MIDI works flawlessley, but the playback and capture interfaces do not. When the device is connected, an error usually appears comething like: [ 15.737868] usb 2-1: Manufacturer: Roland [ 15.737871] usb 2-1: SerialNumber: BQJP352DDA054653353337241943 [ 15.765216] snd-usb-audio: probe of 2-1:1.0 failed with error -5 [ 15.767884] usbcore: registered new interface driver snd-usb-audio [ 15.803631] usb 2-1: Unable to change format on ep #8e: already in use [ 15.803659] usb 2-1: Unable to change format on ep #8e: already in use ... When playing from aplay or arecord, nothing happens and eventually the program exits with an input/output error: $ aplay -vv -D hw:CARD=Boutiq,DEV=0 -t raw -c 2 -r 44100 -f S32_LE /dev/urandom Playing raw data '/dev/urandom' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo Hardware PCM card 2 'Boutiq' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 22050 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 appl_ptr : 0 hw_ptr : 0 ##################################################+| 99%aplay: pcm_write:1939: write error: Input/output error As far as I can tell, most of the Boutique models have the same problem, and it seems to be related to the audio streaming implicit feedback. I was able to get the JP-08 and SE-02 PCM to work by interrupting the code that adds the feedback interface as follows: --- linux-4.13.3/sound/usb/pcm.c 2017-09-20 16:28:17.000000000 +1000 +++ linux-4.13.3_uh/sound/usb/pcm.c 2017-09-24 04:54:17.950406081 +1000 @@ -367,6 +367,10 @@ search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1, altsd->bAlternateSetting, &alts, &ep) >= 0) { + dev_err(&dev->dev, + " > roland fb search found ep %x for %x\n", + ep, subs->ep_num); + return 0; goto add_sync_ep; } When this patch is applied and the unit connected, you can see the search matching the correct interfaces for feedback, but they are not added. Interestingly, the probe error -5 on 1-1:1.0 is no longer present: [ 24.832193] usb 1-1: Manufacturer: Roland [ 24.832194] usb 1-1: SerialNumber: BQJP352DDA054653353337241943 [ 24.846986] usbcore: registered new interface driver snd-usb-audio [ 24.869113] usb 1-1: > roland fb search found ep 8e for d [ 24.882599] usb 1-1: > roland fb search found ep 8e for d Now when you playback aith aplay, it works: $ aplay -vv -D hw:CARD=Boutiq,DEV=0 -t raw -c 2 -r 44100 -f S32_LE /dev/urandom Playing raw data '/dev/urandom' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo Hardware PCM card 2 'Boutiq' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 22050 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 appl_ptr : 0 hw_ptr : 0 ##################################################+| 99%^C Aborted by signal Interrupt... Capture also works, but only if you have a playback in process. If you try to use arecord on its own, you get input/output error as before. However, if you start the playback, and then capture, it works fine: $ aplay -D hw:CARD=Boutiq,DEV=0 -t raw -c 2 -r 44100 -f S32_LE /dev/urandom & Playing raw data '/dev/urandom' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo $ arecord -vv -D hw:CARD=Boutiq,DEV=0 -c 2 -r 44100 -f S32_LE test.wav Recording WAVE 'test.wav' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo Hardware PCM card 2 'Boutiq' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 22050 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 1 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 appl_ptr : 0 hw_ptr : 0 ###########+ | 21%^C Aborted by signal Interrupt... So it seems like even though the feedback system should be working for these units, some part of the setup process is not working quite as Roland expects, and as a result they just seem to hang. Skipping the feedback search for these units doesn't seem like the correct fix, especially when the feedback mechanism seems to be working. I just don't know where to look. If anyone could shed some light on what's going on, or point me toward a possible solution, that would be fantastic. For reference, the USB info for the JP-08 is as follows (the SE-02 has an identical interface layout): $ lsusb -v -d 0582:01b5 Bus 001 Device 003: ID 0582:01b5 Roland Corp. Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x01b5 bcdDevice 1.00 iManufacturer 1 iProduct 2 iSerial 3 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 258 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x0d EP 13 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x0d EP 13 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8e EP 14 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8e EP 14 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 01 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 Thanks, Nathan Fraser --

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[alsa-devel] [PATCH RESEND] ASoC: codecs: msm8916-wcd-digital: fix RX2 MIX1 and RX3 MIX1
by Jean-François Têtu 23 Sep '17

23 Sep '17

On Fri, Sep 22, 2017 at 17:45:25 -0400, Jean-François Têtu wrote: >The kcontrol for the third input (rxN_mix1_inp3) of both RX2 >and RX3 mixers are not using the correct control register. This simple >patch fixes this. > >Signed-off-by: Jean-François Têtu <jean-francois.tetu(a)savoirfairelinux.com> I'm sorry, I did not send this to the actual alsa-devel ML. This time I did. Jean-François Têtu --- sound/soc/codecs/msm8916-wcd-digital.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 66df8f8..a10a724 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -238,7 +238,7 @@ static const struct soc_enum rx_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( static const struct soc_enum rx2_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 0, 6, rx_mix1_text), SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 3, 6, rx_mix1_text), - SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 0, 6, rx_mix1_text), + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B2_CTL, 0, 6, rx_mix1_text), }; /* RX2 MIX2 */ @@ -249,7 +249,7 @@ static const struct soc_enum rx2_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( static const struct soc_enum rx3_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 0, 6, rx_mix1_text), SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 3, 6, rx_mix1_text), - SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 0, 6, rx_mix1_text), + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text), }; /* DEC */ -- 2.7.4

1 0

[alsa-devel] usb/sounds: slab-out-of-bounds read in snd_usb_create_streams
by Andrey Konovalov 22 Sep '17

22 Sep '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit ebb2c2437d8008d46796902ff390653822af6cc4 (Sep 18). It seems that there's no check that control_header has enough space for uac1_ac_header_descriptor in snd_usb_create_streams(). ================================================================== BUG: KASAN: slab-out-of-bounds in usb_audio_probe+0x27b2/0x2ab0 Read of size 1 at addr ffff88006b83a9e8 by task kworker/0:1/24 CPU: 0 PID: 24 Comm: kworker/0:1 Not tainted 4.14.0-rc1-42251-gebb2c2437d80 #224 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:16 dump_stack+0x292/0x395 lib/dump_stack.c:52 print_address_description+0x78/0x280 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 kasan_report+0x22f/0x340 mm/kasan/report.c:409 __asan_report_load1_noabort+0x19/0x20 mm/kasan/report.c:427 snd_usb_create_streams sound/usb/card.c:248 usb_audio_probe+0x27b2/0x2ab0 sound/usb/card.c:605 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Allocated by task 24: save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59 save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551 __kmalloc+0x14e/0x310 mm/slub.c:3782 kmalloc ./include/linux/slab.h:498 usb_get_configuration+0x372/0x2a60 drivers/usb/core/config.c:848 usb_enumerate_device drivers/usb/core/hub.c:2290 usb_new_device+0xaae/0x1020 drivers/usb/core/hub.c:2426 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Freed by task 4048: save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59 save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_slab_free+0x72/0xc0 mm/kasan/kasan.c:524 slab_free_hook mm/slub.c:1390 slab_free_freelist_hook mm/slub.c:1412 slab_free mm/slub.c:2988 kfree+0xf6/0x2f0 mm/slub.c:3919 free_rb_tree_fname+0x8a/0xe0 fs/ext4/dir.c:402 ext4_htree_free_dir_info fs/ext4/dir.c:424 ext4_release_dir+0x49/0x70 fs/ext4/dir.c:622 __fput+0x33e/0x800 fs/file_table.c:210 ____fput+0x1a/0x20 fs/file_table.c:244 task_work_run+0x1af/0x280 kernel/task_work.c:112 tracehook_notify_resume ./include/linux/tracehook.h:191 exit_to_usermode_loop+0x1e1/0x220 arch/x86/entry/common.c:162 prepare_exit_to_usermode arch/x86/entry/common.c:197 syscall_return_slowpath+0x414/0x480 arch/x86/entry/common.c:266 entry_SYSCALL_64_fastpath+0xc0/0xc2 arch/x86/entry/entry_64.S:238 The buggy address belongs to the object at ffff88006b83a9c0 which belongs to the cache kmalloc-64 of size 64 The buggy address is located 40 bytes inside of 64-byte region [ffff88006b83a9c0, ffff88006b83aa00) The buggy address belongs to the page: page:ffffea0001ae0e80 count:1 mapcount:0 mapping: (null) index:0x0 flags: 0x100000000000100(slab) raw: 0100000000000100 0000000000000000 0000000000000000 00000001802a002a raw: 0000000000000000 0000000100000001 ffff88006c403800 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff88006b83a880: fc fc fc fc 00 00 00 00 00 00 00 00 fc fc fc fc ffff88006b83a900: fb fb fb fb fb fb fb fb fc fc fc fc 00 00 00 00 >ffff88006b83a980: 00 00 fc fc fc fc fc fc 00 00 00 00 07 fc fc fc ^ ffff88006b83aa00: fc fc fc fc 00 00 00 00 00 00 00 00 fc fc fc fc ffff88006b83aa80: 00 00 00 00 00 00 00 00 fc fc fc fc 00 00 00 00 ==================================================================

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: Add sample rate quirk for Plantronics C310/C520-M
by Kai-Heng Feng 22 Sep '17

22 Sep '17

Like other Plantronics devices, C310 and C520-M do not support sample rate reading. Add them to the sample rate quirk accordingly. BugLink: https://bugs.launchpad.net/bugs/1708499 BugLink: https://bugs.launchpad.net/bugs/1709282 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 913552078285..b8cb57aeec77 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1137,6 +1137,8 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0x02F7): /* Plantronics BT-600 */ case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ + case USB_ID(0x047F, 0xC022): /* Plantronics C310 */ + case USB_ID(0x047F, 0xC036): /* Plantronics C520-M */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ case USB_ID(0x05A3, 0x9420): /* ELP HD USB Camera */ -- 2.14.1

2 1

[alsa-devel] [PATCH] sound: core: Fix structure definition for X32 ABI
by Baolin Wang 22 Sep '17

22 Sep '17

X32 ABI uses the 64bit timespec in addition to 64bit alignment of 64bit values. We have added compat ABI for these ioctls, but this patch adds one missing padding into 'struct snd_pcm_mmap_status_x32' to fix incompatibilities. Signed-off-by: Baolin Wang <baolin.wang(a)linaro.org> --- sound/core/pcm_compat.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 10f537f..29ab480 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -547,6 +547,7 @@ struct snd_pcm_mmap_status_x32 { u32 pad2; /* alignment */ struct timespec tstamp; s32 suspended_state; + s32 pad3; struct timespec audio_tstamp; } __packed; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 4/4] ALSA: emu10k1: Cocci spatch "alloc_cast"
by Thomas Meyer 21 Sep '17

21 Sep '17

Remove casting the values returned by memory allocation functions like kmalloc, kzalloc, kmem_cache_alloc, kmem_cache_zalloc etc." Found by coccinelle spatch "api/alloc/alloc_cast.cocci" Signed-off-by: Thomas Meyer <thomas(a)m3y3r.de> --- diff -u -p a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1218,7 +1218,7 @@ static int _snd_emu10k1_audigy_init_efx( if (!icode) return err; - icode->gpr_map = (u_int32_t __user *) kcalloc(512 + 256 + 256 + 2 * 1024, + icode->gpr_map = kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), GFP_KERNEL); if (!icode->gpr_map) goto __err_gpr; @@ -1853,7 +1853,7 @@ static int _snd_emu10k1_init_efx(struct if (!icode) return err; - icode->gpr_map = (u_int32_t __user *) kcalloc(256 + 160 + 160 + 2 * 512, + icode->gpr_map = kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t), GFP_KERNEL); if (!icode->gpr_map) goto __err_gpr;

3 3

[alsa-devel] [PATCH] ASoC: rsnd: add rsnd_dma_alloc()
by Kuninori Morimoto 21 Sep '17

21 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car sound DMA will be used from SSI/SRC. dma.c doesn't alloc DMA handler in .probe timing, because we don't know what kind of DMA transfer will be used then. Thus, SSI/SRC have *rsnd_mod for DMA. rsnd_dma_attach() will allocate it and attach it to system. It will be PIO mode if it can't alloc DMA handler. In case of MIX is used, rsnd_dma_attach() will be called twice from SSI. To avoid duplicate allocation, current rsnd_dma_attach() is checking allocated DMA handler. This DMA related operation is a little bit difficult to understand. This patch adds new rsnd_dma_alloc() and separates allocation and attach for readable code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 60 +++++++++++++++++++++++++++---------------------- 1 file changed, 33 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 041ec10..17220c9 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -753,14 +753,15 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, } } -int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, - struct rsnd_mod **dma_mod) +static int rsnd_dma_alloc(struct rsnd_dai_stream *io, struct rsnd_mod *mod, + struct rsnd_mod **dma_mod) { struct rsnd_mod *mod_from = NULL; struct rsnd_mod *mod_to = NULL; struct rsnd_priv *priv = rsnd_io_to_priv(io); struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dma *dma; struct rsnd_mod_ops *ops; enum rsnd_mod_type type; int (*attach)(struct rsnd_dai_stream *io, struct rsnd_dma *dma, @@ -800,40 +801,45 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, type = RSND_MOD_AUDMA; } - if (!(*dma_mod)) { - struct rsnd_dma *dma; + dma = devm_kzalloc(dev, sizeof(*dma), GFP_KERNEL); + if (!dma) + return -ENOMEM; - dma = devm_kzalloc(dev, sizeof(*dma), GFP_KERNEL); - if (!dma) - return -ENOMEM; + *dma_mod = rsnd_mod_get(dma); - *dma_mod = rsnd_mod_get(dma); + ret = rsnd_mod_init(priv, *dma_mod, ops, NULL, + rsnd_mod_get_status, type, dma_id); + if (ret < 0) + return ret; - ret = rsnd_mod_init(priv, *dma_mod, ops, NULL, - rsnd_mod_get_status, type, dma_id); - if (ret < 0) - return ret; + dev_dbg(dev, "%s[%d] %s[%d] -> %s[%d]\n", + rsnd_mod_name(*dma_mod), rsnd_mod_id(*dma_mod), + rsnd_mod_name(mod_from), rsnd_mod_id(mod_from), + rsnd_mod_name(mod_to), rsnd_mod_id(mod_to)); + + ret = attach(io, dma, mod_from, mod_to); + if (ret < 0) + return ret; - dev_dbg(dev, "%s[%d] %s[%d] -> %s[%d]\n", - rsnd_mod_name(*dma_mod), rsnd_mod_id(*dma_mod), - rsnd_mod_name(mod_from), rsnd_mod_id(mod_from), - rsnd_mod_name(mod_to), rsnd_mod_id(mod_to)); + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + dma->mod_from = mod_from; + dma->mod_to = mod_to; + + return 0; +} + +int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, + struct rsnd_mod **dma_mod) +{ + if (!(*dma_mod)) { + int ret = rsnd_dma_alloc(io, mod, dma_mod); - ret = attach(io, dma, mod_from, mod_to); if (ret < 0) return ret; - - dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); - dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); - dma->mod_from = mod_from; - dma->mod_to = mod_to; } - ret = rsnd_dai_connect(*dma_mod, io, type); - if (ret < 0) - return ret; - - return 0; + return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } int rsnd_dma_probe(struct rsnd_priv *priv) -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: Handle return value of devm_kasprintf
by Arvind Yadav 21 Sep '17

21 Sep '17

devm_kasprintf() can fail here and we must check its return value. Arvind Yadav (3): [PATCH 1/3] ASoC: davinci-mcasp: Handle return value of devm_kasprintf [PATCH 2/3] ASoC: fsl-asoc-card: Handle return value of devm_kasprintf [PATCH 3/3] ASoC: omap-hdmi-audio: Handle return value of devm_kasprintf sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ sound/soc/fsl/fsl-asoc-card.c | 2 ++ sound/soc/omap/omap-hdmi-audio.c | 3 +++ 3 files changed, 17 insertions(+) -- 1.9.1

3 6

[alsa-devel] [PATCH v2 2/3] ASoC: fsl-asoc-card: Handle return value of devm_kasprintf
by Arvind Yadav 21 Sep '17

21 Sep '17

devm_kasprintf() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- changes in v2: Set return 'ret' to -ENOMEM. sound/soc/fsl/fsl-asoc-card.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 2db4d0c..a389885 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -639,6 +639,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) devm_kasprintf(&pdev->dev, GFP_KERNEL, "ac97-codec.%u", (unsigned int)idx); + if (!priv->dai_link[0].codec_name) { + ret = -ENOMEM; + goto asrc_fail; + } } priv->dai_link[0].platform_of_node = cpu_np; -- 1.9.1

2 2

[alsa-devel] How to play music through 3 sound cards simultaneously by aplay
by 韩 21 Sep '17

21 Sep '17

There are three wm8960 codecs in my paltform. I want to test playing music of these three cards at the same time. But I do not know how to realize this idea by alsa tool aplay. Can someone give me some hints? Best regards, Yuejuan

2 1

[alsa-devel] [asoc:topic/wm97xx 4/8] sound/soc/codecs/wm9713.c:27:30: fatal error: sound/ac97/codec.h: No such file or directory
by kbuild test robot 20 Sep '17

20 Sep '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm97xx head: 5aa3b03ad0712e3964a0ae2788f61f08da814bfe commit: 9bd400cadae05a26c3a22dd8393081cb96d0e26a [4/8] ASoC: wm9713: add ac97 new bus support config: frv-allmodconfig (attached as .config) compiler: frv-linux-gcc (GCC) 6.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 9bd400cadae05a26c3a22dd8393081cb96d0e26a # save the attached .config to linux build tree make.cross ARCH=frv All errors (new ones prefixed by >>): >> sound/soc/codecs/wm9713.c:27:30: fatal error: sound/ac97/codec.h: No such file or directory #include <sound/ac97/codec.h> ^ compilation terminated. vim +27 sound/soc/codecs/wm9713.c > 27 #include <sound/ac97/codec.h> 28 #include <sound/ac97/compat.h> 29 #include <sound/initval.h> 30 #include <sound/pcm_params.h> 31 #include <sound/tlv.h> 32 #include <sound/soc.h> 33 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 0/7] Refactor audio device tree handling
by Charles Keepax 20 Sep '17

20 Sep '17

Move the audio device tree binding out into its own documentation file, as is already done for the regulator and extcon bindings. Then add support for a new device tree property allowing the volume limits on the CODEC to be configured. It should be possible to put the patches through their separate trees. Thanks, Charles Charles Keepax (7): ASoC: arizona: Add new common Arizona init function ASoC: arizona: Add handling for audio related device tree entries ASoC: arizona: Add audio device tree bindings mfd: arizona: Remove audio related device tree code mfd: arizona: Remove audio bindings from MFD binding document ASoC: arizona: Add support for setting the output volume limits ASoC: arizona: Add device tree binding doc for volume limits Documentation/devicetree/bindings/mfd/arizona.txt | 40 +---- .../devicetree/bindings/sound/wlf,arizona.txt | 53 +++++++ MAINTAINERS | 1 + drivers/mfd/arizona-core.c | 132 +--------------- include/linux/mfd/arizona/pdata.h | 3 + sound/soc/codecs/arizona.c | 166 ++++++++++++++++++++- sound/soc/codecs/arizona.h | 6 +- sound/soc/codecs/cs47l24.c | 14 +- sound/soc/codecs/wm5102.c | 14 +- sound/soc/codecs/wm5110.c | 14 +- sound/soc/codecs/wm8997.c | 15 +- sound/soc/codecs/wm8998.c | 11 +- 12 files changed, 288 insertions(+), 181 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,arizona.txt -- 2.11.0

4 20

[alsa-devel] alsa-lib and floating point support
by Ricard Wanderlof 20 Sep '17

20 Sep '17

We recently came across an issue with the alsa-lib softvol plugin, which requires hardware floating-point (i.e. --with-softfloat not enabled during build) support in order to use anything but a set of default values (min gain -51 dB, max 0 dB, and 256 gain steps). With no hardware floating point support, the plugin just prints an error message when loaded. static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, ... #ifndef HAVE_SOFT_FLOAT ... double v = (pow(10.0, db / 20.0) * (double)(1 << VOL_SCALE_SHIFT)); ... #else SNDERR("Cannot handle the given dB range and resolution"); return -EINVAL; #endif } ... I'm trying to understand why it is critical to have hardware floating point support in this case. The pow() function is very resource hungry, true, but the function is only called when the plugin is loaded, and not for instance on every sample while the stream is running, so on the whole I would expect the impact to be minimal. Is there some other rationale that I'm missing? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 3

[alsa-devel] [PATCH] ASoC: rl6231: make arrays div and pd static const, reduces object code size
by Colin King 20 Sep '17

20 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the read-only arrays div and pd on the stack, instead make them static const. Makes the object code smaller by 210 bytes: Before: text data bss dec hex filename 2869 720 0 3589 e05 sound/soc/codecs/rl6231.o After: text data bss dec hex filename 2495 880 0 3375 d2f sound/soc/codecs/rl6231.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rl6231.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 7b447d0b173a..974a9040651d 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -71,7 +71,7 @@ EXPORT_SYMBOL_GPL(rl6231_get_pre_div); */ int rl6231_calc_dmic_clk(int rate) { - int div[] = {2, 3, 4, 6, 8, 12}; + static const int div[] = {2, 3, 4, 6, 8, 12}; int i; if (rate < 1000000 * div[0]) { @@ -189,7 +189,8 @@ EXPORT_SYMBOL_GPL(rl6231_pll_calc); int rl6231_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.14.1

2 1

[alsa-devel] [PATCH RESEND 0/3] Intel-rt5670 updates
by Pierre-Louis Bossart 20 Sep '17

20 Sep '17

[when I started cleaning-up some patches reviewed by Andy Shevchenko, I realized this initial batch has not been merged, although both Takashi and Vinod reviewed them, see http://mailman.alsa-project.org/pipermail/alsa-devel/2017-September/125331.… no change except for the addition of Takashi's Reviewed-by and Vinod's Acked-by. I don't mean to be pushy or anything, just checking if something is missing ] Instead of submitting all my patches at ones, I broke them by platforms/codecs. This series is only for Intel-rt5670 platforms. It fixes a mistake on the Dell Venue 5585 - corrected with the help from Realtek and Dell who kindly provided schematics and data sheet information needed to fix non-working DMIC capture. Since there are still some reports of non-functional audio on some platforms, I added quirks that can be changed from the kernel command line (and later added as DMI-based quirks). The last two patches are minor corrections to fix ACPI HID naming issues and use a more meaningful card name. Pierre-Louis Bossart (3): ASoC: rt5670: refactor DMI quirks and fix Dell Venue settings ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name sound/soc/codecs/rt5670.c | 124 +++++++++++++++++++++++++++----- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++- 2 files changed, 112 insertions(+), 22 deletions(-) -- 2.11.0

2 6

[alsa-devel] [asoc:topic/wm97xx 5/8] sound/soc/codecs/wm9712.c:23:30: fatal error: sound/ac97/codec.h: No such file or directory
by kbuild test robot 20 Sep '17

20 Sep '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm97xx head: 5aa3b03ad0712e3964a0ae2788f61f08da814bfe commit: 2ed1a8e0ce8db6d36f849526db61ce3c85a9f8d1 [5/8] ASoC: wm9712: add ac97 new bus support config: arm-multi_v7_defconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 6.1.1-9) 6.1.1 20160705 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 2ed1a8e0ce8db6d36f849526db61ce3c85a9f8d1 # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): >> sound/soc/codecs/wm9712.c:23:30: fatal error: sound/ac97/codec.h: No such file or directory #include <sound/ac97/codec.h> ^ compilation terminated. vim +23 sound/soc/codecs/wm9712.c > 23 #include <sound/ac97/codec.h> 24 #include <sound/ac97/compat.h> 25 #include <sound/initval.h> 26 #include <sound/soc.h> 27 #include <sound/tlv.h> 28 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 0/3] ASoC: Kill BUG_ON() usages
by Takashi Iwai 20 Sep '17

20 Sep '17

Hi, I found a few stupid BUG_ON() usages in the code, let's kill or replace them. Takashi === Takashi Iwai (3): ASoC: tas571x: Kill BUG_ON() usage ASoC: davinci: Kill BUG_ON() usage ASoC: intel: Kill BUG_ON() usage sound/soc/codecs/tas571x.c | 3 ++- sound/soc/davinci/davinci-mcasp.c | 3 ++- sound/soc/intel/atom/sst/sst_loader.c | 1 - sound/soc/intel/atom/sst/sst_stream.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-) -- 2.14.1

3 7

[alsa-devel] [Resend] [PATCH 0/4] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 20 Sep '17

20 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. This code change doesn't do much except just refactor snd_soc_dapm_new_pcm function. Now the part of allocating memory for kcontrol is allocated to a new function and it is called only in the case it is needed. anish kumar (4): ASoC: dapm: fix error path in snd_soc_dapm_new_pcm ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm ASoC: dapm: Avoid creating kcontrol for params ASoC: dapm: Remove the redundant check sound/soc/soc-dapm.c | 122 ++++++++++++++++++++++++++++----------------------- 1 file changed, 66 insertions(+), 56 deletions(-) -- 2.5.4 (Apple Git-61)

3 8

[alsa-devel] [patch 2/4] ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm
by anish kumar 20 Sep '17

20 Sep '17

refactor snd_soc_dapm_new_pcm to reduce the size of this function to facilitate further refactoring. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6e10a9a..1687c8a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3821,19 +3821,15 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kasprintf(card->dev, GFP_KERNEL, "Anonymous Configuration %d", count); - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, strlen(config->stream_name) + 1, GFP_KERNEL); - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + } + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; } config++; } -- 2.10.1 (Apple Git-78)

2 1

[alsa-devel] [PATCH] ASoC: tfa9879: Add device tree bindings
by Fabio Estevam 20 Sep '17

20 Sep '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> Even though the tfa9879 driver can probe via device tree trough the I2C core code, it is preferable to have explicit device tree bindings instead [1], so add this support. [1] https://www.spinics.net/lists/devicetree/msg195176.html Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Hi Lukasz, Could you please test this on your hardware? .../devicetree/bindings/sound/tfa9879.txt | 23 ++++++++++++++++++++++ sound/soc/codecs/tfa9879.c | 6 ++++++ 2 files changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tfa9879.txt diff --git a/Documentation/devicetree/bindings/sound/tfa9879.txt b/Documentation/devicetree/bindings/sound/tfa9879.txt new file mode 100644 index 0000000..731ffe8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tfa9879.txt @@ -0,0 +1,23 @@ +NXP TFA9879 class-D audio amplifier + +Required properties: + +- compatible : "nxp,tfa9879" + +- reg : the I2C address of the device + +Example: + +&i2c1 { + clock-frequency = <100000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2c1>; + status = "okay"; + + codec: tfa9879@6c { + #sound-dai-cells = <0>; + compatible = "nxp,tfa9879"; + reg = <0x6c>; + }; +}; + diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 95e0a7a..f8dd67c 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -312,9 +312,15 @@ static const struct i2c_device_id tfa9879_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, tfa9879_i2c_id); +static const struct of_device_id tfa9879_of_match[] = { + { .compatible = "nxp,tfa9879", }, + { } +}; + static struct i2c_driver tfa9879_i2c_driver = { .driver = { .name = "tfa9879", + .of_match_table = tfa9879_of_match, }, .probe = tfa9879_i2c_probe, .remove = tfa9879_i2c_remove, -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: add missing snd_soc_component_set_jack
by Kuninori Morimoto 20 Sep '17

20 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch adds missing definition of snd_soc_component_set_jack(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index d22de97..996bdbc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1487,6 +1487,8 @@ int snd_soc_component_set_sysclk(struct snd_soc_component *component, int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_component_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data); #ifdef CONFIG_REGMAP -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] soc-pcm: check symmetry after hw_params
by Jiada Wang 20 Sep '17

20 Sep '17

hw_params may be fixup by be_hw_params_fixup, calling soc_pcm_params_symmetry() before hw_params will have issue if there is hw_params changes in be_hw_params_fixup. For example, with following use case 1. a dai-link which is able to convert sample rate on BE side 2. set BE playback and capture sample rate to 44100Hz 3. play a 48000Hz audio stream with this dai-link 4. record from this dai-link with 44100Hz sample rate Got following error message when record starts [ 495.013527] be_link_ak4613: ASoC: unmatched rate symmetry: 48000 - 44100 [ 495.021729] be_link_ak4613: ASoC: hw_params BE failed -22 [ 495.028589] rsnd_link0: ASoC: hw_params BE failed -22 Because in soc_pcm_hw_params(), FE rate is still having value before it is fixup by be_hw_params_fixup(), when soc_pcm_params_symmetry() checks symmetry, thus soc_pcm_params_symmetry() complains about the unmatched rate between the active stream and the new stream tries to start. This patch moves soc_pcm_params_symmetry() after hw_params to resolve the above issue. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/soc-pcm.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 94b88b8..f11421f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -855,11 +855,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - - ret = soc_pcm_params_symmetry(substream, params); - if (ret) - goto out; - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -930,6 +925,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); + + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto platform_err; out: mutex_unlock(&rtd->pcm_mutex); return ret; -- 2.9.3

2 2

[alsa-devel] [PATCH v7 0/8] AC97 device/driver model revamp
by Robert Jarzmik 20 Sep '17

20 Sep '17

Hi Lars, Mark, Charles, Lee, This is a revision for Lee and Charles, only targetted at wm97xx-core changes (second round) suggested by Lee and codecs remove path for Charles. I removed the leading 4 patches as Mark has already scheduled them. For easier spotting, I included in [1] the diff from the last serie. As before Charles, I think your ack is not granted anymore for codecs, could you have a look please ? Cheers. -- Robert Robert Jarzmik (8): Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 +++++++++++++++------- drivers/mfd/Kconfig | 14 ++ drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 366 ++++++++++++++++++++++++++++++++ include/linux/mfd/wm97xx.h | 25 +++ sound/arm/Kconfig | 1 - sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/wm9705.c | 66 ++++-- sound/soc/codecs/wm9712.c | 44 ++-- sound/soc/codecs/wm9713.c | 39 ++-- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 46 ++-- 13 files changed, 712 insertions(+), 158 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h -- 2.11.0 [1] Diff from the previous serie ---8>--- diff --git a/drivers/mfd/wm97xx-core.c b/drivers/mfd/wm97xx-core.c index 66e477fffd43..4141ee52a70b 100644 --- a/drivers/mfd/wm97xx-core.c +++ b/drivers/mfd/wm97xx-core.c @@ -109,12 +109,8 @@ static const struct regmap_config wm9705_regmap_config = { }; static struct mfd_cell wm9705_cells[] = { - { - .name = "wm9705-codec", - }, - { - .name = "wm97xx-ts", - }, + { .name = "wm9705-codec", }, + { .name = "wm97xx-ts", }, }; static bool wm9712_volatile_reg(struct device *dev, unsigned int reg) @@ -181,12 +177,8 @@ static const struct regmap_config wm9712_regmap_config = { }; static struct mfd_cell wm9712_cells[] = { - { - .name = "wm9712-codec", - }, - { - .name = "wm97xx-ts", - }, + { .name = "wm9712-codec", }, + { .name = "wm97xx-ts", }, }; static const struct reg_default wm9713_reg_defaults[] = { @@ -256,12 +248,8 @@ static const struct regmap_config wm9713_regmap_config = { }; static struct mfd_cell wm9713_cells[] = { - { - .name = "wm9713-codec", - }, - { - .name = "wm97xx-ts", - }, + { .name = "wm9713-codec", }, + { .name = "wm97xx-ts", }, }; static int wm97xx_ac97_probe(struct ac97_codec_device *adev) @@ -270,7 +258,7 @@ static int wm97xx_ac97_probe(struct ac97_codec_device *adev) const struct regmap_config *config; struct wm97xx_platform_data *codec_pdata; struct mfd_cell *cells; - int ret = 0, nb_cells, i; + int ret = -ENODEV, nb_cells, i; struct wm97xx_pdata *pdata = snd_ac97_codec_get_platdata(adev); wm97xx = devm_kzalloc(ac97_codec_dev2dev(adev), @@ -309,7 +297,7 @@ static int wm97xx_ac97_probe(struct ac97_codec_device *adev) nb_cells = ARRAY_SIZE(wm9713_cells); break; default: - config = NULL; + goto err_free_compat; } for (i = 0; i < nb_cells; i++) { @@ -317,23 +305,22 @@ static int wm97xx_ac97_probe(struct ac97_codec_device *adev) cells[i].pdata_size = sizeof(*codec_pdata); } - if (config) { - codec_pdata->regmap = - devm_regmap_init_ac97(wm97xx->ac97, config); - if (IS_ERR(codec_pdata->regmap)) - ret = PTR_ERR(codec_pdata->regmap); - } else { - ret = -ENODEV; + codec_pdata->regmap = devm_regmap_init_ac97(wm97xx->ac97, config); + if (IS_ERR(codec_pdata->regmap)) { + ret = PTR_ERR(codec_pdata->regmap); + goto err_free_compat; } - if (!ret) - ret = devm_mfd_add_devices(wm97xx->dev, PLATFORM_DEVID_NONE, - cells, nb_cells, NULL, 0, NULL); - + ret = devm_mfd_add_devices(wm97xx->dev, PLATFORM_DEVID_NONE, + cells, nb_cells, NULL, 0, NULL); if (ret) - snd_ac97_compat_release(wm97xx->ac97); + goto err_free_compat; return ret; + +err_free_compat: + snd_ac97_compat_release(wm97xx->ac97); + return ret; } static int wm97xx_ac97_remove(struct ac97_codec_device *adev) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 46b5a77c53e4..68c204e3599f 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -354,11 +354,14 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) static int wm9705_soc_remove(struct snd_soc_codec *codec) { +#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9705_priv *wm9705 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_exit_regmap(codec); - if (!wm9705->mfd_pdata) + if (!wm9705->mfd_pdata) { + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9705->ac97); + } +#endif return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 7ebbb48b18c5..1e228bf9f1ae 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -674,11 +674,14 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) static int wm9712_soc_remove(struct snd_soc_codec *codec) { +#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_exit_regmap(codec); - if (!wm9712->mfd_pdata) + if (!wm9712->mfd_pdata) { + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9712->ac97); + } +#endif return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 3df2c01d751d..df7220656d98 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1238,14 +1238,14 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) static int wm9713_soc_remove(struct snd_soc_codec *codec) { +#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); if (!wm9713->mfd_pdata) { snd_soc_codec_exit_regmap(codec); -#ifdef CONFIG_SND_SOC_AC97_BUS snd_soc_free_ac97_codec(wm9713->ac97); -#endif } +#endif return 0; }

3 23

[alsa-devel] [RESEND][PATCH] ASoC: rsnd: add rsnd_dma_alloc()
by Kuninori Morimoto 20 Sep '17

20 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car sound DMA will be used from SSI/SRC. dma.c doesn't alloc DMA handler in .probe timing, because we don't know what kind of DMA transfer will be used then. Thus, SSI/SRC have *rsnd_mod for DMA. rsnd_dma_attach() will allocate it and attach it to system. It will be PIO mode if it can't alloc DMA handler. In case of MIX is used, rsnd_dma_attach() will be called twice from SSI. To avoid duplicate allocation, current rsnd_dma_attach() is checking allocated DMA handler. This DMA related operation is a little bit difficult to understand. This patch adds new rsnd_dma_alloc() and separates allocation and attach for readable code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 60 +++++++++++++++++++++++++++---------------------- 1 file changed, 33 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 041ec10..17220c9 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -753,14 +753,15 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, } } -int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, - struct rsnd_mod **dma_mod) +static int rsnd_dma_alloc(struct rsnd_dai_stream *io, struct rsnd_mod *mod, + struct rsnd_mod **dma_mod) { struct rsnd_mod *mod_from = NULL; struct rsnd_mod *mod_to = NULL; struct rsnd_priv *priv = rsnd_io_to_priv(io); struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dma *dma; struct rsnd_mod_ops *ops; enum rsnd_mod_type type; int (*attach)(struct rsnd_dai_stream *io, struct rsnd_dma *dma, @@ -800,40 +801,45 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, type = RSND_MOD_AUDMA; } - if (!(*dma_mod)) { - struct rsnd_dma *dma; + dma = devm_kzalloc(dev, sizeof(*dma), GFP_KERNEL); + if (!dma) + return -ENOMEM; - dma = devm_kzalloc(dev, sizeof(*dma), GFP_KERNEL); - if (!dma) - return -ENOMEM; + *dma_mod = rsnd_mod_get(dma); - *dma_mod = rsnd_mod_get(dma); + ret = rsnd_mod_init(priv, *dma_mod, ops, NULL, + rsnd_mod_get_status, type, dma_id); + if (ret < 0) + return ret; - ret = rsnd_mod_init(priv, *dma_mod, ops, NULL, - rsnd_mod_get_status, type, dma_id); - if (ret < 0) - return ret; + dev_dbg(dev, "%s[%d] %s[%d] -> %s[%d]\n", + rsnd_mod_name(*dma_mod), rsnd_mod_id(*dma_mod), + rsnd_mod_name(mod_from), rsnd_mod_id(mod_from), + rsnd_mod_name(mod_to), rsnd_mod_id(mod_to)); + + ret = attach(io, dma, mod_from, mod_to); + if (ret < 0) + return ret; - dev_dbg(dev, "%s[%d] %s[%d] -> %s[%d]\n", - rsnd_mod_name(*dma_mod), rsnd_mod_id(*dma_mod), - rsnd_mod_name(mod_from), rsnd_mod_id(mod_from), - rsnd_mod_name(mod_to), rsnd_mod_id(mod_to)); + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + dma->mod_from = mod_from; + dma->mod_to = mod_to; + + return 0; +} + +int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, + struct rsnd_mod **dma_mod) +{ + if (!(*dma_mod)) { + int ret = rsnd_dma_alloc(io, mod, dma_mod); - ret = attach(io, dma, mod_from, mod_to); if (ret < 0) return ret; - - dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); - dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); - dma->mod_from = mod_from; - dma->mod_to = mod_to; } - ret = rsnd_dai_connect(*dma_mod, io, type); - if (ret < 0) - return ret; - - return 0; + return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } int rsnd_dma_probe(struct rsnd_priv *priv) -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: add missing snd_soc_component_set_jack
by Kuninori Morimoto 20 Sep '17

20 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 44c07365e9e2 ("ASoC: add Component level set_jack") added new snd_soc_component_set_jack(), but it didn't add definition on soc.h. This patch adds it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index d22de97..996bdbc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1487,6 +1487,8 @@ int snd_soc_component_set_sysclk(struct snd_soc_component *component, int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_component_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data); #ifdef CONFIG_REGMAP -- 1.9.1

1 1

[alsa-devel] [PATCH] ALSA: hda - program ICT bits to support HBR audio
by Subhransu S. Prusty 20 Sep '17

20 Sep '17

From: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> On recent Intel platforms (Haswell, Broadwell, Skylake, ApolloLake, KabyLake, ...), the IEC Coding Type (ICT) bitfield in the Digital Converter Control #3 needs to be set explicitly for HDMI/DisplayPort High Bit Rate (HBR) audio playback to work. This was not required in earlier platforms when HBR was first introduced. The ICT bits are defined in Section 7.3.3.9 of the HDaudio 1.0a specification. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=98797 Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- include/sound/hda_verbs.h | 1 + sound/pci/hda/patch_hdmi.c | 18 ++++++++++++++++++ 2 files changed, 19 insertions(+) diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index d0509db6d0ec..d2004d7feb2b 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -54,6 +54,7 @@ enum { #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d #define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ +#define AC_VERB_SET_DIGI_CONVERT_3 0x73e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 53f9311370de..2dcd99d2a757 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -906,6 +906,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; + unsigned int param; int err; err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); @@ -915,6 +916,23 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return err; } + /* + * on recent platforms IEC Coding Type is required for HBR support, + * read current Digital Converter settings and set ICT bitfield if + * needed. + */ + param = snd_hda_codec_read(codec, cvt_nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + + param = (param >> 16) & ~(AC_DIG3_ICT); + + /* on recent platforms ICT mode is required for HBR support */ + if (is_hbr_format(format)) + param |= 0x1; + + snd_hda_codec_write(codec, cvt_nid, 0, + AC_VERB_SET_DIGI_CONVERT_3, param); + snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); return 0; } -- 1.9.1

4 4

[alsa-devel] [PATCH] ALSA: hrtimer: make hrtimer_hw const and __initconst
by Bhumika Goyal 19 Sep '17

19 Sep '17

Make this const as it is only used during a copy operation. Also, make it __initconst as it is only used during the init phase and after this it is not referenced anywhere. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/core/hrtimer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 1ac0c42..5603e49 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -127,7 +127,7 @@ static int snd_hrtimer_stop(struct snd_timer *t) return 0; } -static struct snd_timer_hardware hrtimer_hw = { +static const struct snd_timer_hardware hrtimer_hw __initconst = { .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET, .open = snd_hrtimer_open, .close = snd_hrtimer_close, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: asihpi: fix a potential double-fetch bug when copying puhm
by Meng Xu 19 Sep '17

19 Sep '17

The hm->h.size is intended to hold the actual size of the hm struct that is copied from userspace and should always be <= sizeof(*hm). However, after copy_from_user(hm, puhm, hm->h.size), since userspace process has full control over the memory region pointed by puhm, it is possible that the value of hm->h.size is different from what is fetched-in previously (get_user(hm->h.size, (u16 __user *)puhm)). In other words, hm->h.size is overriden and the relation between hm->h.size and the hm struct is broken. This patch proposes to use a seperate variable, msg_size, to hold the value of the first fetch and override hm->h.size to msg_size after the second fetch to maintain the relation. Signed-off-by: Meng Xu <mengxu.gatech(a)gmail.com> --- sound/pci/asihpi/hpioctl.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 7e3aa50..5badd08 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -103,6 +103,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) void __user *puhr; union hpi_message_buffer_v1 *hm; union hpi_response_buffer_v1 *hr; + u16 msg_size; u16 res_max_size; u32 uncopied_bytes; int err = 0; @@ -127,22 +128,25 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) } /* Now read the message size and data from user space. */ - if (get_user(hm->h.size, (u16 __user *)puhm)) { + if (get_user(msg_size, (u16 __user *)puhm)) { err = -EFAULT; goto out; } - if (hm->h.size > sizeof(*hm)) - hm->h.size = sizeof(*hm); + if (msg_size > sizeof(*hm)) + msg_size = sizeof(*hm); /* printk(KERN_INFO "message size %d\n", hm->h.wSize); */ - uncopied_bytes = copy_from_user(hm, puhm, hm->h.size); + uncopied_bytes = copy_from_user(hm, puhm, msg_size); if (uncopied_bytes) { HPI_DEBUG_LOG(ERROR, "uncopied bytes %d\n", uncopied_bytes); err = -EFAULT; goto out; } + /* Override h.size in case it is changed between two userspace fetches */ + hm->h.size = msg_size; + if (get_user(res_max_size, (u16 __user *)puhr)) { err = -EFAULT; goto out; -- 2.7.4

3 3

[alsa-devel] [asoc:for-next 9/9] sound/soc/codecs/wm9712.c:737:1: error: invalid storage class for function '__inittest'
by kbuild test robot 19 Sep '17

19 Sep '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 8c860a21cc365e822e70f970684179a50d22a77b commit: 8c860a21cc365e822e70f970684179a50d22a77b [9/9] Merge remote-tracking branches 'asoc/topic/wm97xx' and 'asoc/topic/zte' into asoc-next config: tile-allmodconfig (attached as .config) compiler: tilegx-linux-gcc (GCC) 4.6.2 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8c860a21cc365e822e70f970684179a50d22a77b # save the attached .config to linux build tree make.cross ARCH=tile All errors (new ones prefixed by >>): sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:675:12: error: invalid storage class for function 'wm9712_soc_remove' sound/soc/codecs/wm9712.c:675:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] sound/soc/codecs/wm9712.c:690:2: error: initializer element is not constant sound/soc/codecs/wm9712.c:690:2: error: (near initialization for 'soc_codec_dev_wm9712.remove') sound/soc/codecs/wm9712.c:705:12: error: invalid storage class for function 'wm9712_probe' sound/soc/codecs/wm9712.c:722:12: error: invalid storage class for function 'wm9712_remove' sound/soc/codecs/wm9712.c:733:2: error: initializer element is not constant sound/soc/codecs/wm9712.c:733:2: error: (near initialization for 'wm9712_codec_driver.probe') sound/soc/codecs/wm9712.c:734:2: error: initializer element is not constant sound/soc/codecs/wm9712.c:734:2: error: (near initialization for 'wm9712_codec_driver.remove') sound/soc/codecs/wm9712.c:737:1: error: invalid storage class for function 'wm9712_codec_driver_init' >> sound/soc/codecs/wm9712.c:737:1: error: invalid storage class for function '__inittest' sound/soc/codecs/wm9712.c:737:1: warning: 'alias' attribute ignored [-Wattributes] sound/soc/codecs/wm9712.c:737:1: error: invalid storage class for function 'wm9712_codec_driver_exit' sound/soc/codecs/wm9712.c:737:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] >> sound/soc/codecs/wm9712.c:737:1: error: invalid storage class for function '__exittest' sound/soc/codecs/wm9712.c:737:1: warning: 'alias' attribute ignored [-Wattributes] sound/soc/codecs/wm9712.c:739:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] sound/soc/codecs/wm9712.c:741:1: error: expected declaration or statement at end of input sound/soc/codecs/wm9712.c: At top level: sound/soc/codecs/wm9712.c:637:12: warning: 'wm9712_soc_probe' defined but not used [-Wunused-function] vim +/__inittest +737 sound/soc/codecs/wm9712.c 10c5cf304 Richard Purdie 2006-10-06 687 f802d6c02 Julia Lawall 2016-08-31 688 static const struct snd_soc_codec_driver soc_codec_dev_wm9712 = { 10c5cf304 Richard Purdie 2006-10-06 689 .probe = wm9712_soc_probe, 10c5cf304 Richard Purdie 2006-10-06 @690 .remove = wm9712_soc_remove, 10c5cf304 Richard Purdie 2006-10-06 691 .resume = wm9712_soc_resume, f0fba2ad1 Liam Girdwood 2010-03-17 692 .set_bias_level = wm9712_set_bias_level, ab492b86b Lars-Peter Clausen 2014-11-23 693 .suspend_bias_off = true, 9a812c6b7 Lars-Peter Clausen 2014-10-30 694 3123e54b3 Kuninori Morimoto 2016-08-08 695 .component_driver = { 9a812c6b7 Lars-Peter Clausen 2014-10-30 696 .controls = wm9712_snd_ac97_controls, 9a812c6b7 Lars-Peter Clausen 2014-10-30 697 .num_controls = ARRAY_SIZE(wm9712_snd_ac97_controls), 983347785 Lu Guanqun 2011-03-30 698 .dapm_widgets = wm9712_dapm_widgets, 983347785 Lu Guanqun 2011-03-30 699 .num_dapm_widgets = ARRAY_SIZE(wm9712_dapm_widgets), 983347785 Lu Guanqun 2011-03-30 700 .dapm_routes = wm9712_audio_map, 983347785 Lu Guanqun 2011-03-30 701 .num_dapm_routes = ARRAY_SIZE(wm9712_audio_map), 3123e54b3 Kuninori Morimoto 2016-08-08 702 }, 10c5cf304 Richard Purdie 2006-10-06 703 }; f0fba2ad1 Liam Girdwood 2010-03-17 704 7a79e94e9 Bill Pemberton 2012-12-07 705 static int wm9712_probe(struct platform_device *pdev) f0fba2ad1 Liam Girdwood 2010-03-17 706 { cf1f2ebe8 Lars-Peter Clausen 2014-11-03 707 struct wm9712_priv *wm9712; cf1f2ebe8 Lars-Peter Clausen 2014-11-03 708 cf1f2ebe8 Lars-Peter Clausen 2014-11-03 709 wm9712 = devm_kzalloc(&pdev->dev, sizeof(*wm9712), GFP_KERNEL); cf1f2ebe8 Lars-Peter Clausen 2014-11-03 710 if (wm9712 == NULL) cf1f2ebe8 Lars-Peter Clausen 2014-11-03 711 return -ENOMEM; cf1f2ebe8 Lars-Peter Clausen 2014-11-03 712 cf1f2ebe8 Lars-Peter Clausen 2014-11-03 713 mutex_init(&wm9712->lock); cf1f2ebe8 Lars-Peter Clausen 2014-11-03 714 2ed1a8e0c Robert Jarzmik 2017-09-13 715 wm9712->mfd_pdata = dev_get_platdata(&pdev->dev); cf1f2ebe8 Lars-Peter Clausen 2014-11-03 716 platform_set_drvdata(pdev, wm9712); cf1f2ebe8 Lars-Peter Clausen 2014-11-03 717 f0fba2ad1 Liam Girdwood 2010-03-17 718 return snd_soc_register_codec(&pdev->dev, f0fba2ad1 Liam Girdwood 2010-03-17 719 &soc_codec_dev_wm9712, wm9712_dai, ARRAY_SIZE(wm9712_dai)); f0fba2ad1 Liam Girdwood 2010-03-17 720 } f0fba2ad1 Liam Girdwood 2010-03-17 721 7a79e94e9 Bill Pemberton 2012-12-07 722 static int wm9712_remove(struct platform_device *pdev) f0fba2ad1 Liam Girdwood 2010-03-17 723 { f0fba2ad1 Liam Girdwood 2010-03-17 724 snd_soc_unregister_codec(&pdev->dev); f0fba2ad1 Liam Girdwood 2010-03-17 725 return 0; f0fba2ad1 Liam Girdwood 2010-03-17 726 } f0fba2ad1 Liam Girdwood 2010-03-17 727 f0fba2ad1 Liam Girdwood 2010-03-17 728 static struct platform_driver wm9712_codec_driver = { f0fba2ad1 Liam Girdwood 2010-03-17 729 .driver = { f0fba2ad1 Liam Girdwood 2010-03-17 730 .name = "wm9712-codec", f0fba2ad1 Liam Girdwood 2010-03-17 731 }, f0fba2ad1 Liam Girdwood 2010-03-17 732 f0fba2ad1 Liam Girdwood 2010-03-17 @733 .probe = wm9712_probe, 7a79e94e9 Bill Pemberton 2012-12-07 734 .remove = wm9712_remove, f0fba2ad1 Liam Girdwood 2010-03-17 735 }; f0fba2ad1 Liam Girdwood 2010-03-17 736 5bbcc3c0d Mark Brown 2011-11-23 @737 module_platform_driver(wm9712_codec_driver); 10c5cf304 Richard Purdie 2006-10-06 738 :::::: The code at line 737 was first introduced by commit :::::: 5bbcc3c0d0f063318ec83146d1958acf7154c66f ASoC: Convert CODEC drivers to module_platform_driver :::::: TO: Mark Brown <broonie(a)opensource.wolfsonmicro.com> :::::: CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-next 9/9] sound/soc/codecs/wm9712.c:675:12: error: invalid storage class for function 'wm9712_soc_remove'
by kbuild test robot 19 Sep '17

19 Sep '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 8c860a21cc365e822e70f970684179a50d22a77b commit: 8c860a21cc365e822e70f970684179a50d22a77b [9/9] Merge remote-tracking branches 'asoc/topic/wm97xx' and 'asoc/topic/zte' into asoc-next config: xtensa-allmodconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8c860a21cc365e822e70f970684179a50d22a77b # save the attached .config to linux build tree make.cross ARCH=xtensa All error/warnings (new ones prefixed by >>): sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': >> sound/soc/codecs/wm9712.c:675:12: error: invalid storage class for function 'wm9712_soc_remove' static int wm9712_soc_remove(struct snd_soc_codec *codec) ^ >> sound/soc/codecs/wm9712.c:675:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] static int wm9712_soc_remove(struct snd_soc_codec *codec) ^ >> sound/soc/codecs/wm9712.c:690:2: error: initializer element is not constant .remove = wm9712_soc_remove, ^ sound/soc/codecs/wm9712.c:690:2: error: (near initialization for 'soc_codec_dev_wm9712.remove') >> sound/soc/codecs/wm9712.c:705:12: error: invalid storage class for function 'wm9712_probe' static int wm9712_probe(struct platform_device *pdev) ^ >> sound/soc/codecs/wm9712.c:722:12: error: invalid storage class for function 'wm9712_remove' static int wm9712_remove(struct platform_device *pdev) ^ sound/soc/codecs/wm9712.c:733:2: error: initializer element is not constant .probe = wm9712_probe, ^ sound/soc/codecs/wm9712.c:733:2: error: (near initialization for 'wm9712_codec_driver.probe') sound/soc/codecs/wm9712.c:734:2: error: initializer element is not constant .remove = wm9712_remove, ^ sound/soc/codecs/wm9712.c:734:2: error: (near initialization for 'wm9712_codec_driver.remove') In file included from sound/soc/codecs/wm9712.c:18:0: >> sound/soc/codecs/wm9712.c:737:24: error: invalid storage class for function 'wm9712_codec_driver_init' module_platform_driver(wm9712_codec_driver); ^ include/linux/device.h:1502:19: note: in definition of macro 'module_driver' static int __init __driver##_init(void) \ ^ >> sound/soc/codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ In file included from sound/soc/codecs/wm9712.c:16:0: >> include/linux/module.h:129:42: error: invalid storage class for function '__inittest' static inline initcall_t __maybe_unused __inittest(void) \ ^ include/linux/device.h:1506:1: note: in expansion of macro 'module_init' module_init(__driver##_init); \ ^ include/linux/platform_device.h:228:2: note: in expansion of macro 'module_driver' module_driver(__platform_driver, platform_driver_register, \ ^ >> sound/soc/codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ >> sound/soc/codecs/wm9712.c:737:1: warning: 'alias' attribute ignored [-Wattributes] In file included from sound/soc/codecs/wm9712.c:18:0: >> sound/soc/codecs/wm9712.c:737:24: error: invalid storage class for function 'wm9712_codec_driver_exit' module_platform_driver(wm9712_codec_driver); ^ include/linux/device.h:1507:20: note: in definition of macro 'module_driver' static void __exit __driver##_exit(void) \ ^ >> sound/soc/codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ >> include/linux/device.h:1507:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] static void __exit __driver##_exit(void) \ ^ include/linux/platform_device.h:228:2: note: in expansion of macro 'module_driver' module_driver(__platform_driver, platform_driver_register, \ ^ >> sound/soc/codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ In file included from sound/soc/codecs/wm9712.c:16:0: >> include/linux/module.h:135:42: error: invalid storage class for function '__exittest' static inline exitcall_t __maybe_unused __exittest(void) \ ^ include/linux/device.h:1511:1: note: in expansion of macro 'module_exit' module_exit(__driver##_exit); ^ include/linux/platform_device.h:228:2: note: in expansion of macro 'module_driver' module_driver(__platform_driver, platform_driver_register, \ ^ >> sound/soc/codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ >> sound/soc/codecs/wm9712.c:737:1: warning: 'alias' attribute ignored [-Wattributes] In file included from include/linux/module.h:18:0, from sound/soc/codecs/wm9712.c:16: include/linux/moduleparam.h:21:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] static const char __UNIQUE_ID(name)[] \ ^ include/linux/module.h:160:32: note: in expansion of macro '__MODULE_INFO' #define MODULE_INFO(tag, info) __MODULE_INFO(tag, tag, info) ^ include/linux/module.h:207:42: note: in expansion of macro 'MODULE_INFO' #define MODULE_DESCRIPTION(_description) MODULE_INFO(description, _description) ^ >> sound/soc/codecs/wm9712.c:739:1: note: in expansion of macro 'MODULE_DESCRIPTION' MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); ^ >> sound/soc/codecs/wm9712.c:741:1: error: expected declaration or statement at end of input MODULE_LICENSE("GPL"); ^ sound/soc/codecs/wm9712.c: At top level: sound/soc/codecs/wm9712.c:637:12: warning: 'wm9712_soc_probe' defined but not used [-Wunused-function] static int wm9712_soc_probe(struct snd_soc_codec *codec) ^ -- sound/soc//codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc//codecs/wm9712.c:675:12: error: invalid storage class for function 'wm9712_soc_remove' static int wm9712_soc_remove(struct snd_soc_codec *codec) ^ sound/soc//codecs/wm9712.c:675:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] static int wm9712_soc_remove(struct snd_soc_codec *codec) ^ sound/soc//codecs/wm9712.c:690:2: error: initializer element is not constant .remove = wm9712_soc_remove, ^ sound/soc//codecs/wm9712.c:690:2: error: (near initialization for 'soc_codec_dev_wm9712.remove') sound/soc//codecs/wm9712.c:705:12: error: invalid storage class for function 'wm9712_probe' static int wm9712_probe(struct platform_device *pdev) ^ sound/soc//codecs/wm9712.c:722:12: error: invalid storage class for function 'wm9712_remove' static int wm9712_remove(struct platform_device *pdev) ^ sound/soc//codecs/wm9712.c:733:2: error: initializer element is not constant .probe = wm9712_probe, ^ sound/soc//codecs/wm9712.c:733:2: error: (near initialization for 'wm9712_codec_driver.probe') sound/soc//codecs/wm9712.c:734:2: error: initializer element is not constant .remove = wm9712_remove, ^ sound/soc//codecs/wm9712.c:734:2: error: (near initialization for 'wm9712_codec_driver.remove') In file included from sound/soc//codecs/wm9712.c:18:0: sound/soc//codecs/wm9712.c:737:24: error: invalid storage class for function 'wm9712_codec_driver_init' module_platform_driver(wm9712_codec_driver); ^ include/linux/device.h:1502:19: note: in definition of macro 'module_driver' static int __init __driver##_init(void) \ ^ sound/soc//codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ In file included from sound/soc//codecs/wm9712.c:16:0: >> include/linux/module.h:129:42: error: invalid storage class for function '__inittest' static inline initcall_t __maybe_unused __inittest(void) \ ^ include/linux/device.h:1506:1: note: in expansion of macro 'module_init' module_init(__driver##_init); \ ^ include/linux/platform_device.h:228:2: note: in expansion of macro 'module_driver' module_driver(__platform_driver, platform_driver_register, \ ^ sound/soc//codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ sound/soc//codecs/wm9712.c:737:1: warning: 'alias' attribute ignored [-Wattributes] In file included from sound/soc//codecs/wm9712.c:18:0: sound/soc//codecs/wm9712.c:737:24: error: invalid storage class for function 'wm9712_codec_driver_exit' module_platform_driver(wm9712_codec_driver); ^ include/linux/device.h:1507:20: note: in definition of macro 'module_driver' static void __exit __driver##_exit(void) \ ^ sound/soc//codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ >> include/linux/device.h:1507:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] static void __exit __driver##_exit(void) \ ^ include/linux/platform_device.h:228:2: note: in expansion of macro 'module_driver' module_driver(__platform_driver, platform_driver_register, \ ^ sound/soc//codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ In file included from sound/soc//codecs/wm9712.c:16:0: >> include/linux/module.h:135:42: error: invalid storage class for function '__exittest' static inline exitcall_t __maybe_unused __exittest(void) \ ^ include/linux/device.h:1511:1: note: in expansion of macro 'module_exit' module_exit(__driver##_exit); ^ include/linux/platform_device.h:228:2: note: in expansion of macro 'module_driver' module_driver(__platform_driver, platform_driver_register, \ ^ sound/soc//codecs/wm9712.c:737:1: note: in expansion of macro 'module_platform_driver' module_platform_driver(wm9712_codec_driver); ^ sound/soc//codecs/wm9712.c:737:1: warning: 'alias' attribute ignored [-Wattributes] In file included from include/linux/module.h:18:0, from sound/soc//codecs/wm9712.c:16: include/linux/moduleparam.h:21:1: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement] static const char __UNIQUE_ID(name)[] \ ^ include/linux/module.h:160:32: note: in expansion of macro '__MODULE_INFO' #define MODULE_INFO(tag, info) __MODULE_INFO(tag, tag, info) ^ include/linux/module.h:207:42: note: in expansion of macro 'MODULE_INFO' #define MODULE_DESCRIPTION(_description) MODULE_INFO(description, _description) ^ sound/soc//codecs/wm9712.c:739:1: note: in expansion of macro 'MODULE_DESCRIPTION' MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); ^ sound/soc//codecs/wm9712.c:741:1: error: expected declaration or statement at end of input MODULE_LICENSE("GPL"); ^ sound/soc//codecs/wm9712.c: At top level: sound/soc//codecs/wm9712.c:637:12: warning: 'wm9712_soc_probe' defined but not used [-Wunused-function] static int wm9712_soc_probe(struct snd_soc_codec *codec) ^ vim +/wm9712_soc_remove +675 sound/soc/codecs/wm9712.c 10c5cf304 Richard Purdie 2006-10-06 674 f0fba2ad1 Liam Girdwood 2010-03-17 @675 static int wm9712_soc_remove(struct snd_soc_codec *codec) 10c5cf304 Richard Purdie 2006-10-06 676 { 2ed1a8e0c Robert Jarzmik 2017-09-13 677 #ifdef CONFIG_SND_SOC_AC97_BUS 358a8bb56 Lars-Peter Clausen 2014-11-10 678 struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); 358a8bb56 Lars-Peter Clausen 2014-11-10 679 2ed1a8e0c Robert Jarzmik 2017-09-13 680 if (!wm9712->mfd_pdata) { b4f4f2eba Lars-Peter Clausen 2016-09-30 681 snd_soc_codec_exit_regmap(codec); 358a8bb56 Lars-Peter Clausen 2014-11-10 682 snd_soc_free_ac97_codec(wm9712->ac97); 2ed1a8e0c Robert Jarzmik 2017-09-13 683 } 2ed1a8e0c Robert Jarzmik 2017-09-13 684 #endif 10c5cf304 Richard Purdie 2006-10-06 685 return 0; 10c5cf304 Richard Purdie 2006-10-06 686 } 10c5cf304 Richard Purdie 2006-10-06 687 f802d6c02 Julia Lawall 2016-08-31 688 static const struct snd_soc_codec_driver soc_codec_dev_wm9712 = { 10c5cf304 Richard Purdie 2006-10-06 689 .probe = wm9712_soc_probe, 10c5cf304 Richard Purdie 2006-10-06 @690 .remove = wm9712_soc_remove, 10c5cf304 Richard Purdie 2006-10-06 691 .resume = wm9712_soc_resume, f0fba2ad1 Liam Girdwood 2010-03-17 692 .set_bias_level = wm9712_set_bias_level, ab492b86b Lars-Peter Clausen 2014-11-23 693 .suspend_bias_off = true, 9a812c6b7 Lars-Peter Clausen 2014-10-30 694 3123e54b3 Kuninori Morimoto 2016-08-08 695 .component_driver = { 9a812c6b7 Lars-Peter Clausen 2014-10-30 696 .controls = wm9712_snd_ac97_controls, 9a812c6b7 Lars-Peter Clausen 2014-10-30 697 .num_controls = ARRAY_SIZE(wm9712_snd_ac97_controls), 983347785 Lu Guanqun 2011-03-30 698 .dapm_widgets = wm9712_dapm_widgets, 983347785 Lu Guanqun 2011-03-30 699 .num_dapm_widgets = ARRAY_SIZE(wm9712_dapm_widgets), 983347785 Lu Guanqun 2011-03-30 700 .dapm_routes = wm9712_audio_map, 983347785 Lu Guanqun 2011-03-30 701 .num_dapm_routes = ARRAY_SIZE(wm9712_audio_map), 3123e54b3 Kuninori Morimoto 2016-08-08 702 }, 10c5cf304 Richard Purdie 2006-10-06 703 }; f0fba2ad1 Liam Girdwood 2010-03-17 704 7a79e94e9 Bill Pemberton 2012-12-07 @705 static int wm9712_probe(struct platform_device *pdev) f0fba2ad1 Liam Girdwood 2010-03-17 706 { cf1f2ebe8 Lars-Peter Clausen 2014-11-03 707 struct wm9712_priv *wm9712; cf1f2ebe8 Lars-Peter Clausen 2014-11-03 708 cf1f2ebe8 Lars-Peter Clausen 2014-11-03 709 wm9712 = devm_kzalloc(&pdev->dev, sizeof(*wm9712), GFP_KERNEL); cf1f2ebe8 Lars-Peter Clausen 2014-11-03 710 if (wm9712 == NULL) cf1f2ebe8 Lars-Peter Clausen 2014-11-03 711 return -ENOMEM; cf1f2ebe8 Lars-Peter Clausen 2014-11-03 712 cf1f2ebe8 Lars-Peter Clausen 2014-11-03 713 mutex_init(&wm9712->lock); cf1f2ebe8 Lars-Peter Clausen 2014-11-03 714 2ed1a8e0c Robert Jarzmik 2017-09-13 715 wm9712->mfd_pdata = dev_get_platdata(&pdev->dev); cf1f2ebe8 Lars-Peter Clausen 2014-11-03 716 platform_set_drvdata(pdev, wm9712); cf1f2ebe8 Lars-Peter Clausen 2014-11-03 717 f0fba2ad1 Liam Girdwood 2010-03-17 718 return snd_soc_register_codec(&pdev->dev, f0fba2ad1 Liam Girdwood 2010-03-17 719 &soc_codec_dev_wm9712, wm9712_dai, ARRAY_SIZE(wm9712_dai)); f0fba2ad1 Liam Girdwood 2010-03-17 720 } f0fba2ad1 Liam Girdwood 2010-03-17 721 7a79e94e9 Bill Pemberton 2012-12-07 @722 static int wm9712_remove(struct platform_device *pdev) f0fba2ad1 Liam Girdwood 2010-03-17 723 { f0fba2ad1 Liam Girdwood 2010-03-17 724 snd_soc_unregister_codec(&pdev->dev); f0fba2ad1 Liam Girdwood 2010-03-17 725 return 0; f0fba2ad1 Liam Girdwood 2010-03-17 726 } f0fba2ad1 Liam Girdwood 2010-03-17 727 f0fba2ad1 Liam Girdwood 2010-03-17 728 static struct platform_driver wm9712_codec_driver = { f0fba2ad1 Liam Girdwood 2010-03-17 729 .driver = { f0fba2ad1 Liam Girdwood 2010-03-17 730 .name = "wm9712-codec", f0fba2ad1 Liam Girdwood 2010-03-17 731 }, f0fba2ad1 Liam Girdwood 2010-03-17 732 f0fba2ad1 Liam Girdwood 2010-03-17 @733 .probe = wm9712_probe, 7a79e94e9 Bill Pemberton 2012-12-07 @734 .remove = wm9712_remove, f0fba2ad1 Liam Girdwood 2010-03-17 735 }; f0fba2ad1 Liam Girdwood 2010-03-17 736 5bbcc3c0d Mark Brown 2011-11-23 @737 module_platform_driver(wm9712_codec_driver); 10c5cf304 Richard Purdie 2006-10-06 738 10c5cf304 Richard Purdie 2006-10-06 @739 MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); 10c5cf304 Richard Purdie 2006-10-06 740 MODULE_AUTHOR("Liam Girdwood"); 10c5cf304 Richard Purdie 2006-10-06 @741 MODULE_LICENSE("GPL"); :::::: The code at line 675 was first introduced by commit :::::: f0fba2ad1b6b53d5360125c41953b7afcd6deff0 ASoC: multi-component - ASoC Multi-Component Support :::::: TO: Liam Girdwood <lrg(a)slimlogic.co.uk> :::::: CC: Liam Girdwood <lrg(a)slimlogic.co.uk> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: Intel: Atom: Remove unneeded linux/miscdevice.h include
by Corentin Labbe 19 Sep '17

19 Sep '17

No file in sound/soc/intel/ use any miscdevice. This patch remove this uncessary include. Signed-off-by: Corentin Labbe <clabbe.montjoie(a)gmail.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index f4d92bb..a971a58 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -23,7 +23,6 @@ #include <linux/interrupt.h> #include <linux/slab.h> #include <linux/io.h> -#include <linux/miscdevice.h> #include <linux/platform_device.h> #include <linux/firmware.h> #include <linux/pm_runtime.h> -- 2.10.2

2 1

[alsa-devel] [PATCH v2 00/12] AC97 device/driver model revamp
by Robert Jarzmik 19 Sep '17

19 Sep '17

Hi Lars, Mark, Charles, Lee, This serie has been stalled for almost one year, as I wasn't feeling that well lately. I'm quite better now, so let's get back to it. As it's been so long since the last review, I basically withdrew all my former objections and took into account all remarks regardless of my own judgement. This is especially the case with Lars's review on the bus code, where I tried to address all his comments. For this next round, let's sum up what happened : - I rebased against latest Linus's tree => as Lars had made ports to regmap of several impacted wm97xx drivers, all the better for me. - ALSA: ac97: add an ac97 bus => this addresses Lars's comments => history : https://patchwork.kernel.org/patch/9398229/ => there is one remark which was left out, the drop of the table ac97_controller->codecs. The reason is that there are traversals required in the bus scan and controller removal, and the code is way simpler with a table, and the data structure very compact. - Input: wm97xx: split out touchscreen registering => Charles's comment taken into account => history: http://https://patchwork.kernel.org/patch/9398223/ Have a happy review. -- Robert Robert Jarzmik (11): ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support Takashi Iwai (1): ASoC: Fix use-after-free at card unregistration drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 ++++++++++----- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 282 +++++++++++++++++ include/linux/mfd/wm97xx.h | 31 ++ include/sound/ac97/codec.h | 118 +++++++ include/sound/ac97/compat.h | 21 ++ include/sound/ac97/controller.h | 85 ++++++ include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 10 + sound/ac97/bus.c | 526 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 105 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 37 ++- sound/arm/pxa2xx-ac97.c | 35 ++- sound/soc/Kconfig | 4 + sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/wm9705.c | 59 ++-- sound/soc/codecs/wm9712.c | 37 ++- sound/soc/codecs/wm9713.c | 39 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 43 ++- sound/soc/soc-core.c | 5 +- 29 files changed, 1603 insertions(+), 178 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.1.4

4 23

[alsa-devel] [PATCH 0/9] AC97 device/driver model revamp
by Robert Jarzmik 19 Sep '17

19 Sep '17

It all started in the pxa device-tree submission here : https://lkml.org/lkml/2016/2/25/965 Last submission was a RFC in here: http://www.gossamer-threads.com/lists/linux/kernel/2446863 It will be maintained in : git fetch https://github.com/rjarzmik/linux.git work/ac97 The driving ideas are still the same, and I put them in [1] for memory. This is the first post RFC submission. In order to make a full demonstration of the framework, wm97xx was converted to an MFD, see [2] for the "why". As this serie is a complete wm9713 change towards the new AC97 bus, here is the patch organization : - 1/9, 2/9, 3/9 and 5/9: the new AC97 bus, for Mark and Takashi to review - 4/9: wm9713 conversion, this one has ugly #ifdefs, should be reviewed more carefully - 6/9: prerequisite for final 9/9, for Marek and Sebastian to approve => this one is independent and can flow through Sebastian's tree in this next-cycle if accepted - 7/9: prerequisite for final 9/9, for Mark/Dmitry => this one is also independent and can flow in this cycle if accepted - 8/9: prerequisite for final 9/9, for Lee to review => this one has a dependency on 2/9, so it probably can't fit in this next-cycle - 9/9: this last one depends on all the patches before, so it can't fit in this next-cycle As a sum-up, I'd like to push for review for this -next cycle at least the patches 1, 2, 3, 5, 6 and 9. Happy review. Robert Jarzmik (9): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: wm9713: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support power_supply: wm97xx_battery: use power_supply_get_drvdata Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 247 +++++++++++----- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 282 ++++++++++++++++++ drivers/power/supply/wm97xx_battery.c | 25 +- include/linux/mfd/wm97xx.h | 31 ++ include/sound/ac97/codec.h | 115 +++++++ include/sound/ac97/compat.h | 21 ++ include/sound/ac97/controller.h | 85 ++++++ include/sound/ac97/regs.h | 262 ++++++++++++++++ include/sound/ac97_codec.h | 239 +-------------- include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 10 + sound/ac97/bus.c | 510 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 105 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 39 ++- sound/soc/Kconfig | 4 + sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/wm9713.c | 37 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 22 +- 28 files changed, 1738 insertions(+), 382 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.1.4 [1] RFC PATCH reminder: Amongst the driving ideas behind this RFC : - use device/driver model for AC'97 device/driver binding is based on vendor_id{1,2} - use auto-probe of the AC'97 bus to enumerate codecs and create devices - be compatible with previous platform_data model for codecs => this enables a smooth transition, where a codec (here wm9713) can be used either through platform_device or automatic discovery - struct snd_ac97 is not used This structure is really heavy, and doesn't represent an AC97 device, but rather an AC'97 codec internals IIUC. I think it's right place would be in include/sound/ac97/codec.h. This is also a problem for regmap adherence, hence the include/sound/ac97/compat.h. - a new AC'97 bus Kconfig is created This was done for a smooth transition ... let's see if it's a good idea. - split the AC'97 into : - the codec - the digital controller - the bus Amongst the things that are not all touched yet : - ac97_codec.c The generic AC'97 codec in sound/pci/ac97/ac97_codec.c has really a lot features, which are not at all PCI related. They could be transfered to sound/ac97, without the pci specific field. => this deserves a deep ahead thinking, which I'll do if the current approach is accepted by the communauty. The most important patch is 2/7 "ALSA: ac97: add an ac97 bus". This is the main evaluation point for the serie, the others are here so that I can test it all. Well, this is a long term effort, which might need a complete rewrite according to the comments it'll get. Let's expose it for comments and see how I can progress with it. [2] As a corollary, wm97xx is changed into a MFD device, which at least spawns: - the touchscreen device - the ac97 codec - the problem to solve was stated in http://www.gossamer-threads.com/lists/linux/kernel/2510113#2510113

9 38

[alsa-devel] Applied "ASoC: wm9713: add ac97 new bus support" to the asoc tree
by Mark Brown 19 Sep '17

19 Sep '17

The patch ASoC: wm9713: add ac97 new bus support has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9bd400cadae05a26c3a22dd8393081cb96d0e26a Mon Sep 17 00:00:00 2001 From: Robert Jarzmik <robert.jarzmik(a)free.fr> Date: Wed, 13 Sep 2017 21:37:19 +0200 Subject: [PATCH] ASoC: wm9713: add ac97 new bus support Add support for the new ac97 bus model, where devices are automatically discovered on AC-Links. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 3 ++- sound/soc/codecs/wm9713.c | 39 +++++++++++++++++++++++++++------------ 2 files changed, 29 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c367d11079bc..807901cac1b3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -216,7 +216,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9090 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS - select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -1136,6 +1136,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate select REGMAP_AC97 + select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_ZX_AUD96P22 tristate "ZTE ZX AUD96P22 CODEC" diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 7e4822185feb..df7220656d98 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -17,12 +17,15 @@ #include <linux/init.h> #include <linux/slab.h> +#include <linux/mfd/wm97xx.h> #include <linux/module.h> #include <linux/device.h> #include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> +#include <sound/ac97/codec.h> +#include <sound/ac97/compat.h> #include <sound/initval.h> #include <sound/pcm_params.h> #include <sound/tlv.h> @@ -38,6 +41,7 @@ struct wm9713_priv { u32 pll_in; /* PLL input frequency */ unsigned int hp_mixer[2]; struct mutex lock; + struct wm97xx_platform_data *mfd_pdata; }; #define HPL_MIXER 0 @@ -1205,17 +1209,23 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - struct regmap *regmap; + struct regmap *regmap = NULL; - wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, - WM9713_VENDOR_ID_MASK); - if (IS_ERR(wm9713->ac97)) - return PTR_ERR(wm9713->ac97); - - regmap = regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); - if (IS_ERR(regmap)) { - snd_soc_free_ac97_codec(wm9713->ac97); - return PTR_ERR(regmap); + if (wm9713->mfd_pdata) { + wm9713->ac97 = wm9713->mfd_pdata->ac97; + regmap = wm9713->mfd_pdata->regmap; + } else { +#ifdef CONFIG_SND_SOC_AC97_BUS + wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, + WM9713_VENDOR_ID_MASK); + if (IS_ERR(wm9713->ac97)) + return PTR_ERR(wm9713->ac97); + regmap = regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); + if (IS_ERR(regmap)) { + snd_soc_free_ac97_codec(wm9713->ac97); + return PTR_ERR(regmap); + } +#endif } snd_soc_codec_init_regmap(codec, regmap); @@ -1228,10 +1238,14 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) static int wm9713_soc_remove(struct snd_soc_codec *codec) { +#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_exit_regmap(codec); - snd_soc_free_ac97_codec(wm9713->ac97); + if (!wm9713->mfd_pdata) { + snd_soc_codec_exit_regmap(codec); + snd_soc_free_ac97_codec(wm9713->ac97); + } +#endif return 0; } @@ -1262,6 +1276,7 @@ static int wm9713_probe(struct platform_device *pdev) mutex_init(&wm9713->lock); + wm9713->mfd_pdata = dev_get_platdata(&pdev->dev); platform_set_drvdata(pdev, wm9713); return snd_soc_register_codec(&pdev->dev, -- 2.14.1

1 0

[alsa-devel] [PATCH v2 0/3] Fix capture devices functionality on TM2
by Jaechul Lee 19 Sep '17

19 Sep '17

Hello, This patchset makes capture functionality working on TM2. Capture devices can't be detected because I2S doesn't have appropriate dai name. Therefore, TM2 will select unexpected cpu dai which is named "i2s-samsung-sec" that doesn't have a capture functionality. In samsung I2S driver, it tries to register two components without specific name. The driver finally has components having same name: dai name as well. As a result, dai_link doesn't have enough information to select cpu_dai properly. Changes in v2: - defined SAMSUNG_I2S_DAI_* in i2s.h and removed duplicated code - modified commit message about invalid parameter that Krzystof mentioned - rebased code because of conflicts in i2s.c Jaechul Lee (3): ASoC: samsung: i2s: Use specific name for i2s dais ASoC: samsung: Use 'samsung-i2s' cpu_dai for dai_links ASoC: samsung: Fix invalid argument when devm_gpiod_get is called sound/soc/samsung/i2s.c | 4 ++++ sound/soc/samsung/i2s.h | 3 +++ sound/soc/samsung/tm2_wm5110.c | 7 ++++--- 3 files changed, 11 insertions(+), 3 deletions(-) -- 2.14.1

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[alsa-devel] [PATCH 0/3] Fix capture devices functionality on TM2
by Jaechul Lee 19 Sep '17

19 Sep '17

Hello, This patchset makes capture functionality working on TM2. Capture devices can't be detected because I2S doesn't have appropriate dai name. Therefore, TM2 will select unexpected cpu dai which is named "i2s-samsung-sec" that doesn't have a capture functionality. In samsung I2S driver, it tries to register two components without specific name. The driver finally has components having same name: dai name as well. As a result, dai_link doesn't have enough information to select cpu_dai properly. Jaechul Lee (3): ASoC: samsung: i2s: Use specific name for i2s dais ASoC: samsung: Use 'samsung-i2s' cpu_dai for dai_links ASoC: samsung: Fix invalid argument when devm_gpiod_get is called sound/soc/samsung/i2s.c | 6 ++++++ sound/soc/samsung/tm2_wm5110.c | 7 ++++--- 2 files changed, 10 insertions(+), 3 deletions(-) -- 2.14.1

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