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

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

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[alsa-devel] echo-cancellation plugin advice?
by Max Klein 27 Oct '23

27 Oct '23

> Well, asym wouldn't be the answer. Basically you'll need to open two > slave PCMs in one plugin. So, it's no simple filter-type plugin. > It's doable, but not easy as it sounds, I guess. Takashi, can you please further explain why asym is not the solution and what you see as a problem? I am also trying to figure out if it would be possible to create such a plugin for ALSA? I am still searching for some good starting points on how to write an ALSA plugin... Jean-Marc Valin (http://lists.linuxaudio.org/pipermail/linux-audio-dev/2005-June/012454.html) already released his work as part of speex but as far I know there is no plugin that has been made out of it? Thanks for any hint! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

3 2

[alsa-devel] ALSA 1.0.29 release
by Jaroslav Kysela 19 Feb '20

19 Feb '20

Hello all, all userspace packages (except alsa-oss) were released today. You may download them from the ALSA website http://www.alsa-project.org or directly from the FTP server ftp://ftp.alsa-project.org/pub . Full list of changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.28_v1.0.29 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH 1/1] ASoC: bcm2835-i2s: Support 384k sample rate
by Clive Messer 10 Nov '17

10 Nov '17

Add support for sample rates up to 384k to bcm2835-i2s driver, with SNDRV_PCM_RATE_CONTINUOUS and specifying 8k min and 384k max, for both playback and capture. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/bcm/bcm2835-i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 6ba2049..4a67db2 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -578,7 +578,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE @@ -586,7 +588,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE -- 2.5.5

3 2

[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] [RFC PATCH 0/7] AC97 device/driver model revamp
by Robert Jarzmik 04 Sep '17

04 Sep '17

It all started in the pxa device-tree submission here : https://lkml.org/lkml/2016/2/25/965 It will be maintained in : git fetch https://github.com/rjarzmik/linux.git work/ac97 And now it transformed into this RFC, which would bring a ground for AC'97 devices closer to the linux device/driver model. This is just an RFC to see if we understand each other Mark. 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. Cheers. -- Robert Robert Jarzmik (7): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: wm9713: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support ARM: pxa: mioa701 remove wm9713 from platform devices ASoC: mioa701_wm9713: convert to new ac97 bus ASoC: add new ac97 bus support arch/arm/mach-pxa/mioa701.c | 1 - include/sound/ac97/codec.h | 98 ++++++++++++ include/sound/ac97/compat.h | 21 +++ include/sound/ac97/controller.h | 39 +++++ include/sound/ac97/regs.h | 262 +++++++++++++++++++++++++++++++ include/sound/ac97_codec.h | 239 +---------------------------- include/sound/pxa2xx-lib.h | 14 +- sound/Kconfig | 7 + sound/Makefile | 1 + sound/ac97/Kconfig | 9 ++ sound/ac97/Makefile | 5 + sound/ac97/ac97_core.h | 10 ++ sound/ac97/bus.c | 330 ++++++++++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 ++ sound/ac97/snd_ac97_compat.c | 104 +++++++++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 31 ++-- sound/soc/Kconfig | 4 + sound/soc/codecs/wm9713.c | 62 +++++++- sound/soc/pxa/Kconfig | 4 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/pxa2xx-ac97.c | 12 +- 22 files changed, 1000 insertions(+), 273 deletions(-) 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

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Re: [alsa-devel] Dell USB audio driver workaround
by Takashi Iwai 21 Jun '17

21 Jun '17

On Wed, 06 Apr 2016 08:29:12 +0200, Kailang wrote: > > Hi Takashi, > > Dell had USB dock. > Maybe firmware issue, the master volume always keep at 0x57. > Attach patch will fix it to keep at Max volume. > > Could you help us to update it? Well, I don't get how this patch works.. > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c > index 4f85757..84d9a4e 100644 > --- a/sound/usb/mixer.c > +++ b/sound/usb/mixer.c > @@ -846,6 +846,12 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, > { > struct snd_usb_audio *chip = cval->head.mixer->chip; > switch (chip->usb_id) { > + case USB_ID(0x0bda, 0x4014): /* Dell workaround */ > + if (strstr(kctl->id.name, "Playback Volume") != NULL) { > + cval->min = 0x0000; > + cval->max = 0x0000; > + } > + break; So, this means that this value has to be fixed to 0? If yes, can't we simply remove this volume instead? thanks, Takashi

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[alsa-devel] [PATCH 0/7] of_graph: prepare for ALSA graph support
by Kuninori Morimoto 17 May '17

17 May '17

Hi Rob, Mark Cc Guennadi, Laurent Now OF graph is mainly used by V4L2 SoC, and ALSA SoC is using different style for SoC <-> Codec binding. But, for example, HDMI case, V4L2 <-> ALSA need to collaborate, and then ALSA SoC needs to adjust to OF graph. OTOH, V4L2's "OF graph" position is same as ALSA SoC "sound card" position. And ALSA SoC side want to keep existing supported feature on new OF graph style. I'm posting this on ALSA SoC ML now. Now, current of_graph is indicating port/endpoint, but there is no way to understand that it is for video port ? or sound port ? or other device port ? For example, HDMI has video port, and sound port. Because of this reason, ALSA SoC side can't handle OF graph correctly. Thus, this patch-set tries to add new "type" on OF graph. And this patch-set includes small feature which are useful for ALSA SoC side OF graph support. Kuninori Morimoto (7): Documentation: of: add type property of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_port/endpoint_type_is() of_graph: add of_graph_get_endpoint_count() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_top_port() of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() Documentation/devicetree/bindings/graph.txt | 26 ++++ drivers/of/base.c | 196 ++++++++++++++++++++++++++-- include/linux/of_graph.h | 67 ++++++++++ 3 files changed, 279 insertions(+), 10 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2 1/2] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 04 Apr '17

04 Apr '17

From: Zidan Wang <b50113(a)freescale.com> When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK. When current bias_level is SND_SOC_BIAS_ON, it is preparing for a transition away from ON. In this case, disable the codec mclk. When current bias_level is not SND_SOC_BIAS_ON, it preparing for a transition to ON. In this case, enable the codec mclk. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 51 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 48 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 75cd7de..fd25973 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +118,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -618,14 +620,38 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - /* Set VMID to 2x50k */ - snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_STANDBY: + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } + + /* Set VMID to 2x50k */ + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + break; + + case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + break; + + default: + break; + } + break; case SND_SOC_BIAS_STANDBY: @@ -674,7 +700,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; switch (level) { case SND_SOC_BIAS_ON: @@ -715,9 +741,22 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF, WM8960_VREF); msleep(100); + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } break; case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + /* Enable anti-pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -1002,6 +1041,12 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); -- 1.9.1

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[alsa-devel] Intel HDA audio on EEE PC 1101HGo
by Enrico Mioso 20 Jan '17

20 Jan '17

Hello guys. I am writing to you all to talk you about a strange behaviour of the Intel HDA Audio device on an EEE PC 1101H. The audio device at some point simply doesn't react to commands, giving in dmesgo utput like this: [22321.638079] snd_hda_intel 0000:00:1b.0: azx_get_response timeout, switching to polling mode: last cmd=0x00170503 [22322.641410] snd_hda_intel 0000:00:1b.0: azx_get_response timeout, switching to single_cmd mode: last cmd=0x00170503 This may happen when I boot the system up fresh (here there is no Windows and no hibernation), or when the controller goes to power-save. If I don't set a power save timeout, then the controller will not be recoverable if it gets stuck at boot. As a workaround, I set a very short timeout, so that when the controller fails to start audio, waiting some time and retrying will work. Preliminary infos: Card: HDA Intel MID Chip: Realtek ALC269; Thank you in advance for your help. Please keep me in CC - since I am not subscribed to any list. Enrico Here follows a lspci 00:00.0 Host bridge: Intel Corporation System Controller Hub (SCH Poulsbo) (rev 07) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 00:02.0 VGA compatible controller: Intel Corporation System Controller Hub (SCH Poulsbo) Graphics Controller (rev 07) (prog-if 00 [VGA controller]) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 5 Region 0: Memory at f3f80000 (32-bit, non-prefetchable) [size=512K] Region 1: I/O ports at c880 [size=8] Region 2: Memory at d0000000 (32-bit, non-prefetchable) [size=256M] Region 3: Memory at f3f40000 (32-bit, non-prefetchable) [size=256K] Expansion ROM at <unassigned> [disabled] Capabilities: [d0] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [b0] Vendor Specific Information: Len=07 <?> Capabilities: [90] MSI: Enable- Count=1/1 Maskable- 64bit- Address: 00000000 Data: 0000 Kernel modules: gma500_gfx 00:1b.0 Audio device: Intel Corporation System Controller Hub (SCH Poulsbo) HD Audio Controller (rev 07) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 23 Region 0: Memory at f3f38000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend+ Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable- ID=0 ArbSelect=Fixed TC/VC=00 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=00 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=00 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000f0000000 Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation System Controller Hub (SCH Poulsbo) PCI Express Port 1 (rev 07) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 16 Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 I/O behind bridge: 0000e000-0000efff Memory behind bridge: fbf00000-fbffffff Prefetchable memory behind bridge: 80000000-801fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR+ NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #1, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <256ns, L1 <4us ClockPM+ Surprise- LLActRep+ BwNot- ASPMOptComp- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surprise+ Slot #0, PowerLimit 0.000W; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet+ LinkState+ RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna- CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [90] Subsystem: ASUSTeK Computer Inc. Device 83ce Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: pcieport Kernel modules: shpchp 00:1c.1 PCI bridge: Intel Corporation System Controller Hub (SCH Poulsbo) PCI Express Port 2 (rev 07) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin B routed to IRQ 17 Bus: primary=00, secondary=01, subordinate=02, sec-latency=0 I/O behind bridge: 0000d000-0000dfff Memory behind bridge: f4000000-fbefffff Prefetchable memory behind bridge: cc000000-cfffffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR+ NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #2, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <256ns, L1 <4us ClockPM+ Surprise- LLActRep+ BwNot- ASPMOptComp- LnkCtl: ASPM L1 Enabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surprise+ Slot #0, PowerLimit 0.000W; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet+ LinkState+ RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna- CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [90] Subsystem: ASUSTeK Computer Inc. Device 83ce Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: pcieport Kernel modules: shpchp 00:1d.0 USB controller: Intel Corporation System Controller Hub (SCH Poulsbo) USB UHCI #1 (rev 07) (prog-if 00 [UHCI]) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 20 Region 4: I/O ports at b880 [size=32] Kernel driver in use: uhci_hcd Kernel modules: uhci_hcd 00:1d.1 USB controller: Intel Corporation System Controller Hub (SCH Poulsbo) USB UHCI #2 (rev 07) (prog-if 00 [UHCI]) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 21 Region 4: I/O ports at c080 [size=32] Kernel driver in use: uhci_hcd Kernel modules: uhci_hcd 00:1d.2 USB controller: Intel Corporation System Controller Hub (SCH Poulsbo) USB UHCI #3 (rev 07) (prog-if 00 [UHCI]) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin C routed to IRQ 18 Region 4: I/O ports at c480 [size=32] Kernel driver in use: uhci_hcd Kernel modules: uhci_hcd 00:1d.7 USB controller: Intel Corporation System Controller Hub (SCH Poulsbo) USB EHCI #1 (rev 07) (prog-if 20 [EHCI]) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin D routed to IRQ 19 Region 0: Memory at f3f37c00 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Debug port: BAR=1 offset=00a0 Kernel driver in use: ehci-pci Kernel modules: ehci_pci 00:1f.0 ISA bridge: Intel Corporation System Controller Hub (SCH Poulsbo) LPC Bridge (rev 07) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Kernel driver in use: lpc_sch Kernel modules: lpc_sch 00:1f.1 IDE interface: Intel Corporation System Controller Hub (SCH Poulsbo) IDE Controller (rev 07) (prog-if 80 [Master]) Subsystem: ASUSTeK Computer Inc. Device 83ce Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Region 0: [virtual] Memory at 000001f0 (32-bit, non-prefetchable) [size=8] Region 1: [virtual] Memory at 000003f0 (type 3, non-prefetchable) Region 2: [virtual] Memory at 00000170 (32-bit, non-prefetchable) [size=8] Region 3: [virtual] Memory at 00000370 (type 3, non-prefetchable) Region 4: I/O ports at ffa0 [size=16] Kernel driver in use: pata_sch Kernel modules: pata_sch, pata_acpi, ata_generic 01:00.0 Network controller: Qualcomm Atheros AR9285 Wireless Network Adapter (PCI-Express) (rev 01) Subsystem: AzureWave AW-NE785 / AW-NE785H 802.11bgn Wireless Full or Half-size Mini PCIe Card Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at fbef0000 (64-bit, non-prefetchable) [size=64K] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1+ D2- AuxCurrent=375mA PME(D0+,D1+,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] MSI: Enable- Count=1/1 Maskable- 64bit- Address: 00000000 Data: 0pcilib: sysfs_read_vpd: read failed: Connection timed out 000 Capabilities: [60] Express (v2) Legacy Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <512ns, L1 <64us ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr+ UncorrErr- FatalErr- UnsuppReq+ AuxPwr+ TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s <512ns, L1 <64us ClockPM- Surprise- LLActRep- BwNot- ASPMOptComp- LnkCtl: ASPM L1 Enabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Not Supported, TimeoutDis+, LTR-, OBFF Not Supported DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis-, LTR-, OBFF Disabled LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis- Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -6dB, EqualizationComplete-, EqualizationPhase1- EqualizationPhase2-, EqualizationPhase3-, LinkEqualizationRequest- Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES+ TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 00, GenCap+ CGenEn- ChkCap+ ChkEn- Capabilities: [140 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- Capabilities: [160 v1] Device Serial Number 00-00-00-00-00-00-00-00 Capabilities: [170 v1] Power Budgeting <?> Kernel driver in use: ath9k Kernel modules: ath9k 03:00.0 Ethernet controller: Qualcomm Atheros AR8132 Fast Ethernet (rev c0) Subsystem: ASUSTeK Computer Inc. Device 14e5 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 24 Region 0: Memory at fbfc0000 (64-bit, non-prefetchable) [size=256K] Region 2: I/O ports at e880 [size=128] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1+,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [48] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4122 Capabilities: [58] Express (v1) Endpoint, MSI 00 DevCap: MaxPayload 4096 bytes, PhantFunc 0, Latency L0s <4us, L1 unlimited ExtTag- AttnBtn+ AttnInd+ PwrInd+ RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Exit Latency L0s unlimited, L1 unlimited ClockPM+ Surprise- LLActRep- BwNot- ASPMOptComp- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [6c] Vital Product Data Not readable Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq+ ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP- SDES+ TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr- CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 14, GenCap+ CGenEn- ChkCap+ ChkEn- Capabilities: [180 v1] Device Serial Number ff-25-f6-3a-90-e6-ba-ff Kernel driver in use: atl1c Kernel modules: atl1c

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[alsa-devel] ASoC: fsl_sai: Has anybody attempted to run the i.MX6UL in TDM mode with channels > 2?
by Caleb Crome 19 Jan '17

19 Jan '17

Hi all, Now that the MX6 fsl_ssi seems to be getting stable for TDM with channels > 2 (thanks all!), the MX6 UL and MX7 have only the SAI port. Does anybody have experience with running the SAI with more than 2 channels? Any thoughts or comments would be welcome! Thanks, -Caleb

3 4

[alsa-devel] Alsamixer: missing Speakers entry in analog output choice (chipset AV200)
by Germano Massullo 08 Jan '17

08 Jan '17

I have a Asus Xonar Essence ST soundboard (Chip: AV200) on Fedora 23 kernel 4.4.5-300.fc23.x86_64 The audiocard has 3 different outputs - Speakers - Headphones - FP Headphones that have mutual exclusion, so that when you are using one of them, you cannot use the others. Sometimes the Speakers output disappears from alsamixer. By restarting the computer, the system gets back to its normal behaviour. I attach 3 logs taken when the problem was occurring. The problem happened also in past years, but in last a 1-2 years it happened less frequently

3 3

[alsa-devel] [PATCH - alsa-lib 1/1] Change card/pid get functions to return -ENOSYS if the kernel is too old
by Adam Goode 03 Jan '17

03 Jan '17

When trying to get the pid or card of a client, the get functions will return -1 if there is no pid or card for a client. Clients have zero or one of these, so -1 is correct for these cases. But we also need to detect the case where the kernel cannot tell us if there is a card or pid, so that userspace can fallback to probing this information in the old way. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/src/seq/seq.c b/src/seq/seq.c index 4405e68..7de1e81 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,9 +1522,10 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** - * \brief Get the sound card number. + * \brief Get the sound card number if the kernel supports this. * \param info client_info container - * \return card number or -1 if value is not available. + * \return card number, -1 if there is no card for this client, + * or \c -ENOSYS if the kernel does not have support for this operation * * Only available for SND_SEQ_KERNEL_CLIENT clients. */ @@ -1535,9 +1536,10 @@ int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) } /** - * \brief Get the owning PID. + * \brief Get the owning PID if the kernel supports this. * \param info client_info container - * \return pid or -1 if value is not available. + * \return pid, -1 if there is no PID for this client, + * or \c -ENOSYS if the kernel does not have support for this operation * * Only available for SND_SEQ_USER_CLIENT clients. */ diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index 578ef12..a1d1e4a 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -102,8 +102,8 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in return -errno; } if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { - info->card = -1; - info->pid = -1; + info->card = -ENOSYS; + info->pid = -ENOSYS; } return 0; } @@ -374,8 +374,8 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i return -errno; } if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { - info->card = -1; - info->pid = -1; + info->card = -ENOSYS; + info->pid = -ENOSYS; } return 0; } -- 2.8.0.rc3.226.g39d4020

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[alsa-devel] [alsa-utils] alsaucm should come with a man page
by Antonio Ospite 02 Dec '16

02 Dec '16

Hi, I used alsaucm to test mixer use cases for an Intel Baytrail tablet, I wanted to use bare alsa before bringing pulseaudio in, and it took some time to get my head around it. The output from the "alsaucm --help" is too terse, I had to look at the alsa-lib code[1] to see the possible IDENTIFIERs, and even then it was not clear to me that some operations, like for instance "list _devices", can only work after setting a _verb in the same _execution_, for instance this sequence doesn't work: # alsaucm -c bytcr-rt5640 set _verb HiFi # alsaucm -c bytcr-rt5640 list _devices But this command does: # alsaucm -c bytcr-rt5640 -b - <<EOM set _verb HiFi list _devces EOM Now that I know something more it makes sense: UCM just operates on mixer settings, it does not have its own external state. However, my point is that a man page explaining these things can make the life easier to new users. It would be better if alsa/UCM developers wrote the man page, but if no one steps up I guess I could draft a first version myself. Thanks, Antonio [1] http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=include/use-case.h -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

3 9

[alsa-devel] [PATCH v5 1/2] ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem
by Sylwester Nawrocki 01 Dec '16

01 Dec '16

This patch adds DT binding documentation for Exnos5433 based TM2 and TM2E boards sound subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v4: - indentation changes. Changes since v2: - none. Changes since initial version: - dropped clocks, clock-names properties, instead properties from the CODEC node will be used, - property renames: 'samsung,model' -> 'model', 'samsung,i2s-controller' -> 'i2s-controller', 'samsung,speaker-amplifier' -> 'audio-amplifier', - added 'audio-codec' property. --- .../bindings/sound/samsung,tm2-audio.txt | 38 ++++++++++++++++++++++ 1 file changed, 38 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt new file mode 100644 index 0000000..94442e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt @@ -0,0 +1,38 @@ +Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +Required properties: + + - compatible : "samsung,tm2-audio" + - model : the user-visible name of this sound complex + - audio-codec : the phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt + - i2s-controller : the phandle of the I2S controller + - audio-amplifier : the phandle of the MAX98504 amplifier + - samsung,audio-routing : a list of the connections between audio components; + each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source; valid names for sources and sinks are the + WM5110's and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, + Headset Mic + - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator + + +Example: + +sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>; + i2s-controller = <&i2s0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 0>; + model = "wm5110"; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; +}; -- 1.9.1

4 8

[alsa-devel] [PATCH v2] clkdev: add devm_of_clk_get()
by Kuninori Morimoto 01 Dec '16

01 Dec '16

This is based on devm_clk_get() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - added "static" and "inline" on header drivers/clk/clkdev.c | 26 ++++++++++++++++++++++++++ include/linux/clk.h | 7 +++++++ 2 files changed, 33 insertions(+) diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 89cc700..93a613b 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -55,6 +55,32 @@ struct clk *of_clk_get(struct device_node *np, int index) } EXPORT_SYMBOL(of_clk_get); +static void devm_of_clk_release(struct device *dev, void *res) +{ + clk_put(*(struct clk **)res); +} + +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + struct clk **ptr, *clk; + + ptr = devres_alloc(devm_of_clk_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return ERR_PTR(-ENOMEM); + + clk = of_clk_get(np, index); + if (!IS_ERR(clk)) { + *ptr = clk; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return clk; +} +EXPORT_SYMBOL(devm_of_clk_get); + static struct clk *__of_clk_get_by_name(struct device_node *np, const char *dev_id, const char *name) diff --git a/include/linux/clk.h b/include/linux/clk.h index a89ba4e..33cd540 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -502,6 +502,8 @@ struct of_phandle_args; #if defined(CONFIG_OF) && defined(CONFIG_COMMON_CLK) struct clk *of_clk_get(struct device_node *np, int index); +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index); struct clk *of_clk_get_by_name(struct device_node *np, const char *name); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec); #else @@ -509,6 +511,11 @@ static inline struct clk *of_clk_get(struct device_node *np, int index) { return ERR_PTR(-ENOENT); } +static inline struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + return ERR_PTR(-ENOENT); +} static inline struct clk *of_clk_get_by_name(struct device_node *np, const char *name) { -- 1.9.1

4 15

[alsa-devel] [RFC 00/17] clk: Add per-controller locks to fix deadlocks
by Krzysztof Kozlowski 12 Nov '16

12 Nov '16

Hi, RFC, please, do not apply, maybe except patch #1 which is harmless. Introduction ============ The patchset brings new entity: clock controller representing a hardware block. The clock controller comes with its own prepare lock which is used then in many places. The idea is to fix the deadlock mentioned in commit 10ff4c5239a1 ("i2c: exynos5: Fix possible ABBA deadlock by keeping I2C clock prepared") and commit 34e81ad5f0b6 ("i2c: s3c2410: fix ABBA deadlock by keeping clock prepared"). Disclaimer ========== Request for comments, so: 1. Only exynos_defconfig builds, 2. A lot of FIXME/TODO note still, 3. Checkpatch not run, lines not aligned, 4. Other (non-exynos) drivers not converted, 5. Probably not yet bisectable, 6. Locking became quite complex. The previous one lock was simple. Inefficient and dead-lock prone but simple. Because of clock hierarchy spanning through controllers, the new locking became quite complicated. I don't like it but... Details ======= In Exynos-based boards case the deadlock occurs between clock's prepare_lock and regmap-i2c's lock: CPU #0: CPU #1: lock(regmap) s2mps11-clk: clk_prepare_lock() i2c-exynos: clk_prepare_lock() - wait lock(regmap) - wait The clk_prepare_lock() on both CPUs come from different clock drivers and components: 1. I2C clock is part of SoC block and is required by i2c-s3c2410/i2c-exynos5 driver, 2. S2MPS11 clock is separate device, however driver uses I2C regmap. The deadlock was reported by lockdep (always) and was happening in 20% of boots of Odroid XU3 with multi_v7 defconfig. Workaround for deadlock was implemented by removing prepare/unprepare calls from I2C transfers. However these are just workarounds... which after applying this patch can be reverted. Additionally Marek Szyprowski's work on domains/clocks/pinctrl exposed the deadlock again in different configuration. Comments as usual are welcomed. Best regards, Krzysztof Krzysztof Kozlowski (17): clk: bcm2835: Rename clk_register to avoid name conflict clk: Add clock controller to fine-grain the prepare lock clk: s2mps11: Switch to new clock controller API clk: samsung: Allocate a clock controller in context clk: fixed-rate: Switch to new clock controller API clk: gate: Switch to new clock controller API clk: mux: Switch to new clock controller API clk: fixed-factor: Switch to new clock controller API clk: divider: Switch to new clock controller API clk: composite: Switch to new clock controller API clk: gpio: Switch to new clock controller API ASoC: samsung: Switch to new clock controller API clk: samsung: audss: samsung: Switch to new clock controller API clk: samsung: clkout: samsung: Switch to new clock controller API clk: Use per-controller locking Revert "i2c: exynos5: Fix possible ABBA deadlock by keeping I2C clock prepared" Revert "i2c: s3c2410: fix ABBA deadlock by keeping clock prepared" drivers/clk/bcm/clk-bcm2835.c | 8 +- drivers/clk/clk-composite.c | 8 +- drivers/clk/clk-divider.c | 10 +- drivers/clk/clk-fixed-factor.c | 11 +- drivers/clk/clk-fixed-rate.c | 28 +- drivers/clk/clk-fractional-divider.c | 5 +- drivers/clk/clk-gate.c | 8 +- drivers/clk/clk-gpio.c | 29 +- drivers/clk/clk-mux.c | 32 ++- drivers/clk/clk-s2mps11.c | 10 +- drivers/clk/clk.c | 456 +++++++++++++++++++++++++++----- drivers/clk/samsung/clk-exynos-audss.c | 30 ++- drivers/clk/samsung/clk-exynos-clkout.c | 11 +- drivers/clk/samsung/clk.c | 25 +- drivers/clk/samsung/clk.h | 1 + drivers/i2c/busses/i2c-exynos5.c | 24 +- drivers/i2c/busses/i2c-s3c2410.c | 23 +- include/linux/clk-provider.h | 88 ++++-- include/linux/clk.h | 1 + sound/soc/samsung/i2s.c | 13 +- 20 files changed, 612 insertions(+), 209 deletions(-) -- 1.9.1

7 27

[alsa-devel] [PATCH v4 0/3] ASoC/ARM: tegra: apalis t30/tk1/colibri t30: sgtl5000 audio
by Marcel Ziswiler 07 Nov '16

07 Nov '16

This series adds/integrates Freescale SGTL5000 analogue audio codec support. Changes in v4: - simple-audio-card does still not allow for more advanced use cases like Tegra SoCs - further platform drivers have been accepted since my last attempt (e.g. rt5677 one) - relevance for one further board the new Toradex Apalis TK1 - drop unused sound/jack.h include - in tegra_sgtl5000_driver_remove() pass return value of snd_soc_unregister_card() on to caller - no longer set owner property in platform_driver to THIS_MODULE - re-based/resend Changes in v3: - revert to not using simple-audio-card being incompatible with tegra - rebased to for-next Changes in v2: - using simple-audio-card as suggested by Fabio Marcel Ziswiler (3): ASoC: tegra: add tegra sgtl5000 machine driver ARM: tegra: apalis/colibri t30: integrate audio ARM: tegra: enable sgtl5000 audio .../bindings/sound/nvidia,tegra-audio-sgtl5000.txt | 42 ++++ arch/arm/boot/dts/tegra30-apalis.dtsi | 49 +++++ arch/arm/boot/dts/tegra30-colibri.dtsi | 49 +++++ arch/arm/configs/tegra_defconfig | 1 + sound/soc/tegra/Kconfig | 11 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_sgtl5000.c | 212 +++++++++++++++++++++ 7 files changed, 366 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt create mode 100644 sound/soc/tegra/tegra_sgtl5000.c -- 2.5.5

4 5

Re: [alsa-devel] Intel Cherry Trail platform sound issue
by Zhihe Tu 05 Nov '16

05 Nov '16

Can anyone give some suggestions? Thanks Tu 发件人: 涂治河 [mailto:tuzhihe@pcasl.com] 发送时间: 2015年11月5日 19:37 收件人: 'alsa-devel(a)alsa-project.org' 主题: Intel Cherry Trail platform sound issue Hi All, I get cherry trail tablet, and install ubuntu14.04.3, Kernel verison is 3.19. The table using rt5672 codec with no sound output in ubuntu, but it works in win8. I get intel sst firmware fw_sst_22a8.bin from https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/tr… The driver load the firmware successfully and I change some alsa mixer items: amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on amixer -c0 sset 'media0_out mix 0 media1_in' on amixer -c0 sset 'media1_in Gain 0' 80% amixer -c0 sset 'media1_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'media1_in Gain 0' off amixer -c0 sset 'pcm0_in Gain 0' 80% amixer -c0 sset 'pcm0_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'pcm0_in Gain 0' off amixer -c0 sset 'codec_out0 Gain 0' 80% amixer -c0 sset 'codec_out0 Gain 0 Ramp Delay' 50 amixer -c0 sset 'codec_out0 Gain 0' off But there are some errors with dmesg: [ 16.359646] Audio Port: ASoC: no backend DAIs enabled for Audio Port … [ 16.382156] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 16.383984] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 16.541809] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 17.545512] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 17.555207] intel_sst_acpi 808622A8:00: fw returned err -16 [ 18.605859] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 18.615908] intel_sst_acpi 808622A8:00: fw returned err -16 [ 19.621943] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 19.631499] intel_sst_acpi 808622A8:00: fw returned err -16 [ 20.637527] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 20.647083] intel_sst_acpi 808622A8:00: fw returned err -16 [ 20.682979] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 21.681280] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 21.690836] intel_sst_acpi 808622A8:00: fw returned err -16 [ 21.697087] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 22.757082] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 22.766640] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.772867] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 23.782432] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.857938] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 23.989254] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 23.989491] intel_sst_acpi 808622A8:00: FW sent error response 0x40034 [ 23.996864] intel_sst_acpi 808622A8:00: FW alloc failed ret -4 [ 24.003399] intel_sst_acpi 808622A8:00: sst_get_stream returned err -5 [ 24.010737] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 24.018652] Audio Port: ASoC: prepare FE Audio Port failed [ 25.024622] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 25.034180] intel_sst_acpi 808622A8:00: fw returned err -16 [ 26.040399] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 26.049955] intel_sst_acpi 808622A8:00: fw returned err -16 [ 27.056333] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 27.065890] intel_sst_acpi 808622A8:00: fw returned err -16 [ 28.072032] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 28.081591] intel_sst_acpi 808622A8:00: fw returned err -16 [ 28.117427] Audio Port: sst: error code = -22 [ 28.122401] snd_pcm_update_hw_ptr0: 120 callbacks suppressed [ 28.122409] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.132784] Audio Port: sst: error code = -22 [ 28.137749] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.372967] Audio Port: sst: error code = -22 [ 28.377946] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.388317] Audio Port: sst: error code = -22 [ 28.393289] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.403687] Audio Port: sst: error code = -22 lsmod info: snd_soc_sst_cht_bsw_rt5672 16384 0 snd_intel_sst_acpi 16384 1 snd_intel_sst_core 73728 1 snd_intel_sst_acpi snd_soc_sst_mfld_platform 77824 3 snd_intel_sst_core snd_soc_rt5670 106496 1 snd_soc_rl6231 16384 1 snd_soc_rt5670 snd_soc_core 196608 4 snd_soc_rt5670,snd_soc_sst_cht_bsw_rt5672,snd_soc_sst_mfld_platform snd_seq_midi 16384 0 snd_seq_midi_event 16384 1 snd_seq_midi snd_compress 20480 1 snd_soc_core snd_pcm_dmaengine 16384 1 snd_soc_core snd_rawmidi 32768 1 snd_seq_midi snd_pcm 106496 5 snd_soc_rt5670,snd_soc_core,snd_soc_sst_cht_bsw_rt5672,snd_soc_sst_mfld_platform,snd_pcm_dmaengine snd_seq 65536 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 16384 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 32768 2 snd_pcm,snd_seq snd 86016 10 snd_soc_core,snd_timer,snd_pcm,snd_seq,snd_rawmidi,snd_soc_sst_mfld_platform,snd_seq_device,snd_compress soundcore 16384 1 snd I have no idea with the next step, so can anyone give a look? Thanks Tu __________ Information from ESET Smart Security, version of virus signature database 4468 (20090929) __________ The message was checked by ESET Smart Security. http://www.eset.com __________ Information from ESET Smart Security, version of virus signature database 4468 (20090929) __________ The message was checked by ESET Smart Security. http://www.eset.com

2 1

[alsa-devel] [PATCH v1 1/2] sound: lpass-cpu: add module licence and description
by Srinivas Kandagatla 31 Oct '16

31 Oct '16

This patch adds module licence to lpass-cpu driver, without this patch lpass-cpu module would taint with below error, while insmod/modprobe. snd_soc_lpass_cpu: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint snd_soc_lpass_cpu: Unknown symbol regmap_write (err 0) snd_soc_lpass_cpu: Unknown symbol devm_kmalloc (err 0) ... Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 3cde9fb..eff3f9a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -586,3 +586,6 @@ int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_platform_remove); + +MODULE_DESCRIPTION("QTi LPASS CPU Driver"); +MODULE_LICENSE("GPL v2"); -- 2.8.3

3 3

[alsa-devel] [PATCH 0/13] ALSA: Deletion of some unnecessary checks
by SF Markus Elfring 26 Oct '16

26 Oct '16

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 4 Jan 2015 11:50:12 +0100 Further update suggestions were taken into account after several patches were applied from static source code analysis. Markus Elfring (3): seq: Deletion of unnecessary checks before the function call "snd_midi_event_free" oss: Deletion of unnecessary checks before the function call "vfree" emu10k1: Delete an unnecessary check before the function call "kfree" oxygen: Delete an unnecessary check before the function call "snd_pcm_suspend" emux: Delete an unnecessary check before the function call "snd_sf_free" ASoC: Intel: Delete an unnecessary check before the function call "sst_dma_free" ASoC: fsi: Deletion of unnecessary checks before the function call "clk_enable" ASoC: Intel: Delete an unnecessary check before the function call "release_firmware" i2sbus: Delete an unnecessary check before the function call "snd_pcm_suspend_all" sb: Delete an unnecessary check before the function call "snd_emux_free" Deletion of checks before the function call "iounmap" msnd: One function call less in snd_msnd_attach() after error detection msnd: Fix centralized exiting from snd_msnd_attach() sound/aoa/soundbus/i2sbus/core.c | 13 ++++++------- sound/arm/aaci.c | 4 ++-- sound/drivers/ml403-ac97cr.c | 3 +-- sound/isa/msnd/msnd_pinnacle.c | 31 ++++++++++++++++--------------- sound/parisc/harmony.c | 4 +--- sound/pci/ad1889.c | 5 +---- sound/pci/asihpi/hpioctl.c | 6 ++---- sound/pci/atiixp.c | 3 +-- sound/pci/atiixp_modem.c | 3 +-- sound/pci/aw2/aw2-alsa.c | 4 +--- sound/pci/bt87x.c | 3 +-- sound/pci/cs4281.c | 6 ++---- sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/ctxfi/cthw20k1.c | 5 +---- sound/pci/ctxfi/cthw20k2.c | 5 +---- sound/pci/echoaudio/echoaudio.c | 6 +----- sound/pci/hda/hda_intel.c | 3 +-- sound/pci/lola/lola.c | 6 ++---- sound/pci/mixart/mixart.c | 7 +++---- sound/pci/nm256/nm256.c | 6 ++---- sound/pci/rme9652/hdsp.c | 4 +--- sound/pci/rme9652/hdspm.c | 4 +--- sound/pci/rme9652/rme9652.c | 3 +-- sound/pci/sis7019.c | 5 +---- sound/pci/ymfpci/ymfpci_main.c | 3 +-- sound/ppc/pmac.c | 15 +++++---------- 26 files changed, 58 insertions(+), 103 deletions(-) -- 2.2.1

5 25

[alsa-devel] [PATCH 0/8] ASoC: s3c24xx_uda134x card fixes and cleanups
by Sylwester Nawrocki 25 Oct '16

25 Oct '16

This series might have a bit of an archaeological aspect, nevertheless my goal was to have sound working on one of the older S3C24XX SoC based boards for testing any patches touching the older Samsung SoC IP block related code. Together with patches: ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card ARM: S3C24XX: Add dma_mask assignments for DMA devices ARM: S3C24XX: Add missing DMA device for Mini2440 board this series fixes sound on Micro2440SDK/mini2440 board broken since the 3.12 kernel release. I'm not sure how it worked before but currently the uda134x codec always fails to probe due to uninitialized platform_data, which is supposed to be set by the ASoC machine driver. This series addresses the issue by moving all required codec platform_data to the board file and exposing optional GPIO bitbang L3 bus ops which are selected by the codec. ASoC dependency patches: ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card The patch "ARM: S3C24XX: Specify audio codec platform_data for mini2440 board" is for arm-soc but it would be good to merge whole series through one tree to avoid compilation breaks due to unmet dependencies. Sylwester Nawrocki (8): ASoC: L3 bus: Add default gpio ops ARM: S3C24XX: Specify audio codec platform_data for mini2440 board ASoC: uda134x: Optionally initialize L3 ops to default GPIO ops ASoC: s3c24xx_uda134x: Remove unused power() callback ASoC: s3c24xx_uda134x: Drop initialization of codec's platform data ASoC: samsung: Convert s3c24xx_uda134x to use devm_snd_soc_register_card() ASoC: samsung: s3c24xx_uda134x: debug/error trace cleanup ASoC: s3c244_uda134x: Allocate private data dynamically arch/arm/mach-s3c24xx/mach-mini2440.c | 19 ++++ include/sound/l3.h | 15 ++- include/sound/s3c24xx_uda134x.h | 1 - sound/soc/codecs/l3.c | 71 ++++++++++-- sound/soc/codecs/uda134x.c | 7 ++ sound/soc/samsung/s3c24xx_uda134x.c | 205 ++++++++++------------------------ 6 files changed, 158 insertions(+), 160 deletions(-) -- 1.9.1

2 13

[alsa-devel] Potential Audio miniconference at LPC/KS
by Mark Brown 24 Oct '16

24 Oct '16

Hi, As in previous years we're trying to organize an audio miniconference so we can get together and talk through issues, especially design decisons, face to face. This year we're trying a slightly different tack and trying to organize this as part of the workshops day at Kernel Summit and Linux Plumbers in Santa Fe the week 31st October to 4th November (I'm writing this offline so don't have the exact date for the workshop day, sorry). There is a wiki page with an initial list of proposed topics here: http://wiki.linuxplumbersconf.org/2016:workshops In order to get this workshop approved we're going to need both a clear set of things to discuss and visible interest from attendees so if people could both follow up here and add to the wiki with suggestions of topics that'd be especially appreciated. Thanks, Mark

4 14

[alsa-devel] [PATCH 0/2 v2][resend] DesignWare HDMI I2S suport
by Kuninori Morimoto 17 Oct '16

17 Oct '16

Hi Archit, Mark Cc Thierry, Russell These are resend of v2 of DesignWare HDMI I2S support patches. (I added drm-bridge maintainer Archit on To:) It will use ALSA SoC hdmi-codec driver, but we can't use it as-is at this point. 1) patch tidyup hdmi-codec driver to enable dw-hdmi I2S support. 2) patch is based on it. Difficult is that these 2 patches are under different maintainer. 1) Mark ? 2) Archit ? Kuninori Morimoto (2): 1) ASoC: hdmi-codec: enable multi probe for same device 2) drm: bridge: add DesignWare HDMI I2S audio support drivers/gpu/drm/bridge/Kconfig | 8 ++++++++ drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 +++++++ drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 130 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 ++++++++++++++++++-- drivers/gpu/drm/bridge/dw-hdmi.h | 21 +++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++--- 7 files changed, 251 insertions(+), 5 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: topology: Reenable use from userspace
by Mark Brown 11 Oct '16

11 Oct '16

We had inserted a #error into the topology UAPI code to ensure that the ABI was not adopted by userspace while final review and testing was ongoing. Since we are now comfortable that the ABI is suitable for production remove the #error, future changes will include provision for backwards compatibility. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Are we OK with this? Looking for some acks and reviews... include/uapi/sound/asoc.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 247c50b..51b8066 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -18,12 +18,6 @@ #include <linux/types.h> #include <sound/asound.h> -#ifndef __KERNEL__ -#error This API is an early revision and not enabled in the current -#error kernel release, it will be enabled in a future kernel version -#error with incompatible changes to what is here. -#endif - /* * Maximum number of channels topology kcontrol can represent. */ -- 2.6.1

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[alsa-devel] [PATCH v2] ASoC: cs4270: fix DAPM stream name mismatch
by murray foster 07 Oct '16

07 Oct '16

Mismatching stream names in DAPM route and widget definitions are causing compilation errors. Fixing these names allows the cs4270 driver to compile and function. Signed-off-by: Murray Foster <mrafoster(a)gmail.com> --- Changes in v2: - Remove the 's' typo at the beginning of the codec driver. sound/soc/codecs/cs4270.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e07807d..3670086 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -148,11 +148,11 @@ SND_SOC_DAPM_OUTPUT("AOUTR"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { - { "Capture", NULL, "AINA" }, - { "Capture", NULL, "AINB" }, + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, - { "AOUTA", NULL, "Playback" }, - { "AOUTB", NULL, "Playback" }, + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, }; /** -- 2.4.9

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[alsa-devel] [PATCH 0/5] ALSA: soc: fix sparse warnings
by Takashi Sakamoto 28 Sep '16

28 Sep '16

Hi, This patchset is to fix some sparse warnings in ALSA SoC part. Below items are going to be fixed: * fix a condition statement with a logical operation * purge old way to initialize array * missing 'static' qualifier for file local symbols Although I got many warnings from Intel SST drivers due to confusions of address spaces (__iomem and __kernel), I still left them as they are, because they require more codes to fix. I hope Intel developers will do additional work for them... Takashi Sakamoto (5): ALSA: soc-da7219: fix inappropriate condition statement ALSA: soc-sst-bxt-rt298: fix obsoleted initializers for array ALSA: soc-sst-bxt-da7219_max98357a: fix obsoleted initializers for array ALSA: soc-rt5616: add static qualifier for file local symbols ALSA: soc-tpa6130a2: add static qualifier for file local symbols sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/rt5616.c | 4 ++-- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 ++++++------ sound/soc/intel/boards/bxt_rt298.c | 14 +++++++------- 5 files changed, 17 insertions(+), 17 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/9] ASoC: Intel: Skylake: Driver updates
by Vinod Koul 25 Sep '16

25 Sep '16

So we had some driver updates collated, time to send them upstream :) - manifest data size was not checked so that is fixed - previous conversion to use core parsing for HDA capabilities was wrong at one place, that is fixed. - DMA config size was wrong, that is also fixed - Add 32bit format - The module instance IDs are manged in driver now. Dharageswari R (5): ASoC: Intel: Skylake: Unload all the loadable modules ASoC: Intel: Skylake: Create dynamic instance ids for DSP modules ASoC: Intel: Skylake: Use private instance id of modules in IPC ASoC: Intel: Skylake: Table for module instance id and private id ASoC: Intel: Skylake: Override the actual instance id's to pvt_id's Jeeja KP (1): ASoC: Intel: Skylake: Fix DMA control config size Samaga Krishna (1): ASoC: Intel: Skylake: Add 32bit support Vinod Koul (2): ASoC: Intel: Skylake: check manifest size ASoC: Intel: Skylake: Fix the inverted logic check sound/soc/intel/skylake/skl-messages.c | 24 +++-- sound/soc/intel/skylake/skl-pcm.c | 5 +- sound/soc/intel/skylake/skl-sst-dsp.h | 6 ++ sound/soc/intel/skylake/skl-sst-utils.c | 133 ++++++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.c | 44 ++++++++- sound/soc/intel/skylake/skl-topology.h | 13 ++- sound/soc/intel/skylake/skl-tplg-interface.h | 3 +- 7 files changed, 208 insertions(+), 20 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: Chromebook Pixel 2015 support
by John Keeping 25 Sep '16

25 Sep '16

These patches add basic support for audio on the Chromebook Pixel 2015. I took the machine driver from ChromeOS commit v3.14-10186-g7e7d95df2983 and other than removing support for wake on voice (which is not supported by the mainline rt5677 codec driver), the only differences are to cope with API changes since 3.14. Only two authors have committed changes to the machine driver in the ChromeOS tree and all of the commits are signed off; both authors' SoB lines are preserved in the second commit here. The codec driver is sufficiently different that I just added ACPI support from scratch using the ChromeOS source as a reference for the property names. John Keeping (2): ASoC: rt5677: Add ACPI support ASoC: Intel: boards: Add bdw-rt5677 machine driver sound/soc/codecs/rt5677.c | 109 +++++++++--- sound/soc/intel/Kconfig | 11 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5677.c | 341 ++++++++++++++++++++++++++++++++++++ sound/soc/intel/common/sst-acpi.c | 1 + 5 files changed, 443 insertions(+), 21 deletions(-) create mode 100644 sound/soc/intel/boards/bdw-rt5677.c -- 2.9.3.728.g30b24b4.dirty

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[alsa-devel] [snd_ali5451] snd_ali5451 0000:00:06.0: BUG: , pos = 16384, buffer size = 16384, period size = 1024
by Enrico Mioso 21 Sep '16

21 Sep '16

Hello guys. The message on the subject comes from my dmesg and is generated by the said module when using mplayer. (MPlayer SVN-r37379 (C) 2000-2015 MPlayer Team) Including some devices from my lspci: if you wish to see the integral one, let me know. 00:06.0 Multimedia audio controller: ULi Electronics Inc. M5451 PCI AC-Link Controller Audio Device (rev 02) Subsystem: Hewlett-Packard Company Device 002a Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR+ <PERR+ INTx- Latency: 64 (500ns min, 6000ns max) Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at 1000 [size=256] Region 1: Memory at d0001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [dc] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: snd_ali5451 00:07.0 ISA bridge: ULi Electronics Inc. M1533/M1535/M1543 PCI to ISA Bridge [Aladdin IV/V/V+] Subsystem: ULi Electronics Inc. ALi M1533 Aladdin IV/V ISA Bridge Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Capabilities: [a0] Power Management version 1 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- 00:08.0 Modem: ULi Electronics Inc. M5457 AC'97 Modem Controller (prog-if 00 [Generic]) Subsystem: Hewlett-Packard Company Device 002a Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin A routed to IRQ 10 Region 0: Memory at d0002000 (32-bit, non-prefetchable) [size=4K] Region 1: I/O ports at 1400 [size=256] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Thank you. P.S.: sorry for giving no follow-up to my previous hd-audio thing; thanks.

2 4

[alsa-devel] Dell XPS One 2710 HDA Intel PCH - No S/PDIF Output
by Marve 20 Sep '16

20 Sep '16

Hi all, I've been trying stuff myself and got help from someone (debianuser) at #alsa IRC, but without success! I got the advice to explain the issue in this mailing list! I'm running Linux Mint 18 on a all-in-one Dell XPS One 2710. Most stuff works, except sound. The former Mint version also sound from the speakers worked out of the box. Now, I have to switch from the Pulse Audio volume control from "Speakers" to "Headphones (unplugged)" every time I start up to get audio from the internal speakers, which is working well then, annoying, is probably an issue that belongs in another group and not the reason I'm writing you! :) The XPS One 2710 has a S/PDIF Toslink (optical) connection. Back in the days that Microsoft wasn't trying to shove up Windows 10 to everyone, I successfully could get output from this connection to my JVC Receiver from Windows7 of the box. I find it hard to have it configured under Linux. Partly, with help from a anonymous debianuser on #alsa IRC I checked with alsamixer that the S/PDIF was not muted. Also, with the Jack Retasking for HDA tool, I enforced Pin ID: 0x1e is a SPDIF Out connection. After this, the Pulse Audio Tools shows several new interesting Built-in Audio options and I think "Digital Stereo (IEC958) Outout looks most promising. So I select this one and with Output Devices I select "Digital Output (S/PDIF). Then, when I play something on the box, the volume meters actually show that there is something playing, but I can't hear anything from the JVC receiver! The receiver supports PCM to DTS, but no sound is coming out of it! Thanks if you could help me out! I am happy to help you when you need more information or want we to test something. I had this problem on the previous Linux Mint LTS version and tried to install a newer kernel and alsa driver, which didn't work - so I decided to wait for the current LTS. I made a fresh info: http://www.alsa-project.org/db/?f=556ae2bdae66341579831231ef2d90ca0d7c8e24 If you need one with other settings, please let me know! Best wishes, M.

1 3

[alsa-devel] [PATCH 00/15] Line6 POD X3/X3Live suport
by Andrej Krutak 20 Sep '16

20 Sep '16

Hello all, attached is the first version of the driver, based on POD HD. They share a lot of similarities, but at the same time there were some generalizations needed. I think POD HD could follow up on these patches to add hwdep support, perhaps helix too... I tried to not introduce regressions to the old HW. Anyhow, this is probably not the final version of the patches, as it is my first submission to upstream - I'm sure there will be some issues. Especially the patch "Use device_create_file instead of snd_card_add_dev_attr" I have to revisit, it may not be necessarry in the end. Other than that, the patches mostly pass `make checkpatches`. I've been testing the driver for a while now, including lock debugging options etc., and there don't seem to be functional problems. There's one missing thing - the driver uses bulk USB interface of the device, but so far I wasn't able to make the usb_driver_claim_interface() work... I hope this can be added later, if someone (or I) finds time. Thanks for your inputs, greetings, Andrej Krutak (15): ALSA: line6: Make driver configuration more generic. ALSA: line6: Add LINE6_CAP_IN_NEEDS_OUT, a void playback stream during capture ALSA: line6: Distinguish device init (ctrl EP) and MIDI data transfer (int EP) ALSA: line6: Add support for POD X3 ALSA: line6: Use device_create_file instead of snd_card_add_dev_attr ALSA: line6: Allow bulk endpoints instead of interrupt endpoints ALSA: line6: Allow processing of raw incoming messages ALSA: line6: Cleanup initialization ALSA: line6: Add hwdep interface to access the POD control messages ALSA: line6: Add proper locks for hwdep open/release/read ALSA: line6: Only free buffer if it is set. ALSA: line6: Give up on the lock while URBs are released. ALSA: line6: Add support for POD X3 Live (only USB ID differs from POD X3) ALSA: line6: Give up hwdep spinlock temporarily during read operation ALSA: line6: Remove double line6_pcm_release() after failed acquire. include/uapi/sound/asound.h | 3 +- sound/usb/line6/Kconfig | 4 +- sound/usb/line6/capture.c | 49 +++++-- sound/usb/line6/driver.c | 303 ++++++++++++++++++++++++++++++++++------ sound/usb/line6/driver.h | 71 +++++++--- sound/usb/line6/midi.c | 2 +- sound/usb/line6/pcm.c | 83 +++++++---- sound/usb/line6/pcm.h | 19 +-- sound/usb/line6/playback.c | 37 +++-- sound/usb/line6/pod.c | 12 +- sound/usb/line6/podhd.c | 328 +++++++++++++++++++++++++++++++++++++++++--- sound/usb/line6/toneport.c | 6 +- sound/usb/line6/variax.c | 6 +- 13 files changed, 767 insertions(+), 156 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2 0/9] ASoC: simple-card: tidyup and synchronize patches
by Kuninori Morimoto 16 Sep '16

16 Sep '16

Hi Mark These are v2 of cleanup/tidyup and synchronize patches for simple-card. We have now simple-card and simple-scu-card. These are almost same, difference should be very small. But, because of historical reason, these have "code" difference. This patch cleanup and synchronize these. Kuninori Morimoto (9): ASoC: simple-card: call of_node_put() for dai_link ASoC: simple-card: tidyup mclk-fs of_property_read_u32() usage ASoC: simple-card: code sync: tidyup simple_priv_to_xxx() macro ASoC: simple-card: code sync: use simple_priv_to_props() macro ASoC: simple-card: is GPL v2 ASoC: simple-card: code sync: tidyup white line ASoC: simple-card: code sync: use tag ASoC: simple-card: code sync: rename num_link to num ASoC: simple-card: use kzalloc() for dai_props / dai_link sound/soc/generic/simple-card.c | 87 ++++++++++++++++++++--------------------- 1 file changed, 43 insertions(+), 44 deletions(-) -- 1.9.1

2 11

[alsa-devel] [RFC][PATCH] ASoC: rt5659: Fix incorrect register addresses
by Nicolin Chen 14 Sep '16

14 Sep '16

According to the datasheets of Realtek ALC5658 (Rev. 0.9) and ALC5659 (Rev. 0.13), the "IRQ Control 4" register is located at MX-00BAh while the "IRQ control 5" register is at MX-00BBh. There is no "IRQ Control 6" register in the datasheets but it is supposed to be behind the "IRQ Control 5" register. So this patch corrects these addresses. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Hi Mark, This patch needs the confirmation from Realtek side (hoping that at least Bard or Oder would give an Ack) as the datasheets that I have might not reflect the latest register map. And the datasheets don't seem to be directly downloadable from Realtek website. sound/soc/codecs/rt5659.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h index d69b0eb..8f1aeef 100644 --- a/sound/soc/codecs/rt5659.h +++ b/sound/soc/codecs/rt5659.h @@ -180,9 +180,9 @@ #define RT5659_IRQ_CTRL_1 0x00b6 #define RT5659_IRQ_CTRL_2 0x00b7 #define RT5659_IRQ_CTRL_3 0x00b8 -#define RT5659_IRQ_CTRL_4 0x00b9 -#define RT5659_IRQ_CTRL_5 0x00ba -#define RT5659_IRQ_CTRL_6 0x00bb +#define RT5659_IRQ_CTRL_4 0x00ba +#define RT5659_IRQ_CTRL_5 0x00bb +#define RT5659_IRQ_CTRL_6 0x00bc #define RT5659_INT_ST_1 0x00be #define RT5659_INT_ST_2 0x00bf #define RT5659_GPIO_CTRL_1 0x00c0 -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: rt5660: add rt5660 codec driver
by Oder Chiou 14 Sep '16

14 Sep '16

This is the initial codec driver for rt5660 Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5660.txt | 50 + include/sound/rt5660.h | 31 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5660.c | 1364 ++++++++++++++++++++ sound/soc/codecs/rt5660.h | 847 ++++++++++++ 6 files changed, 2300 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5660.txt create mode 100644 include/sound/rt5660.h create mode 100644 sound/soc/codecs/rt5660.c create mode 100644 sound/soc/codecs/rt5660.h diff --git a/Documentation/devicetree/bindings/sound/rt5660.txt b/Documentation/devicetree/bindings/sound/rt5660.txt new file mode 100644 index 0000000..c6cdb0d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5660.txt @@ -0,0 +1,50 @@ +RT5660 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5660". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in3-differential + Boolean. Indicate MIC1/3 input are differential, rather than single-ended. + +- realtek,use-ldo2 + Boolean. Use LDO2 for MICVDD + +- realtek,poweroff-in-suspend + Boolean. If the codec will be powered off in suspend, the resume should be + added delay time for waiting codec power ready. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using IN1P pin as dmic1 data pin + +Pins on the device (for linking into audio routes) for RT5660: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN1N + * IN2P + * IN3P + * IN3N + * SPO + * LOUTL + * LOUTR + +Example: + +rt5660 { + compatible = "realtek,rt5660"; + reg = <0x1c>; +}; diff --git a/include/sound/rt5660.h b/include/sound/rt5660.h new file mode 100644 index 0000000..065f83a --- /dev/null +++ b/include/sound/rt5660.h @@ -0,0 +1,31 @@ +/* + * linux/sound/rt5660.h -- Platform data for RT5660 + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5660_H +#define __LINUX_SND_RT5660_H + +enum rt5660_dmic1_data_pin { + RT5660_DMIC1_NULL, + RT5660_DMIC1_DATA_GPIO2, + RT5660_DMIC1_DATA_IN1P, +}; + +struct rt5660_platform_data { + /* IN1 & IN3 can optionally be differential */ + bool in1_diff; + bool in3_diff; + bool use_ldo2; + bool poweroff_codec_in_suspend; + + enum rt5660_dmic1_data_pin dmic1_data_pin; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1cd6ab3..890affe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -112,6 +112,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5659 if I2C + select SND_SOC_RT5660 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -645,6 +646,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y + default y if SND_SOC_RT5660=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -653,6 +655,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m default m if SND_SOC_RT5659=m + default m if SND_SOC_RT5660=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -697,6 +700,9 @@ config SND_SOC_RT5651 config SND_SOC_RT5659 tristate +config SND_SOC_RT5660 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 58036af..68fed20 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5659-objs := rt5659.o +snd-soc-rt5660-objs := rt5660.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -333,6 +334,7 @@ obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5659) += snd-soc-rt5659.o +obj-$(CONFIG_SND_SOC_RT5660) += snd-soc-rt5660.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c new file mode 100644 index 0000000..4c2c65e --- /dev/null +++ b/sound/soc/codecs/rt5660.c @@ -0,0 +1,1364 @@ +/* + * rt5660.c -- RT5660 ALSA SoC audio codec driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" +#include "rt5660.h" + +#define RT5660_DEVICE_ID 0x6338 + +#define RT5660_PR_RANGE_BASE (0xff + 1) +#define RT5660_PR_SPACING 0x100 + +#define RT5660_PR_BASE (RT5660_PR_RANGE_BASE + (0 * RT5660_PR_SPACING)) + +static const struct regmap_range_cfg rt5660_ranges[] = { + { .name = "PR", .range_min = RT5660_PR_BASE, + .range_max = RT5660_PR_BASE + 0xf3, + .selector_reg = RT5660_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5660_PRIV_DATA, + .window_len = 0x1, }, +}; + +static const struct reg_sequence rt5660_patch[] = { + { RT5660_ALC_PGA_CTRL2, 0x44c3 }, + { RT5660_PR_BASE + 0x3d, 0x2600 }, +}; + +static const struct reg_default rt5660_reg[] = { + { 0x00, 0x0000 }, + { 0x01, 0xc800 }, + { 0x02, 0xc8c8 }, + { 0x0d, 0x1010 }, + { 0x0e, 0x1010 }, + { 0x19, 0xafaf }, + { 0x1c, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x6060 }, + { 0x29, 0x8080 }, + { 0x2a, 0x4242 }, + { 0x2f, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe000 }, + { 0x46, 0x003e }, + { 0x48, 0xf800 }, + { 0x4a, 0x0004 }, + { 0x4d, 0x0000 }, + { 0x4e, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x70, 0x8000 }, + { 0x73, 0x7000 }, + { 0x74, 0x3c00 }, + { 0x75, 0x2800 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x8c, 0x0228 }, + { 0x8d, 0xa000 }, + { 0x8e, 0x0000 }, + { 0x92, 0x0000 }, + { 0x93, 0x3000 }, + { 0xa1, 0x0059 }, + { 0xa2, 0x0001 }, + { 0xa3, 0x5c80 }, + { 0xa4, 0x0146 }, + { 0xa5, 0x1f1f }, + { 0xa6, 0x78c6 }, + { 0xa7, 0xe5ec }, + { 0xa8, 0xba61 }, + { 0xa9, 0x3c78 }, + { 0xaa, 0x8ae2 }, + { 0xab, 0xe5ec }, + { 0xac, 0xc600 }, + { 0xad, 0xba61 }, + { 0xae, 0x17ed }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x020c }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x4000 }, + { 0xbb, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0100 }, + { 0xc0, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xd3, 0xa220 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xe0, 0x8000 }, + { 0xe1, 0x0200 }, + { 0xe2, 0x8000 }, + { 0xe3, 0x0200 }, + { 0xe4, 0x0f20 }, + { 0xe5, 0x001f }, + { 0xe6, 0x020c }, + { 0xe7, 0x1f00 }, + { 0xe8, 0x0000 }, + { 0xe9, 0x4000 }, + { 0xea, 0x00a6 }, + { 0xeb, 0x04c3 }, + { 0xec, 0x27c8 }, + { 0xed, 0x7418 }, + { 0xee, 0xbf50 }, + { 0xef, 0x0045 }, + { 0xf0, 0x0007 }, + { 0xfa, 0x0000 }, + { 0xfd, 0x0000 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6338 }, +}; + +static bool rt5660_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5660_ranges); i++) + if ((reg >= rt5660_ranges[i].window_start && + reg <= rt5660_ranges[i].window_start + + rt5660_ranges[i].window_len) || + (reg >= rt5660_ranges[i].range_min && + reg <= rt5660_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5660_RESET: + case RT5660_PRIV_DATA: + case RT5660_EQ_CTRL1: + case RT5660_IRQ_CTRL2: + case RT5660_INT_IRQ_ST: + case RT5660_VENDOR_ID: + case RT5660_VENDOR_ID1: + case RT5660_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5660_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5660_ranges); i++) + if ((reg >= rt5660_ranges[i].window_start && + reg <= rt5660_ranges[i].window_start + + rt5660_ranges[i].window_len) || + (reg >= rt5660_ranges[i].range_min && + reg <= rt5660_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5660_RESET: + case RT5660_SPK_VOL: + case RT5660_LOUT_VOL: + case RT5660_IN1_IN2: + case RT5660_IN3_IN4: + case RT5660_DAC1_DIG_VOL: + case RT5660_STO1_ADC_DIG_VOL: + case RT5660_ADC_BST_VOL1: + case RT5660_STO1_ADC_MIXER: + case RT5660_AD_DA_MIXER: + case RT5660_STO_DAC_MIXER: + case RT5660_DIG_INF1_DATA: + case RT5660_REC_L1_MIXER: + case RT5660_REC_L2_MIXER: + case RT5660_REC_R1_MIXER: + case RT5660_REC_R2_MIXER: + case RT5660_LOUT_MIXER: + case RT5660_SPK_MIXER: + case RT5660_SPO_MIXER: + case RT5660_SPO_CLSD_RATIO: + case RT5660_OUT_L_GAIN1: + case RT5660_OUT_L_GAIN2: + case RT5660_OUT_L1_MIXER: + case RT5660_OUT_R_GAIN1: + case RT5660_OUT_R_GAIN2: + case RT5660_OUT_R1_MIXER: + case RT5660_PWR_DIG1: + case RT5660_PWR_DIG2: + case RT5660_PWR_ANLG1: + case RT5660_PWR_ANLG2: + case RT5660_PWR_MIXER: + case RT5660_PWR_VOL: + case RT5660_PRIV_INDEX: + case RT5660_PRIV_DATA: + case RT5660_I2S1_SDP: + case RT5660_ADDA_CLK1: + case RT5660_ADDA_CLK2: + case RT5660_DMIC_CTRL1: + case RT5660_GLB_CLK: + case RT5660_PLL_CTRL1: + case RT5660_PLL_CTRL2: + case RT5660_CLSD_AMP_OC_CTRL: + case RT5660_CLSD_AMP_CTRL: + case RT5660_LOUT_AMP_CTRL: + case RT5660_SPK_AMP_SPKVDD: + case RT5660_MICBIAS: + case RT5660_CLSD_OUT_CTRL1: + case RT5660_CLSD_OUT_CTRL2: + case RT5660_DIPOLE_MIC_CTRL1: + case RT5660_DIPOLE_MIC_CTRL2: + case RT5660_DIPOLE_MIC_CTRL3: + case RT5660_DIPOLE_MIC_CTRL4: + case RT5660_DIPOLE_MIC_CTRL5: + case RT5660_DIPOLE_MIC_CTRL6: + case RT5660_DIPOLE_MIC_CTRL7: + case RT5660_DIPOLE_MIC_CTRL8: + case RT5660_DIPOLE_MIC_CTRL9: + case RT5660_DIPOLE_MIC_CTRL10: + case RT5660_DIPOLE_MIC_CTRL11: + case RT5660_DIPOLE_MIC_CTRL12: + case RT5660_EQ_CTRL1: + case RT5660_EQ_CTRL2: + case RT5660_DRC_AGC_CTRL1: + case RT5660_DRC_AGC_CTRL2: + case RT5660_DRC_AGC_CTRL3: + case RT5660_DRC_AGC_CTRL4: + case RT5660_DRC_AGC_CTRL5: + case RT5660_JD_CTRL: + case RT5660_IRQ_CTRL1: + case RT5660_IRQ_CTRL2: + case RT5660_INT_IRQ_ST: + case RT5660_GPIO_CTRL1: + case RT5660_GPIO_CTRL2: + case RT5660_WIND_FILTER_CTRL1: + case RT5660_SV_ZCD1: + case RT5660_SV_ZCD2: + case RT5660_DRC1_LM_CTRL1: + case RT5660_DRC1_LM_CTRL2: + case RT5660_DRC2_LM_CTRL1: + case RT5660_DRC2_LM_CTRL2: + case RT5660_MULTI_DRC_CTRL: + case RT5660_DRC2_CTRL1: + case RT5660_DRC2_CTRL2: + case RT5660_DRC2_CTRL3: + case RT5660_DRC2_CTRL4: + case RT5660_DRC2_CTRL5: + case RT5660_ALC_PGA_CTRL1: + case RT5660_ALC_PGA_CTRL2: + case RT5660_ALC_PGA_CTRL3: + case RT5660_ALC_PGA_CTRL4: + case RT5660_ALC_PGA_CTRL5: + case RT5660_ALC_PGA_CTRL6: + case RT5660_ALC_PGA_CTRL7: + case RT5660_GEN_CTRL1: + case RT5660_GEN_CTRL2: + case RT5660_GEN_CTRL3: + case RT5660_VENDOR_ID: + case RT5660_VENDOR_ID1: + case RT5660_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(rt5660_out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_adc_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_bst_tlv, -1200, 75, 0); + +static const struct snd_kcontrol_new rt5660_snd_controls[] = { + /* Speaker Output Volume */ + SOC_SINGLE("Speaker Playback Switch", RT5660_SPK_VOL, RT5660_L_MUTE_SFT, + 1, 1), + SOC_SINGLE_TLV("Speaker Playback Volume", RT5660_SPK_VOL, + RT5660_L_VOL_SFT, 39, 1, rt5660_out_vol_tlv), + + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5660_LOUT_VOL, RT5660_L_MUTE_SFT, + RT5660_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5660_LOUT_VOL, RT5660_L_VOL_SFT, + RT5660_R_VOL_SFT, 39, 1, rt5660_out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5660_DAC1_DIG_VOL, + RT5660_DAC_L1_VOL_SFT, RT5660_DAC_R1_VOL_SFT, 87, 0, + rt5660_dac_vol_tlv), + + /* IN1/IN2/IN3 Control */ + SOC_SINGLE_TLV("IN1 Boost Volume", RT5660_IN1_IN2, RT5660_BST_SFT1, 69, + 0, rt5660_bst_tlv), + SOC_SINGLE_TLV("IN2 Boost Volume", RT5660_IN1_IN2, RT5660_BST_SFT2, 69, + 0, rt5660_bst_tlv), + SOC_SINGLE_TLV("IN3 Boost Volume", RT5660_IN3_IN4, RT5660_BST_SFT3, 69, + 0, rt5660_bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5660_STO1_ADC_DIG_VOL, + RT5660_L_MUTE_SFT, RT5660_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5660_STO1_ADC_DIG_VOL, + RT5660_ADC_L_VOL_SFT, RT5660_ADC_R_VOL_SFT, 63, 0, + rt5660_adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5660_ADC_BST_VOL1, + RT5660_STO1_ADC_L_BST_SFT, RT5660_STO1_ADC_R_BST_SFT, 3, 0, + rt5660_adc_bst_tlv), +}; + +/** + * rt5660_set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + */ +static int rt5660_set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + int idx, rate; + + rate = rt5660->sysclk / rl6231_get_pre_div(rt5660->regmap, + RT5660_ADDA_CLK1, RT5660_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5660_DMIC_CTRL1, + RT5660_DMIC_CLK_MASK, idx << RT5660_DMIC_CLK_SFT); + + return idx; +} + +static int rt5660_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + unsigned int val; + + val = snd_soc_read(codec, RT5660_GLB_CLK); + val &= RT5660_SCLK_SRC_MASK; + if (val == RT5660_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int rt5660_is_using_ldo2(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + return rt5660->pdata.use_ldo2; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5660_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5660_AD_DA_MIXER, + RT5660_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5660_AD_DA_MIXER, + RT5660_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5660_AD_DA_MIXER, + RT5660_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5660_AD_DA_MIXER, + RT5660_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5660_rec_l_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST3_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5660_REC_L2_MIXER, + RT5660_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_rec_r_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST3_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5660_REC_R2_MIXER, + RT5660_M_OM_R_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_spk_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_SPK_MIXER, + RT5660_M_BST3_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_SPK_MIXER, + RT5660_M_BST1_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_SPK_MIXER, + RT5660_M_DACL_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_SPK_MIXER, + RT5660_M_DACR_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIXL Switch", RT5660_SPK_MIXER, + RT5660_M_OM_L_SM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_out_l_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST3_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIXL Switch", RT5660_OUT_L1_MIXER, + RT5660_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_OUT_L1_MIXER, + RT5660_M_DAC_R_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_OUT_L1_MIXER, + RT5660_M_DAC_L_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5660_OUT_R1_MIXER, + RT5660_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_OUT_R1_MIXER, + RT5660_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Switch", RT5660_OUT_R1_MIXER, + RT5660_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_OUT_R1_MIXER, + RT5660_M_DAC_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_OUT_R1_MIXER, + RT5660_M_DAC_L_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_spo_mix[] = { + SOC_DAPM_SINGLE("DACR Switch", RT5660_SPO_MIXER, + RT5660_M_DAC_R_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_SPO_MIXER, + RT5660_M_DAC_L_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL Switch", RT5660_SPO_MIXER, + RT5660_M_SV_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_SPO_MIXER, + RT5660_M_BST1_SPM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_lout_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT5660_LOUT_MIXER, + RT5660_M_DAC1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX Switch", RT5660_LOUT_MIXER, + RT5660_M_LOVOL_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spk_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_SPK_VOL, + RT5660_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_l_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_LOUT_VOL, + RT5660_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_LOUT_VOL, + RT5660_VOL_R_SFT, 1, 1); + +/* Interface data select */ +static const char * const rt5660_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5660_if1_dac_enum, + RT5660_DIG_INF1_DATA, RT5660_IF1_DAC_IN_SFT, rt5660_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5660_if1_adc_enum, + RT5660_DIG_INF1_DATA, RT5660_IF1_ADC_IN_SFT, rt5660_data_select); + +static const struct snd_kcontrol_new rt5660_if1_dac_swap_mux = + SOC_DAPM_ENUM("IF1 DAC Swap Source", rt5660_if1_dac_enum); + +static const struct snd_kcontrol_new rt5660_if1_adc_swap_mux = + SOC_DAPM_ENUM("IF1 ADC Swap Source", rt5660_if1_adc_enum); + +static int rt5660_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5660_LOUT_AMP_CTRL, + RT5660_LOUT_CO_MASK | RT5660_LOUT_CB_MASK, + RT5660_LOUT_CO_EN | RT5660_LOUT_CB_PU); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5660_LOUT_AMP_CTRL, + RT5660_LOUT_CO_MASK | RT5660_LOUT_CB_MASK, + RT5660_LOUT_CO_DIS | RT5660_LOUT_CB_PD); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5660_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5660_PWR_ANLG1, + RT5660_PWR_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5660_PWR_ANLG2, + RT5660_PWR_PLL_BIT, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5660_PWR_ANLG2, + RT5660_PWR_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5660_PWR_ANLG2, + RT5660_PWR_MB2_BIT, 0, NULL, 0), + + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + rt5660_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Power", RT5660_DMIC_CTRL1, + RT5660_DMIC_1_EN_SFT, 0, NULL, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5660_PWR_ANLG2, RT5660_PWR_BST1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("BST2", RT5660_PWR_ANLG2, RT5660_PWR_BST2_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5660_PWR_ANLG2, RT5660_PWR_BST3_BIT, 0, + NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5660_PWR_MIXER, RT5660_PWR_RM_L_BIT, + 0, rt5660_rec_l_mix, ARRAY_SIZE(rt5660_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5660_PWR_MIXER, RT5660_PWR_RM_R_BIT, + 0, rt5660_rec_r_mix, ARRAY_SIZE(rt5660_rec_r_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC L power", RT5660_PWR_DIG1, + RT5660_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC R power", RT5660_PWR_DIG1, + RT5660_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", RT5660_PR_BASE + RT5660_CHOP_DAC_ADC, + 12, 0, NULL, 0), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5660_PWR_DIG2, + RT5660_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5660_sto1_adc_l_mix, ARRAY_SIZE(rt5660_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5660_sto1_adc_r_mix, ARRAY_SIZE(rt5660_sto1_adc_r_mix)), + + /* ADC */ + SND_SOC_DAPM_ADC("Stereo1 ADC MIXL", NULL, RT5660_STO1_ADC_DIG_VOL, + RT5660_L_MUTE_SFT, 1), + SND_SOC_DAPM_ADC("Stereo1 ADC MIXR", NULL, RT5660_STO1_ADC_DIG_VOL, + RT5660_R_MUTE_SFT, 1), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5660_PWR_DIG1, RT5660_PWR_I2S1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF1 DAC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5660_if1_dac_swap_mux), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF1 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5660_if1_adc_swap_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, rt5660_dac_l_mix, + ARRAY_SIZE(rt5660_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, rt5660_dac_r_mix, + ARRAY_SIZE(rt5660_dac_r_mix)), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5660_PWR_DIG2, + RT5660_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5660_sto_dac_l_mix, ARRAY_SIZE(rt5660_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5660_sto_dac_r_mix, ARRAY_SIZE(rt5660_sto_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5660_PWR_DIG1, + RT5660_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5660_PWR_DIG1, + RT5660_PWR_DAC_R1_BIT, 0), + + /* OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIX", RT5660_PWR_MIXER, RT5660_PWR_SM_BIT, + 0, rt5660_spk_mix, ARRAY_SIZE(rt5660_spk_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5660_PWR_MIXER, RT5660_PWR_OM_L_BIT, + 0, rt5660_out_l_mix, ARRAY_SIZE(rt5660_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5660_PWR_MIXER, RT5660_PWR_OM_R_BIT, + 0, rt5660_out_r_mix, ARRAY_SIZE(rt5660_out_r_mix)), + + /* Output Volume */ + SND_SOC_DAPM_SWITCH("SPKVOL", RT5660_PWR_VOL, + RT5660_PWR_SV_BIT, 0, &spk_vol_control), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("LOUTVOL", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("LOUTVOL L", SND_SOC_NOPM, + RT5660_PWR_LV_L_BIT, 0, &lout_l_vol_control), + SND_SOC_DAPM_SWITCH("LOUTVOL R", SND_SOC_NOPM, + RT5660_PWR_LV_R_BIT, 0, &lout_r_vol_control), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPO MIX", SND_SOC_NOPM, 0, + 0, rt5660_spo_mix, ARRAY_SIZE(rt5660_spo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, + rt5660_lout_mix, ARRAY_SIZE(rt5660_lout_mix)), + SND_SOC_DAPM_SUPPLY("VREF HP", RT5660_GEN_CTRL1, + RT5660_PWR_VREF_HP_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, RT5660_PWR_ANLG1, + RT5660_PWR_HA_BIT, 0, rt5660_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 1, RT5660_PWR_DIG1, + RT5660_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("SPO"), +}; + +static const struct snd_soc_dapm_route rt5660_dapm_routes[] = { + { "MICBIAS1", NULL, "LDO2", rt5660_is_using_ldo2 }, + { "MICBIAS2", NULL, "LDO2", rt5660_is_using_ldo2 }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST2", NULL, "IN2P" }, + { "BST3", NULL, "IN3P" }, + { "BST3", NULL, "IN3N" }, + + { "RECMIXL", "BST3 Switch", "BST3" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + { "RECMIXL", "OUT MIXL Switch", "OUT MIXL" }, + + { "RECMIXR", "BST3 Switch", "BST3" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + { "RECMIXR", "OUT MIXR Switch", "OUT MIXR" }, + + { "ADC L", NULL, "RECMIXL" }, + { "ADC L", NULL, "ADC L power" }, + { "ADC L", NULL, "ADC clock" }, + { "ADC R", NULL, "RECMIXR" }, + { "ADC R", NULL, "ADC R power" }, + { "ADC R", NULL, "ADC clock" }, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC Power"}, + + { "Sto1 ADC MIXL", "ADC1 Switch", "ADC L" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "DMIC L1" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "ADC R" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "DMIC R1" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "IF1 ADC", NULL, "Stereo1 ADC MIXL" }, + { "IF1 ADC", NULL, "Stereo1 ADC MIXR" }, + { "IF1 ADC", NULL, "I2S1" }, + + { "IF1 ADC Swap Mux", "L/R", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "R/L", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "L/L", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "R/R", "IF1 ADC" }, + { "AIF1TX", NULL, "IF1 ADC Swap Mux" }, + + { "IF1 DAC", NULL, "AIF1RX" }, + { "IF1 DAC", NULL, "I2S1" }, + + { "IF1 DAC Swap Mux", "L/R", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "R/L", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "L/L", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "R/R", "IF1 DAC" }, + + { "IF1 DAC L", NULL, "IF1 DAC Swap Mux" }, + { "IF1 DAC R", NULL, "IF1 DAC Swap Mux" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "IF1 DAC L" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "IF1 DAC R" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + + { "SPK MIX", "BST3 Switch", "BST3" }, + { "SPK MIX", "BST1 Switch", "BST1" }, + { "SPK MIX", "DACL Switch", "DAC L1" }, + { "SPK MIX", "DACR Switch", "DAC R1" }, + { "SPK MIX", "OUTMIXL Switch", "OUT MIXL" }, + + { "OUT MIXL", "BST3 Switch", "BST3" }, + { "OUT MIXL", "BST2 Switch", "BST2" }, + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "RECMIXL Switch", "RECMIXL" }, + { "OUT MIXL", "DACR Switch", "DAC R1" }, + { "OUT MIXL", "DACL Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "BST1 Switch", "BST1" }, + { "OUT MIXR", "RECMIXR Switch", "RECMIXR" }, + { "OUT MIXR", "DACR Switch", "DAC R1" }, + { "OUT MIXR", "DACL Switch", "DAC L1" }, + + { "SPO MIX", "DACR Switch", "DAC R1" }, + { "SPO MIX", "DACL Switch", "DAC L1" }, + { "SPO MIX", "SPKVOL Switch", "SPKVOL" }, + { "SPO MIX", "BST1 Switch", "BST1" }, + + { "SPKVOL", "Switch", "SPK MIX" }, + { "LOUTVOL L", "Switch", "OUT MIXL" }, + { "LOUTVOL R", "Switch", "OUT MIXR" }, + + { "LOUTVOL", NULL, "LOUTVOL L" }, + { "LOUTVOL", NULL, "LOUTVOL R" }, + + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + + { "LOUT MIX", "DAC Switch", "DAC 1" }, + { "LOUT MIX", "OUTMIX Switch", "LOUTVOL" }, + + { "LOUT amp", NULL, "LOUT MIX" }, + { "LOUT amp", NULL, "VREF HP" }, + { "LOUTL", NULL, "LOUT amp" }, + { "LOUTR", NULL, "LOUT amp" }, + + { "SPK amp", NULL, "SPO MIX" }, + { "SPO", NULL, "SPK amp" }, +}; + +static int rt5660_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5660->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5660->sysclk, rt5660->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5660->lrck[dai->id], dai->id); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return frame_size; + } + + if (frame_size > 32) + bclk_ms = 1; + else + bclk_ms = 0; + + rt5660->bclk[dai->id] = rt5660->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5660->bclk[dai->id], rt5660->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= RT5660_I2S_DL_20; + break; + case 24: + val_len |= RT5660_I2S_DL_24; + break; + case 8: + val_len |= RT5660_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5660_AIF1: + mask_clk = RT5660_I2S_BCLK_MS1_MASK | RT5660_I2S_PD1_MASK; + val_clk = bclk_ms << RT5660_I2S_BCLK_MS1_SFT | + pre_div << RT5660_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5660_I2S1_SDP, RT5660_I2S_DL_MASK, + val_len); + snd_soc_update_bits(codec, RT5660_ADDA_CLK1, mask_clk, val_clk); + break; + + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5660_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5660->master[dai->id] = 1; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5660_I2S_MS_S; + rt5660->master[dai->id] = 0; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5660_I2S_BP_INV; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5660_I2S_DF_LEFT; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5660_I2S_DF_PCM_A; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5660_I2S_DF_PCM_B; + break; + + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5660_AIF1: + snd_soc_update_bits(codec, RT5660_I2S1_SDP, + RT5660_I2S_MS_MASK | RT5660_I2S_BP_MASK | + RT5660_I2S_DF_MASK, reg_val); + break; + + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5660_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5660->sysclk && clk_id == rt5660->sysclk_src) + return 0; + + switch (clk_id) { + case RT5660_SCLK_S_MCLK: + reg_val |= RT5660_SCLK_SRC_MCLK; + break; + + case RT5660_SCLK_S_PLL1: + reg_val |= RT5660_SCLK_SRC_PLL1; + break; + + case RT5660_SCLK_S_RCCLK: + reg_val |= RT5660_SCLK_SRC_RCCLK; + break; + + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5660_GLB_CLK, RT5660_SCLK_SRC_MASK, + reg_val); + + rt5660->sysclk = freq; + rt5660->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5660_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5660->pll_src && freq_in == rt5660->pll_in && + freq_out == rt5660->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5660->pll_in = 0; + rt5660->pll_out = 0; + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_SCLK_SRC_MASK, RT5660_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5660_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_PLL1_SRC_MASK, RT5660_PLL1_SRC_MCLK); + break; + + case RT5660_PLL1_S_BCLK: + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_PLL1_SRC_MASK, RT5660_PLL1_SRC_BCLK1); + break; + + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5660_PLL_CTRL1, + pll_code.n_code << RT5660_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5660_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5660_PLL_M_SFT | + pll_code.m_bp << RT5660_PLL_M_BP_SFT); + + rt5660->pll_in = freq_in; + rt5660->pll_out = freq_out; + rt5660->pll_src = source; + + return 0; +} + +static int rt5660_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, RT5660_GEN_CTRL1, + RT5660_DIG_GATE_CTRL, RT5660_DIG_GATE_CTRL); + + if (IS_ERR(rt5660->mclk)) + break; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { + clk_disable_unprepare(rt5660->mclk); + } else { + ret = clk_prepare_enable(rt5660->mclk); + if (ret) + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, RT5660_PWR_ANLG1, + RT5660_PWR_VREF1 | RT5660_PWR_MB | + RT5660_PWR_BG | RT5660_PWR_VREF2, + RT5660_PWR_VREF1 | RT5660_PWR_MB | + RT5660_PWR_BG | RT5660_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5660_PWR_ANLG1, + RT5660_PWR_FV1 | RT5660_PWR_FV2, + RT5660_PWR_FV1 | RT5660_PWR_FV2); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5660_GEN_CTRL1, + RT5660_DIG_GATE_CTRL, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5660_probe(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + /* Check if MCLK provided */ + rt5660->mclk = devm_clk_get(codec->dev, "mclk"); + if (PTR_ERR(rt5660->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + rt5660->codec = codec; + + return 0; +} + +static int rt5660_remove(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5660_RESET, 0); +} + +#ifdef CONFIG_PM +static int rt5660_suspend(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5660->regmap, true); + regcache_mark_dirty(rt5660->regmap); + + return 0; +} + +static int rt5660_resume(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + if (rt5660->pdata.poweroff_codec_in_suspend) + usleep_range(350000, 400000); + + regcache_cache_only(rt5660->regmap, false); + regcache_sync(rt5660->regmap); + + return 0; +} +#else +#define rt5660_suspend NULL +#define rt5660_resume NULL +#endif + +#define RT5660_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5660_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5660_aif_dai_ops = { + .hw_params = rt5660_hw_params, + .set_fmt = rt5660_set_dai_fmt, + .set_sysclk = rt5660_set_dai_sysclk, + .set_pll = rt5660_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5660_dai[] = { + { + .name = "rt5660-aif1", + .id = RT5660_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5660_STEREO_RATES, + .formats = RT5660_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5660_STEREO_RATES, + .formats = RT5660_FORMATS, + }, + .ops = &rt5660_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5660 = { + .probe = rt5660_probe, + .remove = rt5660_remove, + .suspend = rt5660_suspend, + .resume = rt5660_resume, + .set_bias_level = rt5660_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt5660_snd_controls, + .num_controls = ARRAY_SIZE(rt5660_snd_controls), + .dapm_widgets = rt5660_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5660_dapm_widgets), + .dapm_routes = rt5660_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5660_dapm_routes), + }, +}; + +static const struct regmap_config rt5660_regmap = { + .reg_bits = 8, + .val_bits = 16, + .use_single_rw = true, + + .max_register = RT5660_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5660_ranges) * + RT5660_PR_SPACING), + .volatile_reg = rt5660_volatile_register, + .readable_reg = rt5660_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5660_reg, + .num_reg_defaults = ARRAY_SIZE(rt5660_reg), + .ranges = rt5660_ranges, + .num_ranges = ARRAY_SIZE(rt5660_ranges), +}; + +static const struct i2c_device_id rt5660_i2c_id[] = { + { "rt5660", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5660_i2c_id); + +static const struct of_device_id rt5660_of_match[] = { + { .compatible = "realtek,rt5660", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5660_of_match); + +static const struct acpi_device_id rt5660_acpi_match[] = { + { "10EC5660", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); + +static int rt5660_parse_dt(struct rt5660_priv *rt5660, struct device *dev) +{ + rt5660->pdata.in1_diff = device_property_read_bool(dev, + "realtek,in1-differential"); + rt5660->pdata.in3_diff = device_property_read_bool(dev, + "realtek,in3-differential"); + rt5660->pdata.use_ldo2 = device_property_read_bool(dev, + "realtek,use-ldo2"); + rt5660->pdata.poweroff_codec_in_suspend = device_property_read_bool(dev, + "realtek,poweroff-in-suspend"); + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5660->pdata.dmic1_data_pin); + + return 0; +} + +static int rt5660_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5660_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5660_priv *rt5660; + int ret; + unsigned int val; + + rt5660 = devm_kzalloc(&i2c->dev, sizeof(struct rt5660_priv), + GFP_KERNEL); + + if (rt5660 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5660); + + if (pdata) + rt5660->pdata = *pdata; + else if (i2c->dev.of_node) + rt5660_parse_dt(rt5660, &i2c->dev); + + rt5660->regmap = devm_regmap_init_i2c(i2c, &rt5660_regmap); + if (IS_ERR(rt5660->regmap)) { + ret = PTR_ERR(rt5660->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5660->regmap, RT5660_VENDOR_ID2, &val); + if (val != RT5660_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5660/39\n", val); + return -ENODEV; + } + + regmap_write(rt5660->regmap, RT5660_RESET, 0); + + ret = regmap_register_patch(rt5660->regmap, rt5660_patch, + ARRAY_SIZE(rt5660_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5660->pdata.dmic1_data_pin) { + regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, + RT5660_GP1_PIN_MASK, RT5660_GP1_PIN_DMIC1_SCL); + + if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_GPIO2) + regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, + RT5660_SEL_DMIC_DATA_MASK, + RT5660_SEL_DMIC_DATA_GPIO2); + else if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_IN1P) + regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, + RT5660_SEL_DMIC_DATA_MASK, + RT5660_SEL_DMIC_DATA_IN1P); + } + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5660, + rt5660_dai, ARRAY_SIZE(rt5660_dai)); +} + +static int rt5660_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5660_i2c_driver = { + .driver = { + .name = "rt5660", + .acpi_match_table = ACPI_PTR(rt5660_acpi_match), + .of_match_table = of_match_ptr(rt5660_of_match), + }, + .probe = rt5660_i2c_probe, + .remove = rt5660_i2c_remove, + .id_table = rt5660_i2c_id, +}; +module_i2c_driver(rt5660_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5660 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5660.h b/sound/soc/codecs/rt5660.h new file mode 100644 index 0000000..6cdb926 --- /dev/null +++ b/sound/soc/codecs/rt5660.h @@ -0,0 +1,847 @@ +/* + * rt5660.h -- RT5660 ALSA SoC audio driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _RT5660_H +#define _RT5660_H + +#include <linux/clk.h> +#include <sound/rt5660.h> + +/* Info */ +#define RT5660_RESET 0x00 +#define RT5660_VENDOR_ID 0xfd +#define RT5660_VENDOR_ID1 0xfe +#define RT5660_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5660_SPK_VOL 0x01 +#define RT5660_LOUT_VOL 0x02 +/* I/O - Input */ +#define RT5660_IN1_IN2 0x0d +#define RT5660_IN3_IN4 0x0e +/* I/O - ADC/DAC/DMIC */ +#define RT5660_DAC1_DIG_VOL 0x19 +#define RT5660_STO1_ADC_DIG_VOL 0x1c +#define RT5660_ADC_BST_VOL1 0x1e +/* Mixer - D-D */ +#define RT5660_STO1_ADC_MIXER 0x27 +#define RT5660_AD_DA_MIXER 0x29 +#define RT5660_STO_DAC_MIXER 0x2a +#define RT5660_DIG_INF1_DATA 0x2f +/* Mixer - ADC */ +#define RT5660_REC_L1_MIXER 0x3b +#define RT5660_REC_L2_MIXER 0x3c +#define RT5660_REC_R1_MIXER 0x3d +#define RT5660_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5660_LOUT_MIXER 0x45 +#define RT5660_SPK_MIXER 0x46 +#define RT5660_SPO_MIXER 0x48 +#define RT5660_SPO_CLSD_RATIO 0x4a +#define RT5660_OUT_L_GAIN1 0x4d +#define RT5660_OUT_L_GAIN2 0x4e +#define RT5660_OUT_L1_MIXER 0x4f +#define RT5660_OUT_R_GAIN1 0x50 +#define RT5660_OUT_R_GAIN2 0x51 +#define RT5660_OUT_R1_MIXER 0x52 +/* Power */ +#define RT5660_PWR_DIG1 0x61 +#define RT5660_PWR_DIG2 0x62 +#define RT5660_PWR_ANLG1 0x63 +#define RT5660_PWR_ANLG2 0x64 +#define RT5660_PWR_MIXER 0x65 +#define RT5660_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5660_PRIV_INDEX 0x6a +#define RT5660_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5660_I2S1_SDP 0x70 +#define RT5660_ADDA_CLK1 0x73 +#define RT5660_ADDA_CLK2 0x74 +#define RT5660_DMIC_CTRL1 0x75 +/* Function - Analog */ +#define RT5660_GLB_CLK 0x80 +#define RT5660_PLL_CTRL1 0x81 +#define RT5660_PLL_CTRL2 0x82 +#define RT5660_CLSD_AMP_OC_CTRL 0x8c +#define RT5660_CLSD_AMP_CTRL 0x8d +#define RT5660_LOUT_AMP_CTRL 0x8e +#define RT5660_SPK_AMP_SPKVDD 0x92 +#define RT5660_MICBIAS 0x93 +#define RT5660_CLSD_OUT_CTRL1 0xa1 +#define RT5660_CLSD_OUT_CTRL2 0xa2 +#define RT5660_DIPOLE_MIC_CTRL1 0xa3 +#define RT5660_DIPOLE_MIC_CTRL2 0xa4 +#define RT5660_DIPOLE_MIC_CTRL3 0xa5 +#define RT5660_DIPOLE_MIC_CTRL4 0xa6 +#define RT5660_DIPOLE_MIC_CTRL5 0xa7 +#define RT5660_DIPOLE_MIC_CTRL6 0xa8 +#define RT5660_DIPOLE_MIC_CTRL7 0xa9 +#define RT5660_DIPOLE_MIC_CTRL8 0xaa +#define RT5660_DIPOLE_MIC_CTRL9 0xab +#define RT5660_DIPOLE_MIC_CTRL10 0xac +#define RT5660_DIPOLE_MIC_CTRL11 0xad +#define RT5660_DIPOLE_MIC_CTRL12 0xae +/* Function - Digital */ +#define RT5660_EQ_CTRL1 0xb0 +#define RT5660_EQ_CTRL2 0xb1 +#define RT5660_DRC_AGC_CTRL1 0xb3 +#define RT5660_DRC_AGC_CTRL2 0xb4 +#define RT5660_DRC_AGC_CTRL3 0xb5 +#define RT5660_DRC_AGC_CTRL4 0xb6 +#define RT5660_DRC_AGC_CTRL5 0xb7 +#define RT5660_JD_CTRL 0xbb +#define RT5660_IRQ_CTRL1 0xbd +#define RT5660_IRQ_CTRL2 0xbe +#define RT5660_INT_IRQ_ST 0xbf +#define RT5660_GPIO_CTRL1 0xc0 +#define RT5660_GPIO_CTRL2 0xc2 +#define RT5660_WIND_FILTER_CTRL1 0xd3 +#define RT5660_SV_ZCD1 0xd9 +#define RT5660_SV_ZCD2 0xda +#define RT5660_DRC1_LM_CTRL1 0xe0 +#define RT5660_DRC1_LM_CTRL2 0xe1 +#define RT5660_DRC2_LM_CTRL1 0xe2 +#define RT5660_DRC2_LM_CTRL2 0xe3 +#define RT5660_MULTI_DRC_CTRL 0xe4 +#define RT5660_DRC2_CTRL1 0xe5 +#define RT5660_DRC2_CTRL2 0xe6 +#define RT5660_DRC2_CTRL3 0xe7 +#define RT5660_DRC2_CTRL4 0xe8 +#define RT5660_DRC2_CTRL5 0xe9 +#define RT5660_ALC_PGA_CTRL1 0xea +#define RT5660_ALC_PGA_CTRL2 0xeb +#define RT5660_ALC_PGA_CTRL3 0xec +#define RT5660_ALC_PGA_CTRL4 0xed +#define RT5660_ALC_PGA_CTRL5 0xee +#define RT5660_ALC_PGA_CTRL6 0xef +#define RT5660_ALC_PGA_CTRL7 0xf0 + +/* General Control */ +#define RT5660_GEN_CTRL1 0xfa +#define RT5660_GEN_CTRL2 0xfb +#define RT5660_GEN_CTRL3 0xfc + +/* Index of Codec Private Register definition */ +#define RT5660_CHOP_DAC_ADC 0x3d + +/* Global Definition */ +#define RT5660_L_MUTE (0x1 << 15) +#define RT5660_L_MUTE_SFT 15 +#define RT5660_VOL_L_MUTE (0x1 << 14) +#define RT5660_VOL_L_SFT 14 +#define RT5660_R_MUTE (0x1 << 7) +#define RT5660_R_MUTE_SFT 7 +#define RT5660_VOL_R_MUTE (0x1 << 6) +#define RT5660_VOL_R_SFT 6 +#define RT5660_L_VOL_MASK (0x3f << 8) +#define RT5660_L_VOL_SFT 8 +#define RT5660_R_VOL_MASK (0x3f) +#define RT5660_R_VOL_SFT 0 + +/* IN1 and IN2 Control (0x0d) */ +#define RT5660_IN_DF1 (0x1 << 15) +#define RT5660_IN_SFT1 15 +#define RT5660_BST_MASK1 (0x7f << 8) +#define RT5660_BST_SFT1 8 +#define RT5660_IN_DF2 (0x1 << 7) +#define RT5660_IN_SFT2 7 +#define RT5660_BST_MASK2 (0x7f << 0) +#define RT5660_BST_SFT2 0 + +/* IN3 and IN4 Control (0x0e) */ +#define RT5660_IN_DF3 (0x1 << 15) +#define RT5660_IN_SFT3 15 +#define RT5660_BST_MASK3 (0x7f << 8) +#define RT5660_BST_SFT3 8 +#define RT5660_IN_DF4 (0x1 << 7) +#define RT5660_IN_SFT4 7 +#define RT5660_BST_MASK4 (0x7f << 0) +#define RT5660_BST_SFT4 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5660_DAC_L1_VOL_MASK (0x7f << 9) +#define RT5660_DAC_L1_VOL_SFT 9 +#define RT5660_DAC_R1_VOL_MASK (0x7f << 1) +#define RT5660_DAC_R1_VOL_SFT 1 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5660_ADC_L_VOL_MASK (0x3f << 9) +#define RT5660_ADC_L_VOL_SFT 9 +#define RT5660_ADC_R_VOL_MASK (0x3f << 1) +#define RT5660_ADC_R_VOL_SFT 1 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5660_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5660_STO1_ADC_L_BST_SFT 14 +#define RT5660_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5660_STO1_ADC_R_BST_SFT 12 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5660_M_ADC_L1 (0x1 << 14) +#define RT5660_M_ADC_L1_SFT 14 +#define RT5660_M_ADC_L2 (0x1 << 13) +#define RT5660_M_ADC_L2_SFT 13 +#define RT5660_M_ADC_R1 (0x1 << 6) +#define RT5660_M_ADC_R1_SFT 6 +#define RT5660_M_ADC_R2 (0x1 << 5) +#define RT5660_M_ADC_R2_SFT 5 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5660_M_ADCMIX_L (0x1 << 15) +#define RT5660_M_ADCMIX_L_SFT 15 +#define RT5660_M_DAC1_L (0x1 << 14) +#define RT5660_M_DAC1_L_SFT 14 +#define RT5660_M_ADCMIX_R (0x1 << 7) +#define RT5660_M_ADCMIX_R_SFT 7 +#define RT5660_M_DAC1_R (0x1 << 6) +#define RT5660_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5660_M_DAC_L1 (0x1 << 14) +#define RT5660_M_DAC_L1_SFT 14 +#define RT5660_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5660_DAC_L1_STO_L_VOL_SFT 13 +#define RT5660_M_DAC_R1_STO_L (0x1 << 9) +#define RT5660_M_DAC_R1_STO_L_SFT 9 +#define RT5660_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5660_DAC_R1_STO_L_VOL_SFT 8 +#define RT5660_M_DAC_R1 (0x1 << 6) +#define RT5660_M_DAC_R1_SFT 6 +#define RT5660_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5660_DAC_R1_STO_R_VOL_SFT 5 +#define RT5660_M_DAC_L1_STO_R (0x1 << 1) +#define RT5660_M_DAC_L1_STO_R_SFT 1 +#define RT5660_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5660_DAC_L1_STO_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5660_IF1_DAC_IN_SEL (0x3 << 14) +#define RT5660_IF1_DAC_IN_SFT 14 +#define RT5660_IF1_ADC_IN_SEL (0x3 << 12) +#define RT5660_IF1_ADC_IN_SFT 12 + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5660_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5660_G_BST3_RM_L_SFT 4 +#define RT5660_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5660_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5660_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5660_G_BST1_RM_L_SFT 13 +#define RT5660_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5660_G_OM_L_RM_L_SFT 10 +#define RT5660_M_BST3_RM_L (0x1 << 3) +#define RT5660_M_BST3_RM_L_SFT 3 +#define RT5660_M_BST2_RM_L (0x1 << 2) +#define RT5660_M_BST2_RM_L_SFT 2 +#define RT5660_M_BST1_RM_L (0x1 << 1) +#define RT5660_M_BST1_RM_L_SFT 1 +#define RT5660_M_OM_L_RM_L (0x1) +#define RT5660_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5660_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5660_G_BST3_RM_R_SFT 4 +#define RT5660_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5660_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5660_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5660_G_BST1_RM_R_SFT 13 +#define RT5660_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5660_G_OM_R_RM_R_SFT 10 +#define RT5660_M_BST3_RM_R (0x1 << 3) +#define RT5660_M_BST3_RM_R_SFT 3 +#define RT5660_M_BST2_RM_R (0x1 << 2) +#define RT5660_M_BST2_RM_R_SFT 2 +#define RT5660_M_BST1_RM_R (0x1 << 1) +#define RT5660_M_BST1_RM_R_SFT 1 +#define RT5660_M_OM_R_RM_R (0x1) +#define RT5660_M_OM_R_RM_R_SFT 0 + +/* LOUTMIX Control (0x45) */ +#define RT5660_M_DAC1_LM (0x1 << 14) +#define RT5660_M_DAC1_LM_SFT 14 +#define RT5660_M_LOVOL_M (0x1 << 13) +#define RT5660_M_LOVOL_LM_SFT 13 + +/* SPK Mixer Control (0x46) */ +#define RT5660_G_BST3_SM_MASK (0x3 << 14) +#define RT5660_G_BST3_SM_SFT 14 +#define RT5660_G_BST1_SM_MASK (0x3 << 12) +#define RT5660_G_BST1_SM_SFT 12 +#define RT5660_G_DACl_SM_MASK (0x3 << 10) +#define RT5660_G_DACl_SM_SFT 10 +#define RT5660_G_DACR_SM_MASK (0x3 << 8) +#define RT5660_G_DACR_SM_SFT 8 +#define RT5660_G_OM_L_SM_MASK (0x3 << 6) +#define RT5660_G_OM_L_SM_SFT 6 +#define RT5660_M_DACR_SM (0x1 << 5) +#define RT5660_M_DACR_SM_SFT 5 +#define RT5660_M_BST1_SM (0x1 << 4) +#define RT5660_M_BST1_SM_SFT 4 +#define RT5660_M_BST3_SM (0x1 << 3) +#define RT5660_M_BST3_SM_SFT 3 +#define RT5660_M_DACL_SM (0x1 << 2) +#define RT5660_M_DACL_SM_SFT 2 +#define RT5660_M_OM_L_SM (0x1 << 1) +#define RT5660_M_OM_L_SM_SFT 1 + +/* SPOMIX Control (0x48) */ +#define RT5660_M_DAC_R_SPM (0x1 << 14) +#define RT5660_M_DAC_R_SPM_SFT 14 +#define RT5660_M_DAC_L_SPM (0x1 << 13) +#define RT5660_M_DAC_L_SPM_SFT 13 +#define RT5660_M_SV_SPM (0x1 << 12) +#define RT5660_M_SV_SPM_SFT 12 +#define RT5660_M_BST1_SPM (0x1 << 11) +#define RT5660_M_BST1_SPM_SFT 11 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5660_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5660_G_BST3_OM_L_SFT 13 +#define RT5660_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5660_G_BST2_OM_L_SFT 10 +#define RT5660_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5660_G_BST1_OM_L_SFT 7 +#define RT5660_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5660_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5660_G_DAC_R1_OM_L_MASK (0x7 << 10) +#define RT5660_G_DAC_R1_OM_L_SFT 10 +#define RT5660_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5660_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5660_M_BST3_OM_L (0x1 << 5) +#define RT5660_M_BST3_OM_L_SFT 5 +#define RT5660_M_BST2_OM_L (0x1 << 4) +#define RT5660_M_BST2_OM_L_SFT 4 +#define RT5660_M_BST1_OM_L (0x1 << 3) +#define RT5660_M_BST1_OM_L_SFT 3 +#define RT5660_M_RM_L_OM_L (0x1 << 2) +#define RT5660_M_RM_L_OM_L_SFT 2 +#define RT5660_M_DAC_R_OM_L (0x1 << 1) +#define RT5660_M_DAC_R_OM_L_SFT 1 +#define RT5660_M_DAC_L_OM_L (0x1) +#define RT5660_M_DAC_L_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5660_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5660_G_BST2_OM_R_SFT 10 +#define RT5660_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5660_G_BST1_OM_R_SFT 7 +#define RT5660_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5660_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5660_G_DAC_L_OM_R_MASK (0x7 << 10) +#define RT5660_G_DAC_L_OM_R_SFT 10 +#define RT5660_G_DAC_R_OM_R_MASK (0x7 << 7) +#define RT5660_G_DAC_R_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5660_M_BST2_OM_R (0x1 << 4) +#define RT5660_M_BST2_OM_R_SFT 4 +#define RT5660_M_BST1_OM_R (0x1 << 3) +#define RT5660_M_BST1_OM_R_SFT 3 +#define RT5660_M_RM_R_OM_R (0x1 << 2) +#define RT5660_M_RM_R_OM_R_SFT 2 +#define RT5660_M_DAC_L_OM_R (0x1 << 1) +#define RT5660_M_DAC_L_OM_R_SFT 1 +#define RT5660_M_DAC_R_OM_R (0x1) +#define RT5660_M_DAC_R_OM_R_SFT 0 + +/* Power Management for Digital 1 (0x61) */ +#define RT5660_PWR_I2S1 (0x1 << 15) +#define RT5660_PWR_I2S1_BIT 15 +#define RT5660_PWR_DAC_L1 (0x1 << 12) +#define RT5660_PWR_DAC_L1_BIT 12 +#define RT5660_PWR_DAC_R1 (0x1 << 11) +#define RT5660_PWR_DAC_R1_BIT 11 +#define RT5660_PWR_ADC_L (0x1 << 2) +#define RT5660_PWR_ADC_L_BIT 2 +#define RT5660_PWR_ADC_R (0x1 << 1) +#define RT5660_PWR_ADC_R_BIT 1 +#define RT5660_PWR_CLS_D (0x1) +#define RT5660_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5660_PWR_ADC_S1F (0x1 << 15) +#define RT5660_PWR_ADC_S1F_BIT 15 +#define RT5660_PWR_DAC_S1F (0x1 << 11) +#define RT5660_PWR_DAC_S1F_BIT 11 + +/* Power Management for Analog 1 (0x63) */ +#define RT5660_PWR_VREF1 (0x1 << 15) +#define RT5660_PWR_VREF1_BIT 15 +#define RT5660_PWR_FV1 (0x1 << 14) +#define RT5660_PWR_FV1_BIT 14 +#define RT5660_PWR_MB (0x1 << 13) +#define RT5660_PWR_MB_BIT 13 +#define RT5660_PWR_BG (0x1 << 11) +#define RT5660_PWR_BG_BIT 11 +#define RT5660_PWR_HP_L (0x1 << 7) +#define RT5660_PWR_HP_L_BIT 7 +#define RT5660_PWR_HP_R (0x1 << 6) +#define RT5660_PWR_HP_R_BIT 6 +#define RT5660_PWR_HA (0x1 << 5) +#define RT5660_PWR_HA_BIT 5 +#define RT5660_PWR_VREF2 (0x1 << 4) +#define RT5660_PWR_VREF2_BIT 4 +#define RT5660_PWR_FV2 (0x1 << 3) +#define RT5660_PWR_FV2_BIT 3 +#define RT5660_PWR_LDO2 (0x1 << 2) +#define RT5660_PWR_LDO2_BIT 2 + +/* Power Management for Analog 2 (0x64) */ +#define RT5660_PWR_BST1 (0x1 << 15) +#define RT5660_PWR_BST1_BIT 15 +#define RT5660_PWR_BST2 (0x1 << 14) +#define RT5660_PWR_BST2_BIT 14 +#define RT5660_PWR_BST3 (0x1 << 13) +#define RT5660_PWR_BST3_BIT 13 +#define RT5660_PWR_MB1 (0x1 << 11) +#define RT5660_PWR_MB1_BIT 11 +#define RT5660_PWR_MB2 (0x1 << 10) +#define RT5660_PWR_MB2_BIT 10 +#define RT5660_PWR_PLL (0x1 << 9) +#define RT5660_PWR_PLL_BIT 9 + +/* Power Management for Mixer (0x65) */ +#define RT5660_PWR_OM_L (0x1 << 15) +#define RT5660_PWR_OM_L_BIT 15 +#define RT5660_PWR_OM_R (0x1 << 14) +#define RT5660_PWR_OM_R_BIT 14 +#define RT5660_PWR_SM (0x1 << 13) +#define RT5660_PWR_SM_BIT 13 +#define RT5660_PWR_RM_L (0x1 << 11) +#define RT5660_PWR_RM_L_BIT 11 +#define RT5660_PWR_RM_R (0x1 << 10) +#define RT5660_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5660_PWR_SV (0x1 << 15) +#define RT5660_PWR_SV_BIT 15 +#define RT5660_PWR_LV_L (0x1 << 11) +#define RT5660_PWR_LV_L_BIT 11 +#define RT5660_PWR_LV_R (0x1 << 10) +#define RT5660_PWR_LV_R_BIT 10 + +/* I2S1 Audio Serial Data Port Control (0x70) */ +#define RT5660_I2S_MS_MASK (0x1 << 15) +#define RT5660_I2S_MS_SFT 15 +#define RT5660_I2S_MS_M (0x0 << 15) +#define RT5660_I2S_MS_S (0x1 << 15) +#define RT5660_I2S_O_CP_MASK (0x3 << 10) +#define RT5660_I2S_O_CP_SFT 10 +#define RT5660_I2S_O_CP_OFF (0x0 << 10) +#define RT5660_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5660_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5660_I2S_I_CP_MASK (0x3 << 8) +#define RT5660_I2S_I_CP_SFT 8 +#define RT5660_I2S_I_CP_OFF (0x0 << 8) +#define RT5660_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5660_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5660_I2S_BP_MASK (0x1 << 7) +#define RT5660_I2S_BP_SFT 7 +#define RT5660_I2S_BP_NOR (0x0 << 7) +#define RT5660_I2S_BP_INV (0x1 << 7) +#define RT5660_I2S_DL_MASK (0x3 << 2) +#define RT5660_I2S_DL_SFT 2 +#define RT5660_I2S_DL_16 (0x0 << 2) +#define RT5660_I2S_DL_20 (0x1 << 2) +#define RT5660_I2S_DL_24 (0x2 << 2) +#define RT5660_I2S_DL_8 (0x3 << 2) +#define RT5660_I2S_DF_MASK (0x3) +#define RT5660_I2S_DF_SFT 0 +#define RT5660_I2S_DF_I2S (0x0) +#define RT5660_I2S_DF_LEFT (0x1) +#define RT5660_I2S_DF_PCM_A (0x2) +#define RT5660_I2S_DF_PCM_B (0x3) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5660_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5660_I2S_BCLK_MS1_SFT 15 +#define RT5660_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5660_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5660_I2S_PD1_MASK (0x7 << 12) +#define RT5660_I2S_PD1_SFT 12 +#define RT5660_I2S_PD1_1 (0x0 << 12) +#define RT5660_I2S_PD1_2 (0x1 << 12) +#define RT5660_I2S_PD1_3 (0x2 << 12) +#define RT5660_I2S_PD1_4 (0x3 << 12) +#define RT5660_I2S_PD1_6 (0x4 << 12) +#define RT5660_I2S_PD1_8 (0x5 << 12) +#define RT5660_I2S_PD1_12 (0x6 << 12) +#define RT5660_I2S_PD1_16 (0x7 << 12) +#define RT5660_DAC_OSR_MASK (0x3 << 2) +#define RT5660_DAC_OSR_SFT 2 +#define RT5660_DAC_OSR_128 (0x0 << 2) +#define RT5660_DAC_OSR_64 (0x1 << 2) +#define RT5660_DAC_OSR_32 (0x2 << 2) +#define RT5660_DAC_OSR_16 (0x3 << 2) +#define RT5660_ADC_OSR_MASK (0x3) +#define RT5660_ADC_OSR_SFT 0 +#define RT5660_ADC_OSR_128 (0x0) +#define RT5660_ADC_OSR_64 (0x1) +#define RT5660_ADC_OSR_32 (0x2) +#define RT5660_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5660_RESET_ADF (0x1 << 13) +#define RT5660_RESET_ADF_SFT 13 +#define RT5660_RESET_DAF (0x1 << 12) +#define RT5660_RESET_DAF_SFT 12 +#define RT5660_DAHPF_EN (0x1 << 11) +#define RT5660_DAHPF_EN_SFT 11 +#define RT5660_ADHPF_EN (0x1 << 10) +#define RT5660_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5660_DMIC_1_EN_MASK (0x1 << 15) +#define RT5660_DMIC_1_EN_SFT 15 +#define RT5660_DMIC_1_DIS (0x0 << 15) +#define RT5660_DMIC_1_EN (0x1 << 15) +#define RT5660_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5660_DMIC_1L_LH_SFT 13 +#define RT5660_DMIC_1L_LH_RISING (0x0 << 13) +#define RT5660_DMIC_1L_LH_FALLING (0x1 << 13) +#define RT5660_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5660_DMIC_1R_LH_SFT 12 +#define RT5660_DMIC_1R_LH_RISING (0x0 << 12) +#define RT5660_DMIC_1R_LH_FALLING (0x1 << 12) +#define RT5660_SEL_DMIC_DATA_MASK (0x1 << 11) +#define RT5660_SEL_DMIC_DATA_SFT 11 +#define RT5660_SEL_DMIC_DATA_GPIO2 (0x0 << 11) +#define RT5660_SEL_DMIC_DATA_IN1P (0x1 << 11) +#define RT5660_DMIC_CLK_MASK (0x7 << 5) +#define RT5660_DMIC_CLK_SFT 5 + +/* Global Clock Control (0x80) */ +#define RT5660_SCLK_SRC_MASK (0x3 << 14) +#define RT5660_SCLK_SRC_SFT 14 +#define RT5660_SCLK_SRC_MCLK (0x0 << 14) +#define RT5660_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5660_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5660_PLL1_SRC_MASK (0x3 << 12) +#define RT5660_PLL1_SRC_SFT 12 +#define RT5660_PLL1_SRC_MCLK (0x0 << 12) +#define RT5660_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5660_PLL1_SRC_RCCLK (0x2 << 12) +#define RT5660_PLL1_PD_MASK (0x1 << 3) +#define RT5660_PLL1_PD_SFT 3 +#define RT5660_PLL1_PD_1 (0x0 << 3) +#define RT5660_PLL1_PD_2 (0x1 << 3) + +#define RT5660_PLL_INP_MAX 40000000 +#define RT5660_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5660_PLL_N_MAX 0x1ff +#define RT5660_PLL_N_MASK (RT5660_PLL_N_MAX << 7) +#define RT5660_PLL_N_SFT 7 +#define RT5660_PLL_K_MAX 0x1f +#define RT5660_PLL_K_MASK (RT5660_PLL_K_MAX) +#define RT5660_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5660_PLL_M_MAX 0xf +#define RT5660_PLL_M_MASK (RT5660_PLL_M_MAX << 12) +#define RT5660_PLL_M_SFT 12 +#define RT5660_PLL_M_BP (0x1 << 11) +#define RT5660_PLL_M_BP_SFT 11 + +/* Class D Over Current Control (0x8c) */ +#define RT5660_CLSD_OC_MASK (0x1 << 9) +#define RT5660_CLSD_OC_SFT 9 +#define RT5660_CLSD_OC_PU (0x0 << 9) +#define RT5660_CLSD_OC_PD (0x1 << 9) +#define RT5660_AUTO_PD_MASK (0x1 << 8) +#define RT5660_AUTO_PD_SFT 8 +#define RT5660_AUTO_PD_DIS (0x0 << 8) +#define RT5660_AUTO_PD_EN (0x1 << 8) +#define RT5660_CLSD_OC_TH_MASK (0x3f) +#define RT5660_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5660_CLSD_RATIO_MASK (0xf << 12) +#define RT5660_CLSD_RATIO_SFT 12 + +/* Lout Amp Control 1 (0x8e) */ +#define RT5660_LOUT_CO_MASK (0x1 << 4) +#define RT5660_LOUT_CO_SFT 4 +#define RT5660_LOUT_CO_DIS (0x0 << 4) +#define RT5660_LOUT_CO_EN (0x1 << 4) +#define RT5660_LOUT_CB_MASK (0x1) +#define RT5660_LOUT_CB_SFT 0 +#define RT5660_LOUT_CB_PD (0x0) +#define RT5660_LOUT_CB_PU (0x1) + +/* SPKVDD detection control (0x92) */ +#define RT5660_SPKVDD_DET_MASK (0x1 << 15) +#define RT5660_SPKVDD_DET_SFT 15 +#define RT5660_SPKVDD_DET_DIS (0x0 << 15) +#define RT5660_SPKVDD_DET_EN (0x1 << 15) +#define RT5660_SPK_AG_MASK (0x1 << 14) +#define RT5660_SPK_AG_SFT 14 +#define RT5660_SPK_AG_DIS (0x0 << 14) +#define RT5660_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5660_MIC1_BS_MASK (0x1 << 15) +#define RT5660_MIC1_BS_SFT 15 +#define RT5660_MIC1_BS_9AV (0x0 << 15) +#define RT5660_MIC1_BS_75AV (0x1 << 15) +#define RT5660_MIC2_BS_MASK (0x1 << 14) +#define RT5660_MIC2_BS_SFT 14 +#define RT5660_MIC2_BS_9AV (0x0 << 14) +#define RT5660_MIC2_BS_75AV (0x1 << 14) +#define RT5660_MIC1_OVCD_MASK (0x1 << 11) +#define RT5660_MIC1_OVCD_SFT 11 +#define RT5660_MIC1_OVCD_DIS (0x0 << 11) +#define RT5660_MIC1_OVCD_EN (0x1 << 11) +#define RT5660_MIC1_OVTH_MASK (0x3 << 9) +#define RT5660_MIC1_OVTH_SFT 9 +#define RT5660_MIC1_OVTH_600UA (0x0 << 9) +#define RT5660_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5660_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5660_MIC2_OVCD_MASK (0x1 << 8) +#define RT5660_MIC2_OVCD_SFT 8 +#define RT5660_MIC2_OVCD_DIS (0x0 << 8) +#define RT5660_MIC2_OVCD_EN (0x1 << 8) +#define RT5660_MIC2_OVTH_MASK (0x3 << 6) +#define RT5660_MIC2_OVTH_SFT 6 +#define RT5660_MIC2_OVTH_600UA (0x0 << 6) +#define RT5660_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5660_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5660_PWR_CLK25M_MASK (0x1 << 4) +#define RT5660_PWR_CLK25M_SFT 4 +#define RT5660_PWR_CLK25M_PD (0x0 << 4) +#define RT5660_PWR_CLK25M_PU (0x1 << 4) + +/* EQ Control 1 (0xb0) */ +#define RT5660_EQ_SRC_MASK (0x1 << 15) +#define RT5660_EQ_SRC_SFT 15 +#define RT5660_EQ_SRC_DAC (0x0 << 15) +#define RT5660_EQ_SRC_ADC (0x1 << 15) +#define RT5660_EQ_UPD (0x1 << 14) +#define RT5660_EQ_UPD_BIT 14 + +/* Jack Detect Control (0xbb) */ +#define RT5660_JD_MASK (0x3 << 14) +#define RT5660_JD_SFT 14 +#define RT5660_JD_DIS (0x0 << 14) +#define RT5660_JD_GPIO1 (0x1 << 14) +#define RT5660_JD_GPIO2 (0x2 << 14) +#define RT5660_JD_LOUT_MASK (0x1 << 11) +#define RT5660_JD_LOUT_SFT 11 +#define RT5660_JD_LOUT_DIS (0x0 << 11) +#define RT5660_JD_LOUT_EN (0x1 << 11) +#define RT5660_JD_LOUT_TRG_MASK (0x1 << 10) +#define RT5660_JD_LOUT_TRG_SFT 10 +#define RT5660_JD_LOUT_TRG_LO (0x0 << 10) +#define RT5660_JD_LOUT_TRG_HI (0x1 << 10) +#define RT5660_JD_SPO_MASK (0x1 << 9) +#define RT5660_JD_SPO_SFT 9 +#define RT5660_JD_SPO_DIS (0x0 << 9) +#define RT5660_JD_SPO_EN (0x1 << 9) +#define RT5660_JD_SPO_TRG_MASK (0x1 << 8) +#define RT5660_JD_SPO_TRG_SFT 8 +#define RT5660_JD_SPO_TRG_LO (0x0 << 8) +#define RT5660_JD_SPO_TRG_HI (0x1 << 8) + +/* IRQ Control 1 (0xbd) */ +#define RT5660_IRQ_JD_MASK (0x1 << 15) +#define RT5660_IRQ_JD_SFT 15 +#define RT5660_IRQ_JD_BP (0x0 << 15) +#define RT5660_IRQ_JD_NOR (0x1 << 15) +#define RT5660_IRQ_OT_MASK (0x1 << 14) +#define RT5660_IRQ_OT_SFT 14 +#define RT5660_IRQ_OT_BP (0x0 << 14) +#define RT5660_IRQ_OT_NOR (0x1 << 14) +#define RT5660_JD_STKY_MASK (0x1 << 13) +#define RT5660_JD_STKY_SFT 13 +#define RT5660_JD_STKY_DIS (0x0 << 13) +#define RT5660_JD_STKY_EN (0x1 << 13) +#define RT5660_OT_STKY_MASK (0x1 << 12) +#define RT5660_OT_STKY_SFT 12 +#define RT5660_OT_STKY_DIS (0x0 << 12) +#define RT5660_OT_STKY_EN (0x1 << 12) +#define RT5660_JD_P_MASK (0x1 << 11) +#define RT5660_JD_P_SFT 11 +#define RT5660_JD_P_NOR (0x0 << 11) +#define RT5660_JD_P_INV (0x1 << 11) +#define RT5660_OT_P_MASK (0x1 << 10) +#define RT5660_OT_P_SFT 10 +#define RT5660_OT_P_NOR (0x0 << 10) +#define RT5660_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5660_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5660_IRQ_MB1_OC_SFT 15 +#define RT5660_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5660_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5660_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5660_IRQ_MB2_OC_SFT 14 +#define RT5660_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5660_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5660_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5660_MB1_OC_STKY_SFT 11 +#define RT5660_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5660_MB1_OC_STKY_EN (0x1 << 11) +#define RT5660_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5660_MB2_OC_STKY_SFT 10 +#define RT5660_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5660_MB2_OC_STKY_EN (0x1 << 10) +#define RT5660_MB1_OC_P_MASK (0x1 << 7) +#define RT5660_MB1_OC_P_SFT 7 +#define RT5660_MB1_OC_P_NOR (0x0 << 7) +#define RT5660_MB1_OC_P_INV (0x1 << 7) +#define RT5660_MB2_OC_P_MASK (0x1 << 6) +#define RT5660_MB2_OC_P_SFT 6 +#define RT5660_MB2_OC_P_NOR (0x0 << 6) +#define RT5660_MB2_OC_P_INV (0x1 << 6) +#define RT5660_MB1_OC_CLR (0x1 << 3) +#define RT5660_MB1_OC_CLR_SFT 3 +#define RT5660_MB2_OC_CLR (0x1 << 2) +#define RT5660_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5660_GP2_PIN_MASK (0x1 << 14) +#define RT5660_GP2_PIN_SFT 14 +#define RT5660_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5660_GP2_PIN_DMIC1_SDA (0x1 << 14) +#define RT5660_GP1_PIN_MASK (0x3 << 12) +#define RT5660_GP1_PIN_SFT 12 +#define RT5660_GP1_PIN_GPIO1 (0x0 << 12) +#define RT5660_GP1_PIN_DMIC1_SCL (0x1 << 12) +#define RT5660_GP1_PIN_IRQ (0x2 << 12) +#define RT5660_GPIO_M_MASK (0x1 << 9) +#define RT5660_GPIO_M_SFT 9 +#define RT5660_GPIO_M_FLT (0x0 << 9) +#define RT5660_GPIO_M_PH (0x1 << 9) + +/* GPIO Control 3 (0xc2) */ +#define RT5660_GP2_PF_MASK (0x1 << 5) +#define RT5660_GP2_PF_SFT 5 +#define RT5660_GP2_PF_IN (0x0 << 5) +#define RT5660_GP2_PF_OUT (0x1 << 5) +#define RT5660_GP2_OUT_MASK (0x1 << 4) +#define RT5660_GP2_OUT_SFT 4 +#define RT5660_GP2_OUT_LO (0x0 << 4) +#define RT5660_GP2_OUT_HI (0x1 << 4) +#define RT5660_GP2_P_MASK (0x1 << 3) +#define RT5660_GP2_P_SFT 3 +#define RT5660_GP2_P_NOR (0x0 << 3) +#define RT5660_GP2_P_INV (0x1 << 3) +#define RT5660_GP1_PF_MASK (0x1 << 2) +#define RT5660_GP1_PF_SFT 2 +#define RT5660_GP1_PF_IN (0x0 << 2) +#define RT5660_GP1_PF_OUT (0x1 << 2) +#define RT5660_GP1_OUT_MASK (0x1 << 1) +#define RT5660_GP1_OUT_SFT 1 +#define RT5660_GP1_OUT_LO (0x0 << 1) +#define RT5660_GP1_OUT_HI (0x1 << 1) +#define RT5660_GP1_P_MASK (0x1) +#define RT5660_GP1_P_SFT 0 +#define RT5660_GP1_P_NOR (0x0) +#define RT5660_GP1_P_INV (0x1) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5660_SV_MASK (0x1 << 15) +#define RT5660_SV_SFT 15 +#define RT5660_SV_DIS (0x0 << 15) +#define RT5660_SV_EN (0x1 << 15) +#define RT5660_SPO_SV_MASK (0x1 << 14) +#define RT5660_SPO_SV_SFT 14 +#define RT5660_SPO_SV_DIS (0x0 << 14) +#define RT5660_SPO_SV_EN (0x1 << 14) +#define RT5660_OUT_SV_MASK (0x1 << 12) +#define RT5660_OUT_SV_SFT 12 +#define RT5660_OUT_SV_DIS (0x0 << 12) +#define RT5660_OUT_SV_EN (0x1 << 12) +#define RT5660_ZCD_DIG_MASK (0x1 << 11) +#define RT5660_ZCD_DIG_SFT 11 +#define RT5660_ZCD_DIG_DIS (0x0 << 11) +#define RT5660_ZCD_DIG_EN (0x1 << 11) +#define RT5660_ZCD_MASK (0x1 << 10) +#define RT5660_ZCD_SFT 10 +#define RT5660_ZCD_PD (0x0 << 10) +#define RT5660_ZCD_PU (0x1 << 10) +#define RT5660_SV_DLY_MASK (0xf) +#define RT5660_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5660_ZCD_SPO_MASK (0x1 << 15) +#define RT5660_ZCD_SPO_SFT 15 +#define RT5660_ZCD_SPO_DIS (0x0 << 15) +#define RT5660_ZCD_SPO_EN (0x1 << 15) +#define RT5660_ZCD_OMR_MASK (0x1 << 8) +#define RT5660_ZCD_OMR_SFT 8 +#define RT5660_ZCD_OMR_DIS (0x0 << 8) +#define RT5660_ZCD_OMR_EN (0x1 << 8) +#define RT5660_ZCD_OML_MASK (0x1 << 7) +#define RT5660_ZCD_OML_SFT 7 +#define RT5660_ZCD_OML_DIS (0x0 << 7) +#define RT5660_ZCD_OML_EN (0x1 << 7) +#define RT5660_ZCD_SPM_MASK (0x1 << 6) +#define RT5660_ZCD_SPM_SFT 6 +#define RT5660_ZCD_SPM_DIS (0x0 << 6) +#define RT5660_ZCD_SPM_EN (0x1 << 6) +#define RT5660_ZCD_RMR_MASK (0x1 << 5) +#define RT5660_ZCD_RMR_SFT 5 +#define RT5660_ZCD_RMR_DIS (0x0 << 5) +#define RT5660_ZCD_RMR_EN (0x1 << 5) +#define RT5660_ZCD_RML_MASK (0x1 << 4) +#define RT5660_ZCD_RML_SFT 4 +#define RT5660_ZCD_RML_DIS (0x0 << 4) +#define RT5660_ZCD_RML_EN (0x1 << 4) + +/* General Control 1 (0xfa) */ +#define RT5660_PWR_VREF_HP (0x1 << 11) +#define RT5660_PWR_VREF_HP_SFT 11 +#define RT5660_DIG_GATE_CTRL (0x1) +#define RT5660_DIG_GATE_CTRL_SFT 0 + +/* System Clock Source */ +#define RT5660_SCLK_S_MCLK 0 +#define RT5660_SCLK_S_PLL1 1 +#define RT5660_SCLK_S_RCCLK 2 + +/* PLL1 Source */ +#define RT5660_PLL1_S_MCLK 0 +#define RT5660_PLL1_S_BCLK 1 + +enum { + RT5660_AIF1, + RT5660_AIFS, +}; + +struct rt5660_priv { + struct snd_soc_codec *codec; + struct rt5660_platform_data pdata; + struct regmap *regmap; + struct clk *mclk; + + int sysclk; + int sysclk_src; + int lrck[RT5660_AIFS]; + int bclk[RT5660_AIFS]; + int master[RT5660_AIFS]; + + int pll_src; + int pll_in; + int pll_out; +}; + +#endif -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [v7 PATCH] amidi: add sysex-interval option
by Felipe F. Tonello 14 Sep '16

14 Sep '16

This patch adds a new option to amidi tool: sysex-interval. It adds a delay (in milliseconds) in between each SysEx message - it searches for a 0xF7 byte. This is very useful when sending firmware updates to a remote device via SysEx or any other use that requires this delay in between SysEx messages. `amidi' manual was updated with an example usage as well. Signed-off-by: Felipe F. Tonello <eu(a)felipetonello.com> --- amidi/amidi.1 | 16 ++++++++++- amidi/amidi.c | 88 +++++++++++++++++++++++++++++++++++++++++++++++------------ 2 files changed, 86 insertions(+), 18 deletions(-) diff --git a/amidi/amidi.1 b/amidi/amidi.1 index 86beb27a1e3b..5bc24ba3ad54 100644 --- a/amidi/amidi.1 +++ b/amidi/amidi.1 @@ -1,4 +1,4 @@ -.TH AMIDI 1 "16 Apr 2016" +.TH AMIDI 1 "30 Aug 2016" .SH NAME amidi \- read from and write to ALSA RawMIDI ports @@ -120,6 +120,12 @@ received MIDI commands. Does not ignore Clock bytes (F8h) when saving or printing received MIDI commands. +.TP +.I \-i, \-\-sysex-interval=mseconds +Adds a delay in between each SysEx message sent to a device. It is +useful when sending firmware updates via SysEx messages to a remote +device. + .SH EXAMPLES .TP @@ -130,6 +136,14 @@ to port .I hw:0. .TP +.B amidi \-p hw:1,0,0 -s firmware.syx \-i 100 +will send the MIDI commands in +.I firmware.syx +to port +.I hw:1,0,0 +with 100 milliseconds delay in between each SysEx message. + +.TP .B amidi \-S 'F0 43 10 4C 00 00 7E 00 F7' sends an XG Reset to the default port. diff --git a/amidi/amidi.c b/amidi/amidi.c index c20512cc96a7..a8264f181cf3 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -52,6 +52,7 @@ static int receive_file; static int dump; static float timeout; static int stop; +static int sysex_interval; static snd_rawmidi_t *input, **inputp; static snd_rawmidi_t *output, **outputp; @@ -70,19 +71,20 @@ static void usage(void) printf( "Usage: amidi options\n" "\n" - "-h, --help this help\n" - "-V, --version print current version\n" - "-l, --list-devices list all hardware ports\n" - "-L, --list-rawmidis list all RawMIDI definitions\n" - "-p, --port=name select port by name\n" - "-s, --send=file send the contents of a (.syx) file\n" - "-r, --receive=file write received data into a file\n" - "-S, --send-hex=\"...\" send hexadecimal bytes\n" - "-d, --dump print received data as hexadecimal bytes\n" - "-t, --timeout=seconds exits when no data has been received\n" - " for the specified duration\n" - "-a, --active-sensing include active sensing bytes\n" - "-c, --clock include clock bytes\n"); + "-h, --help this help\n" + "-V, --version print current version\n" + "-l, --list-devices list all hardware ports\n" + "-L, --list-rawmidis list all RawMIDI definitions\n" + "-p, --port=name select port by name\n" + "-s, --send=file send the contents of a (.syx) file\n" + "-r, --receive=file write received data into a file\n" + "-S, --send-hex=\"...\" send hexadecimal bytes\n" + "-d, --dump print received data as hexadecimal bytes\n" + "-t, --timeout=seconds exits when no data has been received\n" + " for the specified duration\n" + "-a, --active-sensing include active sensing bytes\n" + "-c, --clock include clock bytes\n" + "-i, --sysex-interval=mseconds delay in between each SysEx message\n"); } static void version(void) @@ -230,6 +232,47 @@ static void rawmidi_list(void) snd_output_close(output); } +static int send_midi_interleaved(void) +{ + int err; + char *data = send_data; + size_t buffer_size; + snd_rawmidi_params_t *param; + snd_rawmidi_status_t *st; + + snd_rawmidi_status_alloca(&st); + + snd_rawmidi_params_alloca(&param); + snd_rawmidi_params_current(output, param); + buffer_size = snd_rawmidi_params_get_buffer_size(param); + + while (data < (send_data + send_data_length)) { + int len = send_data + send_data_length - data; + char *temp; + + if (data > send_data) { + snd_rawmidi_status(output, st); + do { + /* 320 µs per byte as noted in Page 1 of MIDI spec */ + usleep((buffer_size - snd_rawmidi_status_get_avail(st)) * 320); + snd_rawmidi_status(output, st); + } while(snd_rawmidi_status_get_avail(st) < buffer_size); + usleep(sysex_interval * 1000); + } + + /* find end of SysEx */ + if ((temp = memchr(data, 0xf7, len)) != NULL) + len = temp - data + 1; + + if ((err = snd_rawmidi_write(output, data, len)) < 0) + return err; + + data += len; + } + + return 0; +} + static void load_file(void) { int fd; @@ -411,7 +454,7 @@ static void add_send_hex_data(const char *str) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlLp:s:r:S::dt:ac"; + static const char short_options[] = "hVlLp:s:r:S::dt:aci:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -425,6 +468,7 @@ int main(int argc, char *argv[]) {"timeout", 1, NULL, 't'}, {"active-sensing", 0, NULL, 'a'}, {"clock", 0, NULL, 'c'}, + {"sysex-interval", 1, NULL, 'i'}, { } }; int c, err, ok = 0; @@ -474,6 +518,9 @@ int main(int argc, char *argv[]) case 'c': ignore_clock = 0; break; + case 'i': + sysex_interval = atoi(optarg); + break; default: error("Try `amidi --help' for more information."); return 1; @@ -549,9 +596,16 @@ int main(int argc, char *argv[]) error("cannot set blocking mode: %s", snd_strerror(err)); goto _exit; } - if ((err = snd_rawmidi_write(output, send_data, send_data_length)) < 0) { - error("cannot send data: %s", snd_strerror(err)); - goto _exit; + if (!sysex_interval) { + if ((err = snd_rawmidi_write(output, send_data, send_data_length)) < 0) { + error("cannot send data: %s", snd_strerror(err)); + return err; + } + } else { + if ((err = send_midi_interleaved()) < 0) { + error("cannot send data: %s", snd_strerror(err)); + return err; + } } } -- 2.9.3

6 7

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Tony Lindgren 13 Sep '16

13 Sep '16

We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Let's fix the issue by adding PM QoS latency requirement for McBSP. Based on my experiments a working latency value is somewhere around 35 ms with 40 ms breaking. So let's use a safer value of 30 ms. Note that this is different from snd_pcm_hw_constraint_step() as that can configure things based on the buffer and period size. However, that does not help to block idle between the fifo fills. Note that in theory some omaps are capable of playing audio while hitting deeper idle states. If somebody needs that and gets it working, we can set the latency requirements accordingly. Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- sound/soc/omap/mcbsp.c | 10 ++++++++++ sound/soc/omap/mcbsp.h | 2 ++ 2 files changed, 12 insertions(+) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -643,6 +644,10 @@ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) int enable_srg = 0; u16 w; + /* Prevent omap hardware from hitting off between fifo fills */ + pm_qos_add_request(&mcbsp->pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + 30 * 1000); + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -731,6 +736,8 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); + + pm_qos_remove_request(&mcbsp->pm_qos_req); } int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) @@ -1098,6 +1105,9 @@ err_thres: void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (dev_pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 61e93b1..e603f33 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -325,6 +325,8 @@ struct omap_mcbsp { unsigned int in_freq; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, -- 2.9.3

4 27

[alsa-devel] [RFC][PATCH 0/4] ALSA: control: return payload length of TLV operation
by Takashi Sakamoto 13 Sep '16

13 Sep '16

Hi, ALSA control interface has a feature named TLV. This is originally designed to transfer information about threshold level. Later, it's used for I/O operation. In current implementation of this feature, length of handled data is not returned to applications. This is not better as a feature related to I/O. This patchset attempts to improve this feature so that the length is returned to application in a 'length' field of tlv packet structure (struct snd_ctl_tlv). Takashi Sakamoto (4): ALSA: control: return payload length for TLV operation ALSA: control: delegate checking the length of data payload to each drivers ALSA: control: add kerneldoc for snd_kcontrol_tlv_rw_t ALSA: control: bump up protocol version to 2.0.8 include/sound/control.h | 22 ++++++++++++++++++++-- include/sound/soc.h | 2 +- include/uapi/sound/asound.h | 2 +- sound/core/control.c | 30 +++++++++++++++++++----------- sound/core/pcm_lib.c | 15 ++++++++------- sound/core/vmaster.c | 2 +- sound/hda/hdmi_chmap.c | 16 +++++++++------- sound/pci/hda/hda_codec.c | 12 ++++++++---- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/lola/lola_mixer.c | 7 +++++-- sound/soc/soc-ops.c | 9 +++++---- sound/usb/mixer.c | 7 +++++-- sound/usb/mixer.h | 2 +- sound/usb/stream.c | 13 ++++++++----- 15 files changed, 94 insertions(+), 51 deletions(-) -- 2.7.4

5 47

[alsa-devel] [PATCH 00/13] ASoC: topology: Remaining kernel patches
by mengdong.lin＠linux.intel.com 09 Sep '16

09 Sep '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series contains all remaining kernel patches of topology, including some ABI update to PCM (FrontEnds) and link (BackEnds) objects. We don't user more series. User will be able to create BE DAIs and DAI links, configure more for FE links. Code are verified and can cover reqeust of Intel pre-release platforms for next year, so ABI should be stable. Current kernel topology code does not really touch Codec-Codec links since there is no user requst atm. We can add support for CC links later by reusing code and data structures for BE links, and no need to revise ABI. Guneshwor Singh (1): ASoC: topology: Add FE DAIs only if not already added Mengdong Lin (12): ASoC: topology: Able to create BE DAIs ASoC: topology: ABI - Add sig_bits to stream caps ASoC: topology: ABI - Define DPCM trigger ordering for PCM ASoC: topology: ABI - Add flags to PCM ASoC: topology: ABI - Add private data to PCM ASoC: topology: ABI - Add name & component info to BE/CC links ASoC: topology: ABI - Define DAI physical PCM data formats ASoC: topology: ABI - Add HW configurations to BE/CC links ASoC: topology: ABI - Add flags and private data to BE/CC links ASoC: Define API to find a dai link by id ASoC: Probe link components after finding new links ASoC: topology: Able to create BE DAI links include/sound/soc-dai.h | 15 +- include/sound/soc-dpcm.h | 13 -- include/sound/soc-topology.h | 1 + include/sound/soc.h | 2 + include/uapi/sound/asoc.h | 106 ++++++++++++- sound/soc/soc-core.c | 36 ++++- sound/soc/soc-topology.c | 359 ++++++++++++++++++++++++++++++++++++++----- 7 files changed, 466 insertions(+), 66 deletions(-) -- 2.5.0

3 25

[alsa-devel] [PATCH] ASoC: sgtl5000: add headphone and LINEOUT mute controls
by Richard Leitner 09 Sep '16

09 Sep '16

These controls mute/unmute the LINEOUT and headphone outputs of SGTL5000 using its CHIP_ANA_CTRL register. Signed-off-by: Richard Leitner <dev(a)g0hl1n.net> --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 527b759..a6af65d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -411,6 +411,8 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 0, 8, 0x7f, 1, headphone_volume), + SOC_SINGLE("Headphone Playback Switch", SGTL5000_CHIP_ANA_CTRL, + 4, 1, 0), SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL, 5, 1, 0), @@ -423,6 +425,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT, 0x1f, 1, lineout_volume), + SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 0), }; /* mute the codec used by alsa core */ -- 2.9.3

2 3

[alsa-devel] [alsa-lib][PATCH] test: fix comment about first two fields of TLV data payload
by Takashi Sakamoto 03 Sep '16

03 Sep '16

I misunderstand TLV packet structure (struct snd_ctl_tlv) nests in its payload. But actually, the payload stores different type of data. This commit corrects a comment in a test program of user control element set including my misunderstanding. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/user-ctl-element-set.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index 9b9dc59..f6f050a 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -418,11 +418,11 @@ static int check_tlv(struct elem_set_trial *trial) int err; /* - * See a layout of 'struct snd_ctl_tlv'. I don't know the reason to - * construct this buffer with the same layout. It should be abstracted - * inner userspace library... + * When transferring threshold level information via TLV feature of + * ALSA control interface, the first two fields of packet payload + * consist of data type and data length. */ - orig[0] = snd_ctl_elem_id_get_numid(trial->id); + orig[0] = SNDRV_CTL_TLVT_CONTAINER; orig[1] = 6 * sizeof(orig[0]); orig[2] = 'a'; orig[3] = 'b'; -- 2.7.4

3 7

[alsa-devel] [PATCH] clk: x86: Add Atom PMC platform clocks
by Pierre-Louis Bossart 03 Sep '16

03 Sep '16

From: Irina Tirdea <irina.tirdea(a)intel.com> The Baytral and CherryTrail platforms provide platform clocks through their Power Management Controller (PMC). The SoC supports up to 6 clocks (PMC_PLT_CLK[5:0]) with a frequency of either 19.2 MHz (PLL) or 25 MHz (XTAL). These clocks are available for general system use, where appropriate and each have Control & Frequency register fields associated with them. For example, the usage for platform clocks could be: PLT_CLK[2:0] - Camera PLT_CLK[3] - Audio Codec PLT_CLK[4] - PLT_CLK[5] - COMMs Signed-off-by: Irina Tirdea <irina.tirdea(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Notes: Submitting this patch through the clock tree as requested by Mark Brown. This patch specifically enables the audio MCLK required by Baytrail CR devices (support already merged in Mark's tree) arch/x86/Kconfig | 1 + arch/x86/include/asm/pmc_atom.h | 6 + arch/x86/platform/atom/pmc_atom.c | 55 ++++- drivers/clk/x86/Makefile | 1 + drivers/clk/x86/clk-byt-plt.c | 413 ++++++++++++++++++++++++++++++++++++++ 5 files changed, 473 insertions(+), 3 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c diff --git a/arch/x86/Kconfig b/arch/x86/Kconfig index c580d8c..8bd50e6 100644 --- a/arch/x86/Kconfig +++ b/arch/x86/Kconfig @@ -2745,6 +2745,7 @@ config X86_DMA_REMAP config PMC_ATOM def_bool y depends on PCI + select COMMON_CLK config VMD depends on PCI_MSI diff --git a/arch/x86/include/asm/pmc_atom.h b/arch/x86/include/asm/pmc_atom.h index aa8744c..59f9516 100644 --- a/arch/x86/include/asm/pmc_atom.h +++ b/arch/x86/include/asm/pmc_atom.h @@ -152,6 +152,12 @@ #define SLEEP_TYPE_S5 0x1C00 #define SLEEP_ENABLE 0x2000 +struct pmc_clk { + const char *name; + unsigned long freq; + const char *parent_name; +}; + extern int pmc_atom_read(int offset, u32 *value); extern int pmc_atom_write(int offset, u32 value); diff --git a/arch/x86/platform/atom/pmc_atom.c b/arch/x86/platform/atom/pmc_atom.c index 964ff4f..e2de3e2 100644 --- a/arch/x86/platform/atom/pmc_atom.c +++ b/arch/x86/platform/atom/pmc_atom.c @@ -21,6 +21,7 @@ #include <linux/debugfs.h> #include <linux/seq_file.h> #include <linux/io.h> +#include <linux/platform_device.h> #include <asm/pmc_atom.h> @@ -37,6 +38,11 @@ struct pmc_reg_map { const struct pmc_bit_map *pss; }; +struct pmc_data { + const struct pmc_reg_map *map; + const struct pmc_clk *clks; +}; + struct pmc_dev { u32 base_addr; void __iomem *regmap; @@ -50,6 +56,29 @@ struct pmc_dev { static struct pmc_dev pmc_device; static u32 acpi_base_addr; +static const struct pmc_clk byt_clks[] = { + { + .name = "xtal", + .freq = 25000000, + .parent_name = NULL, + }, + { + .name = "pll", + .freq = 19200000, + .parent_name = "xtal", + }, + {}, +}; + +static const struct pmc_clk cht_clks[] = { + { + .name = "xtal", + .freq = 19200000, + .parent_name = NULL, + }, + {}, +}; + static const struct pmc_bit_map d3_sts_0_map[] = { {"LPSS1_F0_DMA", BIT_LPSS1_F0_DMA}, {"LPSS1_F1_PWM1", BIT_LPSS1_F1_PWM1}, @@ -169,6 +198,16 @@ static const struct pmc_reg_map cht_reg_map = { .pss = cht_pss_map, }; +static const struct pmc_data byt_data = { + .map = &byt_reg_map, + .clks = byt_clks, +}; + +static const struct pmc_data cht_data = { + .map = &cht_reg_map, + .clks = cht_clks, +}; + static inline u32 pmc_reg_read(struct pmc_dev *pmc, int reg_offset) { return readl(pmc->regmap + reg_offset); @@ -384,8 +423,11 @@ static int pmc_dbgfs_register(struct pmc_dev *pmc) static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) { + struct platform_device *clkdev; struct pmc_dev *pmc = &pmc_device; - const struct pmc_reg_map *map = (struct pmc_reg_map *)ent->driver_data; + const struct pmc_data *data = (struct pmc_data *)ent->driver_data; + const struct pmc_reg_map *map = data->map; + const struct pmc_clk *clks = data->clks; int ret; /* Obtain ACPI base address */ @@ -414,6 +456,13 @@ static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) if (ret) dev_warn(&pdev->dev, "debugfs register failed\n"); + /* Register platform clocks - PMC_PLT_CLK [5:0] */ + clkdev = platform_device_register_data(NULL, "clk-byt-plt", -1, + &clks, sizeof(clks)); + if (IS_ERR(clkdev)) + dev_warn(&pdev->dev, "platform clocks register failed: %ld\n", + PTR_ERR(clkdev)); + pmc->init = true; return ret; } @@ -424,8 +473,8 @@ static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) * used by pci_match_id() call below. */ static const struct pci_device_id pmc_pci_ids[] = { - { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_VLV_PMC), (kernel_ulong_t)&byt_reg_map }, - { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_CHT_PMC), (kernel_ulong_t)&cht_reg_map }, + { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_VLV_PMC), (kernel_ulong_t)&byt_data }, + { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_CHT_PMC), (kernel_ulong_t)&cht_data }, { 0, }, }; diff --git a/drivers/clk/x86/Makefile b/drivers/clk/x86/Makefile index 0478138..cbdc8cc 100644 --- a/drivers/clk/x86/Makefile +++ b/drivers/clk/x86/Makefile @@ -1,2 +1,3 @@ clk-x86-lpss-objs := clk-lpt.o obj-$(CONFIG_X86_INTEL_LPSS) += clk-x86-lpss.o +obj-$(CONFIG_PMC_ATOM) += clk-byt-plt.o diff --git a/drivers/clk/x86/clk-byt-plt.c b/drivers/clk/x86/clk-byt-plt.c new file mode 100644 index 0000000..330cd35 --- /dev/null +++ b/drivers/clk/x86/clk-byt-plt.c @@ -0,0 +1,413 @@ +/* + * Intel Atom platform clocks driver for Baytrail and CherryTrail SoC. + * + * Copyright (C) 2016, Intel Corporation + * Author: Irina Tirdea <irina.tirdea(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#include <linux/clk-provider.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/clkdev.h> + +#include <asm/pmc_atom.h> + +#define PLT_CLK_NAME_BASE "pmc_plt_clk_" +#define PLT_CLK_DRIVER_NAME "clk-byt-plt" + +#define PMC_CLK_CTL_0 0x60 +#define PMC_CLK_CTL_SIZE 4 +#define PMC_CLK_NUM 6 +#define PMC_MASK_CLK_CTL GENMASK(1, 0) +#define PMC_MASK_CLK_FREQ BIT(2) +#define PMC_CLK_CTL_GATED_ON_D3 0x0 +#define PMC_CLK_CTL_FORCE_ON 0x1 +#define PMC_CLK_CTL_FORCE_OFF 0x2 +#define PMC_CLK_CTL_RESERVED 0x3 +#define PMC_CLK_FREQ_XTAL 0x0 /* 25 MHz */ +#define PMC_CLK_FREQ_PLL 0x4 /* 19.2 MHz */ + +struct clk_plt_fixed { + struct clk *clk; + struct clk_lookup *lookup; +}; + +struct clk_plt { + struct clk_hw hw; + u8 id; + u32 offset; + struct clk_lookup *lookup; + spinlock_t lock; +}; + +#define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) + +struct clk_plt_data { + struct clk_plt_fixed **parents; + u8 nparents; + struct clk *clks[PMC_CLK_NUM]; +}; + +static inline int plt_reg_to_parent(int reg) +{ + switch (reg & PMC_MASK_CLK_FREQ) { + case PMC_CLK_FREQ_XTAL: + return 0; /* index 0 in parents[] */ + case PMC_CLK_FREQ_PLL: + return 1; /* index 1 in parents[] */ + } + + return 0; +} + +static inline int plt_parent_to_reg(int index) +{ + switch (index) { + case 0: /* index 0 in parents[] */ + return PMC_CLK_FREQ_XTAL; + case 1: /* index 0 in parents[] */ + return PMC_CLK_FREQ_PLL; + } + + return PMC_CLK_FREQ_XTAL; +} + +static inline int plt_reg_to_enabled(int reg) +{ + switch (reg & PMC_MASK_CLK_CTL) { + case PMC_CLK_CTL_GATED_ON_D3: + case PMC_CLK_CTL_FORCE_ON: + return 1; /* enabled */ + case PMC_CLK_CTL_FORCE_OFF: + case PMC_CLK_CTL_RESERVED: + default: + return 0; /* disabled */ + } +} + +static int plt_pmc_atom_update(struct clk_plt *clk, u32 mask, u32 val) +{ + int ret; + u32 orig, tmp; + unsigned long flags = 0; + + spin_lock_irqsave(&clk->lock, flags); + + ret = pmc_atom_read(clk->offset, &orig); + if (ret) + goto out; + + tmp = orig & ~mask; + tmp |= val & mask; + + if (tmp == orig) + goto out; + + ret = pmc_atom_write(clk->offset, tmp); + if (ret) + goto out; + +out: + spin_unlock_irqrestore(&clk->lock, flags); + + return ret; +} + +static int plt_clk_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_plt *clk = to_clk_plt(hw); + + return plt_pmc_atom_update(clk, PMC_MASK_CLK_FREQ, + plt_parent_to_reg(index)); +} + +static u8 plt_clk_get_parent(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + u32 value; + int ret; + + ret = pmc_atom_read(clk->offset, &value); + if (ret) + return ret; + + return plt_reg_to_parent(value); +} + +static int plt_clk_enable(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + + return plt_pmc_atom_update(clk, PMC_MASK_CLK_CTL, PMC_CLK_CTL_FORCE_ON); +} + +static void plt_clk_disable(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + + plt_pmc_atom_update(clk, PMC_MASK_CLK_CTL, PMC_CLK_CTL_FORCE_OFF); +} + +static int plt_clk_is_enabled(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + u32 value; + int ret; + + ret = pmc_atom_read(clk->offset, &value); + if (ret) + return ret; + + return plt_reg_to_enabled(value); +} + +static const struct clk_ops plt_clk_ops = { + .enable = plt_clk_enable, + .disable = plt_clk_disable, + .is_enabled = plt_clk_is_enabled, + .get_parent = plt_clk_get_parent, + .set_parent = plt_clk_set_parent, + .determine_rate = __clk_mux_determine_rate, +}; + +static struct clk *plt_clk_register(struct platform_device *pdev, int id, + const char **parent_names, int num_parents) +{ + struct clk_plt *pclk; + struct clk *clk; + struct clk_init_data init; + int ret = 0; + + pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); + if (!pclk) + return ERR_PTR(-ENOMEM); + + init.name = kasprintf(GFP_KERNEL, "%s%d", PLT_CLK_NAME_BASE, id); + init.ops = &plt_clk_ops; + init.flags = 0; + init.parent_names = parent_names; + init.num_parents = num_parents; + + pclk->hw.init = &init; + pclk->id = id; + pclk->offset = PMC_CLK_CTL_0 + id * PMC_CLK_CTL_SIZE; + spin_lock_init(&pclk->lock); + + clk = clk_register(&pdev->dev, &pclk->hw); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto err_free_pclk; + } + + pclk->lookup = clkdev_create(clk, init.name, NULL); + if (!pclk->lookup) { + ret = -ENOMEM; + goto err_clk_unregister; + } + + kfree(init.name); + + return clk; + +err_clk_unregister: + clk_unregister(clk); +err_free_pclk: + kfree(init.name); + return ERR_PTR(ret); +} + +static void plt_clk_unregister(struct clk *clk) +{ + struct clk_plt *pclk; + struct clk_hw *hw; + + hw = __clk_get_hw(clk); + if (!hw) + return; + + pclk = to_clk_plt(hw); + + clkdev_drop(pclk->lookup); + clk_unregister(clk); +} + +static struct clk_plt_fixed *plt_clk_register_fixed_rate(struct platform_device *pdev, + const char *name, + const char *parent_name, + unsigned long fixed_rate) +{ + struct clk_plt_fixed *pclk; + int ret = 0; + + pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); + if (!pclk) + return ERR_PTR(-ENOMEM); + + pclk->clk = clk_register_fixed_rate(&pdev->dev, name, parent_name, + 0, fixed_rate); + if (IS_ERR(pclk->clk)) { + ret = PTR_ERR(pclk->clk); + return ERR_PTR(ret); + } + + pclk->lookup = clkdev_create(pclk->clk, name, NULL); + if (!pclk->lookup) { + ret = -ENOMEM; + goto err_clk_unregister; + } + + return pclk; + +err_clk_unregister: + clk_unregister_fixed_rate(pclk->clk); + return ERR_PTR(ret); +} + +static void plt_clk_unregister_fixed_rate(struct clk_plt_fixed *pclk) +{ + clkdev_drop(pclk->lookup); + clk_unregister_fixed_rate(pclk->clk); +} + +static const char **plt_clk_register_parents(struct platform_device *pdev, + struct clk_plt_data *data) +{ + struct pmc_clk **pclks, *clks; + const char **parent_names; + int i, err; + + data->nparents = 0; + pclks = dev_get_platdata(&pdev->dev); + if (!pclks) + return NULL; + + clks = *pclks; + while (clks[data->nparents].name) + data->nparents++; + + data->parents = devm_kzalloc(&pdev->dev, + sizeof(*data->parents) * data->nparents, + GFP_KERNEL); + if (!data->parents) { + err = -ENOMEM; + goto err_out; + } + + parent_names = kcalloc(data->nparents, sizeof(*parent_names), + GFP_KERNEL); + if (!parent_names) { + err = -ENOMEM; + goto err_out; + } + + for (i = 0; i < data->nparents; i++) { + data->parents[i] = + plt_clk_register_fixed_rate(pdev, clks[i].name, + clks[i].parent_name, + clks[i].freq); + if (IS_ERR(data->parents[i])) { + err = PTR_ERR(data->parents[i]); + goto err_unreg; + } + parent_names[i] = kstrdup_const(clks[i].name, GFP_KERNEL); + } + + return parent_names; + +err_unreg: + for (i--; i >= 0; i--) { + plt_clk_unregister_fixed_rate(data->parents[i]); + kfree_const(parent_names[i]); + } + kfree(parent_names); +err_out: + data->nparents = 0; + return ERR_PTR(err); +} + +static void plt_clk_unregister_parents(struct clk_plt_data *data) +{ + int i; + + for (i = 0; i < data->nparents; i++) + plt_clk_unregister_fixed_rate(data->parents[i]); +} + +static int plt_clk_probe(struct platform_device *pdev) +{ + struct clk_plt_data *data; + int i, err; + const char **parent_names; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + parent_names = plt_clk_register_parents(pdev, data); + if (IS_ERR(parent_names)) + return PTR_ERR(parent_names); + + for (i = 0; i < PMC_CLK_NUM; i++) { + data->clks[i] = plt_clk_register(pdev, i, parent_names, + data->nparents); + if (IS_ERR(data->clks[i])) { + err = PTR_ERR(data->clks[i]); + goto err_unreg_clk_plt; + } + } + + for (i = 0; i < data->nparents; i++) + kfree_const(parent_names[i]); + kfree(parent_names); + + dev_set_drvdata(&pdev->dev, data); + return 0; + +err_unreg_clk_plt: + for (i--; i >= 0; i--) + plt_clk_unregister(data->clks[i]); + plt_clk_unregister_parents(data); + for (i = 0; i < data->nparents; i++) + kfree_const(parent_names[i]); + kfree(parent_names); + return err; +} + +static int plt_clk_remove(struct platform_device *pdev) +{ + struct clk_plt_data *data; + int i; + + data = dev_get_drvdata(&pdev->dev); + if (!data) + return 0; + + for (i = 0; i < PMC_CLK_NUM; i++) + plt_clk_unregister(data->clks[i]); + plt_clk_unregister_parents(data); + return 0; +} + +static struct platform_driver plt_clk_driver = { + .driver = { + .name = PLT_CLK_DRIVER_NAME, + }, + .probe = plt_clk_probe, + .remove = plt_clk_remove, +}; +module_platform_driver(plt_clk_driver); + +MODULE_DESCRIPTION("Intel Atom platform clocks driver"); +MODULE_AUTHOR("Irina Tirdea <irina.tirdea(a)intel.com>"); +MODULE_LICENSE("GPL v2"); -- 2.7.4

3 6

[alsa-devel] [PATCH 1/3] ALSA: timer: fix NULL pointer dereference in read()/ioctl() race
by Vegard Nossum 02 Sep '16

02 Sep '16

I got this with syzkaller: ================================================================== BUG: KASAN: null-ptr-deref on address 0000000000000020 Read of size 32 by task syz-executor/22519 CPU: 1 PID: 22519 Comm: syz-executor Not tainted 4.8.0-rc2+ #169 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.9.3-0-ge2fc41e-prebuilt.qemu-project.org 04/01/2 014 0000000000000001 ffff880111a17a00 ffffffff81f9f141 ffff880111a17a90 ffff880111a17c50 ffff880114584a58 ffff880114584a10 ffff880111a17a80 ffffffff8161fe3f ffff880100000000 ffff880118d74a48 ffff880118d74a68 Call Trace: [<ffffffff81f9f141>] dump_stack+0x83/0xb2 [<ffffffff8161fe3f>] kasan_report_error+0x41f/0x4c0 [<ffffffff8161ff74>] kasan_report+0x34/0x40 [<ffffffff82c84b54>] ? snd_timer_user_read+0x554/0x790 [<ffffffff8161e79e>] check_memory_region+0x13e/0x1a0 [<ffffffff8161e9c1>] kasan_check_read+0x11/0x20 [<ffffffff82c84b54>] snd_timer_user_read+0x554/0x790 [<ffffffff82c84600>] ? snd_timer_user_info_compat.isra.5+0x2b0/0x2b0 [<ffffffff817d0831>] ? proc_fault_inject_write+0x1c1/0x250 [<ffffffff817d0670>] ? next_tgid+0x2a0/0x2a0 [<ffffffff8127c278>] ? do_group_exit+0x108/0x330 [<ffffffff8174653a>] ? fsnotify+0x72a/0xca0 [<ffffffff81674dfe>] __vfs_read+0x10e/0x550 [<ffffffff82c84600>] ? snd_timer_user_info_compat.isra.5+0x2b0/0x2b0 [<ffffffff81674cf0>] ? do_sendfile+0xc50/0xc50 [<ffffffff81745e10>] ? __fsnotify_update_child_dentry_flags+0x60/0x60 [<ffffffff8143fec6>] ? kcov_ioctl+0x56/0x190 [<ffffffff81e5ada2>] ? common_file_perm+0x2e2/0x380 [<ffffffff81746b0e>] ? __fsnotify_parent+0x5e/0x2b0 [<ffffffff81d93536>] ? security_file_permission+0x86/0x1e0 [<ffffffff816728f5>] ? rw_verify_area+0xe5/0x2b0 [<ffffffff81675355>] vfs_read+0x115/0x330 [<ffffffff81676371>] SyS_read+0xd1/0x1a0 [<ffffffff816762a0>] ? vfs_write+0x4b0/0x4b0 [<ffffffff82001c2c>] ? __this_cpu_preempt_check+0x1c/0x20 [<ffffffff8150455a>] ? __context_tracking_exit.part.4+0x3a/0x1e0 [<ffffffff816762a0>] ? vfs_write+0x4b0/0x4b0 [<ffffffff81005524>] do_syscall_64+0x1c4/0x4e0 [<ffffffff810052fc>] ? syscall_return_slowpath+0x16c/0x1d0 [<ffffffff83c3276a>] entry_SYSCALL64_slow_path+0x25/0x25 ================================================================== There are a couple of problems that I can see: - ioctl(SNDRV_TIMER_IOCTL_SELECT), which potentially sets tu->queue/tu->tqueue to NULL on memory allocation failure, so read() would get a NULL pointer dereference like the above splat - the same ioctl() can free tu->queue/to->tqueue which means read() could potentially see (and dereference) the freed pointer We can fix the NULL pointer dereference by not touching the pointers until we have allocated the new memory (similar to what is done in ioctl(SNDRV_TIMER_IOCTL_PARAMS)) and we can fix the use-after-free by taking the ioctl_lock mutex when dereferencing ->queue/->tqueue, since that's always held over all the ioctl() code. Just looking at the code I find it likely that there are more problems here such as tu->qhead pointing outside the buffer if the size is changed concurrently using SNDRV_TIMER_IOCTL_PARAMS. Signed-off-by: Vegard Nossum <vegard.nossum(a)oracle.com> --- sound/core/timer.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 9a6157e..7899e37 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1602,15 +1602,25 @@ static int snd_timer_user_tselect(struct file *file, kfree(tu->tqueue); tu->tqueue = NULL; if (tu->tread) { - tu->tqueue = kmalloc(tu->queue_size * sizeof(struct snd_timer_tread), + struct snd_timer_tread *ttr; + ttr = kmalloc(tu->queue_size * sizeof(struct snd_timer_tread), GFP_KERNEL); - if (tu->tqueue == NULL) + if (ttr) { + kfree(tu->tqueue); + tu->tqueue = ttr; + } else { err = -ENOMEM; + } } else { - tu->queue = kmalloc(tu->queue_size * sizeof(struct snd_timer_read), + struct snd_timer_read *tr; + tr = kmalloc(tu->queue_size * sizeof(struct snd_timer_read), GFP_KERNEL); - if (tu->queue == NULL) + if (tr) { + kfree(tu->queue); + tu->queue = tr; + } else { err = -ENOMEM; + } } if (err < 0) { @@ -1958,6 +1968,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, tu->qused--; spin_unlock_irq(&tu->qlock); + mutex_lock(&tu->ioctl_lock); if (tu->tread) { if (copy_to_user(buffer, &tu->tqueue[qhead], sizeof(struct snd_timer_tread))) @@ -1967,6 +1978,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, sizeof(struct snd_timer_read))) err = -EFAULT; } + mutex_unlock(&tu->ioctl_lock); spin_lock_irq(&tu->qlock); if (err < 0) -- 2.10.0.rc0.1.g07c9292

2 8

[alsa-devel] [PATCH] ASoC: arizona: Wait for resume before enabling FLL
by Richard Fitzgerald 01 Sep '16

01 Sep '16

When enabling an FLL use pm_runtime_get_sync() instead of pm_runtime_get() to ensure that all the register settings have been written out and the codec is powered-up before we write the enable bit. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ecfdbfc..8b8f2d2 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2268,7 +2268,7 @@ static int arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_BW); if (!already_enabled) - pm_runtime_get(arizona->dev); + pm_runtime_get_sync(arizona->dev); regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rockchip: use SPI dependency for rt5514
by Arnd Bergmann 01 Sep '16

01 Sep '16

The rk3399 scans the spi_bus_type to find the rt5514 driver, but does not actually have a Kconfig dependency on SPI, so we can end up with a link failure: sound/soc/codecs/snd-soc-rt5514-spi.o: In function `rt5514_spi_driver_init': rt5514-spi.c:(.init.text+0x14): undefined reference to `__spi_register_driver' sound/soc/codecs/snd-soc-rt5514-spi.o: In function `rt5514_spi_burst_read': rt5514-spi.c:(.text.rt5514_spi_burst_read+0x18c): undefined reference to `spi_sync' sound/soc/codecs/snd-soc-rt5514-spi.o: In function `rt5514_spi_burst_write': rt5514-spi.c:(.text.rt5514_spi_burst_write+0x1b4): undefined reference to `spi_sync' sound/soc/rockchip/snd-soc-rk3399-gru-sound.o: In function `rockchip_sound_probe': rk3399_gru_sound.c:(.text.rockchip_sound_probe+0x128): undefined reference to `spi_bus_type' This adds the missing dependency. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: c6eac8a36a84 ("ASoC: rockchip: Add machine driver for RK3399 GRU Boards") --- sound/soc/rockchip/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index d82d763d4854..c783f9a22595 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -44,7 +44,7 @@ config SND_SOC_ROCKCHIP_RT5645 config SND_SOC_RK3399_GRU_SOUND tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP && SPI select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98357A select SND_SOC_RT5514 -- 2.9.0

2 1

[alsa-devel] [PATCH v9 0/2] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 01 Sep '16

01 Sep '16

This patch adds support for some mixer controls: - Line-In - FM-In - Mic 2 - Capture Source - Differential Line-In v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (2): ASoC: sun4i-codec: Distinguish sun4i from sun7i Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 278 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 259 insertions(+), 19 deletions(-)

3 15

[alsa-devel] CIP Discontinuity with snd-firewire-digi00x
by lin-alsa＠noblejury.com 01 Sep '16

01 Sep '16

All, I've been working to set up a Digi002 Rack mixer with Jack using the Alsa backend. Unfortunately, the mixer seems to desync or something after a few minutes - hours. dmesg gives me `snd-firewire-digi00x fw1.0: Detect discontinuity of CIP: XX XX` with XX XX replaced with various hex pairs. I'm not sure how to debug this, and any help would be much appreciated. Also, cat /proc/asound/card1/firewire/clock gives `Optical mode: adat Sampling Rate: 88200 Clock Source: internal` so I don't think it's an external clock issue. Lin

2 7

[alsa-devel] [PATCH] ASoC: constify snd_soc_codec_driver structures
by Julia Lawall 01 Sep '16

01 Sep '16

Check for snd_soc_codec_driver structures that are only passed to snd_soc_register_codec or memcpy (2nd arg), for which the corresponding parameters are declared const. Declare as const snd_soc_codec_driver structures that have these properties. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r disable optional_qualifier@ identifier i; position p; @@ static struct snd_soc_codec_driver i@p = { ... }; @ok@ identifier r.i; expression e1,e2,e3; position p; @@ ( snd_soc_register_codec(e1,&i@p,e2,e3) | memcpy(e1,&i@p,e2) ) @bad@ position p != {r.p,ok.p}; identifier r.i; @@ i@p @depends on !bad disable optional_qualifier@ identifier r.i; @@ static +const struct snd_soc_codec_driver i = { ... }; // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/codecs/wm0010.c | 2 +- sound/soc/codecs/wm1250-ev1.c | 2 +- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8727.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8782.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- 48 files changed, 48 insertions(+), 48 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 4ef6919..0eb5dcf 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -789,7 +789,7 @@ static int wm0010_set_sysclk(struct snd_soc_codec *codec, int source, static int wm0010_probe(struct snd_soc_codec *codec); -static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .probe = wm0010_probe, .set_bias_level = wm0010_set_bias_level, .set_sysclk = wm0010_set_sysclk, diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index caf5012..cf5f058 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -141,7 +141,7 @@ static struct snd_soc_dai_driver wm1250_ev1_dai = { .ops = &wm1250_ev1_ops, }; -static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { .component_driver = { .dapm_widgets = wm1250_ev1_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm1250_ev1_dapm_widgets), diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 199a3d8..23cde3a 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -799,7 +799,7 @@ static int wm2000_remove(struct snd_soc_codec *codec) return wm2000_anc_transition(wm2000, ANC_OFF); } -static struct snd_soc_codec_driver soc_codec_dev_wm2000 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm2000 = { .probe = wm2000_probe, .remove = wm2000_remove, .suspend = wm2000_suspend, diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index d913d0e..606bf88 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2103,7 +2103,7 @@ static struct snd_soc_dai_driver wm2200_dai = { .ops = &wm2200_dai_ops, }; -static struct snd_soc_codec_driver soc_codec_wm2200 = { +static const struct snd_soc_codec_driver soc_codec_wm2200 = { .probe = wm2200_probe, .idle_bias_off = true, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8cbdf17..278467d 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2381,7 +2381,7 @@ static int wm5100_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .probe = wm5100_probe, .remove = wm5100_remove, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 2d6c19d..ed03ed27 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2000,7 +2000,7 @@ static struct regmap *wm5102_get_regmap(struct device *dev) return priv->core.arizona->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_wm5102 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm5102 = { .probe = wm5102_codec_probe, .remove = wm5102_codec_remove, .get_regmap = wm5102_get_regmap, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fa35e19..53d35fe 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2357,7 +2357,7 @@ static struct regmap *wm5110_get_regmap(struct device *dev) return priv->core.arizona->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm5110 = { .probe = wm5110_codec_probe, .remove = wm5110_codec_remove, .get_regmap = wm5110_get_regmap, diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 18bc4ca..2efc5b4 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1587,7 +1587,7 @@ static struct regmap *wm8350_get_regmap(struct device *dev) return wm8350->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .probe = wm8350_codec_probe, .remove = wm8350_codec_remove, .get_regmap = wm8350_get_regmap, diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 4551e54..6c59fb9 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1332,7 +1332,7 @@ static struct regmap *wm8400_get_regmap(struct device *dev) return wm8400->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .probe = wm8400_codec_probe, .remove = wm8400_codec_remove, .get_regmap = wm8400_get_regmap, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 7f93adf..119ceac 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -581,7 +581,7 @@ static int wm8510_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .probe = wm8510_probe, .set_bias_level = wm8510_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 0bb189a..deb2e07 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -413,7 +413,7 @@ static int wm8523_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8523 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8523 = { .probe = wm8523_probe, .set_bias_level = wm8523_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 2e69270..faa7287 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -899,7 +899,7 @@ static int wm8580_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .probe = wm8580_probe, .remove = wm8580_remove, .set_bias_level = wm8580_set_bias_level, diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 4e8ebef..2b376c9 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -367,7 +367,7 @@ static int wm8711_probe(struct snd_soc_codec *codec) } -static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .probe = wm8711_probe, .set_bias_level = wm8711_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index b01e64e..7fde077 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -53,7 +53,7 @@ static struct snd_soc_dai_driver wm8727_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8727 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8727 = { .component_driver = { .dapm_widgets = wm8727_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8727_dapm_widgets), diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index dcd64b3..797cc6e 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -211,7 +211,7 @@ static struct snd_soc_dai_driver wm8728_dai = { .ops = &wm8728_dai_ops, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .set_bias_level = wm8728_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 545fc9a..4f9a1eb 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -628,7 +628,7 @@ err_regulator_enable: return ret; } -static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .set_bias_level = wm8731_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 728c42c..f0cb1c4 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -573,7 +573,7 @@ err_get: return ret; } -static struct snd_soc_codec_driver soc_codec_dev_wm8737 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8737 = { .probe = wm8737_probe, .set_bias_level = wm8737_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3e43272..565d477 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -497,7 +497,7 @@ static int wm8741_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .probe = wm8741_probe, .remove = wm8741_remove, .resume = wm8741_resume, diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index a96a145..0da2bba 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -708,7 +708,7 @@ static int wm8750_probe(struct snd_soc_codec *codec) return ret; } -static struct snd_soc_codec_driver soc_codec_dev_wm8750 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8750 = { .probe = wm8750_probe, .set_bias_level = wm8750_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 1bb272c..9bdf544 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1478,7 +1478,7 @@ static int wm8753_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .probe = wm8753_probe, .resume = wm8753_resume, .set_bias_level = wm8753_set_bias_level, diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 8783a5f..d6edcbb 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -608,7 +608,7 @@ err_reg_enable: return ret; } -static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .probe = wm8770_probe, .set_bias_level = wm8770_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index f17a3a8..ae30480 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -425,7 +425,7 @@ static int wm8776_probe(struct snd_soc_codec *codec) return ret; } -static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .probe = wm8776_probe, .set_bias_level = wm8776_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index 1e9c495..bcda210 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -50,7 +50,7 @@ static struct snd_soc_dai_driver wm8782_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8782 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8782 = { .component_driver = { .dapm_widgets = wm8782_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 724cec4..c77b49a 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1208,7 +1208,7 @@ static int wm8900_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8900 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8900 = { .probe = wm8900_probe, .suspend = wm8900_suspend, .resume = wm8900_resume, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c19ba7c..837c8b9 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1874,7 +1874,7 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903) } #endif -static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .resume = wm8903_resume, .set_bias_level = wm8903_set_bias_level, .seq_notifier = wm8903_seq_notifier, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index edd7a77..4fd350e8 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2086,7 +2086,7 @@ static int wm8904_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .probe = wm8904_probe, .remove = wm8904_remove, .set_bias_level = wm8904_set_bias_level, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 3c9febc..b593562 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -723,7 +723,7 @@ static int wm8940_probe(struct snd_soc_codec *codec) return ret; } -static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .probe = wm8940_probe, .set_bias_level = wm8940_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 575075d..1edc7b1 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -940,7 +940,7 @@ err_enable: return ret; } -static struct snd_soc_codec_driver soc_codec_dev_wm8955 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8955 = { .probe = wm8955_probe, .set_bias_level = wm8955_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index d7f444f..be32443 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1264,7 +1264,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8960 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8960 = { .probe = wm8960_probe, .set_bias_level = wm8960_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 4b51761..e23ceac 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -882,7 +882,7 @@ static int wm8961_resume(struct snd_soc_codec *codec) #define wm8961_resume NULL #endif -static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .probe = wm8961_probe, .resume = wm8961_resume, .set_bias_level = wm8961_set_bias_level, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f3109da..392b920 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3479,7 +3479,7 @@ static int wm8962_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .probe = wm8962_probe, .remove = wm8962_remove, .set_bias_level = wm8962_set_bias_level, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 214fa13..887d31c 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -649,7 +649,7 @@ static int wm8971_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8971 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8971 = { .probe = wm8971_probe, .set_bias_level = wm8971_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index fbda6e3..d414ddd 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -676,7 +676,7 @@ static int wm8974_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8974 = { .probe = wm8974_probe, .set_bias_level = wm8974_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 910b36f..90b2d41 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -993,7 +993,7 @@ static int wm8978_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8978 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8978 = { .probe = wm8978_probe, .suspend = wm8978_suspend, .resume = wm8978_resume, diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9609fc0..bfdbe72 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -976,7 +976,7 @@ static struct snd_soc_dai_driver wm8983_dai = { .symmetric_rates = 1 }; -static struct snd_soc_codec_driver soc_codec_dev_wm8983 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8983 = { .probe = wm8983_probe, .set_bias_level = wm8983_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index bcf3473..05344f9 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1105,7 +1105,7 @@ static struct snd_soc_dai_driver wm8985_dai = { .symmetric_rates = 1 }; -static struct snd_soc_codec_driver soc_codec_dev_wm8985 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8985 = { .probe = wm8985_probe, .set_bias_level = wm8985_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index a412fd0..a894500 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1294,7 +1294,7 @@ static int wm8990_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .probe = wm8990_probe, .set_bias_level = wm8990_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 5822d10..9843790 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1232,7 +1232,7 @@ static struct snd_soc_dai_driver wm8991_dai = { .ops = &wm8991_ops }; -static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8991 = { .set_bias_level = wm8991_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 8668c4c..195f7bf 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1613,7 +1613,7 @@ static const struct regmap_config wm8993_regmap = { .num_reg_defaults = ARRAY_SIZE(wm8993_reg_defaults), }; -static struct snd_soc_codec_driver soc_codec_dev_wm8993 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8993 = { .probe = wm8993_probe, .suspend = wm8993_suspend, .resume = wm8993_resume, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a18aecb..3896523 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4439,7 +4439,7 @@ static struct regmap *wm8994_get_regmap(struct device *dev) return control->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .probe = wm8994_codec_probe, .remove = wm8994_codec_remove, .suspend = wm8994_codec_suspend, diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index abacf6c..5eba8ff 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2684,7 +2684,7 @@ static int wm8996_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .probe = wm8996_probe, .remove = wm8996_remove, .set_bias_level = wm8996_set_bias_level, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index cf23a6b..2f2821b 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1095,7 +1095,7 @@ static struct regmap *wm8997_get_regmap(struct device *dev) return priv->core.arizona->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_wm8997 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8997 = { .probe = wm8997_codec_probe, .remove = wm8997_codec_remove, .get_regmap = wm8997_get_regmap, diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index edaa4cf..bcc2e10 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1361,7 +1361,7 @@ static struct regmap *wm8998_get_regmap(struct device *dev) return priv->core.arizona->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_wm8998 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm8998 = { .probe = wm8998_codec_probe, .remove = wm8998_codec_remove, .get_regmap = wm8998_get_regmap, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 30eaf4c..856867e 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1274,7 +1274,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .probe = wm9081_probe, .set_sysclk = wm9081_set_sysclk, diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 5d73729..5a13138 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -550,7 +550,7 @@ static int wm9090_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm9090 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm9090 = { .probe = wm9090_probe, .set_bias_level = wm9090_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index c88492d..dcdd055 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -352,7 +352,7 @@ static int wm9705_soc_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm9705 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm9705 = { .probe = wm9705_soc_probe, .remove = wm9705_soc_remove, .suspend = wm9705_soc_suspend, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e5aafdd..557709e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -669,7 +669,7 @@ static int wm9712_soc_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm9712 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm9712 = { .probe = wm9712_soc_probe, .remove = wm9712_soc_remove, .resume = wm9712_soc_resume, diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 255a7c2..e4301dd 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1235,7 +1235,7 @@ static int wm9713_soc_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { +static const struct snd_soc_codec_driver soc_codec_dev_wm9713 = { .probe = wm9713_soc_probe, .remove = wm9713_soc_remove, .suspend = wm9713_soc_suspend,

2 1

[alsa-devel] locking looks odd
by Samuel Thibault 01 Sep '16

01 Sep '16

Hello, We are having odd issues with libasound 1.1.2 which we didn't have with libasound 1.1.1, more precisely https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=833950 so I'm having a look at the locking API introduced in 1.1.2, and there are some oddities: - snd_pcm_new seems to initialize pcm->thread_safe to 0 by default, this does not seem safe. The attached patch initializes it to 1, which fixes the bug in our tests. - snd_pcm_hw_open_fd forces it to 1, thus ignoring what snd_pcm_new set. - one can find both __snd_pcm_lock and snd_pcm_lock functions, what is the expected difference between them? - __snd_pcm_lock takes locks when thread_safe >= 0, while snd_pcm_lock takes locks when thread_safe == 0, this looks really odd. - libasound could just not link against libpthread, pthread_mutex_lock/unlock are already provided as empty stubs by libc, the overhead will thus only be hit when the application links against libpthread (libasound will then properly use pthread locks). Samuel

6 11

[alsa-devel] [PATCH] ASoC: rockchip: implement system suspend/resume for i2s
by Sugar Zhang 01 Sep '16

01 Sep '16

restore hw registers after power loss during a suspend/resume cycle. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 574c6af..53970ac 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -614,9 +614,35 @@ static const struct of_device_id rockchip_i2s_match[] = { {}, }; +#ifdef CONFIG_PM_SLEEP +static int rockchip_i2s_suspend(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + + regcache_mark_dirty(i2s->regmap); + + return 0; +} + +static int rockchip_i2s_resume(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(i2s->regmap); + pm_runtime_put(dev); + + return ret; +} +#endif + static const struct dev_pm_ops rockchip_i2s_pm_ops = { SET_RUNTIME_PM_OPS(i2s_runtime_suspend, i2s_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(rockchip_i2s_suspend, rockchip_i2s_resume) }; static struct platform_driver rockchip_i2s_driver = { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Remove deprecated create_singlethread_workqueue
by Bhaktipriya Shridhar 01 Sep '16

01 Sep '16

The workqueue "dac33_wq" queues a single work item &dac33->work and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in dac33_soc_remove to ensure that there are no pending tasks while disconnecting the driver. The workqueue "post_msg_wq" queues a single work item &drv->ipc_post_msg_wq and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in sst_context_cleanup to ensure that there are no pending tasks while disconnecting the driver. System workqueues have been able to handle high level of concurrency for a long time now and hence it's not required to have a singlethreaded workqueue just to gain concurrency. Unlike a dedicated per-cpu workqueue created with create_singlethread_workqueue(), system_wq allows multiple work items to overlap executions even on the same CPU; however, a per-cpu workqueue doesn't have any CPU locality or global ordering guarantee unless the target CPU is explicitly specified and thus the increase of local concurrency shouldn't make any difference. Signed-off-by: Bhaktipriya Shridhar <bhaktipriya96(a)gmail.com> --- sound/soc/codecs/tlv320dac33.c | 17 ++++------------- sound/soc/intel/atom/sst/sst.c | 14 +++++--------- 2 files changed, 9 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f7a6ce7..6822ac1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -90,7 +90,6 @@ static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { struct tlv320dac33_priv { struct mutex mutex; - struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec *codec; struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; @@ -771,7 +770,7 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); return IRQ_HANDLED; } @@ -1127,7 +1126,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (dac33->fifo_mode) { dac33->state = DAC33_PREFILL; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; case SNDRV_PCM_TRIGGER_STOP: @@ -1135,7 +1134,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (dac33->fifo_mode) { dac33->state = DAC33_FLUSH; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; default: @@ -1410,14 +1409,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dac33->irq = -1; } if (dac33->irq != -1) { - /* Setup work queue */ - dac33->dac33_wq = - create_singlethread_workqueue("tlv320dac33"); - if (dac33->dac33_wq == NULL) { - free_irq(dac33->irq, codec); - return -ENOMEM; - } - INIT_WORK(&dac33->work, dac33_work); } } @@ -1437,7 +1428,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) if (dac33->irq >= 0) { free_irq(dac33->irq, dac33->codec); - destroy_workqueue(dac33->dac33_wq); + flush_work(&dac33->work); } return 0; } diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..8cfd0cf 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -76,7 +76,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) spin_unlock(&drv->ipc_spin_lock); /* we can send more messages to DSP so trigger work */ - queue_work(drv->post_msg_wq, &drv->ipc_post_msg_wq); + schedule_work(&drv->ipc_post_msg_wq); retval = IRQ_HANDLED; } @@ -212,10 +212,6 @@ static int sst_workqueue_init(struct intel_sst_drv *ctx) INIT_WORK(&ctx->ipc_post_msg_wq, sst_process_pending_msg); init_waitqueue_head(&ctx->wait_queue); - ctx->post_msg_wq = - create_singlethread_workqueue("sst_post_msg_wq"); - if (!ctx->post_msg_wq) - return -EBUSY; return 0; } @@ -318,7 +314,6 @@ int sst_context_init(struct intel_sst_drv *ctx) return 0; do_free_mem: - destroy_workqueue(ctx->post_msg_wq); return ret; } EXPORT_SYMBOL_GPL(sst_context_init); @@ -330,7 +325,7 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); - destroy_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); pm_qos_remove_request(ctx->qos); kfree(ctx->fw_sg_list.src); kfree(ctx->fw_sg_list.dst); @@ -414,7 +409,7 @@ static int intel_sst_runtime_suspend(struct device *dev) sst_set_fw_state_locked(ctx, SST_RESET); synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ @@ -445,8 +440,9 @@ static int intel_sst_suspend(struct device *dev) return -EBUSY; } } + synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(ctx->ipc_post_msg_wq); /* Move the SST state to Reset */ sst_set_fw_state_locked(ctx, SST_RESET); -- 2.1.4

3 4

[alsa-devel] Setting some clocks back to DUMMY fixes spdif output on imx6q wandboard rev B1
by Xavi Drudis Ferran 31 Aug '16

31 Aug '16

Hello. Sorry for my unfamiliarity with the kernel community. I'm afraid I'll be getting the procedure wrong... I was using linux-libre-3.19 (implies no working sdma) with a wandboard quad (Freescale imx6q). Spidf output worked fine. When I upgraded to linux-libre-4.7 spdif output was supressed without any error (precisely, with the same errors about sdma that 3.19 gave). I saw someone else reporting the same elsewhere with linux-4.4 https://forum.digikey.com/thread/34240 (but I don't have a login there) This patch fixes it for me and sound works again on spdif. But I don't know if it can break (or fix?) something for some other boards or kernels or cases... I hardly know what I'm doing. The commits that might have caused the problem for me might be commit 833f2cbf7091099baee28136dc68678e974c0ac5 Author: Shengjiu Wang <shengjiu.wang(a)freescale.com> Date: Sat Oct 10 18:15:07 2015 +0800 ARM: dts: imx6: change the core clock of spdif The correct core clock of spdif is SPDIF_GCLK, which is added to clock tree. So the dts also need to be updated. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> Signed-off-by: Shawn Guo <shawnguo(a)kernel.org> (the commit changed more clocks than SPDIF_GCLK) It might have to do with vendor commit http://git.freescale.com/git/cgit.cgi/imx/linux-2.6-imx.git/commit/arch/arm… The issue is fixed for me with this patch but I'm not sure what's the best way to help fix any issue someone else may have or what other info or test you might need. Any guidance welcome. Thank you for linux.

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Re: [alsa-devel] [PATCH] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Tony Lindgren 31 Aug '16

31 Aug '16

* kbuild test robot <lkp(a)intel.com> [160829 21:31]: > vim +/dev_pm_qos_request_active +1108 sound/soc/omap/mcbsp.c > > 1102 clk_put(mcbsp->fclk); > 1103 return ret; > 1104 } > 1105 > 1106 void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) > 1107 { > > 1108 if (dev_pm_qos_request_active(&mcbsp->pm_qos_req)) > 1109 pm_qos_remove_request(&mcbsp->pm_qos_req); > 1110 > 1111 if (mcbsp->pdata->buffer_size) Oops sorry about that. Looks like I completely missed that warning after a copy-paste adding of the check to omap_mcbsp_cleanup. Looks like implementing dev_pm_qos needs more work as it needs the set_latency_tolerance implemented somewhere. Updated patch below, Mark let me know if you need an incremental patch to fix the warning. Regards, Tony 8< ----------------- From: Tony Lindgren <tony(a)atomide.com> Date: Mon, 29 Aug 2016 11:27:46 -0700 Subject: [PATCHv2] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Let's fix the issue by adding PM QoS latency requirement for McBSP. Based on my experiments a working latency value is somewhere around 35 ms with 40 ms breaking. So let's use a safer value of 30 ms. Note that this is different from snd_pcm_hw_constraint_step() as that can configure things based on the buffer and period size. However, that does not help to block idle between the fifo fills. Note that in theory some omaps are capable of playing audio while hitting deeper idle states. If somebody needs that and gets it working, we can set the latency requirements accordingly. Signed-off-by: Tony Lindgren <tony(a)atomide.com> diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -643,6 +644,10 @@ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) int enable_srg = 0; u16 w; + /* Prevent omap hardware from hitting off between fifo fills */ + pm_qos_add_request(&mcbsp->pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + 30 * 1000); + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -731,6 +736,8 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); + + pm_qos_remove_request(&mcbsp->pm_qos_req); } int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) @@ -1098,6 +1105,9 @@ err_thres: void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -325,6 +325,8 @@ struct omap_mcbsp { unsigned int in_freq; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp,

2 2

[alsa-devel] [PATCH v3] ASoC: Remove deprecated create_singlethread_workqueue
by Bhaktipriya Shridhar 31 Aug '16

31 Aug '16

The workqueue "dac33_wq" queues a single work item &dac33->work and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in dac33_soc_remove to ensure that there are no pending tasks while disconnecting the driver. The workqueue "post_msg_wq" queues a single work item &drv->ipc_post_msg_wq and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in sst_context_cleanup to ensure that there are no pending tasks while disconnecting the driver. System workqueues have been able to handle high level of concurrency for a long time now and hence it's not required to have a singlethreaded workqueue just to gain concurrency. Unlike a dedicated per-cpu workqueue created with create_singlethread_workqueue(), system_wq allows multiple work items to overlap executions even on the same CPU; however, a per-cpu workqueue doesn't have any CPU locality or global ordering guarantee unless the target CPU is explicitly specified and thus the increase of local concurrency shouldn't make any difference. Signed-off-by: Bhaktipriya Shridhar <bhaktipriya96(a)gmail.com> --- Changes in v3: -Added missing '&' sound/soc/codecs/tlv320dac33.c | 17 ++++------------- sound/soc/intel/atom/sst/sst.c | 14 +++++--------- 2 files changed, 9 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f7a6ce7..6822ac1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -90,7 +90,6 @@ static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { struct tlv320dac33_priv { struct mutex mutex; - struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec *codec; struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; @@ -771,7 +770,7 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); return IRQ_HANDLED; } @@ -1127,7 +1126,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (dac33->fifo_mode) { dac33->state = DAC33_PREFILL; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; case SNDRV_PCM_TRIGGER_STOP: @@ -1135,7 +1134,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (dac33->fifo_mode) { dac33->state = DAC33_FLUSH; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; default: @@ -1410,14 +1409,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dac33->irq = -1; } if (dac33->irq != -1) { - /* Setup work queue */ - dac33->dac33_wq = - create_singlethread_workqueue("tlv320dac33"); - if (dac33->dac33_wq == NULL) { - free_irq(dac33->irq, codec); - return -ENOMEM; - } - INIT_WORK(&dac33->work, dac33_work); } } @@ -1437,7 +1428,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) if (dac33->irq >= 0) { free_irq(dac33->irq, dac33->codec); - destroy_workqueue(dac33->dac33_wq); + flush_work(&dac33->work); } return 0; } diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..1454603 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -76,7 +76,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) spin_unlock(&drv->ipc_spin_lock); /* we can send more messages to DSP so trigger work */ - queue_work(drv->post_msg_wq, &drv->ipc_post_msg_wq); + schedule_work(&drv->ipc_post_msg_wq); retval = IRQ_HANDLED; } @@ -212,10 +212,6 @@ static int sst_workqueue_init(struct intel_sst_drv *ctx) INIT_WORK(&ctx->ipc_post_msg_wq, sst_process_pending_msg); init_waitqueue_head(&ctx->wait_queue); - ctx->post_msg_wq = - create_singlethread_workqueue("sst_post_msg_wq"); - if (!ctx->post_msg_wq) - return -EBUSY; return 0; } @@ -318,7 +314,6 @@ int sst_context_init(struct intel_sst_drv *ctx) return 0; do_free_mem: - destroy_workqueue(ctx->post_msg_wq); return ret; } EXPORT_SYMBOL_GPL(sst_context_init); @@ -330,7 +325,7 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); - destroy_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); pm_qos_remove_request(ctx->qos); kfree(ctx->fw_sg_list.src); kfree(ctx->fw_sg_list.dst); @@ -414,7 +409,7 @@ static int intel_sst_runtime_suspend(struct device *dev) sst_set_fw_state_locked(ctx, SST_RESET); synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ @@ -445,8 +440,9 @@ static int intel_sst_suspend(struct device *dev) return -EBUSY; } } + synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); /* Move the SST state to Reset */ sst_set_fw_state_locked(ctx, SST_RESET); -- 2.1.4

3 2

[alsa-devel] [PATCH v4] ALSA: fireworks: accessing to user space outside spinlock
by Takashi Sakamoto 31 Aug '16

31 Aug '16

Hi, This patch is a revised version of a part of my previous patchset: [alsa-devel] [PATCH v3 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/112366.html In ALSA fireworks driver, current implementation for hwdep interface have issues for page fault handling. For example, when using single core processor, after executing lock_page() in page fault handler and call task scheduler, then the context never regain CPU again, because any interrupts are disabled by these drivers and the context can't catch waiting event. This patchset fixes the issues. The issues were reported by Vaishali Thakkar, with a help of coccinelle. Thanks a lot for her care to these minor drivers. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111887.html Changes in v4: * Give up counting the number of queued response messages to simplify ring-buffer management. Instead, comparing pull/push pointers is used consistently in any parts. * Assign current state to automatic variables in critical section of blocking processing part, then utilize it for later processing. Changes in v3: * Add a workaround against a case that several tasks are going to read from one response buffer. Changes in v2: * improve designated initialization for automatic variable of union type Takashi Sakamoto (1): ALSA: fireworks: accessing to user space outside spinlock sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_hwdep.c | 71 +++++++++++++++++------- sound/firewire/fireworks/fireworks_proc.c | 4 +- sound/firewire/fireworks/fireworks_transaction.c | 5 +- 4 files changed, 56 insertions(+), 25 deletions(-) -- 2.7.4

2 3

[alsa-devel] [PATCH v3 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock
by Takashi Sakamoto 31 Aug '16

31 Aug '16

Hi, This patchset updates my previous one: [alsa-devel] [PATCH v2 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/112236.html In ALSA fireworks driver and firewire-tascam driver, current implementation for hwdep interface have issues for page fault handling. For example, when using single core processor, after executing lock_page() in page fault handler and call task scheduler, then the context never regain CPU again, because any interrupts are disabled by these drivers and the context can't catch waiting event. This patchset fixes the issues. The issues were reported by Vaishali Thakkar, with a help of coccinelle. Thanks a lot for her care to these minor drivers. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111887.html Changes in v3: * Add a workaround against a case that several tasks are going to read from one response buffer. Changes in v2: * improve designated initialization for automatic variable of union type Takashi Sakamoto (2): ALSA: firewire-tascam: accessing to user space outside spinlock ALSA: fireworks: accessing to user space outside spinlock sound/firewire/fireworks/fireworks_hwdep.c | 52 ++++++++++++++++++------ sound/firewire/fireworks/fireworks_transaction.c | 4 +- sound/firewire/tascam/tascam-hwdep.c | 33 +++++---------- 3 files changed, 53 insertions(+), 36 deletions(-) -- 2.7.4

2 5

[alsa-devel] [PATCH v2 0/3] drm/i2c: tda998x ASoC hdmi-codec support + BeagleBoneBlack audio support
by Jyri Sarha 31 Aug '16

31 Aug '16

Changes since first version [1] of this series: - "drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata" - Change audio in tda998x pdata to audio_params to match the naming in private data struct - Skip IEC958_AES2 byte when writing status bytes to registers - Turn AFMT-macros to enum - include linux/hdmi.h in drm/i2c/tda998x.h - "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Fix macronaming naming in dt-bindings/display/tda998x.h - Turn AFMT-macros to enum - Add locking to protect access to audio registers from ASoC thread - "dts/am335x-boneblack: SQUASH" - Use corrected macros from dt-bindings/display/tda998x.h [1] https://lists.freedesktop.org/archives/dri-devel/2016-August/114491.html Jyri Sarha (3): drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 ++ arch/arm/boot/dts/am335x-boneblack.dts | 71 ++++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 297 ++++++++++++++++++--- include/drm/i2c/tda998x.h | 29 +- include/dt-bindings/display/tda998x.h | 7 + 6 files changed, 368 insertions(+), 55 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h -- 1.9.1

2 4

[alsa-devel] [PATCH v2] ALSA: seq: initialize whole fields of automatic variable with union type
by Takashi Sakamoto 31 Aug '16

31 Aug '16

Currently, automatic variable of 'union ioctl_arg' type is initialized by designated initialization. Although, the actual effect is interpretation of early element of int type and initialization of 'int pversion'. Therefore the first field corresponding to int type is initialized to zero. This is against my expectation to initialize whole fields. This commit uses memset() to initialize the variable, instead of designated initialization. Fixes: 04a56dd8ed0d ('ALSA: seq: change ioctl command operation to get data in kernel space') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/seq/seq_clientmgr.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 286394b..811b95b 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2084,7 +2084,7 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, { struct snd_seq_client *client = file->private_data; /* To use kernel stack for ioctl data. */ - union ioctl_arg { + union { int pversion; int client_id; struct snd_seq_system_info system_info; @@ -2100,7 +2100,7 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, struct snd_seq_client_pool client_pool; struct snd_seq_remove_events remove_events; struct snd_seq_query_subs query_subs; - } buf = {0}; + } buf; const struct ioctl_handler *handler; unsigned long size; int err; @@ -2114,6 +2114,9 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, } if (handler->cmd == 0) return -ENOTTY; + + memset(&buf, 0, sizeof(buf)); + /* * All of ioctl commands for ALSA sequencer get an argument of size * within 13 bits. We can safely pick up the size from the command. -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: arizona: Don't change the FLLn_GAIN before entering FREERUN
by Charles Keepax 31 Aug '16

31 Aug '16

From: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> When reclocking an active FLL, to ensure a clean transition, do not change the gain setting until we have entered free run. Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ecfdbfc..0fc3b4b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2218,11 +2218,11 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (already_enabled) { /* Facilitate smooth refclk across the transition */ - regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, - ARIZONA_FLL1_GAIN_MASK, 0); regmap_update_bits(fll->arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); udelay(32); + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, + ARIZONA_FLL1_GAIN_MASK, 0); } /* -- 2.1.4

2 1

[alsa-devel] [PATCH v6 1/2] mfd: Add DT bindings documentation for Samsung Exynos LPASS
by Sylwester Nawrocki 31 Aug '16

31 Aug '16

This patch adds documentation of the DT bindings for the Samsung Exynos SoC Low Power Audio Subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v5: - none. Changes since v4: - text reformatting, added VIC to the list of sub-devices, relative paths used for the reference DT binding documentation files, added example dts. Changes since v3: - moved to Documentation/devicetree/bindings/mfd, - added #address-cells, #size-cells, ranges as required properties, - added description of the IP block subnodes and links to their binding's documentation. Changes since v2: - none. --- .../bindings/mfd/samsung,exynos5433-lpass.txt | 70 ++++++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt diff --git a/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt new file mode 100644 index 0000000..c110e11 --- /dev/null +++ b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt @@ -0,0 +1,70 @@ +Samsung Exynos SoC Low Power Audio Subsystem (LPASS) + +Required properties: + + - compatible : "samsung,exynos5433-lpass" + - reg : should contain the LPASS top SFR region location + and size + - samsung,pmu-syscon : the phandle to the Power Management Unit node + - #address-cells : should be 1 + - #size-cells : should be 1 + - ranges : must be present + +Each IP block of the Low Power Audio Subsystem should be specified as +an optional sub-node. For "samsung,exynos5433-lpass" compatible this includes: +UART, SLIMBUS, PCM, I2S, DMAC, Timers 0...4, VIC, WDT 0...1 devices. + +Bindings of the sub-nodes are described in: + ../serial/samsung_uart.txt + ../sound/samsung-i2s.txt + ../dma/arm-pl330.txt + + +Example: + +audio-subsystem { + compatible = "samsung,exynos5433-lpass"; + reg = <0x11400000 0x100>, <0x11500000 0x08>; + samsung,pmu-syscon = <&pmu_system_controller>; + #address-cells = <1>; + #size-cells = <1>; + ranges; + + adma: adma@11420000 { + compatible = "arm,pl330", "arm,primecell"; + reg = <0x11420000 0x1000>; + interrupts = <0 73 0>; + clocks = <&cmu_aud CLK_ACLK_DMAC>; + clock-names = "apb_pclk"; + #dma-cells = <1>; + #dma-channels = <8>; + #dma-requests = <32>; + }; + + i2s0: i2s0@11440000 { + compatible = "samsung,exynos7-i2s"; + reg = <0x11440000 0x100>; + dmas = <&adma 0 &adma 2>; + dma-names = "tx", "rx"; + interrupts = <0 70 0>; + clocks = <&cmu_aud CLK_PCLK_AUD_I2S>, + <&cmu_aud CLK_SCLK_AUD_I2S>, + <&cmu_aud CLK_SCLK_I2S_BCLK>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + status = "disabled"; + }; + + serial_3: serial@11460000 { + compatible = "samsung,exynos5433-uart"; + reg = <0x11460000 0x100>; + interrupts = <0 67 0>; + clocks = <&cmu_aud CLK_PCLK_AUD_UART>, + <&cmu_aud CLK_SCLK_AUD_UART>; + clock-names = "uart", "clk_uart_baud0"; + pinctrl-names = "default"; + pinctrl-0 = <&uart_aud_bus>; + status = "disabled"; + }; + }; -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: Intel: Atom: Fix message handling during drop stream
by Harsha Priya 31 Aug '16

31 Aug '16

If a stream is being dropped, the period_elapsed received then after need not be processed. Processing of the period_elapsed message causes a time out in drop stream response processing. This patch adds a condition to skip period_elapsed message processing if the stream is in INIT state. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 8afa6fe..bfc8899 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -267,6 +267,9 @@ static void process_fw_async_msg(struct intel_sst_drv *sst_drv_ctx, "Period elapsed rcvd for pipe id 0x%x\n", pipe_id); stream = &sst_drv_ctx->streams[str_id]; + /* If stream is dropped, skip processing this message*/ + if (stream->status == STREAM_INIT) + break; if (stream->period_elapsed) stream->period_elapsed(stream->pcm_substream); if (stream->compr_cb) -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: Remove deprecated create_singlethread_workqueue
by Bhaktipriya Shridhar 30 Aug '16

30 Aug '16

The workqueue "dac33_wq" queues a single work item &dac33->work and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in dac33_soc_remove to ensure that there are no pending tasks while disconnecting the driver. The workqueue "post_msg_wq" queues a single work item &drv->ipc_post_msg_wq and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in sst_context_cleanup to ensure that there are no pending tasks while disconnecting the driver. System workqueues have been able to handle high level of concurrency for a long time now and hence it's not required to have a singlethreaded workqueue just to gain concurrency. Unlike a dedicated per-cpu workqueue created with create_singlethread_workqueue(), system_wq allows multiple work items to overlap executions even on the same CPU; however, a per-cpu workqueue doesn't have any CPU locality or global ordering guarantee unless the target CPU is explicitly specified and thus the increase of local concurrency shouldn't make any difference. Signed-off-by: Bhaktipriya Shridhar <bhaktipriya96(a)gmail.com> --- Changes in v2: - Fixed heading to indicate that changes were made in sound/soc sound/soc/codecs/tlv320dac33.c | 17 ++++------------- sound/soc/intel/atom/sst/sst.c | 14 +++++--------- 2 files changed, 9 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f7a6ce7..6822ac1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -90,7 +90,6 @@ static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { struct tlv320dac33_priv { struct mutex mutex; - struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec *codec; struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; @@ -771,7 +770,7 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); return IRQ_HANDLED; } @@ -1127,7 +1126,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (dac33->fifo_mode) { dac33->state = DAC33_PREFILL; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; case SNDRV_PCM_TRIGGER_STOP: @@ -1135,7 +1134,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (dac33->fifo_mode) { dac33->state = DAC33_FLUSH; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; default: @@ -1410,14 +1409,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dac33->irq = -1; } if (dac33->irq != -1) { - /* Setup work queue */ - dac33->dac33_wq = - create_singlethread_workqueue("tlv320dac33"); - if (dac33->dac33_wq == NULL) { - free_irq(dac33->irq, codec); - return -ENOMEM; - } - INIT_WORK(&dac33->work, dac33_work); } } @@ -1437,7 +1428,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) if (dac33->irq >= 0) { free_irq(dac33->irq, dac33->codec); - destroy_workqueue(dac33->dac33_wq); + flush_work(&dac33->work); } return 0; } diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..8cfd0cf 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -76,7 +76,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) spin_unlock(&drv->ipc_spin_lock); /* we can send more messages to DSP so trigger work */ - queue_work(drv->post_msg_wq, &drv->ipc_post_msg_wq); + schedule_work(&drv->ipc_post_msg_wq); retval = IRQ_HANDLED; } @@ -212,10 +212,6 @@ static int sst_workqueue_init(struct intel_sst_drv *ctx) INIT_WORK(&ctx->ipc_post_msg_wq, sst_process_pending_msg); init_waitqueue_head(&ctx->wait_queue); - ctx->post_msg_wq = - create_singlethread_workqueue("sst_post_msg_wq"); - if (!ctx->post_msg_wq) - return -EBUSY; return 0; } @@ -318,7 +314,6 @@ int sst_context_init(struct intel_sst_drv *ctx) return 0; do_free_mem: - destroy_workqueue(ctx->post_msg_wq); return ret; } EXPORT_SYMBOL_GPL(sst_context_init); @@ -330,7 +325,7 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); - destroy_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); pm_qos_remove_request(ctx->qos); kfree(ctx->fw_sg_list.src); kfree(ctx->fw_sg_list.dst); @@ -414,7 +409,7 @@ static int intel_sst_runtime_suspend(struct device *dev) sst_set_fw_state_locked(ctx, SST_RESET); synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ @@ -445,8 +440,9 @@ static int intel_sst_suspend(struct device *dev) return -EBUSY; } } + synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(ctx->ipc_post_msg_wq); /* Move the SST state to Reset */ sst_set_fw_state_locked(ctx, SST_RESET); -- 2.1.4

1 0

[alsa-devel] [PATCH v9 0/2] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 30 Aug '16

30 Aug '16

This patch adds support for some mixer controls: * Line-In * FM-In * Mic 2 * Capture Source * Differential Line-In v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (2): ASoC: sun4i-codec: Distinguish sun4i from sun7i Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 278 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 259 insertions(+), 19 deletions(-)

3 6

Re: [alsa-devel] Questions about thinkpad_helper
by Takashi Iwai 30 Aug '16

30 Aug '16

[Changed Cc to proper alsa-devel ML] On Tue, 30 Aug 2016 12:02:56 +0200, Dennis Wassenberg wrote: > > Hi all, > > I wrote a driver which handles the function / special keys for > ThinKeyboard of Lenovo ThinkPad X1 Tablet 2016 including LED control. > This Keyboard is connected via USB to the Tablet. I added the driver to > hid-lenovo.c file because there are HID drivers included for different > Lenovo devices. > > For example this driver is able to handle LED control for mute and mic > mute keys. Now I want to make the other direction working. If someone > sets mute or mic mute at amixer, I want to control the mute and mic mute > LEDs. At other Thinkpad devices this is done by thinkpad_helper.c. The > thinkpad_helper let the thinkpad_apci driver to control the LEDs. > > Now I think about how to make this working for Lenovo Thinkpad X1 Tablet > 2016. There it is not possible to use the thinkpad_acpi driver. Instead > we have to use the hid-lenovo driver to make this working. > > Should I add this to thinkpad_helper because this is a thinkpad device > too or should I make a new helper? Can I use the same API than > thinkpad_acpi (static int (*led_set_func)(int, bool)) or should I make a > new one (e.g. in combination with LED triggers)? > > For the first shot I made something at > https://github.com/dwassenberg/linux/tree/development/hid-rmi > (dc05de0add955775434d18d4fe096ea78d1e63bd..5a88c66f2ae8db04147bb75ecf84a4f52e0da1df) > this shot is based on using the thinkpad_helper and the same API. Please > tell me your opinion. > > Maybe one additional remark. In case of using the USB connected > ThinKeyboard there is one additional effort to make this working at the > initial state if the Tablet is finished booting and after that the > ThinKeyboard is connected. So either there is an additional trigger > needed which tells the helper to publish the current mute and mic mute > state when the keyboard is connected. Or an other approach is to update > the hid-lenovo driver even when this keyboard is not connected such that > the hid-lenovo driver always knows the current state and is able to set > this if the keyboard is connected. Or is there there an other solution? I think your changes are OK. A slight concern I found while looking quickly through the git commits is the cosmetic build issues, either only one of CONFIG_THINKPAD_ACPI and CONFIG_HID_LENOVO is enabled. But other than that, it looks straightforward. Of course, another option would be leave everything to user-space. But I don't think it would work fluently on the current desktop, even in near future. So considering practically, we'll likely go to the kernel hook way, I suppose. thanks, Takashi

1 0

[alsa-devel] [PATCH] ASoC: Intel: Atom: Fix message handling during drop stream
by Harsha Priya 30 Aug '16

30 Aug '16

If a stream is being dropped, the period_elapsed received then after need not be processed. Processing of the period_elapsed message causes a time out in drop stream response processing. This patch adds a condition to skip period_elapsed message processing if the stream is in INIT state. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 8afa6fe..bfc8899 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -267,6 +267,9 @@ static void process_fw_async_msg(struct intel_sst_drv *sst_drv_ctx, "Period elapsed rcvd for pipe id 0x%x\n", pipe_id); stream = &sst_drv_ctx->streams[str_id]; + /* If stream is dropped, skip processing this message*/ + if (stream->status == STREAM_INIT) + break; if (stream->period_elapsed) stream->period_elapsed(stream->pcm_substream); if (stream->compr_cb) -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: Add sample rate inquiry quirk for B850V3 CP2114
by Akshay Bhat 30 Aug '16

30 Aug '16

Hi, On Fri, Aug 12, 2016 at 2:08 PM, Akshay Bhat <akshay.bhat(a)timesys.com> wrote: > From: Ken Lin <ken.lin(a)advantech.com.tw> > > Avoid getting sample rate on B850V3 CP2114 as it is unsupported and > causes noisy "current rate is different from the runtime rate" messages > when playback starts. > > Signed-off-by: Ken Lin <ken.lin(a)advantech.com.tw> > Signed-off-by: Akshay Bhat <akshay.bhat(a)timesys.com> > --- Any feedback on the patch? If not can it be applied? Thanks.

2 5

[alsa-devel] [PATCH] ASoC: hdac_hdmi: fix spelling mistake "montior" -> "monitor"
by Colin King 30 Aug '16

30 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_warn message. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdac_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e181b2..c602c49 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -614,7 +614,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, (!pin->eld.eld_valid)) { dev_warn(&hdac->hdac.dev, - "Failed: montior present? %d ELD valid?: %d for pin: %d\n", + "Failed: monitor present? %d ELD valid?: %d for pin: %d\n", pin->eld.monitor_present, pin->eld.eld_valid, pin->nid); return 0; -- 2.9.3

2 1

[alsa-devel] sound: deadlock snd_rawmidi_kernel_open/check_and_subscribe_port
by Dmitry Vyukov 30 Aug '16

30 Aug '16

Hello, While running syzkaller fuzzer on f31494bd05b06b0cdb4da6aebe92eaafab970df6 (Aug 12), I've got the following deadlock report: ====================================================== [ INFO: possible circular locking dependency detected ] 4.8.0-rc1+ #11 Not tainted ------------------------------------------------------- syz-executor/7154 is trying to acquire lock: (register_mutex#5){+.+.+.}, at: [<ffffffff84fd6d4b>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 but task is already holding lock: (&grp->list_mutex){++++.+}, at: [<ffffffff850138bb>] check_and_subscribe_port+0x5b/0x5c0 sound/core/seq/seq_ports.c:495 which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&grp->list_mutex){++++.+}: [<ffffffff8147a3a8>] lock_acquire+0x208/0x430 kernel/locking/lockdep.c:3746 [<ffffffff863f6199>] down_read+0x49/0xc0 kernel/locking/rwsem.c:22 [< inline >] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:681 [<ffffffff85005c5e>] snd_seq_deliver_event+0x35e/0x890 sound/core/seq/seq_clientmgr.c:822 [<ffffffff85006e96>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2418 [<ffffffff85012c52>] snd_seq_system_broadcast+0xb2/0xf0 sound/core/seq/seq_system.c:101 [<ffffffff84fff70a>] snd_seq_create_kernel_client+0x24a/0x330 sound/core/seq/seq_clientmgr.c:2297 [< inline >] snd_virmidi_dev_attach_seq sound/core/seq/seq_virmidi.c:383 [<ffffffff8502d29f>] snd_virmidi_dev_register+0x29f/0x750 sound/core/seq/seq_virmidi.c:450 [<ffffffff84fd208c>] snd_rawmidi_dev_register+0x30c/0xd40 sound/core/rawmidi.c:1645 [<ffffffff84f816d3>] __snd_device_register.part.0+0x63/0xc0 sound/core/device.c:164 [< inline >] __snd_device_register sound/core/device.c:162 [<ffffffff84f8235d>] snd_device_register_all+0xad/0x110 sound/core/device.c:212 [<ffffffff84f7546f>] snd_card_register+0xef/0x6c0 sound/core/init.c:749 [<ffffffff85040b7f>] snd_virmidi_probe+0x3ef/0x590 sound/drivers/virmidi.c:123 [<ffffffff833ebf7b>] platform_drv_probe+0x8b/0x170 drivers/base/platform.c:564 [< inline >] really_probe drivers/base/dd.c:377 [<ffffffff833e5993>] driver_probe_device+0x563/0xcc0 drivers/base/dd.c:499 [<ffffffff833e653d>] __device_attach_driver+0x21d/0x2e0 drivers/base/dd.c:594 [<ffffffff833df38f>] bus_for_each_drv+0x13f/0x1d0 drivers/base/bus.c:463 [<ffffffff833e51ef>] __device_attach+0x1ef/0x300 drivers/base/dd.c:651 [<ffffffff833e669a>] device_initial_probe+0x1a/0x20 drivers/base/dd.c:698 [<ffffffff833e2999>] bus_probe_device+0x1e9/0x290 drivers/base/bus.c:557 [<ffffffff833dc459>] device_add+0xc49/0x14c0 drivers/base/core.c:1120 [<ffffffff833eb629>] platform_device_add+0x2f9/0x7b0 drivers/base/platform.c:403 [<ffffffff833ed3b2>] platform_device_register_full+0x392/0x4b0 drivers/base/platform.c:536 [< inline >] platform_device_register_resndata ./include/linux/platform_device.h:111 [< inline >] platform_device_register_simple ./include/linux/platform_device.h:140 [<ffffffff8870fabd>] alsa_card_virmidi_init+0x104/0x1da sound/drivers/virmidi.c:172 [<ffffffff81002330>] do_one_initcall+0xa0/0x2b0 init/main.c:778 [< inline >] do_initcall_level init/main.c:843 [< inline >] do_initcalls init/main.c:851 [< inline >] do_basic_setup init/main.c:869 [<ffffffff88604d11>] kernel_init_freeable+0x47b/0x534 init/main.c:1016 [<ffffffff863e2f53>] kernel_init+0x13/0x180 init/main.c:942 [<ffffffff863fbdcf>] ret_from_fork+0x1f/0x40 arch/x86/entry/entry_64.S:393 -> #0 (register_mutex#5){+.+.+.}: [< inline >] check_prev_add kernel/locking/lockdep.c:1829 [< inline >] check_prevs_add kernel/locking/lockdep.c:1939 [< inline >] validate_chain kernel/locking/lockdep.c:2266 [<ffffffff814791f4>] __lock_acquire+0x4d44/0x4d80 kernel/locking/lockdep.c:3335 [<ffffffff8147a3a8>] lock_acquire+0x208/0x430 kernel/locking/lockdep.c:3746 [< inline >] __mutex_lock_common kernel/locking/mutex.c:521 [<ffffffff863f0ef1>] mutex_lock_nested+0xb1/0xa20 kernel/locking/mutex.c:621 [<ffffffff84fd6d4b>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 [<ffffffff8502e7c7>] midisynth_subscribe+0xf7/0x350 sound/core/seq/seq_midi.c:188 [< inline >] subscribe_port sound/core/seq/seq_ports.c:427 [<ffffffff85013cc7>] check_and_subscribe_port+0x467/0x5c0 sound/core/seq/seq_ports.c:510 [<ffffffff85015da9>] snd_seq_port_connect+0x2c9/0x500 sound/core/seq/seq_ports.c:579 [<ffffffff850079b8>] snd_seq_ioctl_subscribe_port+0x1d8/0x2b0 sound/core/seq/seq_clientmgr.c:1480 [<ffffffff84ffe9e4>] snd_seq_do_ioctl+0x184/0x1e0 sound/core/seq/seq_clientmgr.c:2225 [<ffffffff84ffeae8>] snd_seq_kernel_client_ctl+0xa8/0x110 sound/core/seq/seq_clientmgr.c:2440 [<ffffffff85027664>] snd_seq_oss_midi_open+0x3b4/0x610 sound/core/seq/oss/seq_oss_midi.c:375 [<ffffffff85023d67>] snd_seq_oss_synth_setup_midi+0x107/0x4c0 sound/core/seq/oss/seq_oss_synth.c:281 [<ffffffff8501b0a8>] snd_seq_oss_open+0x748/0x8d0 sound/core/seq/oss/seq_oss_init.c:274 [<ffffffff85019d8a>] odev_open+0x6a/0x90 sound/core/seq/oss/seq_oss.c:138 [<ffffffff84f7040f>] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [<ffffffff818069ea>] chrdev_open+0x22a/0x4c0 fs/char_dev.c:392 [<ffffffff817f146f>] do_dentry_open+0x69f/0xca0 fs/open.c:736 [<ffffffff817f4aa5>] vfs_open+0x105/0x220 fs/open.c:849 [< inline >] do_last fs/namei.c:3374 [<ffffffff8182c35d>] path_openat+0x1efd/0x2f60 fs/namei.c:3497 [<ffffffff81830b6e>] do_filp_open+0x18e/0x250 fs/namei.c:3532 [<ffffffff817f53c1>] do_sys_open+0x201/0x420 fs/open.c:1036 [< inline >] SYSC_open fs/open.c:1054 [<ffffffff817f560d>] SyS_open+0x2d/0x40 fs/open.c:1049 [<ffffffff863fbb80>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 other info that might help us debug this: Possible unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&grp->list_mutex); lock(register_mutex#5); lock(&grp->list_mutex); lock(register_mutex#5); *** DEADLOCK *** 2 locks held by syz-executor/7154: #0: (register_mutex#4){+.+.+.}, at: [<ffffffff85019d7f>] odev_open+0x5f/0x90 sound/core/seq/oss/seq_oss.c:137 #1: (&grp->list_mutex){++++.+}, at: [<ffffffff850138bb>] check_and_subscribe_port+0x5b/0x5c0 sound/core/seq/seq_ports.c:495 stack backtrace: CPU: 0 PID: 7154 Comm: syz-executor Not tainted 4.8.0-rc1+ #11 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff878b7680 ffff880038b670f0 ffffffff82a5f719 ffffffff00000000 fffffbfff0f16ed0 ffffffff8901a0d0 ffffffff8901a0d0 ffffffff890191a0 ffff88003da2cf08 ffff88003da2c6c0 ffff880038b67140 ffffffff814708a8 Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82a5f719>] dump_stack+0x12e/0x185 lib/dump_stack.c:51 [<ffffffff814708a8>] print_circular_bug+0x288/0x340 kernel/locking/lockdep.c:1202 [< inline >] check_prev_add kernel/locking/lockdep.c:1829 [< inline >] check_prevs_add kernel/locking/lockdep.c:1939 [< inline >] validate_chain kernel/locking/lockdep.c:2266 [<ffffffff814791f4>] __lock_acquire+0x4d44/0x4d80 kernel/locking/lockdep.c:3335 [<ffffffff8147a3a8>] lock_acquire+0x208/0x430 kernel/locking/lockdep.c:3746 [< inline >] __mutex_lock_common kernel/locking/mutex.c:521 [<ffffffff863f0ef1>] mutex_lock_nested+0xb1/0xa20 kernel/locking/mutex.c:621 [<ffffffff84fd6d4b>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 [<ffffffff8502e7c7>] midisynth_subscribe+0xf7/0x350 sound/core/seq/seq_midi.c:188 [< inline >] subscribe_port sound/core/seq/seq_ports.c:427 [<ffffffff85013cc7>] check_and_subscribe_port+0x467/0x5c0 sound/core/seq/seq_ports.c:510 [<ffffffff85015da9>] snd_seq_port_connect+0x2c9/0x500 sound/core/seq/seq_ports.c:579 [<ffffffff850079b8>] snd_seq_ioctl_subscribe_port+0x1d8/0x2b0 sound/core/seq/seq_clientmgr.c:1480 [<ffffffff84ffe9e4>] snd_seq_do_ioctl+0x184/0x1e0 sound/core/seq/seq_clientmgr.c:2225 [<ffffffff84ffeae8>] snd_seq_kernel_client_ctl+0xa8/0x110 sound/core/seq/seq_clientmgr.c:2440 [<ffffffff85027664>] snd_seq_oss_midi_open+0x3b4/0x610 sound/core/seq/oss/seq_oss_midi.c:375 [<ffffffff85023d67>] snd_seq_oss_synth_setup_midi+0x107/0x4c0 sound/core/seq/oss/seq_oss_synth.c:281 [<ffffffff8501b0a8>] snd_seq_oss_open+0x748/0x8d0 sound/core/seq/oss/seq_oss_init.c:274 [<ffffffff85019d8a>] odev_open+0x6a/0x90 sound/core/seq/oss/seq_oss.c:138 [<ffffffff84f7040f>] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [<ffffffff818069ea>] chrdev_open+0x22a/0x4c0 fs/char_dev.c:392 [<ffffffff817f146f>] do_dentry_open+0x69f/0xca0 fs/open.c:736 [<ffffffff817f4aa5>] vfs_open+0x105/0x220 fs/open.c:849 [< inline >] do_last fs/namei.c:3374 [<ffffffff8182c35d>] path_openat+0x1efd/0x2f60 fs/namei.c:3497 [<ffffffff81830b6e>] do_filp_open+0x18e/0x250 fs/namei.c:3532 [<ffffffff817f53c1>] do_sys_open+0x201/0x420 fs/open.c:1036 [< inline >] SYSC_open fs/open.c:1054 [<ffffffff817f560d>] SyS_open+0x2d/0x40 fs/open.c:1049 [<ffffffff863fbb80>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207

2 4

[alsa-devel] [PATCH 0/2] Support non-ssi cpu-dai for sgtl5000
by Winter Wang 30 Aug '16

30 Aug '16

These patches support sgtl5000 to be attached to other non-ssi cpu-dais like SAI. Winter Wang (2): ASoC: imx-sgtl5000: Support non-ssi cpu-dai ARM: dts: modify sgtl5000 property to support non-ssi cpu-dai .../bindings/sound/imx-audio-sgtl5000.txt | 4 +-- arch/arm/boot/dts/imx25-pdk.dts | 2 +- arch/arm/boot/dts/imx51-babbage.dts | 2 +- arch/arm/boot/dts/imx53-m53evk.dts | 2 +- arch/arm/boot/dts/imx53-mba53.dts | 2 +- arch/arm/boot/dts/imx53-qsb-common.dtsi | 2 +- arch/arm/boot/dts/imx53-tx53.dtsi | 2 +- arch/arm/boot/dts/imx53-voipac-bsb.dts | 2 +- arch/arm/boot/dts/imx6dl-riotboard.dts | 2 +- arch/arm/boot/dts/imx6q-bx50v3.dtsi | 2 +- arch/arm/boot/dts/imx6q-gw5400-a.dts | 2 +- arch/arm/boot/dts/imx6q-h100.dts | 2 +- arch/arm/boot/dts/imx6q-icore-rqs.dts | 2 +- arch/arm/boot/dts/imx6q-tbs2910.dts | 2 +- arch/arm/boot/dts/imx6qdl-apalis.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-apf6dev.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw52xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw53xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-hummingboard.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nit6xlite.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nitrogen6_max.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nitrogen6x.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-rex.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-sabrelite.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-tx6.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-wandboard.dtsi | 2 +- sound/soc/fsl/imx-sgtl5000.c | 41 ++++++++++++++-------- 28 files changed, 55 insertions(+), 42 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH v6 0/2] amidi delay option
by Felipe F. Tonello 30 Aug '16

30 Aug '16

Small improvement to amidi, adding useful functionality. changes from v5: * wait until MIDI message is competely sent to sleep changes from v4: * simplified code changes from v3: * fixed a bug if memchr() returns NULL changes from v2: * only check of end of sysex (0xf7) * use memchr() instead of direct loop approach * fixed commit message changes from v1: * remove delay in between any MIDI message, only SysEx * added patch to use getopt_long() arguments Felipe F. Tonello (2): amidi: add delay option amidi: use GNU getopt_long() proper argument names amidi/amidi.1 | 14 ++++++++++ amidi/amidi.c | 82 ++++++++++++++++++++++++++++++++++++++++++++++------------- 2 files changed, 79 insertions(+), 17 deletions(-) -- 2.9.3

3 6

Re: [alsa-devel] No sound for several minutes after reboot
by Takashi Iwai 30 Aug '16

30 Aug '16

On Sat, 27 Aug 2016 13:39:08 +0200, Markus Trippelsdorf wrote: > > On 2016.08.27 at 11:50 +0200, Markus Trippelsdorf wrote: > > Hi, > > > > since several kernel releases I get no sound for several minutes after > > reboot. After this delay it starts working normally. > > > > Is this some kind of speaker protection? Is there a parameter that I can > > set to get sound immediately? > > > > % aplay -l > > **** List of PLAYBACK Hardware Devices **** > > card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: SB [HDA ATI SB], device 3: VT1708S Digital [VT1708S Digital] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 1: HDMI [HDA ATI HDMI], device 3: Generic Digital [Generic Digital] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > > > card 0, device 3 [VT1708S Digital] is the device that I use. It is a > > TOSLINK-S/PDIF link to my DAC. > > Here is the diff (bad (after reboot) vs. good (after a few minutes)): > > @@ -121,7 +121,7 @@ > Control: name="Surround Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 > - Amp-Out vals: [0x1c 0x1c] > + Amp-Out vals: [0x26 0x26] > Converter: stream=0, channel=0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > @@ -174,12 +174,12 @@ > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 > - Power: setting=D0, actual=D0 > + Power: setting=D3, actual=D3 > Connection: 1 > 0x1e So this looks like the only significant change. Others are about the volume level, and should lead to the complete mute. Could you try to adjust the power state manually while it happens? hda-verb /dev/snd/hwC0D0 0xXX SET_POWER 0x00 where 0xXX is filled with the node number of this widget. BTW, at the next time, take a diff with -p option. This will show the node ID in the diff. Takashi

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[alsa-devel] iMX6SX having SSI TXC from external and TXFS from internal, is it possible?
by Roberto Fichera 30 Aug '16

30 Aug '16

Hi, I'm playing with a custom iMX6SX board having the SSI connected to a SLIC in TDM mode. SSI has TXC (PCLK) coming from an external oscillator running at 2.048MHz while TXFS (FSYNC) is set to be generated from internal or at least this is what I'm trying to do. The SSI documentation is not really clear in case, so wondering to know if it is possible to generate FSYNC with this particular configuration: SSI set in I2S normal mode, synchronous transfers, network mode, TXC taken from external oscillator. The relevant part reported in iMX6SXRM doc (62.8.4 SSI Clocking) is: "In master and synchronous mode, the unused port SRCK is used as network clock (oversampling clock) enabled by the SCR register bit 15, SYS_CLK_EN. This network clock is an oversampling clock of the frame sync clock (STFS). In this mode, the word length (WL), Prescaler Range (PSR), Prescaler Modulus (PM) and Frame rate (DC) selects the ratio of network clock to sampling clock STFS. In case of I2S mode, the network clock (oversampling clock) can be made available on this port if the SYS_CLK_EN bit is set." Accordingly with Figure 62-22 in the iMX6SX reference manual network clock on SRCK can be activated only when both TXDIR and SYS_CLK_EN are set. But if TXDIR is set I cannot get STCK from external oscillator because SSI is now generating it from internal. I've also changed the way how audmux is routing FYSNC, internal port is SSI1 and external is SSI6 int audmux_config(unsigned int master, unsigned int slave) { unsigned int ptcr, pdcr; ptcr = IMX_AUDMUX_V2_PTCR_SYN /* Synchronous mode */ | IMX_AUDMUX_V2_PTCR_TCSEL(slave); /* transmit clock select from external port */ pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); audmux_v2_configure_port(master, ptcr, pdcr); /* configure internal port */ ptcr = IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR /* Transmit Frame Sync from Internal */ | IMX_AUDMUX_V2_PTCR_TFSEL(master); /* from master port */ pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); audmux_v2_configure_port(slave, ptcr, pdcr); /* configure external port */ return 0; } and SSI1 is set as: scr = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_SYN | /* Synchronous mode */ CCSR_SSI_SCR_NET; /* Network mode to enable TDM */ stcr = CCSR_SSI_STCR_TXBIT0 | /* LSB Aligned */ CCSR_SSI_STCR_TFDIR | /* Transmit Frame Direction Internal - generated internally */ CCSR_SSI_STCR_TFEN0 | /* Enable TX FIFO0 */ CCSR_SSI_STCR_TSCKP; /* Transmit Clock Polarity - Data Clocked out on falling edge */ Doesn't anyone know if it's possible to have this particular TDM configuration? Thanks in advance, Roberto Fichera.

1 0

[alsa-devel] [PATCH 0/4] ASoC: dapm: Various fixes for dapm kcontrols
by Chen-Yu Tsai 30 Aug '16

30 Aug '16

Hi everyone, This series is a bunch of fixes for some problems I encountered while experimenting with various dapm kcontrols for Allwinner A31 support in sun4i-codec. Patch 1 fixes a possible uninitialized variable usage in snd_soc_dapm_get_volsw(). This doesn't show up as a compiler warning but I'm fairly certain it might happen. Patch 2 fixes setting the value for an enum kcontrol's second channel. Patch 3 fixes kcontrol creation for output driver widgets. An output driver widget is the same as a PGA widget, just with a different priority. Patch 4 adds an error message if anyone attempts to add hardware backed kcontrols on PGA widgets. Originally kcontrols on PGAs weren't supported at all. Support for them on virtual widgets were added later, but a guard for real widgets was missing. Regards ChenYu Chen-Yu Tsai (4): ASoC: dapm: Fix possible uninitialized variable in snd_soc_dapm_get_volsw() ASoC: dapm: Fix value setting for _ENUM_DOUBLE MUX's second channel ASoC: dapm: Fix kcontrol creation for output driver widget ASoC: dapm: Give error when adding hardware backed controls for PGA widgets sound/soc/soc-dapm.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) -- 2.9.3

2 5

[alsa-devel] [PATCH v2] ASoC: core: fix shift used for second item in snd_soc_get_enum_double
by Jaswinder Jassal 30 Aug '16

30 Aug '16

Incorrect shift value was being used to extract the second item. Signed-off-by: Jaswinder Jassal <jjassal(a)opensource.wolfsonmicro.com> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a513a34..9fc1a7b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -77,7 +77,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_r) & e->mask; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } -- 2.1.4

2 1

[alsa-devel] Range of pitchbend MIDI events
by Maurizio Berti 30 Aug '16

30 Aug '16

Hello, I was playing with pyalsa and I realized an unexpected behavior with pitchbend events. The pitchbend range should be, by MIDI standard, between -8192 and +8191, or 0-16384. Being a 3 byte event, it should ignore any subsequent byte received. Using pyalsa I'm able to create events with values greater than 127 and they are translated using as many bytes as possible; in fact, if I create a note event with a note value of 8193, I receive a value 1 (using the first byte of a 2 byte value with 7bit format). That's fine, I suppose it's by architecture (I'm not a programmer, I just play around with python, I'm just assuming this). The problem comes with pitchbend values, since it looks like alsa accepts values greater than 8191 and does not ignore the exceeding bytes as it should; that is, I'm able to send 8192, and that's what vkeybd does. The transformation here is the issue, since receiving programs seem to translate only the latest two bytes, that's why vkeybd (and other software too) shouldn't have a 8192 value. That can be confusing, for example, using a2jmidi: the end of the pitchbend scale is suddenly a 0 value; I know that vkeybd is not the ultimate software in music production, but, still, it shouldn't be able to do so, I think :) I don't know if, maybe, there's some less known aspect regarding pitchbend, and, as I mentioned before, I don't know a thing about programming, nor how actually works ALSA under the hood. Anyway, I think there should be some kind of range control, at least with pyalsa, if not by alsa at all. What do you think? Regards, Maurizio -- È difficile avere una convinzione precisa quando si parla delle ragioni del cuore. - "Sostiene Pereira", Antonio Tabucchi http://www.jidesk.net

3 5

[alsa-devel] [PATCH v2] ALSA: hda - Enable subwoofer on Dell Inspiron 7559
by Kai-Heng Feng 30 Aug '16

30 Aug '16

The subwoofer on Inspiron 7559 was disabled originally. Applying a pin fixup to node 0x1b can enable it and make it work. Old pin: 0x411111f0 New pin: 0x90170151 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7100f05..2a3dd18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4855,6 +4855,7 @@ enum { ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, ALC298_FIXUP_SPK_VOLUME, + ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, }; static const struct hda_fixup alc269_fixups[] = { @@ -5516,6 +5517,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170151 }, + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5560,6 +5570,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), -- 2.9.3

2 1

[alsa-devel] [PATCH 00/12] ASoC: hdac: Add hdac generic driver
by Subhransu S. Prusty 30 Aug '16

30 Aug '16

HDA devices generically can be modelled with DAPM in ASoC. We would like to create a framework for the HDA devices in ASoC. This will follow HDA standard. This will be registered as generic "virtual" class device (Not implemented in this series). If there is no match found for vendor id and device id, then the class driver will be loaded. For vendor specific devices, vendor quirks will be loaded based on the match. Additional widgets, controls and route will be created in a vendor specific way through vendor ops. This series adds a framework in ASoC to model HDA codecs. HDA widgets are enumerated and one or multiple DAPM widget(s) are created. Connection list is queried for each widget to identify the connection between two endpoints and modelled using DAPM graph. Set of event handlers are defined for each widget type. Based on DAPM events required verbs are sent to program codec. Finally a function is exported to query for the device endpoint configuration to create machine controls. Features yet to be added: - Optional features of widgets will be added using additional dapm widgets or controls - Pin retasking will be done by adding all possible dapm route map - Class driver to load the generic driver if no match found - Jack detection Subhransu S. Prusty (12): ALSA: hdac: Add codec helper library ALSA: hda - Add macro to test pin widget's input capability ASoC: hdac: Add a generic hdac driver framework ASoC: hdac: Create DAPM model for HDA widgets ALSA: hdac: Move 'to_hda_ext_device' helper to core ASoC: hdac_hdmi: use get_edev helper from core ASoC: hdac: Build DAPM graph by querying through widget connection list ASoC: hdac: Register widget event handlers ALSA: hda - macro to get default config device of pin widgets ASoC: hdac: Export API to create machine controls ALSA: hdac: Implement a match function ASoC: hdac: Add coefficient init callback for RT286 include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 1 + sound/hda/ext/Makefile | 3 +- sound/hda/ext/hdac_codec.c | 182 ++++ sound/hda/ext/hdac_codec.h | 60 ++ sound/hda/ext/hdac_ext_bus.c | 12 + sound/hda/local.h | 13 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_generic.c | 1754 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_generic.h | 36 + sound/soc/codecs/hdac_hdmi.c | 8 +- 12 files changed, 2069 insertions(+), 8 deletions(-) create mode 100644 sound/hda/ext/hdac_codec.c create mode 100644 sound/hda/ext/hdac_codec.h create mode 100644 sound/soc/codecs/hdac_generic.c create mode 100644 sound/soc/codecs/hdac_generic.h -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Enable subwoofer on Dell Inspiron 7559
by Kai-Heng Feng 30 Aug '16

30 Aug '16

The subwoofer on Inspiron 7559 does not work originally. Applying a pin fixup can make it work. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7100f05..2a3dd18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4855,6 +4855,7 @@ enum { ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, ALC298_FIXUP_SPK_VOLUME, + ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, }; static const struct hda_fixup alc269_fixups[] = { @@ -5516,6 +5517,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170151 }, + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5560,6 +5570,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), -- 2.9.3

3 4

[alsa-devel] [PATCH v2 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock
by Takashi Sakamoto 30 Aug '16

30 Aug '16

Hi, In ALSA fireworks driver and firewire-tascam driver, current implementation for hwdep interface have issues for page fault handling. For example, when using single core processor, after executing lock_page() in page fault handler and call task scheduler, then the context never regain CPU again, because any interrupts are disabled by these drivers and the context can't catch waiting event. This patchset fixes the issues. The issues were reported by Vaishali Thakkar, with a help of coccinelle. Thanks a lot for her care to these minor drivers. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111887.html Takashi Sakamoto (2): ALSA: fireworks: accessing to user space outside spinlock ALSA: firewire-tascam: accessing to user space outside spinlock sound/firewire/fireworks/fireworks_hwdep.c | 21 ++++++++++++++----- sound/firewire/tascam/tascam-hwdep.c | 33 ++++++++++-------------------- 2 files changed, 27 insertions(+), 27 deletions(-) -- 2.7.4

3 6

[alsa-devel] [PATCH] ALSA: seq: initialize whole fields of automatic variable with union type
by Takashi Sakamoto 30 Aug '16

30 Aug '16

Currently, automatic variable of 'union ioctl_arg' type is initialized by designated initialization. Although, the actual effect is interpretation of early element of int type and initialization of 'int pversion'. Therefore the first field corresponding to int type is initialized to zero. This is against my expectation to initialize whole fields. This commit uses memset() to initialize the variable, instead of designated initialization. Fixes: 04a56dd8ed0d ('ALSA: seq: change ioctl command operation to get data in kernel space') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/seq/seq_clientmgr.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 286394b..fd50c38 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2100,7 +2100,7 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, struct snd_seq_client_pool client_pool; struct snd_seq_remove_events remove_events; struct snd_seq_query_subs query_subs; - } buf = {0}; + } buf; const struct ioctl_handler *handler; unsigned long size; int err; @@ -2114,6 +2114,9 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, } if (handler->cmd == 0) return -ENOTTY; + + memset(&buf, 0, sizeof(union ioctl_arg)); + /* * All of ioctl commands for ALSA sequencer get an argument of size * within 13 bits. We can safely pick up the size from the command. -- 2.7.4

2 2

[alsa-devel] [PATCH 0/2] alsabat: add power management S3 test
by keqiao.zhang＠intel.com 30 Aug '16

30 Aug '16

From: "Keqiao, Zhang" <keqiao.zhang(a)intel.com> The patch set provide a method for QA to quick test audio during system S3 The first patch fix -86 error when system suspend/resume The second patch provide a test script Keqiao, Zhang (2): alsabat: fix alsabat -86 error alsabat: add system power management S3 test bat/alsa.c | 10 ++++++++++ bat/alsabat-test.sh | 34 ++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+) -- 2.5.0

2 3

[alsa-devel] [PATCH] ALSA: hda - Add headset mic quirk for Dell Inspiron 5468
by Shrirang Bagul 30 Aug '16

30 Aug '16

This patch enables headset microphone on some variants of Dell Inspiron 5468. (Dell SSID 0x07ad) BugLink: https://bugs.launchpad.net/bugs/1617900 Signed-off-by: Shrirang Bagul <shrirang.bagul(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7100f05..02fd252 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5895,6 +5895,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60170}, {0x14, 0x90170120}, {0x21, 0x02211030}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell Inspiron 5468", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60180}, + {0x14, 0x90170120}, + {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, -- 2.7.4

3 2

[alsa-devel] [PATCH v9 0/2] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 30 Aug '16

30 Aug '16

This patch adds support for some mixer controls: - Line-In - FM-In - Mic 2 - Capture Source - Differential Line-In v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (2): ASoC: sun4i-codec: Distinguish sun4i from sun7i Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 278 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 259 insertions(+), 19 deletions(-)

3 5

[alsa-devel] [PATCH 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock
by Takashi Sakamoto 30 Aug '16

30 Aug '16

Hi, In ALSA fireworks driver and firewire-tascam driver, current implementation for hwdep interface have issues for page fault handling. For example, when using single core processor, after executing lock_page() in page fault handler and call task scheduler, then the context never run again, because any interrupts are disabled by these drivers and the context can't catch waiting event. This patchset fixes the issues. The issues were reported by Vaishali Thakkar, with a help of coccinelle. Thanks a lot for his care to these minor drivers. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111887.html Takashi Sakamoto (2): ALSA: fireworks: accessing to user space outside spinlock ALSA: firewire-tascam: accessing to user space outside spinlock sound/firewire/fireworks/fireworks_hwdep.c | 20 ++++++++++++++----- sound/firewire/tascam/tascam-hwdep.c | 32 ++++++++++-------------------- 2 files changed, 25 insertions(+), 27 deletions(-) -- 2.7.4

2 4

[alsa-devel] Translating USB device names to ALSA client numbers
by Dejay 30 Aug '16

30 Aug '16

Hello, I'm trying to write some udev wrappers to automatically connect USB MIDI devices to each other. My logic could be really robust if I had a way of mapping ALSA client numbers (e.g. from 'aconnect -l', such as '20:0') to physical USB devices (either by bus ID, /dev/ symlink name, /proc/asound/ files, etc.) My ultimate goal is so that when a udev rule executes upon hotplugging a MIDI device, I can reference that MIDI device by client number and then execute an aconnect command. Thus, I need some way to map USB device information to the ALSA client numbers that get reported by aconnect. Is there any way to accomplish this? I tried to review the ALSA source code, but my eyes got jumbly after awhile. Thanks, Dejay

1 1

[alsa-devel] [PATCH] ASoC: wm8962: fix spelling mistake "mesurement" -> "measurement"
by Colin King 29 Aug '16

29 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f3109da..97b0228 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3713,7 +3713,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, ARRAY_SIZE(wm8962_dc_measure)); if (ret != 0) dev_err(&i2c->dev, - "Failed to configure for DC mesurement: %d\n", + "Failed to configure for DC measurement: %d\n", ret); } -- 2.9.3

2 1

[alsa-devel] [RFC PATCH v2 0/7] AC97 device/driver model revamp
by Robert Jarzmik 29 Aug '16

29 Aug '16

It all started in the pxa device-tree submission here : https://lkml.org/lkml/2016/2/25/965 It will be maintained in : git fetch https://github.com/rjarzmik/linux.git work/ac97 And now it transformed into this RFC, which would bring a ground for AC'97 devices closer to the linux device/driver model. The driving ideas are still the same, and I put them in [1] for memory. This is the second opus of the RFC. I'm intending to stop the RFC cycle here if possible and have a true PATCH v1 submission after this RFC v2 unless there is a deep design flaw uncovered. I took into account Mark's and Takashi's remarks, I hope I forgot none. All the changes should be documented in the first 2 patches mainly. I also added : - the AC97 link clock For now, bus code doesn't disable it in suspend and re-enable in resume, leaving the controller decide. I'm still pondering if for S2RAM (as opposed to runtime suspend), bus code should disable the clock. - the .h guards I'm not particularly happy with my naming even for v2, feel free to propose anything better, the codec.h is particularly ugly. - statics and namespace pollution I made more functions static, limiting further the namespace pollution. - Kconfig/Makefile layout change I get the feeling that the KConfig "select" flavor would have better been a "depends on" one. And yet I didn't find a good way to enforce it, because of the way sound/soc/codecs/Kconfig is designed, and I'm afraid to break the structure by doing a "depends on AC97_BUS_NEW" in WM9713 configuration. Let's have another review cycle, that will let me test this serie more thoroughly, especially the suspend/resume and the reset in the resume, which doesn't work, and requires me to test more deeply. I'll also test by removing the wm9713 change to see how robust the implementation is. Happy review. -- Robert Robert Jarzmik (7): 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 ARM: pxa: mioa701 convert to the new AC97 bus ASoC: mioa701_wm9713: convert to new ac97 bus arch/arm/mach-pxa/mioa701.c | 12 +- include/sound/ac97/codec.h | 111 ++++++++++++ include/sound/ac97/compat.h | 21 +++ include/sound/ac97/controller.h | 80 +++++++++ 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 | 18 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 10 ++ sound/ac97/bus.c | 381 ++++++++++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 ++ sound/ac97/snd_ac97_compat.c | 104 +++++++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 39 ++-- sound/soc/Kconfig | 4 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9713.c | 62 ++++++- sound/soc/pxa/Kconfig | 4 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/pxa2xx-ac97.c | 20 ++- 23 files changed, 1132 insertions(+), 282 deletions(-) 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 10

[alsa-devel] [PATCH 1/1] ASoC: Intel: Atom: add a missing star in a memcpy call
by Nicolas Iooss 29 Aug '16

29 Aug '16

In sst_prepare_and_post_msg(), when a response is received in "block", the following code gets executed: *data = kzalloc(block->size, GFP_KERNEL); memcpy(data, (void *) block->data, block->size); The memcpy() call overwrites the content of the *data pointer instead of filling the newly-allocated memory (which pointer is hold by *data). Fix this by using *data in the memcpy() call. Fixes: 60dc8dbacb00 ("ASoC: Intel: sst: Add some helper functions") Cc: stable(a)vger.kernel.org # 3.19.x Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index adb32fefd693..7c398b7c9d4b 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -289,7 +289,7 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, ret = -ENOMEM; goto out; } else - memcpy(data, (void *) block->data, block->size); + memcpy(*data, (void *) block->data, block->size); } } out: -- 2.9.3

3 8

[alsa-devel] sound: division by 0 in snd_hrtimer_callback
by Dmitry Vyukov 28 Aug '16

28 Aug '16

Hello, The following program triggers divide error in snd_hrtimer_callback: divide error: 0000 [#1] SMP DEBUG_PAGEALLOC KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 3 PID: 17469 Comm: syz-executor Not tainted 4.8.0-rc3+ #33 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88003dc4a640 task.stack: ffff88003c690000 RIP: 0010:[<ffffffff8575b49c>] [< inline >] ktime_divns ./include/linux/ktime.h:195 [<ffffffff8575b49c>] snd_hrtimer_callback+0x1bc/0x3c0 sound/core/hrtimer.c:62 RSP: 0018:ffff88006d507d98 EFLAGS: 00010006 RAX: 0000000000005f28 RBX: ffff88006c4acb48 RCX: 0000000000000000 RDX: 0000000000000000 RSI: ffff88006c4acb60 RDI: ffffffff884b7a68 RBP: ffff88006d507df0 R08: 000000687d030efd R09: ffff88007fff70c8 R10: ffff88007fff70d8 R11: ffff88007fff70d0 R12: ffff88003cda70c0 R13: ffff88003cda7174 R14: ffff88003cda71f0 R15: ffff88006c4acb40 FS: 00007f131f945700(0000) GS:ffff88006d500000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000451ed0 CR3: 000000005bb88000 CR4: 00000000000006e0 Stack: 0000000000000000 ffff88006c4acb60 ffff88003cda7178 0000000000005f28 ffff88006c4acba0 0000000000000000 ffff88006c4acb48 0000000000000003 ffff88006d51b400 dffffc0000000000 dffffc0000000000 ffff88006d507ee8 Call Trace: <IRQ> [< inline >] __run_hrtimer kernel/time/hrtimer.c:1238 [<ffffffff814f48b5>] __hrtimer_run_queues+0x325/0xe70 kernel/time/hrtimer.c:1302 [<ffffffff814f726b>] hrtimer_interrupt+0x18b/0x420 kernel/time/hrtimer.c:1336 [<ffffffff81268f8f>] local_apic_timer_interrupt+0x6f/0xe0 arch/x86/kernel/apic/apic.c:933 [<ffffffff86c27d96>] smp_apic_timer_interrupt+0x76/0xa0 arch/x86/kernel/apic/apic.c:957 [<ffffffff86c26e4c>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:455 <EOI> [< inline >] spin_unlock_irqrestore ./include/linux/spinlock.h:362 [<ffffffff85750f7d>] snd_timer_start1+0xdd/0x590 sound/core/timer.c:464 [<ffffffff857532b5>] snd_timer_continue+0x45/0x80 sound/core/timer.c:593 [< inline >] snd_timer_user_continue sound/core/timer.c:1821 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1888 [<ffffffff85758ce9>] snd_timer_user_ioctl+0xe99/0x2470 sound/core/timer.c:1903 [< inline >] vfs_ioctl fs/ioctl.c:43 [<ffffffff8186f1fc>] do_vfs_ioctl+0x18c/0x1080 fs/ioctl.c:675 [< inline >] SYSC_ioctl fs/ioctl.c:690 [<ffffffff8187017f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:681 [<ffffffff86c25480>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 Code: 6b 45 e5 fb 8b 0d e5 ca 64 05 48 0f af 4d d0 48 85 c9 0f 88 6a 01 00 00 48 89 4d a8 e8 4e 45 e5 fb 48 8b 45 c0 48 8b 4d a8 48 99 <48> f7 f9 48 01 45 d0 e8 38 45 e5 fb 4c 89 fa 48 b8 00 00 00 00 RIP [< inline >] ktime_divns ./include/linux/ktime.h:195 RIP [<ffffffff8575b49c>] snd_hrtimer_callback+0x1bc/0x3c0 sound/core/hrtimer.c:62 RSP <ffff88006d507d98> ---[ end trace b12f88d370b1ccee ]--- Kernel panic - not syncing: Fatal exception in interrupt On commit 61c04572de404e52a655a36752e696bbcb483cf5 (Aug 25). // autogenerated by syzkaller (http://github.com/google/syzkaller) #ifndef __NR_mmap #define __NR_mmap 9 #endif #ifndef __NR_syz_open_dev #define __NR_syz_open_dev 1000001 #endif #ifndef __NR_ioctl #define __NR_ioctl 16 #endif #ifndef __NR_syz_fuse_mount #define __NR_syz_fuse_mount 1000003 #endif #ifndef __NR_syz_fuseblk_mount #define __NR_syz_fuseblk_mount 1000004 #endif #ifndef __NR_syz_open_pts #define __NR_syz_open_pts 1000002 #endif #include <fcntl.h> #include <pthread.h> #include <setjmp.h> #include <signal.h> #include <stddef.h> #include <stdint.h> #include <stdio.h> #include <stdlib.h> #include <string.h> #include <sys/ioctl.h> #include <sys/stat.h> #include <sys/syscall.h> #include <sys/types.h> #include <unistd.h> __thread int skip_segv; __thread jmp_buf segv_env; static void segv_handler(int sig, siginfo_t* info, void* uctx) { if (__atomic_load_n(&skip_segv, __ATOMIC_RELAXED)) _longjmp(segv_env, 1); exit(sig); } static void install_segv_handler() { struct sigaction sa; memset(&sa, 0, sizeof(sa)); sa.sa_sigaction = segv_handler; sa.sa_flags = SA_NODEFER | SA_SIGINFO; sigaction(SIGSEGV, &sa, NULL); sigaction(SIGBUS, &sa, NULL); } #define NONFAILING(...) \ { \ __atomic_fetch_add(&skip_segv, 1, __ATOMIC_SEQ_CST); \ if (_setjmp(segv_env) == 0) { \ __VA_ARGS__; \ } \ __atomic_fetch_sub(&skip_segv, 1, __ATOMIC_SEQ_CST); \ } static uintptr_t syz_open_dev(uintptr_t a0, uintptr_t a1, uintptr_t a2) { if (a0 == 0xc || a0 == 0xb) { char buf[128]; sprintf(buf, "/dev/%s/%d:%d", a0 == 0xc ? "char" : "block", (uint8_t)a1, (uint8_t)a2); return open(buf, O_RDWR, 0); } else { char buf[1024]; char* hash; strncpy(buf, (char*)a0, sizeof(buf)); buf[sizeof(buf) - 1] = 0; while ((hash = strchr(buf, '#'))) { *hash = '0' + (char)(a1 % 10); a1 /= 10; } return open(buf, a2, 0); } } static uintptr_t syz_open_pts(uintptr_t a0, uintptr_t a1) { int ptyno = 0; if (ioctl(a0, TIOCGPTN, &ptyno)) return -1; char buf[128]; sprintf(buf, "/dev/pts/%d", ptyno); return open(buf, a1, 0); } static uintptr_t syz_fuse_mount(uintptr_t a0, uintptr_t a1, uintptr_t a2, uintptr_t a3, uintptr_t a4, uintptr_t a5) { uint64_t target = a0; uint64_t mode = a1; uint64_t uid = a2; uint64_t gid = a3; uint64_t maxread = a4; uint64_t flags = a5; int fd = open("/dev/fuse", O_RDWR); if (fd == -1) return fd; char buf[1024]; sprintf(buf, "fd=%d,user_id=%ld,group_id=%ld,rootmode=0%o", fd, (long)uid, (long)gid, (unsigned)mode & ~3u); if (maxread != 0) sprintf(buf + strlen(buf), ",max_read=%ld", (long)maxread); if (mode & 1) strcat(buf, ",default_permissions"); if (mode & 2) strcat(buf, ",allow_other"); syscall(SYS_mount, "", target, "fuse", flags, buf); return fd; } static uintptr_t syz_fuseblk_mount(uintptr_t a0, uintptr_t a1, uintptr_t a2, uintptr_t a3, uintptr_t a4, uintptr_t a5, uintptr_t a6, uintptr_t a7) { uint64_t target = a0; uint64_t blkdev = a1; uint64_t mode = a2; uint64_t uid = a3; uint64_t gid = a4; uint64_t maxread = a5; uint64_t blksize = a6; uint64_t flags = a7; int fd = open("/dev/fuse", O_RDWR); if (fd == -1) return fd; if (syscall(SYS_mknodat, AT_FDCWD, blkdev, S_IFBLK, makedev(7, 199))) return fd; char buf[256]; sprintf(buf, "fd=%d,user_id=%ld,group_id=%ld,rootmode=0%o", fd, (long)uid, (long)gid, (unsigned)mode & ~3u); if (maxread != 0) sprintf(buf + strlen(buf), ",max_read=%ld", (long)maxread); if (blksize != 0) sprintf(buf + strlen(buf), ",blksize=%ld", (long)blksize); if (mode & 1) strcat(buf, ",default_permissions"); if (mode & 2) strcat(buf, ",allow_other"); syscall(SYS_mount, blkdev, target, "fuseblk", flags, buf); return fd; } static uintptr_t execute_syscall(int nr, uintptr_t a0, uintptr_t a1, uintptr_t a2, uintptr_t a3, uintptr_t a4, uintptr_t a5, uintptr_t a6, uintptr_t a7, uintptr_t a8) { switch (nr) { default: return syscall(nr, a0, a1, a2, a3, a4, a5); case __NR_syz_open_dev: return syz_open_dev(a0, a1, a2); case __NR_syz_open_pts: return syz_open_pts(a0, a1); case __NR_syz_fuse_mount: return syz_fuse_mount(a0, a1, a2, a3, a4, a5); case __NR_syz_fuseblk_mount: return syz_fuseblk_mount(a0, a1, a2, a3, a4, a5, a6, a7); } } long r[42]; int main() { install_segv_handler(); memset(r, -1, sizeof(r)); r[0] = execute_syscall(__NR_mmap, 0x20000000ul, 0xd000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul, 0, 0, 0); NONFAILING(memcpy( (void*)0x20003000, "\x2f\x64\x65\x76\x2f\x73\x6e\x64\x2f\x74\x69\x6d\x65\x72\x00", 15)); r[2] = execute_syscall(__NR_syz_open_dev, 0x20003000ul, 0x0ul, 0x1ul, 0, 0, 0, 0, 0, 0); NONFAILING(*(uint32_t*)0x2000bfcc = (uint32_t)0x1); NONFAILING(*(uint32_t*)0x2000bfd0 = (uint32_t)0x4); NONFAILING(*(uint32_t*)0x2000bfd4 = (uint32_t)0x2); NONFAILING(*(uint32_t*)0x2000bfd8 = (uint32_t)0x3); NONFAILING(*(uint32_t*)0x2000bfdc = (uint32_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe0 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe1 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe2 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe3 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe4 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe5 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe6 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe7 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe8 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe9 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfea = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfeb = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfec = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfed = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfee = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfef = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff0 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff1 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff2 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff3 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff4 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff5 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff6 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff7 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff8 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff9 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffa = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffb = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffc = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffd = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffe = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfff = (uint8_t)0x0); r[40] = execute_syscall(__NR_ioctl, r[2], 0x40345410ul, 0x2000bfccul, 0, 0, 0, 0, 0, 0); r[41] = execute_syscall(__NR_ioctl, r[2], 0x54a2ul, 0, 0, 0, 0, 0, 0, 0); return 0; }

2 1

[alsa-devel] [PATCH v2 1/1] ASoC: Intel: Atom: add a missing star in a memcpy call
by Nicolas Iooss 28 Aug '16

28 Aug '16

In sst_prepare_and_post_msg(), when a response is received in "block", the following code gets executed: *data = kzalloc(block->size, GFP_KERNEL); memcpy(data, (void *) block->data, block->size); The memcpy() call overwrites the content of the *data pointer instead of filling the newly-allocated memory (which pointer is hold by *data). Fix this by merging kzalloc+memcpy into a single kmemdup() call. Thanks Joe Perches for suggesting using kmemdup() Fixes: 60dc8dbacb00 ("ASoC: Intel: sst: Add some helper functions") Cc: stable(a)vger.kernel.org # 3.19.x Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index adb32fefd693..b1e6b8f34a6a 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -279,17 +279,15 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, if (response) { ret = sst_wait_timeout(sst, block); - if (ret < 0) { + if (ret < 0) goto out; - } else if(block->data) { - if (!data) - goto out; - *data = kzalloc(block->size, GFP_KERNEL); - if (!(*data)) { + + if (data && block->data) { + *data = kmemdup(block->data, block->size, GFP_KERNEL); + if (!*data) { ret = -ENOMEM; goto out; - } else - memcpy(data, (void *) block->data, block->size); + } } } out: -- 2.9.3

1 0

[alsa-devel] [PATCH] ARM: dts: am335x-wega: Add audio support
by Stefan Müller-Klieser 26 Aug '16

26 Aug '16

The wega board has a TLV320AIC3007 connected via McASP0. In the default configuration, no external crystal is mounted. We run a system clock of 25 MHz, so we use the audio codec PLL for audio clock generation. Signed-off-by: Stefan Müller-Klieser <s.mueller-klieser(a)phytec.de> --- arch/arm/boot/dts/am335x-wega.dtsi | 62 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 62 insertions(+) diff --git a/arch/arm/boot/dts/am335x-wega.dtsi b/arch/arm/boot/dts/am335x-wega.dtsi index 282f6d4..7f5f8ed 100644 --- a/arch/arm/boot/dts/am335x-wega.dtsi +++ b/arch/arm/boot/dts/am335x-wega.dtsi @@ -11,6 +11,10 @@ model = "Phytec AM335x phyBOARD-WEGA"; compatible = "phytec,am335x-wega", "phytec,am335x-phycore-som", "ti,am33xx"; + sound: sound_iface { + compatible = "ti,da830-evm-audio"; + }; + regulators { compatible = "simple-bus"; @@ -24,6 +28,58 @@ }; }; +/* Audio */ +&am33xx_pinmux { + mcasp0_pins: pinmux_mcasp0 { + pinctrl-single,pins = < + AM33XX_IOPAD(0x9AC, PIN_OUTPUT_PULLDOWN | MUX_MODE0) /* mcasp0_ahclkx.mcasp0_ahclkx */ + AM33XX_IOPAD(0x990, PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_aclkx.mcasp0_aclkx */ + AM33XX_IOPAD(0x994, PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_fsx.mcasp0_fsx */ + AM33XX_IOPAD(0x998, PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr0.mcasp0_axr0 */ + AM33XX_IOPAD(0x9A8, PIN_OUTPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr1.mcasp0_axr1 */ + >; + }; +}; + +&i2c0 { + tlv320aic3007: tlv320aic3007@18 { + compatible = "ti,tlv320aic3007"; + reg = <0x18>; + AVDD-supply = <&vcc3v3>; + IOVDD-supply = <&vcc3v3>; + DRVDD-supply = <&vcc3v3>; + DVDD-supply = <&vdig1_reg>; + status = "okay"; + }; +}; + +&mcasp0 { + pinctrl-names = "default"; + pinctrl-0 = <&mcasp0_pins>; + op-mode = <0>; /* DAVINCI_MCASP_IIS_MODE */ + tdm-slots = <2>; + serial-dir = < + 2 1 0 0 /* # 0: INACTIVE, 1: TX, 2: RX */ + >; + tx-num-evt = <16>; + rt-num-evt = <16>; + status = "okay"; +}; + +&sound { + ti,model = "AM335x-Wega"; + ti,audio-codec = <&tlv320aic3007>; + ti,mcasp-controller = <&mcasp0>; + ti,audio-routing = + "Line Out", "LLOUT", + "Line Out", "RLOUT", + "LINE1L", "Line In", + "LINE1R", "Line In"; + clocks = <&mcasp0_fck>; + clock-names = "mclk"; + status = "okay"; +}; + /* CAN Busses */ &am33xx_pinmux { dcan1_pins: pinmux_dcan1 { @@ -99,6 +155,12 @@ status = "okay"; }; +/* Power */ +&vdig1_reg { + regulator-boot-on; + regulator-always-on; +}; + /* UARTs */ &am33xx_pinmux { uart0_pins: pinmux_uart0 { -- 1.9.1

3 2

[alsa-devel] Toggle GPIO if ALSA playbacks begins or ends
by Jörg Krause 26 Aug '16

26 Aug '16

Hi, I am looking for a way to toggle a GPIO if ALSA is about to play or has stopped to turn an external amplifier on/off. Can it be done using libasound or can I use a plugin or write one? Best regards Jörg Krause

1 1

[alsa-devel] [PATCH v9 0/2] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 26 Aug '16

26 Aug '16

This patch adds support for some mixer controls: - Line-In - FM-In - Mic 2 - Capture Source - Differential Line-In v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (2): ASoC: sun4i-codec: Distinguish sun4i from sun7i Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 278 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 259 insertions(+), 19 deletions(-)

2 3

Re: [alsa-devel] [PATCH 8/9] ASoC: simple-card: code sync: rename num_link to num
by Kuninori Morimoto 26 Aug '16

26 Aug '16

Hi Mark This is my fault. I will post v2 patch. > [auto build test ERROR on asoc/for-next] > [also build test ERROR on next-20160824] > [cannot apply to v4.8-rc3] > [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] > [Suggest to use git(>=2.9.0) format-patch --base=<commit> (or --base=auto for convenience) to record what (public, well-known) commit your patch series was built on] > [Check https://git-scm.com/docs/git-format-patch for more information] > > url: https://github.com/0day-ci/linux/commits/Kuninori-Morimoto/ASoC-simple-card… > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > config: i386-randconfig-x011-201634 (attached as .config) > compiler: gcc-6 (Debian 6.1.1-9) 6.1.1 20160705 > reproduce: > # save the attached .config to linux build tree > make ARCH=i386 > > Note: the linux-review/Kuninori-Morimoto/ASoC-simple-card-tidyup-and-synchronize-patches/20160825-100058 HEAD 36fb10f61ffb92eb287328e22cad00a486495747 builds fine. > It only hurts bisectibility. > > All errors (new ones prefixed by >>): > > sound/soc/generic/simple-card.c: In function 'asoc_simple_card_probe': > >> sound/soc/generic/simple-card.c:398:40: error: 'num_links' undeclared (first use in this function) > sizeof(*priv) + sizeof(*dai_link) * num_links, > ^~~~~~~~~ > sound/soc/generic/simple-card.c:398:40: note: each undeclared identifier is reported only once for each function it appears in > > vim +/num_links +398 sound/soc/generic/simple-card.c > > 6805a759b Kuninori Morimoto 2016-08-25 392 num = of_get_child_count(np); > 2080437d3 Xiubo Li 2014-09-03 393 else > 6805a759b Kuninori Morimoto 2016-08-25 394 num = 1; > f2390880e Kuninori Morimoto 2012-04-08 395 > 0dd4fc3c2 Xiubo Li 2014-09-10 396 /* Allocate the private data and the DAI link array */ > cf7dc23cb Jean-Francois Moine 2014-03-20 397 priv = devm_kzalloc(dev, > 6a91a17bd Jean-Francois Moine 2014-03-20 @398 sizeof(*priv) + sizeof(*dai_link) * num_links, > cf7dc23cb Jean-Francois Moine 2014-03-20 399 GFP_KERNEL); > ca65b492c Jean-Francois Moine 2014-01-15 400 if (!priv) > ca919fe4b Xiubo Li 2014-01-14 401 return -ENOMEM; > > :::::: The code at line 398 was first introduced by commit > :::::: 6a91a17bd7b92b2d2aa9ece85457f52a62fd7708 ASoC: simple-card: Handle many DAI links > > :::::: TO: Jean-Francois Moine <moinejf(a)free.fr> > :::::: CC: Mark Brown <broonie(a)linaro.org> > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation > [2 .config.gz <application/octet-stream (base64)>] > Best regards --- Kuninori Morimoto

1 0

[alsa-devel] ALSA hooks: dlopen failed: snd_dlsym_start: symbol not found
by Jörg Krause 26 Aug '16

26 Aug '16

Hi, I am trying to build a custom hook library. The minimal hook is: #include <alsa/asoundlib.h> int _snd_pcm_hook_audio_status_install(snd_pcm_t *pcm, snd_config_t *conf) { return 0; } SND_DLSYM_BUILD_VERSION(_snd_pcm_hook_audio_status_install, SND_CONFIG_DLSYM_VERSION_HOOK); I compile this as a shared library: $(CC) -fPIC -lasound -shared -Wl,-soname,lib$(NAME).so $^ -o $@ If I try to load this library using aplay I get the following error: dlopen failed: Error relocating /usr/lib/alsa-lib/libaudiostatus.so: snd_dlsym_start: symbol not found This is the output of the preprocessor: int _snd_pcm_hook_audio_status_install(snd_pcm_t *pcm, snd_config_t *conf) { return 0; } static struct snd_dlsym_link _snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook_001; void _snd_dlsym_constructor__snd_pcm_hook_audio_status_install_dlsym_conf ig_hook_001 (void) __attribute__ ((constructor)); void _snd_dlsym_constructor__snd_pcm_hook_audio_status_install_dlsym_conf ig_hook_001 (void) { _snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook _001.next = snd_dlsym_start; _snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook _001.dlsym_name = "_snd_pcm_hook_audio_status_install"; _snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook _001.dlsym_ptr = (void *)&_snd_pcm_hook_audio_status_install; snd_dlsym_start = &_snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook_001 ; }; What do I miss? Best regards Jörg Krause

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[alsa-devel] Alsa-lib 'cancelability'
by Alan Horstmann 25 Aug '16

25 Aug '16

Hi, I am wondering if there is a policy on the thread 'cancelability' / 'cancellability' of the functions in Alsa-lib? We are dissecting an issue on the Portaudio list which has appeared with the 'Locking API' version 1.1.2, when using the Portaudio callback API. The callback runs in it's own thread; when a user requests 'AbortStream' a thread cancel is requested. Sometimes it seems this can take effect during snd_pcm_mmap_commit(), so the function never completes and a lock is not released. Then, during the cleanup, snd_pcm_drop() attempts to aquire the lock, which is already held, and a lock-up occurs. Is it up to the application to ensure each Alsa function completes, eg by disabling cancelability on the thread for those periods, or is it reasonable to expect the functions to ensure they complete adequately? Any comment would be appreciated. Regards Alan

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[alsa-devel] [GIT PULL] ASoC fixes for v4.8
by Mark Brown 25 Aug '16

25 Aug '16

The following changes since commit fa8410b355251fd30341662a40ac6b22d3e38468: Linux 4.8-rc3 (2016-08-21 16:14:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.8-rc4 for you to fetch changes up to cfb89f2e7505c6823020a18bbdc5410284305234: Merge remote-tracking branches 'asoc/fix/max98371', 'asoc/fix/nau8825', 'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple' and 'asoc/fix/wm2000' into asoc-linus (2016-08-24 19:05:25 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.8 A clutch of fixes for v4.8. These are mainly driver specific, the most notable ones being those for OMAP which fix a series of issues that broke boot on some platforms there when deferred probe kicked in. There's also one core fix for an issue when unbinding a card which for some reason had managed to not manifest until recently. ---------------------------------------------------------------- Adam Thomson (1): ASoC: da7213: Default to 64 BCLKs per WCLK to support all formats Charles Keepax (3): ASoC: Fix leak of rtd in soc_bind_dai_link ASoC: wm2000: Fix return of uninitialised varible ASoC: compress: Fix leak of a widget list in soc_compr_open_fe Christoph Huber (1): ASoC: atmel_ssc_dai: Don't unconditionally reset SSC on stream startup Hiroyuki Yokoyama (1): ASoC: rsnd: Fixup SRCm_IFSVR calculate method John Hsu (2): ASoC: nau8825: fix bug in playback when suspend ASoC: nau8825: fix static check error about semaphone control Kuninori Morimoto (1): ASoC: simple-card-utils: add missing MODULE_xxx() Mark Brown (7): ASoC: core: Clean up DAPM before the card debugfs Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/compress', 'asoc/fix/da7213' and 'asoc/fix/debugfs' into asoc-linus Merge remote-tracking branches 'asoc/fix/max98371', 'asoc/fix/nau8825', 'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple' and 'asoc/fix/wm2000' into asoc-linus Nicolin Chen (1): ASoC: dapm: Add a dummy snd_pcm_runtime to avoid NULL pointer access Peter Ujfalusi (3): ASoC: omap-mcpdm: Drop pdmclk clock handling ASoC: omap-abe-twl6040: Correct dmic-codec device registration ASoC: omap-mcpdm: Fix irq resource handling Sylwester Nawrocki (1): ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card Vinod Koul (1): ASoC: Intel: Skylake: Check list empty while getting module info Wei Yongjun (2): ASoC: Intel: Skylake: Fix error return code in skl_probe() ASoC: max98371: Add terminate entry for i2c_device_id tables .../devicetree/bindings/sound/omap-mcpdm.txt | 10 --- sound/soc/atmel/atmel_ssc_dai.c | 5 +- sound/soc/codecs/da7213.c | 4 +- sound/soc/codecs/max98371.c | 1 + sound/soc/codecs/nau8825.c | 78 +++++++++------------- sound/soc/codecs/wm2000.c | 2 +- sound/soc/generic/Makefile | 6 +- sound/soc/generic/simple-card-utils.c | 6 ++ sound/soc/intel/skylake/skl-sst-utils.c | 5 ++ sound/soc/intel/skylake/skl.c | 4 +- sound/soc/omap/omap-abe-twl6040.c | 61 +++++++++-------- sound/soc/omap/omap-mcpdm.c | 22 +----- sound/soc/samsung/s3c24xx_uda134x.c | 7 +- sound/soc/sh/rcar/src.c | 6 +- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 10 +++ 17 files changed, 115 insertions(+), 121 deletions(-)

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[alsa-devel] [PATCH] ALSA: snd-aoa: enable sound on PowerBook G4 12"
by Aaro Koskinen 25 Aug '16

25 Aug '16

Enable sound on PowerBook G4 12". Signed-off-by: Aaro Koskinen <aaro.koskinen(a)iki.fi> --- sound/aoa/fabrics/layout.c | 8 ++++++++ sound/aoa/soundbus/i2sbus/core.c | 2 +- 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 8f71f7e..edc8681 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -112,6 +112,7 @@ MODULE_ALIAS("sound-layout-100"); MODULE_ALIAS("aoa-device-id-14"); MODULE_ALIAS("aoa-device-id-22"); +MODULE_ALIAS("aoa-device-id-31"); MODULE_ALIAS("aoa-device-id-35"); MODULE_ALIAS("aoa-device-id-44"); @@ -362,6 +363,13 @@ static struct layout layouts[] = { .connections = tas_connections_nolineout, }, }, + /* PowerBook6,1 */ + { .device_id = 31, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, /* PowerBook6,5 */ { .device_id = 44, .codecs[0] = { diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 1cbf210..000b585 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -197,7 +197,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * so restrict to those we do handle for now. */ if (id && (*id == 22 || *id == 14 || *id == 35 || - *id == 44)) { + *id == 31 || *id == 44)) { snprintf(dev->sound.modalias, 32, "aoa-device-id-%d", *id); ok = 1; -- 2.9.2

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[alsa-devel] [RFC PATCH] mfd: arizona: Add gating of external MCLKn clocks
by Sylwester Nawrocki 25 Aug '16

25 Aug '16

This patch adds handling of the CODEC's external MCLK{1,2} clocks needed for board configurations where these clocks are not always on oscillators. The 32k source MCLKn clock is basically kept permanently enabled while the other MCLKn clock is being gated in the MFD's runtime suspend/resume callbacks. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- I'm not sure it's a correct approach hence only an RFC patch. --- drivers/mfd/arizona-core.c | 105 ++++++++++++++++++++++++++++++++------- include/linux/mfd/arizona/core.h | 9 ++++ 2 files changed, 96 insertions(+), 18 deletions(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index e4f97b3..a733b61 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -10,6 +10,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/delay.h> #include <linux/err.h> #include <linux/gpio.h> @@ -49,7 +50,15 @@ int arizona_clk32k_enable(struct arizona *arizona) case ARIZONA_32KZ_MCLK1: ret = pm_runtime_get_sync(arizona->dev); if (ret != 0) - goto out; + goto err_ref; + ret = clk_prepare_enable(arizona->mclk[ARIZONA_MCLK1]); + if (ret != 0) + goto err_pm; + break; + case ARIZONA_32KZ_MCLK2: + ret = clk_prepare_enable(arizona->mclk[ARIZONA_MCLK2]); + if (ret != 0) + goto err_ref; break; } @@ -58,7 +67,9 @@ int arizona_clk32k_enable(struct arizona *arizona) ARIZONA_CLK_32K_ENA); } -out: +err_pm: + pm_runtime_put_sync(arizona->dev); +err_ref: if (ret != 0) arizona->clk32k_ref--; @@ -83,6 +94,10 @@ int arizona_clk32k_disable(struct arizona *arizona) switch (arizona->pdata.clk32k_src) { case ARIZONA_32KZ_MCLK1: pm_runtime_put_sync(arizona->dev); + clk_disable_unprepare(arizona->mclk[ARIZONA_MCLK1]); + break; + case ARIZONA_32KZ_MCLK2: + clk_disable_unprepare(arizona->mclk[ARIZONA_MCLK2]); break; } } @@ -93,6 +108,34 @@ int arizona_clk32k_disable(struct arizona *arizona) } EXPORT_SYMBOL_GPL(arizona_clk32k_disable); +/* Enable all external MCLKn clocks except the 32k source clock */ +int arizona_mclk_enable(struct arizona *arizona) +{ + int ret; + + if (arizona->pdata.clk32k_src != ARIZONA_32KZ_MCLK1) { + ret = clk_prepare_enable(arizona->mclk[ARIZONA_MCLK1]); + if (ret != 0) + return ret; + } + if (arizona->pdata.clk32k_src != ARIZONA_32KZ_MCLK2) { + ret = clk_prepare_enable(arizona->mclk[ARIZONA_MCLK2]); + if (ret != 0) + return ret; + } + return 0; +} + +/* Disable all external MCLKn clocks except the 32k source clock */ +void arizona_mclk_disable(struct arizona *arizona) +{ + if (arizona->pdata.clk32k_src != ARIZONA_32KZ_MCLK1) + clk_disable_unprepare(arizona->mclk[ARIZONA_MCLK1]); + + if (arizona->pdata.clk32k_src != ARIZONA_32KZ_MCLK2) + clk_disable_unprepare(arizona->mclk[ARIZONA_MCLK2]); +} + static irqreturn_t arizona_clkgen_err(int irq, void *data) { struct arizona *arizona = data; @@ -533,6 +576,12 @@ static int arizona_runtime_resume(struct device *dev) return ret; } + ret = arizona_mclk_enable(arizona); + if (ret != 0) { + dev_err(dev, "Failed to enable mclk: %d\n", ret); + goto err_reg; + } + if (arizona->has_fully_powered_off) { arizona_disable_reset(arizona); enable_irq(arizona->irq); @@ -546,14 +595,14 @@ static int arizona_runtime_resume(struct device *dev) if (arizona->external_dcvdd) { ret = arizona_connect_dcvdd(arizona); if (ret != 0) - goto err; + goto err_clk; } ret = wm5102_patch(arizona); if (ret != 0) { dev_err(arizona->dev, "Failed to apply patch: %d\n", ret); - goto err; + goto err_clk; } ret = wm5102_apply_hardware_patch(arizona); @@ -561,19 +610,19 @@ static int arizona_runtime_resume(struct device *dev) dev_err(arizona->dev, "Failed to apply hardware patch: %d\n", ret); - goto err; + goto err_clk; } break; case WM5110: case WM8280: ret = arizona_wait_for_boot(arizona); if (ret) - goto err; + goto err_clk; if (arizona->external_dcvdd) { ret = arizona_connect_dcvdd(arizona); if (ret != 0) - goto err; + goto err_clk; } else { /* * As this is only called for the internal regulator @@ -586,7 +635,7 @@ static int arizona_runtime_resume(struct device *dev) dev_err(arizona->dev, "Failed to set resume voltage: %d\n", ret); - goto err; + goto err_clk; } } @@ -595,24 +644,24 @@ static int arizona_runtime_resume(struct device *dev) dev_err(arizona->dev, "Failed to re-apply sleep patch: %d\n", ret); - goto err; + goto err_clk; } break; case WM1831: case CS47L24: ret = arizona_wait_for_boot(arizona); if (ret != 0) - goto err; + goto err_clk; break; default: ret = arizona_wait_for_boot(arizona); if (ret != 0) - goto err; + goto err_clk; if (arizona->external_dcvdd) { ret = arizona_connect_dcvdd(arizona); if (ret != 0) - goto err; + goto err_clk; } break; } @@ -620,12 +669,14 @@ static int arizona_runtime_resume(struct device *dev) ret = regcache_sync(arizona->regmap); if (ret != 0) { dev_err(arizona->dev, "Failed to restore register cache\n"); - goto err; + goto err_clk; } return 0; -err: +err_clk: + arizona_mclk_disable(arizona); +err_reg: regcache_cache_only(arizona->regmap, true); regulator_disable(arizona->dcvdd); return ret; @@ -707,6 +758,7 @@ static int arizona_runtime_suspend(struct device *dev) regcache_cache_only(arizona->regmap, true); regcache_mark_dirty(arizona->regmap); regulator_disable(arizona->dcvdd); + arizona_mclk_disable(arizona); /* Allow us to completely power down if no jack detection */ if (!jd_active) { @@ -1000,6 +1052,7 @@ static const struct mfd_cell wm8998_devs[] = { int arizona_dev_init(struct arizona *arizona) { + const char * const mclk_name[] = { "mclk1", "mclk2" }; struct device *dev = arizona->dev; const char *type_name = NULL; unsigned int reg, val, mask; @@ -1016,6 +1069,15 @@ int arizona_dev_init(struct arizona *arizona) else arizona_of_get_core_pdata(arizona); + for (i = 0; i < ARRAY_SIZE(arizona->mclk); i++) { + arizona->mclk[i] = devm_clk_get(arizona->dev, mclk_name[i]); + if (IS_ERR(arizona->mclk[i])) { + dev_info(arizona->dev, "Failed to get mclk%d: %ld\n", + i, PTR_ERR(arizona->mclk[i])); + arizona->mclk[i] = NULL; + } + } + regcache_cache_only(arizona->regmap, true); switch (arizona->type) { @@ -1101,6 +1163,16 @@ int arizona_dev_init(struct arizona *arizona) goto err_enable; } + /* Chip default */ + if (!arizona->pdata.clk32k_src) + arizona->pdata.clk32k_src = ARIZONA_32KZ_MCLK2; + + ret = arizona_mclk_enable(arizona); + if (ret != 0) { + dev_err(dev, "Failed to enable mclk: %d\n", ret); + goto err_enable; + } + arizona_disable_reset(arizona); regcache_cache_only(arizona->regmap, false); @@ -1332,10 +1404,6 @@ int arizona_dev_init(struct arizona *arizona) arizona->pdata.gpio_defaults[i]); } - /* Chip default */ - if (!arizona->pdata.clk32k_src) - arizona->pdata.clk32k_src = ARIZONA_32KZ_MCLK2; - switch (arizona->pdata.clk32k_src) { case ARIZONA_32KZ_MCLK1: case ARIZONA_32KZ_MCLK2: @@ -1490,6 +1558,7 @@ err_pm: pm_runtime_disable(arizona->dev); err_reset: arizona_enable_reset(arizona); + arizona_mclk_disable(arizona); regulator_disable(arizona->dcvdd); err_enable: regulator_bulk_disable(arizona->num_core_supplies, diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 58ab4c0..b9909bb 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -13,6 +13,7 @@ #ifndef _WM_ARIZONA_CORE_H #define _WM_ARIZONA_CORE_H +#include <linux/clk.h> #include <linux/interrupt.h> #include <linux/notifier.h> #include <linux/regmap.h> @@ -21,6 +22,12 @@ #define ARIZONA_MAX_CORE_SUPPLIES 2 +enum { + ARIZONA_MCLK1, + ARIZONA_MCLK2, + ARIZONA_NUM_MCLK +}; + enum arizona_type { WM5102 = 1, WM5110 = 2, @@ -139,6 +146,8 @@ struct arizona { struct mutex clk_lock; int clk32k_ref; + struct clk *mclk[ARIZONA_NUM_MCLK]; + bool ctrlif_error; struct snd_soc_dapm_context *dapm; -- 1.9.1

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