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[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

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[alsa-devel] Problem in using few ladspa plugins
by varun sood 13 Sep '16

13 Sep '16

Hi Alsa Developers, I am working on ladspa plugin using alsa lib . There is problem with alsa library with those ladspa plugins where there is change in number of input and output audio ports from the ladspa plugin.E,g for ladspa plugin, if I do analyseplugin surround_encoder_1401.soPorts: "L" input, audio "R" input, audio "C" input, audio "S" input, audio "Lt" output, audio "Rt" output, audio There are 4 input ports("L","R","C","S") for ladspa and 2 output port from ladspa("Lt","Rt"). This is equivalent to 4 input channels and 2 output channels. The alsa library does not handle this plugin properly. I receive segmentation fault on using this plugin in alsa chain scenario at line if (pchannels[chn] == NULL && chn < ichannels) in function snd_pcm_ladspa_allocate_memory in file pcm_ladspa.c Similar issue I have seen in split_1406.so, which takes 1 channel input( 1 input port) and gives 2 channel output( 2 output ports).No segmentation fault in this case but it ignores the 2nd channel . Is there a way to solve these problem ? RegardsVarun Sood

1 0

[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 00/26] constify local structures
by Julia Lawall 13 Sep '16

13 Sep '16

Constify local structures. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> // The first rule ignores some cases that posed problems @r disable optional_qualifier@ identifier s != {peri_clk_data,threshold_attr,tracer_flags,tracer}; identifier i != {s5k5baf_cis_rect,smtcfb_fix}; position p; @@ static struct s i@p = { ... }; @lstruct@ identifier r.s; @@ struct s { ... }; @used depends on lstruct@ identifier r.i; @@ i @bad1@ expression e; identifier r.i; assignment operator a; @@ (<+...i...+>) a e @bad2@ identifier r.i; @@ &(<+...i...+>) @bad3@ identifier r.i; declarer d; @@ d(...,<+...i...+>,...); @bad4@ identifier r.i; type T; T[] e; identifier f; position p; @@ f@p(..., ( (<+...i...+>) & e ) ,...) @bad4a@ identifier r.i; type T; T *e; identifier f; position p; @@ f@p(..., ( (<+...i...+>) & e ) ,...) @ok5@ expression *e; identifier r.i; position p; @@ e =@p i @bad5@ expression *e; identifier r.i; position p != ok5.p; @@ e =@p (<+...i...+>) @rr depends on used && !bad1 && !bad2 && !bad3 && !bad4 && !bad4a && !bad5@ identifier s,r.i; position r.p; @@ static +const struct s i@p = { ... }; @depends on used && !bad1 && !bad2 && !bad3 && !bad4 && !bad4a && !bad5 disable optional_qualifier@ identifier rr.s,r.i; @@ static +const struct s i; // </smpl> --- drivers/acpi/acpi_apd.c | 8 +++--- drivers/char/tpm/tpm-interface.c | 10 ++++---- drivers/char/tpm/tpm-sysfs.c | 2 - drivers/cpufreq/intel_pstate.c | 8 +++--- drivers/infiniband/hw/i40iw/i40iw_uk.c | 6 ++--- drivers/media/i2c/tvp514x.c | 2 - drivers/media/pci/ddbridge/ddbridge-core.c | 18 +++++++-------- drivers/media/pci/ngene/ngene-cards.c | 14 ++++++------ drivers/media/pci/smipcie/smipcie-main.c | 8 +++--- drivers/misc/sgi-xp/xpc_uv.c | 2 - drivers/net/arcnet/com20020-pci.c | 10 ++++---- drivers/net/can/c_can/c_can_pci.c | 4 +-- drivers/net/can/sja1000/plx_pci.c | 20 ++++++++--------- drivers/net/ethernet/mellanox/mlx4/main.c | 4 +-- drivers/net/ethernet/oki-semi/pch_gbe/pch_gbe_main.c | 2 - drivers/net/ethernet/renesas/sh_eth.c | 14 ++++++------ drivers/net/ethernet/stmicro/stmmac/stmmac_pci.c | 2 - drivers/net/wireless/ath/dfs_pattern_detector.c | 2 - drivers/net/wireless/intel/iwlegacy/3945.c | 4 +-- drivers/net/wireless/realtek/rtlwifi/rtl8188ee/sw.c | 2 - drivers/net/wireless/realtek/rtlwifi/rtl8192ce/sw.c | 2 - drivers/net/wireless/realtek/rtlwifi/rtl8192de/sw.c | 2 - drivers/net/wireless/realtek/rtlwifi/rtl8192ee/sw.c | 2 - drivers/net/wireless/realtek/rtlwifi/rtl8192se/sw.c | 2 - drivers/net/wireless/realtek/rtlwifi/rtl8723ae/sw.c | 2 - drivers/net/wireless/realtek/rtlwifi/rtl8723be/sw.c | 2 - drivers/net/wireless/realtek/rtlwifi/rtl8821ae/sw.c | 2 - drivers/platform/chrome/chromeos_laptop.c | 22 +++++++++---------- drivers/platform/x86/intel_scu_ipc.c | 6 ++--- drivers/platform/x86/intel_telemetry_debugfs.c | 2 - drivers/scsi/esas2r/esas2r_flash.c | 2 - drivers/scsi/hptiop.c | 6 ++--- drivers/spi/spi-dw-pci.c | 4 +-- drivers/staging/rtl8192e/rtl8192e/rtl_core.c | 2 - drivers/usb/misc/ezusb.c | 2 - drivers/video/fbdev/matrox/matroxfb_g450.c | 2 - lib/crc64_ecma.c | 2 - sound/pci/ctxfi/ctatc.c | 2 - sound/pci/hda/patch_ca0132.c | 10 ++++---- sound/pci/riptide/riptide.c | 2 - 40 files changed, 110 insertions(+), 110 deletions(-)

6 14

[alsa-devel] alsactl segfault
by Orion Poplawski 12 Sep '16

12 Sep '16

There are reports of alsactl segfaulting: https://bugzilla.redhat.com/show_bug.cgi?id=1335431 https://bugzilla.redhat.com/show_bug.cgi?id=1283737 Core was generated by `/usr/sbin/alsactl -s -n 19 -c -E ALSA_CONFIG_PATH=/etc/alsa/alsactl.conf --init'. Program terminated with signal SIGSEGV, Segmentation fault. #0 free_list (list=list@entry=0x20) at daemon.c:74 74 for (i = 0; i < list->size; i++) Thread 1 (Thread 0x7fd798589700 (LWP 1015)): #0 free_list (list=list@entry=0x20) at daemon.c:74 i = 0 #1 0x000055e3c8704935 in card_free (card=0x55e3ca6c5018) at daemon.c:83 c = 0x0 #2 0x000055e3c87054a4 in state_daemon (file=0x55e3c8705ced "/var/lib/alsa/asound.state", cardname=0x0, period=300, pidfile=0x55e3c8705cbc "/var/run/alsactl.pid") at daemon.c:455 count = 2 pcount = <optimized out> psize = <optimized out> i = 1 j = <optimized out> k = <optimized out> changed = 0 last_write = 1463036017 now = 1463037368 revents = 0 cards = 0x55e3ca6c5010 pfd = <optimized out> pfdn = <optimized out> __FUNCTION__ = "state_daemon" #3 0x000055e3c86f90e7 in main (argc=<optimized out>, argv=<optimized out>) at alsactl.c:370 devfiles = {0x55e3c8705d08 "/dev/snd/controlC", 0x55e3c8705db3 "/dev/snd/pcmC", 0x55e3c8705dc1 "/dev/snd/midiC", 0x55e3c8705dd0 "/dev/snd/hwC", 0x0} cfgfile = 0x55e3c8705ced "/var/lib/alsa/asound.state" initfile = 0x7ffc87ee1f80 "/lib/alsa/init/00main" pidfile = 0x55e3c8705cbc "/var/run/alsactl.pid" cardname = 0x0 ncardname = "(\201Z\230\327\177\000\000\000\000\000\000\000\000\000" cmd = 0x7ffc87ee1f96 "rdaemon" tmp = <optimized out> removestate = 0 init_fallback = 1 period = 300 background = 0 daemoncmd = <optimized out> use_nice = 19 sched_idle = 1 o = <optimized out> j = <optimized out> k = <optimized out> res = 0 long_option = 0x0 short_option = 0x0 Any idea what might be causing this? Any other debug info that would be helpful? It doesn't crash all the time. -- Orion Poplawski Technical Manager 303-415-9701 x222 NWRA, Boulder/CoRA Office FAX: 303-415-9702 3380 Mitchell Lane orion(a)nwra.com Boulder, CO 80301 http://www.nwra.com

2 2

[alsa-devel] [PATCH] ASoC: rt5514: make the volume TLV to match the units 0.01dB
by Oder Chiou 12 Sep '16

12 Sep '16

The volume have a step of 0.375dB, but TLV uses the units 0.01dB. It should be changed to a step of 0.75dB to match the units of TLV. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 6 +++--- sound/soc/codecs/rt5514.h | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index e0606a5..f24b7cf 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -278,7 +278,7 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, 8, 8, TLV_DB_SCALE_ITEM(1700, 0, 0) ); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static int rt5514_dsp_voice_wake_up_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -352,10 +352,10 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), SOC_DOUBLE_R_TLV("ADC1 Capture Volume", RT5514_DOWNFILTER0_CTRL1, - RT5514_DOWNFILTER0_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, + RT5514_DOWNFILTER0_CTRL2, RT5514_AD_GAIN_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_R_TLV("ADC2 Capture Volume", RT5514_DOWNFILTER1_CTRL1, - RT5514_DOWNFILTER1_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, + RT5514_DOWNFILTER1_CTRL2, RT5514_AD_GAIN_SFT, 63, 0, adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 68883c6..229de0e 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -196,8 +196,8 @@ #define RT5514_AD_AD_MIX_BIT 10 #define RT5514_AD_AD_MUTE (0x1 << 7) #define RT5514_AD_AD_MUTE_BIT 7 -#define RT5514_AD_GAIN_MASK (0x7f << 0) -#define RT5514_AD_GAIN_SFT 0 +#define RT5514_AD_GAIN_MASK (0x3f << 1) +#define RT5514_AD_GAIN_SFT 1 /* RT5514_ANA_CTRL_MICBST (0x2220) */ #define RT5514_SEL_BSTL_MASK (0xf << 4) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3] ASoC: rockchip: i2s: restore register during runtime_suspend/resume cycle
by Sugar Zhang 12 Sep '16

12 Sep '16

when step into runtime_suspend, i2s pd will be disabled and loss state. so need to restore register when runtime_resume. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v3: - remove unneeded initialization Changes in v2: - remove system suspend/resume and restore register in runtime_suspend/resume per Doug Anderson's suggestion sound/soc/rockchip/rockchip_i2s.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 652e8c5..974915c 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -57,6 +57,7 @@ static int i2s_runtime_suspend(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->mclk); return 0; @@ -73,7 +74,14 @@ static int i2s_runtime_resume(struct device *dev) return ret; } - return 0; + regcache_cache_only(i2s->regmap, false); + regcache_mark_dirty(i2s->regmap); + + ret = regcache_sync(i2s->regmap); + if (ret) + clk_disable_unprepare(i2s->mclk); + + return ret; } static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ASoC: rockchip: spdif: restore register during runtime_suspend/resume cycle
by Sugar Zhang 12 Sep '16

12 Sep '16

when step into runtime_suspend, spdif pd will be disabled and loss state. so need to restore register when runtime_resume. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v3: - remove unneeded initialization Changes in v2: - remove system suspend/resume and restore register in runtime_suspend/resume per Doug Anderson's suggestion sound/soc/rockchip/rockchip_spdif.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 4ca2657..fa8101d 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -65,6 +65,7 @@ static int __maybe_unused rk_spdif_runtime_suspend(struct device *dev) { struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + regcache_cache_only(spdif->regmap, true); clk_disable_unprepare(spdif->mclk); clk_disable_unprepare(spdif->hclk); @@ -88,7 +89,16 @@ static int __maybe_unused rk_spdif_runtime_resume(struct device *dev) return ret; } - return 0; + regcache_cache_only(spdif->regmap, false); + regcache_mark_dirty(spdif->regmap); + + ret = regcache_sync(spdif->regmap); + if (ret) { + clk_disable_unprepare(spdif->mclk); + clk_disable_unprepare(spdif->hclk); + } + + return ret; } static int rk_spdif_hw_params(struct snd_pcm_substream *substream, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/6] constify gpio_chip structures
by Julia Lawall 12 Sep '16

12 Sep '16

2 3

[alsa-devel] [PATCH 3/4] hid-lenovo: Add support for X1 Tablet special keys and, LED control
by Dennis Wassenberg 12 Sep '16

12 Sep '16

The Lenovo X1 Tablet Cover is connected via USB. It consists of 1 device with 3 usb interfaces. Interface 0 represents keyboard, interface 1 the function / special keys and LED control, interface 2 is the Synaptics touchpad and pointing stick. This driver will bind to interfaces 0 and 1 and handle function / special keys including LED control. Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- drivers/hid/hid-lenovo.c | 549 +++++++++++++++++++++++++++++++++++++++++++++ include/linux/hid-lenovo.h | 15 ++ 2 files changed, 564 insertions(+) create mode 100644 include/linux/hid-lenovo.h diff --git a/drivers/hid/hid-lenovo.c b/drivers/hid/hid-lenovo.c index 1ac4ff4..4251aac 100644 --- a/drivers/hid/hid-lenovo.c +++ b/drivers/hid/hid-lenovo.c @@ -3,9 +3,11 @@ * - ThinkPad USB Keyboard with TrackPoint (tpkbd) * - ThinkPad Compact Bluetooth Keyboard with TrackPoint (cptkbd) * - ThinkPad Compact USB Keyboard with TrackPoint (cptkbd) + * - ThinkPad X1 Cover USB Keyboard with TrackPoint and Touchpad (tpx1cover) * * Copyright (c) 2012 Bernhard Seibold * Copyright (c) 2014 Jamie Lentin <jm(a)lentin.co.uk> + * Copyright (c) 2016 Dennis Wassenberg <dennis.wassenberg(a)secunet.com> */ /* @@ -19,11 +21,19 @@ #include <linux/sysfs.h> #include <linux/device.h> #include <linux/hid.h> +#include <linux/hid-lenovo.h> #include <linux/input.h> #include <linux/leds.h> #include "hid-ids.h" +struct led_table_entry { + struct led_classdev *dev; + uint8_t state; +}; + +static struct led_table_entry hid_lenovo_led_table[HID_LENOVO_LED_MAX]; + struct lenovo_drvdata_tpkbd { int led_state; struct led_classdev led_mute; @@ -42,6 +52,37 @@ struct lenovo_drvdata_cptkbd { int sensitivity; }; +struct lenovo_drvdata_tpx1cover { + uint16_t led_state; + uint8_t fnlock_state; + uint8_t led_present; + struct led_classdev led_mute; + struct led_classdev led_micmute; + struct led_classdev led_fnlock; +}; + +int hid_lenovo_led_set(int led_num, bool on) +{ + struct led_classdev *dev; + + if (led_num >= HID_LENOVO_LED_MAX) + return -EINVAL; + + dev = hid_lenovo_led_table[led_num].dev; + hid_lenovo_led_table[led_num].state = on; + + if (!dev) + return -ENODEV; + + if (!dev->brightness_set) + return -ENODEV; + + dev->brightness_set(dev, on ? LED_FULL : LED_OFF); + + return 0; +} +EXPORT_SYMBOL_GPL(hid_lenovo_led_set); + #define map_key_clear(c) hid_map_usage_clear(hi, usage, bit, max, EV_KEY, (c)) static const __u8 lenovo_pro_dock_need_fixup_collection[] = { @@ -86,6 +127,84 @@ static int lenovo_input_mapping_tpkbd(struct hid_device *hdev, return 0; } +static int lenovo_input_mapping_tpx1cover(struct hid_device *hdev, + struct hid_input *hi, struct hid_field *field, + struct hid_usage *usage, unsigned long **bit, int *max) +{ + if ((usage->hid & HID_USAGE_PAGE) == HID_UP_CONSUMER) { + switch (usage->hid & HID_USAGE) { + case 0x0001: // Unknown keys -> Idenditied by usage index! + map_key_clear(KEY_UNKNOWN); + switch (usage->usage_index) { + case 0x8: + input_set_capability(hi->input, EV_KEY, KEY_FN); + break; + + case 0x9: + input_set_capability(hi->input, EV_KEY, KEY_MICMUTE); + break; + + case 0xa: + input_set_capability(hi->input, EV_KEY, KEY_CONFIG); + break; + + case 0xb: + input_set_capability(hi->input, EV_KEY, KEY_SEARCH); + break; + + case 0xc: + input_set_capability(hi->input, EV_KEY, KEY_SETUP); + break; + + case 0xd: + input_set_capability(hi->input, EV_KEY, KEY_SWITCHVIDEOMODE); + break; + + case 0xe: + input_set_capability(hi->input, EV_KEY, KEY_RFKILL); + break; + + default: + return -1; + } + + return 1; + + case 0x006f: // Consumer.006f ---> Key.BrightnessUp + map_key_clear(KEY_BRIGHTNESSUP); + return 1; + + case 0x0070: // Consumer.0070 ---> Key.BrightnessDown + map_key_clear(KEY_BRIGHTNESSDOWN); + return 1; + + case 0x00b7:// Consumer.00b7 ---> Key.StopCD + map_key_clear(KEY_STOPCD); + return 1; + + case 0x00cd: // Consumer.00cd ---> Key.PlayPause + map_key_clear(KEY_PLAYPAUSE); + return 1; + + case 0x00e0: // Consumer.00e0 ---> Absolute.Volume + return 0; + case 0x00e2: // Consumer.00e2 ---> Key.Mute + map_key_clear(KEY_MUTE); + return 1; + + case 0x00e9: // Consumer.00e9 ---> Key.VolumeUp + map_key_clear(KEY_VOLUMEUP); + return 1; + + case 0x00ea: // Consumer.00ea ---> Key.VolumeDown + map_key_clear(KEY_VOLUMEDOWN); + return 1; + } + } + + return 0; +} + static int lenovo_input_mapping_cptkbd(struct hid_device *hdev, struct hid_input *hi, struct hid_field *field, struct hid_usage *usage, unsigned long **bit, int *max) @@ -172,6 +291,9 @@ static int lenovo_input_mapping(struct hid_device *hdev, case USB_DEVICE_ID_LENOVO_CBTKBD: return lenovo_input_mapping_cptkbd(hdev, hi, field, usage, bit, max); + case USB_DEVICE_ID_LENOVO_X1_COVER: + return lenovo_input_mapping_tpx1cover(hdev, hi, field, + usage, bit, max); default: return 0; } @@ -362,6 +484,143 @@ static int lenovo_event_cptkbd(struct hid_device *hdev, return 0; } +static enum led_brightness lenovo_led_brightness_get_tpx1cover( + struct led_classdev *led_cdev) +{ + struct device *dev = led_cdev->dev->parent; + struct hid_device *hdev = to_hid_device(dev); + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + int led_nr = 0; + + if (led_cdev == &drv_data->led_mute) + led_nr = 0; + else if (led_cdev == &drv_data->led_micmute) + led_nr = 1; + else if (led_cdev == &drv_data->led_fnlock) + led_nr = 2; + else + return LED_OFF; + + return drv_data->led_state & (1 << led_nr) + ? LED_FULL + : LED_OFF; +} + +static void lenovo_led_brightness_set_tpx1cover(struct led_classdev *led_cdev, + enum led_brightness value) +{ + struct device *dev = led_cdev->dev->parent; + struct hid_device *hdev = to_hid_device(dev); + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + struct hid_report *report; + int led_nr = -1; + int led_nr_hw = -1; + + if (led_cdev == &drv_data->led_mute) { + led_nr = 0; + led_nr_hw = 0x64; + } else if (led_cdev == &drv_data->led_micmute) { + led_nr = 1; + led_nr_hw = 0x74; + } else if (led_cdev == &drv_data->led_fnlock) { + led_nr = 2; + led_nr_hw = 0x54; + } else { + hid_warn(hdev, "Invalid LED to set.\n"); + return; + } + + if (value == LED_OFF) + drv_data->led_state &= ~(1 << led_nr); + else + drv_data->led_state |= 1 << led_nr; + + report = hdev->report_enum[HID_OUTPUT_REPORT].report_id_hash[9]; + if (report) { + report->field[0]->value[0] = led_nr_hw; + report->field[0]->value[1] = (drv_data->led_state & (1 << led_nr)) + ? 0x02 : 0x01; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + } +} + +static int lenovo_event_tpx1cover(struct hid_device *hdev, + struct hid_field *field, struct hid_usage *usage, __s32 value) +{ + int ret = 0; + + if ((usage->hid & HID_USAGE_PAGE) == HID_UP_CONSUMER + && (usage->hid & HID_USAGE) == 0x0001) { + + if (usage->usage_index == 0x8 && value == 1) { + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + + if (drv_data && drv_data->led_present) { + drv_data->fnlock_state = lenovo_led_brightness_get_tpx1cover( + &drv_data->led_fnlock) == LED_OFF ? 1 : 0; + lenovo_led_brightness_set_tpx1cover( + &drv_data->led_fnlock, + drv_data->fnlock_state ? LED_FULL : LED_OFF); + } + } + + if (usage->usage_index == 0x9 && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_MICMUTE, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_MICMUTE, 0); + input_sync(field->hidinput->input); + ret = 1; + } + + if (usage->usage_index == 0xa && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_CONFIG, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_CONFIG, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + + if (usage->usage_index == 0xb && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_SEARCH, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_SEARCH, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + + if (usage->usage_index == 0xc && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_SETUP, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_SETUP, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + + if (usage->usage_index == 0xd && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_SWITCHVIDEOMODE, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_SWITCHVIDEOMODE, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + + if (usage->usage_index == 0xe && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_RFKILL, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_RFKILL, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + } + + return ret; +} + static int lenovo_event(struct hid_device *hdev, struct hid_field *field, struct hid_usage *usage, __s32 value) { @@ -369,6 +628,8 @@ static int lenovo_event(struct hid_device *hdev, struct hid_field *field, case USB_DEVICE_ID_LENOVO_CUSBKBD: case USB_DEVICE_ID_LENOVO_CBTKBD: return lenovo_event_cptkbd(hdev, field, usage, value); + case USB_DEVICE_ID_LENOVO_X1_COVER: + return lenovo_event_tpx1cover(hdev, field, usage, value); default: return 0; } @@ -731,6 +992,251 @@ static int lenovo_probe_tpkbd(struct hid_device *hdev) return ret; } +static int lenovo_probe_tpx1cover_configure(struct hid_device *hdev) +{ + struct hid_report *report = hdev->report_enum[HID_OUTPUT_REPORT].report_id_hash[9]; + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + + if (!drv_data) + return -ENODEV; + + if (!report) + return -ENOENT; + + report->field[0]->value[0] = 0x54; + report->field[0]->value[1] = 0x20; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + report->field[0]->value[0] = 0x54; + report->field[0]->value[1] = 0x08; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + report->field[0]->value[0] = 0xA0; + report->field[0]->value[1] = 0x02; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + lenovo_led_brightness_set_tpx1cover(&drv_data->led_mute, + hid_lenovo_led_table[HID_LENOVO_LED_MUTE].state ? LED_FULL : LED_OFF); + hid_hw_wait(hdev); + + lenovo_led_brightness_set_tpx1cover(&drv_data->led_micmute, + hid_lenovo_led_table[HID_LENOVO_LED_MICMUTE].state ? LED_FULL : LED_OFF); + hid_hw_wait(hdev); + + lenovo_led_brightness_set_tpx1cover(&drv_data->led_fnlock, LED_FULL); + + return 0; +} + +static int lenovo_probe_tpx1cover_special_functions(struct hid_device *hdev) +{ + struct device *dev = &hdev->dev; + struct lenovo_drvdata_tpx1cover *drv_data = NULL; + + size_t name_sz = strlen(dev_name(dev)) + 16; + char *name_led = NULL; + + struct hid_report *report; + bool report_match = 1; + + int ret = 0; + + report = hid_validate_values(hdev, HID_INPUT_REPORT, 2, 0, 3); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 3, 0, 16); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_OUTPUT_REPORT, 9, 0, 2); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 32, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 84, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 100, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 116, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 132, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 144, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 162, 0, 1); + report_match &= report ? 1 : 0; + + if (!report_match) { + ret = -ENODEV; + goto err; + } + + drv_data = devm_kzalloc(&hdev->dev, + sizeof(struct lenovo_drvdata_tpx1cover), + GFP_KERNEL); + + if (!drv_data) { + hid_err(hdev, + "Could not allocate memory for tpx1cover driver data\n"); + ret = -ENOMEM; + goto err; + } + + name_led = devm_kzalloc(&hdev->dev, name_sz, GFP_KERNEL); + if (!name_led) { + hid_err(hdev, "Could not allocate memory for mute led data\n"); + ret = -ENOMEM; + goto err_cleanup; + } + snprintf(name_led, name_sz, "%s:amber:mute", dev_name(dev)); + + drv_data->led_mute.name = name_led; + drv_data->led_mute.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_mute.brightness_set = lenovo_led_brightness_set_tpx1cover; + drv_data->led_mute.dev = dev; + hid_lenovo_led_table[HID_LENOVO_LED_MUTE].dev = &drv_data->led_mute; + led_classdev_register(dev, &drv_data->led_mute); + + + name_led = devm_kzalloc(&hdev->dev, name_sz, GFP_KERNEL); + if (!name_led) { + hid_err(hdev, + "Could not allocate memory for mic mute led data\n"); + ret = -ENOMEM; + goto err_cleanup; + } + snprintf(name_led, name_sz, "%s:amber:micmute", dev_name(dev)); + + drv_data->led_micmute.name = name_led; + drv_data->led_micmute.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_micmute.brightness_set = lenovo_led_brightness_set_tpx1cover; + drv_data->led_micmute.dev = dev; + hid_lenovo_led_table[HID_LENOVO_LED_MICMUTE].dev = &drv_data->led_micmute; + led_classdev_register(dev, &drv_data->led_micmute); + + + name_led = devm_kzalloc(&hdev->dev, name_sz, GFP_KERNEL); + if (!name_led) { + hid_err(hdev, + "Could not allocate memory for FN lock led data\n"); + ret = -ENOMEM; + goto err_cleanup; + } + + snprintf(name_led, name_sz, "%s:amber:fnlock", dev_name(dev)); + + drv_data->led_fnlock.name = name_led; + drv_data->led_fnlock.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_fnlock.brightness_set = lenovo_led_brightness_set_tpx1cover; + drv_data->led_fnlock.dev = dev; + hid_lenovo_led_table[HID_LENOVO_LED_FNLOCK].dev = &drv_data->led_fnlock; + led_classdev_register(dev, &drv_data->led_fnlock); + + drv_data->led_state = 0; + drv_data->fnlock_state = 1; + drv_data->led_present = 1; + + hid_set_drvdata(hdev, drv_data); + + return lenovo_probe_tpx1cover_configure(hdev); + +err_cleanup: + if (drv_data->led_fnlock.name) { + led_classdev_unregister(&drv_data->led_fnlock); + devm_kfree(&hdev->dev, (void *) drv_data->led_fnlock.name); + } + + if (drv_data->led_micmute.name) { + led_classdev_unregister(&drv_data->led_micmute); + devm_kfree(&hdev->dev, (void *) drv_data->led_micmute.name); + } + + if (drv_data->led_mute.name) { + led_classdev_unregister(&drv_data->led_mute); + devm_kfree(&hdev->dev, (void *) drv_data->led_mute.name); + } + + if (drv_data) + kfree(drv_data); + +err: + return ret; +} + +static int lenovo_probe_tpx1cover_touch(struct hid_device *hdev) +{ + struct hid_report *report; + bool report_match = 1; + int ret = 0; + + report = hid_validate_values(hdev, HID_INPUT_REPORT, 2, 0, 2); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 2, 1, 2); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 11, 0, 61); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 12, 0, 61); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 16, 0, 3); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 16, 1, 2); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_OUTPUT_REPORT, 9, 0, 20); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_OUTPUT_REPORT, 10, 0, 20); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 14, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 15, 0, 3); + report_match &= report ? 1 : 0; + + if (!report_match) + ret = -ENODEV; + + return ret; +} + +static int lenovo_probe_tpx1cover(struct hid_device *hdev) +{ + int ret = 0; + + /* + * Probing for special function keys and LED control -> usb intf 1 + * Probing for touch input -> usb intf 2 (handled by rmi4 driver) + * Other (keyboard) -> usb intf 0 + */ + if (!lenovo_probe_tpx1cover_special_functions(hdev)) { + // special function keys and LED control + ret = 0; + } else if (!lenovo_probe_tpx1cover_touch(hdev)) { + // handled by rmi + ret = -ENODEV; + } else { + // keyboard + struct lenovo_drvdata_tpx1cover *drv_data; + + drv_data = devm_kzalloc(&hdev->dev, + sizeof(struct lenovo_drvdata_tpx1cover), + GFP_KERNEL); + + if (!drv_data) { + hid_err(hdev, + "Could not allocate memory for tpx1cover driver data\n"); + ret = -ENOMEM; + goto out; + } + + drv_data->led_state = 0; + drv_data->led_present = 0; + drv_data->fnlock_state = 0; + hid_set_drvdata(hdev, drv_data); + + ret = 0; + } + +out: + return ret; +} + static int lenovo_probe_cptkbd(struct hid_device *hdev) { int ret; @@ -803,6 +1309,9 @@ static int lenovo_probe(struct hid_device *hdev, case USB_DEVICE_ID_LENOVO_CBTKBD: ret = lenovo_probe_cptkbd(hdev); break; + case USB_DEVICE_ID_LENOVO_X1_COVER: + ret = lenovo_probe_tpx1cover(hdev); + break; default: ret = 0; break; @@ -843,6 +1352,42 @@ static void lenovo_remove_cptkbd(struct hid_device *hdev) &lenovo_attr_group_cptkbd); } +static void lenovo_remove_tpx1cover(struct hid_device *hdev) +{ + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + + if (!drv_data) + return; + + if (drv_data->led_present) { + if (drv_data->led_fnlock.name) { + hid_lenovo_led_table[HID_LENOVO_LED_FNLOCK].dev = NULL; + + led_classdev_unregister(&drv_data->led_fnlock); + devm_kfree(&hdev->dev, (void *) drv_data->led_fnlock.name); + } + + if (drv_data->led_micmute.name) { + hid_lenovo_led_table[HID_LENOVO_LED_MICMUTE].dev = NULL; + + led_classdev_unregister(&drv_data->led_micmute); + devm_kfree(&hdev->dev, (void *) drv_data->led_micmute.name); + } + + if (drv_data->led_mute.name) { + hid_lenovo_led_table[HID_LENOVO_LED_MUTE].dev = NULL; + + led_classdev_unregister(&drv_data->led_mute); + devm_kfree(&hdev->dev, (void *) drv_data->led_mute.name); + } + } + + if (drv_data) + devm_kfree(&hdev->dev, drv_data); + + hid_set_drvdata(hdev, NULL); +} + static void lenovo_remove(struct hid_device *hdev) { switch (hdev->product) { @@ -853,6 +1398,9 @@ static void lenovo_remove(struct hid_device *hdev) case USB_DEVICE_ID_LENOVO_CBTKBD: lenovo_remove_cptkbd(hdev); break; + case USB_DEVICE_ID_LENOVO_X1_COVER: + lenovo_remove_tpx1cover(hdev); + break; } hid_hw_stop(hdev); @@ -883,6 +1431,7 @@ static int lenovo_input_configured(struct hid_device *hdev, { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_CUSBKBD) }, { HID_BLUETOOTH_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_CBTKBD) }, { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_TPPRODOCK) }, + { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_X1_COVER) }, { } }; diff --git a/include/linux/hid-lenovo.h b/include/linux/hid-lenovo.h new file mode 100644 index 0000000..d0b0145 --- /dev/null +++ b/include/linux/hid-lenovo.h @@ -0,0 +1,15 @@ + +#ifndef __HID_LENOVO_H__ +#define __HID_LENOVO_H__ + + +enum { + HID_LENOVO_LED_MUTE, + HID_LENOVO_LED_MICMUTE, + HID_LENOVO_LED_FNLOCK, + HID_LENOVO_LED_MAX, +}; + +int hid_lenovo_led_set(int led_num, bool on); + +#endif /* __HID_LENOVO_H_ */ -- 1.9.1

1 0

[alsa-devel] [PATCH v2] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 12 Sep '16

12 Sep '16

Introduce mechanism to detect sun7i and provide a different regmap and different controls for it - different compared to sun4i Allwinner A10. The controls will be extended in a forthcoming patch - it is necessary to distinguish between sun4i and sun7i controls because the mic gains are in different registers. Renamed SUN4I_CODEC_AC_SYS_VERI to SUN7I_CODEC_AC_DAC_CAL and renamed SUN4I_CODEC_AC_MIC_PHONE_CAL to SUN7I_CODEC_AC_MIC_PHONE_CAL because these are actually not present on Allwinner A10. Handle quirks by regmap config and codec and select the correct quirks automatically. Signed-off-by: Danny Milosavljevic <dannym(a)scratchpost.org> --- sound/soc/sunxi/sun4i-codec.c | 78 ++++++++++++++++++++++++++++++++----------- 1 file changed, 58 insertions(+), 20 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 0e19c50..239e330 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -96,8 +96,8 @@ /* Other various ADC registers */ #define SUN4I_CODEC_DAC_TXCNT (0x30) #define SUN4I_CODEC_ADC_RXCNT (0x34) -#define SUN4I_CODEC_AC_SYS_VERI (0x38) -#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) +#define SUN7I_CODEC_AC_DAC_CAL (0x38) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL (0x3c) struct sun4i_codec { struct device *dev; @@ -509,7 +509,7 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); -static const struct snd_kcontrol_new sun4i_codec_widgets[] = { +static const struct snd_kcontrol_new sun4i_codec_controls[] = { SOC_SINGLE_TLV("Power Amplifier Volume", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, sun4i_codec_pa_volume_scale), @@ -627,10 +627,60 @@ static const struct snd_soc_dapm_route sun4i_codec_codec_dapm_routes[] = { { "Mic1", NULL, "VMIC" }, }; +static const struct regmap_config sun4i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN4I_CODEC_ADC_RXCNT, +}; + +static const struct regmap_config sun7i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, +}; + +struct sun4i_quirks { + const struct snd_kcontrol_new *controls; + unsigned int num_controls; + const struct regmap_config *regmap_config; +}; + +static const struct sun4i_quirks sun4i_quirks = { + .controls = NULL, + .num_controls = 0U, + .regmap_config = &sun4i_codec_regmap_config, +}; + +static const struct sun4i_quirks sun7i_quirks = { + .controls = NULL, + .num_controls = 0U, + .regmap_config = &sun7i_codec_regmap_config, +}; + +static const struct of_device_id sun4i_codec_of_match[] = { + { .compatible = "allwinner,sun4i-a10-codec", .data = &sun4i_quirks }, + { .compatible = "allwinner,sun7i-a20-codec", .data = &sun7i_quirks }, + {} +}; +MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); + +static int sun4i_codec_codec_probe(struct snd_soc_codec *scodec) +{ + const struct sun4i_quirks *quirks; + + quirks = of_device_get_match_data(scodec->dev); + return snd_soc_add_codec_controls(scodec, + quirks->controls, + quirks->num_controls); +} + static struct snd_soc_codec_driver sun4i_codec_codec = { + .probe = sun4i_codec_codec_probe, .component_driver = { - .controls = sun4i_codec_widgets, - .num_controls = ARRAY_SIZE(sun4i_codec_widgets), + .controls = sun4i_codec_controls, + .num_controls = ARRAY_SIZE(sun4i_codec_controls), .dapm_widgets = sun4i_codec_codec_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_codec_dapm_widgets), .dapm_routes = sun4i_codec_codec_dapm_routes, @@ -678,20 +728,6 @@ static struct snd_soc_dai_driver dummy_cpu_dai = { }, }; -static const struct regmap_config sun4i_codec_regmap_config = { - .reg_bits = 32, - .reg_stride = 4, - .val_bits = 32, - .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, -}; - -static const struct of_device_id sun4i_codec_of_match[] = { - { .compatible = "allwinner,sun4i-a10-codec" }, - { .compatible = "allwinner,sun7i-a20-codec" }, - {} -}; -MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); - static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, int *num_links) { @@ -760,6 +796,7 @@ static int sun4i_codec_probe(struct platform_device *pdev) { struct snd_soc_card *card; struct sun4i_codec *scodec; + const struct sun4i_quirks *quirks; struct resource *res; void __iomem *base; int ret; @@ -777,8 +814,9 @@ static int sun4i_codec_probe(struct platform_device *pdev) return PTR_ERR(base); } + quirks = of_device_get_match_data(&pdev->dev); scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base, - &sun4i_codec_regmap_config); + quirks->regmap_config); if (IS_ERR(scodec->regmap)) { dev_err(&pdev->dev, "Failed to create our regmap\n"); return PTR_ERR(scodec->regmap); -- 2.1.4

3 2

[alsa-devel] [PATCH] treewide: remove redundant #include <linux/kconfig.h>
by Masahiro Yamada 12 Sep '16

12 Sep '16

Kernel source files need not include <linux/kconfig.h> explicitly because the top Makefile forces to include it with: -include $(srctree)/include/linux/kconfig.h This commit removes explicit includes except the following: * arch/s390/include/asm/facilities_src.h * tools/testing/radix-tree/linux/kernel.h These two are used for host programs. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- I thought including <asm/facilities_src.h> from a host tool is weird. I am planning to fix it later. arch/arm/include/asm/trusted_foundations.h | 1 - arch/arm64/include/asm/alternative.h | 1 - arch/mips/include/asm/mach-loongson64/loongson.h | 1 - arch/mips/math-emu/cp1emu.c | 1 - arch/mips/net/bpf_jit.c | 1 - drivers/char/virtio_console.c | 1 - drivers/input/rmi4/rmi_bus.c | 1 - drivers/input/rmi4/rmi_driver.c | 1 - drivers/input/rmi4/rmi_f01.c | 1 - drivers/input/rmi4/rmi_f11.c | 1 - drivers/irqchip/irq-bcm6345-l1.c | 1 - drivers/irqchip/irq-bcm7038-l1.c | 1 - drivers/irqchip/irq-bcm7120-l2.c | 1 - drivers/irqchip/irq-brcmstb-l2.c | 1 - drivers/media/dvb-frontends/af9013.h | 1 - drivers/media/dvb-frontends/af9033.h | 2 -- drivers/media/dvb-frontends/ascot2e.h | 1 - drivers/media/dvb-frontends/atbm8830.h | 1 - drivers/media/dvb-frontends/au8522.h | 1 - drivers/media/dvb-frontends/cx22702.h | 1 - drivers/media/dvb-frontends/cx24113.h | 2 -- drivers/media/dvb-frontends/cx24116.h | 1 - drivers/media/dvb-frontends/cx24117.h | 1 - drivers/media/dvb-frontends/cx24120.h | 1 - drivers/media/dvb-frontends/cx24123.h | 1 - drivers/media/dvb-frontends/cxd2820r.h | 1 - drivers/media/dvb-frontends/cxd2841er.h | 1 - drivers/media/dvb-frontends/dib3000mc.h | 2 -- drivers/media/dvb-frontends/dib7000m.h | 2 -- drivers/media/dvb-frontends/dib7000p.h | 2 -- drivers/media/dvb-frontends/drxd.h | 1 - drivers/media/dvb-frontends/drxk.h | 1 - drivers/media/dvb-frontends/ds3000.h | 1 - drivers/media/dvb-frontends/dvb_dummy_fe.h | 1 - drivers/media/dvb-frontends/ec100.h | 1 - drivers/media/dvb-frontends/hd29l2.h | 1 - drivers/media/dvb-frontends/helene.h | 1 - drivers/media/dvb-frontends/horus3a.h | 1 - drivers/media/dvb-frontends/ix2505v.h | 1 - drivers/media/dvb-frontends/lg2160.h | 1 - drivers/media/dvb-frontends/lgdt3305.h | 1 - drivers/media/dvb-frontends/lgs8gl5.h | 1 - drivers/media/dvb-frontends/lgs8gxx.h | 1 - drivers/media/dvb-frontends/lnbh24.h | 2 -- drivers/media/dvb-frontends/lnbh25.h | 1 - drivers/media/dvb-frontends/lnbp21.h | 2 -- drivers/media/dvb-frontends/lnbp22.h | 2 -- drivers/media/dvb-frontends/m88rs2000.h | 1 - drivers/media/dvb-frontends/mb86a20s.h | 1 - drivers/media/dvb-frontends/s5h1409.h | 1 - drivers/media/dvb-frontends/s5h1411.h | 1 - drivers/media/dvb-frontends/s5h1432.h | 1 - drivers/media/dvb-frontends/s921.h | 1 - drivers/media/dvb-frontends/si21xx.h | 1 - drivers/media/dvb-frontends/sp2.h | 1 - drivers/media/dvb-frontends/stb6000.h | 1 - drivers/media/dvb-frontends/stv0288.h | 1 - drivers/media/dvb-frontends/stv0367.h | 1 - drivers/media/dvb-frontends/stv0900.h | 1 - drivers/media/dvb-frontends/stv6110.h | 1 - drivers/media/dvb-frontends/tda10048.h | 1 - drivers/media/dvb-frontends/tda18271c2dd.h | 2 -- drivers/media/dvb-frontends/ts2020.h | 1 - drivers/media/dvb-frontends/zl10036.h | 1 - drivers/media/dvb-frontends/zl10039.h | 2 -- drivers/media/pci/cx23885/altera-ci.h | 2 -- drivers/media/tuners/fc0011.h | 1 - drivers/media/tuners/fc0012.h | 1 - drivers/media/tuners/fc0013.h | 1 - drivers/media/tuners/max2165.h | 2 -- drivers/media/tuners/mc44s803.h | 2 -- drivers/media/tuners/mxl5005s.h | 2 -- drivers/media/tuners/r820t.h | 1 - drivers/media/tuners/si2157.h | 1 - drivers/media/tuners/tda18212.h | 1 - drivers/media/tuners/tda18218.h | 1 - drivers/media/tuners/xc5000.h | 1 - drivers/media/usb/dvb-usb-v2/mxl111sf-demod.h | 1 - drivers/media/usb/dvb-usb-v2/mxl111sf-tuner.h | 1 - drivers/media/usb/dvb-usb/dibusb-common.c | 1 - drivers/media/usb/hdpvr/hdpvr-video.c | 1 - drivers/mtd/mtdcore.c | 1 - drivers/mtd/mtdpart.c | 1 - drivers/net/dsa/b53/b53_mmap.c | 1 - drivers/net/ethernet/sun/ldmvsw.c | 1 - drivers/net/ethernet/wiznet/w5100.c | 1 - drivers/net/ethernet/wiznet/w5300.c | 1 - drivers/usb/early/ehci-dbgp.c | 1 - drivers/usb/gadget/udc/bcm63xx_udc.c | 1 - drivers/usb/host/pci-quirks.c | 1 - fs/lockd/procfs.h | 2 -- include/linux/export.h | 1 - include/linux/gpio/driver.h | 1 - net/batman-adv/debugfs.h | 2 -- sound/soc/intel/common/sst-acpi.h | 1 - tools/testing/nvdimm/config_check.c | 1 - 96 files changed, 112 deletions(-) diff --git a/arch/arm/include/asm/trusted_foundations.h b/arch/arm/include/asm/trusted_foundations.h index 624e1d4..0074835 100644 --- a/arch/arm/include/asm/trusted_foundations.h +++ b/arch/arm/include/asm/trusted_foundations.h @@ -26,7 +26,6 @@ #ifndef __ASM_ARM_TRUSTED_FOUNDATIONS_H #define __ASM_ARM_TRUSTED_FOUNDATIONS_H -#include <linux/kconfig.h> #include <linux/printk.h> #include <linux/bug.h> #include <linux/of.h> diff --git a/arch/arm64/include/asm/alternative.h b/arch/arm64/include/asm/alternative.h index 8746ff6..e998a1f 100644 --- a/arch/arm64/include/asm/alternative.h +++ b/arch/arm64/include/asm/alternative.h @@ -6,7 +6,6 @@ #ifndef __ASSEMBLY__ #include <linux/init.h> -#include <linux/kconfig.h> #include <linux/types.h> #include <linux/stddef.h> #include <linux/stringify.h> diff --git a/arch/mips/include/asm/mach-loongson64/loongson.h b/arch/mips/include/asm/mach-loongson64/loongson.h index d1ff774..c68c0cc 100644 --- a/arch/mips/include/asm/mach-loongson64/loongson.h +++ b/arch/mips/include/asm/mach-loongson64/loongson.h @@ -14,7 +14,6 @@ #include <linux/io.h> #include <linux/init.h> #include <linux/irq.h> -#include <linux/kconfig.h> #include <boot_param.h> /* loongson internal northbridge initialization */ diff --git a/arch/mips/math-emu/cp1emu.c b/arch/mips/math-emu/cp1emu.c index 36775d2..f8b7bf8 100644 --- a/arch/mips/math-emu/cp1emu.c +++ b/arch/mips/math-emu/cp1emu.c @@ -35,7 +35,6 @@ */ #include <linux/sched.h> #include <linux/debugfs.h> -#include <linux/kconfig.h> #include <linux/percpu-defs.h> #include <linux/perf_event.h> diff --git a/arch/mips/net/bpf_jit.c b/arch/mips/net/bpf_jit.c index 39e7b47..49a2e22 100644 --- a/arch/mips/net/bpf_jit.c +++ b/arch/mips/net/bpf_jit.c @@ -14,7 +14,6 @@ #include <linux/errno.h> #include <linux/filter.h> #include <linux/if_vlan.h> -#include <linux/kconfig.h> #include <linux/moduleloader.h> #include <linux/netdevice.h> #include <linux/string.h> diff --git a/drivers/char/virtio_console.c b/drivers/char/virtio_console.c index 5da47e26..65a6e04 100644 --- a/drivers/char/virtio_console.c +++ b/drivers/char/virtio_console.c @@ -38,7 +38,6 @@ #include <linux/workqueue.h> #include <linux/module.h> #include <linux/dma-mapping.h> -#include <linux/kconfig.h> #include "../tty/hvc/hvc_console.h" #define is_rproc_enabled IS_ENABLED(CONFIG_REMOTEPROC) diff --git a/drivers/input/rmi4/rmi_bus.c b/drivers/input/rmi4/rmi_bus.c index a735806..e0b5a45 100644 --- a/drivers/input/rmi4/rmi_bus.c +++ b/drivers/input/rmi4/rmi_bus.c @@ -9,7 +9,6 @@ #include <linux/kernel.h> #include <linux/device.h> -#include <linux/kconfig.h> #include <linux/list.h> #include <linux/pm.h> #include <linux/rmi.h> diff --git a/drivers/input/rmi4/rmi_driver.c b/drivers/input/rmi4/rmi_driver.c index c83bce8..4a88312 100644 --- a/drivers/input/rmi4/rmi_driver.c +++ b/drivers/input/rmi4/rmi_driver.c @@ -17,7 +17,6 @@ #include <linux/bitmap.h> #include <linux/delay.h> #include <linux/fs.h> -#include <linux/kconfig.h> #include <linux/pm.h> #include <linux/slab.h> #include <linux/of.h> diff --git a/drivers/input/rmi4/rmi_f01.c b/drivers/input/rmi4/rmi_f01.c index fac81fc..b5d2dfc 100644 --- a/drivers/input/rmi4/rmi_f01.c +++ b/drivers/input/rmi4/rmi_f01.c @@ -8,7 +8,6 @@ */ #include <linux/kernel.h> -#include <linux/kconfig.h> #include <linux/rmi.h> #include <linux/slab.h> #include <linux/uaccess.h> diff --git a/drivers/input/rmi4/rmi_f11.c b/drivers/input/rmi4/rmi_f11.c index 20c7134..f798f42 100644 --- a/drivers/input/rmi4/rmi_f11.c +++ b/drivers/input/rmi4/rmi_f11.c @@ -12,7 +12,6 @@ #include <linux/device.h> #include <linux/input.h> #include <linux/input/mt.h> -#include <linux/kconfig.h> #include <linux/rmi.h> #include <linux/slab.h> #include <linux/of.h> diff --git a/drivers/irqchip/irq-bcm6345-l1.c b/drivers/irqchip/irq-bcm6345-l1.c index b844c89..daa4ae8 100644 --- a/drivers/irqchip/irq-bcm6345-l1.c +++ b/drivers/irqchip/irq-bcm6345-l1.c @@ -52,7 +52,6 @@ #include <linux/bitops.h> #include <linux/cpumask.h> -#include <linux/kconfig.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/interrupt.h> diff --git a/drivers/irqchip/irq-bcm7038-l1.c b/drivers/irqchip/irq-bcm7038-l1.c index 0fea985..353c549 100644 --- a/drivers/irqchip/irq-bcm7038-l1.c +++ b/drivers/irqchip/irq-bcm7038-l1.c @@ -12,7 +12,6 @@ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt #include <linux/bitops.h> -#include <linux/kconfig.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/interrupt.h> diff --git a/drivers/irqchip/irq-bcm7120-l2.c b/drivers/irqchip/irq-bcm7120-l2.c index 0ec9263..64c2692 100644 --- a/drivers/irqchip/irq-bcm7120-l2.c +++ b/drivers/irqchip/irq-bcm7120-l2.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/slab.h> #include <linux/module.h> -#include <linux/kconfig.h> #include <linux/kernel.h> #include <linux/platform_device.h> #include <linux/of.h> diff --git a/drivers/irqchip/irq-brcmstb-l2.c b/drivers/irqchip/irq-brcmstb-l2.c index 1d4a5b4..bddf169 100644 --- a/drivers/irqchip/irq-brcmstb-l2.c +++ b/drivers/irqchip/irq-brcmstb-l2.c @@ -18,7 +18,6 @@ #include <linux/init.h> #include <linux/slab.h> #include <linux/module.h> -#include <linux/kconfig.h> #include <linux/platform_device.h> #include <linux/spinlock.h> #include <linux/of.h> diff --git a/drivers/media/dvb-frontends/af9013.h b/drivers/media/dvb-frontends/af9013.h index 1dcc936..dcdd163 100644 --- a/drivers/media/dvb-frontends/af9013.h +++ b/drivers/media/dvb-frontends/af9013.h @@ -25,7 +25,6 @@ #ifndef AF9013_H #define AF9013_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> /* AF9013/5 GPIOs (mostly guessed) diff --git a/drivers/media/dvb-frontends/af9033.h b/drivers/media/dvb-frontends/af9033.h index 6ad22b6..5b83e4f 100644 --- a/drivers/media/dvb-frontends/af9033.h +++ b/drivers/media/dvb-frontends/af9033.h @@ -22,8 +22,6 @@ #ifndef AF9033_H #define AF9033_H -#include <linux/kconfig.h> - /* * I2C address (TODO: are these in 8-bit format?) * 0x38, 0x3a, 0x3c, 0x3e diff --git a/drivers/media/dvb-frontends/ascot2e.h b/drivers/media/dvb-frontends/ascot2e.h index 6da4ae6..dc61bf7 100644 --- a/drivers/media/dvb-frontends/ascot2e.h +++ b/drivers/media/dvb-frontends/ascot2e.h @@ -22,7 +22,6 @@ #ifndef __DVB_ASCOT2E_H__ #define __DVB_ASCOT2E_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include <linux/i2c.h> diff --git a/drivers/media/dvb-frontends/atbm8830.h b/drivers/media/dvb-frontends/atbm8830.h index 5446d13..bb86238 100644 --- a/drivers/media/dvb-frontends/atbm8830.h +++ b/drivers/media/dvb-frontends/atbm8830.h @@ -22,7 +22,6 @@ #ifndef __ATBM8830_H__ #define __ATBM8830_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include <linux/i2c.h> diff --git a/drivers/media/dvb-frontends/au8522.h b/drivers/media/dvb-frontends/au8522.h index 78bf3f7..21c51a4 100644 --- a/drivers/media/dvb-frontends/au8522.h +++ b/drivers/media/dvb-frontends/au8522.h @@ -22,7 +22,6 @@ #ifndef __AU8522_H__ #define __AU8522_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> enum au8522_if_freq { diff --git a/drivers/media/dvb-frontends/cx22702.h b/drivers/media/dvb-frontends/cx22702.h index 68b69a7..a1956a9 100644 --- a/drivers/media/dvb-frontends/cx22702.h +++ b/drivers/media/dvb-frontends/cx22702.h @@ -28,7 +28,6 @@ #ifndef CX22702_H #define CX22702_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct cx22702_config { diff --git a/drivers/media/dvb-frontends/cx24113.h b/drivers/media/dvb-frontends/cx24113.h index 962919b..194c703 100644 --- a/drivers/media/dvb-frontends/cx24113.h +++ b/drivers/media/dvb-frontends/cx24113.h @@ -22,8 +22,6 @@ #ifndef CX24113_H #define CX24113_H -#include <linux/kconfig.h> - struct dvb_frontend; struct cx24113_config { diff --git a/drivers/media/dvb-frontends/cx24116.h b/drivers/media/dvb-frontends/cx24116.h index f6dbabc..9ff8df8d 100644 --- a/drivers/media/dvb-frontends/cx24116.h +++ b/drivers/media/dvb-frontends/cx24116.h @@ -21,7 +21,6 @@ #ifndef CX24116_H #define CX24116_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct cx24116_config { diff --git a/drivers/media/dvb-frontends/cx24117.h b/drivers/media/dvb-frontends/cx24117.h index 1648ab4..445f13f 100644 --- a/drivers/media/dvb-frontends/cx24117.h +++ b/drivers/media/dvb-frontends/cx24117.h @@ -22,7 +22,6 @@ #ifndef CX24117_H #define CX24117_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct cx24117_config { diff --git a/drivers/media/dvb-frontends/cx24120.h b/drivers/media/dvb-frontends/cx24120.h index f097042..de4ca9a 100644 --- a/drivers/media/dvb-frontends/cx24120.h +++ b/drivers/media/dvb-frontends/cx24120.h @@ -20,7 +20,6 @@ #ifndef CX24120_H #define CX24120_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include <linux/firmware.h> diff --git a/drivers/media/dvb-frontends/cx24123.h b/drivers/media/dvb-frontends/cx24123.h index 975f3c9..aac2344 100644 --- a/drivers/media/dvb-frontends/cx24123.h +++ b/drivers/media/dvb-frontends/cx24123.h @@ -21,7 +21,6 @@ #ifndef CX24123_H #define CX24123_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct cx24123_config { diff --git a/drivers/media/dvb-frontends/cxd2820r.h b/drivers/media/dvb-frontends/cxd2820r.h index 56d4276..297a71a 100644 --- a/drivers/media/dvb-frontends/cxd2820r.h +++ b/drivers/media/dvb-frontends/cxd2820r.h @@ -22,7 +22,6 @@ #ifndef CXD2820R_H #define CXD2820R_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #define CXD2820R_GPIO_D (0 << 0) /* disable */ diff --git a/drivers/media/dvb-frontends/cxd2841er.h b/drivers/media/dvb-frontends/cxd2841er.h index 62ad5f0..7f1acfb 100644 --- a/drivers/media/dvb-frontends/cxd2841er.h +++ b/drivers/media/dvb-frontends/cxd2841er.h @@ -22,7 +22,6 @@ #ifndef CXD2841ER_H #define CXD2841ER_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> enum cxd2841er_xtal { diff --git a/drivers/media/dvb-frontends/dib3000mc.h b/drivers/media/dvb-frontends/dib3000mc.h index b37e69e..67a6d50 100644 --- a/drivers/media/dvb-frontends/dib3000mc.h +++ b/drivers/media/dvb-frontends/dib3000mc.h @@ -13,8 +13,6 @@ #ifndef DIB3000MC_H #define DIB3000MC_H -#include <linux/kconfig.h> - #include "dibx000_common.h" struct dib3000mc_config { diff --git a/drivers/media/dvb-frontends/dib7000m.h b/drivers/media/dvb-frontends/dib7000m.h index 6468c27..8f84dfa 100644 --- a/drivers/media/dvb-frontends/dib7000m.h +++ b/drivers/media/dvb-frontends/dib7000m.h @@ -1,8 +1,6 @@ #ifndef DIB7000M_H #define DIB7000M_H -#include <linux/kconfig.h> - #include "dibx000_common.h" struct dib7000m_config { diff --git a/drivers/media/dvb-frontends/dib7000p.h b/drivers/media/dvb-frontends/dib7000p.h index baa2789..205fbbf 100644 --- a/drivers/media/dvb-frontends/dib7000p.h +++ b/drivers/media/dvb-frontends/dib7000p.h @@ -1,8 +1,6 @@ #ifndef DIB7000P_H #define DIB7000P_H -#include <linux/kconfig.h> - #include "dibx000_common.h" struct dib7000p_config { diff --git a/drivers/media/dvb-frontends/drxd.h b/drivers/media/dvb-frontends/drxd.h index a47c22d..f0507cd 100644 --- a/drivers/media/dvb-frontends/drxd.h +++ b/drivers/media/dvb-frontends/drxd.h @@ -24,7 +24,6 @@ #ifndef _DRXD_H_ #define _DRXD_H_ -#include <linux/kconfig.h> #include <linux/types.h> #include <linux/i2c.h> diff --git a/drivers/media/dvb-frontends/drxk.h b/drivers/media/dvb-frontends/drxk.h index 8f0b9ee..a629897 100644 --- a/drivers/media/dvb-frontends/drxk.h +++ b/drivers/media/dvb-frontends/drxk.h @@ -1,7 +1,6 @@ #ifndef _DRXK_H_ #define _DRXK_H_ -#include <linux/kconfig.h> #include <linux/types.h> #include <linux/i2c.h> diff --git a/drivers/media/dvb-frontends/ds3000.h b/drivers/media/dvb-frontends/ds3000.h index 153169d..82e8c25 100644 --- a/drivers/media/dvb-frontends/ds3000.h +++ b/drivers/media/dvb-frontends/ds3000.h @@ -22,7 +22,6 @@ #ifndef DS3000_H #define DS3000_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct ds3000_config { diff --git a/drivers/media/dvb-frontends/dvb_dummy_fe.h b/drivers/media/dvb-frontends/dvb_dummy_fe.h index 15e4cea..50f1af5 100644 --- a/drivers/media/dvb-frontends/dvb_dummy_fe.h +++ b/drivers/media/dvb-frontends/dvb_dummy_fe.h @@ -22,7 +22,6 @@ #ifndef DVB_DUMMY_FE_H #define DVB_DUMMY_FE_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/ec100.h b/drivers/media/dvb-frontends/ec100.h index 9544bab..e894bdc 100644 --- a/drivers/media/dvb-frontends/ec100.h +++ b/drivers/media/dvb-frontends/ec100.h @@ -22,7 +22,6 @@ #ifndef EC100_H #define EC100_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct ec100_config { diff --git a/drivers/media/dvb-frontends/hd29l2.h b/drivers/media/dvb-frontends/hd29l2.h index 48e9ab7..a14d6f3 100644 --- a/drivers/media/dvb-frontends/hd29l2.h +++ b/drivers/media/dvb-frontends/hd29l2.h @@ -23,7 +23,6 @@ #ifndef HD29L2_H #define HD29L2_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct hd29l2_config { diff --git a/drivers/media/dvb-frontends/helene.h b/drivers/media/dvb-frontends/helene.h index e1b9224..3336154 100644 --- a/drivers/media/dvb-frontends/helene.h +++ b/drivers/media/dvb-frontends/helene.h @@ -21,7 +21,6 @@ #ifndef __DVB_HELENE_H__ #define __DVB_HELENE_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include <linux/i2c.h> diff --git a/drivers/media/dvb-frontends/horus3a.h b/drivers/media/dvb-frontends/horus3a.h index c1e2d18..672a556 100644 --- a/drivers/media/dvb-frontends/horus3a.h +++ b/drivers/media/dvb-frontends/horus3a.h @@ -22,7 +22,6 @@ #ifndef __DVB_HORUS3A_H__ #define __DVB_HORUS3A_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include <linux/i2c.h> diff --git a/drivers/media/dvb-frontends/ix2505v.h b/drivers/media/dvb-frontends/ix2505v.h index af107a2..5eab397 100644 --- a/drivers/media/dvb-frontends/ix2505v.h +++ b/drivers/media/dvb-frontends/ix2505v.h @@ -20,7 +20,6 @@ #ifndef DVB_IX2505V_H #define DVB_IX2505V_H -#include <linux/kconfig.h> #include <linux/i2c.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/lg2160.h b/drivers/media/dvb-frontends/lg2160.h index d20bd90..8c74ddc 100644 --- a/drivers/media/dvb-frontends/lg2160.h +++ b/drivers/media/dvb-frontends/lg2160.h @@ -22,7 +22,6 @@ #ifndef _LG2160_H_ #define _LG2160_H_ -#include <linux/kconfig.h> #include <linux/i2c.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/lgdt3305.h b/drivers/media/dvb-frontends/lgdt3305.h index f91a1b4..e7dceb6 100644 --- a/drivers/media/dvb-frontends/lgdt3305.h +++ b/drivers/media/dvb-frontends/lgdt3305.h @@ -22,7 +22,6 @@ #ifndef _LGDT3305_H_ #define _LGDT3305_H_ -#include <linux/kconfig.h> #include <linux/i2c.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/lgs8gl5.h b/drivers/media/dvb-frontends/lgs8gl5.h index a5b3faf..f36a7fd 100644 --- a/drivers/media/dvb-frontends/lgs8gl5.h +++ b/drivers/media/dvb-frontends/lgs8gl5.h @@ -23,7 +23,6 @@ #ifndef LGS8GL5_H #define LGS8GL5_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct lgs8gl5_config { diff --git a/drivers/media/dvb-frontends/lgs8gxx.h b/drivers/media/dvb-frontends/lgs8gxx.h index 368c992..7519c02 100644 --- a/drivers/media/dvb-frontends/lgs8gxx.h +++ b/drivers/media/dvb-frontends/lgs8gxx.h @@ -26,7 +26,6 @@ #ifndef __LGS8GXX_H__ #define __LGS8GXX_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include <linux/i2c.h> diff --git a/drivers/media/dvb-frontends/lnbh24.h b/drivers/media/dvb-frontends/lnbh24.h index a088b8e..24431df 100644 --- a/drivers/media/dvb-frontends/lnbh24.h +++ b/drivers/media/dvb-frontends/lnbh24.h @@ -23,8 +23,6 @@ #ifndef _LNBH24_H #define _LNBH24_H -#include <linux/kconfig.h> - /* system register bits */ #define LNBH24_OLF 0x01 #define LNBH24_OTF 0x02 diff --git a/drivers/media/dvb-frontends/lnbh25.h b/drivers/media/dvb-frontends/lnbh25.h index 1f329ef..f13fd03 100644 --- a/drivers/media/dvb-frontends/lnbh25.h +++ b/drivers/media/dvb-frontends/lnbh25.h @@ -22,7 +22,6 @@ #define LNBH25_H #include <linux/i2c.h> -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> /* 22 kHz tone enabled. Tone output controlled by DSQIN pin */ diff --git a/drivers/media/dvb-frontends/lnbp21.h b/drivers/media/dvb-frontends/lnbp21.h index cd9101f..4bb6439 100644 --- a/drivers/media/dvb-frontends/lnbp21.h +++ b/drivers/media/dvb-frontends/lnbp21.h @@ -27,8 +27,6 @@ #ifndef _LNBP21_H #define _LNBP21_H -#include <linux/kconfig.h> - /* system register bits */ /* [RO] 0=OK; 1=over current limit flag */ #define LNBP21_OLF 0x01 diff --git a/drivers/media/dvb-frontends/lnbp22.h b/drivers/media/dvb-frontends/lnbp22.h index 5d01d92..0cb7212 100644 --- a/drivers/media/dvb-frontends/lnbp22.h +++ b/drivers/media/dvb-frontends/lnbp22.h @@ -28,8 +28,6 @@ #ifndef _LNBP22_H #define _LNBP22_H -#include <linux/kconfig.h> - /* Enable */ #define LNBP22_EN 0x10 /* Voltage selection */ diff --git a/drivers/media/dvb-frontends/m88rs2000.h b/drivers/media/dvb-frontends/m88rs2000.h index de74301..1a313b0 100644 --- a/drivers/media/dvb-frontends/m88rs2000.h +++ b/drivers/media/dvb-frontends/m88rs2000.h @@ -20,7 +20,6 @@ #ifndef M88RS2000_H #define M88RS2000_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/mb86a20s.h b/drivers/media/dvb-frontends/mb86a20s.h index a113282..dfb02db 100644 --- a/drivers/media/dvb-frontends/mb86a20s.h +++ b/drivers/media/dvb-frontends/mb86a20s.h @@ -16,7 +16,6 @@ #ifndef MB86A20S_H #define MB86A20S_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> /** diff --git a/drivers/media/dvb-frontends/s5h1409.h b/drivers/media/dvb-frontends/s5h1409.h index f58b9ca..b38557c 100644 --- a/drivers/media/dvb-frontends/s5h1409.h +++ b/drivers/media/dvb-frontends/s5h1409.h @@ -22,7 +22,6 @@ #ifndef __S5H1409_H__ #define __S5H1409_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct s5h1409_config { diff --git a/drivers/media/dvb-frontends/s5h1411.h b/drivers/media/dvb-frontends/s5h1411.h index f3a87f7..791bab0 100644 --- a/drivers/media/dvb-frontends/s5h1411.h +++ b/drivers/media/dvb-frontends/s5h1411.h @@ -22,7 +22,6 @@ #ifndef __S5H1411_H__ #define __S5H1411_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #define S5H1411_I2C_TOP_ADDR (0x32 >> 1) diff --git a/drivers/media/dvb-frontends/s5h1432.h b/drivers/media/dvb-frontends/s5h1432.h index f490c5e..b81c9bd 100644 --- a/drivers/media/dvb-frontends/s5h1432.h +++ b/drivers/media/dvb-frontends/s5h1432.h @@ -22,7 +22,6 @@ #ifndef __S5H1432_H__ #define __S5H1432_H__ -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #define S5H1432_I2C_TOP_ADDR (0x02 >> 1) diff --git a/drivers/media/dvb-frontends/s921.h b/drivers/media/dvb-frontends/s921.h index f5b722d..a47ed89 100644 --- a/drivers/media/dvb-frontends/s921.h +++ b/drivers/media/dvb-frontends/s921.h @@ -17,7 +17,6 @@ #ifndef S921_H #define S921_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct s921_config { diff --git a/drivers/media/dvb-frontends/si21xx.h b/drivers/media/dvb-frontends/si21xx.h index ef5f351..b1be62f 100644 --- a/drivers/media/dvb-frontends/si21xx.h +++ b/drivers/media/dvb-frontends/si21xx.h @@ -1,7 +1,6 @@ #ifndef SI21XX_H #define SI21XX_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/sp2.h b/drivers/media/dvb-frontends/sp2.h index 6cceea0..3901cd7 100644 --- a/drivers/media/dvb-frontends/sp2.h +++ b/drivers/media/dvb-frontends/sp2.h @@ -17,7 +17,6 @@ #ifndef SP2_H #define SP2_H -#include <linux/kconfig.h> #include "dvb_ca_en50221.h" /* diff --git a/drivers/media/dvb-frontends/stb6000.h b/drivers/media/dvb-frontends/stb6000.h index da581b6..78e75df 100644 --- a/drivers/media/dvb-frontends/stb6000.h +++ b/drivers/media/dvb-frontends/stb6000.h @@ -23,7 +23,6 @@ #ifndef __DVB_STB6000_H__ #define __DVB_STB6000_H__ -#include <linux/kconfig.h> #include <linux/i2c.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/stv0288.h b/drivers/media/dvb-frontends/stv0288.h index b58603c..803acb9 100644 --- a/drivers/media/dvb-frontends/stv0288.h +++ b/drivers/media/dvb-frontends/stv0288.h @@ -27,7 +27,6 @@ #ifndef STV0288_H #define STV0288_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/stv0367.h b/drivers/media/dvb-frontends/stv0367.h index 92b3e85..b88166a 100644 --- a/drivers/media/dvb-frontends/stv0367.h +++ b/drivers/media/dvb-frontends/stv0367.h @@ -26,7 +26,6 @@ #ifndef STV0367_H #define STV0367_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/stv0900.h b/drivers/media/dvb-frontends/stv0900.h index c90bf00..9ca2da9 100644 --- a/drivers/media/dvb-frontends/stv0900.h +++ b/drivers/media/dvb-frontends/stv0900.h @@ -26,7 +26,6 @@ #ifndef STV0900_H #define STV0900_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/stv6110.h b/drivers/media/dvb-frontends/stv6110.h index f3c8a5c..4604f79 100644 --- a/drivers/media/dvb-frontends/stv6110.h +++ b/drivers/media/dvb-frontends/stv6110.h @@ -25,7 +25,6 @@ #ifndef __DVB_STV6110_H__ #define __DVB_STV6110_H__ -#include <linux/kconfig.h> #include <linux/i2c.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/tda10048.h b/drivers/media/dvb-frontends/tda10048.h index bc77a73..a2cebb0 100644 --- a/drivers/media/dvb-frontends/tda10048.h +++ b/drivers/media/dvb-frontends/tda10048.h @@ -22,7 +22,6 @@ #ifndef TDA10048_H #define TDA10048_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> #include <linux/firmware.h> diff --git a/drivers/media/dvb-frontends/tda18271c2dd.h b/drivers/media/dvb-frontends/tda18271c2dd.h index 7ebd8ea..e6ccf24 100644 --- a/drivers/media/dvb-frontends/tda18271c2dd.h +++ b/drivers/media/dvb-frontends/tda18271c2dd.h @@ -1,8 +1,6 @@ #ifndef _TDA18271C2DD_H_ #define _TDA18271C2DD_H_ -#include <linux/kconfig.h> - #if IS_REACHABLE(CONFIG_DVB_TDA18271C2DD) struct dvb_frontend *tda18271c2dd_attach(struct dvb_frontend *fe, struct i2c_adapter *i2c, u8 adr); diff --git a/drivers/media/dvb-frontends/ts2020.h b/drivers/media/dvb-frontends/ts2020.h index 9220e5c..facc54f 100644 --- a/drivers/media/dvb-frontends/ts2020.h +++ b/drivers/media/dvb-frontends/ts2020.h @@ -22,7 +22,6 @@ #ifndef TS2020_H #define TS2020_H -#include <linux/kconfig.h> #include <linux/dvb/frontend.h> struct ts2020_config { diff --git a/drivers/media/dvb-frontends/zl10036.h b/drivers/media/dvb-frontends/zl10036.h index 670e76a..c568d8d 100644 --- a/drivers/media/dvb-frontends/zl10036.h +++ b/drivers/media/dvb-frontends/zl10036.h @@ -21,7 +21,6 @@ #ifndef DVB_ZL10036_H #define DVB_ZL10036_H -#include <linux/kconfig.h> #include <linux/i2c.h> #include "dvb_frontend.h" diff --git a/drivers/media/dvb-frontends/zl10039.h b/drivers/media/dvb-frontends/zl10039.h index 0709294..66e7085 100644 --- a/drivers/media/dvb-frontends/zl10039.h +++ b/drivers/media/dvb-frontends/zl10039.h @@ -22,8 +22,6 @@ #ifndef ZL10039_H #define ZL10039_H -#include <linux/kconfig.h> - #if IS_REACHABLE(CONFIG_DVB_ZL10039) struct dvb_frontend *zl10039_attach(struct dvb_frontend *fe, u8 i2c_addr, diff --git a/drivers/media/pci/cx23885/altera-ci.h b/drivers/media/pci/cx23885/altera-ci.h index 6c51172..57a40c8 100644 --- a/drivers/media/pci/cx23885/altera-ci.h +++ b/drivers/media/pci/cx23885/altera-ci.h @@ -20,8 +20,6 @@ #ifndef __ALTERA_CI_H #define __ALTERA_CI_H -#include <linux/kconfig.h> - #define ALT_DATA 0x000000ff #define ALT_TDI 0x00008000 #define ALT_TDO 0x00004000 diff --git a/drivers/media/tuners/fc0011.h b/drivers/media/tuners/fc0011.h index 81bb568..438cf89 100644 --- a/drivers/media/tuners/fc0011.h +++ b/drivers/media/tuners/fc0011.h @@ -1,7 +1,6 @@ #ifndef LINUX_FC0011_H_ #define LINUX_FC0011_H_ -#include <linux/kconfig.h> #include "dvb_frontend.h" diff --git a/drivers/media/tuners/fc0012.h b/drivers/media/tuners/fc0012.h index 9ad3285..4a23e41 100644 --- a/drivers/media/tuners/fc0012.h +++ b/drivers/media/tuners/fc0012.h @@ -21,7 +21,6 @@ #ifndef _FC0012_H_ #define _FC0012_H_ -#include <linux/kconfig.h> #include "dvb_frontend.h" #include "fc001x-common.h" diff --git a/drivers/media/tuners/fc0013.h b/drivers/media/tuners/fc0013.h index e130bd7..8c34105 100644 --- a/drivers/media/tuners/fc0013.h +++ b/drivers/media/tuners/fc0013.h @@ -22,7 +22,6 @@ #ifndef _FC0013_H_ #define _FC0013_H_ -#include <linux/kconfig.h> #include "dvb_frontend.h" #include "fc001x-common.h" diff --git a/drivers/media/tuners/max2165.h b/drivers/media/tuners/max2165.h index 5054f01..aadd9fe 100644 --- a/drivers/media/tuners/max2165.h +++ b/drivers/media/tuners/max2165.h @@ -22,8 +22,6 @@ #ifndef __MAX2165_H__ #define __MAX2165_H__ -#include <linux/kconfig.h> - struct dvb_frontend; struct i2c_adapter; diff --git a/drivers/media/tuners/mc44s803.h b/drivers/media/tuners/mc44s803.h index b3e614b..6b40df3 100644 --- a/drivers/media/tuners/mc44s803.h +++ b/drivers/media/tuners/mc44s803.h @@ -22,8 +22,6 @@ #ifndef MC44S803_H #define MC44S803_H -#include <linux/kconfig.h> - struct dvb_frontend; struct i2c_adapter; diff --git a/drivers/media/tuners/mxl5005s.h b/drivers/media/tuners/mxl5005s.h index 5764b12..d842734 100644 --- a/drivers/media/tuners/mxl5005s.h +++ b/drivers/media/tuners/mxl5005s.h @@ -23,8 +23,6 @@ #ifndef __MXL5005S_H #define __MXL5005S_H -#include <linux/kconfig.h> - #include <linux/i2c.h> #include "dvb_frontend.h" diff --git a/drivers/media/tuners/r820t.h b/drivers/media/tuners/r820t.h index b1e5661..fdcab91 100644 --- a/drivers/media/tuners/r820t.h +++ b/drivers/media/tuners/r820t.h @@ -21,7 +21,6 @@ #ifndef R820T_H #define R820T_H -#include <linux/kconfig.h> #include "dvb_frontend.h" enum r820t_chip { diff --git a/drivers/media/tuners/si2157.h b/drivers/media/tuners/si2157.h index 5f1a60b..76807f5 100644 --- a/drivers/media/tuners/si2157.h +++ b/drivers/media/tuners/si2157.h @@ -17,7 +17,6 @@ #ifndef SI2157_H #define SI2157_H -#include <linux/kconfig.h> #include <media/media-device.h> #include "dvb_frontend.h" diff --git a/drivers/media/tuners/tda18212.h b/drivers/media/tuners/tda18212.h index e58c909..6391daf 100644 --- a/drivers/media/tuners/tda18212.h +++ b/drivers/media/tuners/tda18212.h @@ -21,7 +21,6 @@ #ifndef TDA18212_H #define TDA18212_H -#include <linux/kconfig.h> #include "dvb_frontend.h" struct tda18212_config { diff --git a/drivers/media/tuners/tda18218.h b/drivers/media/tuners/tda18218.h index 1eacb4f..076b5f2 100644 --- a/drivers/media/tuners/tda18218.h +++ b/drivers/media/tuners/tda18218.h @@ -21,7 +21,6 @@ #ifndef TDA18218_H #define TDA18218_H -#include <linux/kconfig.h> #include "dvb_frontend.h" struct tda18218_config { diff --git a/drivers/media/tuners/xc5000.h b/drivers/media/tuners/xc5000.h index 00ba29e..336bd49 100644 --- a/drivers/media/tuners/xc5000.h +++ b/drivers/media/tuners/xc5000.h @@ -22,7 +22,6 @@ #ifndef __XC5000_H__ #define __XC5000_H__ -#include <linux/kconfig.h> #include <linux/firmware.h> struct dvb_frontend; diff --git a/drivers/media/usb/dvb-usb-v2/mxl111sf-demod.h b/drivers/media/usb/dvb-usb-v2/mxl111sf-demod.h index 7065aca..e6eae9d 100644 --- a/drivers/media/usb/dvb-usb-v2/mxl111sf-demod.h +++ b/drivers/media/usb/dvb-usb-v2/mxl111sf-demod.h @@ -21,7 +21,6 @@ #ifndef __MXL111SF_DEMOD_H__ #define __MXL111SF_DEMOD_H__ -#include <linux/kconfig.h> #include "dvb_frontend.h" #include "mxl111sf.h" diff --git a/drivers/media/usb/dvb-usb-v2/mxl111sf-tuner.h b/drivers/media/usb/dvb-usb-v2/mxl111sf-tuner.h index 509b550..e96d9a4 100644 --- a/drivers/media/usb/dvb-usb-v2/mxl111sf-tuner.h +++ b/drivers/media/usb/dvb-usb-v2/mxl111sf-tuner.h @@ -21,7 +21,6 @@ #ifndef __MXL111SF_TUNER_H__ #define __MXL111SF_TUNER_H__ -#include <linux/kconfig.h> #include "dvb_frontend.h" #include "mxl111sf.h" diff --git a/drivers/media/usb/dvb-usb/dibusb-common.c b/drivers/media/usb/dvb-usb/dibusb-common.c index 6eea4e6..9e94b17 100644 --- a/drivers/media/usb/dvb-usb/dibusb-common.c +++ b/drivers/media/usb/dvb-usb/dibusb-common.c @@ -9,7 +9,6 @@ * see Documentation/dvb/README.dvb-usb for more information */ -#include <linux/kconfig.h> #include "dibusb.h" /* Max transfer size done by I2C transfer functions */ diff --git a/drivers/media/usb/hdpvr/hdpvr-video.c b/drivers/media/usb/hdpvr/hdpvr-video.c index 2a3a8b4..f3325b0 100644 --- a/drivers/media/usb/hdpvr/hdpvr-video.c +++ b/drivers/media/usb/hdpvr/hdpvr-video.c @@ -10,7 +10,6 @@ */ #include <linux/kernel.h> -#include <linux/kconfig.h> #include <linux/errno.h> #include <linux/init.h> #include <linux/slab.h> diff --git a/drivers/mtd/mtdcore.c b/drivers/mtd/mtdcore.c index e3936b8..44da7b8 100644 --- a/drivers/mtd/mtdcore.c +++ b/drivers/mtd/mtdcore.c @@ -39,7 +39,6 @@ #include <linux/gfp.h> #include <linux/slab.h> #include <linux/reboot.h> -#include <linux/kconfig.h> #include <linux/leds.h> #include <linux/mtd/mtd.h> diff --git a/drivers/mtd/mtdpart.c b/drivers/mtd/mtdpart.c index 1f13e32..3f6034c 100644 --- a/drivers/mtd/mtdpart.c +++ b/drivers/mtd/mtdpart.c @@ -30,7 +30,6 @@ #include <linux/mtd/mtd.h> #include <linux/mtd/partitions.h> #include <linux/err.h> -#include <linux/kconfig.h> #include "mtdcore.h" diff --git a/drivers/net/dsa/b53/b53_mmap.c b/drivers/net/dsa/b53/b53_mmap.c index 77ffc43..4432681 100644 --- a/drivers/net/dsa/b53/b53_mmap.c +++ b/drivers/net/dsa/b53/b53_mmap.c @@ -17,7 +17,6 @@ */ #include <linux/kernel.h> -#include <linux/kconfig.h> #include <linux/module.h> #include <linux/io.h> #include <linux/platform_device.h> diff --git a/drivers/net/ethernet/sun/ldmvsw.c b/drivers/net/ethernet/sun/ldmvsw.c index e15bf84..0ac449a 100644 --- a/drivers/net/ethernet/sun/ldmvsw.c +++ b/drivers/net/ethernet/sun/ldmvsw.c @@ -11,7 +11,6 @@ #include <linux/highmem.h> #include <linux/if_vlan.h> #include <linux/init.h> -#include <linux/kconfig.h> #include <linux/kernel.h> #include <linux/module.h> #include <linux/mutex.h> diff --git a/drivers/net/ethernet/wiznet/w5100.c b/drivers/net/ethernet/wiznet/w5100.c index 37ab46c..d2349a1 100644 --- a/drivers/net/ethernet/wiznet/w5100.c +++ b/drivers/net/ethernet/wiznet/w5100.c @@ -9,7 +9,6 @@ #include <linux/kernel.h> #include <linux/module.h> -#include <linux/kconfig.h> #include <linux/netdevice.h> #include <linux/etherdevice.h> #include <linux/platform_device.h> diff --git a/drivers/net/ethernet/wiznet/w5300.c b/drivers/net/ethernet/wiznet/w5300.c index 0b37ce9..ca31a57 100644 --- a/drivers/net/ethernet/wiznet/w5300.c +++ b/drivers/net/ethernet/wiznet/w5300.c @@ -10,7 +10,6 @@ #include <linux/kernel.h> #include <linux/module.h> -#include <linux/kconfig.h> #include <linux/netdevice.h> #include <linux/etherdevice.h> #include <linux/platform_device.h> diff --git a/drivers/usb/early/ehci-dbgp.c b/drivers/usb/early/ehci-dbgp.c index 12731e6..ea73afb 100644 --- a/drivers/usb/early/ehci-dbgp.c +++ b/drivers/usb/early/ehci-dbgp.c @@ -20,7 +20,6 @@ #include <linux/usb/ehci_def.h> #include <linux/delay.h> #include <linux/serial_core.h> -#include <linux/kconfig.h> #include <linux/kgdb.h> #include <linux/kthread.h> #include <asm/io.h> diff --git a/drivers/usb/gadget/udc/bcm63xx_udc.c b/drivers/usb/gadget/udc/bcm63xx_udc.c index f5fccb3..f785032 100644 --- a/drivers/usb/gadget/udc/bcm63xx_udc.c +++ b/drivers/usb/gadget/udc/bcm63xx_udc.c @@ -21,7 +21,6 @@ #include <linux/errno.h> #include <linux/interrupt.h> #include <linux/ioport.h> -#include <linux/kconfig.h> #include <linux/kernel.h> #include <linux/list.h> #include <linux/module.h> diff --git a/drivers/usb/host/pci-quirks.c b/drivers/usb/host/pci-quirks.c index 35af362..d793f54 100644 --- a/drivers/usb/host/pci-quirks.c +++ b/drivers/usb/host/pci-quirks.c @@ -9,7 +9,6 @@ */ #include <linux/types.h> -#include <linux/kconfig.h> #include <linux/kernel.h> #include <linux/pci.h> #include <linux/delay.h> diff --git a/fs/lockd/procfs.h b/fs/lockd/procfs.h index 2257a13..184a15e 100644 --- a/fs/lockd/procfs.h +++ b/fs/lockd/procfs.h @@ -6,8 +6,6 @@ #ifndef _LOCKD_PROCFS_H #define _LOCKD_PROCFS_H -#include <linux/kconfig.h> - #if IS_ENABLED(CONFIG_PROC_FS) int lockd_create_procfs(void); void lockd_remove_procfs(void); diff --git a/include/linux/export.h b/include/linux/export.h index c565f87..d7df492 100644 --- a/include/linux/export.h +++ b/include/linux/export.h @@ -78,7 +78,6 @@ extern struct module __this_module; #elif defined(CONFIG_TRIM_UNUSED_KSYMS) -#include <linux/kconfig.h> #include <generated/autoksyms.h> #define __EXPORT_SYMBOL(sym, sec) \ diff --git a/include/linux/gpio/driver.h b/include/linux/gpio/driver.h index 50882e0..073d9c5 100644 --- a/include/linux/gpio/driver.h +++ b/include/linux/gpio/driver.h @@ -9,7 +9,6 @@ #include <linux/irqdomain.h> #include <linux/lockdep.h> #include <linux/pinctrl/pinctrl.h> -#include <linux/kconfig.h> struct gpio_desc; struct of_phandle_args; diff --git a/net/batman-adv/debugfs.h b/net/batman-adv/debugfs.h index 1ab4e2e6..992ec26 100644 --- a/net/batman-adv/debugfs.h +++ b/net/batman-adv/debugfs.h @@ -20,8 +20,6 @@ #include "main.h" -#include <linux/kconfig.h> - struct net_device; #define BATADV_DEBUGFS_SUBDIR "batman_adv" diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h index 5d29493..0127422 100644 --- a/sound/soc/intel/common/sst-acpi.h +++ b/sound/soc/intel/common/sst-acpi.h @@ -12,7 +12,6 @@ * */ -#include <linux/kconfig.h> #include <linux/stddef.h> #include <linux/acpi.h> diff --git a/tools/testing/nvdimm/config_check.c b/tools/testing/nvdimm/config_check.c index 878daf3..7dc5a0a 100644 --- a/tools/testing/nvdimm/config_check.c +++ b/tools/testing/nvdimm/config_check.c @@ -1,4 +1,3 @@ -#include <linux/kconfig.h> #include <linux/bug.h> void check(void) -- 1.9.1

1 0

[alsa-devel] No speaker output on Skylake SoC
by infernix 12 Sep '16

12 Sep '16

Hi, On a Skylake based Samsung Tabpro S tablet I am having two alsa issues on 4.7.0 and 4.8.0-rc5: - no output on speaker with any possible configuration, but audio over headphone jack works - playback speed/pitch is doubled (audible over headphones) Sound works fine in Windows obviously, but rebooting from Windows to Linux (as opposed to cold boot) doesn't seem to have any influence on these issues. I've tried modprobing with all the various ALC model quirks to no avail. I've tried toggling various options with amixer (Auto Mute, various speaker related values with amixer cset) and even played with hda-analyzer.py but was unable to get anything from the speakers whatsoever. Since this is Sunrise Point-LP HD Audio, I realize it's fairly new and there may be new quirks required to make this work. I've attached alsa-info.txt; I'm happy to patch and test if someone can give me some direction on what to try, but any help would be much appreciated. Thanks.

1 0

Re: [alsa-devel] Switching from default to plughw:0,0
by Clemens Ladisch 11 Sep '16

11 Sep '16

Gonzalo Garramuño wrote: > El 10/09/16 a las 20:46, Clemens Ladisch escribió: >> Gonzalo Garramuño wrote: >>> My problem is that when I choose the default, which is equivalent to plughw:0,0 >> >> The default actually is PulseAudio or dmix. > > dmix is not installed in my system It's part of ALSA. > Do I have to deal with the pulseaudio api, instead of the ALSA api? When PulseAudio is configured as the default, it can be accessed through the ALSA API. Anyway, the correct name for the default device is "default". To get a list of devices, see "aplay -L". Regards, Clemens

2 1

[alsa-devel] Fwd: HT Omega Claro II support
by German Gomez 11 Sep '16

11 Sep '16

Hello, I've sent this email to Clemens Ladisch, that appears as the developer of the snd-oxygen driver, but just in case he is not the actual maintainer of the driver, I'm forwarding it here. Thanks, - german -------- Forwarded Message -------- Subject: HT Omega Claro II support Date: Thu, 8 Sep 2016 00:32:36 -0700 From: German Gomez <gergomez(a)gmail.com> To: Clemens Ladisch <clemens(a)ladisch.de> Hello, I've read in the alsa-users mailing-list (https://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg29284.html) that there is no support for the HT Omega Claro II card and that you should have to reverse engineer the windows driver to add it. I have that soundcard, although I has been able to use the onboard HDA audio when booting to Linux. But now my onboard audio is not working (it is dead, not working either under Windows) so I'm trying to make the Claro II working under Linux. I've contacted the HT Omega support and to my surprise they actually answered with some useful information. I've attached the email. The summary is that the Claro II should be exactly the same as the other cards by HT Omega using the CMI8788 except that the GPIO for initializing the card is #6. I have compared specs for both Claro (http://www.htomega.com/claro.html) and Claro II (http://www.htomega.com/claro2.html), and there are more differences, the Claro uses AK4396VF and WM8785G chips while the Claro II uses AK4396VF and CS4362 (+ CS5361 for line input). I've checked the code under sound/pci/oxygen and it seems that we have support for the CS4362 for the Xonar D1/DX soundcard. I don't know if having the GPIO pin for initialization and all the other chips already there is enough to add support, of course I'm willing to test any patch that you could provide. I've taken a look at the code, as I'm not bad with C, but audio driver development is completely outside my skills and domain, so I'm quite lost, if you don't have time or if it is not trivial to add support for the Claro II I would understand it, but the card is a really good one and it is quite easy to find in the US, it would be awesome to have support for it in ALSA. Thank you for your time! - german* *

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix headset mic detection problem for several Dell laptops
by Hui Wang 11 Sep '16

11 Sep '16

One of the laptops has the codec ALC256 on it, applying the ALC255_FIXUP_DELL1_MIC_NO_PRESENCE can fix the problem, the rest of laptops have the codec ALC295 on them, they are similar to machines with ALC225, applying the ALC269_FIXUP_DELL1_MIC_NO_PRESENCE can fix the problem. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 575cefd..32cbcf8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5806,6 +5806,13 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x14, 0x90170110}, \ {0x15, 0x0221401f} +#define ALC295_STANDARD_PINS \ + {0x12, 0xb7a60130}, \ + {0x14, 0x90170110}, \ + {0x17, 0x21014020}, \ + {0x18, 0x21a19030}, \ + {0x21, 0x04211020} + #define ALC298_STANDARD_PINS \ {0x12, 0x90a60130}, \ {0x21, 0x03211020} @@ -5911,6 +5918,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170120}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0xb7a60130}, + {0x14, 0x90170110}, + {0x21, 0x02211020}), + 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, {0x12, 0x90a60130}, @@ -6021,6 +6032,8 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x13, 0x90a60140}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC295_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_STANDARD_PINS, {0x17, 0x90170110}), -- 1.9.1

2 1

[alsa-devel] Switching from default to plughw:0,0
by Gonzalo Garramuño 11 Sep '16

11 Sep '16

I have coded an application that uses ALSA and plays back movie's audios, together with their videos. I have pulseaudio installed by default as it comes with Ubuntu Xenian. My goal is to provide the user with an easy pull-down where they can select the audio device to play the audio in. The first audio device is "default" and defaults to ALSA's default in the Ubuntu Sound Preferences. All that works fine. I can change the Ubuntu Sound Preferences and the default driver immediately picks up the other cards or outputs in the system. My problem is that when I choose the default, which is equivalent to plughw:0,0, when I go to force select in my program the plughw:0,0 (HDA Intel PCH: ALC1150 Analog), I get a Device or resource busy. Sure enough, if I do: $ fuser -v /dev/snd/* USUARIO ORDEN DE ACCESO PID /dev/snd/controlC0: gga 3594 F.... pulseaudio /dev/snd/controlC1: gga 3594 F.... pulseaudio /dev/snd/pcmC0D0p: gga 4821 F...m pulseaudio I am very careful in closing the audio before reopening it again. However the resource remains busy for about 7 seconds. Then it turns into: USUARIO ORDEN DE ACCESO PID /dev/snd/controlC0: gga 3594 F.... pulseaudio /dev/snd/controlC1: gga 3594 F.... pulseaudio /dev/snd/pcmC0D0p: gga 4821 F...m mrViewer and the audio works again. My close function looks like: bool ALSAEngine::close() { if ( _pcm_handle ) { int err = snd_pcm_drop( _pcm_handle ); if ( err < 0 ) { std::cerr << "ERROR: [alsa] snd_pcm_drop failed with " << snd_strerror(err) << std::endl; } err = snd_pcm_close( _pcm_handle ); if ( err < 0 ) { std::cerr << "ERROR: [alsa] snd_pcm_close failed with " << snd_strerror(err) << std::endl; } _pcm_handle = NULL; } return true; } and the open function (simplified till it fails). bool ALSAEngine::open( const unsigned channels, const unsigned freq, const AudioFormat format, const unsigned bits ) { try { close(); int status; unsigned int test_format = (unsigned int) format; char buf[256]; /* Open the audio device */ /* Name of device should depend on # channels in spec */ _pcm_handle = NULL; status = snd_pcm_open(&_pcm_handle, device().c_str(), SND_PCM_STREAM_PLAYBACK, 0); if ( status < 0 || _pcm_handle == NULL ) { sprintf( buf, _("Couldn't open audio device %s: %s"), device().c_str(), snd_strerror(status)); THROW(buf); } ....snip..... }

2 2

[alsa-devel] [RFC][PATCH 0/2] ALSA: control: limit life time of user-defined element set
by Takashi Sakamoto 10 Sep '16

10 Sep '16

Hi, This patchset is an example to implement an idea mentioned in this thread: [alsa-devel] [PATCH] Softvol: Allow per process volume control. http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095026.html ALSA control core allows applications to add arbitrary element sets to an instance of control device, while removal of the element sets is voluntarily done by some applications. This is not convenient to certain situations. For example, an application adds some element sets, then aborted due to programming mistakes, and restart. Needless elements can be added one after another. This patchset limits life time of user-defined element set. Basically, the life time is corresponding to a file descriptor opened by a process to add element sets. However, elements can be locked by the other processes. For this case, this patchset counts locked elements, then judge to remove an element set including these elements. My large concern is whether there's any applications to open a file descriptor just to add any element sets. In this case, added element sets are removed immediately at closing the file descriptor unexpectedly to the application, and unavailable to the other processes anymore. In this point, this patchset is intrusive. (But I'm a bit optimistic because use-defined element set is long abandoned not to work well in several items, and there may be probably few consumers in user space. Perhaps, this change is acceptable, I hope.) To reviewers, my aim for this patchset is to show attempts to synchronize life time of user-defined element set to file descriptors. Any objections are welcome. Then, I'd like to receive alternative ideas, especially for definition of the life time. Takashi Sakamoto (2): ALSA: control: limit life time of user-defined element set ALSA: control: bump up protocol version to 2.0.8 include/uapi/sound/asound.h | 2 +- sound/core/control.c | 101 +++++++++++++++++++++++++++++++++++++------- 2 files changed, 87 insertions(+), 16 deletions(-) -- 2.7.4

3 7

[alsa-devel] [RFC][PATCH 0/2] ALSA: control: export all of TLV related macros to user land
by Takashi Sakamoto 10 Sep '16

10 Sep '16

2 7

[alsa-devel] [PATCH] ASoC: da7219: make sure the valid event when startup
by Xing Zheng 09 Sep '16

09 Sep '16

We need to ensure the master bias and jack detection to be enabled before reporting event at the da7219_aad_irq_thread. Otherwise, we may acquire the incorrect the unplug event when the DUT startup with a plugged headphone. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- sound/soc/codecs/da7219-aad.c | 7 +++++++ sound/soc/codecs/da7219-aad.h | 1 + 2 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 4e369a1..cef17c0d 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -38,6 +38,7 @@ void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack) da7219->aad->jack = jack; da7219->aad->jack_inserted = false; + da7219->aad->accdet_en = false; /* Send an initial empty report */ snd_soc_jack_report(jack, 0, DA7219_AAD_REPORT_ALL_MASK); @@ -46,6 +47,8 @@ void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack) snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, DA7219_ACCDET_EN_MASK, (jack ? DA7219_ACCDET_EN_MASK : 0)); + + da7219->aad->accdet_en = true; } EXPORT_SYMBOL_GPL(da7219_aad_jack_det); @@ -293,6 +296,10 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) u8 statusa; int i, report = 0, mask = 0; + /* Ensure the master bias to be enabled */ + if (!da7219_aad->accdet_en) + return IRQ_NONE; + /* Read current IRQ events */ regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, events, DA7219_AAD_IRQ_REG_MAX); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h index 4fccf67..5641965 100644 --- a/sound/soc/codecs/da7219-aad.h +++ b/sound/soc/codecs/da7219-aad.h @@ -200,6 +200,7 @@ struct da7219_aad_priv { struct snd_soc_jack *jack; bool jack_inserted; + bool accdet_en; }; /* AAD control */ -- 1.9.1

2 1

[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] [PATCH] ASoC: add rt5663 codec driver
by Bard Liao 09 Sep '16

09 Sep '16

This is the initial codec driver for both rt5663 rt5668 codec. Signed-off-by: John Lin <john.lin(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5663.txt | 30 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5663.c | 3216 ++++++++++++++++++++ sound/soc/codecs/rt5663.h | 1121 +++++++ 5 files changed, 4375 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5663.txt create mode 100644 sound/soc/codecs/rt5663.c create mode 100644 sound/soc/codecs/rt5663.h diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt new file mode 100644 index 0000000..ca1f875 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -0,0 +1,30 @@ +RT5663/RT5668 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5663" or "realtek,rt5668". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +Pins on the device (for linking into audio routes) for RT5645/RT5650: + + * IN1P + * IN1N + * IN2P + * IN2N + * HPOL + * HPOR + +Example: + +codec: rt5663@12 { + compatible = "realtek,rt5663"; + reg = <0x12>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3fd110..4a86fcf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -113,6 +113,7 @@ 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_RT5663 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -646,6 +647,7 @@ 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_RT5663=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -654,6 +656,7 @@ 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_RT5663=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -698,6 +701,9 @@ config SND_SOC_RT5651 config SND_SOC_RT5659 tristate +config SND_SOC_RT5663 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ffaa523..ab19c79 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -113,6 +113,7 @@ 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-rt5663-objs := rt5663.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -335,6 +336,7 @@ 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-$(CONFIG_SND_SOC_RT5663) += snd-soc-rt5663.o obj-$(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/rt5663.c b/sound/soc/codecs/rt5663.c new file mode 100644 index 0000000..7963a27 --- /dev/null +++ b/sound/soc/codecs/rt5663.c @@ -0,0 +1,3216 @@ +/* + * rt5663.c -- RT5668/RT5663 ALSA SoC audio codec driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Jack Yu <jack.yu(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/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5663.h" +#include "rl6231.h" + +#define RT5668_DEVICE_ID 0x6451 +#define RT5663_DEVICE_ID 0x6406 + +enum { + CODEC_TYPE_RT5668, + CODEC_TYPE_RT5663, +}; + +struct rt5663_priv { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct delayed_work jack_detect_work; + struct snd_soc_jack *hs_jack; + struct timer_list btn_check_timer; + + int codec_type; + int sysclk; + int sysclk_src; + int lrck; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_default rt5668_reg[] = { + { 0x0000, 0x0000 }, + { 0x0001, 0xc8c8 }, + { 0x0002, 0x8080 }, + { 0x0003, 0x8000 }, + { 0x0004, 0xc80a }, + { 0x0005, 0x0000 }, + { 0x0006, 0x0000 }, + { 0x0007, 0x0000 }, + { 0x000a, 0x0000 }, + { 0x000b, 0x0000 }, + { 0x000c, 0x0000 }, + { 0x000d, 0x0000 }, + { 0x000f, 0x0808 }, + { 0x0010, 0x4000 }, + { 0x0011, 0x0000 }, + { 0x0012, 0x1404 }, + { 0x0013, 0x1000 }, + { 0x0014, 0xa00a }, + { 0x0015, 0x0404 }, + { 0x0016, 0x0404 }, + { 0x0017, 0x0011 }, + { 0x0018, 0xafaf }, + { 0x0019, 0xafaf }, + { 0x001a, 0xafaf }, + { 0x001b, 0x0011 }, + { 0x001c, 0x2f2f }, + { 0x001d, 0x2f2f }, + { 0x001e, 0x2f2f }, + { 0x001f, 0x0000 }, + { 0x0020, 0x0000 }, + { 0x0021, 0x0000 }, + { 0x0022, 0x5757 }, + { 0x0023, 0x0039 }, + { 0x0024, 0x000b }, + { 0x0026, 0xc0c0 }, + { 0x0027, 0xc0c0 }, + { 0x0028, 0xc0c0 }, + { 0x0029, 0x8080 }, + { 0x002a, 0xaaaa }, + { 0x002b, 0xaaaa }, + { 0x002c, 0xaba8 }, + { 0x002d, 0x0000 }, + { 0x002e, 0x0000 }, + { 0x002f, 0x0000 }, + { 0x0030, 0x0000 }, + { 0x0031, 0x5000 }, + { 0x0032, 0x0000 }, + { 0x0033, 0x0000 }, + { 0x0034, 0x0000 }, + { 0x0035, 0x0000 }, + { 0x003a, 0x0000 }, + { 0x003b, 0x0000 }, + { 0x003c, 0x00ff }, + { 0x003d, 0x0000 }, + { 0x003e, 0x00ff }, + { 0x003f, 0x0000 }, + { 0x0040, 0x0000 }, + { 0x0041, 0x00ff }, + { 0x0042, 0x0000 }, + { 0x0043, 0x00ff }, + { 0x0044, 0x0c0c }, + { 0x0049, 0xc00b }, + { 0x004a, 0x0000 }, + { 0x004b, 0x031f }, + { 0x004d, 0x0000 }, + { 0x004e, 0x001f }, + { 0x004f, 0x0000 }, + { 0x0050, 0x001f }, + { 0x0052, 0xf000 }, + { 0x0061, 0x0000 }, + { 0x0062, 0x0000 }, + { 0x0063, 0x003e }, + { 0x0064, 0x0000 }, + { 0x0065, 0x0000 }, + { 0x0066, 0x003f }, + { 0x0067, 0x0000 }, + { 0x006b, 0x0000 }, + { 0x006d, 0xff00 }, + { 0x006e, 0x2808 }, + { 0x006f, 0x000a }, + { 0x0070, 0x8000 }, + { 0x0071, 0x8000 }, + { 0x0072, 0x8000 }, + { 0x0073, 0x7000 }, + { 0x0074, 0x7770 }, + { 0x0075, 0x0002 }, + { 0x0076, 0x0001 }, + { 0x0078, 0x00f0 }, + { 0x0079, 0x0000 }, + { 0x007a, 0x0000 }, + { 0x007b, 0x0000 }, + { 0x007c, 0x0000 }, + { 0x007d, 0x0123 }, + { 0x007e, 0x4500 }, + { 0x007f, 0x8003 }, + { 0x0080, 0x0000 }, + { 0x0081, 0x0000 }, + { 0x0082, 0x0000 }, + { 0x0083, 0x0000 }, + { 0x0084, 0x0000 }, + { 0x0085, 0x0000 }, + { 0x0086, 0x0008 }, + { 0x0087, 0x0000 }, + { 0x0088, 0x0000 }, + { 0x0089, 0x0000 }, + { 0x008a, 0x0000 }, + { 0x008b, 0x0000 }, + { 0x008c, 0x0003 }, + { 0x008e, 0x0060 }, + { 0x008f, 0x1000 }, + { 0x0091, 0x0c26 }, + { 0x0092, 0x0073 }, + { 0x0093, 0x0000 }, + { 0x0094, 0x0080 }, + { 0x0098, 0x0000 }, + { 0x0099, 0x0000 }, + { 0x009a, 0x0007 }, + { 0x009f, 0x0000 }, + { 0x00a0, 0x0000 }, + { 0x00a1, 0x0002 }, + { 0x00a2, 0x0001 }, + { 0x00a3, 0x0002 }, + { 0x00a4, 0x0001 }, + { 0x00ae, 0x2040 }, + { 0x00af, 0x0000 }, + { 0x00b6, 0x0000 }, + { 0x00b7, 0x0000 }, + { 0x00b8, 0x0000 }, + { 0x00b9, 0x0000 }, + { 0x00ba, 0x0002 }, + { 0x00bb, 0x0000 }, + { 0x00be, 0x0000 }, + { 0x00c0, 0x0000 }, + { 0x00c1, 0x0aaa }, + { 0x00c2, 0xaa80 }, + { 0x00c3, 0x0003 }, + { 0x00c4, 0x0000 }, + { 0x00d0, 0x0000 }, + { 0x00d1, 0x2244 }, + { 0x00d2, 0x0000 }, + { 0x00d3, 0x3300 }, + { 0x00d4, 0x2200 }, + { 0x00d9, 0x0809 }, + { 0x00da, 0x0000 }, + { 0x00db, 0x0008 }, + { 0x00dc, 0x00c0 }, + { 0x00dd, 0x6724 }, + { 0x00de, 0x3131 }, + { 0x00df, 0x0008 }, + { 0x00e0, 0x4000 }, + { 0x00e1, 0x3131 }, + { 0x00e2, 0x600c }, + { 0x00ea, 0xb320 }, + { 0x00eb, 0x0000 }, + { 0x00ec, 0xb300 }, + { 0x00ed, 0x0000 }, + { 0x00ee, 0xb320 }, + { 0x00ef, 0x0000 }, + { 0x00f0, 0x0201 }, + { 0x00f1, 0x0ddd }, + { 0x00f2, 0x0ddd }, + { 0x00f6, 0x0000 }, + { 0x00f7, 0x0000 }, + { 0x00f8, 0x0000 }, + { 0x00fa, 0x0000 }, + { 0x00fb, 0x0000 }, + { 0x00fc, 0x0000 }, + { 0x00fd, 0x0000 }, + { 0x00fe, 0x10ec }, + { 0x00ff, 0x6451 }, + { 0x0100, 0xaaaa }, + { 0x0101, 0x000a }, + { 0x010a, 0xaaaa }, + { 0x010b, 0xa0a0 }, + { 0x010c, 0xaeae }, + { 0x010d, 0xaaaa }, + { 0x010e, 0xaaaa }, + { 0x010f, 0xaaaa }, + { 0x0110, 0xe002 }, + { 0x0111, 0xa602 }, + { 0x0112, 0xaaaa }, + { 0x0113, 0x2000 }, + { 0x0117, 0x0f00 }, + { 0x0125, 0x0420 }, + { 0x0132, 0x0000 }, + { 0x0133, 0x0000 }, + { 0x0136, 0x5555 }, + { 0x0137, 0x5540 }, + { 0x0138, 0x3700 }, + { 0x0139, 0x79a1 }, + { 0x013a, 0x2020 }, + { 0x013b, 0x2020 }, + { 0x013c, 0x2005 }, + { 0x013f, 0x0000 }, + { 0x0145, 0x0002 }, + { 0x0146, 0x0000 }, + { 0x0147, 0x0000 }, + { 0x0148, 0x0000 }, + { 0x0160, 0x4ec0 }, + { 0x0161, 0x0080 }, + { 0x0162, 0x0200 }, + { 0x0163, 0x0800 }, + { 0x0164, 0x0000 }, + { 0x0165, 0x0000 }, + { 0x0166, 0x0000 }, + { 0x0167, 0x000f }, + { 0x0168, 0x000f }, + { 0x0170, 0x4e80 }, + { 0x0171, 0x0080 }, + { 0x0172, 0x0200 }, + { 0x0173, 0x0800 }, + { 0x0174, 0x00ff }, + { 0x0175, 0x0000 }, + { 0x0190, 0x4131 }, + { 0x0191, 0x4131 }, + { 0x0192, 0x4131 }, + { 0x0193, 0x4131 }, + { 0x0194, 0x0000 }, + { 0x0195, 0x0000 }, + { 0x0196, 0x0000 }, + { 0x0197, 0x0000 }, + { 0x0198, 0x0000 }, + { 0x0199, 0x0000 }, + { 0x01a0, 0x1e64 }, + { 0x01a1, 0x06a3 }, + { 0x01a2, 0x0000 }, + { 0x01a3, 0x0000 }, + { 0x01a4, 0x0000 }, + { 0x01a5, 0x0000 }, + { 0x01a6, 0x0000 }, + { 0x01a7, 0x0000 }, + { 0x01a8, 0x0000 }, + { 0x01a9, 0x0000 }, + { 0x01aa, 0x0000 }, + { 0x01ab, 0x0000 }, + { 0x01b5, 0x0000 }, + { 0x01b6, 0x01c3 }, + { 0x01b7, 0x02a0 }, + { 0x01b8, 0x03e9 }, + { 0x01b9, 0x1389 }, + { 0x01ba, 0xc351 }, + { 0x01bb, 0x0009 }, + { 0x01bc, 0x0018 }, + { 0x01bd, 0x002a }, + { 0x01be, 0x004c }, + { 0x01bf, 0x0097 }, + { 0x01c0, 0x433d }, + { 0x01c1, 0x0000 }, + { 0x01c2, 0x0000 }, + { 0x01c3, 0x0000 }, + { 0x01c4, 0x0000 }, + { 0x01c5, 0x0000 }, + { 0x01c6, 0x0000 }, + { 0x01c7, 0x0000 }, + { 0x01c8, 0x40af }, + { 0x01c9, 0x0702 }, + { 0x01ca, 0x0000 }, + { 0x01cb, 0x0000 }, + { 0x01cc, 0x5757 }, + { 0x01cd, 0x5757 }, + { 0x01ce, 0x5757 }, + { 0x01cf, 0x5757 }, + { 0x01d0, 0x5757 }, + { 0x01d1, 0x5757 }, + { 0x01d2, 0x5757 }, + { 0x01d3, 0x5757 }, + { 0x01d4, 0x5757 }, + { 0x01d5, 0x5757 }, + { 0x01d6, 0x003c }, + { 0x01da, 0x0000 }, + { 0x01db, 0x0000 }, + { 0x01dc, 0x0000 }, + { 0x01de, 0x7c00 }, + { 0x01df, 0x0320 }, + { 0x01e0, 0x06a1 }, + { 0x01e1, 0x0000 }, + { 0x01e2, 0x0000 }, + { 0x01e3, 0x0000 }, + { 0x01e4, 0x0000 }, + { 0x01e5, 0x0000 }, + { 0x01e6, 0x0001 }, + { 0x01e7, 0x0000 }, + { 0x01e8, 0x0000 }, + { 0x01ea, 0x0000 }, + { 0x01eb, 0x0000 }, + { 0x01ec, 0x0000 }, + { 0x01ed, 0x0000 }, + { 0x01ee, 0x0000 }, + { 0x01ef, 0x0000 }, + { 0x01f0, 0x0000 }, + { 0x01f1, 0x0000 }, + { 0x01f2, 0x0000 }, + { 0x01f3, 0x0000 }, + { 0x01f4, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x0000 }, + { 0x0202, 0x0000 }, + { 0x0203, 0x0000 }, + { 0x0204, 0x0000 }, + { 0x0205, 0x0000 }, + { 0x0206, 0x0000 }, + { 0x0207, 0x0000 }, + { 0x0208, 0x0000 }, + { 0x0210, 0x60b1 }, + { 0x0211, 0xa000 }, + { 0x0212, 0x024c }, + { 0x0213, 0xf7ff }, + { 0x0214, 0x024c }, + { 0x0215, 0x0102 }, + { 0x0216, 0x00a3 }, + { 0x0217, 0x0048 }, + { 0x0218, 0x92c0 }, + { 0x0219, 0x0000 }, + { 0x021a, 0x00c8 }, + { 0x021b, 0x0020 }, + { 0x02fa, 0x0000 }, + { 0x02fb, 0x0000 }, + { 0x02fc, 0x0000 }, + { 0x02ff, 0x0110 }, + { 0x0300, 0x001f }, + { 0x0301, 0x032c }, + { 0x0302, 0x5f21 }, + { 0x0303, 0x4000 }, + { 0x0304, 0x4000 }, + { 0x0305, 0x06d5 }, + { 0x0306, 0x8000 }, + { 0x0307, 0x0700 }, + { 0x0310, 0x4560 }, + { 0x0311, 0xa4a8 }, + { 0x0312, 0x7418 }, + { 0x0313, 0x0000 }, + { 0x0314, 0x0006 }, + { 0x0315, 0xffff }, + { 0x0316, 0xc400 }, + { 0x0317, 0x0000 }, + { 0x0330, 0x00a6 }, + { 0x0331, 0x04c3 }, + { 0x0332, 0x27c8 }, + { 0x0333, 0xbf50 }, + { 0x0334, 0x0045 }, + { 0x0335, 0x0007 }, + { 0x0336, 0x7418 }, + { 0x0337, 0x0501 }, + { 0x0338, 0x0000 }, + { 0x0339, 0x0010 }, + { 0x033a, 0x1010 }, + { 0x03c0, 0x7e00 }, + { 0x03c1, 0x8000 }, + { 0x03c2, 0x8000 }, + { 0x03c3, 0x8000 }, + { 0x03c4, 0x8000 }, + { 0x03c5, 0x8000 }, + { 0x03c6, 0x8000 }, + { 0x03c7, 0x8000 }, + { 0x03c8, 0x8000 }, + { 0x03c9, 0x8000 }, + { 0x03ca, 0x8000 }, + { 0x03cb, 0x8000 }, + { 0x03cc, 0x8000 }, + { 0x03d0, 0x0000 }, + { 0x03d1, 0x0000 }, + { 0x03d2, 0x0000 }, + { 0x03d3, 0x0000 }, + { 0x03d4, 0x2000 }, + { 0x03d5, 0x2000 }, + { 0x03d6, 0x0000 }, + { 0x03d7, 0x0000 }, + { 0x03d8, 0x2000 }, + { 0x03d9, 0x2000 }, + { 0x03da, 0x2000 }, + { 0x03db, 0x2000 }, + { 0x03dc, 0x0000 }, + { 0x03dd, 0x0000 }, + { 0x03de, 0x0000 }, + { 0x03df, 0x2000 }, + { 0x03e0, 0x0000 }, + { 0x03e1, 0x0000 }, + { 0x03e2, 0x0000 }, + { 0x03e3, 0x0000 }, + { 0x03e4, 0x0000 }, + { 0x03e5, 0x0000 }, + { 0x03e6, 0x0000 }, + { 0x03e7, 0x0000 }, + { 0x03e8, 0x0000 }, + { 0x03e9, 0x0000 }, + { 0x03ea, 0x0000 }, + { 0x03eb, 0x0000 }, + { 0x03ec, 0x0000 }, + { 0x03ed, 0x0000 }, + { 0x03ee, 0x0000 }, + { 0x03ef, 0x0000 }, + { 0x03f0, 0x0800 }, + { 0x03f1, 0x0800 }, + { 0x03f2, 0x0800 }, + { 0x03f3, 0x0800 }, + { 0x03fe, 0x0000 }, + { 0x03ff, 0x0000 }, + { 0x07f0, 0x0000 }, + { 0x07fa, 0x0000 }, +}; + +static const struct reg_default rt5663_reg[] = { + { 0x0000, 0x0000 }, + { 0x0002, 0x0008 }, + { 0x0005, 0x1000 }, + { 0x0006, 0x1000 }, + { 0x000a, 0x0000 }, + { 0x0010, 0x000f }, + { 0x0015, 0x42c1 }, + { 0x0016, 0x0000 }, + { 0x0018, 0x000b }, + { 0x0019, 0xafaf }, + { 0x001c, 0x2f2f }, + { 0x001f, 0x0000 }, + { 0x0022, 0x5757 }, + { 0x0023, 0x0039 }, + { 0x0026, 0xc0c0 }, + { 0x0029, 0x8080 }, + { 0x002a, 0xa0a0 }, + { 0x002c, 0x000c }, + { 0x002d, 0x0000 }, + { 0x0040, 0x0808 }, + { 0x0061, 0x0000 }, + { 0x0062, 0x0000 }, + { 0x0063, 0x003e }, + { 0x0064, 0x0000 }, + { 0x0065, 0x0000 }, + { 0x0066, 0x0000 }, + { 0x006b, 0x0000 }, + { 0x006e, 0x0000 }, + { 0x006f, 0x0000 }, + { 0x0070, 0x8020 }, + { 0x0073, 0x1000 }, + { 0x0074, 0xe400 }, + { 0x0075, 0x0002 }, + { 0x0076, 0x0001 }, + { 0x0077, 0x00f0 }, + { 0x0078, 0x0000 }, + { 0x0079, 0x0000 }, + { 0x007a, 0x0123 }, + { 0x007b, 0x8003 }, + { 0x0080, 0x0000 }, + { 0x0081, 0x0000 }, + { 0x0082, 0x0000 }, + { 0x0083, 0x0000 }, + { 0x0084, 0x0000 }, + { 0x0086, 0x0008 }, + { 0x0087, 0x0000 }, + { 0x008a, 0x0000 }, + { 0x008b, 0x0000 }, + { 0x008c, 0x0003 }, + { 0x008e, 0x0004 }, + { 0x008f, 0x1000 }, + { 0x0090, 0x0646 }, + { 0x0091, 0x0e3e }, + { 0x0092, 0x1071 }, + { 0x0093, 0x0000 }, + { 0x0094, 0x0080 }, + { 0x0097, 0x0000 }, + { 0x0098, 0x0000 }, + { 0x009a, 0x0000 }, + { 0x009f, 0x0000 }, + { 0x00ae, 0x2000 }, + { 0x00af, 0x0000 }, + { 0x00b6, 0x0000 }, + { 0x00b7, 0x0000 }, + { 0x00b8, 0x0000 }, + { 0x00ba, 0x0000 }, + { 0x00bb, 0x0000 }, + { 0x00be, 0x0000 }, + { 0x00bf, 0x0000 }, + { 0x00c0, 0x0000 }, + { 0x00c1, 0x0000 }, + { 0x00c5, 0x0000 }, + { 0x00cb, 0xa02f }, + { 0x00cc, 0x0000 }, + { 0x00cd, 0x0e02 }, + { 0x00d9, 0x08f9 }, + { 0x00db, 0x0008 }, + { 0x00dc, 0x00c0 }, + { 0x00dd, 0x6724 }, + { 0x00de, 0x3131 }, + { 0x00df, 0x0008 }, + { 0x00e0, 0x4000 }, + { 0x00e1, 0x3131 }, + { 0x00e2, 0x0043 }, + { 0x00e4, 0x400b }, + { 0x00e5, 0x8031 }, + { 0x00e6, 0x3080 }, + { 0x00e7, 0x4100 }, + { 0x00e8, 0x1400 }, + { 0x00e9, 0xe00a }, + { 0x00ea, 0x0404 }, + { 0x00eb, 0x0404 }, + { 0x00ec, 0xb320 }, + { 0x00ed, 0x0000 }, + { 0x00f4, 0x0000 }, + { 0x00f6, 0x0000 }, + { 0x00f8, 0x0000 }, + { 0x00fa, 0x8000 }, + { 0x00fd, 0x0001 }, + { 0x00fe, 0x10ec }, + { 0x00ff, 0x6406 }, + { 0x0100, 0xa0a0 }, + { 0x0108, 0x4444 }, + { 0x0109, 0x4444 }, + { 0x010a, 0xaaaa }, + { 0x010b, 0x00a0 }, + { 0x010c, 0x8aaa }, + { 0x010d, 0xaaaa }, + { 0x010e, 0x2aaa }, + { 0x010f, 0x002a }, + { 0x0110, 0xa0a4 }, + { 0x0111, 0x4602 }, + { 0x0112, 0x0101 }, + { 0x0113, 0x2000 }, + { 0x0114, 0x0000 }, + { 0x0116, 0x0000 }, + { 0x0117, 0x0f00 }, + { 0x0118, 0x0006 }, + { 0x0125, 0x2224 }, + { 0x0126, 0x5550 }, + { 0x0127, 0x0400 }, + { 0x0128, 0x7711 }, + { 0x0132, 0x0004 }, + { 0x0137, 0x5441 }, + { 0x0139, 0x79a1 }, + { 0x013a, 0x30c0 }, + { 0x013b, 0x2000 }, + { 0x013c, 0x2005 }, + { 0x013d, 0x30c0 }, + { 0x013e, 0x0000 }, + { 0x0140, 0x3700 }, + { 0x0141, 0x1f00 }, + { 0x0144, 0x0000 }, + { 0x0145, 0x0002 }, + { 0x0146, 0x0000 }, + { 0x0160, 0x0e80 }, + { 0x0161, 0x0020 }, + { 0x0162, 0x0080 }, + { 0x0163, 0x0800 }, + { 0x0164, 0x0000 }, + { 0x0165, 0x0000 }, + { 0x0166, 0x0000 }, + { 0x0167, 0x1417 }, + { 0x0168, 0x0017 }, + { 0x0169, 0x0017 }, + { 0x0180, 0x2000 }, + { 0x0181, 0x0000 }, + { 0x0182, 0x0000 }, + { 0x0183, 0x2000 }, + { 0x0184, 0x0000 }, + { 0x0185, 0x0000 }, + { 0x01b0, 0x4b30 }, + { 0x01b1, 0x0000 }, + { 0x01b2, 0xd870 }, + { 0x01b3, 0x0000 }, + { 0x01b4, 0x0030 }, + { 0x01b5, 0x5757 }, + { 0x01b6, 0x5757 }, + { 0x01b7, 0x5757 }, + { 0x01b8, 0x5757 }, + { 0x01c0, 0x433d }, + { 0x01c1, 0x0540 }, + { 0x01c2, 0x0000 }, + { 0x01c3, 0x0000 }, + { 0x01c4, 0x0000 }, + { 0x01c5, 0x0009 }, + { 0x01c6, 0x0018 }, + { 0x01c7, 0x002a }, + { 0x01c8, 0x004c }, + { 0x01c9, 0x0097 }, + { 0x01ca, 0x01c3 }, + { 0x01cb, 0x03e9 }, + { 0x01cc, 0x1389 }, + { 0x01cd, 0xc351 }, + { 0x01ce, 0x0000 }, + { 0x01cf, 0x0000 }, + { 0x01d0, 0x0000 }, + { 0x01d1, 0x0000 }, + { 0x01d2, 0x0000 }, + { 0x01d3, 0x003c }, + { 0x01d4, 0x5757 }, + { 0x01d5, 0x5757 }, + { 0x01d6, 0x5757 }, + { 0x01d7, 0x5757 }, + { 0x01d8, 0x5757 }, + { 0x01d9, 0x5757 }, + { 0x01da, 0x0000 }, + { 0x01db, 0x0000 }, + { 0x01dd, 0x0009 }, + { 0x01de, 0x7f00 }, + { 0x01df, 0x00c8 }, + { 0x01e0, 0x0691 }, + { 0x01e1, 0x0000 }, + { 0x01e2, 0x0000 }, + { 0x01e3, 0x0000 }, + { 0x01e4, 0x0000 }, + { 0x01e5, 0x0040 }, + { 0x01e6, 0x0000 }, + { 0x01e7, 0x0000 }, + { 0x01e8, 0x0000 }, + { 0x01ea, 0x0000 }, + { 0x01eb, 0x0000 }, + { 0x01ec, 0x0000 }, + { 0x01ed, 0x0000 }, + { 0x01ee, 0x0000 }, + { 0x01ef, 0x0000 }, + { 0x01f0, 0x0000 }, + { 0x01f1, 0x0000 }, + { 0x01f2, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x2244 }, + { 0x0202, 0xaaaa }, + { 0x0250, 0x8010 }, + { 0x0251, 0x0000 }, + { 0x0252, 0x028a }, + { 0x02fa, 0x0000 }, + { 0x02fb, 0x0000 }, + { 0x02fc, 0x0000 }, + { 0x0300, 0x0000 }, + { 0x03d0, 0x0000 }, + { 0x03d1, 0x0000 }, + { 0x03d2, 0x0000 }, + { 0x03d3, 0x0000 }, + { 0x03d4, 0x2000 }, + { 0x03d5, 0x2000 }, + { 0x03d6, 0x0000 }, + { 0x03d7, 0x0000 }, + { 0x03d8, 0x2000 }, + { 0x03d9, 0x2000 }, + { 0x03da, 0x2000 }, + { 0x03db, 0x2000 }, + { 0x03dc, 0x0000 }, + { 0x03dd, 0x0000 }, + { 0x03de, 0x0000 }, + { 0x03df, 0x2000 }, + { 0x03e0, 0x0000 }, + { 0x03e1, 0x0000 }, + { 0x03e2, 0x0000 }, + { 0x03e3, 0x0000 }, + { 0x03e4, 0x0000 }, + { 0x03e5, 0x0000 }, + { 0x03e6, 0x0000 }, + { 0x03e7, 0x0000 }, + { 0x03e8, 0x0000 }, + { 0x03e9, 0x0000 }, + { 0x03ea, 0x0000 }, + { 0x03eb, 0x0000 }, + { 0x03ec, 0x0000 }, + { 0x03ed, 0x0000 }, + { 0x03ee, 0x0000 }, + { 0x03ef, 0x0000 }, + { 0x03f0, 0x0800 }, + { 0x03f1, 0x0800 }, + { 0x03f2, 0x0800 }, + { 0x03f3, 0x0800 }, +}; + +static bool rt5663_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5663_RESET: + case RT5663_SIL_DET_CTL: + case RT5663_HP_IMP_GAIN_2: + case RT5663_AD_DA_MIXER: + case RT5663_FRAC_DIV_2: + case RT5663_MICBIAS_1: + case RT5663_ASRC_11_2: + case RT5663_ADC_EQ_1: + case RT5663_INT_ST_1: + case RT5663_INT_ST_2: + case RT5663_GPIO_STA: + case RT5663_SIN_GEN_1: + case RT5663_IL_CMD_1: + case RT5663_IL_CMD_5: + case RT5663_IL_CMD_PWRSAV1: + case RT5663_EM_JACK_TYPE_1: + case RT5663_EM_JACK_TYPE_2: + case RT5663_EM_JACK_TYPE_3: + case RT5663_JD_CTRL2: + case RT5663_VENDOR_ID: + case RT5663_VENDOR_ID_1: + case RT5663_VENDOR_ID_2: + case RT5663_PLL_INT_REG: + case RT5663_SOFT_RAMP: + case RT5663_STO_DRE_1: + case RT5663_STO_DRE_5: + case RT5663_STO_DRE_6: + case RT5663_STO_DRE_7: + case RT5663_MIC_DECRO_1: + case RT5663_MIC_DECRO_4: + case RT5663_HP_IMP_SEN_1: + case RT5663_HP_IMP_SEN_3: + case RT5663_HP_IMP_SEN_4: + case RT5663_HP_IMP_SEN_5: + case RT5663_HP_CALIB_1_1: + case RT5663_HP_CALIB_9: + case RT5663_HP_CALIB_ST1: + case RT5663_HP_CALIB_ST2: + case RT5663_HP_CALIB_ST3: + case RT5663_HP_CALIB_ST4: + case RT5663_HP_CALIB_ST5: + case RT5663_HP_CALIB_ST6: + case RT5663_HP_CALIB_ST7: + case RT5663_HP_CALIB_ST8: + case RT5663_HP_CALIB_ST9: + case RT5663_ANA_JD: + return true; + default: + return false; + } +} + +static bool rt5663_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5663_RESET: + case RT5663_HP_OUT_EN: + case RT5663_HP_LCH_DRE: + case RT5663_HP_RCH_DRE: + case RT5663_CALIB_BST: + case RT5663_RECMIX: + case RT5663_SIL_DET_CTL: + case RT5663_PWR_SAV_SILDET: + case RT5663_SIDETONE_CTL: + case RT5663_STO1_DAC_DIG_VOL: + case RT5663_STO1_ADC_DIG_VOL: + case RT5663_STO1_BOOST: + case RT5663_HP_IMP_GAIN_1: + case RT5663_HP_IMP_GAIN_2: + case RT5663_STO1_ADC_MIXER: + case RT5663_AD_DA_MIXER: + case RT5663_STO_DAC_MIXER: + case RT5663_DIG_SIDE_MIXER: + case RT5663_BYPASS_STO_DAC: + case RT5663_CALIB_REC_MIX: + case RT5663_PWR_DIG_1: + case RT5663_PWR_DIG_2: + case RT5663_PWR_ANLG_1: + case RT5663_PWR_ANLG_2: + case RT5663_PWR_ANLG_3: + case RT5663_PWR_MIXER: + case RT5663_SIG_CLK_DET: + case RT5663_PRE_DIV_GATING_1: + case RT5663_PRE_DIV_GATING_2: + case RT5663_I2S1_SDP: + case RT5663_ADDA_CLK_1: + case RT5663_ADDA_RST: + case RT5663_FRAC_DIV_1: + case RT5663_FRAC_DIV_2: + case RT5663_TDM_1: + case RT5663_TDM_2: + case RT5663_TDM_3: + case RT5663_TDM_4: + case RT5663_TDM_5: + case RT5663_GLB_CLK: + case RT5663_PLL_1: + case RT5663_PLL_2: + case RT5663_ASRC_1: + case RT5663_ASRC_2: + case RT5663_ASRC_4: + case RT5663_DUMMY_REG: + case RT5663_ASRC_8: + case RT5663_ASRC_9: + case RT5663_ASRC_11: + case RT5663_DEPOP_1: + case RT5663_DEPOP_2: + case RT5663_DEPOP_3: + case RT5663_HP_CHARGE_PUMP_1: + case RT5663_HP_CHARGE_PUMP_2: + case RT5663_MICBIAS_1: + case RT5663_RC_CLK: + case RT5663_ASRC_11_2: + case RT5663_DUMMY_REG_2: + case RT5663_REC_PATH_GAIN: + case RT5663_AUTO_1MRC_CLK: + case RT5663_ADC_EQ_1: + case RT5663_ADC_EQ_2: + case RT5663_IRQ_1: + case RT5663_IRQ_2: + case RT5663_IRQ_3: + case RT5663_IRQ_4: + case RT5663_IRQ_5: + case RT5663_INT_ST_1: + case RT5663_INT_ST_2: + case RT5663_GPIO_1: + case RT5663_GPIO_2: + case RT5663_GPIO_STA: + case RT5663_SIN_GEN_1: + case RT5663_SIN_GEN_2: + case RT5663_SIN_GEN_3: + case RT5663_SOF_VOL_ZC1: + case RT5663_IL_CMD_1: + case RT5663_IL_CMD_2: + case RT5663_IL_CMD_3: + case RT5663_IL_CMD_4: + case RT5663_IL_CMD_5: + case RT5663_IL_CMD_6: + case RT5663_IL_CMD_7: + case RT5663_IL_CMD_8: + case RT5663_IL_CMD_PWRSAV1: + case RT5663_IL_CMD_PWRSAV2: + case RT5663_EM_JACK_TYPE_1: + case RT5663_EM_JACK_TYPE_2: + case RT5663_EM_JACK_TYPE_3: + case RT5663_EM_JACK_TYPE_4: + case RT5663_EM_JACK_TYPE_5: + case RT5663_EM_JACK_TYPE_6: + case RT5663_STO1_HPF_ADJ1: + case RT5663_STO1_HPF_ADJ2: + case RT5663_FAST_OFF_MICBIAS: + case RT5663_JD_CTRL1: + case RT5663_JD_CTRL2: + case RT5663_DIG_MISC: + case RT5663_VENDOR_ID: + case RT5663_VENDOR_ID_1: + case RT5663_VENDOR_ID_2: + case RT5663_DIG_VOL_ZCD: + case RT5663_ANA_BIAS_CUR_1: + case RT5663_ANA_BIAS_CUR_2: + case RT5663_ANA_BIAS_CUR_3: + case RT5663_ANA_BIAS_CUR_4: + case RT5663_ANA_BIAS_CUR_5: + case RT5663_ANA_BIAS_CUR_6: + case RT5663_BIAS_CUR_5: + case RT5663_BIAS_CUR_6: + case RT5663_BIAS_CUR_7: + case RT5663_BIAS_CUR_8: + case RT5663_DACREF_LDO: + case RT5663_DUMMY_REG_3: + case RT5663_BIAS_CUR_9: + case RT5663_DUMMY_REG_4: + case RT5663_VREFADJ_OP: + case RT5663_VREF_RECMIX: + case RT5663_CHARGE_PUMP_1: + case RT5663_CHARGE_PUMP_1_2: + case RT5663_CHARGE_PUMP_1_3: + case RT5663_CHARGE_PUMP_2: + case RT5663_DIG_IN_PIN1: + case RT5663_PAD_DRV_CTL: + case RT5663_PLL_INT_REG: + case RT5663_CHOP_DAC_L: + case RT5663_CHOP_ADC: + case RT5663_CALIB_ADC: + case RT5663_CHOP_DAC_R: + case RT5663_DUMMY_CTL_DACLR: + case RT5663_DUMMY_REG_5: + case RT5663_SOFT_RAMP: + case RT5663_TEST_MODE_1: + case RT5663_TEST_MODE_2: + case RT5663_TEST_MODE_3: + case RT5663_STO_DRE_1: + case RT5663_STO_DRE_2: + case RT5663_STO_DRE_3: + case RT5663_STO_DRE_4: + case RT5663_STO_DRE_5: + case RT5663_STO_DRE_6: + case RT5663_STO_DRE_7: + case RT5663_STO_DRE_8: + case RT5663_STO_DRE_9: + case RT5663_STO_DRE_10: + case RT5663_MIC_DECRO_1: + case RT5663_MIC_DECRO_2: + case RT5663_MIC_DECRO_3: + case RT5663_MIC_DECRO_4: + case RT5663_MIC_DECRO_5: + case RT5663_MIC_DECRO_6: + case RT5663_HP_DECRO_1: + case RT5663_HP_DECRO_2: + case RT5663_HP_DECRO_3: + case RT5663_HP_DECRO_4: + case RT5663_HP_DECOUP: + case RT5663_HP_IMP_SEN_MAP8: + case RT5663_HP_IMP_SEN_MAP9: + case RT5663_HP_IMP_SEN_MAP10: + case RT5663_HP_IMP_SEN_MAP11: + case RT5663_HP_IMP_SEN_1: + case RT5663_HP_IMP_SEN_2: + case RT5663_HP_IMP_SEN_3: + case RT5663_HP_IMP_SEN_4: + case RT5663_HP_IMP_SEN_5: + case RT5663_HP_IMP_SEN_6: + case RT5663_HP_IMP_SEN_7: + case RT5663_HP_IMP_SEN_8: + case RT5663_HP_IMP_SEN_9: + case RT5663_HP_IMP_SEN_10: + case RT5663_HP_IMP_SEN_11: + case RT5663_HP_IMP_SEN_12: + case RT5663_HP_IMP_SEN_13: + case RT5663_HP_IMP_SEN_14: + case RT5663_HP_IMP_SEN_15: + case RT5663_HP_IMP_SEN_16: + case RT5663_HP_IMP_SEN_17: + case RT5663_HP_IMP_SEN_18: + case RT5663_HP_IMP_SEN_19: + case RT5663_HP_IMPSEN_DIG5: + case RT5663_HP_IMPSEN_MAP1: + case RT5663_HP_IMPSEN_MAP2: + case RT5663_HP_IMPSEN_MAP3: + case RT5663_HP_IMPSEN_MAP4: + case RT5663_HP_IMPSEN_MAP5: + case RT5663_HP_IMPSEN_MAP7: + case RT5663_HP_LOGIC_1: + case RT5663_HP_LOGIC_2: + case RT5663_HP_CALIB_1: + case RT5663_HP_CALIB_1_1: + case RT5663_HP_CALIB_2: + case RT5663_HP_CALIB_3: + case RT5663_HP_CALIB_4: + case RT5663_HP_CALIB_5: + case RT5663_HP_CALIB_5_1: + case RT5663_HP_CALIB_6: + case RT5663_HP_CALIB_7: + case RT5663_HP_CALIB_9: + case RT5663_HP_CALIB_10: + case RT5663_HP_CALIB_11: + case RT5663_HP_CALIB_ST1: + case RT5663_HP_CALIB_ST2: + case RT5663_HP_CALIB_ST3: + case RT5663_HP_CALIB_ST4: + case RT5663_HP_CALIB_ST5: + case RT5663_HP_CALIB_ST6: + case RT5663_HP_CALIB_ST7: + case RT5663_HP_CALIB_ST8: + case RT5663_HP_CALIB_ST9: + case RT5663_HP_AMP_DET: + case RT5663_DUMMY_REG_6: + case RT5663_HP_BIAS: + case RT5663_CBJ_1: + case RT5663_CBJ_2: + case RT5663_CBJ_3: + case RT5663_DUMMY_1: + case RT5663_DUMMY_2: + case RT5663_DUMMY_3: + case RT5663_ANA_JD: + case RT5663_ADC_LCH_LPF1_A1: + case RT5663_ADC_RCH_LPF1_A1: + case RT5663_ADC_LCH_LPF1_H0: + case RT5663_ADC_RCH_LPF1_H0: + case RT5663_ADC_LCH_BPF1_A1: + case RT5663_ADC_RCH_BPF1_A1: + case RT5663_ADC_LCH_BPF1_A2: + case RT5663_ADC_RCH_BPF1_A2: + case RT5663_ADC_LCH_BPF1_H0: + case RT5663_ADC_RCH_BPF1_H0: + case RT5663_ADC_LCH_BPF2_A1: + case RT5663_ADC_RCH_BPF2_A1: + case RT5663_ADC_LCH_BPF2_A2: + case RT5663_ADC_RCH_BPF2_A2: + case RT5663_ADC_LCH_BPF2_H0: + case RT5663_ADC_RCH_BPF2_H0: + case RT5663_ADC_LCH_BPF3_A1: + case RT5663_ADC_RCH_BPF3_A1: + case RT5663_ADC_LCH_BPF3_A2: + case RT5663_ADC_RCH_BPF3_A2: + case RT5663_ADC_LCH_BPF3_H0: + case RT5663_ADC_RCH_BPF3_H0: + case RT5663_ADC_LCH_BPF4_A1: + case RT5663_ADC_RCH_BPF4_A1: + case RT5663_ADC_LCH_BPF4_A2: + case RT5663_ADC_RCH_BPF4_A2: + case RT5663_ADC_LCH_BPF4_H0: + case RT5663_ADC_RCH_BPF4_H0: + case RT5663_ADC_LCH_HPF1_A1: + case RT5663_ADC_RCH_HPF1_A1: + case RT5663_ADC_LCH_HPF1_H0: + case RT5663_ADC_RCH_HPF1_H0: + case RT5663_ADC_EQ_PRE_VOL_L: + case RT5663_ADC_EQ_PRE_VOL_R: + case RT5663_ADC_EQ_POST_VOL_L: + case RT5663_ADC_EQ_POST_VOL_R: + return true; + default: + return false; + } +} + +static bool rt5668_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5663_RESET: + case RT5668_CBJ_TYPE_2: + case RT5668_PDM_OUT_CTL: + case RT5668_PDM_I2C_DATA_CTL1: + case RT5668_PDM_I2C_DATA_CTL4: + case RT5668_ALC_BK_GAIN: + case RT5663_PLL_2: + case RT5663_MICBIAS_1: + case RT5663_ADC_EQ_1: + case RT5663_INT_ST_1: + case RT5668_GPIO_STA: + case RT5663_IL_CMD_1: + case RT5663_IL_CMD_5: + case RT5668_A_JD_CTRL: + case RT5663_JD_CTRL2: + case RT5663_VENDOR_ID: + case RT5663_VENDOR_ID_1: + case RT5663_VENDOR_ID_2: + case RT5663_STO_DRE_1: + case RT5663_STO_DRE_5: + case RT5663_STO_DRE_6: + case RT5663_STO_DRE_7: + case RT5668_MONO_DYNA_6: + case RT5668_STO1_SIL_DET: + case RT5668_MONOL_SIL_DET: + case RT5668_MONOR_SIL_DET: + case RT5668_STO2_DAC_SIL: + case RT5668_MONO_AMP_CAL_ST1: + case RT5668_MONO_AMP_CAL_ST2: + case RT5668_MONO_AMP_CAL_ST3: + case RT5668_MONO_AMP_CAL_ST4: + case RT5663_HP_IMP_SEN_2: + case RT5663_HP_IMP_SEN_3: + case RT5663_HP_IMP_SEN_4: + case RT5663_HP_IMP_SEN_10: + case RT5663_HP_CALIB_1: + case RT5663_HP_CALIB_10: + case RT5663_HP_CALIB_ST1: + case RT5663_HP_CALIB_ST4: + case RT5663_HP_CALIB_ST5: + case RT5663_HP_CALIB_ST6: + case RT5663_HP_CALIB_ST7: + case RT5663_HP_CALIB_ST8: + case RT5663_HP_CALIB_ST9: + case RT5668_HP_CALIB_ST10: + case RT5668_HP_CALIB_ST11: + return true; + default: + return false; + } +} + +static bool rt5668_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5668_LOUT_CTRL: + case RT5668_HP_AMP_2: + case RT5668_MONO_OUT: + case RT5668_MONO_GAIN: + case RT5668_AEC_BST: + case RT5668_IN1_IN2: + case RT5668_IN3_IN4: + case RT5668_INL1_INR1: + case RT5668_CBJ_TYPE_2: + case RT5668_CBJ_TYPE_3: + case RT5668_CBJ_TYPE_4: + case RT5668_CBJ_TYPE_5: + case RT5668_CBJ_TYPE_8: + case RT5668_DAC3_DIG_VOL: + case RT5668_DAC3_CTRL: + case RT5668_MONO_ADC_DIG_VOL: + case RT5668_STO2_ADC_DIG_VOL: + case RT5668_MONO_ADC_BST_GAIN: + case RT5668_STO2_ADC_BST_GAIN: + case RT5668_SIDETONE_CTRL: + case RT5668_MONO1_ADC_MIXER: + case RT5668_STO2_ADC_MIXER: + case RT5668_MONO_DAC_MIXER: + case RT5668_DAC2_SRC_CTRL: + case RT5668_IF_3_4_DATA_CTL: + case RT5668_IF_5_DATA_CTL: + case RT5668_PDM_OUT_CTL: + case RT5668_PDM_I2C_DATA_CTL1: + case RT5668_PDM_I2C_DATA_CTL2: + case RT5668_PDM_I2C_DATA_CTL3: + case RT5668_PDM_I2C_DATA_CTL4: + case RT5668_RECMIX1_NEW: + case RT5668_RECMIX1L_0: + case RT5668_RECMIX1L: + case RT5668_RECMIX1R_0: + case RT5668_RECMIX1R: + case RT5668_RECMIX2_NEW: + case RT5668_RECMIX2_L_2: + case RT5668_RECMIX2_R: + case RT5668_RECMIX2_R_2: + case RT5668_CALIB_REC_LR: + case RT5668_ALC_BK_GAIN: + case RT5668_MONOMIX_GAIN: + case RT5668_MONOMIX_IN_GAIN: + case RT5668_OUT_MIXL_GAIN: + case RT5668_OUT_LMIX_IN_GAIN: + case RT5668_OUT_RMIX_IN_GAIN: + case RT5668_OUT_RMIX_IN_GAIN1: + case RT5668_LOUT_MIXER_CTRL: + case RT5668_PWR_VOL: + case RT5668_ADCDAC_RST: + case RT5668_I2S34_SDP: + case RT5668_I2S5_SDP: + case RT5668_TDM_5: + case RT5668_TDM_6: + case RT5668_TDM_7: + case RT5668_TDM_8: + case RT5668_ASRC_3: + case RT5668_ASRC_6: + case RT5668_ASRC_7: + case RT5668_PLL_TRK_13: + case RT5668_I2S_M_CLK_CTL: + case RT5668_FDIV_I2S34_M_CLK: + case RT5668_FDIV_I2S34_M_CLK2: + case RT5668_FDIV_I2S5_M_CLK: + case RT5668_FDIV_I2S5_M_CLK2: + case RT5668_IRQ_4: + case RT5668_GPIO_3: + case RT5668_GPIO_4: + case RT5668_GPIO_STA: + case RT5668_HP_AMP_DET1: + case RT5668_HP_AMP_DET2: + case RT5668_HP_AMP_DET3: + case RT5668_MID_BD_HP_AMP: + case RT5668_LOW_BD_HP_AMP: + case RT5668_SOF_VOL_ZC2: + case RT5668_ADC_STO2_ADJ1: + case RT5668_ADC_STO2_ADJ2: + case RT5668_A_JD_CTRL: + case RT5668_JD1_TRES_CTRL: + case RT5668_JD2_TRES_CTRL: + case RT5668_JD_CTRL2: + case RT5668_DUM_REG_2: + case RT5668_DUM_REG_3: + case RT5663_VENDOR_ID: + case RT5663_VENDOR_ID_1: + case RT5663_VENDOR_ID_2: + case RT5668_DACADC_DIG_VOL2: + case RT5668_DIG_IN_PIN2: + case RT5668_PAD_DRV_CTL1: + case RT5668_SOF_RAM_DEPOP: + case RT5668_VOL_TEST: + case RT5668_TEST_MODE_3: + case RT5668_TEST_MODE_4: + case RT5663_STO_DRE_9: + case RT5668_MONO_DYNA_1: + case RT5668_MONO_DYNA_2: + case RT5668_MONO_DYNA_3: + case RT5668_MONO_DYNA_4: + case RT5668_MONO_DYNA_5: + case RT5668_MONO_DYNA_6: + case RT5668_STO1_SIL_DET: + case RT5668_MONOL_SIL_DET: + case RT5668_MONOR_SIL_DET: + case RT5668_STO2_DAC_SIL: + case RT5668_PWR_SAV_CTL1: + case RT5668_PWR_SAV_CTL2: + case RT5668_PWR_SAV_CTL3: + case RT5668_PWR_SAV_CTL4: + case RT5668_PWR_SAV_CTL5: + case RT5668_PWR_SAV_CTL6: + case RT5668_MONO_AMP_CAL1: + case RT5668_MONO_AMP_CAL2: + case RT5668_MONO_AMP_CAL3: + case RT5668_MONO_AMP_CAL4: + case RT5668_MONO_AMP_CAL5: + case RT5668_MONO_AMP_CAL6: + case RT5668_MONO_AMP_CAL7: + case RT5668_MONO_AMP_CAL_ST1: + case RT5668_MONO_AMP_CAL_ST2: + case RT5668_MONO_AMP_CAL_ST3: + case RT5668_MONO_AMP_CAL_ST4: + case RT5668_MONO_AMP_CAL_ST5: + case RT5668_HP_IMP_SEN_13: + case RT5668_HP_IMP_SEN_14: + case RT5668_HP_IMP_SEN_6: + case RT5668_HP_IMP_SEN_7: + case RT5668_HP_IMP_SEN_8: + case RT5668_HP_IMP_SEN_9: + case RT5668_HP_IMP_SEN_10: + case RT5668_HP_LOGIC_3: + case RT5668_HP_CALIB_ST10: + case RT5668_HP_CALIB_ST11: + case RT5668_PRO_REG_TBL_4: + case RT5668_PRO_REG_TBL_5: + case RT5668_PRO_REG_TBL_6: + case RT5668_PRO_REG_TBL_7: + case RT5668_PRO_REG_TBL_8: + case RT5668_PRO_REG_TBL_9: + case RT5668_SAR_ADC_INL_1: + case RT5668_SAR_ADC_INL_2: + case RT5668_SAR_ADC_INL_3: + case RT5668_SAR_ADC_INL_4: + case RT5668_SAR_ADC_INL_5: + case RT5668_SAR_ADC_INL_6: + case RT5668_SAR_ADC_INL_7: + case RT5668_SAR_ADC_INL_8: + case RT5668_SAR_ADC_INL_9: + case RT5668_SAR_ADC_INL_10: + case RT5668_SAR_ADC_INL_11: + case RT5668_SAR_ADC_INL_12: + case RT5668_DRC_CTRL_1: + case RT5668_DRC1_CTRL_2: + case RT5668_DRC1_CTRL_3: + case RT5668_DRC1_CTRL_4: + case RT5668_DRC1_CTRL_5: + case RT5668_DRC1_CTRL_6: + case RT5668_DRC1_HD_CTRL_1: + case RT5668_DRC1_HD_CTRL_2: + case RT5668_DRC1_PRI_REG_1: + case RT5668_DRC1_PRI_REG_2: + case RT5668_DRC1_PRI_REG_3: + case RT5668_DRC1_PRI_REG_4: + case RT5668_DRC1_PRI_REG_5: + case RT5668_DRC1_PRI_REG_6: + case RT5668_DRC1_PRI_REG_7: + case RT5668_DRC1_PRI_REG_8: + case RT5668_ALC_PGA_CTL_1: + case RT5668_ALC_PGA_CTL_2: + case RT5668_ALC_PGA_CTL_3: + case RT5668_ALC_PGA_CTL_4: + case RT5668_ALC_PGA_CTL_5: + case RT5668_ALC_PGA_CTL_6: + case RT5668_ALC_PGA_CTL_7: + case RT5668_ALC_PGA_CTL_8: + case RT5668_ALC_PGA_REG_1: + case RT5668_ALC_PGA_REG_2: + case RT5668_ALC_PGA_REG_3: + case RT5668_ADC_EQ_RECOV_1: + case RT5668_ADC_EQ_RECOV_2: + case RT5668_ADC_EQ_RECOV_3: + case RT5668_ADC_EQ_RECOV_4: + case RT5668_ADC_EQ_RECOV_5: + case RT5668_ADC_EQ_RECOV_6: + case RT5668_ADC_EQ_RECOV_7: + case RT5668_ADC_EQ_RECOV_8: + case RT5668_ADC_EQ_RECOV_9: + case RT5668_ADC_EQ_RECOV_10: + case RT5668_ADC_EQ_RECOV_11: + case RT5668_ADC_EQ_RECOV_12: + case RT5668_ADC_EQ_RECOV_13: + case RT5668_VID_HIDDEN: + case RT5668_VID_CUSTOMER: + case RT5668_SCAN_MODE: + case RT5668_I2C_BYPA: + return true; + case RT5663_TDM_1: + case RT5663_DEPOP_3: + case RT5663_ASRC_11_2: + case RT5663_INT_ST_2: + case RT5663_GPIO_STA: + case RT5663_SIN_GEN_1: + case RT5663_SIN_GEN_2: + case RT5663_SIN_GEN_3: + case RT5663_IL_CMD_PWRSAV1: + case RT5663_IL_CMD_PWRSAV2: + case RT5663_EM_JACK_TYPE_1: + case RT5663_EM_JACK_TYPE_2: + case RT5663_EM_JACK_TYPE_3: + case RT5663_EM_JACK_TYPE_4: + case RT5663_FAST_OFF_MICBIAS: + case RT5663_ANA_BIAS_CUR_1: + case RT5663_ANA_BIAS_CUR_2: + case RT5663_BIAS_CUR_9: + case RT5663_DUMMY_REG_4: + case RT5663_VREF_RECMIX: + case RT5663_CHARGE_PUMP_1_2: + case RT5663_CHARGE_PUMP_1_3: + case RT5663_CHARGE_PUMP_2: + case RT5663_CHOP_DAC_R: + case RT5663_DUMMY_CTL_DACLR: + case RT5663_DUMMY_REG_5: + case RT5663_SOFT_RAMP: + case RT5663_TEST_MODE_1: + case RT5663_STO_DRE_10: + case RT5663_MIC_DECRO_1: + case RT5663_MIC_DECRO_2: + case RT5663_MIC_DECRO_3: + case RT5663_MIC_DECRO_4: + case RT5663_MIC_DECRO_5: + case RT5663_MIC_DECRO_6: + case RT5663_HP_DECRO_1: + case RT5663_HP_DECRO_2: + case RT5663_HP_DECRO_3: + case RT5663_HP_DECRO_4: + case RT5663_HP_DECOUP: + case RT5663_HP_IMPSEN_MAP4: + case RT5663_HP_IMPSEN_MAP5: + case RT5663_HP_IMPSEN_MAP7: + case RT5663_HP_CALIB_1: + case RT5663_CBJ_1: + case RT5663_CBJ_2: + case RT5663_CBJ_3: + return false; + default: + return rt5663_readable_register(dev, reg); + } +} + +static const DECLARE_TLV_DB_SCALE(rt5663_hp_vol_tlv, -2400, 150, 0); +static const DECLARE_TLV_DB_SCALE(rt5668_hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(in_bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5663_if1_adc_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5663_if1_adc_enum, RT5663_TDM_2, + RT5663_DATA_SWAP_ADCDAT1_SHIFT, rt5663_if1_adc_data_select); + +static void rt5663_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + if (enable) { + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_EN_4BTN_INL_MASK, RT5668_EN_4BTN_INL_EN); + /* reset in-line command */ + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_RESET); + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_NOR); + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + snd_soc_update_bits(codec, RT5663_IRQ_3, + RT5668_EN_IRQ_INLINE_MASK, + RT5668_EN_IRQ_INLINE_NOR); + break; + case CODEC_TYPE_RT5663: + snd_soc_update_bits(codec, RT5663_IRQ_2, + RT5663_EN_IRQ_INLINE_MASK, + RT5663_EN_IRQ_INLINE_NOR); + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + } else { + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + snd_soc_update_bits(codec, RT5663_IRQ_3, + RT5668_EN_IRQ_INLINE_MASK, + RT5668_EN_IRQ_INLINE_BYP); + break; + case CODEC_TYPE_RT5663: + snd_soc_update_bits(codec, RT5663_IRQ_2, + RT5663_EN_IRQ_INLINE_MASK, + RT5663_EN_IRQ_INLINE_BYP); + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_EN_4BTN_INL_MASK, RT5668_EN_4BTN_INL_DIS); + /* reset in-line command */ + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_RESET); + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_NOR); + } +} + +/** + * rt5668_jack_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5668_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5663_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30}; + + dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert); + if (jack_insert) { + snd_soc_write(codec, RT5668_CBJ_TYPE_2, 0x8040); + snd_soc_write(codec, RT5668_CBJ_TYPE_3, 0x1484); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_update_bits(codec, RT5663_RC_CLK, + RT5668_DIG_1M_CLK_MASK, RT5668_DIG_1M_CLK_EN); + snd_soc_update_bits(codec, RT5663_RECMIX, 0x8, 0x8); + + while (i < 5) { + msleep(sleep_time[i]); + val = snd_soc_read(codec, RT5668_CBJ_TYPE_2) & 0x0003; + if (val == 0x1 || val == 0x2 || val == 0x3) + break; + dev_dbg(codec->dev, "%s: MX-0011 val=%x sleep %d\n", + __func__, val, sleep_time[i]); + i++; + } + dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + switch (val) { + case 1: + case 2: + rt5668->jack_type = SND_JACK_HEADSET; + rt5663_enable_push_button_irq(codec, true); + break; + default: + snd_soc_dapm_disable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + rt5668->jack_type = SND_JACK_HEADPHONE; + break; + } + } else { + snd_soc_update_bits(codec, RT5663_RECMIX, 0x8, 0x0); + + if (rt5668->jack_type == SND_JACK_HEADSET) { + rt5663_enable_push_button_irq(codec, false); + snd_soc_dapm_disable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + } + rt5668->jack_type = 0; + } + + dev_dbg(codec->dev, "jack_type = %d\n", rt5668->jack_type); + return rt5668->jack_type; +} + +/** + * rt5663_jack_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30}; + + dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert); + + if (jack_insert) { + snd_soc_update_bits(codec, RT5663_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, RT5668_DIG_GATE_CTRL_EN); + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5663_SI_HP_MASK | RT5668_OSW_HP_L_MASK | + RT5668_OSW_HP_R_MASK, RT5663_SI_HP_EN | + RT5668_OSW_HP_L_DIS | RT5668_OSW_HP_R_DIS); + snd_soc_update_bits(codec, RT5663_DUMMY_1, + RT5663_EMB_CLK_MASK | RT5663_HPA_CPL_BIAS_MASK | + RT5663_HPA_CPR_BIAS_MASK, RT5663_EMB_CLK_EN | + RT5663_HPA_CPL_BIAS_1 | RT5663_HPA_CPR_BIAS_1); + snd_soc_update_bits(codec, RT5663_CBJ_1, + RT5663_INBUF_CBJ_BST1_MASK | RT5663_CBJ_SENSE_BST1_MASK, + RT5663_INBUF_CBJ_BST1_ON | RT5663_CBJ_SENSE_BST1_L); + snd_soc_update_bits(codec, RT5663_IL_CMD_2, + RT5663_PWR_MIC_DET_MASK, RT5663_PWR_MIC_DET_ON); + /* BST1 power on for JD */ + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5668_PWR_BST1_MASK, RT5668_PWR_BST1_ON); + snd_soc_update_bits(codec, RT5663_EM_JACK_TYPE_1, + RT5663_CBJ_DET_MASK | RT5663_EXT_JD_MASK | + RT5663_POL_EXT_JD_MASK, RT5663_CBJ_DET_EN | + RT5663_EXT_JD_EN | RT5663_POL_EXT_JD_EN); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5668_PWR_MB_MASK | RT5668_LDO1_DVO_MASK | + RT5668_AMP_HP_MASK, RT5668_PWR_MB | + RT5668_LDO1_DVO_0_9V | RT5668_AMP_HP_3X); + snd_soc_update_bits(codec, RT5663_AUTO_1MRC_CLK, + RT5668_IRQ_POW_SAV_MASK, RT5668_IRQ_POW_SAV_EN); + snd_soc_update_bits(codec, RT5663_IRQ_1, + RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN); + while (i < 5) { + msleep(sleep_time[i]); + val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) & + 0x0003; + i++; + if (val == 0x1 || val == 0x2 || val == 0x3) + break; + dev_dbg(codec->dev, "%s: MX-00e7 val=%x sleep %d\n", + __func__, val, sleep_time[i]); + } + dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + switch (val) { + case 1: + case 2: + rt5663->jack_type = SND_JACK_HEADSET; + rt5663_enable_push_button_irq(codec, true); + break; + default: + rt5663->jack_type = SND_JACK_HEADPHONE; + break; + } + } else { + if (rt5663->jack_type == SND_JACK_HEADSET) + rt5663_enable_push_button_irq(codec, false); + rt5663->jack_type = 0; + } + + dev_dbg(codec->dev, "jack_type = %d\n", rt5663->jack_type); + return rt5663->jack_type; +} + +int rt5663_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5663_IL_CMD_5); + dev_dbg(codec->dev, "%s: val=0x%x\n", __func__, val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5663_IL_CMD_5, val); + + return btn_type; +} + +static irqreturn_t rt5663_irq(int irq, void *data) +{ + struct rt5663_priv *rt5663 = data; + + dev_dbg(rt5663->codec->dev, "%s IRQ queue work\n", __func__); + + queue_delayed_work(system_wq, &rt5663->jack_detect_work, + msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +int rt5663_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + rt5663->hs_jack = hs_jack; + + rt5663_irq(0, rt5663); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5663_set_jack_detect); + +static bool rt5663_check_jd_status(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + int val = snd_soc_read(codec, RT5663_INT_ST_1); + + dev_dbg(codec->dev, "%s val=%x\n", __func__, val); + + /* JD1 */ + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + return !(val & 0x2000); + case CODEC_TYPE_RT5663: + return !(val & 0x1000); + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + + return false; +} + +static void rt5663_jack_detect_work(struct work_struct *work) +{ + struct rt5663_priv *rt5663 = + container_of(work, struct rt5663_priv, jack_detect_work.work); + struct snd_soc_codec *codec = rt5663->codec; + int btn_type, report = 0; + + if (!codec) + return; + + if (rt5663_check_jd_status(codec)) { + /* jack in */ + if (rt5663->jack_type == 0) { + /* jack was out, report jack type */ + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + report = rt5668_jack_detect(rt5663->codec, 1); + break; + case CODEC_TYPE_RT5663: + report = rt5663_jack_detect(rt5663->codec, 1); + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + } else { + /* jack is already in, report button event */ + report = SND_JACK_HEADSET; + btn_type = rt5663_button_detect(rt5663->codec); + /** + * rt5663 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5663->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + /* button release or spurious interrput*/ + if (btn_type == 0) + report = rt5663->jack_type; + } + } else { + /* jack out */ + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + report = rt5668_jack_detect(rt5663->codec, 0); + break; + case CODEC_TYPE_RT5663: + report = rt5663_jack_detect(rt5663->codec, 0); + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + } + dev_dbg(codec->dev, "%s jack report: 0x%04x\n", __func__, report); + snd_soc_jack_report(rt5663->hs_jack, report, SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); +} + +static const struct snd_kcontrol_new rt5663_snd_controls[] = { + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC Playback Volume", RT5663_STO1_DAC_DIG_VOL, + RT5668_DAC_L1_VOL_SHIFT + 1, RT5668_DAC_R1_VOL_SHIFT + 1, + 87, 0, dac_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5663_STO1_ADC_DIG_VOL, + RT5668_ADC_L_MUTE_SHIFT, RT5668_ADC_R_MUTE_SHIFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5663_STO1_ADC_DIG_VOL, + RT5668_ADC_L_VOL_SHIFT + 1, RT5668_ADC_R_VOL_SHIFT + 1, + 63, 0, adc_vol_tlv), +}; + +static const struct snd_kcontrol_new rt5668_specific_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_HP_LCH_DRE, + RT5663_HP_RCH_DRE, RT5668_GAIN_HP_SHIFT, 15, 1, + rt5668_hp_vol_tlv), + /* Mic Boost Volume */ + SOC_SINGLE_TLV("IN1 Capture Volume", RT5668_AEC_BST, + RT5668_GAIN_CBJ_SHIFT, 8, 0, in_bst_tlv), +}; + +static const struct snd_kcontrol_new rt5663_specific_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, + RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, + rt5663_hp_vol_tlv), + /* Mic Boost Volume*/ + SOC_SINGLE_TLV("IN1 Capture Volume", RT5663_CBJ_2, + RT5663_GAIN_BST1_SHIFT, 8, 0, in_bst_tlv), + /* Data Swap for Slot0/1 in ADCDAT1 */ + SOC_ENUM("IF1 ADC Data Swap", rt5663_if1_adc_enum), +}; + +static int rt5663_is_sys_clk_from_pll(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + val = snd_soc_read(codec, RT5663_GLB_CLK); + val &= RT5663_SCLK_SRC_MASK; + if (val == RT5663_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int rt5663_is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + switch (w->shift) { + case RT5668_ADC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_3; + shift = RT5668_AD_STO1_TRACK_SHIFT; + break; + case RT5668_DAC_STO1_ASRC_SHIFT: + reg = RT5663_ASRC_2; + shift = RT5668_DA_STO1_TRACK_SHIFT; + break; + default: + return 0; + } + } else { + switch (w->shift) { + case RT5663_ADC_STO1_ASRC_SHIFT: + reg = RT5663_ASRC_2; + shift = RT5663_AD_STO1_TRACK_SHIFT; + break; + case RT5663_DAC_STO1_ASRC_SHIFT: + reg = RT5663_ASRC_2; + shift = RT5663_DA_STO1_TRACK_SHIFT; + break; + default: + return 0; + } + } + + val = (snd_soc_read(codec, reg) >> shift) & 0x7; + + if (val) + return 1; + + return 0; +} + +static int rt5663_i2s_use_asrc(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 rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + int da_asrc_en, ad_asrc_en; + + da_asrc_en = (snd_soc_read(codec, RT5663_ASRC_2) & + RT5663_DA_STO1_TRACK_MASK) ? 1 : 0; + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + ad_asrc_en = (snd_soc_read(codec, RT5668_ASRC_3) & + RT5668_AD_STO1_TRACK_MASK) ? 1 : 0; + break; + case CODEC_TYPE_RT5663: + ad_asrc_en = (snd_soc_read(codec, RT5663_ASRC_2) & + RT5663_AD_STO1_TRACK_MASK) ? 1 : 0; + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + + if (da_asrc_en || ad_asrc_en) + if (rt5663->sysclk > rt5663->lrck * 384) + return 1; + + dev_err(codec->dev, "sysclk < 384 x fs, disable i2s asrc\n"); + + return 0; +} + +/** + * rt5663_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5668 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5663_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + struct rt5663_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc2_mask = 0; + unsigned int asrc2_value = 0; + unsigned int asrc3_mask = 0; + unsigned int asrc3_value = 0; + + switch (clk_src) { + case RT5663_CLK_SEL_SYS: + case RT5663_CLK_SEL_I2S1_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5663_DA_STEREO_FILTER) { + asrc2_mask |= RT5668_DA_STO1_TRACK_MASK; + asrc2_value |= clk_src << RT5668_DA_STO1_TRACK_SHIFT; + } + + if (filter_mask & RT5663_AD_STEREO_FILTER) { + switch (rt5668->codec_type) { + case CODEC_TYPE_RT5668: + asrc3_mask |= RT5668_AD_STO1_TRACK_MASK; + asrc3_value |= clk_src << RT5668_AD_STO1_TRACK_SHIFT; + break; + case CODEC_TYPE_RT5663: + asrc2_mask |= RT5663_AD_STO1_TRACK_MASK; + asrc2_value |= clk_src << RT5663_AD_STO1_TRACK_SHIFT; + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5663_ASRC_2, asrc2_mask, + asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5668_ASRC_3, asrc3_mask, + asrc3_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5663_sel_asrc_clk_src); + +/* Analog Mixer */ +static const struct snd_kcontrol_new rt5668_recmix1l[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5668_RECMIX1L, + RT5668_RECMIX1L_BST2_SHIFT, 1, 1), + SOC_DAPM_SINGLE("BST1 CBJ Switch", RT5668_RECMIX1L, + RT5668_RECMIX1L_BST1_CBJ_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_recmix1r[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5668_RECMIX1R, + RT5668_RECMIX1R_BST2_SHIFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5663_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5663_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L1_SHIFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5663_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L2_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5663_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R1_SHIFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5663_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R2_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5663_adda_l_mix[] = { + SOC_DAPM_SINGLE("ADC L Switch", RT5663_AD_DA_MIXER, + RT5668_M_ADCMIX_L_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC L Switch", RT5663_AD_DA_MIXER, + RT5668_M_DAC1_L_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5663_adda_r_mix[] = { + SOC_DAPM_SINGLE("ADC R Switch", RT5663_AD_DA_MIXER, + RT5668_M_ADCMIX_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5663_AD_DA_MIXER, + RT5668_M_DAC1_R_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, + RT5668_M_DAC_L1_STO_L_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, + RT5668_M_DAC_R1_STO_L_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5663_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, + RT5668_M_DAC_L1_STO_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, + RT5668_M_DAC_R1_STO_R_SHIFT, 1, 1), +}; + +/* Out Switch */ +static const struct snd_kcontrol_new rt5668_hpo_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5668_HP_AMP_2, + RT5668_EN_DAC_HPO_SHIFT, 1, 0); + +/* Stereo ADC source */ +static const char * const rt5668_sto1_adc_src[] = { + "ADC L", "ADC R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5668_sto1_adcl_enum, RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_L_SRC_SHIFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcl_mux = + SOC_DAPM_ENUM("STO1 ADC L Mux", rt5668_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5668_sto1_adcr_enum, RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_R_SRC_SHIFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcr_mux = + SOC_DAPM_ENUM("STO1 ADC R Mux", rt5668_sto1_adcr_enum); + +/* RT5663: Analog DACL1 input source */ +static const char * const rt5663_alg_dacl_src[] = { + "DAC L", "STO DAC MIXL" +}; + +static SOC_ENUM_SINGLE_DECL(rt5663_alg_dacl_enum, RT5663_BYPASS_STO_DAC, + RT5663_DACL1_SRC_SHIFT, rt5663_alg_dacl_src); + +static const struct snd_kcontrol_new rt5663_alg_dacl_mux = + SOC_DAPM_ENUM("DAC L Mux", rt5663_alg_dacl_enum); + +/* RT5663: Analog DACR1 input source */ +static const char * const rt5663_alg_dacr_src[] = { + "DAC R", "STO DAC MIXR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5663_alg_dacr_enum, RT5663_BYPASS_STO_DAC, + RT5663_DACR1_SRC_SHIFT, rt5663_alg_dacr_src); + +static const struct snd_kcontrol_new rt5663_alg_dacr_mux = + SOC_DAPM_ENUM("DAC R Mux", rt5663_alg_dacr_enum); + +static int rt5663_hp_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); + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5668_SEL_PM_HP_SHIFT, RT5668_SEL_PM_HP_HIGH); + snd_soc_update_bits(codec, RT5663_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, + RT5668_HP_SIG_SRC1_SILENCE); + } else { + snd_soc_write(codec, RT5663_DEPOP_2, 0x3003); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x000b, + 0x000b); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, + 0x0030); + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5668_OVCD_HP_MASK, RT5668_OVCD_HP_DIS); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_2, 0x1371); + snd_soc_write(codec, RT5663_HP_BIAS, 0xabba); + snd_soc_write(codec, RT5663_CHARGE_PUMP_1, 0x2224); + snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7766); + snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa); + snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, + 0x3000); + } + break; + + case SND_SOC_DAPM_PRE_PMD: + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5663_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, + RT5668_HP_SIG_SRC1_REG); + } else { + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0x0); + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5668_OVCD_HP_MASK, RT5668_OVCD_HP_EN); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0x0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x000b, + 0x000b); + } + break; + + default: + return 0; + } + + return 0; +} + +static int rt5668_bst2_power(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, RT5663_PWR_ANLG_2, + RT5668_PWR_BST2_MASK | RT5668_PWR_BST2_OP_MASK, + RT5668_PWR_BST2 | RT5668_PWR_BST2_OP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5668_PWR_BST2_MASK | RT5668_PWR_BST2_OP_MASK, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5663_pre_div_power(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_write(codec, RT5663_PRE_DIV_GATING_1, 0xff00); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0xfffc); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0x0000); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0x0000); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5663_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5663_PWR_ANLG_3, RT5668_PWR_PLL_SHIFT, 0, + NULL, 0), + + /* micbias */ + SND_SOC_DAPM_MICBIAS("MICBIAS1", RT5663_PWR_ANLG_2, + RT5668_PWR_MB1_SHIFT, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS2", RT5663_PWR_ANLG_2, + RT5668_PWR_MB2_SHIFT, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + + /* REC Mixer Power */ + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5663_PWR_ANLG_2, + RT5668_PWR_RECMIX1_SHIFT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC L Power", RT5663_PWR_DIG_1, + RT5668_PWR_ADC_L1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", RT5663_CHOP_ADC, + RT5668_CKGEN_ADCC_SHIFT, 0, NULL, 0), + + /* ADC Mixer */ + SND_SOC_DAPM_MIXER("STO1 ADC MIXL", SND_SOC_NOPM, + 0, 0, rt5663_sto1_adc_l_mix, + ARRAY_SIZE(rt5663_sto1_adc_l_mix)), + + /* ADC Filter Power */ + SND_SOC_DAPM_SUPPLY("STO1 ADC Filter", RT5663_PWR_DIG_2, + RT5668_PWR_ADC_S1F_SHIFT, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S", RT5663_PWR_DIG_1, RT5668_PWR_I2S1_SHIFT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("IF DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFTX", "AIF Capture", 0, SND_SOC_NOPM, 0, 0), + + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("ADDA MIXL", SND_SOC_NOPM, 0, 0, rt5663_adda_l_mix, + ARRAY_SIZE(rt5663_adda_l_mix)), + SND_SOC_DAPM_MIXER("ADDA MIXR", SND_SOC_NOPM, 0, 0, rt5663_adda_r_mix, + ARRAY_SIZE(rt5663_adda_r_mix)), + SND_SOC_DAPM_PGA("DAC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("STO1 DAC Filter", RT5663_PWR_DIG_2, + RT5668_PWR_DAC_S1F_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("STO1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5663_sto1_dac_l_mix, ARRAY_SIZE(rt5663_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("STO1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5663_sto1_dac_r_mix, ARRAY_SIZE(rt5663_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("STO1 DAC L Power", RT5663_PWR_DIG_1, + RT5668_PWR_DAC_L1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("STO1 DAC R Power", RT5663_PWR_DIG_1, + RT5668_PWR_DAC_R1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Headphone*/ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5663_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_widget rt5668_specific_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5663_PWR_ANLG_3, + RT5668_PWR_LDO2_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5668_PWR_VOL, + RT5668_PWR_MIC_DET_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO DAC", RT5663_PWR_DIG_1, + RT5668_PWR_LDO_DACREF_SHIFT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY("I2S ASRC", RT5663_ASRC_1, + RT5668_I2S1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC ASRC", RT5663_ASRC_1, + RT5668_DAC_STO1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC ASRC", RT5663_ASRC_1, + RT5668_ADC_STO1_ASRC_SHIFT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5663_PWR_ANLG_3, + RT5668_PWR_CBJ_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BST2 Power", SND_SOC_NOPM, 0, 0, + rt5668_bst2_power, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5668_recmix1l, + ARRAY_SIZE(rt5668_recmix1l)), + SND_SOC_DAPM_MIXER("RECMIX1R", SND_SOC_NOPM, 0, 0, rt5668_recmix1r, + ARRAY_SIZE(rt5668_recmix1r)), + SND_SOC_DAPM_SUPPLY("RECMIX1R Power", RT5663_PWR_ANLG_2, + RT5668_PWR_RECMIX2_SHIFT, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC R Power", RT5663_PWR_DIG_1, + RT5668_PWR_ADC_R1_SHIFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_PGA("STO1 ADC L1", RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_L1_SRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC R1", RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_R1_SRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC L2", RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_L2_SRC_SHIFT, 1, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC R2", RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_R2_SRC_SHIFT, 1, NULL, 0), + + SND_SOC_DAPM_MUX("STO1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcl_mux), + SND_SOC_DAPM_MUX("STO1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcr_mux), + + /* ADC Mix */ + SND_SOC_DAPM_MIXER("STO1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5668_sto1_adc_r_mix, ARRAY_SIZE(rt5668_sto1_adc_r_mix)), + + /* Analog DAC Clock */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RT5663_CHOP_DAC_L, + RT5668_CKGEN_DAC1_SHIFT, 0, NULL, 0), + + /* Headphone out */ + SND_SOC_DAPM_SWITCH("HPO Playback", SND_SOC_NOPM, 0, 0, + &rt5668_hpo_switch), +}; + +static const struct snd_soc_dapm_widget rt5663_specific_dapm_widgets[] = { + /* System Clock Pre Divider Gating */ + SND_SOC_DAPM_SUPPLY("Pre Div Power", SND_SOC_NOPM, 0, 0, + rt5663_pre_div_power, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + /* LDO */ + SND_SOC_DAPM_SUPPLY("LDO ADC", RT5663_PWR_DIG_1, + RT5668_PWR_LDO_DACREF_SHIFT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY("I2S ASRC", RT5663_ASRC_1, + RT5663_I2S1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC ASRC", RT5663_ASRC_1, + RT5663_DAC_STO1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC ASRC", RT5663_ASRC_1, + RT5663_ADC_STO1_ASRC_SHIFT, 0, NULL, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* STO ADC */ + SND_SOC_DAPM_PGA("STO1 ADC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC L2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog DAC source */ + SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0, &rt5663_alg_dacl_mux), + SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0, &rt5663_alg_dacr_mux), +}; + +static const struct snd_soc_dapm_route rt5663_dapm_routes[] = { + /* PLL */ + { "I2S", NULL, "PLL", rt5663_is_sys_clk_from_pll }, + + /* ASRC */ + { "STO1 ADC Filter", NULL, "ADC ASRC", rt5663_is_using_asrc }, + { "STO1 DAC Filter", NULL, "DAC ASRC", rt5663_is_using_asrc }, + { "I2S", NULL, "I2S ASRC", rt5663_i2s_use_asrc }, + + { "ADC L", NULL, "ADC L Power" }, + { "ADC L", NULL, "ADC Clock" }, + + { "STO1 ADC L2", NULL, "STO1 DAC MIXL" }, + + { "STO1 ADC MIXL", "ADC1 Switch", "STO1 ADC L1" }, + { "STO1 ADC MIXL", "ADC2 Switch", "STO1 ADC L2" }, + { "STO1 ADC MIXL", NULL, "STO1 ADC Filter" }, + + { "IF1 ADC1", NULL, "STO1 ADC MIXL" }, + { "IF ADC", NULL, "IF1 ADC1" }, + { "AIFTX", NULL, "IF ADC" }, + { "AIFTX", NULL, "I2S" }, + + { "AIFRX", NULL, "I2S" }, + { "IF DAC", NULL, "AIFRX" }, + { "IF1 DAC1 L", NULL, "IF DAC" }, + { "IF1 DAC1 R", NULL, "IF DAC" }, + + { "ADDA MIXL", "ADC L Switch", "STO1 ADC MIXL" }, + { "ADDA MIXL", "DAC L Switch", "IF1 DAC1 L" }, + { "ADDA MIXL", NULL, "STO1 DAC Filter" }, + { "ADDA MIXL", NULL, "STO1 DAC L Power" }, + { "ADDA MIXR", "DAC R Switch", "IF1 DAC1 R" }, + { "ADDA MIXR", NULL, "STO1 DAC Filter" }, + { "ADDA MIXR", NULL, "STO1 DAC R Power" }, + + { "DAC L1", NULL, "ADDA MIXL" }, + { "DAC R1", NULL, "ADDA MIXR" }, + + { "STO1 DAC MIXL", "DAC L Switch", "DAC L1" }, + { "STO1 DAC MIXL", "DAC R Switch", "DAC R1" }, + { "STO1 DAC MIXL", NULL, "STO1 DAC L Power" }, + { "STO1 DAC MIXL", NULL, "STO1 DAC Filter" }, + { "STO1 DAC MIXR", "DAC R Switch", "DAC R1" }, + { "STO1 DAC MIXR", "DAC L Switch", "DAC L1" }, + { "STO1 DAC MIXR", NULL, "STO1 DAC R Power" }, + { "STO1 DAC MIXR", NULL, "STO1 DAC Filter" }, + + { "HP Amp", NULL, "DAC L" }, + { "HP Amp", NULL, "DAC R" }, +}; + +static const struct snd_soc_dapm_route rt5668_specific_dapm_routes[] = { + { "MICBIAS1", NULL, "LDO2" }, + { "MICBIAS2", NULL, "LDO2" }, + + { "BST1 CBJ", NULL, "IN1P" }, + { "BST1 CBJ", NULL, "IN1N" }, + { "BST1 CBJ", NULL, "CBJ Power" }, + + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + { "BST2", NULL, "BST2 Power" }, + + { "RECMIX1L", "BST2 Switch", "BST2" }, + { "RECMIX1L", "BST1 CBJ Switch", "BST1 CBJ" }, + { "RECMIX1L", NULL, "RECMIX1L Power" }, + { "RECMIX1R", "BST2 Switch", "BST2" }, + { "RECMIX1R", NULL, "RECMIX1R Power" }, + + { "ADC L", NULL, "RECMIX1L" }, + { "ADC R", NULL, "RECMIX1R" }, + { "ADC R", NULL, "ADC R Power" }, + { "ADC R", NULL, "ADC Clock" }, + + { "STO1 ADC L Mux", "ADC L", "ADC L" }, + { "STO1 ADC L Mux", "ADC R", "ADC R" }, + { "STO1 ADC L1", NULL, "STO1 ADC L Mux" }, + + { "STO1 ADC R Mux", "ADC L", "ADC L" }, + { "STO1 ADC R Mux", "ADC R", "ADC R" }, + { "STO1 ADC R1", NULL, "STO1 ADC R Mux" }, + { "STO1 ADC R2", NULL, "STO1 DAC MIXR" }, + + { "STO1 ADC MIXR", "ADC1 Switch", "STO1 ADC R1" }, + { "STO1 ADC MIXR", "ADC2 Switch", "STO1 ADC R2" }, + { "STO1 ADC MIXR", NULL, "STO1 ADC Filter" }, + + { "IF1 ADC1", NULL, "STO1 ADC MIXR" }, + + { "ADDA MIXR", "ADC R Switch", "STO1 ADC MIXR" }, + + { "DAC L", NULL, "STO1 DAC MIXL" }, + { "DAC L", NULL, "LDO DAC" }, + { "DAC L", NULL, "DAC Clock" }, + { "DAC R", NULL, "STO1 DAC MIXR" }, + { "DAC R", NULL, "LDO DAC" }, + { "DAC R", NULL, "DAC Clock" }, + + { "HPO Playback", "Switch", "HP Amp" }, + { "HPOL", NULL, "HPO Playback" }, + { "HPOR", NULL, "HPO Playback" }, +}; + +static const struct snd_soc_dapm_route rt5663_specific_dapm_routes[] = { + { "I2S", NULL, "Pre Div Power" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "RECMIX1L Power" }, + + { "ADC L", NULL, "BST1" }, + + { "STO1 ADC L1", NULL, "ADC L" }, + + { "DAC L Mux", "DAC L", "DAC L1" }, + { "DAC L Mux", "STO DAC MIXL", "STO1 DAC MIXL" }, + { "DAC R Mux", "DAC R", "DAC R1"}, + { "DAC R Mux", "STO DAC MIXR", "STO1 DAC MIXR" }, + + { "DAC L", NULL, "DAC L Mux" }, + { "DAC R", NULL, "DAC R Mux" }, + + { "HPOL", NULL, "HP Amp" }, + { "HPOR", NULL, "HP Amp" }, +}; + +static int rt5663_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 rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + int pre_div; + + rt5663->lrck = params_rate(params); + + dev_dbg(dai->dev, "bclk is %dHz and sysclk is %dHz\n", + rt5663->lrck, rt5663->sysclk); + + pre_div = rl6231_get_clk_info(rt5663->sysclk, rt5663->lrck); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5663->lrck, dai->id); + return -EINVAL; + } + + dev_dbg(dai->dev, "pre_div is %d for iis %d\n", pre_div, dai->id); + + switch (params_width(params)) { + case 8: + val_len = RT5668_I2S_DL_8; + break; + case 16: + val_len = RT5668_I2S_DL_16; + break; + case 20: + val_len = RT5668_I2S_DL_20; + break; + case 24: + val_len = RT5668_I2S_DL_24; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5663_I2S1_SDP, + RT5668_I2S_DL_MASK, val_len); + + snd_soc_update_bits(codec, RT5663_ADDA_CLK_1, + RT5668_I2S_PD1_MASK, pre_div << RT5668_I2S_PD1_SHIFT); + + return 0; +} + +static int rt5663_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5668_I2S_MS_S; + 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 |= RT5668_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 |= RT5668_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5668_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5668_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5663_I2S1_SDP, RT5668_I2S_MS_MASK | + RT5668_I2S_BP_MASK | RT5668_I2S_DF_MASK, reg_val); + + return 0; +} + +static int rt5663_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5663->sysclk && clk_id == rt5663->sysclk_src) + return 0; + + switch (clk_id) { + case RT5663_SCLK_S_MCLK: + reg_val |= RT5663_SCLK_SRC_MCLK; + break; + case RT5663_SCLK_S_PLL1: + reg_val |= RT5663_SCLK_SRC_PLL1; + break; + case RT5663_SCLK_S_RCCLK: + reg_val |= RT5663_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5663_GLB_CLK, RT5668_SCLK_SRC_MASK, + reg_val); + rt5663->sysclk = freq; + rt5663->sysclk_src = clk_id; + + dev_dbg(codec->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5663_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 rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + int mask, shift, val; + + if (source == rt5663->pll_src && freq_in == rt5663->pll_in && + freq_out == rt5663->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5663->pll_in = 0; + rt5663->pll_out = 0; + snd_soc_update_bits(codec, RT5663_GLB_CLK, + RT5663_SCLK_SRC_MASK, RT5663_SCLK_SRC_MCLK); + return 0; + } + + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + mask = RT5668_PLL1_SRC_MASK; + shift = RT5668_PLL1_SRC_SHIFT; + break; + case CODEC_TYPE_RT5663: + mask = RT5663_PLL1_SRC_MASK; + shift = RT5663_PLL1_SRC_SHIFT; + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + + switch (source) { + case RT5663_PLL1_S_MCLK: + val = 0x0; + break; + case RT5663_PLL1_S_BCLK1: + val = 0x1; + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5663_GLB_CLK, mask, (val << shift)); + + 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, RT5663_PLL_1, + pll_code.n_code << RT5668_PLL_N_SHIFT | pll_code.k_code); + snd_soc_write(codec, RT5663_PLL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5668_PLL_M_SHIFT | + pll_code.m_bp << RT5668_PLL_M_BP_SHIFT); + + rt5663->pll_in = freq_in; + rt5663->pll_out = freq_out; + rt5663->pll_src = source; + + return 0; +} + +static int rt5663_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, reg; + + if (rx_mask || tx_mask) + val |= RT5668_TDM_MODE_TDM; + + switch (slots) { + case 4: + val |= RT5668_TDM_IN_CH_4; + val |= RT5668_TDM_OUT_CH_4; + break; + case 6: + val |= RT5668_TDM_IN_CH_6; + val |= RT5668_TDM_OUT_CH_6; + break; + case 8: + val |= RT5668_TDM_IN_CH_8; + val |= RT5668_TDM_OUT_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= RT5668_TDM_IN_LEN_20; + val |= RT5668_TDM_OUT_LEN_20; + break; + case 24: + val |= RT5668_TDM_IN_LEN_24; + val |= RT5668_TDM_OUT_LEN_24; + break; + case 32: + val |= RT5668_TDM_IN_LEN_32; + val |= RT5668_TDM_OUT_LEN_32; + break; + case 16: + break; + default: + return -EINVAL; + } + + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + reg = RT5663_TDM_2; + break; + case CODEC_TYPE_RT5663: + reg = RT5663_TDM_1; + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + + snd_soc_update_bits(codec, reg, RT5668_TDM_MODE_MASK | + RT5668_TDM_IN_CH_MASK | RT5668_TDM_OUT_CH_MASK | + RT5668_TDM_IN_LEN_MASK | RT5668_TDM_OUT_LEN_MASK, val); + + return 0; +} + +static int rt5663_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + dev_dbg(codec->dev, "%s ratio = %d\n", __func__, ratio); + + if (rt5663->codec_type == CODEC_TYPE_RT5668) + reg = RT5668_TDM_8; + else + reg = RT5663_TDM_5; + + switch (ratio) { + case 32: + snd_soc_update_bits(codec, reg, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_16); + break; + case 40: + snd_soc_update_bits(codec, reg, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_20); + break; + case 48: + snd_soc_update_bits(codec, reg, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_24); + break; + case 64: + snd_soc_update_bits(codec, reg, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_32); + break; + default: + dev_err(codec->dev, "Invalid ratio!\n"); + return -EINVAL; + } + + return 0; +} + +static int rt5663_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5668_PWR_FV1_MASK | RT5668_PWR_FV2_MASK, + RT5668_PWR_FV1 | RT5668_PWR_FV2); + break; + + case SND_SOC_BIAS_PREPARE: + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5663_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, + RT5668_DIG_GATE_CTRL_EN); + snd_soc_update_bits(codec, RT5663_SIG_CLK_DET, + RT5668_EN_ANA_CLK_DET_MASK | + RT5668_PWR_CLK_DET_MASK, + RT5668_EN_ANA_CLK_DET_AUTO | + RT5668_PWR_CLK_DET_EN); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (rt5663->codec_type == CODEC_TYPE_RT5668) + snd_soc_update_bits(codec, RT5663_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, + RT5668_DIG_GATE_CTRL_DIS); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5668_PWR_VREF1_MASK | RT5668_PWR_VREF2_MASK | + RT5668_PWR_FV1_MASK | RT5668_PWR_FV2_MASK | + RT5668_PWR_MB_MASK, RT5668_PWR_VREF1 | + RT5668_PWR_VREF2 | RT5668_PWR_MB); + usleep_range(10000, 10005); + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5663_SIG_CLK_DET, + RT5668_EN_ANA_CLK_DET_MASK | + RT5668_PWR_CLK_DET_MASK, + RT5668_EN_ANA_CLK_DET_DIS | + RT5668_PWR_CLK_DET_DIS); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5668_PWR_VREF1_MASK | RT5668_PWR_VREF2_MASK | + RT5668_PWR_FV1 | RT5668_PWR_FV2, 0x0); + break; + + default: + break; + } + + return 0; +} + +static int rt5663_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + rt5663->codec = codec; + + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + snd_soc_dapm_new_controls(dapm, + rt5668_specific_dapm_widgets, + ARRAY_SIZE(rt5668_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, + rt5668_specific_dapm_routes, + ARRAY_SIZE(rt5668_specific_dapm_routes)); + snd_soc_add_codec_controls(codec, rt5668_specific_controls, + ARRAY_SIZE(rt5668_specific_controls)); + break; + case CODEC_TYPE_RT5663: + snd_soc_dapm_new_controls(dapm, + rt5663_specific_dapm_widgets, + ARRAY_SIZE(rt5663_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, + rt5663_specific_dapm_routes, + ARRAY_SIZE(rt5663_specific_dapm_routes)); + snd_soc_add_codec_controls(codec, rt5663_specific_controls, + ARRAY_SIZE(rt5663_specific_controls)); + break; + } + + return 0; +} + +static int rt5663_remove(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5663->regmap, RT5663_RESET, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5663_suspend(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5663->regmap, true); + regcache_mark_dirty(rt5663->regmap); + + return 0; +} + +static int rt5663_resume(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5663->regmap, false); + regcache_sync(rt5663->regmap); + + return 0; +} +#else +#define rt5663_suspend NULL +#define rt5663_resume NULL +#endif + +#define RT5663_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5663_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5663_aif_dai_ops = { + .hw_params = rt5663_hw_params, + .set_fmt = rt5663_set_dai_fmt, + .set_sysclk = rt5663_set_dai_sysclk, + .set_pll = rt5663_set_dai_pll, + .set_tdm_slot = rt5663_set_tdm_slot, + .set_bclk_ratio = rt5663_set_bclk_ratio, +}; + +struct snd_soc_dai_driver rt5663_dai[] = { + { + .name = "rt5663-aif", + .id = RT5663_AIF, + .playback = { + .stream_name = "AIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5663_STEREO_RATES, + .formats = RT5663_FORMATS, + }, + .capture = { + .stream_name = "AIF Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5663_STEREO_RATES, + .formats = RT5663_FORMATS, + }, + .ops = &rt5663_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5663 = { + .probe = rt5663_probe, + .remove = rt5663_remove, + .suspend = rt5663_suspend, + .resume = rt5663_resume, + .set_bias_level = rt5663_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt5663_snd_controls, + .num_controls = ARRAY_SIZE(rt5663_snd_controls), + .dapm_widgets = rt5663_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5663_dapm_widgets), + .dapm_routes = rt5663_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5663_dapm_routes), + } +}; + +static const struct regmap_config rt5668_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x07fa, + .volatile_reg = rt5668_volatile_register, + .readable_reg = rt5668_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5668_reg, + .num_reg_defaults = ARRAY_SIZE(rt5668_reg), +}; + +static const struct regmap_config rt5663_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x03f3, + .volatile_reg = rt5663_volatile_register, + .readable_reg = rt5663_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5663_reg, + .num_reg_defaults = ARRAY_SIZE(rt5663_reg), +}; + +static const struct regmap_config temp_regmap = { + .name = "nocache", + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x03f3, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt5663_i2c_id[] = { + { "rt5668", 0 }, + { "rt5663", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5663_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt5663_of_match[] = { + { .compatible = "realtek,rt5668", }, + { .compatible = "realtek,rt5663", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5663_of_match); +#endif + +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5663_acpi_match[] = { + { "10EC5668", 0}, + { "10EC5663", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5663_acpi_match); +#endif + +static void rt5668_calibrate(struct rt5663_priv *rt5668) +{ + regmap_write(rt5668->regmap, RT5663_BIAS_CUR_8, 0xa402); + regmap_write(rt5668->regmap, RT5663_PWR_DIG_1, 0x0100); + regmap_write(rt5668->regmap, RT5663_RECMIX, 0x4040); + regmap_write(rt5668->regmap, RT5663_DIG_MISC, 0x0001); + regmap_write(rt5668->regmap, RT5663_RC_CLK, 0x0380); + regmap_write(rt5668->regmap, RT5663_GLB_CLK, 0x8000); + regmap_write(rt5668->regmap, RT5663_ADDA_CLK_1, 0x1000); + regmap_write(rt5668->regmap, RT5663_CHOP_DAC_L, 0x3030); + regmap_write(rt5668->regmap, RT5663_CALIB_ADC, 0x3c05); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_1, 0xa23e); + msleep(40); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_1, 0xf23e); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_2, 0x0321); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_1, 0xfc00); + msleep(500); +} + +static void rt5663_calibrate(struct rt5663_priv *rt5668) +{ + int value, count; + + regmap_write(rt5668->regmap, RT5663_RC_CLK, 0x0280); + regmap_write(rt5668->regmap, RT5663_GLB_CLK, 0x8000); + regmap_write(rt5668->regmap, RT5663_DIG_MISC, 0x8001); + regmap_write(rt5668->regmap, RT5663_VREF_RECMIX, 0x0032); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_1, 0xa2be); + msleep(20); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_1, 0xf2be); + regmap_write(rt5668->regmap, RT5663_PWR_DIG_2, 0x8400); + regmap_write(rt5668->regmap, RT5663_CHOP_ADC, 0x3000); + regmap_write(rt5668->regmap, RT5663_DEPOP_1, 0x003b); + regmap_write(rt5668->regmap, RT5663_PWR_DIG_1, 0x8df8); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_2, 0x0003); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_3, 0x018c); + regmap_write(rt5668->regmap, RT5663_ADDA_CLK_1, 0x1111); + regmap_write(rt5668->regmap, RT5663_PRE_DIV_GATING_1, 0xffff); + regmap_write(rt5668->regmap, RT5663_PRE_DIV_GATING_2, 0xffff); + regmap_write(rt5668->regmap, RT5663_DEPOP_2, 0x3003); + regmap_write(rt5668->regmap, RT5663_DEPOP_1, 0x003b); + regmap_write(rt5668->regmap, RT5663_HP_CHARGE_PUMP_1, 0x1e32); + regmap_write(rt5668->regmap, RT5663_HP_CHARGE_PUMP_2, 0x1371); + regmap_write(rt5668->regmap, RT5663_DACREF_LDO, 0x3b0b); + regmap_write(rt5668->regmap, RT5663_STO_DAC_MIXER, 0x2080); + regmap_write(rt5668->regmap, RT5663_BYPASS_STO_DAC, 0x000c); + regmap_write(rt5668->regmap, RT5663_HP_BIAS, 0xabba); + regmap_write(rt5668->regmap, RT5663_CHARGE_PUMP_1, 0x2224); + regmap_write(rt5668->regmap, RT5663_HP_OUT_EN, 0x8088); + regmap_write(rt5668->regmap, RT5663_STO_DRE_9, 0x0017); + regmap_write(rt5668->regmap, RT5663_STO_DRE_10, 0x0017); + regmap_write(rt5668->regmap, RT5663_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5668->regmap, RT5663_RECMIX, 0x0005); + regmap_write(rt5668->regmap, RT5663_ADDA_RST, 0xc000); + regmap_write(rt5668->regmap, RT5663_STO1_HPF_ADJ1, 0x3320); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_2, 0x00c9); + regmap_write(rt5668->regmap, RT5663_DUMMY_1, 0x004c); + regmap_write(rt5668->regmap, RT5663_ANA_BIAS_CUR_1, 0x7766); + regmap_write(rt5668->regmap, RT5663_BIAS_CUR_8, 0x4702); + msleep(200); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_1, 0x0069); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_3, 0x06c2); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_1_1, 0x7b00); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_1_1, 0xfb00); + count = 0; + while (true) { + regmap_read(rt5668->regmap, RT5663_HP_CALIB_1_1, &value); + if (value & 0x8000) + usleep_range(10000, 10005); + else + break; + + if (count > 200) + return; + count++; + } +} + +static int rt5663_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5663_priv *rt5663; + int ret; + unsigned int val; + struct regmap *regmap; + + rt5663 = devm_kzalloc(&i2c->dev, sizeof(struct rt5663_priv), + GFP_KERNEL); + + if (rt5663 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5663); + + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&i2c->dev, "Failed to allocate temp register map: %d\n", + ret); + return ret; + } + regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + switch (val) { + case RT5668_DEVICE_ID: + rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5668_regmap); + rt5663->codec_type = CODEC_TYPE_RT5668; + break; + case RT5663_DEVICE_ID: + rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5663_regmap); + rt5663->codec_type = CODEC_TYPE_RT5663; + break; + default: + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663 or rt5668\n", + val); + return -ENODEV; + } + + if (IS_ERR(rt5663->regmap)) { + ret = PTR_ERR(rt5663->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + /* reset and calibrate */ + regmap_write(rt5663->regmap, RT5663_RESET, 0); + regcache_cache_bypass(rt5663->regmap, true); + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + rt5668_calibrate(rt5663); + break; + case CODEC_TYPE_RT5663: + rt5663_calibrate(rt5663); + break; + default: + dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); + } + regcache_cache_bypass(rt5663->regmap, false); + regmap_write(rt5663->regmap, RT5663_RESET, 0); + dev_dbg(&i2c->dev, "calibrate done\n"); + + /* GPIO1 as IRQ */ + regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, RT5668_GP1_PIN_MASK, + RT5668_GP1_PIN_IRQ); + /* 4btn inline command debounce */ + regmap_update_bits(rt5663->regmap, RT5663_IL_CMD_5, + RT5668_4BTN_CLK_DEB_MASK, RT5668_4BTN_CLK_DEB_65MS); + + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + regmap_write(rt5663->regmap, RT5663_BIAS_CUR_8, 0xa402); + /* JD1 */ + regmap_update_bits(rt5663->regmap, RT5663_AUTO_1MRC_CLK, + RT5668_IRQ_POW_SAV_MASK | RT5668_IRQ_POW_SAV_JD1_MASK, + RT5668_IRQ_POW_SAV_EN | RT5668_IRQ_POW_SAV_JD1_EN); + regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_2, + RT5668_PWR_JD1_MASK, RT5668_PWR_JD1); + regmap_update_bits(rt5663->regmap, RT5663_IRQ_1, + RT5668_EN_CB_JD_MASK, RT5668_EN_CB_JD_EN); + + regmap_update_bits(rt5663->regmap, RT5663_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, RT5668_HP_SIG_SRC1_REG); + regmap_update_bits(rt5663->regmap, RT5663_RECMIX, + RT5668_VREF_BIAS_MASK | RT5668_CBJ_DET_MASK | + RT5668_DET_TYPE_MASK, RT5668_VREF_BIAS_REG | + RT5668_CBJ_DET_EN | RT5668_DET_TYPE_QFN); + /* Set GPIO4 and GPIO8 as input for combo jack */ + regmap_update_bits(rt5663->regmap, RT5663_GPIO_2, + RT5668_GP4_PIN_CONF_MASK, RT5668_GP4_PIN_CONF_INPUT); + regmap_update_bits(rt5663->regmap, RT5668_GPIO_3, + RT5668_GP8_PIN_CONF_MASK, RT5668_GP8_PIN_CONF_INPUT); + regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_1, + RT5668_LDO1_DVO_MASK | RT5668_AMP_HP_MASK, + RT5668_LDO1_DVO_0_9V | RT5668_AMP_HP_3X); + break; + case CODEC_TYPE_RT5663: + regmap_write(rt5663->regmap, RT5663_VREF_RECMIX, 0x0032); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xa2be); + msleep(20); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xf2be); + regmap_update_bits(rt5663->regmap, RT5663_GPIO_2, + RT5663_GP1_PIN_CONF_MASK, RT5663_GP1_PIN_CONF_OUTPUT); + /* DACREF LDO control */ + regmap_update_bits(rt5663->regmap, RT5663_DACREF_LDO, 0x3e0e, + 0x3a0a); + regmap_update_bits(rt5663->regmap, RT5663_RECMIX, + RT5663_RECMIX1_BST1_MASK, RT5663_RECMIX1_BST1_ON); + regmap_update_bits(rt5663->regmap, RT5663_TDM_2, + RT5663_DATA_SWAP_ADCDAT1_MASK, + RT5663_DATA_SWAP_ADCDAT1_LL); + break; + default: + dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); + } + + INIT_DELAYED_WORK(&rt5663->jack_detect_work, rt5663_jack_detect_work); + + if (i2c->irq) { + ret = request_irq(i2c->irq, rt5663_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5663", rt5663); + if (ret) + dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", + __func__, ret); + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5663, + rt5663_dai, ARRAY_SIZE(rt5663_dai)); + + if (ret) { + if (i2c->irq) + free_irq(i2c->irq, rt5663); + } + + return ret; +} + +static int rt5663_i2c_remove(struct i2c_client *i2c) +{ + struct rt5663_priv *rt5663 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt5663); + + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +void rt5663_i2c_shutdown(struct i2c_client *client) +{ + struct rt5663_priv *rt5663 = i2c_get_clientdata(client); + + regmap_write(rt5663->regmap, RT5663_RESET, 0); +} + +struct i2c_driver rt5663_i2c_driver = { + .driver = { + .name = "rt5663", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5663_acpi_match), + .of_match_table = of_match_ptr(rt5663_of_match), + }, + .probe = rt5663_i2c_probe, + .remove = rt5663_i2c_remove, + .shutdown = rt5663_i2c_shutdown, + .id_table = rt5663_i2c_id, +}; +module_i2c_driver(rt5663_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5663 driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h new file mode 100644 index 0000000..2cc8f28 --- /dev/null +++ b/sound/soc/codecs/rt5663.h @@ -0,0 +1,1121 @@ +/* + * rt5663.h -- RT5663 ALSA SoC audio driver + * + * Copyright 2016 Realtek Microelectronics + * Author: Jack Yu <jack.yu(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 __RT5663_H__ +#define __RT5663_H__ + +/* Info */ +#define RT5663_RESET 0x0000 +#define RT5663_VENDOR_ID 0x00fd +#define RT5663_VENDOR_ID_1 0x00fe +#define RT5663_VENDOR_ID_2 0x00ff + +#define RT5668_LOUT_CTRL 0x0001 +#define RT5668_HP_AMP_2 0x0003 +#define RT5668_MONO_OUT 0x0004 +#define RT5668_MONO_GAIN 0x0007 + +#define RT5668_AEC_BST 0x000b +#define RT5668_IN1_IN2 0x000c +#define RT5668_IN3_IN4 0x000d +#define RT5668_INL1_INR1 0x000f +#define RT5668_CBJ_TYPE_2 0x0011 +#define RT5668_CBJ_TYPE_3 0x0012 +#define RT5668_CBJ_TYPE_4 0x0013 +#define RT5668_CBJ_TYPE_5 0x0014 +#define RT5668_CBJ_TYPE_8 0x0017 + +/* I/O - ADC/DAC/DMIC */ +#define RT5668_DAC3_DIG_VOL 0x001a +#define RT5668_DAC3_CTRL 0x001b +#define RT5668_MONO_ADC_DIG_VOL 0x001d +#define RT5668_STO2_ADC_DIG_VOL 0x001e +#define RT5668_MONO_ADC_BST_GAIN 0x0020 +#define RT5668_STO2_ADC_BST_GAIN 0x0021 +#define RT5668_SIDETONE_CTRL 0x0024 +/* Mixer - D-D */ +#define RT5668_MONO1_ADC_MIXER 0x0027 +#define RT5668_STO2_ADC_MIXER 0x0028 +#define RT5668_MONO_DAC_MIXER 0x002b +#define RT5668_DAC2_SRC_CTRL 0x002e +#define RT5668_IF_3_4_DATA_CTL 0x002f +#define RT5668_IF_5_DATA_CTL 0x0030 +#define RT5668_PDM_OUT_CTL 0x0031 +#define RT5668_PDM_I2C_DATA_CTL1 0x0032 +#define RT5668_PDM_I2C_DATA_CTL2 0x0033 +#define RT5668_PDM_I2C_DATA_CTL3 0x0034 +#define RT5668_PDM_I2C_DATA_CTL4 0x0035 + +/*Mixer - Analog*/ +#define RT5668_RECMIX1_NEW 0x003a +#define RT5668_RECMIX1L_0 0x003b +#define RT5668_RECMIX1L 0x003c +#define RT5668_RECMIX1R_0 0x003d +#define RT5668_RECMIX1R 0x003e +#define RT5668_RECMIX2_NEW 0x003f +#define RT5668_RECMIX2_L_2 0x0041 +#define RT5668_RECMIX2_R 0x0042 +#define RT5668_RECMIX2_R_2 0x0043 +#define RT5668_CALIB_REC_LR 0x0044 +#define RT5668_ALC_BK_GAIN 0x0049 +#define RT5668_MONOMIX_GAIN 0x004a +#define RT5668_MONOMIX_IN_GAIN 0x004b +#define RT5668_OUT_MIXL_GAIN 0x004d +#define RT5668_OUT_LMIX_IN_GAIN 0x004e +#define RT5668_OUT_RMIX_IN_GAIN 0x004f +#define RT5668_OUT_RMIX_IN_GAIN1 0x0050 +#define RT5668_LOUT_MIXER_CTRL 0x0052 +/* Power */ +#define RT5668_PWR_VOL 0x0067 + +#define RT5668_ADCDAC_RST 0x006d +/* Format - ADC/DAC */ +#define RT5668_I2S34_SDP 0x0071 +#define RT5668_I2S5_SDP 0x0072 +/* Format - TDM Control */ +#define RT5668_TDM_5 0x007c +#define RT5668_TDM_6 0x007d +#define RT5668_TDM_7 0x007e +#define RT5668_TDM_8 0x007f + +/* Function - Analog */ +#define RT5668_ASRC_3 0x0085 +#define RT5668_ASRC_6 0x0088 +#define RT5668_ASRC_7 0x0089 +#define RT5668_PLL_TRK_13 0x0099 +#define RT5668_I2S_M_CLK_CTL 0x00a0 +#define RT5668_FDIV_I2S34_M_CLK 0x00a1 +#define RT5668_FDIV_I2S34_M_CLK2 0x00a2 +#define RT5668_FDIV_I2S5_M_CLK 0x00a3 +#define RT5668_FDIV_I2S5_M_CLK2 0x00a4 + +/* Function - Digital */ +#define RT5668_IRQ_4 0x00b9 +#define RT5668_GPIO_3 0x00c2 +#define RT5668_GPIO_4 0x00c3 +#define RT5668_GPIO_STA 0x00c4 +#define RT5668_HP_AMP_DET1 0x00d0 +#define RT5668_HP_AMP_DET2 0x00d1 +#define RT5668_HP_AMP_DET3 0x00d2 +#define RT5668_MID_BD_HP_AMP 0x00d3 +#define RT5668_LOW_BD_HP_AMP 0x00d4 +#define RT5668_SOF_VOL_ZC2 0x00da +#define RT5668_ADC_STO2_ADJ1 0x00ee +#define RT5668_ADC_STO2_ADJ2 0x00ef +/* General Control */ +#define RT5668_A_JD_CTRL 0x00f0 +#define RT5668_JD1_TRES_CTRL 0x00f1 +#define RT5668_JD2_TRES_CTRL 0x00f2 +#define RT5668_JD_CTRL2 0x00f7 +#define RT5668_DUM_REG_2 0x00fb +#define RT5668_DUM_REG_3 0x00fc + + +#define RT5668_DACADC_DIG_VOL2 0x0101 +#define RT5668_DIG_IN_PIN2 0x0133 +#define RT5668_PAD_DRV_CTL1 0x0136 +#define RT5668_SOF_RAM_DEPOP 0x0138 +#define RT5668_VOL_TEST 0x013f +#define RT5668_TEST_MODE_3 0x0147 +#define RT5668_TEST_MODE_4 0x0148 +#define RT5668_MONO_DYNA_1 0x0170 +#define RT5668_MONO_DYNA_2 0x0171 +#define RT5668_MONO_DYNA_3 0x0172 +#define RT5668_MONO_DYNA_4 0x0173 +#define RT5668_MONO_DYNA_5 0x0174 +#define RT5668_MONO_DYNA_6 0x0175 +#define RT5668_STO1_SIL_DET 0x0190 +#define RT5668_MONOL_SIL_DET 0x0191 +#define RT5668_MONOR_SIL_DET 0x0192 +#define RT5668_STO2_DAC_SIL 0x0193 +#define RT5668_PWR_SAV_CTL1 0x0194 +#define RT5668_PWR_SAV_CTL2 0x0195 +#define RT5668_PWR_SAV_CTL3 0x0196 +#define RT5668_PWR_SAV_CTL4 0x0197 +#define RT5668_PWR_SAV_CTL5 0x0198 +#define RT5668_PWR_SAV_CTL6 0x0199 +#define RT5668_MONO_AMP_CAL1 0x01a0 +#define RT5668_MONO_AMP_CAL2 0x01a1 +#define RT5668_MONO_AMP_CAL3 0x01a2 +#define RT5668_MONO_AMP_CAL4 0x01a3 +#define RT5668_MONO_AMP_CAL5 0x01a4 +#define RT5668_MONO_AMP_CAL6 0x01a5 +#define RT5668_MONO_AMP_CAL7 0x01a6 +#define RT5668_MONO_AMP_CAL_ST1 0x01a7 +#define RT5668_MONO_AMP_CAL_ST2 0x01a8 +#define RT5668_MONO_AMP_CAL_ST3 0x01a9 +#define RT5668_MONO_AMP_CAL_ST4 0x01aa +#define RT5668_MONO_AMP_CAL_ST5 0x01ab +#define RT5668_HP_IMP_SEN_13 0x01b9 +#define RT5668_HP_IMP_SEN_14 0x01ba +#define RT5668_HP_IMP_SEN_6 0x01bb +#define RT5668_HP_IMP_SEN_7 0x01bc +#define RT5668_HP_IMP_SEN_8 0x01bd +#define RT5668_HP_IMP_SEN_9 0x01be +#define RT5668_HP_IMP_SEN_10 0x01bf +#define RT5668_HP_LOGIC_3 0x01dc +#define RT5668_HP_CALIB_ST10 0x01f3 +#define RT5668_HP_CALIB_ST11 0x01f4 +#define RT5668_PRO_REG_TBL_4 0x0203 +#define RT5668_PRO_REG_TBL_5 0x0204 +#define RT5668_PRO_REG_TBL_6 0x0205 +#define RT5668_PRO_REG_TBL_7 0x0206 +#define RT5668_PRO_REG_TBL_8 0x0207 +#define RT5668_PRO_REG_TBL_9 0x0208 +#define RT5668_SAR_ADC_INL_1 0x0210 +#define RT5668_SAR_ADC_INL_2 0x0211 +#define RT5668_SAR_ADC_INL_3 0x0212 +#define RT5668_SAR_ADC_INL_4 0x0213 +#define RT5668_SAR_ADC_INL_5 0x0214 +#define RT5668_SAR_ADC_INL_6 0x0215 +#define RT5668_SAR_ADC_INL_7 0x0216 +#define RT5668_SAR_ADC_INL_8 0x0217 +#define RT5668_SAR_ADC_INL_9 0x0218 +#define RT5668_SAR_ADC_INL_10 0x0219 +#define RT5668_SAR_ADC_INL_11 0x021a +#define RT5668_SAR_ADC_INL_12 0x021b +#define RT5668_DRC_CTRL_1 0x02ff +#define RT5668_DRC1_CTRL_2 0x0301 +#define RT5668_DRC1_CTRL_3 0x0302 +#define RT5668_DRC1_CTRL_4 0x0303 +#define RT5668_DRC1_CTRL_5 0x0304 +#define RT5668_DRC1_CTRL_6 0x0305 +#define RT5668_DRC1_HD_CTRL_1 0x0306 +#define RT5668_DRC1_HD_CTRL_2 0x0307 +#define RT5668_DRC1_PRI_REG_1 0x0310 +#define RT5668_DRC1_PRI_REG_2 0x0311 +#define RT5668_DRC1_PRI_REG_3 0x0312 +#define RT5668_DRC1_PRI_REG_4 0x0313 +#define RT5668_DRC1_PRI_REG_5 0x0314 +#define RT5668_DRC1_PRI_REG_6 0x0315 +#define RT5668_DRC1_PRI_REG_7 0x0316 +#define RT5668_DRC1_PRI_REG_8 0x0317 +#define RT5668_ALC_PGA_CTL_1 0x0330 +#define RT5668_ALC_PGA_CTL_2 0x0331 +#define RT5668_ALC_PGA_CTL_3 0x0332 +#define RT5668_ALC_PGA_CTL_4 0x0333 +#define RT5668_ALC_PGA_CTL_5 0x0334 +#define RT5668_ALC_PGA_CTL_6 0x0335 +#define RT5668_ALC_PGA_CTL_7 0x0336 +#define RT5668_ALC_PGA_CTL_8 0x0337 +#define RT5668_ALC_PGA_REG_1 0x0338 +#define RT5668_ALC_PGA_REG_2 0x0339 +#define RT5668_ALC_PGA_REG_3 0x033a +#define RT5668_ADC_EQ_RECOV_1 0x03c0 +#define RT5668_ADC_EQ_RECOV_2 0x03c1 +#define RT5668_ADC_EQ_RECOV_3 0x03c2 +#define RT5668_ADC_EQ_RECOV_4 0x03c3 +#define RT5668_ADC_EQ_RECOV_5 0x03c4 +#define RT5668_ADC_EQ_RECOV_6 0x03c5 +#define RT5668_ADC_EQ_RECOV_7 0x03c6 +#define RT5668_ADC_EQ_RECOV_8 0x03c7 +#define RT5668_ADC_EQ_RECOV_9 0x03c8 +#define RT5668_ADC_EQ_RECOV_10 0x03c9 +#define RT5668_ADC_EQ_RECOV_11 0x03ca +#define RT5668_ADC_EQ_RECOV_12 0x03cb +#define RT5668_ADC_EQ_RECOV_13 0x03cc +#define RT5668_VID_HIDDEN 0x03fe +#define RT5668_VID_CUSTOMER 0x03ff +#define RT5668_SCAN_MODE 0x07f0 +#define RT5668_I2C_BYPA 0x07fa + +/* Headphone Amp Control 2 (0x0003) */ +#define RT5668_EN_DAC_HPO_MASK (0x1 << 14) +#define RT5668_EN_DAC_HPO_SHIFT 14 +#define RT5668_EN_DAC_HPO_DIS (0x0 << 14) +#define RT5668_EN_DAC_HPO_EN (0x1 << 14) + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5668_GAIN_HP (0x1f << 8) +#define RT5668_GAIN_HP_SHIFT 8 + +/* AEC BST Control (0x000b) */ +#define RT5668_GAIN_CBJ_MASK (0xf << 8) +#define RT5668_GAIN_CBJ_SHIFT 8 + +/* IN1 Control / MIC GND REF (0x000c) */ +#define RT5668_IN1_DF_MASK (0x1 << 15) +#define RT5668_IN1_DF_SHIFT 15 + +/* Combo Jack and Type Detection Control 1 (0x0010) */ +#define RT5668_CBJ_DET_MASK (0x1 << 15) +#define RT5668_CBJ_DET_SHIFT 15 +#define RT5668_CBJ_DET_DIS (0x0 << 15) +#define RT5668_CBJ_DET_EN (0x1 << 15) +#define RT5668_DET_TYPE_MASK (0x1 << 12) +#define RT5668_DET_TYPE_SHIFT 12 +#define RT5668_DET_TYPE_WLCSP (0x0 << 12) +#define RT5668_DET_TYPE_QFN (0x1 << 12) +#define RT5668_VREF_BIAS_MASK (0x1 << 6) +#define RT5668_VREF_BIAS_SHIFT 6 +#define RT5668_VREF_BIAS_FSM (0x0 << 6) +#define RT5668_VREF_BIAS_REG (0x1 << 6) + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5668_RECMIX1L_BST1_CBJ (0x1 << 7) +#define RT5668_RECMIX1L_BST1_CBJ_SHIFT 7 +#define RT5668_RECMIX1L_BST2 (0x1 << 4) +#define RT5668_RECMIX1L_BST2_SHIFT 4 + +/* REC Right Mixer Control 2 (0x003e) */ +#define RT5668_RECMIX1R_BST2 (0x1 << 4) +#define RT5668_RECMIX1R_BST2_SHIFT 4 + +/* DAC1 Digital Volume (0x0019) */ +#define RT5668_DAC_L1_VOL_MASK (0xff << 8) +#define RT5668_DAC_L1_VOL_SHIFT 8 +#define RT5668_DAC_R1_VOL_MASK (0xff) +#define RT5668_DAC_R1_VOL_SHIFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5668_ADC_L_MUTE_MASK (0x1 << 15) +#define RT5668_ADC_L_MUTE_SHIFT 15 +#define RT5668_ADC_L_VOL_MASK (0x7f << 8) +#define RT5668_ADC_L_VOL_SHIFT 8 +#define RT5668_ADC_R_MUTE_MASK (0x1 << 7) +#define RT5668_ADC_R_MUTE_SHIFT 7 +#define RT5668_ADC_R_VOL_MASK (0x7f) +#define RT5668_ADC_R_VOL_SHIFT 0 + +/* Stereo ADC Mixer Control (0x0026) */ +#define RT5668_M_STO1_ADC_L1 (0x1 << 15) +#define RT5668_M_STO1_ADC_L1_SHIFT 15 +#define RT5668_M_STO1_ADC_L2 (0x1 << 14) +#define RT5668_M_STO1_ADC_L2_SHIFT 14 +#define RT5668_STO1_ADC_L1_SRC (0x1 << 13) +#define RT5668_STO1_ADC_L1_SRC_SHIFT 13 +#define RT5668_STO1_ADC_L2_SRC (0x1 << 12) +#define RT5668_STO1_ADC_L2_SRC_SHIFT 12 +#define RT5668_STO1_ADC_L_SRC (0x3 << 10) +#define RT5668_STO1_ADC_L_SRC_SHIFT 10 +#define RT5668_M_STO1_ADC_R1 (0x1 << 7) +#define RT5668_M_STO1_ADC_R1_SHIFT 7 +#define RT5668_M_STO1_ADC_R2 (0x1 << 6) +#define RT5668_M_STO1_ADC_R2_SHIFT 6 +#define RT5668_STO1_ADC_R1_SRC (0x1 << 5) +#define RT5668_STO1_ADC_R1_SRC_SHIFT 5 +#define RT5668_STO1_ADC_R2_SRC (0x1 << 4) +#define RT5668_STO1_ADC_R2_SRC_SHIFT 4 +#define RT5668_STO1_ADC_R_SRC (0x3 << 2) +#define RT5668_STO1_ADC_R_SRC_SHIFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5668_M_ADCMIX_L (0x1 << 15) +#define RT5668_M_ADCMIX_L_SHIFT 15 +#define RT5668_M_DAC1_L (0x1 << 14) +#define RT5668_M_DAC1_L_SHIFT 14 +#define RT5668_M_ADCMIX_R (0x1 << 7) +#define RT5668_M_ADCMIX_R_SHIFT 7 +#define RT5668_M_DAC1_R (0x1 << 6) +#define RT5668_M_DAC1_R_SHIFT 6 + +/* Stereo DAC Mixer Control (0x002a) */ +#define RT5668_M_DAC_L1_STO_L (0x1 << 15) +#define RT5668_M_DAC_L1_STO_L_SHIFT 15 +#define RT5668_M_DAC_R1_STO_L (0x1 << 13) +#define RT5668_M_DAC_R1_STO_L_SHIFT 13 +#define RT5668_M_DAC_L1_STO_R (0x1 << 7) +#define RT5668_M_DAC_L1_STO_R_SHIFT 7 +#define RT5668_M_DAC_R1_STO_R (0x1 << 5) +#define RT5668_M_DAC_R1_STO_R_SHIFT 5 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5668_PWR_I2S1 (0x1 << 15) +#define RT5668_PWR_I2S1_SHIFT 15 +#define RT5668_PWR_DAC_L1 (0x1 << 11) +#define RT5668_PWR_DAC_L1_SHIFT 11 +#define RT5668_PWR_DAC_R1 (0x1 << 10) +#define RT5668_PWR_DAC_R1_SHIFT 10 +#define RT5668_PWR_LDO_DACREF_MASK (0x1 << 8) +#define RT5668_PWR_LDO_DACREF_SHIFT 8 +#define RT5668_PWR_LDO_DACREF_ON (0x1 << 8) +#define RT5668_PWR_LDO_DACREF_DOWN (0x0 << 8) +#define RT5668_PWR_LDO_SHIFT 8 +#define RT5668_PWR_ADC_L1 (0x1 << 4) +#define RT5668_PWR_ADC_L1_SHIFT 4 +#define RT5668_PWR_ADC_R1 (0x1 << 3) +#define RT5668_PWR_ADC_R1_SHIFT 3 + +/* Power Management for Digital 2 (0x0062) */ +#define RT5668_PWR_ADC_S1F (0x1 << 15) +#define RT5668_PWR_ADC_S1F_SHIFT 15 +#define RT5668_PWR_DAC_S1F (0x1 << 10) +#define RT5668_PWR_DAC_S1F_SHIFT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5668_PWR_VREF1 (0x1 << 15) +#define RT5668_PWR_VREF1_MASK (0x1 << 15) +#define RT5668_PWR_VREF1_SHIFT 15 +#define RT5668_PWR_FV1 (0x1 << 14) +#define RT5668_PWR_FV1_MASK (0x1 << 14) +#define RT5668_PWR_FV1_SHIFT 14 +#define RT5668_PWR_VREF2 (0x1 << 13) +#define RT5668_PWR_VREF2_MASK (0x1 << 13) +#define RT5668_PWR_VREF2_SHIFT 13 +#define RT5668_PWR_FV2 (0x1 << 12) +#define RT5668_PWR_FV2_MASK (0x1 << 12) +#define RT5668_PWR_FV2_SHIFT 12 +#define RT5668_PWR_MB (0x1 << 9) +#define RT5668_PWR_MB_MASK (0x1 << 9) +#define RT5668_PWR_MB_SHIFT 9 +#define RT5668_AMP_HP_MASK (0x3 << 2) +#define RT5668_AMP_HP_SHIFT 2 +#define RT5668_AMP_HP_1X (0x0 << 2) +#define RT5668_AMP_HP_3X (0x1 << 2) +#define RT5668_AMP_HP_5X (0x3 << 2) +#define RT5668_LDO1_DVO_MASK (0x3) +#define RT5668_LDO1_DVO_SHIFT 0 +#define RT5668_LDO1_DVO_0_9V (0x0) +#define RT5668_LDO1_DVO_1_0V (0x1) +#define RT5668_LDO1_DVO_1_2V (0x2) +#define RT5668_LDO1_DVO_1_4V (0x3) + +/* Power Management for Analog 2 (0x0064) */ +#define RT5668_PWR_BST1 (0x1 << 15) +#define RT5668_PWR_BST1_MASK (0x1 << 15) +#define RT5668_PWR_BST1_SHIFT 15 +#define RT5668_PWR_BST1_OFF (0x0 << 15) +#define RT5668_PWR_BST1_ON (0x1 << 15) +#define RT5668_PWR_BST2 (0x1 << 14) +#define RT5668_PWR_BST2_MASK (0x1 << 14) +#define RT5668_PWR_BST2_SHIFT 14 +#define RT5668_PWR_MB1 (0x1 << 11) +#define RT5668_PWR_MB1_SHIFT 11 +#define RT5668_PWR_MB2 (0x1 << 10) +#define RT5668_PWR_MB2_SHIFT 10 +#define RT5668_PWR_BST2_OP (0x1 << 6) +#define RT5668_PWR_BST2_OP_MASK (0x1 << 6) +#define RT5668_PWR_BST2_OP_SHIFT 6 +#define RT5668_PWR_JD1 (0x1 << 3) +#define RT5668_PWR_JD1_MASK (0x1 << 3) +#define RT5668_PWR_JD1_SHIFT 3 +#define RT5668_PWR_JD2 (0x1 << 2) +#define RT5668_PWR_JD2_MASK (0x1 << 2) +#define RT5668_PWR_JD2_SHIFT 2 +#define RT5668_PWR_RECMIX1 (0x1 << 1) +#define RT5668_PWR_RECMIX1_SHIFT 1 +#define RT5668_PWR_RECMIX2 (0x1) +#define RT5668_PWR_RECMIX2_SHIFT 0 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5668_PWR_CBJ_MASK (0x1 << 9) +#define RT5668_PWR_CBJ_SHIFT 9 +#define RT5668_PWR_CBJ_OFF (0x0 << 9) +#define RT5668_PWR_CBJ_ON (0x1 << 9) +#define RT5668_PWR_PLL (0x1 << 6) +#define RT5668_PWR_PLL_SHIFT 6 +#define RT5668_PWR_LDO2 (0x1 << 2) +#define RT5668_PWR_LDO2_SHIFT 2 + +/* Power Management for Volume (0x0067) */ +#define RT5668_PWR_MIC_DET (0x1 << 5) +#define RT5668_PWR_MIC_DET_SHIFT 5 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5668_EN_ANA_CLK_DET_MASK (0x1 << 15) +#define RT5668_EN_ANA_CLK_DET_SHIFT 15 +#define RT5668_EN_ANA_CLK_DET_DIS (0x0 << 15) +#define RT5668_EN_ANA_CLK_DET_AUTO (0x1 << 15) +#define RT5668_PWR_CLK_DET_MASK (0x1) +#define RT5668_PWR_CLK_DET_SHIFT 0 +#define RT5668_PWR_CLK_DET_DIS (0x0) +#define RT5668_PWR_CLK_DET_EN (0x1) + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5668_I2S_MS_MASK (0x1 << 15) +#define RT5668_I2S_MS_SHIFT 15 +#define RT5668_I2S_MS_M (0x0 << 15) +#define RT5668_I2S_MS_S (0x1 << 15) +#define RT5668_I2S_BP_MASK (0x1 << 8) +#define RT5668_I2S_BP_SHIFT 8 +#define RT5668_I2S_BP_NOR (0x0 << 8) +#define RT5668_I2S_BP_INV (0x1 << 8) +#define RT5668_I2S_DL_MASK (0x3 << 4) +#define RT5668_I2S_DL_SHIFT 4 +#define RT5668_I2S_DL_16 (0x0 << 4) +#define RT5668_I2S_DL_20 (0x1 << 4) +#define RT5668_I2S_DL_24 (0x2 << 4) +#define RT5668_I2S_DL_8 (0x3 << 4) +#define RT5668_I2S_DF_MASK (0x7) +#define RT5668_I2S_DF_SHIFT 0 +#define RT5668_I2S_DF_I2S (0x0) +#define RT5668_I2S_DF_LEFT (0x1) +#define RT5668_I2S_DF_PCM_A (0x2) +#define RT5668_I2S_DF_PCM_B (0x3) +#define RT5668_I2S_DF_PCM_A_N (0x6) +#define RT5668_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5668_I2S_PD1_MASK (0x7 << 12) +#define RT5668_I2S_PD1_SHIFT 12 +#define RT5668_M_I2S_DIV_MASK (0x7 << 8) +#define RT5668_M_I2S_DIV_SHIFT 8 +#define RT5668_CLK_SRC_MASK (0x3 << 4) +#define RT5668_CLK_SRC_MCLK (0x0 << 4) +#define RT5668_CLK_SRC_PLL_OUT (0x1 << 4) +#define RT5668_CLK_SRC_DIV (0x2 << 4) +#define RT5668_CLK_SRC_RC (0x3 << 4) +#define RT5668_DAC_OSR_MASK (0x3 << 2) +#define RT5668_DAC_OSR_SHIFT 2 +#define RT5668_DAC_OSR_128 (0x0 << 2) +#define RT5668_DAC_OSR_64 (0x1 << 2) +#define RT5668_DAC_OSR_32 (0x2 << 2) +#define RT5668_ADC_OSR_MASK (0x3) +#define RT5668_ADC_OSR_SHIFT 0 +#define RT5668_ADC_OSR_128 (0x0) +#define RT5668_ADC_OSR_64 (0x1) +#define RT5668_ADC_OSR_32 (0x2) + +/* TDM1 control 1 (0x0078) */ +#define RT5668_TDM_MODE_MASK (0x1 << 15) +#define RT5668_TDM_MODE_SHIFT 15 +#define RT5668_TDM_MODE_I2S (0x0 << 15) +#define RT5668_TDM_MODE_TDM (0x1 << 15) +#define RT5668_TDM_IN_CH_MASK (0x3 << 10) +#define RT5668_TDM_IN_CH_SHIFT 10 +#define RT5668_TDM_IN_CH_2 (0x0 << 10) +#define RT5668_TDM_IN_CH_4 (0x1 << 10) +#define RT5668_TDM_IN_CH_6 (0x2 << 10) +#define RT5668_TDM_IN_CH_8 (0x3 << 10) +#define RT5668_TDM_OUT_CH_MASK (0x3 << 8) +#define RT5668_TDM_OUT_CH_SHIFT 8 +#define RT5668_TDM_OUT_CH_2 (0x0 << 8) +#define RT5668_TDM_OUT_CH_4 (0x1 << 8) +#define RT5668_TDM_OUT_CH_6 (0x2 << 8) +#define RT5668_TDM_OUT_CH_8 (0x3 << 8) +#define RT5668_TDM_IN_LEN_MASK (0x3 << 6) +#define RT5668_TDM_IN_LEN_SHIFT 6 +#define RT5668_TDM_IN_LEN_16 (0x0 << 6) +#define RT5668_TDM_IN_LEN_20 (0x1 << 6) +#define RT5668_TDM_IN_LEN_24 (0x2 << 6) +#define RT5668_TDM_IN_LEN_32 (0x3 << 6) +#define RT5668_TDM_OUT_LEN_MASK (0x3 << 4) +#define RT5668_TDM_OUT_LEN_SHIFT 4 +#define RT5668_TDM_OUT_LEN_16 (0x0 << 4) +#define RT5668_TDM_OUT_LEN_20 (0x1 << 4) +#define RT5668_TDM_OUT_LEN_24 (0x2 << 4) +#define RT5668_TDM_OUT_LEN_32 (0x3 << 4) + +/* Global Clock Control (0x0080) */ +#define RT5668_SCLK_SRC_MASK (0x3 << 14) +#define RT5668_SCLK_SRC_SHIFT 14 +#define RT5668_SCLK_SRC_MCLK (0x0 << 14) +#define RT5668_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5668_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5668_PLL1_SRC_MASK (0x7 << 8) +#define RT5668_PLL1_SRC_SHIFT 8 +#define RT5668_PLL1_SRC_MCLK (0x0 << 8) +#define RT5668_PLL1_SRC_BCLK1 (0x1 << 8) +#define RT5668_PLL1_PD_MASK (0x1 << 4) +#define RT5668_PLL1_PD_SHIFT 4 + +#define RT5668_PLL_INP_MAX 40000000 +#define RT5668_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5668_PLL_N_MAX 0x001ff +#define RT5668_PLL_N_MASK (RT5668_PLL_N_MAX << 7) +#define RT5668_PLL_N_SHIFT 7 +#define RT5668_PLL_K_MAX 0x001f +#define RT5668_PLL_K_MASK (RT5668_PLL_K_MAX) +#define RT5668_PLL_K_SHIFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5668_PLL_M_MAX 0x00f +#define RT5668_PLL_M_MASK (RT5668_PLL_M_MAX << 12) +#define RT5668_PLL_M_SHIFT 12 +#define RT5668_PLL_M_BP (0x1 << 11) +#define RT5668_PLL_M_BP_SHIFT 11 + +/* PLL tracking mode 1 (0x0083) */ +#define RT5668_I2S1_ASRC_MASK (0x1 << 13) +#define RT5668_I2S1_ASRC_SHIFT 13 +#define RT5668_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5668_DAC_STO1_ASRC_SHIFT 12 +#define RT5668_ADC_STO1_ASRC_MASK (0x1 << 4) +#define RT5668_ADC_STO1_ASRC_SHIFT 4 + +/* PLL tracking mode 2 (0x0084)*/ +#define RT5668_DA_STO1_TRACK_MASK (0x7 << 12) +#define RT5668_DA_STO1_TRACK_SHIFT 12 +#define RT5668_DA_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5668_DA_STO1_TRACK_I2S1 (0x1 << 12) + +/* PLL tracking mode 3 (0x0085)*/ +#define RT5668_AD_STO1_TRACK_MASK (0x7 << 12) +#define RT5668_AD_STO1_TRACK_SHIFT 12 +#define RT5668_AD_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5668_AD_STO1_TRACK_I2S1 (0x1 << 12) + +/* HPOUT Charge pump control 1 (0x0091) */ +#define RT5668_OSW_HP_L_MASK (0x1 << 11) +#define RT5668_OSW_HP_L_SHIFT 11 +#define RT5668_OSW_HP_L_EN (0x1 << 11) +#define RT5668_OSW_HP_L_DIS (0x0 << 11) +#define RT5668_OSW_HP_R_MASK (0x1 << 10) +#define RT5668_OSW_HP_R_SHIFT 10 +#define RT5668_OSW_HP_R_EN (0x1 << 10) +#define RT5668_OSW_HP_R_DIS (0x0 << 10) +#define RT5668_SEL_PM_HP_MASK (0x3 << 8) +#define RT5668_SEL_PM_HP_SHIFT 8 +#define RT5668_SEL_PM_HP_0_6 (0x0 << 8) +#define RT5668_SEL_PM_HP_0_9 (0x1 << 8) +#define RT5668_SEL_PM_HP_1_8 (0x2 << 8) +#define RT5668_SEL_PM_HP_HIGH (0x3 << 8) +#define RT5668_OVCD_HP_MASK (0x1 << 2) +#define RT5668_OVCD_HP_SHIFT 2 +#define RT5668_OVCD_HP_EN (0x1 << 2) +#define RT5668_OVCD_HP_DIS (0x0 << 2) + +/* RC Clock Control (0x0094) */ +#define RT5668_DIG_25M_CLK_MASK (0x1 << 9) +#define RT5668_DIG_25M_CLK_SHIFT 9 +#define RT5668_DIG_25M_CLK_DIS (0x0 << 9) +#define RT5668_DIG_25M_CLK_EN (0x1 << 9) +#define RT5668_DIG_1M_CLK_MASK (0x1 << 8) +#define RT5668_DIG_1M_CLK_SHIFT 8 +#define RT5668_DIG_1M_CLK_DIS (0x0 << 8) +#define RT5668_DIG_1M_CLK_EN (0x1 << 8) + +/* Auto Turn On 1M RC CLK (0x009f) */ +#define RT5668_IRQ_POW_SAV_MASK (0x1 << 15) +#define RT5668_IRQ_POW_SAV_SHIFT 15 +#define RT5668_IRQ_POW_SAV_DIS (0x0 << 15) +#define RT5668_IRQ_POW_SAV_EN (0x1 << 15) +#define RT5668_IRQ_POW_SAV_JD1_MASK (0x1 << 14) +#define RT5668_IRQ_POW_SAV_JD1_SHIFT 14 +#define RT5668_IRQ_POW_SAV_JD1_DIS (0x0 << 14) +#define RT5668_IRQ_POW_SAV_JD1_EN (0x1 << 14) + +/* IRQ Control 1 (0x00b6) */ +#define RT5668_EN_CB_JD_MASK (0x1 << 3) +#define RT5668_EN_CB_JD_SHIFT 3 +#define RT5668_EN_CB_JD_EN (0x1 << 3) +#define RT5668_EN_CB_JD_DIS (0x0 << 3) + +/* IRQ Control 3 (0x00b8) */ +#define RT5668_EN_IRQ_INLINE_MASK (0x1 << 6) +#define RT5668_EN_IRQ_INLINE_SHIFT 6 +#define RT5668_EN_IRQ_INLINE_BYP (0x0 << 6) +#define RT5668_EN_IRQ_INLINE_NOR (0x1 << 6) + +/* GPIO Control 1 (0x00c0) */ +#define RT5668_GP1_PIN_MASK (0x1 << 15) +#define RT5668_GP1_PIN_SHIFT 15 +#define RT5668_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5668_GP1_PIN_IRQ (0x1 << 15) + +/* GPIO Control 2 (0x00c1) */ +#define RT5668_GP4_PIN_CONF_MASK (0x1 << 5) +#define RT5668_GP4_PIN_CONF_SHIFT 5 +#define RT5668_GP4_PIN_CONF_INPUT (0x0 << 5) +#define RT5668_GP4_PIN_CONF_OUTPUT (0x1 << 5) + +/* GPIO Control 2 (0x00c2) */ +#define RT5668_GP8_PIN_CONF_MASK (0x1 << 13) +#define RT5668_GP8_PIN_CONF_SHIFT 13 +#define RT5668_GP8_PIN_CONF_INPUT (0x0 << 13) +#define RT5668_GP8_PIN_CONF_OUTPUT (0x1 << 13) + +/* 4 Buttons Inline Command Function 1 (0x00df) */ +#define RT5668_4BTN_CLK_DEB_MASK (0x3 << 2) +#define RT5668_4BTN_CLK_DEB_SHIFT 2 +#define RT5668_4BTN_CLK_DEB_8MS (0x0 << 2) +#define RT5668_4BTN_CLK_DEB_16MS (0x1 << 2) +#define RT5668_4BTN_CLK_DEB_32MS (0x2 << 2) +#define RT5668_4BTN_CLK_DEB_65MS (0x3 << 2) + +/* Inline Command Function 6 (0x00e0) */ +#define RT5668_EN_4BTN_INL_MASK (0x1 << 15) +#define RT5668_EN_4BTN_INL_SHIFT 15 +#define RT5668_EN_4BTN_INL_DIS (0x0 << 15) +#define RT5668_EN_4BTN_INL_EN (0x1 << 15) +#define RT5668_RESET_4BTN_INL_MASK (0x1 << 14) +#define RT5668_RESET_4BTN_INL_SHIFT 14 +#define RT5668_RESET_4BTN_INL_RESET (0x0 << 14) +#define RT5668_RESET_4BTN_INL_NOR (0x1 << 14) + +/* Digital Misc Control (0x00fa) */ +#define RT5668_DIG_GATE_CTRL_MASK 0x1 +#define RT5668_DIG_GATE_CTRL_SHIFT (0) +#define RT5668_DIG_GATE_CTRL_DIS 0x0 +#define RT5668_DIG_GATE_CTRL_EN 0x1 + +/* Chopper and Clock control for DAC L (0x013a)*/ +#define RT5668_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5668_CKXEN_DAC1_SHIFT 13 +#define RT5668_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5668_CKGEN_DAC1_SHIFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5668_CKXEN_ADCC_MASK (0x1 << 13) +#define RT5668_CKXEN_ADCC_SHIFT 13 +#define RT5668_CKGEN_ADCC_MASK (0x1 << 12) +#define RT5668_CKGEN_ADCC_SHIFT 12 + +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5668_HP_SIG_SRC1_MASK (0x3) +#define RT5668_HP_SIG_SRC1_SHIFT 0 +#define RT5668_HP_SIG_SRC1_HP_DC (0x0) +#define RT5668_HP_SIG_SRC1_HP_CALIB (0x1) +#define RT5668_HP_SIG_SRC1_REG (0x2) +#define RT5668_HP_SIG_SRC1_SILENCE (0x3) + +/* RT5663 specific register */ +#define RT5663_HP_OUT_EN 0x0002 +#define RT5663_HP_LCH_DRE 0x0005 +#define RT5663_HP_RCH_DRE 0x0006 +#define RT5663_CALIB_BST 0x000a +#define RT5663_RECMIX 0x0010 +#define RT5663_SIL_DET_CTL 0x0015 +#define RT5663_PWR_SAV_SILDET 0x0016 +#define RT5663_SIDETONE_CTL 0x0018 +#define RT5663_STO1_DAC_DIG_VOL 0x0019 +#define RT5663_STO1_ADC_DIG_VOL 0x001c +#define RT5663_STO1_BOOST 0x001f +#define RT5663_HP_IMP_GAIN_1 0x0022 +#define RT5663_HP_IMP_GAIN_2 0x0023 +#define RT5663_STO1_ADC_MIXER 0x0026 +#define RT5663_AD_DA_MIXER 0x0029 +#define RT5663_STO_DAC_MIXER 0x002a +#define RT5663_DIG_SIDE_MIXER 0x002c +#define RT5663_BYPASS_STO_DAC 0x002d +#define RT5663_CALIB_REC_MIX 0x0040 +#define RT5663_PWR_DIG_1 0x0061 +#define RT5663_PWR_DIG_2 0x0062 +#define RT5663_PWR_ANLG_1 0x0063 +#define RT5663_PWR_ANLG_2 0x0064 +#define RT5663_PWR_ANLG_3 0x0065 +#define RT5663_PWR_MIXER 0x0066 +#define RT5663_SIG_CLK_DET 0x006b +#define RT5663_PRE_DIV_GATING_1 0x006e +#define RT5663_PRE_DIV_GATING_2 0x006f +#define RT5663_I2S1_SDP 0x0070 +#define RT5663_ADDA_CLK_1 0x0073 +#define RT5663_ADDA_RST 0x0074 +#define RT5663_FRAC_DIV_1 0x0075 +#define RT5663_FRAC_DIV_2 0x0076 +#define RT5663_TDM_1 0x0077 +#define RT5663_TDM_2 0x0078 +#define RT5663_TDM_3 0x0079 +#define RT5663_TDM_4 0x007a +#define RT5663_TDM_5 0x007b +#define RT5663_GLB_CLK 0x0080 +#define RT5663_PLL_1 0x0081 +#define RT5663_PLL_2 0x0082 +#define RT5663_ASRC_1 0x0083 +#define RT5663_ASRC_2 0x0084 +#define RT5663_ASRC_4 0x0086 +#define RT5663_DUMMY_REG 0x0087 +#define RT5663_ASRC_8 0x008a +#define RT5663_ASRC_9 0x008b +#define RT5663_ASRC_11 0x008c +#define RT5663_DEPOP_1 0x008e +#define RT5663_DEPOP_2 0x008f +#define RT5663_DEPOP_3 0x0090 +#define RT5663_HP_CHARGE_PUMP_1 0x0091 +#define RT5663_HP_CHARGE_PUMP_2 0x0092 +#define RT5663_MICBIAS_1 0x0093 +#define RT5663_RC_CLK 0x0094 +#define RT5663_ASRC_11_2 0x0097 +#define RT5663_DUMMY_REG_2 0x0098 +#define RT5663_REC_PATH_GAIN 0x009a +#define RT5663_AUTO_1MRC_CLK 0x009f +#define RT5663_ADC_EQ_1 0x00ae +#define RT5663_ADC_EQ_2 0x00af +#define RT5663_IRQ_1 0x00b6 +#define RT5663_IRQ_2 0x00b7 +#define RT5663_IRQ_3 0x00b8 +#define RT5663_IRQ_4 0x00ba +#define RT5663_IRQ_5 0x00bb +#define RT5663_INT_ST_1 0x00be +#define RT5663_INT_ST_2 0x00bf +#define RT5663_GPIO_1 0x00c0 +#define RT5663_GPIO_2 0x00c1 +#define RT5663_GPIO_STA 0x00c5 +#define RT5663_SIN_GEN_1 0x00cb +#define RT5663_SIN_GEN_2 0x00cc +#define RT5663_SIN_GEN_3 0x00cd +#define RT5663_SOF_VOL_ZC1 0x00d9 +#define RT5663_IL_CMD_1 0x00db +#define RT5663_IL_CMD_2 0x00dc +#define RT5663_IL_CMD_3 0x00dd +#define RT5663_IL_CMD_4 0x00de +#define RT5663_IL_CMD_5 0x00df +#define RT5663_IL_CMD_6 0x00e0 +#define RT5663_IL_CMD_7 0x00e1 +#define RT5663_IL_CMD_8 0x00e2 +#define RT5663_IL_CMD_PWRSAV1 0x00e4 +#define RT5663_IL_CMD_PWRSAV2 0x00e5 +#define RT5663_EM_JACK_TYPE_1 0x00e6 +#define RT5663_EM_JACK_TYPE_2 0x00e7 +#define RT5663_EM_JACK_TYPE_3 0x00e8 +#define RT5663_EM_JACK_TYPE_4 0x00e9 +#define RT5663_EM_JACK_TYPE_5 0x00ea +#define RT5663_EM_JACK_TYPE_6 0x00eb +#define RT5663_STO1_HPF_ADJ1 0x00ec +#define RT5663_STO1_HPF_ADJ2 0x00ed +#define RT5663_FAST_OFF_MICBIAS 0x00f4 +#define RT5663_JD_CTRL1 0x00f6 +#define RT5663_JD_CTRL2 0x00f8 +#define RT5663_DIG_MISC 0x00fa +#define RT5663_DIG_VOL_ZCD 0x0100 +#define RT5663_ANA_BIAS_CUR_1 0x0108 +#define RT5663_ANA_BIAS_CUR_2 0x0109 +#define RT5663_ANA_BIAS_CUR_3 0x010a +#define RT5663_ANA_BIAS_CUR_4 0x010b +#define RT5663_ANA_BIAS_CUR_5 0x010c +#define RT5663_ANA_BIAS_CUR_6 0x010d +#define RT5663_BIAS_CUR_5 0x010e +#define RT5663_BIAS_CUR_6 0x010f +#define RT5663_BIAS_CUR_7 0x0110 +#define RT5663_BIAS_CUR_8 0x0111 +#define RT5663_DACREF_LDO 0x0112 +#define RT5663_DUMMY_REG_3 0x0113 +#define RT5663_BIAS_CUR_9 0x0114 +#define RT5663_DUMMY_REG_4 0x0116 +#define RT5663_VREFADJ_OP 0x0117 +#define RT5663_VREF_RECMIX 0x0118 +#define RT5663_CHARGE_PUMP_1 0x0125 +#define RT5663_CHARGE_PUMP_1_2 0x0126 +#define RT5663_CHARGE_PUMP_1_3 0x0127 +#define RT5663_CHARGE_PUMP_2 0x0128 +#define RT5663_DIG_IN_PIN1 0x0132 +#define RT5663_PAD_DRV_CTL 0x0137 +#define RT5663_PLL_INT_REG 0x0139 +#define RT5663_CHOP_DAC_L 0x013a +#define RT5663_CHOP_ADC 0x013b +#define RT5663_CALIB_ADC 0x013c +#define RT5663_CHOP_DAC_R 0x013d +#define RT5663_DUMMY_CTL_DACLR 0x013e +#define RT5663_DUMMY_REG_5 0x0140 +#define RT5663_SOFT_RAMP 0x0141 +#define RT5663_TEST_MODE_1 0x0144 +#define RT5663_TEST_MODE_2 0x0145 +#define RT5663_TEST_MODE_3 0x0146 +#define RT5663_STO_DRE_1 0x0160 +#define RT5663_STO_DRE_2 0x0161 +#define RT5663_STO_DRE_3 0x0162 +#define RT5663_STO_DRE_4 0x0163 +#define RT5663_STO_DRE_5 0x0164 +#define RT5663_STO_DRE_6 0x0165 +#define RT5663_STO_DRE_7 0x0166 +#define RT5663_STO_DRE_8 0x0167 +#define RT5663_STO_DRE_9 0x0168 +#define RT5663_STO_DRE_10 0x0169 +#define RT5663_MIC_DECRO_1 0x0180 +#define RT5663_MIC_DECRO_2 0x0181 +#define RT5663_MIC_DECRO_3 0x0182 +#define RT5663_MIC_DECRO_4 0x0183 +#define RT5663_MIC_DECRO_5 0x0184 +#define RT5663_MIC_DECRO_6 0x0185 +#define RT5663_HP_DECRO_1 0x01b0 +#define RT5663_HP_DECRO_2 0x01b1 +#define RT5663_HP_DECRO_3 0x01b2 +#define RT5663_HP_DECRO_4 0x01b3 +#define RT5663_HP_DECOUP 0x01b4 +#define RT5663_HP_IMP_SEN_MAP8 0x01b5 +#define RT5663_HP_IMP_SEN_MAP9 0x01b6 +#define RT5663_HP_IMP_SEN_MAP10 0x01b7 +#define RT5663_HP_IMP_SEN_MAP11 0x01b8 +#define RT5663_HP_IMP_SEN_1 0x01c0 +#define RT5663_HP_IMP_SEN_2 0x01c1 +#define RT5663_HP_IMP_SEN_3 0x01c2 +#define RT5663_HP_IMP_SEN_4 0x01c3 +#define RT5663_HP_IMP_SEN_5 0x01c4 +#define RT5663_HP_IMP_SEN_6 0x01c5 +#define RT5663_HP_IMP_SEN_7 0x01c6 +#define RT5663_HP_IMP_SEN_8 0x01c7 +#define RT5663_HP_IMP_SEN_9 0x01c8 +#define RT5663_HP_IMP_SEN_10 0x01c9 +#define RT5663_HP_IMP_SEN_11 0x01ca +#define RT5663_HP_IMP_SEN_12 0x01cb +#define RT5663_HP_IMP_SEN_13 0x01cc +#define RT5663_HP_IMP_SEN_14 0x01cd +#define RT5663_HP_IMP_SEN_15 0x01ce +#define RT5663_HP_IMP_SEN_16 0x01cf +#define RT5663_HP_IMP_SEN_17 0x01d0 +#define RT5663_HP_IMP_SEN_18 0x01d1 +#define RT5663_HP_IMP_SEN_19 0x01d2 +#define RT5663_HP_IMPSEN_DIG5 0x01d3 +#define RT5663_HP_IMPSEN_MAP1 0x01d4 +#define RT5663_HP_IMPSEN_MAP2 0x01d5 +#define RT5663_HP_IMPSEN_MAP3 0x01d6 +#define RT5663_HP_IMPSEN_MAP4 0x01d7 +#define RT5663_HP_IMPSEN_MAP5 0x01d8 +#define RT5663_HP_IMPSEN_MAP7 0x01d9 +#define RT5663_HP_LOGIC_1 0x01da +#define RT5663_HP_LOGIC_2 0x01db +#define RT5663_HP_CALIB_1 0x01dd +#define RT5663_HP_CALIB_1_1 0x01de +#define RT5663_HP_CALIB_2 0x01df +#define RT5663_HP_CALIB_3 0x01e0 +#define RT5663_HP_CALIB_4 0x01e1 +#define RT5663_HP_CALIB_5 0x01e2 +#define RT5663_HP_CALIB_5_1 0x01e3 +#define RT5663_HP_CALIB_6 0x01e4 +#define RT5663_HP_CALIB_7 0x01e5 +#define RT5663_HP_CALIB_9 0x01e6 +#define RT5663_HP_CALIB_10 0x01e7 +#define RT5663_HP_CALIB_11 0x01e8 +#define RT5663_HP_CALIB_ST1 0x01ea +#define RT5663_HP_CALIB_ST2 0x01eb +#define RT5663_HP_CALIB_ST3 0x01ec +#define RT5663_HP_CALIB_ST4 0x01ed +#define RT5663_HP_CALIB_ST5 0x01ee +#define RT5663_HP_CALIB_ST6 0x01ef +#define RT5663_HP_CALIB_ST7 0x01f0 +#define RT5663_HP_CALIB_ST8 0x01f1 +#define RT5663_HP_CALIB_ST9 0x01f2 +#define RT5663_HP_AMP_DET 0x0200 +#define RT5663_DUMMY_REG_6 0x0201 +#define RT5663_HP_BIAS 0x0202 +#define RT5663_CBJ_1 0x0250 +#define RT5663_CBJ_2 0x0251 +#define RT5663_CBJ_3 0x0252 +#define RT5663_DUMMY_1 0x02fa +#define RT5663_DUMMY_2 0x02fb +#define RT5663_DUMMY_3 0x02fc +#define RT5663_ANA_JD 0x0300 +#define RT5663_ADC_LCH_LPF1_A1 0x03d0 +#define RT5663_ADC_RCH_LPF1_A1 0x03d1 +#define RT5663_ADC_LCH_LPF1_H0 0x03d2 +#define RT5663_ADC_RCH_LPF1_H0 0x03d3 +#define RT5663_ADC_LCH_BPF1_A1 0x03d4 +#define RT5663_ADC_RCH_BPF1_A1 0x03d5 +#define RT5663_ADC_LCH_BPF1_A2 0x03d6 +#define RT5663_ADC_RCH_BPF1_A2 0x03d7 +#define RT5663_ADC_LCH_BPF1_H0 0x03d8 +#define RT5663_ADC_RCH_BPF1_H0 0x03d9 +#define RT5663_ADC_LCH_BPF2_A1 0x03da +#define RT5663_ADC_RCH_BPF2_A1 0x03db +#define RT5663_ADC_LCH_BPF2_A2 0x03dc +#define RT5663_ADC_RCH_BPF2_A2 0x03dd +#define RT5663_ADC_LCH_BPF2_H0 0x03de +#define RT5663_ADC_RCH_BPF2_H0 0x03df +#define RT5663_ADC_LCH_BPF3_A1 0x03e0 +#define RT5663_ADC_RCH_BPF3_A1 0x03e1 +#define RT5663_ADC_LCH_BPF3_A2 0x03e2 +#define RT5663_ADC_RCH_BPF3_A2 0x03e3 +#define RT5663_ADC_LCH_BPF3_H0 0x03e4 +#define RT5663_ADC_RCH_BPF3_H0 0x03e5 +#define RT5663_ADC_LCH_BPF4_A1 0x03e6 +#define RT5663_ADC_RCH_BPF4_A1 0x03e7 +#define RT5663_ADC_LCH_BPF4_A2 0x03e8 +#define RT5663_ADC_RCH_BPF4_A2 0x03e9 +#define RT5663_ADC_LCH_BPF4_H0 0x03ea +#define RT5663_ADC_RCH_BPF4_H0 0x03eb +#define RT5663_ADC_LCH_HPF1_A1 0x03ec +#define RT5663_ADC_RCH_HPF1_A1 0x03ed +#define RT5663_ADC_LCH_HPF1_H0 0x03ee +#define RT5663_ADC_RCH_HPF1_H0 0x03ef +#define RT5663_ADC_EQ_PRE_VOL_L 0x03f0 +#define RT5663_ADC_EQ_PRE_VOL_R 0x03f1 +#define RT5663_ADC_EQ_POST_VOL_L 0x03f2 +#define RT5663_ADC_EQ_POST_VOL_R 0x03f3 + +/* RT5663: RECMIX Control (0x0010) */ +#define RT5663_RECMIX1_BST1_MASK (0x1) +#define RT5663_RECMIX1_BST1_SHIFT 0 +#define RT5663_RECMIX1_BST1_ON (0x0) +#define RT5663_RECMIX1_BST1_OFF (0x1) + +/* RT5663: Bypass Stereo1 DAC Mixer Control (0x002d) */ +#define RT5663_DACL1_SRC_MASK (0x1 << 3) +#define RT5663_DACL1_SRC_SHIFT 3 +#define RT5663_DACR1_SRC_MASK (0x1 << 2) +#define RT5663_DACR1_SRC_SHIFT 2 + +/* RT5663: TDM control 2 (0x0078) */ +#define RT5663_DATA_SWAP_ADCDAT1_MASK (0x3 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_SHIFT 14 +#define RT5663_DATA_SWAP_ADCDAT1_LR (0x0 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_RL (0x1 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_LL (0x2 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_RR (0x3 << 14) + +/* RT5663: TDM control 5 (0x007b) */ +#define RT5663_TDM_LENGTN_MASK (0x3) +#define RT5663_TDM_LENGTN_SHIFT 0 +#define RT5663_TDM_LENGTN_16 (0x0) +#define RT5663_TDM_LENGTN_20 (0x1) +#define RT5663_TDM_LENGTN_24 (0x2) +#define RT5663_TDM_LENGTN_32 (0x3) + +/* RT5663: Global Clock Control (0x0080) */ +#define RT5663_SCLK_SRC_MASK (0x3 << 14) +#define RT5663_SCLK_SRC_SHIFT 14 +#define RT5663_SCLK_SRC_MCLK (0x0 << 14) +#define RT5663_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5663_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5663_PLL1_SRC_MASK (0x7 << 11) +#define RT5663_PLL1_SRC_SHIFT 11 +#define RT5663_PLL1_SRC_MCLK (0x0 << 11) +#define RT5663_PLL1_SRC_BCLK1 (0x1 << 11) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5663_I2S1_ASRC_MASK (0x1 << 11) +#define RT5663_I2S1_ASRC_SHIFT 11 +#define RT5663_DAC_STO1_ASRC_MASK (0x1 << 10) +#define RT5663_DAC_STO1_ASRC_SHIFT 10 +#define RT5663_ADC_STO1_ASRC_MASK (0x1 << 3) +#define RT5663_ADC_STO1_ASRC_SHIFT 3 + +/* PLL tracking mode 2 (0x0084)*/ +#define RT5663_DA_STO1_TRACK_MASK (0x7 << 12) +#define RT5663_DA_STO1_TRACK_SHIFT 12 +#define RT5663_DA_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5663_DA_STO1_TRACK_I2S1 (0x1 << 12) +#define RT5663_AD_STO1_TRACK_MASK (0x7) +#define RT5663_AD_STO1_TRACK_SHIFT 0 +#define RT5663_AD_STO1_TRACK_SYSCLK (0x0) +#define RT5663_AD_STO1_TRACK_I2S1 (0x1) + +/* RT5663: HPOUT Charge pump control 1 (0x0091) */ +#define RT5663_SI_HP_MASK (0x1 << 12) +#define RT5663_SI_HP_SHIFT 12 +#define RT5663_SI_HP_EN (0x1 << 12) +#define RT5663_SI_HP_DIS (0x0 << 12) + +/* RT5663: GPIO Control 2 (0x00b6) */ +#define RT5663_GP1_PIN_CONF_MASK (0x1 << 2) +#define RT5663_GP1_PIN_CONF_SHIFT 2 +#define RT5663_GP1_PIN_CONF_OUTPUT (0x1 << 2) +#define RT5663_GP1_PIN_CONF_INPUT (0x0 << 2) + +/* RT5663: GPIO Control 2 (0x00b7) */ +#define RT5663_EN_IRQ_INLINE_MASK (0x1 << 3) +#define RT5663_EN_IRQ_INLINE_SHIFT 3 +#define RT5663_EN_IRQ_INLINE_NOR (0x1 << 3) +#define RT5663_EN_IRQ_INLINE_BYP (0x0 << 3) + +/* RT5663: IRQ Control 1 (0x00c1) */ +#define RT5663_EN_IRQ_JD1_MASK (0x1 << 6) +#define RT5663_EN_IRQ_JD1_SHIFT 6 +#define RT5663_EN_IRQ_JD1_EN (0x1 << 6) +#define RT5663_EN_IRQ_JD1_DIS (0x0 << 6) + +/* RT5663: Inline Command Function 2 (0x00dc) */ +#define RT5663_PWR_MIC_DET_MASK (0x1) +#define RT5663_PWR_MIC_DET_SHIFT 0 +#define RT5663_PWR_MIC_DET_ON (0x1) +#define RT5663_PWR_MIC_DET_OFF (0x0) + +/* RT5663: Embeeded Jack and Type Detection Control 1 (0x00e6)*/ +#define RT5663_CBJ_DET_MASK (0x1 << 15) +#define RT5663_CBJ_DET_SHIFT 15 +#define RT5663_CBJ_DET_DIS (0x0 << 15) +#define RT5663_CBJ_DET_EN (0x1 << 15) +#define RT5663_EXT_JD_MASK (0x1 << 11) +#define RT5663_EXT_JD_SHIFT 11 +#define RT5663_EXT_JD_EN (0x1 << 11) +#define RT5663_EXT_JD_DIS (0x0 << 11) +#define RT5663_POL_EXT_JD_MASK (0x1 << 10) +#define RT5663_POL_EXT_JD_SHIFT 10 +#define RT5663_POL_EXT_JD_EN (0x1 << 10) +#define RT5663_POL_EXT_JD_DIS (0x0 << 10) + +/* RT5663: DACREF LDO Control (0x0112)*/ +#define RT5663_PWR_LDO_DACREFL_MASK (0x1 << 9) +#define RT5663_PWR_LDO_DACREFL_SHIFT 9 +#define RT5663_PWR_LDO_DACREFR_MASK (0x1 << 1) +#define RT5663_PWR_LDO_DACREFR_SHIFT 1 + +/* RT5663: Stereo Dynamic Range Enhancement Control 9 (0x0168, 0x0169)*/ +#define RT5663_DRE_GAIN_HP_MASK (0x1f) +#define RT5663_DRE_GAIN_HP_SHIFT 0 + +/* RT5663: Combo Jack Control (0x0250) */ +#define RT5663_INBUF_CBJ_BST1_MASK (0x1 << 11) +#define RT5663_INBUF_CBJ_BST1_SHIFT 11 +#define RT5663_INBUF_CBJ_BST1_ON (0x1 << 11) +#define RT5663_INBUF_CBJ_BST1_OFF (0x0 << 11) +#define RT5663_CBJ_SENSE_BST1_MASK (0x1 << 10) +#define RT5663_CBJ_SENSE_BST1_SHIFT 10 +#define RT5663_CBJ_SENSE_BST1_L (0x1 << 10) +#define RT5663_CBJ_SENSE_BST1_R (0x0 << 10) + +/* RT5663: Combo Jack Control (0x0251) */ +#define RT5663_GAIN_BST1_MASK (0xf) +#define RT5663_GAIN_BST1_SHIFT 0 + +/* RT5663: Dummy register 1 (0x02fa) */ +#define RT5663_EMB_CLK_MASK (0x1 << 9) +#define RT5663_EMB_CLK_SHIFT 9 +#define RT5663_EMB_CLK_EN (0x1 << 9) +#define RT5663_EMB_CLK_DIS (0x0 << 9) +#define RT5663_HPA_CPL_BIAS_MASK (0x7 << 6) +#define RT5663_HPA_CPL_BIAS_SHIFT 6 +#define RT5663_HPA_CPL_BIAS_0_5 (0x0 << 6) +#define RT5663_HPA_CPL_BIAS_1 (0x1 << 6) +#define RT5663_HPA_CPL_BIAS_2 (0x2 << 6) +#define RT5663_HPA_CPL_BIAS_3 (0x3 << 6) +#define RT5663_HPA_CPL_BIAS_4_1 (0x4 << 6) +#define RT5663_HPA_CPL_BIAS_4_2 (0x5 << 6) +#define RT5663_HPA_CPL_BIAS_6 (0x6 << 6) +#define RT5663_HPA_CPL_BIAS_8 (0x7 << 6) +#define RT5663_HPA_CPR_BIAS_MASK (0x7 << 3) +#define RT5663_HPA_CPR_BIAS_SHIFT 3 +#define RT5663_HPA_CPR_BIAS_0_5 (0x0 << 3) +#define RT5663_HPA_CPR_BIAS_1 (0x1 << 3) +#define RT5663_HPA_CPR_BIAS_2 (0x2 << 3) +#define RT5663_HPA_CPR_BIAS_3 (0x3 << 3) +#define RT5663_HPA_CPR_BIAS_4_1 (0x4 << 3) +#define RT5663_HPA_CPR_BIAS_4_2 (0x5 << 3) +#define RT5663_HPA_CPR_BIAS_6 (0x6 << 3) +#define RT5663_HPA_CPR_BIAS_8 (0x7 << 3) +#define RT5663_DUMMY_BIAS_MASK (0x7) +#define RT5663_DUMMY_BIAS_SHIFT 0 +#define RT5663_DUMMY_BIAS_0_5 (0x0) +#define RT5663_DUMMY_BIAS_1 (0x1) +#define RT5663_DUMMY_BIAS_2 (0x2) +#define RT5663_DUMMY_BIAS_3 (0x3) +#define RT5663_DUMMY_BIAS_4_1 (0x4) +#define RT5663_DUMMY_BIAS_4_2 (0x5) +#define RT5663_DUMMY_BIAS_6 (0x6) +#define RT5663_DUMMY_BIAS_8 (0x7) + + +/* System Clock Source */ +enum { + RT5663_SCLK_S_MCLK, + RT5663_SCLK_S_PLL1, + RT5663_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5663_PLL1_S_MCLK, + RT5663_PLL1_S_BCLK1, +}; + +enum { + RT5663_AIF, + RT5663_AIFS, +}; + +/* asrc clock source */ +enum { + RT5663_CLK_SEL_SYS = 0x0, + RT5663_CLK_SEL_I2S1_ASRC = 0x1, +}; + +/* filter mask */ +enum { + RT5663_DA_STEREO_FILTER = 0x1, + RT5663_AD_STEREO_FILTER = 0x2, +}; + +int rt5663_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack); +int rt5663_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5663_H__ */ -- 1.8.1.1.439.g50a6b54

1 0

Re: [alsa-devel] [PATCH v2] clk: x86: Add Atom PMC platform clocks
by Stephen Boyd 08 Sep '16

08 Sep '16

On 09/07, Irina Tirdea wrote: > @@ -152,6 +155,17 @@ > #define SLEEP_TYPE_S5 0x1C00 > #define SLEEP_ENABLE 0x2000 > > +struct pmc_clk { > + const char *name; > + unsigned long freq; > + const char *parent_name; > +}; > + > +struct pmc_clk_data { > + void __iomem *base; > + const struct pmc_clk *clks; > +}; Can you please put these structures in include/linux/platform_data/ with some new file? That way we don't need to have any architecture specific configuration to test build the clk driver and we can drop the asm include fro mthe clk driver too. > 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..ec98bdf > --- /dev/null > +++ b/drivers/clk/x86/clk-byt-plt.c > @@ -0,0 +1,387 @@ > +/* > + * 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_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; Hopefully this can be a clk_hw pointer instead. > + struct clk_lookup *lookup; > +}; > + [..] > + > +static void plt_clk_reg_update(struct clk_plt *clk, u32 mask, u32 val) > +{ > + u32 orig, tmp; > + unsigned long flags = 0; > + > + spin_lock_irqsave(&clk->lock, flags); > + > + orig = readl(clk->reg); > + > + tmp = orig & ~mask; > + tmp |= val & mask; > + > + if (tmp == orig) > + goto out; > + > + writel(tmp, clk->reg); Or just if (tmp != orig) writel(..) > + > +out: > + spin_unlock_irqrestore(&clk->lock, flags); > +} > + [..] > + > +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, clk_hw_register_fixed_rate? > + 0, fixed_rate); > + if (IS_ERR(pclk->clk)) { > + ret = PTR_ERR(pclk->clk); > + return ERR_PTR(ret); Could be simplified to return ERR_CAST(pclk->clk); -- Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, a Linux Foundation Collaborative Project

2 2

[alsa-devel] sound: divide by 0 in snd_hrtimer_callback (or hang)
by Dmitry Vyukov 08 Sep '16

08 Sep '16

Hello, The following program causes either division error or hangs kernel: https://gist.githubusercontent.com/dvyukov/b1f33be3bfcc15d629e10db483bee1e4… divide error: 0000 [#1] SMP DEBUG_PAGEALLOC KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 1 PID: 27320 Comm: syz-executor Not tainted 4.8.0-rc3-next-20160825+ #8 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88003c9b2280 task.stack: ffff880027280000 RIP: 0010:[<ffffffff858e1a6c>] [< inline >] ktime_divns include/linux/ktime.h:195 RIP: 0010:[<ffffffff858e1a6c>] [<ffffffff858e1a6c>] snd_hrtimer_callback+0x1bc/0x3c0 sound/core/hrtimer.c:62 RSP: 0018:ffff88003ed07dd8 EFLAGS: 00010006 RAX: 0000000000004801 RBX: ffff88003d3a0a88 RCX: 0000000000000000 RDX: 0000000000000000 RSI: ffff88003d3a0aa0 RDI: ffffffff886b69a8 RBP: ffff88003ed07e30 R08: 0000010e85417665 R09: ffff88007fff7048 R10: ffff88007fff7058 R11: ffff88007fff7050 R12: ffff88003d00f840 R13: ffff88003d00f8f4 R14: ffff88003d00f970 R15: ffff88003d3a0a80 FS: 0000000000000000(0000) GS:ffff88003ed00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007f4152126db8 CR3: 000000006b896000 CR4: 00000000000006e0 DR0: 000000000000001e DR1: 000000000000001e DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000600 Stack: 0000000000000000 ffff88003d3a0aa0 ffff88003d00f8f8 0000000000004801 ffff88003d3a0ae0 0000000000000000 ffff88003d3a0a88 0000000000000001 ffff88003ed1b400 dffffc0000000000 0000000000000001 ffff88003ed07f28 Call Trace: <IRQ> [< inline >] __run_hrtimer kernel/time/hrtimer.c:1238 [<ffffffff81504335>] __hrtimer_run_queues+0x325/0xe70 kernel/time/hrtimer.c:1302 [<ffffffff81506ceb>] hrtimer_interrupt+0x18b/0x420 kernel/time/hrtimer.c:1336 [<ffffffff8126d8df>] local_apic_timer_interrupt+0x6f/0xe0 arch/x86/kernel/apic/apic.c:933 [<ffffffff86e13056>] smp_apic_timer_interrupt+0x76/0xa0 arch/x86/kernel/apic/apic.c:957 [<ffffffff86e1210c>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:487 <EOI> [< inline >] rcu_lock_release include/linux/rcupdate.h:494 [< inline >] rcu_read_unlock include/linux/rcupdate.h:927 [<ffffffff81839f55>] unlock_page_memcg+0x95/0x130 mm/memcontrol.c:1680 [< inline >] page_remove_file_rmap mm/rmap.c:1343 [<ffffffff817b742c>] page_remove_rmap+0x1cc/0x8e0 mm/rmap.c:1394 [< inline >] zap_pte_range mm/memory.c:1170 [< inline >] zap_pmd_range mm/memory.c:1257 [< inline >] zap_pud_range mm/memory.c:1278 [<ffffffff81789119>] unmap_page_range+0xf89/0x1ba0 mm/memory.c:1299 [<ffffffff81789e31>] unmap_single_vma+0x101/0x260 mm/memory.c:1344 [<ffffffff8178a741>] unmap_vmas+0xf1/0x1b0 [<ffffffff817a5aab>] exit_mmap+0x22b/0x420 mm/mmap.c:2787 [< inline >] __mmput kernel/fork.c:770 [<ffffffff8137c576>] mmput+0xd6/0x3c0 kernel/fork.c:790 [< inline >] exit_mm kernel/exit.c:512 [<ffffffff813940e8>] do_exit+0x738/0x2e70 kernel/exit.c:815 [<ffffffff81396998>] do_group_exit+0x108/0x330 kernel/exit.c:958 [<ffffffff813ba4aa>] get_signal+0x62a/0x15d0 kernel/signal.c:2307 [<ffffffff81202333>] do_signal+0x83/0x1f60 arch/x86/kernel/signal.c:805 [<ffffffff81006345>] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:163 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:198 [< inline >] syscall_return_slowpath arch/x86/entry/common.c:267 [<ffffffff81008be0>] do_syscall_64+0x4c0/0x640 arch/x86/entry/common.c:293 [<ffffffff86e107c3>] entry_SYSCALL64_slow_path+0x25/0x25 Code: cb a2 ce fb 8b 0d f5 21 70 05 48 0f af 4d d0 48 85 c9 0f 88 6a 01 00 00 48 89 4d a8 e8 ae a2 ce fb 48 8b 45 c0 48 8b 4d a8 48 99 <48> f7 f9 48 01 45 d0 e8 98 a2 ce fb 4c 89 fa 48 b8 00 00 00 00 RIP [< inline >] ktime_divns include/linux/ktime.h:195 RIP [<ffffffff858e1a6c>] snd_hrtimer_callback+0x1bc/0x3c0 sound/core/hrtimer.c:62 RSP <ffff88003ed07dd8> ---[ end trace e1a6d1e5a6136d50 ]--- or: INFO: rcu_sched detected stalls on CPUs/tasks: 0-...: (1 GPs behind) idle=203/140000000000001/0 softirq=923090/923091 fqs=6490 (detected by 1, t=26002 jiffies, g=508022, c=508021, q=20) Task dump for CPU 0: a.out R running task 28848 31756 4047 0x0000000a 0000000000000000 0000000000000282 0000000000000000 ffff880066a9b430 ffff880066a9b430 ffff8800667cfae0 0000000000000282 0000000000000000 0000000000000000 0000000000000000 000003a0184674dd dffffc0000000000 Call Trace: [< inline >] spin_unlock_irqrestore ./include/linux/spinlock.h:362 [<ffffffff858db3dd>] snd_timer_start1+0xdd/0x740 sound/core/timer.c:477 [<ffffffff858dbb25>] snd_timer_continue+0x45/0x80 sound/core/timer.c:606 [< inline >] snd_timer_user_continue sound/core/timer.c:1835 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1902 [<ffffffff858df299>] snd_timer_user_ioctl+0xe99/0x2470 sound/core/timer.c:1917 [< inline >] vfs_ioctl fs/ioctl.c:43 [<ffffffff818a1dfc>] do_vfs_ioctl+0x18c/0x1080 fs/ioctl.c:675 [< inline >] SYSC_ioctl fs/ioctl.c:690 [<ffffffff818a2d7f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:681 [<ffffffff86e10700>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:208 I am on 0f98f121e1670eaa2a2fbb675e07d6ba7f0e146f of linux-next with the following diff in sound/core/timer.c (one is a fix for another division by 0): diff --git a/sound/core/timer.c b/sound/core/timer.c index 9a6157e..3e55c6d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -813,6 +813,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, timer->tmr_subdevice = tid->subdevice; if (id) strlcpy(timer->id, id, sizeof(timer->id)); + timer->sticks = 1; INIT_LIST_HEAD(&timer->device_list); INIT_LIST_HEAD(&timer->open_list_head); INIT_LIST_HEAD(&timer->active_list_head); @@ -1958,6 +1959,7 @@ 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 +1969,7 @@ 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 9

[alsa-devel] [PATCH] ALSA: hda - Skip Realtek SKU check for Lenovo machines
by Takashi Iwai 08 Sep '16

08 Sep '16

Realtek codec driver expects an implicit rule where either the codec SSID or the pincfg NID 0x1d contains the some information encoded in some bits. One of the expected information is there is the availability of PC beep, and the driver doesn't build up the PC beep control if this is *supposed* to be disabled there. Meanwhile, Lenovo doesn't seem to follow this requirement (yes it's non-standard after all), and the BIOS sets just the normal SSID and the pincfg values. This resulted in the lack of PC beep on a few machines, purely with a lucky or unlucky number. It didn't bother most people, but some people still demand the PC beep, as found in bug reports. This patch just adds the fixup chain to Lenovo machines to skip the SKU checks. Then the beep control will show up in the mixer, and user can still decide to enable / disable it via the standard mixer interface. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=156311 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 574b1b48996f..4be2c77031fa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5263,6 +5263,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, .v.func = hda_fixup_thinkpad_acpi, + .chained = true, + .chain_id = ALC269_FIXUP_SKU_IGNORE, }, [ALC269_FIXUP_DMIC_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, -- 2.10.0

1 0

[alsa-devel] [PATCH] ALSA: usb: constify snd_pcm_ops structures
by Julia Lawall 08 Sep '16

08 Sep '16

Check for snd_pcm_ops structures that are only stored in the ops field of a snd_soc_platform_driver structure or passed as the third argument to snd_pcm_set_ops. The corresponding field or parameter is declared const, so snd_pcm_ops structures that have this property can be declared as const also. 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_pcm_ops i@p = { ... }; @ok1@ identifier r.i; struct snd_soc_platform_driver e; position p; @@ e.ops = &i@p; @ok2@ identifier r.i; expression e1, e2; position p; @@ snd_pcm_set_ops(e1, e2, &i@p) @bad@ position p != {r.p,ok1.p,ok2.p}; identifier r.i; struct snd_pcm_ops e; @@ e@i@p @depends on !bad disable optional_qualifier@ identifier r.i; @@ static +const struct snd_pcm_ops i = { ... }; // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/usb/6fire/pcm.c | 2 +- sound/usb/caiaq/audio.c | 2 +- sound/usb/hiface/pcm.c | 2 +- sound/usb/misc/ua101.c | 4 ++-- sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- 5 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 36f4115..224a6a5 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -555,7 +555,7 @@ static snd_pcm_uframes_t usb6fire_pcm_pointer( return ret; } -static struct snd_pcm_ops pcm_ops = { +static const struct snd_pcm_ops pcm_ops = { .open = usb6fire_pcm_open, .close = usb6fire_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 2c44139..23e66ed 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -511,7 +511,7 @@ static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) return bytes_to_frames(alsa_sub->runtime, dma_offset); } -static struct snd_pcm_ops pcm_ops = { +static const struct snd_pcm_ops pcm_ops = { .open = hiface_pcm_open, .close = hiface_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 90766a9..1f28cb2a 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -587,7 +587,7 @@ static int snd_usX2Y_usbpcm_close(struct snd_pcm_substream *substream) } -static struct snd_pcm_ops snd_usX2Y_usbpcm_ops = +static const struct snd_pcm_ops snd_usX2Y_usbpcm_ops = { .open = snd_usX2Y_usbpcm_open, .close = snd_usX2Y_usbpcm_close, diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index c19a5dd..71a0e9e 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -890,7 +890,7 @@ static snd_pcm_uframes_t playback_pcm_pointer(struct snd_pcm_substream *subs) return ua101_pcm_pointer(ua, &ua->playback); } -static struct snd_pcm_ops capture_pcm_ops = { +static const struct snd_pcm_ops capture_pcm_ops = { .open = capture_pcm_open, .close = capture_pcm_close, .ioctl = snd_pcm_lib_ioctl, @@ -903,7 +903,7 @@ static struct snd_pcm_ops capture_pcm_ops = { .mmap = snd_pcm_lib_mmap_vmalloc, }; -static struct snd_pcm_ops playback_pcm_ops = { +static const struct snd_pcm_ops playback_pcm_ops = { .open = playback_pcm_open, .close = playback_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 8f66ba7..af5ad84 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -338,7 +338,7 @@ unlock: } /* operators for both playback and capture */ -static struct snd_pcm_ops snd_usb_caiaq_ops = { +static const struct snd_pcm_ops snd_usb_caiaq_ops = { .open = snd_usb_caiaq_substream_open, .close = snd_usb_caiaq_substream_close, .ioctl = snd_pcm_lib_ioctl,

2 1

[alsa-devel] [PATCH] ASoC: constify snd_pcm_ops structures
by Julia Lawall 08 Sep '16

08 Sep '16

Check for snd_pcm_ops structures that are only stored in the ops field of a snd_soc_platform_driver structure or passed as the third argument to snd_pcm_set_ops. The corresponding field or parameter is declared const, so snd_pcm_ops structures that have this property can be declared as const also. 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_pcm_ops i@p = { ... }; @ok1@ identifier r.i; struct snd_soc_platform_driver e; position p; @@ e.ops = &i@p; @ok2@ identifier r.i; expression e1, e2; position p; @@ snd_pcm_set_ops(e1, e2, &i@p) @bad@ position p != {r.p,ok1.p,ok2.p}; identifier r.i; struct snd_pcm_ops e; @@ e@i@p @depends on !bad disable optional_qualifier@ identifier r.i; @@ static +const struct snd_pcm_ops i = { ... }; // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/soc-utils.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 3154525..9e4094e 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -871,7 +871,7 @@ out: return ret; } -static struct snd_pcm_ops hsw_pcm_ops = { +static const struct snd_pcm_ops hsw_pcm_ops = { .open = hsw_pcm_open, .close = hsw_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index da861b4..91b7069 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -381,7 +381,7 @@ static int atmel_pcm_close(struct snd_pcm_substream *substream) return 0; } -static struct snd_pcm_ops atmel_pcm_ops = { +static const struct snd_pcm_ops atmel_pcm_ops = { .open = atmel_pcm_open, .close = atmel_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index a144c14..e2ff538 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -372,7 +372,7 @@ static int lpass_platform_pcmops_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -static struct snd_pcm_ops lpass_platform_pcm_ops = { +static const struct snd_pcm_ops lpass_platform_pcm_ops = { .open = lpass_platform_pcmops_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = lpass_platform_pcmops_hw_params, diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 77ff8eb..09103aa 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -236,7 +236,7 @@ static snd_pcm_uframes_t rt5514_spi_pcm_pointer( return bytes_to_frames(runtime, rt5514_dsp->dma_offset); } -static struct snd_pcm_ops rt5514_spi_pcm_ops = { +static const struct snd_pcm_ops rt5514_spi_pcm_ops = { .open = rt5514_spi_pcm_open, .hw_params = rt5514_spi_hw_params, .hw_free = rt5514_spi_hw_free, diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index d1fb035..504c7cd 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -897,7 +897,7 @@ static int acp_dma_close(struct snd_pcm_substream *substream) return 0; } -static struct snd_pcm_ops acp_dma_ops = { +static const struct snd_pcm_ops acp_dma_ops = { .open = acp_dma_open, .close = acp_dma_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 52ed434..25c6d87 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -670,7 +670,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer return str_info->buffer_ptr; } -static struct snd_pcm_ops sst_platform_ops = { +static const struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, .ioctl = snd_pcm_lib_ioctl, .trigger = sst_platform_pcm_trigger, diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 77bfd40..86c125f 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1093,7 +1093,7 @@ static int skl_get_time_info(struct snd_pcm_substream *substream, return 0; } -static struct snd_pcm_ops skl_platform_ops = { +static const struct snd_pcm_ops skl_platform_ops = { .open = skl_platform_open, .ioctl = snd_pcm_lib_ioctl, .trigger = skl_platform_pcm_trigger, diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index dbfdfe9..dafd22e 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -242,7 +242,7 @@ static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream return count; } -static struct snd_pcm_ops kirkwood_dma_ops = { +static const struct snd_pcm_ops kirkwood_dma_ops = { .open = kirkwood_dma_open, .close = kirkwood_dma_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 53dd085..393e8f0 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -80,7 +80,7 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops dummy_dma_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index ffc000b..dc30d78 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -322,7 +322,7 @@ static snd_pcm_uframes_t fsl_asrc_dma_pcm_pointer(struct snd_pcm_substream *subs return bytes_to_frames(substream->runtime, pair->pos); } -static struct snd_pcm_ops fsl_asrc_dma_pcm_ops = { +static const struct snd_pcm_ops fsl_asrc_dma_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = fsl_asrc_dma_hw_params, .hw_free = fsl_asrc_dma_hw_free,

1 0

[alsa-devel] [PATCH] ALSA: squash lines for simple wrapper functions
by Masahiro Yamada 07 Sep '16

07 Sep '16

Remove unneeded variables and assignments. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- sound/aoa/fabrics/layout.c | 7 +------ sound/pci/asihpi/hpifunc.c | 7 ++----- sound/pci/ctxfi/ctvmem.c | 6 +----- sound/pci/emu10k1/p16v.c | 16 ++++------------ sound/pci/ice1712/ice1724.c | 4 +--- sound/ppc/snd_ps3.c | 4 +--- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 12 +++--------- 7 files changed, 13 insertions(+), 43 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 8f71f7e..b6ba091 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1161,12 +1161,7 @@ static struct soundbus_driver aoa_soundbus_driver = { static int __init aoa_fabric_layout_init(void) { - int err; - - err = soundbus_register_driver(&aoa_soundbus_driver); - if (err) - return err; - return 0; + return soundbus_register_driver(&aoa_soundbus_driver); } static void __exit aoa_fabric_layout_exit(void) diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 510e56c..f9b5764 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -2323,11 +2323,8 @@ u16 hpi_sample_clock_get_source_index(u32 h_control, u16 *pw_source_index) u16 hpi_sample_clock_query_local_rate(const u32 h_clock, const u32 index, u32 *prate) { - u16 err; - err = hpi_control_query(h_clock, HPI_SAMPLECLOCK_LOCAL_SAMPLERATE, - index, 0, prate); - - return err; + return hpi_control_query(h_clock, HPI_SAMPLECLOCK_LOCAL_SAMPLERATE, + index, 0, prate); } u16 hpi_sample_clock_set_local_rate(u32 h_control, u32 sample_rate) diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 419306e..520e19b 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -166,11 +166,7 @@ static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block) static dma_addr_t ct_get_ptp_phys(struct ct_vm *vm, int index) { - dma_addr_t addr; - - addr = (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr; - - return addr; + return (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr; } int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci) diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 3c60b43..87f6622 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -300,37 +300,29 @@ static int snd_p16v_pcm_open_capture(struct snd_pcm_substream *substream) static int snd_p16v_pcm_hw_params_playback(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - int result; - result = snd_pcm_lib_malloc_pages(substream, + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); - return result; } /* hw_params callback */ static int snd_p16v_pcm_hw_params_capture(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - int result; - result = snd_pcm_lib_malloc_pages(substream, + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); - return result; } /* hw_free callback */ static int snd_p16v_pcm_hw_free_playback(struct snd_pcm_substream *substream) { - int result; - result = snd_pcm_lib_free_pages(substream); - return result; + return snd_pcm_lib_free_pages(substream); } /* hw_free callback */ static int snd_p16v_pcm_hw_free_capture(struct snd_pcm_substream *substream) { - int result; - result = snd_pcm_lib_free_pages(substream); - return result; + return snd_pcm_lib_free_pages(substream); } diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 0b22c00..4731e9c 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -620,9 +620,7 @@ static const unsigned int stdclock_rate_list[16] = { static unsigned int stdclock_get_rate(struct snd_ice1712 *ice) { - unsigned int rate; - rate = stdclock_rate_list[inb(ICEMT1724(ice, RATE)) & 15]; - return rate; + return stdclock_rate_list[inb(ICEMT1724(ice, RATE)) & 15]; } static void stdclock_set_rate(struct snd_ice1712 *ice, unsigned int rate) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 3682425..b84d7d3 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -564,9 +564,7 @@ static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream, static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream) { - int ret; - ret = snd_pcm_lib_free_pages(substream); - return ret; + return snd_pcm_lib_free_pages(substream); }; static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream, diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 5bbaa66..c8455b4 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -184,15 +184,9 @@ struct sst_byt { static inline u64 sst_byt_header(int msg_id, int data, bool large, int str_id) { - u64 header; - - header = IPC_HEADER_MSG_ID(msg_id) | - IPC_HEADER_STR_ID(str_id) | - IPC_HEADER_LARGE(large) | - IPC_HEADER_DATA(data) | - SST_BYT_IPCX_BUSY; - - return header; + return IPC_HEADER_MSG_ID(msg_id) | IPC_HEADER_STR_ID(str_id) | + IPC_HEADER_LARGE(large) | IPC_HEADER_DATA(data) | + SST_BYT_IPCX_BUSY; } static inline u16 sst_byt_header_msg_id(u64 header) -- 1.9.1

2 1

[alsa-devel] [PATCH v4 0/6] ASoC: Add support to Qualcomm msm8916-wcd multi codec
by Srinivas Kandagatla 06 Sep '16

06 Sep '16

This patchset aims at adding msm8916-wcd multi codec support. msm8916-wcd codec is found in Qualcomm msm8916 and apq8016 processors. This codec IP is split in to two parts(Digital & Analog), Analog part is integrated in to PMIC PM8916 and the digital part is integrated into Application processor. Register access to the analog part is done via SPMI interface to PMIC, and registers on the Application processor are memory mapped. Data transfer between Analog and Digital Die is done via a internal bus called PDM. Most of this driver is ported from downstream Andriod v3.10 kernel. Jack support and Headset button detection code is not added in to this series. First two patches add digital and analog codec support and the follow on patches add routes and multicodec support to the sound card and finally dts patch for board. This codec support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary Mic. - 2 Digital Microphones. - 2 Mic Bias Circuits. - Earpiece - Headset - Loud Speaker. - Jack Detect. - Headset Button detection circuit. Tested this code on v4.8-rc3 with for Headset (Mic and speakers), Secondary Mic and Loudspeaker on DB410C. DMIC should work too. Thanks, srini Changes since v3: - Split the codec to 2 seperate codecs digital and analog, suggested by Mark and Kenneth. - Addressed various suggestions from Mark and Kenneth. Srinivas Kandagatla (6): ASoC: codecs: Add msm8916-wcd analog codec ASoC: codecs: Add msm8916-wcd digital codec ASoC: qcom: apq8016-sbc: add board specific dapm mic widgets ASoC: qcom: apq8016-sbc: add support to routing via DT ASoC: qcom: apq8016-sbc: Add support to multi codec. arm64: dts: apq8016-sbc: add analog audio support with multicodec .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 30 + .../bindings/sound/qcom,msm8916-wcd-analog.txt | 83 ++ .../bindings/sound/qcom,msm8916-wcd-digital.txt | 20 + arch/arm64/boot/dts/qcom/apq8016-sbc.dtsi | 63 ++ arch/arm64/boot/dts/qcom/msm8916.dtsi | 9 + arch/arm64/boot/dts/qcom/pm8916.dtsi | 45 +- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 5 +- sound/soc/codecs/msm8916-wcd-analog.c | 888 +++++++++++++++++++ sound/soc/codecs/msm8916-wcd-digital.c | 954 +++++++++++++++++++++ sound/soc/qcom/apq8016_sbc.c | 31 +- 11 files changed, 2125 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt create mode 100644 sound/soc/codecs/msm8916-wcd-analog.c create mode 100644 sound/soc/codecs/msm8916-wcd-digital.c -- 2.7.4

2 14

[alsa-devel] [PATCH] ASoC: Intel: Atom: fix IOSF_MBI dependency
by Pierre-Louis Bossart 06 Sep '16

06 Sep '16

fix issue reported by 0-Day with randconfig The commit a68bc0d43e1b ("ASoC: Intel: Atom: auto-detection of Baytrail-CR") added a dependency on IOSF_MBI. The code used the IS_ENABLED macro to check for this dependency but this is not enough in case the SST driver is built-in. This could be fixed with IS_REACHABLE but Baytrail-CR would not be detected depending on combinations of options Add dependency in Kconfig to solve this for good Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a20c3df..fc6e780 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -25,6 +25,7 @@ config SND_SST_IPC_ACPI tristate select SND_SST_IPC select SND_SOC_INTEL_SST + select IOSF_MBI config SND_SOC_INTEL_SST tristate -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: rockchip: spdif: restore register during runtime_suspend/resume cycle
by Sugar Zhang 06 Sep '16

06 Sep '16

when step into runtime_suspend, spdif pd will be disabled and loss state. so need to restore register when runtime_resume. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - remove system suspend/resume and restore register in runtime_suspend/resume per Doug Anderson's suggestion sound/soc/rockchip/rockchip_spdif.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 4ca2657..0eb8d49 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -65,6 +65,7 @@ static int __maybe_unused rk_spdif_runtime_suspend(struct device *dev) { struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + regcache_cache_only(spdif->regmap, true); clk_disable_unprepare(spdif->mclk); clk_disable_unprepare(spdif->hclk); @@ -74,7 +75,7 @@ static int __maybe_unused rk_spdif_runtime_suspend(struct device *dev) static int __maybe_unused rk_spdif_runtime_resume(struct device *dev) { struct rk_spdif_dev *spdif = dev_get_drvdata(dev); - int ret; + int ret = 0; ret = clk_prepare_enable(spdif->mclk); if (ret) { @@ -88,7 +89,16 @@ static int __maybe_unused rk_spdif_runtime_resume(struct device *dev) return ret; } - return 0; + regcache_cache_only(spdif->regmap, false); + regcache_mark_dirty(spdif->regmap); + + ret = regcache_sync(spdif->regmap); + if (ret) { + clk_disable_unprepare(spdif->mclk); + clk_disable_unprepare(spdif->hclk); + } + + return ret; } static int rk_spdif_hw_params(struct snd_pcm_substream *substream, -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: rockchip: i2s: restore register during runtime_suspend/resume cycle
by Sugar Zhang 06 Sep '16

06 Sep '16

when step into runtime_suspend, i2s pd will be disabled and loss state. so need to restore register when runtime_resume. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - remove system suspend/resume and restore register in runtime_suspend/resume per Doug Anderson's suggestion sound/soc/rockchip/rockchip_i2s.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 652e8c5..98b6822 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -57,6 +57,7 @@ static int i2s_runtime_suspend(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->mclk); return 0; @@ -65,7 +66,7 @@ static int i2s_runtime_suspend(struct device *dev) static int i2s_runtime_resume(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); - int ret; + int ret = 0; ret = clk_prepare_enable(i2s->mclk); if (ret) { @@ -73,7 +74,14 @@ static int i2s_runtime_resume(struct device *dev) return ret; } - return 0; + regcache_cache_only(i2s->regmap, false); + regcache_mark_dirty(i2s->regmap); + + ret = regcache_sync(i2s->regmap); + if (ret) + clk_disable_unprepare(i2s->mclk); + + return ret; } static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 06 Sep '16

06 Sep '16

Here's just the part that allows distinguishing sun4i from sun7i. Since I'm waiting for Chen-Yu's DAPM_DOUBLE patch series I figured I'd post the preliminaries in the mean time which will not change and can be applied independently without problems. Danny Milosavljevic (1): ASoC: sun4i-codec: Distinguish sun4i from sun7i sound/soc/sunxi/sun4i-codec.c | 66 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 54 insertions(+), 12 deletions(-) -- 2.1.4

4 5

[alsa-devel] [PATCH] ALSA: pcm: Fix avail to return error if stream is suspended
by jeeja.kp＠intel.com 06 Sep '16

06 Sep '16

From: Jeeja KP <jeeja.kp(a)intel.com> When the stream is in suspended state some applications wait on "Stream Pipe Error" in response to snd_pcm_avail call to resume the stream. In the current implementation snd_pcm_avail() returns zero when the stream is in suspended state. This causes application to enter in infinite loop for frames to be available. "Stream pipe Error" code is getting returned for read/write call when the stream is in suspended state. Similarly update snd_pcm_avail to return -ESTRPIPE. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- sound/core/pcm_native.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c61fd50..9d33c1e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2637,9 +2637,11 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream) break; /* Fall through */ case SNDRV_PCM_STATE_PREPARED: - case SNDRV_PCM_STATE_SUSPENDED: err = 0; break; + case SNDRV_PCM_STATE_SUSPENDED: + err = -ESTRPIPE; + break; case SNDRV_PCM_STATE_XRUN: err = -EPIPE; break; -- 2.5.0

2 1

[alsa-devel] missing header in last alsa-lib-1.1.2
by DIDIER Philippe 04 Sep '16

04 Sep '16

Since my distribution (Mageia) updated to the last alsa-lib (1.1.2) some packages won't build with it For instance Bristol needs an header that exists in the previous version (1.1.1) and disappeared in the last tarball there was : /alsa-lib-1.1.1.tar.bz2/alsa-lib-1.1.1/include/iatomic.h and it exists no more in the last one Is it a mistake ? Thanks

2 1

[alsa-devel] [PATCH v4 0/2] ASoc Remove deprecated create_singlethread_workqueue
by Bhaktipriya Shridhar 04 Sep '16

04 Sep '16

This patch set removes deprecated create_singlethread_workqueue instances in sound/soc/. Bhaktipriya Shridhar (2): ASoC: tlv320dac33: Remove deprecated create_singlethread_workqueue ASoC: Intel: Remove deprecated create_singlethread_workqueue sound/soc/codecs/tlv320dac33.c | 17 ++++------------- sound/soc/intel/atom/sst/sst.c | 15 +++++---------- 2 files changed, 9 insertions(+), 23 deletions(-) -- 2.1.4

1 2

[alsa-devel] [PATCH v7 1/2] ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem
by Sylwester Nawrocki 03 Sep '16

03 Sep '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 v6: - none. 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

2 2

[alsa-devel] [PATCH] ASoC: Intel: sst: fix to spelling mistake: "susupend" -> "suspend"
by Colin King 03 Sep '16

03 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err message and replace "cant" with "can't" Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst/sst.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 498b5f7..9b6e273 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -442,7 +442,7 @@ static int intel_sst_suspend(struct device *dev) struct stream_info *stream = &ctx->streams[i]; if (stream->status == STREAM_RUNNING) { - dev_err(dev, "stream %d is running, cant susupend, abort\n", i); + dev_err(dev, "stream %d is running, can't suspend, abort\n", i); return -EBUSY; } } -- 2.9.3

2 1

[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] ASoC: fsl_esai: fix spelling mistake "Transmition" -> "Transmission"
by Colin King 02 Sep '16

02 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistakes in dev_dbg messages Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/fsl_esai.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 26a90e1..e927955 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -77,19 +77,19 @@ static irqreturn_t esai_isr(int irq, void *devid) regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr); if (esr & ESAI_ESR_TINIT_MASK) - dev_dbg(&pdev->dev, "isr: Transmition Initialized\n"); + dev_dbg(&pdev->dev, "isr: Transmission Initialized\n"); if (esr & ESAI_ESR_RFF_MASK) dev_warn(&pdev->dev, "isr: Receiving overrun\n"); if (esr & ESAI_ESR_TFE_MASK) - dev_warn(&pdev->dev, "isr: Transmition underrun\n"); + dev_warn(&pdev->dev, "isr: Transmission underrun\n"); if (esr & ESAI_ESR_TLS_MASK) dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); if (esr & ESAI_ESR_TDE_MASK) - dev_dbg(&pdev->dev, "isr: Transmition data exception\n"); + dev_dbg(&pdev->dev, "isr: Transmission data exception\n"); if (esr & ESAI_ESR_TED_MASK) dev_dbg(&pdev->dev, "isr: Transmitting even slots\n"); -- 2.9.3

2 1

[alsa-devel] [PATCH v6 1/2] ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem
by Sylwester Nawrocki 02 Sep '16

02 Sep '16

This patch adds DT binding documentation for Exynos5433 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 v5: - none. 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

1 2

[alsa-devel] [PATCH] ASoC: ux500: fix spelling mistake "Unsopported" -> "Unsupported"
by Colin King 02 Sep '16

02 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistakes in dev_err messages Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/ux500/ux500_msp_dai.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 6d5698b..b343efd 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -187,7 +187,7 @@ static int setup_clocking(struct snd_soc_dai *dai, default: dev_err(dai->dev, - "%s: Error: Unsopported inversion (fmt = 0x%x)!\n", + "%s: Error: Unsupported inversion (fmt = 0x%x)!\n", __func__, fmt); return -EINVAL; @@ -218,7 +218,7 @@ static int setup_clocking(struct snd_soc_dai *dai, break; default: - dev_err(dai->dev, "%s: Error: Unsopported master (fmt = 0x%x)!\n", + dev_err(dai->dev, "%s: Error: Unsupported master (fmt = 0x%x)!\n", __func__, fmt); return -EINVAL; @@ -374,7 +374,7 @@ static int setup_msp_config(struct snd_pcm_substream *substream, break; default: - dev_err(dai->dev, "%s: Error: Unsopported format (%d)!\n", + dev_err(dai->dev, "%s: Error: Unsupported format (%d)!\n", __func__, fmt); return -EINVAL; } -- 2.9.3

1 0

[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 v2 00/11] ASoC: hdac: Add hdac generic driver
by Subhransu S. Prusty 02 Sep '16

02 Sep '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 changes in v2: - Addressed review comment from Takashi - Moved the widget_list to hdac_ext_device - helper to fill the widget list is now called from hdac_ext_bus_device_init instead of driver probe - Fixed a NULL pointer dereferencing issue - Reorder patch to fix 'auto build test error' - Playback with hs/spk and amic capture now working for rt286 - And some more cleanup and code refactoring to avoid long function definition Subhransu S. Prusty (11): 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 ALSA: hdac: Move 'to_hda_ext_device' helper to core ASoC: hdac: Create DAPM model for HDA widgets 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_ext.h | 40 + sound/hda/ext/Makefile | 3 +- sound/hda/ext/hdac_ext_bus.c | 24 + sound/hda/ext/hdac_ext_codec.c | 156 ++++ sound/hda/ext/hdac_ext_codec.h | 26 + sound/hda/local.h | 13 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_generic.c | 1743 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_generic.h | 36 + sound/soc/codecs/hdac_hdmi.c | 8 +- 11 files changed, 2048 insertions(+), 8 deletions(-) create mode 100644 sound/hda/ext/hdac_ext_codec.c create mode 100644 sound/hda/ext/hdac_ext_codec.h create mode 100644 sound/soc/codecs/hdac_generic.c create mode 100644 sound/soc/codecs/hdac_generic.h -- 1.9.1

1 11

[alsa-devel] [PATCH 0/6] constify snd_pcm_ops structures
by Julia Lawall 02 Sep '16

02 Sep '16

4 11

[alsa-devel] [PATCH] pcm_hw: Respect LIBASOUND_THREAD_SAFE env var
by David Henningsson 02 Sep '16

02 Sep '16

Without this patch, there no way to revert to the old behaviour with pcm_hw, so currently there will always be some extra locking. This seems to be an oversight? Reported-by: Samuel Thibault <samuel.thibault(a)ens-lyon.org> Signed-off-by: David Henningsson <diwic(a)ubuntu.com> --- src/pcm/pcm_hw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 3a5634c..e6a8a4f 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1514,7 +1514,8 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, const char *name, pcm->poll_events = info.stream == SND_PCM_STREAM_PLAYBACK ? POLLOUT : POLLIN; pcm->tstamp_type = tstamp_type; #ifdef THREAD_SAFE_API - pcm->thread_safe = 1; + if (pcm->thread_safe != -1) + pcm->thread_safe = 1; #endif ret = snd_pcm_hw_mmap_status(pcm); -- 1.9.1

3 6

[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] ALSA: cs46xx: fix typo "seconadry" -> "secondary"
by Colin King 01 Sep '16

01 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to typos dev_dbg messages and comment. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/cs46xx/dsp_spos.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 2706f27..528102cc 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2460,7 +2460,7 @@ static int cs46xx_detect_codec(struct snd_cs46xx *chip, int codec) udelay(10); if (snd_cs46xx_codec_read(chip, AC97_RESET, codec) & 0x8000) { dev_dbg(chip->card->dev, - "seconadry codec not present\n"); + "secondary codec not present\n"); return -ENXIO; } } @@ -2503,7 +2503,7 @@ int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device) chip->nr_ac97_codecs = 1; #ifdef CONFIG_SND_CS46XX_NEW_DSP - dev_dbg(chip->card->dev, "detecting seconadry codec\n"); + dev_dbg(chip->card->dev, "detecting secondary codec\n"); /* try detect a secondary codec */ if (! cs46xx_detect_codec(chip, CS46XX_SECONDARY_CODEC_INDEX)) chip->nr_ac97_codecs = 2; diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index d2951ed..4a0cbd2 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -1441,7 +1441,7 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) if (chip->nr_ac97_codecs == 2) { /* create CODEC tasklet for rear Center/LFE output - slot 6 and 9 on seconadry CODEC */ + slot 6 and 9 on secondary CODEC */ clfe_codec_out_scb = cs46xx_dsp_create_codec_out_scb(chip,"CodecOutSCB_CLFE",0x0030,0x0030, CLFE_MIXER_SCB_ADDR, CLFE_CODEC_SCB_ADDR, -- 2.9.3

2 1

[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: davinci-mcasp: off-by-one in 'davinci_mcasp_hw_rule_format'
by Christophe JAILLET 01 Sep '16

01 Sep '16

When calling 'snd_pcm_format_width', SNDRV_PCM_FORMAT_LAST is a valid value, so don't skip it. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Un-tested --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 05c2d33aa74d..3c5a9804d3f5 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1218,7 +1218,7 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, snd_mask_none(&nfmt); - for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { if (snd_mask_test(fmt, i)) { uint sbits = snd_pcm_format_width(i); int ppm; -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: davinci-mcasp: off-by-one in davinci_mcasp_hw_rule_format()
by Peter Ujfalusi 01 Sep '16

01 Sep '16

The SNDRV_PCM_FORMAT_LAST is valid, we should not skip it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e10cfda21d89..31b4eac87aa2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1239,7 +1239,7 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, snd_mask_none(&nfmt); - for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { if (snd_mask_test(fmt, i)) { uint sbits = snd_pcm_format_width(i); int ppm; -- 2.9.3

1 0

[alsa-devel] [PATCH] ASoC: omap-pcm: off-by-one in 'omap_pcm_limit_supported_formats'
by Christophe JAILLET 01 Sep '16

01 Sep '16

When calling 'snd_pcm_format_physical_width', SNDRV_PCM_FORMAT_LAST is a valid value, so don't skip it. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Un-tested --- sound/soc/omap/omap-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index a84f677234f0..94e9ff791f3a 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -58,7 +58,7 @@ static void omap_pcm_limit_supported_formats(void) { int i; - for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { switch (snd_pcm_format_physical_width(i)) { case 8: case 16: -- 2.7.4

2 1

[alsa-devel] [PATCH] test/pcm-multi-thread: Remove pthread_cancel, add snd_pcm_close
by David Henningsson 01 Sep '16

01 Sep '16

Because 1) Apparently, our thread cancellation policy is "don't even think about it" and we shouldn't break our own rules 2) It is good practice to close pcm after usage Signed-off-by: David Henningsson <diwic(a)ubuntu.com> --- test/pcm-multi-thread.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/test/pcm-multi-thread.c b/test/pcm-multi-thread.c index da1b87c..26de7db 100644 --- a/test/pcm-multi-thread.c +++ b/test/pcm-multi-thread.c @@ -219,18 +219,22 @@ int main(int argc, char **argv) return 1; } - if (setup_params()) + if (setup_params()) { + snd_pcm_close(pcm); return 1; + } buf = calloc(1, snd_pcm_format_size(format, bufsize) * channels); if (!buf) { fprintf(stderr, "cannot alloc buffer\n"); + snd_pcm_close(pcm); return 1; } for (i = 0; i < num_threads; i++) { if (pthread_create(&peeper_threads[i], NULL, peeper, (void *)(long)i)) { fprintf(stderr, "pthread_create error\n"); + snd_pcm_close(pcm); return 1; } } @@ -255,9 +259,8 @@ int main(int argc, char **argv) running = 0; for (i = 0; i < num_threads; i++) - pthread_cancel(peeper_threads[i]); - for (i = 0; i < num_threads; i++) pthread_join(peeper_threads[i], NULL); + snd_pcm_close(pcm); return 1; } -- 1.9.1

2 1

[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

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[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(-)

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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

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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

2 26

[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

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[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

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[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

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[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(-)

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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

1 0

[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

2 2

[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(-)

2 1

[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

3 3

[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

3 8

[alsa-devel] [PATCH 0/8] ASoC: simple-scu-card: tidyup and synchronize patches
by Kuninori Morimoto 25 Aug '16

25 Aug '16

Hi Mark These are cleanup/tidyup and synchronize patches for simple-scu-card. We have now simple-card and simple-scu-card. These are almost same, difference is very small. But, because of historical reason, these have "code" difference. This patch cleanup and synchronize these. These are based on mark/topic/rcar Kuninori Morimoto (8): ASoC: simple-scu-card: remove unused definitions ASoC: simple-scu-card: is GPL v2 ASoC: simple-scu-card: tidyup codec daifmt handling method ASoC: simple-scu-card: tidyup asoc_simple_card_parse_links() method ASoC: simple-scu-card: depends on CONFIG_OF ASoC: simple-scu-card: add 2 CPU 1 Codec example ASoC: simple-scu-card: code sync: move asoc_simple_card_of_match ASoC: simple-scu-card: code sync: tidyup TDM setting position .../devicetree/bindings/sound/simple-scu-card.txt | 32 +++++++++- sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-scu-card.c | 72 +++++++++------------- 3 files changed, 60 insertions(+), 45 deletions(-) -- 1.9.1

1 8

[alsa-devel] [PATCH 0/9] ASoC: simple-card: tidyup and synchronize patches
by Kuninori Morimoto 25 Aug '16

25 Aug '16

Hi Mark These are cleanup/tidyup and synchronize patches for simple-card. We have now simple-card and simple-scu-card. These are almost same, difference is 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

1 9

[alsa-devel] [PATCH] ASoC: nau8810: Fix memory leak in nau8810_eq_put error path
by Axel Lin 25 Aug '16

25 Aug '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/nau8810.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index f9bcdc3..e455186 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -221,6 +221,7 @@ static int nau8810_eq_put(struct snd_kcontrol *kcontrol, if (ret) { dev_err(codec->dev, "EQ configuration fail, register: %x ret: %d\n", reg + i, ret); + kfree(data); return ret; } } -- 2.7.4

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: simple-card-utils: add __printf attribute
by Nicolas Iooss 25 Aug '16

25 Aug '16

asoc_simple_card_set_dailink_name() uses devm_kvasprintf() to format some of its arguments. Adding a __printf attribute to this function makes it possible to detect at compile-time errors related to format strings. Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- include/sound/simple_card_utils.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 86088aed9002..3207b1a70d38 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -27,6 +27,7 @@ int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *codec, char *prefix, unsigned int *retfmt); +__printf(3, 4) int asoc_simple_card_set_dailink_name(struct device *dev, struct snd_soc_dai_link *dai_link, const char *fmt, ...); -- 2.9.3

2 2

[alsa-devel] [PATCH] ASoC: tegra_rt5640: Correct a copy and paste typo in the comments
by Nicolin Chen 25 Aug '16

25 Aug '16

This patch corrects a copy and paste typo. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/tegra/tegra_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 773daec..e5ef4e9 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -1,5 +1,5 @@ /* -* tegra_rt5640.c - Tegra machine ASoC driver for boards using WM8903 codec. +* tegra_rt5640.c - Tegra machine ASoC driver for boards using RT5640 codec. * * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. * -- 2.1.4

1 0

[alsa-devel] Spam in the wiki
by Tanu Kaskinen 24 Aug '16

24 Aug '16

I got a notification, because a wiki page that I had created was edited by a spammer. Looking at the recent changes, it seems that there's quite a lot of spam activity: http://www.alsa-project.org/main/index.php/Special:RecentChanges Can the wiki user registrations be restricted? There's already a captcha, but that doesn't seem very effective. -- Tanu

1 0

[alsa-devel] [PATCH] ASoC: rt5659: Enable IRQ output for GPIO1 pin
by Nicolin Chen 24 Aug '16

24 Aug '16

Since it's possible to have an IRQ without enabling JD3, the GPIO1 pin then would remain its default GPIO function which is set as an input direction (seeing from rt5659). Meanwhile, CPU would also listen this connection via its own GPIO input: [input] [input] CPU GPIO <--------> RT5659 GPIO1 The result for the IRQ on the CPU side will be unexpectable. So this patch enables the IRQ output for GPIO1 pin any way as long as there's an IRQ assigned from platform data or DT: [input] [IRQ output] CPU GPIO <--------< RT5659 GPIO1 Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/rt5659.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 4c0d2b8..db54550 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4189,6 +4189,9 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, if (ret) dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + /* Enable IRQ output for GPIO1 pin any way */ + regmap_update_bits(rt5659->regmap, RT5659_GPIO_CTRL_1, + RT5659_GP1_PIN_MASK, RT5659_GP1_PIN_IRQ); } return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659, -- 2.1.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: simple-card-utils: add __printf attribute
by Nicolas Iooss 24 Aug '16

24 Aug '16

asoc_simple_card_set_dailink_name() uses devm_kvasprintf() to format some of its arguments. Adding a __printf attribute to this function makes it possible to detect at compile-time errors related to format strings. Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- include/sound/simple_card_utils.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 86088aed9002..3207b1a70d38 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -27,6 +27,7 @@ int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *codec, char *prefix, unsigned int *retfmt); +__printf(3, 4) int asoc_simple_card_set_dailink_name(struct device *dev, struct snd_soc_dai_link *dai_link, const char *fmt, ...); -- 2.9.3

3 4

[alsa-devel] [PATCH -next] sound: rockchip: fix build when SPI_MASTER is not enabled
by Randy Dunlap 24 Aug '16

24 Aug '16

From: Randy Dunlap <rdunlap(a)infradead.org> Fix multiple build errors in sound/soc/rockchip/ drivers by adding "depends on SPI_MASTER". Fixes these build errors: sound/built-in.o: In function `rt5514_spi_burst_read': (.text+0x6c26c): undefined reference to `spi_sync' sound/built-in.o: In function `rt5514_spi_burst_write': (.text+0x6c96e): undefined reference to `spi_sync' sound/built-in.o: In function `rockchip_sound_probe': rk3399_gru_sound.c:(.text+0x9f579): undefined reference to `spi_bus_type' sound/built-in.o: In function `rt5514_spi_driver_init': rt5514-spi.c:(.init.text+0x983): undefined reference to `__spi_register_driver' Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-rockchip(a)lists.infradead.org --- sound/soc/rockchip/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20160824.orig/sound/soc/rockchip/Kconfig +++ linux-next-20160824/sound/soc/rockchip/Kconfig @@ -45,6 +45,7 @@ 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 SPI_MASTER select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98357A select SND_SOC_RT5514

1 0

[alsa-devel] [PATCH -next] sound: rockchip: fix build when SPI is not enabled
by Randy Dunlap 24 Aug '16

24 Aug '16

From: Randy Dunlap <rdunlap(a)infradead.org> Fix build errors found in rt5514 drivers when SPI is not enabled. Fixes these build errors: ERROR: "__spi_register_driver" [sound/soc/codecs/snd-soc-rt5514-spi.ko] undefined! ERROR: "spi_sync" [sound/soc/codecs/snd-soc-rt5514-spi.ko] undefined! Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/rockchip/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) This builds without errors, but if you always want to have the SPI interface available in this driver, change the first "depends" to include "&& SPI_MASTER". --- linux-next-20160822.orig/sound/soc/rockchip/Kconfig +++ linux-next-20160822/sound/soc/rockchip/Kconfig @@ -49,7 +49,7 @@ config SND_SOC_RK3399_GRU_SOUND select SND_SOC_MAX98357A select SND_SOC_RT5514 select SND_SOC_DA7219 - select SND_SOC_RT5514_SPI + select SND_SOC_RT5514_SPI if SPI_MASTER help Say Y or M here if you want to add support multiple codecs for SoC audio on Rockchip RK3399 GRU boards.

1 1

[alsa-devel] Questions about DAPM Routing for STA309A DSP Driver
by Mauricio Cirelli 24 Aug '16

24 Aug '16

Dear ALSA Community, I am new to this community and have a basic knowledge on ASoC architecture. I have worked a few times with Freescale's boards and the SGTL5000 Codec driver. Now, time has come for me to develop a ASoC Codec Driver for an Audio DSP (ST Microelectronics STA309A). Obviously, this driver should work only on the Playback stream. This chip supports up to 8 playback channels (I2S format) and has up to 8 output channels. There are also 8 internal channels, whose inputs can be one of the 8 playback channels and outputs are connected to one of each 8 output channels like this: Playback Channel X --> Input Channel Y --> Output Channel Z This DSP has several controls to apply effects and modify volume on each channel internal channel individually. I have been able to implement snd_kcontrol_new for them successfully and I am able to control them from the userspace correctly. The part I am still in doubt on how to proceed is related to this channel routing. From what I have researched, this could be designed on top of DAPM Widgets and DAPM Routes in a way that the machine-level driver could configure this routing according to the specific board and the userspace would not be required to deal with this stuff. I want to develop a device driver following the ASoC architecture so I can use it in different boards and in a way it could be useful for other developers. Here is what I was sketching up for this snd_soc_codec_driver: static const char *input_names[8] = { "DIN1", "DIN2", "DIN3", "DIN4", "DIN5", "DIN6", "DIN7", "DIN8" }; static const char *internal_names[8] = { "INT1", "INT2", "INT3", "INT4", "INT5", "INT6", "INT7", "INT8" }; /* Playback Channel to Internal DSP Channel Mapping */ static SOC_ENUM_SINGLE_DECL(int1mux, STA309A_C12IM, 0, input_names); static const struct snd_kcontrol_new int1mux_ctrl = SOC_DAPM_ENUM("INT1MUX", int1mux); .... replicated to all 8 internal channels /* Internal DSP Channel to Output Channel Mapping */ static SOC_ENUM_SINGLE_DECL(out1mux, STA309A_C12OM, 0, internal_names); static const struct snd_kcontrol_new out1mux_ctrl = SOC_DAPM_ENUM("OUT1MUX", out1mux); .... replicated to all 8 output channels static const struct snd_soc_dapm_widget sta309a_dsp_dapm_widgets[] = { SND_SOC_DAPM_MUX("INT1MUX", SND_SOC_NOPM, 0, 0, &int1mux_ctrl), .... added 8 internal channel muxes SND_SOC_DAPM_MUX("OUT1MUX", SND_SOC_NOPM, 0, 0, &out1mux_ctrl), .... added 8 output channel muxes }; Now I am lost on how to declare the possible routes. It should be possible to route each playback channel to any of the 8 possible internal channels and the internal channels to any of the 8 possible output channels. Note that this DSP supports Cross-Overs, so one can route a playback channel to one or more internal channels. Analogously, one can route an internal channel to one or more output channels. I have not started to develop the machine-level driver, but any thoughts on how to design it would be very helpful. In the machine-level driver, I would like to define the DSP routing for an specific board. How can I do it? Thank you very much. Mauricio Cirelli

1 0

[alsa-devel] [PATCH 0/2] ALSA: bebob: merge vendor-dependent codes for Yamaha/Terratec
by Takashi Sakamoto 24 Aug '16

24 Aug '16

Hi, This patchset merges vendor-dependent codes for Yamaha/Terratec, because each of these two pairs has the same configuration: - Yamaha GO44 / Terratec Phase 24 FW - Yamaha GO46 / Terratec Phase X24 FW The two companies might once have cooperated to develop some audio and music units on IEEE 1394 bus, based on DM1000E ASIC and BeBoB firmware. This conclusion comes from a reporter of a forum of Ubuntu Japanese Community. Thanks for his help to identify subunit formation and supported stream formats: Jack & Firewireオーディオインターフェイスの設定に関する質問です。 https://forums.ubuntulinux.jp/viewtopic.php?id=18657 Terratec has also produced the other models with DM ASIC series and BeBoB firmware. For the models, vendor-dependent codes are still left. Takashi Sakamoto (2): ALSA: bebob: rename file with vendor-dependent code so that it's for Yamaha/Terratec ALSA: bebob: unify configurations for some models produced by Yamaha/Terratec sound/firewire/bebob/Makefile | 5 ++-- sound/firewire/bebob/bebob.c | 8 +++--- sound/firewire/bebob/bebob.h | 3 +-- sound/firewire/bebob/bebob_terratec.c | 31 ---------------------- .../{bebob_yamaha.c => bebob_yamaha_terratec.c} | 9 ++++--- 5 files changed, 14 insertions(+), 42 deletions(-) rename sound/firewire/bebob/{bebob_yamaha.c => bebob_yamaha_terratec.c} (85%) -- 2.7.4

2 3

[alsa-devel] [PATCH -next] ASoC: samsung: i2s: Add missing clk_disable_unprepare() on error in samsung_i2s_probe()
by Wei Yongjun 24 Aug '16

24 Aug '16

Add the missing clk_disable_unprepare() before return from samsung_i2s_probe() in the error handling case. Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/samsung/i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index fa3ff03..7e32cf4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1318,7 +1318,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); - return -ENOMEM; + ret = -ENOMEM; + goto err_disable_clk; } sec_dai->lock = &pri_dai->spinlock; @@ -1342,7 +1343,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); - return -EINVAL; + ret = -EINVAL; + goto err_disable_clk; } ret = devm_snd_soc_register_component(&pri_dai->pdev->dev, @@ -1366,6 +1368,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) err_free_dai: if (sec_dai) i2s_free_sec_dai(sec_dai); +err_disable_clk: + clk_disable_unprepare(pri_dai->clk); return ret; }

3 2

[alsa-devel] [PATCH] ASoC: tas5086: fix typo: "Inavlid" -> "Invalid"
by Colin King 24 Aug '16

24 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> trivial typo fix in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index c297b9f..b7de857 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -387,7 +387,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), priv->mclk / priv->rate); if (val < 0) { - dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + dev_err(codec->dev, "Invalid MCLK / Fs ratio\n"); return -EINVAL; } -- 2.9.3

2 1

[alsa-devel] [PATCH -next] ASoC: omap-abe-twl6040: Use devm_snd_soc_register_card()
by Wei Yongjun 24 Aug '16

24 Aug '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Using devm_snd_soc_register_card() can make the code shorter and cleaner. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/omap/omap-abe-twl6040.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index f61b3b5..89fe95e 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -305,23 +305,14 @@ static int omap_abe_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", ret); return ret; } -static int omap_abe_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id omap_abe_of_match[] = { {.compatible = "ti,abe-twl6040", }, { }, @@ -335,7 +326,6 @@ static struct platform_driver omap_abe_driver = { .of_match_table = omap_abe_of_match, }, .probe = omap_abe_probe, - .remove = omap_abe_remove, }; static int __init omap_abe_init(void)

3 2

[alsa-devel] [PATCH] ASoC: sunxi: depend on ARCH_SUNXI
by Peter Robinson 24 Aug '16

24 Aug '16

Depend on ARCH_SUNXI or COMPILE_TEST as it doesn't make sense to enable these if the appropriate SoC they're embedded in isn't enabled. Signed-off-by: Peter Robinson <pbrobinson(a)gmail.com> --- sound/soc/sunxi/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 2a954bd..a2b62dc5 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -2,6 +2,7 @@ menu "Allwinner SoC Audio support" config SND_SUN4I_CODEC tristate "Allwinner A10 Codec Support" + depends on ARCH_SUNXI || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help @@ -10,6 +11,7 @@ config SND_SUN4I_CODEC config SND_SUN4I_I2S tristate "Allwinner A10 I2S Support" + depends on ARCH_SUNXI || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help @@ -19,7 +21,7 @@ config SND_SUN4I_I2S config SND_SUN4I_SPDIF tristate "Allwinner A10 SPDIF Support" - depends on OF + depends on OF && (ARCH_SUNXI || COMPILE_TEST) select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help -- 2.9.3

4 5

[alsa-devel] [PATCH v2 0/4] ASoC: rsrc-card to simple-scu-card
by Kuninori Morimoto 24 Aug '16

24 Aug '16

Hi Mark Thesee are v2 of rsrc-card to simple-scu-card. rsrc-card was created for DPCM sound card for Renesas, but we can reuse this driver as more generic purpose. This driver is supporting Multi CPU and Single Codec DPCM system. Kuninori Morimoto (4): ASoC: rsrc-card: remove board specific compatible ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase1 ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase2 ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase3 .../bindings/sound/renesas,rsrc-card.txt | 75 --------- .../devicetree/bindings/sound/simple-scu-card.txt | 80 ++++++++++ sound/soc/generic/Kconfig | 7 + sound/soc/generic/Makefile | 2 + .../rcar/rsrc-card.c => generic/simple-scu-card.c} | 172 +++++++++------------ sound/soc/sh/Kconfig | 6 - sound/soc/sh/rcar/Makefile | 3 - 7 files changed, 162 insertions(+), 183 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-scu-card.txt rename sound/soc/{sh/rcar/rsrc-card.c => generic/simple-scu-card.c} (57%) -- 1.9.1

2 7

Re: [alsa-devel] good ideas
by lemeunier gilles 24 Aug '16

24 Aug '16

Hi! I have some brilliant ideas regarding the issues we've discussed last time, please read 'em here <http://ndicrugafru.whonue.net/e6wutsxl> Pardon my monkey thumbs, lemeunier gilles

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - fix headset mic detection for MSI MS-B120
by Anisse Astier 24 Aug '16

24 Aug '16

MSI Cubi MS-B120 needs the same fixup as the Gigabyte BXBT-2807 for its mic to work. They both use a single 3-way jack for both mic and headset with an ALC283 codec, with the same pins used. Cc: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- sound/pci/hda/patch_realtek.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 574b1b4..7100f05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4828,7 +4828,7 @@ enum { ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, ALC292_FIXUP_TPT440, - ALC283_FIXUP_BXBT2807_MIC, + ALC283_FIXUP_HEADSET_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, ALC280_FIXUP_HP_GPIO4, @@ -5321,7 +5321,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC292_FIXUP_TPT440_DOCK, }, - [ALC283_FIXUP_BXBT2807_MIC] = { + [ALC283_FIXUP_HEADSET_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x04a110f0 }, @@ -5651,7 +5651,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), - SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC), + SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), -- 2.5.5

2 1

[alsa-devel] [PATCH] ASoC: sunxi: depend on ARCH_SUNXI
by Peter Robinson 23 Aug '16

23 Aug '16

Depend on ARCH_SUNXI or COMPILE_TEST as it doesn't make sense to enable these if the appropriate SoC they're embedded in isn't enabled. Signed-off-by: Peter Robinson <pbrobinson(a)gmail.com> --- sound/soc/sunxi/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 2a954bd..dd23682 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -1,4 +1,5 @@ menu "Allwinner SoC Audio support" + depends on ARCH_SUNXI || COMPILE_TEST config SND_SUN4I_CODEC tristate "Allwinner A10 Codec Support" -- 2.9.3

1 0

[alsa-devel] CVE-2014-9892 fix is not upstream
by Luis Henriques 23 Aug '16

23 Aug '16

Hi! Digging through some old CVEs I came across this one that doesn't seem be in mainline. Was there a good reason for not being sent upstream? Maybe it was rejected for some reason and I failed to find the discussion. References: https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2014-9892 https://source.android.com/security/bulletin/2016-08-01.html https://source.codeaurora.org/quic/la/kernel/msm-3.10/commit/?id=591b1f455c… Cheers, -- Luís

3 3

[alsa-devel] [PATCH v5 0/2] amidi delay option
by Felipe F. Tonello 23 Aug '16

23 Aug '16

Small improvement to amidi, adding useful functionality. 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 | 61 +++++++++++++++++++++++++++++++++++++++++++---------------- 2 files changed, 59 insertions(+), 16 deletions(-) -- 2.9.3

5 13

[alsa-devel] [PATCH] ASoC: nau8810: fix compile warning in loopback switch control
by John Hsu 23 Aug '16

23 Aug '16

Thank Stephen Rothwell for the message. The patch is to fix the following compile error. sound/soc/codecs/nau8810.c:441:3: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, &nau8810_loopback), ^ sound/soc/codecs/nau8810.c:441:3: note: (near initialization for 'nau8810_dapm_widgets[11].kcontrol_news') Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 7617dda..f9bcdc3 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -391,10 +391,9 @@ static const struct snd_kcontrol_new nau8810_mic_boost_controls = NAU8810_PMICBSTGAIN_SFT, 0x7, 0); /* Loopback Switch */ -static const struct snd_kcontrol_new nau8810_loopback[] = { +static const struct snd_kcontrol_new nau8810_loopback = SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, - NAU8810_ADDAP_SFT, 1, 0), -}; + NAU8810_ADDAP_SFT, 1, 0); static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) -- 2.6.4

2 1

[alsa-devel] [PATCH for 4.8] ASoC: omap-mcpdm: Fix irq resource handling
by Peter Ujfalusi 23 Aug '16

23 Aug '16

Fixes: ddd17531ad908 ("ASoC: omap-mcpdm: Clean up with devm_* function") Managed irq request will not doing any good in ASoC probe level as it is not going to free up the irq when the driver is unbound from the sound card. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reported-by: Russell King <linux(a)armlinux.org.uk> --- sound/soc/omap/omap-mcpdm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 74d6e6fdcfd0..64609c77a79d 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -394,8 +394,8 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", + (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -420,6 +420,7 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; -- 2.9.2

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Parse manifest data
by Vinod Koul 23 Aug '16

23 Aug '16

From: Shreyas NC <shreyas.nc(a)intel.com> Topology manifest has lib names and lib count info. So, define tokens to represent module private data and parse these tokens to fill up the manifest structure in the driver accordingly. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/snd_sst_tokens.h | 8 +- sound/soc/intel/skylake/skl-topology.c | 194 ++++++++++++++++++++++++++++++++- 2 files changed, 200 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h index f56a932736ca..1ee2e943d66a 100644 --- a/include/uapi/sound/snd_sst_tokens.h +++ b/include/uapi/sound/snd_sst_tokens.h @@ -153,6 +153,10 @@ * * %SKL_TKN_U32_PROC_DOMAIN: Specify processing domain * + * %SKL_TKN_U32_LIB_COUNT: Specifies the number of libraries + * + * %SKL_TKN_STR_LIB_NAME: Specifies the library name + * * module_id and loadable flags dont have tokens as these values will be * read from the DSP FW manifest */ @@ -202,7 +206,9 @@ enum SKL_TKNS { SKL_TKN_U32_CAPS_PARAMS_ID, SKL_TKN_U32_CAPS_SIZE, SKL_TKN_U32_PROC_DOMAIN, - SKL_TKN_MAX = SKL_TKN_U32_PROC_DOMAIN, + SKL_TKN_U32_LIB_COUNT, + SKL_TKN_STR_LIB_NAME, + SKL_TKN_MAX = SKL_TKN_STR_LIB_NAME, }; #endif diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 000482e6e2f7..108ebb9ab329 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2201,6 +2201,197 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, return 0; } +static int skl_tplg_fill_str_mfest_tkn(struct device *dev, + struct snd_soc_tplg_vendor_string_elem *str_elem, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0; + static int ref_count; + + switch (str_elem->token) { + case SKL_TKN_STR_LIB_NAME: + if (ref_count > minfo->lib_count - 1) { + ref_count = 0; + return -EINVAL; + } + + strncpy(minfo->lib[ref_count].name, str_elem->string, + ARRAY_SIZE(minfo->lib[ref_count].name)); + ref_count++; + tkn_count++; + break; + + default: + dev_err(dev, "Not a string token %d", str_elem->token); + break; + } + + return tkn_count; +} + +static int skl_tplg_get_str_tkn(struct device *dev, + struct snd_soc_tplg_vendor_array *array, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0, ret; + struct snd_soc_tplg_vendor_string_elem *str_elem; + + str_elem = (struct snd_soc_tplg_vendor_string_elem *)array->value; + while (tkn_count < array->num_elems) { + ret = skl_tplg_fill_str_mfest_tkn(dev, str_elem, minfo); + str_elem++; + + if (ret < 0) + return ret; + + tkn_count = tkn_count + ret; + } + + return tkn_count; +} + +static int skl_tplg_get_int_tkn(struct device *dev, + struct snd_soc_tplg_vendor_value_elem *tkn_elem, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0; + + switch (tkn_elem->token) { + case SKL_TKN_U32_LIB_COUNT: + minfo->lib_count = tkn_elem->value; + tkn_count++; + break; + + default: + dev_err(dev, "Not a manifest token %d", tkn_elem->token); + return -EINVAL; + } + + return tkn_count; +} + +/* + * Fill the manifest structure by parsing the tokens based on the + * type. + */ +static int skl_tplg_get_manifest_tkn(struct device *dev, + char *pvt_data, struct skl_dfw_manifest *minfo, + int block_size) +{ + int tkn_count = 0, ret; + int off = 0, tuple_size = 0; + struct snd_soc_tplg_vendor_array *array; + struct snd_soc_tplg_vendor_value_elem *tkn_elem; + + if (block_size <= 0) + return -EINVAL; + + while (tuple_size < block_size) { + array = (struct snd_soc_tplg_vendor_array *)(pvt_data + off); + off += array->size; + switch (array->type) { + case SND_SOC_TPLG_TUPLE_TYPE_STRING: + ret = skl_tplg_get_str_tkn(dev, array, minfo); + + if (ret < 0) + return ret; + tkn_count += ret; + + tuple_size += tkn_count * + sizeof(struct snd_soc_tplg_vendor_string_elem); + continue; + + case SND_SOC_TPLG_TUPLE_TYPE_UUID: + dev_warn(dev, "no uuid tokens for skl tplf manifest"); + continue; + + default: + tkn_elem = array->value; + tkn_count = 0; + break; + } + + while (tkn_count <= array->num_elems - 1) { + ret = skl_tplg_get_int_tkn(dev, + tkn_elem, minfo); + if (ret < 0) + return ret; + + tkn_count = tkn_count + ret; + tkn_elem++; + tuple_size += tkn_count * + sizeof(struct snd_soc_tplg_vendor_value_elem); + break; + } + tkn_count = 0; + } + + return 0; +} + +/* + * Parse manifest private data for tokens. The private data block is + * preceded by descriptors for type and size of data block. + */ +static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, + struct device *dev, struct skl_dfw_manifest *minfo) +{ + struct snd_soc_tplg_vendor_array *array; + int num_blocks, block_size = 0, block_type, off = 0; + char *data; + int ret; + + /* Read the NUM_DATA_BLOCKS descriptor */ + array = (struct snd_soc_tplg_vendor_array *)manifest->priv.data; + ret = skl_tplg_get_desc_blocks(dev, array); + if (ret < 0) + return ret; + num_blocks = ret; + + off += array->size; + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + /* Read the BLOCK_TYPE and BLOCK_SIZE descriptor */ + while (num_blocks > 0) { + ret = skl_tplg_get_desc_blocks(dev, array); + + if (ret < 0) + return ret; + block_type = ret; + off += array->size; + + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + ret = skl_tplg_get_desc_blocks(dev, array); + + if (ret < 0) + return ret; + block_size = ret; + off += array->size; + + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + data = (manifest->priv.data + off); + + if (block_type == SKL_TYPE_TUPLE) { + ret = skl_tplg_get_manifest_tkn(dev, data, minfo, + block_size); + + if (ret < 0) + return ret; + + --num_blocks; + } else { + return -EINVAL; + } + } + + return 0; +} + static int skl_manifest_load(struct snd_soc_component *cmpnt, struct snd_soc_tplg_manifest *manifest) { @@ -2211,7 +2402,8 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int ret = 0; minfo = &skl->skl_sst->manifest; - memcpy(minfo, manifest->priv.data, sizeof(struct skl_dfw_manifest)); + + skl_tplg_get_manifest_data(manifest, bus->dev, minfo); if (minfo->lib_count > HDA_MAX_LIB) { dev_err(bus->dev, "Exceeding max Library count. Got:%d\n", -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: rename rsrc-card to simple-dpcm-card
by Kuninori Morimoto 23 Aug '16

23 Aug '16

Hi Mark These patches moves current Renesas specific rsrc-card to more generic simple-dpcm-card, but by 1 patch was difficult. So, it uses 3 phases Kuninori Morimoto (3): ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase1 ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase2 ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase3 .../bindings/sound/renesas,rsrc-card.txt | 75 --------- .../devicetree/bindings/sound/simple-dpcm-card.txt | 79 ++++++++++ sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + .../rsrc-card.c => generic/simple-dpcm-card.c} | 173 +++++++++------------ sound/soc/sh/Kconfig | 6 - sound/soc/sh/rcar/Makefile | 3 - 7 files changed, 160 insertions(+), 184 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-dpcm-card.txt rename sound/soc/{sh/rcar/rsrc-card.c => generic/simple-dpcm-card.c} (57%) -- 1.9.1

2 10

[alsa-devel] [PATCH RESEND 0/3] conf: Add vendor tuples in conf
by Vinod Koul 23 Aug '16

23 Aug '16

Topology changes in alsa-lib allow driver data for modules to be be passed in topology conf file using tuples. These changes allow private data to represented by tuples of type UUID, byte, short, word and strings. This patch series adds changes to : - include tuple and token list in the conf file to define module private data. - Private data blobs for modules are no longer required. So, remove the binary blobs and the tool used to generate the blobs. - Then add bxt conf file which is based on tuples Shreyas NC (3): conf: sklrt286: define module private data through tuples in skylake conf: sklrt286: Remove tool to generate private data blobs conf: bxtrt298: Add topology conf file for bxt configure.ac | 2 +- src/conf/topology/Makefile.am | 2 +- src/conf/topology/bxtrt298/Makefile.am | 4 + src/conf/topology/bxtrt298/bxt_i2s.conf | 3323 ++++++++++++++++++++ src/conf/topology/sklrt286/Makefile.am | 3 +- src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/data/Makefile.am | 6 - src/conf/topology/sklrt286/data/pvt.c | 1815 ----------- src/conf/topology/sklrt286/data/pvt_data.c | 90 - src/conf/topology/sklrt286/data/pvt_local.h | 9 - .../topology/sklrt286/data/skl-tplg-interface.h | 232 -- .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/skl_i2s.conf | 2694 +++++++++++++++- 29 files changed, 5988 insertions(+), 2192 deletions(-) create mode 100644 src/conf/topology/bxtrt298/Makefile.am create mode 100644 src/conf/topology/bxtrt298/bxt_i2s.conf delete mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin delete mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin delete mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin delete mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin delete mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin delete mode 100644 src/conf/topology/sklrt286/data/Makefile.am delete mode 100644 src/conf/topology/sklrt286/data/pvt.c delete mode 100644 src/conf/topology/sklrt286/data/pvt_data.c delete mode 100644 src/conf/topology/sklrt286/data/pvt_local.h delete mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h delete mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin delete mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin delete mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin delete mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin delete mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin delete mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin delete mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin delete mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin delete mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin delete mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin -- 1.9.1

1 3

[alsa-devel] [PATCH 0/2] ALSA: compress: Fine-tuning for snd_compr_set_params()
by SF Markus Elfring 23 Aug '16

23 Aug '16

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 21 Aug 2016 21:35:43 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (2): Use memdup_user() Reduce the scope for two variables sound/core/compress_offload.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) -- 2.9.3

6 13

[alsa-devel] [PATCH 0/4] ASoC: Intel: Skylake: Add token support for topology data
by Vinod Koul 23 Aug '16

23 Aug '16

With topology changes in alsa-lib, driver data for modules can now be passed in topology conf file using tuples. These changes allow private data to represented by tuples of type UUID, byte, short, word and strings. This patch series adds changes to: - Define vendor tokens to best describe module private data - Parse these token value to populate module private data - Remove the skl_dfw_config structure as it would no longer be required. - Define tokens for manifest private data and parse them to populate library names for library loading. Shreyas NC (4): ASoC: uapi: Intel: Skylake: Define vendor specific tokens ASoC: Intel: Skylake: Parse vendor tokens to build module data ASoC: Intel: Skylake: Remove dfw config and associated structures ASoC: Intel: Skylake: Parse manifest data include/uapi/sound/Kbuild | 1 + include/uapi/sound/snd_sst_tokens.h | 214 +++++++ sound/soc/intel/skylake/skl-sst-utils.c | 5 - sound/soc/intel/skylake/skl-topology.c | 839 +++++++++++++++++++++++---- sound/soc/intel/skylake/skl-tplg-interface.h | 83 +-- 5 files changed, 953 insertions(+), 189 deletions(-) create mode 100644 include/uapi/sound/snd_sst_tokens.h -- 1.9.1

4 14

[alsa-devel] [PATCH 0/3] conf: Add vendor tuples in conf
by Vinod Koul 23 Aug '16

23 Aug '16

Topology changes in alsa-lib allow driver data for modules to be be passed in topology conf file using tuples. These changes allow private data to represented by tuples of type UUID, byte, short, word and strings. This patch series adds changes to : - include tuple and token list in the conf file to define module private data. - Private data blobs for modules are no longer required. So, remove the binary blobs and the tool used to generate the blobs. - Then add bxt conf file which is based on tuples Takashi, This should be merged after the kernel changes are merged. Shreyas NC (3): conf: sklrt286: define module private data through tuples in skylake conf: sklrt286: Remove tool to generate private data blobs conf: bxtrt298: Add topology conf file for bxt configure.ac | 2 +- src/conf/topology/Makefile.am | 2 +- src/conf/topology/bxtrt298/Makefile.am | 4 + src/conf/topology/bxtrt298/bxt_i2s.conf | 3323 ++++++++++++++++++++ src/conf/topology/sklrt286/Makefile.am | 3 +- src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/data/Makefile.am | 6 - src/conf/topology/sklrt286/data/pvt.c | 1815 ----------- src/conf/topology/sklrt286/data/pvt_data.c | 90 - src/conf/topology/sklrt286/data/pvt_local.h | 9 - .../topology/sklrt286/data/skl-tplg-interface.h | 232 -- .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/skl_i2s.conf | 2694 +++++++++++++++- 29 files changed, 5988 insertions(+), 2192 deletions(-) create mode 100644 src/conf/topology/bxtrt298/Makefile.am create mode 100644 src/conf/topology/bxtrt298/bxt_i2s.conf delete mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin delete mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin delete mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin delete mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin delete mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin delete mode 100644 src/conf/topology/sklrt286/data/Makefile.am delete mode 100644 src/conf/topology/sklrt286/data/pvt.c delete mode 100644 src/conf/topology/sklrt286/data/pvt_data.c delete mode 100644 src/conf/topology/sklrt286/data/pvt_local.h delete mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h delete mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin delete mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin delete mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin delete mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin delete mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin delete mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin delete mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin delete mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin delete mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin delete mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin -- 1.9.1

2 3

[alsa-devel] [PATCH v05 63/72] include/uapi/sound/Kbuild: remove sound/asoc.h from userspace
by Mikko Rapeli 22 Aug '16

22 Aug '16

Based on git commit messages and the #error macro messages it should not be used in userspace. Fixes userspace compile errors: sound/asoc.h:21:2: error: #error This API is an early revision and not enabled in the current #error This API is an early revision and not enabled in the current ^~~~~ sound/asoc.h:22:2: error: #error kernel release, it will be enabled in a futur e kernel version #error kernel release, it will be enabled in a future kernel version ^~~~~ sound/asoc.h:23:2: error: #error with incompatible changes to what is here. #error with incompatible changes to what is here. ^~~~~ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/Kbuild | 1 - 1 file changed, 1 deletion(-) diff --git a/include/uapi/sound/Kbuild b/include/uapi/sound/Kbuild index 691984c..6d8b203 100644 --- a/include/uapi/sound/Kbuild +++ b/include/uapi/sound/Kbuild @@ -1,6 +1,5 @@ # UAPI Header export list header-y += asequencer.h -header-y += asoc.h header-y += asound.h header-y += asound_fm.h header-y += compress_offload.h -- 2.8.1

1 0

[alsa-devel] [PATCH -next] ASoC: max98371: Add terminate entry for i2c_device_id tables
by Wei Yongjun 22 Aug '16

22 Aug '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Make sure i2c_device_id tables are NULL terminated. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/max98371.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 3053879..781be9ba 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -412,6 +412,7 @@ static int max98371_i2c_remove(struct i2c_client *client) static const struct i2c_device_id max98371_i2c_id[] = { { "max98371", 0 }, + { } }; MODULE_DEVICE_TABLE(i2c, max98371_i2c_id);

2 1

[alsa-devel] [PATCH] ASoC: nau8810: Add driver for Nuvoton codec chip NAU88C10
by John Hsu 22 Aug '16

22 Aug '16

The driver is for codec NAU88C10 of Nuvoton Technology Corporation. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8810.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8810.c | 884 +++++++++++++++++++++ sound/soc/codecs/nau8810.h | 281 +++++++ 5 files changed, 1188 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8810.txt create mode 100644 sound/soc/codecs/nau8810.c create mode 100644 sound/soc/codecs/nau8810.h diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt new file mode 100644 index 0000000..05830e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8810.txt @@ -0,0 +1,16 @@ +NAU8810 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8810" + + - reg : the I2C address of the device. + +Example: + +codec: nau8810@1a { + compatible = "nuvoton,nau8810"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1cd6ab3..b3fd110 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -92,6 +92,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C @@ -1064,6 +1065,10 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8810 + tristate "Nuvoton Technology Corporation NAU88C10 CODEC" + depends on I2C + config SND_SOC_NAU8825 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 58036af..ffaa523 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -86,6 +86,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8810-objs := nau8810.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o @@ -307,6 +308,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c new file mode 100644 index 0000000..7617dda --- /dev/null +++ b/sound/soc/codecs/nau8810.c @@ -0,0 +1,884 @@ +/* + * nau8810.c -- NAU8810 ALSA Soc Audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * Based on WM8974.c + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "nau8810.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + + +static const int nau8810_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8810_reg_defaults[] = { + { NAU8810_REG_POWER1, 0x0000 }, + { NAU8810_REG_POWER2, 0x0000 }, + { NAU8810_REG_POWER3, 0x0000 }, + { NAU8810_REG_IFACE, 0x0050 }, + { NAU8810_REG_COMP, 0x0000 }, + { NAU8810_REG_CLOCK, 0x0140 }, + { NAU8810_REG_SMPLR, 0x0000 }, + { NAU8810_REG_DAC, 0x0000 }, + { NAU8810_REG_DACGAIN, 0x00FF }, + { NAU8810_REG_ADC, 0x0100 }, + { NAU8810_REG_ADCGAIN, 0x00FF }, + { NAU8810_REG_EQ1, 0x012C }, + { NAU8810_REG_EQ2, 0x002C }, + { NAU8810_REG_EQ3, 0x002C }, + { NAU8810_REG_EQ4, 0x002C }, + { NAU8810_REG_EQ5, 0x002C }, + { NAU8810_REG_DACLIM1, 0x0032 }, + { NAU8810_REG_DACLIM2, 0x0000 }, + { NAU8810_REG_NOTCH1, 0x0000 }, + { NAU8810_REG_NOTCH2, 0x0000 }, + { NAU8810_REG_NOTCH3, 0x0000 }, + { NAU8810_REG_NOTCH4, 0x0000 }, + { NAU8810_REG_ALC1, 0x0038 }, + { NAU8810_REG_ALC2, 0x000B }, + { NAU8810_REG_ALC3, 0x0032 }, + { NAU8810_REG_NOISEGATE, 0x0000 }, + { NAU8810_REG_PLLN, 0x0008 }, + { NAU8810_REG_PLLK1, 0x000C }, + { NAU8810_REG_PLLK2, 0x0093 }, + { NAU8810_REG_PLLK3, 0x00E9 }, + { NAU8810_REG_ATTEN, 0x0000 }, + { NAU8810_REG_INPUT_SIGNAL, 0x0003 }, + { NAU8810_REG_PGAGAIN, 0x0010 }, + { NAU8810_REG_ADCBOOST, 0x0100 }, + { NAU8810_REG_OUTPUT, 0x0002 }, + { NAU8810_REG_SPKMIX, 0x0001 }, + { NAU8810_REG_SPKGAIN, 0x0039 }, + { NAU8810_REG_MONOMIX, 0x0001 }, + { NAU8810_REG_POWER4, 0x0000 }, + { NAU8810_REG_TSLOTCTL1, 0x0000 }, + { NAU8810_REG_TSLOTCTL2, 0x0020 }, + { NAU8810_REG_DEVICE_REVID, 0x0000 }, + { NAU8810_REG_I2C_DEVICEID, 0x001A }, + { NAU8810_REG_ADDITIONID, 0x00CA }, + { NAU8810_REG_RESERVE, 0x0124 }, + { NAU8810_REG_OUTCTL, 0x0001 }, + { NAU8810_REG_ALC1ENHAN1, 0x0010 }, + { NAU8810_REG_ALC1ENHAN2, 0x0000 }, + { NAU8810_REG_MISCCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFF, 0x0000 }, + { NAU8810_REG_AGCP2POUT, 0x0000 }, + { NAU8810_REG_AGCPOUT, 0x0000 }, + { NAU8810_REG_AMTCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFFMAN, 0x0000 }, +}; + +static bool nau8810_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + case NAU8810_REG_OUTCTL ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_OUTCTL ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + return true; + default: + return false; + } +} + +/* The EQ parameters get function is to get the 5 band equalizer control. + * The regmap raw read can't work here because regmap doesn't provide + * value format for value width of 9 bits. Therefore, the driver reads data + * from cache and makes value format according to the endianness of + * bytes type control element. + */ +static int nau8810_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg, reg_val; + u16 *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8810_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + regmap_read(nau8810->regmap, reg + i, &reg_val); + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, &reg_val, sizeof(reg_val)); + } + + return 0; +} + +/* The EQ parameters put function is to make configuration of 5 band equalizer + * control. These configuration includes central frequency, equalizer gain, + * cut-off frequency, bandwidth control, and equalizer path. + * The regmap raw write can't work here because regmap doesn't provide + * register and value format for register with address 7 bits and value 9 bits. + * Therefore, the driver makes value format according to the endianness of + * bytes type control element and writes data to codec. + */ +static int nau8810_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8810_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + value = be16_to_cpu(*(val + i)); + ret = regmap_write(nau8810->regmap, reg + i, value); + if (ret) { + dev_err(codec->dev, "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8810_companding[] = { + "Off", "NC", "u-law", "A-law" }; + +static const struct soc_enum nau8810_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_ADCCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding); + +static const struct soc_enum nau8810_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_DACCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding); + +static const char * const nau8810_deemp[] = { + "None", "32kHz", "44.1kHz", "48kHz" }; + +static const struct soc_enum nau8810_deemp_enum = + SOC_ENUM_SINGLE(NAU8810_REG_DAC, NAU8810_DEEMP_SFT, + ARRAY_SIZE(nau8810_deemp), nau8810_deemp); + +static const char * const nau8810_eqmode[] = {"Capture", "Playback" }; + +static const struct soc_enum nau8810_eqmode_enum = + SOC_ENUM_SINGLE(NAU8810_REG_EQ1, NAU8810_EQM_SFT, + ARRAY_SIZE(nau8810_eqmode), nau8810_eqmode); + +static const char * const nau8810_alc[] = {"Normal", "Limiter" }; + +static const struct soc_enum nau8810_alc_enum = + SOC_ENUM_SINGLE(NAU8810_REG_ALC3, NAU8810_ALCM_SFT, + ARRAY_SIZE(nau8810_alc), nau8810_alc); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); + +static const struct snd_kcontrol_new nau8810_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8810_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8810_companding_dac_enum), + SOC_ENUM("DAC De-emphasis", nau8810_deemp_enum), + + SOC_ENUM("EQ Function", nau8810_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8810_eq_get, nau8810_eq_put), + + SOC_SINGLE("DAC Inversion Switch", NAU8810_REG_DAC, + NAU8810_DACPL_SFT, 1, 0), + SOC_SINGLE_TLV("Playback Volume", NAU8810_REG_DACGAIN, + NAU8810_DACGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", NAU8810_REG_ADC, + NAU8810_HPFEN_SFT, 1, 0), + SOC_SINGLE("High Pass Cut Off", NAU8810_REG_ADC, + NAU8810_HPF_SFT, 0x7, 0), + + SOC_SINGLE("ADC Inversion Switch", NAU8810_REG_ADC, + NAU8810_ADCPL_SFT, 1, 0), + SOC_SINGLE_TLV("Capture Volume", NAU8810_REG_ADCGAIN, + NAU8810_ADCGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE_TLV("EQ1 Volume", NAU8810_REG_EQ1, + NAU8810_EQ1GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ2 Volume", NAU8810_REG_EQ2, + NAU8810_EQ2GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ3 Volume", NAU8810_REG_EQ3, + NAU8810_EQ3GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ4 Volume", NAU8810_REG_EQ4, + NAU8810_EQ4GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ5 Volume", NAU8810_REG_EQ5, + NAU8810_EQ5GC_SFT, 0x18, 1, eq_tlv), + + SOC_SINGLE("DAC Limiter Switch", NAU8810_REG_DACLIM1, + NAU8810_DACLIMEN_SFT, 1, 0), + SOC_SINGLE("DAC Limiter Decay", NAU8810_REG_DACLIM1, + NAU8810_DACLIMDCY_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Attack", NAU8810_REG_DACLIM1, + NAU8810_DACLIMATK_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Threshold", NAU8810_REG_DACLIM2, + NAU8810_DACLIMTHL_SFT, 0x7, 0), + SOC_SINGLE("DAC Limiter Boost", NAU8810_REG_DACLIM2, + NAU8810_DACLIMBST_SFT, 0xf, 0), + + SOC_ENUM("ALC Mode", nau8810_alc_enum), + SOC_SINGLE("ALC Enable Switch", NAU8810_REG_ALC1, + NAU8810_ALCEN_SFT, 1, 0), + SOC_SINGLE("ALC Max Volume", NAU8810_REG_ALC1, + NAU8810_ALCMXGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC Min Volume", NAU8810_REG_ALC1, + NAU8810_ALCMINGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC ZC Switch", NAU8810_REG_ALC2, + NAU8810_ALCZC_SFT, 1, 0), + SOC_SINGLE("ALC Hold", NAU8810_REG_ALC2, + NAU8810_ALCHT_SFT, 0xf, 0), + SOC_SINGLE("ALC Target", NAU8810_REG_ALC2, + NAU8810_ALCSL_SFT, 0xf, 0), + SOC_SINGLE("ALC Decay", NAU8810_REG_ALC3, + NAU8810_ALCDCY_SFT, 0xf, 0), + SOC_SINGLE("ALC Attack", NAU8810_REG_ALC3, + NAU8810_ALCATK_SFT, 0xf, 0), + SOC_SINGLE("ALC Noise Gate Switch", NAU8810_REG_NOISEGATE, + NAU8810_ALCNEN_SFT, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", NAU8810_REG_NOISEGATE, + NAU8810_ALCNTH_SFT, 0x7, 0), + + SOC_SINGLE("PGA ZC Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAZC_SFT, 1, 0), + SOC_SINGLE_TLV("PGA Volume", NAU8810_REG_PGAGAIN, + NAU8810_PGAGAIN_SFT, 0x3f, 0, inpga_tlv), + + SOC_SINGLE("Speaker ZC Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKZC_SFT, 1, 0), + SOC_SINGLE("Speaker Mute Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKMT_SFT, 1, 0), + SOC_SINGLE_TLV("Speaker Volume", NAU8810_REG_SPKGAIN, + NAU8810_SPKGAIN_SFT, 0x3f, 0, spk_tlv), + + SOC_SINGLE("Capture Boost(+20dB)", NAU8810_REG_ADCBOOST, + NAU8810_PGABST_SFT, 1, 0), + SOC_SINGLE("Mono Mute Switch", NAU8810_REG_MONOMIX, + NAU8810_MOUTMXMT_SFT, 1, 0), + + SOC_SINGLE("DAC Oversampling Rate(128x) Switch", NAU8810_REG_DAC, + NAU8810_DACOS_SFT, 1, 0), + SOC_SINGLE("ADC Oversampling Rate(128x) Switch", NAU8810_REG_ADC, + NAU8810_ADCOS_SFT, 1, 0), +}; + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_SPKMIX, + NAU8810_BYPSPK_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_SPKMIX, + NAU8810_DACSPK_SFT, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_MONOMIX, + NAU8810_BYPMOUT_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_MONOMIX, + NAU8810_DACMOUT_SFT, 1, 0), +}; + +/* PGA Mute */ +static const struct snd_kcontrol_new nau8810_inpga_mute[] = { + SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAMT_SFT, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8810_inpga[] = { + SOC_DAPM_SINGLE("MicN Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_NMICPGA_SFT, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_PMICPGA_SFT, 1, 0), +}; + +/* Mic Input boost vol */ +static const struct snd_kcontrol_new nau8810_mic_boost_controls = + SOC_DAPM_SINGLE("Mic Volume", NAU8810_REG_ADCBOOST, + NAU8810_PMICBSTGAIN_SFT, 0x7, 0); + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8810_loopback[] = { + SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, + NAU8810_ADDAP_SFT, 1, 0), +}; + +static int check_mclk_select_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); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + regmap_read(nau8810->regmap, NAU8810_REG_CLOCK, &value); + return (value & NAU8810_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Speaker Mixer", NAU8810_REG_POWER3, + NAU8810_SPKMX_EN_SFT, 0, &nau8810_speaker_mixer_controls[0], + ARRAY_SIZE(nau8810_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, + NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], + ARRAY_SIZE(nau8810_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + NAU8810_DAC_EN_SFT, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + NAU8810_ADC_EN_SFT, 0), + SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, + NAU8810_NSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", NAU8810_REG_POWER3, + NAU8810_PSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", NAU8810_REG_POWER3, + NAU8810_MOUT_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Input PGA", NAU8810_REG_POWER2, + NAU8810_PGA_EN_SFT, 0, nau8810_inpga, + ARRAY_SIZE(nau8810_inpga)), + SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, + NAU8810_BST_EN_SFT, 0, nau8810_inpga_mute, + ARRAY_SIZE(nau8810_inpga_mute)), + + SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, + NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", NAU8810_REG_POWER1, + NAU8810_PLL_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8810_loopback), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { + {"DAC", NULL, "PLL", check_mclk_select_pll}, + + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Input Boost Stage */ + {"ADC", NULL, "Input Boost Stage"}, + {"ADC", NULL, "PLL", check_mclk_select_pll}, + {"Input Boost Stage", NULL, "Input PGA"}, + {"Input Boost Stage", NULL, "MICP"}, + + /* Input PGA */ + {"Input PGA", NULL, "Mic Bias"}, + {"Input PGA", "MicN Switch", "MICN"}, + {"Input PGA", "MicP Switch", "MICP"}, + + /* Digital Looptack */ + {"Digital Loopback", "Switch", "ADC"}, + {"DAC", NULL, "Digital Loopback"}, +}; + +static int nau8810_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + + nau8810->clk_id = clk_id; + nau8810->sysclk = freq; + dev_dbg(nau8810->dev, "master sysclk %dHz, source %s\n", + freq, clk_id == NAU8810_SCLK_PLL ? "PLL" : "MCLK"); + + return 0; +} + +static int nau88l0_calc_pll(unsigned int pll_in, + unsigned int fs, struct nau8810_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); + for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + if (ARRAY_SIZE(nau8810_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8810_set_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + struct nau8810_pll *pll_param = &nau8810->pll; + int ret, fs; + + fs = freq_out / 256; + ret = nau88l0_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(nau8810->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_info(nau8810->dev, "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, + pll_param->pre_factor); + + regmap_update_bits(map, NAU8810_REG_PLLN, + NAU8810_PLLMCLK_DIV2 | NAU8810_PLLN_MASK, + (pll_param->pre_factor ? NAU8810_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + regmap_write(map, NAU8810_REG_PLLK1, + (pll_param->pll_frac >> NAU8810_PLLK1_SFT) & + NAU8810_PLLK1_MASK); + regmap_write(map, NAU8810_REG_PLLK2, + (pll_param->pll_frac >> NAU8810_PLLK2_SFT) & + NAU8810_PLLK2_MASK); + regmap_write(map, NAU8810_REG_PLLK3, + pll_param->pll_frac & NAU8810_PLLK3_MASK); + regmap_update_bits(map, NAU8810_REG_CLOCK, NAU8810_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8810_MCLKSEL_SFT); + regmap_update_bits(map, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_PLL); + + return 0; +} + +static int nau8810_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + u16 ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8810_CLKIO_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8810_AIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8810_AIFMT_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8810_AIFMT_PCM_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= NAU8810_BCLKP_IB | NAU8810_FSP_IF; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8810_BCLKP_IB; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= NAU8810_FSP_IF; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_AIFMT_MASK | NAU8810_FSP_IF | + NAU8810_BCLKP_IB, ctrl1_val); + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_CLKIO_MASK, ctrl2_val); + + return 0; +} + +static int nau8810_mclk_clkdiv(struct nau8810 *nau8810, int rate) +{ + int i, sclk, imclk = rate * 256, div = 0; + + if (!nau8810->sysclk) { + dev_err(nau8810->dev, "Make mclk div configuration fail because of invalid system clock\n"); + return -EINVAL; + } + + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + for (i = 1; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + sclk = (nau8810->sysclk * 10) / + nau8810_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(nau8810->dev, + "master clock prescaler %x for fs %d\n", div, rate); + + /* master clock from MCLK and disable PLL */ + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_MCLK); + + return 0; +} + +static int nau8810_pcm_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 nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + int val_len = 0, val_rate = 0, ret = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8810_WLEN_20; + break; + case 24: + val_len |= NAU8810_WLEN_24; + break; + case 32: + val_len |= NAU8810_WLEN_32; + break; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8810_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8810_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8810_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8810_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8810_SMPLR_32K; + break; + case 44100: + case 48000: + break; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_WLEN_MASK, val_len); + regmap_update_bits(nau8810->regmap, NAU8810_REG_SMPLR, + NAU8810_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8810->clk_id == NAU8810_SCLK_MCLK) { + ret = nau8810_mclk_clkdiv(nau8810, params_rate(params)); + if (ret < 0) + dev_err(nau8810->dev, "MCLK div configuration fail\n"); + } + + return ret; +} + +static int nau8810_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN); + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + regcache_sync(map); + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_3K); + mdelay(100); + } + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + regmap_write(map, NAU8810_REG_POWER1, 0); + regmap_write(map, NAU8810_REG_POWER2, 0); + regmap_write(map, NAU8810_REG_POWER3, 0); + break; + } + + return 0; +} + + +#define NAU8810_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8810_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8810_ops = { + .hw_params = nau8810_pcm_hw_params, + .set_fmt = nau8810_set_dai_fmt, + .set_sysclk = nau8810_set_sysclk, + .set_pll = nau8810_set_pll, +}; + +static struct snd_soc_dai_driver nau8810_dai = { + .name = "nau8810-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .ops = &nau8810_ops, + .symmetric_rates = 1, +}; + +static const struct regmap_config nau8810_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8810_REG_MAX, + .readable_reg = nau8810_readable_reg, + .writeable_reg = nau8810_writeable_reg, + .volatile_reg = nau8810_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8810_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8810_reg_defaults), +}; + +static struct snd_soc_codec_driver nau8810_codec_driver = { + .set_bias_level = nau8810_set_bias_level, + .suspend_bias_off = true, + + .component_driver = { + .controls = nau8810_snd_controls, + .num_controls = ARRAY_SIZE(nau8810_snd_controls), + .dapm_widgets = nau8810_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8810_dapm_widgets), + .dapm_routes = nau8810_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8810_dapm_routes), + }, +}; + +static int nau8810_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8810 *nau8810 = dev_get_platdata(dev); + + if (!nau8810) { + nau8810 = devm_kzalloc(dev, sizeof(*nau8810), GFP_KERNEL); + if (!nau8810) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8810); + + nau8810->regmap = devm_regmap_init_i2c(i2c, &nau8810_regmap_config); + if (IS_ERR(nau8810->regmap)) + return PTR_ERR(nau8810->regmap); + nau8810->dev = dev; + + regmap_write(nau8810->regmap, NAU8810_REG_RESET, 0x00); + + return snd_soc_register_codec(dev, + &nau8810_codec_driver, &nau8810_dai, 1); +} + +static int nau8810_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id nau8810_i2c_id[] = { + { "nau8810", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8810_of_match[] = { + { .compatible = "nuvoton,nau8810", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8810_of_match); +#endif + +static struct i2c_driver nau8810_i2c_driver = { + .driver = { + .name = "nau8810", + .of_match_table = of_match_ptr(nau8810_of_match), + }, + .probe = nau8810_i2c_probe, + .remove = nau8810_i2c_remove, + .id_table = nau8810_i2c_id, +}; + +module_i2c_driver(nau8810_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8810 driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8810.h b/sound/soc/codecs/nau8810.h new file mode 100644 index 0000000..df88265 --- /dev/null +++ b/sound/soc/codecs/nau8810.h @@ -0,0 +1,281 @@ +/* + * NAU8810 ALSA SoC audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * Author: David Lin <ctlin0(a)nuvoton.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 __NAU8810_H__ +#define __NAU8810_H__ + +#define NAU8810_REG_RESET 0x00 +#define NAU8810_REG_POWER1 0x01 +#define NAU8810_REG_POWER2 0x02 +#define NAU8810_REG_POWER3 0x03 +#define NAU8810_REG_IFACE 0x04 +#define NAU8810_REG_COMP 0x05 +#define NAU8810_REG_CLOCK 0x06 +#define NAU8810_REG_SMPLR 0x07 +#define NAU8810_REG_DAC 0x0A +#define NAU8810_REG_DACGAIN 0x0B +#define NAU8810_REG_ADC 0x0E +#define NAU8810_REG_ADCGAIN 0x0F +#define NAU8810_REG_EQ1 0x12 +#define NAU8810_REG_EQ2 0x13 +#define NAU8810_REG_EQ3 0x14 +#define NAU8810_REG_EQ4 0x15 +#define NAU8810_REG_EQ5 0x16 +#define NAU8810_REG_DACLIM1 0x18 +#define NAU8810_REG_DACLIM2 0x19 +#define NAU8810_REG_NOTCH1 0x1B +#define NAU8810_REG_NOTCH2 0x1C +#define NAU8810_REG_NOTCH3 0x1D +#define NAU8810_REG_NOTCH4 0x1E +#define NAU8810_REG_ALC1 0x20 +#define NAU8810_REG_ALC2 0x21 +#define NAU8810_REG_ALC3 0x22 +#define NAU8810_REG_NOISEGATE 0x23 +#define NAU8810_REG_PLLN 0x24 +#define NAU8810_REG_PLLK1 0x25 +#define NAU8810_REG_PLLK2 0x26 +#define NAU8810_REG_PLLK3 0x27 +#define NAU8810_REG_ATTEN 0x28 +#define NAU8810_REG_INPUT_SIGNAL 0x2C +#define NAU8810_REG_PGAGAIN 0x2D +#define NAU8810_REG_ADCBOOST 0x2F +#define NAU8810_REG_OUTPUT 0x31 +#define NAU8810_REG_SPKMIX 0x32 +#define NAU8810_REG_SPKGAIN 0x36 +#define NAU8810_REG_MONOMIX 0x38 +#define NAU8810_REG_POWER4 0x3A +#define NAU8810_REG_TSLOTCTL1 0x3B +#define NAU8810_REG_TSLOTCTL2 0x3C +#define NAU8810_REG_DEVICE_REVID 0x3E +#define NAU8810_REG_I2C_DEVICEID 0x3F +#define NAU8810_REG_ADDITIONID 0x40 +#define NAU8810_REG_RESERVE 0x41 +#define NAU8810_REG_OUTCTL 0x45 +#define NAU8810_REG_ALC1ENHAN1 0x46 +#define NAU8810_REG_ALC1ENHAN2 0x47 +#define NAU8810_REG_MISCCTL 0x49 +#define NAU8810_REG_OUTTIEOFF 0x4B +#define NAU8810_REG_AGCP2POUT 0x4C +#define NAU8810_REG_AGCPOUT 0x4D +#define NAU8810_REG_AMTCTL 0x4E +#define NAU8810_REG_OUTTIEOFFMAN 0x4F +#define NAU8810_REG_MAX NAU8810_REG_OUTTIEOFFMAN + + +/* NAU8810_REG_POWER1 (0x1) */ +#define NAU8810_DCBUF_EN (0x1 << 8) +#define NAU8810_PLL_EN_SFT 5 +#define NAU8810_MICBIAS_EN_SFT 4 +#define NAU8810_ABIAS_EN (0x1 << 3) +#define NAU8810_IOBUF_EN (0x1 << 2) +#define NAU8810_REFIMP_MASK 0x3 +#define NAU8810_REFIMP_DIS 0x0 +#define NAU8810_REFIMP_80K 0x1 +#define NAU8810_REFIMP_300K 0x2 +#define NAU8810_REFIMP_3K 0x3 + +/* NAU8810_REG_POWER2 (0x2) */ +#define NAU8810_BST_EN_SFT 4 +#define NAU8810_PGA_EN_SFT 2 +#define NAU8810_ADC_EN_SFT 0 + +/* NAU8810_REG_POWER3 (0x3) */ +#define NAU8810_DAC_EN_SFT 0 +#define NAU8810_SPKMX_EN_SFT 2 +#define NAU8810_MOUTMX_EN_SFT 3 +#define NAU8810_PSPK_EN_SFT 5 +#define NAU8810_NSPK_EN_SFT 6 +#define NAU8810_MOUT_EN_SFT 7 + +/* NAU8810_REG_IFACE (0x4) */ +#define NAU8810_AIFMT_SFT 3 +#define NAU8810_AIFMT_MASK (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_RIGHT (0x0 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_LEFT (0x1 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_I2S (0x2 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_PCM_A (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_WLEN_SFT 5 +#define NAU8810_WLEN_MASK (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_16 (0x0 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_20 (0x1 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_24 (0x2 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_32 (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_FSP_IF (0x1 << 7) +#define NAU8810_BCLKP_IB (0x1 << 8) + +/* NAU8810_REG_COMP (0x5) */ +#define NAU8810_ADDAP_SFT 0 +#define NAU8810_ADCCM_SFT 1 +#define NAU8810_DACCM_SFT 3 + +/* NAU8810_REG_CLOCK (0x6) */ +#define NAU8810_CLKIO_MASK 0x1 +#define NAU8810_CLKIO_SLAVE 0x0 +#define NAU8810_CLKIO_MASTER 0x1 +#define NAU8810_BCLKSEL_SFT 2 +#define NAU8810_BCLKSEL_MASK (0x7 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_1 (0x0 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_2 (0x1 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_4 (0x2 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_8 (0x3 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_16 (0x4 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_32 (0x5 << NAU8810_BCLKSEL_SFT) +#define NAU8810_MCLKSEL_SFT 5 +#define NAU8810_MCLKSEL_MASK (0x7 << NAU8810_MCLKSEL_SFT) +#define NAU8810_CLKM_SFT 8 +#define NAU8810_CLKM_MASK (0x1 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_MCLK (0x0 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_PLL (0x1 << NAU8810_CLKM_SFT) + +/* NAU8810_REG_SMPLR (0x7) */ +#define NAU8810_SMPLR_SFT 1 +#define NAU8810_SMPLR_MASK (0x7 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_48K (0x0 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_32K (0x1 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_24K (0x2 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_16K (0x3 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_12K (0x4 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_8K (0x5 << NAU8810_SMPLR_SFT) + +/* NAU8810_REG_DAC (0xA) */ +#define NAU8810_DACPL_SFT 0 +#define NAU8810_DACOS_SFT 3 +#define NAU8810_DEEMP_SFT 4 + +/* NAU8810_REG_DACGAIN (0xB) */ +#define NAU8810_DACGAIN_SFT 0 + +/* NAU8810_REG_ADC (0xE) */ +#define NAU8810_ADCPL_SFT 0 +#define NAU8810_ADCOS_SFT 3 +#define NAU8810_HPF_SFT 4 +#define NAU8810_HPFEN_SFT 8 + +/* NAU8810_REG_ADCGAIN (0xF) */ +#define NAU8810_ADCGAIN_SFT 0 + +/* NAU8810_REG_EQ1 (0x12) */ +#define NAU8810_EQ1GC_SFT 0 +#define NAU8810_EQ1CF_SFT 5 +#define NAU8810_EQM_SFT 8 + +/* NAU8810_REG_EQ2 (0x13) */ +#define NAU8810_EQ2GC_SFT 0 +#define NAU8810_EQ2CF_SFT 5 +#define NAU8810_EQ2BW_SFT 8 + +/* NAU8810_REG_EQ3 (0x14) */ +#define NAU8810_EQ3GC_SFT 0 +#define NAU8810_EQ3CF_SFT 5 +#define NAU8810_EQ3BW_SFT 8 + +/* NAU8810_REG_EQ4 (0x15) */ +#define NAU8810_EQ4GC_SFT 0 +#define NAU8810_EQ4CF_SFT 5 +#define NAU8810_EQ4BW_SFT 8 + +/* NAU8810_REG_EQ5 (0x16) */ +#define NAU8810_EQ5GC_SFT 0 +#define NAU8810_EQ5CF_SFT 5 + +/* NAU8810_REG_DACLIM1 (0x18) */ +#define NAU8810_DACLIMATK_SFT 0 +#define NAU8810_DACLIMDCY_SFT 4 +#define NAU8810_DACLIMEN_SFT 8 + +/* NAU8810_REG_DACLIM2 (0x19) */ +#define NAU8810_DACLIMBST_SFT 0 +#define NAU8810_DACLIMTHL_SFT 4 + +/* NAU8810_REG_ALC1 (0x20) */ +#define NAU8810_ALCMINGAIN_SFT 0 +#define NAU8810_ALCMXGAIN_SFT 3 +#define NAU8810_ALCEN_SFT 8 + +/* NAU8810_REG_ALC2 (0x21) */ +#define NAU8810_ALCSL_SFT 0 +#define NAU8810_ALCHT_SFT 4 +#define NAU8810_ALCZC_SFT 8 + +/* NAU8810_REG_ALC3 (0x22) */ +#define NAU8810_ALCATK_SFT 0 +#define NAU8810_ALCDCY_SFT 4 +#define NAU8810_ALCM_SFT 8 + +/* NAU8810_REG_NOISEGATE (0x23) */ +#define NAU8810_ALCNTH_SFT 0 +#define NAU8810_ALCNEN_SFT 3 + +/* NAU8810_REG_PLLN (0x24) */ +#define NAU8810_PLLN_MASK 0xF +#define NAU8810_PLLMCLK_DIV2 (0x1 << 4) + +/* NAU8810_REG_PLLK1 (0x25) */ +#define NAU8810_PLLK1_SFT 18 +#define NAU8810_PLLK1_MASK 0x3F + +/* NAU8810_REG_PLLK2 (0x26) */ +#define NAU8810_PLLK2_SFT 9 +#define NAU8810_PLLK2_MASK 0x1FF + +/* NAU8810_REG_PLLK3 (0x27) */ +#define NAU8810_PLLK3_MASK 0x1FF + +/* NAU8810_REG_INPUT_SIGNAL (0x2C) */ +#define NAU8810_PMICPGA_SFT 0 +#define NAU8810_NMICPGA_SFT 1 + +/* NAU8810_REG_PGAGAIN (0x2D) */ +#define NAU8810_PGAGAIN_SFT 0 +#define NAU8810_PGAMT_SFT 6 +#define NAU8810_PGAZC_SFT 7 + +/* NAU8810_REG_ADCBOOST (0x2F) */ +#define NAU8810_PMICBSTGAIN_SFT 4 +#define NAU8810_PGABST_SFT 8 + +/* NAU8810_REG_SPKMIX (0x32) */ +#define NAU8810_DACSPK_SFT 0 +#define NAU8810_BYPSPK_SFT 1 + +/* NAU8810_REG_SPKGAIN (0x36) */ +#define NAU8810_SPKGAIN_SFT 0 +#define NAU8810_SPKMT_SFT 6 +#define NAU8810_SPKZC_SFT 7 + +/* NAU8810_REG_MONOMIX (0x38) */ +#define NAU8810_DACMOUT_SFT 0 +#define NAU8810_BYPMOUT_SFT 1 +#define NAU8810_MOUTMXMT_SFT 6 + + +/* System Clock Source */ +enum { + NAU8810_SCLK_MCLK, + NAU8810_SCLK_PLL, +}; + +struct nau8810_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +struct nau8810 { + struct device *dev; + struct regmap *regmap; + struct nau8810_pll pll; + int sysclk; + int clk_id; +}; + +#endif -- 2.6.4

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[alsa-devel] [PATCH v2 0/2] alsabat: test scripts and reference amixer config files
by focus.luo＠linux.intel.com 22 Aug '16

22 Aug '16

From: Focus Luo <focus.luo(a)linux.intel.com> The patch set supplies some automated test scripts and reference amixer config files for linux audio driver testing by using alsa-bat as test tool on Intel platforms. The first patch includes automated test scripts for linux audio driver based on alsa-lib interface by using alsabat as test tool. It supports analog audio and display audio(HDMI/DP) interface. The second patch supplies some amixer reference config file on Intel platforms. Focus Luo (2): alsabat: automation test scripts alsabat: add amixer config files bat/Makefile.am | 1 + bat/tests/Makefile.am | 11 + bat/tests/README | 32 ++ bat/tests/alsabat_main.sh | 178 ++++++++++ bat/tests/analog_audio_playback_and_capture.sh | 84 +++++ bat/tests/asound_state/Makefile.am | 6 + bat/tests/asound_state/asound.state.Broadwell | 439 +++++++++++++++++++++++++ bat/tests/asound_state/asound.state.Haswell | 342 +++++++++++++++++++ bat/tests/asound_state/asound.state.Skylake | 437 ++++++++++++++++++++++++ bat/tests/dp_audio_playback.sh | 83 +++++ bat/tests/dp_audio_subdevice_number.sh | 68 ++++ bat/tests/hdmi_audio_playback.sh | 81 +++++ bat/tests/hdmi_audio_subdevice_number.sh | 65 ++++ bat/tests/map_test_case | 8 + configure.ac | 3 +- 15 files changed, 1837 insertions(+), 1 deletion(-) create mode 100644 bat/tests/Makefile.am create mode 100644 bat/tests/README create mode 100755 bat/tests/alsabat_main.sh create mode 100755 bat/tests/analog_audio_playback_and_capture.sh create mode 100644 bat/tests/asound_state/Makefile.am create mode 100644 bat/tests/asound_state/asound.state.Broadwell create mode 100644 bat/tests/asound_state/asound.state.Haswell create mode 100644 bat/tests/asound_state/asound.state.Skylake create mode 100755 bat/tests/dp_audio_playback.sh create mode 100755 bat/tests/dp_audio_subdevice_number.sh create mode 100755 bat/tests/hdmi_audio_playback.sh create mode 100755 bat/tests/hdmi_audio_subdevice_number.sh create mode 100644 bat/tests/map_test_case -- 1.9.1

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[alsa-devel] [PATCH] pcm: Fix shm initialization race-condition
by Ismael Luceno 22 Aug '16

22 Aug '16

Easily seen when two threads try at the same time, one of them will fail. The bug was identified by using apulse with Skype. Fixes: dec428c35221 ("pcm: fix 'unable to create IPC shm instance' caused by fork from a thread") Fixes: https://github.com/i-rinat/apulse/issues/38 Signed-off-by: Ismael Luceno <ismael(a)iodev.co.uk> --- src/pcm/pcm_direct.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index c3925cc20fd3..643498375b34 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -96,11 +96,12 @@ int snd_pcm_direct_shm_create_or_connect(snd_pcm_direct_t *dmix) retryget: dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), dmix->ipc_perm); - if (dmix->shmid < 0) { - if (errno == ENOENT) + if (dmix->shmid < 0 && errno == ENOENT) { if ((dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), IPC_CREAT | IPC_EXCL | dmix->ipc_perm)) != -1) first_instance = 1; + else if (errno == EEXIST) + goto retryget; } err = -errno; if (dmix->shmid < 0) { -- 2.9.2

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[alsa-devel] [PATCH 0/2] ASoC: Extended SPDIF support to Allwinner A31 SoCs
by codekipper＠gmail.com 22 Aug '16

22 Aug '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set extends the current support of the Allwinner SPDIF transceiver to include the A31 SoC. For now just the SPDIF transmitter has been tested on the Mele I7. In order for this patch set to be functional then we require changes to the clocks and the device tree. For those that are interested I've pushed all the required patches here https://github.com/codekipper/linux-sunxi/commits/a31-spdif Thanks in advance, CK Marcus Cooper (2): ASoC: sunxi: Add bindings for sun6i to SPDIF ASoC: sunxi: compatibility for sun6i to SPDIF .../devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 3 +++ sound/soc/sunxi/sun4i-spdif.c | 17 ++++++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) -- 2.9.2

8 15

Re: [alsa-devel] [PATCH] pcm: Fix shm initialization race-condition
by Takashi Iwai 22 Aug '16

22 Aug '16

On Sun, 14 Aug 2016 02:28:52 +0200, Ismael Luceno wrote: > > Easily seen when two threads try at the same time, one of them will fail. > > The bug was identified by using apulse with Skype. > > Fixes: dec428c35221 ("pcm: fix 'unable to create IPC shm instance' caused by fork from a thread") > Fixes: https://github.com/i-rinat/apulse/issues/38 > Signed-off-by: Ismael Luceno <ismael(a)iodev.co.uk> > --- > src/pcm/pcm_direct.c | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c > index c3925cc20fd3..b5215ba35406 100644 > --- a/src/pcm/pcm_direct.c > +++ b/src/pcm/pcm_direct.c > @@ -101,6 +101,8 @@ retryget: > if ((dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), > IPC_CREAT | IPC_EXCL | dmix->ipc_perm)) != -1) > first_instance = 1; > + if (dmix->shmid < 0 && errno == EEXIST) > + goto retryget; Hrm, but this would result in an endless loop if the shm was already taken persistently. Also, which call does give a negative shmid, actually? It's from the first shmget() or the second shmget()? (The existing code is ugly and we need to fix the indentation there...) thanks, Takashi

2 3

[alsa-devel] [PATCH] ASoC: core: Clean up DAPM before the card debugfs
by Mark Brown 22 Aug '16

22 Aug '16

Both the card and DAPM cleanups recursively delete their debugfs directories. Since the DAPM debugfs subdirectory for the card is located within the card debugfs this means we end up trying to double free the DAPM subdirectory. Reorder the cleanup to free the card debugfs after we've cleaned up DAPM and it has deleted its own subdirectory. Reported-by: Russell King - ARM Linux <linux(a)armlinux.org.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Not tested at all yet, just about done for today. sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16369cad4803..66a33f1e4881 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2083,14 +2083,13 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove auxiliary devices */ soc_remove_aux_devices(card); + snd_soc_dapm_free(&card->dapm); soc_cleanup_card_debugfs(card); /* remove the card */ if (card->remove) card->remove(card); - snd_soc_dapm_free(&card->dapm); - snd_card_free(card->snd_card); return 0; -- 2.8.1

3 7

[alsa-devel] [PATCH] ALSA: hdspm: fix spelling mistake "Externel" -> "External"
by Colin King 22 Aug '16

22 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/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a4a999a..0e51a00 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1666,7 +1666,7 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally) HDSPM_AUTOSYNC_FROM_NONE) { dev_warn(hdspm->card->dev, - "Detected no Externel Sync\n"); + "Detected no External Sync\n"); not_set = 1; } else if (rate != external_freq) { -- 2.9.3

2 1

Re: [alsa-devel] ASoC: sun4i-codec: playback stall and I/O error with DAPM paths all disabled
by Maxime Ripard 22 Aug '16

22 Aug '16

On Tue, Aug 16, 2016 at 12:45:39PM +0800, wens Tsai wrote: > On Mon, Aug 15, 2016 at 7:42 PM, Mark Brown <broonie(a)kernel.org> wrote: > > On Mon, Aug 15, 2016 at 05:43:55PM +0800, wens Tsai wrote: > > > >> What is unexpected is any attempt to play anything under this state makes > >> the playback software (in my case mpg321) stall, and later report an I/O > >> error. My guess is that the DAC is still disabled by DAPM, so it doesn't > >> send any DRQs, and thus the DMA engine is not consuming any data from > >> userspace. > > > > This is normal for ASoC - like you say it'll be becasue the hardware > > isn't powered up. > > > >> I think we should just enable the digital bits of the DAC/ADC all the > >> time. Or maybe transfer and then discard data if the DAC is off. Not > >> sure if this is doable though. I expect playback software to work, and > >> not block, regardless of the hardware status. > > > > Powering things up all the time will have a major effect on battery life > > for systems that care about that. The expectation is that systems with > > this sort of hardware won't normally be offering end users direct > > control of the routing, it'll be something that's handled during system > > integration. > > Ok. So I guess one solution would be to move the mute controls out of > DAPM, and maybe change some other mux like paths into actual muxes, so > there's at least one usable path at all times. I'm not sure that's even a requirement. IIRC, but Mark might correct me, you just need to expose whatever the reset state is. Maxime -- Maxime Ripard, Free Electrons Embedded Linux and Kernel engineering http://free-electrons.com

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: rmove print for failure of kmalloc
by Shawn Lin 22 Aug '16

22 Aug '16

kmalloc already print similar error once failing to alloc enough memory, so let's remove this dump here. Signed-off-by: Shawn Lin <shawn.lin(a)rock-chips.com> --- sound/usb/format.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 789d19e..2c44386 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -176,10 +176,8 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof int r, idx; fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); - if (fp->rate_table == NULL) { - usb_audio_err(chip, "cannot malloc\n"); + if (fp->rate_table == NULL) return -ENOMEM; - } fp->nr_rates = 0; fp->rate_min = fp->rate_max = 0; -- 2.3.7

2 1

[alsa-devel] [PATCH 1/3] ALSA: usb: move udh01_fb_quirk setting to quirks.c
by Daniel Mack 22 Aug '16

22 Aug '16

That's a quirk, after all, so move it where to all the other quirks live. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/endpoint.c | 4 ---- sound/usb/quirks.c | 5 +++++ 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c07a7ed..4b4ffa0 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -502,10 +502,6 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, ep->syncinterval = 3; ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); - - if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && - ep->syncmaxsize == 4) - ep->udh01_fb_quirk = 1; } list_add_tail(&ep->list, &chip->ep_list); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6cf1f35..299813f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1216,6 +1216,11 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->chip->usb_id == USB_ID(0x0763, 0x2031)) && ep->type == SND_USB_ENDPOINT_TYPE_DATA) ep->skip_packets = 16; + + /* Work around devices that report unreasonable feedback data */ + if (ep->chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && + ep->syncmaxsize == 4) + ep->udh01_fb_quirk = 1; } void snd_usb_set_interface_quirk(struct usb_device *dev) -- 2.5.5

2 3

[alsa-devel] [PATCH] mixer: Fix rounding mode documentation
by David Henningsson 22 Aug '16

22 Aug '16

The documentation specifies three different rounding modes, but this is wrong, because they all end up calling snd_tlv_convert_from_dB, which only has two rounding modes. Signed-off-by: David Henningsson <diwic(a)ubuntu.com> --- src/mixer/simple.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/src/mixer/simple.c b/src/mixer/simple.c index fd9ba93..82476d5 100644 --- a/src/mixer/simple.c +++ b/src/mixer/simple.c @@ -376,7 +376,7 @@ int snd_mixer_selem_ask_playback_vol_dB(snd_mixer_elem_t *elem, long value, long * \brief Return corresponding integer playback volume for given dB value for a mixer simple element * \param elem Mixer simple element handle * \param value value to be converted to dB range - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \param dBvalue pointer to returned dB value * \return 0 on success otherwise a negative error code */ @@ -454,7 +454,7 @@ int snd_mixer_selem_set_playback_volume(snd_mixer_elem_t *elem, snd_mixer_selem_ * \param elem Mixer simple element handle * \param channel mixer simple element channel identifier * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_playback_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -491,7 +491,7 @@ int snd_mixer_selem_set_playback_volume_all(snd_mixer_elem_t *elem, long value) * \brief Set value in dB of playback volume control for all channels of a mixer simple element * \param elem Mixer simple element handle * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t *elem, long value, int dir) @@ -706,7 +706,7 @@ int snd_mixer_selem_ask_capture_vol_dB(snd_mixer_elem_t *elem, long value, long * \param elem Mixer simple element handle * \param dBvalue dB value to be converted to integer range * \param value pointer to returned integer value - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_ask_capture_dB_vol(snd_mixer_elem_t *elem, long dBvalue, int dir, long *value) @@ -777,7 +777,7 @@ int snd_mixer_selem_set_capture_volume(snd_mixer_elem_t *elem, snd_mixer_selem_c * \param elem Mixer simple element handle * \param channel mixer simple element channel identifier * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_capture_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -814,7 +814,7 @@ int snd_mixer_selem_set_capture_volume_all(snd_mixer_elem_t *elem, long value) * \brief Set value in dB of capture volume control for all channels of a mixer simple element * \param elem Mixer simple element handle * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_capture_dB_all(snd_mixer_elem_t *elem, long value, int dir) -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] amidi: fix handling of ignored messages
by Clemens Ladisch 22 Aug '16

22 Aug '16

Clemens Ladisch (2): amidi: ignore not only Active Sensing but also Clock bytes amidi: fix timeout handling amidi/amidi.1 | 19 ++++++++--- amidi/amidi.c | 82 +++++++++++++++++++++++++++++++++++++++----------- 2 files changed, 78 insertions(+), 23 deletions(-)

4 11

[alsa-devel] [PATCH v4 0/4] ALSA: seq: obsolete change of address limit
by Takashi Sakamoto 22 Aug '16

22 Aug '16

ALSA sequencer core has two types of clients; application and kernel. For kernel clients, the core should handle data in kernel space, meanwhile this is achieved with a hack of change of address limit. This should be purged. This patchset is a revised version of my previous one: [alsa-devel] obsolete change of address limit http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/110937.html Changes in v2: - Use '_IOC_SIZE' macro to calculate the size of argument of ioctl command. - Keep redundant longer name. - Improve comments. - Merge some patches relevant to the same features. Changes in v3: - Improve series of patch - Use 'dir' field of ioctl command to detect the length of argument Changes in v4: - Update kernel API documentation - Use logical AND to detect ioctl direction Takashi Sakamoto (4): ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: add an alternative way to handle ioctl requests ALSA: seq: change ioctl command operation to get data in kernel space ALSA: seq: obsolete change of address limit sound/core/seq/seq_clientmgr.c | 721 ++++++++++++++++++----------------------- sound/core/seq/seq_compat.c | 7 +- 2 files changed, 317 insertions(+), 411 deletions(-) -- 2.7.4

2 6

[alsa-devel] [PATCH -next] ASoC: sun4i-codec: Fix error return code in sun4i_codec_probe()
by Wei Yongjun 22 Aug '16

22 Aug '16

Fix to return error code -ENOMEM instead of 0 when create card failed, as done elsewhere in this function. Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/sunxi/sun4i-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 0e19c50..e741659 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -846,6 +846,7 @@ static int sun4i_codec_probe(struct platform_device *pdev) card = sun4i_codec_create_card(&pdev->dev); if (!card) { dev_err(&pdev->dev, "Failed to create our card\n"); + ret = -ENOMEM; goto err_unregister_codec; }

3 3

[alsa-devel] [PATCH 000/127] ASoC: codec cleanup - codec probe/remove
by Kuninori Morimoto 22 Aug '16

22 Aug '16

Hi Mark These are codec duplicate functions cleanup patches - v2 We would like to switch to "component" style base ASoC in the future instead of current "CPU/Codec/Platform/Card" style. Current "codec" has its .probe/.remove, and "component" has it too. codec side is just relayed it to component side. This is the time to cleanup. These are very big patch-set, but it doesn't add new features. [001/127] - [126/127] move callback functions from codec to component, and last [127/127] will remove codec side .probe/.remove. We need this kind of cleanup for Platform too. I build-tested these patches with make allyesconfig on x86, and tested on Lager board. Best regards --- Kuninori Morimoto

3 137

Re: [alsa-devel] [PATCH] Try to support analog audio codec at firefly reload
by John Keeping 20 Aug '16

20 Aug '16

On Tue, Aug 16, 2016 at 04:42:06AM +0800, Randy Li wrote: > This patch could enabled the es8328 codec at firefly reload board, > but it doesn't not really work. I disable the mute of channel > Left mix and Right mix in alsamixer, I could hear the noise from > my headphone, but it never be a music. > I just send this patch here to see whether somebody could figure > out what is wrong, I am really not familiar with sound. Have you tried adding an mclk-fs setting to the simple-audio-card? You probably want to set it to 256, which will allow 44.1kHz, 48kHz, 88.2kHz and 96kHz at the codec. This will give more functionality than pinning the clock to a specific rate. For the Rock2 Square, I write a machine driver for the sound because the headphone GPIO setup cannot be made to work with simple-audio-card; I found the Firefly Reload schematic and it looks like the configuration hardware is very similar. It's a bit of a mess but you can find it on Github [1] if you want to take a look. [1] https://github.com/johnkeeping/linux/tree/topic/audio

1 0

[alsa-devel] [PATCH 1/3] ASoC: fsl_asrc: Propagate the real error code
by Fabio Estevam 19 Aug '16

19 Aug '16

Instead of returning -EINVAL on error, return the real error code. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index c1a0e01..15f8ba5 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -892,7 +892,7 @@ static int fsl_asrc_probe(struct platform_device *pdev) ret = fsl_asrc_init(asrc_priv); if (ret) { dev_err(&pdev->dev, "failed to init asrc %d\n", ret); - return -EINVAL; + return ret; } asrc_priv->channel_avail = 10; @@ -901,14 +901,14 @@ static int fsl_asrc_probe(struct platform_device *pdev) &asrc_priv->asrc_rate); if (ret) { dev_err(&pdev->dev, "failed to get output rate\n"); - return -EINVAL; + return ret; } ret = of_property_read_u32(np, "fsl,asrc-width", &asrc_priv->asrc_width); if (ret) { dev_err(&pdev->dev, "failed to get output width\n"); - return -EINVAL; + return ret; } if (asrc_priv->asrc_width != 16 && asrc_priv->asrc_width != 24) { -- 1.9.1

2 4

[alsa-devel] [PATCH for 4.8] ASoC: omap-abe-twl6040: Correct dmic-codec device registration
by Peter Ujfalusi 19 Aug '16

19 Aug '16

The dmic-codec was registered within the platform_driver's probe function, which can cause deferred probe to run in loops as reported and analyzed by Russell King. Use module_init/exit in the driver and handle the dmic-codec device registration and removal at that level instead of the platform_driver probe/remove. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reported-by: Russell King <linux(a)arm.linux.org.uk> --- sound/soc/omap/omap-abe-twl6040.c | 61 ++++++++++++++++++++------------------- 1 file changed, 32 insertions(+), 29 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 0843a68f277c..f61b3b58083b 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -38,10 +38,10 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ - - struct platform_device *dmic_codec_dev; }; +struct platform_device *dmic_codec_dev; + static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -258,8 +258,6 @@ static int omap_abe_probe(struct platform_device *pdev) if (priv == NULL) return -ENOMEM; - priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -284,13 +282,6 @@ static int omap_abe_probe(struct platform_device *pdev) num_links = 2; abe_twl6040_dai_links[1].cpu_of_node = dai_node; abe_twl6040_dai_links[1].platform_of_node = dai_node; - - priv->dmic_codec_dev = platform_device_register_simple( - "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } } else { num_links = 1; } @@ -299,16 +290,14 @@ static int omap_abe_probe(struct platform_device *pdev) of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + return -EINVAL; } card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - ret = -ENODEV; - goto err_unregister; + return -ENODEV; } card->dai_link = abe_twl6040_dai_links; @@ -317,17 +306,9 @@ static int omap_abe_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err_unregister; - } - - return 0; - -err_unregister: - if (!IS_ERR(priv->dmic_codec_dev)) - platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -335,13 +316,9 @@ err_unregister: static int omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); - if (!IS_ERR(priv->dmic_codec_dev)) - platform_device_unregister(priv->dmic_codec_dev); - return 0; } @@ -361,7 +338,33 @@ static struct platform_driver omap_abe_driver = { .remove = omap_abe_remove, }; -module_platform_driver(omap_abe_driver); +static int __init omap_abe_init(void) +{ + int ret; + + dmic_codec_dev = platform_device_register_simple("dmic-codec", -1, NULL, + 0); + if (IS_ERR(dmic_codec_dev)) { + pr_err("%s: dmic-codec device registration failed\n", __func__); + return PTR_ERR(dmic_codec_dev); + } + + ret = platform_driver_register(&omap_abe_driver); + if (ret) { + pr_err("%s: platform driver registration failed\n", __func__); + platform_device_unregister(dmic_codec_dev); + } + + return ret; +} +module_init(omap_abe_init); + +static void __exit omap_abe_exit(void) +{ + platform_driver_unregister(&omap_abe_driver); + platform_device_unregister(dmic_codec_dev); +} +module_exit(omap_abe_exit); MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec"); -- 2.9.2

3 3

Re: [alsa-devel] [PATCH] Try to support analog audio codec at firefly reload
by Heiko Stuebner 19 Aug '16

19 Aug '16

Am Dienstag, 16. August 2016, 04:42:06 CEST schrieb Randy Li: > Hello: > This patch could enabled the es8328 codec at firefly reload board, > but it doesn't not really work. I disable the mute of channel > Left mix and Right mix in alsamixer, I could hear the noise from > my headphone, but it never be a music. > I just send this patch here to see whether somebody could figure > out what is wrong, I am really not familiar with sound. I can't really help here as I have neither the board nor that much sound experience, but it's definitly customary to label patches as rfc when they're expected to need more work, so make that [PATCH RFC] ... for stuff that requires more work before it can actually be merged. Heiko

1 0

[alsa-devel] [PATCH] ASoC: rockchip: Add support rt5514 dsp summy dailink
by Xing Zheng 19 Aug '16

19 Aug '16

This patch can attach automaticlly rt5514 spi DAI with driver name "rt5514" in the snd_soc_find_dai process. Turn this feature on, we can enable the voice wake up via rt5514 dsp for RK3399 Gru Boards. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- sound/soc/rockchip/Kconfig | 1 + sound/soc/rockchip/rk3399_gru_sound.c | 39 ++++++++++++++++++++++++++++++++- 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 6d39032..d82d763 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_RK3399_GRU_SOUND select SND_SOC_MAX98357A select SND_SOC_RT5514 select SND_SOC_DA7219 + select SND_SOC_RT5514_SPI help Say Y or M here if you want to add support multiple codecs for SoC audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 9933703..164b6da 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -230,8 +230,11 @@ enum { DAILINK_MAX98357A, DAILINK_RT5514, DAILINK_DA7219, + DAILINK_RT5514_DSP, }; +#define DAILINK_ENTITIES (DAILINK_DA7219 + 1) + static struct snd_soc_dai_link rockchip_dailinks[] = { [DAILINK_MAX98357A] = { .name = "MAX98357A", @@ -261,6 +264,13 @@ static struct snd_soc_dai_link rockchip_dailinks[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, }, + /* RT5514 DSP for voice wakeup via spi bus */ + [DAILINK_RT5514_DSP] = { + .name = "RT5514 DSP", + .stream_name = "Wake on Voice", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + }, }; static struct snd_soc_card rockchip_sound_card = { @@ -276,10 +286,17 @@ static struct snd_soc_card rockchip_sound_card = { .num_controls = ARRAY_SIZE(rockchip_controls), }; +static int rockchip_sound_match_stub(struct device *dev, void *data) +{ + return 1; +} + static int rockchip_sound_probe(struct platform_device *pdev) { struct snd_soc_card *card = &rockchip_sound_card; struct device_node *cpu_node; + struct device *dev; + struct device_driver *drv; int i, ret; cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); @@ -288,7 +305,7 @@ static int rockchip_sound_probe(struct platform_device *pdev) return -EINVAL; } - for (i = 0; i < card->num_links; i++) { + for (i = 0; i < DAILINK_ENTITIES; i++) { rockchip_dailinks[i].platform_of_node = cpu_node; rockchip_dailinks[i].cpu_of_node = cpu_node; @@ -301,6 +318,26 @@ static int rockchip_sound_probe(struct platform_device *pdev) } } + /** + * To acquire the spi driver of the rt5514 and set the dai-links names + * for soc_bind_dai_link + */ + drv = driver_find("rt5514", &spi_bus_type); + if (!drv) { + dev_err(&pdev->dev, "Can not find the rt5514 driver at the spi bus\n"); + return -EINVAL; + } + + dev = driver_find_device(drv, NULL, NULL, rockchip_sound_match_stub); + if (!dev) { + dev_err(&pdev->dev, "Can not find the rt5514 device\n"); + return -ENODEV; + } + + rockchip_dailinks[DAILINK_RT5514_DSP].cpu_name = kstrdup_const(dev_name(dev), GFP_KERNEL); + rockchip_dailinks[DAILINK_RT5514_DSP].cpu_dai_name = kstrdup_const(dev_name(dev), GFP_KERNEL); + rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL); + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

1 0

[alsa-devel] Use of copy_to_user in fireworks_hwdep.c while holding a spin_lock
by Vaishali Thakkar 19 Aug '16

19 Aug '16

Hello, I was wondering about the call to copy_to_user in function hwdep_read_locked and hwdep_read_resp_buf for driver sound/firewire/fireworks/fireworks_hwdep.c. The function hwdep_read calls both of these functions while holding a spinlock[1], which is not normally allowed due to the possibility of a deadlock. This seems to be coming from the commit 555e8a8f7f149544eb7d4aa3a6420bc4c3055638 while adding a command/response functionality into hwdep interface. Is there some reason that I am overlooking, why it is OK in this case? Is there some code in the same file which ensures that page fault will not occur when we are calling these functions while holding a spin_lock_irq? The same issue is there with the driver sound/firewire/tascam/tascam-hwdep.c for obvious reasons. Coccinelle script is used to detect this issue. Thank you. [1] http://lxr.free-electrons.com/source/sound/firewire/fireworks/fireworks_hwd… -- Vaishali

2 1

[alsa-devel] [PATCH 0/3] line6 locking and segfault fixes
by Andrej Krutak 19 Aug '16

19 Aug '16

These are already seen patches fixing a few locking issues, now split out of the POD-X3 patchset. In addition, pod sysfs attributes touched something else than they thought, causing segfaults - fixed it (tested with the same code in podhd)... Andrej Krutak (3): ALSA: line6: Remove double line6_pcm_release() after failed acquire. ALSA: line6: Give up on the lock while URBs are released. ALSA: line6: Fix POD sysfs attributes segfault sound/usb/line6/pcm.c | 3 ++- sound/usb/line6/pod.c | 12 ++++++------ 2 files changed, 8 insertions(+), 7 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH] ASoC: nau8810: Add driver for Nuvoton codec chip NAU88C10
by John Hsu 19 Aug '16

19 Aug '16

The driver is for codec NAU88C10 of Nuvoton Technology Corporation. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8810.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8810.c | 908 +++++++++++++++++++++ sound/soc/codecs/nau8810.h | 288 +++++++ 5 files changed, 1219 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8810.txt create mode 100644 sound/soc/codecs/nau8810.c create mode 100644 sound/soc/codecs/nau8810.h diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt new file mode 100644 index 0000000..05830e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8810.txt @@ -0,0 +1,16 @@ +NAU8810 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8810" + + - reg : the I2C address of the device. + +Example: + +codec: nau8810@1a { + compatible = "nuvoton,nau8810"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb57f8f..bbd7324 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -91,6 +91,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C @@ -1046,6 +1047,10 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8810 + tristate "Nuvoton Technology Corporation NAU88C10 CODEC" + depends on I2C + config SND_SOC_NAU8825 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 550a174..3c15bc5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -84,6 +84,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8810-objs := nau8810.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o @@ -301,6 +302,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c new file mode 100644 index 0000000..aa25286 --- /dev/null +++ b/sound/soc/codecs/nau8810.c @@ -0,0 +1,908 @@ +/* + * nau8810.c -- NAU8810 ALSA Soc Audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * Based on WM8974.c + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "nau8810.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + + +static const int nau8810_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8810_reg_defaults[] = { + { NAU8810_REG_POWER1, 0x0000 }, + { NAU8810_REG_POWER2, 0x0000 }, + { NAU8810_REG_POWER3, 0x0000 }, + { NAU8810_REG_IFACE, 0x0050 }, + { NAU8810_REG_COMP, 0x0000 }, + { NAU8810_REG_CLOCK, 0x0140 }, + { NAU8810_REG_SMPLR, 0x0000 }, + { NAU8810_REG_DAC, 0x0000 }, + { NAU8810_REG_DACGAIN, 0x00FF }, + { NAU8810_REG_ADC, 0x0100 }, + { NAU8810_REG_ADCGAIN, 0x00FF }, + { NAU8810_REG_EQ1, 0x012C }, + { NAU8810_REG_EQ2, 0x002C }, + { NAU8810_REG_EQ3, 0x002C }, + { NAU8810_REG_EQ4, 0x002C }, + { NAU8810_REG_EQ5, 0x002C }, + { NAU8810_REG_DACLIM1, 0x0032 }, + { NAU8810_REG_DACLIM2, 0x0000 }, + { NAU8810_REG_NOTCH1, 0x0000 }, + { NAU8810_REG_NOTCH2, 0x0000 }, + { NAU8810_REG_NOTCH3, 0x0000 }, + { NAU8810_REG_NOTCH4, 0x0000 }, + { NAU8810_REG_ALC1, 0x0038 }, + { NAU8810_REG_ALC2, 0x000B }, + { NAU8810_REG_ALC3, 0x0032 }, + { NAU8810_REG_NOISEGATE, 0x0000 }, + { NAU8810_REG_PLLN, 0x0008 }, + { NAU8810_REG_PLLK1, 0x000C }, + { NAU8810_REG_PLLK2, 0x0093 }, + { NAU8810_REG_PLLK3, 0x00E9 }, + { NAU8810_REG_ATTEN, 0x0000 }, + { NAU8810_REG_INPUT_SIGNAL, 0x0003 }, + { NAU8810_REG_PGAGAIN, 0x0010 }, + { NAU8810_REG_ADCBOOST, 0x0100 }, + { NAU8810_REG_OUTPUT, 0x0002 }, + { NAU8810_REG_SPKMIX, 0x0001 }, + { NAU8810_REG_SPKGAIN, 0x0039 }, + { NAU8810_REG_MONOMIX, 0x0001 }, + { NAU8810_REG_POWER4, 0x0000 }, + { NAU8810_REG_TSLOTCTL1, 0x0000 }, + { NAU8810_REG_TSLOTCTL2, 0x0020 }, + { NAU8810_REG_DEVICE_REVID, 0x0000 }, + { NAU8810_REG_I2C_DEVICEID, 0x001A }, + { NAU8810_REG_ADDITIONID, 0x00CA }, + { NAU8810_REG_RESERVE, 0x0124 }, + { NAU8810_REG_OUTCTL, 0x0001 }, + { NAU8810_REG_ALC1ENHAN1, 0x0010 }, + { NAU8810_REG_ALC1ENHAN2, 0x0000 }, + { NAU8810_REG_MISCCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFF, 0x0000 }, + { NAU8810_REG_AGCP2POUT, 0x0000 }, + { NAU8810_REG_AGCPOUT, 0x0000 }, + { NAU8810_REG_AMTCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFFMAN, 0x0000 }, +}; + +static bool nau8810_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + case NAU8810_REG_OUTCTL ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_OUTCTL ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + return true; + default: + return false; + } +} + +static int nau8810_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg, reg_val; + u16 *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8810_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + regmap_read(nau8810->regmap, reg + i, &reg_val); + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, &reg_val, sizeof(reg_val)); + } + + return 0; +} + +static int nau8810_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8810_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + value = be16_to_cpu(*(val + i)); + ret = regmap_write(nau8810->regmap, reg + i, value); + if (ret) { + dev_err(codec->dev, "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8810_companding[] = { + "Off", "NC", "u-law", "A-law" }; + +static const struct soc_enum nau8810_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_ADCCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding); + +static const struct soc_enum nau8810_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_DACCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding); + +static const char * const nau8810_deemp[] = { + "None", "32kHz", "44.1kHz", "48kHz" }; + +static const struct soc_enum nau8810_deemp_enum = + SOC_ENUM_SINGLE(NAU8810_REG_DAC, NAU8810_DEEMP_SFT, + ARRAY_SIZE(nau8810_deemp), nau8810_deemp); + +static const char * const nau8810_eqmode[] = {"Capture", "Playback" }; + +static const struct soc_enum nau8810_eqmode_enum = + SOC_ENUM_SINGLE(NAU8810_REG_EQ1, NAU8810_EQM_SFT, + ARRAY_SIZE(nau8810_eqmode), nau8810_eqmode); + +static const char * const nau8810_alc[] = {"Normal", "Limiter" }; + +static const struct soc_enum nau8810_alc_enum = + SOC_ENUM_SINGLE(NAU8810_REG_ALC3, NAU8810_ALCM_SFT, + ARRAY_SIZE(nau8810_alc), nau8810_alc); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); + +static const struct snd_kcontrol_new nau8810_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8810_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8810_companding_dac_enum), + SOC_ENUM("DAC De-emphasis", nau8810_deemp_enum), + + SOC_ENUM("EQ Function", nau8810_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8810_eq_get, nau8810_eq_put), + + SOC_SINGLE("DAC Inversion Switch", NAU8810_REG_DAC, + NAU8810_DACPL_SFT, 1, 0), + SOC_SINGLE_TLV("Playback Volume", NAU8810_REG_DACGAIN, + NAU8810_DACGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", NAU8810_REG_ADC, + NAU8810_HPFEN_SFT, 1, 0), + SOC_SINGLE("High Pass Cut Off", NAU8810_REG_ADC, + NAU8810_HPF_SFT, 0x7, 0), + + SOC_SINGLE("ADC Inversion Switch", NAU8810_REG_ADC, + NAU8810_ADCPL_SFT, 1, 0), + SOC_SINGLE_TLV("Capture Volume", NAU8810_REG_ADCGAIN, + NAU8810_ADCGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE_TLV("EQ1 Volume", NAU8810_REG_EQ1, + NAU8810_EQ1GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ2 Volume", NAU8810_REG_EQ2, + NAU8810_EQ2GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ3 Volume", NAU8810_REG_EQ3, + NAU8810_EQ3GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ4 Volume", NAU8810_REG_EQ4, + NAU8810_EQ4GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ5 Volume", NAU8810_REG_EQ5, + NAU8810_EQ5GC_SFT, 0x18, 1, eq_tlv), + + SOC_SINGLE("DAC Limiter Switch", NAU8810_REG_DACLIM1, + NAU8810_DACLIMEN_SFT, 1, 0), + SOC_SINGLE("DAC Limiter Decay", NAU8810_REG_DACLIM1, + NAU8810_DACLIMDCY_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Attack", NAU8810_REG_DACLIM1, + NAU8810_DACLIMATK_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Threshold", NAU8810_REG_DACLIM2, + NAU8810_DACLIMTHL_SFT, 0x7, 0), + SOC_SINGLE("DAC Limiter Boost", NAU8810_REG_DACLIM2, + NAU8810_DACLIMBST_SFT, 0xf, 0), + + SOC_ENUM("ALC Mode", nau8810_alc_enum), + SOC_SINGLE("ALC Enable Switch", NAU8810_REG_ALC1, + NAU8810_ALCEN_SFT, 1, 0), + SOC_SINGLE("ALC Max Volume", NAU8810_REG_ALC1, + NAU8810_ALCMXGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC Min Volume", NAU8810_REG_ALC1, + NAU8810_ALCMINGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC ZC Switch", NAU8810_REG_ALC2, + NAU8810_ALCZC_SFT, 1, 0), + SOC_SINGLE("ALC Hold", NAU8810_REG_ALC2, + NAU8810_ALCHT_SFT, 0xf, 0), + SOC_SINGLE("ALC Target", NAU8810_REG_ALC2, + NAU8810_ALCSL_SFT, 0xf, 0), + SOC_SINGLE("ALC Decay", NAU8810_REG_ALC3, + NAU8810_ALCDCY_SFT, 0xf, 0), + SOC_SINGLE("ALC Attack", NAU8810_REG_ALC3, + NAU8810_ALCATK_SFT, 0xf, 0), + SOC_SINGLE("ALC Noise Gate Switch", NAU8810_REG_NOISEGATE, + NAU8810_ALCNEN_SFT, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", NAU8810_REG_NOISEGATE, + NAU8810_ALCNTH_SFT, 0x7, 0), + + SOC_SINGLE("PGA ZC Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAZC_SFT, 1, 0), + SOC_SINGLE_TLV("PGA Volume", NAU8810_REG_PGAGAIN, + NAU8810_PGAGAIN_SFT, 0x3f, 0, inpga_tlv), + + SOC_SINGLE("Speaker ZC Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKZC_SFT, 1, 0), + SOC_SINGLE("Speaker Mute Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKMT_SFT, 1, 0), + SOC_SINGLE_TLV("Speaker Volume", NAU8810_REG_SPKGAIN, + NAU8810_SPKGAIN_SFT, 0x3f, 0, spk_tlv), + + SOC_SINGLE("Capture Boost(+20dB)", NAU8810_REG_ADCBOOST, + NAU8810_PGABST_SFT, 1, 0), + SOC_SINGLE("Mono Mute Switch", NAU8810_REG_MONOMIX, + NAU8810_MOUTMXMT_SFT, 1, 0), + + SOC_SINGLE("DAC Oversampling Rate(128x) Switch", NAU8810_REG_DAC, + NAU8810_DACOS_SFT, 1, 0), + SOC_SINGLE("ADC Oversampling Rate(128x) Switch", NAU8810_REG_ADC, + NAU8810_ADCOS_SFT, 1, 0), +}; + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_SPKMIX, + NAU8810_BYPSPK_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_SPKMIX, + NAU8810_DACSPK_SFT, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_MONOMIX, + NAU8810_BYPMOUT_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_MONOMIX, + NAU8810_DACMOUT_SFT, 1, 0), +}; + +/* PGA Mute */ +static const struct snd_kcontrol_new nau8810_inpga_mute[] = { + SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAMT_SFT, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8810_inpga[] = { + SOC_DAPM_SINGLE("MicN Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_NMICPGA_SFT, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_PMICPGA_SFT, 1, 0), +}; + +/* Mic Input boost vol */ +static const struct snd_kcontrol_new nau8810_mic_boost_controls = + SOC_DAPM_SINGLE("Mic Volume", NAU8810_REG_ADCBOOST, + NAU8810_PMICBSTGAIN_SFT, 0x7, 0); + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8810_loopback[] = { + SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, + NAU8810_ADDAP_SFT, 1, 0), +}; + +static int check_mclk_select_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); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + regmap_read(nau8810->regmap, NAU8810_REG_CLOCK, &value); + return (value & NAU8810_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Speaker Mixer", NAU8810_REG_POWER3, + NAU8810_SPKMX_EN_SFT, 0, &nau8810_speaker_mixer_controls[0], + ARRAY_SIZE(nau8810_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, + NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], + ARRAY_SIZE(nau8810_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + NAU8810_DAC_EN_SFT, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + NAU8810_ADC_EN_SFT, 0), + SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, + NAU8810_NSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", NAU8810_REG_POWER3, + NAU8810_PSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", NAU8810_REG_POWER3, + NAU8810_MOUT_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Input PGA", NAU8810_REG_POWER2, + NAU8810_PGA_EN_SFT, 0, nau8810_inpga, + ARRAY_SIZE(nau8810_inpga)), + SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, + NAU8810_BST_EN_SFT, 0, nau8810_inpga_mute, + ARRAY_SIZE(nau8810_inpga_mute)), + + SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, + NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", NAU8810_REG_POWER1, + NAU8810_PLL_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8810_loopback), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { + {"DAC", NULL, "PLL", check_mclk_select_pll}, + + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Input Boost Stage */ + {"ADC", NULL, "Input Boost Stage"}, + {"ADC", NULL, "PLL", check_mclk_select_pll}, + {"Input Boost Stage", NULL, "Input PGA"}, + {"Input Boost Stage", NULL, "MICP"}, + + /* Input PGA */ + {"Input PGA", NULL, "Mic Bias"}, + {"Input PGA", "MicN Switch", "MICN"}, + {"Input PGA", "MicP Switch", "MICP"}, + + /* Digital Looptack */ + {"Digital Loopback", "Switch", "ADC"}, + {"DAC", NULL, "Digital Loopback"}, +}; + +static int nau8810_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + + nau8810->sysclk = freq; + dev_dbg(nau8810->dev, "master sysclk %dHz\n", nau8810->sysclk); + + return 0; +} + +static int nau8810_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8810->regmap; + struct nau8810_pll *pll = &nau8810->pll; + + switch (nau8810->div_id) { + case NAU8810_MCLK_DIV_PLL: + /* master clock from PLL and enable PLL */ + if (pll->mclk_scaler != div) { + dev_err(nau8810->dev, "Master clock prescaler not meet PLL parameters\n"); + return -EINVAL; + } + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_PLL); + break; + + case NAU8810_MCLK_DIV_MCLK: + /* Defer the master clock prescaler configuration to DAI + * hardware parameter if master clock from MCLK because + * it needs runtime fs information to get the proper div. + */ + break; + + case NAU8810_BCLK_DIV: + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_BCLKSEL_MASK, (div << NAU8810_BCLKSEL_SFT)); + break; + + default: + return -EINVAL; + } + nau8810->div_id = div_id; + + return 0; +} + +static int nau8810_mclk_clkdiv(struct nau8810 *nau8810, int rate) +{ + struct regmap *regmap = nau8810->regmap; + int i, sclk, imclk = rate * 256, div = 0; + + if (!nau8810->sysclk) { + dev_err(nau8810->dev, "Make mclk div configuration fail because of invalid system clock\n"); + return -EINVAL; + } + + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + for (i = 1; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + sclk = (nau8810->sysclk * 10) / + nau8810_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(nau8810->dev, + "master clock prescaler %x for fs %d\n", div, rate); + + /* master clock from MCLK and disable PLL */ + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_MCLK); + + return 0; +} + +static int nau88l0_calc_pll(unsigned int pll_in, + unsigned int fs, struct nau8810_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); + for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + if (ARRAY_SIZE(nau8810_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8810_set_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + struct nau8810_pll *pll_param = &nau8810->pll; + int ret, fs; + + fs = freq_out / 256; + ret = nau88l0_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(nau8810->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_info(nau8810->dev, "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, + pll_param->pre_factor); + + regmap_update_bits(map, NAU8810_REG_PLLN, + NAU8810_PLLMCLK_DIV2 | NAU8810_PLLN_MASK, + (pll_param->pre_factor ? NAU8810_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + regmap_write(map, NAU8810_REG_PLLK1, + (pll_param->pll_frac >> NAU8810_PLLK1_SFT) & + NAU8810_PLLK1_MASK); + regmap_write(map, NAU8810_REG_PLLK2, + (pll_param->pll_frac >> NAU8810_PLLK2_SFT) & + NAU8810_PLLK2_MASK); + regmap_write(map, NAU8810_REG_PLLK3, + pll_param->pll_frac & NAU8810_PLLK3_MASK); + regmap_update_bits(map, NAU8810_REG_CLOCK, NAU8810_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8810_MCLKSEL_SFT); + regmap_update_bits(map, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_PLL); + + return 0; +} + +static int nau8810_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + u16 ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8810_CLKIO_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8810_AIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8810_AIFMT_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8810_AIFMT_PCM_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= NAU8810_BCLKP_IB | NAU8810_FSP_IF; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8810_BCLKP_IB; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= NAU8810_FSP_IF; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_AIFMT_MASK | NAU8810_FSP_IF | + NAU8810_BCLKP_IB, ctrl1_val); + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_CLKIO_MASK, ctrl2_val); + + return 0; +} + +static int nau8810_pcm_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 nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + int val_len = 0, val_rate = 0, ret = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8810_WLEN_20; + break; + case 24: + val_len |= NAU8810_WLEN_24; + break; + case 32: + val_len |= NAU8810_WLEN_32; + break; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8810_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8810_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8810_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8810_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8810_SMPLR_32K; + break; + case 44100: + case 48000: + break; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_WLEN_MASK, val_len); + regmap_update_bits(nau8810->regmap, NAU8810_REG_SMPLR, + NAU8810_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8810->div_id == NAU8810_MCLK_DIV_MCLK) { + ret = nau8810_mclk_clkdiv(nau8810, params_rate(params)); + if (ret < 0) + dev_err(nau8810->dev, "MCLK div configuration fail\n"); + } + + return ret; +} + +static int nau8810_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN); + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + regcache_sync(map); + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_3K); + mdelay(100); + } + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + regmap_write(map, NAU8810_REG_POWER1, 0); + regmap_write(map, NAU8810_REG_POWER2, 0); + regmap_write(map, NAU8810_REG_POWER3, 0); + break; + } + + return 0; +} + + +#define NAU8810_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8810_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8810_ops = { + .hw_params = nau8810_pcm_hw_params, + .set_fmt = nau8810_set_dai_fmt, + .set_sysclk = nau8810_set_sysclk, + .set_clkdiv = nau8810_set_clkdiv, + .set_pll = nau8810_set_pll, +}; + +static struct snd_soc_dai_driver nau8810_dai = { + .name = "nau8810-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .ops = &nau8810_ops, + .symmetric_rates = 1, +}; + +static const struct regmap_config nau8810_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8810_REG_MAX, + .readable_reg = nau8810_readable_reg, + .writeable_reg = nau8810_writeable_reg, + .volatile_reg = nau8810_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8810_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8810_reg_defaults), +}; + +static int nau8810_probe(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_nau8810 = { + .probe = nau8810_probe, + .set_bias_level = nau8810_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8810_snd_controls, + .num_controls = ARRAY_SIZE(nau8810_snd_controls), + .dapm_widgets = nau8810_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8810_dapm_widgets), + .dapm_routes = nau8810_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8810_dapm_routes), +}; + +static int nau8810_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8810 *nau8810 = dev_get_platdata(dev); + + if (!nau8810) { + nau8810 = devm_kzalloc(dev, sizeof(*nau8810), GFP_KERNEL); + if (!nau8810) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8810); + + nau8810->regmap = devm_regmap_init_i2c(i2c, &nau8810_regmap_config); + if (IS_ERR(nau8810->regmap)) + return PTR_ERR(nau8810->regmap); + nau8810->dev = dev; + + regmap_write(nau8810->regmap, NAU8810_REG_RESET, 0x00); + + return snd_soc_register_codec(dev, + &soc_codec_dev_nau8810, &nau8810_dai, 1); +} + +static int nau8810_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id nau8810_i2c_id[] = { + { "nau8810", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8810_of_match[] = { + { .compatible = "nuvoton,nau8810", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8810_of_match); +#endif + +static struct i2c_driver nau8810_i2c_driver = { + .driver = { + .name = "nau8810", + .of_match_table = of_match_ptr(nau8810_of_match), + }, + .probe = nau8810_i2c_probe, + .remove = nau8810_i2c_remove, + .id_table = nau8810_i2c_id, +}; + +module_i2c_driver(nau8810_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8810 driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8810.h b/sound/soc/codecs/nau8810.h new file mode 100644 index 0000000..14b8ae3 --- /dev/null +++ b/sound/soc/codecs/nau8810.h @@ -0,0 +1,288 @@ +/* + * NAU8810 ALSA SoC audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * Author: David Lin <ctlin0(a)nuvoton.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 __NAU8810_H__ +#define __NAU8810_H__ + +#define NAU8810_REG_RESET 0x00 +#define NAU8810_REG_POWER1 0x01 +#define NAU8810_REG_POWER2 0x02 +#define NAU8810_REG_POWER3 0x03 +#define NAU8810_REG_IFACE 0x04 +#define NAU8810_REG_COMP 0x05 +#define NAU8810_REG_CLOCK 0x06 +#define NAU8810_REG_SMPLR 0x07 +#define NAU8810_REG_DAC 0x0A +#define NAU8810_REG_DACGAIN 0x0B +#define NAU8810_REG_ADC 0x0E +#define NAU8810_REG_ADCGAIN 0x0F +#define NAU8810_REG_EQ1 0x12 +#define NAU8810_REG_EQ2 0x13 +#define NAU8810_REG_EQ3 0x14 +#define NAU8810_REG_EQ4 0x15 +#define NAU8810_REG_EQ5 0x16 +#define NAU8810_REG_DACLIM1 0x18 +#define NAU8810_REG_DACLIM2 0x19 +#define NAU8810_REG_NOTCH1 0x1B +#define NAU8810_REG_NOTCH2 0x1C +#define NAU8810_REG_NOTCH3 0x1D +#define NAU8810_REG_NOTCH4 0x1E +#define NAU8810_REG_ALC1 0x20 +#define NAU8810_REG_ALC2 0x21 +#define NAU8810_REG_ALC3 0x22 +#define NAU8810_REG_NOISEGATE 0x23 +#define NAU8810_REG_PLLN 0x24 +#define NAU8810_REG_PLLK1 0x25 +#define NAU8810_REG_PLLK2 0x26 +#define NAU8810_REG_PLLK3 0x27 +#define NAU8810_REG_ATTEN 0x28 +#define NAU8810_REG_INPUT_SIGNAL 0x2C +#define NAU8810_REG_PGAGAIN 0x2D +#define NAU8810_REG_ADCBOOST 0x2F +#define NAU8810_REG_OUTPUT 0x31 +#define NAU8810_REG_SPKMIX 0x32 +#define NAU8810_REG_SPKGAIN 0x36 +#define NAU8810_REG_MONOMIX 0x38 +#define NAU8810_REG_POWER4 0x3A +#define NAU8810_REG_TSLOTCTL1 0x3B +#define NAU8810_REG_TSLOTCTL2 0x3C +#define NAU8810_REG_DEVICE_REVID 0x3E +#define NAU8810_REG_I2C_DEVICEID 0x3F +#define NAU8810_REG_ADDITIONID 0x40 +#define NAU8810_REG_RESERVE 0x41 +#define NAU8810_REG_OUTCTL 0x45 +#define NAU8810_REG_ALC1ENHAN1 0x46 +#define NAU8810_REG_ALC1ENHAN2 0x47 +#define NAU8810_REG_MISCCTL 0x49 +#define NAU8810_REG_OUTTIEOFF 0x4B +#define NAU8810_REG_AGCP2POUT 0x4C +#define NAU8810_REG_AGCPOUT 0x4D +#define NAU8810_REG_AMTCTL 0x4E +#define NAU8810_REG_OUTTIEOFFMAN 0x4F +#define NAU8810_REG_MAX NAU8810_REG_OUTTIEOFFMAN + + +/* NAU8810_REG_POWER1 (0x1) */ +#define NAU8810_DCBUF_EN (0x1 << 8) +#define NAU8810_PLL_EN_SFT 5 +#define NAU8810_MICBIAS_EN_SFT 4 +#define NAU8810_ABIAS_EN (0x1 << 3) +#define NAU8810_IOBUF_EN (0x1 << 2) +#define NAU8810_REFIMP_MASK 0x3 +#define NAU8810_REFIMP_DIS 0x0 +#define NAU8810_REFIMP_80K 0x1 +#define NAU8810_REFIMP_300K 0x2 +#define NAU8810_REFIMP_3K 0x3 + +/* NAU8810_REG_POWER2 (0x2) */ +#define NAU8810_BST_EN_SFT 4 +#define NAU8810_PGA_EN_SFT 2 +#define NAU8810_ADC_EN_SFT 0 + +/* NAU8810_REG_POWER3 (0x3) */ +#define NAU8810_DAC_EN_SFT 0 +#define NAU8810_SPKMX_EN_SFT 2 +#define NAU8810_MOUTMX_EN_SFT 3 +#define NAU8810_PSPK_EN_SFT 5 +#define NAU8810_NSPK_EN_SFT 6 +#define NAU8810_MOUT_EN_SFT 7 + +/* NAU8810_REG_IFACE (0x4) */ +#define NAU8810_AIFMT_SFT 3 +#define NAU8810_AIFMT_MASK (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_RIGHT (0x0 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_LEFT (0x1 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_I2S (0x2 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_PCM_A (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_WLEN_SFT 5 +#define NAU8810_WLEN_MASK (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_16 (0x0 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_20 (0x1 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_24 (0x2 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_32 (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_FSP_IF (0x1 << 7) +#define NAU8810_BCLKP_IB (0x1 << 8) + +/* NAU8810_REG_COMP (0x5) */ +#define NAU8810_ADDAP_SFT 0 +#define NAU8810_ADCCM_SFT 1 +#define NAU8810_DACCM_SFT 3 + +/* NAU8810_REG_CLOCK (0x6) */ +#define NAU8810_CLKIO_MASK 0x1 +#define NAU8810_CLKIO_SLAVE 0x0 +#define NAU8810_CLKIO_MASTER 0x1 +#define NAU8810_BCLKSEL_SFT 2 +#define NAU8810_BCLKSEL_MASK (0x7 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_1 (0x0 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_2 (0x1 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_4 (0x2 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_8 (0x3 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_16 (0x4 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_32 (0x5 << NAU8810_BCLKSEL_SFT) +#define NAU8810_MCLKSEL_SFT 5 +#define NAU8810_MCLKSEL_MASK (0x7 << NAU8810_MCLKSEL_SFT) +#define NAU8810_CLKM_SFT 8 +#define NAU8810_CLKM_MASK (0x1 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_MCLK (0x0 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_PLL (0x1 << NAU8810_CLKM_SFT) + +/* NAU8810_REG_SMPLR (0x7) */ +#define NAU8810_SMPLR_SFT 1 +#define NAU8810_SMPLR_MASK (0x7 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_48K (0x0 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_32K (0x1 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_24K (0x2 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_16K (0x3 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_12K (0x4 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_8K (0x5 << NAU8810_SMPLR_SFT) + +/* NAU8810_REG_DAC (0xA) */ +#define NAU8810_DACPL_SFT 0 +#define NAU8810_DACOS_SFT 3 +#define NAU8810_DEEMP_SFT 4 + +/* NAU8810_REG_DACGAIN (0xB) */ +#define NAU8810_DACGAIN_SFT 0 + +/* NAU8810_REG_ADC (0xE) */ +#define NAU8810_ADCPL_SFT 0 +#define NAU8810_ADCOS_SFT 3 +#define NAU8810_HPF_SFT 4 +#define NAU8810_HPFEN_SFT 8 + +/* NAU8810_REG_ADCGAIN (0xF) */ +#define NAU8810_ADCGAIN_SFT 0 + +/* NAU8810_REG_EQ1 (0x12) */ +#define NAU8810_EQ1GC_SFT 0 +#define NAU8810_EQ1CF_SFT 5 +#define NAU8810_EQM_SFT 8 + +/* NAU8810_REG_EQ2 (0x13) */ +#define NAU8810_EQ2GC_SFT 0 +#define NAU8810_EQ2CF_SFT 5 +#define NAU8810_EQ2BW_SFT 8 + +/* NAU8810_REG_EQ3 (0x14) */ +#define NAU8810_EQ3GC_SFT 0 +#define NAU8810_EQ3CF_SFT 5 +#define NAU8810_EQ3BW_SFT 8 + +/* NAU8810_REG_EQ4 (0x15) */ +#define NAU8810_EQ4GC_SFT 0 +#define NAU8810_EQ4CF_SFT 5 +#define NAU8810_EQ4BW_SFT 8 + +/* NAU8810_REG_EQ5 (0x16) */ +#define NAU8810_EQ5GC_SFT 0 +#define NAU8810_EQ5CF_SFT 5 + +/* NAU8810_REG_DACLIM1 (0x18) */ +#define NAU8810_DACLIMATK_SFT 0 +#define NAU8810_DACLIMDCY_SFT 4 +#define NAU8810_DACLIMEN_SFT 8 + +/* NAU8810_REG_DACLIM2 (0x19) */ +#define NAU8810_DACLIMBST_SFT 0 +#define NAU8810_DACLIMTHL_SFT 4 + +/* NAU8810_REG_ALC1 (0x20) */ +#define NAU8810_ALCMINGAIN_SFT 0 +#define NAU8810_ALCMXGAIN_SFT 3 +#define NAU8810_ALCEN_SFT 8 + +/* NAU8810_REG_ALC2 (0x21) */ +#define NAU8810_ALCSL_SFT 0 +#define NAU8810_ALCHT_SFT 4 +#define NAU8810_ALCZC_SFT 8 + +/* NAU8810_REG_ALC3 (0x22) */ +#define NAU8810_ALCATK_SFT 0 +#define NAU8810_ALCDCY_SFT 4 +#define NAU8810_ALCM_SFT 8 + +/* NAU8810_REG_NOISEGATE (0x23) */ +#define NAU8810_ALCNTH_SFT 0 +#define NAU8810_ALCNEN_SFT 3 + +/* NAU8810_REG_PLLN (0x24) */ +#define NAU8810_PLLN_MASK 0xF +#define NAU8810_PLLMCLK_DIV2 (0x1 << 4) + +/* NAU8810_REG_PLLK1 (0x25) */ +#define NAU8810_PLLK1_SFT 18 +#define NAU8810_PLLK1_MASK 0x3F + +/* NAU8810_REG_PLLK2 (0x26) */ +#define NAU8810_PLLK2_SFT 9 +#define NAU8810_PLLK2_MASK 0x1FF + +/* NAU8810_REG_PLLK3 (0x27) */ +#define NAU8810_PLLK3_MASK 0x1FF + +/* NAU8810_REG_INPUT_SIGNAL (0x2C) */ +#define NAU8810_PMICPGA_SFT 0 +#define NAU8810_NMICPGA_SFT 1 + +/* NAU8810_REG_PGAGAIN (0x2D) */ +#define NAU8810_PGAGAIN_SFT 0 +#define NAU8810_PGAMT_SFT 6 +#define NAU8810_PGAZC_SFT 7 + +/* NAU8810_REG_ADCBOOST (0x2F) */ +#define NAU8810_PMICBSTGAIN_SFT 4 +#define NAU8810_PGABST_SFT 8 + +/* NAU8810_REG_SPKMIX (0x32) */ +#define NAU8810_DACSPK_SFT 0 +#define NAU8810_BYPSPK_SFT 1 + +/* NAU8810_REG_SPKGAIN (0x36) */ +#define NAU8810_SPKGAIN_SFT 0 +#define NAU8810_SPKMT_SFT 6 +#define NAU8810_SPKZC_SFT 7 + +/* NAU8810_REG_MONOMIX (0x38) */ +#define NAU8810_DACMOUT_SFT 0 +#define NAU8810_BYPMOUT_SFT 1 +#define NAU8810_MOUTMXMT_SFT 6 + + +/* System Clock Source */ +enum { + NAU8810_CLK_MCLK, + NAU8810_CLK_PLL, +}; + +/* Clock divider Id's */ +enum { + NAU8810_MCLK_DIV_PLL, + NAU8810_MCLK_DIV_MCLK, + NAU8810_BCLK_DIV, +}; + +struct nau8810_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +struct nau8810 { + struct device *dev; + struct regmap *regmap; + struct nau8810_pll pll; + int sysclk; + int div_id; +}; + +#endif -- 2.6.4

2 8

[alsa-devel] [PATCH 00/17] Baytrail audio fixes
by Pierre-Louis Bossart 18 Aug '16

18 Aug '16

This second set of patches solves problems with Baytrail-CR, specifically the auto-detection of IRQ index and SSP0 routing. These patches enable sound on the Asus T100TAF with the support of the MCLK handled by the clock framework (Credits to Irina Tirdea) If you prefer git access, the patches are available at https://github.com/plbossart/sound/commits/t100taf-6 The corresponding UCM file is at https://github.com/plbossart/UCM/commits/t100-test Thanks to Luka Karinja and Antonio Ospite for early tests and suggestions, as well as Liam Girdwood and Vinod Koul for internal code reviews. Additional fixes for UCM files will be coming soon to support mono speakers, IN3 input and other fixes. Irina Tirdea (3): clk: x86: Add Atom PMC platform clocks ASoC: Intel: bytcr_rt5640: add MCLK support AsoC: Intel: Add quirks for MinnowBoard MAX Pierre-Louis Bossart (14): ASoC: Intel: bytcr-rt5640: add Asus T100TAF quirks ASoC: Intel: bytcr_rt5640: quirk for mono speaker ASoC: Intel: bytcr_rt5640: enable differential mic quirk ASoC: Intel: Atom: auto-detection of Baytrail-CR ASoC: Intel: Atom: add definitions for modem/SSP0 interface ASoC: Intel: atom: enable configuration of SSP0 ASoC: Intel: bytcr_rt5640: add SSP2_AIF2 routing ASoC: Intel: bytcr_rt56040: additional routing quirks ASoC: Intel: bytcr_rt5640: fix dai/clock setup for SSP0 routing ASoC: Intel: bytcr_rt5640: default routing and quirks on Baytrail-CR ASoC: Intel: bytcr_rt5640: add IN3 map ASoC: rt5640: add internal clock source support ASoC: bytcr_rt5640: Add quirk for Teclast X98 Air 3G tablet ASoC: Intel: bytcr_rt5640: log quirks 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 +++++++++++++++++++++++++ sound/soc/codecs/rt5640.c | 3 + sound/soc/codecs/rt5640.h | 1 + sound/soc/intel/atom/sst-atom-controls.c | 32 +- sound/soc/intel/atom/sst-atom-controls.h | 6 + sound/soc/intel/atom/sst/sst_acpi.c | 82 ++++- sound/soc/intel/boards/bytcr_rt5640.c | 496 ++++++++++++++++++++++++++++--- 11 files changed, 1036 insertions(+), 60 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c -- 2.5.0

2 39

[alsa-devel] [PATCH for 4.8] Revert "ASoC: omap-mcpdm: Add support for pdmclk clock handling"
by Peter Ujfalusi 18 Aug '16

18 Aug '16

This reverts commit 65aca64d05b5eaa5ce15e18b458a8d338ddbd478. The patches for twl6040 MFD and clk missed the merge window and causing the McPDM driver to never probe since it is put back to the deferred list because the missing drivers. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Mark, OMAP4/5 audio regressed in 4.8 due to the hack that the twl6040 MFD and clk driver changes were not take in time. The only clean solution I can think is to revert the pdmclk clock handling from the McPDM driver for now. Regards, Peter Documentation/devicetree/bindings/sound/omap-mcpdm.txt | 10 ---------- sound/soc/omap/omap-mcpdm.c | 17 ----------------- 2 files changed, 27 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt index 6f6c2f8e908d..0741dff048dd 100644 --- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt +++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt @@ -8,8 +8,6 @@ Required properties: - interrupts: Interrupt number for McPDM - interrupt-parent: The parent interrupt controller - ti,hwmods: Name of the hwmod associated to the McPDM -- clocks: phandle for the pdmclk provider, likely <&twl6040> -- clock-names: Must be "pdmclk" Example: @@ -21,11 +19,3 @@ mcpdm: mcpdm@40132000 { interrupt-parent = <&gic>; ti,hwmods = "mcpdm"; }; - -In board DTS file the pdmclk needs to be added: - -&mcpdm { - clocks = <&twl6040>; - clock-names = "pdmclk"; - status = "okay"; -}; diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index e7cdc51fd806..74d6e6fdcfd0 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -31,7 +31,6 @@ #include <linux/err.h> #include <linux/io.h> #include <linux/irq.h> -#include <linux/clk.h> #include <linux/slab.h> #include <linux/pm_runtime.h> #include <linux/of_device.h> @@ -55,7 +54,6 @@ struct omap_mcpdm { unsigned long phys_base; void __iomem *io_base; int irq; - struct clk *pdmclk; struct mutex mutex; @@ -390,7 +388,6 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int ret; - clk_prepare_enable(mcpdm->pdmclk); pm_runtime_enable(mcpdm->dev); /* Disable lines while request is ongoing */ @@ -425,7 +422,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) pm_runtime_disable(mcpdm->dev); - clk_disable_unprepare(mcpdm->pdmclk); return 0; } @@ -445,8 +441,6 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai) mcpdm->pm_active_count++; } - clk_disable_unprepare(mcpdm->pdmclk); - return 0; } @@ -454,8 +448,6 @@ static int omap_mcpdm_resume(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - clk_prepare_enable(mcpdm->pdmclk); - if (mcpdm->pm_active_count) { while (mcpdm->pm_active_count--) pm_runtime_get_sync(mcpdm->dev); @@ -549,15 +541,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; - mcpdm->pdmclk = devm_clk_get(&pdev->dev, "pdmclk"); - if (IS_ERR(mcpdm->pdmclk)) { - if (PTR_ERR(mcpdm->pdmclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; - dev_warn(&pdev->dev, "Error getting pdmclk (%ld)!\n", - PTR_ERR(mcpdm->pdmclk)); - mcpdm->pdmclk = NULL; - } - ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, &omap_mcpdm_dai, 1); -- 2.9.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: sst: fix ix spelling mistake: "capablities" -> "capabilites"
by Colin King 18 Aug '16

18 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err message and reformat code to avoid being over 80 chars wide per line Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst/sst.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..498b5f7 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -190,7 +190,8 @@ int sst_driver_ops(struct intel_sst_drv *sst) default: dev_err(sst->dev, - "SST Driver capablities missing for dev_id: %x", sst->dev_id); + "SST Driver capabilities missing for dev_id: %x", + sst->dev_id); return -EINVAL; }; } -- 2.8.1

2 1

[alsa-devel] [PATCH 0/3] Baytrail followup patches
by Pierre-Louis Bossart 17 Aug '16

17 Aug '16

fixes for warnings detected by 0-Day New quirk for Acer Aspire platforms Resending clock framework support (for-next audio not working on all platforms enabling MCLK if this patch is not applied) Irina Tirdea (1): clk: x86: Add Atom PMC platform clocks Pierre-Louis Bossart (2): ASoC: Intel: atom: fix 0-day warnings ASoC: Intel: bytcr_rt5640: quirk for Acer Aspire SWS-012 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 ++++++++++++++++++++++++++++++++++ sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 15 +- 7 files changed, 487 insertions(+), 6 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c -- 2.7.4

2 6

[alsa-devel] [PATCH] ASoC: fsl_sai: Use 'np' variable
by Fabio Estevam 17 Aug '16

17 Aug '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> The 'np' variable is already assigned to 'pdev->dev.of_node', so use it to improve readability. No functional change. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 2147994..9fadf7e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -801,8 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || - of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) + if (of_device_is_compatible(np, "fsl,imx6sx-sai") || + of_device_is_compatible(np, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -883,7 +883,7 @@ static int fsl_sai_probe(struct platform_device *pdev) } if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && - of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) { + of_device_is_compatible(np, "fsl,imx6ul-sai")) { gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); if (IS_ERR(gpr)) { dev_err(&pdev->dev, "cannot find iomuxc registers\n"); -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: dapm: Fix typo in comment
by Peter Meerwald-Stadler 16 Aug '16

16 Aug '16

Signed-off-by: Peter Meerwald-Stadler <pmeerw(a)pmeerw.net> --- sound/soc/soc-dapm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8698c26..c46b1ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1169,7 +1169,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @custom_stop_condition: (optional) a function meant to stop the widget graph * walk based on custom logic. * - * Queries DAPM graph as to whether an valid audio stream path exists for + * Queries DAPM graph as to whether a valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * @@ -1294,8 +1294,7 @@ static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) return w->new_power; } -/* Generic check to see if a widget should be powered. - */ +/* Generic check to see if a widget should be powered. */ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; @@ -1646,7 +1645,7 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) struct snd_soc_dapm_context *d = data; int ret; - /* If we're off and we're not supposed to be go into STANDBY */ + /* If we're off and we're not supposed to go into STANDBY */ if (d->bias_level == SND_SOC_BIAS_OFF && d->target_bias_level != SND_SOC_BIAS_OFF) { if (d->dev) @@ -1798,7 +1797,7 @@ static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm) * A complete path is a route that has valid endpoints i.e.:- * * o DAC to output pin. - * o Input Pin to ADC. + * o Input pin to ADC. * o Input pin to Output pin (bypass, sidetone) * o DAC to ADC (loopback). */ @@ -2114,7 +2113,7 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) * soc_dapm_connect_path() - Connects or disconnects a path * @path: The path to update * @connect: The new connect state of the path. True if the path is connected, - * false if it is disconneted. + * false if it is disconnected. * @reason: The reason why the path changed (for debugging only) */ static void soc_dapm_connect_path(struct snd_soc_dapm_path *path, @@ -2233,7 +2232,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, if (w->dapm != dapm) continue; - /* only display widgets that burnm power */ + /* only display widgets that burn power */ switch (w->id) { case snd_soc_dapm_hp: case snd_soc_dapm_mic: @@ -2461,7 +2460,7 @@ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w) switch (w->id) { case snd_soc_dapm_input: - /* On a fully routed card a input is never a source */ + /* On a fully routed card an input is never a source */ if (w->dapm->card->fully_routed) return; ep = SND_SOC_DAPM_EP_SOURCE; @@ -3445,7 +3444,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->endpoints[dir] = -1; } - /* machine layer set ups unconnected pins and insertions */ + /* machine layer sets up unconnected pins and insertions */ w->connected = 1; return w; } -- 1.7.10.4

2 3

Re: [alsa-devel] linux-next: Tree for Aug 16 (mellanox/mlx5 & sound/oss/)
by Randy Dunlap 16 Aug '16

16 Aug '16

On 08/15/16 21:01, Stephen Rothwell wrote: > Hi all, > > Changes since 20160815: > on i386: sound/built-in.o:(.data+0x5c): multiple definition of `dev_list' drivers/built-in.o:(.data+0x6aee8): first defined here ld: Warning: size of symbol `dev_list' changed from 8 in drivers/built-in.o to 16 in sound/built-in.o sound/ is an old oss driver (soundcard.c): struct oss_minor_dev { unsigned short minor; unsigned int enabled; } dev_list[] = { { SND_DEV_DSP16 }, { SND_DEV_AUDIO }, }; ===== drivers/ is net/ethernet/mellanox/mlx5/core/mlx5_core.h: extern struct list_head dev_list; -- ~Randy

1 0

[alsa-devel] [PATCH] ASoC: compress: Fix leak of a widget list in soc_compr_open_fe
by Charles Keepax 16 Aug '16

16 Aug '16

After we have called dpcm_path_get we should make sure to call dpcm_path_put on all error paths. This was not happening causing the allocated widget list to be leaked, this patch corrects this by adding a dpcm_path_put to the error path. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d2df46c..bf7b52f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -121,7 +121,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) dpcm_be_disconnect(fe, stream); fe->dpcm[stream].runtime = NULL; - goto fe_err; + goto path_err; } dpcm_clear_pending_state(fe, stream); @@ -136,6 +136,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) return 0; +path_err: + dpcm_path_put(&list); fe_err: if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) fe->dai_link->compr_ops->shutdown(cstream); -- 2.1.4

2 1

[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Don't unconditionally reset SSC on stream startup
by Peter Meerwald-Stadler 16 Aug '16

16 Aug '16

From: Christoph Huber <c.huber(a)bct-electronic.com> commit cbaadf0f90d6 ("ASoC: atmel_ssc_dai: refactor the startup and shutdown") refactored code such that the SSC is reset on every startup; this breaks duplex audio (e.g. first start audio playback, then start record, causing the playback to stop/hang) Signed-off-by: Christoph Huber <c.huber(a)bct-electronic.com> Signed-off-by: Peter Meerwald-Stadler <p.meerwald(a)bct-electronic.com> Cc: Songjun Wu <songjun.wu(a)microchip.com> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- v2 fixes formatting issues and rephrases the comment sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 54c09ac..16e459a 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -299,8 +299,9 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, clk_enable(ssc_p->ssc->clk); ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); - /* Reset the SSC to keep it at a clean status */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Reset the SSC unless initialized to keep it in a clean state */ + if (!ssc_p->initialized) + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: max98371: Delete unnecessary assignment for the field "owner"
by SF Markus Elfring 16 Aug '16

16 Aug '16

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 15 Aug 2016 16:16:00 +0200 The field "owner" is set by the core. Thus delete an unneeded initialisation. Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/max98371.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index cf0a39b..3053879 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -425,7 +425,6 @@ MODULE_DEVICE_TABLE(of, max98371_of_match); static struct i2c_driver max98371_i2c_driver = { .driver = { .name = "max98371", - .owner = THIS_MODULE, .pm = NULL, .of_match_table = of_match_ptr(max98371_of_match), }, -- 2.9.2

2 1

[alsa-devel] [PATCH 1149/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 16 Aug '16

16 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/core/oss/pcm_oss.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index ebc9fdf..0ff0dd8 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -53,11 +53,11 @@ static bool nonblock_open = 1; MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); MODULE_DESCRIPTION("PCM OSS emulation for ALSA."); MODULE_LICENSE("GPL"); -module_param_array(dsp_map, int, NULL, 0444); +module_param_array(dsp_map, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(dsp_map, "PCM device number assigned to 1st OSS device."); -module_param_array(adsp_map, int, NULL, 0444); +module_param_array(adsp_map, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(adsp_map, "PCM device number assigned to 2nd OSS device."); -module_param(nonblock_open, bool, 0644); +module_param(nonblock_open, bool, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(nonblock_open, "Don't block opening busy PCM devices."); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_PCM); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_PCM1); -- 2.9.2

2 1

[alsa-devel] [PATCH v1 00/14] clk/mfd/ASoC/ARM: OMAP4/5: McPDM/twl6040 pdmclk support
by Peter Ujfalusi 15 Aug '16

15 Aug '16

Hi, Changes since v1: - Added Rob's Acked-by to patch 5 and 14 - Check the return value of clk_prepare_enable() calls in the last patch this is something I wanted to do for a long time... First of all: sorry for the cross domain series. I did tested alone all the domain patches and they are not causing any regression. When they come together we will have ability to control the pdmclk needed by McPDM. CLK patches: The driver was w/o DT support and needed some cleanup so we can use the driver. I have also renamed it to twl6040-pdmclk from twl6040-clk to be precise. MFD patches: The regmap_sync() was not working since the twl6040 does not support bulk access and now we are going to create the needed child for the pdmclk. ARM patches: Updatding the board DTS files and add the needed lines for the pdmclk. ASoC patches: The machine driver needs to select the twl6040-clk driver as it will be used by the omap-mcpdm. I needed to do some cleanup and add also suspend/resume support before adding the code for pdmclk handling since I don't want to keep the twl6040 powered on when the board is suspended. At the moment it is not possible to do true dynamic twl6040 power up/down due to pop noises, but I will be looking at that later. The driver in the future will fail if it is not able to get the pdmclk to avoid kernel crash (McPDM registers are not accessible when pdmclk is not available). I have tested the patches by domain and also in all (I hope) permutation they could be. I have not experienced any regression. Regards, Peter --- Peter Ujfalusi (14): clk: twl6040: Correct clk_ops clk: twl6040: Register the clock as of_clk_provider clk: twl6040: Rename the driver and use consistent names in the code mfd: twl6040: The chip does not support bulk access mfd: twl6040: Register child device for twl6040-pdmclk ARM: dts: omap5-board-common: Add pdmclk binding for audio ARM: dts: omap4-panda-common: Add pdmclk binding for audio ARM: dts: omap4-sdp: Add pdmclk binding for audio ARM: dts: omap4-var-som-om44: Add pdmclk binding for audio ARM: dts: omap4-duovero: Add pdmclk binding for audio ASoC: omap: Kconfig: SND_OMAP_SOC_OMAP_ABE_TWL6040 to select CLK_TWL6040 ASoC: omap-mcpdm: Move the WD enable write inside omap_mcpdm_open_streams() ASoC: omap-mcpdm: Support for suspend resume ASoC: omap-mcpdm: Add support for pdmclk clock handling Documentation/devicetree/bindings/mfd/twl6040.txt | 1 + .../devicetree/bindings/sound/omap-mcpdm.txt | 10 +++ arch/arm/boot/dts/omap4-duovero.dtsi | 5 ++ arch/arm/boot/dts/omap4-panda-common.dtsi | 5 ++ arch/arm/boot/dts/omap4-sdp.dts | 5 ++ arch/arm/boot/dts/omap4-var-som-om44.dtsi | 5 ++ arch/arm/boot/dts/omap5-board-common.dtsi | 5 ++ drivers/clk/clk-twl6040.c | 80 ++++++++++++--------- drivers/mfd/twl6040.c | 6 ++ include/linux/mfd/twl6040.h | 2 +- sound/soc/omap/Kconfig | 1 + sound/soc/omap/omap-mcpdm.c | 84 ++++++++++++++++++++-- 12 files changed, 170 insertions(+), 39 deletions(-) -- 2.8.3

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[alsa-devel] [PATCH v4 0/2] amidi delay option
by Felipe F. Tonello 15 Aug '16

15 Aug '16

Small improvement to amidi, adding useful functionality. 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 | 63 ++++++++++++++++++++++++++++++++++++++++++++--------------- 2 files changed, 61 insertions(+), 16 deletions(-) -- 2.9.3

2 3

[alsa-devel] [PATCH v3 0/2] amidi delay option
by Felipe F. Tonello 15 Aug '16

15 Aug '16

Small improvement to amidi, adding useful functionality. change 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 | 60 +++++++++++++++++++++++++++++++++++++++++++---------------- 2 files changed, 58 insertions(+), 16 deletions(-) -- 2.9.3

2 3

[alsa-devel] [PATCH v2 0/2] amidi delay option
by Felipe F. Tonello 15 Aug '16

15 Aug '16

Small improvement to amidi, adding useful functionality. 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 | 66 ++++++++++++++++++++++++++++++++++++++++++++--------------- 2 files changed, 64 insertions(+), 16 deletions(-) -- 2.9.2

3 4

[alsa-devel] [PATCH v8 0/5] ASoC: MT8173 HDMI jack detection
by Philipp Zabel 15 Aug '16

15 Aug '16

Hi, now that kernel v4.8-rc1 is released and includes the new CEC framework in drivers/staging/media/cec, I wonder how to proceed with this. To allow the ASoC drivers to present HDMI cable connection state as an ALSA jack, and to allow them to update sink capabilities when it signals an EDID change, we need a notification mechanism between CEC/HDMI, and ALSA drivers. I had used Russell's hdmi notification prototype to let the CEC/HDMI drivers notify the hdmi-codec and ASoC machine drivers of changes. The mtk_cec driver currently is just a placeholder that only handles the HPD interrupts. Should the notifications be integrated with the CEC framework for those devices that do have CEC support? Changes since v7 ("ASoC: MT8173 HDMI codec support"): - Dropped the already applied ELD control and machine driver patches - Dropped the N/CTS helper patch for now, as the helpers don't exist yet. - Lock ELD mutex while copying ELD in hdmi_eld_ctl_get regards Philipp Philipp Zabel (5): video: rmk's HDMI notification prototype ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver drm/mediatek: hdmi: issue notifications drivers/gpu/drm/mediatek/mtk_cec.c | 11 +++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 3 + drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 ++++++++++++++++ include/linux/hdmi-notifier.h | 44 ++++++++++++ include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 88 +++++++++++++++++++++--- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 21 ++++++ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 21 ++++++ 11 files changed, 252 insertions(+), 8 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.8.1

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[alsa-devel] [GIT PULL]: intel audio firmware updates
by Vinod Koul 15 Aug '16

15 Aug '16

Hi, Please PULL to receive the latest Broxton and Kabylake audio firmware. The following changes since commit c170c8d95794d6aedbaeea44674daaa96baf04f7: linux-firmware: intel: Update Skylake audio firmware (2016-08-04 16:09:21 +0530) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git apl for you to fetch changes up to 92b8db544104bf8f4fa8b5b2bea2f3681ed2e713: linux-firmware: intel: Update Broxton audio firmware (2016-08-12 16:07:01 +0530) ---------------------------------------------------------------- Vinod Koul (2): linux-firmware: intel: Update Kabylake audio firmware linux-firmware: intel: Update Broxton audio firmware WHENCE | 8 ++++---- intel/dsp_fw_bxtn.bin | 2 +- intel/dsp_fw_bxtn_v1118.bin | Bin 0 -> 505608 bytes intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v1037.bin | Bin 0 -> 238920 bytes 5 files changed, 6 insertions(+), 6 deletions(-) create mode 100644 intel/dsp_fw_bxtn_v1118.bin create mode 100644 intel/dsp_fw_kbl_v1037.bin Thanks -- ~Vinod

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Re: [alsa-devel] ASoC: sun4i-codec: playback stall and I/O error with DAPM paths all disabled
by Mark Brown 15 Aug '16

15 Aug '16

On Mon, Aug 15, 2016 at 05:43:55PM +0800, wens Tsai wrote: > What is unexpected is any attempt to play anything under this state makes > the playback software (in my case mpg321) stall, and later report an I/O > error. My guess is that the DAC is still disabled by DAPM, so it doesn't > send any DRQs, and thus the DMA engine is not consuming any data from > userspace. This is normal for ASoC - like you say it'll be becasue the hardware isn't powered up. > I think we should just enable the digital bits of the DAC/ADC all the > time. Or maybe transfer and then discard data if the DAC is off. Not > sure if this is doable though. I expect playback software to work, and > not block, regardless of the hardware status. Powering things up all the time will have a major effect on battery life for systems that care about that. The expectation is that systems with this sort of hardware won't normally be offering end users direct control of the routing, it'll be something that's handled during system integration.

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[alsa-devel] Bug in 3.10 alsa routing?
by noman pouigt 15 Aug '16

15 Aug '16

Hello folks, I am working on enabling a new alsa codec driver and I am stuck with this below routing problem. SND_SOC_DAPM_AIF_OUT("AIF1TX1",) { "AIF3RX1", NULL, "AIF3 Playback" }, { "AIF1TX1 Input 1", NULL, "AIF3RX1" }, { "AIF1TX1", AIF1TX1, "AIF1TX1 Input 1" }, { "AIF1TX1", AIF1TX2, "AIF1TX1 Input 1" }, { "AIF1 Capture", NULL, "AIF1TX1" }, AIF3 Playback and AIF1 Capture is stream name of AIF3 and AIF1 dai respectively. After setting the paths using mixer command and triggering AIF3 playback dai, I see that it is not powering AIF3 Playback widget. After debugging I found the problem to be that ALSA core searches for physically connected or active widget path before it can power on the widget. So I think the codec developer wanted me start the AIF1 capturing to turn on the whole path. However in our system we just want to connect AIF3RX1 to AIF1TX1. So i changed to below and now it is powering on the AIF3 Playback widget. SND_SOC_DAPM_OUTPUT("DUMMY_NODE"), // { "AIF1 Capture", NULL, "AIF1TX1" }, { "DUMMY_NODE", NULL, "AIF1TX1" }, Is ALSA core working as designed as I am using 3.11? Thanks, Variksla

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[alsa-devel] [PATCH v2 0/2] alsabat: test scripts and reference amixer config files
by focus.luo＠linux.intel.com 15 Aug '16

15 Aug '16

From: Focus Luo <focus.luo(a)linux.intel.com> The patch set supplies some automated test scripts and reference amixer config files for linux audio driver testing by using alsa-bat as test tool on Intel platforms. The first patch includes automated test scripts for linux audio driver based on alsa-lib interface by using alsabat as test tool. It supports analog audio and display audio(HDMI/DP) interface. The second patch supplies some amixer reference config file on Intel platforms. Focus Luo (2): alsabat: automation test scripts alsabat: add amixer config files bat/Makefile.am | 1 + bat/tests/Makefile.am | 11 + bat/tests/README | 32 ++ bat/tests/alsabat_main.sh | 178 ++++++++++ bat/tests/analog_audio_playback_and_capture.sh | 84 +++++ bat/tests/asound_state/Makefile.am | 6 + bat/tests/asound_state/asound.state.Broadwell | 439 +++++++++++++++++++++++++ bat/tests/asound_state/asound.state.Haswell | 342 +++++++++++++++++++ bat/tests/asound_state/asound.state.Skylake | 437 ++++++++++++++++++++++++ bat/tests/dp_audio_playback.sh | 83 +++++ bat/tests/dp_audio_subdevice_number.sh | 68 ++++ bat/tests/hdmi_audio_playback.sh | 81 +++++ bat/tests/hdmi_audio_subdevice_number.sh | 65 ++++ bat/tests/map_test_case | 8 + configure.ac | 3 +- 15 files changed, 1837 insertions(+), 1 deletion(-) create mode 100644 bat/tests/Makefile.am create mode 100644 bat/tests/README create mode 100755 bat/tests/alsabat_main.sh create mode 100755 bat/tests/analog_audio_playback_and_capture.sh create mode 100644 bat/tests/asound_state/Makefile.am create mode 100644 bat/tests/asound_state/asound.state.Broadwell create mode 100644 bat/tests/asound_state/asound.state.Haswell create mode 100644 bat/tests/asound_state/asound.state.Skylake create mode 100755 bat/tests/dp_audio_playback.sh create mode 100755 bat/tests/dp_audio_subdevice_number.sh create mode 100755 bat/tests/hdmi_audio_playback.sh create mode 100755 bat/tests/hdmi_audio_subdevice_number.sh create mode 100644 bat/tests/map_test_case -- 1.9.1

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Re: [alsa-devel] twl4030: runtime audio capture channel swapping (kernel 3.18+)
by Peter Ujfalusi 15 Aug '16

15 Aug '16

Hi Akram, On 08/11/16 04:07, Akram Hameed wrote: > Hei Peter, > > Sorry to bother you directly, but I got no response to my questions on the > alsa-devel list and since you seem to be a maintainer of the omap mcbsp and > pcm code, I thought you would: Sorry, it looks like I have missed your mail in alsa-devel. Adding it to CC and also looping Jarkko. > a) Want to be aware of my issues > > b) Might help point me in a direction to fix things. > > In short: I am capturing a stereo audio stream, 16-bit, 48kHz using a gumstix > overo that is quite similar to the older beagleboards (we are using two > variants - one is OMAP3530 based and the other, DM3730). Periodically, left > channel audio will move to right channel and vice versa over time. > > Previously my company was using a branch from kernel 3.0 (made by Steve > Sakoman) and our audio capture was fine - no swapping, ever. Due to > instabilities with the old mmc interface (unpredictable failures using newer > UHS-3 cards), we have been forced to migrate to a newer kernel, and here I > encounter the channel swap issue. > > To me, it seems the problem is like what is described > here: http://processors.wiki.ti.com/index.php/McBSP_Channel_Swapping > Which I presume draws on a discussion you had years ago with one > Ying: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-February/036895.h… > > After spending quite some time reading and investigating the various > registers, I come to the following conclusions: > > 1) Audio codec (TWL) appears to be configured in the same between working > (3.0) and non working (3.18) kernels, except that voice mode is active also > and some gain settings are different (different defaults, I suppose). I have a > comparison of these register maps if you want to see. > > 2) McBSP2 is also configured the same between old and new kernels, though I > note for some reason the ID pulled by dev_dbg looks spurious, though I guess > unrelated to my issue. > Configuring McBSP255 phys_base: 0x49022000 > **** McBSP255 regs **** > DRR2: 0xf88a > DRR1: 0x0000 > DXR2: 0x0000 > DXR1: 0x0000 > SPCR2: 0x0230 > SPCR1: 0x0031 > RCR2: 0x8041 > RCR1: 0x0040 > XCR2: 0x8041 > XCR1: 0x0040 > SRGR2: 0x001f > SRGR1: 0x0f00 > PCR0: 0x000f > *********************** We must drop the DRR1/2 (and probably DXR1/2) dump... By reading the DRR register with the debug code we introduce underflow. The data must be only moved by the DMA from the DATA registers. If we have low threshold it is more likely to cause channel swap and it is for sure going to introduce missing data - the data is going to the kernel log instead of the receive buffer. > 3) My channel swapping occurs when using dma_op_mode element OR threshold mode > in kernel 3.18+ (I have only tested on 3.18 and 4.4, for the record). Setting > a threshold of 960 with period size 10ms reduces the frequency of the swap. > > I did notice that DMA CCR is different between kernel 3 and 3.18 when > operating in ELEMENT mode. > > Version 3 kernel: > > DMA4:Chan5 @ 48056200:CCR=/dev/mem opened. > Memory mapped at address 0x4013b000. > Read at address 0x48056200 (0x4013b200): 0x01084482 > > > Version 3.18 kernel: > > DMA4:Chan5 @ 48056200:CCR=/dev/mem opened. > Memory mapped at address 0xb6fa1000. > Read at address 0x48056200 (0xb6fa1200): 0x010C44A2 <- McBSP2 RX request. > > > Interestingly, it appears the new kernel (3.18+) is using packet mode even > when 'element' is selected. If I choose dma_op_mode THRESHOLD in the older > kernel 3.0, I see the same 0x010C44A2 in the CCR as I do in 3.18+. Yes, I have made a change that if we are in element mode we will still use packet mode to transfer one sample per DMA request. Real element mode will be used only in case of mono audio. If the audio is 2+ channels we are using packet mode to transfer the sample. This does helped to avoid swaps as we send/receive one sample at a time from/to McBSP. > 4) I have not observed playback channel swapping. I am unsure why. > > 5) I enabled underflow and overflow reporting in the McBSP IRQ handler (kernel > 3.18+ since older 3 kernel did not have such a handler and I have not patched > one in yet), and found several interesting things. I can not recall the history, but between 3.0 and 3.18 we might moved to dmaengine based PCM for OMAPs. But I still have no recollection to experience capture side channel swap. > a) Swap definitely is happening due to Overflow, but so far I only observe an > overflow when using dma_op_mode THRESHOLD. When using dma_op_mode THRESHOLD, > the swap does not seem to coincide with an underflow, so I am at a loss to > explain further. How does you application works? What are the parameters ALSA is configured for capture/playback (cat /proc/asound/card0/pcm0p/sub0/hw_params; cat /proc/asound/card0/pcm0c/sub0/hw_params) - number of periods, etc? > b) In dma_op_mode ELEMENT, I get Underflow reported almost every second in > this newer kernel (again, using 10ms period size). Channel swap only seems to > occur ever few hours, however, and I do not believe I have observed the > overflow at all in ELEMENT mode. So, the swap occurs due to underflow, maybe? Underflow in McBSP or Underflow by ALSA? Underflow in McBSP RX tells that DMA is trying to read data when the FIFO is empty. This can only happen if something else is reading data also from the McBSP since we configure the McBSP and sDMA in sync. Overflow in McBSP happens when DAM is failing to read the data out from McBSP FIFO in time - FIFO is full and McBSP discards the incoming data) ALSA underflow is different, it happens when the application fails to process the period in time and the DMA starts to rewrite the buffer. This happens when you have one thread to capture/process/play. If the process/play takes more time than we go round in the boffer (buffer time) we have ALSA underflow. You can try to increase the number of periods or separate the capture and process/play jobs into separate threads. > My plan from now is to get my JTAG up and running and try and make the change > "If you are working with 16-bit stereo data a nice solution is to configure > the McBSP for a single 32-bit element instead of 2 16-bit elements. " The McBSP driver does not have support for this ATM. There is a side effect of this AFAIK: channels will be swapped, but I have not tested it. Using 32bit element instead of 2x16bit helps in case when you do not have FIFO for sure to give more time for the DMA. Also in case of underrun/overrun you will be loosing both channel's data so the swap would not happen. > If you have any helpful suggestions, I would really appreciate them. I am > operating under the assumption I can leave the TWL codec settings as they are > and just 'lie' to McBSP about the data framing. Namely, specify single phase > frame with 1 word of 32 bits, and set DMA packet size to a 32bit word also. > Since 16 bit mode operates by default in dual phase, I am not sure if the > signalling will work correctly, but I guess I will find out. Yeah, this is also going to be a problem. As twl4030 is using I2S when in stereo mode, with fake mono 32bit you can not use I2S signals. But since McBSP is looking for the start condition, configuring it for DSP_A mode will probably going to start the reception/playback at the correct time. > Again, my apologies for contacting you directly. Hopefully the information I > have provided can help diagnosing the real problem in the kernel. No problem. Sorry for that I have missed your mail in alsa-devel. > For me, I > must find a solution in the near-term so I will try to figure out some hacks > as described in the Wiki article. The wiki page was written for daVinci devices AFAIK where they don't have FIFO for their McBSP. -- Péter

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[alsa-devel] [PATCH 0/2] ALSA: rawmidi: add runtime error state for disorder of data transmission
by Takashi Sakamoto 14 Aug '16

14 Aug '16

Hi, This patchset is based on an idea addressed in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104589.h… For packet-oriented drivers such as USB Audio Device class, message transmission can fail due to data bus issue. In this case, corresponding rawmidi runtime should be closed. Although, current ALSA rawmidi core has no mechanism to help applications/drivers to handle this situation. This commit adds a helper functions for this aim. Takashi Sakamoto (2): ALSA: rawmidi: handle error state for broken logical channel ALSA: rawmidi: add a helper to set runtime error include/sound/rawmidi.h | 3 ++ sound/core/rawmidi.c | 86 ++++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 84 insertions(+), 5 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2] drm/i915/dp: DP audio API changes for MST
by Dhinakaran Pandiyan 13 Aug '16

13 Aug '16

DP MST provides the capability to send multiple video and audio streams through a single port. This requires the API's between i915 and audio drivers to distinguish between multiple audio capable displays that can be connected to a port. Currently only the port identity is shared in the APIs. This patch adds support for MST with an additional parameter 'int pipe'. The existing parameter 'port' does not change it's meaning. pipe = MST : display pipe that the stream originates from Non-MST : -1 Affected APIs: struct i915_audio_component_ops - int (*sync_audio_rate)(struct device *, int port, int rate); + int (*sync_audio_rate)(struct device *, int port, int pipe, + int rate); - int (*get_eld)(struct device *, int port, bool *enabled, - unsigned char *buf, int max_bytes); + int (*get_eld)(struct device *, int port, int pipe, + bool *enabled, unsigned char *buf, int max_bytes); struct i915_audio_component_audio_ops - void (*pin_eld_notify)(void *audio_ptr, int port); + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); This patch makes dummy changes in the audio drivers (Libin) for build to succeed. The audio side drivers will send the right 'pipe' values in patches that will follow. v2: Renamed the new API parameter from 'dev_id' to 'pipe'. (Jim, Ville) Included Asoc driver API compatibility changes from Jeeja. Added WARN_ON() for invalid pipe in get_saved_encoder(). (Takashi) Added comment for av_enc_map[] definition. (Takashi) Signed-off-by: Dhinakaran Pandiyan <dhinakaran.pandiyan(a)intel.com> --- drivers/gpu/drm/i915/i915_drv.h | 3 +- drivers/gpu/drm/i915/intel_audio.c | 92 ++++++++++++++++++++++++++------------ include/drm/i915_component.h | 6 +-- include/sound/hda_i915.h | 11 ++--- sound/hda/hdac_i915.c | 9 ++-- sound/pci/hda/patch_hdmi.c | 7 +-- sound/soc/codecs/hdac_hdmi.c | 2 +- 7 files changed, 85 insertions(+), 45 deletions(-) diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h index c36d176..8e4a88f 100644 --- a/drivers/gpu/drm/i915/i915_drv.h +++ b/drivers/gpu/drm/i915/i915_drv.h @@ -2036,7 +2036,8 @@ struct drm_i915_private { /* perform PHY state sanity checks? */ bool chv_phy_assert[2]; - struct intel_encoder *dig_port_map[I915_MAX_PORTS]; + /* Used to save the pipe-to-encoder mapping for audio */ + struct intel_encoder *av_enc_map[I915_MAX_PIPES]; /* * NOTE: This is the dri1/ums dungeon, don't add stuff here. Your patch diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index ef20875..240dad2 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -500,6 +500,7 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) struct i915_audio_component *acomp = dev_priv->audio_component; struct intel_digital_port *intel_dig_port = enc_to_dig_port(encoder); enum port port = intel_dig_port->port; + enum pipe pipe = -1; connector = drm_select_eld(encoder); if (!connector) @@ -524,12 +525,18 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) mutex_lock(&dev_priv->av_mutex); intel_encoder->audio_connector = connector; + /* referred in audio callbacks */ - dev_priv->dig_port_map[port] = intel_encoder; + dev_priv->av_enc_map[pipe] = intel_encoder; mutex_unlock(&dev_priv->av_mutex); + + if (intel_encoder->type == INTEL_OUTPUT_DP_MST) + pipe = crtc->pipe; + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + (int) port, (int) pipe); } /** @@ -542,22 +549,28 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) void intel_audio_codec_disable(struct intel_encoder *intel_encoder) { struct drm_encoder *encoder = &intel_encoder->base; + struct intel_crtc *crtc = to_intel_crtc(encoder->crtc); struct drm_device *dev = encoder->dev; struct drm_i915_private *dev_priv = to_i915(dev); struct i915_audio_component *acomp = dev_priv->audio_component; struct intel_digital_port *intel_dig_port = enc_to_dig_port(encoder); enum port port = intel_dig_port->port; + enum pipe pipe = -1; if (dev_priv->display.audio_codec_disable) dev_priv->display.audio_codec_disable(intel_encoder); mutex_lock(&dev_priv->av_mutex); intel_encoder->audio_connector = NULL; - dev_priv->dig_port_map[port] = NULL; + dev_priv->av_enc_map[pipe] = NULL; mutex_unlock(&dev_priv->av_mutex); + if (intel_encoder->type == INTEL_OUTPUT_DP_MST) + pipe = crtc->pipe; + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + (int) port, (int) pipe); } /** @@ -632,15 +645,39 @@ static int i915_audio_component_get_cdclk_freq(struct device *dev) return dev_priv->cdclk_freq; } -static int i915_audio_component_sync_audio_rate(struct device *dev, - int port, int rate) +static struct intel_encoder *get_saved_encoder(struct intel_encoder *av_enc_map[], + int port, int pipe) +{ + struct drm_encoder *encoder; + + if (WARN_ON(pipe >= I915_MAX_PIPES)) + return NULL; + + /* MST */ + if (pipe != -1) + return av_enc_map[pipe]; + + /* Non-MST */ + for (pipe = PIPE_A; pipe < I915_MAX_PIPES; pipe++) { + if (!av_enc_map[pipe]) + continue; + + encoder = &av_enc_map[pipe]->base; + if (port == enc_to_dig_port(encoder)->port) + return av_enc_map[pipe]; + } + + return NULL; +} + +static int i915_audio_component_sync_audio_rate(struct device *dev, int port, + int pipe, int rate) { struct drm_i915_private *dev_priv = dev_to_i915(dev); struct intel_encoder *intel_encoder; struct intel_crtc *crtc; struct drm_display_mode *mode; struct i915_audio_component *acomp = dev_priv->audio_component; - enum pipe pipe = INVALID_PIPE; u32 tmp; int n; int err = 0; @@ -654,25 +691,20 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, i915_audio_component_get_power(dev); mutex_lock(&dev_priv->av_mutex); + /* 1. get the pipe */ - intel_encoder = dev_priv->dig_port_map[port]; - /* intel_encoder might be NULL for DP MST */ + intel_encoder = get_saved_encoder(dev_priv->av_enc_map, port, pipe); if (!intel_encoder || !intel_encoder->base.crtc || intel_encoder->type != INTEL_OUTPUT_HDMI) { - DRM_DEBUG_KMS("no valid port %c\n", port_name(port)); + DRM_DEBUG_KMS("Not valid for port %c\n", port_name(port)); err = -ENODEV; goto unlock; } + + /* pipe passed from the audio driver will be -1 for non-mst case */ crtc = to_intel_crtc(intel_encoder->base.crtc); pipe = crtc->pipe; - if (pipe == INVALID_PIPE) { - DRM_DEBUG_KMS("no pipe for the port %c\n", port_name(port)); - err = -ENODEV; - goto unlock; - } - DRM_DEBUG_KMS("pipe %c connects port %c\n", - pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; /* port must be valid now, otherwise the pipe will be invalid */ @@ -708,7 +740,7 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, } static int i915_audio_component_get_eld(struct device *dev, int port, - bool *enabled, + int pipe, bool *enabled, unsigned char *buf, int max_bytes) { struct drm_i915_private *dev_priv = dev_to_i915(dev); @@ -717,16 +749,20 @@ static int i915_audio_component_get_eld(struct device *dev, int port, int ret = -EINVAL; mutex_lock(&dev_priv->av_mutex); - intel_encoder = dev_priv->dig_port_map[port]; - /* intel_encoder might be NULL for DP MST */ - if (intel_encoder) { - ret = 0; - *enabled = intel_encoder->audio_connector != NULL; - if (*enabled) { - eld = intel_encoder->audio_connector->eld; - ret = drm_eld_size(eld); - memcpy(buf, eld, min(max_bytes, ret)); - } + + intel_encoder = get_saved_encoder(dev_priv->av_enc_map, port, pipe); + if (!intel_encoder) { + DRM_DEBUG_KMS("Not valid for port %c\n", port_name(port)); + mutex_unlock(&dev_priv->av_mutex); + return ret; + } + + ret = 0; + *enabled = intel_encoder->audio_connector != NULL; + if (*enabled) { + eld = intel_encoder->audio_connector->eld; + ret = drm_eld_size(eld); + memcpy(buf, eld, min(max_bytes, ret)); } mutex_unlock(&dev_priv->av_mutex); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index b46fa0e..545c6e0 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -64,7 +64,7 @@ struct i915_audio_component_ops { * Called from audio driver. After audio driver sets the * sample rate, it will call this function to set n/cts */ - int (*sync_audio_rate)(struct device *, int port, int rate); + int (*sync_audio_rate)(struct device *, int port, int pipe, int rate); /** * @get_eld: fill the audio state and ELD bytes for the given port * @@ -77,7 +77,7 @@ struct i915_audio_component_ops { * Note that the returned size may be over @max_bytes. Then it * implies that only a part of ELD has been copied to the buffer. */ - int (*get_eld)(struct device *, int port, bool *enabled, + int (*get_eld)(struct device *, int port, int pipe, bool *enabled, unsigned char *buf, int max_bytes); }; @@ -97,7 +97,7 @@ struct i915_audio_component_audio_ops { * status accordingly (even when the HDA controller is in power save * mode). */ - void (*pin_eld_notify)(void *audio_ptr, int port); + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); }; /** diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index 796cabf..07fd64e 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -10,8 +10,9 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); void snd_hdac_i915_set_bclk(struct hdac_bus *bus); -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate); -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int pipe, int rate); +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int pipe, bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); @@ -29,13 +30,13 @@ static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { } static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, - hda_nid_t nid, int rate) + hda_nid_t nid, int pipe, int rate) { return 0; } static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, - bool *audio_enabled, char *buffer, - int max_bytes) + int pipe, bool *audio_enabled, + char *buffer, int max_bytes) { return -ENODEV; } diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index c9af022..b99994b 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -201,7 +201,8 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) * This function sets N/CTS value based on the given sample rate. * Returns zero for success, or a negative error code. */ -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int pipe, int rate) { struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; @@ -212,7 +213,7 @@ int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) port = pin2port(codec, nid); if (port < 0) return -EINVAL; - return acomp->ops->sync_audio_rate(acomp->dev, port, rate); + return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); } EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); @@ -236,7 +237,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * thus it may be over @max_bytes. If it's over @max_bytes, it implies * that only a part of ELD bytes have been fetched. */ -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int pipe, bool *audio_enabled, char *buffer, int max_bytes) { struct hdac_bus *bus = codec->bus; @@ -249,7 +250,7 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, port = pin2port(codec, nid); if (port < 0) return -EINVAL; - return acomp->ops->get_eld(acomp->dev, port, audio_enabled, + return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, buffer, max_bytes); } EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d0d5ad8..67890df 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1485,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, -1, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); if (size > 0) { @@ -1739,7 +1739,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ if (codec_has_acomp(codec)) - snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); + snd_hdac_sync_audio_rate(&codec->core, pin_nid, -1, + runtime->rate); non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -2285,7 +2286,7 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state); } -static void intel_pin_eld_notify(void *audio_ptr, int port) +static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) { struct hda_codec *codec = audio_ptr; int pin_nid; diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 2abb742..cf57ab3 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1366,7 +1366,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, return hdac_hdmi_init_dai_map(edev); } -static void hdac_hdmi_eld_notify_cb(void *aptr, int port) +static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) { struct hdac_ext_device *edev = aptr; struct hdac_hdmi_priv *hdmi = edev->private_data; -- 2.5.0

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Re: [alsa-devel] Correct modules for Bay Trail MAX98090 soc?
by Pierre-Louis Bossart 12 Aug '16

12 Aug '16

On 8/11/16 3:42 PM, Laura Abbott wrote: > Hi, > > Fedora got a bug report[1] that sound was failing on Intel Bay Trail Socs > that use the MAX98090 module. I found > > Author: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Date: Mon Feb 8 10:45:38 2016 +0530 > > ASoC: Intel: Load the atom DPCM driver only > DPCM driver is recommended for BYT, CHT based platforms, so if > CONFIG_SND_SST_IPC_ACPI is selected then don't compile the BYT > Device IDs in common ACPI driver to avoid probe conflicts. > Signed-off-by: Pierre-Louis Bossart > <pierre-louis.bossart(a)linux.intel.com> > Acked-by: Jie Yang <yang.jie(a)intel.com> > Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > > which changed the dependencies for CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH. > The set of options Fedora selects means that > CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH > can't be selected. Is there another driver that's supposed to replace > CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH on Bay Trail or do the dependencies > need to be updated? The bugzilla has alsa-info for working and non- > working cases and the Fedora config is attached. If you remove support for all other baytrail options this driver should still be there and selectable. We just can't support both this driver for Chromebooks and the rest for other machines with the same distribution at the moment.

3 6

[alsa-devel] [PATCH] amidi: add delay option
by Felipe F. Tonello 12 Aug '16

12 Aug '16

This patch adds a new options to amidi tool to add a delay (in milliseconds) to each MIDI message. This is very useful when sending firmware updates to a remote device via SysEx or any other use that requires this delay in between messages. Signed-off-by: Felipe F. Tonello <eu(a)felipetonello.com> --- amidi/amidi.1 | 14 ++++++++++++ amidi/amidi.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 81 insertions(+), 5 deletions(-) diff --git a/amidi/amidi.1 b/amidi/amidi.1 index 1b4cfb1ea956..40402c17e374 100644 --- a/amidi/amidi.1 +++ b/amidi/amidi.1 @@ -111,6 +111,12 @@ to stop receiving data. Does not ignore Active Sensing bytes (FEh) when saving or printing received MIDI commands. +.TP +.I \-D, \-\-delay=mseconds +Adds a delay in between each MIDI message sent to a device. It is +useful when sending firmware updates via SysEx messages to a remote +device. + .SH EXAMPLES .TP @@ -121,6 +127,14 @@ to port .I hw:0. .TP +.B amidi \-p hw:1,0,0 -s firmware.syx \-D 100 +will send the MIDI commands in +.I firmware.syx +to port +.I hw:1,0,0 +with 100 milliseconds delay in between MIDI messages. + +.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 cedf18c5fd39..f45abfa7eac0 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -48,6 +48,7 @@ static int receive_file; static int dump; static int timeout; static int stop; +static int delay; static snd_rawmidi_t *input, **inputp; static snd_rawmidi_t *output, **outputp; @@ -77,7 +78,8 @@ static void usage(void) "-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 don't ignore active sensing bytes\n"); + "-a, --active-sensing don't ignore active sensing bytes\n" + "-D, --delay=mseconds delay in between each sent MIDI message\n"); } static void version(void) @@ -305,6 +307,35 @@ static void parse_data(void) send_data_length = i; } +static int message_length(unsigned char byte) +{ + static int rstate_len = 0; + + switch (byte) { + case 0xf8 ... 0xff: + return 1; + case 0xf0: + return -1; /* variable size */ + case 0xf1: + case 0xf3: + return 2; + case 0xf2: + return 3; + case 0xf4: + case 0xf5: + return 0; /* ignore */ + case 0xf6: + return 1; + case 0xf7: + return 0; /* ignore */ + case 0x80 ... 0xef: + rstate_len = (byte >= 0xc0 && byte <= 0xdf) ? 2 : 3; + return rstate_len; + case 0x00 ... 0x7f: + return rstate_len ?: rstate_len - 1; + } +} + /* * prints MIDI commands, formatting them nicely */ @@ -406,7 +437,7 @@ 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:a"; + static const char short_options[] = "hVlLp:s:r:S::dt:aD:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -419,6 +450,7 @@ int main(int argc, char *argv[]) {"dump", 0, NULL, 'd'}, {"timeout", 1, NULL, 't'}, {"active-sensing", 0, NULL, 'a'}, + {"delay", 1, NULL, 'D'}, { } }; int c, err, ok = 0; @@ -463,6 +495,9 @@ int main(int argc, char *argv[]) case 'a': ignore_active_sensing = 0; break; + case 'D': + delay = atoi(optarg); + break; default: error("Try `amidi --help' for more information."); return 1; @@ -538,9 +573,36 @@ 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 (!delay) { + if ((err = snd_rawmidi_write(output, send_data, send_data_length)) < 0) { + error("cannot send data: %s", snd_strerror(err)); + return err; + } + } else { + unsigned char *data = (unsigned char *)send_data; + + while (data < ((unsigned char *)send_data + send_data_length)) { + + if (data > (unsigned char *)send_data) + usleep(delay * 1000); + + int len = message_length(*data); + + /* check for variable length */ + if (len == -1) { + int i = 0; + while (data[i++] != 0xf7); + len = i; + } + + if (len > 0 && + (err = snd_rawmidi_write(output, data, len)) < 0) { + error("cannot send data: %s", snd_strerror(err)); + return err; + } + + data += len ?: 1; + } } } -- 2.9.2

4 5

[alsa-devel] [PATCH] ALSA: hda - repoll jack presence state on resume
by Jaroslav Kysela 12 Aug '16

12 Aug '16

The jack detection on resume might fail on some hardware (Lenovo T430s, T440s and X1 Carbon 2016). The codec registers reports that the jack is not used and the unsolicited event is not sent on change (or perhaps, it's lost?). This patch will repoll the jack presence state after 100ms again. Because the jack status is updated from two workqueues now, create a new codec->jack_mutex to protect the concurrent access. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/hda_bind.c | 2 ++ sound/pci/hda/hda_codec.c | 9 ++++++++- sound/pci/hda/hda_codec.h | 1 + 3 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 6efadbf..81a9284 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -42,8 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + mutex_lock(&codec->jack_mutex); if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); + mutex_unlock(&codec->jack_mutex); } /** diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9913be8..4339e98 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -592,8 +592,10 @@ static void hda_jackpoll_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); + mutex_lock(&codec->jack_mutex); snd_hda_jack_set_dirty_all(codec); snd_hda_jack_poll_all(codec); + mutex_unlock(&codec->jack_mutex); if (!codec->jackpoll_interval) return; @@ -837,6 +839,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); + mutex_init(&codec->jack_mutex); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); @@ -3002,8 +3005,12 @@ static void hda_call_codec_resume(struct hda_codec *codec) if (codec->jackpoll_interval) hda_jackpoll_work(&codec->jackpoll_work.work); - else + else { snd_hda_jack_report_sync(codec); + /* Some hw requires a little time to settle things */ + schedule_delayed_work(&codec->jackpoll_work, + msecs_to_jiffies(100)); + } atomic_dec(&codec->core.in_pm); } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 373fcad..6a4a631 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -213,6 +213,7 @@ struct hda_codec { struct mutex spdif_mutex; struct mutex control_mutex; + struct mutex jack_mutex; struct snd_array spdif_out; unsigned int spdif_in_enable; /* SPDIF input enable? */ const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ -- 2.5.5

2 1

[alsa-devel] [PATCH -next] ASoC: atmel-pdmic: add missing clk_disable_unprepare() on error in atmel_pdmic_cpu_dai_startup()
by Wei Yongjun 12 Aug '16

12 Aug '16

Add the missing clk_disable_unprepare() before return from atmel_pdmic_cpu_dai_startup() in the error handling case. Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/atmel/atmel-pdmic.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 2a3a41f..a67e31b 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -115,8 +115,10 @@ static int atmel_pdmic_cpu_dai_startup(struct snd_pcm_substream *substream, return ret; ret = clk_prepare_enable(dd->pclk); - if (ret) + if (ret) { + clk_disable_unprepare(dd->gclk); return ret; + } /* Clear all bits in the Control Register(PDMIC_CR) */ regmap_write(dd->regmap, PDMIC_CR, 0);

2 1

[alsa-devel] [PATCH -next] ASoC: topology: Fix error return code in soc_tplg_dapm_widget_create()
by Wei Yongjun 12 Aug '16

12 Aug '16

Fix to return error code -ENOMEM instead of 0 when failed to create widget, as done elsewhere in this function. Fixes: 8a9782346dcc ("ASoC: topology: Add topology core") Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/soc-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 05a18f6..a9e83a2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1476,6 +1476,7 @@ widget: if (widget == NULL) { dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", w->name); + ret = -ENOMEM; goto hdr_err; }

2 1

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Enable TX/RX under and overflow interrupts
by Peter Ujfalusi 12 Aug '16

12 Aug '16

FIFO under or overflow can cause channel swaps and data loss. Reporting them can help to identify such events. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/mcbsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 76ce33199bf9..06fec5699cc8 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -221,7 +221,8 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, /* Enable TX/RX sync error interrupts by default */ if (mcbsp->irq) - MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN); + MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN | + RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN); } /** -- 2.9.2

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: Extended SPDIF support to Allwinner A31 SoCs
by codekipper＠gmail.com 12 Aug '16

12 Aug '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set extends the current support of the Allwinner SPDIF transceiver to include the A31 SoC. For now just the SPDIF transmitter has been tested on the Mele I7. In order for this patch set to be functional then we require changes to the clocks and the device tree. For those that are interested I've pushed all the required patches here https://github.com/codekipper/linux-sunxi/commits/a31-spdif Thanks in advance, CK --- Changes since v1: - Corrected mistake in bindings documentation Marcus Cooper (2): ASoC: sunxi: Add bindings for sun6i to SPDIF ASoC: sunxi: compatibility for sun6i to SPDIF .../devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 3 +++ sound/soc/sunxi/sun4i-spdif.c | 17 ++++++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) -- 2.9.2

2 3

[alsa-devel] [PATCH -next] ASoC: Intel: Skylake: Fix error return code in skl_probe()
by Wei Yongjun 12 Aug '16

12 Aug '16

Fix to return error code -ENODEV from the error handling case instead of 0, as done elsewhere in this function. Fixes: 87b2bdf02278 ("ASoC: Intel: Skylake: Initialize NHLT table") Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/intel/skylake/skl.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index a893ca1..2989c16 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -672,8 +672,10 @@ static int skl_probe(struct pci_dev *pci, skl->nhlt = skl_nhlt_init(bus->dev); - if (skl->nhlt == NULL) + if (skl->nhlt == NULL) { + err = -ENODEV; goto out_free; + } skl_nhlt_update_topology_bin(skl);

2 1

Re: [alsa-devel] Underrun with 2 periods on skylake HDA driver
by Dharageswari R 12 Aug '16

12 Aug '16

On Tue, Aug 09, 2016 at 11:54:22PM -0700, Clemens Ladisch wrote: > Dharageswari R wrote: > > While doing the validation of Skylake HDA driver using aplay and > > following parameters, we noticed underrun issues. > > > > buffer_size : 9600 > > period_size : 4800 > > > > After doing further debugging, what I found is > > > > 1.DMA complete interrupt occurs for the first period. > > 2.Driver reads DMA position in buffer pointer and gets value of 4784. > > The interrupt says that the buffer descriptor list entry has been > completed, but the position in buffer pointer says that the entry has > not yet been completed. > yes > > I checked with HDA architects and they are saying that when the DMA > > completion interrupt occurs there can be slight delay before the DMA > > Position In Buffer Pointer gets updated. > > The driver's interrupt handler takes care to read the interrupt status > register first, so that any pending memory writes from the device are > flushed. > > Of course, this does not work if the device posts writes _after_ the > interrupt has been raised. This is a bug in the hardware. > > > So driver is expected to refill data based on whatever content is > > reflected in the DMA Position In Buffer. > > No, that's not how it works. The driver waits until all 4800 frames > are available, and the only way to wake it up is with an interrupt. > There is no other interrupt apart from the buffer completion interrupt to know the post write completion. The only option here would be to poll till the position is equal or more than the period boundary with timeout before the period elapsed reporting. I thought of avoiding this if possible but doesnt seem to be true. Thanks for your reply. I will make sure that the driver reports period ellapsed at or above the period boundary. > > It appears the driver must work around this bug by scheduling another > interrupt after the delay. Is there an upper limit (time or frames) for > this delay? The upper limit is not deterministic in this case. > > > Regards, > Clemens --

2 2

Re: [alsa-devel] [PATCH] ALSA: usb: caiaq: audio: don't print error when allocating urb fails
by Takashi Iwai 12 Aug '16

12 Aug '16

On Thu, 11 Aug 2016 22:40:11 +0200, Wolfram Sang wrote: > > kmalloc will print enough information in case of failure. > > Signed-off-by: Wolfram Sang <wsa-dev(a)sang-engineering.com> Applied, thanks. Takashi > --- > sound/usb/caiaq/audio.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c > index 327f8642ca80e6..8f66ba730d69d9 100644 > --- a/sound/usb/caiaq/audio.c > +++ b/sound/usb/caiaq/audio.c > @@ -739,7 +739,6 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) > for (i = 0; i < N_URBS; i++) { > urbs[i] = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL); > if (!urbs[i]) { > - dev_err(dev, "unable to usb_alloc_urb(), OOM!?\n"); > *ret = -ENOMEM; > return urbs; > } > -- > 2.8.1 > >

1 0

[alsa-devel] [PATCH v3 0/4] ALSA: seq: obsolete change of address limit
by Takashi Sakamoto 11 Aug '16

11 Aug '16

Hi, ALSA sequencer core has two types of clients; application and kernel. For kernel clients, the core should handle data in kernel space, meanwhile this is achieved with a hack of change of address limit. This should be purged. This patchset is a revised version of my previous one: [alsa-devel] obsolete change of address limit http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/110937.html Changes in v2: - Use '_IOC_SIZE' macro to calculate the size of argument of ioctl command. - Keep redundant longer name. - Improve comments. - Merge some patches relevant to the same features. Changes in v3: - Improve series of patch - Use 'dir' field of ioctl command to detect the length of argument Regards Takashi Sakamoto (4): ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: add an alternative way to handle ioctl requests ALSA: seq: change ioctl command operation to get data in kernel space ALSA: seq: obsolete change of address limit sound/core/seq/seq_clientmgr.c | 721 ++++++++++++++++++----------------------- sound/core/seq/seq_compat.c | 7 +- 2 files changed, 317 insertions(+), 411 deletions(-) -- 2.7.4

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Re: [alsa-devel] [PATCH v8 1/5] video: rmk's HDMI notification prototype
by Russell King - ARM Linux 11 Aug '16

11 Aug '16

On Thu, Aug 11, 2016 at 04:49:28PM +0200, Hans Verkuil wrote: > On 08/11/16 16:16, Russell King - ARM Linux wrote: > > On Thu, Aug 11, 2016 at 12:49:21PM +0200, Hans Verkuil wrote: > >> I thought this was the raw EDID data. But if you pass a struct edid around, > >> then why not make this const struct edid *edid? I fail to see why you would > >> want to use a void pointer here. > > > > struct edid is a DRM thing - it's not generic. Using struct edid here > > would force everyone to use the DRM structure, whereas other subsystems > > use, eg, unsigned char. > > So how the edid pointer should be interpreted depends on which subsystem > sends it? That doesn't sound right. It makes it really hard for the drivers > receiving the notifications. No, that's wrong... and I really don't see why you're trying to make a meal of this. In both cases, it's the EDID as received from the device. One subsystem happens to represent it internally as a "struct edid" in its drivers, another subsystem uses "unsigned char []". In order to _cleanly_ accept both styles, without having to resort to driver authors needing to litter their code with lots of idiotic casts, I'm accepting a void * pointer, so that we can accept _both_ a struct edid and an unsigned char array. Both are compatible with each other. Both have the same bytes at the same offset. There is no difference between them other than the C types that are used by each subsystem. > But I understand that currently the main purpose of the hdmi_event_new_edid > is to get the physical address from the EDID for CEC? Or do you have other > uses as well? That is one use, but if an audio driver wants to parse the raw EDID instead of the ELD, it can. What I want to avoid is having the video driver parse lots of different bits from the EDID and then send lots of events "just in case" something wants some of the parsed information - it's much more efficient to give drivers the raw information and allow them to parse what they need from the EDID. I'm not convinced that parsing out the physical address, and then passing that from the video side is a good idea. The reason I made an exception for the ELD is because (a) it's already passed between video and audio drivers, (b) the code we have present to convert an EDID to ELD is in DRM, and requires DRM data structures which won't be available in an audio driver. If you want a mechanism to pass just the physical address, we could add it, and require CEC drivers to make use of both the EDID or physical address events. That would mean situations where the physical address is known to be fixed before hand can be easily catered for. -- RMK's Patch system: http://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line: currently at 9.6Mbps down 400kbps up according to speedtest.net.

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Re: [alsa-devel] SND_PCM_STREAM_CAPTURE and SND_PCM_NONBLOCK producing strange artifacts
by Takashi Iwai 11 Aug '16

11 Aug '16

Please don't drop Cc to ML. Also avoid top-post. On Tue, 05 Jul 2016 09:02:12 +0200, Trent Reed wrote: > > Thanks Takashi, > > My implementation does only use the "hw:N" devices, unfortunately - I am > not using MMAP because I was under the impression that you must poll when > directly accessing the hardware. And yes, 48kHz is the exact rate I'm > attempting. :( > I do have an update, though - I'm able to reproduce the issue quite simply > by running the ALSA test application latency with the following command: > > `latency -P <playback-device> -C <capture-device> -r 48000` > > This, as far as I understand from the source code, should open R/W > interleaved streams on ("hw:2" is what I'm using for both, though the > problem persists in non-duplex mode, e.g. "hw:2" "hw:0") in non-blocking > mode without polling. > > What happens is the latency application does normalize around some value > (say, 16ms roundtrip time), but the audio is awful and broken, a lot of > this "static" I'm talking about. It seems to me that it's the same problem > that I have in my above-mentioned sample code. > > I would love to help more, so if I can in any way, please let me know! But > I feel at a loss for how to further debug. The latency program is too complex to analyze the issue. Check arecord with --nonblock --test-nowait options and with the hw device. Does the issue happen always? Or does it depend on the buffer or period size? Takashi > > Thanks, > - Trent Reed > > On Mon, Jul 4, 2016 at 11:52 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 02 Jul 2016 23:58:28 +0200, > > Trent Reed wrote: > > > > > > (Ping) > > > > > > I'm afraid I still don't understand why this is failing. I found that a > > > call to sdn_pcm_wait() when -EAGAIN is returned fixes the problem because > > > it waits for samples to be ready, but I don't understand why it is a > > > problem in the first place. Much of the sample code is written in a way > > > that suggests that the call to snd_pcm_wait() is optional, and only > > > required when directly reading/writing from the device (MMAP, I'm not > > doing > > > this). > > > > > > I'm just wondering what is wrong with basically looping over a call to > > > snd_pcm_readi() in non-blocking mode, reacting to errors, and capturing > > > frames. This produces awful static (which is basically garbage samples), > > is > > > it a requirement that I call snd_pcm_wait() when -EAGAIN is passed back > > to > > > wait for a full period to be ready? > > > > Well, in the API level, it should work as you expected. So there must > > be definitely some bug(s). But it's hard to spot out, as there are > > several layers behind the scene. > > > > As a first shot, try to reproduce without alsa-lib plugins, i.e. only > > with "hw" PCM device, if not done yet. Also, it's interesting to see > > whether it happens with or without mmap r/w. > > > > Another point is whether it depends on the parameter. Did you try > > 48kHz instead of 44.1kHz? > > > > > > Takashi > > > > > > > > > > Thanks, > > > - Trent Reed > > > > > > > > > On Tue, Jun 28, 2016 at 7:27 PM, Trent Reed <treed0803(a)gmail.com> wrote: > > > > > > > Hey All, > > > > > > > > I've been banging my head against the wall for about a week on this > > bug. > > > > The following gist shows my sample reproduction code: > > > > https://gist.github.com/TReed0803/985c5d5c3d295245e19006a27be447c3 > > > > > > > > I'm simply opening up a non-blocking PCM capture stream and writing the > > > > contents of the reads to stdout. > > > > (Originally, I was writing to the playback stream, but I was hearing > > this > > > > strange static occasionally!) > > > > > > > > It's the static I'm trying to debug. It doesn't happen every time. In > > > > fact, sometimes I'll go a few consecutive executions without hearing > > it. > > > > I was able to capture some of the bad data, and I loaded it up in > > Audacity > > > > for visualization: > > > > > > > > > > https://drive.google.com/file/d/0B-1aumGKQcQTcUJoZzIwRWhYSFE/view?usp=shari… > > > > > > > > It looks like the internal buffer occasionally is sending me more data > > > > than it actually captured, and I end up either reading old PCM data > > from > > > > the internal ring buffer, or (at the very beginning) a bunch of zeros. > > > > > > > > Can anyone help me understand what is going on? What could cause these > > > > definitely incorrect samples to be recorded? (I get the same effect > > > > regardless of hardware, but I will list hardware just in case.) > > > > > > > > I hope I have all the information you might need: > > > > Hardware: Samson Meteor Mic (USB-Audio, USB Mixer) [Though, it even > > > > happens with my built-in microphone] > > > > ALSA version: Advanced Linux Sound Architecture Driver Version > > > > k4.4.0-28-generic. > > > > apt-cache policy (installed alsa-lib version): 1.1.0-0ubuntu1 > > > > > > > > Thanks, > > > > - Trent Reed > > > > > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

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