December 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Extend FLL function
by John Hsu 26 Feb '16

26 Feb '16

Extend FFL clock source from MCLK/BCLK/FS; And modify some FLL parameter setting. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 100 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/nau8825.h | 14 ++++++- 2 files changed, 86 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 7fc7b4e..16da8ef 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -926,7 +926,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -939,10 +940,20 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FILTER_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FILTER_SW_REF); + /* Disable free-running mode */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_DCO_EN, 0); + /* FLL sigma delta modulator enable */ + if (fll_param->fll_frac) + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + else + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, 0); } /* freq_out must be 256*Fs in order to achieve the best performance */ @@ -970,6 +981,37 @@ static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, return 0; } +static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) +{ + int ret = 0; + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + goto done; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + goto done; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + goto done; + } + } + +done: + return ret; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -981,29 +1023,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); - - /* We selected MCLK source but the clock itself managed externally */ - if (!nau8825->mclk) - break; - - if (!nau8825->mclk_freq) { - ret = clk_prepare_enable(nau8825->mclk); - if (ret) { - dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); - return ret; - } - } - - if (nau8825->mclk_freq != freq) { - nau8825->mclk_freq = freq; - - freq = clk_round_rate(nau8825->mclk, freq); - ret = clk_set_rate(nau8825->mclk, freq); - if (ret) { - dev_err(nau8825->dev, "Unable to set mclk rate\n"); - return ret; - } - } + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; break; case NAU8825_CLK_INTERNAL: @@ -1018,6 +1040,30 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, } break; + case NAU8825_CLK_FLL_MCLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_BLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_FS: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; default: dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); return -EINVAL; diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..e8f7ca5 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -112,12 +112,21 @@ /* FLL3 (0x06) */ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8825_FLL_CLK_SRC_SFT 10 +#define NAU8825_FLL_CLK_SRC_MASK (0x3 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_MCLK (0 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_BLK (0x2 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 13) +#define NAU8825_FLL_FILTER_SW_N2 (0x1 << 13) +#define NAU8825_FLL_FILTER_SW_REF (0x0 << 13) /* FLL6 (0x9) */ #define NAU8825_DCO_EN_MASK (0x1 << 15) @@ -306,6 +315,9 @@ enum { NAU8825_CLK_MCLK = 0, NAU8825_CLK_INTERNAL, + NAU8825_CLK_FLL_MCLK, + NAU8825_CLK_FLL_BLK, + NAU8825_CLK_FLL_FS, }; struct nau8825 { -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: pxa: remove incorrect do_div() call
by Arnd Bergmann 26 Feb '16

26 Feb '16

The new optimized do_div implementation (now in asm-generic/next) exposes a glitch in the brownstone audio driver by producing a compile-time warning: sound/soc/pxa/brownstone.c: In function 'brownstone_wm8994_hw_params': sound/soc/pxa/brownstone.c:67:85: warning: comparison of distinct pointer types lacks a cast sound/soc/pxa/brownstone.c:67:10125: warning: right shift count >= width of type [-Wshift-count-overflow] sound/soc/pxa/brownstone.c:67:10254: warning: passing argument 1 of '__div64_32' from incompatible pointer type [-Wincompatible-pointer-types] The driver just divides two plain integer values, so it should not use do_div to start with, but has apparently done so ever since the code was first merged. This replaces do_div with a simple division operator. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- I think this one is harmless, so we only need it in 4.5 to avoid the warning, but not backported to earlier kernels. Found today on ARM allmodconfig diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 6147e86e9b0f..416ea646c3b1 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -63,8 +63,7 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, sysclk = params_rate(params) * 512; sspa_mclk = params_rate(params) * 64; } - sspa_div = freq_out; - do_div(sspa_div, sspa_mclk); + sspa_div = freq_out / sspa_mclk; snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);

3 4

[alsa-devel] [PATCH 3/3] ASoC: fsl_ssi: remove register defaults
by Maciej S. Szmigiero 22 Feb '16

22 Feb '16

There is no guarantee that on fsl_ssi module load SSI registers will have their power-on-reset values. In fact, if the driver is reloaded the values in registers will be whatever they were set to previously. This fixes hard lockup on fsl_ssi module reload, at least in AC'97 mode. Fixes: 05cf237972fe ("ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast") Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 40dfd8a36484..f6ef7d64acb3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,20 +112,6 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; -static const struct reg_default fsl_ssi_reg_defaults[] = { - {CCSR_SSI_SCR, 0x00000000}, - {CCSR_SSI_SIER, 0x00003003}, - {CCSR_SSI_STCR, 0x00000200}, - {CCSR_SSI_SRCR, 0x00000200}, - {CCSR_SSI_STCCR, 0x00040000}, - {CCSR_SSI_SRCCR, 0x00040000}, - {CCSR_SSI_SACNT, 0x00000000}, - {CCSR_SSI_STMSK, 0x00000000}, - {CCSR_SSI_SRMSK, 0x00000000}, - {CCSR_SSI_SACCEN, 0x00000000}, - {CCSR_SSI_SACCDIS, 0x00000000}, -}; - static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -190,8 +176,6 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .reg_defaults = fsl_ssi_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg,

4 27

[alsa-devel] [PATCH v2 00/14] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 15 Feb '16

15 Feb '16

This patch series adds DP audio and hotplug notification support. On Skylake two DP ports are available and to enable DP on both ports all pins need to be enabled. There is a special vendor widget which need to be programmed to enable all pins and converters. This series adds hotplug notification, read/set constraint based on ELD, enable all pin/cvts, DP1.2, programs the audio infoframe for DP. There is a one to one mapping between converter and stream, so the dais are created based on the no of streams supported on hdmi codec. Even though cvts can be mapped dynamically to the streams, currently it is statically mapped as simultaneous playback on both DP and HDMI is not supported as of now. Pin muxes and controls are created dynamically to map converter to pin widget. So at run time specific pin is mapped to the dai based on the control selected (based on the display type DP/HDMI connected). Finally the DP audio infoframe programming is added to support the DP feature. Also with hotplug notification support, ELD is read and capabilities are set for rate, formats and channels. drm_eld sound/core framework is updated to limit the formats based on ELD. There are few fixes one fixing the static checker warning and other one not to fail if no connection list is found for a pin widget. Pls note, the 11th patch is adding a small macro for getting connection type in drm header, we have CCed drm folks on that and this one. Pls ack so that we can have this series merged thru sound trees changes in v2: - Address review comments - Dropped devm_xxx for cvt and pin memory allocation. - Dropped the ELD helper APIs from core and copied to driver. - Updated notification patch to add PM fixes from legacy hda. - commit message and change log updated for display power fix. - Updated DRM helper APIs and the caller Jeeja KP (1): ASoC: hdac_hdmi: Add codec suspend/resume handler Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep display active while enumerating codec Subhransu S. Prusty (12): ASoC: hdac_hdmi: Fix to check num nodes correctly ASoC: hdac_hdmi: Fix to warn instead of err for no connected nids ASoC: hdac_hdmi - Use list to add pins and converters ASoC: hdac_hdmi: Add hotplug notification and read eld ALSA: pcm: Add DRM helper to set constraint for format ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi - create dais based on number of streams ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Assign pin for stream based on dapm connection drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio include/drm/drm_edid.h | 12 + sound/core/pcm_drm_eld.c | 42 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1019 ++++++++++++++++++++++++++++++++++++------ 4 files changed, 925 insertions(+), 149 deletions(-) -- 1.9.1

4 32

[alsa-devel] [PATCH 0/5] ASoC: topology: Add support for FE DAI & DAI links
by mengdong.lin＠linux.intel.com 15 Feb '16

15 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows topology to create FE DAI and DAI links from the PCM topology objects defined by the user space. Mengdong Lin (5): ASoC: Define soc_add_dai() to add a DAI to a component ASoC: Support registering a DAI dynamically ALSA: pcm: Add snd_pcm_rate_range_to_bits() ASoC: topology: Add FE DAIs dynamically ASoC: topology: Add FE DAI links dynamically include/sound/pcm.h | 2 + include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 21 +++-- include/sound/soc.h | 5 +- sound/core/pcm_misc.c | 27 ++++++ sound/soc/soc-core.c | 128 ++++++++++++++++++++-------- sound/soc/soc-topology.c | 195 ++++++++++++++++++++++++++++++++++--------- 7 files changed, 291 insertions(+), 88 deletions(-) -- 2.5.0

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[alsa-devel] Usb audio - unsuported sampling rates
by Gordon Dilan 12 Feb '16

12 Feb '16

Hello, I'm using an usb mic which(according to it's producer should support up to 48kHz sampling rate). The problem is ALSA allows me to use only 16 kHz for this device. I do not want to resample the data, I want to sample it at 48kHz in order to be able to record some high pitch sounds 18-20kHz. I would want to try to do any modifications necessary in order to try support 48kHz, but I'm not sure where to start. Some information: cat /proc/asound/modules ... 1 snd_usb_audio arecord --list-devices **** List of CAPTURE Hardware Devices **** .... card 1: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 when I try to use: arecord -v -D hw:1,0 -f S16_LE -c 4 -r 48000 gigi.wav I get this: Recording WAVE 'gigi.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Warning: rate is not accurate (requested = 48000Hz, got = 16000Hz) please, try the plug plugin Hardware PCM card 1 'USB Camera-B4.09.24.1' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 1 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 Any sampling rate I would try higher or lower than 16kHz, I will always get 16000 kHz. Some more information: lsusb Bus 001 Device 007: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 7, If 0, Class=vend., Driver=ov534, 480M |__ Port 2: Dev 7, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 7, If 2, Class=audio, Driver=snd-usb-audio, 480M At the moment I'm looking inside ALSA's source code to try to see from where this limitation of 16000kHz for my device appears, but I have to admit that I'm kind of lost. Any pointers on what direction should I go to try make my desired rate supported would be greatly appreciated! Thank you and best regards, Gordon

3 6

Re: [alsa-devel] [PATCH 18/19] ASoC: Intel: Atom: add support for RT5642
by Luka Karinja 11 Feb '16

11 Feb '16

> The machine driver is not loaded when the BIOS uses the 10EC5642 > _HID. Add it to the white list of known _HIDs, codec_name is > already taken care of by previous commit > > Tested on Asus T100TAF. > I'm trying to get sound working on my T100TAF i did apply all the 19 patches on top of broonie/sound and added UCM files to my system. Does the TAF need different UCM file than the T100TA? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: bytcrrt5640 [bytcr-rt5640], device 0: Baytrail Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: bytcrrt5640 [bytcr-rt5640], device 1: Deep-Buffer Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 dmesg: [ 5.209239] Adding 1049596k swap on /dev/mmcblk0p4. Priority:-1 extents:1 across:1049596k SSFS [ 5.219099] input: PC Speaker as /devices/platform/pcspkr/input/input6 [ 5.273802] i2c i2c-10: Added multiplexed i2c bus 13 [ 5.274629] intel_sst_acpi 80860F28:00: LPE base: 0x50a00000 size:0x200000 [ 5.274638] intel_sst_acpi 80860F28:00: IRAM base: 0x50ac0000 [ 5.274671] intel_sst_acpi 80860F28:00: DRAM base: 0x50b00000 [ 5.274685] intel_sst_acpi 80860F28:00: SHIM base: 0x50b40000 [ 5.274700] intel_sst_acpi 80860F28:00: Mailbox base: 0x50b44000 [ 5.274712] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [ 5.274842] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [ 5.290156] Error: Driver 'pcspkr' is already registered, aborting... [ 5.311924] random: nonblocking pool is initialized [ 5.397912] EXT4-fs (mmcblk0p2): mounted filesystem with ordered data mode. Opts: data=ordered [ 5.506882] genirq: Flags mismatch irq 8. 00000083 (volume_down) vs. 00000000 (rtc0) [ 5.507025] gpio-keys gpio-keys.0.auto: Unable to claim irq 8; error -16 [ 5.507283] gpio-keys: probe of gpio-keys.0.auto failed with error -16 [ 5.507518] input: gpio-keys as /devices/platform/gpio-keys.1.auto/input/input8 [ 5.519513] iTCO_vendor_support: vendor-support=0 [ 5.591021] iTCO_wdt: Intel TCO WatchDog Timer Driver v1.11 [ 5.591153] iTCO_wdt: Found a Bay Trail SoC TCO device (Version=3, TCOBASE=0x0460) [ 5.604447] iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) [ 5.613586] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 5.613794] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [ 5.613922] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [ 5.615555] bytcr_rt5640 bytcr_rt5640: rt5640-aif1 <-> ssp2-port mapping ok [ 5.642183] asus_wmi: ASUS WMI generic driver loaded [ 5.663786] usbcore: registered new interface driver usbmouse [ 5.665362] usbcore: registered new interface driver usbkbd [ 5.681990] asus_wmi: Initialization: 0x1 [ 5.682121] asus_wmi: BIOS WMI version: 7.9 [ 5.682219] asus_wmi: SFUN value: 0x37 [ 5.685337] input: Asus WMI hotkeys as /devices/platform/asus-nb-wmi/input/input9 [ 5.708007] asus_wmi: Number of fans: 1 [ 5.801413] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.801529] input: SIS0817:00 0457:1071 as /devices/platform/80860F41:03/i2c-11/i2c-SIS0817:00/0018:0457:1071.0004/input/input10 [ 5.802167] hid-multitouch 0018:0457:1071.0004: input,hidraw3: I2C HID v1.00 Device [SIS0817:00 0457:1071] on i2c-SIS0817:00 [ 5.816775] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.816996] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.817204] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.817405] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.817616] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.818884] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.819109] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! [ 5.911351] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.911578] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.911787] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.911997] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.912201] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.912417] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.912634] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.915142] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! [ 6.042680] intel_rapl: Found RAPL domain package [ 6.042692] intel_rapl: Found RAPL domain core [ 6.247467] Bluetooth: BNEP (Ethernet Emulation) ver 1.3 [ 6.247489] Bluetooth: BNEP filters: protocol multicast [ 6.247515] Bluetooth: BNEP socket layer initialized [ 7.524852] brcmf_sdio_drivestrengthinit: No SDIO Drive strength init done for chip 43340 rev 2 pmurev 20 [ 7.525341] usbcore: registered new interface driver brcmfmac [ 7.722004] brcmf_c_preinit_dcmds: Firmware version = wl0: Jun 17 2014 11:48:43 version 6.10.190.49 (r485840) FWID 01-8744bcd2 [ 7.739335] brcmf_cfg80211_reg_notifier: not a ISO3166 code [ 7.899345] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 7.921532] audit: type=1006 audit(1450805770.516:2): pid=1018 uid=0 old-auid=4294967295 auid=1000 old-ses=4294967295 ses=1 res=1 [ 7.984230] audit: type=1006 audit(1450805770.580:3): pid=1091 uid=0 old-auid=4294967295 auid=1000 old-ses=4294967295 ses=2 res=1 [ 7.996410] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 8.151283] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 9.038571] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 11.245916] Bluetooth: hci1: BCM: chip id 82 [ 11.245928] Bluetooth: hci1: BCM43341B0 (002.001.014) build 0000 [ 11.755123] Bluetooth: hci1: BCM (002.001.014) build 0164 [ 14.513027] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x0 fw_state 0x1 [ 14.513045] intel_sst_acpi 80860F28:00: fw download failed -16 [ 14.513067] intel_sst_acpi 80860F28:00: FW download fail -16 [ 14.513864] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.514676] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.515574] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.518058] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.518941] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.520293] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.525383] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.527248] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.528335] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.534111] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port

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[alsa-devel] [PATCH] ASoC: cs42xx8: fix the noise in the right dac channel when playback mono wav with record in background
by Zidan Wang 04 Feb '16

04 Feb '16

When playback mono wav with record in background, there will be some nosie in the right dac channel. It seems that the ADC data has been routed to the dac channel. The cs42888 have 8 dac channels, it's appropriate to mute the unused dac channels, and the noise will disappear. Steps to reproduce this issue: arecord -D hw:0,0 -f S16_LE -r 48000 -c 1 a.wav & aplay -Dhw:0,0 audio48k16M.wav Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/cs42xx8.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index d562e1b..1179101 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -44,6 +44,7 @@ struct cs42xx8_priv { bool slave_mode; unsigned long sysclk; + u32 tx_channels; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -257,6 +258,9 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, u32 ratio = cs42xx8->sysclk / params_rate(params); u32 i, fm, val, mask; + if (tx) + cs42xx8->tx_channels = params_channels(params); + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { if (cs42xx8_ratios[i].ratio == ratio) break; @@ -283,9 +287,11 @@ static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + u8 dac_unmute = cs42xx8->tx_channels ? + ~((0x1 << cs42xx8->tx_channels) - 1) : 0; - regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE, - CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0); + regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, + mute ? CS42XX8_DACMUTE_ALL : dac_unmute); return 0; } -- 1.9.1

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[alsa-devel] [PATCH v2 1/3] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format
by Lars-Peter Clausen 29 Jan '16

29 Jan '16

The DAI link format should be specified for the whole link rather than just one component on the link. So move the format specification for the HDMI audio link from the CPU component to the link itself. Since the sh-mobile-hdmi DAI driver doesn't implement the set_fmt() callback in this case there is no functional difference between only specifying the the format for the CPU side or for the whole link, but the later it will allow us to remove support for just specifying the format for one component. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1. --- arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 6d949f1..75de26c 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1040,9 +1040,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", -- 1.8.0

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[alsa-devel] [PATCH 0/4] ASoC: fix saif record unwork issue
by Dong Aisheng 16 Jan '16

16 Jan '16

This patch series mainly fixes the saif record unwork issue. The root cause consists of: 1) clk_prepare and clk_enable un-balanced 2) missed setting a proper initial clock rate for saif in dt 3) sgtl5000 dapm widget kcontrol does not work So can not test record on mx28evk. 4) missed setting clkmux mode for saif in dt With this patch series it can fix sgtl5000 widget kcontrol unwork and saif record unwork issue. BTW, just let people konw, i still met a few overruns during record, that's another issue and will be fixed in separte patches later. Patch based on sound-2.6/for-3.6. Dong Aisheng (4): ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ARM: mx28evk: add missed clkmux select for saif arch/arm/mach-mxs/mach-mxs.c | 3 +++ sound/soc/codecs/sgtl5000.c | 2 -- sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ 3 files changed, 27 insertions(+), 2 deletions(-)

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