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

[alsa-devel] Applied "ASoC: tas2552: fix dBscale-min declaration" to the asoc tree
by Mark Brown 06 Oct '15

06 Oct '15

The patch ASoC: tas2552: fix dBscale-min declaration has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e2600460bc3aa14ca1df86318a327cbbabedf9a8 Mon Sep 17 00:00:00 2001 From: Andreas Dannenberg <dannenberg(a)ti.com> Date: Mon, 5 Oct 2015 15:00:14 -0500 Subject: [PATCH] ASoC: tas2552: fix dBscale-min declaration The minimum volume level for the TAS2552 (control register value 0x00) is -7dB however the driver declares it as -0.07dB. Running amixer before the patch reports: dBscale-min=-0.07dB,step=1.00dB,mute=0 Running amixer with the patch applied reports: dBscale-min=-7.00dB,step=1.00dB,mute=0 Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/tas2552.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index e3a0bca..cc1d398 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = { /* * DAC digital volumes. From -7 to 24 dB in 1 dB steps */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0); static const char * const tas2552_din_source_select[] = { "Muted", -- 2.5.0

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[alsa-devel] [PATCH v2] ASoC: rt5645: Add the HW EQ for the customized speaker output of Google Celes
by Oder Chiou 06 Oct '15

06 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 1 + sound/soc/codecs/rt5645.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5645.h | 5 ++++ 3 files changed, 75 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf615..cf944e4 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,7 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Invert JD when jack insert */ bool jd_invert; + unsigned int eq_mode; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c4fe6b..def9caa 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -224,6 +224,36 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5647_eq_parameter { + unsigned int reg; + unsigned int val; +}; + +#define EQ_REG_NUM 56 +struct hweq_s { + struct rt5647_eq_parameter par[EQ_REG_NUM]; + unsigned int ctrl; +}; + +static const struct hweq_s hweq_param[] = { + { /* EQ_NONE */ + { + {0, 0}, + }, + 0x0000, + }, + { /* EQ_GOOGLE_CELES */ + { + {0x1c0, 0x04b8}, + {0x1c1, 0xfd02}, + {0x1c2, 0x04b8}, + {0x1c3, 0xfd02}, + {0, 0}, + }, + 0x0020, + }, +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -631,6 +661,27 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_update_eqmode(struct snd_soc_codec *codec, + unsigned int mode) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int i; + + for (i = 0; i < EQ_REG_NUM; i++) { + if (hweq_param[mode].par[i].reg) + regmap_write(rt5645->regmap, + hweq_param[mode].par[i].reg, + hweq_param[mode].par[i].val); + else + break; + } + + snd_soc_update_bits(codec, RT5645_EQ_CTRL2, 0x33fe, + hweq_param[mode].ctrl); + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1532,9 +1583,11 @@ static int rt5645_spk_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 rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: + rt5645_update_eqmode(codec, rt5645->pdata.eq_mode); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1596,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + rt5645_update_eqmode(codec, EQ_NONE); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3205,6 +3259,20 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } +static struct rt5645_platform_data celes_platform_data = { + .dmic1_data_pin = RT5645_DMIC1_DISABLE, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, + .eq_mode = EQ_GOOGLE_CELES, +}; + +static int celes_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &celes_platform_data; + + return 1; +} + static const struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", @@ -3215,7 +3283,7 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { }, { .ident = "Google Celes", - .callback = strago_quirk_cb, + .callback = celes_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 13ac732..89f25b8 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2187,6 +2187,11 @@ enum { RT5645_AD_MONO_R_FILTER = (0x1 << 5), }; +enum { + EQ_NONE, + EQ_GOOGLE_CELES, +}; + int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec, unsigned int filter_mask, unsigned int clk_src); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: Document DAI signal polarity
by Anatol Pomozov 06 Oct '15

06 Oct '15

Currently there is no clear definition of what is FSYNC polarity. Different drivers use its own definition of what is "normal" and what is "inverted" fsync. This leads to compatibility problems between drivers. For example TegraX1 driver assumes that DSP-A format with frames starting at rising FSYNC edge has "negative" polarity, while RT5677 assumes it is "positive" polarity. Explicitly specify meaning of BCLK/FSYNC polarity to avoid future compatibility problems. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/sound/soc-dai.h | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..45198db 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,22 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * - * Specifies whether the DAI can also support inverted clocks for the specified - * format. + * BCLK: + * - "normal" polarity means signal is available at rising edge of BCLK + * - "inverted" polarity means signal is available at falling edge of BCLK + * + * FSYNC "normal" polarity depends on the frame format: + * - I2S: frame consists of left then right channel data. Left channel starts + * with falling FSYNC edge, right channel starts with rising FSYNC edge. + * - I2S right/left justified: frame consists of left then right channel data. + * Left channel starts with rising FSYNC edge, right channel starts with + * falling FSYNC edge. + * - DSP A/B: Frame starts with rising FSYNC edge. + * - AC97: Frame starts with rising FSYNC edge. + * + * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- 2.6.0.rc2.230.g3dd15c0

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[alsa-devel] [PATCH alsa-lib] hwdep: add FireWire digi00x/tascam driver support
by Takashi Sakamoto 05 Oct '15

05 Oct '15

Linux 4.4 newly support these drivers. Their hardware dependent interfaces give functionalities to get hardware information, to lock/unlock kernel streaming and to receive lock status event. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/hwdep.h | 4 +++- include/sound/asound.h | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/include/hwdep.h b/include/hwdep.h index 3d3c31b..2fe78cd 100644 --- a/include/hwdep.h +++ b/include/hwdep.h @@ -74,8 +74,10 @@ typedef enum _snd_hwdep_iface { SND_HWDEP_IFACE_FW_FIREWORKS, /**< Echo Audio Fireworks based device */ SND_HWDEP_IFACE_FW_BEBOB, /**< BridgeCo BeBoB based device */ SND_HWDEP_IFACE_FW_OXFW, /**< Oxford OXFW970/971 based device */ + SND_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ + SND_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */ - SND_HWDEP_IFACE_LAST = SND_HWDEP_IFACE_FW_OXFW /**< last known hwdep interface */ + SND_HWDEP_IFACE_LAST = SND_HWDEP_IFACE_FW_TASCAM /**< last known hwdep interface */ } snd_hwdep_iface_t; /** open for reading */ diff --git a/include/sound/asound.h b/include/sound/asound.h index a45be6b..a82108e 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -100,9 +100,11 @@ enum { SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */ SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */ + SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ + SNDRV_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_OXFW + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM }; struct snd_hwdep_info { -- 2.1.4

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[alsa-devel] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered
by Rick Mann 05 Oct '15

05 Oct '15

UPDATE: I was able to read the McASP registers (turns out, its clocks are only enabled while speaker-test is running, so that's the only time the registers are mapped). AHCLKXCTL is 0x00008000, which corresponds to having it set as an output, not inverted, and clock divide-by-1. This seems wrong. I'm having trouble determining what the AUXCLK that feeds the McASP is, but it's not 12 MHz, and without division, I don't see how to get there. For reference, here are the actual values of the McASP transmit registers: GBLCTL 0x00000000 Lots of stuff held in reset (red flag?) XGBLCTL 0x00000000 Same XMASK 0x0000FFFF I think this masks out 16-bits? XFMT 0x00008074 MSB bit first, slot size 16-bit, rot right 16 bits AFSXCTL 0x00000100 2-slot TDM (I2S mode) ACLKXCTL 0x00000000 Sync xmit, ACLKX is from pin AHCLKXCTL 0x00008000 Internal high-freq clock from clock divider XTDM 0x00000003 2 TDM slots active XINTCTL 0x00000000 XSTAT 0x0000010C Xmit error occurred, cur TDM slot is even, xmit clock failure XSLOT 0x0000017F Current time slot (shouldn't this be 0 or 1?) XCLKCHK 0x00000000 XEVTCTL 0x00000000 Thanks, Rick ---- I've been at this for weeks and I'm still unable to get linux to work correctly with my TI codec. This is on a Beaglebone Black running a 4.1.4 kernel (later versions are worse). I've tried multiple versions of the kernel trying to narrow it down. I keep coming back to 4.1.4 as the most recent that at least generates some audio clocks. The AM335x mcasp0 should be speaking i2s with the tlv320aic3104. I have my own build from source of the BBB 4.1.4-ti-r9 release (mainline plus TI plus Beaglebone stuff). I've sprinkled some logging around the audio drivers, but so far I don't see much, and I really don't understand how all the pieces fit together. One thing I found with my logging is the mcasp MCLK (AHCLKXDIV) is never set. I have logging calls in __davinci_mcasp_set_clkdiv() (sound/soc/davinci/davinci-mcasp.c), which is the only code I could find that sets the McASP clock, and it's never called. I tried to use devmem2 to read the registers in the SoC to see if they were set correctly, but I just get a bus error (other reads do work). Errors in dmesg when I load the device tree overlay for my cape suggest I still don't have something configured correctly (see below). I'd sure appreciate any insight into these latest issues. Thanks! Symptoms: MCLK should be 12 MHz, but is 24.00 MHz. Buffer underflows when running speaker-test. Errors to console upon loading cape: [ 38.502299] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.517917] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.530425] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.536996] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.564279] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT dmesg upon loading cape overlay (https://github.com/JetForMe/podtique/blob/v1/bbb/cape/Podtique1/BB-ENABLE-P…): [ 568.323587] bone_capemgr bone_capemgr: part_number 'BB-ENABLE-PRU', version 'N/A' [ 568.323629] bone_capemgr bone_capemgr: slot #4: override [ 568.323646] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 568.323663] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-ENABLE-PRU' [ 568.323804] bone_capemgr:capemgr_load_slot:1148: bone_capemgr bone_capemgr: slot #4: Requesting part number/version based 'BB-ENABLE-PRU-00A0.dtbo [ 568.323825] bone_capemgr:capemgr_load_slot:1162: bone_capemgr bone_capemgr: slot #4: Requesting firmware 'BB-ENABLE-PRU-00A0.dtbo' for board-name 'Override Board Name', version '00A0' [ 568.330369] bone_capemgr:capemgr_load_slot:1171: bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; converting to live tree [ 568.341685] gpio-of-helper ocp:gpio_helper: ready [ 568.344695] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; overlay id #0 [ 568.387640] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.387695] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.387712] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.394284] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.410560] snd_soc_core:snd_soc_register_codec:3064: tlv320aic3x-codec 2-0018: codec register 2-0018 [ 568.410614] snd_soc_core:snd_soc_register_dais:2590: tlv320aic3x-codec 2-0018: ASoC: dai register 2-0018 #1 [ 568.410633] snd_soc_core:snd_soc_register_dais:2634: tlv320aic3x-codec 2-0018: ASoC: Registered DAI 'tlv320aic3x-hifi' [ 568.410653] snd_soc_core:snd_soc_register_codec:3138: tlv320aic3x-codec 2-0018: ASoC: Registered codec 'tlv320aic3x-codec.2-0018' [ 568.421529] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.421588] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.421605] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.431455] **** RLM davinci-mcasp.c:1734: davinci_mcasp_probe [ 568.431507] base:of_property_match_string:1398: comparing tx with tx [ 568.431544] base:of_property_match_string:1398: comparing rx with tx [ 568.431555] base:of_property_match_string:1398: comparing rx with rx [ 568.431608] base:of_property_match_string:1398: comparing common with tx [ 568.431621] base:of_property_match_string:1398: comparing common with rx [ 568.431633] base:of_property_match_string:1398: comparing rx with tx [ 568.431643] base:of_property_match_string:1398: comparing rx with rx [ 568.434643] base:of_property_match_string:1398: comparing tx with tx [ 568.437971] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.446651] snd_soc_core:snd_soc_register_dais:2590: davinci-mcasp 48038000.mcasp: ASoC: dai register 48038000.mcasp #1 [ 568.446682] snd_soc_core:snd_soc_register_dais:2634: davinci-mcasp 48038000.mcasp: ASoC: Registered DAI '48038000.mcasp' [ 568.446840] snd_soc_core:snd_soc_add_platform:2892: davinci-mcasp 48038000.mcasp: ASoC: Registered platform '48038000.mcasp' [ 568.450046] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.450100] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.450425] snd_soc_core:snd_soc_dapm_new_dai_widgets:3494: tlv320aic3x-codec 2-0018: ASoC: adding Playback widget [ 568.450448] snd_soc_core:snd_soc_dapm_new_dai_widgets:3513: tlv320aic3x-codec 2-0018: ASoC: adding Capture widget [ 568.450995] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -2 [ 568.451015] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -1 [ 568.451030] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 0 [ 568.451045] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 1 [ 568.451059] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 2 [ 568.451099] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT [ 568.465530] snd_soc_core:soc_new_pcm:2510: davinci_evm ocp:sound: ASoC: registered pcm #0 AIC3X tlv320aic3x-hifi-0 [ 568.466155] davinci_evm ocp:sound: tlv320aic3x-hifi <-> 48038000.mcasp mapping ok [ 568.466185] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Right ADC -> Capture [ 568.466206] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Left ADC -> Capture [ 568.466226] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Right DAC [ 568.466241] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Left DAC upon starting speaker-test: [ 572.786071] snd_soc_core:soc_pcm_open:535: ASoC: tlv320aic3x-hifi <-> 48038000.mcasp info: [ 572.786103] snd_soc_core:soc_pcm_open:536: ASoC: rate mask 0x7fe [ 572.786114] snd_soc_core:soc_pcm_open:538: ASoC: min ch 2 max ch 2 [ 572.786126] snd_soc_core:soc_pcm_open:540: ASoC: min rate 8000 max rate 96000 [ 572.841514] davinci-mcasp 48038000.mcasp: Transmit buffer underflow ...repeated [ 574.087162] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 579.135283] snd_soc_core:close_delayed_work:602: TLV320AIC3X: ASoC: pop wq checking: Playback status: inactive waiting: yes -- Rick Mann rmann(a)latencyzero.com

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[alsa-devel] [asoc:topic/fsl-mega-fast 4/4] sound/soc/fsl/fsl_esai.c:903:5: warning: "CONFIG_PM_SLEEP" is not defined
by kbuild test robot 05 Oct '15

05 Oct '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/fsl-mega-fast head: c64c60763b4e3c72a3520c8d51be858cd67bacb5 commit: c64c60763b4e3c72a3520c8d51be858cd67bacb5 [4/4] ASoC: fsl_esai: Add driver suspend and resume to support MEGA Fast config: x86_64-randconfig-x015-201540 (attached as .config) reproduce: git checkout c64c60763b4e3c72a3520c8d51be858cd67bacb5 # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): >> sound/soc/fsl/fsl_esai.c:903:5: warning: "CONFIG_PM_SLEEP" is not defined [-Wundef] #if CONFIG_PM_SLEEP ^ vim +/CONFIG_PM_SLEEP +903 sound/soc/fsl/fsl_esai.c 887 } 888 889 ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); 890 if (ret) 891 dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); 892 893 return ret; 894 } 895 896 static const struct of_device_id fsl_esai_dt_ids[] = { 897 { .compatible = "fsl,imx35-esai", }, 898 { .compatible = "fsl,vf610-esai", }, 899 {} 900 }; 901 MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); 902 > 903 #if CONFIG_PM_SLEEP 904 static int fsl_esai_suspend(struct device *dev) 905 { 906 struct fsl_esai *esai = dev_get_drvdata(dev); 907 908 regcache_cache_only(esai->regmap, true); 909 regcache_mark_dirty(esai->regmap); 910 911 return 0; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 1/4] ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver
by Zidan Wang 05 Oct '15

05 Oct '15

Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume be called in suspend/resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/imx-spdif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 33da26a..a407e83 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -89,6 +89,7 @@ MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); static struct platform_driver imx_spdif_driver = { .driver = { .name = "imx-spdif", + .pm = &snd_soc_pm_ops, .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, -- 1.9.1

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[alsa-devel] [PATCH v3 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 05 Oct '15

05 Oct '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..5c737f1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

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[alsa-devel] [PATCH linux-next v3 1/1] ASoC: ad193x: add support to ad1934
by Cyrille Pitchen 05 Oct '15

05 Oct '15

The AD1934 codec has no DAC feature. Hence it register mapping is slightly different from the register mapping of other members of the AD193x family. Some ASoC controls and widgets are related to the DAC feature so are not relevant in the case of an AD1934 codec. Signed-off-by: Cyrille Pitchen <cyrille.pitchen(a)atmel.com> --- Hi all, This patch adds support to the AD1934 codec, which is DAC only. Best Regards, Cyrille ChangeLog v3: - rebase patch for linux-next: don't use the CODEC's dapm field, which no longer exists, but call snd_soc_codec_get_dapm() instead. v2: - replace AD193x by AD193X in ad193x-i2c.c - add ad193x_ prefix to adc_audio_paths --- sound/soc/codecs/ad193x-i2c.c | 8 +-- sound/soc/codecs/ad193x-spi.c | 16 +++++- sound/soc/codecs/ad193x.c | 122 ++++++++++++++++++++++++++++++++---------- sound/soc/codecs/ad193x.h | 9 +++- 4 files changed, 121 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..171313664bc8 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..364e82112e8c 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, + { "ad1938", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..76d7966bd72c 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,8 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +365,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(ad193x_adc_audio_paths); + ret = snd_soc_dapm_add_routes(dapm, + ad193x_adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -367,7 +429,8 @@ const struct regmap_config ad193x_regmap_config = { }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +442,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -- 1.8.2.2

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[alsa-devel] [PATCH 0/4] ALSA: hdac: fixes and support for hda codecs
by Vinod Koul 05 Oct '15

05 Oct '15

We had two fixes found on testing. First one checks if stream is in use or not before releasing and second fixs wrong stread id map Then we add structure defination for hdac_ext_dma_params and lastly add helpers for read/write for codecs Jeeja KP (1): ALSA: hdac: Fix to check if stream not in use in release Subhransu S. Prusty (3): ALSA: hdac: Fix incorrect update of stream id mapping ALSA: hdac: structure definition for ext_dma_params ALSA: hdac: Add codec read/write and check power state for widgets include/sound/hdaudio.h | 6 ++++++ include/sound/hdaudio_ext.h | 6 ++++++ sound/hda/ext/hdac_ext_stream.c | 9 ++++---- sound/hda/hdac_device.c | 47 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 63 insertions(+), 5 deletions(-) -- 2.4.3

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