June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Wandboard Quad - SPDIF issue, no sound
by Michael Howard 26 Jun '14

26 Jun '14

Hello All, I'm resending this email as orignally I wasn't subcribed and I've been told the message hasn't made it through. I was pointed here from the Wandboard Google Groups forum. If I'm in the wrong place please accept my appologies. With known good cabling and receiver I'm not getting sound from the Wandboard Quad's S/PDIF output. I have built various kernels from mainline, the last two sources I tried are 3.15.1 and 3.16-rc2. With sound built as modules I get nothing on any outputs, errors from dmesg are; dmesg | grep snd 5.206990] imx-spdif sound-spdif.13: snd_soc_register_card failed: -517 [ 5.241976] imx-spdif sound-spdif.13: snd_soc_register_card failed: -517 [ 5.258174] imx-spdif sound-spdif.13: snd-soc-dummy-dai <-> 2004000.spdif mapping ok [ 5.281704] imx-sgtl5000 sound.12: snd_soc_register_card failed (-517) With sound built in I get sound on the 'line out' output and from dmesg; dmesg | grep snd imx-spdif sound-spdif: snd-soc-dummy-dai <-> 2004000.spdif mapping ok In both cases I get (or similar); dmesg | grep firm imx-sdma 20ec000.sdma: Direct firmware load failed with error -2 imx-sdma 20ec000.sdma: loaded firmware 1.1 Not sure if the firmware message above is a problem or not. aplay output; aplay -l **** List of PLAYBACK Hardware Devices **** card 0: imx6wandboardsg [imx6-wandboard-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: imxspdif [imx-spdif], device 0: S/PDIF PCM snd-soc-dummy-dai-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server sysdefault:CARD=imx6wandboardsg imx6-wandboard-sgtl5000, Default Audio Device sysdefault:CARD=imxspdif imx-spdif, Default Audio Device aplay -vv ImABeliever.wav Playing WAVE 'ImABeliever.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Hardware PCM card 1 'imx-spdif' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 ################# + | 55% but no sound form SPDIF. If this is right place for this and if I can be of any help debuuging or you need more info I'm more than happy to help in any way I can. Since I sent the original message Ive realised that data is being sent to and received at the SPDIF output of the wandboard quad it can't seem to handle it. If I connect a DAC to the SPDIF output and then connect the DAC to my recievers aux input I hear the sound in all it's glory. The same receiver and cabling works with a cubietruck and the sunxi kernel. Cheers, Mike --

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[alsa-devel] [PATCH] ALSA: usb-audio: Remove from list in snd_usb_endpoint_free()
by Takashi Iwai 26 Jun '14

26 Jun '14

snd_usb_endpoint_free() deallocates the object but it doesn't unlink from the list. This is actually no problem in the current usage because all elements are freed at once, but it's more consistent to do a proper unlinking there. Reported-by: Julia Lawall <julia.lawall(a)lip6.fr> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a 3.17 material. sound/usb/endpoint.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 114e3e7ff511..304f36bf29cb 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1011,6 +1011,7 @@ void snd_usb_endpoint_free(struct list_head *head) struct snd_usb_endpoint *ep; ep = list_entry(head, struct snd_usb_endpoint, list); + list_del(head); kfree(ep); } -- 2.0.0

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[alsa-devel] [PATCH 1/1] sound/soc/omap/mcbsp.c: remove unnecessary null test before kfree
by Fabian Frederick 26 Jun '14

26 Jun '14

Fix checkpatch warning: WARNING: kfree(NULL) is safe this check is probably not required Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/soc/omap/mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 86c7538..68a1252 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -621,8 +621,7 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = NULL; spin_unlock(&mcbsp->lock); - if (reg_cache) - kfree(reg_cache); + kfree(reg_cache); } /* -- 1.8.4.5

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[alsa-devel] [PATCH] ALSA: hda - restore BCLK M/N values when resuming HSW/BDW display controller
by mengdong.lin＠intel.com 26 Jun '14

26 Jun '14

From: Mengdong Lin <mengdong.lin(a)intel.com> For Intel Haswell/Broadwell display HD-A controller, the 24MHz HD-A link BCLK is converted from Core Display Clock (CDCLK): BCLK = CDCLK * M / N And there are two registers EM4 and EM5 to program M, N value respectively. The EM4/EM5 values will be lost and when the display power well is disabled. BIOS programs CDCLK selected by OEM and EM4/EM5, but BIOS has no idea about display power well on/off at runtime. So the M/N can be wrong if non-default CDCLK is used when the audio controller resumes, which results in an invalid BCLK and abnormal audio playback rate. So this patch saves and restores valid M/N values on controller suspend/resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bb65a124..90a2750 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -580,6 +580,18 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ +static void haswell_save_bclk(struct azx *chip) +{ + chip->bclk_m = azx_readw(chip, EM4); + chip->bclk_n = azx_readw(chip, EM5); +} + +static void haswell_restore_bclk(struct azx *chip) +{ + azx_writew(chip, EM4, chip->bclk_m); + azx_writew(chip, EM5, chip->bclk_n); +} + #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -606,6 +618,13 @@ static int azx_suspend(struct device *dev) free_irq(chip->irq, chip); chip->irq = -1; } + + /* Save BCLK M/N values before they become invalid in D3. + * Will test if display power well can be released now. + */ + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + haswell_save_bclk(chip); + if (chip->msi) pci_disable_msi(chip->pci); pci_disable_device(pci); @@ -625,8 +644,10 @@ static int azx_resume(struct device *dev) if (chip->disabled) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); + haswell_restore_bclk(chip); + } pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); if (pci_enable_device(pci) < 0) { @@ -670,8 +691,10 @@ static int azx_runtime_suspend(struct device *dev) azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + haswell_save_bclk(chip); hda_display_power(false); + } return 0; } @@ -689,8 +712,10 @@ static int azx_runtime_resume(struct device *dev) if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); + haswell_restore_bclk(chip); + } /* Read STATESTS before controller reset */ status = azx_readw(chip, STATESTS); diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index 4a7cb01..1469edb 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -96,6 +96,14 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ICH6_REG_SD_BDLPL 0x18 #define ICH6_REG_SD_BDLPU 0x1c +/* Intel HSW/BDW display HD-A controller Extended Mode registers. + * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display + * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled. + */ +#define ICH6_REG_EM4 0x100c +#define ICH6_REG_EM5 0x1010 + /* PCI space */ #define ICH6_PCIREG_TCSEL 0x44 @@ -417,6 +425,10 @@ struct azx { /* secondary power domain for hdmi audio under vga device */ struct dev_pm_domain hdmi_pm_domain; + + /* for HSW/BDW display HDA controller to restore BCLK from CDCLK */ + unsigned int bclk_m; + unsigned int bclk_n; }; #ifdef CONFIG_SND_VERBOSE_PRINTK -- 1.8.1.2

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[alsa-devel] [PATCH v7 0/3] Add ES8328 audio codec
by Sean Cross 26 Jun '14

26 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks, in addition to the i.MX audmux routing. Changes since v6: - Minor clarification of AUDMUX in imx-audio-es8328.txt - Use u32 instead of int in imx-es8328.c for int-mux and ext-mux - Bounds checking on mux-int-port and mux-ext-port in imx-es8328.c - Remove NULL checks before calling of_node_put() Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 61 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 719 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 233 +++++++ 12 files changed, 1510 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] alsa-tools 1.0.28 regression with sblive/emu10k1: init_live shell script broken.
by Roc Vallès 25 Jun '14

25 Jun '14

I've recently upgraded to alsa-tools 1.0.28. There's apparently a regression: The script that is bundled for sound blaster live (emu10k1 in my case) soundcard initialization has been broken. Without this, the soundcard can only be used at a very basic level (just 2 speakers, no equalization or other effects, most inputs unavailable). Reporting here as the bugtracker is down. Correct run (1.0.17): # sh /usr/bin/init_live No Emux synth hwdep device is found Found hardware: "EMU10K1" "SigmaTel STAC9721,23" "AC97a:83847609" "0x1102" "0x8040" Hardware is initialized using a generic method Broken run (1.0.28): # /usr/bin/init_live /usr/bin/init_live: line 49: log: command not found /usr/bin/init_live: line 51: local: can only be used in a function /usr/bin/init_live: line 55: [: 0: unary operator expected /usr/bin/init_live: line 60: runlo10k1: command not found /usr/bin/init_live: line 62: log: command not found rename: not enough arguments Usage: rename [options] expression replacement file... Options: -v, --verbose explain what is being done -s, --symlink act on symlink target -h, --help display this help and exit -V, --version output version information and exit For more details see rename(1). /usr/bin/init_live: line 79: log: command not found /usr/bin/init_live: line 81: runlo10k1: command not found /usr/bin/init_live: line 84: connect: command not found /usr/bin/init_live: line 90: log: command not found /usr/bin/init_live: line 92: runlo10k1: command not found /usr/bin/init_live: line 95: copy: command not found /usr/bin/init_live: line 97: local: can only be used in a function /usr/bin/init_live: line 98: local: can only be used in a function /usr/bin/init_live: line 100: log: command not found /usr/bin/init_live: line 103: runlo10k1: command not found /usr/bin/init_live: line 105: connect: command not found /usr/bin/init_live: line 108: volume: command not found /usr/bin/init_live: line 110: local: can only be used in a function /usr/bin/init_live: line 111: local: can only be used in a function /usr/bin/init_live: line 112: local: can only be used in a function /usr/bin/init_live: line 113: local: can only be used in a function /usr/bin/init_live: line 115: log: command not found /usr/bin/init_live: line 118: runlo10k1: command not found /usr/bin/init_live: line 120: connect: command not found /usr/bin/init_live: line 123: master_volume: command not found /usr/bin/init_live: line 125: local: can only be used in a function /usr/bin/init_live: line 126: local: can only be used in a function /usr/bin/init_live: line 127: local: can only be used in a function /usr/bin/init_live: line 128: local: can only be used in a function /usr/bin/init_live: line 130: log: command not found /usr/bin/init_live: line 133: runlo10k1: command not found /usr/bin/init_live: line 135: connect: command not found /usr/bin/init_live: line 138: switch: command not found /usr/bin/init_live: line 140: local: can only be used in a function /usr/bin/init_live: line 141: local: can only be used in a function /usr/bin/init_live: line 142: local: can only be used in a function /usr/bin/init_live: line 144: log: command not found /usr/bin/init_live: line 148: runlo10k1: command not found /usr/bin/init_live: line 150: connect: command not found /usr/bin/init_live: line 153: switch_2x2: command not found /usr/bin/init_live: line 155: local: can only be used in a function /usr/bin/init_live: line 156: local: can only be used in a function /usr/bin/init_live: line 157: local: can only be used in a function /usr/bin/init_live: line 158: local: can only be used in a function /usr/bin/init_live: line 160: log: command not found /usr/bin/init_live: line 163: runlo10k1: command not found /usr/bin/init_live: line 165: connect: command not found /usr/bin/init_live: line 166: connect: command not found /usr/bin/init_live: line 169: switch_5x2: command not found /usr/bin/init_live: line 171: local: can only be used in a function /usr/bin/init_live: line 172: local: can only be used in a function /usr/bin/init_live: line 173: local: can only be used in a function /usr/bin/init_live: line 174: local: can only be used in a function /usr/bin/init_live: line 176: log: command not found /usr/bin/init_live: line 179: runlo10k1: command not found /usr/bin/init_live: line 181: connect: command not found /usr/bin/init_live: line 182: connect: command not found /usr/bin/init_live: line 185: master_switch: command not found /usr/bin/init_live: line 187: local: can only be used in a function /usr/bin/init_live: line 188: local: can only be used in a function /usr/bin/init_live: line 189: local: can only be used in a function /usr/bin/init_live: line 190: local: can only be used in a function /usr/bin/init_live: line 192: log: command not found /usr/bin/init_live: line 195: runlo10k1: command not found /usr/bin/init_live: line 197: connect: command not found /usr/bin/init_live: line 200: playback: command not found /usr/bin/init_live: line 205: log: command not found /usr/bin/init_live: line 208: switch: command not found /usr/bin/init_live: line 209: volume: command not found /usr/bin/init_live: line 211: add_output: command not found /usr/bin/init_live: line 214: playback_noswitch: command not found /usr/bin/init_live: line 219: log: command not found /usr/bin/init_live: line 222: volume: command not found /usr/bin/init_live: line 224: add_output: command not found /usr/bin/init_live: line 227: capture: command not found /usr/bin/init_live: line 232: log: command not found /usr/bin/init_live: line 235: switch: command not found /usr/bin/init_live: line 236: volume: command not found /usr/bin/init_live: line 238: connect: command not found /usr/bin/init_live: line 241: capture_noswitch: command not found /usr/bin/init_live: line 246: log: command not found /usr/bin/init_live: line 249: volume: command not found /usr/bin/init_live: line 251: connect: command not found /usr/bin/init_live: line 254: master: command not found /usr/bin/init_live: line 256: local: can only be used in a function /usr/bin/init_live: line 257: local: can only be used in a function /usr/bin/init_live: line 258: local: can only be used in a function /usr/bin/init_live: line 260: log: command not found /usr/bin/init_live: line 263: master_volume: command not found /usr/bin/init_live: line 264: master_switch: command not found /usr/bin/init_live: line 267: dump_patches: command not found

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[alsa-devel] snd_usb_endpoint_free
by Julia Lawall 25 Jun '14

25 Jun '14

The function snd_usb_endpoint_free in sound/usb/endpoint.c is defined as follows: void snd_usb_endpoint_free(struct list_head *head) { struct snd_usb_endpoint *ep; ep = list_entry(head, struct snd_usb_endpoint, list); release_urbs(ep, 1); kfree(ep); } I wonder if the final kfree should be list_del? In practice, this function is only used from snd_usb_audio_disconnect in sound/usb/card.c where the entire list is destroyed, but it seems like quite a generic function that someone may someday want to use for just freeing one entry. julia

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[alsa-devel] [PATCH] sound: oss: mpu401.c: Cleaning up variable is set more than once
by Rickard Strandqvist 25 Jun '14

25 Jun '14

A struct member variable is set to the same value more than once This was found using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/oss/mpu401.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 3bbc3ec..31d7f72 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -972,7 +972,6 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner) devc->m_busy = 0; devc->m_state = ST_INIT; devc->shared_irq = hw_config->always_detect; - devc->irq = hw_config->irq; spin_lock_init(&devc->lock); if (devc->irq < 0) -- 1.7.10.4

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[alsa-devel] [PATCH v6 0/3] Add ES8328 audio codec
by Sean Cross 25 Jun '14

25 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks. Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 61 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 719 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 224 +++++++ 12 files changed, 1501 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: aic32x4_set_dai_fmt(): drop unneded register access
by Peter Korsgaard 24 Jun '14

24 Jun '14

There's no point in reading the IFACE2 register when the result isn't used, so drop it. Signed-off-by: Peter Korsgaard <peter(a)korsgaard.com> --- sound/soc/codecs/tlv320aic32x4.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 89e41d2..630367f 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -336,7 +336,6 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1); iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2); - iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2); iface_reg_2 = 0; iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3); iface_reg_3 = iface_reg_3 & ~(1 << 3); -- 2.0.0

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