August 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: fsl: Drop SND_SOC_FSL_UTILS from i.mx machine code
by Fabio Estevam 18 Aug '13

18 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> SND_SOC_FSL_UTILS is only used by PowerPC machines, so let's drop it in the i.mx case. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e15f771..3a4808d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -175,7 +175,6 @@ config SND_SOC_IMX_WM8962 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. @@ -187,7 +186,6 @@ config SND_SOC_IMX_SGTL5000 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. -- 1.8.1.2

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[alsa-devel] [PATCH v4 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 16 Aug '13

16 Aug '13

Changelog: v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 100 ++ .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1350 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 223 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1854 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] [PATCH v6 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 16 Aug '13

16 Aug '13

Changelog: v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 56 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1272 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1733 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] [PATCH 1/2] ALSA: hda - Fix internal mic boost on three Thinkpad machines
by David Henningsson 16 Aug '13

16 Aug '13

The internal mic boost is so noisy on boosts 2 and 3 so they are unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213055 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ae9504..6ec96f9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3704,6 +3704,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 1.7.9.5

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[alsa-devel] [git pull] [PATCH 0/7] add support for many Roland and Yamaha devices
by Clemens Ladisch 16 Aug '13

16 Aug '13

The following changes since commit f722406faae2d073cc1d01063d1123c35425939e: Linux 3.10-rc1 (2013-05-11 17:14:08 -0700) are available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git full-roland-support for you to fetch changes up to b7f33917bcd993ff81f3f80b9dc1890fb7410c6d: ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE (2013-06-27 21:59:50 +0200) ---------------------------------------------------------------- Clemens Ladisch (7): ALSA: usb-audio: store protocol version in struct audioformat ALSA: usb-audio: detect implicit feedback on Roland devices ALSA: usb-audio: add support for many Roland/Yamaha devices ALSA: usb-audio: add MIDI port names for some Roland devices ALSA: usb-audio: remove superfluous Roland quirks ALSA: usb-audio: claim autodetected PCM interfaces all at once ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE sound/usb/card.h | 1 + sound/usb/clock.c | 4 +- sound/usb/format.c | 34 +--- sound/usb/format.h | 2 +- sound/usb/midi.c | 74 +++++++ sound/usb/pcm.c | 45 ++++- sound/usb/quirks-table.h | 497 +++++++++-------------------------------------- sound/usb/quirks.c | 209 ++++++++++++++++++++ sound/usb/stream.c | 18 +- sound/usb/usbaudio.h | 2 + 10 files changed, 445 insertions(+), 441 deletions(-)

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[alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Remove direct use of internal I/O functions
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> Use the core to do I/O rather than directly calling the driver operations in order to support further refactoring. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic26.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b192cd4..a4f9360 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -174,9 +174,9 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg); reg = dval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); @@ -185,13 +185,13 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, reg |= 0x0800; if (fsref == 48000) reg |= 0x2000; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Audio Control 1 (FSref divisor) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); reg &= ~0x0fff; reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); return 0; } @@ -212,7 +212,7 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) reg |= 0x8080; else reg &= ~0x8080; - aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg); + snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); return 0; } @@ -348,7 +348,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -368,20 +368,20 @@ static int aic26_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ - aic26_reg_write(codec, AIC26_REG_RESET, 0xBB00); + snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); /* Power up CODEC */ - aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0); + snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read(codec, AIC26_REG_AUDIO_CTRL3); + reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3); reg &= ~0xf800; reg |= 0x0800; /* set master mode */ - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Fill register cache */ for (i = 0; i < codec->driver->reg_cache_size; i++) - aic26_reg_read(codec, i); + snd_soc_read(codec, i); /* Register the sysfs files for debugging */ /* Create SysFS files */ -- 1.8.4.rc2

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[alsa-devel] [PATCH] ASoC: imx: Add MODULE_LICENSE to DMA drivers
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> Reported-by: Ben Hutchings <ben(a)decadent.org.uk> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/imx-pcm-dma.c | 2 ++ sound/soc/fsl/imx-pcm-fiq.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index f323ce0..aa6de5c 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -73,3 +73,5 @@ void imx_pcm_dma_exit(struct platform_device *pdev) snd_dmaengine_pcm_unregister(&pdev->dev); } EXPORT_SYMBOL_GPL(imx_pcm_dma_exit); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 3b2ba99..34043c5 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -408,3 +408,5 @@ void imx_pcm_fiq_exit(struct platform_device *pdev) snd_soc_unregister_platform(&pdev->dev); } EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit); + +MODULE_LICENSE("GPL"); -- 1.8.4.rc2

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[alsa-devel] [PATCH] ASoC: si476x: Add DAPM support
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> This ensures the driver continues to work with DAPM mandatory and makes it easier to connect the device up to other components in the subsystem. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/si476x.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 73e205c..38f3b10 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -102,6 +102,16 @@ static int si476x_codec_write(struct snd_soc_codec *codec, return err; } +static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +}; + +static const struct snd_soc_dapm_route si476x_dapm_routes[] = { + { "Capture", NULL, "LOUT" }, + { "Capture", NULL, "ROUT" }, +}; + static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -260,6 +270,10 @@ static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, .read = si476x_codec_read, .write = si476x_codec_write, + .dapm_widgets = si476x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), + .dapm_routes = si476x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(si476x_dapm_routes), }; static int si476x_platform_probe(struct platform_device *pdev) -- 1.8.4.rc2

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[alsa-devel] [PATCH] ASoC: imx-audmux: Move definitions to dt-bindings
by Markus Pargmann 16 Aug '13

16 Aug '13

Move imx-audmux macro definitions to include/dt-bindings, so they can be used for devicetree. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 56 ++++++++++++++++++++++++++++++ sound/soc/fsl/imx-audmux.h | 52 +-------------------------- 2 files changed, 57 insertions(+), 51 deletions(-) create mode 100644 include/dt-bindings/sound/fsl-imx-audmux.h diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h new file mode 100644 index 0000000..50b09e9 --- /dev/null +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -0,0 +1,56 @@ +#ifndef __DT_FSL_IMX_AUDMUX_H +#define __DT_FSL_IMX_AUDMUX_H + +#define MX27_AUDMUX_HPCR1_SSI0 0 +#define MX27_AUDMUX_HPCR2_SSI1 1 +#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 +#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 +#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 +#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 + +#define MX31_AUDMUX_PORT1_SSI0 0 +#define MX31_AUDMUX_PORT2_SSI1 1 +#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 +#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 +#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 +#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 + +#define MX51_AUDMUX_PORT1_SSI0 0 +#define MX51_AUDMUX_PORT2_SSI1 1 +#define MX51_AUDMUX_PORT3 2 +#define MX51_AUDMUX_PORT4 3 +#define MX51_AUDMUX_PORT5 4 +#define MX51_AUDMUX_PORT6 5 +#define MX51_AUDMUX_PORT7 6 + +/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) +#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) +#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) +#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) +#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) +#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) +#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) +#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) +#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) +#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) + +/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) +#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) +#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) +#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) +#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) +#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) +#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) +#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) +#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) + +#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) +#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) +#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) + +#endif /* __DT_FSL_IMX_AUDMUX_H */ diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index b8ff44b..38a4209 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -1,57 +1,7 @@ #ifndef __IMX_AUDMUX_H #define __IMX_AUDMUX_H -#define MX27_AUDMUX_HPCR1_SSI0 0 -#define MX27_AUDMUX_HPCR2_SSI1 1 -#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 -#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 -#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 -#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 - -#define MX31_AUDMUX_PORT1_SSI0 0 -#define MX31_AUDMUX_PORT2_SSI1 1 -#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 -#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 -#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 -#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 -#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 - -#define MX51_AUDMUX_PORT1_SSI0 0 -#define MX51_AUDMUX_PORT2_SSI1 1 -#define MX51_AUDMUX_PORT3 2 -#define MX51_AUDMUX_PORT4 3 -#define MX51_AUDMUX_PORT5 4 -#define MX51_AUDMUX_PORT6 5 -#define MX51_AUDMUX_PORT7 6 - -/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) -#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) -#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) -#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) -#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) -#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) -#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) -#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) -#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) - -/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) -#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) -#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) -#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) -#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) -#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) -#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) -#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) -#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) - -#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) -#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) -#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) +#include <dt-bindings/sound/fsl-imx-audmux.h> int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr); -- 1.8.3.2

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Re: [alsa-devel] RealTek Inverted Mic Patch
by Takashi Iwai 16 Aug '13

16 Aug '13

At Thu, 15 Aug 2013 14:27:07 -0600, Nathanael D. Noblet wrote: > > Hello, > > I have hardware that suffers from an internal phase inverted mic that > isn't properly detected. Below is a patch that fixes the issue for my > particular hardware. > > Pre patch alsa-info: > > http://www.alsa-project.org/db/?f=a922c77ddf473a8f478ec334fa9dca684f213b59 > > Post patch alsa-info: > http://www.alsa-project.org/db/?f=14cd25b323d3bbb02fabbac78e62851569893c3f > > Incase you care. The patch is small. I don't know if you need any kind > of signed off/written by etc for something so small. In any case there > is no intellectual property in this at all. Feel free to do as you wish > with it. However including it in the kernel would make my life easier. Thanks, I fixed it in sound git tree now. Takashi > > Thanks, > -- > Nathanael d. Noblet > t 403.875.4613 > [2 alsa-realtek-inverted-mic.patch <text/x-patch (7bit)>] > --- a/sound/pci/hda/patch_realtek.c.org 2013-08-15 13:17:49.804492932 -0600 > +++ b/sound/pci/hda/patch_realtek.c 2013-08-15 13:18:40.582364931 -0600 > @@ -4208,6 +4208,7 @@ > SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), > SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), > SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), > + SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), > > #if 0 > /* Below is a quirk table taken from the old code.

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