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

[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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[alsa-devel] ASoC updates for v3.11
by Mark Brown 15 Aug '13

15 Aug '13

The following changes since commit d4e4ab86bcba5a72779c43dc1459f71fea3d89c8: Linux 3.11-rc5 (2013-08-11 18:04:20 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc5 for you to fetch changes up to 44ffb69ec617f64cd83182e8b64d93b84758d636: Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus (2013-08-15 11:37:54 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.11 A few driver specific fixes here plus one core fix for a memory corruption issue in DAPM initialisation which could lead to crashes. ---------------------------------------------------------------- Brian Austin (2): ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume ASoC: cs42l52: Add new TLV for Beep Volume Lars-Peter Clausen (1): ASoC: dapm: Fix empty list check in dapm_new_mux() Lothar Waßmann (2): ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control Mark Brown (4): Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus Stephen Warren (1): ASoC: tegra: fix Tegra30 I2S capture parameter setup sound/soc/codecs/cs42l52.c | 5 ++++- sound/soc/codecs/sgtl5000.c | 18 ++++++++++++++---- sound/soc/soc-dapm.c | 7 ++++--- sound/soc/tegra/tegra30_i2s.c | 2 +- 4 files changed, 23 insertions(+), 9 deletions(-)

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[alsa-devel] [PATCH] ALSA: core: allow SND_DMAENGINE_PCM use from modules
by Daniel Mack 15 Aug '13

15 Aug '13

When users of SND_DMAENGINE_PCM are built as module, the config symbol SND_DMAENGINE_PCM must be tristate, otherwise the linker will fail. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sound/core/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 94ce1c4..313f22e 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -7,7 +7,7 @@ config SND_PCM select SND_TIMER config SND_DMAENGINE_PCM - bool + tristate config SND_HWDEP tristate -- 1.8.3.1

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Re: [alsa-devel] [PATCH 0/2] add sysfs for acpi interfaces of thinkpad hardware mute
by David Henningsson 15 Aug '13

15 Aug '13

(Added alsa-devel to CC) On 08/14/2013 09:16 AM, Alex Hung wrote: > Added in David and Tim who also has some ideas to share. > > On Wed, Aug 14, 2013 at 3:00 PM, Matthew Garrett > <matthew.garrett(a)nebula.com> wrote: >> On Wed, 2013-08-14 at 14:41 +0800, Alex Hung wrote: >>> Hi Matthew, >>> >>> The problem is that thinkpad-acpi does not aware of mute is changed by >>> either hotkey presses or in desktop. >> >> Ok, how about this. Register the LED with the kernel LED subsystem. Add >> a new LED trigger for volume-mute. Have the HDA driver call that trigger >> whenever the mute state changes. >> >> -- >> Matthew Garrett <matthew.garrett(a)nebula.com> > > > So, to take a wider grip on this issue. First, the hardware mute. Because this is directly connected to the internal speaker (and headphones?), it needs to turn into a ALSA kcontrol on the sound card that controls speaker and/or headphones. And it needs to named accordingly, e g "Speaker Playback Switch" if it controls the speaker only. Exactly how this is going to work out given the arbitrary module load order of snd-hda-intel and thinkpad-acpi, I'm not sure, but the way of creating another sound card (which I've seen on some thinkpads) just isn't working out for userspace. Second, about the mute LEDs and mic-mute LEDs, we currently have several interfaces to choose from: a) HP LEDs currently uses a ALSA kcontrol to control them. For Mute LED there is also a limited automatic control of this LED from kernel space (by default), so it follows the status of "Master Playback Switch" IIRC. b) thinkpad-acpi currently uses custom sysfs nodes to control the LEDs. c) and we also have the /sys/class/leds interface. There might be even more solutions. It's important we get a consistent interfaces towards userspace for the LEDs. And we should also make sure we don't have a permissions problem: if we want the mute/mic-mute LED to follow the status of the currently selected sound card in PulseAudio, which IMO should be our goal, writes cannot be root-only. To have the HDA driver call into the LED subsystem is better than nothing, but in the end we would want the LEDs to follow whatever sound input/output the user currently uses, whether that is bluetooth, USB, HDMI, or internal audio. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH 0/5] ASoC: pxa dmaengine preparation patches
by Daniel Mack 15 Aug '13

15 Aug '13

Mark, This series is an excerpt of my pxa-dma rework that can be merged independently in the next merge window. The rest of the series will take some time to get sorted out, and given that -rc5 is released already, I don't think the entire series will make it in time. Thanks, Daniel Daniel Mack (5): ALSA: move dmaengine implementation from ASoC to ALSA core ASoC: pxa: pxa-ssp: add DT bindings ASoC: pxa: use snd_dmaengine_dai_dma_data ASoC: pxa: pxa-ssp: set dma filter data from startup hook ASoC: pxa: add DT bindings for pxa2xx-pcm .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 ++ .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 + include/sound/pxa2xx-lib.h | 7 - sound/arm/pxa2xx-ac97.c | 26 +- sound/arm/pxa2xx-pcm-lib.c | 52 ++- sound/arm/pxa2xx-pcm.c | 5 +- sound/arm/pxa2xx-pcm.h | 6 +- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + sound/core/pcm_dmaengine.c | 367 +++++++++++++++++++++ sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/omap/Kconfig | 2 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/mmp-pcm.c | 8 +- sound/soc/pxa/mmp-sspa.c | 12 +- sound/soc/pxa/pxa-ssp.c | 76 +++-- sound/soc/pxa/pxa2xx-ac97.c | 67 ++-- sound/soc/pxa/pxa2xx-i2s.c | 28 +- sound/soc/pxa/pxa2xx-pcm.c | 21 +- sound/soc/soc-dmaengine-pcm.c | 367 --------------------- sound/soc/spear/Kconfig | 2 +- 22 files changed, 607 insertions(+), 499 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 sound/core/pcm_dmaengine.c delete mode 100644 sound/soc/soc-dmaengine-pcm.c -- 1.8.3.1

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[alsa-devel] [PATCH 1/2] ASoC: pcm3008: Add DAPM support
by Mark Brown 15 Aug '13

15 Aug '13

From: Mark Brown <broonie(a)linaro.org> Make it possible to connect external devices to the CODEC and ensure continued operation with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm3008.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 8b9b378..19f5028 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,6 +28,22 @@ #include "pcm3008.h" +static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINL"), +SND_SOC_DAPM_INPUT("VINR"), + +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { + { "PCM3008 Capture", NULL, "VINL" }, + { "PCM3008 Capture", NULL, "VINR" }, + + { "VOUTL", NULL, "PCM3008 Playback" }, + { "VOUTR", NULL, "PCM3008 Playback" }, +}; + #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -77,6 +93,10 @@ static int pcm3008_soc_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { .suspend = pcm3008_soc_suspend, .resume = pcm3008_soc_resume, + .dapm_widgets = pcm3008_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), + .dapm_routes = pcm3008_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes), }; static int pcm3008_codec_probe(struct platform_device *pdev) -- 1.8.4.rc2

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[alsa-devel] [PATCH] ASoC: tegra: support a Mic Jack in the Tegra+RT5640 machine driver
by Stephen Warren 15 Aug '13

15 Aug '13

From: Stephen Warren <swarren(a)nvidia.com> Add a Mic Jack widget to the Tegra+RT5640 machine driver, and document this in the DT binding. This enables the DT to include the Mic Jack in the audio routing table, and hence enables capture of audio, in addition to the previously-working playback. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt | 1 + sound/soc/tegra/tegra_rt5640.c | 1 + 2 files changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt index cba4f88..dc62249 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt @@ -16,6 +16,7 @@ Required properties: * Headphones * Speakers + * Mic Jack - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 08794f9..4511c5a 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -99,6 +99,7 @@ static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), }; static const struct snd_kcontrol_new tegra_rt5640_controls[] = { -- 1.8.1.5

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[alsa-devel] [PATCH] ASoC: tegra: fix Tegra30 I2S capture parameter setup
by Stephen Warren 15 Aug '13

15 Aug '13

From: Stephen Warren <swarren(a)nvidia.com> The Tegra30 I2S driver was writing the AHUB interface parameters to the playback path register rather than the capture path register. This caused the capture parameters not to be configured at all, so if capturing using non-HW-default parameters (e.g. 16-bit stereo rather than 8-bit mono) the audio would be corrupted. With this fixed, audio capture from an analog microphone works correctly on the Cardhu board. Cc: stable(a)vger.kernel.org Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d04146c..47565fd04 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - reg = TEGRA30_I2S_CIF_RX_CTRL; + reg = TEGRA30_I2S_CIF_TX_CTRL; } regmap_write(i2s->regmap, reg, val); -- 1.8.1.5

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[alsa-devel] [PATCH v4 resent 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 15 Aug '13

15 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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