August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add rt5514 spi dailink
by Harsha Priya 10 Aug '17

10 Aug '17

This patch adds a dai to rt5514-spi driver for wake on voice functionality. Signed-off-by: Hsin-yu Chao <hychao(a)google.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/Kconfig | 1 + sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 9 +++++++++ 2 files changed, 10 insertions(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b301bff..e3ca40d 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -260,6 +260,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH select SND_SOC_INTEL_SKYLAKE select SND_SOC_RT5663 select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI help diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 5ec9428..b51dff6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -65,6 +65,7 @@ enum { KBL_DPCM_AUDIO_HS_PB, KBL_DPCM_AUDIO_ECHO_REF_CP, KBL_DPCM_AUDIO_DMIC_CP, + KBL_DPCM_AUDIO_RT5514_DSP, KBL_DPCM_AUDIO_HDMI1_PB, KBL_DPCM_AUDIO_HDMI2_PB, }; @@ -480,6 +481,14 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) .capture_only = 1, .nonatomic = 1, }, + [KBL_DPCM_AUDIO_RT5514_DSP] = { + .name = "rt5514 dsp", + .stream_name = "Wake on Voice", + .cpu_dai_name = "spi-PRP0001:00", + .platform_name = "spi-PRP0001:00", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", .stream_name = "dmiccap", -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: dapm: Adjustments for some function implementations
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 10 Aug 2017 14:54:32 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (5): Delete error messages for a failed memory allocation in snd_soc_dapm_new_pcm() Improve a size determination in two functions Use kcalloc() in snd_soc_dapm_new_widgets() Adjust one function call together with a variable assignment Adjust seven checks for null pointers sound/soc/soc-dapm.c | 36 ++++++++++++++++-------------------- 1 file changed, 16 insertions(+), 20 deletions(-) -- 2.14.0

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[alsa-devel] [PATCH] ASoC: spear: Delete an error message for a failed memory allocation in two functions
by SF Markus Elfring 10 Aug '17

10 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 10 Aug 2017 09:50:51 +0200 Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/spear/spdif_in.c | 4 +--- sound/soc/spear/spdif_out.c | 4 +--- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 7f32527fc3c8..01b28e59b4c0 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -216,10 +216,8 @@ static int spdif_in_probe(struct platform_device *pdev) return -EINVAL; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); - if (!host) { - dev_warn(&pdev->dev, "kzalloc fail\n"); + if (!host) return -ENOMEM; - } host->io_base = io_base; host->irq = platform_get_irq(pdev, 0); diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 0a72d52d533e..58d5843811f9 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -282,10 +282,8 @@ static int spdif_out_probe(struct platform_device *pdev) int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); - if (!host) { - dev_warn(&pdev->dev, "kzalloc fail\n"); + if (!host) return -ENOMEM; - } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); host->io_base = devm_ioremap_resource(&pdev->dev, res); -- 2.13.4

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[alsa-devel] sidebar link to Bug Tracker is 404
by 積丹尼 Dan Jacobson 10 Aug '17

10 Aug '17

In the sidebar of https://www.alsa-project.org/ there is a link to Bug Tracker. Alas that link is broken.

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[alsa-devel] speaker-test man page should mention the proper way to exit
by 積丹尼 Dan Jacobson 10 Aug '17

10 Aug '17

The speaker-test man page should mention the proper way to exit speaker-test. ^C gives Write error: -4,Interrupted system call xrun_recovery failed: -4,Interrupted system call Transfer failed: Interrupted system call except when run with -c 2. See also https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=802935

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[alsa-devel] [PATCH v5 0/7] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 10 Aug '17

10 Aug '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. I suggest this whole series going through the clk subsystem. v5: - replaced clk_audio_pll_frac_round_rate by clk_audio_pll_frac_determine_rate to also return min and max rate for the clk, suggested by Stephen Boyd during v3, v4: - initialized the clk_init_data structure, - switched to of_clk_parent_fill instead of manually setting parent_names, - removed include of clkdev.h in favor of slab.h, - removed unnecessary second cast in operation, - used clamp instead of two conditions, - removed dependency on clock-output-names, the name is gotten from DT node, - merged all three drivers for audio PLLs (FRAC, PMC, PAD) into one, v3: - added a flag per generated clock to know which one is allowed to modify audio_pll rate, - set the flag in setup function depending on the compatible and the clock ID, v2: - split big patch containing dt-binding, pll drivers and classd modifications, - removed unused AUDIO_PLL_*FOUT* defines from clk-audio-pll-pmc, - added conditions for audio pll rate setting restriction for SSC and I2S, - renamed all mentions of ACLK to GCLK in classD driver, Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop dt-bindings: clk: at91: add audio plls to the compatible list clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock Documentation/devicetree/bindings/clock/at91-clock.txt | 10 +- Documentation/devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/mach-at91/Kconfig | 4 +- drivers/clk/at91/Makefile | 1 +- drivers/clk/at91/clk-audio-pll.c | 536 ++++++++- drivers/clk/at91/clk-generated.c | 101 +- include/linux/clk/at91_pmc.h | 25 +- sound/soc/atmel/atmel-classd.c | 47 +- 8 files changed, 675 insertions(+), 58 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll.c base-commit: 22bbe310b60b37b029659460843265f190364e48 -- git-series 0.9.1

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[alsa-devel] [PATCH 2/2] ASoC: cs43130: Add devicetree bindings for CS43130
by Li Xu 10 Aug '17

10 Aug '17

Add devicetree bindings documentation file for Cirrus Logic CS43130 codec. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs43130.txt | 72 ++++++++++++++++++++++ 1 file changed, 72 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs43130.txt diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt new file mode 100644 index 0000000..51ea38a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs43130.txt @@ -0,0 +1,72 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : Active low GPIO used to reset the device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + + - cirrus,dc-measure: + Boolean, define to enable headphone DC impedance measurement. + + - cirrus,ac-measure: + Boolean, define to enable headphone AC impedance measurement. + DC impedance must also be enabled for AC impedance measurement. + + - cirrus,dc-threshold: + Define 2 DC impedance thresholds in ohms for HP output control. + Default values are 50 and 120 Ohms. + + - cirrus,ac-freq: + Define the frequencies at which to measure HP AC impedance. + Only used if "cirrus,dc-measure" is defined. + Exactly 10 frequencies must be defined. + If this properties is undefined, by default, + following frequencies are used: + <24 43 93 200 431 928 2000 4309 9283 20000> + The above frequencies are logarithmically equally spaced. + Log base is 10. + + - cirrus,idle-bias-off: + Boolean, define this flag such that the device is only powered on when + playing audio. Otherwise, the device is turned off. + If this flag is defined, HP detection and impedance measurement is disabled. + +Example: + +cs43130: audio-codec@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 0>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + cirrus,idle-bias-off; + cirrus,dc-measure; + cirrus,ac-measure; + cirrus,dc-threshold = /bits/ 16 <20 100>; + cirrus,ac-freq = /bits/ 16 <24 43 93 200 431 928 2000 4309 9283 20000>; +}; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5514: Eliminate the noise in the ASRC case
by Oder Chiou 09 Aug '17

09 Aug '17

In the ASRC case, there is the noise if the clock is asynchronous with the LRCK. The default settings will eliminate the noise. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 21cbb56..0eabe56 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,6 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, }; static const struct reg_default rt5514_reg[] = { @@ -89,10 +91,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_DELAY_BUF_CTRL3, 0x00000000}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x00000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x00000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: remove 16ch support for now
by Kuninori Morimoto 09 Aug '17

09 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car Gen3 sound device can use TDM 16ch, but its usage is very limited. It is not yet supported. Let's remove unexpected/unsupported channel settings for now. This patch makes xx_ADINR register settings sanity without runtime check. R-Car Gen3 datasheet 40.4.7 Note on Usage of the SSIU Module When an SSIU module is used in "TDM-16ch mode", only the transfer with the external memory is available. Reported-by: Nguyen Viet Dung <dung.nguyen.aj(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4117b56..cd8db39 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -726,7 +726,6 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, case 2: case 6: case 8: - case 16: /* TDM Extend Mode */ rsnd_rdai_channels_set(rdai, slots); rsnd_rdai_ssi_lane_set(rdai, 1); @@ -740,7 +739,7 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, } static unsigned int rsnd_soc_hw_channels_list[] = { - 2, 6, 8, 16, + 2, 6, 8, }; static unsigned int rsnd_soc_hw_rate_list[] = { @@ -1025,7 +1024,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, drv->playback.rates = RSND_RATES; drv->playback.formats = RSND_FMTS; drv->playback.channels_min = 2; - drv->playback.channels_max = 16; + drv->playback.channels_max = 8; drv->playback.stream_name = rdai->playback.name; snprintf(rdai->capture.name, RSND_DAI_NAME_SIZE, @@ -1033,7 +1032,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, drv->capture.rates = RSND_RATES; drv->capture.formats = RSND_FMTS; drv->capture.channels_min = 2; - drv->capture.channels_max = 16; + drv->capture.channels_max = 8; drv->capture.stream_name = rdai->capture.name; rdai->playback.rdai = rdai; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: avoid duplicate free_irq()
by Kuninori Morimoto 09 Aug '17

09 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 1f8754d4daea5f ("ASoC: rsnd: don't call free_irq() on Parent SSI") fixed Parent SSI duplicate free_irq(). But on Renesas Sound, not only Parent SSI but also Multi SSI have same issue. This patch avoid duplicate free_irq() if it was not pure SSI. Fixes: 1f8754d4daea5f ("ASoC: rsnd: don't call free_irq() on Parent SSI") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 25c351f..0309de5 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -846,10 +846,10 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - struct rsnd_mod *ssi_parent_mod = rsnd_io_to_mod_ssip(io); + struct rsnd_mod *pure_ssi_mod = rsnd_io_to_mod_ssi(io); - /* Do nothing for SSI parent mod */ - if (ssi_parent_mod == mod) + /* Do nothing if non SSI (= SSI parent, multi SSI) mod */ + if (pure_ssi_mod != mod) return 0; /* PIO will request IRQ again */ -- 1.9.1

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