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

[alsa-devel] [PATCH] ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4
by Howard Mitchell 24 Mar '15

24 Mar '15

Currently GPIO4 is hardcoded to output the pll-lock signal. Unfortunately this is after the pll-out GPIO is configured which is selectable in the device tree. Therefore it is not possible to use GPIO4 for pll-out. Therefore this patch removes the configuration of GPIO4. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- sound/soc/codecs/pcm512x.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 8472099..5a30fdd 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1296,25 +1296,6 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, ret, pcm512x->pll_out); return ret; } - - gpio = PCM512x_G1OE << (4 - 1); - ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, - gpio, gpio); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable gpio %d: %d\n", - 4, ret); - return ret; - } - - gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1; - ret = regmap_update_bits(pcm512x->regmap, gpio, - PCM512x_GxSL, PCM512x_GxSL_PLLLK); - if (ret != 0) { - dev_err(codec->dev, - "Failed to output pll lock on %d: %d\n", - ret, 4); - return ret; - } } ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE, -- 1.7.9.5

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[alsa-devel] [TINYCOMPRESS][PATCH 0/6] Convert build system to autotools
by Qais Yousef 24 Mar '15

24 Mar '15

This series converts the build system to use autotools and splits the files into lib and utils. Also patch 5 fixes error handling code. I can't test Android builds but both cross and native compiling worked for me. I left AUTHORS, ChangeLog and NEWS files empty. I used the license header to fill COPYING. Qais Yousef (6): Convert the build system to autotools add .gitignore file makefile.linux: delete as no longer necessary/used Android.mk: Update to use the new location of source files compress.c: fix check for errors from poll(), read() and write() src/lib/utils.c: remove this empty file .gitignore | 39 ++++ AUTHORS | 0 Android.mk | 6 +- COPYING | 51 ++++ ChangeLog | 0 INSTALL | 365 +++++++++++++++++++++++++++++ Makefile.am | 3 + NEWS | 0 autogen.sh | 6 + compress.c | 632 ------------------------------------------------- configure.ac | 31 +++ cplay.c | 330 -------------------------- crec.c | 489 -------------------------------------- m4/.place_holder | 0 makefile.linux | 41 ---- src/Makefile.am | 1 + src/lib/Makefile.am | 5 + src/lib/compress.c | 634 ++++++++++++++++++++++++++++++++++++++++++++++++++ src/utils/Makefile.am | 11 + src/utils/cplay.c | 330 ++++++++++++++++++++++++++ src/utils/crec.c | 489 ++++++++++++++++++++++++++++++++++++++ utils.c | 62 ----- 22 files changed, 1968 insertions(+), 1557 deletions(-) create mode 100644 .gitignore create mode 100644 AUTHORS create mode 100644 COPYING create mode 100644 ChangeLog create mode 100644 INSTALL create mode 100644 Makefile.am create mode 100644 NEWS create mode 100755 autogen.sh delete mode 100644 compress.c create mode 100644 configure.ac delete mode 100644 cplay.c delete mode 100644 crec.c create mode 100644 m4/.place_holder delete mode 100644 makefile.linux create mode 100644 src/Makefile.am create mode 100644 src/lib/Makefile.am create mode 100644 src/lib/compress.c create mode 100644 src/utils/Makefile.am create mode 100644 src/utils/cplay.c create mode 100644 src/utils/crec.c delete mode 100644 utils.c -- 2.1.0

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[alsa-devel] [PATCH] MAINTAINERS: Add maintainer for maxim codecs
by Anish Kumar 24 Mar '15

24 Mar '15

This patch adds maintainer for maxim audio codecs. Signed-off-by: Anish Kumar <yesanishhere(a)gmail.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 8bdd7a7..2128586 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7387,6 +7387,13 @@ S: Supported F: sound/soc/ F: include/sound/soc* +MAXIM INTEGRATED SOC CODEC DRIVER +M: Anish Kumar <yesanishhere(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/codecs/max* +F: include/sound/max* + SPARC + UltraSPARC (sparc/sparc64) M: "David S. Miller" <davem(a)davemloft.net> L: sparclinux(a)vger.kernel.org -- 1.9.3

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Re: [alsa-devel] Alsa-devel Digest, Vol 97, Issue 173
by noman pouigt 24 Mar '15

24 Mar '15

> I would like to know which file I can use to connect/enable > DPAM widgets permanently so a simple aplay/record can work? > > My environment is: > - Yocto Linux Kernel 3.10 - (DTS based) > - Max98090 Codec > - imx6SL SoC > > I thought about doing this in alsa.conf or asound.conf but I couldn't > find a single example with DAPM widgets . I am also stuck with the same problem. I am sure ALSA experts can help here. What i want to know is how to set the dapm widgets in open embedded? Is it alsa.conf or asound.conf ? I need to set something like this: amixer cset name="LEFT MIXER DAC" 1 Using amixer i am able to playback but i don't know how can i modify conf file to have this dapm widgets associated with a device. Any example would suffice.

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[alsa-devel] [RFC][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 24 Mar '15

24 Mar '15

This patchset adds an experimental driver for Digidesign 002/003 family. If you have any models, please test with this driver and report the result. Currently, this driver correctly supports these major functionalities: (as long as I tested) * PCM sample capture * MIDI messages capture * MIDI messages playback And functionalities which need to be tested: * streaming with S/PDIF, ADAT and work clock source * MIDI device control And includes a bug which should be fixed: * PCM sample playback For a driver for these models, there was a preceding implementation by Robin Gareus and Damien Zammit. [alsa-devel] draft -- ALSA firewire + digi003 http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058177.ht… This patchset includes their implementation, called-as double-oh-three, to multiplex PCM samples into CIP packet. As long as I test, their implementation may includes any bugs and this driver still causes noisy sound in several cases, while I confirm to get the same data block pattern which Robin explains. http://gareus.org/wiki/digi003 If you interested in this bug and tries to fix it, my packet dump may be helpfull. Please see: http://subversion.ffado.org/wiki/digi003rack * 96khz-24bit-1ch.log (27.5 kB) * 44khz-24bit-1ch.log (49.8 kB) * 44khz-24bit-2ch.log (49.8 kB) And DKMS package is available on my github repository. https://github.com/takaswie/snd-firewire-improve Takashi Sakamoto (11): ALSA: digi00x: add skelton for Digi 002/003 device driver ALSA: digi00x: add streaming functionality ALSA: digi00x: add proc node for clock status ALSA: digi00x: add PCM functionality ALSA: digi00x: add MIDI functionality ALSA: digi00x: add hwdep interface ALSA: digi00x: support unknown asynchronous message ALSA: digi00x: support MIDI ports for device control ALSA: firewire-lib: allows to implement external MIDI callback function digi00x: improve MIDI capture/playback ALSA: digi00x: apply double-oh-three algorism to multiplex PCM samples include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 10 + sound/firewire/Makefile | 1 + sound/firewire/amdtp.c | 30 ++- sound/firewire/amdtp.h | 5 + sound/firewire/digi00x/Makefile | 3 + sound/firewire/digi00x/digi00x-hwdep.c | 192 ++++++++++++++ sound/firewire/digi00x/digi00x-midi.c | 199 +++++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 336 +++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 71 ++++++ sound/firewire/digi00x/digi00x-protocol.c | 361 +++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-stream.c | 398 ++++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x.c | 179 ++++++++++++++ sound/firewire/digi00x/digi00x.h | 126 ++++++++++ 15 files changed, 1907 insertions(+), 8 deletions(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-protocol.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.0

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[alsa-devel] [PATCH v2] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 24 Mar '15

24 Mar '15

Add MAX98090 audio codec, I2S interface and the sound nodes to support audio on Exynos5422 SoC Based Odroid-XU3 board. Now we can support audio in Odroid-XU3 board using simple-audio-card DT binding. Signed-off-by: Inha Song <ideal.song(a)samsung.com> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos5420.dtsi | 9 +++++ arch/arm/boot/dts/exynos5422-odroidxu3.dts | 59 ++++++++++++++++++++++++++++++ 2 files changed, 68 insertions(+) diff --git a/arch/arm/boot/dts/exynos5420.dtsi b/arch/arm/boot/dts/exynos5420.dtsi index 4eaeabe..e459c1d 100644 --- a/arch/arm/boot/dts/exynos5420.dtsi +++ b/arch/arm/boot/dts/exynos5420.dtsi @@ -415,6 +415,9 @@ <&clock_audss EXYNOS_I2S_BUS>, <&clock_audss EXYNOS_SCLK_I2S>; clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk0"; + #sound-dai-cells = <1>; samsung,idma-addr = <0x03000000>; pinctrl-names = "default"; pinctrl-0 = <&i2s0_bus>; @@ -429,6 +432,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S1>, <&clock CLK_SCLK_I2S1>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk1"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s1_bus>; status = "disabled"; @@ -442,6 +448,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S2>, <&clock CLK_SCLK_I2S2>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk2"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s2_bus>; status = "disabled"; diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3.dts b/arch/arm/boot/dts/exynos5422-odroidxu3.dts index edc25cf..c038f0d 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3.dts +++ b/arch/arm/boot/dts/exynos5422-odroidxu3.dts @@ -11,6 +11,7 @@ */ /dts-v1/; +#include <dt-bindings/sound/samsung-i2s.h> #include "exynos5800.dtsi" / { @@ -285,6 +286,64 @@ rtc@101E0000 { status = "okay"; }; + + sound: sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "Odroid-XU3"; + simple-audio-card,widgets = + "Headphone", "Headphone Jack", + "Speakers", "Speakers"; + simple-audio-card,routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "Headphone Jack", "MICBIAS", + "IN1", "Headphone Jack", + "Speakers", "SPKL", + "Speakers", "SPKR"; + + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&link0_codec>; + simple-audio-card,frame-master = <&link0_codec>; + + simple-audio-card,cpu { + sound-dai = <&i2s0 0>; + system-clock-frequency = <19200000>; + }; + + link0_codec: simple-audio-card,codec { + sound-dai = <&max98090>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + }; + }; +}; + +&clock_audss { + assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>, + <&clock_audss EXYNOS_MOUT_I2S>, + <&clock_audss EXYNOS_DOUT_AUD_BUS>; + assigned-clock-parents = <&clock CLK_FIN_PLL>, + <&clock_audss EXYNOS_MOUT_AUDSS>; + assigned-clock-rates = <0>, + <0>, + <19200000>; +}; + +&hsi2c_5 { + status = "okay"; + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 0>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; +}; + +&i2s0 { + status = "okay"; }; &hdmi { -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH 00/10 v3] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 24 Mar '15

24 Mar '15

Hi Mark These are v3 of DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, basically it was copied from simple-card. v2 patch-set had ak4642 patch, but, it is already applied. v3 doesn't include it. 1 : prepare patch for Renesas new sound card driver 2 - 3 : new Renesas sound card driver 4 - 8 : fixup/tidyup patches for rsnd 9 - 10 : DPCM support patch for rsnd Kuninori Morimoto (10): 1) ASoC: core: add snd_soc_get_dai_drv() to get DAI driver 2) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 3) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 4) ASoC: rsnd: no more SSI restart when unusual situation 5) ASoC: rsnd: no more SRC restart when unusual situation 6) ASoC: rsnd: tidyup error message format 7) ASoC: rsnd: show debug info for sampling rate convert 8) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 9) ASoC: rsnd: add dai_link stream name 10) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ include/sound/soc.h | 1 + sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 +++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd.h | 16 +++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 69 ++++++++++---- 12 files changed, 777 insertions(+), 54 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] ASoC:pcm512x: Make PLL lock output selectable via device tree.
by Howard Mitchell 23 Mar '15

23 Mar '15

Currently the PLL Lock output signal is hardcoded to GPIO4. This makes it seletable in the same way as pll-in and pll-out. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- .../devicetree/bindings/sound/pcm512x.txt | 3 ++ sound/soc/codecs/pcm512x.c | 47 +++++++++++++------- 2 files changed, 33 insertions(+), 17 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt index 3aae3b4..432f186 100644 --- a/Documentation/devicetree/bindings/sound/pcm512x.txt +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -26,6 +26,8 @@ Optional properties: given pll-in pin and PLL output on the given pll-out pin. An external connection from the pll-out pin to the SCLK pin is assumed. + - pll-lock : gpio pin used to output the PLL lock flag. + Examples: pcm5122: pcm5122@4c { @@ -49,4 +51,5 @@ Examples: clocks = <&sck>; pll-in = <3>; pll-out = <6>; + pll-lock = <4>; }; diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 8472099..a4217d7 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -49,6 +49,7 @@ struct pcm512x_priv { int fmt; int pll_in; int pll_out; + int pll_lock; int pll_r; int pll_j; int pll_d; @@ -1296,24 +1297,26 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, ret, pcm512x->pll_out); return ret; } + } - gpio = PCM512x_G1OE << (4 - 1); - ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, - gpio, gpio); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable gpio %d: %d\n", - 4, ret); - return ret; - } - - gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1; - ret = regmap_update_bits(pcm512x->regmap, gpio, - PCM512x_GxSL, PCM512x_GxSL_PLLLK); - if (ret != 0) { - dev_err(codec->dev, - "Failed to output pll lock on %d: %d\n", - ret, 4); - return ret; + if (pcm512x->pll_lock) { + gpio = PCM512x_G1OE << (pcm512x->pll_lock - 1); + ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, + gpio, gpio); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable gpio %d: %d\n", + pcm512x->pll_lock, ret); + return ret; + } + + gpio = PCM512x_GPIO_OUTPUT_1 + pcm512x->pll_lock - 1; + ret = regmap_update_bits(pcm512x->regmap, gpio, + PCM512x_GxSL, PCM512x_GxSL_PLLLK); + if (ret != 0) { + dev_err(codec->dev, + "Failed to output pll lock on %d: %d\n", + ret, pcm512x->pll_lock); + return ret; } } @@ -1518,6 +1521,16 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) ret = -EINVAL; goto err_clk; } + + if (of_property_read_u32(np, "pll-lock", &val) >= 0) { + if (val > 6) { + dev_err(dev, "Invalid pll-lock\n"); + ret = -EINVAL; + goto err_clk; + } + pcm512x->pll_lock = val; + } + } #endif -- 1.7.9.5

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[alsa-devel] [PATCH v2] ALSA: hda - Add dock support for Thinkpad T450s (17aa:5036)
by Sebastian Wicki 23 Mar '15

23 Mar '15

This model uses the same dock port as the previous generation. Signed-off-by: Sebastian Wicki <gandro(a)gmx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eee4532..8775389 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4995,6 +4995,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "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), -- 2.3.3

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[alsa-devel] [PATCH] ALSA: hda - Add dock support for Thinkpad T450s (17aa:5036)
by Sebastian Wicki 23 Mar '15

23 Mar '15

This model uses the same dock port as the previous generation. Signed-off-by: Sebastian Wicki <gandro(a)gmx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eee4532..a9e2e6e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4988,6 +4988,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.3.3

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