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

[alsa-devel] Applied "ASoC: rsnd: gen: add rsnd_force_write()" to the asoc tree
by Mark Brown 10 Jul '15

10 Jul '15

The patch ASoC: rsnd: gen: add rsnd_force_write() has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7b47ab47b3938e2274834dbde7915df98fc74368 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 16 Jun 2015 08:53:11 +0000 Subject: [PATCH] ASoC: rsnd: gen: add rsnd_force_write() rsnd_force_write() is similar to rsnd_write(), but rsnd_force_write() write data to register even though it is same value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/gen.c | 16 ++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 4 ++++ 2 files changed, 20 insertions(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 8c7dc51b1c4f..48f704b1d80e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -103,6 +103,22 @@ void rsnd_write(struct rsnd_priv *priv, regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); } +void rsnd_force_write(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + if (!rsnd_is_accessible_reg(priv, gen, reg)) + return; + + dev_dbg(dev, "w %s[%d] - %4d : %08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data); + + regmap_fields_force_write(gen->regs[reg], rsnd_mod_id(mod), data); +} + void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data) { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 09fcc54a8ee0..f729646e3d18 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -157,12 +157,16 @@ struct rsnd_dai_stream; rsnd_read(rsnd_mod_to_priv(m), m, RSND_REG_##r) #define rsnd_mod_write(m, r, d) \ rsnd_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d) +#define rsnd_mod_force_write(m, r, d) \ + rsnd_force_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d) #define rsnd_mod_bset(m, r, s, d) \ rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d) u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 data); +void rsnd_force_write(struct rsnd_priv *priv, struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data); void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data); u32 rsnd_get_adinr(struct rsnd_mod *mod, struct rsnd_dai_stream *io); -- 2.1.4

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[alsa-devel] [PATCH] ALSA: aoa: Drop owner assignment from i2c_driver
by Krzysztof Kozlowski 10 Jul '15

10 Jul '15

i2c_driver does not need to set an owner because i2c_register_driver() will set it. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- The coccinelle script which generated the patch was sent here: http://www.spinics.net/lists/kernel/msg2029903.html --- sound/aoa/codecs/onyx.c | 1 - sound/aoa/codecs/tas.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 23c371ecfb6b..a04edff8b729 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1050,7 +1050,6 @@ MODULE_DEVICE_TABLE(i2c,onyx_i2c_id); static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", - .owner = THIS_MODULE, }, .probe = onyx_i2c_probe, .remove = onyx_i2c_remove, diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 364c7c4416e8..78ed1ffbf786 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -939,7 +939,6 @@ MODULE_DEVICE_TABLE(i2c,tas_i2c_id); static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", - .owner = THIS_MODULE, }, .probe = tas_i2c_probe, .remove = tas_i2c_remove, -- 1.9.1

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Re: [alsa-devel] Intel Baytrail Linux Audio
by Tyler Yeomans 10 Jul '15

10 Jul '15

Vinod, I am using the ssp0 firmware from: https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/tree/intel?…. Thanks- Tyler On 07/06/2015 11:19 PM, Vinod Koul wrote: > On Mon, Jul 06, 2015 at 10:29:34AM -0400, Tyler Yeomans wrote: >> Hello all, >> >> Apparently I'm in the same boat as Nicolas is. >> I've attached my dmesg output. >> Everything seems to be generating but I am not able to get any sound >> output. >> Between the 37 second mark and the 617 second mark I had routed the >> FE to BE via the commands: >> >> amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on >> amixer -c0 sset 'media0_out mix 0 media1_in' on > ok that seems fine >> Although it appears that did not work: >> [ 719.834922] intel_sst_acpi 80860F28:00: Wait timed-out >> condition:0x0, msg_id:0x1 fw_state 0x3 >> [ 719.834932] intel_sst_acpi 80860F28:00: fw returned err -16 > I don't think so, since you were able to start the card, that part worked :) > > The error is coming from FW here and seems unresponsive. > > Which firmware are you using? > > Second suspicion is for interrupts, if they are not mapped well we might see > these. > > Jarkko I remember that some board have to be configured properly for > interrupts, can you explaian the subtleties involved here > > -- > ~Vinod

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[alsa-devel] [PATCH v3 0/4] ASoC: Intel: Skylake - Add IPC routines
by Vinod Koul 10 Jul '15

10 Jul '15

The SKL aDSP needs to send and receive IPCs to and from aDSP. To aid this we add IPC and aDSP management routines as a library for driver to use. For most of IPC handling we use the common IPC Intel code in place and add the messaging and message parsing bits on top. First 2 patches are against common IPC code and can be applied independently Last two add IPC handling and aDSP management routines v3 changes: - sst_dsp_register_poll uses opportunistic polling and polls for first few instances using mdelay and later sleeps - s/skl_ipc_reply_find_msg/skl_ipc_reply_get_msg - return IRQ_NONE when interrupt in not handled - removed trace_printk and superfluous cast to/from void * Subhransu S. Prusty (4): ASoC: Intel: Add helper to poll register for DSP status ASoC: Intel: Add helper to update register bits with attr RWC ASoC: Intel: Add Skylake IPC library ASoC: Intel: Add DSP init and boot up functionality for SKL sound/soc/intel/Kconfig | 1 + sound/soc/intel/common/sst-dsp-priv.h | 8 + sound/soc/intel/common/sst-dsp.c | 71 ++++ sound/soc/intel/common/sst-dsp.h | 6 + sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-sst-dsp.c | 337 +++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 57 +++ sound/soc/intel/skylake/skl-sst-ipc.c | 760 ++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 125 ++++++ 9 files changed, 1370 insertions(+) create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h -- 1.9.1

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Re: [alsa-devel] [RFT PATCH 1/2] ASoC: zx: i2s: Fix devm_ioremap_resource return value check
by Mark Brown 09 Jul '15

09 Jul '15

On Thu, Jul 09, 2015 at 10:19:28PM +0900, Krzysztof Kozlowski wrote: > Signed-off-by: Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com> > Cc: <stable(a)vger.kernel.org> Why are you adding a Cc to stable? The zx driver was added in the merge window...

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[alsa-devel] [PATCH alsa-plugins] include speexdsp_types.h, not speex_types.h
by Tanu Kaskinen 09 Jul '15

09 Jul '15

Speexdsp was separated from speex in 1.2rc2. speex_types.h is not shipped by speexdsp, so alsa-plugins shouldn't use that file. speexdsp has speexdsp_types.h, which has the same contents as speex_types.h. speexdsp_types.h is a new file introduced in 1.2rc2, so this change bumps the minimum supported speexdsp version. The version check in configure.ac will actually break if speexdsp 1.2 ever gets released, because pkg-config thinks that "1.2" < "1.2rc2", but I think it's useful to fail if the installed speexdsp version is 1.2rc1 (which I believe is very common on current distributions). If a non-rc version of speexdsp will ever get released, I hope version number 1.2 will be skipped for this reason. (A non-rc version seems unlikely, since 1.2rc1 was released years ago, so it's pretty likely that the project is stuck on so called "release candidates" forever...) Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> --- configure.ac | 2 +- pph/arch.h | 2 +- pph/speex_resampler.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/configure.ac b/configure.ac index 902a6d7..c554d22 100644 --- a/configure.ac +++ b/configure.ac @@ -108,7 +108,7 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) -PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) +PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2rc2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes") AC_ARG_WITH([speex], diff --git a/pph/arch.h b/pph/arch.h index e2d731a..a07d0d9 100644 --- a/pph/arch.h +++ b/pph/arch.h @@ -36,7 +36,7 @@ #define ARCH_H #ifndef OUTSIDE_SPEEX -#include "speex/speex_types.h" +#include "speex/speexdsp_types.h" #endif #define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */ diff --git a/pph/speex_resampler.h b/pph/speex_resampler.h index c44fbcd..aa85abb 100644 --- a/pph/speex_resampler.h +++ b/pph/speex_resampler.h @@ -82,7 +82,7 @@ #else /* OUTSIDE_SPEEX */ -#include "speex/speex_types.h" +#include "speex/speexdsp_types.h" #endif /* OUTSIDE_SPEEX */ -- 1.9.3

2 2

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Sklyake - add IPC routines
by Vinod Koul 09 Jul '15

09 Jul '15

The SKL aDSP needs to send and receive IPCs to and from aDSP. To aid this we add IPC and aDSP management routines as a library for driver to use. For most of IPC handling we use the common IPC Intel code in place and add the messaging and message parsing bits on top. First 4 patches are against common IPC code and can be applied independently Last two add IPC handling and aDSP management routines, they are not dependent upon anything and can be applied independently but they are generated against SKL PCM driver series so Makefile updates expect those to be there v2 folds few fixes seen and make function name consistent Subhransu S. Prusty (6): ASoC: Intel: Add helper to poll register for DSP status ASoC: Intel: Define SKL ADSP IPC and general purpose registers ASoC: Intel: Reorganize the common dsp structure ASoC: Intel: Add helper to update register bits with attr RWC ASoC: Intel: Add Skylake IPC library ASoC: Intel: Add DSP init and boot up functionality for SKL sound/soc/intel/Kconfig | 1 + sound/soc/intel/common/sst-dsp-priv.h | 20 +- sound/soc/intel/common/sst-dsp.c | 51 +++ sound/soc/intel/common/sst-dsp.h | 6 + sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-sst-dsp.c | 337 +++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 119 ++++++ sound/soc/intel/skylake/skl-sst-ipc.c | 752 ++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 125 ++++++ 9 files changed, 1412 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h -- 1.9.1

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[alsa-devel] debugging kernel module: ASoC sun7i A20 I2S DAI driver for simple sound card setup (with UDA1380 codec)
by Andrea Venturi 09 Jul '15

09 Jul '15

hello, i'm writing here because i'm facing an issue on a new driver and like a feedback from the community. wht place better then this ML! i'm using an Allwinner A20 based card with I2S/I2C wires exposed (Olimex A20 SOM + EVB), and an external UDA1380 codec (dirty cheap purchase from those china techno junk sites) i'd like to make this bundle work together; my aim is to produce the mainline kernel DAI module for the actually missing I2S HW feature for Allwinner SOC (at least sun4i A10 and sun7i A20, don't know if the ) i gathered as more info as possible about Alsa ASoC implementation; but, boys, is not very easy to catch on as there's a huge bulk of documentation scattered all over the web with varying degree of obsolescence in place.. and usually the newer stuff is the less documented :-) anyway, as my "awareness" of the whole Alsa scenario went increasing (at least in my opinion..), i found the Freescale SAI (see .../sound/soc/fsl/fsl_sai.c) module to be the nearest/better/more modern "example" from which starting modeling the sunxi I2S module. so, now, i've shaped a Sunxi I2S kernel module that loads someway on my HW testbed. all the tech details (with source and tests) are published here: https://gist.github.com/aventuri/c40d08b946b0d1da5081 you'll find: - sunxi-i2s.c - a DTS with the sound entry for a simple sound card - a slightly modded uda1380.c codec driver; as a sidenote, i had to modify for params loading from DTS. if more details are needed, i'll add stuff on request, of course.. the starting mainline kernel is linux 4.1.rc sunxi-wip from here: https://github.com/jwrdegoede/linux-sunxi/tree/sunxi-wip where the i2s clocks and dmaengine (and the analog codec of AW sunxi, but it's not needed here) are already in place. the "gory details" about HW registers are coming from the my reading of A20 PDF and allwinner I2S linux-sunxi legacy I2S driver for 3.4: https://github.com/linux-sunxi/linux-sunxi/tree/sunxi-3.4/sound/soc/sunxi/i… these registers CAN be way off as i'm still debugging a bit before starting checking them carefully.. now, the kernel modules (snd-soc-simple-card snd-soc-sunxi-i2s and snd-soc-uda1380) are loaded ok and doing someway a proper initialization of HW and kernel data structures (check on the gist before). i can see the alsa device nodes # ls -lR /dev/snd /dev/snd: total 0 crw-rw---- 1 root root 116, 0 Jan 1 03:00 controlC0 crw-rw---- 1 root root 116, 24 Jan 1 03:00 pcmC0D0c crw-rw---- 1 root root 116, 16 Jan 1 03:00 pcmC0D0p crw-rw---- 1 root root 116, 33 Jan 1 03:00 timer and alsamixer is more or less showing all the controllers (many are from the UDA codec, of course, already working..) the problem i'm facing now is that the when i launch aplay with a wav file, when it accesses the dev node "/dev/snd/pcmC0D0p" is spitting out: "open error: Invalid argument" these are the last few rows of a "strace aplay wav/equinox-48KHz.wav" execution: # aplay wav/equinox-48KHz.wav [ 9018.638837] [I2S]Entered sunxi_i2s_startup: hw is subdevice #0, stream is: playback [ 9018.646858] [I2S]Entered sunxi_i2s_shutdown, need to clear some HW regs? aplay: main:722: audio open error: Invalid argument # ===================== ... open("/dev/snd/controlC0", O_RDONLY|O_LARGEFILE|O_CLOEXEC) = 3 fcntl64(3, F_SETFD, FD_CLOEXEC) = 0 ioctl(3, SNDRV_CTL_IOCTL_CARD_INFO or UI_DEV_CREATE, 0xbeac43b8) = 0 close(3) = 0 open("/dev/snd/controlC0", O_RDWR|O_LARGEFILE|O_CLOEXEC) = 3 fcntl64(3, F_SETFD, FD_CLOEXEC) = 0 ioctl(3, SNDRV_CTL_IOCTL_PVERSION or USBDEVFS_CONTROL, 0xbeac4540) = 0 ioctl(3, SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE, 0xbeac457c) = 0 open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_LARGEFILE|O_CLOEXEC) = -1 EINVAL (Invalid argument) close(3) = 0 write(2, "aplay: main:722: ", 17aplay: main:722: ) = 17 write(2, "audio open error: Invalid argume"..., 34audio open error: Invalid argument) = 34 write(2, "\n", 1 ) = 1 exit_group(1) = ? +++ exited with 1 +++ # ================== now my question is: how can i start debugging this step? which kernel tools i could use to pinpoint quickly who is answering back this "invalid argument" in the called function chain. i suppose some "sound params" in the my own kernel driver is not filled correctly; i tried with many wav files (44.1 48 etc..) but always get this reply. i tried with the "kernel function tracing" tool (/sys/kernel/debug/tracing/...) but it doesn't help a lot.. any suggestion? any indication on documents about effective Alsa ASoC debugging or architecture? bests andrea

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[alsa-devel] [PATCH] ALSA: hda: Delete an unnecessary check before the function call "kobject_put"
by SF Markus Elfring 09 Jul '15

09 Jul '15

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 8 Jul 2015 21:26:02 +0200 The kobject_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/hda/hdac_sysfs.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 0a6ce3b..089b35f 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -321,8 +321,7 @@ static void widget_tree_free(struct hdac_device *codec) free_widget_node(*p, &widget_node_group); kfree(tree->nodes); } - if (tree->root) - kobject_put(tree->root); + kobject_put(tree->root); kfree(tree); codec->widgets = NULL; } -- 2.4.5

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[alsa-devel] [PATCH] ASoC: mediatek: Use platform_of_node for machine drivers
by Koro Chen 09 Jul '15

09 Jul '15

This replaces the platform_name in snd_soc_dai_link by device tree node. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- .../devicetree/bindings/sound/mt8173-max98090.txt | 2 ++ .../bindings/sound/mt8173-rt5650-rt5676.txt | 2 ++ sound/soc/mediatek/mt8173-max98090.c | 17 +++++++++++++---- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 19 +++++++++++++++---- 4 files changed, 32 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt index 829bd26..519e97c 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt @@ -3,11 +3,13 @@ MT8173 with MAX98090 CODEC Required properties: - compatible : "mediatek,mt8173-max98090" - mediatek,audio-codec: the phandle of the MAX98090 audio codec +- mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-max98090"; mediatek,audio-codec = <&max98090>; + mediatek,platform = <&afe>; }; diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt index 61e98c9..f205ce9 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -3,11 +3,13 @@ MT8173 with RT5650 RT5676 CODECS Required properties: - compatible : "mediatek,mt8173-rt5650-rt5676" - mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-rt5650-rt5676"; mediatek,audio-codec = <&rt5650 &rt5676>; + mediatek,platform = <&afe>; }; diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 4d44b58..2d2536a 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -103,7 +103,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Playback", .stream_name = "MAX98090 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -114,7 +113,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Capture", .stream_name = "MAX98090 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -125,7 +123,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codec_dai_name = "HiFi", .init = mt8173_max98090_init, @@ -152,9 +149,21 @@ static struct snd_soc_card mt8173_max98090_card = { static int mt8173_max98090_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_max98090_card; - struct device_node *codec_node; + struct device_node *codec_node, *platform_node; int ret, i; + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + for (i = 0; i < card->num_links; i++) { + if (mt8173_max98090_dais[i].platform_name) + continue; + mt8173_max98090_dais[i].platform_of_node = platform_node; + } + codec_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); if (!codec_node) { diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 0940553..6f52eca 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -138,7 +138,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Playback", .stream_name = "rt5650_rt5676 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -149,7 +148,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Capture", .stream_name = "rt5650_rt5676 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -161,7 +159,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codecs = mt8173_rt5650_rt5676_codecs, .num_codecs = 2, @@ -209,7 +206,21 @@ static struct snd_soc_card mt8173_rt5650_rt5676_card = { static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5676_card; - int ret; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_rt5676_dais[i].platform_name) + continue; + mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node; + } mt8173_rt5650_rt5676_codecs[0].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); -- 1.8.1.1.dirty

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