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[alsa-devel] [PATCH] ALSA: hda - Fix a wrong FIXUP for alc289 on Dell machines
by Hui Wang 03 Mar '18

03 Mar '18

With the alc289, the Pin 0x1b is Headphone-Mic, so we should assign ALC269_FIXUP_DELL4_MIC_NO_PRESENCE rather than ALC225_FIXUP_DELL1_MIC_NO_PRESENCE to it. And this change is suggested by Kailang of Realtek and is verified on the machine. (This fixes the commit 3f2f7c55) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b9c93fa..7a9a867 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6872,7 +6872,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60120}, {0x14, 0x90170110}, {0x21, 0x0321101f}), - SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, {0x12, 0xb7a60130}, {0x14, 0x90170110}, {0x21, 0x04211020}), -- 2.7.4

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[alsa-devel] regression v4.16 on Nokia N900:/dev/input/event6 aka AV Jack support disappeared
by Pavel Machek 02 Mar '18

02 Mar '18

Hi! In v4.16, AV jack support disappeared. This one is suspect: commit 7be4b5dc7ffa9499ac6ef33a5ffa9ff43f9b7057 Author: Andrew F. Davis <afd(a)ti.com> Date: Wed Nov 29 11:13:59 2017 -0600 ARM: dts: omap3-n900: Fix the audio CODEC's reset pin The correct DT property for specifying a GPIO used for reset is "reset-gpios", fix this here. Fixes: 14e3e295b2b9 ("ARM: dts: omap3-n900: Add TLV320AIC3X support") Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Tony Lindgren <tony(a)atomide.com> How was it tested? It reverts polarity of reset pin, but sound/soc/codecs/tlv320aic3x.c treats those as aliases: ret = of_get_named_gpio(np, "reset-gpios", 0); if (ret >= 0) { aic3x->gpio_reset = ret; } else { ret = of_get_named_gpio(np, "gpio-reset", 0); if (ret > 0) { dev_warn(&i2c->dev, "Using deprecated property \"gpio-r\eset\", please update your DT"); aic3x->gpio_reset = ret; Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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[alsa-devel] [asoc:topic/intel 22/22] sound/soc/intel/boards/cht_bsw_nau8824.c:137:36: sparse: incorrect type in argument 1 (different base types)
by kbuild test robot 02 Mar '18

02 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 3bf045d15f52529e8abe26543dbe22af44efb41a commit: 3bf045d15f52529e8abe26543dbe22af44efb41a [22/22] ASoC: Intel: fix argument error in nau8824 machine reproduce: # apt-get install sparse git checkout 3bf045d15f52529e8abe26543dbe22af44efb41a make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/intel/boards/cht_bsw_nau8824.c:137:36: sparse: incorrect type in argument 1 (different base types) @@ expected struct snd_soc_codec *codec @@ got strustruct snd_soc_codec *codec @@ sound/soc/intel/boards/cht_bsw_nau8824.c:137:36: expected struct snd_soc_codec *codec sound/soc/intel/boards/cht_bsw_nau8824.c:137:36: got struct snd_soc_component *component sound/soc/intel/boards/cht_bsw_nau8824.c: In function 'cht_codec_init': sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type [-Werror=incompatible-pointer-types] nau8824_enable_jack_detect(component, jack); ^~~~~~~~~ In file included from sound/soc/intel/boards/cht_bsw_nau8824.c:31:0: sound/soc/intel/boards/../../codecs/nau8824.h:474:5: note: expected 'struct snd_soc_codec *' but argument is of type 'struct snd_soc_component *' int nau8824_enable_jack_detect(struct snd_soc_codec *codec, ^~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +137 sound/soc/intel/boards/cht_bsw_nau8824.c 101 102 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) 103 { 104 struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); 105 struct snd_soc_jack *jack = &ctx->jack; 106 struct snd_soc_dai *codec_dai = runtime->codec_dai; 107 struct snd_soc_component *component = codec_dai->component; 108 int ret, jack_type; 109 110 /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ 111 ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); 112 if (ret < 0) { 113 dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); 114 return ret; 115 } 116 117 /* NAU88L24 supports 4 butons headset detection 118 * KEY_MEDIA 119 * KEY_VOICECOMMAND 120 * KEY_VOLUMEUP 121 * KEY_VOLUMEDOWN 122 */ 123 jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | 124 SND_JACK_BTN_2 | SND_JACK_BTN_3; 125 ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, 126 cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); 127 if (ret) { 128 dev_err(runtime->dev, 129 "Headset Jack creation failed %d\n", ret); 130 return ret; 131 } 132 snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); 133 snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); 134 snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); 135 snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); 136 > 137 nau8824_enable_jack_detect(component, jack); 138 139 return ret; 140 } 141 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 0/6] ASoC: max9867: Driver update
by Ladislav Michl 02 Mar '18

02 Mar '18

Hi, here are first few patches for max9867 codec driver. It is still not enough to make it work (at least for me :)), but lets send cleanups first. I would be happy to read clarification about using continuous rate - see patch 6. Thank you, ladis Ladislav Michl (6): ASoC: max9867: Show Kconfig entry ASoC: max9867: Improve error logging ASoC: max9867: Drop probe function ASoC: max9867: Fix codec capabilities ASoC: max9867: DSP mode ASoC: max9867: Use continuous sample rate sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/max9867.c | 122 ++++++++++++--------------------------------- sound/soc/codecs/max9867.h | 1 - 3 files changed, 34 insertions(+), 92 deletions(-) -- 2.16.2

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[alsa-devel] [PATCH RESEND] ASoC: max9867: Drop probe function
by Ladislav Michl 02 Mar '18

02 Mar '18

Driver probe function has no use and can be deleted. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- sound/soc/codecs/max9867.c | 10 ---------- sound/soc/codecs/max9867.h | 1 - 2 files changed, 11 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 99b8035d2fda..3620b38291f0 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -406,17 +406,7 @@ static int max9867_resume(struct device *dev) } #endif -static int max9867_probe(struct snd_soc_component *component) -{ - struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - - dev_dbg(component->dev, "max98090_probe\n"); - max9867->component = component; - return 0; -} - static const struct snd_soc_component_driver max9867_component = { - .probe = max9867_probe, .controls = max9867_snd_controls, .num_controls = ARRAY_SIZE(max9867_snd_controls), .dapm_routes = max9867_audio_map, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index c0aea3d1f2ba..55cd9976ff47 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -75,7 +75,6 @@ /* codec private data */ struct max9867_priv { struct regmap *regmap; - struct snd_soc_component *component; unsigned int sysclk; unsigned int pclk; unsigned int master; -- 2.16.2

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[alsa-devel] [PATCH] ASoC: Intel: fix argument error in nau8824 machine
by John Hsu 02 Mar '18

02 Mar '18

Fix the error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type nau8824_enable_jack_detect(codec, jack); ^~~~~ Which expects 'struct snd_soc_component *' but argument is of type 'struct snd_soc_codec *' Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 08d0b81..680f2b3 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -103,8 +103,8 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); struct snd_soc_jack *jack = &ctx->jack; - struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_component *component = codec_dai->component; int ret, jack_type; /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ @@ -134,7 +134,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - nau8824_enable_jack_detect(codec, jack); + nau8824_enable_jack_detect(component, jack); return ret; } -- 2.6.4

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[alsa-devel] [PATCH v2] ASoC: max9867: Drop probe function
by Ladislav Michl 02 Mar '18

02 Mar '18

Driver probe function has no use and can be deleted. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- Changes: - v2: Rebased against Mark's current tree Mark, is this what you asked for? It does not make much sense to me as you have to redo the work I wanted to save you... sound/soc/codecs/max9867.c | 10 ---------- sound/soc/codecs/max9867.h | 1 - 2 files changed, 11 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 2f60924fe919..6b128dd6c5dd 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -404,17 +404,7 @@ static int max9867_resume(struct device *dev) } #endif -static int max9867_probe(struct snd_soc_codec *codec) -{ - struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); - - dev_dbg(codec->dev, "max98090_probe\n"); - max9867->codec = codec; - return 0; -} - static const struct snd_soc_codec_driver max9867_codec = { - .probe = max9867_probe, .component_driver = { .controls = max9867_snd_controls, .num_controls = ARRAY_SIZE(max9867_snd_controls), diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 65590b4ad62a..55cd9976ff47 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -75,7 +75,6 @@ /* codec private data */ struct max9867_priv { struct regmap *regmap; - struct snd_soc_codec *codec; unsigned int sysclk; unsigned int pclk; unsigned int master; -- 2.16.2

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[alsa-devel] [PATCH] ASoC: Intel: Fix cht_bsw_nau8824 machine driver build error
by Hans de Goede 02 Mar '18

02 Mar '18

The cht_bsw_nau8824 machine driver is passing snd_soc_codec to nau8824_enable_jack_detect(), but that expects a snd_soc_component as argument, causing a compile error. This commit fixes this. Fixes: 5b00ce63e07f ("ASoC: Intel: add nau8824 sound card") Cc: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_nau8824.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 08d0b8128fab..5913ad159cc1 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -103,7 +103,6 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); struct snd_soc_jack *jack = &ctx->jack; - struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dai *codec_dai = runtime->codec_dai; int ret, jack_type; @@ -134,7 +133,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - nau8824_enable_jack_detect(codec, jack); + nau8824_enable_jack_detect(codec_dai->component, jack); return ret; } -- 2.14.3

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[alsa-devel] [PATCH v2 0/3] Enable HDA Codec support on Intel Platforms (Series1)
by Rakesh Ughreja 02 Mar '18

02 Mar '18

Enhanced HD Audio capabilities introduced in HD Audio controllers are added in backward compatible way i.e. it does not change the behavior of the controller with respect to HDA Bus and HDA devices. Since there is no change in the hardware with respect to the HDA bus and HDA device it is more appropriate to represent the same in software as well. In order to align software representation with hardware it makes more sense to have common data structures across device, bus and driver for both enhanced HDA controllers and legacy HDA controllers. This patch series removes the hdac_ext_device structure, hdac_ext_bus and hdac_ext_driver data structures so that legacy and enhanced HDaudio capabilities can be handled in a backward-compatible way without separate definitions. Once this clean-up is complete, additional patches will allow for HDaudio codec support in an ASoC framework without the need to develop new codec drivers, thereby enabling the use of the Intel DSP on more platforms (currently limited to hdmi). Changes v2: - No functionality changes in this series except rebase. Rakesh Ughreja (3): ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead include/sound/hdaudio.h | 20 ++ include/sound/hdaudio_ext.h | 119 +++------- sound/hda/ext/hdac_ext_bus.c | 76 +++--- sound/hda/ext/hdac_ext_controller.c | 55 +++-- sound/hda/ext/hdac_ext_stream.c | 104 ++++----- sound/soc/codecs/hdac_hdmi.c | 416 ++++++++++++++++----------------- sound/soc/intel/skylake/skl-messages.c | 50 ++-- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 112 +++++---- sound/soc/intel/skylake/skl-topology.c | 20 +- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 178 +++++++------- sound/soc/intel/skylake/skl.h | 7 +- 13 files changed, 522 insertions(+), 649 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 00/10] Enable HDA Codec support on Intel Platforms (Series2)
by Rakesh Ughreja 02 Mar '18

02 Mar '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. This patch carry forwards the work that is done in the previous series to enable HD Audio codecs on such platforms. This patch series adds ASoC HDA codec driver for Intel platforms. It is written by reusing the legacy HDA ALSA codec driver. Intention is to maximize the reuse and minimize the changes in the legacy HDA codec driver. Changes in v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Rakesh Ughreja (10): ALSA - Ext hda: add wait for codec to respond after link reset ASoC: Intel: Boards: Machine driver for Intel platforms ASoC: Intel: Skylake: Add entry in sst_acpi_mach for HDA codecs ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ALSA: hda - split snd_hda_codec_new function ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init ALSA: hdac: add extended ops in the hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers ASoC: Intel: Boards: add support for HDA codecs include/sound/hdaudio.h | 9 + include/sound/hdaudio_ext.h | 6 +- sound/hda/ext/hdac_ext_bus.c | 12 +- sound/hda/ext/hdac_ext_controller.c | 8 + sound/pci/hda/hda_bind.c | 12 + sound/pci/hda/hda_codec.c | 68 +++- sound/pci/hda/hda_codec.h | 2 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 464 +++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 23 ++ sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 131 ++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 35 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 175 ++++++++++ sound/soc/intel/skylake/skl-pcm.c | 70 +++- sound/soc/intel/skylake/skl.c | 119 ++++++- sound/soc/intel/skylake/skl.h | 12 +- 19 files changed, 1115 insertions(+), 49 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c -- 2.7.4

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[alsa-devel] [asoc:for-next 32/53] sound/soc/intel/boards/cht_bsw_nau8824.c:137:36: sparse: incorrect type in argument 1 (different base types)
by kbuild test robot 02 Mar '18

02 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 392b22b9bc76706334efdfe11d3747c9ef8962bf commit: 77348ef86e9b9fdab2fbb059b8e00c80adfa98ca [32/53] Merge remote-tracking branches 'asoc/topic/msm8916-wcd-digital', 'asoc/topic/nau8540', 'asoc/topic/nau8810' and 'asoc/topic/nau8824' into asoc-next reproduce: # apt-get install sparse git checkout 77348ef86e9b9fdab2fbb059b8e00c80adfa98ca make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/intel/boards/cht_bsw_nau8824.c:137:36: sparse: incorrect type in argument 1 (different base types) @@ expected struct snd_soc_component *component @@ got ct snd_soc_component *component @@ sound/soc/intel/boards/cht_bsw_nau8824.c:137:36: expected struct snd_soc_component *component sound/soc/intel/boards/cht_bsw_nau8824.c:137:36: got struct snd_soc_codec *codec sound/soc/intel/boards/cht_bsw_nau8824.c: In function 'cht_codec_init': sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type [-Werror=incompatible-pointer-types] nau8824_enable_jack_detect(codec, jack); ^~~~~ In file included from sound/soc/intel/boards/cht_bsw_nau8824.c:31:0: sound/soc/intel/boards/../../codecs/nau8824.h:474:5: note: expected 'struct snd_soc_component *' but argument is of type 'struct snd_soc_codec *' int nau8824_enable_jack_detect(struct snd_soc_component *component, ^~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +137 sound/soc/intel/boards/cht_bsw_nau8824.c 5b00ce63 John Hsu 2018-02-06 101 5b00ce63 John Hsu 2018-02-06 102 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) 5b00ce63 John Hsu 2018-02-06 103 { 5b00ce63 John Hsu 2018-02-06 104 struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); 5b00ce63 John Hsu 2018-02-06 105 struct snd_soc_jack *jack = &ctx->jack; 5b00ce63 John Hsu 2018-02-06 106 struct snd_soc_codec *codec = runtime->codec; 5b00ce63 John Hsu 2018-02-06 107 struct snd_soc_dai *codec_dai = runtime->codec_dai; 5b00ce63 John Hsu 2018-02-06 108 int ret, jack_type; 5b00ce63 John Hsu 2018-02-06 109 5b00ce63 John Hsu 2018-02-06 110 /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ 5b00ce63 John Hsu 2018-02-06 111 ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); 5b00ce63 John Hsu 2018-02-06 112 if (ret < 0) { 5b00ce63 John Hsu 2018-02-06 113 dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); 5b00ce63 John Hsu 2018-02-06 114 return ret; 5b00ce63 John Hsu 2018-02-06 115 } 5b00ce63 John Hsu 2018-02-06 116 5b00ce63 John Hsu 2018-02-06 117 /* NAU88L24 supports 4 butons headset detection 5b00ce63 John Hsu 2018-02-06 118 * KEY_MEDIA 5b00ce63 John Hsu 2018-02-06 119 * KEY_VOICECOMMAND 5b00ce63 John Hsu 2018-02-06 120 * KEY_VOLUMEUP 5b00ce63 John Hsu 2018-02-06 121 * KEY_VOLUMEDOWN 5b00ce63 John Hsu 2018-02-06 122 */ 5b00ce63 John Hsu 2018-02-06 123 jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | 5b00ce63 John Hsu 2018-02-06 124 SND_JACK_BTN_2 | SND_JACK_BTN_3; 5b00ce63 John Hsu 2018-02-06 125 ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, 5b00ce63 John Hsu 2018-02-06 126 cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); 5b00ce63 John Hsu 2018-02-06 127 if (ret) { 5b00ce63 John Hsu 2018-02-06 128 dev_err(runtime->dev, 5b00ce63 John Hsu 2018-02-06 129 "Headset Jack creation failed %d\n", ret); 5b00ce63 John Hsu 2018-02-06 130 return ret; 5b00ce63 John Hsu 2018-02-06 131 } 5b00ce63 John Hsu 2018-02-06 132 snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); 5b00ce63 John Hsu 2018-02-06 133 snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); 5b00ce63 John Hsu 2018-02-06 134 snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); 5b00ce63 John Hsu 2018-02-06 135 snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); 5b00ce63 John Hsu 2018-02-06 136 5b00ce63 John Hsu 2018-02-06 @137 nau8824_enable_jack_detect(codec, jack); 5b00ce63 John Hsu 2018-02-06 138 5b00ce63 John Hsu 2018-02-06 139 return ret; 5b00ce63 John Hsu 2018-02-06 140 } 5b00ce63 John Hsu 2018-02-06 141 :::::: The code at line 137 was first introduced by commit :::::: 5b00ce63e07fac692400db80d3e1fbd1f1f65701 ASoC: Intel: add nau8824 sound card :::::: TO: John Hsu <KCHSU0(a)nuvoton.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v2] ASoC: max9867: Improve error logging
by Ladislav Michl 02 Mar '18

02 Mar '18

Tell user what are clock rate limits and reindent log messages. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- Changes: - v2: Rebase against current branch sound/soc/codecs/max9867.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 6b128dd6c5dd..b2a613e54fc7 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -291,7 +291,9 @@ static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, value |= MAX9867_PSCLK_40_60; max9867->pclk = freq/4; } else { - pr_err("bad clock frequency %d", freq); + dev_err(codec->dev, + "Invalid clock frequency %uHz (required 10-60MHz)\n", + freq); return -EINVAL; } value = value << MAX9867_PSCLK_SHIFT; @@ -476,8 +478,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c, max9867->regmap = devm_regmap_init_i2c(i2c, &max9867_regmap); if (IS_ERR(max9867->regmap)) { ret = PTR_ERR(max9867->regmap); - dev_err(&i2c->dev, - "Failed to allocate regmap: %d\n", ret); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); return ret; } ret = regmap_read(max9867->regmap, -- 2.16.2

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[alsa-devel] [asoc:topic/max9867 2/4] sound/soc/codecs/max9867.c:294:11: error: 'component' undeclared; did you mean 'complete'?
by kbuild test robot 01 Mar '18

01 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/max9867 head: 79e139744508105bd749300f364563934bff23f2 commit: 8b9c716aada77e72e2fe82320aba5a95b75ca400 [2/4] ASoC: max9867: Improve error logging config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 8b9c716aada77e72e2fe82320aba5a95b75ca400 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/codecs/max9867.c: In function 'max9867_set_dai_sysclk': >> sound/soc/codecs/max9867.c:294:11: error: 'component' undeclared (first use in this function); did you mean 'complete'? dev_err(component->dev, ^~~~~~~~~ complete sound/soc/codecs/max9867.c:294:11: note: each undeclared identifier is reported only once for each function it appears in vim +294 sound/soc/codecs/max9867.c 275 276 static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, 277 int clk_id, unsigned int freq, int dir) 278 { 279 struct snd_soc_codec *codec = codec_dai->codec; 280 struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); 281 int value = 0; 282 283 /* Set the prescaler based on the master clock frequency*/ 284 if (freq >= 10000000 && freq <= 20000000) { 285 value |= MAX9867_PSCLK_10_20; 286 max9867->pclk = freq; 287 } else if (freq >= 20000000 && freq <= 40000000) { 288 value |= MAX9867_PSCLK_20_40; 289 max9867->pclk = freq/2; 290 } else if (freq >= 40000000 && freq <= 60000000) { 291 value |= MAX9867_PSCLK_40_60; 292 max9867->pclk = freq/4; 293 } else { > 294 dev_err(component->dev, 295 "Invalid clock frequency %uHz (required 10-60MHz)\n", 296 freq); 297 return -EINVAL; 298 } 299 value = value << MAX9867_PSCLK_SHIFT; 300 max9867->sysclk = freq; 301 /* exact integer mode is not supported */ 302 value &= ~MAX9867_FREQ_MASK; 303 regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, 304 MAX9867_PSCLK_MASK, value); 305 return 0; 306 } 307 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:for-next 32/53] sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type
by kbuild test robot 01 Mar '18

01 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 90d3c54575fc6488e912870c3d1a8b4af6a5a7b7 commit: bdc6f82ed1a68bdedea3c899e4ac35c88e02e63a [32/53] Merge remote-tracking branches 'asoc/topic/msm8916-wcd-digital', 'asoc/topic/nau8540', 'asoc/topic/nau8810' and 'asoc/topic/nau8824' into asoc-next config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout bdc6f82ed1a68bdedea3c899e4ac35c88e02e63a # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/intel/boards/cht_bsw_nau8824.c: In function 'cht_codec_init': >> sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type [-Werror=incompatible-pointer-types] nau8824_enable_jack_detect(codec, jack); ^~~~~ In file included from sound/soc/intel/boards/cht_bsw_nau8824.c:31:0: sound/soc/intel/boards/../../codecs/nau8824.h:474:5: note: expected 'struct snd_soc_component *' but argument is of type 'struct snd_soc_codec *' int nau8824_enable_jack_detect(struct snd_soc_component *component, ^~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/nau8824_enable_jack_detect +137 sound/soc/intel/boards/cht_bsw_nau8824.c 5b00ce63 John Hsu 2018-02-06 101 5b00ce63 John Hsu 2018-02-06 102 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) 5b00ce63 John Hsu 2018-02-06 103 { 5b00ce63 John Hsu 2018-02-06 104 struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); 5b00ce63 John Hsu 2018-02-06 105 struct snd_soc_jack *jack = &ctx->jack; 5b00ce63 John Hsu 2018-02-06 106 struct snd_soc_codec *codec = runtime->codec; 5b00ce63 John Hsu 2018-02-06 107 struct snd_soc_dai *codec_dai = runtime->codec_dai; 5b00ce63 John Hsu 2018-02-06 108 int ret, jack_type; 5b00ce63 John Hsu 2018-02-06 109 5b00ce63 John Hsu 2018-02-06 110 /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ 5b00ce63 John Hsu 2018-02-06 111 ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); 5b00ce63 John Hsu 2018-02-06 112 if (ret < 0) { 5b00ce63 John Hsu 2018-02-06 113 dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); 5b00ce63 John Hsu 2018-02-06 114 return ret; 5b00ce63 John Hsu 2018-02-06 115 } 5b00ce63 John Hsu 2018-02-06 116 5b00ce63 John Hsu 2018-02-06 117 /* NAU88L24 supports 4 butons headset detection 5b00ce63 John Hsu 2018-02-06 118 * KEY_MEDIA 5b00ce63 John Hsu 2018-02-06 119 * KEY_VOICECOMMAND 5b00ce63 John Hsu 2018-02-06 120 * KEY_VOLUMEUP 5b00ce63 John Hsu 2018-02-06 121 * KEY_VOLUMEDOWN 5b00ce63 John Hsu 2018-02-06 122 */ 5b00ce63 John Hsu 2018-02-06 123 jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | 5b00ce63 John Hsu 2018-02-06 124 SND_JACK_BTN_2 | SND_JACK_BTN_3; 5b00ce63 John Hsu 2018-02-06 125 ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, 5b00ce63 John Hsu 2018-02-06 126 cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); 5b00ce63 John Hsu 2018-02-06 127 if (ret) { 5b00ce63 John Hsu 2018-02-06 128 dev_err(runtime->dev, 5b00ce63 John Hsu 2018-02-06 129 "Headset Jack creation failed %d\n", ret); 5b00ce63 John Hsu 2018-02-06 130 return ret; 5b00ce63 John Hsu 2018-02-06 131 } 5b00ce63 John Hsu 2018-02-06 132 snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); 5b00ce63 John Hsu 2018-02-06 133 snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); 5b00ce63 John Hsu 2018-02-06 134 snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); 5b00ce63 John Hsu 2018-02-06 135 snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); 5b00ce63 John Hsu 2018-02-06 136 5b00ce63 John Hsu 2018-02-06 @137 nau8824_enable_jack_detect(codec, jack); 5b00ce63 John Hsu 2018-02-06 138 5b00ce63 John Hsu 2018-02-06 139 return ret; 5b00ce63 John Hsu 2018-02-06 140 } 5b00ce63 John Hsu 2018-02-06 141 :::::: The code at line 137 was first introduced by commit :::::: 5b00ce63e07fac692400db80d3e1fbd1f1f65701 ASoC: Intel: add nau8824 sound card :::::: TO: John Hsu <KCHSU0(a)nuvoton.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] Applied "ASoC: rt5514: Fix rt5514_spi_burst_read() buffer passing" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: rt5514: Fix rt5514_spi_burst_read() buffer passing has been applied to the asoc tree at https://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 29838a5c1e77ed93d41fb9837390e243b9b73518 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven <geert(a)linux-m68k.org> Date: Tue, 30 Jan 2018 14:19:42 +0100 Subject: [PATCH] ASoC: rt5514: Fix rt5514_spi_burst_read() buffer passing MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit rt5514_dsp_voice_wake_up_put() passes "(u8 *)&buf" to rt5514_spi_burst_read(), where the cast is needed to silence a compiler warning: sound/soc/codecs/rt5514.c: In function ‘rt5514_dsp_voice_wake_up_put’: sound/soc/codecs/rt5514.c:357: warning: passing argument 2 of ‘rt5514_spi_burst_read’ from incompatible pointer type However, as the variable is declared as an array ("u8 buf[8]"), the function should just pass "buf" instead. Fixes: fc9cab05837639ce ("ASoC: rt5514: The DSP clock can be calibrated by the other clock source") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 2d79571098ac..b358704ee19c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -353,8 +353,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, msleep(20); #if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) rt5514_spi_burst_read(RT5514_PLL3_CALIB_CTRL6 | - RT5514_DSP_MAPPING, - (u8 *)&buf, sizeof(buf)); + RT5514_DSP_MAPPING, buf, sizeof(buf)); #else dev_err(component->dev, "There is no SPI driver for" " loading the firmware\n"); -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: rt5514: Fix uninitialized calibration value" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: rt5514: Fix uninitialized calibration value has been applied to the asoc tree at https://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 890861aa85043cc9d99a01e1bd26883722e95243 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven <geert(a)linux-m68k.org> Date: Tue, 30 Jan 2018 14:19:43 +0100 Subject: [PATCH] ASoC: rt5514: Fix uninitialized calibration value MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With gcc-4.1.2, if CONFIG_SND_SOC_RT5514_SPI is not set: sound/soc/codecs/rt5514.c: In function ‘rt5514_dsp_voice_wake_up_put’: sound/soc/codecs/rt5514.c:363: warning: ‘buf[0]’ is used uninitialized in this function sound/soc/codecs/rt5514.c:363: warning: ‘buf[1]’ is used uninitialized in this function sound/soc/codecs/rt5514.c:363: warning: ‘buf[2]’ is used uninitialized in this function sound/soc/codecs/rt5514.c:363: warning: ‘buf[3]’ is used uninitialized in this function Fix this by initializing the buffer with zeroes, to avoid random bits being written to the calibration register later. Fixes: fc9cab05837639ce ("ASoC: rt5514: The DSP clock can be calibrated by the other clock source") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index b358704ee19c..e8a66b03faab 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -357,6 +357,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, #else dev_err(component->dev, "There is no SPI driver for" " loading the firmware\n"); + memset(buf, 0, sizeof(buf)); #endif rt5514->pll3_cal_value = buf[0] | buf[1] << 8 | buf[2] << 16 | buf[3] << 24; -- 2.16.2

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[alsa-devel] [PATCH 0/5] ASoC: Intel: Skylake fixes
by Guneshwor Singh 01 Mar '18

01 Mar '18

This includes fixes for Skylake+ driver which are improving widget type check, fixes for simultaneous wake-on-voice (WoV) and capture use-case, fix to improve firmware download latency and lastly, fix to use uuid instead of module id for library modules. Guneshwor Singh (1): ASoC: Intel: Skylake: Refine skl widget type check Pawse, GuruprasadX (2): ASoC: Intel: Skylake: Add BE DAI for WoV usecase ASoC: Intel: board: Add BE DAI link for WoV and update DAPM machine map. Sanyog Kale (1): ASoC: Intel: Skylake: Disable clock and power gating during fw/lib download Sriram Periyasamy (1): ASoC: Intel: Skylake: Find module id from UUID for bind params sound/soc/intel/boards/bxt_rt298.c | 15 +++++ sound/soc/intel/skylake/skl-messages.c | 7 ++- sound/soc/intel/skylake/skl-pcm.c | 20 +++++- sound/soc/intel/skylake/skl-sst-ipc.h | 3 + sound/soc/intel/skylake/skl-topology.c | 107 +++++++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-topology.h | 13 +++- sound/soc/intel/skylake/skl.c | 21 +++++++ sound/soc/intel/skylake/skl.h | 4 ++ 8 files changed, 176 insertions(+), 14 deletions(-) -- 2.16.1

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[alsa-devel] Applied "ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection has been applied to the asoc tree at https://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 1310e737a68741c9b06cf63be0084bae6018b308 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 25 Feb 2018 11:46:55 +0100 Subject: [PATCH] ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection To determine if a plugged in jack is a headset (speakers + mic) or headphones (mic contact shorted to ground) we use the micbias1 OVer Current Detect (OVCD) functionality. For this to work we need to have a micbias current to actually cause an overcurrent condition when headphones are plugged in, so jack-type detection requires both the LDO and micbias1 supplies to be on. Before this commit there were 2 issues with the handling of this: 1) The LDO supply was force-enabled twice and never disabled again even though it only needs to be forced on when doing jack-type detection 2) micbias1 was not force-enabled, and thus may be off when doing jack-type detection This commit fixes both by force-enabling the LDO and micbias1 supplies before checking for an overcurrent condition and disabling them afterwards. Note that both supplies will still get turned on normally (and OVCD will protect against overcurrent) when the micbias1 is enabled normally because the user has activated a sound stream recording from the mic. Tested-by: Carlo Caione <carlo(a)endlessm.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5651.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index cbf9f56ecb14..be942886700d 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1561,6 +1561,28 @@ static int rt5651_set_bias_level(struct snd_soc_component *component, return 0; } +static void rt5651_enable_micbias1_for_ovcd(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "micbias1"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "micbias1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "LDO"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + static irqreturn_t rt5651_irq(int irq, void *data) { struct rt5651_priv *rt5651 = data; @@ -1574,7 +1596,6 @@ static irqreturn_t rt5651_irq(int irq, void *data) static int rt5651_set_jack(struct snd_soc_component *component, struct snd_soc_jack *hp_jack, void *data) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); int ret; @@ -1616,9 +1637,6 @@ static int rt5651_set_jack(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5651_PWR_ANLG2, RT5651_PWR_JD_M, RT5651_PWR_JD_M); - snd_soc_dapm_force_enable_pin(dapm, "LDO"); - snd_soc_dapm_sync(dapm); - snd_soc_component_update_bits(component, RT5651_MICBIAS, 0x38, 0x38); rt5651->hp_jack = hp_jack; @@ -1819,13 +1837,10 @@ static const struct dmi_system_id rt5651_quirk_table[] = { static int rt5651_jack_detect(struct snd_soc_component *component, int jack_insert) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); int jack_type; if (jack_insert) { - snd_soc_dapm_force_enable_pin(dapm, "LDO"); - snd_soc_dapm_sync(dapm); - + rt5651_enable_micbias1_for_ovcd(component); snd_soc_component_update_bits(component, RT5651_MICBIAS, RT5651_MIC1_OVCD_MASK | RT5651_MIC1_OVTH_MASK | @@ -1842,6 +1857,7 @@ static int rt5651_jack_detect(struct snd_soc_component *component, int jack_inse jack_type = SND_JACK_HEADSET; snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, RT5651_MB1_OC_CLR, 0); + rt5651_disable_micbias1_for_ovcd(component); } else { /* jack out */ jack_type = 0; -- 2.16.2

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[alsa-devel] xhci: Not enough bandwidth for new device state
by Waldemar Brodkorb 01 Mar '18

01 Mar '18

Hi, I am getting these messages from a Debian 9.3 system using Linux 4.9.x: [ 547.352746] usb 3-5.2: 3:1: usb_set_interface failed (-28) [ 548.352865] usb 3-5.3: Not enough bandwidth for new device state. [ 548.352868] usb 3-5.3: Not enough bandwidth for altsetting 1 I have connected a 7-Port USB 3.0 Hub to my machine. I have 6 different Headsets and USB Soundcards connected to the HUB. As soon as I start a C++ application which opens all ALSA devices I get these errors and the USB stopps working correctly. I think the maximum count of USB sound devices, which behave stable is three or four. Is this some miscalculation of the Linux kernel regarding the USB bandwidth usage? How I can determine how much bandwidth the USB sound card driver preallocates? Or are five USB soundcards really too much for an USB 3.0 bus? Thanks in advance for any advise, best regards Waldemar P.s.: I sent this message to LKML, but maybe it is more appropriate to ask here.

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[alsa-devel] Applied "ASoC: Intel: sst: Fix error-code check in sst_pause_stream()" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Fix error-code check in sst_pause_stream() has been applied to the asoc tree at https://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 6f5062773747b3d3eb9f957030ac1be98e983ac2 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:38 +0100 Subject: [PATCH] ASoC: Intel: sst: Fix error-code check in sst_pause_stream() The value returned by sst_prepare_and_post_msg() is a negated SST_ERR_* value, so we must check for -SST_ERR_INVALID_STREAM_ID. Note that sst_pause_resume() already has the correct check. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index 7ee6aeb7e0af..b082b0922a7a 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -253,7 +253,7 @@ int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) if (retval == 0) { str_info->prev = str_info->status; str_info->status = STREAM_PAUSED; - } else if (retval == SST_ERR_INVALID_STREAM_ID) { + } else if (retval == -SST_ERR_INVALID_STREAM_ID) { retval = -EINVAL; mutex_lock(&sst_drv_ctx->sst_lock); sst_clean_stream(str_info); -- 2.16.2

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[alsa-devel] Applied "ASoC: Intel: sst: Remove 2 unused members from stream_info struct" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove 2 unused members from stream_info struct has been applied to the asoc tree at https://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 bd47469c9985b50831a55bdda5a36f4672452764 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:39 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove 2 unused members from stream_info struct Remove the unused ops and str_id members from the stream_info struct. While at it also remove some kernel-doc comments for members which have already been removed in the past. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 7 ------- sound/soc/intel/atom/sst/sst_pvt.c | 1 - 2 files changed, 8 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index e02e2b4cc08f..1f3107909044 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -181,21 +181,15 @@ struct sst_block { * * @status : stream current state * @prev : stream prev state - * @ops : stream operation pb/cp/drm... - * @bufs: stream buffer list * @lock : stream mutex for protecting state * @pcm_substream : PCM substream * @period_elapsed : PCM period elapsed callback * @sfreq : stream sampling freq - * @str_type : stream type * @cumm_bytes : cummulative bytes decoded - * @str_type : stream type - * @src : stream source */ struct stream_info { unsigned int status; unsigned int prev; - unsigned int ops; struct mutex lock; void *pcm_substream; @@ -212,7 +206,6 @@ struct stream_info { unsigned int num_ch; unsigned int pipe_id; - unsigned int str_id; unsigned int task_id; }; diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index b1e6b8f34a6a..d7d38b0f68db 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -365,7 +365,6 @@ void sst_init_stream(struct stream_info *stream, { stream->status = STREAM_INIT; stream->prev = STREAM_UN_INIT; - stream->ops = ops; } int sst_validate_strid( -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: Intel: sst: Remove unnecessary sst_init_stream() function" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove unnecessary sst_init_stream() function has been applied to the asoc tree at https://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 8cf732bbd8d6bc9a28ac48c7bac0ccc9f55cb151 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:40 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove unnecessary sst_init_stream() function sst_init_stream() has only one caller and all its function arguments are unused. Inline it on the one call site and remove it. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 2 -- sound/soc/intel/atom/sst/sst_pvt.c | 7 ------- sound/soc/intel/atom/sst/sst_stream.c | 5 ++--- 3 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 1f3107909044..164b0f674c20 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -494,8 +494,6 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, void sst_process_pending_msg(struct work_struct *work); int sst_assign_pvt_id(struct intel_sst_drv *sst_drv_ctx); -void sst_init_stream(struct stream_info *stream, - int codec, int sst_id, int ops, u8 slot); int sst_validate_strid(struct intel_sst_drv *sst_drv_ctx, int str_id); struct stream_info *get_stream_info(struct intel_sst_drv *sst_drv_ctx, int str_id); diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index d7d38b0f68db..af93244b4868 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -360,13 +360,6 @@ int sst_assign_pvt_id(struct intel_sst_drv *drv) return local; } -void sst_init_stream(struct stream_info *stream, - int codec, int sst_id, int ops, u8 slot) -{ - stream->status = STREAM_INIT; - stream->prev = STREAM_UN_INIT; -} - int sst_validate_strid( struct intel_sst_drv *sst_drv_ctx, int str_id) { diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index b082b0922a7a..7cf5dd1993bc 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -83,6 +83,8 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) pipe_id = str_params->device_type; task_id = str_params->task; + sst_drv_ctx->streams[str_id].status = STREAM_INIT; + sst_drv_ctx->streams[str_id].prev = STREAM_UN_INIT; sst_drv_ctx->streams[str_id].pipe_id = pipe_id; sst_drv_ctx->streams[str_id].task_id = task_id; sst_drv_ctx->streams[str_id].num_ch = num_ch; @@ -100,9 +102,6 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) pipe_id, task_id); /* allocate device type context */ - sst_init_stream(&sst_drv_ctx->streams[str_id], alloc_param.codec_type, - str_id, alloc_param.operation, 0); - dev_dbg(sst_drv_ctx->dev, "Alloc for str %d pipe %#x\n", str_id, pipe_id); ret = sst_prepare_and_post_msg(sst_drv_ctx, task_id, IPC_CMD, -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states has been applied to the asoc tree at https://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 de9833476de7eb1d4b8bdc599cbbd616fb0e1a28 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:41 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states STREAM_DECODE is completely unused, status == STREAM_RESET was checked for, but never set, remove both. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 4 +--- sound/soc/intel/atom/sst/sst_drv_interface.c | 19 ------------------- 2 files changed, 1 insertion(+), 22 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 164b0f674c20..f4fd442080b2 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -65,9 +65,7 @@ enum sst_stream_states { STREAM_UN_INIT = 0, /* Freed/Not used stream */ STREAM_RUNNING = 1, /* Running */ STREAM_PAUSED = 2, /* Paused stream */ - STREAM_DECODE = 3, /* stream is in decoding only state */ - STREAM_INIT = 4, /* stream init, waiting for data */ - STREAM_RESET = 5, /* force reset on recovery */ + STREAM_INIT = 3, /* stream init, waiting for data */ }; enum sst_ram_type { diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 71af5449be90..6a8b253c58d2 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -238,16 +238,7 @@ static int sst_cdev_close(struct device *dev, unsigned int str_id) return -EINVAL; } - if (stream->status == STREAM_RESET) { - dev_dbg(dev, "stream in reset state...\n"); - stream->status = STREAM_UN_INIT; - - retval = 0; - goto put; - } - retval = sst_free_stream(ctx, str_id); -put: stream->compr_cb_param = NULL; stream->compr_cb = NULL; @@ -256,7 +247,6 @@ static int sst_cdev_close(struct device *dev, unsigned int str_id) dev_dbg(dev, "End\n"); return retval; - } static int sst_cdev_ack(struct device *dev, unsigned int str_id, @@ -486,16 +476,7 @@ static int sst_close_pcm_stream(struct device *dev, unsigned int str_id) return -EINVAL; } - if (stream->status == STREAM_RESET) { - /* silently fail here as we have cleaned the stream earlier */ - dev_dbg(ctx->dev, "stream in reset state...\n"); - - retval = 0; - goto put; - } - retval = free_stream_context(ctx, str_id); -put: stream->pcm_substream = NULL; stream->status = STREAM_UN_INIT; stream->period_elapsed = NULL; -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: Intel: sst: Add sst_realloc_stream() function" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Add sst_realloc_stream() function has been applied to the asoc tree at https://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 473858cad1ea3d4e6c287bad23961cd5fb7382d9 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:42 +0100 Subject: [PATCH] ASoC: Intel: sst: Add sst_realloc_stream() function Move the struct snd_sst_alloc_mrfld alloc parameters from the stack into struct stream_info and add a new sst_realloc_stream() function which can re-alloc a stream with the same parameters as before. This is a preparation patch for fixing suspend/resume issues with some SST / DSP firmware versions. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 3 ++ sound/soc/intel/atom/sst/sst_stream.c | 88 +++++++++++++++++++++-------------- 2 files changed, 56 insertions(+), 35 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index f4fd442080b2..a357cd615b59 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -180,6 +180,7 @@ struct sst_block { * @status : stream current state * @prev : stream prev state * @lock : stream mutex for protecting state + * @alloc_param : parameters used for stream (re-)allocation * @pcm_substream : PCM substream * @period_elapsed : PCM period elapsed callback * @sfreq : stream sampling freq @@ -189,6 +190,7 @@ struct stream_info { unsigned int status; unsigned int prev; struct mutex lock; + struct snd_sst_alloc_mrfld alloc_param; void *pcm_substream; void (*period_elapsed)(void *pcm_substream); @@ -429,6 +431,7 @@ struct intel_sst_ops { void (*post_download)(struct intel_sst_drv *sst); }; +int sst_realloc_stream(struct intel_sst_drv *sst_drv_ctx, int str_id); int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int id); int sst_resume_stream(struct intel_sst_drv *sst_drv_ctx, int id); int sst_drop_stream(struct intel_sst_drv *sst_drv_ctx, int id); diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index 7cf5dd1993bc..fcedaa237505 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -35,29 +35,31 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) { - struct snd_sst_alloc_mrfld alloc_param; + struct snd_pcm_params *pcm_params; struct snd_sst_params *str_params; struct snd_sst_tstamp fw_tstamp; struct stream_info *str_info; - struct snd_sst_alloc_response *response; - unsigned int str_id, pipe_id, task_id; - int i, num_ch, ret = 0; - void *data = NULL; + int i, num_ch, str_id; dev_dbg(sst_drv_ctx->dev, "Enter\n"); str_params = (struct snd_sst_params *)params; - memset(&alloc_param, 0, sizeof(alloc_param)); - alloc_param.operation = str_params->ops; - alloc_param.codec_type = str_params->codec; - alloc_param.sg_count = str_params->aparams.sg_count; - alloc_param.ring_buf_info[0].addr = + str_id = str_params->stream_id; + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; + + memset(&str_info->alloc_param, 0, sizeof(str_info->alloc_param)); + str_info->alloc_param.operation = str_params->ops; + str_info->alloc_param.codec_type = str_params->codec; + str_info->alloc_param.sg_count = str_params->aparams.sg_count; + str_info->alloc_param.ring_buf_info[0].addr = str_params->aparams.ring_buf_info[0].addr; - alloc_param.ring_buf_info[0].size = + str_info->alloc_param.ring_buf_info[0].size = str_params->aparams.ring_buf_info[0].size; - alloc_param.frag_size = str_params->aparams.frag_size; + str_info->alloc_param.frag_size = str_params->aparams.frag_size; - memcpy(&alloc_param.codec_params, &str_params->sparams, + memcpy(&str_info->alloc_param.codec_params, &str_params->sparams, sizeof(struct snd_sst_stream_params)); /* @@ -67,46 +69,62 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) * Currently hardcoding as per FW reqm. */ num_ch = sst_get_num_channel(str_params); + pcm_params = &str_info->alloc_param.codec_params.uc.pcm_params; for (i = 0; i < 8; i++) { if (i < num_ch) - alloc_param.codec_params.uc.pcm_params.channel_map[i] = i; + pcm_params->channel_map[i] = i; else - alloc_param.codec_params.uc.pcm_params.channel_map[i] = 0xFF; + pcm_params->channel_map[i] = 0xff; } - str_id = str_params->stream_id; - str_info = get_stream_info(sst_drv_ctx, str_id); - if (str_info == NULL) { - dev_err(sst_drv_ctx->dev, "get stream info returned null\n"); - return -EINVAL; - } - - pipe_id = str_params->device_type; - task_id = str_params->task; sst_drv_ctx->streams[str_id].status = STREAM_INIT; sst_drv_ctx->streams[str_id].prev = STREAM_UN_INIT; - sst_drv_ctx->streams[str_id].pipe_id = pipe_id; - sst_drv_ctx->streams[str_id].task_id = task_id; + sst_drv_ctx->streams[str_id].pipe_id = str_params->device_type; + sst_drv_ctx->streams[str_id].task_id = str_params->task; sst_drv_ctx->streams[str_id].num_ch = num_ch; if (sst_drv_ctx->info.lpe_viewpt_rqd) - alloc_param.ts = sst_drv_ctx->info.mailbox_start + + str_info->alloc_param.ts = sst_drv_ctx->info.mailbox_start + sst_drv_ctx->tstamp + (str_id * sizeof(fw_tstamp)); else - alloc_param.ts = sst_drv_ctx->mailbox_add + + str_info->alloc_param.ts = sst_drv_ctx->mailbox_add + sst_drv_ctx->tstamp + (str_id * sizeof(fw_tstamp)); dev_dbg(sst_drv_ctx->dev, "alloc tstamp location = 0x%x\n", - alloc_param.ts); + str_info->alloc_param.ts); dev_dbg(sst_drv_ctx->dev, "assigned pipe id 0x%x to task %d\n", - pipe_id, task_id); + str_info->pipe_id, str_info->task_id); + + return sst_realloc_stream(sst_drv_ctx, str_id); +} + +/** + * sst_realloc_stream - Send msg for (re-)allocating a stream using the + * @sst_drv_ctx intel_sst_drv context pointer + * @str_id: stream ID + * + * Send a msg for (re-)allocating a stream using the parameters previously + * passed to sst_alloc_stream_mrfld() for the same stream ID. + * Return: 0 or negative errno value. + */ +int sst_realloc_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + struct snd_sst_alloc_response *response; + struct stream_info *str_info; + void *data = NULL; + int ret; + + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; - /* allocate device type context */ dev_dbg(sst_drv_ctx->dev, "Alloc for str %d pipe %#x\n", - str_id, pipe_id); - ret = sst_prepare_and_post_msg(sst_drv_ctx, task_id, IPC_CMD, - IPC_IA_ALLOC_STREAM_MRFLD, pipe_id, sizeof(alloc_param), - &alloc_param, &data, true, true, false, true); + str_id, str_info->pipe_id); + + ret = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, + IPC_IA_ALLOC_STREAM_MRFLD, str_info->pipe_id, + sizeof(str_info->alloc_param), &str_info->alloc_param, + &data, true, true, false, true); if (ret < 0) { dev_err(sst_drv_ctx->dev, "FW alloc failed ret %d\n", ret); -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: Intel: sst: Free streams on suspend, re-alloc on resume" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Free streams on suspend, re-alloc on resume has been applied to the asoc tree at https://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 bf642bf57398572327072d306e26e38aebd94fa7 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:43 +0100 Subject: [PATCH] ASoC: Intel: sst: Free streams on suspend, re-alloc on resume The Bay Trail SST-DSP firmware version looses track of all streams over a suspend/resume, failing any attempts to resume and/or free streams, with a SST_ERR_INVALID_STREAM_ID error. This commit adds support for free-ing the streams on suspend and re-allocating them on resume, fixing suspend/resume issues on devices using this firmware version. This new behavior gets triggered by a new flag in sst_platform_info which only gets set on Bay Trail platforms. This has been tested on the following devices: -Asus T100TA, Bay Trail + ALC5642 codec -Ployer MOMO7W, Bay Trail CR + ALC5652 codec Tested-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/x86/include/asm/platform_sst_audio.h | 1 + sound/soc/intel/atom/sst/sst.c | 24 +++++++++++++++++++++++- sound/soc/intel/atom/sst/sst.h | 4 ++++ sound/soc/intel/atom/sst/sst_acpi.c | 3 ++- sound/soc/intel/atom/sst/sst_stream.c | 24 +++++++++++++++++++++++- 5 files changed, 53 insertions(+), 3 deletions(-) diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h index 5973a2f3db3d..059823bb8af7 100644 --- a/arch/x86/include/asm/platform_sst_audio.h +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -135,6 +135,7 @@ struct sst_platform_info { const struct sst_res_info *res_info; const struct sst_lib_dnld_info *lib_info; const char *platform; + bool streams_lost_on_suspend; }; int add_sst_platform_device(void); #endif diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 8afdff457579..0962bc9adc62 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -449,6 +449,13 @@ static int intel_sst_suspend(struct device *dev) dev_err(dev, "stream %d is running, can't suspend, abort\n", i); return -EBUSY; } + + if (ctx->pdata->streams_lost_on_suspend) { + stream->resume_status = stream->status; + stream->resume_prev = stream->prev; + if (stream->status != STREAM_UN_INIT) + sst_free_stream(ctx, i); + } } synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); @@ -509,8 +516,8 @@ static int intel_sst_resume(struct device *dev) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); struct sst_fw_save *fw_save = ctx->fw_save; - int ret = 0; struct sst_block *block; + int i, ret = 0; if (!fw_save) return 0; @@ -550,6 +557,21 @@ static int intel_sst_resume(struct device *dev) sst_set_fw_state_locked(ctx, SST_FW_RUNNING); } + if (ctx->pdata->streams_lost_on_suspend) { + for (i = 1; i <= ctx->info.max_streams; i++) { + struct stream_info *stream = &ctx->streams[i]; + + if (stream->resume_status != STREAM_UN_INIT) { + dev_dbg(ctx->dev, "Re-allocing stream %d status %d prev %d\n", + i, stream->resume_status, + stream->resume_prev); + sst_realloc_stream(ctx, i); + stream->status = stream->resume_status; + stream->prev = stream->resume_prev; + } + } + } + sst_free_block(ctx, block); return ret; } diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index a357cd615b59..b2a705dc9304 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -179,6 +179,8 @@ struct sst_block { * * @status : stream current state * @prev : stream prev state + * @resume_status : stream current state to restore on resume + * @resume_prev : stream prev state to restore on resume * @lock : stream mutex for protecting state * @alloc_param : parameters used for stream (re-)allocation * @pcm_substream : PCM substream @@ -189,6 +191,8 @@ struct sst_block { struct stream_info { unsigned int status; unsigned int prev; + unsigned int resume_status; + unsigned int resume_prev; struct mutex lock; struct snd_sst_alloc_mrfld alloc_param; diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 6cd481bec275..c90b04cc071d 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -143,10 +143,11 @@ static struct sst_platform_info byt_rvp_platform_data = { .lib_info = &byt_lib_dnld_info, .res_info = &byt_rvp_res_info, .platform = "sst-mfld-platform", + .streams_lost_on_suspend = true, }; /* Cherryview (Cherrytrail and Braswell) uses same mrfld dpcm fw as Baytrail, - * so pdata is same as Baytrail. + * so pdata is same as Baytrail, minus the streams_lost_on_suspend quirk. */ static struct sst_platform_info chv_platform_data = { .probe_data = &byt_fwparse_info, diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index fcedaa237505..107271f7dd63 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -302,7 +302,29 @@ int sst_resume_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) return -EINVAL; if (str_info->status == STREAM_RUNNING) return 0; - if (str_info->status == STREAM_PAUSED) { + + if (str_info->resume_status == STREAM_PAUSED && + str_info->resume_prev == STREAM_RUNNING) { + /* + * Stream was running before suspend and re-created on resume, + * start it to get back to running state. + */ + dev_dbg(sst_drv_ctx->dev, "restart recreated stream after resume\n"); + str_info->status = STREAM_RUNNING; + str_info->prev = STREAM_PAUSED; + retval = sst_start_stream(sst_drv_ctx, str_id); + str_info->resume_status = STREAM_UN_INIT; + } else if (str_info->resume_status == STREAM_PAUSED && + str_info->resume_prev == STREAM_INIT) { + /* + * Stream was idle before suspend and re-created on resume, + * keep it as is. + */ + dev_dbg(sst_drv_ctx->dev, "leaving recreated stream idle after resume\n"); + str_info->status = STREAM_INIT; + str_info->prev = STREAM_PAUSED; + str_info->resume_status = STREAM_UN_INIT; + } else if (str_info->status == STREAM_PAUSED) { retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, IPC_IA_RESUME_STREAM_MRFLD, str_info->pipe_id, 0, NULL, NULL, -- 2.16.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: kbl: Enable mclk and ssp sclk early for rt5663
by Naveen Manohar 01 Mar '18

01 Mar '18

rt5663 requires mclk/sclk early to synchronize its internal clocks. Enable these clocks early. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Kevin Cheng <kevin.cheng(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3c9585eb6860..0e6b7e79441c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -204,13 +204,18 @@ static const struct snd_soc_dapm_widget kabylake_5663_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("DP", NULL), SND_SOC_DAPM_SPK("HDMI", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_5663_map[] = { + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: Intel: add nau8824 sound card" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: add nau8824 sound card has been applied to the asoc tree at https://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 5b00ce63e07fac692400db80d3e1fbd1f1f65701 Mon Sep 17 00:00:00 2001 From: John Hsu <KCHSU0(a)nuvoton.com> Date: Tue, 6 Feb 2018 15:02:43 +0800 Subject: [PATCH] ASoC: Intel: add nau8824 sound card Create sound card with NAU88L24 codec in Intel Cherryview-based platforms, Cherrytrail and Braswell. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cht_bsw_nau8824.c | 282 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cht-match.c | 9 + 4 files changed, 304 insertions(+) create mode 100644 sound/soc/intel/boards/cht_bsw_nau8824.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index eedc83b96fbb..24797482a3d2 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -125,6 +125,17 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_CHT_BSW_NAU8824_MACH + tristate "Cherrytrail & Braswell with NAU88L24 codec" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ACPI + select SND_SOC_NAU8824 + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with NAU88L24 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_BYT_CHT_DA7213_MACH tristate "Baytrail & Cherrytrail with DA7212/7213 codec" depends on X86_INTEL_LPSS && I2C && ACPI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 6fae506c40fa..92b5507291af 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -11,6 +11,7 @@ snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-sst-cht-bsw-nau8824-objs := cht_bsw_nau8824.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o @@ -33,6 +34,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_NAU8824_MACH) += snd-soc-sst-cht-bsw-nau8824.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c new file mode 100644 index 000000000000..08d0b8128fab --- /dev/null +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -0,0 +1,282 @@ +/* + * cht-bsw-nau8824.c - ASoc Machine driver for Intel Cherryview-based + * platforms Cherrytrail and Braswell, with nau8824 codec. + * + * Copyright (C) 2018 Intel Corp + * Copyright (C) 2018 Nuvoton Technology Corp + * + * Author: Wang, Joseph C <joequant(a)gmail.com> + * Co-author: John Hsu <KCHSU0(a)nuvoton.com> + * This file is based on cht_bsw_rt5672.c and cht-bsw-max98090.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <linux/input.h> +#include "../atom/sst-atom-controls.h" +#include "../../codecs/nau8824.h" + +struct cht_mc_private { + struct snd_soc_jack jack; +}; + +static struct snd_soc_jack_pin cht_bsw_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"Ext Spk", NULL, "SPKOUTL"}, + {"Ext Spk", NULL, "SPKOUTR"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"MIC1", NULL, "Int Mic"}, + {"MIC2", NULL, "Int Mic"}, + {"HSMIC1", NULL, "Headset Mic"}, + {"HSMIC2", NULL, "Headset Mic"}, + {"Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8824_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + return ret; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, params_rate(params), + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + struct snd_soc_jack *jack = &ctx->jack; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + int ret, jack_type; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* NAU88L24 supports 4 butons headset detection + * KEY_MEDIA + * KEY_VOICECOMMAND + * KEY_VOLUMEUP + * KEY_VOLUMEDOWN + */ + jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, + cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); + if (ret) { + dev_err(runtime->dev, + "Headset Jack creation failed %d\n", ret); + return ret; + } + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + nau8824_enable_jack_detect(codec, jack); + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = + hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_none(fmt); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 48000); +} + +static const struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static const struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = NAU8824_CODEC_DAI, + .codec_name = "i2c-10508824:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "chtnau8824", + .owner = THIS_MODULE, + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + struct cht_mc_private *drv; + int ret_val; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-bsw-nau8824", + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Wang, Joseph C <joequant(a)gmail.com>"); +MODULE_AUTHOR("John Hsu <KCHSU0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-nau8824"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index b50a0d53846b..ad1eb2d644be 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -117,6 +117,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .sof_tplg_filename = "intel/reef-cht-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, + { + .id = "10508824", + .drv_name = "cht-bsw-nau8824", + .fw_filename = "intel/fw_sst_22a8.bin", + .board = "cht-bsw", + .sof_fw_filename = "intel/reef-cht.ri", + .sof_tplg_filename = "intel/reef-cht-nau8824.tplg", + .asoc_plat_name = "sst-mfld-platform", + }, { .id = "DLGS7212", .drv_name = "bytcht_da7213", -- 2.16.2

1 0

[alsa-devel] [RFC 0/7] ASoC: rt5651: Drop pdata, move over to device-properties
by Hans de Goede 01 Mar '18

01 Mar '18

Hi Mark, et al., Here is a second attempt to stop having duplicated DMI quirk entries between the rt5651 codec and machine drivers. Note this is only compile tested, I'm looking for some quick feedback if this seems to be going in the right direction, as I'm planning on rebuilding the rest of my jack-detect cleanup / fix series on top of this. The diffstat for this entire set is: Documentation/devicetree/bindings/sound/rt5651.txt | 6 include/sound/rt5651.h | 12 sound/soc/codecs/rt5651.c | 375 +++++++++------------ sound/soc/codecs/rt5651.h | 7 sound/soc/intel/boards/bytcr_rt5651.c | 46 ++ 5 files changed, 213 insertions(+), 233 deletions(-) So besides making the code cleaner (IMHO) it also makes it somewhat smaller and as an added bonus it fixes support the DMIC_EN quirk which was broken. Regards, Hans

2 9

[alsa-devel] [PATCH 0/7] Let Atmel use simple-audio-card
by Ladislav Michl 01 Mar '18

01 Mar '18

Alexandre, following patchset is sent as a bare minimum I need to use MAX9867 codec in master mode on AT91SAM9G20 based board. Codec's MCLK is driven by PCK0 and SSC is wired as follows: RK0 <-- BCLK RF0 <-- LRCLK TD0 --> SDIN RD0 <-- SDOUT Following DT framgment was used: i2c-gpio-0 { codec: max9867@18 { compatible = "maxim,max9867"; reg = <0x18>; #sound-dai-cells = <0>; }; }; sound { compatible = "simple-audio-card"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_pck0_as_mck>; simple-audio-card,name = "simear4-max9867"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&codec_dai>; simple-audio-card,frame-master = <&codec_dai>; simple-audio-card,routing = "LINE_IN", "Line In Jack", "Speaker", "HPOUT"; simple-audio-card,widgets = "Line", "Line In Jack", "Speaker", "Speaker"; cpu_dai: simple-audio-card,cpu { sound-dai = <&ssc0>; }; codec_dai: simple-audio-card,codec { sound-dai = <&codec>; clocks = <&pck0>; clock-frequency = <12384000>; }; }; &ssc0 { status = "okay"; pinctrl-0 = <&pinctrl_board_ssc0_tx &pinctrl_ssc0_rx>; atmel,clk-from-rk-pin; #sound-dai-cells = <0>; }; Codec code itself seems to be unused since it was merged and I'm wondering whenever it worked at all. A comment from codec author would help a lot, just to be sure I did not break anything. By no means this code is ready to be merged (only except those patches with commit log and after carefull consideration :)), I'm posting it as a base for discussion. Thank you, ladis Ladislav Michl (7): ASoC: atmel: Remove redundant dev_err() call in probe function ASoC: atmel_ssc_dai: Fix TCMR settings in I2S slave mode ASoC: simple_card_utils: Set clock frequency ASoC: max9867: Show Kconfig entry ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix BSEL value in master mode. ASoC: max9867: Take chip out of shutdown sound/soc/atmel/atmel-classd.c | 7 +-- sound/soc/atmel/atmel-pdmic.c | 7 +-- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/max9867.c | 110 +++------------------------------- sound/soc/generic/simple-card-utils.c | 4 ++ 6 files changed, 19 insertions(+), 114 deletions(-) -- 2.15.1

4 19

[alsa-devel] [PATCH - THREAD_SAFE_API 1/1] pcm: Do not access lock_enabled if thread safe API
by twischer＠de.adit-jv.com 01 Mar '18

01 Mar '18

From: Timo Wischer <twischer(a)de.adit-jv.com> is not enabled. Without this commit compiling fails when THREAD_SAFE_API is not enabled Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1be3833..d53ed98 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -823,8 +823,11 @@ int snd_pcm_async(snd_pcm_t *pcm, int sig, pid_t pid) sig = SIGIO; if (pid == 0) pid = getpid(); + +#ifdef THREAD_SAFE_API /* async handler may lead to a deadlock; suppose no multi thread */ pcm->lock_enabled = 0; +#endif return pcm->ops->async(pcm->op_arg, sig, pid); } #endif -- 2.7.4

2 1

[alsa-devel] [PATCH] sound/pci/ice1712: replace strcpy() with scnprintf()
by Joey Pabalinas 01 Mar '18

01 Mar '18

Replace unsafe uses of strcpy() to copy the name argument into the sid.name buffer with scnprintf() to guard against possible buffer overflows. Even though we don't actually care about the return value in this specific case, scnprintf() is still preferred over snprintf() due to scnprintf() returning the much more logical length of what was *actually* encoded into the destination array instead of returning length the resulting string *would* be, assuming it all fit into the destination array as snprintf() does. Maybe one day someone will use the return value, and since the behavior is exactly the same apart from the return value it would be better to account for that possibility and program defensively. Signed-off-by: Joey Pabalinas <joeypabalinas(a)gmail.com> 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 0dbaccf61f33270608..0abacc64168f895d53 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -26,6 +26,7 @@ #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/kernel.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/tlv.h> @@ -425,10 +426,9 @@ DECLARE_TLV_DB_SCALE(juli_master_db_scale, -6350, 50, 1); static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name) { - struct snd_ctl_elem_id sid; - memset(&sid, 0, sizeof(sid)); - /* FIXME: strcpy is bad. */ - strcpy(sid.name, name); + struct snd_ctl_elem_id sid = {0}; + + scnprintf(sid.name, sizeof(sid.name), "%s", name); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; return snd_ctl_find_id(card, &sid); } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d145b5eb7ff86d978d..332f6226548c6a089a 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -25,6 +25,7 @@ #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/kernel.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/tlv.h> @@ -785,10 +786,9 @@ DECLARE_TLV_DB_SCALE(qtet_master_db_scale, -6350, 50, 1); static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name) { - struct snd_ctl_elem_id sid; - memset(&sid, 0, sizeof(sid)); - /* FIXME: strcpy is bad. */ - strcpy(sid.name, name); + struct snd_ctl_elem_id sid = {0}; + + scnprintf(sid.name, sizeof(sid.name), "%s", name); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; return snd_ctl_find_id(card, &sid); } -- 2.16.2

3 6

[alsa-devel] [PATCH v1 0/4] ASoC: Add support for DAC PCM1789
by Mylène Josserand 01 Mar '18

01 Mar '18

Hello everyone, You will find in this series the support of Texas Instrument's DAC PCM1789. This DAC is very minimalist and is similar to PCM1792a except for some differences in registers. Series based on asoc tree, "for-next" branch (last commit 130c3888dfdc). It is important to notice that this DAC needs to always have clocks enabled (even without any data) otherwise it will be in a "desynchronized" state and can not send data correctly. This issue has been solved by performing a reset each time a sound is played (see patch 04). This reset can produce a "pop" noise. Thank you in advance for any review. Best regards, Mylène Mylène Josserand (4): ASoC: codecs: pcm179x: Add PCM1789 id ASoC: codecs: pcm179x: Add support for PCM1789 ASoC: codecs: pcm179x: Add reset gpio ASoC: codecs: pcm179x: Add trigger function to perform a reset .../devicetree/bindings/sound/pcm179x.txt | 2 +- sound/soc/codecs/pcm179x-i2c.c | 17 +- sound/soc/codecs/pcm179x.c | 242 ++++++++++++++++++++- sound/soc/codecs/pcm179x.h | 10 +- 4 files changed, 264 insertions(+), 7 deletions(-) -- 2.11.0

4 14

[alsa-devel] [PATCH v2] ASoC: atmel: Remove redundant dev_err() call in probe function
by Ladislav Michl 01 Mar '18

01 Mar '18

There is an error message within devm_ioremap_resource already, so remove the redundant dev_err() call. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Acked-by: Nicolas Ferre <nicolas.ferre(a)microchip.com> --- Changes: - v2: None, just added Ack. sound/soc/atmel/atmel-classd.c | 7 ++----- sound/soc/atmel/atmel-pdmic.c | 7 ++----- 2 files changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 9baaf6bc5fbd..3d70061901a0 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -583,11 +583,8 @@ static int atmel_classd_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); io_base = devm_ioremap_resource(dev, res); - if (IS_ERR(io_base)) { - ret = PTR_ERR(io_base); - dev_err(dev, "failed to remap register memory: %d\n", ret); - return ret; - } + if (IS_ERR(io_base)) + return PTR_ERR(io_base); dd->phy_base = res->start; dd->dev = dev; diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 94f852574a44..915c2b064de1 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -633,11 +633,8 @@ static int atmel_pdmic_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); io_base = devm_ioremap_resource(dev, res); - if (IS_ERR(io_base)) { - ret = PTR_ERR(io_base); - dev_err(dev, "failed to remap register memory: %d\n", ret); - return ret; - } + if (IS_ERR(io_base)) + return PTR_ERR(io_base); dd->phy_base = res->start; -- 2.16.2

2 1

[alsa-devel] [PATCH v4] ASoC: imx-wm8962: Remove machine driver
by Fabio Estevam 28 Feb '18

28 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> fsl-asoc-card machine driver also handles wm8962, so there is really no need for keeping the dedicated imx-wm8962 driver anymore. Remove the imx-wm8962 machine driver. Suggested-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v3: - Remove the imx-audio-wm8962.txt reference from fsl-asoc-card.txt. (Nicolin) .../devicetree/bindings/sound/fsl-asoc-card.txt | 1 - .../devicetree/bindings/sound/imx-audio-wm8962.txt | 53 ---- sound/soc/fsl/Kconfig | 11 - sound/soc/fsl/Makefile | 2 - sound/soc/fsl/imx-wm8962.c | 312 --------------------- 5 files changed, 379 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt delete mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index f749e27..c60a573 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -28,7 +28,6 @@ The compatible list for this generic sound card currently: (compatible with CS4271 and CS4272) "fsl,imx-audio-wm8962" - (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt) "fsl,imx-audio-sgtl5000" (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt) diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt deleted file mode 100644 index acea71b..0000000 --- a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt +++ /dev/null @@ -1,53 +0,0 @@ -Freescale i.MX audio complex with WM8962 codec - -Required properties: - - - compatible : "fsl,imx-audio-wm8962" - - - model : The user-visible name of this sound complex - - - ssi-controller : The phandle of the i.MX SSI controller - - - audio-codec : The phandle of the WM8962 audio codec - - - audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names could be power supplies, WM8962 - pins, and the jacks on the board: - - Power supplies: - * Mic Bias - - Board connectors: - * Mic Jack - * Headphone Jack - * Ext Spk - - - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) - - - mux-ext-port : The external port of the i.MX audio muxer - -Note: The AUDMUX port numbering should start at 1, which is consistent with -hardware manual. - -Example: - -sound { - compatible = "fsl,imx6q-sabresd-wm8962", - "fsl,imx-audio-wm8962"; - model = "wm8962-audio"; - ssi-controller = <&ssi2>; - audio-codec = <&codec>; - audio-routing = - "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "Ext Spk", "SPKOUTL", - "Ext Spk", "SPKOUTR", - "MICBIAS", "AMIC", - "IN3R", "MICBIAS", - "DMIC", "MICBIAS", - "DMICDAT", "DMIC"; - mux-int-port = <2>; - mux-ext-port = <3>; -}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37f9b62..6ec19fb 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -231,17 +231,6 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface -config SND_SOC_IMX_WM8962 - tristate "SoC Audio support for i.MX boards with wm8962" - depends on OF && I2C && INPUT - select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI - help - Say Y if you want to add support for SoC audio on an i.MX board with - a wm8962 codec. - config SND_SOC_IMX_ES8328 tristate "SoC Audio support for i.MX boards with the ES8328 codec" depends on OF && (I2C || SPI) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index c67bf11..de94fa0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -55,7 +55,6 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o -snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,6 +64,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o -obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c deleted file mode 100644 index 206b898..0000000 --- a/sound/soc/fsl/imx-wm8962.c +++ /dev/null @@ -1,312 +0,0 @@ -/* - * Copyright 2013 Freescale Semiconductor, Inc. - * - * Based on imx-sgtl5000.c - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ - -#include <linux/module.h> -#include <linux/of_platform.h> -#include <linux/i2c.h> -#include <linux/slab.h> -#include <linux/clk.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/soc-dapm.h> -#include <linux/pinctrl/consumer.h> - -#include "../codecs/wm8962.h" -#include "imx-audmux.h" - -#define DAI_NAME_SIZE 32 - -struct imx_wm8962_data { - struct snd_soc_dai_link dai; - struct snd_soc_card card; - char codec_dai_name[DAI_NAME_SIZE]; - char platform_name[DAI_NAME_SIZE]; - unsigned int clk_frequency; -}; - -struct imx_priv { - struct platform_device *pdev; - int sample_rate; - snd_pcm_format_t sample_format; -}; - -static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static int imx_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card); - - priv->sample_rate = params_rate(params); - priv->sample_format = params_format(params); - - return 0; -} - -static const struct snd_soc_ops imx_hifi_ops = { - .hw_params = imx_hifi_hw_params, -}; - -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int imx_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - data->clk_frequency, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(dev, "failed to set sysclk in %s\n", __func__); - - return ret; -} - -static int imx_wm8962_probe(struct platform_device *pdev) -{ - struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; - struct platform_device *ssi_pdev; - struct i2c_client *codec_dev; - struct imx_wm8962_data *data; - struct imx_priv *priv; - struct clk *codec_clk; - int int_port, ext_port; - int ret; - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->pdev = pdev; - priv->sample_rate = 44100; - priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; - - ret = of_property_read_u32(np, "mux-int-port", &int_port); - if (ret) { - dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); - return ret; - } - ret = of_property_read_u32(np, "mux-ext-port", &ext_port); - if (ret) { - dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); - return ret; - } - - /* - * The port numbering in the hardware manual starts at 1, while - * the audmux API expects it starts at 0. - */ - int_port--; - ext_port--; - ret = imx_audmux_v2_configure_port(int_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TCLKDIR, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(&pdev->dev, "audmux internal port setup failed\n"); - return ret; - } - ret = imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(&pdev->dev, "audmux external port setup failed\n"); - return ret; - } - - ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); - codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); - if (!ssi_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); - ret = -EINVAL; - goto fail; - } - - ssi_pdev = of_find_device_by_node(ssi_np); - if (!ssi_pdev) { - dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; - goto fail; - } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev || !codec_dev->dev.driver) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } - - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto fail; - } - - codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(codec_clk)) { - ret = PTR_ERR(codec_clk); - dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); - goto fail; - } - - data->clk_frequency = clk_get_rate(codec_clk); - clk_put(codec_clk); - - data->dai.name = "HiFi"; - data->dai.stream_name = "HiFi"; - data->dai.codec_dai_name = "wm8962"; - data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); - data->dai.platform_of_node = ssi_np; - data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - data->card.dev = &pdev->dev; - ret = snd_soc_of_parse_card_name(&data->card, "model"); - if (ret) - goto fail; - ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); - if (ret) - goto fail; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - data->card.dai_link = &data->dai; - data->card.dapm_widgets = imx_wm8962_dapm_widgets; - data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); - - data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; - - platform_set_drvdata(pdev, &data->card); - snd_soc_card_set_drvdata(&data->card, data); - - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto fail; - } - -fail: - of_node_put(ssi_np); - of_node_put(codec_np); - - return ret; -} - -static const struct of_device_id imx_wm8962_dt_ids[] = { - { .compatible = "fsl,imx-audio-wm8962", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); - -static struct platform_driver imx_wm8962_driver = { - .driver = { - .name = "imx-wm8962", - .pm = &snd_soc_pm_ops, - .of_match_table = imx_wm8962_dt_ids, - }, - .probe = imx_wm8962_probe, -}; -module_platform_driver(imx_wm8962_driver); - -MODULE_AUTHOR("Freescale Semiconductor, Inc."); -MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:imx-wm8962"); -- 2.7.4

4 3

[alsa-devel] [PATCH v2 0/2] ASoC: add support for ROHM BD28623 codec
by Katsuhiro Suzuki 28 Feb '18

28 Feb '18

This patch adds support for ROHM BD28623MUV class D speaker amplifier codec driver. This driver only refers information of HW specification document that can be derivered at website of ROHM. http://www.rohm.com/web/global/products/-/product/BD28623MUV --- Changes in v2: - Change gpio properties to optional - Change VCCA, VCCP-supply to mandatory - Fix lisence comment style - Use _cansleep() version for manipulating GPIO - Use ASoC component remove() function instead of platform remove() function to power off this IC - Remove update function that was just alias of gpiod_set_value Katsuhiro Suzuki (2): ASoC: add DT bindings documentation for ROHM BD28623 codec ASoC: support ROHM BD28623 codec .../devicetree/bindings/sound/rohm,bd28623.txt | 29 +++ sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/bd28623.c | 242 +++++++++++++++++++++ 4 files changed, 281 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 sound/soc/codecs/bd28623.c -- 2.16.1

2 3

[alsa-devel] Applied "ASoC: support ROHM BD28623 codec" to the asoc tree
by Mark Brown 28 Feb '18

28 Feb '18

The patch ASoC: support ROHM BD28623 codec has been applied to the asoc tree at https://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 f6d4b052175d84c6f2b73e884f3ab8c83c5f4d45 Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> Date: Fri, 23 Feb 2018 10:17:17 +0900 Subject: [PATCH] ASoC: support ROHM BD28623 codec This patch adds support of the ROHM BD28623MUV Class D speaker amplifier for Flat-panel TVs. This IC delivers an output power of 20W + 20W. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 8 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/bd28623.c | 242 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 252 insertions(+) create mode 100644 sound/soc/codecs/bd28623.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2b331f7266ab..4545f8fde220 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -45,6 +45,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_BT_SCO + select SND_SOC_BD28623 select SND_SOC_CQ0093VC select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C @@ -405,6 +406,13 @@ config SND_SOC_ALC5623 config SND_SOC_ALC5632 tristate +config SND_SOC_BD28623 + tristate "ROHM BD28623 CODEC" + help + Enable support for ROHM BD28623MUV Class D speaker amplifier. + This codec does not have any control buses such as I2C, it + detect format of I2S automatically. + config SND_SOC_BT_SCO tristate "Dummy BT SCO codec driver" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index da1571336f1e..6df2fb32fc7e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -35,6 +35,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o +snd-soc-bd28623-objs := bd28623.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o @@ -280,6 +281,7 @@ obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o +obj-$(CONFIG_SND_SOC_BD28623) += snd-soc-bd28623.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o diff --git a/sound/soc/codecs/bd28623.c b/sound/soc/codecs/bd28623.c new file mode 100644 index 000000000000..31904ef5c88b --- /dev/null +++ b/sound/soc/codecs/bd28623.c @@ -0,0 +1,242 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ROHM BD28623MUV class D speaker amplifier codec driver. +// +// Copyright (c) 2018 Socionext Inc. + +#include <linux/delay.h> +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#define BD28623_NUM_SUPPLIES 3 + +static const char *const bd28623_supply_names[BD28623_NUM_SUPPLIES] = { + "VCCA", + "VCCP1", + "VCCP2", +}; + +struct bd28623_priv { + struct device *dev; + struct regulator_bulk_data supplies[BD28623_NUM_SUPPLIES]; + struct gpio_desc *reset_gpio; + struct gpio_desc *mute_gpio; + + int switch_spk; +}; + +static const struct snd_soc_dapm_widget bd28623_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT1P"), + SND_SOC_DAPM_OUTPUT("OUT1N"), + SND_SOC_DAPM_OUTPUT("OUT2P"), + SND_SOC_DAPM_OUTPUT("OUT2N"), +}; + +static const struct snd_soc_dapm_route bd28623_routes[] = { + { "OUT1P", NULL, "DAC" }, + { "OUT1N", NULL, "DAC" }, + { "OUT2P", NULL, "DAC" }, + { "OUT2N", NULL, "DAC" }, +}; + +static int bd28623_power_on(struct bd28623_priv *bd) +{ + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(bd->supplies), bd->supplies); + if (ret) { + dev_err(bd->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + gpiod_set_value_cansleep(bd->reset_gpio, 0); + usleep_range(300000, 400000); + + return 0; +} + +static void bd28623_power_off(struct bd28623_priv *bd) +{ + gpiod_set_value_cansleep(bd->reset_gpio, 1); + + regulator_bulk_disable(ARRAY_SIZE(bd->supplies), bd->supplies); +} + +static int bd28623_get_switch_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = bd->switch_spk; + + return 0; +} + +static int bd28623_set_switch_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + + if (bd->switch_spk == ucontrol->value.integer.value[0]) + return 0; + + bd->switch_spk = ucontrol->value.integer.value[0]; + + gpiod_set_value_cansleep(bd->mute_gpio, bd->switch_spk ? 0 : 1); + + return 0; +} + +static const struct snd_kcontrol_new bd28623_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Switch", 0, + bd28623_get_switch_spk, bd28623_set_switch_spk), +}; + +static int bd28623_codec_probe(struct snd_soc_component *component) +{ + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + int ret; + + bd->switch_spk = 1; + + ret = bd28623_power_on(bd); + if (ret) + return ret; + + gpiod_set_value_cansleep(bd->mute_gpio, bd->switch_spk ? 0 : 1); + + return 0; +} + +static void bd28623_codec_remove(struct snd_soc_component *component) +{ + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + + bd28623_power_off(bd); +} + +static int bd28623_codec_suspend(struct snd_soc_component *component) +{ + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + + bd28623_power_off(bd); + + return 0; +} + +static int bd28623_codec_resume(struct snd_soc_component *component) +{ + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + int ret; + + ret = bd28623_power_on(bd); + if (ret) + return ret; + + gpiod_set_value_cansleep(bd->mute_gpio, bd->switch_spk ? 0 : 1); + + return 0; +} + +static const struct snd_soc_component_driver soc_codec_bd = { + .probe = bd28623_codec_probe, + .remove = bd28623_codec_remove, + .suspend = bd28623_codec_suspend, + .resume = bd28623_codec_resume, + .dapm_widgets = bd28623_widgets, + .num_dapm_widgets = ARRAY_SIZE(bd28623_widgets), + .dapm_routes = bd28623_routes, + .num_dapm_routes = ARRAY_SIZE(bd28623_routes), + .controls = bd28623_controls, + .num_controls = ARRAY_SIZE(bd28623_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct snd_soc_dai_driver soc_dai_bd = { + .name = "bd28623-speaker", + .playback = { + .stream_name = "Playback", + .formats = SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_32000, + .channels_min = 2, + .channels_max = 2, + }, +}; + +static int bd28623_probe(struct platform_device *pdev) +{ + struct bd28623_priv *bd; + struct device *dev = &pdev->dev; + int i, ret; + + bd = devm_kzalloc(&pdev->dev, sizeof(struct bd28623_priv), GFP_KERNEL); + if (!bd) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(bd->supplies); i++) + bd->supplies[i].supply = bd28623_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(bd->supplies), + bd->supplies); + if (ret) { + dev_err(dev, "Failed to get supplies: %d\n", ret); + return ret; + } + + bd->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(bd->reset_gpio)) { + dev_err(dev, "Failed to request reset_gpio: %ld\n", + PTR_ERR(bd->reset_gpio)); + return PTR_ERR(bd->reset_gpio); + } + + bd->mute_gpio = devm_gpiod_get_optional(dev, "mute", + GPIOD_OUT_HIGH); + if (IS_ERR(bd->mute_gpio)) { + dev_err(dev, "Failed to request mute_gpio: %ld\n", + PTR_ERR(bd->mute_gpio)); + return PTR_ERR(bd->mute_gpio); + } + + platform_set_drvdata(pdev, bd); + bd->dev = dev; + + return devm_snd_soc_register_component(dev, &soc_codec_bd, + &soc_dai_bd, 1); +} + +static const struct of_device_id bd28623_of_match[] = { + { .compatible = "rohm,bd28623", }, + {} +}; +MODULE_DEVICE_TABLE(of, bd28623_of_match); + +static struct platform_driver bd28623_codec_driver = { + .driver = { + .name = "bd28623", + .of_match_table = of_match_ptr(bd28623_of_match), + }, + .probe = bd28623_probe, +}; +module_platform_driver(bd28623_codec_driver); + +MODULE_AUTHOR("Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com>"); +MODULE_DESCRIPTION("ROHM BD28623 speaker amplifier driver"); +MODULE_LICENSE("GPL v2"); -- 2.16.1

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: For TLV controls only register TLV get/set
by Richard Fitzgerald 28 Feb '18

28 Feb '18

Normal 512-byte get/set of a TLV isn't supported but we were registering the normal get/set anyway and relying on omitting the SNDRV_CTL_ELEM_ACCESS_[READ|WRITE] flags to prevent them being called. Trouble is if this gets broken in the core ALSA code - as it has been since at least 4.14 - the standard get/set can be called unexpectedly and corrupt memory. There's no point providing functions that won't be called and it's a trivial change. The benefit is that if the ALSA core gets broken again we get a big fat immediate NULL dereference instead of a memory corruption timebomb. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9b27390cf4f1..82b0927e6ed7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1204,12 +1204,14 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->put = wm_coeff_put_acked; break; default: - kcontrol->get = wm_coeff_get; - kcontrol->put = wm_coeff_put; - - ctl->bytes_ext.max = ctl->len; - ctl->bytes_ext.get = wm_coeff_tlv_get; - ctl->bytes_ext.put = wm_coeff_tlv_put; + if (kcontrol->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + } else { + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + } break; } -- 2.11.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: x86: Fix missing spinlock and mutex initializations
by Takashi Iwai 28 Feb '18

28 Feb '18

The commit change for supporting the multiple ports moved involved some code shuffling, and there the initializations of spinlock and mutex in snd_intelhad object were dropped mistakenly. This patch adds the missing initializations again for each port. Fixes: b4eb0d522fcb ("ALSA: x86: Split snd_intelhad into card and PCM specific structures") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/x86/intel_hdmi_audio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 96115c401292..cec62f9b7085 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1835,6 +1835,8 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) ctx->port = single_port ? -1 : port; ctx->pipe = -1; + spin_lock_init(&ctx->had_spinlock); + mutex_init(&ctx->mutex); INIT_WORK(&ctx->hdmi_audio_wq, had_audio_wq); ret = snd_pcm_new(card, INTEL_HAD, port, MAX_PB_STREAMS, -- 2.16.2

1 1

[alsa-devel] [PATCH] ALSA: control: Fix a bunch of whitespace errors
by Richard Fitzgerald 28 Feb '18

28 Feb '18

Remove a bunch of trailing whitespace errors. They are fairly annoying if you have your editor set to strip trailing whitespace because you find you've introduced more changes than you were trying to make. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/core/control.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 8a77620a3854..69734b0eafd0 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -105,7 +105,7 @@ static void snd_ctl_empty_read_queue(struct snd_ctl_file * ctl) { unsigned long flags; struct snd_kctl_event *cread; - + spin_lock_irqsave(&ctl->read_lock, flags); while (!list_empty(&ctl->events)) { cread = snd_kctl_event(ctl->events.next); @@ -159,7 +159,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, unsigned long flags; struct snd_ctl_file *ctl; struct snd_kctl_event *ev; - + if (snd_BUG_ON(!card || !id)) return; if (card->shutdown) @@ -213,7 +213,7 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, { unsigned int size; unsigned int idx; - + if (count == 0 || count > MAX_CONTROL_COUNT) return -EINVAL; @@ -238,7 +238,7 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, * @ncontrol: the initialization record * @private_data: the private data to set * - * Allocates a new struct snd_kcontrol instance and initialize from the given + * Allocates a new struct snd_kcontrol instance and initialize from the given * template. When the access field of ncontrol is 0, it's assumed as * READWRITE access. When the count field is 0, it's assumes as one. * @@ -251,7 +251,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, unsigned int count; unsigned int access; int err; - + if (snd_BUG_ON(!ncontrol || !ncontrol->info)) return NULL; @@ -753,7 +753,7 @@ static int snd_ctl_elem_list(struct snd_card *card, struct snd_ctl_elem_id id; unsigned int offset, space, jidx; int err = 0; - + if (copy_from_user(&list, _list, sizeof(list))) return -EFAULT; offset = list.offset; @@ -827,7 +827,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, struct snd_kcontrol_volatile *vd; unsigned int index_offset; int result; - + down_read(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &info->id); if (kctl == NULL) { @@ -992,7 +992,7 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; int result; - + if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; down_write(&card->controls_rwsem); @@ -1020,7 +1020,7 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; int result; - + if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; down_write(&card->controls_rwsem); -- 2.11.0

2 1

[alsa-devel] [PATCH v1 0/9] Enable HDA Codec support on Intel Platforms (Series2)
by Rakesh Ughreja 27 Feb '18

27 Feb '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. This patch carry forwards the work that is done in the previous series to enable HD Audio codecs on such platforms. This patch series adds ASoC HDA codec driver for Intel platforms. It is written by reusing the legacy HDA ALSA codec driver. Intention is to maximize the reuse and minimize the changes in the legacy HDA codec driver. Rakesh Ughreja (9): ASoC: Intel: Boards: Machine driver for Intel platforms ASoC: Intel: Skylake: Add entry in sst_acpi_mach for HDA codecs ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ALSA: hda - split snd_hda_codec_new function ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init ALSA: hdac: add extended ops in the hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers ASoC: Intel: Boards: add support for HDA codecs include/sound/hdaudio.h | 9 + include/sound/hdaudio_ext.h | 6 +- sound/hda/ext/hdac_ext_bus.c | 12 +- sound/pci/hda/hda_bind.c | 6 + sound/pci/hda/hda_codec.c | 67 +++- sound/pci/hda/hda_codec.h | 2 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 448 +++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 23 ++ sound/soc/intel/Kconfig | 1 + sound/soc/intel/boards/Kconfig | 10 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 131 ++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 34 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 143 +++++++++ sound/soc/intel/skylake/skl-pcm.c | 50 ++- sound/soc/intel/skylake/skl.c | 98 +++++- sound/soc/intel/skylake/skl.h | 11 +- 19 files changed, 1011 insertions(+), 49 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c -- 2.7.4

4 44

[alsa-devel] [PATCH] ALSA: control: Fix memory corruption risk in snd_ctl_elem_read
by Richard Fitzgerald 27 Feb '18

27 Feb '18

The patch "ALSA: control: code refactoring for ELEM_READ/ELEM_WRITE operations" introduced a potential for kernel memory corruption due to an incorrect if statement allowing non-readable controls to fall through and call the get function. For TLV controls a driver can omit SNDRV_CTL_ELEM_ACCESS_READ to ensure that only the TLV get function can be called. Instead the normal get() can be invoked unexpectedly and as the driver expects that this will only be called for controls <= 512 bytes, potentially try to copy >512 bytes into the 512 byte return array, so corrupting kernel memory. The problem is an attempt to refactor the snd_ctl_elem_read function to invert the logic so that it conditionally aborted if the control is unreadable instead of conditionally executing. But the if statement wasn't inverted correctly. The correct inversion of if (a && !b) is if (!a || b) Fixes: becf9e5d553c2 ("ALSA: control: code refactoring for ELEM_READ/ELEM_WRITE operations") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 0b3026d937b1..8a77620a3854 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -889,7 +889,7 @@ static int snd_ctl_elem_read(struct snd_card *card, index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; - if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) && kctl->get == NULL) + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL) return -EPERM; snd_ctl_build_ioff(&control->id, kctl, index_offset); -- 2.11.0

2 1

[alsa-devel] aplay\arecord compare utility
by Muni Sekhar 27 Feb '18

27 Feb '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, I’m using an alsa utilities(aplay & arecord) for sound loopback testing. From Linux host , sending & receiving the wav file. At end I want to compare the sent & received wav file. Does Linux has any utility to support this one? -- Thanks, Sekhar

2 3

[alsa-devel] [PATCH] ALSA: Use scnprintf() instead of snprintf()
by Jaejoong Kim 27 Feb '18

27 Feb '18

The show() method should use scnprintf() not snprintf() because snprintf() may returns a value that exceeds its second argument. Signed-off-by: Jaejoong Kim <climbbb.kim(a)gmail.com> --- sound/core/init.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 168ae03..4a51a06 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -670,7 +670,7 @@ card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); - return snprintf(buf, PAGE_SIZE, "%s\n", card->id); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->id); } static ssize_t @@ -710,7 +710,7 @@ card_number_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); - return snprintf(buf, PAGE_SIZE, "%i\n", card->number); + return scnprintf(buf, PAGE_SIZE, "%i\n", card->number); } static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); -- 2.7.4

3 4

[alsa-devel] No sound on MacbookPro 13,3 (A1707)
by Freaky 27 Feb '18

27 Feb '18

Hi there, we're trying to get sound working on a MacbookPro 13,3 (A1707). So far I've been unable to get the sound working. Did use the HDA analyzer and flipped the out-dir, enable and data on all 8 GPIO's, but none of them got the sound working on either the onboard speakers nor the jack output for headphone. Is there anyone willing to help on this issue? Driver development is a bit beyond my skills. I can provide hands, debug logs, etc. tho' :). Currently trying with a default Ubuntu 17.10.1 live USB, but a 4.15.3 kernel didn't help either. Alsa-info has been uploaded to http://www.alsa-project.org/db/?f=33cc432d3b648d1aa6adc997630817046656a086 The system was booted with an external USB-C dock from Dell, otherwise there's no network and no regular (USB-A) ports available (booting from stick) as the WiFi doesn't work and the device has only USB-C ports. The card 0 is from the dock - it's not relevant (and working). Before alsa-info ran there was also a USB headphone plugged in, that was card 1. Without those devices card 2 = 0 and 3 = 1 at boot. Issues are with what's listed as card 2 currently, the HDA-Intel - HDA Intel PCH. Any help is greatly appreciated. Some people already have some trackers for the current state of the devices. Not sure if they are of use, but might help: https://github.com/Dunedan/mbp-2016-linux/ https://gist.github.com/roadrunner2/1289542a748d9a104e7baec6a92f9cd7 Kind regards,

2 1

[alsa-devel] [PATCH v3] ASoC: imx-wm8962: Remove machine driver
by Fabio Estevam 27 Feb '18

27 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> fsl-asoc-card machine driver also handles wm8962, so there is really no need for keeping the dedicated imx-wm8962 driver anymore. Remove the imx-wm8962 machine driver. Suggested-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v2: - Also remove the dt-bindings .../devicetree/bindings/sound/imx-audio-wm8962.txt | 53 ---- sound/soc/fsl/Kconfig | 11 - sound/soc/fsl/Makefile | 2 - sound/soc/fsl/imx-wm8962.c | 312 --------------------- 4 files changed, 378 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt delete mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt deleted file mode 100644 index acea71b..0000000 --- a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt +++ /dev/null @@ -1,53 +0,0 @@ -Freescale i.MX audio complex with WM8962 codec - -Required properties: - - - compatible : "fsl,imx-audio-wm8962" - - - model : The user-visible name of this sound complex - - - ssi-controller : The phandle of the i.MX SSI controller - - - audio-codec : The phandle of the WM8962 audio codec - - - audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names could be power supplies, WM8962 - pins, and the jacks on the board: - - Power supplies: - * Mic Bias - - Board connectors: - * Mic Jack - * Headphone Jack - * Ext Spk - - - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) - - - mux-ext-port : The external port of the i.MX audio muxer - -Note: The AUDMUX port numbering should start at 1, which is consistent with -hardware manual. - -Example: - -sound { - compatible = "fsl,imx6q-sabresd-wm8962", - "fsl,imx-audio-wm8962"; - model = "wm8962-audio"; - ssi-controller = <&ssi2>; - audio-codec = <&codec>; - audio-routing = - "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "Ext Spk", "SPKOUTL", - "Ext Spk", "SPKOUTR", - "MICBIAS", "AMIC", - "IN3R", "MICBIAS", - "DMIC", "MICBIAS", - "DMICDAT", "DMIC"; - mux-int-port = <2>; - mux-ext-port = <3>; -}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37f9b62..6ec19fb 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -231,17 +231,6 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface -config SND_SOC_IMX_WM8962 - tristate "SoC Audio support for i.MX boards with wm8962" - depends on OF && I2C && INPUT - select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI - help - Say Y if you want to add support for SoC audio on an i.MX board with - a wm8962 codec. - config SND_SOC_IMX_ES8328 tristate "SoC Audio support for i.MX boards with the ES8328 codec" depends on OF && (I2C || SPI) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index c67bf11..de94fa0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -55,7 +55,6 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o -snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,6 +64,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o -obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c deleted file mode 100644 index 206b898..0000000 --- a/sound/soc/fsl/imx-wm8962.c +++ /dev/null @@ -1,312 +0,0 @@ -/* - * Copyright 2013 Freescale Semiconductor, Inc. - * - * Based on imx-sgtl5000.c - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ - -#include <linux/module.h> -#include <linux/of_platform.h> -#include <linux/i2c.h> -#include <linux/slab.h> -#include <linux/clk.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/soc-dapm.h> -#include <linux/pinctrl/consumer.h> - -#include "../codecs/wm8962.h" -#include "imx-audmux.h" - -#define DAI_NAME_SIZE 32 - -struct imx_wm8962_data { - struct snd_soc_dai_link dai; - struct snd_soc_card card; - char codec_dai_name[DAI_NAME_SIZE]; - char platform_name[DAI_NAME_SIZE]; - unsigned int clk_frequency; -}; - -struct imx_priv { - struct platform_device *pdev; - int sample_rate; - snd_pcm_format_t sample_format; -}; - -static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static int imx_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card); - - priv->sample_rate = params_rate(params); - priv->sample_format = params_format(params); - - return 0; -} - -static const struct snd_soc_ops imx_hifi_ops = { - .hw_params = imx_hifi_hw_params, -}; - -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int imx_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - data->clk_frequency, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(dev, "failed to set sysclk in %s\n", __func__); - - return ret; -} - -static int imx_wm8962_probe(struct platform_device *pdev) -{ - struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; - struct platform_device *ssi_pdev; - struct i2c_client *codec_dev; - struct imx_wm8962_data *data; - struct imx_priv *priv; - struct clk *codec_clk; - int int_port, ext_port; - int ret; - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->pdev = pdev; - priv->sample_rate = 44100; - priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; - - ret = of_property_read_u32(np, "mux-int-port", &int_port); - if (ret) { - dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); - return ret; - } - ret = of_property_read_u32(np, "mux-ext-port", &ext_port); - if (ret) { - dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); - return ret; - } - - /* - * The port numbering in the hardware manual starts at 1, while - * the audmux API expects it starts at 0. - */ - int_port--; - ext_port--; - ret = imx_audmux_v2_configure_port(int_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TCLKDIR, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(&pdev->dev, "audmux internal port setup failed\n"); - return ret; - } - ret = imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(&pdev->dev, "audmux external port setup failed\n"); - return ret; - } - - ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); - codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); - if (!ssi_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); - ret = -EINVAL; - goto fail; - } - - ssi_pdev = of_find_device_by_node(ssi_np); - if (!ssi_pdev) { - dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; - goto fail; - } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev || !codec_dev->dev.driver) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } - - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto fail; - } - - codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(codec_clk)) { - ret = PTR_ERR(codec_clk); - dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); - goto fail; - } - - data->clk_frequency = clk_get_rate(codec_clk); - clk_put(codec_clk); - - data->dai.name = "HiFi"; - data->dai.stream_name = "HiFi"; - data->dai.codec_dai_name = "wm8962"; - data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); - data->dai.platform_of_node = ssi_np; - data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - data->card.dev = &pdev->dev; - ret = snd_soc_of_parse_card_name(&data->card, "model"); - if (ret) - goto fail; - ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); - if (ret) - goto fail; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - data->card.dai_link = &data->dai; - data->card.dapm_widgets = imx_wm8962_dapm_widgets; - data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); - - data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; - - platform_set_drvdata(pdev, &data->card); - snd_soc_card_set_drvdata(&data->card, data); - - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto fail; - } - -fail: - of_node_put(ssi_np); - of_node_put(codec_np); - - return ret; -} - -static const struct of_device_id imx_wm8962_dt_ids[] = { - { .compatible = "fsl,imx-audio-wm8962", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); - -static struct platform_driver imx_wm8962_driver = { - .driver = { - .name = "imx-wm8962", - .pm = &snd_soc_pm_ops, - .of_match_table = imx_wm8962_dt_ids, - }, - .probe = imx_wm8962_probe, -}; -module_platform_driver(imx_wm8962_driver); - -MODULE_AUTHOR("Freescale Semiconductor, Inc."); -MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:imx-wm8962"); -- 2.7.4

2 2

[alsa-devel] Applied "ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling"" to the asoc tree
by Mark Brown 26 Feb '18

26 Feb '18

The patch ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling" has been applied to the asoc tree at https://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 b84acf44d5b86597cebcb6aca16e7dd9ca2b0294 Mon Sep 17 00:00:00 2001 From: Fabio Estevam <fabio.estevam(a)nxp.com> Date: Mon, 26 Feb 2018 15:55:25 -0300 Subject: [PATCH] ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling" This reverts commit f91b1e73ccde71d4bc69ae10d475196df38844ab. As explained by Lars-Peter Clausen: "This creates a resource leak. dmaengine_pcm_request_chan_of() requests bothtransmit and receive channels. It might return with an error if one of them fails, but the other one succeeded. In this case we need to call dmaengine_pcm_release_chan() to free the requested channel" So revert the commit. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 567fbdfd1ca9..56a541b9ff9e 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -504,7 +504,7 @@ int snd_dmaengine_pcm_register(struct device *dev, ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) - goto err_free_pcm; + goto err_free_dma; if (config && config->process) ret = snd_soc_add_component(dev, &pcm->component, @@ -520,7 +520,6 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); -err_free_pcm: kfree(pcm); return ret; } -- 2.16.1

1 0

[alsa-devel] [PATCH v2] ASoC: imx-wm8962: Remove machine driver
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> fsl-asoc-card machine driver also handles wm8962, so there is really no need for keeping the dedicated imx-wm8962 driver anymore. Remove the imx-wm8962 machine driver. Suggested-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Instead of fixing the imx-wm8962 driver private strcuc, better remove it as wm8962 can be handled by fsl-asoc-card. (Nicolin) sound/soc/fsl/Kconfig | 11 -- sound/soc/fsl/Makefile | 2 - sound/soc/fsl/imx-wm8962.c | 312 --------------------------------------------- 3 files changed, 325 deletions(-) delete mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37f9b62..6ec19fb 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -231,17 +231,6 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface -config SND_SOC_IMX_WM8962 - tristate "SoC Audio support for i.MX boards with wm8962" - depends on OF && I2C && INPUT - select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI - help - Say Y if you want to add support for SoC audio on an i.MX board with - a wm8962 codec. - config SND_SOC_IMX_ES8328 tristate "SoC Audio support for i.MX boards with the ES8328 codec" depends on OF && (I2C || SPI) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index c67bf11..de94fa0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -55,7 +55,6 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o -snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,6 +64,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o -obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c deleted file mode 100644 index 206b898..0000000 --- a/sound/soc/fsl/imx-wm8962.c +++ /dev/null @@ -1,312 +0,0 @@ -/* - * Copyright 2013 Freescale Semiconductor, Inc. - * - * Based on imx-sgtl5000.c - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ - -#include <linux/module.h> -#include <linux/of_platform.h> -#include <linux/i2c.h> -#include <linux/slab.h> -#include <linux/clk.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/soc-dapm.h> -#include <linux/pinctrl/consumer.h> - -#include "../codecs/wm8962.h" -#include "imx-audmux.h" - -#define DAI_NAME_SIZE 32 - -struct imx_wm8962_data { - struct snd_soc_dai_link dai; - struct snd_soc_card card; - char codec_dai_name[DAI_NAME_SIZE]; - char platform_name[DAI_NAME_SIZE]; - unsigned int clk_frequency; -}; - -struct imx_priv { - struct platform_device *pdev; - int sample_rate; - snd_pcm_format_t sample_format; -}; - -static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static int imx_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card); - - priv->sample_rate = params_rate(params); - priv->sample_format = params_format(params); - - return 0; -} - -static const struct snd_soc_ops imx_hifi_ops = { - .hw_params = imx_hifi_hw_params, -}; - -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int imx_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - data->clk_frequency, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(dev, "failed to set sysclk in %s\n", __func__); - - return ret; -} - -static int imx_wm8962_probe(struct platform_device *pdev) -{ - struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; - struct platform_device *ssi_pdev; - struct i2c_client *codec_dev; - struct imx_wm8962_data *data; - struct imx_priv *priv; - struct clk *codec_clk; - int int_port, ext_port; - int ret; - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->pdev = pdev; - priv->sample_rate = 44100; - priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; - - ret = of_property_read_u32(np, "mux-int-port", &int_port); - if (ret) { - dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); - return ret; - } - ret = of_property_read_u32(np, "mux-ext-port", &ext_port); - if (ret) { - dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); - return ret; - } - - /* - * The port numbering in the hardware manual starts at 1, while - * the audmux API expects it starts at 0. - */ - int_port--; - ext_port--; - ret = imx_audmux_v2_configure_port(int_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TCLKDIR, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(&pdev->dev, "audmux internal port setup failed\n"); - return ret; - } - ret = imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(&pdev->dev, "audmux external port setup failed\n"); - return ret; - } - - ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); - codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); - if (!ssi_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); - ret = -EINVAL; - goto fail; - } - - ssi_pdev = of_find_device_by_node(ssi_np); - if (!ssi_pdev) { - dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; - goto fail; - } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev || !codec_dev->dev.driver) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } - - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto fail; - } - - codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(codec_clk)) { - ret = PTR_ERR(codec_clk); - dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); - goto fail; - } - - data->clk_frequency = clk_get_rate(codec_clk); - clk_put(codec_clk); - - data->dai.name = "HiFi"; - data->dai.stream_name = "HiFi"; - data->dai.codec_dai_name = "wm8962"; - data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); - data->dai.platform_of_node = ssi_np; - data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - data->card.dev = &pdev->dev; - ret = snd_soc_of_parse_card_name(&data->card, "model"); - if (ret) - goto fail; - ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); - if (ret) - goto fail; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - data->card.dai_link = &data->dai; - data->card.dapm_widgets = imx_wm8962_dapm_widgets; - data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); - - data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; - - platform_set_drvdata(pdev, &data->card); - snd_soc_card_set_drvdata(&data->card, data); - - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto fail; - } - -fail: - of_node_put(ssi_np); - of_node_put(codec_np); - - return ret; -} - -static const struct of_device_id imx_wm8962_dt_ids[] = { - { .compatible = "fsl,imx-audio-wm8962", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); - -static struct platform_driver imx_wm8962_driver = { - .driver = { - .name = "imx-wm8962", - .pm = &snd_soc_pm_ops, - .of_match_table = imx_wm8962_dt_ids, - }, - .probe = imx_wm8962_probe, -}; -module_platform_driver(imx_wm8962_driver); - -MODULE_AUTHOR("Freescale Semiconductor, Inc."); -MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:imx-wm8962"); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: imx-wm8962: Use a single private structure
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 8f7206d69ab8c ("ASoC: imx-wm8962: Remove global variables") ended up by assigning the same value for two different structs: struct imx_priv *priv = snd_soc_card_get_drvdata(card); struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); Fix it by consolidating all the private data into a single structure. Fixes: 8f7206d69ab8c ("ASoC: imx-wm8962: Remove global variables") Reported-by: Leonard Crestez <leonard.crestez(a)nxp.com> Suggested-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/imx-wm8962.c | 64 +++++++++++++++++++--------------------------- 1 file changed, 26 insertions(+), 38 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 206b898..29d7e9b 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -28,15 +28,12 @@ #define DAI_NAME_SIZE 32 -struct imx_wm8962_data { +struct imx_priv { struct snd_soc_dai_link dai; struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; unsigned int clk_frequency; -}; - -struct imx_priv { struct platform_device *pdev; int sample_rate; snd_pcm_format_t sample_format; @@ -72,7 +69,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; @@ -91,7 +87,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, pll_out = priv->sample_rate * 256; ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, + WM8962_FLL_MCLK, priv->clk_frequency, pll_out); if (ret < 0) { dev_err(dev, "failed to start FLL: %d\n", ret); @@ -111,7 +107,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, case SND_SOC_BIAS_STANDBY: if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, + WM8962_SYSCLK_MCLK, priv->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) { dev_err(dev, @@ -141,14 +137,13 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); codec_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - data->clk_frequency, SND_SOC_CLOCK_IN); + priv->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(dev, "failed to set sysclk in %s\n", __func__); @@ -161,7 +156,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_wm8962_data *data; struct imx_priv *priv; struct clk *codec_clk; int int_port, ext_port; @@ -232,12 +226,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto fail; - } - codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(codec_clk)) { ret = PTR_ERR(codec_clk); @@ -245,39 +233,39 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } - data->clk_frequency = clk_get_rate(codec_clk); + priv->clk_frequency = clk_get_rate(codec_clk); clk_put(codec_clk); - data->dai.name = "HiFi"; - data->dai.stream_name = "HiFi"; - data->dai.codec_dai_name = "wm8962"; - data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); - data->dai.platform_of_node = ssi_np; - data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + priv->dai.name = "HiFi"; + priv->dai.stream_name = "HiFi"; + priv->dai.codec_dai_name = "wm8962"; + priv->dai.codec_of_node = codec_np; + priv->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + priv->dai.platform_of_node = ssi_np; + priv->dai.ops = &imx_hifi_ops; + priv->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; - data->card.dev = &pdev->dev; - ret = snd_soc_of_parse_card_name(&data->card, "model"); + priv->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) goto fail; - ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); if (ret) goto fail; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - data->card.dai_link = &data->dai; - data->card.dapm_widgets = imx_wm8962_dapm_widgets; - data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + priv->card.num_links = 1; + priv->card.owner = THIS_MODULE; + priv->card.dai_link = &priv->dai; + priv->card.dapm_widgets = imx_wm8962_dapm_widgets; + priv->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); - data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; + priv->card.late_probe = imx_wm8962_late_probe; + priv->card.set_bias_level = imx_wm8962_set_bias_level; - platform_set_drvdata(pdev, &data->card); - snd_soc_card_set_drvdata(&data->card, data); + platform_set_drvdata(pdev, &priv->card); + snd_soc_card_set_drvdata(&priv->card, priv); - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling"
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> This reverts commit f91b1e73ccde71d4bc69ae10d475196df38844ab. As explained by Lars-Peter Clausen: "This creates a resource leak. dmaengine_pcm_request_chan_of() requests bothtransmit and receive channels. It might return with an error if one of them fails, but the other one succeeded. In this case we need to call dmaengine_pcm_release_chan() to free the requested channel" So revert the commit. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 32ea16d..768247f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -450,7 +450,7 @@ int snd_dmaengine_pcm_register(struct device *dev, ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) - goto err_free_pcm; + goto err_free_dma; ret = snd_soc_add_component(dev, &pcm->component, &dmaengine_pcm_component, NULL, 0); @@ -461,7 +461,6 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); -err_free_pcm: kfree(pcm); return ret; } -- 2.7.4

1 0

[alsa-devel] Commit 8f7206d69a ("ASoC: imx-wm8962: Remove global variables") looks strange
by Daniel Baluta 26 Feb '18

26 Feb '18

Hi Fabio, Leonard (Cc'ed) noticed that commit in $subject looks very strange. For example: @@ -139,7 +140,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; - struct imx_priv *priv = &card_priv; + struct imx_priv *priv = snd_soc_card_get_drvdata(card); struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); So, now data and priv point to the same memory area but they are completely different structures! Unless we are missing something this is obviously a bug. I will try to have a look at the end of the week, but any comments are welcome. I think the fix would be to have one global priv structure and to unify imx_wm8962_data with imx_priv. thanks, Daniel.

2 3

[alsa-devel] [PATCH] ALSA: hda - Fix pincfg at resume on Lenovo T470 dock
by Takashi Iwai 26 Feb '18

26 Feb '18

We've added a quirk to enable the recent Lenovo dock support, where it overwrites the pin configs of NID 0x17 and 19, not only updating the pin config cache. It works right after the boot, but the problem is that the pin configs are occasionally cleared when the machine goes to PM. Meanwhile the quirk writes the pin configs only at the pre-probe, so this won't be applied any longer. For addressing that issue, this patch moves the code to overwrite the pin configs into HDA_FIXUP_ACT_INIT section so that it's always applied at both probe and resume time. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195161 Fixes: 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce28f7ce64e6..b9c93fa0a51c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4997,13 +4997,14 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; + snd_hda_apply_pincfgs(codec, pincfgs); + } else if (action == HDA_FIXUP_ACT_INIT) { /* Enable DOCK device */ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0); /* Enable DOCK device */ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0); - snd_hda_apply_pincfgs(codec, pincfgs); } } -- 2.16.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: soc-generic-dmaengine-pcm: Fix error handling
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> When dmaengine_pcm_request_chan_of() fails it should release the previously acquired resources, which in this case is to call kfree(pcm), so jump to the correct point in the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 768247f..32ea16d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -450,7 +450,7 @@ int snd_dmaengine_pcm_register(struct device *dev, ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) - goto err_free_dma; + goto err_free_pcm; ret = snd_soc_add_component(dev, &pcm->component, &dmaengine_pcm_component, NULL, 0); @@ -461,6 +461,7 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); +err_free_pcm: kfree(pcm); return ret; } -- 2.7.4

3 5

[alsa-devel] [RFC PATCH 0/3] ASoC: stm32: add S/PDIF support to SAI
by Olivier Moysan 26 Feb '18

26 Feb '18

This patch-set adds S/PDIF IEC6958 protocol support to STM32 SAI. Only S/PDIF playback is supported here, as SAI peripheral does not support S/PDIF capture. SAI peripheral does not take as input full IEC60958 frames as it generates preamble and parity bit. However, it cannot handle raw PCM data, as it is not designed to insert channel status and user data in the generated IEC frames. Moreover, SAI data register format does not match exactly IEC60958 subframe format. (see figure below). Channel status (C) and User data (U) are at 26-25 bits position, in data register, instead of 30-29 bits position. IEC60958 subframe bits: 31-28 27-4 3-0 P C U V data (16|24 bits) preamble SAI data register format: 27-24 23-0 P C U V data (16|24 bits) The impact in software is that the data have to be processed before DMA transfer. These constraints should be managed at driver level to minimize the impact on application. This means that a callback is necessary to perform data processing (shift and channel status insertion) This patch-set introduces a callback to allow processing on samples. The implementation is based on previous discussions available here: https://patchwork.kernel.org/patch/9570255/ BRs Olivier Olivier Moysan (3): ASoC: stm32: Add S/PDIF to SAI bindings ASoC: dmaengine_pcm: add processing support ASoC: stm32: sai: Add support of S/PDIF playback .../devicetree/bindings/sound/st,stm32-sai.txt | 6 + include/sound/dmaengine_pcm.h | 3 + sound/soc/soc-generic-dmaengine-pcm.c | 62 ++++++++- sound/soc/stm/stm32_sai.c | 2 + sound/soc/stm/stm32_sai.h | 2 + sound/soc/stm/stm32_sai_sub.c | 153 +++++++++++++++++---- 6 files changed, 202 insertions(+), 26 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: uniphier: fix broken sound if use SRC in replay
by Katsuhiro Suzuki 26 Feb '18

26 Feb '18

This patch fixes settings for ports with SRC. These ports need to set the fixed audio rate and clock. If not, the SRC outputs broken sound. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 1e5f053d9f6b..1711361fc0c2 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -501,7 +501,7 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub) OPORTMXCTR2_MSSEL_MASTER | OPORTMXCTR2_EXTLSIFSSEL_36 | OPORTMXCTR2_DACCKSEL_1_2; - } else { + } else if (sub->swm->type == PORT_TYPE_SPDIF) { if (sub->aio->pll_out >= ARRAY_SIZE(v_pll)) { dev_err(dev, "PLL(%d) is invalid\n", sub->aio->pll_out); @@ -521,6 +521,11 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub) v |= OPORTMXCTR2_EXTLSIFSSEL_24; break; } + } else { + v = OPORTMXCTR2_ACLKSEL_A1 | + OPORTMXCTR2_MSSEL_MASTER | + OPORTMXCTR2_EXTLSIFSSEL_36 | + OPORTMXCTR2_DACCKSEL_1_2; } regmap_write(r, OPORTMXCTR2(sub->swm->oport.map), v); } else { @@ -550,11 +555,19 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params) { struct regmap *r = sub->aio->chip->regmap; + unsigned int rate; u32 v; int ret; if (!pass_through) { - ret = aio_port_set_rate(sub, params_rate(params)); + if (sub->swm->type == PORT_TYPE_EVE || + sub->swm->type == PORT_TYPE_CONV) { + rate = 48000; + } else { + rate = params_rate(params); + } + + ret = aio_port_set_rate(sub, rate); if (ret) return ret; -- 2.16.1

2 1

[alsa-devel] [PATCH] ASoC: sirf: remove duplicated bit-wise or of USP_RXFIFO_THD_INT
by Colin King 26 Feb '18

26 Feb '18

From: Colin Ian King <colin.king(a)canonical.com> Bit pattern USP_RXFIFO_THD_INT is being bit-wise or'd twice; remove the redundant 2nd USP_RXFIFO_THD_INT Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sirf/sirf-usp.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sirf/sirf-usp.h b/sound/soc/sirf/sirf-usp.h index bf0201cb15bc..e22e13a619e8 100644 --- a/sound/soc/sirf/sirf-usp.h +++ b/sound/soc/sirf/sirf-usp.h @@ -154,7 +154,7 @@ USP_RX_IO_DMA_INT|\ USP_RXFIFO_FULL_INT|\ USP_RXFIFO_THD_INT|\ - USP_RXFIFO_THD_INT|USP_RX_TIMEOUT_INT) + USP_RX_TIMEOUT_INT) #define USP_INT_ALL 0x1FFF -- 2.15.1

2 1

[alsa-devel] [PATCH] ASoC: pxa: remove duplicated bit-wise or of SNDRV_PCM_FMTBIT_S24_LE
by Colin King 26 Feb '18

26 Feb '18

From: Colin Ian King <colin.king(a)canonical.com> Bit pattern SNDRV_PCM_FMTBIT_S24_LE is being bit-wise or'd twice; remove the redundant 2nd SNDRV_PCM_FMTBIT_S24_LE Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/pxa/mmp-sspa.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 64b85e30c1f8..7c998ea4ebee 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -377,7 +377,6 @@ static int mmp_sspa_probe(struct snd_soc_dai *dai) #define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops mmp_sspa_dai_ops = { -- 2.15.1

2 1

[alsa-devel] Applied "ASoC: sun4i-i2s: Fix RX slot number of SUN8I" to the asoc tree
by Mark Brown 26 Feb '18

26 Feb '18

The patch ASoC: sun4i-i2s: Fix RX slot number of SUN8I has been applied to the asoc tree at https://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 5a3386790a172cf738194e1574f631cd43c6140a Mon Sep 17 00:00:00 2001 From: Yong Deng <yong.deng(a)magewell.com> Date: Mon, 26 Feb 2018 10:43:52 +0800 Subject: [PATCH] ASoC: sun4i-i2s: Fix RX slot number of SUN8I I2S's RX slot number of SUN8I should be shifted 4 bit to left. Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3") Signed-off-by: Yong Deng <yong.deng(a)magewell.com> Reviewed-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/sunxi/sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index dca1143c1150..a4aa931ebfae 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -104,7 +104,7 @@ #define SUN8I_I2S_CHAN_CFG_REG 0x30 #define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK GENMASK(6, 4) -#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) (chan - 1) +#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) ((chan - 1) << 4) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK GENMASK(2, 0) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(chan) (chan - 1) -- 2.16.1

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: Support streams which can start/stop with DSP active
by Charles Keepax 26 Feb '18

26 Feb '18

Clear the buffer data structure on each trigger start such that the buffer is in a sensible state even if the DSP itself didn't restart. This is necessary to support voice control streams which can trigger multiple times without reloading the firmware. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 28 ++++++++++++++++++---------- 1 file changed, 18 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b6e7c614301f..9b27390cf4f1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3258,6 +3258,13 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) return 0; } +static void wm_adsp_buffer_clear(struct wm_adsp_compr_buf *buf) +{ + buf->irq_count = 0xFFFFFFFF; + buf->read_index = -1; + buf->avail = 0; +} + static int wm_adsp_buffer_init(struct wm_adsp *dsp) { struct wm_adsp_compr_buf *buf; @@ -3268,8 +3275,8 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) return -ENOMEM; buf->dsp = dsp; - buf->read_index = -1; - buf->irq_count = 0xFFFFFFFF; + + wm_adsp_buffer_clear(buf); ret = wm_adsp_buffer_locate(buf); if (ret < 0) { @@ -3327,16 +3334,17 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (wm_adsp_compr_attached(compr)) - break; - - ret = wm_adsp_compr_attach(compr); - if (ret < 0) { - adsp_err(dsp, "Failed to link buffer and stream: %d\n", - ret); - break; + if (!wm_adsp_compr_attached(compr)) { + ret = wm_adsp_compr_attach(compr); + if (ret < 0) { + adsp_err(dsp, "Failed to link buffer and stream: %d\n", + ret); + break; + } } + wm_adsp_buffer_clear(compr->buf); + /* Trigger the IRQ at one fragment of data */ ret = wm_adsp_buffer_write(compr->buf, HOST_BUFFER_FIELD(high_water_mark), -- 2.11.0

2 1

[alsa-devel] Applied "ASoC: dmaengine_pcm: add processing support" to the asoc tree
by Mark Brown 26 Feb '18

26 Feb '18

The patch ASoC: dmaengine_pcm: add processing support has been applied to the asoc tree at https://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 78648092ef46255e6dc6685202164199c86cf930 Mon Sep 17 00:00:00 2001 From: Olivier Moysan <olivier.moysan(a)st.com> Date: Mon, 19 Feb 2018 16:00:36 +0100 Subject: [PATCH] ASoC: dmaengine_pcm: add processing support Allow dmaengine client to optionally register a processing callback. This callback is intended to apply processing on samples in buffer copied from/to user space, before/after DMA transfer. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/dmaengine_pcm.h | 3 ++ sound/soc/soc-generic-dmaengine-pcm.c | 62 +++++++++++++++++++++++++++++++++-- 2 files changed, 63 insertions(+), 2 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 8a5a8404966e..47ef486852ed 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -140,6 +140,9 @@ struct snd_dmaengine_pcm_config { struct dma_chan *(*compat_request_channel)( struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream); + int (*process)(struct snd_pcm_substream *substream, + int channel, unsigned long hwoff, + void *buf, unsigned long bytes); dma_filter_fn compat_filter_fn; struct device *dma_dev; const char *chan_names[SNDRV_PCM_STREAM_LAST + 1]; diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 785f25ede3e5..567fbdfd1ca9 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -341,6 +341,41 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer( return snd_dmaengine_pcm_pointer(substream); } +static int dmaengine_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long hwoff, + void *buf, unsigned long bytes) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, SND_DMAENGINE_PCM_DRV_NAME); + struct snd_pcm_runtime *runtime = substream->runtime; + struct dmaengine_pcm *pcm = soc_component_to_pcm(component); + int (*process)(struct snd_pcm_substream *substream, + int channel, unsigned long hwoff, + void *buf, unsigned long bytes) = pcm->config->process; + bool is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + void *dma_ptr = runtime->dma_area + hwoff + + channel * (runtime->dma_bytes / runtime->channels); + int ret; + + if (is_playback) + if (copy_from_user(dma_ptr, (void __user *)buf, bytes)) + return -EFAULT; + + if (process) { + ret = process(substream, channel, hwoff, + (void __user *)buf, bytes); + if (ret < 0) + return ret; + } + + if (!is_playback) + if (copy_to_user((void __user *)buf, dma_ptr, bytes)) + return -EFAULT; + + return 0; +} + static const struct snd_pcm_ops dmaengine_pcm_ops = { .open = dmaengine_pcm_open, .close = snd_dmaengine_pcm_close, @@ -351,6 +386,17 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = { .pointer = dmaengine_pcm_pointer, }; +static const struct snd_pcm_ops dmaengine_pcm_process_ops = { + .open = dmaengine_pcm_open, + .close = snd_dmaengine_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = dmaengine_pcm_hw_params, + .hw_free = snd_pcm_lib_free_pages, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = dmaengine_pcm_pointer, + .copy_user = dmaengine_copy_user, +}; + static const struct snd_soc_component_driver dmaengine_pcm_component = { .name = SND_DMAENGINE_PCM_DRV_NAME, .probe_order = SND_SOC_COMP_ORDER_LATE, @@ -358,6 +404,13 @@ static const struct snd_soc_component_driver dmaengine_pcm_component = { .pcm_new = dmaengine_pcm_new, }; +static const struct snd_soc_component_driver dmaengine_pcm_component_process = { + .name = SND_DMAENGINE_PCM_DRV_NAME, + .probe_order = SND_SOC_COMP_ORDER_LATE, + .ops = &dmaengine_pcm_process_ops, + .pcm_new = dmaengine_pcm_new, +}; + static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_PLAYBACK] = "tx", [SNDRV_PCM_STREAM_CAPTURE] = "rx", @@ -453,8 +506,13 @@ int snd_dmaengine_pcm_register(struct device *dev, if (ret) goto err_free_pcm; - ret = snd_soc_add_component(dev, &pcm->component, - &dmaengine_pcm_component, NULL, 0); + if (config && config->process) + ret = snd_soc_add_component(dev, &pcm->component, + &dmaengine_pcm_component_process, + NULL, 0); + else + ret = snd_soc_add_component(dev, &pcm->component, + &dmaengine_pcm_component, NULL, 0); if (ret) goto err_free_dma; -- 2.16.1

1 0

[alsa-devel] [PATCH] ASoC: sun4i-i2s: Fix offset tune for SUN8I
by Yong Deng 26 Feb '18

26 Feb '18

SUN8I_I2S_TX_CHAN_OFFSET_MASK should be GENMASK(13, 12). And offset tune of RX should also be updated. Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3") Signed-off-by: Yong Deng <yong.deng(a)magewell.com> --- sound/soc/sunxi/sun4i-i2s.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..797cafffaac8 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -110,12 +110,15 @@ #define SUN8I_I2S_TX_CHAN_MAP_REG 0x44 #define SUN8I_I2S_TX_CHAN_SEL_REG 0x34 -#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 11) +#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 12) #define SUN8I_I2S_TX_CHAN_OFFSET(offset) (offset << 12) #define SUN8I_I2S_TX_CHAN_EN_MASK GENMASK(11, 4) #define SUN8I_I2S_TX_CHAN_EN(num_chan) (((1 << num_chan) - 1) << 4) #define SUN8I_I2S_RX_CHAN_SEL_REG 0x54 +#define SUN8I_I2S_RX_CHAN_OFFSET_MASK GENMASK(13, 12) +#define SUN8I_I2S_RX_CHAN_OFFSET(offset) (offset << 12) + #define SUN8I_I2S_RX_CHAN_MAP_REG 0x58 /** @@ -460,6 +463,9 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG, SUN8I_I2S_TX_CHAN_OFFSET_MASK, SUN8I_I2S_TX_CHAN_OFFSET(offset)); + regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG, + SUN8I_I2S_RX_CHAN_OFFSET_MASK, + SUN8I_I2S_RX_CHAN_OFFSET(offset)); } regmap_field_write(i2s->field_fmt_mode, val); -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: sun4i-i2s: Fix RX slot number of SUN8I
by Yong Deng 26 Feb '18

26 Feb '18

I2S's RX slot number of SUN8I should be shifted 4 bit to left. Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3") Signed-off-by: Yong Deng <yong.deng(a)magewell.com> --- sound/soc/sunxi/sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index dca1143c1150..a4aa931ebfae 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -104,7 +104,7 @@ #define SUN8I_I2S_CHAN_CFG_REG 0x30 #define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK GENMASK(6, 4) -#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) (chan - 1) +#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) ((chan - 1) << 4) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK GENMASK(2, 0) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(chan) (chan - 1) -- 1.8.3.1

2 1

[alsa-devel] [PATCH 0/7] Modernize vga_switcheroo by using device link for HDA
by Lukas Wunner 26 Feb '18

26 Feb '18

Modernize vga_switcheroo by using a "device link" to enforce a runtime PM dependency from an HDA controller to the GPU it's integrated into. Remove thereby obsoleted code and fix a bunch of bugs. Device links were introduced in v4.10. Users might see a small power saving if the discrete GPU is in use and its HDA controller is not, because the HDA controller is now allowed to runtime suspend to D3hot. Probing and accessing the HDA controller while the GPU is in D3cold should be very robust, unlike before. Under the hood things become quite a bit leaner. Also, this series gets us one step closer to supporting runtime PM on muxed laptops such as the MacBook Pro because it fixes a deadlock occurring when runtime resuming the discrete GPU on switching the mux. (The deadlock occurs in vga_switcheroo_set_dynamic_switch() and that function is obsoleted and removed by this series.) The meat of the series is in patch [5/7], read its commit message for details. The other patches contain prep and cleanup work. Patches [1/7], [2/7] and [5/7] require an ack from Bjorn (and Rafael?), patch [5/7] requires an ack from Takashi. Additionally I'd appreciate a Tested-by and/or Acked-by from Peter Wu, the resident Nvidia Optimus expert, and from Alex for AMD PowerXpress because my own testing only covers the MacBook Pro. Testing and comments from anyone else are most welcome of course. The series is based on 4.16-rc1. To test it on 4.15, you need to cherry-pick 7506dc798993 and 2fa6d6cdaf28. For your convenience I've pushed a 4.15-based branch to: https://github.com/l1k/linux/commits/switcheroo_devlink_v1 Minimal test procedure for non-Macs: - Note well: Recent Optimus require that a Mini-DP or HDMI cable is plugged in on boot for the HDA device to be present. - Check that HDA, GPU and root port autosuspend when not in use: cat /sys/bus/pci/devices/0000:01:00.1/power/runtime_status # HDA cat /sys/bus/pci/devices/0000:01:00.0/power/runtime_status # GPU cat /sys/bus/pci/devices/0000:00:01.0/power/runtime_status # Root Port - Check that all three autoresume when accessing the HDA: hdajacksensetest -c 1 - Unbind the HDA controller: echo 0000:01:00.1 > /sys/bus/pci/drivers/snd_hda_intel/unbind Wait for GPU to power off, then rebind the HDA controller: echo 0000:01:00.1 > /sys/bus/pci/drivers/snd_hda_intel/bind Check dmesg for errors, try accessing HDA with hdajacksensetest. - If your laptop uses the root port's _PR3 to cut power to the GPU: Unbind the GPU: echo 0000:01:00.0 > /sys/bus/pci/drivers/{nouveau,amdgpu,radeon}/unbind Allow runtime PM on the GPU: echo auto > /sys/bus/pci/devices/0000:01:00.0/power/control Wait for GPU to power off, then rebind it: echo 0000:01:00.0 > /sys/bus/pci/drivers/{nouveau,amdgpu,radeon}/bind Check dmesg for errors. If you see any then we may need to perform further actions in pci_pm_runtime_resume(), see patch [1/7]. Thanks, Lukas Lukas Wunner (7): PCI: Restore BARs on runtime resume despite being unbound PCI: Make pci_wakeup_bus() & pci_bus_set_current_state() public vga_switcheroo: Update PCI current_state on power change vga_switcheroo: Deduplicate power state tracking vga_switcheroo: Use device link for HDA controller vga_switcheroo: Let HDA autosuspend on mux change drm/nouveau: Runtime suspend despite HDA being unbound drivers/gpu/drm/amd/amdgpu/amdgpu_drv.c | 2 - drivers/gpu/drm/nouveau/nouveau_drm.c | 46 ---------- drivers/gpu/drm/nouveau/nouveau_drv.h | 1 - drivers/gpu/drm/radeon/radeon_drv.c | 2 - drivers/gpu/vga/vga_switcheroo.c | 152 ++++++++------------------------ drivers/pci/pci-driver.c | 8 +- drivers/pci/pci.c | 10 +-- drivers/pci/pci.h | 1 + drivers/pci/quirks.c | 39 ++++++++ include/linux/pci.h | 2 + include/linux/pci_ids.h | 1 + include/linux/vga_switcheroo.h | 6 -- include/sound/hdaudio.h | 3 - sound/pci/hda/hda_intel.c | 36 +++++--- sound/pci/hda/hda_intel.h | 3 - 15 files changed, 114 insertions(+), 198 deletions(-) -- 2.15.1

6 18

[alsa-devel] [PATCH 01/11] ASoC: rt5651: Avoid duplicating DMI quirks between codec and machine driver
by Hans de Goede 24 Feb '18

24 Feb '18

Instead of duplicating the DMI quirks between the codec and machine driver, add a rt5651_set_pdata() codec private function which the machine driver can use to pass in pdata. This means we need to delay setting up the jack-detect registers, so this commits moves this to rt5651_set_jack() code. Note this is actually a good thing as before we would register the irq handler before rt5651->hp_jack was assigned, leading to a potential NULL deref if the jack_detect work runs before the machine driver has called set_jack. While at it also move over to the standard snd_soc_codec_set_jack() function instead of using a codec private function for this. Note this commit causes the machine-driver to now actually honor the BYT_RT5651_DMIC_EN quirk, which was ignored before. To avoid this causing a functional change in what is purely a refactor commit, this commit removes the quirk from the defaults. If we really want the BYT_RT5651_DMIC_EN behavior anywhere it should be specifically enabled by follow up commits. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5651.c | 232 +++++++++++++++------------------- sound/soc/codecs/rt5651.h | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 43 +++++-- 3 files changed, 139 insertions(+), 142 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 45a73049cf64..c5d19a80506b 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -19,7 +19,6 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/acpi.h> -#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -32,10 +31,6 @@ #include "rl6231.h" #include "rt5651.h" -#define RT5651_JD_MAP(quirk) ((quirk) & GENMASK(7, 0)) -#define RT5651_IN2_DIFF BIT(16) -#define RT5651_DMIC_EN BIT(17) - #define RT5651_DEVICE_ID_VALUE 0x6281 #define RT5651_PR_RANGE_BASE (0xff + 1) @@ -43,8 +38,6 @@ #define RT5651_PR_BASE (RT5651_PR_RANGE_BASE + (0 * RT5651_PR_SPACING)) -static unsigned long rt5651_quirk; - static const struct regmap_range_cfg rt5651_ranges[] = { { .name = "PR", .range_min = RT5651_PR_BASE, .range_max = RT5651_PR_BASE + 0xb4, @@ -1585,10 +1578,88 @@ static int rt5651_set_bias_level(struct snd_soc_codec *codec, return 0; } +static irqreturn_t rt5651_irq(int irq, void *data) +{ + struct rt5651_priv *rt5651 = data; + + queue_delayed_work(system_power_efficient_wq, + &rt5651->jack_detect_work, msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static int rt5651_set_jack(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, void *data) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + int ret; + + if (!rt5651->pdata.jd_src || !rt5651->irq) + return -EINVAL; + + /* IRQ output on GPIO1 */ + regmap_update_bits(rt5651->regmap, RT5651_GPIO_CTRL1, + RT5651_GP1_PIN_MASK, RT5651_GP1_PIN_IRQ); + + switch (rt5651->pdata.jd_src) { + case RT5651_JD1_1: + regmap_update_bits(rt5651->regmap, RT5651_JD_CTRL2, + RT5651_JD_TRG_SEL_MASK, + RT5651_JD_TRG_SEL_JD1_1); + regmap_update_bits(rt5651->regmap, RT5651_IRQ_CTRL1, + RT5651_JD1_1_IRQ_EN, + RT5651_JD1_1_IRQ_EN); + break; + case RT5651_JD1_2: + regmap_update_bits(rt5651->regmap, RT5651_JD_CTRL2, + RT5651_JD_TRG_SEL_MASK, + RT5651_JD_TRG_SEL_JD1_2); + regmap_update_bits(rt5651->regmap, RT5651_IRQ_CTRL1, + RT5651_JD1_2_IRQ_EN, + RT5651_JD1_2_IRQ_EN); + break; + case RT5651_JD2: + regmap_update_bits(rt5651->regmap, RT5651_JD_CTRL2, + RT5651_JD_TRG_SEL_MASK, + RT5651_JD_TRG_SEL_JD2); + regmap_update_bits(rt5651->regmap, RT5651_IRQ_CTRL1, + RT5651_JD2_IRQ_EN, + RT5651_JD2_IRQ_EN); + break; + case RT5651_JD_NULL: + break; + default: + dev_warn(codec->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n"); + break; + } + + snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(dapm, "LDO"); + snd_soc_dapm_sync(dapm); + + regmap_update_bits(rt5651->regmap, RT5651_MICBIAS, + 0x38, 0x38); + + ret = devm_request_threaded_irq(codec->dev, rt5651->irq, NULL, + rt5651_irq, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING | + IRQF_ONESHOT, "rt5651", rt5651); + if (ret) { + dev_err(codec->dev, "Failed to reguest IRQ: %d\n", ret); + return ret; + } + + rt5651->hp_jack = hp_jack; + rt5651_irq(0, rt5651); + + return 0; +} + static int rt5651_probe(struct snd_soc_codec *codec) { struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); rt5651->codec = codec; @@ -1604,15 +1675,6 @@ static int rt5651_probe(struct snd_soc_codec *codec) snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); - if (rt5651->pdata.jd_src) { - snd_soc_dapm_force_enable_pin(dapm, "JD Power"); - snd_soc_dapm_force_enable_pin(dapm, "LDO"); - snd_soc_dapm_sync(dapm); - - regmap_update_bits(rt5651->regmap, RT5651_MICBIAS, - 0x38, 0x38); - } - return 0; } @@ -1698,6 +1760,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_rt5651 = { .resume = rt5651_resume, .set_bias_level = rt5651_set_bias_level, .idle_bias_off = true, + .set_jack = rt5651_set_jack, .component_driver = { .controls = rt5651_snd_controls, .num_controls = ARRAY_SIZE(rt5651_snd_controls), @@ -1747,54 +1810,16 @@ static const struct i2c_device_id rt5651_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5651_i2c_id); -static int rt5651_quirk_cb(const struct dmi_system_id *id) -{ - rt5651_quirk = (unsigned long) id->driver_data; - return 1; -} - -static const struct dmi_system_id rt5651_quirk_table[] = { - { - .callback = rt5651_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), - DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), - }, - .driver_data = (unsigned long *) RT5651_JD1_1, - }, - {} -}; - static int rt5651_parse_dt(struct rt5651_priv *rt5651, struct device_node *np) { - if (of_property_read_bool(np, "realtek,in2-differential")) - rt5651_quirk |= RT5651_IN2_DIFF; - if (of_property_read_bool(np, "realtek,dmic-en")) - rt5651_quirk |= RT5651_DMIC_EN; + rt5651->pdata.in2_diff = + of_property_read_bool(np, "realtek,in2-differential"); + rt5651->pdata.dmic_en = + of_property_read_bool(np, "realtek,dmic-en"); return 0; } -static void rt5651_set_pdata(struct rt5651_priv *rt5651) -{ - if (rt5651_quirk & RT5651_IN2_DIFF) - rt5651->pdata.in2_diff = true; - if (rt5651_quirk & RT5651_DMIC_EN) - rt5651->pdata.dmic_en = true; - if (RT5651_JD_MAP(rt5651_quirk)) - rt5651->pdata.jd_src = RT5651_JD_MAP(rt5651_quirk); -} - -static irqreturn_t rt5651_irq(int irq, void *data) -{ - struct rt5651_priv *rt5651 = data; - - queue_delayed_work(system_power_efficient_wq, - &rt5651->jack_detect_work, msecs_to_jiffies(250)); - - return IRQ_HANDLED; -} - static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); @@ -1860,17 +1885,26 @@ static void rt5651_jack_detect_work(struct work_struct *work) snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); } -int rt5651_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack) +static void rt5651_apply_pdata(struct rt5651_priv *rt5651) { - struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + if (rt5651->pdata.in2_diff) + regmap_update_bits(rt5651->regmap, RT5651_IN1_IN2, + RT5651_IN_DF2, RT5651_IN_DF2); - rt5651->hp_jack = hp_jack; - rt5651_irq(0, rt5651); + if (rt5651->pdata.dmic_en) + regmap_update_bits(rt5651->regmap, RT5651_GPIO_CTRL1, + RT5651_GP2_PIN_MASK, RT5651_GP2_PIN_DMIC1_SCL); +} - return 0; +void rt5651_set_pdata(struct snd_soc_codec *codec, + struct rt5651_platform_data *pdata) +{ + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + + rt5651->pdata = *pdata; + rt5651_apply_pdata(rt5651); } -EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); +EXPORT_SYMBOL_GPL(rt5651_set_pdata); static int rt5651_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -1890,10 +1924,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, rt5651->pdata = *pdata; else if (i2c->dev.of_node) rt5651_parse_dt(rt5651, i2c->dev.of_node); - else - dmi_check_system(rt5651_quirk_table); - - rt5651_set_pdata(rt5651); rt5651->regmap = devm_regmap_init_i2c(i2c, &rt5651_regmap); if (IS_ERR(rt5651->regmap)) { @@ -1917,69 +1947,13 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); - if (rt5651->pdata.in2_diff) - regmap_update_bits(rt5651->regmap, RT5651_IN1_IN2, - RT5651_IN_DF2, RT5651_IN_DF2); - - if (rt5651->pdata.dmic_en) - regmap_update_bits(rt5651->regmap, RT5651_GPIO_CTRL1, - RT5651_GP2_PIN_MASK, RT5651_GP2_PIN_DMIC1_SCL); + if (pdata || i2c->dev.of_node) + rt5651_apply_pdata(rt5651); + rt5651->irq = i2c->irq; rt5651->hp_mute = 1; - - if (rt5651->pdata.jd_src) { - - /* IRQ output on GPIO1 */ - regmap_update_bits(rt5651->regmap, RT5651_GPIO_CTRL1, - RT5651_GP1_PIN_MASK, RT5651_GP1_PIN_IRQ); - - switch (rt5651->pdata.jd_src) { - case RT5651_JD1_1: - regmap_update_bits(rt5651->regmap, RT5651_JD_CTRL2, - RT5651_JD_TRG_SEL_MASK, - RT5651_JD_TRG_SEL_JD1_1); - regmap_update_bits(rt5651->regmap, RT5651_IRQ_CTRL1, - RT5651_JD1_1_IRQ_EN, - RT5651_JD1_1_IRQ_EN); - break; - case RT5651_JD1_2: - regmap_update_bits(rt5651->regmap, RT5651_JD_CTRL2, - RT5651_JD_TRG_SEL_MASK, - RT5651_JD_TRG_SEL_JD1_2); - regmap_update_bits(rt5651->regmap, RT5651_IRQ_CTRL1, - RT5651_JD1_2_IRQ_EN, - RT5651_JD1_2_IRQ_EN); - break; - case RT5651_JD2: - regmap_update_bits(rt5651->regmap, RT5651_JD_CTRL2, - RT5651_JD_TRG_SEL_MASK, - RT5651_JD_TRG_SEL_JD2); - regmap_update_bits(rt5651->regmap, RT5651_IRQ_CTRL1, - RT5651_JD2_IRQ_EN, - RT5651_JD2_IRQ_EN); - break; - case RT5651_JD_NULL: - break; - default: - dev_warn(&i2c->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n"); - break; - } - } - INIT_DELAYED_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work); - if (i2c->irq) { - ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, - rt5651_irq, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING | - IRQF_ONESHOT, "rt5651", rt5651); - if (ret) { - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); - return ret; - } - } - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5651, rt5651_dai, ARRAY_SIZE(rt5651_dai)); diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h index 4f8b202121d7..151ac92f6bad 100644 --- a/sound/soc/codecs/rt5651.h +++ b/sound/soc/codecs/rt5651.h @@ -2065,6 +2065,7 @@ struct rt5651_priv { struct snd_soc_jack *hp_jack; struct delayed_work jack_detect_work; + int irq; int sysclk; int sysclk_src; int lrck[RT5651_AIFS]; @@ -2079,6 +2080,7 @@ struct rt5651_priv { bool hp_mute; }; -int rt5651_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack); +void rt5651_set_pdata(struct snd_soc_codec *codec, + struct rt5651_platform_data *pdata); + #endif /* __RT5651_H__ */ diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 49c538f2770a..8ef5b5500fb7 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -42,7 +42,15 @@ enum { BYT_RT5651_IN3_MAP, }; -#define BYT_RT5651_MAP(quirk) ((quirk) & GENMASK(7, 0)) +enum { + BYT_RT5651_JD_NULL = (RT5651_JD_NULL << 4), + BYT_RT5651_JD1_1 = (RT5651_JD1_1 << 4), + BYT_RT5651_JD1_2 = (RT5651_JD1_2 << 4), + BYT_RT5651_JD2 = (RT5651_JD2 << 4), +}; + +#define BYT_RT5651_MAP(quirk) ((quirk) & GENMASK(3, 0)) +#define BYT_RT5651_JDSRC(quirk) (((quirk) & GENMASK(7, 4)) >> 4) #define BYT_RT5651_DMIC_EN BIT(16) #define BYT_RT5651_MCLK_EN BIT(17) #define BYT_RT5651_MCLK_25MHZ BIT(18) @@ -53,7 +61,6 @@ struct byt_rt5651_private { }; static unsigned long byt_rt5651_quirk = BYT_RT5651_DMIC_MAP | - BYT_RT5651_DMIC_EN | BYT_RT5651_MCLK_EN; static void log_quirks(struct device *dev) @@ -66,6 +73,9 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN3_MAP) dev_info(dev, "quirk IN3_MAP enabled"); + if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) + dev_info(dev, "quirk jack-detect src %ld\n", + BYT_RT5651_JDSRC(byt_rt5651_quirk)); if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) dev_info(dev, "quirk DMIC enabled"); if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) @@ -288,6 +298,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), }, .driver_data = (void *)(BYT_RT5651_MCLK_EN | + BYT_RT5651_JD1_1 | BYT_RT5651_IN1_IN2_MAP), }, {} @@ -299,6 +310,7 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_codec *codec = runtime->codec; struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); const struct snd_soc_dapm_route *custom_map; + struct rt5651_platform_data pdata = {}; int num_routes; int ret; @@ -360,17 +372,26 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) dev_err(card->dev, "unable to set MCLK rate\n"); } - ret = snd_soc_card_jack_new(runtime->card, "Headset", - SND_JACK_HEADSET, &priv->jack, - bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins)); - if (ret) { - dev_err(runtime->dev, "Headset jack creation failed %d\n", ret); - return ret; - } + pdata.jd_src = BYT_RT5651_JDSRC(byt_rt5651_quirk); + if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) + pdata.dmic_en = true; + rt5651_set_pdata(codec, &pdata); + + if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &priv->jack, + bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins)); + if (ret) { + dev_err(runtime->dev, "Jack creation failed %d\n", ret); + return ret; + } - rt5651_set_jack_detect(codec, &priv->jack); + ret = snd_soc_codec_set_jack(codec, &priv->jack, NULL); + if (ret) + return ret; + } - return ret; + return 0; } static const struct snd_soc_pcm_stream byt_rt5651_dai_params = { -- 2.14.3

4 34

[alsa-devel] [PATCH - JACK PCM plugin] jack: Use boundary as hw_ptr wrap around
by twischer＠de.adit-jv.com 24 Feb '18

24 Feb '18

From: Timo Wischer <twischer(a)de.adit-jv.com> instead of using buffer_size as wrap around. This is required to detect Xruns. It is also required to allow the JACK thread to processes the whole ALSA audio buffer at once without calling snd_pcm_avail_update() in between. For example when the hw_ptr will be updated with hw_ptr += buffer_size and it is using the buffer_size as wrap around hw_ptr %= buffer_size would result in the same value as before the add operation. Due to that the user application would not recognize that the complete audio buffer was copied. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- This patch is depending on http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/130942.ht… diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 3aed332..c22a5d0 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -40,6 +40,7 @@ typedef struct { char **port_names; unsigned int num_ports; + snd_pcm_uframes_t boundary; unsigned int hw_ptr; unsigned int sample_bits; snd_pcm_uframes_t min_avail; @@ -130,6 +131,21 @@ static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io, static snd_pcm_sframes_t snd_pcm_jack_pointer(snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; + + /* ALSA library is calulating the delta between the last pointer and + * the current one. + * Normally it is expecting a value between 0 and buffer_size. + * The following example would result in an negative delta + * which would result in a hw_ptr which will be reduced. + * last_hw = jack->boundary - io->buffer_size + * hw = 0 + * But we cannot use + * return jack->hw_ptr % io->buffer_size; + * because in this case an update of + * hw_ptr += io->buffer_size + * would not be recognized by the ALSA library. + * Therefore we are using jack->boundary as the wrap around. + */ return jack->hw_ptr; } @@ -162,7 +178,7 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) while (xfer < nframes) { snd_pcm_uframes_t frames = nframes - xfer; - snd_pcm_uframes_t offset = hw_ptr; + snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; snd_pcm_uframes_t cont = io->buffer_size - offset; if (cont < frames) @@ -176,7 +192,8 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) } hw_ptr += frames; - hw_ptr %= io->buffer_size; + if (hw_ptr >= jack->boundary) + hw_ptr -= jack->boundary; xfer += frames; } jack->hw_ptr = hw_ptr; @@ -200,6 +217,8 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) err = snd_pcm_sw_params_current(io->pcm, swparams); if (err == 0) { snd_pcm_sw_params_get_avail_min(swparams, &jack->min_avail); + /* get boundary for available calulation */ + snd_pcm_sw_params_get_boundary(swparams, &jack->boundary); } /* deactivate jack connections if this is XRUN recovery */ -- 2.7.4

3 11

[alsa-devel] [PATCH] ALSA: usb-audio: Add a quirck for B&W PX headphones
by erik.veijola＠gmail.com 24 Feb '18

24 Feb '18

From: Erik Veijola <erik.veijola(a)gmail.com> The capture interface doesn't work and the playback interface only supports 48 kHz sampling rate even though it advertises more rates. Signed-off-by: Erik Veijola <erik.veijola(a)gmail.com> --- sound/usb/quirks-table.h | 47 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 50252046b01d..754e632a27bd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3325,4 +3325,51 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +{ + /* + * Bower's & Wilkins PX headphones only support the 48 kHz sample rate + * even though it advertises more. The capture interface doesn't work + * even on windows. + */ + USB_DEVICE(0x19b5, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + /* Capture */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_FILL_MAX | + UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x03, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + } + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 2.11.0

2 1

[alsa-devel] [RFC v3] ALSA: hda: Add a power_save blacklist
by Hans de Goede 23 Feb '18

23 Feb '18

On some boards setting power_save to a non 0 value leads to clicking / popping sounds when ever we enter/leave powersaving mode. Ideally we would figure out how to avoid these sounds, but that is not always feasible. This commit adds a blacklist for devices where powersaving is known to cause problems and disables it on these devices. Note I tried to put this blacklist in userspace first: https://github.com/systemd/systemd/pull/8128 But the systemd maintainers rightfully pointed out that it would be impossible to then later remove entries once we actually find a way to make power-saving work on listed boards without issues. Having this list in the kernel will allow removal of the blacklist entry in the same commit which fixes the clicks / plops. The blacklist only applies to the default power_save module-option value, if a user explicitly sets the module-option then the blacklist is not used. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=198611 Cc: stable(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in RFC v2: -Only use the blacklist if the power_save module-option is not explicitly set by the user Changes in RFC v2: -Fix logic to work on boards with more then 1 HDA device --- sound/pci/hda/hda_intel.c | 33 +++++++++++++++++++++++++++++++-- 1 file changed, 31 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c71dcacea807..9d67b6b0ec95 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -181,7 +181,7 @@ static const struct kernel_param_ops param_ops_xint = { }; #define param_check_xint param_check_int -static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +static int power_save = -1; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -2186,6 +2186,22 @@ static int azx_probe(struct pci_dev *pci, return err; } +/* On some boards setting power_save to a non 0 value leads to clicking / + * popping sounds when ever we enter/leave powersaving mode. Ideally we would + * figure out how to avoid these sounds, but that is not always feasible. + * So we keep a list of devices where we disable powersaving as its known + * to causes problems on these devices. + */ +static struct snd_pci_quirk power_save_blacklist[] = { + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), + /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ + SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), + {} +}; + /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { [AZX_DRIVER_NVIDIA] = 8, @@ -2197,7 +2213,9 @@ static int azx_probe_continue(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; + const struct snd_pci_quirk *q; int dev = chip->dev_index; + int val; int err; hda->probe_continued = 1; @@ -2278,7 +2296,18 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); - snd_hda_set_power_save(&chip->bus, power_save * 1000); + + val = power_save; + if (val == -1) { + val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && val) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + val = 0; + } + } + snd_hda_set_power_save(&chip->bus, val * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_autosuspend(&pci->dev); -- 2.14.3

2 2

[alsa-devel] [PATCHv4 0/4] Motorola Droid 4 Audio Support
by Sebastian Reichel 23 Feb '18

23 Feb '18

Hi, This adds audio support to Motorola Droid 4. I dropped the regulator from the DT binding as requested by Mark. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (needs different DAI/I2S config) Changes since PATCHv3: * Drop regulator from DT binding, request VAUDIO from global regulator namespace instead Changes since PATCHv2: * Fix a whitespace issue * Fix const notes Takashi provided * Fix a DAPM route issue I accidently introduced in v2 Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (4): dt-bindings: sound: add motorola,cpcap-audio-codec ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: omap4-droid4: add soundcard .../bindings/sound/motorola,cpcap-audio-codec.txt | 17 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 5 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1588 ++++++++++++++++++++ 6 files changed, 1701 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.15.1

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[alsa-devel] [PATCH v1 0/3] Enable HDA Codec support on Intel Platforms (Series1)
by Rakesh Ughreja 23 Feb '18

23 Feb '18

Enhanced HD Audio capabilities introduced in HD Audio controllers are added in backward compatible way i.e. it does not change the behavior of the controller with respect to HDA Bus and HDA devices. Since there is no change in the hardware with respect to the HDA bus and HDA device it is more appropriate to represent the same in software as well. In order to align software representation with hardware it makes more sense to have common data structures across device, bus and driver for both enhanced HDA controllers and legacy HDA controllers. This patch series removes the hdac_ext_device structure, hdac_ext_bus and hdac_ext_driver data structures so that legacy and enhanced HDaudio capabilities can be handled in a backward-compatible way without separate definitions. Once this clean-up is complete, additional patches will allow for HDaudio codec support in an ASoC framework without the need to develop new codec drivers, thereby enabling the use of the Intel DSP on more platforms (currently limited to hdmi). Rakesh Ughreja (3): ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead include/sound/hdaudio.h | 20 ++ include/sound/hdaudio_ext.h | 119 +++------- sound/hda/ext/hdac_ext_bus.c | 76 +++--- sound/hda/ext/hdac_ext_controller.c | 55 +++-- sound/hda/ext/hdac_ext_stream.c | 104 ++++----- sound/soc/codecs/hdac_hdmi.c | 416 ++++++++++++++++----------------- sound/soc/intel/skylake/skl-messages.c | 50 ++-- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 112 +++++---- sound/soc/intel/skylake/skl-topology.c | 18 +- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 178 +++++++------- sound/soc/intel/skylake/skl.h | 7 +- 13 files changed, 521 insertions(+), 648 deletions(-) -- 2.7.4

1 3

[alsa-devel] [PATCH V3 0/2 RESEND] ARM: dts: bcm283x: Fix probing of bcm2835-i2s
by Stefan Wahren 22 Feb '18

22 Feb '18

This small series fixes the probing of bcm2835-i2s, which is broken since 517e7a1537a ("ASoC: bcm2835: move to use the clock framework"). Changes in V3: * split up patch into dt-binding and functional part Changes in V2: * add Fixes, Reviewed-by and Tested-by tags Stefan Wahren (2): dt-bindings: bcm283x: Fix register ranges of bcm2835-i2s ARM: dts: bcm283x: Fix probing of bcm2835-i2s Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt | 4 ++-- Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 9 ++++----- arch/arm/boot/dts/bcm283x.dtsi | 4 ++-- 3 files changed, 8 insertions(+), 9 deletions(-) -- 2.7.4

3 4

[alsa-devel] [asoc:topic/generic-dmaengine 1/1] sound/soc/soc-generic-dmaengine-pcm.c:435:5: error: 'struct dmaengine_pcm' has no member named 'component'
by kbuild test robot 22 Feb '18

22 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/generic-dmaengine head: f0b3bdbd5c63c3f69f3b6a869355c676d22fd348 commit: f0b3bdbd5c63c3f69f3b6a869355c676d22fd348 [1/1] ASoC: soc-generic-dmaengine-pcm: Add a DMA debugfs_prefix entry config: i386-randconfig-x004-201807 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout f0b3bdbd5c63c3f69f3b6a869355c676d22fd348 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/soc-generic-dmaengine-pcm.c: In function 'snd_dmaengine_pcm_register': >> sound/soc/soc-generic-dmaengine-pcm.c:435:5: error: 'struct dmaengine_pcm' has no member named 'component' pcm->component.debugfs_prefix = "dma"; ^~ vim +435 sound/soc/soc-generic-dmaengine-pcm.c 417 418 /** 419 * snd_dmaengine_pcm_register - Register a dmaengine based PCM device 420 * @dev: The parent device for the PCM device 421 * @config: Platform specific PCM configuration 422 * @flags: Platform specific quirks 423 */ 424 int snd_dmaengine_pcm_register(struct device *dev, 425 const struct snd_dmaengine_pcm_config *config, unsigned int flags) 426 { 427 struct dmaengine_pcm *pcm; 428 int ret; 429 430 pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); 431 if (!pcm) 432 return -ENOMEM; 433 434 #ifdef CONFIG_DEBUG_FS > 435 pcm->component.debugfs_prefix = "dma"; 436 #endif 437 pcm->config = config; 438 pcm->flags = flags; 439 440 ret = dmaengine_pcm_request_chan_of(pcm, dev, config); 441 if (ret) 442 goto err_free_dma; 443 444 ret = snd_soc_add_platform(dev, &pcm->platform, 445 &dmaengine_pcm_platform); 446 if (ret) 447 goto err_free_dma; 448 449 return 0; 450 451 err_free_dma: 452 dmaengine_pcm_release_chan(pcm); 453 kfree(pcm); 454 return ret; 455 } 456 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); 457 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [PATCH v3 00/17] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

==Change log== v3 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to make RX and TX more clearly defined + Replaced "bool dir" with "int dir" in PATCH-04 + Replaced "!dir" with "int adir" in PATCH-05 + Put CBM_CFS behind the baudclk check to keep the same program flow in PATCH-14 + Removed all cpu_dai_drv changes in PATCH-15 v2 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to keep all ssi->i2s_net updated + Replaced bool tx with bool dir in PATCH-03 and PATCH-06 + Moved all initial register configurations from dai probe() to platform probe() so as to let AC97 CODEC successfully probe. * Added Tested-by from Caleb for TDM test cases. ==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Caleb has tested v1 with TDM lookback tests on i.MX6. Example of uncovered tests: AC97, PowerPC and FIQ. Nicolin Chen (17): ASoC: fsl_ssi: Redefine RX and TX macros ASoC: fsl_ssi: Keep ssi->i2s_net updated ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to probe() ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Add bool synchronous to mark synchronous mode ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 733 ++++++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 3 - 2 files changed, 370 insertions(+), 366 deletions(-) -- 2.7.4

4 27

[alsa-devel] [PATCH v2 00/16] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

==Change log== v2 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to keep all ssi->i2s_net updated + Replaced bool tx with bool dir in PATCH-03 and PATCH-06 + Moved all initial register configurations from dai probe() to platform probe() so as to let AC97 CODEC successfully probe. * Added Tested-by from Caleb for TDM test cases. ==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Caleb has tested v1 with TDM lookback tests on i.MX6. Example of uncovered tests: AC97, PowerPC and FIQ. Nicolin Chen (16): ASoC: fsl_ssi: Keep ssi->i2s_net updated ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to probe() ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Remove cpu_dai_drv from fsl_ssi structure ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 740 ++++++++++++++++++++++++------------------------ 1 file changed, 369 insertions(+), 371 deletions(-) -- 2.7.4

4 28

[alsa-devel] [PATCH v1 00/15] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Example of uncovered tests: TDM, AC97, PowerPC and FIQ. Nicolin Chen (15): ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to dai_probe() ASoC: fsl_ssi: Setup AC97 in dai_probe() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Remove cpu_dai_drv from fsl_ssi structure ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 710 ++++++++++++++++++++++++------------------------ 1 file changed, 348 insertions(+), 362 deletions(-) -- 2.7.4

5 34

[alsa-devel] Applied "ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config()" to the asoc tree
by Mark Brown 22 Feb '18

22 Feb '18

The patch ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() has been applied to the asoc tree at https://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 2e1327403b2543bc4dd639571b84bf6174212cee Mon Sep 17 00:00:00 2001 From: Nicolin Chen <nicoleotsuka(a)gmail.com> Date: Mon, 12 Feb 2018 14:03:14 -0800 Subject: [PATCH] ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() The FIFO clear helper function is just one line of code now. So it could be cleaned up by removing it and calling regmap directly. Meanwhile, FIFO clear could be applied to all use cases, not confined to AC97. So this patch also moves FIFO clear in the trigger() to fsl_ssi_config() and removes the AC97 check. Note that SOR register is safe from offline_config HW limit. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Tested-by: Caleb Crome <caleb(a)crome.org> Tested-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Reviewed-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_ssi.c | 33 ++++++++++----------------------- 1 file changed, 10 insertions(+), 23 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0d8c800db0b3..d276b78684e4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -409,17 +409,6 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi *ssi, bool enable) } } -/** - * Clear remaining data in the FIFO to avoid dirty data or channel slipping - */ -static void fsl_ssi_fifo_clear(struct fsl_ssi *ssi, bool is_rx) -{ - bool tx = !is_rx; - - regmap_update_bits(ssi->regs, REG_SSI_SOR, - SSI_SOR_xX_CLR(tx), SSI_SOR_xX_CLR(tx)); -} - /** * Exclude bits that are used by the opposite stream * @@ -446,10 +435,11 @@ static void fsl_ssi_fifo_clear(struct fsl_ssi *ssi, bool is_rx) static void fsl_ssi_config(struct fsl_ssi *ssi, bool enable, struct fsl_ssi_regvals *vals) { - int adir = (&ssi->regvals[TX] == vals) ? RX : TX; - int dir = (&ssi->regvals[TX] == vals) ? TX : RX; + bool tx = &ssi->regvals[TX] == vals; struct regmap *regs = ssi->regs; struct fsl_ssi_regvals *avals; + int adir = tx ? RX : TX; + int dir = tx ? TX : RX; bool aactive; /* Check if the opposite stream is active */ @@ -489,7 +479,9 @@ static void fsl_ssi_config(struct fsl_ssi *ssi, bool enable, /* Online configure single direction while SSI is running */ if (enable) { - fsl_ssi_fifo_clear(ssi, vals->scr & SSI_SCR_RE); + /* Clear FIFO to prevent dirty data or channel slipping */ + regmap_update_bits(ssi->regs, REG_SSI_SOR, + SSI_SOR_xX_CLR(tx), SSI_SOR_xX_CLR(tx)); regmap_update_bits(regs, REG_SSI_SRCR, vals->srcr, vals->srcr); regmap_update_bits(regs, REG_SSI_STCR, vals->stcr, vals->stcr); @@ -511,6 +503,10 @@ static void fsl_ssi_config(struct fsl_ssi *ssi, bool enable, regmap_update_bits(regs, REG_SSI_SRCR, srcr, 0); regmap_update_bits(regs, REG_SSI_STCR, stcr, 0); regmap_update_bits(regs, REG_SSI_SIER, sier, 0); + + /* Clear FIFO to prevent dirty data or channel slipping */ + regmap_update_bits(ssi->regs, REG_SSI_SOR, + SSI_SOR_xX_CLR(tx), SSI_SOR_xX_CLR(tx)); } config_done: @@ -1091,7 +1087,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct regmap *regs = ssi->regs; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -1116,14 +1111,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return -EINVAL; } - /* Clear corresponding FIFO */ - if (fsl_ssi_is_ac97(ssi)) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - regmap_write(regs, REG_SSI_SOR, SSI_SOR_TX_CLR); - else - regmap_write(regs, REG_SSI_SOR, SSI_SOR_RX_CLR); - } - return 0; } -- 2.16.1

1 0

[alsa-devel] Applied "ASoC: fsl_ssi: Move one-time configurations to probe()" to the asoc tree
by Mark Brown 22 Feb '18

22 Feb '18

The patch ASoC: fsl_ssi: Move one-time configurations to probe() has been applied to the asoc tree at https://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 a1d154ac2765cf982be0d4d378bc6ba342c467a3 Mon Sep 17 00:00:00 2001 From: Nicolin Chen <nicoleotsuka(a)gmail.com> Date: Mon, 12 Feb 2018 14:03:20 -0800 Subject: [PATCH] ASoC: fsl_ssi: Move one-time configurations to probe() The probe() could handle some one-time configurations since they will not be changed once being configured. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Tested-by: Caleb Crome <caleb(a)crome.org> Tested-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Reviewed-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_ssi.c | 39 ++++++++++++++++++++++++++------------- 1 file changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7e15b30d4f94..36f3d51c5c43 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -865,7 +865,6 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, { struct regmap *regs = ssi->regs; u32 strcr = 0, stcr, srcr, scr, mask; - u8 wm; ssi->dai_fmt = fmt; @@ -874,8 +873,6 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, return -EINVAL; } - fsl_ssi_setup_regvals(ssi); - regmap_read(regs, REG_SSI_SCR, &scr); scr &= ~(SSI_SCR_SYN | SSI_SCR_I2S_MODE_MASK); /* Synchronize frame sync clock for TE to avoid data slipping */ @@ -990,16 +987,6 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, regmap_write(regs, REG_SSI_SRCR, srcr); regmap_write(regs, REG_SSI_SCR, scr); - wm = ssi->fifo_watermark; - - regmap_write(regs, REG_SSI_SFCSR, - SSI_SFCSR_TFWM0(wm) | SSI_SFCSR_RFWM0(wm) | - SSI_SFCSR_TFWM1(wm) | SSI_SFCSR_RFWM1(wm)); - - if (ssi->use_dual_fifo) - regmap_update_bits(regs, REG_SSI_SCR, - SSI_SCR_TCH_EN, SSI_SCR_TCH_EN); - if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi); @@ -1248,6 +1235,29 @@ static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = { .write = fsl_ssi_ac97_write, }; +/** + * Initialize SSI registers + */ +static int fsl_ssi_hw_init(struct fsl_ssi *ssi) +{ + u32 wm = ssi->fifo_watermark; + + /* Initialize regvals */ + fsl_ssi_setup_regvals(ssi); + + /* Set watermarks */ + regmap_write(ssi->regs, REG_SSI_SFCSR, + SSI_SFCSR_TFWM0(wm) | SSI_SFCSR_RFWM0(wm) | + SSI_SFCSR_TFWM1(wm) | SSI_SFCSR_RFWM1(wm)); + + /* Enable Dual FIFO mode */ + if (ssi->use_dual_fifo) + regmap_update_bits(ssi->regs, REG_SSI_SCR, + SSI_SCR_TCH_EN, SSI_SCR_TCH_EN); + + return 0; +} + /** * Make every character in a string lower-case */ @@ -1533,6 +1543,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ssi->dai_fmt) _fsl_ssi_set_dai_fmt(dev, ssi, ssi->dai_fmt); + /* Initially configures SSI registers */ + fsl_ssi_hw_init(ssi); + if (fsl_ssi_is_ac97(ssi)) { u32 ssi_idx; -- 2.16.1

1 0

[alsa-devel] Applied "ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init()" to the asoc tree
by Mark Brown 22 Feb '18

22 Feb '18

The patch ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() has been applied to the asoc tree at https://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 37ac30a4bdc397cf1311111264ac35dc8473e1de Mon Sep 17 00:00:00 2001 From: Nicolin Chen <nicoleotsuka(a)gmail.com> Date: Mon, 12 Feb 2018 14:03:21 -0800 Subject: [PATCH] ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() AC97 configures most of registers earlier to start a communication with CODECs in order to successfully initialize CODEC. Currently, _fsl_ssi_set_dai_fmt() and fsl_ssi_setup_ac97() are called to get all SSI registers properly set. Since now the driver has a fsl_ssi_hw_init() to handle all register initial settings, this patch moves those register settings of AC97 to the fsl_ssi_hw_init() as well. Meanwhile it applies _fsl_ssi_set_dai_fmt() call to AC97 only since other formats would be configured via normal set_dai_fmt() directly. This patch also adds fsl_ssi_hw_clean() to cleanup control bits for AC97 in the platform remote() function. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Tested-by: Caleb Crome <caleb(a)crome.org> Tested-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Reviewed-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_ssi.c | 33 +++++++++++++++++++++++++++------ 1 file changed, 27 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 36f3d51c5c43..dfb0da3cedda 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -987,9 +987,6 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, regmap_write(regs, REG_SSI_SRCR, srcr); regmap_write(regs, REG_SSI_SCR, scr); - if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) - fsl_ssi_setup_ac97(ssi); - return 0; } @@ -1255,9 +1252,33 @@ static int fsl_ssi_hw_init(struct fsl_ssi *ssi) regmap_update_bits(ssi->regs, REG_SSI_SCR, SSI_SCR_TCH_EN, SSI_SCR_TCH_EN); + /* AC97 should start earlier to communicate with CODECs */ + if (fsl_ssi_is_ac97(ssi)) { + _fsl_ssi_set_dai_fmt(ssi->dev, ssi, ssi->dai_fmt); + fsl_ssi_setup_ac97(ssi); + } + return 0; } +/** + * Clear SSI registers + */ +static void fsl_ssi_hw_clean(struct fsl_ssi *ssi) +{ + /* Disable registers for AC97 */ + if (fsl_ssi_is_ac97(ssi)) { + /* Disable TE and RE bits first */ + regmap_update_bits(ssi->regs, REG_SSI_SCR, + SSI_SCR_TE | SSI_SCR_RE, 0); + /* Disable AC97 mode */ + regmap_write(ssi->regs, REG_SSI_SACNT, 0); + /* Unset WAIT bits */ + regmap_write(ssi->regs, REG_SSI_SOR, 0); + /* Disable SSI -- software reset */ + regmap_update_bits(ssi->regs, REG_SSI_SCR, SSI_SCR_SSIEN, 0); + } +} /** * Make every character in a string lower-case */ @@ -1540,9 +1561,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) } done: - if (ssi->dai_fmt) - _fsl_ssi_set_dai_fmt(dev, ssi, ssi->dai_fmt); - /* Initially configures SSI registers */ fsl_ssi_hw_init(ssi); @@ -1592,6 +1610,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi->pdev) platform_device_unregister(ssi->pdev); + /* Clean up SSI registers */ + fsl_ssi_hw_clean(ssi); + if (ssi->soc->imx) fsl_ssi_imx_clean(pdev, ssi); -- 2.16.1

1 0

[alsa-devel] [PATCH v4 00/17] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

==Change log== v4 * Reworked the series by taking suggestions from Maciej + Added TXBIT0 bit back to play safe in PATCH-14 + Made bool synchronous exclusive with AC97 mode in PATCH-16 v3 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to make RX and TX more clearly defined + Replaced "bool dir" with "int dir" in PATCH-04 + Replaced "!dir" with "int adir" in PATCH-05 + Put CBM_CFS behind the baudclk check to keep the same program flow in PATCH-14 + Removed all cpu_dai_drv changes in PATCH-15 v2 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to keep all ssi->i2s_net updated + Replaced bool tx with bool dir in PATCH-03 and PATCH-06 + Moved all initial register configurations from dai probe() to platform probe() so as to let AC97 CODEC successfully probe. * Added Tested-by from Caleb for TDM test cases. ==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Caleb has tested v1 with TDM lookback tests on i.MX6. Example of uncovered tests: AC97, PowerPC and FIQ. Nicolin Chen (17): ASoC: fsl_ssi: Redefine RX and TX macros ASoC: fsl_ssi: Keep ssi->i2s_net updated ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to probe() ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Add bool synchronous to mark synchronous mode ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 741 +++++++++++++++++++++++++----------------------- sound/soc/fsl/fsl_ssi.h | 3 - 2 files changed, 379 insertions(+), 365 deletions(-) -- 2.7.4

4 24

[alsa-devel] [PATCH v6 00/17] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

[ The v6 just rebased the series and fixed the comments in probe(). There is no need to re-test it except the uncovered tests listed at the end of this cover letter. Timur, these patches have been in the list for quite a long time and there has been no review for nearly a month since last ver. Would you please give a review and send an ack? Thanks a lot. ] ==Change log== v6 * Added Tested-by and Reviewed-by from Maciej. * Fixed one line of comments in the probe() suggested by Maciej. * Rebased all patches since there's a change of fsl_ssi merged recently. v5 * Reworked the series by taking suggestions from Maciej for AC97 + Fixed SSI lockup issue by changing cleanup sequence in PATCH-13 + Moved fsl_ssi_hw_clean() after unregistering the CODEC device in PATCH-13 + Set NULL as the parent of CODEC platform device to fix a NULL pointer dereference bug in PATCH-16 * Updated comments of three variables/pointers in struct fsl_ssi to describe them more accurately in PATCH-16 v4 * Reworked the series by taking suggestions from Maciej + Added TXBIT0 bit back to play safe in PATCH-14 + Made bool synchronous exclusive with AC97 mode in PATCH-16 v3 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to make RX and TX more clearly defined + Replaced "bool dir" with "int dir" in PATCH-04 + Replaced "!dir" with "int adir" in PATCH-05 + Put CBM_CFS behind the baudclk check to keep the same program flow in PATCH-14 + Removed all cpu_dai_drv changes in PATCH-15 v2 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to keep all ssi->i2s_net updated + Replaced bool tx with bool dir in PATCH-03 and PATCH-06 + Moved all initial register configurations from dai probe() to platform probe() so as to let AC97 CODEC successfully probe. * Added Tested-by from Caleb for TDM test cases. ==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require to be fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Caleb has tested all versions with TDM lookback tests on i.MX6. Maciej has tested v5 with AC97 tests on i.MX6. Example of uncovered tests: PowerPC and FIQ. Nicolin Chen (17): ASoC: fsl_ssi: Redefine RX and TX macros ASoC: fsl_ssi: Keep ssi->i2s_net updated ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to probe() ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Add bool synchronous to mark synchronous mode ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 756 +++++++++++++++++++++++++----------------------- sound/soc/fsl/fsl_ssi.h | 3 - 2 files changed, 395 insertions(+), 364 deletions(-) -- 2.1.4

2 18

[alsa-devel] [PATCH] ASoC: uniphier: evea: fix lisence comment style
by Katsuhiro Suzuki 22 Feb '18

22 Feb '18

This patch changes lisence comment style from C to C++. And use SPDX instead of boiler plate of GPLv2, as same as other audio codec drivers. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/evea.c | 22 ++++------------------ 1 file changed, 4 insertions(+), 18 deletions(-) diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index ac2d30b55fb6..439f14f91b23 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -1,22 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * Socionext UniPhier EVEA ADC/DAC codec driver. - * - * Copyright (c) 2016-2017 Socionext Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * as published by the Free Software Foundation; version 2 - * of the License. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see <http://www.gnu.org/licenses/>. - */ +// +// Socionext UniPhier EVEA ADC/DAC codec driver. +// +// Copyright (c) 2016-2017 Socionext Inc. #include <linux/clk.h> #include <linux/module.h> -- 2.16.1

2 1

[alsa-devel] [PATCH] ASoC: soc-generic-dmaengine-pcm: Add a DMA debugfs_prefix entry
by Fabio Estevam 22 Feb '18

22 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> After the conversion from platform to component the following warnings are seen: fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory fsl-spdif-dai 2004000.spdif: ASoC: Failed to createcomponent debugfs directory As explained by Mark Brown: "It's a legit warning - we shouldn't really be creating two components for the same device. However this is a bit of a corner case as it's the dmaengine platform driver which is kind of a virtual device, it's not really the device that we use with DMA that's being represented but rather the link between that and the DMA controller." Disambiguate the DMA component name by adding a "dma" debugfs_prefix entry. Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index c07d5c7..768247f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -442,6 +442,9 @@ int snd_dmaengine_pcm_register(struct device *dev, if (!pcm) return -ENOMEM; +#ifdef CONFIG_DEBUG_FS + pcm->component.debugfs_prefix = "dma"; +#endif pcm->config = config; pcm->flags = flags; -- 2.7.4

2 1

[alsa-devel] [RFC v2] ALSA: hda: Add a power_save blacklist
by Hans de Goede 22 Feb '18

22 Feb '18

On some boards setting power_save to a non 0 value leads to clicking / popping sounds when ever we enter/leave powersaving mode. Ideally we would figure out how to avoid these sounds, but that is not always feasible. This commit adds a blacklist for devices where powersaving is known to cause problems and disables it on these devices. Note I tried to put this blacklist in userspace first: https://github.com/systemd/systemd/pull/8128 But the systemd maintainers rightfully pointed out that it would be impossible to then later remove entries once we actually find a way to make power-saving work on listed boards without issues. Having this list in the kernel will allow removal of the blacklist entry in the same commit which fixes the clicks / plops. The blacklist only applies to the default power_save module-option value, if an user explicitly sets the module-option then the blacklist is not used. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=198611 Cc: stable(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in RFC v2: -Only use the blacklist if the power_save module-option is not explicitly set by the user --- sound/pci/hda/hda_intel.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c71dcacea807..7c588b0d6c96 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -181,7 +181,7 @@ static const struct kernel_param_ops param_ops_xint = { }; #define param_check_xint param_check_int -static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +static int power_save = -1; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -2186,6 +2186,22 @@ static int azx_probe(struct pci_dev *pci, return err; } +/* On some boards setting power_save to a non 0 value leads to clicking / + * popping sounds when ever we enter/leave powersaving mode. Ideally we would + * figure out how to avoid these sounds, but that is not always feasible. + * So we keep a list of devices where we disable powersaving as its known + * to causes problems on these devices. + */ +static struct snd_pci_quirk power_save_blacklist[] = { + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), + /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ + SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), + {} +}; + /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { [AZX_DRIVER_NVIDIA] = 8, @@ -2197,6 +2213,7 @@ static int azx_probe_continue(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; + const struct snd_pci_quirk *q; int dev = chip->dev_index; int err; @@ -2278,6 +2295,16 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); + + if (power_save == -1) { + power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && power_save) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + power_save = 0; + } + } snd_hda_set_power_save(&chip->bus, power_save * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_autosuspend(&pci->dev); -- 2.14.3

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[alsa-devel] [RFC 0/1] ALSA: hda: Add a power_save blacklist
by Hans de Goede 22 Feb '18

22 Feb '18

Hi All, On some boards setting power_save to a non 0 value leads to clicking / popping sounds when ever we enter/leave powersaving mode. Ideally we would figure out how to avoid these sounds, but that is not always feasible. This commit adds a blacklist for devices where powersaving is known to cause problems and disables it on these devices. Note this patch ATM is a RFC because I still need to get test-feedback that it actually works on affected boards. I tried to put this blacklist in userspace first: https://github.com/systemd/systemd/pull/8128 But the systemd maintainers rightfully pointed out that it would be impossible to then later remove entries once we actually find a way to make power-saving work on listed boards without issues. Having this list in the kernel will allow removal of the blacklist entry in the same commit which fixes the clicks / plops. I've chosen to also apply the blacklist to changes made through /sys/module/snd_hda_intel/parameters/power_save, since tools such as powertop and TLP do this automatically. About the choice to also apply this to changes made through /sys/module/snd_hda_intel/parameters/power_save, one could argue that this will get in the way of users who want the power-saving anyways and are willing to live with the clicks/plops. An alternative approach would be to only apply this to the value of the module-parameter at boot and then only when it matches CONFIG_SND_HDA_POWER_SAVE_DEFAULT. I'm open to changing this. Regards, Hans

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[alsa-devel] [PATCH v13 0/8] TDA1997x HDMI video reciver
by Tim Harvey 22 Feb '18

22 Feb '18

This is a v4l2 subdev driver supporting the TDA1997x HDMI video receiver. I've tested this on a Gateworks GW54xx/GW551x with an IMX6Q/IMX6DL which uses the TDA19971 with 16bits connected to the IMX6 CSI and single-lane I2S audio providing 2-channel audio. For this configuration I've tested both 16bit YUV422 and 8bit BT656 parallel video bus modes. While the driver should support the TDA1993 I do not have one for testing. Further potential development efforts include: - CEC support - HDCP support - TDA19972 support (2 inputs) Media graphs can be found at http://dev.gateworks.com/docs/linux/media See details below for configuration and compliance tests History: v13: - fix coccinelle warnings v12: - fix coccinelle warnings v11: - return -ERANGE from tda1997x_detect_std (Hans) - clean up tda1997x_g_input_status (Hans) - show detected timings on resolution change if debug enabled - fix unitialized ret var in tda1997x_query_dv_timings - update log status to show detected timings v10: - removed unnecessary check for !timings in get/set/query dv timings (Hans) - dropped pointless s_stream handler (Hans) - remove need for detected_timings and always use set timings (Hans) v9: - add digital video decoder video interface entity function v8: - fix clearing pad for VIDIOC_DV_TIMIGNS_CAP - support full range of input modes based on timings_cap - add patch to fix clearing pad for VIDIOC_DV_TIMIGINGS - fix available formats for tda19971 bt656 bus width >12 - fix set_format (compliance) - fixed get/set edid (compliance) - add init_cfg to setup default pad config (compliance) - added missing pad checks to get_dv_timings_cap/enum_dv_timings (compliance) - fix alignment of if statement and whitespace in comment (Hans) - move regs to tda1997x_regs.h to clean up (Hans) - add define and sanity check for num of mbus_codes (Hans) v7: - fix interlaced mode - support no AVI infoframe (ie DVI) (Hans) - add support for multiple output formats (Hans) v6: - tda1997x: fix return on regulator enablei in tda1997x_set_power() (Fabio) - tda1997x: fix colorspace handling (Hans) - bindings: added Robs's ack (Rob) - replace copyright with SPDX tag (Philippe) v5: - added v4l2_hdmi_colorimetry() patch from Hans to series - bindings: added Sakari's ack - tda1997x: uppercase string constants - tda1997x: use v4l2_hdmi_rx_coloriemtry to fill format - tda1997x: fix V4L2_CID_DV_RX_RGB_RANGE - tda1997x: fix interlaced mode format - dts: remove leading 0 from unit address - dts: add newline between property list and child node - dts: added missing audmux in GW551x dts v4: - move include/dt-bindings/media/tda1997x.h to bindings patch - clarify port node details in bindings - fix typos - fix default quant range for VGA - fix quant range handling and conv matrix - add additional standards and capabilities to timings_cap v3: - fix typo in dt bindings - added dt bindings for GW551x - use V4L2_DV_BT_FRAME_WIDTH/HEIGHT macros - fixed missing break - use only hdmi_infoframe_log for infoframe logging - simplify tda1997x_s_stream error handling - add delayed work proc to handle hotplug enable/disable - fix set_edid (disable HPD before writing, enable after) - remove enabling edid by default - initialize timings - take quant range into account in colorspace conversion - remove vendor/product tracking (we provide this in log_status via infoframes) - add v4l_controls - add more detail to log_status - calculate vhref generator timings - timing detection fixes (rounding errors, hswidth errors) - rename configure_input/configure_conv functions v2: - encorporate feedback into dt bindings - change audio dt bindings - implement dv timings enum/cap - remove deprecated g_mbus_config op - fix dv_query_timings - add EDID get/set handling - remove max-pixel-rate support - add audio codec DAI support - added media-ctl and v4l2-compliance details v1: - initial RFC Pipeline configuration: $ media-ctl -e 'tda19971 2-0048' /dev/v4l-subdev1 $ v4l2-ctl -d /dev/v4l-subdev1 --set-dv-bt-timings=query BT timings set $ media-ctl --get-v4l2 '"tda19971 2-0048":0' [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] $ media-ctl --link "tda19971 2-0048":0 -> "ipu1_csi0_mux":1[1] $ media-ctl --link "ipu1_csi0_mux":2 -> "ipu1_csi0":0[1] $ media-ctl --link "ipu1_csi0":2 -> "ipu1_csi0 capture":0[1] $ media-ctl --set-v4l2 'tda19971 2-0048':0[fmt:UYVY8_2X8/1280x720] $ media-ctl --set-v4l2 'ipu1_csi0_mux':2[fmt:UYVY8_2X8/1280x720] $ media-ctl --set-v4l2 'ipu1_csi0':0[fmt:UYVY8_2X8/1280x720] $ gst-launch-1.0 v4l2src device=/dev/video4 ! video/x-raw,width=1280,height=720,format=UYVY ! jpegenc ! rtpjpegpay ! udpsink host=172.24.40.6 port=5000 $ media-ctl -d /dev/media0 -p Media controller API version 4.15.0 Media device information ------------------------ driver imx-media model imx-media serial bus info hw revision 0x0 driver version 4.15.0 Device topology - entity 1: adv7180 2-0020 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 20004 device node name /dev/v4l-subdev0 pad0: Source [fmt:UYVY8_2X8/720x480 field:interlaced colorspace:smpte170m] -> "ipu2_csi1_mux":1 [] - entity 3: tda19971 2-0048 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev1 pad0: Source [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] [dv.caps:BT.656/1120 min:640x350@13000000 max:1920x1200@165000000 stds:CEA-861,DMT,CVT,GTF caps:interlaced,progressive,reduced-blanking,custom] [dv.detect:BT.656/1120 1280x720p60 (1650x750) stds:CEA-861 flags:can-reduce-fps,CE-video,has-cea861-vic] [dv.current:BT.656/1120 1280x720p60 (1650x750) stds:CEA-861 flags:can-reduce-fps,CE-video,has-cea861-vic] -> "ipu1_csi0_mux":1 [ENABLED] - entity 5: ipu1_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev2 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] - entity 9: ipu2_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev3 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] - entity 13: ipu1_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev4 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_vdic":2 [] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf":0 [] - entity 17: ipu1_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev5 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc capture":0 [] - entity 20: ipu1_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video0 pad0: Sink <- "ipu1_ic_prpenc":1 [] - entity 26: ipu1_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev6 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf capture":0 [] - entity 29: ipu1_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video1 pad0: Sink <- "ipu1_ic_prpvf":1 [] - entity 35: ipu2_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev7 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_vdic":2 [] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf":0 [] - entity 39: ipu2_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev8 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc capture":0 [] - entity 42: ipu2_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video2 pad0: Sink <- "ipu2_ic_prpenc":1 [] - entity 48: ipu2_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev9 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf capture":0 [] - entity 51: ipu2_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video3 pad0: Sink <- "ipu2_ic_prpvf":1 [] - entity 57: ipu1_csi0 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev10 pad0: Sink [fmt:UYVY8_2X8/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range crop.bounds:(0,0)/1280x720 crop:(0,0)/1280x720 compose.bounds:(0,0)/1280x720 compose:(0,0)/1280x720] <- "ipu1_csi0_mux":2 [ENABLED] pad1: Source [fmt:AYUV8_1X32/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range] -> "ipu1_csi0 capture":0 [ENABLED] - entity 61: ipu1_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video4 pad0: Sink <- "ipu1_csi0":2 [ENABLED] - entity 67: ipu1_csi1 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev11 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_csi1 capture":0 [] - entity 71: ipu1_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video5 pad0: Sink <- "ipu1_csi1":2 [] - entity 77: ipu2_csi0 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev12 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi0 capture":0 [] - entity 81: ipu2_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video6 pad0: Sink <- "ipu2_csi0":2 [] - entity 87: ipu2_csi1 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev13 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] <- "ipu2_csi1_mux":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi1 capture":0 [] - entity 91: ipu2_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video7 pad0: Sink <- "ipu2_csi1":2 [] - entity 97: ipu1_csi0_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev14 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] <- "tda19971 2-0048":0 [ENABLED] pad2: Source [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] -> "ipu1_csi0":0 [ENABLED] - entity 101: ipu2_csi1_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev15 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:unknown/0x0] <- "adv7180 2-0020":0 [] pad2: Source [fmt:unknown/0x0] -> "ipu2_csi1":0 [] v4l2-compliance test results: - with the following kernel patches: v4l2-subdev: clear reserved fields v4l2-subdev: without controls return -ENOTTY $ v4l2-compliance -u1 v4l2-compliance SHA : b2f8f9049056eb6f9e028927dacb2c715a062df8 Compliance test for device /dev/v4l-subdev1: Media Driver Info: Driver name : imx-media Model : imx-media Serial : Bus info : Media version : 4.15.0 Hardware revision: 0x00000000 (0) Driver version : 4.15.0 Interface Info: ID : 0x0300008f Type : V4L Sub-Device Entity Info: ID : 0x00000003 (3) Name : tda19971 2-0048 Function : WARNING: Unknown Function (00006001), is v4l2-compliance out-of-date? Pad 0x01000004 : Source Link 0x0200006f: to remote pad 0x1000063 of entity 'ipu1_csi0_mux': Data, Enabled Required ioctls: test MC information (see 'Media Driver Info' above): OK Allow for multiple opens: test second /dev/v4l-subdev1 open: OK test for unlimited opens: OK Debug ioctls: test VIDIOC_LOG_STATUS: OK Input ioctls: test VIDIOC_G/S_TUNER/ENUM_FREQ_BANDS: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_S_HW_FREQ_SEEK: OK (Not Supported) test VIDIOC_ENUMAUDIO: OK (Not Supported) test VIDIOC_G/S/ENUMINPUT: OK (Not Supported) test VIDIOC_G/S_AUDIO: OK (Not Supported) Inputs: 0 Audio Inputs: 0 Tuners: 0 Output ioctls: test VIDIOC_G/S_MODULATOR: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_ENUMAUDOUT: OK (Not Supported) test VIDIOC_G/S/ENUMOUTPUT: OK (Not Supported) test VIDIOC_G/S_AUDOUT: OK (Not Supported) Outputs: 0 Audio Outputs: 0 Modulators: 0 Input/Output configuration ioctls: test VIDIOC_ENUM/G/S/QUERY_STD: OK (Not Supported) test VIDIOC_ENUM/G/S/QUERY_DV_TIMINGS: OK test VIDIOC_DV_TIMINGS_CAP: OK test VIDIOC_G/S_EDID: OK Sub-Device ioctls (Source Pad 0): test Try VIDIOC_SUBDEV_ENUM_MBUS_CODE/FRAME_SIZE/FRAME_INTERVAL: OK test Try VIDIOC_SUBDEV_G/S_FMT: OK test Try VIDIOC_SUBDEV_G/S_SELECTION/CROP: OK (Not Supported) test Active VIDIOC_SUBDEV_ENUM_MBUS_CODE/FRAME_SIZE/FRAME_INTERVAL: OK test Active VIDIOC_SUBDEV_G/S_FMT: OK test Active VIDIOC_SUBDEV_G/S_SELECTION/CROP: OK (Not Supported) test VIDIOC_SUBDEV_G/S_FRAME_INTERVAL: OK (Not Supported) Control ioctls: test VIDIOC_QUERY_EXT_CTRL/QUERYMENU: OK test VIDIOC_QUERYCTRL: OK test VIDIOC_G/S_CTRL: OK test VIDIOC_G/S/TRY_EXT_CTRLS: OK test VIDIOC_(UN)SUBSCRIBE_EVENT/DQEVENT: OK test VIDIOC_G/S_JPEGCOMP: OK (Not Supported) Standard Controls: 4 Private Controls: 0 Format ioctls: test VIDIOC_ENUM_FMT/FRAMESIZES/FRAMEINTERVALS: OK (Not Supported) test VIDIOC_G/S_PARM: OK (Not Supported) test VIDIOC_G_FBUF: OK (Not Supported) test VIDIOC_G_FMT: OK (Not Supported) test VIDIOC_TRY_FMT: OK (Not Supported) test VIDIOC_S_FMT: OK (Not Supported) test VIDIOC_G_SLICED_VBI_CAP: OK (Not Supported) test Cropping: OK (Not Supported) test Composing: OK (Not Supported) test Scaling: OK (Not Supported) Codec ioctls: test VIDIOC_(TRY_)ENCODER_CMD: OK (Not Supported) test VIDIOC_G_ENC_INDEX: OK (Not Supported) test VIDIOC_(TRY_)DECODER_CMD: OK (Not Supported) Buffer ioctls: test VIDIOC_REQBUFS/CREATE_BUFS/QUERYBUF: OK (Not Supported) test VIDIOC_EXPBUF: OK (Not Supported) Total: 47, Succeeded: 47, Failed: 0, Warnings: 0 $ v4l2-compliance -M0 v4l2-compliance SHA : b2f8f9049056eb6f9e028927dacb2c715a062df8 Compliance test for device /dev/media0: Media Driver Info: Driver name : imx-media Model : imx-media Serial : Bus info : Media version : 4.15.0 Hardware revision: 0x00000000 (0) Driver version : 4.15.0 Required ioctls: test MEDIA_IOC_DEVICE_INFO: OK Allow for multiple opens: test second /dev/media0 open: OK test MEDIA_IOC_DEVICE_INFO: OK test for unlimited opens: OK Media Controller ioctls: fail: v4l2-test-media.cpp(96): function == MEDIA_ENT_F_V4L2_SUBDEV_UNKNOWN fail: v4l2-test-media.cpp(161): checkFunction(ent.function, true) test MEDIA_IOC_G_TOPOLOGY: FAIL fail: v4l2-test-media.cpp(290): num_data_links != num_links test MEDIA_IOC_ENUM_ENTITIES/LINKS: FAIL test MEDIA_IOC_SETUP_LINK: OK Total: 7, Succeeded: 5, Failed: 2, Warnings: 0 ^^^^ above failures are due to imx/adv7180 not setting a valid entity function Hans Verkuil (1): v4l2-dv-timings: add v4l2_hdmi_colorimetry() Hans Verkuil (1): v4l2-dv-timings: add v4l2_hdmi_colorimetry() Tim Harvey (7): media: v4l-ioctl: fix clearing pad for VIDIOC_DV_TIMIGNS_CAP media: add digital video decoder video interface entity functions MAINTAINERS: add entry for NXP TDA1997x driver media: dt-bindings: Add bindings for TDA1997X media: i2c: Add TDA1997x HDMI receiver driver ARM: dts: imx: Add TDA19971 HDMI Receiver to GW54xx ARM: dts: imx: Add TDA19971 HDMI Receiver to GW551x .../devicetree/bindings/media/i2c/tda1997x.txt | 179 ++ Documentation/media/uapi/mediactl/media-types.rst | 11 + MAINTAINERS | 8 + arch/arm/boot/dts/imx6q-gw54xx.dts | 105 + arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 29 +- arch/arm/boot/dts/imx6qdl-gw551x.dtsi | 138 + drivers/media/i2c/Kconfig | 9 + drivers/media/i2c/Makefile | 1 + drivers/media/i2c/tda1997x.c | 2820 ++++++++++++++++++++ drivers/media/i2c/tda1997x_regs.h | 641 +++++ drivers/media/v4l2-core/v4l2-dv-timings.c | 141 + drivers/media/v4l2-core/v4l2-ioctl.c | 2 +- include/dt-bindings/media/tda1997x.h | 74 + include/media/i2c/tda1997x.h | 42 + include/media/v4l2-dv-timings.h | 21 + include/uapi/linux/media.h | 5 + 16 files changed, 4222 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/media/i2c/tda1997x.txt create mode 100644 drivers/media/i2c/tda1997x.c create mode 100644 drivers/media/i2c/tda1997x_regs.h create mode 100644 include/dt-bindings/media/tda1997x.h create mode 100644 include/media/i2c/tda1997x.h -- 2.7.4

3 11

[alsa-devel] snd_hda_intel: live-reconfiguring codec possible while in use?
by Dominique Bureau 21 Feb '18

21 Feb '18

Hi, I'm trying to figure out if there would be any sane way to perform codec reconfiguration live, without requiring to stop/disconnect all running applications. Currently, sound/pci/hda/hda_codec.c: snd_hda_codec_reset() doesn't allow to be called while the device is in use (returns -EBUSY). I'm looking for some advice before digging further. We're currently in the final phase of a product design and noticed some audio instability when the system is subjected to ESD. The issue seems to be that the codec (Realtek ALC892) falls in an unknown state and stops communicating properly with the audio controller (Intel Sunrise Point-H HD Audio, as reported by lspci). Whenever this happens, all capture device buffers and pointers from in /proc/asound/card0/pcm0c/sub0/status fall to 0 like so: ----------------------- state: RUNNING owner_pid : 1384 trigger_time: 104807.286533210 tstamp : 104807.292766671 delay : 0 avail : 0 avail_max : 0 ----- hw_ptr : 0 appl_ptr : 0 ----------------------- Reloading snd_hda_intel or reconfiguring the codec (using CONFIG_SND_HDA_RECONFIG' /sys/class/sound/hwC0D1/reconfig) brings it back on its feet, but this requires stopping all applications. Kernel is 4.9.61, and we are using jackd version 0.125.0. If anyone has any pointers about this, I'm all ears! Thanks, Dominique Bureau

1 0

[alsa-devel] [RESEND PATCH v2 0/2] ASoC: topology: Add parsing clock gating
by Kirill Marinushkin 21 Feb '18

21 Feb '18

This patch series adds parsing the topology clock gating parameter, which is currently can be set by alsa-lib, but ignored by kernel. The patch 1/2 modifies the UAPI header. If it is applied, the corresponded patch should be applied in alsa-lib [2]. Related, but not required alsa-lib patch: [1]. [1]. [alsa-lib,v2,1/2] topology: Fix parsing config with multiple hw_configs [2]. [alsa-lib,v2,2/2] ASoC: topology: Add defines for snd_soc_tplg_hw_config.clock_gated Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Kirill Marinushkin (2): ASoC: topology: Add defines for snd_soc_tplg_hw_config.clock_gated ASoC: topology: Add missing clock gating parameter when parsing hw_configs include/uapi/sound/asoc.h | 7 ++++++- sound/soc/soc-topology.c | 7 +++++++ 2 files changed, 13 insertions(+), 1 deletion(-) -- 2.13.6

2 4

[alsa-devel] [PATCH] ALSA: x86: hdmi: Add single_port option for compatible behavior
by Takashi Iwai 21 Feb '18

21 Feb '18

The recent support for the multiple PCM devices allowed user to use multiple HDMI/DP outputs, but at the same time, the PCM stream assignment has been changed, too. Due to that, the former PCM#0 (there was only one stream in the past) is likely assigned to a different one (e.g. PCM#2), and it ends up with the regression when user sticks with the fixed configuration using the device#0. Although the multiple monitor support shouldn't matter when user deploys the backend like PulseAudio that checks the jack detection state, the behavior change isn't always acceptable for some users. As a mitigation, this patch introduces an option to switch the behavior back to the old-good-days: when the new option, single_port=1, is passed, the driver creates only a single PCM device, and it's assigned to the first connected one, like the earlier versions did. The option is turned off as default still to support the multiple monitors. Fixes: 8a2d6ae1f737 ("ALSA: x86: Register multiple PCM devices for the LPE audio card") Reported-and-tested-by: Hubert Mantel <mantel(a)metadox.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/x86/intel_hdmi_audio.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index a0951505c7f5..96115c401292 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -50,6 +50,7 @@ /*standard module options for ALSA. This module supports only one card*/ static int hdmi_card_index = SNDRV_DEFAULT_IDX1; static char *hdmi_card_id = SNDRV_DEFAULT_STR1; +static bool single_port; module_param_named(index, hdmi_card_index, int, 0444); MODULE_PARM_DESC(index, @@ -57,6 +58,9 @@ MODULE_PARM_DESC(index, module_param_named(id, hdmi_card_id, charp, 0444); MODULE_PARM_DESC(id, "ID string for INTEL Intel HDMI Audio controller."); +module_param(single_port, bool, 0444); +MODULE_PARM_DESC(single_port, + "Single-port mode (for compatibility)"); /* * ELD SA bits in the CEA Speaker Allocation data block @@ -1579,7 +1583,11 @@ static irqreturn_t display_pipe_interrupt_handler(int irq, void *dev_id) static void notify_audio_lpe(struct platform_device *pdev, int port) { struct snd_intelhad_card *card_ctx = platform_get_drvdata(pdev); - struct snd_intelhad *ctx = &card_ctx->pcm_ctx[port]; + struct snd_intelhad *ctx; + + ctx = &card_ctx->pcm_ctx[single_port ? 0 : port]; + if (single_port) + ctx->port = port; schedule_work(&ctx->hdmi_audio_wq); } @@ -1816,7 +1824,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) init_channel_allocations(); card_ctx->num_pipes = pdata->num_pipes; - card_ctx->num_ports = pdata->num_ports; + card_ctx->num_ports = single_port ? 1 : pdata->num_ports; for_each_port(card_ctx, port) { struct snd_intelhad *ctx = &card_ctx->pcm_ctx[port]; @@ -1824,7 +1832,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) ctx->card_ctx = card_ctx; ctx->dev = card_ctx->dev; - ctx->port = port; + ctx->port = single_port ? -1 : port; ctx->pipe = -1; INIT_WORK(&ctx->hdmi_audio_wq, had_audio_wq); -- 2.16.1

2 2

[alsa-devel] [PATCH v2] ASoC: soc-generic-dmaengine-pcm: Add a DMA debugfs_prefix entry
by Fabio Estevam 21 Feb '18

21 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> After the conversion from platform to component the following warnings are seen: fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory fsl-spdif-dai 2004000.spdif: ASoC: Failed to create component debugfs directory As explained by Mark Brown: "It's a legit warning - we shouldn't really be creating two components for the same device. However this is a bit of a corner case as it's the dmaengine platform driver which is kind of a virtual device, it's not really the device that we use with DMA that's being represented but rather the link between that and the DMA controller." Disambiguate the DMA component name by adding a "dma" debugfs_prefix entry. Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Fixed typo in commit log "createcomponent" sound/soc/soc-generic-dmaengine-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index c07d5c7..768247f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -442,6 +442,9 @@ int snd_dmaengine_pcm_register(struct device *dev, if (!pcm) return -ENOMEM; +#ifdef CONFIG_DEBUG_FS + pcm->component.debugfs_prefix = "dma"; +#endif pcm->config = config; pcm->flags = flags; -- 2.7.4

1 0

[alsa-devel] ASoC: Failed to create component debugfs directory
by Fabio Estevam 21 Feb '18

21 Feb '18

Hi, On linux-next I see the following errors on imx6 wandboard: [ 3.040897] fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory [ 3.065722] fsl-spdif-dai 2004000.spdif: ASoC: Failed to create component debugfs directory I haven't started debugging this yet and not sure if this is something fsl driver specific. Any ideas? Thanks

5 14

[alsa-devel] [PATCH 0/2] ASoC: add support for ROHM BD28623 codec
by Katsuhiro Suzuki 21 Feb '18

21 Feb '18

This patch adds support for ROHM BD28623MUV class D speaker amplifier codec driver. This driver only refers information of HW specification document that can be derivered at website of ROHM. http://www.rohm.com/web/global/products/-/product/BD28623MUV Katsuhiro Suzuki (2): ASoC: add DT bindings documentation for ROHM BD28623 codec ASoC: support ROHM BD28623 codec .../devicetree/bindings/sound/rohm,bd28623.txt | 26 +++ sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/bd28623.c | 258 +++++++++++++++++++++ 4 files changed, 294 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 sound/soc/codecs/bd28623.c -- 2.16.1

2 4

[alsa-devel] [PATCH] ASoC: dmic: don't use codec anymore
by Kuninori Morimoto 21 Feb '18

21 Feb '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 6d6c3946d877 ("ASoC: dmic: replace codec to component") replaced codec to component, but commit 05c9b302eda7 ("ASoC: dmic: Add optional wakeup delay") used codec again. This patch fix it up. Fixes: 05c9b302eda7 ("ASoC: dmic: Add optional wakeup delay") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/dmic.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index bac5bbb..8c4926d 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -37,8 +37,8 @@ struct dmic { static int dmic_aif_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - struct dmic *dmic = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct dmic *dmic = snd_soc_component_get_drvdata(component); switch (event) { case SND_SOC_DAPM_POST_PMU: -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: arizona: Fixup some minor formatting issues
by Charles Keepax 21 Feb '18

21 Feb '18

Repair any formatting/style issues that can be fixed without major code refactoring. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/arizona.c | 59 ++++++++++++++++++++++++------------------- sound/soc/codecs/cs47l24.c | 22 ++++++++-------- sound/soc/codecs/wm5102.c | 20 +++++++-------- sound/soc/codecs/wm5110.c | 62 +++++++++++++++++++++++++--------------------- sound/soc/codecs/wm8997.c | 14 +++++------ sound/soc/codecs/wm8998.c | 33 +++++++++++++----------- 6 files changed, 113 insertions(+), 97 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ffcfe8ea1848..5727ea079ad7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -90,7 +90,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - val = snd_soc_component_read32(component, ARIZONA_INTERRUPT_RAW_STATUS_3); + val = snd_soc_component_read32(component, + ARIZONA_INTERRUPT_RAW_STATUS_3); if (val & ARIZONA_SPK_OVERHEAT_STS) { dev_crit(arizona->dev, "Speaker not enabled due to temperature\n"); @@ -641,7 +642,6 @@ const unsigned int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { }; EXPORT_SYMBOL_GPL(arizona_rate_val); - const struct soc_enum arizona_isrc_fsh[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_1, ARIZONA_ISRC1_FSH_SHIFT, 0xf, @@ -923,7 +923,8 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, priv->in_pending++; break; case SND_SOC_DAPM_POST_PMU: - snd_soc_component_update_bits(component, reg, ARIZONA_IN1L_MUTE, 0); + snd_soc_component_update_bits(component, reg, + ARIZONA_IN1L_MUTE, 0); /* If this is the last input pending then allow VU */ priv->in_pending--; @@ -1070,9 +1071,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_out_ev); -int arizona_hp_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) +int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, + int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct arizona_priv *priv = snd_soc_component_get_drvdata(component); @@ -1275,8 +1275,8 @@ static unsigned int arizona_opclk_ref_44k1_rates[] = { 45158400, }; -static int arizona_set_opclk(struct snd_soc_component *component, unsigned int clk, - unsigned int freq) +static int arizona_set_opclk(struct snd_soc_component *component, + unsigned int clk, unsigned int freq) { struct arizona_priv *priv = snd_soc_component_get_drvdata(component); unsigned int reg; @@ -1302,7 +1302,7 @@ static int arizona_set_opclk(struct snd_soc_component *component, unsigned int c rates = arizona_opclk_ref_48k_rates; for (ref = 0; ref < ARRAY_SIZE(arizona_opclk_ref_48k_rates) && - rates[ref] <= refclk; ref++) { + rates[ref] <= refclk; ref++) { div = 1; while (rates[ref] / div >= freq && div < 32) { if (rates[ref] / div == freq) { @@ -1727,18 +1727,23 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, } snd_soc_component_update_bits(component, ARIZONA_SAMPLE_RATE_1, - ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + ARIZONA_SAMPLE_RATE_1_MASK, + sr_val); if (base) - snd_soc_component_update_bits(component, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 0); + snd_soc_component_update_bits(component, + base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); break; case ARIZONA_CLK_ASYNCCLK: - snd_soc_component_update_bits(component, ARIZONA_ASYNC_SAMPLE_RATE_1, - ARIZONA_ASYNC_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_component_update_bits(component, + ARIZONA_ASYNC_SAMPLE_RATE_1, + ARIZONA_ASYNC_SAMPLE_RATE_1_MASK, + sr_val); if (base) - snd_soc_component_update_bits(component, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, - 8 << ARIZONA_AIF1_RATE_SHIFT); + snd_soc_component_update_bits(component, + base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); @@ -1849,9 +1854,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, base + ARIZONA_AIF_RX_ENABLES); /* Disable AIF TX/RX before reconfiguring it */ regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_TX_ENABLES, 0xff, 0x0); + base + ARIZONA_AIF_TX_ENABLES, + 0xff, 0x0); regmap_update_bits(arizona->regmap, - base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0); + base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0); } ret = arizona_hw_params_rate(substream, params, dai); @@ -1960,8 +1966,9 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) else reg = 0; - return snd_soc_component_update_bits(component, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_TRI, reg); + return snd_soc_component_update_bits(component, + base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_TRI, reg); } static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, @@ -2319,7 +2326,6 @@ static int arizona_calc_fll(struct arizona_fll *fll, arizona_fll_dbg(fll, "GAIN=0x%x(%d)\n", cfg->gain, 1 << cfg->gain); return 0; - } static void arizona_apply_fll(struct arizona *arizona, unsigned int base, @@ -2565,9 +2571,8 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; - if (fll->fout && Fref > 0) { + if (fll->fout && Fref > 0) ret = arizona_enable_fll(fll); - } return ret; } @@ -2656,7 +2661,8 @@ EXPORT_SYMBOL_GPL(arizona_init_fll); * Most systems have a single static configuration and should use * platform data instead. */ -int arizona_set_output_mode(struct snd_soc_component *component, int output, bool diff) +int arizona_set_output_mode(struct snd_soc_component *component, int output, + bool diff) { unsigned int reg, val; @@ -2670,7 +2676,8 @@ int arizona_set_output_mode(struct snd_soc_component *component, int output, boo else val = 0; - return snd_soc_component_update_bits(component, reg, ARIZONA_OUT1_MONO, val); + return snd_soc_component_update_bits(component, reg, + ARIZONA_OUT1_MONO, val); } EXPORT_SYMBOL_GPL(arizona_set_output_mode); diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index c93d3f7e8e36..196e9c343aeb 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -196,14 +196,13 @@ SOC_SINGLE("HPOUT1 SC Protect Switch", ARIZONA_HP1_SHORT_CIRCUIT_CTRL, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, - ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, - ARIZONA_OUT4L_VOL_SHIFT, - 0xbf, 0, digital_tlv), + ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), @@ -494,8 +493,7 @@ SND_SOC_DAPM_PGA("ISRC3DEC4", ARIZONA_ISRC_3_CTRL_3, ARIZONA_ISRC3_DEC3_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &cs47l24_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &cs47l24_aec_loopback_mux), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), @@ -931,8 +929,8 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = { { "DSP3 Voice Trigger", "Switch", "DSP3" }, }; -static int cs47l24_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int cs47l24_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct cs47l24_priv *cs47l24 = snd_soc_component_get_drvdata(component); @@ -1144,7 +1142,8 @@ static int cs47l24_component_probe(struct snd_soc_component *component) goto err_adsp2_codec_probe; ret = snd_soc_add_component_controls(component, - &arizona_adsp2_rate_controls[1], 2); + &arizona_adsp2_rate_controls[1], + 2); if (ret) goto err_adsp2_codec_probe; @@ -1214,7 +1213,7 @@ static int cs47l24_probe(struct platform_device *pdev) BUILD_BUG_ON(ARRAY_SIZE(cs47l24_dai) > ARIZONA_MAX_DAI); cs47l24 = devm_kzalloc(&pdev->dev, sizeof(struct cs47l24_priv), - GFP_KERNEL); + GFP_KERNEL); if (!cs47l24) return -ENOMEM; @@ -1294,8 +1293,9 @@ static int cs47l24_probe(struct platform_device *pdev) goto err_dsp_irq; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_cs47l24, - cs47l24_dai, ARRAY_SIZE(cs47l24_dai)); + &soc_component_dev_cs47l24, + cs47l24_dai, + ARRAY_SIZE(cs47l24_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 24f19cfc98f5..1ac83388d1b8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -620,7 +620,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, } static int wm5102_adsp_power_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct arizona *arizona = dev_get_drvdata(component->dev->parent); @@ -717,7 +717,7 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, return 0; } -static const char *wm5102_osr_text[] = { +static const char * const wm5102_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -1062,7 +1062,7 @@ ARIZONA_MIXER_ENUMS(DSP1R, ARIZONA_DSP1RMIX_INPUT_1_SOURCE); ARIZONA_DSP_AUX_ENUMS(DSP1, ARIZONA_DSP1AUX1MIX_INPUT_1_SOURCE); -static const char *wm5102_aec_loopback_texts[] = { +static const char * const wm5102_aec_loopback_texts[] = { "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "EPOUT", "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", }; @@ -1317,8 +1317,7 @@ SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, @@ -1736,8 +1735,8 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "DRC1 Signal Activity", NULL, "DRC1R" }, }; -static int wm5102_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int wm5102_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct wm5102_priv *wm5102 = snd_soc_component_get_drvdata(component); @@ -1947,7 +1946,7 @@ static int wm5102_component_probe(struct snd_soc_component *component) return ret; ret = snd_soc_add_component_controls(component, - arizona_adsp2_rate_controls, 1); + arizona_adsp2_rate_controls, 1); if (ret) goto err_adsp2_codec_probe; @@ -2105,8 +2104,9 @@ static int wm5102_probe(struct platform_device *pdev) goto err_dsp_irq; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_wm5102, - wm5102_dai, ARRAY_SIZE(wm5102_dai)); + &soc_component_dev_wm5102, + wm5102_dai, + ARRAY_SIZE(wm5102_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b33cbd0161d3..fb9835dcd836 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -332,19 +332,25 @@ static int wm5110_hp_pre_disable(struct snd_soc_dapm_widget *w) switch (w->shift) { case ARIZONA_OUT1L_ENA_SHIFT: if (!(val & ARIZONA_DRE1L_ENA_MASK)) { - snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, - ARIZONA_WS_TRG1, ARIZONA_WS_TRG1); - snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, - ARIZONA_WS_TRG1, 0); + snd_soc_component_update_bits(component, + ARIZONA_SPARE_TRIGGERS, + ARIZONA_WS_TRG1, + ARIZONA_WS_TRG1); + snd_soc_component_update_bits(component, + ARIZONA_SPARE_TRIGGERS, + ARIZONA_WS_TRG1, 0); priv->out_down_delay += 27; } break; case ARIZONA_OUT1R_ENA_SHIFT: if (!(val & ARIZONA_DRE1R_ENA_MASK)) { - snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, - ARIZONA_WS_TRG2, ARIZONA_WS_TRG2); - snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, - ARIZONA_WS_TRG2, 0); + snd_soc_component_update_bits(component, + ARIZONA_SPARE_TRIGGERS, + ARIZONA_WS_TRG2, + ARIZONA_WS_TRG2); + snd_soc_component_update_bits(component, + ARIZONA_SPARE_TRIGGERS, + ARIZONA_WS_TRG2, 0); priv->out_down_delay += 27; } break; @@ -539,7 +545,7 @@ static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: snd_soc_component_update_bits(component, reg, mask, - wm5110->in_pga_cache[w->shift]); + wm5110->in_pga_cache[w->shift]); wm5110->in_post_pending--; if (wm5110->in_post_pending == 0) @@ -863,14 +869,14 @@ SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, ARIZONA_SPK2R_MUTE_SHIFT, 1, 1), SOC_DOUBLE_EXT("HPOUT1 DRE Switch", ARIZONA_DRE_ENABLE, - ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0, - snd_soc_get_volsw, wm5110_put_dre), + ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0, + snd_soc_get_volsw, wm5110_put_dre), SOC_DOUBLE_EXT("HPOUT2 DRE Switch", ARIZONA_DRE_ENABLE, - ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0, - snd_soc_get_volsw, wm5110_put_dre), + ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0, + snd_soc_get_volsw, wm5110_put_dre), SOC_DOUBLE_EXT("HPOUT3 DRE Switch", ARIZONA_DRE_ENABLE, - ARIZONA_DRE3L_ENA_SHIFT, ARIZONA_DRE3R_ENA_SHIFT, 1, 0, - snd_soc_get_volsw, wm5110_put_dre), + ARIZONA_DRE3L_ENA_SHIFT, ARIZONA_DRE3R_ENA_SHIFT, 1, 0, + snd_soc_get_volsw, wm5110_put_dre), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), @@ -1036,7 +1042,7 @@ ARIZONA_MUX_ENUMS(ISRC3DEC2, ARIZONA_ISRC3DEC2MIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ISRC3DEC3, ARIZONA_ISRC3DEC3MIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ISRC3DEC4, ARIZONA_ISRC3DEC4MIX_INPUT_1_SOURCE); -static const char *wm5110_aec_loopback_texts[] = { +static const char * const wm5110_aec_loopback_texts[] = { "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "HPOUT3L", "HPOUT3R", "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", "SPKDAT2L", "SPKDAT2R", }; @@ -1274,18 +1280,17 @@ SND_SOC_DAPM_PGA("ISRC3DEC4", ARIZONA_ISRC_3_CTRL_3, ARIZONA_ISRC3_DEC3_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm5110_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm5110_aec_loopback_mux), SND_SOC_DAPM_SUPPLY("RXANC NG External Clock", SND_SOC_NOPM, - ARIZONA_EXT_NG_SEL_SET_SHIFT, 0, arizona_anc_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + ARIZONA_EXT_NG_SEL_SET_SHIFT, 0, arizona_anc_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("RXANCL NG External", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("RXANCR NG External", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("RXANC NG Clock", SND_SOC_NOPM, - ARIZONA_CLK_NG_ENA_SET_SHIFT, 0, arizona_anc_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + ARIZONA_CLK_NG_ENA_SET_SHIFT, 0, arizona_anc_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("RXANCL NG Internal", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("RXANCR NG Internal", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -2034,8 +2039,8 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "DSP3 Voice Trigger", "Switch", "DSP3" }, }; -static int wm5110_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int wm5110_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct wm5110_priv *wm5110 = snd_soc_component_get_drvdata(component); @@ -2302,8 +2307,8 @@ static int wm5110_component_probe(struct snd_soc_component *component) } ret = snd_soc_add_component_controls(component, - arizona_adsp2_rate_controls, - WM5110_NUM_ADSP); + arizona_adsp2_rate_controls, + WM5110_NUM_ADSP); if (ret) goto err_adsp2_codec_probe; @@ -2460,8 +2465,9 @@ static int wm5110_probe(struct platform_device *pdev) goto err_dsp_irq; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_wm5110, - wm5110_dai, ARRAY_SIZE(wm5110_dai)); + &soc_component_dev_wm5110, + wm5110_dai, + ARRAY_SIZE(wm5110_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 45ee1e6bf715..df5b36b8fc5a 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -118,7 +118,7 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, return arizona_dvfs_sysclk_ev(w, kcontrol, event); } -static const char *wm8997_osr_text[] = { +static const char * const wm8997_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -609,8 +609,7 @@ SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, ARIZONA_SLIMRX8_ENA_SHIFT, 0), SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm8997_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm8997_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, @@ -927,8 +926,8 @@ static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { { "MICSUPP", NULL, "SYSCLK" }, }; -static int wm8997_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int wm8997_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct wm8997_priv *wm8997 = snd_soc_component_get_drvdata(component); @@ -1173,8 +1172,9 @@ static int wm8997_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_wm8997, - wm8997_dai, ARRAY_SIZE(wm8997_dai)); + &soc_component_dev_wm8997, + wm8997_dai, + ARRAY_SIZE(wm8997_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 6f3fdd7b095e..61294c787f27 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -137,11 +137,13 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, if (inmode & ARIZONA_INMODE_SE) src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT; - snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); + snd_soc_component_update_bits(component, mode_reg, + ARIZONA_IN1_MODE_MASK, mode_val); snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | ARIZONA_IN1L_SRC_SE_MASK, - src_val); + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); return snd_soc_dapm_mux_update_power(dapm, kcontrol, ucontrol->value.enumerated.item[0], @@ -322,7 +324,7 @@ SOC_DOUBLE_R("HPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, SOC_DOUBLE_R("LINEOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, - ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("SPKDAT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, @@ -335,7 +337,7 @@ SOC_DOUBLE_R_TLV("LINEOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, - ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), @@ -615,12 +617,12 @@ SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_MUX("AEC1 Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm8998_aec_loopback_mux[0]), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8998_aec_loopback_mux[0]), SND_SOC_DAPM_MUX("AEC2 Loopback", ARIZONA_DAC_AEC_CONTROL_2, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm8998_aec_loopback_mux[1]), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8998_aec_loopback_mux[1]), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), @@ -739,9 +741,9 @@ SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA("SPD1TX1", ARIZONA_SPD1_TX_CONTROL, - ARIZONA_SPD1_VAL1_SHIFT, 0, NULL, 0), + ARIZONA_SPD1_VAL1_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("SPD1TX2", ARIZONA_SPD1_TX_CONTROL, - ARIZONA_SPD1_VAL2_SHIFT, 0, NULL, 0), + ARIZONA_SPD1_VAL2_SHIFT, 0, NULL, 0), SND_SOC_DAPM_OUT_DRV("SPD1", ARIZONA_SPD1_TX_CONTROL, ARIZONA_SPD1_ENA_SHIFT, 0, NULL, 0), @@ -1249,8 +1251,8 @@ static struct snd_soc_dai_driver wm8998_dai[] = { }, }; -static int wm8998_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int wm8998_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct wm8998_priv *wm8998 = snd_soc_component_get_drvdata(component); @@ -1379,8 +1381,9 @@ static int wm8998_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_wm8998, - wm8998_dai, ARRAY_SIZE(wm8998_dai)); + &soc_component_dev_wm8998, + wm8998_dai, + ARRAY_SIZE(wm8998_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; -- 2.11.0

2 1

[alsa-devel] audio CODEC(CS42888) test
by Muni Sekhar 21 Feb '18

21 Feb '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, I’m having a sound hardware which contains an analogue audio CODEC(CS42888) and configuring the CODEC is performed over SPI bus. I noticed that linux kernel already had a driver(snd-soc-cs42xx8.ko) for this, I planned to use this driver. Basically I’d like to perform loopback test and compare the sent and received files to verify data fidelity. Also the test to be repeated for various data rates and data formats. I’d like to know how to test this loopback scenario and compare the sent\received files? I appreciate any help. -- Thanks, Sekhar

1 0

[alsa-devel] [PATCH] drm/i915: Remove unused IRQ chip data of HDMI LPE audio
by Augustine.Chen 20 Feb '18

20 Feb '18

The chip data of HDMI LPE audio is set to drm_i915_private which is not consistent with the expectation by x86 APIC driver. In the case of not enabling CONFIG_CPUMASK_OFFSTACK, this would cause kernel panic while doing CPU hotplug. Since the dependency of IRQ chip data was removed from HDMI LPE audio by Commit 9bd9590997b92fbd79fd028f704f6c584b4439d7 ("drm/i915: Stop pretending to mask/unmask LPE audio interrupts"), remove the code of setting IRQ chip data to resolve this issue. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=103731 Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> Cc: Jerome Anand <jerome.anand(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Augustine.Chen <augustine.chen(a)intel.com> --- drivers/gpu/drm/i915/intel_lpe_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_lpe_audio.c b/drivers/gpu/drm/i915/intel_lpe_audio.c index 3bf6528..56176f9 100644 --- a/drivers/gpu/drm/i915/intel_lpe_audio.c +++ b/drivers/gpu/drm/i915/intel_lpe_audio.c @@ -176,7 +176,7 @@ static int lpe_audio_irq_init(struct drm_i915_private *dev_priv) handle_simple_irq, "hdmi_lpe_audio_irq_handler"); - return irq_set_chip_data(irq, dev_priv); + return 0; } static bool lpe_audio_detect(struct drm_i915_private *dev_priv) -- 1.9.1

6 15

[alsa-devel] [PATCH, alsa-lib 1/2] topology: Fix parsing config with multiple hw_configs
by Kirill Marinushkin 20 Feb '18

20 Feb '18

Currently, if the config file includes several hw_configs sections, parse_hw_config_refs() returns after parsing only the first section. For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is parsed incorrectly: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. bclk "master" # Platform is master of bit clock fsync "master" # platform is master of fsync } SectionHWConfig."CodecHWConfig2" { id "2" format "AC97" } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" "CodecHWConfig2" ] default_hw_conf_id "2" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: patch(a)alsa-project.org --- src/topology/pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 58cee31d..d3836677 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -882,6 +882,7 @@ static int parse_hw_config_refs(snd_tplg_t *tplg, snd_config_t *cfg, /* refer to a list of HW configs */ snd_config_for_each(i, next, cfg) { const char *val; + int err; n = snd_config_iterator_entry(i); if (snd_config_get_string(n, &val) < 0) @@ -893,7 +894,9 @@ static int parse_hw_config_refs(snd_tplg_t *tplg, snd_config_t *cfg, } link->num_hw_configs++; - return tplg_ref_add(elem, SND_TPLG_TYPE_HW_CONFIG, val); + err = tplg_ref_add(elem, SND_TPLG_TYPE_HW_CONFIG, val); + if (err < 0) + return err; } return 0; -- 2.13.6

2 6

[alsa-devel] [PATCH 1/2] ASoC: topology: Rename clock_gated to clock_cont in snd_soc_tplg_hw_config
by Kirill Marinushkin 20 Feb '18

20 Feb '18

In kernel `soc-dai.h`, DAI clock gating is defined as following: ~~~~ \#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ \#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ ~~~~ Therefore, the corresponding field of struct snd_soc_tplg_hw_config should be inverted compared to the current logic: clock_count = 1 => SND_SOC_DAIFMT_CONT clock_count = 0 => SND_SOC_DAIFMT_GATED Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- include/uapi/sound/asoc.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 69c37ecbff7e..10188850dede 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -312,7 +312,9 @@ struct snd_soc_tplg_hw_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - - used to match */ __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ - __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 clock_cont; /* 1 if clock is continuous, and can not be + * gated to save power + */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ -- 2.13.6

3 10

[alsa-devel] [PATCH] ALSA: hda-beep: add SPDX identifiers
by Matt Ranostay 20 Feb '18

20 Feb '18

Add SPDX GPLv2.0+ identifiers and update authors email Signed-off-by: Matt Ranostay <matt.ranostay(a)konsulko.com> --- sound/pci/hda/hda_beep.c | 17 ++--------------- sound/pci/hda/hda_beep.h | 17 ++--------------- 2 files changed, 4 insertions(+), 30 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eac..066b5b59c4d7 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -1,22 +1,9 @@ +// SPDX-License-Identifier: GPL-2.0+ /* * Digital Beep Input Interface for HD-audio codec * - * Author: Matt Ranostay <mranostay(a)gmail.com> + * Author: Matt Ranostay <matt.ranostay(a)konsulko.com> * Copyright (c) 2008 Embedded Alley Solutions Inc - * - * This driver is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This driver is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include <linux/input.h> diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 1052ad380e97..d1a6a9c1329a 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -1,22 +1,9 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ /* * Digital Beep Input Interface for HD-audio codec * - * Author: Matt Ranostay <mranostay(a)gmail.com> + * Author: Matt Ranostay <matt.ranostay(a)konsulko.com> * Copyright (c) 2008 Embedded Alley Solutions Inc - * - * This driver is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This driver is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #ifndef __SOUND_HDA_BEEP_H -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: codecs: ak5558: Fix style for SPDX identifier
by Daniel Baluta 20 Feb '18

20 Feb '18

Use // for .c and /* for .h files according to: Documentation/process/license-rules.rst file. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/codecs/ak5558.c | 11 +++++------ sound/soc/codecs/ak5558.h | 4 ++-- 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index f224831..f4ed5cc 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -1,10 +1,9 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * Audio driver for AK5558 ADC - * - * Copyright (C) 2015 Asahi Kasei Microdevices Corporation - * Copyright 2018 NXP - */ +// +// Audio driver for AK5558 ADC +// +// Copyright (C) 2015 Asahi Kasei Microdevices Corporation +// Copyright 2018 NXP #include <linux/delay.h> #include <linux/gpio/consumer.h> diff --git a/sound/soc/codecs/ak5558.h b/sound/soc/codecs/ak5558.h index 7c9e373..6105908 100644 --- a/sound/soc/codecs/ak5558.h +++ b/sound/soc/codecs/ak5558.h @@ -1,5 +1,5 @@ -// SPDX-License-Identifier: GPL-2.0 -/* +/* SPDX-License-Identifier: GPL-2.0 + * * Audio driver header for AK5558 * * Copyright (C) 2016 Asahi Kasei Microdevices Corporation -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: hdmi-codec: Fix module unloading caused kernel crash
by Peter Ujfalusi 20 Feb '18

20 Feb '18

The hcp->chmap_info must not be freed up in the hdmi_codec_remove() function as it leads to kernel crash due ALSA core's pcm_chmap_ctl_private_free() is trying to free it up again when the card destroyed via snd_card_free. Since the driver was converted to devm_snd_soc_register_component() the whole platform_device.remove can be removed. Commit cd6111b26280a ("ASoC: hdmi-codec: add channel mapping control") should not have added the kfree(hcp->chmap_info); to the hdmi_codec_remove function. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/hdmi-codec.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 42df579cb507..6fa11888672d 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -798,23 +798,11 @@ static int hdmi_codec_probe(struct platform_device *pdev) return 0; } -static int hdmi_codec_remove(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - struct hdmi_codec_priv *hcp; - - hcp = dev_get_drvdata(dev); - kfree(hcp->chmap_info); - - return 0; -} - static struct platform_driver hdmi_codec_driver = { .driver = { .name = HDMI_CODEC_DRV_NAME, }, .probe = hdmi_codec_probe, - .remove = hdmi_codec_remove, }; module_platform_driver(hdmi_codec_driver); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 3

[alsa-devel] [PATCH v2] ASoC: hdmi-codec: Fix module unloading caused kernel crash
by Peter Ujfalusi 20 Feb '18

20 Feb '18

The hcp->chmap_info must not be freed up in the hdmi_codec_remove() function as it leads to kernel crash due ALSA core's pcm_chmap_ctl_private_free() is trying to free it up again when the card destroyed via snd_card_free. Commit cd6111b26280a ("ASoC: hdmi-codec: add channel mapping control") should not have added the kfree(hcp->chmap_info); to the hdmi_codec_remove function. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reviewed-by: Jyri Sarha <jsarha(a)ti.com> Tested-by: Jyri Sarha <jsarha(a)ti.com> --- Hi, Changes since v1: - Rebased on mainline (4.16-rc2) Mainline does not have devm_ version of snd_soc_register_codec(), the hdmi_codec_remove callback can not be removed. Regards, Peter sound/soc/codecs/hdmi-codec.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 5672e516bec3..c1830ccd3bb8 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -798,12 +798,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) static int hdmi_codec_remove(struct platform_device *pdev) { - struct device *dev = &pdev->dev; - struct hdmi_codec_priv *hcp; - - hcp = dev_get_drvdata(dev); - kfree(hcp->chmap_info); - snd_soc_unregister_codec(dev); + snd_soc_unregister_codec(&pdev->dev); return 0; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [asoc:topic/topology 1/1] sound/soc/soc-topology.c:1985:33: error: 'SND_SOC_TPLG_DAI_CLK_GATE_GATED' undeclared; did you mean 'SND_SOC_TPLG_DAPM_AIF_OUT'?
by kbuild test robot 20 Feb '18

20 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/topology head: bdb86c40e8c5d2abe2ec7be964d621c1b7c27081 commit: bdb86c40e8c5d2abe2ec7be964d621c1b7c27081 [1/1] ASoC: topology: Add missing clock gating parameter when parsing hw_configs config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout bdb86c40e8c5d2abe2ec7be964d621c1b7c27081 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/soc-topology.c: In function 'set_link_hw_format': >> sound/soc/soc-topology.c:1985:33: error: 'SND_SOC_TPLG_DAI_CLK_GATE_GATED' undeclared (first use in this function); did you mean 'SND_SOC_TPLG_DAPM_AIF_OUT'? if (hw_config->clock_gated == SND_SOC_TPLG_DAI_CLK_GATE_GATED) ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ SND_SOC_TPLG_DAPM_AIF_OUT sound/soc/soc-topology.c:1985:33: note: each undeclared identifier is reported only once for each function it appears in >> sound/soc/soc-topology.c:1988:5: error: 'SND_SOC_TPLG_DAI_CLK_GATE_CONT' undeclared (first use in this function); did you mean 'SND_SOC_TPLG_DAI_CLK_GATE_GATED'? SND_SOC_TPLG_DAI_CLK_GATE_CONT) ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ SND_SOC_TPLG_DAI_CLK_GATE_GATED vim +1985 sound/soc/soc-topology.c 1959 1960 /** 1961 * set_link_hw_format - Set the HW audio format of the physical DAI link. 1962 * @link: &snd_soc_dai_link which should be updated 1963 * @cfg: physical link configs. 1964 * 1965 * Topology context contains a list of supported HW formats (configs) and 1966 * a default format ID for the physical link. This function will use this 1967 * default ID to choose the HW format to set the link's DAI format for init. 1968 */ 1969 static void set_link_hw_format(struct snd_soc_dai_link *link, 1970 struct snd_soc_tplg_link_config *cfg) 1971 { 1972 struct snd_soc_tplg_hw_config *hw_config; 1973 unsigned char bclk_master, fsync_master; 1974 unsigned char invert_bclk, invert_fsync; 1975 int i; 1976 1977 for (i = 0; i < cfg->num_hw_configs; i++) { 1978 hw_config = &cfg->hw_config[i]; 1979 if (hw_config->id != cfg->default_hw_config_id) 1980 continue; 1981 1982 link->dai_fmt = hw_config->fmt & SND_SOC_DAIFMT_FORMAT_MASK; 1983 1984 /* clock gating */ > 1985 if (hw_config->clock_gated == SND_SOC_TPLG_DAI_CLK_GATE_GATED) 1986 link->dai_fmt |= SND_SOC_DAIFMT_GATED; 1987 else if (hw_config->clock_gated == > 1988 SND_SOC_TPLG_DAI_CLK_GATE_CONT) 1989 link->dai_fmt |= SND_SOC_DAIFMT_CONT; 1990 1991 /* clock signal polarity */ 1992 invert_bclk = hw_config->invert_bclk; 1993 invert_fsync = hw_config->invert_fsync; 1994 if (!invert_bclk && !invert_fsync) 1995 link->dai_fmt |= SND_SOC_DAIFMT_NB_NF; 1996 else if (!invert_bclk && invert_fsync) 1997 link->dai_fmt |= SND_SOC_DAIFMT_NB_IF; 1998 else if (invert_bclk && !invert_fsync) 1999 link->dai_fmt |= SND_SOC_DAIFMT_IB_NF; 2000 else 2001 link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; 2002 2003 /* clock masters */ 2004 bclk_master = hw_config->bclk_master; 2005 fsync_master = hw_config->fsync_master; 2006 if (!bclk_master && !fsync_master) 2007 link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 2008 else if (bclk_master && !fsync_master) 2009 link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; 2010 else if (!bclk_master && fsync_master) 2011 link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; 2012 else 2013 link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; 2014 } 2015 } 2016 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 0/2] Fix coding style issues in AK4458 DAC
by Cosmin-Gabriel Samoila 20 Feb '18

20 Feb '18

Modify SPDX license comment style and sort headers alphabetically. Cosmin-Gabriel Samoila (2): ASoC: ak4458: Modify SPDX license format ASoC: ak4458: Sort headers alphabetically sound/soc/codecs/ak4458.c | 23 +++++++++++------------ sound/soc/codecs/ak4458.h | 2 +- 2 files changed, 12 insertions(+), 13 deletions(-) -- 2.7.4

2 4

[alsa-devel] Applied "ASoC: Add DT bindings documentation for max9759 amplifier" to the asoc tree
by Mark Brown 20 Feb '18

20 Feb '18

The patch ASoC: Add DT bindings documentation for max9759 amplifier has been applied to the asoc tree at https://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 01182d6529ae499bb73129f4766b45c2ae8e700b Mon Sep 17 00:00:00 2001 From: Neil Armstrong <narmstrong(a)baylibre.com> Date: Mon, 19 Feb 2018 14:45:57 +0100 Subject: [PATCH] ASoC: Add DT bindings documentation for max9759 amplifier This patch adds DT bindings documentation for Maxim MAX9759 speaker amplifier. Signed-off-by: Neil Armstrong <narmstrong(a)baylibre.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/maxim,max9759.txt | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max9759.txt diff --git a/Documentation/devicetree/bindings/sound/maxim,max9759.txt b/Documentation/devicetree/bindings/sound/maxim,max9759.txt new file mode 100644 index 000000000000..737a996374d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max9759.txt @@ -0,0 +1,18 @@ +Maxim MAX9759 Speaker Amplifier +=============================== + +Required properties: +- compatible : "maxim,max9759" +- shutdown-gpios : the gpio connected to the shutdown pin +- mute-gpios : the gpio connected to the mute pin +- gain-gpios : the 2 gpios connected to the g1 and g2 pins + +Example: + +max9759: analog-amplifier { + compatible = "maxim,max9759"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio3 19 GPIO_ACTIVE_LOW>; + gain-gpios = <&gpio3 23 GPIO_ACTIVE_LOW>, + <&gpio3 25 GPIO_ACTIVE_LOW>; +}; -- 2.16.1

1 0

[alsa-devel] Applied "ASoC: max9759: Add Amplifier Driver" to the asoc tree
by Mark Brown 20 Feb '18

20 Feb '18

The patch ASoC: max9759: Add Amplifier Driver has been applied to the asoc tree at https://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 fa8d915172b8c10ec0734c4021e99e9705023b07 Mon Sep 17 00:00:00 2001 From: Neil Armstrong <narmstrong(a)baylibre.com> Date: Mon, 19 Feb 2018 14:45:58 +0100 Subject: [PATCH] ASoC: max9759: Add Amplifier Driver The max9759 is a gpio controlled amplifier. Tested on a Variscite Dart MX6 SoM based custom board. Signed-off-by: Neil Armstrong <narmstrong(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 5 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9759.c | 207 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 214 insertions(+) create mode 100644 sound/soc/codecs/max9759.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2b331f7266ab..f9ea4603a903 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98373 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9860 if I2C + select SND_SOC_MAX9759 select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX @@ -1187,6 +1188,10 @@ config SND_SOC_ZX_AUD96P22 config SND_SOC_LM4857 tristate +config SND_SOC_MAX9759 + tristate "Maxim MAX9759 speaker Amplifier" + select GPIOLIB + config SND_SOC_MAX9768 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index da1571336f1e..af4238d90c99 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -80,6 +80,7 @@ snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o +snd-soc-max9759-objs := max9759.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o @@ -325,6 +326,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o +obj-$(CONFIG_SND_SOC_MAX9759) += snd-soc-max9759.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o diff --git a/sound/soc/codecs/max9759.c b/sound/soc/codecs/max9759.c new file mode 100644 index 000000000000..ecfb4a80424b --- /dev/null +++ b/sound/soc/codecs/max9759.c @@ -0,0 +1,207 @@ +// SPDX-Licence-Identifier: GPL-2.0 +/* + * MAX9759 Amplifier Driver + * + * Copyright (c) 2017 BayLibre, SAS. + * Author: Neil Armstrong <narmstrong(a)baylibre.com> + */ + +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#define DRV_NAME "max9759" + +struct max9759 { + struct gpio_desc *gpiod_shutdown; + struct gpio_desc *gpiod_mute; + struct gpio_descs *gpiod_gain; + bool is_mute; + unsigned int gain; +}; + +static int pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->gpiod_shutdown, 0); + else + gpiod_set_value_cansleep(priv->gpiod_shutdown, 1); + + return 0; +} + +/* From 6dB to 24dB in steps of 6dB */ +static const DECLARE_TLV_DB_SCALE(speaker_gain_tlv, 600, 600, 0); + +static int speaker_gain_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + ucontrol->value.integer.value[0] = priv->gain; + + return 0; +} + +static const bool speaker_gain_table[4][2] = { + /* G1, G2 */ + {true, true}, /* +6dB */ + {false, true}, /* +12dB */ + {true, false}, /* +18dB */ + {false, false}, /* +24dB */ +}; + +static int speaker_gain_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + if (ucontrol->value.integer.value[0] > 3) + return -EINVAL; + + priv->gain = ucontrol->value.integer.value[0]; + + /* G1 */ + gpiod_set_value_cansleep(priv->gpiod_gain->desc[0], + speaker_gain_table[priv->gain][0]); + /* G2 */ + gpiod_set_value_cansleep(priv->gpiod_gain->desc[1], + speaker_gain_table[priv->gain][1]); + + return 1; +} + +static int speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + ucontrol->value.integer.value[0] = !priv->is_mute; + + return 0; +} + +static int speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + priv->is_mute = !ucontrol->value.integer.value[0]; + + gpiod_set_value_cansleep(priv->gpiod_mute, priv->is_mute); + + return 1; +} + +static const struct snd_kcontrol_new max9759_dapm_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Gain Volume", 0, 0, 3, 0, + speaker_gain_control_get, speaker_gain_control_put, + speaker_gain_tlv), + SOC_SINGLE_BOOL_EXT("Playback Switch", 0, + speaker_mute_get, speaker_mute_put), +}; + +static const struct snd_soc_dapm_widget max9759_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), + SND_SOC_DAPM_PGA_E("PGA", SND_SOC_NOPM, 0, 0, NULL, 0, pga_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route max9759_dapm_routes[] = { + { "PGA", NULL, "INL" }, + { "PGA", NULL, "INR" }, + { "OUTL", NULL, "PGA" }, + { "OUTR", NULL, "PGA" }, +}; + +static const struct snd_soc_component_driver max9759_component_driver = { + .controls = max9759_dapm_controls, + .num_controls = ARRAY_SIZE(max9759_dapm_controls), + .dapm_widgets = max9759_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9759_dapm_widgets), + .dapm_routes = max9759_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9759_dapm_routes), +}; + +static int max9759_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct max9759 *priv; + int err; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + + priv->gpiod_shutdown = devm_gpiod_get(dev, "shutdown", GPIOD_OUT_HIGH); + if (IS_ERR(priv->gpiod_shutdown)) { + err = PTR_ERR(priv->gpiod_shutdown); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'shutdown' gpio: %d", err); + return err; + } + + priv->gpiod_mute = devm_gpiod_get(dev, "mute", GPIOD_OUT_HIGH); + if (IS_ERR(priv->gpiod_mute)) { + err = PTR_ERR(priv->gpiod_mute); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'mute' gpio: %d", err); + return err; + } + priv->is_mute = true; + + priv->gpiod_gain = devm_gpiod_get_array(dev, "gain", GPIOD_OUT_HIGH); + if (IS_ERR(priv->gpiod_gain)) { + err = PTR_ERR(priv->gpiod_gain); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'gain' gpios: %d", err); + return err; + } + priv->gain = 0; + + if (priv->gpiod_gain->ndescs != 2) { + dev_err(dev, "Invalid 'gain' gpios count: %d", + priv->gpiod_gain->ndescs); + return -EINVAL; + } + + return devm_snd_soc_register_component(dev, &max9759_component_driver, + NULL, 0); +} + +#ifdef CONFIG_OF +static const struct of_device_id max9759_ids[] = { + { .compatible = "maxim,max9759", }, + { } +}; +MODULE_DEVICE_TABLE(of, max9759_ids); +#endif + +static struct platform_driver max9759_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(max9759_ids), + }, + .probe = max9759_probe, +}; + +module_platform_driver(max9759_driver); + +MODULE_DESCRIPTION("ASoC MAX9759 amplifier driver"); +MODULE_AUTHOR("Neil Armstrong <narmstrong(a)baylibre.com>"); +MODULE_LICENSE("GPL"); -- 2.16.1

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[alsa-devel] [PATCH 0/2] ASoC: Add support for max9759 Amplifier Driver
by Neil Armstrong 20 Feb '18

20 Feb '18

The MAX9759 is a gpio controlled speaker amplifier, this patchset add the DT Bindings and the corresponding ASoC component driver. Neil Armstrong (2): ASoC: Add DT bindings documentation for max9759 amplifier ASoC: max9759: Add Amplifier Driver .../devicetree/bindings/sound/maxim,max9759.txt | 18 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9759.c | 207 +++++++++++++++++++++ 4 files changed, 232 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max9759.txt create mode 100644 sound/soc/codecs/max9759.c -- 2.7.4

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[alsa-devel] [4.4-stable 10/22] ASoC: Intel: Kconfig: fix build when ACPI is not enabled
by Arnd Bergmann 20 Feb '18

20 Feb '18

From: Vinod Koul <vinod.koul(a)intel.com> commit 3493d4a86457c7de9f1e602b4267c9b0f9ec1c9f upstream. Randy reported following error when ACPI is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI +which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) causing these build errors: In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:40:0: ../include/acpi/acpi_bus.h:65:20: error: conflicting types for 'acpi_evaluate_dsm' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:31:0: ../include/linux/acpi.h:676:34: note: previous definition of 'acpi_evaluate_dsm' was here static inline union acpi_object *acpi_evaluate_dsm(acpi_handle handle, CONFIG_SND_SST_IPC_ACPI was already dependent upon ACPI, but that was not solving it. So move the depends up to machine drivers and remove from CONFIG_SND_SST_IPC_ACPI. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> [arnd: rebased to 4.4-stable kernel] Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/Kconfig | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index d430ef5a4f38..79c29330c56a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -24,7 +24,6 @@ config SND_SST_IPC_PCI config SND_SST_IPC_ACPI tristate select SND_SST_IPC - depends on ACPI config SND_SOC_INTEL_SST tristate @@ -91,7 +90,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio DSP Support for MID BYT Platform" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -103,7 +102,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -115,7 +114,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI -- 2.9.0

1 0

[alsa-devel] [PATCH] ASoC: uniphier: evea: fix typo 'eva' -> 'evea'
by Katsuhiro Suzuki 20 Feb '18

20 Feb '18

Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/evea.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index cad7e60de21d..ac2d30b55fb6 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -292,7 +292,7 @@ static int evea_set_switch_hp(struct snd_kcontrol *kcontrol, return evea_update_switch_hp(evea); } -static const struct snd_kcontrol_new eva_controls[] = { +static const struct snd_kcontrol_new evea_controls[] = { SOC_SINGLE_BOOL_EXT("Line Capture Switch", 0, evea_get_switch_lin, evea_set_switch_lin), SOC_SINGLE_BOOL_EXT("Line Playback Switch", 0, @@ -384,8 +384,8 @@ static struct snd_soc_component_driver soc_codec_evea = { .num_dapm_widgets = ARRAY_SIZE(evea_widgets), .dapm_routes = evea_routes, .num_dapm_routes = ARRAY_SIZE(evea_routes), - .controls = eva_controls, - .num_controls = ARRAY_SIZE(eva_controls), + .controls = evea_controls, + .num_controls = ARRAY_SIZE(evea_controls), .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, -- 2.16.1

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[alsa-devel] Need help implementing an external PCM/control plugin pair
by Dimitris Papavasiliou 20 Feb '18

20 Feb '18

Hi, I'm in the process of putting together a loudness compensation plugin for ALSA. The filtering part is implemented as an external PCM plugin and it is, more or less, ready, meaning that you can set its parameters via configuration options and it'll happily filter away in real time. I'm trying now to create a matching mixer plugin, to allow on-the-fly adjustment of the parameters, and I'm not sure what the best way of accomplishing that would be. My main concerns are: 1. I've figured out how to create a basic external control plugin, that would allow me to expose a set of controls, which the user can fiddle with, using alsamixer et al., but what would be the best way of communicating changes in the values to the PCM plugin? I could imagine using methods external to ALSA, such as a file with stored values, which the ctl plugin updates and which the PCM plugin monitors for changes, or perhaps I could try to open the mixer from inside the PCM plugin, set callbacks for each of its elements and enter a snd_mixer_wait/snd_mixer_handle_events loop, in a separate thread of course. Is there reason to prefer one way over the other? Perhaps there is a better way, I'm not aware of? 2. I'd like to support event subscription/notification, but I'm not sure how to make that work. My understanding (from searching the net and looking at ALSA's source), is that I need to provide some sort of file descriptor, which the subscribing code can poll, so as to block until there's an update. Since I have no obvious candidates for such a file, I've tried creating a pipe, handing one end to the control plugin upon initialization and then writing single bytes to the other end when there's a change in one of the controls. This works but, as far as I can see, the pipe is never actually read from, from the subscribing code, so that I would have to manually read from it somehwere, say in the read_event callback, in order to flush it. In the end, I'll probably use the descriptor from an inotify watch on the file where the plugin stores the control settings, but is this a valid approach overall? Is there some documentation specifying the exact way in which poll descriptors are handled by ALSA (in the case of an extrnal CTL plugin), that I've missed? If not, are my assumptions above (that the descriptor is only polled and that its contents are never actually read) correct? 3. Finally, and although this is not essential, I'd like to provide metadata for some controls, to support dB ranges, as that's the most natural way to set them. I see hints that this is supported for external control plugins (SND_CTL_EXT_ACCESS_TLV_READ/WRITE, the snd_ctl_ext_tlv_rw_t callback), but I'm not sure how to implement it. Is it documented somewhere? If not, could someone describe the basic approach? Thanks, Dimitris

2 3

[alsa-devel] [PATCH v4 0/2] Add support for AK5558 ADC
by Daniel Baluta 19 Feb '18

19 Feb '18

We support normal mode, TDM mode and pm. Changes since v3: [addressed Mark's comments] * use C++ comments for SPDX identifier * removed some unused widgets and controls (channel enable MUXes, TDM, DSD related stuff) * removed set_dai_mute, set_bias_level, set_dai_sysclk as they are not used. * refactored probe/remove to have symmetric operations * use snd_soc_update_bits instead of hardcode read/write operation. Changes since v2: [addressed comments from Andy, Fabio and Rob] * sort include files * use probe_new * reword the binding document * use adc@10 instead of ak5558@10 * remove file name at the beginning of codec source code. * make i2c_probe and i2c_remove parameters naming consistent. Changes since v1: [addressed comments from Andy and Fabio] * fix GPIO polarity from active high to active low for correct documentation * fix license header by using SPDX identifier * remove debug prints at the beginning of functions. * only support auto clock switching (manual switching was dead code anyway) (in the future we could add a DT property to choose between manual and auto) * Use gpiod API * use GENMASK * introduce power_off/power_on Daniel Baluta (2): ASoC: codecs: Add support for AK5558 ADC driver ASoC: ak5558: Add bindings for AK5558 ADC Documentation/devicetree/bindings/sound/ak5558.txt | 22 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5558.c | 417 +++++++++++++++++++++ sound/soc/codecs/ak5558.h | 52 +++ 5 files changed, 499 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5558.txt create mode 100644 sound/soc/codecs/ak5558.c create mode 100644 sound/soc/codecs/ak5558.h -- 2.7.4

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[alsa-devel] [PATCH] ASoC: ADAU7002: Adding ACPI id
by Akshu Agrawal 19 Feb '18

19 Feb '18

Adding ACPI entry id for probing the driver. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/codecs/adau7002.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/adau7002.c b/sound/soc/codecs/adau7002.c index 49475e3..fdff868 100644 --- a/sound/soc/codecs/adau7002.c +++ b/sound/soc/codecs/adau7002.c @@ -7,6 +7,7 @@ * Licensed under the GPL-2. */ +#include <linux/acpi.h> #include <linux/init.h> #include <linux/module.h> #include <linux/of.h> @@ -69,10 +70,19 @@ static int adau7002_remove(struct platform_device *pdev) MODULE_DEVICE_TABLE(of, adau7002_dt_ids); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id adau7002_acpi_match[] = { + { "ADAU7002", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, adau7002_acpi_match); +#endif + static struct platform_driver adau7002_driver = { .driver = { .name = "adau7002", .of_match_table = of_match_ptr(adau7002_dt_ids), + .acpi_match_table = ACPI_PTR(adau7002_acpi_match), }, .probe = adau7002_probe, .remove = adau7002_remove, -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: tlv320aic31xx: Optimize the master mode I2S clock handling
by Peter Ujfalusi 19 Feb '18

19 Feb '18

Hi, Instead of forcing DAC/ADC to be turned on, we can use P0 R29, Bit2 to let the I2S clocks running, but we do not want it to run all the time and we do not want I2S clocks when it is not needed (analog loopback mode). Use DAPM_SUPPLY for the bit and connect it to the in/out AIF. The DAPM routes can be modify to not include the DAC/ADC. Regards, Peter --- Peter Ujfalusi (2): ASoC: tlv320aic31xx: Rename AIF_IN from 'DAC IN' to 'AIF IN' ASoC: tlv320aic31xx: Do not force power on the DAC/ADC in clock master mode sound/soc/codecs/tlv320aic31xx.c | 46 +++++++++++++++++++++++++--------------- sound/soc/codecs/tlv320aic31xx.h | 1 + 2 files changed, 30 insertions(+), 17 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH 0/4] ASoC: missing replace codec to component
by Kuninori Morimoto 19 Feb '18

19 Feb '18

Hi Mark These are new added driver on mark/for-next branch, but based on old "codec" framework. We want to convert it to component. These patches are do it Kuninori Morimoto (4): ASoC: intel: kbl_da7219_max98357: replace codec to component ASoC: amd: acp-da7219-max98357: replace codec to component ASoC: ak4458: replace codec to component ASoC: ak5558: replace codec to component sound/soc/amd/acp-da7219-max98357a.c | 4 +- sound/soc/codecs/ak4458.c | 105 +++++++++++++------------- sound/soc/codecs/ak5558.c | 65 ++++++++-------- sound/soc/intel/boards/kbl_da7219_max98357a.c | 12 +-- 4 files changed, 94 insertions(+), 92 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/4 v2] ASoC: missing replace codec to component
by Kuninori Morimoto 19 Feb '18

19 Feb '18

Hi Mark These are v2 of replace patch for new added driver on mark/for-next branch. The diff v1 -> v2 is that compile warning issue was solved on ak driver Kuninori Morimoto (4): ASoC: intel: kbl_da7219_max98357: replace codec to component ASoC: amd: acp-da7219-max98357: replace codec to component ASoC: ak4458: replace codec to component ASoC: ak5558: replace codec to component sound/soc/amd/acp-da7219-max98357a.c | 4 +- sound/soc/codecs/ak4458.c | 107 +++++++++++++------------- sound/soc/codecs/ak5558.c | 67 ++++++++-------- sound/soc/intel/boards/kbl_da7219_max98357a.c | 12 +-- 4 files changed, 94 insertions(+), 96 deletions(-) -- 1.9.1

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[alsa-devel] Applied "ASoC: sgtl5000: add 'Audio Switch' block implementation" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: sgtl5000: add 'Audio Switch' block implementation has been applied to the asoc tree at https://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 7f7d5239f56cfd25ba8470354d9705815c079157 Mon Sep 17 00:00:00 2001 From: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Date: Mon, 19 Feb 2018 10:03:45 +0100 Subject: [PATCH] ASoC: sgtl5000: add 'Audio Switch' block implementation Add mux for choosing DAC source: ADC, I2S or DAP. DAP - digital audio processing block. Moreover create choosing mux for DAP main source (ADC or I2S) and DAP mixer source (ADC or I2S). Mixer block allows summing DAP main and mixer sources togheter with defined volumes. Signed-off-by: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 78 ++++++++++++++++++++++++++++++++++++++------- 1 file changed, 67 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 05592ddcf0e6..1724c4622052 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -216,17 +216,54 @@ static SOC_ENUM_SINGLE_DECL(adc_enum, static const struct snd_kcontrol_new adc_mux = SOC_DAPM_ENUM("Capture Mux", adc_enum); +/* input sources for headphone */ +static const char *hp_mux_text[] = { + "DAC", "LINE_IN" +}; + +static SOC_ENUM_SINGLE_DECL(hp_enum, + SGTL5000_CHIP_ANA_CTRL, 6, + hp_mux_text); + +static const struct snd_kcontrol_new hp_mux = +SOC_DAPM_ENUM("Headphone Mux", hp_enum); + /* input sources for DAC */ static const char *dac_mux_text[] = { - "DAC", "LINE_IN" + "ADC", "I2S", "Rsvrd", "DAP" }; static SOC_ENUM_SINGLE_DECL(dac_enum, - SGTL5000_CHIP_ANA_CTRL, 6, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAC_SEL_SHIFT, dac_mux_text); static const struct snd_kcontrol_new dac_mux = -SOC_DAPM_ENUM("Headphone Mux", dac_enum); +SOC_DAPM_ENUM("Digital Input Mux", dac_enum); + +/* input sources for DAP */ +static const char *dap_mux_text[] = { + "ADC", "I2S" +}; + +static SOC_ENUM_SINGLE_DECL(dap_enum, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAP_SEL_SHIFT, + dap_mux_text); + +static const struct snd_kcontrol_new dap_mux = +SOC_DAPM_ENUM("DAP Mux", dap_enum); + +/* input sources for DAP mix */ +static const char *dapmix_mux_text[] = { + "ADC", "I2S" +}; + +static SOC_ENUM_SINGLE_DECL(dapmix_enum, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAP_MIX_SEL_SHIFT, + dapmix_mux_text); + +static const struct snd_kcontrol_new dapmix_mux = +SOC_DAPM_ENUM("DAP MIX Mux", dapmix_enum); + static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LINE_IN"), @@ -243,7 +280,12 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), - SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &hp_mux), + SND_SOC_DAPM_MUX("Digital Input Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + SND_SOC_DAPM_MUX("DAP Mux", SGTL5000_DAP_CTRL, 0, 0, &dap_mux), + SND_SOC_DAPM_MUX("DAP MIX Mux", SGTL5000_DAP_CTRL, 4, 0, &dapmix_mux), + SND_SOC_DAPM_MIXER("DAP", SGTL5000_CHIP_DIG_POWER, 4, 0, NULL, 0), + /* aif for i2s input */ SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", @@ -270,7 +312,19 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"DAP Mux", "ADC", "ADC"}, /* adc --> DAP mux */ + {"DAP Mux", NULL, "AIFIN"}, /* i2s --> DAP mux */ + {"DAP", NULL, "DAP Mux"}, /* DAP mux --> dap */ + + {"DAP MIX Mux", "ADC", "ADC"}, /* adc --> DAP MIX mux */ + {"DAP MIX Mux", NULL, "AIFIN"}, /* i2s --> DAP MIX mux */ + {"DAP", NULL, "DAP MIX Mux"}, /* DAP MIX mux --> dap */ + + {"Digital Input Mux", "ADC", "ADC"}, /* adc --> audio mux */ + {"Digital Input Mux", NULL, "AIFIN"}, /* i2s --> audio mux */ + {"Digital Input Mux", NULL, "DAP"}, /* dap --> audio mux */ + {"DAC", NULL, "Digital Input Mux"}, /* audio mux --> dac */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ @@ -463,6 +517,9 @@ static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0) ); +/* tlv for DAP channels, 0% - 100% - 200% */ +static const DECLARE_TLV_DB_SCALE(dap_volume, 0, 1, 0); + /* tlv for hp volume, -51.5db to 12.0db, step .5db */ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); @@ -514,6 +571,11 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + SOC_SINGLE_TLV("DAP Main channel", SGTL5000_DAP_MAIN_CHAN, + 0, 0xffff, 0, dap_volume), + + SOC_SINGLE_TLV("DAP Mix channel", SGTL5000_DAP_MIX_CHAN, + 0, 0xffff, 0, dap_volume), /* Automatic Volume Control (DAP AVC) */ SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), @@ -1217,12 +1279,6 @@ static int sgtl5000_probe(struct snd_soc_component *component) /* disable short cut detector */ snd_soc_component_write(component, SGTL5000_CHIP_SHORT_CTRL, 0); - /* - * set i2s as default input of sound switch - * TODO: add sound switch to control and dapm widge. - */ - snd_soc_component_write(component, SGTL5000_CHIP_SSS_CTRL, - SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); snd_soc_component_write(component, SGTL5000_CHIP_DIG_POWER, SGTL5000_ADC_EN | SGTL5000_DAC_EN); -- 2.16.1

1 0

[alsa-devel] Applied "ASoC: sgtl5000: add 5 band graphic equalizer" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: sgtl5000: add 5 band graphic equalizer has been applied to the asoc tree at https://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 92a2742f9813ef65da19145a77e8067d6abbc822 Mon Sep 17 00:00:00 2001 From: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Date: Mon, 19 Feb 2018 10:03:46 +0100 Subject: [PATCH] ASoC: sgtl5000: add 5 band graphic equalizer Enable 5-band graphic equalizer which allows manipulation of gain in range -11.75dB to 12dB for 5 specific sound bands: 115Hz, 330Hz, 990Hz, 3000Hz, 9900Hz. Signed-off-by: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 24 +++++++++++++++++++++--- sound/soc/codecs/sgtl5000.h | 7 +++++++ 2 files changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1724c4622052..7c1d65830c05 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -520,6 +520,9 @@ static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, /* tlv for DAP channels, 0% - 100% - 200% */ static const DECLARE_TLV_DB_SCALE(dap_volume, 0, 1, 0); +/* tlv for bass bands, -11.75db to 12.0db, step .25db */ +static const DECLARE_TLV_DB_SCALE(bass_band, -1175, 25, 0); + /* tlv for hp volume, -51.5db to 12.0db, step .5db */ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); @@ -585,6 +588,21 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, 0, 96, 0, avc_get_threshold, avc_put_threshold, avc_threshold), + + SOC_SINGLE_TLV("BASS 0", SGTL5000_DAP_EQ_BASS_BAND0, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 1", SGTL5000_DAP_EQ_BASS_BAND1, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 2", SGTL5000_DAP_EQ_BASS_BAND2, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 3", SGTL5000_DAP_EQ_BASS_BAND3, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 4", SGTL5000_DAP_EQ_BASS_BAND4, + 0, 0x5F, 0, bass_band), }; /* mute the codec used by alsa core */ @@ -1303,11 +1321,11 @@ static int sgtl5000_probe(struct snd_soc_component *component) SGTL5000_BIAS_VOLT_MASK, sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); /* - * disable DAP + * enable DAP Graphic EQ * TODO: - * Enable DAP in kcontrol and dapm. + * Add control for changing between PEQ/Tone Control/GEQ */ - snd_soc_component_write(component, SGTL5000_DAP_CTRL, 0); + snd_soc_component_write(component, SGTL5000_DAP_AUDIO_EQ, SGTL5000_DAP_SEL_GEQ); /* Unmute DAC after start */ snd_soc_component_update_bits(component, SGTL5000_CHIP_ADCDAC_CTRL, diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 22f3442af982..28cf637155bb 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -399,4 +399,11 @@ #define SGTL5000_SYSCLK 0x00 #define SGTL5000_LRCLK 0x01 +/* + * SGTL5000_DAP_AUDIO_EQ + */ +#define SGTL5000_DAP_SEL_PEQ 1 +#define SGTL5000_DAP_SEL_TONE_CTRL 2 +#define SGTL5000_DAP_SEL_GEQ 3 + #endif -- 2.16.1

1 0

[alsa-devel] Applied "ASoC: rt5651: Fix regcache sync errors on resume" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: rt5651: Fix regcache sync errors on resume has been applied to the asoc tree at https://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 2d30e9494f1ea320aaaad0cff9ddd92c87eac355 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:44 +0100 Subject: [PATCH] ASoC: rt5651: Fix regcache sync errors on resume The ALC5651 does not like multi-write accesses, avoid them. This fixes: rt5651 i2c-10EC5651:00: Unable to sync registers 0x27-0x28. -121 Errors on resume (and all registers after the registers in the error not being synced). Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/rt5651.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 831b297978a4..45a73049cf64 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1722,6 +1722,7 @@ static const struct regmap_config rt5651_regmap = { .num_reg_defaults = ARRAY_SIZE(rt5651_reg), .ranges = rt5651_ranges, .num_ranges = ARRAY_SIZE(rt5651_ranges), + .use_single_rw = true, }; #if defined(CONFIG_OF) -- 2.16.1

1 0

[alsa-devel] [PATCH v3] ASoC: dmic: Add optional wakeup delay
by Matthias Kaehlcke 19 Feb '18

19 Feb '18

On some systems a delay is needed after switching on the clocks, to allow the output to stabilize and avoid a popping noise at the beginning of the recording. Add the optional device tree property 'wakeup-delay-ms' and apply the specified delay after enabling the mic. A blocking delay can't be applied in dmic_daiops_trigger() since the function is called in atomic context. Instead use a DAPM event handler to set the enable GPIO and apply the delay in the handler. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v3: - rebased on latest dmic topic branch Changes in v2: - use DAPM event handler instead of _prepare() and get rid of _trigger() - skip error check for optional 'wakeup-delay-ms' property - updated commit message .../devicetree/bindings/sound/dmic.txt | 2 + sound/soc/codecs/dmic.c | 63 ++++++++++--------- 2 files changed, 37 insertions(+), 28 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt index f7bf65611453..e957b4136716 100644 --- a/Documentation/devicetree/bindings/sound/dmic.txt +++ b/Documentation/devicetree/bindings/sound/dmic.txt @@ -8,6 +8,7 @@ Required properties: Optional properties: - dmicen-gpios: GPIO specifier for dmic to control start and stop - num-channels: Number of microphones on this DAI + - wakeup-delay-ms: Delay (in ms) after enabling the DMIC Example node: @@ -15,4 +16,5 @@ Example node: compatible = "dmic-codec"; dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; num-channels = <1>; + wakeup-delay-ms <50>; }; diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 32a8f71d51f1..bac5bbb2dd59 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,7 @@ * */ +#include <linux/delay.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <linux/platform_device.h> @@ -29,34 +30,33 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -static int dmic_daiops_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); +struct dmic { + struct gpio_desc *gpio_en; + int wakeup_delay; +}; - if (!dmic_en) - return 0; +static int dmic_aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct dmic *dmic = snd_soc_codec_get_drvdata(codec); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(dmic_en, 1); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 1); + + if (dmic->wakeup_delay) + msleep(dmic->wakeup_delay); break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(dmic_en, 0); + case SND_SOC_DAPM_POST_PMD: + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 0); break; } return 0; } -static const struct snd_soc_dai_ops dmic_dai_ops = { - .trigger = dmic_daiops_trigger, -}; - static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -68,26 +68,33 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &dmic_dai_ops, }; static int dmic_component_probe(struct snd_soc_component *component) { - struct gpio_desc *dmic_en; + struct dmic *dmic; + + dmic = devm_kzalloc(component->dev, sizeof(*dmic), GFP_KERNEL); + if (!dmic) + return -ENOMEM; + + dmic->gpio_en = devm_gpiod_get_optional(component->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic->gpio_en)) + return PTR_ERR(dmic->gpio_en); - dmic_en = devm_gpiod_get_optional(component->dev, - "dmicen", GPIOD_OUT_LOW); - if (IS_ERR(dmic_en)) - return PTR_ERR(dmic_en); + device_property_read_u32(component->dev, "wakeup-delay-ms", + &dmic->wakeup_delay); - snd_soc_component_set_drvdata(component, dmic_en); + snd_soc_component_set_drvdata(component, dmic); return 0; } static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { - SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("DMIC AIF", "Capture", 0, + SND_SOC_NOPM, 0, 0, dmic_aif_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_INPUT("DMic"), }; -- 2.16.1.291.g4437f3f132-goog

3 2

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix compiler warning -Wmaybe-uninitialized
by Kirill Marinushkin 19 Feb '18

19 Feb '18

Configuration: SND_SOC_INTEL_SKYLAKE=y PM_SLEEP=y Warning message: sound/soc/intel/skylake/skl.c: In function 'skl_resume': sound/soc/intel/skylake/skl.c:326:6: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] int ret; ^~~ Fixes: 4557c305d4fc ("ASoC: Intel: Skylake: Add support for active suspend") Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Guneshwor Singh <guneshwor.o.singh(a)intel.com> Cc: Naveen Manohar <naveen.m(a)intel.com> Cc: Harsha Priya N <harshapriya.n(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/intel/skylake/skl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 31d8634e8aa1..273a9ab75489 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -323,7 +323,7 @@ static int skl_resume(struct device *dev) struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); struct hdac_ext_link *hlink = NULL; - int ret; + int ret = 0; /* Turned OFF in HDMI codec driver after codec reconfiguration */ if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { -- 2.13.6

2 1

[alsa-devel] [PATCH 0/1] Use of shared resource in designware
by Akshu Agrawal 19 Feb '18

19 Feb '18

This patch will allow the driver to map same registers to multiple instances of driver. It is useful when playback and capture have same regsiters and are required to be present on separate cpu dais. This patch is dependant on "Add devm_ioremap_shared_resource()" patch. https://patchwork.kernel.org/patch/10177127/ Akshu Agrawal (1): ASoC: dwc: Use devm_ioremap_shared_resource include/sound/designware_i2s.h | 1 + sound/soc/dwc/dwc-i2s.c | 18 +++++++++++++++--- 2 files changed, 16 insertions(+), 3 deletions(-) -- 1.9.1

4 5

[alsa-devel] [PATCH] Makefile.am: Add missing code for enhanced configure option of alsa plugins
by Lars Wendler 18 Feb '18

18 Feb '18

Commit 1201e158fb00bf15c353d088c94c1cb3eff9653c added the pieces for configure but did not do the same for Makefile. As result all plugins were still built regardless of corresponding --disable-* option being called with configure. --- Makefile.am | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/Makefile.am b/Makefile.am index 69cfe0d..76cefef 100644 --- a/Makefile.am +++ b/Makefile.am @@ -1,4 +1,4 @@ -SUBDIRS = oss mix usb_stream arcam-av doc +SUBDIRS = doc if HAVE_JACK SUBDIRS += jack endif @@ -20,6 +20,18 @@ endif if HAVE_SPEEXDSP SUBDIRS += speex endif +if HAVE_OSS +SUBDIRS += oss +endif +if HAVE_MIX +SUBDIRS += mix +endif +if HAVE_USBSTREAM +SUBDIRS += usb_stream +endif +if HAVE_ARCAMAV +SUBDIRS += arcam-av +endif EXTRA_DIST = gitcompile version COPYING.GPL m4/attributes.m4 AUTOMAKE_OPTIONS = foreign -- 2.16.2

1 0

[alsa-devel] [asoc:for-next 19/53] sound/soc//amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type
by kbuild test robot 16 Feb '18

16 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 5372fcd763f83d0e4d38ac008a2076f36664cbae commit: b2eeca7e9dc8f413d446f397a0220e21d6eb01d1 [19/53] Merge remote-tracking branches 'asoc/topic/ak5386', 'asoc/topic/ak5558', 'asoc/topic/alc5623', 'asoc/topic/alc5632' and 'asoc/topic/amd' into asoc-next config: x86_64-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout b2eeca7e9dc8f413d446f397a0220e21d6eb01d1 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc//amd/acp-da7219-max98357a.c: In function 'cz_da7219_init': >> sound/soc//amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type [-Werror=incompatible-pointer-types] da7219_aad_jack_det(codec, &cz_jack); ^~~~~ In file included from sound/soc//amd/acp-da7219-max98357a.c:38:0: sound/soc//amd/../codecs/da7219-aad.h:209:6: note: expected 'struct snd_soc_component *' but argument is of type 'struct snd_soc_codec *' void da7219_aad_jack_det(struct snd_soc_component *component, struct snd_soc_jack *jack); ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors sparse warnings: (new ones prefixed by >>) >> sound/soc/amd/acp-da7219-max98357a.c:79:29: sparse: incorrect type in argument 1 (different base types) @@ expected struct snd_soc_component @@ got ct snd_soc_component @@ sound/soc/amd/acp-da7219-max98357a.c:79:29: expected struct snd_soc_component sound/soc/amd/acp-da7219-max98357a.c:79:29: got struct snd_soc_codec sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_da7219_init': sound/soc/amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type da7219_aad_jack_det(codec, &cz_jack); ^~~~~ In file included from sound/soc/amd/acp-da7219-max98357a.c:38:0: sound/soc/amd/../codecs/da7219-aad.h:209:6: note: expected 'struct snd_soc_component but argument is of type 'struct snd_soc_codec void da7219_aad_jack_det(struct snd_soc_component struct snd_soc_jack ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/da7219_aad_jack_det +79 sound/soc//amd/acp-da7219-max98357a.c 608a300f Akshu Agrawal 2018-02-16 45 608a300f Akshu Agrawal 2018-02-16 46 static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) 608a300f Akshu Agrawal 2018-02-16 47 { 608a300f Akshu Agrawal 2018-02-16 48 int ret; 608a300f Akshu Agrawal 2018-02-16 49 struct snd_soc_card *card = rtd->card; 608a300f Akshu Agrawal 2018-02-16 50 struct snd_soc_codec *codec = rtd->codec; 608a300f Akshu Agrawal 2018-02-16 51 struct snd_soc_dai *codec_dai = rtd->codec_dai; 608a300f Akshu Agrawal 2018-02-16 52 608a300f Akshu Agrawal 2018-02-16 53 dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); 608a300f Akshu Agrawal 2018-02-16 54 608a300f Akshu Agrawal 2018-02-16 55 ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 608a300f Akshu Agrawal 2018-02-16 56 CZ_PLAT_CLK, SND_SOC_CLOCK_IN); 608a300f Akshu Agrawal 2018-02-16 57 if (ret < 0) { 608a300f Akshu Agrawal 2018-02-16 58 dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); 608a300f Akshu Agrawal 2018-02-16 59 return ret; 608a300f Akshu Agrawal 2018-02-16 60 } 608a300f Akshu Agrawal 2018-02-16 61 608a300f Akshu Agrawal 2018-02-16 62 ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL, 608a300f Akshu Agrawal 2018-02-16 63 CZ_PLAT_CLK, MCLK_RATE); 608a300f Akshu Agrawal 2018-02-16 64 if (ret < 0) { 608a300f Akshu Agrawal 2018-02-16 65 dev_err(rtd->dev, "can't set codec pll: %d\n", ret); 608a300f Akshu Agrawal 2018-02-16 66 return ret; 608a300f Akshu Agrawal 2018-02-16 67 } 608a300f Akshu Agrawal 2018-02-16 68 608a300f Akshu Agrawal 2018-02-16 69 ret = snd_soc_card_jack_new(card, "Headset Jack", 608a300f Akshu Agrawal 2018-02-16 70 SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | 608a300f Akshu Agrawal 2018-02-16 71 SND_JACK_BTN_0 | SND_JACK_BTN_1 | 608a300f Akshu Agrawal 2018-02-16 72 SND_JACK_BTN_2 | SND_JACK_BTN_3, 608a300f Akshu Agrawal 2018-02-16 73 &cz_jack, NULL, 0); 608a300f Akshu Agrawal 2018-02-16 74 if (ret) { 608a300f Akshu Agrawal 2018-02-16 75 dev_err(card->dev, "HP jack creation failed %d\n", ret); 608a300f Akshu Agrawal 2018-02-16 76 return ret; 608a300f Akshu Agrawal 2018-02-16 77 } 608a300f Akshu Agrawal 2018-02-16 78 608a300f Akshu Agrawal 2018-02-16 @79 da7219_aad_jack_det(codec, &cz_jack); 608a300f Akshu Agrawal 2018-02-16 80 608a300f Akshu Agrawal 2018-02-16 81 return 0; 608a300f Akshu Agrawal 2018-02-16 82 } 608a300f Akshu Agrawal 2018-02-16 83 :::::: The code at line 79 was first introduced by commit :::::: 608a300fc1f0e088446cbef8e13ded686c5172b6 ASoC: AMD: Add machine driver for ST DA7219 MAX98357 :::::: TO: Akshu Agrawal <akshu.agrawal(a)amd.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: wm_adsp: Fix some signedness errors in register access
by Richard Fitzgerald 16 Feb '18

16 Feb '18

base and sysclk_reg should be unsigned int, as that is what regmap takes as a register address. sysclk_mask is used to mask unsigned register values so should be unsigned. sysclk_shift is a shift value so is not allowed to be negative. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 41cc11c19b83..d4042c95315e 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -64,10 +64,10 @@ struct wm_adsp { struct regmap *regmap; struct snd_soc_codec *codec; - int base; - int sysclk_reg; - int sysclk_mask; - int sysclk_shift; + unsigned int base; + unsigned int sysclk_reg; + unsigned int sysclk_mask; + unsigned int sysclk_shift; struct list_head alg_regions; -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: dmic: Add optional wakeup delay
by Matthias Kaehlcke 16 Feb '18

16 Feb '18

On some systems a delay is needed after switching on the clocks, to allow the output to stabilize and avoid a popping noise at the beginning of the recording. Add the optional device tree property 'wakeup-delay-ms' and apply the specified delay after enabling the mic. A blocking delay can't be applied in dmic_daiops_trigger() since the function is called in atomic context. Instead use a DAPM event handler to set the enable GPIO and apply the delay in the handler. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v2: - use DAPM event handler instead of _prepare() and get rid of _trigger() - skip error check for optional 'wakeup-delay-ms' property - updated commit message .../devicetree/bindings/sound/dmic.txt | 2 + sound/soc/codecs/dmic.c | 63 ++++++++++--------- 2 files changed, 37 insertions(+), 28 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt index f7bf65611453..e957b4136716 100644 --- a/Documentation/devicetree/bindings/sound/dmic.txt +++ b/Documentation/devicetree/bindings/sound/dmic.txt @@ -8,6 +8,7 @@ Required properties: Optional properties: - dmicen-gpios: GPIO specifier for dmic to control start and stop - num-channels: Number of microphones on this DAI + - wakeup-delay-ms: Delay (in ms) after enabling the DMIC Example node: @@ -15,4 +16,5 @@ Example node: compatible = "dmic-codec"; dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; num-channels = <1>; + wakeup-delay-ms <50>; }; diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index cf83c423394d..b5f60ac22ace 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,7 @@ * */ +#include <linux/delay.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <linux/platform_device.h> @@ -29,34 +30,33 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -static int dmic_daiops_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); +struct dmic { + struct gpio_desc *gpio_en; + int wakeup_delay; +}; - if (!dmic_en) - return 0; +static int dmic_aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct dmic *dmic = snd_soc_codec_get_drvdata(codec); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(dmic_en, 1); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 1); + + if (dmic->wakeup_delay) + msleep(dmic->wakeup_delay); break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(dmic_en, 0); + case SND_SOC_DAPM_POST_PMD: + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 0); break; } return 0; } -static const struct snd_soc_dai_ops dmic_dai_ops = { - .trigger = dmic_daiops_trigger, -}; - static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -68,26 +68,33 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &dmic_dai_ops, }; static int dmic_codec_probe(struct snd_soc_codec *codec) { - struct gpio_desc *dmic_en; + struct dmic *dmic; + + dmic = devm_kzalloc(codec->dev, sizeof(*dmic), GFP_KERNEL); + if (!dmic) + return -ENOMEM; + + dmic->gpio_en = devm_gpiod_get_optional(codec->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic->gpio_en)) + return PTR_ERR(dmic->gpio_en); - dmic_en = devm_gpiod_get_optional(codec->dev, - "dmicen", GPIOD_OUT_LOW); - if (IS_ERR(dmic_en)) - return PTR_ERR(dmic_en); + device_property_read_u32(codec->dev, "wakeup-delay-ms", + &dmic->wakeup_delay); - snd_soc_codec_set_drvdata(codec, dmic_en); + snd_soc_codec_set_drvdata(codec, dmic); return 0; } static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { - SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("DMIC AIF", "Capture", 0, + SND_SOC_NOPM, 0, 0, dmic_aif_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_INPUT("DMic"), }; -- 2.16.1.291.g4437f3f132-goog

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[alsa-devel] [asoc:topic/ak4458 1/2] sound/soc/codecs/ak4458.c:577:28: sparse: symbol 'ak4458_regmap' was not declared. Should it be static?
by kbuild test robot 16 Feb '18

16 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/ak4458 head: 17e2304f0ad31d3f9bba4285b7024bf3c3626600 commit: 08660086eff9e9e69246a65ee48aba5c3901bf5e [1/2] ASoC: ak4458: Add support for AK4458 DAC driver reproduce: # apt-get install sparse git checkout 08660086eff9e9e69246a65ee48aba5c3901bf5e make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/ak4458.c:577:28: sparse: symbol 'ak4458_regmap' was not declared. Should it be >> sound/soc/codecs/ak4458.c:587:25: sparse: symbol 'ak4458_pm' was not declared. Should it be Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] MAINTAINERS: Add myself as sgtl5000 maintainer
by Fabio Estevam 16 Feb '18

16 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> I would like helping maintaining and reviewing/testing sgtl5000 related patches. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 9a7f76e..c488c8a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -9922,6 +9922,13 @@ F: Documentation/ABI/stable/sysfs-bus-nvmem F: include/linux/nvmem-consumer.h F: include/linux/nvmem-provider.h +NXP SGTL5000 DRIVER +M: Fabio Estevam <fabio.estevam(a)nxp.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/sgtl5000.txt +F: sound/soc/codecs/sgtl5000* + NXP TDA998X DRM DRIVER M: Russell King <linux(a)armlinux.org.uk> S: Supported -- 2.7.4

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[alsa-devel] [PATCH v3] ASoC: sgtl5000: Fix suspend/resume
by Fabio Estevam 16 Feb '18

16 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 8419caa72702 ("ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF") causes the sgtl5000 to fail after a suspend/resume sequence: # aplay /media/a2002011001-e02.wav Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2051: write error: Input/output error The problem is caused by the fact that the aforementioned commit dropped the cache handling, so re-introduce the register map resync to fix the problem. Cc: <stable(a)vger.kernel.org> Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v2: - Rebased against Linus' tree sound/soc/codecs/sgtl5000.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e1ab553..7ff5cb7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -871,15 +871,26 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct sgtl5000_priv *sgtl = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: + regcache_cache_only(sgtl->regmap, false); + ret = regcache_sync(sgtl->regmap); + if (ret) { + regcache_cache_only(sgtl->regmap, true); + return ret; + } + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, SGTL5000_REFTOP_POWERUP); break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl->regmap, true); snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, 0); break; -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] Add support for AK5558 ADC
by Daniel Baluta 16 Feb '18

16 Feb '18

We support normal mode, TDM mode and pm. Changes since v2: [addressed comments from Andy, Fabio and Rob] * sort include files * use probe_new * reword the binding document * use adc@10 instead of ak5558@10 * remove file name at the beginning of codec source code. * make i2c_probe and i2c_remove parameters naming consistent. Changes since v1: [addressed comments from Andy and Fabio] * fix GPIO polarity from active high to active low for correct documentation * fix license header by using SPDX identifier * remove debug prints at the beginning of functions. * only support auto clock switching (manual switching was dead code anyway) (in the future we could add a DT property to choose between manual and auto) * Use gpiod API * use GENMASK * introduce power_off/power_on One open question is the resume sequence which appears to need power_off/power_on. Just power_on alone isn't enough. With just power_on after resume aplay plays a song for 1 seconds and then the sound stops. Datasheet says, page 55" (1) The PDN pin should be held to “L” for more than 150 ns after AVDD and TVDD are powered up. Daniel Baluta (2): ASoC: codecs: Add support for AK5558 ADC driver ASoC: ak5558: Add bindings for AK5558 ADC Documentation/devicetree/bindings/sound/ak5558.txt | 22 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5558.c | 618 +++++++++++++++++++++ sound/soc/codecs/ak5558.h | 52 ++ 5 files changed, 700 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5558.txt create mode 100644 sound/soc/codecs/ak5558.c create mode 100644 sound/soc/codecs/ak5558.h -- 2.7.4

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[alsa-devel] [PATCH 0/2] ASoC: Intel: kabylake: Adds da7219+max98537 machine driver
by Naveen Manohar 16 Feb '18

16 Feb '18

Add kabylake machine driver & corresponding driver entry in machine table for maxim speaker codec - max98357a, configured to ssp0 and dialog headset codec - da7219, configured to ssp1. Naveen Manohar (2): ASoC: Intel: Add Kabylake-y Dialog Maxim machine driver ASoC: Intel: Add Kabylake Dialog+Maxim machine driver entry sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 613 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 13 + 4 files changed, 642 insertions(+) create mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c -- 1.9.1

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[alsa-devel] [PATCH v3 0/2] Add support for AK4458 DAC
by Cosmin-Gabriel Samoila 16 Feb '18

16 Feb '18

We support Normal Mode, TDM Mode and PM. Changes since V2: - remove unused code left from V1 - rewrite set_dai_fmt to only remember the format since - the format is rewritten in hw_params - use defined constants for mode in set_tdm_slot function - use GENMASK - add commit message for bindings patch - modify bindings description text since we only support I2C for now - rename ak4458_codec_probe to ak4458_probe - write whole implementation of regmap init and gpiod get in i2c_probe - added ak4458_remove where we power_off the chip Changes since V1: - support only i2c - i2c probe / remove/suspend/ resume moved into shared code - change enum array to separate variables - aggregate Volume control for Left and Right Channel - remove mux from DAPM - remove unused functions - remove unused controls - remove unused macros - use snd_soc_update_bits instead of soc_read + soc_write - use GPIOD - use SPDX identifier - use dac@10 instgitead of ak4458@10 - use explicit power_on/off functions - add __maybe_unused attribute - use probe_new for i2c_probe Cosmin-Gabriel Samoila (2): ASoC: codecs: Add support for AK4458 DAC driver ASoC: ak4458: Add bindings for AK4458 DAC Documentation/devicetree/bindings/sound/ak4458.txt | 23 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4458.c | 659 +++++++++++++++++++++ sound/soc/codecs/ak4458.h | 86 +++ 5 files changed, 776 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4458.txt create mode 100644 sound/soc/codecs/ak4458.c create mode 100644 sound/soc/codecs/ak4458.h -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix suspend/resume
by Fabio Estevam 16 Feb '18

16 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 8419caa72702 ("ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF") causes the sgtl5000 to fail after a suspend/resume sequence: # aplay /media/a2002011001-e02.wav Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2051: write error: Input/output error The problem is caused by the fact that the aforementioned commit dropped the cache handling, so re-introduce the register map resync to fix the problem. Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Re-introduce cache handling (Mark). Mark, If this gets applied into next, then I can generate the fixes for the 4.9 and 4.14 stable versions later. Thanks sound/soc/codecs/sgtl5000.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c26ac22..7514c77 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -871,15 +871,26 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, static int sgtl5000_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { + struct sgtl5000_priv *sgtl = snd_soc_component_get_drvdata(component); + int ret; + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: + regcache_cache_only(sgtl->regmap, false); + ret = regcache_sync(sgtl->regmap); + if (ret) { + regcache_cache_only(sgtl->regmap, true); + return ret; + } + snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, SGTL5000_REFTOP_POWERUP); break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl->regmap, true); snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, 0); break; -- 2.7.4

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[alsa-devel] WM8985 Machine Driver Write Error
by Mohammad, Jamal M 16 Feb '18

16 Feb '18

Hi Guys, We have a custom i.MX6 Board which has WM8985 audio codec. I developed machine level driver ( imx-wm8985.c ) looking at imx-wm8962 code. Driver loads fine without any error. "aplay -L" list the sound card. I2C interface works fine but it is write only peripheral. But when I am trying to play a Wav file using aplay, it fails with "aplay: pcm_write:1939: write error: Input/output error". Reading Online found that this happens when routing is incorrect. I am unable to proceed after this..Please help me to fix this issue. Here is my device tree. sound { compatible = "fsl,imx6q-sabresd-wm8985", "fsl,imx-audio-wm8985"; model = "wm8985-audio"; cpu-dai = <&ssi2>; audio-codec = <&codec>; audio-routing = "Ext Spk", "SPKL", "Ext Spk", "SPKR"; mux-int-port = <2>; mux-ext-port = <3>; }; Please provide me clue/hint to proceed and solve the issue. I can provide you more details if you need. Thanks for your help. Mohammad Jamal Mohiuddin Software Engineer, Retail Hosp/HW NCR Corporation India Pvt Ltd. MohammadJamal.Mohiuddin(a)ncr.com | www.ncr.com<http://www.ncr.com>

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[alsa-devel] sgtl5000 fails after suspend/resume
by Fabio Estevam 15 Feb '18

15 Feb '18

Hi, I noticed that sgtl5000 does not work after a suspend/resume cycle: # aplay /media/a2002011001-e02.wav Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2051: write error: Input/output error Tested several 4.16-rc1 (and also previous kernels like 4.4) and all of them failed. I think it worked in some point in time. Tested on a imx53qsb and imx6 wandboard. Does anyone have any suggestions? Thanks, Fabio Estevam

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[alsa-devel] [PATCH] ASoC: dmic: Add optional wakeup delay
by Matthias Kaehlcke 15 Feb '18

15 Feb '18

On some systems a delay is needed after switching on the clocks, to allow the DMIC output to stabilize and avoid a popping noise at the beginning of the recording. Add the optional device tree property 'wakeup-delay-ms' and apply the specified delay in the new dmic_daiops_prepare(). Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- .../devicetree/bindings/sound/dmic.txt | 2 + sound/soc/codecs/dmic.c | 54 ++++++++++++++----- 2 files changed, 42 insertions(+), 14 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt index 54c8ef6498a8..de741c6609d0 100644 --- a/Documentation/devicetree/bindings/sound/dmic.txt +++ b/Documentation/devicetree/bindings/sound/dmic.txt @@ -7,10 +7,12 @@ Required properties: Optional properties: - dmicen-gpios: GPIO specifier for dmic to control start and stop + - wakeup-delay-ms: Delay (in ms) after enabling the DMIC Example node: dmic_codec: dmic@0 { compatible = "dmic-codec"; dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; + wakeup-delay-ms <50>; }; diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index b88a1ee66f80..11f6abf11074 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,7 @@ * */ +#include <linux/delay.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <linux/platform_device.h> @@ -29,24 +30,38 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> +struct dmic { + struct gpio_desc *gpio_en; + int wakeup_delay; +}; + +static int dmic_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct dmic *dmic = snd_soc_dai_get_drvdata(dai); + + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 1); + + if (dmic->wakeup_delay) + msleep(dmic->wakeup_delay); + + return 0; +} + static int dmic_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); + struct dmic *dmic = snd_soc_dai_get_drvdata(dai); - if (!dmic_en) + if (!dmic->gpio_en) return 0; switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(dmic_en, 1); - break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(dmic_en, 0); + gpiod_set_value(dmic->gpio_en, 0); break; } @@ -54,6 +69,7 @@ static int dmic_daiops_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops dmic_dai_ops = { + .prepare = dmic_daiops_prepare, .trigger = dmic_daiops_trigger, }; @@ -73,14 +89,24 @@ static struct snd_soc_dai_driver dmic_dai = { static int dmic_codec_probe(struct snd_soc_codec *codec) { - struct gpio_desc *dmic_en; + struct dmic *dmic; + int err; + + dmic = devm_kzalloc(codec->dev, sizeof(*dmic), GFP_KERNEL); + if (!dmic) + return -ENOMEM; + + dmic->gpio_en = devm_gpiod_get_optional(codec->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic->gpio_en)) + return PTR_ERR(dmic->gpio_en); - dmic_en = devm_gpiod_get_optional(codec->dev, - "dmicen", GPIOD_OUT_LOW); - if (IS_ERR(dmic_en)) - return PTR_ERR(dmic_en); + err = device_property_read_u32(codec->dev, "wakeup-delay-ms", + &dmic->wakeup_delay); + if (err && (err != -EINVAL)) + return err; - snd_soc_codec_set_drvdata(codec, dmic_en); + snd_soc_codec_set_drvdata(codec, dmic); return 0; } -- 2.16.1.291.g4437f3f132-goog

3 4

[alsa-devel] [PATCH v12 0/8] TDA1997x HDMI video reciver
by Tim Harvey 15 Feb '18

15 Feb '18

This is a v4l2 subdev driver supporting the TDA1997x HDMI video receiver. I've tested this on a Gateworks GW54xx/GW551x with an IMX6Q/IMX6DL which uses the TDA19971 with 16bits connected to the IMX6 CSI and single-lane I2S audio providing 2-channel audio. For this configuration I've tested both 16bit YUV422 and 8bit BT656 parallel video bus modes. While the driver should support the TDA1993 I do not have one for testing. Further potential development efforts include: - CEC support - HDCP support - TDA19972 support (2 inputs) Media graphs can be found at http://dev.gateworks.com/docs/linux/media See details below for configuration and compliance tests History: v12: - fix coccinelle warnings v11: - return -ERANGE from tda1997x_detect_std (Hans) - clean up tda1997x_g_input_status (Hans) - show detected timings on resolution change if debug enabled - fix unitialized ret var in tda1997x_query_dv_timings - update log status to show detected timings v10: - removed unnecessary check for !timings in get/set/query dv timings (Hans) - dropped pointless s_stream handler (Hans) - remove need for detected_timings and always use set timings (Hans) v9: - add digital video decoder video interface entity function v8: - fix clearing pad for VIDIOC_DV_TIMIGNS_CAP - support full range of input modes based on timings_cap - add patch to fix clearing pad for VIDIOC_DV_TIMIGINGS - fix available formats for tda19971 bt656 bus width >12 - fix set_format (compliance) - fixed get/set edid (compliance) - add init_cfg to setup default pad config (compliance) - added missing pad checks to get_dv_timings_cap/enum_dv_timings (compliance) - fix alignment of if statement and whitespace in comment (Hans) - move regs to tda1997x_regs.h to clean up (Hans) - add define and sanity check for num of mbus_codes (Hans) v7: - fix interlaced mode - support no AVI infoframe (ie DVI) (Hans) - add support for multiple output formats (Hans) v6: - tda1997x: fix return on regulator enablei in tda1997x_set_power() (Fabio) - tda1997x: fix colorspace handling (Hans) - bindings: added Robs's ack (Rob) - replace copyright with SPDX tag (Philippe) v5: - added v4l2_hdmi_colorimetry() patch from Hans to series - bindings: added Sakari's ack - tda1997x: uppercase string constants - tda1997x: use v4l2_hdmi_rx_coloriemtry to fill format - tda1997x: fix V4L2_CID_DV_RX_RGB_RANGE - tda1997x: fix interlaced mode format - dts: remove leading 0 from unit address - dts: add newline between property list and child node - dts: added missing audmux in GW551x dts v4: - move include/dt-bindings/media/tda1997x.h to bindings patch - clarify port node details in bindings - fix typos - fix default quant range for VGA - fix quant range handling and conv matrix - add additional standards and capabilities to timings_cap v3: - fix typo in dt bindings - added dt bindings for GW551x - use V4L2_DV_BT_FRAME_WIDTH/HEIGHT macros - fixed missing break - use only hdmi_infoframe_log for infoframe logging - simplify tda1997x_s_stream error handling - add delayed work proc to handle hotplug enable/disable - fix set_edid (disable HPD before writing, enable after) - remove enabling edid by default - initialize timings - take quant range into account in colorspace conversion - remove vendor/product tracking (we provide this in log_status via infoframes) - add v4l_controls - add more detail to log_status - calculate vhref generator timings - timing detection fixes (rounding errors, hswidth errors) - rename configure_input/configure_conv functions v2: - encorporate feedback into dt bindings - change audio dt bindings - implement dv timings enum/cap - remove deprecated g_mbus_config op - fix dv_query_timings - add EDID get/set handling - remove max-pixel-rate support - add audio codec DAI support - added media-ctl and v4l2-compliance details v1: - initial RFC Pipeline configuration: $ media-ctl -e 'tda19971 2-0048' /dev/v4l-subdev1 $ v4l2-ctl -d /dev/v4l-subdev1 --set-dv-bt-timings=query BT timings set $ media-ctl --get-v4l2 '"tda19971 2-0048":0' [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] $ media-ctl --link "tda19971 2-0048":0 -> "ipu1_csi0_mux":1[1] $ media-ctl --link "ipu1_csi0_mux":2 -> "ipu1_csi0":0[1] $ media-ctl --link "ipu1_csi0":2 -> "ipu1_csi0 capture":0[1] $ media-ctl --set-v4l2 'tda19971 2-0048':0[fmt:UYVY8_2X8/1280x720] $ media-ctl --set-v4l2 'ipu1_csi0_mux':2[fmt:UYVY8_2X8/1280x720] $ media-ctl --set-v4l2 'ipu1_csi0':0[fmt:UYVY8_2X8/1280x720] $ gst-launch-1.0 v4l2src device=/dev/video4 ! video/x-raw,width=1280,height=720,format=UYVY ! jpegenc ! rtpjpegpay ! udpsink host=172.24.40.6 port=5000 $ media-ctl -d /dev/media0 -p Media controller API version 4.15.0 Media device information ------------------------ driver imx-media model imx-media serial bus info hw revision 0x0 driver version 4.15.0 Device topology - entity 1: adv7180 2-0020 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 20004 device node name /dev/v4l-subdev0 pad0: Source [fmt:UYVY8_2X8/720x480 field:interlaced colorspace:smpte170m] -> "ipu2_csi1_mux":1 [] - entity 3: tda19971 2-0048 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev1 pad0: Source [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] [dv.caps:BT.656/1120 min:640x350@13000000 max:1920x1200@165000000 stds:CEA-861,DMT,CVT,GTF caps:interlaced,progressive,reduced-blanking,custom] [dv.detect:BT.656/1120 1280x720p60 (1650x750) stds:CEA-861 flags:can-reduce-fps,CE-video,has-cea861-vic] [dv.current:BT.656/1120 1280x720p60 (1650x750) stds:CEA-861 flags:can-reduce-fps,CE-video,has-cea861-vic] -> "ipu1_csi0_mux":1 [ENABLED] - entity 5: ipu1_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev2 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] - entity 9: ipu2_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev3 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] - entity 13: ipu1_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev4 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_vdic":2 [] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf":0 [] - entity 17: ipu1_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev5 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc capture":0 [] - entity 20: ipu1_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video0 pad0: Sink <- "ipu1_ic_prpenc":1 [] - entity 26: ipu1_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev6 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf capture":0 [] - entity 29: ipu1_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video1 pad0: Sink <- "ipu1_ic_prpvf":1 [] - entity 35: ipu2_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev7 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_vdic":2 [] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf":0 [] - entity 39: ipu2_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev8 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc capture":0 [] - entity 42: ipu2_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video2 pad0: Sink <- "ipu2_ic_prpenc":1 [] - entity 48: ipu2_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev9 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf capture":0 [] - entity 51: ipu2_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video3 pad0: Sink <- "ipu2_ic_prpvf":1 [] - entity 57: ipu1_csi0 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev10 pad0: Sink [fmt:UYVY8_2X8/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range crop.bounds:(0,0)/1280x720 crop:(0,0)/1280x720 compose.bounds:(0,0)/1280x720 compose:(0,0)/1280x720] <- "ipu1_csi0_mux":2 [ENABLED] pad1: Source [fmt:AYUV8_1X32/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range] -> "ipu1_csi0 capture":0 [ENABLED] - entity 61: ipu1_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video4 pad0: Sink <- "ipu1_csi0":2 [ENABLED] - entity 67: ipu1_csi1 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev11 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_csi1 capture":0 [] - entity 71: ipu1_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video5 pad0: Sink <- "ipu1_csi1":2 [] - entity 77: ipu2_csi0 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev12 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi0 capture":0 [] - entity 81: ipu2_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video6 pad0: Sink <- "ipu2_csi0":2 [] - entity 87: ipu2_csi1 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev13 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] <- "ipu2_csi1_mux":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi1 capture":0 [] - entity 91: ipu2_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video7 pad0: Sink <- "ipu2_csi1":2 [] - entity 97: ipu1_csi0_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev14 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] <- "tda19971 2-0048":0 [ENABLED] pad2: Source [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] -> "ipu1_csi0":0 [ENABLED] - entity 101: ipu2_csi1_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev15 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:unknown/0x0] <- "adv7180 2-0020":0 [] pad2: Source [fmt:unknown/0x0] -> "ipu2_csi1":0 [] v4l2-compliance test results: - with the following kernel patches: v4l2-subdev: clear reserved fields v4l2-subdev: without controls return -ENOTTY $ v4l2-compliance -u1 v4l2-compliance SHA : b2f8f9049056eb6f9e028927dacb2c715a062df8 Compliance test for device /dev/v4l-subdev1: Media Driver Info: Driver name : imx-media Model : imx-media Serial : Bus info : Media version : 4.15.0 Hardware revision: 0x00000000 (0) Driver version : 4.15.0 Interface Info: ID : 0x0300008f Type : V4L Sub-Device Entity Info: ID : 0x00000003 (3) Name : tda19971 2-0048 Function : WARNING: Unknown Function (00006001), is v4l2-compliance out-of-date? Pad 0x01000004 : Source Link 0x0200006f: to remote pad 0x1000063 of entity 'ipu1_csi0_mux': Data, Enabled Required ioctls: test MC information (see 'Media Driver Info' above): OK Allow for multiple opens: test second /dev/v4l-subdev1 open: OK test for unlimited opens: OK Debug ioctls: test VIDIOC_LOG_STATUS: OK Input ioctls: test VIDIOC_G/S_TUNER/ENUM_FREQ_BANDS: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_S_HW_FREQ_SEEK: OK (Not Supported) test VIDIOC_ENUMAUDIO: OK (Not Supported) test VIDIOC_G/S/ENUMINPUT: OK (Not Supported) test VIDIOC_G/S_AUDIO: OK (Not Supported) Inputs: 0 Audio Inputs: 0 Tuners: 0 Output ioctls: test VIDIOC_G/S_MODULATOR: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_ENUMAUDOUT: OK (Not Supported) test VIDIOC_G/S/ENUMOUTPUT: OK (Not Supported) test VIDIOC_G/S_AUDOUT: OK (Not Supported) Outputs: 0 Audio Outputs: 0 Modulators: 0 Input/Output configuration ioctls: test VIDIOC_ENUM/G/S/QUERY_STD: OK (Not Supported) test VIDIOC_ENUM/G/S/QUERY_DV_TIMINGS: OK test VIDIOC_DV_TIMINGS_CAP: OK test VIDIOC_G/S_EDID: OK Sub-Device ioctls (Source Pad 0): test Try VIDIOC_SUBDEV_ENUM_MBUS_CODE/FRAME_SIZE/FRAME_INTERVAL: OK test Try VIDIOC_SUBDEV_G/S_FMT: OK test Try VIDIOC_SUBDEV_G/S_SELECTION/CROP: OK (Not Supported) test Active VIDIOC_SUBDEV_ENUM_MBUS_CODE/FRAME_SIZE/FRAME_INTERVAL: OK test Active VIDIOC_SUBDEV_G/S_FMT: OK test Active VIDIOC_SUBDEV_G/S_SELECTION/CROP: OK (Not Supported) test VIDIOC_SUBDEV_G/S_FRAME_INTERVAL: OK (Not Supported) Control ioctls: test VIDIOC_QUERY_EXT_CTRL/QUERYMENU: OK test VIDIOC_QUERYCTRL: OK test VIDIOC_G/S_CTRL: OK test VIDIOC_G/S/TRY_EXT_CTRLS: OK test VIDIOC_(UN)SUBSCRIBE_EVENT/DQEVENT: OK test VIDIOC_G/S_JPEGCOMP: OK (Not Supported) Standard Controls: 4 Private Controls: 0 Format ioctls: test VIDIOC_ENUM_FMT/FRAMESIZES/FRAMEINTERVALS: OK (Not Supported) test VIDIOC_G/S_PARM: OK (Not Supported) test VIDIOC_G_FBUF: OK (Not Supported) test VIDIOC_G_FMT: OK (Not Supported) test VIDIOC_TRY_FMT: OK (Not Supported) test VIDIOC_S_FMT: OK (Not Supported) test VIDIOC_G_SLICED_VBI_CAP: OK (Not Supported) test Cropping: OK (Not Supported) test Composing: OK (Not Supported) test Scaling: OK (Not Supported) Codec ioctls: test VIDIOC_(TRY_)ENCODER_CMD: OK (Not Supported) test VIDIOC_G_ENC_INDEX: OK (Not Supported) test VIDIOC_(TRY_)DECODER_CMD: OK (Not Supported) Buffer ioctls: test VIDIOC_REQBUFS/CREATE_BUFS/QUERYBUF: OK (Not Supported) test VIDIOC_EXPBUF: OK (Not Supported) Total: 47, Succeeded: 47, Failed: 0, Warnings: 0 $ v4l2-compliance -M0 v4l2-compliance SHA : b2f8f9049056eb6f9e028927dacb2c715a062df8 Compliance test for device /dev/media0: Media Driver Info: Driver name : imx-media Model : imx-media Serial : Bus info : Media version : 4.15.0 Hardware revision: 0x00000000 (0) Driver version : 4.15.0 Required ioctls: test MEDIA_IOC_DEVICE_INFO: OK Allow for multiple opens: test second /dev/media0 open: OK test MEDIA_IOC_DEVICE_INFO: OK test for unlimited opens: OK Media Controller ioctls: fail: v4l2-test-media.cpp(96): function == MEDIA_ENT_F_V4L2_SUBDEV_UNKNOWN fail: v4l2-test-media.cpp(161): checkFunction(ent.function, true) test MEDIA_IOC_G_TOPOLOGY: FAIL fail: v4l2-test-media.cpp(290): num_data_links != num_links test MEDIA_IOC_ENUM_ENTITIES/LINKS: FAIL test MEDIA_IOC_SETUP_LINK: OK Total: 7, Succeeded: 5, Failed: 2, Warnings: 0 ^^^^ above failures are due to imx/adv7180 not setting a valid entity function Hans Verkuil (1): v4l2-dv-timings: add v4l2_hdmi_colorimetry() Tim Harvey (7): media: v4l-ioctl: fix clearing pad for VIDIOC_DV_TIMIGNS_CAP media: add digital video decoder video interface entity functions MAINTAINERS: add entry for NXP TDA1997x driver media: dt-bindings: Add bindings for TDA1997X media: i2c: Add TDA1997x HDMI receiver driver ARM: dts: imx: Add TDA19971 HDMI Receiver to GW54xx ARM: dts: imx: Add TDA19971 HDMI Receiver to GW551x .../devicetree/bindings/media/i2c/tda1997x.txt | 179 ++ Documentation/media/uapi/mediactl/media-types.rst | 11 + MAINTAINERS | 8 + arch/arm/boot/dts/imx6q-gw54xx.dts | 105 + arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 29 +- arch/arm/boot/dts/imx6qdl-gw551x.dtsi | 138 + drivers/media/i2c/Kconfig | 9 + drivers/media/i2c/Makefile | 1 + drivers/media/i2c/tda1997x.c | 2822 ++++++++++++++++++++ drivers/media/i2c/tda1997x_regs.h | 641 +++++ drivers/media/v4l2-core/v4l2-dv-timings.c | 141 + drivers/media/v4l2-core/v4l2-ioctl.c | 2 +- include/dt-bindings/media/tda1997x.h | 74 + include/media/i2c/tda1997x.h | 42 + include/media/v4l2-dv-timings.h | 21 + include/uapi/linux/media.h | 5 + 16 files changed, 4224 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/media/i2c/tda1997x.txt create mode 100644 drivers/media/i2c/tda1997x.c create mode 100644 drivers/media/i2c/tda1997x_regs.h create mode 100644 include/dt-bindings/media/tda1997x.h create mode 100644 include/media/i2c/tda1997x.h -- 2.7.4

2 10

[alsa-devel] [PATCH] ASoC: Intel: Fix build
by Mark Brown 15 Feb '18

15 Feb '18

One of the recently introduced commits hadn't been build tested, fix that. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdac_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 60bea9d69fc0..84f7a7a36e4b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -722,12 +722,12 @@ static void hdac_hdmi_set_power_state(struct hdac_ext_device *edev, unsigned int state; if (get_wcaps(&edev->hdev, nid) & AC_WCAP_POWER) { - if (!snd_hdac_check_power_state(&edev->hdac, nid, pwr_state)) { + if (!snd_hdac_check_power_state(&edev->hdev, nid, pwr_state)) { for (count = 0; count < 10; count++) { - snd_hdac_codec_read(&edev->hdac, nid, 0, + snd_hdac_codec_read(&edev->hdev, nid, 0, AC_VERB_SET_POWER_STATE, pwr_state); - state = snd_hdac_sync_power_state(&edev->hdac, + state = snd_hdac_sync_power_state(&edev->hdev, nid, pwr_state); if (!(state & AC_PWRST_ERROR)) break; -- 2.16.1

1 1

[alsa-devel] [asoc:topic/intel 4/4] sound/soc/codecs/hdac_hdmi.c:725:42: error: 'struct hdac_ext_device' has no member named 'hdac'; did you mean 'hdev'?
by kbuild test robot 15 Feb '18

15 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 753597fbb7031ff147d4f2862426699e4ad8efca commit: 753597fbb7031ff147d4f2862426699e4ad8efca [4/4] ASoC: hdac_hdmi : Ensuring proper setting of output widget power state config: i386-randconfig-x075-201806 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 753597fbb7031ff147d4f2862426699e4ad8efca # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/codecs/hdac_hdmi.c: In function 'hdac_hdmi_set_power_state': >> sound/soc/codecs/hdac_hdmi.c:725:42: error: 'struct hdac_ext_device' has no member named 'hdac'; did you mean 'hdev'? if (!snd_hdac_check_power_state(&edev->hdac, nid, pwr_state)) { ^~~~ hdev sound/soc/codecs/hdac_hdmi.c:727:32: error: 'struct hdac_ext_device' has no member named 'hdac'; did you mean 'hdev'? snd_hdac_codec_read(&edev->hdac, nid, 0, ^~~~ hdev sound/soc/codecs/hdac_hdmi.c:730:46: error: 'struct hdac_ext_device' has no member named 'hdac'; did you mean 'hdev'? state = snd_hdac_sync_power_state(&edev->hdac, ^~~~ hdev vim +725 sound/soc/codecs/hdac_hdmi.c 717 718 static void hdac_hdmi_set_power_state(struct hdac_ext_device *edev, 719 hda_nid_t nid, unsigned int pwr_state) 720 { 721 int count; 722 unsigned int state; 723 724 if (get_wcaps(&edev->hdev, nid) & AC_WCAP_POWER) { > 725 if (!snd_hdac_check_power_state(&edev->hdac, nid, pwr_state)) { 726 for (count = 0; count < 10; count++) { 727 snd_hdac_codec_read(&edev->hdac, nid, 0, 728 AC_VERB_SET_POWER_STATE, 729 pwr_state); 730 state = snd_hdac_sync_power_state(&edev->hdac, 731 nid, pwr_state); 732 if (!(state & AC_PWRST_ERROR)) 733 break; 734 } 735 } 736 737 } 738 } 739 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix suspend/resume
by Fabio Estevam 15 Feb '18

15 Feb '18

Commit 8419caa72702 ("ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF") causes the sgtl5000 to fail after a suspend/resume sequence: # aplay /media/a2002011001-e02.wav Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2051: write error: Input/output error Clearing the SGTL5000_REFTOP_POWERUP bit causes the issue after suspend/resume, so do not clear this bit in the same way we used to do previously to the aforementioned commit. As sgtl5000_set_bias_level() only deals with SGTL5000_REFTOP_POWERUP, we can simply remove it as this bit is already set inside sgtl5000_set_power_regs(). Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/sgtl5000.c | 31 ------------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c26ac22..7bae303 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -858,36 +858,6 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -/* - * set dac bias - * common state changes: - * startup: - * off --> standby --> prepare --> on - * standby --> prepare --> on - * - * stop: - * on --> prepare --> standby - */ -static int sgtl5000_set_bias_level(struct snd_soc_component *component, - enum snd_soc_bias_level level) -{ - switch (level) { - case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: - case SND_SOC_BIAS_STANDBY: - snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER, - SGTL5000_REFTOP_POWERUP, - SGTL5000_REFTOP_POWERUP); - break; - case SND_SOC_BIAS_OFF: - snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER, - SGTL5000_REFTOP_POWERUP, 0); - break; - } - - return 0; -} - #define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE |\ @@ -1245,7 +1215,6 @@ static int sgtl5000_probe(struct snd_soc_component *component) static const struct snd_soc_component_driver sgtl5000_driver = { .probe = sgtl5000_probe, - .set_bias_level = sgtl5000_set_bias_level, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), .dapm_widgets = sgtl5000_dapm_widgets, -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: hdac_hdmi: Ensuring proper setting of output widget power state
by abhijeet.kumar＠intel.com 15 Feb '18

15 Feb '18

From: Abhijeet Kumar <abhijeet.kumar(a)intel.com> When we change the resolution of DP pannel or hot plug-unplug it while playing an audio clip,sometimes we observe a silent playback(no audio). During no audio condition, we have noticed that the power state of the pin or the connector is D3. Optimzing the way we set the power could mitigate the issue.With this changes the verb is sent to set the power state and response is received. Thus ensuring power state is set. Signed-off-by: Abhijeet Kumar <abhijeet.kumar(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index f3b4f4dfae6a..e24caecf0a4f 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -718,7 +718,7 @@ static void hdac_hdmi_set_power_state(struct hdac_ext_device *edev, { if (get_wcaps(&edev->hdac, nid) & AC_WCAP_POWER) { if (!snd_hdac_check_power_state(&edev->hdac, nid, pwr_state)) - snd_hdac_codec_write(&edev->hdac, nid, 0, + snd_hdac_codec_read(&edev->hdac, nid, 0, AC_VERB_SET_POWER_STATE, pwr_state); } } -- 1.9.1

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[alsa-devel] [PATCH][next] ASoC: uniphier: remove redundant check of blk_id
by Colin King 15 Feb '18

15 Feb '18

From: Colin Ian King <colin.king(a)canonical.com> The check of blk_id == AUD_CLK_IO is redundant as it also being performed in the following switch statement with the same return of -ENOTSUPP. Fix this by removing the redundant comparison. Detected by CoverityScan, CID#1465227 ("Logically dead code") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/uniphier/aio-cpu.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 55f3248a31fd..7cf2316c69a2 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -134,9 +134,6 @@ static int uniphier_aio_set_sysclk(struct snd_soc_dai *dai, int clk_id, bool pll_auto = false; int pll_id, div_id; - if (clk_id == AUD_CLK_IO) - return -ENOTSUPP; - switch (clk_id) { case AUD_CLK_IO: return -ENOTSUPP; -- 2.15.1

2 1

[alsa-devel] Problem with microphone of ALC3236
by Tobias Bora 15 Feb '18

15 Feb '18

Hello, I've been having lot's of troubles with the microphones with my ALC3236 sound card. By default (tested on ubuntu, fedora, and currently on debian sid) the internal microphone stops working when the mic boost is enabled (via alsamixer or when you set the sound to more than something like 30% on pulseaudio). After reading this discution <http://mailman.alsa-project.org/pipermail/alsa-devel/2015-April/091325.html> that was talking about a problem very similar to mine, I tried to change the file /etc/modprobe.d/alsa-base.conf into options snd-hda-intel model=,dell-headset-multi (I also tried lot's of other models) But none of them is really working. Or only the internal microphone is working and I can't use headsets, or only the headsets are usable and not the internal microphone, or sometimes neigther work... Do you have a solution for that? I'm pretty sure that it's a problem of "bad pin assignment", but I'm not an expert at all ^^ You can find here <https://www.youtube.com/watch?v=MTbVfRXMJ2I> someone that has a similar bug on Mac and that manage to solve it using strange tricks involving pin configuration, Do you know how I could finally get a working microphone ? Thank you!

1 0

[alsa-devel] [PATCH v2] ASoC: rt5645: Introduce mapping for ACPI-defined GPIO
by Carlo Caione 15 Feb '18

15 Feb '18

From: Carlo Caione <carlo(a)endlessm.com> On at least one notebook (ECS EF20EA) the 'hp-detect' GPIO is defined in the DSDT table by the ACPI GpioIo resources in _CRS. Add this mapping to the rt5645 driver to enable the jack detection also on non-DT platforms. Method (_CRS, 0, NotSerialized) { Name (SBUF, ResourceTemplate () { I2cSerialBusV2 (0x001A, ControllerInitiated, 0x00061A80, AddressingMode7Bit, "\\_SB.PCI0.I2C2", 0x00, ResourceConsumer, , Exclusive, ) GpioInt (Edge, ActiveBoth, SharedAndWake, PullNone, 0x0000, "\\_SB.GPO3", 0x00, ResourceConsumer, , ) { // Pin list 0x004F } GpioIo (Shared, PullDefault, 0x0000, 0x0000, IoRestrictionInputOnly, "\\_SB.GPO3", 0x00, ResourceConsumer, , ) { // Pin list 0x004F } }) Return (SBUF) /* \_SB_.PCI0.I2C2.RTK2._CRS.SBUF */ } Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- Changelog: v2: - s/ACPI_HANDLE/has_acpi_companion/ --- sound/soc/codecs/rt5645.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 8f140c8b93ac..8e3d1e4ddb63 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3764,6 +3764,13 @@ static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) return 0; } +static const struct acpi_gpio_params hp_detect_gpios = { 1, 0, false }; + +static const struct acpi_gpio_mapping cht_rt5645_gpios[] = { + { "hp-detect-gpios", &hp_detect_gpios, 1 }, + {}, +}; + static int rt5645_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3779,6 +3786,10 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645 == NULL) return -ENOMEM; + if (has_acpi_companion(&i2c->dev)) + if (devm_acpi_dev_add_driver_gpios(&i2c->dev, cht_rt5645_gpios)) + dev_dbg(&i2c->dev, "Failed to add driver gpios\n"); + rt5645->i2c = i2c; i2c_set_clientdata(i2c, rt5645); -- 2.14.1

3 3

[alsa-devel] [PATCH v11 0/8] TDA1997x HDMI video reciver
by Tim Harvey 15 Feb '18

15 Feb '18

This is a v4l2 subdev driver supporting the TDA1997x HDMI video receiver. I've tested this on a Gateworks GW54xx/GW551x with an IMX6Q/IMX6DL which uses the TDA19971 with 16bits connected to the IMX6 CSI and single-lane I2S audio providing 2-channel audio. For this configuration I've tested both 16bit YUV422 and 8bit BT656 parallel video bus modes. While the driver should support the TDA1993 I do not have one for testing. Further potential development efforts include: - CEC support - HDCP support - TDA19972 support (2 inputs) Media graphs can be found at http://dev.gateworks.com/docs/linux/media See details below for configuration and compliance tests History: v11: - return -ERANGE from tda1997x_detect_std (Hans) - clean up tda1997x_g_input_status (Hans) - show detected timings on resolution change if debug enabled - fix unitialized ret var in tda1997x_query_dv_timings - update log status to show detected timings v10: - removed unnecessary check for !timings in get/set/query dv timings (Hans) - dropped pointless s_stream handler (Hans) - remove need for detected_timings and always use set timings (Hans) v9: - add digital video decoder video interface entity function v8: - fix clearing pad for VIDIOC_DV_TIMIGNS_CAP - support full range of input modes based on timings_cap - add patch to fix clearing pad for VIDIOC_DV_TIMIGINGS - fix available formats for tda19971 bt656 bus width >12 - fix set_format (compliance) - fixed get/set edid (compliance) - add init_cfg to setup default pad config (compliance) - added missing pad checks to get_dv_timings_cap/enum_dv_timings (compliance) - fix alignment of if statement and whitespace in comment (Hans) - move regs to tda1997x_regs.h to clean up (Hans) - add define and sanity check for num of mbus_codes (Hans) v7: - fix interlaced mode - support no AVI infoframe (ie DVI) (Hans) - add support for multiple output formats (Hans) v6: - tda1997x: fix return on regulator enablei in tda1997x_set_power() (Fabio) - tda1997x: fix colorspace handling (Hans) - bindings: added Robs's ack (Rob) - replace copyright with SPDX tag (Philippe) v5: - added v4l2_hdmi_colorimetry() patch from Hans to series - bindings: added Sakari's ack - tda1997x: uppercase string constants - tda1997x: use v4l2_hdmi_rx_coloriemtry to fill format - tda1997x: fix V4L2_CID_DV_RX_RGB_RANGE - tda1997x: fix interlaced mode format - dts: remove leading 0 from unit address - dts: add newline between property list and child node - dts: added missing audmux in GW551x dts v4: - move include/dt-bindings/media/tda1997x.h to bindings patch - clarify port node details in bindings - fix typos - fix default quant range for VGA - fix quant range handling and conv matrix - add additional standards and capabilities to timings_cap v3: - fix typo in dt bindings - added dt bindings for GW551x - use V4L2_DV_BT_FRAME_WIDTH/HEIGHT macros - fixed missing break - use only hdmi_infoframe_log for infoframe logging - simplify tda1997x_s_stream error handling - add delayed work proc to handle hotplug enable/disable - fix set_edid (disable HPD before writing, enable after) - remove enabling edid by default - initialize timings - take quant range into account in colorspace conversion - remove vendor/product tracking (we provide this in log_status via infoframes) - add v4l_controls - add more detail to log_status - calculate vhref generator timings - timing detection fixes (rounding errors, hswidth errors) - rename configure_input/configure_conv functions v2: - encorporate feedback into dt bindings - change audio dt bindings - implement dv timings enum/cap - remove deprecated g_mbus_config op - fix dv_query_timings - add EDID get/set handling - remove max-pixel-rate support - add audio codec DAI support - added media-ctl and v4l2-compliance details v1: - initial RFC Pipeline configuration: $ media-ctl -e 'tda19971 2-0048' /dev/v4l-subdev1 $ v4l2-ctl -d /dev/v4l-subdev1 --set-dv-bt-timings=query BT timings set $ media-ctl --get-v4l2 '"tda19971 2-0048":0' [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] $ media-ctl --link "tda19971 2-0048":0 -> "ipu1_csi0_mux":1[1] $ media-ctl --link "ipu1_csi0_mux":2 -> "ipu1_csi0":0[1] $ media-ctl --link "ipu1_csi0":2 -> "ipu1_csi0 capture":0[1] $ media-ctl --set-v4l2 'tda19971 2-0048':0[fmt:UYVY8_2X8/1280x720] $ media-ctl --set-v4l2 'ipu1_csi0_mux':2[fmt:UYVY8_2X8/1280x720] $ media-ctl --set-v4l2 'ipu1_csi0':0[fmt:UYVY8_2X8/1280x720] $ gst-launch-1.0 v4l2src device=/dev/video4 ! video/x-raw,width=1280,height=720,format=UYVY ! jpegenc ! rtpjpegpay ! udpsink host=172.24.40.6 port=5000 $ media-ctl -d /dev/media0 -p Media controller API version 4.15.0 Media device information ------------------------ driver imx-media model imx-media serial bus info hw revision 0x0 driver version 4.15.0 Device topology - entity 1: adv7180 2-0020 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 20004 device node name /dev/v4l-subdev0 pad0: Source [fmt:UYVY8_2X8/720x480 field:interlaced colorspace:smpte170m] -> "ipu2_csi1_mux":1 [] - entity 3: tda19971 2-0048 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev1 pad0: Source [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] [dv.caps:BT.656/1120 min:640x350@13000000 max:1920x1200@165000000 stds:CEA-861,DMT,CVT,GTF caps:interlaced,progressive,reduced-blanking,custom] [dv.detect:BT.656/1120 1280x720p60 (1650x750) stds:CEA-861 flags:can-reduce-fps,CE-video,has-cea861-vic] [dv.current:BT.656/1120 1280x720p60 (1650x750) stds:CEA-861 flags:can-reduce-fps,CE-video,has-cea861-vic] -> "ipu1_csi0_mux":1 [ENABLED] - entity 5: ipu1_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev2 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] - entity 9: ipu2_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev3 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] - entity 13: ipu1_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev4 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_vdic":2 [] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf":0 [] - entity 17: ipu1_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev5 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc capture":0 [] - entity 20: ipu1_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video0 pad0: Sink <- "ipu1_ic_prpenc":1 [] - entity 26: ipu1_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev6 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf capture":0 [] - entity 29: ipu1_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video1 pad0: Sink <- "ipu1_ic_prpvf":1 [] - entity 35: ipu2_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev7 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_vdic":2 [] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf":0 [] - entity 39: ipu2_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev8 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc capture":0 [] - entity 42: ipu2_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video2 pad0: Sink <- "ipu2_ic_prpenc":1 [] - entity 48: ipu2_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev9 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf capture":0 [] - entity 51: ipu2_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video3 pad0: Sink <- "ipu2_ic_prpvf":1 [] - entity 57: ipu1_csi0 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev10 pad0: Sink [fmt:UYVY8_2X8/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range crop.bounds:(0,0)/1280x720 crop:(0,0)/1280x720 compose.bounds:(0,0)/1280x720 compose:(0,0)/1280x720] <- "ipu1_csi0_mux":2 [ENABLED] pad1: Source [fmt:AYUV8_1X32/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range] -> "ipu1_csi0 capture":0 [ENABLED] - entity 61: ipu1_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video4 pad0: Sink <- "ipu1_csi0":2 [ENABLED] - entity 67: ipu1_csi1 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev11 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_csi1 capture":0 [] - entity 71: ipu1_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video5 pad0: Sink <- "ipu1_csi1":2 [] - entity 77: ipu2_csi0 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev12 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi0 capture":0 [] - entity 81: ipu2_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video6 pad0: Sink <- "ipu2_csi0":2 [] - entity 87: ipu2_csi1 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev13 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] <- "ipu2_csi1_mux":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi1 capture":0 [] - entity 91: ipu2_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video7 pad0: Sink <- "ipu2_csi1":2 [] - entity 97: ipu1_csi0_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev14 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] <- "tda19971 2-0048":0 [ENABLED] pad2: Source [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] -> "ipu1_csi0":0 [ENABLED] - entity 101: ipu2_csi1_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev15 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:unknown/0x0] <- "adv7180 2-0020":0 [] pad2: Source [fmt:unknown/0x0] -> "ipu2_csi1":0 [] v4l2-compliance test results: - with the following kernel patches: v4l2-subdev: clear reserved fields v4l2-subdev: without controls return -ENOTTY $ v4l2-compliance -u1 v4l2-compliance SHA : b2f8f9049056eb6f9e028927dacb2c715a062df8 Compliance test for device /dev/v4l-subdev1: Media Driver Info: Driver name : imx-media Model : imx-media Serial : Bus info : Media version : 4.15.0 Hardware revision: 0x00000000 (0) Driver version : 4.15.0 Interface Info: ID : 0x0300008f Type : V4L Sub-Device Entity Info: ID : 0x00000003 (3) Name : tda19971 2-0048 Function : WARNING: Unknown Function (00006001), is v4l2-compliance out-of-date? Pad 0x01000004 : Source Link 0x0200006f: to remote pad 0x1000063 of entity 'ipu1_csi0_mux': Data, Enabled Required ioctls: test MC information (see 'Media Driver Info' above): OK Allow for multiple opens: test second /dev/v4l-subdev1 open: OK test for unlimited opens: OK Debug ioctls: test VIDIOC_LOG_STATUS: OK Input ioctls: test VIDIOC_G/S_TUNER/ENUM_FREQ_BANDS: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_S_HW_FREQ_SEEK: OK (Not Supported) test VIDIOC_ENUMAUDIO: OK (Not Supported) test VIDIOC_G/S/ENUMINPUT: OK (Not Supported) test VIDIOC_G/S_AUDIO: OK (Not Supported) Inputs: 0 Audio Inputs: 0 Tuners: 0 Output ioctls: test VIDIOC_G/S_MODULATOR: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_ENUMAUDOUT: OK (Not Supported) test VIDIOC_G/S/ENUMOUTPUT: OK (Not Supported) test VIDIOC_G/S_AUDOUT: OK (Not Supported) Outputs: 0 Audio Outputs: 0 Modulators: 0 Input/Output configuration ioctls: test VIDIOC_ENUM/G/S/QUERY_STD: OK (Not Supported) test VIDIOC_ENUM/G/S/QUERY_DV_TIMINGS: OK test VIDIOC_DV_TIMINGS_CAP: OK test VIDIOC_G/S_EDID: OK Sub-Device ioctls (Source Pad 0): test Try VIDIOC_SUBDEV_ENUM_MBUS_CODE/FRAME_SIZE/FRAME_INTERVAL: OK test Try VIDIOC_SUBDEV_G/S_FMT: OK test Try VIDIOC_SUBDEV_G/S_SELECTION/CROP: OK (Not Supported) test Active VIDIOC_SUBDEV_ENUM_MBUS_CODE/FRAME_SIZE/FRAME_INTERVAL: OK test Active VIDIOC_SUBDEV_G/S_FMT: OK test Active VIDIOC_SUBDEV_G/S_SELECTION/CROP: OK (Not Supported) test VIDIOC_SUBDEV_G/S_FRAME_INTERVAL: OK (Not Supported) Control ioctls: test VIDIOC_QUERY_EXT_CTRL/QUERYMENU: OK test VIDIOC_QUERYCTRL: OK test VIDIOC_G/S_CTRL: OK test VIDIOC_G/S/TRY_EXT_CTRLS: OK test VIDIOC_(UN)SUBSCRIBE_EVENT/DQEVENT: OK test VIDIOC_G/S_JPEGCOMP: OK (Not Supported) Standard Controls: 4 Private Controls: 0 Format ioctls: test VIDIOC_ENUM_FMT/FRAMESIZES/FRAMEINTERVALS: OK (Not Supported) test VIDIOC_G/S_PARM: OK (Not Supported) test VIDIOC_G_FBUF: OK (Not Supported) test VIDIOC_G_FMT: OK (Not Supported) test VIDIOC_TRY_FMT: OK (Not Supported) test VIDIOC_S_FMT: OK (Not Supported) test VIDIOC_G_SLICED_VBI_CAP: OK (Not Supported) test Cropping: OK (Not Supported) test Composing: OK (Not Supported) test Scaling: OK (Not Supported) Codec ioctls: test VIDIOC_(TRY_)ENCODER_CMD: OK (Not Supported) test VIDIOC_G_ENC_INDEX: OK (Not Supported) test VIDIOC_(TRY_)DECODER_CMD: OK (Not Supported) Buffer ioctls: test VIDIOC_REQBUFS/CREATE_BUFS/QUERYBUF: OK (Not Supported) test VIDIOC_EXPBUF: OK (Not Supported) Total: 47, Succeeded: 47, Failed: 0, Warnings: 0 $ v4l2-compliance -M0 v4l2-compliance SHA : b2f8f9049056eb6f9e028927dacb2c715a062df8 Compliance test for device /dev/media0: Media Driver Info: Driver name : imx-media Model : imx-media Serial : Bus info : Media version : 4.15.0 Hardware revision: 0x00000000 (0) Driver version : 4.15.0 Required ioctls: test MEDIA_IOC_DEVICE_INFO: OK Allow for multiple opens: test second /dev/media0 open: OK test MEDIA_IOC_DEVICE_INFO: OK test for unlimited opens: OK Media Controller ioctls: fail: v4l2-test-media.cpp(96): function == MEDIA_ENT_F_V4L2_SUBDEV_UNKNOWN fail: v4l2-test-media.cpp(161): checkFunction(ent.function, true) test MEDIA_IOC_G_TOPOLOGY: FAIL fail: v4l2-test-media.cpp(290): num_data_links != num_links test MEDIA_IOC_ENUM_ENTITIES/LINKS: FAIL test MEDIA_IOC_SETUP_LINK: OK Total: 7, Succeeded: 5, Failed: 2, Warnings: 0 ^^^^ above failures are due to imx/adv7180 not setting a valid entity function Hans Verkuil (1): v4l2-dv-timings: add v4l2_hdmi_colorimetry() Tim Harvey (7): media: v4l-ioctl: fix clearing pad for VIDIOC_DV_TIMIGNS_CAP media: add digital video decoder video interface entity functions MAINTAINERS: add entry for NXP TDA1997x driver media: dt-bindings: Add bindings for TDA1997X media: i2c: Add TDA1997x HDMI receiver driver ARM: dts: imx: Add TDA19971 HDMI Receiver to GW54xx ARM: dts: imx: Add TDA19971 HDMI Receiver to GW551x .../devicetree/bindings/media/i2c/tda1997x.txt | 179 ++ Documentation/media/uapi/mediactl/media-types.rst | 11 + MAINTAINERS | 8 + arch/arm/boot/dts/imx6q-gw54xx.dts | 105 + arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 29 +- arch/arm/boot/dts/imx6qdl-gw551x.dtsi | 138 + drivers/media/i2c/Kconfig | 9 + drivers/media/i2c/Makefile | 1 + drivers/media/i2c/tda1997x.c | 2822 ++++++++++++++++++++ drivers/media/i2c/tda1997x_regs.h | 641 +++++ drivers/media/v4l2-core/v4l2-dv-timings.c | 141 + drivers/media/v4l2-core/v4l2-ioctl.c | 2 +- include/dt-bindings/media/tda1997x.h | 74 + include/media/i2c/tda1997x.h | 42 + include/media/v4l2-dv-timings.h | 21 + include/uapi/linux/media.h | 5 + 16 files changed, 4224 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/media/i2c/tda1997x.txt create mode 100644 drivers/media/i2c/tda1997x.c create mode 100644 drivers/media/i2c/tda1997x_regs.h create mode 100644 include/dt-bindings/media/tda1997x.h create mode 100644 include/media/i2c/tda1997x.h -- 2.7.4

1 8

[alsa-devel] [PATCH 1/3] ASoC: adau17x1: Use empty struct initializer
by Fabio Estevam 14 Feb '18

14 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> { 0 } only clears the first member of the structure. The first member of the snd_soc_dapm_update struct is a pointer, and writing 0 to a pointer results in a sparse warning. Use the empty struct initializer that clears all the struct members and fixes the sparse warning. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/adau17x1.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index df34349..80c2a06 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -181,7 +181,7 @@ static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct adau *adau = snd_soc_component_get_drvdata(component); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - struct snd_soc_dapm_update update = { 0 }; + struct snd_soc_dapm_update update = {}; unsigned int stream = e->shift_l; unsigned int val, change; int reg; -- 2.7.4

4 13

[alsa-devel] [PATCH v10 0/8] TDA1997x HDMI video reciver
by Tim Harvey 14 Feb '18

14 Feb '18

This is a v4l2 subdev driver supporting the TDA1997x HDMI video receiver. I've tested this on a Gateworks GW54xx/GW551x with an IMX6Q/IMX6DL which uses the TDA19971 with 16bits connected to the IMX6 CSI and single-lane I2S audio providing 2-channel audio. For this configuration I've tested both 16bit YUV422 and 8bit BT656 parallel video bus modes. While the driver should support the TDA1993 I do not have one for testing. Further potential development efforts include: - CEC support - HDCP support - TDA19972 support (2 inputs) Media graphs can be found at http://dev.gateworks.com/docs/linux/media See details below for configuration and compliance tests History: v10: - removed unnecessary check for !timings in get/set/query dv timings (Hans) - dropped pointless s_stream handler (Hans) - remove need for detected_timings and always use set timings (Hans) v9: - add digital video decoder video interface entity function v8: - fix clearing pad for VIDIOC_DV_TIMIGNS_CAP - support full range of input modes based on timings_cap - add patch to fix clearing pad for VIDIOC_DV_TIMIGINGS - fix available formats for tda19971 bt656 bus width >12 - fix set_format (compliance) - fixed get/set edid (compliance) - add init_cfg to setup default pad config (compliance) - added missing pad checks to get_dv_timings_cap/enum_dv_timings (compliance) - fix alignment of if statement and whitespace in comment (Hans) - move regs to tda1997x_regs.h to clean up (Hans) - add define and sanity check for num of mbus_codes (Hans) v7: - fix interlaced mode - support no AVI infoframe (ie DVI) (Hans) - add support for multiple output formats (Hans) v6: - tda1997x: fix return on regulator enablei in tda1997x_set_power() (Fabio) - tda1997x: fix colorspace handling (Hans) - bindings: added Robs's ack (Rob) - replace copyright with SPDX tag (Philippe) v5: - added v4l2_hdmi_colorimetry() patch from Hans to series - bindings: added Sakari's ack - tda1997x: uppercase string constants - tda1997x: use v4l2_hdmi_rx_coloriemtry to fill format - tda1997x: fix V4L2_CID_DV_RX_RGB_RANGE - tda1997x: fix interlaced mode format - dts: remove leading 0 from unit address - dts: add newline between property list and child node - dts: added missing audmux in GW551x dts v4: - move include/dt-bindings/media/tda1997x.h to bindings patch - clarify port node details in bindings - fix typos - fix default quant range for VGA - fix quant range handling and conv matrix - add additional standards and capabilities to timings_cap v3: - fix typo in dt bindings - added dt bindings for GW551x - use V4L2_DV_BT_FRAME_WIDTH/HEIGHT macros - fixed missing break - use only hdmi_infoframe_log for infoframe logging - simplify tda1997x_s_stream error handling - add delayed work proc to handle hotplug enable/disable - fix set_edid (disable HPD before writing, enable after) - remove enabling edid by default - initialize timings - take quant range into account in colorspace conversion - remove vendor/product tracking (we provide this in log_status via infoframes) - add v4l_controls - add more detail to log_status - calculate vhref generator timings - timing detection fixes (rounding errors, hswidth errors) - rename configure_input/configure_conv functions v2: - encorporate feedback into dt bindings - change audio dt bindings - implement dv timings enum/cap - remove deprecated g_mbus_config op - fix dv_query_timings - add EDID get/set handling - remove max-pixel-rate support - add audio codec DAI support - added media-ctl and v4l2-compliance details v1: - initial RFC Pipeline configuration: $ media-ctl -e 'tda19971 2-0048' /dev/v4l-subdev1 $ v4l2-ctl -d /dev/v4l-subdev1 --set-dv-bt-timings=query BT timings set $ media-ctl --get-v4l2 '"tda19971 2-0048":0' [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] $ media-ctl --link "tda19971 2-0048":0 -> "ipu1_csi0_mux":1[1] $ media-ctl --link "ipu1_csi0_mux":2 -> "ipu1_csi0":0[1] $ media-ctl --link "ipu1_csi0":2 -> "ipu1_csi0 capture":0[1] $ media-ctl --set-v4l2 'tda19971 2-0048':0[fmt:UYVY8_2X8/1280x720] $ media-ctl --set-v4l2 'ipu1_csi0_mux':2[fmt:UYVY8_2X8/1280x720] $ media-ctl --set-v4l2 'ipu1_csi0':0[fmt:UYVY8_2X8/1280x720] $ gst-launch-1.0 v4l2src device=/dev/video4 ! video/x-raw,width=1280,height=720,format=UYVY ! jpegenc ! rtpjpegpay ! udpsink host=172.24.40.6 port=5000 $ media-ctl -d /dev/media0 -p Media controller API version 4.15.0 Media device information ------------------------ driver imx-media model imx-media serial bus info hw revision 0x0 driver version 4.15.0 Device topology - entity 1: adv7180 2-0020 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 20004 device node name /dev/v4l-subdev0 pad0: Source [fmt:UYVY8_2X8/720x480 field:interlaced colorspace:smpte170m] -> "ipu2_csi1_mux":1 [] - entity 3: tda19971 2-0048 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev1 pad0: Source [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] [dv.caps:BT.656/1120 min:640x350@13000000 max:1920x1200@165000000 stds:CEA-861,DMT,CVT,GTF caps:interlaced,progressive,reduced-blanking,custom] [dv.detect:BT.656/1120 1280x720p60 (1650x750) stds:CEA-861 flags:can-reduce-fps,CE-video,has-cea861-vic] [dv.current:BT.656/1120 1280x720p60 (1650x750) stds:CEA-861 flags:can-reduce-fps,CE-video,has-cea861-vic] -> "ipu1_csi0_mux":1 [ENABLED] - entity 5: ipu1_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev2 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] - entity 9: ipu2_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev3 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] - entity 13: ipu1_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev4 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_vdic":2 [] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf":0 [] - entity 17: ipu1_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev5 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc capture":0 [] - entity 20: ipu1_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video0 pad0: Sink <- "ipu1_ic_prpenc":1 [] - entity 26: ipu1_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev6 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf capture":0 [] - entity 29: ipu1_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video1 pad0: Sink <- "ipu1_ic_prpvf":1 [] - entity 35: ipu2_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev7 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_vdic":2 [] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf":0 [] - entity 39: ipu2_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev8 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc capture":0 [] - entity 42: ipu2_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video2 pad0: Sink <- "ipu2_ic_prpenc":1 [] - entity 48: ipu2_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev9 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf capture":0 [] - entity 51: ipu2_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video3 pad0: Sink <- "ipu2_ic_prpvf":1 [] - entity 57: ipu1_csi0 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev10 pad0: Sink [fmt:UYVY8_2X8/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range crop.bounds:(0,0)/1280x720 crop:(0,0)/1280x720 compose.bounds:(0,0)/1280x720 compose:(0,0)/1280x720] <- "ipu1_csi0_mux":2 [ENABLED] pad1: Source [fmt:AYUV8_1X32/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/1280x720@1/30 field:none colorspace:srgb xfer:srgb ycbcr:601 quantization:lim-range] -> "ipu1_csi0 capture":0 [ENABLED] - entity 61: ipu1_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video4 pad0: Sink <- "ipu1_csi0":2 [ENABLED] - entity 67: ipu1_csi1 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev11 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_csi1 capture":0 [] - entity 71: ipu1_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video5 pad0: Sink <- "ipu1_csi1":2 [] - entity 77: ipu2_csi0 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev12 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi0 capture":0 [] - entity 81: ipu2_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video6 pad0: Sink <- "ipu2_csi0":2 [] - entity 87: ipu2_csi1 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev13 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] <- "ipu2_csi1_mux":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi1 capture":0 [] - entity 91: ipu2_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video7 pad0: Sink <- "ipu2_csi1":2 [] - entity 97: ipu1_csi0_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev14 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] <- "tda19971 2-0048":0 [ENABLED] pad2: Source [fmt:UYVY8_2X8/1280x720 field:none colorspace:srgb] -> "ipu1_csi0":0 [ENABLED] - entity 101: ipu2_csi1_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev15 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:unknown/0x0] <- "adv7180 2-0020":0 [] pad2: Source [fmt:unknown/0x0] -> "ipu2_csi1":0 [] v4l2-compliance test results: - with the following kernel patches: v4l2-subdev: clear reserved fields v4l2-subdev: without controls return -ENOTTY $ v4l2-compliance -u1 v4l2-compliance SHA : b2f8f9049056eb6f9e028927dacb2c715a062df8 Compliance test for device /dev/v4l-subdev1: Media Driver Info: Driver name : imx-media Model : imx-media Serial : Bus info : Media version : 4.15.0 Hardware revision: 0x00000000 (0) Driver version : 4.15.0 Interface Info: ID : 0x0300008f Type : V4L Sub-Device Entity Info: ID : 0x00000003 (3) Name : tda19971 2-0048 Function : WARNING: Unknown Function (00006001), is v4l2-compliance out-of-date? Pad 0x01000004 : Source Link 0x0200006f: to remote pad 0x1000063 of entity 'ipu1_csi0_mux': Data, Enabled Required ioctls: test MC information (see 'Media Driver Info' above): OK Allow for multiple opens: test second /dev/v4l-subdev1 open: OK test for unlimited opens: OK Debug ioctls: test VIDIOC_LOG_STATUS: OK Input ioctls: test VIDIOC_G/S_TUNER/ENUM_FREQ_BANDS: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_S_HW_FREQ_SEEK: OK (Not Supported) test VIDIOC_ENUMAUDIO: OK (Not Supported) test VIDIOC_G/S/ENUMINPUT: OK (Not Supported) test VIDIOC_G/S_AUDIO: OK (Not Supported) Inputs: 0 Audio Inputs: 0 Tuners: 0 Output ioctls: test VIDIOC_G/S_MODULATOR: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_ENUMAUDOUT: OK (Not Supported) test VIDIOC_G/S/ENUMOUTPUT: OK (Not Supported) test VIDIOC_G/S_AUDOUT: OK (Not Supported) Outputs: 0 Audio Outputs: 0 Modulators: 0 Input/Output configuration ioctls: test VIDIOC_ENUM/G/S/QUERY_STD: OK (Not Supported) test VIDIOC_ENUM/G/S/QUERY_DV_TIMINGS: OK test VIDIOC_DV_TIMINGS_CAP: OK test VIDIOC_G/S_EDID: OK Sub-Device ioctls (Source Pad 0): test Try VIDIOC_SUBDEV_ENUM_MBUS_CODE/FRAME_SIZE/FRAME_INTERVAL: OK test Try VIDIOC_SUBDEV_G/S_FMT: OK test Try VIDIOC_SUBDEV_G/S_SELECTION/CROP: OK (Not Supported) test Active VIDIOC_SUBDEV_ENUM_MBUS_CODE/FRAME_SIZE/FRAME_INTERVAL: OK test Active VIDIOC_SUBDEV_G/S_FMT: OK test Active VIDIOC_SUBDEV_G/S_SELECTION/CROP: OK (Not Supported) test VIDIOC_SUBDEV_G/S_FRAME_INTERVAL: OK (Not Supported) Control ioctls: test VIDIOC_QUERY_EXT_CTRL/QUERYMENU: OK test VIDIOC_QUERYCTRL: OK test VIDIOC_G/S_CTRL: OK test VIDIOC_G/S/TRY_EXT_CTRLS: OK test VIDIOC_(UN)SUBSCRIBE_EVENT/DQEVENT: OK test VIDIOC_G/S_JPEGCOMP: OK (Not Supported) Standard Controls: 4 Private Controls: 0 Format ioctls: test VIDIOC_ENUM_FMT/FRAMESIZES/FRAMEINTERVALS: OK (Not Supported) test VIDIOC_G/S_PARM: OK (Not Supported) test VIDIOC_G_FBUF: OK (Not Supported) test VIDIOC_G_FMT: OK (Not Supported) test VIDIOC_TRY_FMT: OK (Not Supported) test VIDIOC_S_FMT: OK (Not Supported) test VIDIOC_G_SLICED_VBI_CAP: OK (Not Supported) test Cropping: OK (Not Supported) test Composing: OK (Not Supported) test Scaling: OK (Not Supported) Codec ioctls: test VIDIOC_(TRY_)ENCODER_CMD: OK (Not Supported) test VIDIOC_G_ENC_INDEX: OK (Not Supported) test VIDIOC_(TRY_)DECODER_CMD: OK (Not Supported) Buffer ioctls: test VIDIOC_REQBUFS/CREATE_BUFS/QUERYBUF: OK (Not Supported) test VIDIOC_EXPBUF: OK (Not Supported) Total: 47, Succeeded: 47, Failed: 0, Warnings: 0 $ v4l2-compliance -M0 v4l2-compliance SHA : b2f8f9049056eb6f9e028927dacb2c715a062df8 Compliance test for device /dev/media0: Media Driver Info: Driver name : imx-media Model : imx-media Serial : Bus info : Media version : 4.15.0 Hardware revision: 0x00000000 (0) Driver version : 4.15.0 Required ioctls: test MEDIA_IOC_DEVICE_INFO: OK Allow for multiple opens: test second /dev/media0 open: OK test MEDIA_IOC_DEVICE_INFO: OK test for unlimited opens: OK Media Controller ioctls: fail: v4l2-test-media.cpp(96): function == MEDIA_ENT_F_V4L2_SUBDEV_UNKNOWN fail: v4l2-test-media.cpp(161): checkFunction(ent.function, true) test MEDIA_IOC_G_TOPOLOGY: FAIL fail: v4l2-test-media.cpp(290): num_data_links != num_links test MEDIA_IOC_ENUM_ENTITIES/LINKS: FAIL test MEDIA_IOC_SETUP_LINK: OK Total: 7, Succeeded: 5, Failed: 2, Warnings: 0 ^^^^ above failures are due to imx/adv7180 not setting a valid entity function Hans Verkuil (1): v4l2-dv-timings: add v4l2_hdmi_colorimetry() Tim Harvey (7): media: v4l-ioctl: fix clearing pad for VIDIOC_DV_TIMIGNS_CAP media: add digital video decoder video interface entity functions MAINTAINERS: add entry for NXP TDA1997x driver media: dt-bindings: Add bindings for TDA1997X media: i2c: Add TDA1997x HDMI receiver driver ARM: dts: imx: Add TDA19971 HDMI Receiver to GW54xx ARM: dts: imx: Add TDA19971 HDMI Receiver to GW551x .../devicetree/bindings/media/i2c/tda1997x.txt | 179 ++ Documentation/media/uapi/mediactl/media-types.rst | 11 + MAINTAINERS | 8 + arch/arm/boot/dts/imx6q-gw54xx.dts | 105 + arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 29 +- arch/arm/boot/dts/imx6qdl-gw551x.dtsi | 138 + drivers/media/i2c/Kconfig | 9 + drivers/media/i2c/Makefile | 1 + drivers/media/i2c/tda1997x.c | 2807 ++++++++++++++++++++ drivers/media/i2c/tda1997x_regs.h | 641 +++++ drivers/media/v4l2-core/v4l2-dv-timings.c | 141 + drivers/media/v4l2-core/v4l2-ioctl.c | 2 +- include/dt-bindings/media/tda1997x.h | 74 + include/media/i2c/tda1997x.h | 42 + include/media/v4l2-dv-timings.h | 21 + include/uapi/linux/media.h | 5 + 16 files changed, 4209 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/media/i2c/tda1997x.txt create mode 100644 drivers/media/i2c/tda1997x.c create mode 100644 drivers/media/i2c/tda1997x_regs.h create mode 100644 include/dt-bindings/media/tda1997x.h create mode 100644 include/media/i2c/tda1997x.h -- 2.7.4

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[alsa-devel] [asoc:topic/sgtl5000 2/2] sound/soc//codecs/sgtl5000.c:1246:22: error: 'codec' undeclared; did you mean 'node'?
by kbuild test robot 14 Feb '18

14 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sgtl5000 head: c7c9a1f60b0aa367ec49eed3589d6dd98dd74e75 commit: c7c9a1f60b0aa367ec49eed3589d6dd98dd74e75 [2/2] Merge branch 'fix/sgtl5000' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-sgtl5000 config: x86_64-randconfig-x008-201806 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout c7c9a1f60b0aa367ec49eed3589d6dd98dd74e75 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc//codecs/sgtl5000.c: In function 'sgtl5000_probe': >> sound/soc//codecs/sgtl5000.c:1246:22: error: 'codec' undeclared (first use in this function); did you mean 'node'? snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, ^~~~~ node sound/soc//codecs/sgtl5000.c:1246:22: note: each undeclared identifier is reported only once for each function it appears in vim +1246 sound/soc//codecs/sgtl5000.c 9b34e6cc3 Zeng Zhaoming 2011-02-24 1190 2f8b31880 Kuninori Morimoto 2018-01-29 1191 static int sgtl5000_probe(struct snd_soc_component *component) 9b34e6cc3 Zeng Zhaoming 2011-02-24 1192 { 9b34e6cc3 Zeng Zhaoming 2011-02-24 1193 int ret; 570c70a60 Fabio Estevam 2017-04-05 1194 u16 reg; 2f8b31880 Kuninori Morimoto 2018-01-29 1195 struct sgtl5000_priv *sgtl5000 = snd_soc_component_get_drvdata(component); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1196 9b34e6cc3 Zeng Zhaoming 2011-02-24 1197 /* power up sgtl5000 */ 2f8b31880 Kuninori Morimoto 2018-01-29 1198 ret = sgtl5000_set_power_regs(component); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1199 if (ret) 9b34e6cc3 Zeng Zhaoming 2011-02-24 1200 goto err; 9b34e6cc3 Zeng Zhaoming 2011-02-24 1201 9b34e6cc3 Zeng Zhaoming 2011-02-24 1202 /* enable small pop, introduce 400ms delay in turning off */ 2f8b31880 Kuninori Morimoto 2018-01-29 1203 snd_soc_component_update_bits(component, SGTL5000_CHIP_REF_CTRL, c251ea7bd Fabio Estevam 2014-11-14 1204 SGTL5000_SMALL_POP, 1); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1205 9b34e6cc3 Zeng Zhaoming 2011-02-24 1206 /* disable short cut detector */ 2f8b31880 Kuninori Morimoto 2018-01-29 1207 snd_soc_component_write(component, SGTL5000_CHIP_SHORT_CTRL, 0); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1208 9b34e6cc3 Zeng Zhaoming 2011-02-24 1209 /* 9b34e6cc3 Zeng Zhaoming 2011-02-24 1210 * set i2s as default input of sound switch 9b34e6cc3 Zeng Zhaoming 2011-02-24 1211 * TODO: add sound switch to control and dapm widge. 9b34e6cc3 Zeng Zhaoming 2011-02-24 1212 */ 2f8b31880 Kuninori Morimoto 2018-01-29 1213 snd_soc_component_write(component, SGTL5000_CHIP_SSS_CTRL, 9b34e6cc3 Zeng Zhaoming 2011-02-24 1214 SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); 2f8b31880 Kuninori Morimoto 2018-01-29 1215 snd_soc_component_write(component, SGTL5000_CHIP_DIG_POWER, 9b34e6cc3 Zeng Zhaoming 2011-02-24 1216 SGTL5000_ADC_EN | SGTL5000_DAC_EN); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1217 9b34e6cc3 Zeng Zhaoming 2011-02-24 1218 /* enable dac volume ramp by default */ 2f8b31880 Kuninori Morimoto 2018-01-29 1219 snd_soc_component_write(component, SGTL5000_CHIP_ADCDAC_CTRL, 9b34e6cc3 Zeng Zhaoming 2011-02-24 1220 SGTL5000_DAC_VOL_RAMP_EN | 9b34e6cc3 Zeng Zhaoming 2011-02-24 1221 SGTL5000_DAC_MUTE_RIGHT | 9b34e6cc3 Zeng Zhaoming 2011-02-24 1222 SGTL5000_DAC_MUTE_LEFT); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1223 570c70a60 Fabio Estevam 2017-04-05 1224 reg = ((sgtl5000->lrclk_strength) << SGTL5000_PAD_I2S_LRCLK_SHIFT | 0x5f); 2f8b31880 Kuninori Morimoto 2018-01-29 1225 snd_soc_component_write(component, SGTL5000_CHIP_PAD_STRENGTH, reg); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1226 2f8b31880 Kuninori Morimoto 2018-01-29 1227 snd_soc_component_write(component, SGTL5000_CHIP_ANA_CTRL, 9b34e6cc3 Zeng Zhaoming 2011-02-24 1228 SGTL5000_HP_ZCD_EN | 9b34e6cc3 Zeng Zhaoming 2011-02-24 1229 SGTL5000_ADC_ZCD_EN); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1230 2f8b31880 Kuninori Morimoto 2018-01-29 1231 snd_soc_component_update_bits(component, SGTL5000_CHIP_MIC_CTRL, bd0593f5f Jean-Michel Hautbois 2014-10-14 1232 SGTL5000_BIAS_R_MASK, bd0593f5f Jean-Michel Hautbois 2014-10-14 1233 sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1234 2f8b31880 Kuninori Morimoto 2018-01-29 1235 snd_soc_component_update_bits(component, SGTL5000_CHIP_MIC_CTRL, e256da84a Gianluca Renzi 2015-09-25 1236 SGTL5000_BIAS_VOLT_MASK, e256da84a Gianluca Renzi 2015-09-25 1237 sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1238 /* 9b34e6cc3 Zeng Zhaoming 2011-02-24 1239 * disable DAP 9b34e6cc3 Zeng Zhaoming 2011-02-24 1240 * TODO: 9b34e6cc3 Zeng Zhaoming 2011-02-24 1241 * Enable DAP in kcontrol and dapm. 9b34e6cc3 Zeng Zhaoming 2011-02-24 1242 */ 2f8b31880 Kuninori Morimoto 2018-01-29 1243 snd_soc_component_write(component, SGTL5000_DAP_CTRL, 0); 9b34e6cc3 Zeng Zhaoming 2011-02-24 1244 c5489f9fc Michal Oleszczyk 2018-02-02 1245 /* Unmute DAC after start */ c5489f9fc Michal Oleszczyk 2018-02-02 @1246 snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, c5489f9fc Michal Oleszczyk 2018-02-02 1247 SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT, 0); c5489f9fc Michal Oleszczyk 2018-02-02 1248 9b34e6cc3 Zeng Zhaoming 2011-02-24 1249 return 0; 9b34e6cc3 Zeng Zhaoming 2011-02-24 1250 9b34e6cc3 Zeng Zhaoming 2011-02-24 1251 err: 9b34e6cc3 Zeng Zhaoming 2011-02-24 1252 return ret; 9b34e6cc3 Zeng Zhaoming 2011-02-24 1253 } 9b34e6cc3 Zeng Zhaoming 2011-02-24 1254 :::::: The code at line 1246 was first introduced by commit :::::: c5489f9fc053c744c609f34b32efca395cc2fdad sgtl5000: change digital_mute policy :::::: TO: Michal Oleszczyk <oleszczyk.m(a)gmail.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] Applied "sgtl5000: change digital_mute policy" to the asoc tree
by Mark Brown 14 Feb '18

14 Feb '18

The patch sgtl5000: change digital_mute policy has been applied to the asoc tree at https://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 c5489f9fc053c744c609f34b32efca395cc2fdad Mon Sep 17 00:00:00 2001 From: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Date: Fri, 2 Feb 2018 13:10:29 +0100 Subject: [PATCH] sgtl5000: change digital_mute policy Current implementation mute codec in global way (DAC block). That means when user routes sound not from I2S but from AUX source (LINE_IN) it also will be muted by alsa core. This should not happen. Signed-off-by: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e1ab5537d27a..c445a0794a27 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -529,10 +529,15 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; - u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + u16 i2s_pwr = SGTL5000_I2S_IN_POWERUP; - snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, - adcdac_ctrl, mute ? adcdac_ctrl : 0); + /* + * During 'digital mute' do not mute DAC + * because LINE_IN would be muted aswell. We want to mute + * only I2S block - this can be done by powering it off + */ + snd_soc_update_bits(codec, SGTL5000_CHIP_DIG_POWER, + i2s_pwr, mute ? 0 : i2s_pwr); return 0; } @@ -1237,6 +1242,10 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) */ snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + /* Unmute DAC after start */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT, 0); + return 0; err: -- 2.16.1

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[alsa-devel] [PATCH v4] ASoC: use DEFINE_SHOW_ATTRIBUTE() to decrease code duplication
by Peng Donglin 14 Feb '18

14 Feb '18

There is some duplicate code in soc-core.c, and the kernel provides DEFINE_SHOW_ATTRIBUTE() helper macro to decrease it in seq_file.h. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- v4: * resend patch using outlook email, because I can't disable gmail word wrap in plain text mode v3: [addressed comments from Andy] * add missing tag v2: [addressed comments from Andy] * modify code change description * do not rename m -> s --- sound/soc/soc-core.c | 45 ++++++--------------------------------------- 1 file changed, 6 insertions(+), 39 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96c44f6..f372f34 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -349,7 +349,7 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) "ASoC: Failed to create codec register debugfs file\n"); } -static int codec_list_seq_show(struct seq_file *m, void *v) +static int codec_list_show(struct seq_file *m, void *v) { struct snd_soc_codec *codec; @@ -362,20 +362,9 @@ static int codec_list_seq_show(struct seq_file *m, void *v) return 0; } +DEFINE_SHOW_ATTRIBUTE(codec_list); -static int codec_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, codec_list_seq_show, NULL); -} - -static const struct file_operations codec_list_fops = { - .open = codec_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static int dai_list_seq_show(struct seq_file *m, void *v) +static int dai_list_show(struct seq_file *m, void *v) { struct snd_soc_component *component; struct snd_soc_dai *dai; @@ -390,20 +379,9 @@ static int dai_list_seq_show(struct seq_file *m, void *v) return 0; } +DEFINE_SHOW_ATTRIBUTE(dai_list); -static int dai_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, dai_list_seq_show, NULL); -} - -static const struct file_operations dai_list_fops = { - .open = dai_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static int platform_list_seq_show(struct seq_file *m, void *v) +static int platform_list_show(struct seq_file *m, void *v) { struct snd_soc_platform *platform; @@ -416,18 +394,7 @@ static int platform_list_seq_show(struct seq_file *m, void *v) return 0; } - -static int platform_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, platform_list_seq_show, NULL); -} - -static const struct file_operations platform_list_fops = { - .open = platform_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +DEFINE_SHOW_ATTRIBUTE(platform_list); static void soc_init_card_debugfs(struct snd_soc_card *card) { -- 2.7.4

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Re: [alsa-devel] INFO: rcu detected stall in memcpy
by Dmitry Vyukov 14 Feb '18

14 Feb '18

On Thu, Jan 4, 2018 at 1:03 PM, syzbot <syzbot+387f48da65cb522abfe8(a)syzkaller.appspotmail.com> wrote: > Hello, > > syzkaller hit the following crash on > 30a7acd573899fd8b8ac39236eff6468b195ac7d > git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > Unfortunately, I don't have any reproducer for this bug yet. > > > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+387f48da65cb522abfe8(a)syzkaller.appspotmail.com > It will help syzbot understand when the bug is fixed. See footer for > details. > If you forward the report, please keep this part and the footer. This looks ALSA-related. +ALSA maintainers. > INFO: rcu_sched detected stalls on CPUs/tasks: > (detected by 0, t=125002 jiffies, g=15654, c=15653, q=133) > All QSes seen, last rcu_sched kthread activity 125002 > (4294863411-4294738409), jiffies_till_next_fqs=3, root ->qsmask 0x0 > syz-executor1 R running task 24936 11943 3739 0x0000000c > Call Trace: > <IRQ> > sched_show_task+0x4a3/0x5e0 kernel/sched/core.c:5198 > print_other_cpu_stall+0x996/0x1090 kernel/rcu/tree.c:1564 > check_cpu_stall.isra.61+0x6e6/0x15b0 kernel/rcu/tree.c:1682 > __rcu_pending kernel/rcu/tree.c:3440 [inline] > rcu_pending kernel/rcu/tree.c:3502 [inline] > rcu_check_callbacks+0x256/0xd00 kernel/rcu/tree.c:2842 > update_process_times+0x30/0x60 kernel/time/timer.c:1628 > tick_sched_handle+0x85/0x160 kernel/time/tick-sched.c:162 > tick_sched_timer+0x42/0x120 kernel/time/tick-sched.c:1194 > __run_hrtimer kernel/time/hrtimer.c:1211 [inline] > __hrtimer_run_queues+0x358/0xe20 kernel/time/hrtimer.c:1275 > hrtimer_interrupt+0x1c2/0x5e0 kernel/time/hrtimer.c:1309 > local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1025 [inline] > smp_apic_timer_interrupt+0x14a/0x700 arch/x86/kernel/apic/apic.c:1050 > apic_timer_interrupt+0xa9/0xb0 arch/x86/entry/entry_64.S:920 > </IRQ> > RIP: 0010:kasan_mem_to_shadow include/linux/kasan.h:30 [inline] > RIP: 0010:memory_is_poisoned_n mm/kasan/kasan.c:210 [inline] > RIP: 0010:memory_is_poisoned mm/kasan/kasan.c:241 [inline] > RIP: 0010:check_memory_region_inline mm/kasan/kasan.c:257 [inline] > RIP: 0010:check_memory_region+0x38/0x190 mm/kasan/kasan.c:267 > RSP: 0000:ffff8801bd2ff868 EFLAGS: 00000212 ORIG_RAX: ffffffffffffff11 > RAX: ffff7fffffffffff RBX: ffffc9000160020b RCX: ffffffff841fceaf > RDX: 0000000000000001 RSI: 0000000000000002 RDI: ffffc9000160020a > RBP: ffff8801bd2ff878 R08: ffffc9000160020a R09: dffffc0000000000 > R10: 0000000000000001 R11: ffffed0037a5ff2e R12: ffffc9000160020a > R13: ffff8801bd2ff970 R14: dffffc0000000000 R15: ffffc9000160020a > memcpy+0x37/0x50 mm/kasan/kasan.c:303 > memcpy include/linux/string.h:344 [inline] > cvt_s16_to_native sound/core/oss/mulaw.c:164 [inline] > mulaw_decode+0x52f/0x770 sound/core/oss/mulaw.c:195 > mulaw_transfer+0x222/0x270 sound/core/oss/mulaw.c:273 > snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 > snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 > snd_pcm_oss_write1 sound/core/oss/pcm_oss.c:1372 [inline] > snd_pcm_oss_write+0x5fe/0x830 sound/core/oss/pcm_oss.c:2646 > __vfs_write+0xef/0x970 fs/read_write.c:480 > vfs_write+0x189/0x510 fs/read_write.c:544 > SYSC_write fs/read_write.c:589 [inline] > SyS_write+0xef/0x220 fs/read_write.c:581 > entry_SYSCALL_64_fastpath+0x23/0x9a > RIP: 0033:0x452ac9 > RSP: 002b:00007fa354a13c58 EFLAGS: 00000212 ORIG_RAX: 0000000000000001 > RAX: ffffffffffffffda RBX: cccccccccccccccd RCX: 0000000000452ac9 > RDX: 00000000fffffeb2 RSI: 0000000020083fc6 RDI: 0000000000000014 > RBP: 00000000000005b5 R08: 0000000000000000 R09: 0000000000000000 > R10: 0000000000000000 R11: 0000000000000212 R12: 00000000006f6998 > R13: 00000000ffffffff R14: 00007fa354a146d4 R15: 0000000000000000 > rcu_sched kthread starved for 125002 jiffies! g15654 c15653 f0x2 > RCU_GP_WAIT_FQS(3) ->state=0x0 ->cpu=0 > rcu_sched R running task 23456 8 2 0x80000000 > Call Trace: > context_switch kernel/sched/core.c:2799 [inline] > __schedule+0x8eb/0x2060 kernel/sched/core.c:3375 > schedule+0xf5/0x430 kernel/sched/core.c:3434 > schedule_timeout+0x118/0x230 kernel/time/timer.c:1793 > rcu_gp_kthread+0x9e5/0x1930 kernel/rcu/tree.c:2314 > kthread+0x33c/0x400 kernel/kthread.c:238 > ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:524 > > > --- > This bug is generated by a dumb bot. It may contain errors. > See https://goo.gl/tpsmEJ for details. > Direct all questions to syzkaller(a)googlegroups.com. > > syzbot will keep track of this bug report. > If you forgot to add the Reported-by tag, once the fix for this bug is > merged > into any tree, please reply to this email with: > #syz fix: exact-commit-title > To mark this as a duplicate of another syzbot report, please reply with: > #syz dup: exact-subject-of-another-report > If it's a one-off invalid bug report, please reply with: > #syz invalid > Note: if the crash happens again, it will cause creation of a new bug > report. > Note: all commands must start from beginning of the line in the email body. > > -- > You received this message because you are subscribed to the Google Groups > "syzkaller-bugs" group. > To unsubscribe from this group and stop receiving emails from it, send an > email to syzkaller-bugs+unsubscribe(a)googlegroups.com. > To view this discussion on the web visit > https://groups.google.com/d/msgid/syzkaller-bugs/001a113f711a23fe310561f21d…. > > For more options, visit https://groups.google.com/d/optout.

2 7

[alsa-devel] [PATCH v4] ASoC: use DEFINE_SHOW_ATTRIBUTE() to decrease code duplication
by Peng Donglin 14 Feb '18

14 Feb '18

There is some duplicate code in soc-core.c, and the kernel provides DEFINE_SHOW_ATTRIBUTE() helper macro to decrease it in seq_file.h. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- v4: * resend patch using outlook email, because I can't disable gmail word wrap in plain text mode v3: [addressed comments from Andy] * add missing tag v2: [addressed comments from Andy] * modify code change description * do not rename m -> s --- sound/soc/soc-core.c | 45 ++++++--------------------------------------- 1 file changed, 6 insertions(+), 39 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96c44f6..f372f34 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -349,7 +349,7 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) "ASoC: Failed to create codec register debugfs file\n"); } -static int codec_list_seq_show(struct seq_file *m, void *v) +static int codec_list_show(struct seq_file *m, void *v) { struct snd_soc_codec *codec; @@ -362,20 +362,9 @@ static int codec_list_seq_show(struct seq_file *m, void *v) return 0; } +DEFINE_SHOW_ATTRIBUTE(codec_list); -static int codec_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, codec_list_seq_show, NULL); -} - -static const struct file_operations codec_list_fops = { - .open = codec_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static int dai_list_seq_show(struct seq_file *m, void *v) +static int dai_list_show(struct seq_file *m, void *v) { struct snd_soc_component *component; struct snd_soc_dai *dai; @@ -390,20 +379,9 @@ static int dai_list_seq_show(struct seq_file *m, void *v) return 0; } +DEFINE_SHOW_ATTRIBUTE(dai_list); -static int dai_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, dai_list_seq_show, NULL); -} - -static const struct file_operations dai_list_fops = { - .open = dai_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static int platform_list_seq_show(struct seq_file *m, void *v) +static int platform_list_show(struct seq_file *m, void *v) { struct snd_soc_platform *platform; @@ -416,18 +394,7 @@ static int platform_list_seq_show(struct seq_file *m, void *v) return 0; } - -static int platform_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, platform_list_seq_show, NULL); -} - -static const struct file_operations platform_list_fops = { - .open = platform_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +DEFINE_SHOW_ATTRIBUTE(platform_list); static void soc_init_card_debugfs(struct snd_soc_card *card) { -- 2.7.4

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[alsa-devel] ASoC: rt5640, rt5651 jack-detect and BYT SST suspend/resume patches / status update
by Hans de Goede 14 Feb '18

14 Feb '18

Hi Pierre-Louis, Carlo, et al, This mail is mostly just FYI, I've been doing a lot of work on $subject lately and the purpose of this mail is to avoid people doing double work. As Pierre-Louis knows already, I've been working on adding support for rt5640 jack-detect, this is harder then it should be really, since the micbias overcurrent detect on the rt5640 seems to be very finicky if I don't enable things in exactly the right order its gets in some sort of "unhappy" state where it still works but only sometimes / it is unreliable. This "unhappy" state looks exactly the same as normal working state from a reg-dump pov. So I've jack-detect including headphones vs headset detect working reliable now, after a reboot, but it becomes wonky again after a suspend/resume. Which let me deeper into the rabbit hole. Suspend/resume while streams are active / allocated in the SST-DSP is broken on Bay Trail. I've written a fix for this and I can now suspend/ resume BYT devices while audio is playing over the speakers and the audio properly resumes playing properly. Also these related errors are gone now: intel_sst_acpi 80860F28:00: FW sent error response 0x4000e intel_sst_acpi 80860F28:00: FW sent error response 0x4000f I've added a quirk to the SST driver for this and I was wondering if I needed this at the platform (byt) level or at the machine-driver (so platform+codec combo) level. I've a Ployer momo7w tablet which uses the bytcr-rt5651 machine-driver, so I started testing with that. So the bytcr-rt5651 machine-driver despite its name does not actually work with bytcr devices, which only have SSP0 available, so I fixed that, then had some issues with the jack-detection Carlo recently added, so I've done a bunch of fixes for those too. I also have some UCM fixes for bytcr-rt5651 + SSP0, I will send a pull-req to Pierre-Louis' UCM github repo for those when I get around to it. The kernel patches for all this can be found here: https://github.com/jwrdegoede/linux-sunxi/commits/master Noe this is somewhat in a state of flux, e.g. the BYT SST suspend/resume fixes are good, but currently only enabled on the bytcr-rt5640 combo, testing has shown I need them on the ployer too, but I still need to rework the quirk handling to move the quirk to the sst_platform_info struct. Also this needs to be rebased on top of 4.16-rc1, I know there will be some minor conflicts in the rt5651 code. Note the rt5651 jack change to fix headset vs headphone detection does not work yet, now it sees everything as headphones ... I probably won't have much time to continue work on this today and as said the main purpose of this mail is to get you aware of these patches to avoid double work. I hope to be able to rebase on top of 4.16-rc1, and do an upstream submission of everything except for the rt5640 jack-detect code sometime the next week. Regards, Hans p.s. Carlo or Pierre-Louis can one of you check using e.g. the left + right speaker test in gnome-control-panel's sound config-applet if that left only goes to the left speaker and right only goes to the right speaker? My bytcr-rt5651 tablet has only one speaker, but I here sound for both the left and right speaker tests which suggests some right-to-left mixing is happening which we don't want on devices with stereo speakers. Could be there is some mixing happening outside of the codec on this tablet, still it would be good to test this.

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[alsa-devel] [PATCH v3] ASoC: use DEFINE_SHOW_ATTRIBUTE() to decrease code duplication
by Donglin Peng 14 Feb '18

14 Feb '18

There is some duplicate code in soc-core.c, and the kernel provides DEFINE_SHOW_ATTRIBUTE() helper macro to decrease it in seq_file.h. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- v3: [addressed comments from Andy] * add missing tag v2: [addressed comments from Andy] * modify code change description * do not rename m -> s --- sound/soc/soc-core.c | 45 ++++++--------------------------------------- 1 file changed, 6 insertions(+), 39 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96c44f6576c9..f372f34d7702 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -349,7 +349,7 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) "ASoC: Failed to create codec register debugfs file\n"); } -static int codec_list_seq_show(struct seq_file *m, void *v) +static int codec_list_show(struct seq_file *m, void *v) { struct snd_soc_codec *codec; @@ -362,20 +362,9 @@ static int codec_list_seq_show(struct seq_file *m, void *v) return 0; } +DEFINE_SHOW_ATTRIBUTE(codec_list); -static int codec_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, codec_list_seq_show, NULL); -} - -static const struct file_operations codec_list_fops = { - .open = codec_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static int dai_list_seq_show(struct seq_file *m, void *v) +static int dai_list_show(struct seq_file *m, void *v) { struct snd_soc_component *component; struct snd_soc_dai *dai; @@ -390,20 +379,9 @@ static int dai_list_seq_show(struct seq_file *m, void *v) return 0; } +DEFINE_SHOW_ATTRIBUTE(dai_list); -static int dai_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, dai_list_seq_show, NULL); -} - -static const struct file_operations dai_list_fops = { - .open = dai_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static int platform_list_seq_show(struct seq_file *m, void *v) +static int platform_list_show(struct seq_file *m, void *v) { struct snd_soc_platform *platform; @@ -416,18 +394,7 @@ static int platform_list_seq_show(struct seq_file *m, void *v) return 0; } - -static int platform_list_seq_open(struct inode *inode, struct file *file) -{ - return single_open(file, platform_list_seq_show, NULL); -} - -static const struct file_operations platform_list_fops = { - .open = platform_list_seq_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +DEFINE_SHOW_ATTRIBUTE(platform_list); static void soc_init_card_debugfs(struct snd_soc_card *card) { -- 2.16.1

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[alsa-devel] [asoc:topic/core 3/3] sound//soc/soc-core.c:3856:8: error: too many arguments to function 'snd_soc_register_dais'
by kbuild test robot 14 Feb '18

14 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/core head: a8768c52c511e92d9c9816cecaec4bca77da8a89 commit: a8768c52c511e92d9c9816cecaec4bca77da8a89 [3/3] ASoC: soc-core: remove legacy_dai_naming from snd_soc_register_dais() config: x86_64-randconfig-x018-201806 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout a8768c52c511e92d9c9816cecaec4bca77da8a89 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound//soc/soc-core.c: In function 'snd_soc_register_codec': >> sound//soc/soc-core.c:3856:8: error: too many arguments to function 'snd_soc_register_dais' ret = snd_soc_register_dais(&codec->component, dai_drv, num_dai, false); ^~~~~~~~~~~~~~~~~~~~~ sound//soc/soc-core.c:3152:12: note: declared here static int snd_soc_register_dais(struct snd_soc_component *component, ^~~~~~~~~~~~~~~~~~~~~ vim +/snd_soc_register_dais +3856 sound//soc/soc-core.c a39f75f79 Xiubo Li 2014-03-26 3847 a39f75f79 Xiubo Li 2014-03-26 3848 if (codec_drv->get_regmap) 886f56922 Lars-Peter Clausen 2014-08-19 3849 codec->component.regmap = codec_drv->get_regmap(dev); a39f75f79 Xiubo Li 2014-03-26 3850 f0fba2ad1 Liam Girdwood 2010-03-17 3851 for (i = 0; i < num_dai; i++) { 273d778ef Kuninori Morimoto 2017-10-11 3852 convert_endianness_formats(&dai_drv[i].playback); 273d778ef Kuninori Morimoto 2017-10-11 3853 convert_endianness_formats(&dai_drv[i].capture); 151ab22cf Mark Brown 2009-05-09 3854 } 151ab22cf Mark Brown 2009-05-09 3855 bb13109d8 Lars-Peter Clausen 2014-06-16 @3856 ret = snd_soc_register_dais(&codec->component, dai_drv, num_dai, false); 5acd7dfbd Kuninori Morimoto 2013-02-25 3857 if (ret < 0) { f42cf8d6a Masanari Iida 2015-02-24 3858 dev_err(dev, "ASoC: Failed to register DAIs: %d\n", ret); bb13109d8 Lars-Peter Clausen 2014-06-16 3859 goto err_cleanup; 26b01ccdc Mark Brown 2010-08-18 3860 } f0fba2ad1 Liam Girdwood 2010-03-17 3861 bb13109d8 Lars-Peter Clausen 2014-06-16 3862 list_for_each_entry(dai, &codec->component.dai_list, list) bb13109d8 Lars-Peter Clausen 2014-06-16 3863 dai->codec = codec; f0fba2ad1 Liam Girdwood 2010-03-17 3864 e1328a832 Kuninori Morimoto 2013-03-07 3865 mutex_lock(&client_mutex); bb13109d8 Lars-Peter Clausen 2014-06-16 3866 snd_soc_component_add_unlocked(&codec->component); 054880feb Mark Brown 2012-04-09 3867 list_add(&codec->list, &codec_list); e1328a832 Kuninori Morimoto 2013-03-07 3868 mutex_unlock(&client_mutex); e1328a832 Kuninori Morimoto 2013-03-07 3869 f4333203e Lars-Peter Clausen 2014-06-16 3870 dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", f4333203e Lars-Peter Clausen 2014-06-16 3871 codec->component.name); 0d0cf00a7 Mark Brown 2008-12-10 3872 return 0; f0fba2ad1 Liam Girdwood 2010-03-17 3873 bb13109d8 Lars-Peter Clausen 2014-06-16 3874 err_cleanup: bb13109d8 Lars-Peter Clausen 2014-06-16 3875 snd_soc_component_cleanup(&codec->component); bb13109d8 Lars-Peter Clausen 2014-06-16 3876 err_free: f0fba2ad1 Liam Girdwood 2010-03-17 3877 kfree(codec); f0fba2ad1 Liam Girdwood 2010-03-17 3878 return ret; 0d0cf00a7 Mark Brown 2008-12-10 3879 } 0d0cf00a7 Mark Brown 2008-12-10 3880 EXPORT_SYMBOL_GPL(snd_soc_register_codec); 0d0cf00a7 Mark Brown 2008-12-10 3881 :::::: The code at line 3856 was first introduced by commit :::::: bb13109d85ba5f8009f1a26d840e70198537a4e3 ASoC: Split component registration into two steps :::::: TO: Lars-Peter Clausen <lars(a)metafoo.de> :::::: CC: Mark Brown <broonie(a)linaro.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v4] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
by Peter Ujfalusi 14 Feb '18

14 Feb '18

In the reset state of the codec we do not have complete playback or capture routes. The audio playback/capture will not work due to missing clock signals on the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down. To make sure that even if all output/input is disconnected the codec is generating clocks, we need to have valid DAPM route in every case to power up the must needed parts of the codec. I have verified that switching DAC (during playback) or ADC (during capture) will stop the I2S clocks, so the only solution is to connect the 'Off' routes as well to output/input. The routes will be only added if the codec is clock master. In case the role changes runtime, the applied routes are removed. Tested on am43x-epos-evm with aic3111 codec in master mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v3: - install or remove the master mode DAPM routes if needed - move the clock master DAPM route 'management' to a separate function Changes since v2: - Leftover debug prints removed. Changes since v1: - Only apply the master mode DAPM routes when the codec is clock master - comments added to explain the need. Regards, Peter sound/soc/codecs/tlv320aic31xx.c | 73 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 72 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 858cb8be445f..bd659c803f14 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -166,6 +166,7 @@ struct aic31xx_priv { unsigned int sysclk; u8 p_div; int rate_div_line; + bool master_dapm_route_applied; }; struct aic31xx_rate_divs { @@ -670,6 +671,29 @@ aic310x_audio_map[] = { {"SPK", NULL, "SPK ClassD"}, }; +/* + * Always connected DAPM routes for codec clock master modes. + * If the codec is the master on the I2S bus, we need to power on components + * to have valid DAC_CLK and also the DACs and ADC for playback/capture. + * Otherwise the codec will not generate clocks on the bus. + */ +static const struct snd_soc_dapm_route +common31xx_cm_audio_map[] = { + {"DAC Left Input", "Off", "DAC IN"}, + {"DAC Right Input", "Off", "DAC IN"}, + + {"HPL", NULL, "DAC Left"}, + {"HPR", NULL, "DAC Right"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_cm_audio_map[] = { + {"MIC1LP P-Terminal", "Off", "MIC1LP"}, + {"MIC1RP P-Terminal", "Off", "MIC1RP"}, + {"MIC1LM P-Terminal", "Off", "MIC1LM"}, + {"MIC1LM M-Terminal", "Off", "MIC1LM"}, +}; + static int aic31xx_add_controls(struct snd_soc_codec *codec) { int ret = 0; @@ -912,6 +936,53 @@ static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static int aic31xx_clock_master_routes(struct snd_soc_codec *codec, + unsigned int fmt) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int ret; + + fmt &= SND_SOC_DAIFMT_MASTER_MASK; + if (fmt == SND_SOC_DAIFMT_CBS_CFS && + aic31xx->master_dapm_route_applied) { + /* + * Remove the DAPM route(s) for codec clock master modes, + * if applied + */ + ret = snd_soc_dapm_del_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_del_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = false; + } else if (fmt != SND_SOC_DAIFMT_CBS_CFS && + !aic31xx->master_dapm_route_applied) { + /* + * Add the needed DAPM route(s) for codec clock master modes, + * if it is not done already + */ + ret = snd_soc_dapm_add_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_add_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = true; + } + + return 0; +} + static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -992,7 +1063,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, AIC31XX_BCLKINV_MASK, iface_reg2); - return 0; + return aic31xx_clock_master_routes(codec, fmt); } static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v2] ASoC: replace platform/codec component
by Kuninori Morimoto 14 Feb '18

14 Feb '18

Hi Mark These are v2 of replace platform/codec to component. Main diff between v1 -> v2 are - Based on v4.15 + asoc/for-next - Added missing new driver's replace - Added Tested-by / Acked-by from each driver author I will post both "replace platform" and "replace codec". Kuninori Morimoto (225): ASoC: remove rtd->platform checck ASoC: soc-utils: replace platform to component ASoC: soc-generic-dmaengine-pcm: replace platform to component ASoC: intel: atom: replace platform to component ASoC: intel: skylake: replace platform to component ASoC: intel: baytrail: replace platform to component ASoC: intel: haswell: replace platform to component ASoC: bcm: cygnus: replace platform to component ASoC: sh: rsnd: replace platform to component ASoC: sh: dma-sh7760: replace platform to component ASoC: sh: fsi: replace platform to component ASoC: sh: siu: replace platform to component ASoC: atmel: replace platform to component ASoC: amd: replace platform to component ASoC: cs47l24: replace platform to component ASoC: rt5514-spi: replace platform to component ASoC: wm5110: replace platform to component ASoC: wm5102: replace platform to component ASoC: davinci-i2s: replace platform to component ASoC: fsl: dma: replace platform to component ASoC: fsl: asrc: replace platform to component ASoC: fsl: imx: replace platform to component ASoC: fsl: mpc5200: replace platform to component ASoC: dwc-pcm: replace platform to component ASoC: kirkwood: replace platform to component ASoC: pxa: mmp: replace platform to component ASoC: pxa: pxa2xx: replace platform to component ASoC: qcom: lpass-platform: replace platform to component ASoC: samsung: idma: replace platform to component ASoC: xtfpga-i2s: replace platform to component ASoC: blackfin: bf5xx-ac97-pcm: replace platform to component ASoC: blackfin: bf5xx-i2s-pcm: replace platform to component ASoC: omap-pcm: replace platform to component ASoC: au1x: dbdma2: replace platform to component ASoC: au1x: dma: replace platform to component ASoC: nuc900-pcm: replace platform to component ASoC: txx9aclc: replace platform to component ASoC: mt2701/mt8173: replace platform to component ASoC: stm: stm32_adfsdm: replace platform to component ASoC: soc-utils: replace codec to component ASoC: ac97: replace codec to component ASoC: wm0010: replace codec to component ASoC: wm2000: replace codec to component ASoC: wm2200: replace codec to component ASoC: wm5100: replace codec to component ASoC: wm8350: replace codec to component ASoC: wm8400: replace codec to component ASoC: wm8580: replace codec to component ASoC: wm8524: replace codec to component ASoC: wm8510: replace codec to component ASoC: wm8523: replace codec to component ASoC: wm8711: replace codec to component ASoC: wm8750: replace codec to component ASoC: wm8737: replace codec to component ASoC: wm8776: replace codec to component ASoC: wm8770: replace codec to component ASoC: wm8727: replace codec to component ASoC: wm8731: replace codec to component ASoC: wm8782: replace codec to component ASoC: wm8728: replace codec to component ASoC: wm8741: replace codec to component ASoC: wm8753: replace codec to component ASoC: wm8804: replace codec to component ASoC: wm8900: replace codec to component ASoC: wm8903: replace codec to component ASoC: wm8955: replace codec to component ASoC: wm8960: replace codec to component ASoC: wm8985: replace codec to component ASoC: wm8971: replace codec to component ASoC: wm8978: replace codec to component ASoC: wm8974: replace codec to component ASoC: wm8990: replace codec to component ASoC: wm8988: replace codec to component ASoC: wm8983: replace codec to component ASoC: wm8962: replace codec to component ASoC: wm8996: replace codec to component ASoC: wm8991: replace codec to component ASoC: wm8995: replace codec to component ASoC: wm8961: replace codec to component ASoC: wm8940: replace codec to component ASoC: wm8904: replace codec to component ASoC: wm9081: replace codec to component ASoC: wm9090: replace codec to component ASoC: wm9705: replace codec to component ASoC: wm9712: replace codec to component ASoC: wm9713: replace codec to component ASoC: wm9867: replace codec to component ASoC: wm1250-ev1: replace codec to component ASoC: wm8993/wm8994/wm8958: replace codec to component ASoC: ak4613: replace codec to component ASoC: ak4642: replace codec to component ASoC: ak5386: replace codec to component ASoC: ak4671: replace codec to component ASoC: ak4104: replace codec to component ASoC: ak4535: replace codec to component ASoC: ak4641: replace codec to component ASoC: ak4554: replace codec to component ASoC: rt274: replace codec to component ASoC: rt5616: replace codec to component ASoC: rt5640: replace codec to component ASoC: rt5651: replace codec to component ASoC: rt5514: replace codec to component ASoC: rt5659: replace codec to component ASoC: rt5670: replace codec to component ASoC: rt5660: replace codec to component ASoC: rt5631: replace codec to component ASoC: rt5665: replace codec to component ASoC: rt5645/rt5677: replace codec to component ASoC: cs4271: replace codec to component ASoC: cs4270: replace codec to component ASoC: cs4349: replace codec to component ASoC: cs4265: replace codec to component ASoC: cs35l32: replace codec to component ASoC: cs35l33: replace codec to component ASoC: cs35l35: replace codec to component ASoC: cs35l34: replace codec to component ASoC: cs42xx8: replace codec to component ASoC: cs42l73: replace codec to component ASoC: cs42l52: replace codec to component ASoC: cs42l56: replace codec to component ASoC: cs42l51: replace codec to component ASoC: cs42l42: replace codec to component ASoC: cs43130: replace codec to component ASoC: cs53l30: replace codec to component ASoC: da732x: replace codec to component ASoC: da7210: replace codec to component ASoC: da7218: replace codec to component ASoC: da7213: replace codec to component ASoC: da9055: replace codec to component ASoC: max9860: replace codec to component ASoC: max9850: replace codec to component ASoC: max98371: replace codec to component ASoC: max98095: replace codec to component ASoC: max98090: replace codec to component ASoC: max98926: replace codec to component ASoC: max98088: replace codec to component ASoC: max98925: replace codec to component ASoC: max98927: replace codec to component ASoC: max98357a: replace codec to component ASoC: max98373: replace codec to component ASoC: adav80x: replace codec to component ASoC: adau1373: replace codec to component ASoC: adau7002: replace codec to component ASoC: adau1977: replace codec to component ASoC: adau1701: replace codec to component ASoC: adau17x1/adau1761/adau1781: replace codec to component ASoC: tlv320aic23: replace codec to component ASoC: tlv320aic26: replace codec to component ASoC: tlv320aic3x: replace codec to component ASoC: tlv320dac33: replace codec to component ASoC: tlv320aic32x4: replace codec to component ASoC: tlv320aic31xx: replace codec to component ASoC: cx20442: replace codec to component ASoC: tscs42xx: replace codec to component ASoC: pcm179x: replace codec to component ASoC: pcm3008: replace codec to component ASoC: pcm1681: replace codec to component ASoC: pcm512x: replace codec to component ASoC: pcm5102a: replace codec to component ASoC: pcm3168a: replace codec to component ASoC: twl6040: replace codec to component ASoC: twl4030: replace codec to component ASoC: msm8916-wcd-analog: replace codec to component ASoC: msm8916-wcd-digital: replace codec to component ASoC: spdif_transmitter: replace codec to component ASoC: spdif_receiver: replace codec to component ASoC: tas5720: replace codec to component ASoC: tas2552: replace codec to component ASoC: sun4i: replace codec to component ASoC: sun8i: replace codec to component ASoC: ads117x: replace codec to component ASoC: ab8500: replace codec to component ASoC: ad193x: replace codec to component ASoC: ad1836: replace codec to component ASoC: ad1980: replace codec to component ASoC: ad73311: replace codec to component ASoC: atmel-pdmic: replace codec to component ASoC: atmel-classd: replace codec to component ASoC: ssm4567: replace codec to component ASoC: ssm2602: replace codec to component ASoC: ssm2518: replace codec to component ASoC: sta350: replace codec to component ASoC: sta32x: replace codec to component ASoC: sta529: replace codec to component ASoC: tas5086: replace codec to component ASoC: tas571x: replace codec to component ASoC: nau8824: replace codec to component ASoC: nau8810: replace codec to component ASoC: nau8540: replace codec to component ASoC: es8316: replace codec to component ASoC: es7134: replace codec to component ASoC: es8328: replace codec to component ASoC: alc5632: replace codec to component ASoC: alc5623: replace codec to component ASoC: hdmi-codec: replace codec to component ASoC: bt-sco: replace codec to component ASoC: vc4_hdmi: replace codec to component ASoC: zx_aud96p22: replace codec to component ASoC: wl1273: replace codec to component ASoC: hdac_hdmi/nau8825/rt286/rt298/rt5663/da7219: replace codec to component ASoC: arizona: replace codec to component ASoC: mc13783: replace codec to component ASoC: sgtl5000: replace codec to component ASoC: stac9766: replace codec to component ASoC: sirf-audio: replace codec to component ASoC: inno_rk3036: replace codec to component ASoC: isabelle: replace codec to component ASoC: tfa9879: replace codec to component ASoC: dmic: replace codec to component ASoC: sti-sas: replace codec to component ASoC: gtm601: replace codec to component ASoC: 88pm860x: replace codec to component ASoC: pistachio: replace codec to component ASoC: lm49453: replace codec to component ASoC: cq93vc: replace codec to component ASoC: jz4740: replace codec to component ASoC: uda1380: replace codec to component ASoC: ml26124: replace codec to component ASoC: si476x: replace codec to component ASoC: uda134x: replace codec to component ASoC: ics43432: replace codec to component ASoC: pxa/mioa701_wm9713: replace codec to component ASoC: uniphier: evea: replace codec to component ASoC: pcm186x: replace codec to component ASoC: tas6424: replace codec to component drivers/gpu/drm/vc4/vc4_hdmi.c | 47 +- include/sound/dmaengine_pcm.h | 2 + include/sound/soc.h | 13 +- sound/soc/amd/acp-pcm-dma.c | 29 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/atmel/atmel-classd.c | 58 +- sound/soc/atmel/atmel-pcm-pdc.c | 6 +- sound/soc/atmel/atmel-pdmic.c | 60 +- sound/soc/au1x/dbdma2.c | 13 +- sound/soc/au1x/dma.c | 12 +- sound/soc/bcm/cygnus-pcm.c | 7 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 6 +- sound/soc/codecs/88pm860x-codec.c | 166 ++--- sound/soc/codecs/88pm860x-codec.h | 4 +- sound/soc/codecs/ab8500-codec.c | 295 ++++---- sound/soc/codecs/ac97.c | 46 +- sound/soc/codecs/ad1836.c | 68 +- sound/soc/codecs/ad193x-i2c.c | 7 - sound/soc/codecs/ad193x-spi.c | 7 - sound/soc/codecs/ad193x.c | 46 +- sound/soc/codecs/ad1980.c | 87 +-- sound/soc/codecs/ad73311.c | 27 +- sound/soc/codecs/adau1373.c | 84 +-- sound/soc/codecs/adau1701.c | 101 ++- sound/soc/codecs/adau1761.c | 88 +-- sound/soc/codecs/adau1781.c | 51 +- sound/soc/codecs/adau17x1.c | 67 +- sound/soc/codecs/adau17x1.h | 8 +- sound/soc/codecs/adau1977-i2c.c | 7 - sound/soc/codecs/adau1977-spi.c | 7 - sound/soc/codecs/adau1977.c | 59 +- sound/soc/codecs/adau7002.c | 20 +- sound/soc/codecs/adav801.c | 7 - sound/soc/codecs/adav803.c | 7 - sound/soc/codecs/adav80x.c | 130 ++-- sound/soc/codecs/ads117x.c | 27 +- sound/soc/codecs/ak4104.c | 68 +- sound/soc/codecs/ak4535.c | 78 +- sound/soc/codecs/ak4554.c | 27 +- sound/soc/codecs/ak4613.c | 73 +- sound/soc/codecs/ak4641.c | 91 +-- sound/soc/codecs/ak4642.c | 105 ++- sound/soc/codecs/ak4671.c | 67 +- sound/soc/codecs/ak5386.c | 62 +- sound/soc/codecs/alc5623.c | 147 ++-- sound/soc/codecs/alc5632.c | 138 ++-- sound/soc/codecs/arizona.c | 208 +++--- sound/soc/codecs/arizona.h | 24 +- sound/soc/codecs/bt-sco.c | 21 +- sound/soc/codecs/cq93vc.c | 33 +- sound/soc/codecs/cs35l32.c | 45 +- sound/soc/codecs/cs35l33.c | 186 +++-- sound/soc/codecs/cs35l34.c | 127 ++-- sound/soc/codecs/cs35l35.c | 104 ++- sound/soc/codecs/cs4265.c | 98 ++- sound/soc/codecs/cs4270.c | 124 ++-- sound/soc/codecs/cs4271-i2c.c | 7 - sound/soc/codecs/cs4271-spi.c | 7 - sound/soc/codecs/cs4271.c | 105 ++- sound/soc/codecs/cs42l42.c | 154 ++-- sound/soc/codecs/cs42l42.h | 2 +- sound/soc/codecs/cs42l51-i2c.c | 8 - sound/soc/codecs/cs42l51.c | 81 ++- sound/soc/codecs/cs42l52.c | 146 ++-- sound/soc/codecs/cs42l56.c | 158 ++-- sound/soc/codecs/cs42l73.c | 122 ++-- sound/soc/codecs/cs42xx8-i2c.c | 1 - sound/soc/codecs/cs42xx8.c | 57 +- sound/soc/codecs/cs43130.c | 243 ++++--- sound/soc/codecs/cs43130.h | 2 +- sound/soc/codecs/cs4349.c | 39 +- sound/soc/codecs/cs47l24.c | 120 ++- sound/soc/codecs/cs53l30.c | 57 +- sound/soc/codecs/cx20442.c | 89 ++- sound/soc/codecs/da7210.c | 181 +++-- sound/soc/codecs/da7213.c | 328 ++++----- sound/soc/codecs/da7218.c | 498 +++++++------ sound/soc/codecs/da7218.h | 2 +- sound/soc/codecs/da7219-aad.c | 236 +++--- sound/soc/codecs/da7219-aad.h | 12 +- sound/soc/codecs/da7219.c | 307 ++++---- sound/soc/codecs/da7219.h | 2 +- sound/soc/codecs/da732x.c | 241 +++--- sound/soc/codecs/da9055.c | 186 +++-- sound/soc/codecs/dmic.c | 33 +- sound/soc/codecs/es7134.c | 27 +- sound/soc/codecs/es8316.c | 80 +- sound/soc/codecs/es8328-i2c.c | 7 - sound/soc/codecs/es8328-spi.c | 7 - sound/soc/codecs/es8328.c | 147 ++-- sound/soc/codecs/gtm601.c | 28 +- sound/soc/codecs/hdac_hdmi.c | 47 +- sound/soc/codecs/hdac_hdmi.h | 2 +- sound/soc/codecs/hdmi-codec.c | 23 +- sound/soc/codecs/ics43432.c | 16 +- sound/soc/codecs/inno_rk3036.c | 70 +- sound/soc/codecs/isabelle.c | 57 +- sound/soc/codecs/jz4740.c | 50 +- sound/soc/codecs/lm49453.c | 65 +- sound/soc/codecs/max98088.c | 314 ++++---- sound/soc/codecs/max98090.c | 319 ++++---- sound/soc/codecs/max98090.h | 4 +- sound/soc/codecs/max98095.c | 448 ++++++------ sound/soc/codecs/max98095.h | 2 +- sound/soc/codecs/max98357a.c | 29 +- sound/soc/codecs/max98371.c | 43 +- sound/soc/codecs/max98371.h | 1 - sound/soc/codecs/max98373.c | 75 +- sound/soc/codecs/max9850.c | 80 +- sound/soc/codecs/max9860.c | 70 +- sound/soc/codecs/max9867.c | 65 +- sound/soc/codecs/max9867.h | 2 +- sound/soc/codecs/max98925.c | 72 +- sound/soc/codecs/max98925.h | 2 +- sound/soc/codecs/max98926.c | 56 +- sound/soc/codecs/max98926.h | 2 +- sound/soc/codecs/max98927.c | 82 +-- sound/soc/codecs/max98927.h | 2 +- sound/soc/codecs/mc13783.c | 68 +- sound/soc/codecs/ml26124.c | 103 ++- sound/soc/codecs/msm8916-wcd-analog.c | 207 +++--- sound/soc/codecs/msm8916-wcd-digital.c | 115 +-- sound/soc/codecs/nau8540.c | 79 +- sound/soc/codecs/nau8810.c | 73 +- sound/soc/codecs/nau8824.c | 113 ++- sound/soc/codecs/nau8824.h | 2 +- sound/soc/codecs/nau8825.c | 107 ++- sound/soc/codecs/nau8825.h | 2 +- sound/soc/codecs/pcm1681.c | 62 +- sound/soc/codecs/pcm179x-i2c.c | 6 - sound/soc/codecs/pcm179x-spi.c | 6 - sound/soc/codecs/pcm179x.c | 45 +- sound/soc/codecs/pcm179x.h | 1 - sound/soc/codecs/pcm186x-i2c.c | 8 - sound/soc/codecs/pcm186x-spi.c | 8 - sound/soc/codecs/pcm186x.c | 143 ++-- sound/soc/codecs/pcm186x.h | 1 - sound/soc/codecs/pcm3008.c | 36 +- sound/soc/codecs/pcm3168a.c | 51 +- sound/soc/codecs/pcm5102a.c | 16 +- sound/soc/codecs/pcm512x.c | 162 ++--- sound/soc/codecs/rt274.c | 206 +++--- sound/soc/codecs/rt286.c | 198 +++-- sound/soc/codecs/rt286.h | 2 +- sound/soc/codecs/rt298.c | 210 +++--- sound/soc/codecs/rt298.h | 2 +- sound/soc/codecs/rt5514-spi.c | 36 +- sound/soc/codecs/rt5514.c | 124 ++-- sound/soc/codecs/rt5514.h | 2 +- sound/soc/codecs/rt5616.c | 219 +++--- sound/soc/codecs/rt5631.c | 324 +++++---- sound/soc/codecs/rt5640.c | 260 ++++--- sound/soc/codecs/rt5640.h | 6 +- sound/soc/codecs/rt5645.c | 370 +++++----- sound/soc/codecs/rt5645.h | 4 +- sound/soc/codecs/rt5651.c | 220 +++--- sound/soc/codecs/rt5651.h | 4 +- sound/soc/codecs/rt5659.c | 307 ++++---- sound/soc/codecs/rt5659.h | 4 +- sound/soc/codecs/rt5660.c | 145 ++-- sound/soc/codecs/rt5660.h | 2 +- sound/soc/codecs/rt5663.c | 600 ++++++++------- sound/soc/codecs/rt5663.h | 4 +- sound/soc/codecs/rt5665.c | 345 +++++---- sound/soc/codecs/rt5665.h | 2 +- sound/soc/codecs/rt5670.c | 295 ++++---- sound/soc/codecs/rt5670.h | 10 +- sound/soc/codecs/rt5677.c | 205 +++--- sound/soc/codecs/rt5677.h | 4 +- sound/soc/codecs/sgtl5000.c | 173 +++-- sound/soc/codecs/si476x.c | 42 +- sound/soc/codecs/sirf-audio-codec.c | 58 +- sound/soc/codecs/spdif_receiver.c | 26 +- sound/soc/codecs/spdif_transmitter.c | 26 +- sound/soc/codecs/ssm2518.c | 57 +- sound/soc/codecs/ssm2602-i2c.c | 7 - sound/soc/codecs/ssm2602-spi.c | 7 - sound/soc/codecs/ssm2602.c | 77 +- sound/soc/codecs/ssm4567.c | 44 +- sound/soc/codecs/sta32x.c | 134 ++-- sound/soc/codecs/sta350.c | 117 ++- sound/soc/codecs/sta529.c | 69 +- sound/soc/codecs/stac9766.c | 77 +- sound/soc/codecs/sti-sas.c | 84 +-- sound/soc/codecs/tas2552.c | 178 +++-- sound/soc/codecs/tas5086.c | 94 +-- sound/soc/codecs/tas571x.c | 55 +- sound/soc/codecs/tas5720.c | 151 ++-- sound/soc/codecs/tas6424.c | 99 ++- sound/soc/codecs/tfa9879.c | 50 +- sound/soc/codecs/tlv320aic23-i2c.c | 7 - sound/soc/codecs/tlv320aic23-spi.c | 7 - sound/soc/codecs/tlv320aic23.c | 120 +-- sound/soc/codecs/tlv320aic26.c | 83 +-- sound/soc/codecs/tlv320aic31xx.c | 240 +++--- sound/soc/codecs/tlv320aic32x4.c | 207 +++--- sound/soc/codecs/tlv320aic3x.c | 293 ++++---- sound/soc/codecs/tlv320dac33.c | 402 +++++----- sound/soc/codecs/tscs42xx.c | 222 +++--- sound/soc/codecs/twl4030.c | 383 +++++----- sound/soc/codecs/twl6040.c | 269 ++++--- sound/soc/codecs/twl6040.h | 10 +- sound/soc/codecs/uda134x.c | 78 +- sound/soc/codecs/uda1380.c | 160 ++-- sound/soc/codecs/wl1273.c | 93 ++- sound/soc/codecs/wl1273.h | 2 +- sound/soc/codecs/wm0010.c | 181 +++-- sound/soc/codecs/wm1250-ev1.c | 27 +- sound/soc/codecs/wm2000.c | 78 +- sound/soc/codecs/wm2200.c | 157 ++-- sound/soc/codecs/wm5100.c | 260 ++++--- sound/soc/codecs/wm5100.h | 2 +- sound/soc/codecs/wm5102.c | 140 ++-- sound/soc/codecs/wm5110.c | 174 ++--- sound/soc/codecs/wm8350.c | 183 +++-- sound/soc/codecs/wm8350.h | 4 +- sound/soc/codecs/wm8400.c | 211 +++--- sound/soc/codecs/wm8510.c | 149 ++-- sound/soc/codecs/wm8523.c | 98 ++- sound/soc/codecs/wm8524.c | 56 +- sound/soc/codecs/wm8580.c | 150 ++-- sound/soc/codecs/wm8711.c | 116 ++- sound/soc/codecs/wm8727.c | 29 +- sound/soc/codecs/wm8728.c | 82 +-- sound/soc/codecs/wm8731.c | 101 ++- sound/soc/codecs/wm8737.c | 109 ++- sound/soc/codecs/wm8741.c | 133 ++-- sound/soc/codecs/wm8750.c | 115 ++- sound/soc/codecs/wm8753.c | 289 ++++---- sound/soc/codecs/wm8770.c | 131 ++-- sound/soc/codecs/wm8776.c | 105 ++- sound/soc/codecs/wm8782.c | 27 +- sound/soc/codecs/wm8804.c | 100 ++- sound/soc/codecs/wm8900.c | 251 ++++--- sound/soc/codecs/wm8903.c | 211 +++--- sound/soc/codecs/wm8903.h | 2 +- sound/soc/codecs/wm8904.c | 355 +++++---- sound/soc/codecs/wm8940.c | 159 ++-- sound/soc/codecs/wm8955.c | 189 +++-- sound/soc/codecs/wm8958-dsp2.c | 278 +++---- sound/soc/codecs/wm8960.c | 249 +++---- sound/soc/codecs/wm8961.c | 241 +++--- sound/soc/codecs/wm8962.c | 397 +++++----- sound/soc/codecs/wm8962.h | 2 +- sound/soc/codecs/wm8971.c | 108 ++- sound/soc/codecs/wm8974.c | 143 ++-- sound/soc/codecs/wm8978.c | 185 +++-- sound/soc/codecs/wm8983.c | 170 ++--- sound/soc/codecs/wm8985.c | 201 +++-- sound/soc/codecs/wm8988.c | 127 ++-- sound/soc/codecs/wm8990.c | 179 +++-- sound/soc/codecs/wm8991.c | 167 ++--- sound/soc/codecs/wm8993.c | 237 +++--- sound/soc/codecs/wm8994.c | 810 +++++++++++---------- sound/soc/codecs/wm8994.h | 8 +- sound/soc/codecs/wm8995.c | 286 ++++---- sound/soc/codecs/wm8996.c | 401 +++++----- sound/soc/codecs/wm8996.h | 4 +- sound/soc/codecs/wm8997.c | 62 +- sound/soc/codecs/wm8998.c | 88 ++- sound/soc/codecs/wm9081.c | 228 +++--- sound/soc/codecs/wm9090.c | 109 ++- sound/soc/codecs/wm9705.c | 83 +-- sound/soc/codecs/wm9712.c | 109 ++- sound/soc/codecs/wm9713.c | 201 +++-- sound/soc/codecs/wm_adsp.c | 58 +- sound/soc/codecs/wm_adsp.h | 6 +- sound/soc/codecs/wm_hubs.c | 254 +++---- sound/soc/codecs/wm_hubs.h | 18 +- sound/soc/codecs/zx_aud96p22.c | 35 +- sound/soc/davinci/davinci-i2s.c | 13 +- sound/soc/dwc/dwc-pcm.c | 5 +- sound/soc/fsl/fsl_asrc.c | 10 - sound/soc/fsl/fsl_asrc.h | 3 +- sound/soc/fsl/fsl_asrc_dma.c | 14 +- sound/soc/fsl/fsl_dma.c | 27 +- sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/mpc5200_dma.c | 16 +- sound/soc/fsl/wm1133-ev1.c | 6 +- sound/soc/img/pistachio-internal-dac.c | 34 +- sound/soc/intel/atom/sst-atom-controls.c | 36 +- sound/soc/intel/atom/sst-mfld-platform-compress.c | 4 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 32 +- sound/soc/intel/atom/sst-mfld-platform.h | 4 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 66 +- sound/soc/intel/boards/bdw-rt5677.c | 41 +- sound/soc/intel/boards/broadwell.c | 17 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 +- sound/soc/intel/boards/bxt_rt298.c | 12 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 8 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 18 +- sound/soc/intel/boards/haswell.c | 3 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 14 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 14 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 12 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 12 +- sound/soc/intel/boards/skl_rt286.c | 12 +- sound/soc/intel/haswell/sst-haswell-ipc.h | 2 + sound/soc/intel/haswell/sst-haswell-pcm.c | 118 ++- sound/soc/intel/skylake/skl-debug.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 52 +- sound/soc/intel/skylake/skl-topology.c | 26 +- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/intel/skylake/skl.h | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 3 +- sound/soc/kirkwood/kirkwood-i2s.c | 20 +- sound/soc/kirkwood/kirkwood.h | 4 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 16 +- .../soc/mediatek/common/mtk-afe-platform-driver.c | 9 +- .../soc/mediatek/common/mtk-afe-platform-driver.h | 3 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 50 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 42 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 12 +- sound/soc/nuc900/nuc900-pcm.c | 5 +- sound/soc/omap/ams-delta.c | 20 +- sound/soc/omap/omap-abe-twl6040.c | 8 +- sound/soc/omap/omap-pcm.c | 5 +- sound/soc/pxa/mioa701_wm9713.c | 30 +- sound/soc/pxa/mmp-pcm.c | 11 +- sound/soc/pxa/pxa2xx-pcm.c | 5 +- sound/soc/pxa/ttc-dkb.c | 6 +- sound/soc/qcom/lpass-platform.c | 42 +- sound/soc/rockchip/rk3399_gru_sound.c | 4 +- sound/soc/rockchip/rockchip_rt5645.c | 8 +- sound/soc/samsung/idma.c | 5 +- sound/soc/samsung/littlemill.c | 8 +- sound/soc/samsung/lowland.c | 12 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tm2_wm5110.c | 54 +- sound/soc/samsung/tobermory.c | 6 +- sound/soc/sh/dma-sh7760.c | 5 +- sound/soc/sh/fsi.c | 24 +- sound/soc/sh/rcar/core.c | 22 +- sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_dai.c | 10 +- sound/soc/sh/siu_pcm.c | 9 +- sound/soc/soc-ac97.c | 84 ++- sound/soc/soc-core.c | 5 - sound/soc/soc-generic-dmaengine-pcm.c | 55 +- sound/soc/soc-utils.c | 28 +- sound/soc/stm/stm32_adfsdm.c | 7 +- sound/soc/sunxi/sun4i-codec.c | 71 +- sound/soc/sunxi/sun8i-codec.c | 23 +- sound/soc/tegra/tegra_wm8903.c | 8 +- sound/soc/txx9/txx9aclc.c | 26 +- sound/soc/uniphier/evea.c | 67 +- sound/soc/xtensa/xtfpga-i2s.c | 17 +- 354 files changed, 14618 insertions(+), 15702 deletions(-) Best regards --- Kuninori Morimoto

5 420

[alsa-devel] [PATCH] ASoC: ak4613: don't have .use_pmdown_time
by Kuninori Morimoto 14 Feb '18

14 Feb '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_runtime_ignore_pmdown_time() returns "ignore pmdown time" decision. Then, it collects each component driver's .use_pmdown_time settings by using AND operation. But, ak4613 has no reason to ignore/use pmdown time. Thus, It should has use_pmdown_time=0, it means it doesn't have decision right of "ignore pmdown time". Otherwise, ak4613 paired driver's pmdown_time settings doesn't have meaning. In other words, "ignore pmdown time" is determined irrespective of the value of paired driver's settings (= ignore will be 0 by ak4613). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index a7b1e3a..8523ff9 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -610,7 +610,6 @@ static int ak4613_resume(struct snd_soc_component *component) .dapm_routes = ak4613_intercon, .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), .idle_bias_on = 1, - .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ak4642: don't have .use_pmdown_time
by Kuninori Morimoto 14 Feb '18

14 Feb '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_runtime_ignore_pmdown_time() returns "ignore pmdown time" decision. Then, it collects each component driver's .use_pmdown_time settings by using AND operation. But, ak4613 has no reason to ignore/use pmdown time. Thus, It should has use_pmdown_time=0, it means it doesn't have decision right of "ignore pmdown time". Otherwise, ak4642 paired driver's pmdown_time settings doesn't have meaning. In other words, "ignore pmdown time" is determined irrespective of the value of paired driver's settings (= ignore will be 0 by ak4642). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 9c35e91..6050559 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -562,7 +562,6 @@ static int ak4642_probe(struct snd_soc_component *component) .dapm_routes = ak4642_intercon, .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), .idle_bias_on = 1, - .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, }; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm9712: Use empty struct initializer
by Fabio Estevam 14 Feb '18

14 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> { 0 } only clears the first member of the structure. The first member of the snd_soc_dapm_update struct is pointer, so wrting 0 to a pointer results in the following sparse warning: sound/soc/codecs/wm9712.c:229:47: warning: Using plain integer as NULL pointer Use the empty struct initializer that clears all the struct members and fixes the sparse warning. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/wm9712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 7aeeeec..953d94d 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -226,7 +226,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int mixer, mask, shift, old; - struct snd_soc_dapm_update update = { 0 }; + struct snd_soc_dapm_update update = {}; bool change; mixer = mc->shift >> 8; -- 2.7.4

3 4

[alsa-devel] [PATCH] ASoC: uniphier: fix an error code in uniphier_aio_comprdma_new()
by Dan Carpenter 14 Feb '18

14 Feb '18

The dma_mapping_error() returns true or false, but we want to return -ENOMEM if there was an error. Fixes: e98131222ff9 ("ASoC: uniphier: add support for UniPhier AIO compress audio") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/uniphier/aio-compress.c b/sound/soc/uniphier/aio-compress.c index 7f7abe3ae99d..4751c8bdabec 100644 --- a/sound/soc/uniphier/aio-compress.c +++ b/sound/soc/uniphier/aio-compress.c @@ -54,12 +54,11 @@ static int uniphier_aio_comprdma_new(struct snd_soc_pcm_runtime *rtd) dma_dir = DMA_TO_DEVICE; sub->compr_addr = dma_map_single(dev, sub->compr_area, size, dma_dir); - ret = dma_mapping_error(dev, sub->compr_addr); - if (ret) { + if (dma_mapping_error(dev, sub->compr_addr)) { kfree(sub->compr_area); sub->compr_area = NULL; - return ret; + return -ENOMEM; } sub->compr_bytes = size;

2 1

[alsa-devel] [PATCH][RFC] ASoC: soc-core: soc_probe_dai() code simplification
by Kuninori Morimoto 14 Feb '18

14 Feb '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current soc_probe_dai() is using deep nested condition. Thus, it is difficult to read/understand. This patch simplification it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9c65477..7687515 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1619,22 +1619,21 @@ static int soc_probe_link_components(struct snd_soc_card *card, static int soc_probe_dai(struct snd_soc_dai *dai, int order) { - int ret; + if (dai->probed || + dai->driver->probe_order != order) + return 0; - if (!dai->probed && dai->driver->probe_order == order) { - if (dai->driver->probe) { - ret = dai->driver->probe(dai); - if (ret < 0) { - dev_err(dai->dev, - "ASoC: failed to probe DAI %s: %d\n", - dai->name, ret); - return ret; - } + if (dai->driver->probe) { + int ret = dai->driver->probe(dai); + if (ret < 0) { + dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n", + dai->name, ret); + return ret; } - - dai->probed = 1; } + dai->probed = 1; + return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/7] ASoC: samsung: Add HDMI sound support for tm2 board
by Sylwester Nawrocki 14 Feb '18

14 Feb '18

This series adds support for HDMI audio on exynos5433 tm2 board, it includes a few fixes for the I2S driver and an extension of the existing card driver to support the I2S1<->HDMI transmitter path. In v2 only support for 48k/96k/192k samples rate is left, as adjusting the AUD PLL output frequency might cause regressions. Sylwester Nawrocki (7): ASoC: samsung: i2s: Update clock-output-names property documentation ASoC: samsung: i2s: Ensure names of supplied clocks are unique ASoC: samsung: i2s: Prevent external abort on exynos5433 I2S1 access ASoC: samsung: i2s: Define the parameters list for SAMSUNG_I2S_OPCLK ASoC: samsung,tm2-audio DT binding documentation update ASoC: samsung: Add support for HDMI audio on TM2 board ASoC: samsung: Add missing #sound-dai-cells property documentation .../bindings/sound/samsung,tm2-audio.txt | 14 +- .../devicetree/bindings/sound/samsung-i2s.txt | 22 +-- sound/soc/samsung/i2s-regs.h | 11 +- sound/soc/samsung/i2s.c | 39 ++++-- sound/soc/samsung/i2s.h | 11 +- sound/soc/samsung/tm2_wm5110.c | 152 +++++++++++++++++---- 6 files changed, 191 insertions(+), 58 deletions(-) -- 2.14.2

3 16

[alsa-devel] [PATCH 0/8] ASoC: samsung: Add HDMI sound support for tm2 board
by Sylwester Nawrocki 14 Feb '18

14 Feb '18

This series adds support for HDMI audio on exynos5433 tm2 board, it includes a few fixes for the I2S driver and an extension of the existing card driver to support the I2S1<->HDMI transmitter path. Sylwester Nawrocki (8): ASoC: samsung: i2s: Ensure the RCLK rate is properly determined ASoC: samsung: i2s: Ensure names of supplied clocks are unique ASoC: samsung: i2s: Prevent external abort on exynos5433 I2S1 access ASoC: samsung: i2s: Define the parameters list for SAMSUNG_I2S_OPCLK ASoC: samsung: Add support for HDMI audio on TM2 board ASoC: samsung: i2s: Update clock-output-names property documentation ASoC: samsung: Add missing #sound-dai-cells property documentation ASoC: samsung,tm2-audio DT binding documentation update .../bindings/sound/samsung,tm2-audio.txt | 14 +- .../devicetree/bindings/sound/samsung-i2s.txt | 22 +-- sound/soc/samsung/i2s-regs.h | 11 +- sound/soc/samsung/i2s.c | 50 ++++-- sound/soc/samsung/i2s.h | 11 +- sound/soc/samsung/tm2_wm5110.c | 180 ++++++++++++++++++--- 6 files changed, 229 insertions(+), 59 deletions(-) -- 2.14.2

4 28

[alsa-devel] [PATCH v5] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
by Peter Ujfalusi 14 Feb '18

14 Feb '18

In the reset state of the codec we do not have complete playback or capture routes. The audio playback/capture will not work due to missing clock signals on the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down. To make sure that even if all output/input is disconnected the codec is generating clocks, we need to have valid DAPM route in every case to power up the must needed parts of the codec. I have verified that switching DAC (during playback) or ADC (during capture) will stop the I2S clocks, so the only solution is to connect the 'Off' routes as well to output/input. The routes will be only added if the codec is clock master. In case the role changes runtime, the applied routes are removed. Tested on am43x-epos-evm with aic3111 codec in master mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reviewed-by: Jyri Sarha <jsarha(a)ti.com> --- Hi, CHanges since v4: - Rebased on top of CODEC to component conversion - Added Reviewed-by from Jyri Changes since v3: - install or remove the master mode DAPM routes if needed - move the clock master DAPM route 'management' to a separate function Changes since v2: - Leftover debug prints removed. Changes since v1: - Only apply the master mode DAPM routes when the codec is clock master - comments added to explain the need. Regards, Peter sound/soc/codecs/tlv320aic31xx.c | 73 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 72 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3cd924dc300..7090342e8285 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -166,6 +166,7 @@ struct aic31xx_priv { unsigned int sysclk; u8 p_div; int rate_div_line; + bool master_dapm_route_applied; }; struct aic31xx_rate_divs { @@ -670,6 +671,29 @@ aic310x_audio_map[] = { {"SPK", NULL, "SPK ClassD"}, }; +/* + * Always connected DAPM routes for codec clock master modes. + * If the codec is the master on the I2S bus, we need to power on components + * to have valid DAC_CLK and also the DACs and ADC for playback/capture. + * Otherwise the codec will not generate clocks on the bus. + */ +static const struct snd_soc_dapm_route +common31xx_cm_audio_map[] = { + {"DAC Left Input", "Off", "DAC IN"}, + {"DAC Right Input", "Off", "DAC IN"}, + + {"HPL", NULL, "DAC Left"}, + {"HPR", NULL, "DAC Right"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_cm_audio_map[] = { + {"MIC1LP P-Terminal", "Off", "MIC1LP"}, + {"MIC1RP P-Terminal", "Off", "MIC1RP"}, + {"MIC1LM P-Terminal", "Off", "MIC1LM"}, + {"MIC1LM M-Terminal", "Off", "MIC1LM"}, +}; + static int aic31xx_add_controls(struct snd_soc_component *component) { int ret = 0; @@ -912,6 +936,53 @@ static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static int aic31xx_clock_master_routes(struct snd_soc_component *component, + unsigned int fmt) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + int ret; + + fmt &= SND_SOC_DAIFMT_MASTER_MASK; + if (fmt == SND_SOC_DAIFMT_CBS_CFS && + aic31xx->master_dapm_route_applied) { + /* + * Remove the DAPM route(s) for codec clock master modes, + * if applied + */ + ret = snd_soc_dapm_del_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_del_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = false; + } else if (fmt != SND_SOC_DAIFMT_CBS_CFS && + !aic31xx->master_dapm_route_applied) { + /* + * Add the needed DAPM route(s) for codec clock master modes, + * if it is not done already + */ + ret = snd_soc_dapm_add_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_add_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = true; + } + + return 0; +} + static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -992,7 +1063,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, AIC31XX_BCLKINV_MASK, iface_reg2); - return 0; + return aic31xx_clock_master_routes(component, fmt); } static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [RfC][PATCH] ALSA: hda/realtek Pin quirk for Fujitsu U7[25]7
by Jan-Marek Glogowski 14 Feb '18

14 Feb '18

Hi everybody, I got new HW from our supplier, a Fujitsu U727 and a U757 laptop, which have a headset jack. Attaching a headset, the microphone doesn't work. After some reading and trying various tools to understand the problem, I found hdajacksensetest, which showed a sense change for a seemingly wrong unconnected pin. Overriding this pins' config has fixed the headsets microphone, so now I have the attached kernel patch with a SND_HDA_PIN_QUIRK. After compiling the patched module (for 4.4), everything works as expected, but hdajackretask still shows the pin as unconnected, so I'm wondering, if I need to do something in addition? Is this normal behaviour? This is just a port to master, as we (Munich) are on Ubuntu Trusty with Xenial HWE (4.4 kernel). Thanks for the review. Jan-Marek Glogowski P.S. checkpatch.pl warned about too long lines for the SND_HDA_PIN_QUIRK. Should I fix that? >From 69f1c296b22ab6dce93ba50de9cf05e279d52088 Mon Sep 17 00:00:00 2001 From: Jan-Marek Glogowski <glogow(a)fbihome.de> Date: Wed, 24 Jan 2018 18:57:38 +0100 Subject: [PATCH] ALSA: hda/realtek Pin quirk for Fujitsu U7[25]7 These laptops have a combined jack to attach headsets, the U727 on the left, the U757 on the right, but a headsets microphone doesn't work. Using hdajacksensetest I found that pin 0x19 changed the present state when plugging the headset, in addition to 0x21, but didn't have the correct configuration (shown as "Not connected"). So this sets the configuration to the same values as the headphone pin 0x21 except for the device type microphone, which makes it work correctly. With the patch the configured pins for U727 are Pin 0x12 (Internal Mic, Mobile-In): present = No Pin 0x14 (Internal Speaker): present = No Pin 0x19 (Black Mic, Left side): present = No Pin 0x1d (Internal Aux): present = No Pin 0x21 (Black Headphone, Left side): present = No Signed-off-by: Jan-Marek Glogowski <glogow(a)fbihome.de> --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9aafc6c..52c6beb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5275,6 +5275,8 @@ enum { ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC255_FIXUP_FUJITSU_U727_MIC, + ALC255_FIXUP_FUJITSU_U757_MIC, ALC255_FIXUP_HEADSET_MODE, ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -5786,6 +5788,20 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC255_FIXUP_FUJITSU_U727_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* detect headset mic, left */ + { } + }, + }, + [ALC255_FIXUP_FUJITSU_U757_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11020 }, /* detect headset mic, right */ + { } + }, + }, [ALC255_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, @@ -6589,6 +6605,18 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x1b, 0x01011020}, {0x21, 0x02211010}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x10cf, "Fujitsu Lifebook U727", ALC255_FIXUP_FUJITSU_U727_MIC, + {0x12, 0xb7a60130}, + {0x14, 0x90170110}, + {0x19, 0x411111f0}, + {0x1d, 0x909701f0}, + {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x10cf, "Fujitsu Livebook U757", ALC255_FIXUP_FUJITSU_U757_MIC, + {0x12, 0xb7a60130}, + {0x14, 0x90170110}, + {0x19, 0x411111f0}, + {0x1d, 0x909701f0}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x14, 0x90170110}, -- 2.1.4

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[alsa-devel] kernel crash in snd_ctl_notify
by Sven Schnelle 14 Feb '18

14 Feb '18

Hi List, i just got external screens for my Thinkpad T470p. They are connected via Displayport to the docking station of the thinkpad. However, when i put the Thinkpad in the docking station, i see the following kernel crash: Feb 5 08:39:21 t470p kernel: [ 106.616102] BUG: unable to handle kernel NULL pointer dereference at 0000000000000 Feb 5 08:39:21 t470p kernel: [ 106.616119] IP: snd_ctl_notify.part.9+0xb3/0x190 Feb 5 08:39:21 t470p kernel: [ 106.616123] PGD 0 P4D 0 Feb 5 08:39:21 t470p kernel: [ 106.616129] Oops: 0000 [#1] SMP PTI Feb 5 08:39:21 t470p kernel: [ 106.616133] Modules linked in: e1000e Feb 5 08:39:21 t470p kernel: [ 106.616140] CPU: 6 PID: 1298 Comm: Xorg Not tainted 4.15.0+ #148 Feb 5 08:39:21 t470p kernel: [ 106.616143] Hardware name: LENOVO 20J6003DGE/20J6003DGE, BIOS R0FET39W (1.19 ) 12/ Feb 5 08:39:21 t470p kernel: [ 106.616149] RIP: 0010:snd_ctl_notify.part.9+0xb3/0x190 Feb 5 08:39:21 t470p kernel: [ 106.616152] RSP: 0018:ffffb7d2c068fa88 EFLAGS: 00010086 Feb 5 08:39:21 t470p kernel: [ 106.616156] RAX: ffff949ab97edf60 RBX: ffff949abb4ca4c8 RCX: 0000000000000000 Feb 5 08:39:21 t470p kernel: [ 106.616159] RDX: 0000000000000060 RSI: 0000000000000000 RDI: ffff949ab97edfc0 Feb 5 08:39:21 t470p kernel: [ 106.616162] RBP: ffff949abb4ca000 R08: ffff949abaac3a10 R09: ffff949ab97edf60 Feb 5 08:39:21 t470p kernel: [ 106.616164] R10: ffff949abaac3a10 R11: 00000000000003d8 R12: ffff949aaea132c0 Feb 5 08:39:21 t470p kernel: [ 106.616167] R13: 0000000000000202 R14: 0000000000000010 R15: ffff949aaea13280 Feb 5 08:39:21 t470p kernel: [ 106.616171] FS: 00007f57ded37500(0000) GS:ffff949ad1580000(0000) knlGS:0000000000 Feb 5 08:39:21 t470p kernel: [ 106.616174] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 Feb 5 08:39:21 t470p kernel: [ 106.616176] CR2: 0000000000000010 CR3: 000000043cf06005 CR4: 00000000003606e0 Feb 5 08:39:21 t470p kernel: [ 106.616179] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 Feb 5 08:39:21 t470p kernel: [ 106.616182] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Feb 5 08:39:21 t470p kernel: [ 106.616184] Call Trace: Feb 5 08:39:21 t470p kernel: [ 106.616194] hdmi_present_sense+0xdc/0x370 Feb 5 08:39:21 t470p kernel: [ 106.616200] check_presence_and_report+0x5e/0x80 Feb 5 08:39:21 t470p kernel: [ 106.616206] intel_audio_codec_disable+0xab/0xd0 Feb 5 08:39:21 t470p kernel: [ 106.616213] intel_encoders_disable.isra.97+0x70/0x90 Feb 5 08:39:21 t470p kernel: [ 106.616219] haswell_crtc_disable+0x47/0x140 Feb 5 08:39:21 t470p kernel: [ 106.616225] intel_atomic_commit_tail+0x70e/0xcb0 Feb 5 08:39:21 t470p kernel: [ 106.616231] ? intel_atomic_commit_ready+0x44/0x4c Feb 5 08:39:21 t470p kernel: [ 106.616236] intel_atomic_commit+0x21f/0x2d0 Feb 5 08:39:21 t470p kernel: [ 106.616241] drm_atomic_connector_commit_dpms+0xe5/0xf0 Feb 5 08:39:21 t470p kernel: [ 106.616248] drm_mode_obj_set_property_ioctl+0x153/0x260 Feb 5 08:39:21 t470p kernel: [ 106.616253] ? drm_mode_connector_set_obj_prop+0x70/0x70 Feb 5 08:39:21 t470p kernel: [ 106.616258] drm_mode_connector_property_set_ioctl+0x2e/0x40 Feb 5 08:39:21 t470p kernel: [ 106.616264] drm_ioctl_kernel+0x59/0xb0 Feb 5 08:39:21 t470p kernel: [ 106.616270] drm_ioctl+0x29f/0x340 Feb 5 08:39:21 t470p kernel: [ 106.616275] ? drm_mode_connector_set_obj_prop+0x70/0x70 Feb 5 08:39:21 t470p kernel: [ 106.616282] ? signal_setup_done+0x57/0x90 Feb 5 08:39:21 t470p kernel: [ 106.616288] ? __fpu__restore_sig+0x80/0x400 Feb 5 08:39:21 t470p kernel: [ 106.616295] do_vfs_ioctl+0x8d/0x5d0 Feb 5 08:39:21 t470p kernel: [ 106.616301] SyS_ioctl+0x3b/0x70 Feb 5 08:39:21 t470p kernel: [ 106.616307] entry_SYSCALL_64_fastpath+0x22/0x8a Feb 5 08:39:21 t470p kernel: [ 106.616312] RIP: 0033:0x7f57dc1b7307 Feb 5 08:39:21 t470p kernel: [ 106.616314] RSP: 002b:00007ffe3459f138 EFLAGS: 00003246 Feb 5 08:39:21 t470p kernel: [ 106.616317] Code: 8b 00 48 39 c8 75 ef ba 58 00 00 00 be 20 80 08 01 48 89 4c 24 0 Feb 5 08:39:21 t470p kernel: [ 106.616384] RIP: snd_ctl_notify.part.9+0xb3/0x190 RSP: ffffb7d2c068fa88 Feb 5 08:39:21 t470p kernel: [ 106.616387] CR2: 0000000000000010 Feb 5 08:39:21 t470p kernel: [ 106.616391] ---[ end trace 0e8742968e349514 ]--- RIP is at snd_ctl_notify, because the id parameter is NULL. Looking a bit further, i see the following in the kernel log: Feb 6 08:14:20 t470p kernel: [ 6.266169] snd_hda_intel 0000:00:1f.3: Too many HDMI devices Feb 6 08:14:20 t470p kernel: [ 6.266170] snd_hda_intel 0000:00:1f.3: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y Feb 6 08:14:20 t470p kernel: [ 6.266171] snd_hda_intel 0000:00:1f.3: Too many HDMI devices Feb 6 08:14:20 t470p kernel: [ 6.266172] snd_hda_intel 0000:00:1f.3: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y Feb 6 08:14:20 t470p kernel: [ 6.266172] snd_hda_intel 0000:00:1f.3: Too many HDMI devices Feb 6 08:14:20 t470p kernel: [ 6.266173] snd_hda_intel 0000:00:1f.3: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y So it seems like commit 1f7f51a63114bab3a05920f4b1343154e95e2cb6 ("Fix regression of hdmi eld control created based on invalid pcm") might be causing this issue. While it makes the probe work, it doesn't assign the pcm pointer in the hdmi data structure. Later it get's referenced and the kernel crashes. Enabling CONFIG_SND_DYNAMIC_MINORS makes the issue go away, but the kernel shouldn't crash if it's disabled.I made the attached patch to fix this. However, i'm not sure whether this is the right way to fix it and whether it needs to be patched in other places. Regards Sven

2 1

[alsa-devel] [PATCH v3] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
by Peter Ujfalusi 14 Feb '18

14 Feb '18

In the reset state of the codec we do not have complete playback or capture routes. The audio playback/capture will not work due to missing clock signals on the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down. To make sure that even if all output/input is disconnected the codec is generating clocks, we need to have valid DAPM route in every case to power up the must needed parts of the codec. I have verified that switching DAC (during playback) or ADC (during capture) will stop the I2S clocks, so the only solution is to connect the 'Off' routes as well to output/input. The routes only going to be added if the codec is configured as clock master. Tested on am43x-epos-evm with aic3111 codec in master mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v2: - Leftover debug prints removed. Changes since v1: - Only apply the master mode DAPM routes when the codec is clock master - comments added to explain the need. Regards, Peter sound/soc/codecs/tlv320aic31xx.c | 44 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 858cb8be445f..088648791111 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -166,6 +166,7 @@ struct aic31xx_priv { unsigned int sysclk; u8 p_div; int rate_div_line; + bool master_dapm_route_applied; }; struct aic31xx_rate_divs { @@ -670,6 +671,29 @@ aic310x_audio_map[] = { {"SPK", NULL, "SPK ClassD"}, }; +/* + * Always connected DAPM routes for codec clock master modes. + * If the codec is the master on the I2S bus, we need to power on components + * to have valid DAC_CLK and also the DACs and ADC for playback/capture. + * Otherwise the codec will not generate clocks on the bus. + */ +static const struct snd_soc_dapm_route +common31xx_cm_audio_map[] = { + {"DAC Left Input", "Off", "DAC IN"}, + {"DAC Right Input", "Off", "DAC IN"}, + + {"HPL", NULL, "DAC Left"}, + {"HPR", NULL, "DAC Right"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_cm_audio_map[] = { + {"MIC1LP P-Terminal", "Off", "MIC1LP"}, + {"MIC1RP P-Terminal", "Off", "MIC1RP"}, + {"MIC1LM P-Terminal", "Off", "MIC1LM"}, + {"MIC1LM M-Terminal", "Off", "MIC1LM"}, +}; + static int aic31xx_add_controls(struct snd_soc_codec *codec) { int ret = 0; @@ -916,6 +940,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); u8 iface_reg1 = 0; u8 iface_reg2 = 0; u8 dsp_a_val = 0; @@ -992,6 +1017,25 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, AIC31XX_BCLKINV_MASK, iface_reg2); + /* Add the needed DAPM route(s) for codec clock master modes, once */ + if (iface_reg1 & (AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER) && + !aic31xx->master_dapm_route_applied) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int ret; + + ret = snd_soc_dapm_add_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_add_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = true; + } + return 0; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 2

[alsa-devel] [PATCH] ALSA: seq: Fix racy pool initializations
by Takashi Iwai 14 Feb '18

14 Feb '18

ALSA sequencer core initializes the event pool on demand by invoking snd_seq_pool_init() when the first write happens and the pool is empty. Meanwhile user can reset the pool size manually via ioctl concurrently, and this may lead to UAF or out-of-bound accesses since the function tries to vmalloc / vfree the buffer. A simple fix is to just wrap the snd_seq_pool_init() call with the recently introduced client->ioctl_mutex; as the calls for snd_seq_pool_init() from other side are always protected with this mutex, we can avoid the race. Reported-by: 范龙飞 <long7573(a)126.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 60db32785f62..04d4db44fae5 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1003,7 +1003,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, { struct snd_seq_client *client = file->private_data; int written = 0, len; - int err = -EINVAL; + int err; struct snd_seq_event event; if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT)) @@ -1018,11 +1018,15 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, /* allocate the pool now if the pool is not allocated yet */ if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) { - if (snd_seq_pool_init(client->pool) < 0) + mutex_lock(&client->ioctl_mutex); + err = snd_seq_pool_init(client->pool); + mutex_unlock(&client->ioctl_mutex); + if (err < 0) return -ENOMEM; } /* only process whole events */ + err = -EINVAL; while (count >= sizeof(struct snd_seq_event)) { /* Read in the event header from the user */ len = sizeof(event); -- 2.15.1

1 0

[alsa-devel] [PATCH v3 1/5] ALSA: emu10k1: remove reserved_page
by Maciej S. Szmigiero 14 Feb '18

14 Feb '18

The emu10k1-family chips need the first page (index 0) reserved in their page tables for some reason (every emu10k1 driver I've checked does this without much of an explanation). Using the first page for normal samples results in a broken playback. However, we already have a dummy page allocated - so called "silent page" and, in fact, had always been setting it as the first page in the chip page table because an initialization of every entry of the page table to point to a silent page happens after and overwrites the reserved_page allocation. So the only thing remaining to remove the reserved_page allocation is a trivial change to the page allocation logic to ignore the first page entry and start its allocations from the second entry (index 1). Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1, v2: None in this patch. include/sound/emu10k1.h | 1 - sound/pci/emu10k1/emu10k1_main.c | 11 ----------- sound/pci/emu10k1/memory.c | 8 ++++++-- 3 files changed, 6 insertions(+), 14 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 4f42affe777c..db32b7de52e0 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1718,7 +1718,6 @@ struct snd_emu10k1 { struct snd_dma_buffer p16v_buffer; struct snd_util_memhdr *memhdr; /* page allocation list */ - struct snd_emu10k1_memblk *reserved_page; /* reserved page */ struct list_head mapped_link_head; struct list_head mapped_order_link_head; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index ccf4415a1c7b..a0a4d31ded53 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1272,12 +1272,6 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) release_firmware(emu->dock_fw); if (emu->irq >= 0) free_irq(emu->irq, emu); - /* remove reserved page */ - if (emu->reserved_page) { - snd_emu10k1_synth_free(emu, - (struct snd_util_memblk *)emu->reserved_page); - emu->reserved_page = NULL; - } snd_util_memhdr_free(emu->memhdr); if (emu->silent_page.area) snd_dma_free_pages(&emu->silent_page); @@ -1993,11 +1987,6 @@ int snd_emu10k1_create(struct snd_card *card, SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT; - emu->reserved_page = (struct snd_emu10k1_memblk *) - snd_emu10k1_synth_alloc(emu, 4096); - if (emu->reserved_page) - emu->reserved_page->map_locked = 1; - /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, PAGE_SIZE); silent_page = emu->silent_page.addr << emu->address_mode; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 4f1f69be1865..eaa61024ac7f 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -102,7 +102,7 @@ static void emu10k1_memblk_init(struct snd_emu10k1_memblk *blk) */ static int search_empty_map_area(struct snd_emu10k1 *emu, int npages, struct list_head **nextp) { - int page = 0, found_page = -ENOMEM; + int page = 1, found_page = -ENOMEM; int max_size = npages; int size; struct list_head *candidate = &emu->mapped_link_head; @@ -147,6 +147,10 @@ static int map_memblk(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) page = search_empty_map_area(emu, blk->pages, &next); if (page < 0) /* not found */ return page; + if (page == 0) { + dev_err(emu->card->dev, "trying to map zero (reserved) page\n"); + return -EINVAL; + } /* insert this block in the proper position of mapped list */ list_add_tail(&blk->mapped_link, next); /* append this as a newest block in order list */ @@ -177,7 +181,7 @@ static int unmap_memblk(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) q = get_emu10k1_memblk(p, mapped_link); start_page = q->mapped_page + q->pages; } else - start_page = 0; + start_page = 1; if ((p = blk->mapped_link.next) != &emu->mapped_link_head) { q = get_emu10k1_memblk(p, mapped_link); end_page = q->mapped_page;

2 1

[alsa-devel] [PATCH][RFC] ASoC: soc-utils: don't have .use_pmdown_time
by Kuninori Morimoto 14 Feb '18

14 Feb '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_runtime_ignore_pmdown_time() returns "ignore pmdown time" decision. Then, it collects each component driver's .use_pmdown_time settings by using AND operation. But, dummy codec has no reason to ignore/use pmdown time. Thus, it should has use_pmdown_time=0, it means it doesn't have decision right of "ignore pmdown time". Otherwise, dummy codec paired driver's pmdown_time settings doesn't have meaning. In other words, "ignore pmdown time" is determined irrespective of the value of paired driver's settings (= ignore will be 0 by dummy codec). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-utils.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2d9e98b..b70d524 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -290,7 +290,6 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) static const struct snd_soc_component_driver dummy_codec = { .idle_bias_on = 1, - .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, }; -- 1.9.1

1 0

[alsa-devel] [PATCH v3 5/5] ALSA: emu10k1: add a IOMMU workaround
by Maciej S. Szmigiero 14 Feb '18

14 Feb '18

The Audigy 2 CA0102 chip (but most likely others from the emu10k1 family, too) has a problem that from time to time it likes to do few DMA reads a bit beyond its normal allocation and gets very confused if these reads get blocked by a IOMMU. For the first (reserved) page this happens multiple times at every playback, for various synth pages it happens randomly, rarely for PCM playback buffers and the page table memory itself. All these reads seem to follow a similar pattern, observed read offsets beyond the allocation end were 0x00, 0x40, 0x80 and 0xc0 (PCI cache line multiples), so it looks like the device tries to accesses up to 256 extra bytes. As a workaround let's widen these DMA allocations by an extra page if we detect that the device is behind a non-passthrough IOMMU (the DMA memory should be relatively plenty on IOMMU systems). Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Apply this workaround also to PCM playback buffers since it seems they are affected, too. Changes from v2: Use a common snd_emu10k1_alloc_pages_maybe_wider() helper function, check if rounding the allocation size up to the nearest page size already covers the necessary extra memory space. include/sound/emu10k1.h | 3 +++ sound/pci/emu10k1/emu10k1_main.c | 49 ++++++++++++++++++++++++++++++++++++---- sound/pci/emu10k1/emupcm.c | 10 +++++++- sound/pci/emu10k1/memory.c | 40 +++++++++++++++++++++++++++++--- 4 files changed, 93 insertions(+), 9 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index db32b7de52e0..5ebcc51c0a6a 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1710,6 +1710,7 @@ struct snd_emu10k1 { unsigned int ecard_ctrl; /* ecard control bits */ unsigned int address_mode; /* address mode */ unsigned long dma_mask; /* PCI DMA mask */ + bool iommu_workaround; /* IOMMU workaround needed */ unsigned int delay_pcm_irq; /* in samples */ int max_cache_pages; /* max memory size / PAGE_SIZE */ struct snd_dma_buffer silent_page; /* silent page */ @@ -1877,6 +1878,8 @@ void snd_p16v_resume(struct snd_emu10k1 *emu); /* memory allocation */ struct snd_util_memblk *snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *substream); int snd_emu10k1_free_pages(struct snd_emu10k1 *emu, struct snd_util_memblk *blk); +int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size, + struct snd_dma_buffer *dmab); struct snd_util_memblk *snd_emu10k1_synth_alloc(struct snd_emu10k1 *emu, unsigned int size); int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *blk); int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 8decd2a7a404..18267de3a269 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -36,6 +36,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/interrupt.h> +#include <linux/iommu.h> #include <linux/pci.h> #include <linux/slab.h> #include <linux/vmalloc.h> @@ -1758,6 +1759,38 @@ static struct snd_emu_chip_details emu_chip_details[] = { { } /* terminator */ }; +/* + * The chip (at least the Audigy 2 CA0102 chip, but most likely others, too) + * has a problem that from time to time it likes to do few DMA reads a bit + * beyond its normal allocation and gets very confused if these reads get + * blocked by a IOMMU. + * + * This behaviour has been observed for the first (reserved) page + * (for which it happens multiple times at every playback), often for various + * synth pages and sometimes for PCM playback buffers and the page table + * memory itself. + * + * As a workaround let's widen these DMA allocations by an extra page if we + * detect that the device is behind a non-passthrough IOMMU. + */ +static void snd_emu10k1_detect_iommu(struct snd_emu10k1 *emu) +{ + struct iommu_domain *domain; + + emu->iommu_workaround = false; + + if (!iommu_present(emu->card->dev->bus)) + return; + + domain = iommu_get_domain_for_dev(emu->card->dev); + if (domain && domain->type == IOMMU_DOMAIN_IDENTITY) + return; + + dev_notice(emu->card->dev, + "non-passthrough IOMMU detected, widening DMA allocations"); + emu->iommu_workaround = true; +} + int snd_emu10k1_create(struct snd_card *card, struct pci_dev *pci, unsigned short extin_mask, @@ -1770,6 +1803,7 @@ int snd_emu10k1_create(struct snd_card *card, struct snd_emu10k1 *emu; int idx, err; int is_audigy; + size_t page_table_size; unsigned int silent_page; const struct snd_emu_chip_details *c; static struct snd_device_ops ops = { @@ -1867,6 +1901,8 @@ int snd_emu10k1_create(struct snd_card *card, is_audigy = emu->audigy = c->emu10k2_chip; + snd_emu10k1_detect_iommu(emu); + /* set addressing mode */ emu->address_mode = is_audigy ? 0 : 1; /* set the DMA transfer mask */ @@ -1893,8 +1929,11 @@ int snd_emu10k1_create(struct snd_card *card, emu->port = pci_resource_start(pci, 0); emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT; - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - (emu->address_mode ? 32 : 16) * 1024, &emu->ptb_pages) < 0) { + + page_table_size = sizeof(u32) * (emu->address_mode ? MAXPAGES1 : + MAXPAGES0); + if (snd_emu10k1_alloc_pages_maybe_wider(emu, page_table_size, + &emu->ptb_pages) < 0) { err = -ENOMEM; goto error; } @@ -1910,8 +1949,8 @@ int snd_emu10k1_create(struct snd_card *card, goto error; } - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - EMUPAGESIZE, &emu->silent_page) < 0) { + if (snd_emu10k1_alloc_pages_maybe_wider(emu, EMUPAGESIZE, + &emu->silent_page) < 0) { err = -ENOMEM; goto error; } @@ -1995,7 +2034,7 @@ int snd_emu10k1_create(struct snd_card *card, 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT; /* Clear silent pages and set up pointers */ - memset(emu->silent_page.area, 0, PAGE_SIZE); + memset(emu->silent_page.area, 0, emu->silent_page.bytes); silent_page = emu->silent_page.addr << emu->address_mode; for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++) ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 2683b9717215..cefe613ef7b7 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -411,12 +411,20 @@ static int snd_emu10k1_playback_hw_params(struct snd_pcm_substream *substream, struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm = runtime->private_data; + size_t alloc_size; int err; if ((err = snd_emu10k1_pcm_channel_alloc(epcm, params_channels(hw_params))) < 0) return err; - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) + + alloc_size = params_buffer_bytes(hw_params); + if (emu->iommu_workaround) + alloc_size += EMUPAGESIZE; + err = snd_pcm_lib_malloc_pages(substream, alloc_size); + if (err < 0) return err; + if (emu->iommu_workaround && runtime->dma_bytes >= EMUPAGESIZE) + runtime->dma_bytes -= EMUPAGESIZE; if (err > 0) { /* change */ int mapped; if (epcm->memblk != NULL) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 1d0ce7356bbd..5865f3b90b34 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -377,6 +377,33 @@ int snd_emu10k1_free_pages(struct snd_emu10k1 *emu, struct snd_util_memblk *blk) return snd_emu10k1_synth_free(emu, blk); } +/* + * allocate DMA pages, widening the allocation if necessary + * + * See the comment above snd_emu10k1_detect_iommu() in emu10k1_main.c why + * this might be needed. + * + * If you modify this function check whether __synth_free_pages() also needs + * changes. + */ +int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size, + struct snd_dma_buffer *dmab) +{ + if (emu->iommu_workaround) { + size_t npages = (size + PAGE_SIZE - 1) / PAGE_SIZE; + size_t size_real = npages * PAGE_SIZE; + + /* + * The device has been observed to accesses up to 256 extra + * bytes, but use 1k to be safe. + */ + if (size_real < size + 1024) + size += PAGE_SIZE; + } + + return snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), size, dmab); +} /* * memory allocation using multiple pages (for synth) @@ -472,7 +499,15 @@ static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, continue; dmab.area = emu->page_ptr_table[page]; dmab.addr = emu->page_addr_table[page]; + + /* + * please keep me in sync with logic in + * snd_emu10k1_alloc_pages_maybe_wider() + */ dmab.bytes = PAGE_SIZE; + if (emu->iommu_workaround) + dmab.bytes *= 2; + snd_dma_free_pages(&dmab); emu->page_addr_table[page] = 0; emu->page_ptr_table[page] = NULL; @@ -491,9 +526,8 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk get_single_page_range(emu->memhdr, blk, &first_page, &last_page); /* allocate kernel pages */ for (page = first_page; page <= last_page; page++) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), - PAGE_SIZE, &dmab) < 0) + if (snd_emu10k1_alloc_pages_maybe_wider(emu, PAGE_SIZE, + &dmab) < 0) goto __fail; if (!is_valid_page(emu, dmab.addr)) { snd_dma_free_pages(&dmab);

1 0

[alsa-devel] [PATCH v3 4/5] ALSA: emu10k1: make sure synth DMA pages are allocated with DMA functions
by Maciej S. Szmigiero 14 Feb '18

14 Feb '18

Commit a5003fc04113 ("[ALSA] emu10k1 - simplify page allocation for synth") switched from using the DMA allocator for synth DMA pages to manually calling alloc_page(). However, this usage has an implicit assumption that the DMA address space for the emu10k1-family chip is the same as the CPU physical address space which is not true for a system with a IOMMU. Since this made the synth part of the driver non-functional on such systems let's effectively revert that commit (while keeping the __synth_free_pages() simplification). Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: None in this patch. Changes from v2: Keep the __synth_free_pages() simplification. sound/pci/emu10k1/memory.c | 44 +++++++++++++++++++++++++++----------------- 1 file changed, 27 insertions(+), 17 deletions(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index fcb04cbbc9ab..1d0ce7356bbd 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -461,10 +461,19 @@ static void get_single_page_range(struct snd_util_memhdr *hdr, static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, int last_page) { + struct snd_dma_buffer dmab; int page; + dmab.dev.type = SNDRV_DMA_TYPE_DEV; + dmab.dev.dev = snd_dma_pci_data(emu->pci); + for (page = first_page; page <= last_page; page++) { - free_page((unsigned long)emu->page_ptr_table[page]); + if (emu->page_ptr_table[page] == NULL) + continue; + dmab.area = emu->page_ptr_table[page]; + dmab.addr = emu->page_addr_table[page]; + dmab.bytes = PAGE_SIZE; + snd_dma_free_pages(&dmab); emu->page_addr_table[page] = 0; emu->page_ptr_table[page] = NULL; } @@ -476,30 +485,31 @@ static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { int page, first_page, last_page; + struct snd_dma_buffer dmab; emu10k1_memblk_init(blk); get_single_page_range(emu->memhdr, blk, &first_page, &last_page); /* allocate kernel pages */ for (page = first_page; page <= last_page; page++) { - /* first try to allocate from <4GB zone */ - struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | - __GFP_NOWARN); - if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) { - if (p) - __free_page(p); - /* try to allocate from <16MB zone */ - p = alloc_page(GFP_ATOMIC | GFP_DMA | - __GFP_NORETRY | /* no OOM-killer */ - __GFP_NOWARN); - } - if (!p) { - __synth_free_pages(emu, first_page, page - 1); - return -ENOMEM; + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), + PAGE_SIZE, &dmab) < 0) + goto __fail; + if (!is_valid_page(emu, dmab.addr)) { + snd_dma_free_pages(&dmab); + goto __fail; } - emu->page_addr_table[page] = page_to_phys(p); - emu->page_ptr_table[page] = page_address(p); + emu->page_addr_table[page] = dmab.addr; + emu->page_ptr_table[page] = dmab.area; } return 0; + +__fail: + /* release allocated pages */ + last_page = page - 1; + __synth_free_pages(emu, first_page, last_page); + + return -ENOMEM; } /*

1 0

[alsa-devel] [PATCH v3 3/5] ALSA: emu10k1: add optional debug printouts with DMA addresses
by Maciej S. Szmigiero 14 Feb '18

14 Feb '18

When we get a IOMMU page fault for a emu10k1 device it is very hard to discover which of chip many DMA allocations triggered it (since on a IOMMU system the DMA address space is often very different from the CPU one). Let's add optional debug printouts providing this information. These debug printouts are only enabled on an explicit request via the kernel dynamic debug mechanism. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Fix a sparse warning by changing a variable type in __{get,set}_ptb_entry from u32 to __le32, don't open code EMUPAGESIZE. Changes from v2: None in this patch. sound/pci/emu10k1/emu10k1_main.c | 8 ++++++++ sound/pci/emu10k1/memory.c | 15 +++++++++++---- 2 files changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 11e868f569d6..8decd2a7a404 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1898,6 +1898,9 @@ int snd_emu10k1_create(struct snd_card *card, err = -ENOMEM; goto error; } + dev_dbg(card->dev, "page table address range is %.8lx:%.8lx\n", + (unsigned long)emu->ptb_pages.addr, + (unsigned long)(emu->ptb_pages.addr + emu->ptb_pages.bytes)); emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *)); emu->page_addr_table = vmalloc(emu->max_cache_pages * @@ -1912,6 +1915,11 @@ int snd_emu10k1_create(struct snd_card *card, err = -ENOMEM; goto error; } + dev_dbg(card->dev, "silent page range is %.8lx:%.8lx\n", + (unsigned long)emu->silent_page.addr, + (unsigned long)(emu->silent_page.addr + + emu->silent_page.bytes)); + emu->memhdr = snd_util_memhdr_new(emu->max_cache_pages * PAGE_SIZE); if (emu->memhdr == NULL) { err = -ENOMEM; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index eaa61024ac7f..fcb04cbbc9ab 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -34,7 +34,10 @@ * aligned pages in others */ #define __set_ptb_entry(emu,page,addr) \ - (((u32 *)(emu)->ptb_pages.area)[page] = cpu_to_le32(((addr) << (emu->address_mode)) | (page))) + (((__le32 *)(emu)->ptb_pages.area)[page] = \ + cpu_to_le32(((addr) << (emu->address_mode)) | (page))) +#define __get_ptb_entry(emu, page) \ + (le32_to_cpu(((__le32 *)(emu)->ptb_pages.area)[page])) #define UNIT_PAGES (PAGE_SIZE / EMUPAGESIZE) #define MAX_ALIGN_PAGES0 (MAXPAGES0 / UNIT_PAGES) @@ -44,8 +47,7 @@ /* get offset address from aligned page */ #define aligned_page_offset(page) ((page) << PAGE_SHIFT) -#if PAGE_SIZE == 4096 -/* page size == EMUPAGESIZE */ +#if PAGE_SIZE == EMUPAGESIZE && !IS_ENABLED(CONFIG_DYNAMIC_DEBUG) /* fill PTB entrie(s) corresponding to page with addr */ #define set_ptb_entry(emu,page,addr) __set_ptb_entry(emu,page,addr) /* fill PTB entrie(s) corresponding to page with silence pointer */ @@ -58,6 +60,8 @@ static inline void set_ptb_entry(struct snd_emu10k1 *emu, int page, dma_addr_t a page *= UNIT_PAGES; for (i = 0; i < UNIT_PAGES; i++, page++) { __set_ptb_entry(emu, page, addr); + dev_dbg(emu->card->dev, "mapped page %d to entry %.8x\n", page, + (unsigned int)__get_ptb_entry(emu, page)); addr += EMUPAGESIZE; } } @@ -65,9 +69,12 @@ static inline void set_silent_ptb(struct snd_emu10k1 *emu, int page) { int i; page *= UNIT_PAGES; - for (i = 0; i < UNIT_PAGES; i++, page++) + for (i = 0; i < UNIT_PAGES; i++, page++) { /* do not increment ptr */ __set_ptb_entry(emu, page, emu->silent_page.addr); + dev_dbg(emu->card->dev, "mapped silent page %d to entry %.8x\n", + page, (unsigned int)__get_ptb_entry(emu, page)); + } } #endif /* PAGE_SIZE */

1 0

[alsa-devel] [PATCH v3 2/5] ALSA: emu10k1: use dma_set_mask_and_coherent()
by Maciej S. Szmigiero 14 Feb '18

14 Feb '18

We have been calling dma_set_mask() and then dma_set_coherent_mask() with the same value, but there is a dma_set_mask_and_coherent() function that does exactly that so let's use it instead. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1, v2: None in this patch. sound/pci/emu10k1/emu10k1_main.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index a0a4d31ded53..11e868f569d6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1871,8 +1871,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->address_mode = is_audigy ? 0 : 1; /* set the DMA transfer mask */ emu->dma_mask = emu->address_mode ? EMU10K1_DMA_MASK : AUDIGY_DMA_MASK; - if (dma_set_mask(&pci->dev, emu->dma_mask) < 0 || - dma_set_coherent_mask(&pci->dev, emu->dma_mask) < 0) { + if (dma_set_mask_and_coherent(&pci->dev, emu->dma_mask) < 0) { dev_err(card->dev, "architecture does not support PCI busmaster DMA with mask 0x%lx\n", emu->dma_mask);

1 0

[alsa-devel] [PATCH v2 5/5] ALSA: emu10k1: add a IOMMU workaround
by Maciej S. Szmigiero 14 Feb '18

14 Feb '18

The Audigy 2 CA0102 chip (but most likely others from the emu10k1 family, too) has a problem that from time to time it likes to do few DMA reads a bit beyond its normal allocation and gets very confused if these reads get blocked by a IOMMU. For the first (reserved) page this happens multiple times at every playback, for various synth pages it happens randomly, rarely for PCM playback buffers and the page table memory itself. All these reads seem to follow a similar pattern, observed read offsets beyond the allocation end were 0x00, 0x40, 0x80 and 0xc0 (PCI cache line multiples), so it looks like the device tries to accesses up to 256 extra bytes. As a workaround let's widen these DMA allocations by an extra page if we detect that the device is behind a non-passthrough IOMMU (the DMA memory should be relatively plenty on IOMMU systems). Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Apply this workaround also to PCM playback buffers since it seems they are affected, too. include/sound/emu10k1.h | 1 + sound/pci/emu10k1/emu10k1_main.c | 50 +++++++++++++++++++++++++++++++++++++--- sound/pci/emu10k1/emupcm.c | 9 +++++++- sound/pci/emu10k1/memory.c | 16 ++++++++++--- 4 files changed, 69 insertions(+), 7 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index db32b7de52e0..ba27abf65408 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1710,6 +1710,7 @@ struct snd_emu10k1 { unsigned int ecard_ctrl; /* ecard control bits */ unsigned int address_mode; /* address mode */ unsigned long dma_mask; /* PCI DMA mask */ + bool iommu_workaround; /* IOMMU workaround needed */ unsigned int delay_pcm_irq; /* in samples */ int max_cache_pages; /* max memory size / PAGE_SIZE */ struct snd_dma_buffer silent_page; /* silent page */ diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 8decd2a7a404..3638bff26d23 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -36,6 +36,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/interrupt.h> +#include <linux/iommu.h> #include <linux/pci.h> #include <linux/slab.h> #include <linux/vmalloc.h> @@ -1758,6 +1759,38 @@ static struct snd_emu_chip_details emu_chip_details[] = { { } /* terminator */ }; +/* + * The chip (at least the Audigy 2 CA0102 chip, but most likely others, too) + * has a problem that from time to time it likes to do few DMA reads a bit + * beyond its normal allocation and gets very confused if these reads get + * blocked by a IOMMU. + * + * This behaviour has been observed for the first (reserved) page + * (for which it happens multiple times at every playback), often for various + * synth pages and sometimes for PCM playback buffers and the page table + * memory itself. + * + * As a workaround let's widen these DMA allocations by an extra page if we + * detect that the device is behind a non-passthrough IOMMU. + */ +static void snd_emu10k1_detect_iommu(struct snd_emu10k1 *emu) +{ + struct iommu_domain *domain; + + emu->iommu_workaround = false; + + if (!iommu_present(emu->card->dev->bus)) + return; + + domain = iommu_get_domain_for_dev(emu->card->dev); + if (domain && domain->type == IOMMU_DOMAIN_IDENTITY) + return; + + dev_notice(emu->card->dev, + "non-passthrough IOMMU detected, widening DMA allocations"); + emu->iommu_workaround = true; +} + int snd_emu10k1_create(struct snd_card *card, struct pci_dev *pci, unsigned short extin_mask, @@ -1770,6 +1803,7 @@ int snd_emu10k1_create(struct snd_card *card, struct snd_emu10k1 *emu; int idx, err; int is_audigy; + size_t page_table_size, silent_page_size; unsigned int silent_page; const struct snd_emu_chip_details *c; static struct snd_device_ops ops = { @@ -1867,6 +1901,8 @@ int snd_emu10k1_create(struct snd_card *card, is_audigy = emu->audigy = c->emu10k2_chip; + snd_emu10k1_detect_iommu(emu); + /* set addressing mode */ emu->address_mode = is_audigy ? 0 : 1; /* set the DMA transfer mask */ @@ -1893,8 +1929,13 @@ int snd_emu10k1_create(struct snd_card *card, emu->port = pci_resource_start(pci, 0); emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT; + + page_table_size = sizeof(u32) * (emu->address_mode ? MAXPAGES1 : + MAXPAGES0); + if (emu->iommu_workaround) + page_table_size += PAGE_SIZE; if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - (emu->address_mode ? 32 : 16) * 1024, &emu->ptb_pages) < 0) { + page_table_size, &emu->ptb_pages) < 0) { err = -ENOMEM; goto error; } @@ -1910,8 +1951,11 @@ int snd_emu10k1_create(struct snd_card *card, goto error; } + silent_page_size = EMUPAGESIZE; + if (emu->iommu_workaround) + silent_page_size *= 2; if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - EMUPAGESIZE, &emu->silent_page) < 0) { + silent_page_size, &emu->silent_page) < 0) { err = -ENOMEM; goto error; } @@ -1995,7 +2039,7 @@ int snd_emu10k1_create(struct snd_card *card, 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT; /* Clear silent pages and set up pointers */ - memset(emu->silent_page.area, 0, PAGE_SIZE); + memset(emu->silent_page.area, 0, silent_page_size); silent_page = emu->silent_page.addr << emu->address_mode; for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++) ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 2683b9717215..80b3279692b8 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -411,12 +411,19 @@ static int snd_emu10k1_playback_hw_params(struct snd_pcm_substream *substream, struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm = runtime->private_data; + size_t alloc_size; int err; if ((err = snd_emu10k1_pcm_channel_alloc(epcm, params_channels(hw_params))) < 0) return err; - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) + + alloc_size = params_buffer_bytes(hw_params); + if (emu->iommu_workaround) + alloc_size += EMUPAGESIZE; + if ((err = snd_pcm_lib_malloc_pages(substream, alloc_size)) < 0) return err; + if (emu->iommu_workaround && runtime->dma_bytes >= EMUPAGESIZE) + runtime->dma_bytes -= EMUPAGESIZE; if (err > 0) { /* change */ int mapped; if (epcm->memblk != NULL) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 5cdffe2d31e1..6a5371d10dcf 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -457,6 +457,16 @@ static void get_single_page_range(struct snd_util_memhdr *hdr, *last_page_ret = last_page; } +static size_t synth_get_alloc_size(struct snd_emu10k1 *emu) +{ + size_t alloc_size = PAGE_SIZE; + + if (emu->iommu_workaround) + alloc_size *= 2; + + return alloc_size; +} + /* * allocate kernel pages */ @@ -471,7 +481,7 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk for (page = first_page; page <= last_page; page++) { if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), - PAGE_SIZE, &dmab) < 0) + synth_get_alloc_size(emu), &dmab) < 0) goto __fail; if (!is_valid_page(emu, dmab.addr)) { snd_dma_free_pages(&dmab); @@ -488,7 +498,7 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk for (page = first_page; page <= last_page; page++) { dmab.area = emu->page_ptr_table[page]; dmab.addr = emu->page_addr_table[page]; - dmab.bytes = PAGE_SIZE; + dmab.bytes = synth_get_alloc_size(emu); snd_dma_free_pages(&dmab); emu->page_addr_table[page] = 0; emu->page_ptr_table[page] = NULL; @@ -513,7 +523,7 @@ static int synth_free_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk * continue; dmab.area = emu->page_ptr_table[page]; dmab.addr = emu->page_addr_table[page]; - dmab.bytes = PAGE_SIZE; + dmab.bytes = synth_get_alloc_size(emu); snd_dma_free_pages(&dmab); emu->page_addr_table[page] = 0; emu->page_ptr_table[page] = NULL;

2 4

[alsa-devel] [PATCH v2 1/2] ASoC: wm9712: Use empty struct initializer
by Fabio Estevam 13 Feb '18

13 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> { 0 } only clears the first member of the structure. The first member of the snd_soc_dapm_update struct is a pointer, and writing 0 to a pointer results in the following sparse warning: sound/soc/codecs/wm9712.c:229:47: warning: Using plain integer as NULL pointer Use the empty struct initializer that clears all the struct members and fixes the sparse warning. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Fix typos in commit log sound/soc/codecs/wm9712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 7aeeeec..953d94d 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -226,7 +226,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int mixer, mask, shift, old; - struct snd_soc_dapm_update update = { 0 }; + struct snd_soc_dapm_update update = {}; bool change; mixer = mc->shift >> 8; -- 2.7.4

1 1

[alsa-devel] [PATCH v2 0/2] Add support for AK4458 DAC
by Cosmin-Gabriel Samoila 13 Feb '18

13 Feb '18

We support Normal Mode, TDM Mode and PM. Changes since V1: - support only i2c - i2c probe / remove/suspend/ resume moved into shared code - change enum array to separate variables - aggregate Volume control for Left and Right Channel - remove mux from DAPM - remove unused functions - remove unused controls - remove unused macros - use snd_soc_update_bits instead of soc_read + soc_write - use GPIOD - use SPDX identifier - use dac@10 instgitead of ak4458@10 - use explicit power_on/off functions - add __maybe_unused attribute - use probe_new for i2c_probe Cosmin-Gabriel Samoila (2): ASoC: codecs: Add support for AK4458 DAC driver ASoC: ak4458: Add bindings for AK4458 DAC Documentation/devicetree/bindings/sound/ak4458.txt | 23 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4458.c | 684 +++++++++++++++++++++ sound/soc/codecs/ak4458.h | 75 +++ 5 files changed, 790 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4458.txt create mode 100644 sound/soc/codecs/ak4458.c create mode 100644 sound/soc/codecs/ak4458.h -- 2.7.4

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[alsa-devel] [PATCH 0/2] ASoC: missing replace platform/codec to component
by Kuninori Morimoto 13 Feb '18

13 Feb '18

Hi Mark These are missing replace platform/codec patches which are not yet accepted (except ac97 issue) uniphier: aio-dmac arizona -- 1.9.1

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Re: [alsa-devel] [PATCH] ALSA: usb-audio: Fix UAC2 get_ctl request with a RANGE attribute
by Takashi Iwai 13 Feb '18

13 Feb '18

On Thu, 08 Feb 2018 07:21:36 +0100, Kirill Marinushkin wrote: > > On 02/07/18 06:45, Takashi Iwai wrote: > > On Mon, 29 Jan 2018 06:37:55 +0100, > > Kirill Marinushkin wrote: > >> The layout of the UAC2 Control request and response varies depending on > >> the request type. With the current implementation, only the Layout 2 > >> Parameter Block (with the 2-byte sized RANGE attribute) is handled > >> properly. For the Control requests with the 1-byte sized RANGE attribute > >> (Bass Control, Mid Control, Tremble Control), the response is parsed > >> incorrectly. > >> > >> This commit: > >> * fixes the wLength field value in the request > >> * fixes parsing the range values from the response > >> > >> Fixes: 23caaf19b11e ("ALSA: usb-mixer: Add support for Audio Class v2.0") > >> Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> > >> Cc: Jaroslav Kysela <perex(a)perex.cz> > >> Cc: Takashi Iwai <tiwai(a)suse.com> > >> Cc: Jaejoong Kim <climbbb.kim(a)gmail.com> > >> Cc: Bhumika Goyal <bhumirks(a)gmail.com> > >> Cc: Stephen Barber <smbarber(a)chromium.org> > >> Cc: Julian Scheel <julian(a)jusst.de> > >> Cc: alsa-devel(a)alsa-project.org > >> Cc: linux-kernel(a)vger.kernel.org > > Sorry for the late reply, as I've been (and still) off. > > > > Does this bug actually hit on any real devices, or is it only a > > logical error so far? In the former case, a Cc to stable is > > mandatory. > > > > In anyway, I'll review and merge it properly once after I back to > > work. > > > > > > thanks, > > > > Takashi > > Hello Takashi, > > Thank you for your answer. I will wait until you are back to work, don't > worry about the late replies. > > I did not hit the issue on a real device. > > During my UAC2 experiments, I reproduced this issue on the development > board, and then tested my solution on it. OK, if it happened on a development system, it's a real issue, and we should put Cc to stable. Now I applied the patch. Thanks! Takashi

2 3

[alsa-devel] [PATCH v3 0/3] add UniPhier audio nodes
by Katsuhiro Suzuki 13 Feb '18

13 Feb '18

This series adds device-tree nodes for Socionext audio system for UniPhier LD11/LD20 SoCs. This one enables I2S output for Line-In, Line-Out, S/PDIF output and compress audio output. Katsuhiro Suzuki (3): arm64: dts: uniphier: add sound node for UniPhier arm64: dts: uniphier: add speaker out for UniPhier LD11/LD20 boards arm64: dts: uniphier: add compress audio out for UniPhier LD11/LD20 .../boot/dts/socionext/uniphier-ld11-global.dts | 62 ++++++++++++++ arch/arm64/boot/dts/socionext/uniphier-ld11.dtsi | 92 +++++++++++++++++++++ .../boot/dts/socionext/uniphier-ld20-global.dts | 62 ++++++++++++++ arch/arm64/boot/dts/socionext/uniphier-ld20.dtsi | 96 ++++++++++++++++++++++ 4 files changed, 312 insertions(+) -- 2.15.1

2 4

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
by Peter Ujfalusi 13 Feb '18

13 Feb '18

In the reset state of the codec we do not have complete playback or capture routes. The audio playback/capture will not work due to missing clock signals on the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down. To make sure that even if all output/input is disconnected the codec is generating clocks, we need to have valid DAPM route in every case to power up the must needed parts of the codec. I have verified that switching DAC (during playback) or ADC (during capture) will stop the I2S clocks, so the only solution is to connect the 'Off' routes as well to output/input. Tested on am43x-epos-evm with aic3111 codec in master mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 858cb8be445f..d42a23eb916e 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -586,9 +586,11 @@ common31xx_audio_map[] = { {"DAC Left Input", "Left Data", "DAC IN"}, {"DAC Left Input", "Right Data", "DAC IN"}, {"DAC Left Input", "Mono", "DAC IN"}, + {"DAC Left Input", "Off", "DAC IN"}, {"DAC Right Input", "Left Data", "DAC IN"}, {"DAC Right Input", "Right Data", "DAC IN"}, {"DAC Right Input", "Mono", "DAC IN"}, + {"DAC Right Input", "Off", "DAC IN"}, {"DAC Left", NULL, "DAC Left Input"}, {"DAC Right", NULL, "DAC Right Input"}, @@ -601,6 +603,9 @@ common31xx_audio_map[] = { {"HP Right", "Switch", "Output Right"}, {"HPR Driver", NULL, "HP Right"}, {"HPR", NULL, "HPR Driver"}, + + {"HPL", NULL, "DAC Left"}, + {"HPR", NULL, "DAC Right"}, }; static const struct snd_soc_dapm_route @@ -621,16 +626,20 @@ aic31xx_audio_map[] = { {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "Off", "MIC1LP"}, {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"}, {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"}, {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "Off", "MIC1RP"}, {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"}, {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"}, {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "Off", "MIC1LM"}, {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"}, {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"}, {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "Off", "MIC1LM"}, {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"}, {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"}, -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] ASoC: rsnd: status debug message and its suppress
by Kuninori Morimoto 13 Feb '18

13 Feb '18

Hi Mark Renesas sound driver want to know status when debugging, but sometimes it is too much information. These patches indicate status information when debugging, but we can suppress it by special define Kuninori Morimoto (2): ASoC: rsnd: indicate IRQ error status for debug ASoC: rsnd: suppress rsnd_dai_call() debug message sound/soc/sh/rcar/core.c | 11 ++++++++++- sound/soc/sh/rcar/rsnd.h | 20 ++++++++++++++++++++ sound/soc/sh/rcar/src.c | 22 ++++++++++++++++++++-- sound/soc/sh/rcar/ssi.c | 16 +++++++++++++++- 4 files changed, 65 insertions(+), 4 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [asoc:topic/uniphier 3/7] sound/soc/uniphier/aio-core.c:907:5: sparse: symbol 'aiodma_rb_get_rp' was not declared. Should it be static?
by kbuild test robot 13 Feb '18

13 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/uniphier head: db4cb3d04eba91db48da0032c44b83ed7185973c commit: f37fe2f9987b0460f25a87b1380f8e97a5959121 [3/7] ASoC: uniphier: add support for UniPhier AIO common driver reproduce: # apt-get install sparse git checkout f37fe2f9987b0460f25a87b1380f8e97a5959121 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/uniphier/aio-core.c:907:5: sparse: symbol 'aiodma_rb_get_rp' was not declared. Should it be Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/rt5659 1/3] sound/soc/codecs/rt5659.c:4048:6: sparse: symbol 'rt5659_intel_hd_header_probe_setup' was not declared. Should it be static?
by kbuild test robot 13 Feb '18

13 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5659 head: de86a33253d07675abbb1fa294ffb4f3df6f219e commit: 041e74b71491acadad74b275e8b05add165d92b4 [1/3] ASoC: rt5659: Add the support of Intel HDA Header reproduce: # apt-get install sparse git checkout 041e74b71491acadad74b275e8b05add165d92b4 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/rt5659.c:4048:6: sparse: symbol 'rt5659_intel_hd_header_probe_setup' was not declared. Should it be Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: wm9081: fixup wm9081_digital_mute() reg read
by Kuninori Morimoto 13 Feb '18

13 Feb '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 48c338764296 ("ASoC: wm9081: replace codec to component") replaced codec to component, but it didn't keep WM9081_DAC_DIGITAL_2 value to reg. This patch fixup it. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm9081.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 52dd610..5a0ea7b 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1155,7 +1155,7 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) struct snd_soc_component *component = codec_dai->component; unsigned int reg; - snd_soc_component_read32(component, WM9081_DAC_DIGITAL_2); + reg = snd_soc_component_read32(component, WM9081_DAC_DIGITAL_2); if (mute) reg |= WM9081_DAC_MUTE; -- 1.9.1

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[alsa-devel] [asoc:topic/wm9705 1/1] sound/soc/codecs/wm9705.c:330:18: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'?
by kbuild test robot 13 Feb '18

13 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm9705 head: 93ada23cd28b04d50ddd7b3e68a93706dc28b59c commit: 93ada23cd28b04d50ddd7b3e68a93706dc28b59c [1/1] ASoC: wm9705: replace codec to component config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 93ada23cd28b04d50ddd7b3e68a93706dc28b59c # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): sound/soc/codecs/wm9705.c: In function 'wm9705_soc_probe': >> sound/soc/codecs/wm9705.c:330:18: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'? [-Werror=implicit-function-declaration] wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, ^~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_new_ac97_codec >> sound/soc/codecs/wm9705.c:330:16: warning: assignment makes pointer from integer without a cast [-Wint-conversion] wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, ^ >> sound/soc/codecs/wm9705.c:339:4: error: implicit declaration of function 'snd_soc_free_ac97_component'; did you mean 'snd_soc_free_ac97_codec'? [-Werror=implicit-function-declaration] snd_soc_free_ac97_component(wm9705->ac97); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_free_ac97_codec cc1: some warnings being treated as errors vim +330 sound/soc/codecs/wm9705.c 319 320 static int wm9705_soc_probe(struct snd_soc_component *component) 321 { 322 struct wm9705_priv *wm9705 = snd_soc_component_get_drvdata(component); 323 struct regmap *regmap; 324 325 if (wm9705->mfd_pdata) { 326 wm9705->ac97 = wm9705->mfd_pdata->ac97; 327 regmap = wm9705->mfd_pdata->regmap; 328 } else { 329 #ifdef CONFIG_SND_SOC_AC97_BUS > 330 wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, 331 WM9705_VENDOR_ID_MASK); 332 if (IS_ERR(wm9705->ac97)) { 333 dev_err(component->dev, "Failed to register AC97 codec\n"); 334 return PTR_ERR(wm9705->ac97); 335 } 336 337 regmap = regmap_init_ac97(wm9705->ac97, &wm9705_regmap_config); 338 if (IS_ERR(regmap)) { > 339 snd_soc_free_ac97_component(wm9705->ac97); 340 return PTR_ERR(regmap); 341 } 342 #endif 343 } 344 345 snd_soc_component_set_drvdata(component, wm9705->ac97); 346 snd_soc_component_init_regmap(component, regmap); 347 348 return 0; 349 } 350 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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Re: [alsa-devel] [PATCH] ASoC: codecs: Add support for AK4458 DAC driver
by Mark Brown 13 Feb '18

13 Feb '18

On Wed, Jan 31, 2018 at 03:20:09PM +0200, Cosmin-Gabriel Samoila wrote: This looks pretty good overall, I've got some issues below but nothing too major: > +static int ak4458_i2c_remove(struct i2c_client *i2c) > +{ > + ak4458_remove(&i2c->dev); > + pm_runtime_disable(&i2c->dev); It's weird that the runtime PM handling is here and not in the shared code, and that it only exists in the I2C version. Why is this? > +static const struct soc_enum ak4458_dac_enum[] = { > +/*0*/ SOC_ENUM_SINGLE(AK4458_01_CONTROL2, 1, > + ARRAY_SIZE(ak4458_dem_select_texts), > + ak4458_dem_select_texts), > +/*1*/ SOC_ENUM_SINGLE(AK4458_0A_CONTROL6, 0, The fact that you need these comments is why these arrays are a bad idea - just use individually named variables as other drivers do. > +static const struct snd_kcontrol_new ak4458_snd_controls[] = { > + SOC_SINGLE_TLV("AK4458 L1ch Digital Volume", > + AK4458_03_LCHATT, 0/*shift*/, 0xFF/*max value*/, > + 0/*invert*/, latt_tlv), > + SOC_SINGLE_TLV("AK4458 R1ch Digital Volume", > + AK4458_04_RCHATT, 0, 0xFF, 0, ratt_tlv), It'd be more idiomatic to combine these into stereo pairs than have them as single channel controls. > +static const char * const ak4458_dac_select_texts[] = { "OFF", "ON" }; This looks like the users should be switch controls - what's the goal here? > + rstn = snd_soc_read(codec, AK4458_00_CONTROL1); > + rstn &= ~AK4458_RSTN_MASK; > + > + if (bit) > + rstn |= AK4458_RSTN; > + > + snd_soc_write(codec, AK4458_00_CONTROL1, rstn); This looks like an open coded snd_soc_update_bits()? > +static int ak4458_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct ak4458_priv *ak4458 = snd_soc_codec_get_drvdata(codec); > + u8 format; > + int pcm_width = max(params_physical_width(params), ak4458->slot_width); > + > +#ifdef AK4458_ACKS_USE_MANUAL_MODE > + u8 dfs1, dfs2; > +#endif What's this and why is it a compile time option? > + case 32: > + if (ak4458->fmt == SND_SOC_DAIFMT_I2S) > + format |= AK4458_DIF_32BIT_I2S; > + else if (ak4458->fmt == SND_SOC_DAIFMT_LEFT_J) > + format |= AK4458_DIF_32BIT_MSB; > + else if (ak4458->fmt == SND_SOC_DAIFMT_RIGHT_J) > + format |= AK4458_DIF_32BIT_LSB; > + else if (ak4458->fmt == SND_SOC_DAIFMT_DSP_B) > + format |= AK4458_DIF_32BIT_MSB; > + else > + return -EINVAL; This should be a switch statement. > + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_I2S: > + case SND_SOC_DAIFMT_LEFT_J: > + case SND_SOC_DAIFMT_RIGHT_J: > + ak4458->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; > + break; > + case SND_SOC_DAIFMT_DSP_B: > + ak4458->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; > + break; All these cases seem to be the same? > +static int ak4458_trigger(struct snd_pcm_substream *substream, int cmd, > + struct snd_soc_dai *codec_dai) > +{ > + int ret = 0; > + > + dev_dbg(codec_dai->dev, "%s(%d)\n", __func__, __LINE__); > + return ret; > +} Remove empty functions. > + ak4458->mute_gpio = of_get_named_gpio(np, "ak4458,mute_gpio", 0); > + if (gpio_is_valid(ak4458->mute_gpio)) { Given that this is new code it'd be better to use GPIO descriptors, this will also allow support for non-DT systems - use devm_gpiod_get() and the matching APIs.

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[alsa-devel] [PATCH v2] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
by Peter Ujfalusi 13 Feb '18

13 Feb '18

In the reset state of the codec we do not have complete playback or capture routes. The audio playback/capture will not work due to missing clock signals on the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down. To make sure that even if all output/input is disconnected the codec is generating clocks, we need to have valid DAPM route in every case to power up the must needed parts of the codec. I have verified that switching DAC (during playback) or ADC (during capture) will stop the I2S clocks, so the only solution is to connect the 'Off' routes as well to output/input. The routes only going to be added if the codec is configured as clock master. Tested on am43x-epos-evm with aic3111 codec in master mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Only apply the master mode DAPM routes when the codec is clock master - comments added to explain the need. Regards, Peter sound/soc/codecs/tlv320aic31xx.c | 46 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 858cb8be445f..15de89318ff5 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -166,6 +166,7 @@ struct aic31xx_priv { unsigned int sysclk; u8 p_div; int rate_div_line; + bool master_dapm_route_applied; }; struct aic31xx_rate_divs { @@ -670,6 +671,29 @@ aic310x_audio_map[] = { {"SPK", NULL, "SPK ClassD"}, }; +/* + * Always connected DAPM routes for codec clock master modes. + * If the codec is the master on the I2S bus, we need to power on components + * to have valid DAC_CLK and also the DACs and ADC for playback/capture. + * Otherwise the codec will not generate clocks on the bus. + */ +static const struct snd_soc_dapm_route +common31xx_cm_audio_map[] = { + {"DAC Left Input", "Off", "DAC IN"}, + {"DAC Right Input", "Off", "DAC IN"}, + + {"HPL", NULL, "DAC Left"}, + {"HPR", NULL, "DAC Right"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_cm_audio_map[] = { + {"MIC1LP P-Terminal", "Off", "MIC1LP"}, + {"MIC1RP P-Terminal", "Off", "MIC1RP"}, + {"MIC1LM P-Terminal", "Off", "MIC1LM"}, + {"MIC1LM M-Terminal", "Off", "MIC1LM"}, +}; + static int aic31xx_add_controls(struct snd_soc_codec *codec) { int ret = 0; @@ -916,6 +940,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); u8 iface_reg1 = 0; u8 iface_reg2 = 0; u8 dsp_a_val = 0; @@ -992,6 +1017,27 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, AIC31XX_BCLKINV_MASK, iface_reg2); + /* Add the needed DAPM route(s) for codec clock master modes, once */ + if (iface_reg1 & (AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER) && + !aic31xx->master_dapm_route_applied) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int ret; + + pr_err("%s: trying to add master routes\n", __func__); + ret = snd_soc_dapm_add_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_add_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + pr_err("%s: master routes added with %d\n", __func__, ret); + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = true; + } + return 0; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] pyalsa support for Python 3
by Maurizio Berti 13 Feb '18

13 Feb '18

Hello, I used pyalsa's alsaseq module for a project of mine which used Python 2. Now I'm switching to Python 3 and I'd need to use the alsaseq module with that version. Unfortunally, it seems that Python 3 support has not been added yet. I found this message: http://linux-audio.4202.n7.nabble.com/Python3-PyAlsa-Build-fails-td83590.ht… It is older than the latest release of pyalsa, but, even after trying to follow the instructions, I still get errors and cannot build pyalsa for Python3: while the "pyalsa/alsacontrol.c:184:6: error: ‘struct pyalsacontrol’ has no member named ‘ob_type’" error is gone, it still doesn't build. The latest message reports: error: command 'i686-pc-linux-gnu-gcc' failed with exit status 1 Is there any chance that pyalsa gets an update for it? Thank you, Maurizio -- È difficile avere una convinzione precisa quando si parla delle ragioni del cuore. - "Sostiene Pereira", Antonio Tabucchi http://www.jidesk.net

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[alsa-devel] [asoc:topic/rt5514 1/1] sound/soc/codecs/rt5514-spi.c:477:70: error: 'DRV_NAME' undeclared; did you mean 'IRQ_NONE'?
by kbuild test robot 13 Feb '18

13 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5514 head: 04e8262ac8c3f59599ea6dfe5cd1a039828ea4ad commit: 04e8262ac8c3f59599ea6dfe5cd1a039828ea4ad [1/1] ASoC: rt5514: replace codec to component config: xtensa-allyesconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 04e8262ac8c3f59599ea6dfe5cd1a039828ea4ad # save the attached .config to linux build tree make.cross ARCH=xtensa All errors (new ones prefixed by >>): sound/soc/codecs/rt5514-spi.c: In function 'rt5514_resume': >> sound/soc/codecs/rt5514-spi.c:477:70: error: 'DRV_NAME' undeclared (first use in this function); did you mean 'IRQ_NONE'? struct snd_soc_component *component = snd_soc_lookup_component(dev, DRV_NAME); ^~~~~~~~ IRQ_NONE sound/soc/codecs/rt5514-spi.c:477:70: note: each undeclared identifier is reported only once for each function it appears in vim +477 sound/soc/codecs/rt5514-spi.c 474 475 static int __maybe_unused rt5514_resume(struct device *dev) 476 { > 477 struct snd_soc_component *component = snd_soc_lookup_component(dev, DRV_NAME); 478 struct rt5514_dsp *rt5514_dsp = 479 snd_soc_component_get_drvdata(component); 480 int irq = to_spi_device(dev)->irq; 481 u8 buf[8]; 482 483 if (device_may_wakeup(dev)) 484 disable_irq_wake(irq); 485 486 if (rt5514_dsp) { 487 if (rt5514_dsp->substream) { 488 rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, 489 sizeof(buf)); 490 if (buf[0] & RT5514_IRQ_STATUS_BIT) 491 rt5514_schedule_copy(rt5514_dsp); 492 } 493 } 494 495 return 0; 496 } 497 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v2 0/9] add UniPhier audio system support
by Katsuhiro Suzuki 13 Feb '18

13 Feb '18

This series adds support for Socionext audio system for UniPhier LD11/LD20 SoCs. This driver supports I2S output for Line-In, Line-Out and S/PDIF output. UniPhier AIO DAI driver provides sound devices such as I2S, S/PDIF. Since the AIO has mixed register map for those I/Os, it is hard to split register areas for each sound devices. --- Changes in v2: - Add comments to aiodma_irq() - Add members to struct uniphier_aio_sub and add methods for compress audio - Expose clocking to userspace - Fix bad name 'srcport' to 'src' - Split DMA, DAI patches from large one - Validate parameters in hw_params() - Add error checks - Fix typo in error messages - Change license comment style to C++ from C Katsuhiro Suzuki (9): ASoC: uniphier: add DT bindings documentation for UniPhier AIO ASoC: uniphier: add support for UniPhier AIO common driver ASoC: uniphier: add support for UniPhier AIO DMA driver ASoC: uniphier: add support for UniPhier AIO CPU DAI driver ASoC: uniphier: add support for UniPhier AIO compress audio ASoC: uniphier: add support for UniPhier LD11/LD20 AIO driver arm64: dts: uniphier: add sound node for UniPhier arm64: dts: uniphier: add speaker out for UniPhier LD11/LD20 boards arm64: dts: uniphier: add compress audio out for UniPhier LD11/LD20 .../devicetree/bindings/sound/uniphier,aio.txt | 36 + .../boot/dts/socionext/uniphier-ld11-global.dts | 120 +++ arch/arm64/boot/dts/socionext/uniphier-ld11.dtsi | 31 + .../boot/dts/socionext/uniphier-ld20-global.dts | 120 +++ arch/arm64/boot/dts/socionext/uniphier-ld20.dtsi | 35 + sound/soc/uniphier/Kconfig | 22 + sound/soc/uniphier/Makefile | 6 + sound/soc/uniphier/aio-compress.c | 440 ++++++++ sound/soc/uniphier/aio-core.c | 1104 ++++++++++++++++++++ sound/soc/uniphier/aio-cpu.c | 570 ++++++++++ sound/soc/uniphier/aio-dma.c | 318 ++++++ sound/soc/uniphier/aio-ld11.c | 431 ++++++++ sound/soc/uniphier/aio-reg.h | 462 ++++++++ sound/soc/uniphier/aio.h | 355 +++++++ 14 files changed, 4050 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt create mode 100644 sound/soc/uniphier/aio-compress.c create mode 100644 sound/soc/uniphier/aio-core.c create mode 100644 sound/soc/uniphier/aio-cpu.c create mode 100644 sound/soc/uniphier/aio-dma.c create mode 100644 sound/soc/uniphier/aio-ld11.c create mode 100644 sound/soc/uniphier/aio-reg.h create mode 100644 sound/soc/uniphier/aio.h -- 2.15.0

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[alsa-devel] [asoc:topic/wm9712 1/1] sound/soc//codecs/wm9712.c:648:18: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'?
by kbuild test robot 13 Feb '18

13 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm9712 head: 164c976a9493991ebe7e838931ff895e6e04d7c2 commit: 164c976a9493991ebe7e838931ff895e6e04d7c2 [1/1] ASoC: wm9712: replace codec to component config: sparc64-allyesconfig (attached as .config) compiler: sparc64-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 164c976a9493991ebe7e838931ff895e6e04d7c2 # save the attached .config to linux build tree make.cross ARCH=sparc64 All error/warnings (new ones prefixed by >>): sound/soc//codecs/wm9712.c: In function 'wm9712_soc_probe': >> sound/soc//codecs/wm9712.c:648:18: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'? [-Werror=implicit-function-declaration] wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, ^~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_new_ac97_codec >> sound/soc//codecs/wm9712.c:648:16: warning: assignment makes pointer from integer without a cast [-Wint-conversion] wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, ^ >> sound/soc//codecs/wm9712.c:659:4: error: implicit declaration of function 'snd_soc_free_ac97_component'; did you mean 'snd_soc_free_ac97_codec'? [-Werror=implicit-function-declaration] snd_soc_free_ac97_component(wm9712->ac97); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_free_ac97_codec cc1: some warnings being treated as errors vim +648 sound/soc//codecs/wm9712.c 636 637 static int wm9712_soc_probe(struct snd_soc_component *component) 638 { 639 struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); 640 struct regmap *regmap; 641 int ret; 642 643 if (wm9712->mfd_pdata) { 644 wm9712->ac97 = wm9712->mfd_pdata->ac97; 645 regmap = wm9712->mfd_pdata->regmap; 646 } else { 647 #ifdef CONFIG_SND_SOC_AC97_BUS > 648 wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, 649 WM9712_VENDOR_ID_MASK); 650 if (IS_ERR(wm9712->ac97)) { 651 ret = PTR_ERR(wm9712->ac97); 652 dev_err(component->dev, 653 "Failed to register AC97 codec: %d\n", ret); 654 return ret; 655 } 656 657 regmap = regmap_init_ac97(wm9712->ac97, &wm9712_regmap_config); 658 if (IS_ERR(regmap)) { > 659 snd_soc_free_ac97_component(wm9712->ac97); 660 return PTR_ERR(regmap); 661 } 662 #endif 663 } 664 665 snd_soc_component_init_regmap(component, regmap); 666 667 /* set alc mux to none */ 668 snd_soc_component_update_bits(component, AC97_VIDEO, 0x3000, 0x3000); 669 670 return 0; 671 } 672 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/wm9713 1/1] sound/soc/codecs/wm9713.c:1219:18: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'?
by kbuild test robot 13 Feb '18

13 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm9713 head: 807cef2d327a5148d456d9c4966bfb9b2785ffc8 commit: 807cef2d327a5148d456d9c4966bfb9b2785ffc8 [1/1] ASoC: wm9713: replace codec to component config: cris-allmodconfig (attached as .config) compiler: cris-linux-gcc (GCC) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 807cef2d327a5148d456d9c4966bfb9b2785ffc8 # save the attached .config to linux build tree make.cross ARCH=cris All errors (new ones prefixed by >>): sound/soc/codecs/wm9713.c: In function 'wm9713_soc_probe': >> sound/soc/codecs/wm9713.c:1219:18: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'? [-Werror=implicit-function-declaration] wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, ^~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_new_ac97_codec sound/soc/codecs/wm9713.c:1219:16: warning: assignment makes pointer from integer without a cast [-Wint-conversion] wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, ^ >> sound/soc/codecs/wm9713.c:1225:4: error: implicit declaration of function 'snd_soc_free_ac97_component'; did you mean 'snd_soc_free_ac97_codec'? [-Werror=implicit-function-declaration] snd_soc_free_ac97_component(wm9713->ac97); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_free_ac97_codec cc1: some warnings being treated as errors vim +1219 sound/soc/codecs/wm9713.c 1208 1209 static int wm9713_soc_probe(struct snd_soc_component *component) 1210 { 1211 struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); 1212 struct regmap *regmap = NULL; 1213 1214 if (wm9713->mfd_pdata) { 1215 wm9713->ac97 = wm9713->mfd_pdata->ac97; 1216 regmap = wm9713->mfd_pdata->regmap; 1217 } else { 1218 #ifdef CONFIG_SND_SOC_AC97_BUS > 1219 wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, 1220 WM9713_VENDOR_ID_MASK); 1221 if (IS_ERR(wm9713->ac97)) 1222 return PTR_ERR(wm9713->ac97); 1223 regmap = regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); 1224 if (IS_ERR(regmap)) { > 1225 snd_soc_free_ac97_component(wm9713->ac97); 1226 return PTR_ERR(regmap); 1227 } 1228 #endif 1229 } 1230 1231 snd_soc_component_init_regmap(component, regmap); 1232 1233 /* unmute the adc - move to kcontrol */ 1234 snd_soc_component_update_bits(component, AC97_CD, 0x7fff, 0x0000); 1235 1236 return 0; 1237 } 1238 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v2 4/5] ALSA: emu10k1: make sure synth DMA pages are allocated with DMA functions
by Maciej S. Szmigiero 13 Feb '18

13 Feb '18

Commit a5003fc04113 ("[ALSA] emu10k1 - simplify page allocation for synth") switched from using the DMA allocator for synth DMA pages to manually calling alloc_page(). However, this usage has an implicit assumption that the DMA address space for the emu10k1-family chip is the same as the CPU physical address space which is not true for a system with a IOMMU. Since this made the synth part of the driver non-functional on such systems let's effectively revert that commit. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: None in this patch. sound/pci/emu10k1/memory.c | 70 ++++++++++++++++++++++++++-------------------- 1 file changed, 39 insertions(+), 31 deletions(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index fcb04cbbc9ab..5cdffe2d31e1 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -457,49 +457,44 @@ static void get_single_page_range(struct snd_util_memhdr *hdr, *last_page_ret = last_page; } -/* release allocated pages */ -static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, - int last_page) -{ - int page; - - for (page = first_page; page <= last_page; page++) { - free_page((unsigned long)emu->page_ptr_table[page]); - emu->page_addr_table[page] = 0; - emu->page_ptr_table[page] = NULL; - } -} - /* * allocate kernel pages */ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { int page, first_page, last_page; + struct snd_dma_buffer dmab; emu10k1_memblk_init(blk); get_single_page_range(emu->memhdr, blk, &first_page, &last_page); /* allocate kernel pages */ for (page = first_page; page <= last_page; page++) { - /* first try to allocate from <4GB zone */ - struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | - __GFP_NOWARN); - if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) { - if (p) - __free_page(p); - /* try to allocate from <16MB zone */ - p = alloc_page(GFP_ATOMIC | GFP_DMA | - __GFP_NORETRY | /* no OOM-killer */ - __GFP_NOWARN); - } - if (!p) { - __synth_free_pages(emu, first_page, page - 1); - return -ENOMEM; + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), + PAGE_SIZE, &dmab) < 0) + goto __fail; + if (!is_valid_page(emu, dmab.addr)) { + snd_dma_free_pages(&dmab); + goto __fail; } - emu->page_addr_table[page] = page_to_phys(p); - emu->page_ptr_table[page] = page_address(p); + emu->page_addr_table[page] = dmab.addr; + emu->page_ptr_table[page] = dmab.area; } return 0; + +__fail: + /* release allocated pages */ + last_page = page - 1; + for (page = first_page; page <= last_page; page++) { + dmab.area = emu->page_ptr_table[page]; + dmab.addr = emu->page_addr_table[page]; + dmab.bytes = PAGE_SIZE; + snd_dma_free_pages(&dmab); + emu->page_addr_table[page] = 0; + emu->page_ptr_table[page] = NULL; + } + + return -ENOMEM; } /* @@ -507,10 +502,23 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk */ static int synth_free_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { - int first_page, last_page; + int page, first_page, last_page; + struct snd_dma_buffer dmab; get_single_page_range(emu->memhdr, blk, &first_page, &last_page); - __synth_free_pages(emu, first_page, last_page); + dmab.dev.type = SNDRV_DMA_TYPE_DEV; + dmab.dev.dev = snd_dma_pci_data(emu->pci); + for (page = first_page; page <= last_page; page++) { + if (emu->page_ptr_table[page] == NULL) + continue; + dmab.area = emu->page_ptr_table[page]; + dmab.addr = emu->page_addr_table[page]; + dmab.bytes = PAGE_SIZE; + snd_dma_free_pages(&dmab); + emu->page_addr_table[page] = 0; + emu->page_ptr_table[page] = NULL; + } + return 0; }

2 2

[alsa-devel] usleep_range without a range
by Joe Perches 13 Feb '18

13 Feb '18

scheduling can generally be better when these values are not identical. Perhaps these ranges should be expanded. $ git grep -P -n "usleep_range\s*$\s*([\w\.\>\-]+)\s*,\s*\1\s*$" drivers/clk/ux500/clk-sysctrl.c:45: usleep_range(clk->enable_delay_us, clk->enable_delay_us); drivers/cpufreq/pmac64-cpufreq.c:140: usleep_range(1000, 1000); drivers/cpufreq/pmac64-cpufreq.c:239: usleep_range(10000, 10000); /* should be faster , to fix */ drivers/cpufreq/pmac64-cpufreq.c:284: usleep_range(500, 500); drivers/media/i2c/smiapp/smiapp-core.c:1228: usleep_range(1000, 1000); drivers/media/i2c/smiapp/smiapp-core.c:1235: usleep_range(1000, 1000); drivers/media/i2c/smiapp/smiapp-core.c:1240: usleep_range(sleep, sleep); drivers/media/i2c/smiapp/smiapp-core.c:1387: usleep_range(5000, 5000); drivers/media/i2c/smiapp/smiapp-quirk.c:205: usleep_range(2000, 2000); drivers/media/i2c/smiapp/smiapp-regs.c:279: usleep_range(2000, 2000); drivers/power/supply/ab8500_fg.c:643: usleep_range(100, 100); drivers/staging/rtl8192u/r819xU_phy.c:180: usleep_range(1000, 1000); drivers/staging/rtl8192u/r819xU_phy.c:736: usleep_range(1000, 1000); drivers/staging/rtl8192u/r819xU_phy.c:740: usleep_range(1000, 1000); sound/soc/codecs/ab8500-codec.c:1065: usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); sound/soc/codecs/ab8500-codec.c:1068: usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY);

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[alsa-devel] [asoc:topic/wm9713 1/1] sound/soc/codecs/wm9713.c:1219:18: error: implicit declaration of function 'snd_soc_new_ac97_component'
by kbuild test robot 12 Feb '18

12 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm9713 head: 807cef2d327a5148d456d9c4966bfb9b2785ffc8 commit: 807cef2d327a5148d456d9c4966bfb9b2785ffc8 [1/1] ASoC: wm9713: replace codec to component config: openrisc-allmodconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 807cef2d327a5148d456d9c4966bfb9b2785ffc8 # save the attached .config to linux build tree make.cross ARCH=openrisc All error/warnings (new ones prefixed by >>): sound/soc/codecs/wm9713.c: In function 'wm9713_soc_probe': >> sound/soc/codecs/wm9713.c:1219:18: error: implicit declaration of function 'snd_soc_new_ac97_component' [-Werror=implicit-function-declaration] wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, ^~~~~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc/codecs/wm9713.c:1219:16: warning: assignment makes pointer from integer without a cast [-Wint-conversion] wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, ^ >> sound/soc/codecs/wm9713.c:1225:4: error: implicit declaration of function 'snd_soc_free_ac97_component' [-Werror=implicit-function-declaration] snd_soc_free_ac97_component(wm9713->ac97); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/snd_soc_new_ac97_component +1219 sound/soc/codecs/wm9713.c 1208 1209 static int wm9713_soc_probe(struct snd_soc_component *component) 1210 { 1211 struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); 1212 struct regmap *regmap = NULL; 1213 1214 if (wm9713->mfd_pdata) { 1215 wm9713->ac97 = wm9713->mfd_pdata->ac97; 1216 regmap = wm9713->mfd_pdata->regmap; 1217 } else { 1218 #ifdef CONFIG_SND_SOC_AC97_BUS > 1219 wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, 1220 WM9713_VENDOR_ID_MASK); 1221 if (IS_ERR(wm9713->ac97)) 1222 return PTR_ERR(wm9713->ac97); 1223 regmap = regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); 1224 if (IS_ERR(regmap)) { > 1225 snd_soc_free_ac97_component(wm9713->ac97); 1226 return PTR_ERR(regmap); 1227 } 1228 #endif 1229 } 1230 1231 snd_soc_component_init_regmap(component, regmap); 1232 1233 /* unmute the adc - move to kcontrol */ 1234 snd_soc_component_update_bits(component, AC97_CD, 0x7fff, 0x0000); 1235 1236 return 0; 1237 } 1238 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/uniphier 6/7] sound/soc/uniphier/aio-compress.c:314:34: sparse: cast to restricted __be16
by kbuild test robot 12 Feb '18

12 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/uniphier head: db4cb3d04eba91db48da0032c44b83ed7185973c commit: e98131222ff931a670739cede00d1fb9d6f83763 [6/7] ASoC: uniphier: add support for UniPhier AIO compress audio reproduce: # apt-get install sparse git checkout e98131222ff931a670739cede00d1fb9d6f83763 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/uniphier/aio-compress.c:314:34: sparse: cast to restricted __be16 >> sound/soc/uniphier/aio-compress.c:314:34: sparse: cast to restricted __be16 >> sound/soc/uniphier/aio-compress.c:314:34: sparse: cast to restricted __be16 >> sound/soc/uniphier/aio-compress.c:314:34: sparse: cast to restricted __be16 vim +314 sound/soc/uniphier/aio-compress.c 290 291 static int aio_compr_send_to_hw(struct uniphier_aio_sub *sub, 292 char __user *buf, size_t dstsize) 293 { 294 u32 __user *srcbuf = (u32 __user *)buf; 295 u32 *dstbuf = (u32 *)(sub->compr_area + sub->wr_offs); 296 int src = 0, dst = 0, ret; 297 u32 frm, frm_a, frm_b; 298 299 while (dstsize > 0) { 300 ret = get_user(frm, srcbuf + src); 301 if (ret) 302 return ret; 303 src++; 304 305 frm_a = frm & 0xffff; 306 frm_b = (frm >> 16) & 0xffff; 307 308 if (frm == IEC61937_HEADER_SIGN) { 309 frm_a |= 0x01000000; 310 311 /* Next data is Pc and Pd */ 312 sub->iec_header = true; 313 } else { > 314 u16 pc = be16_to_cpu((__be16)frm_a); 315 316 if (sub->iec_header && sub->iec_pc != pc) { 317 /* Force overwrite IEC frame type */ 318 sub->iec_pc = pc; 319 ret = aio_oport_set_stream_type(sub, pc); 320 if (ret) 321 return ret; 322 } 323 sub->iec_header = false; 324 } 325 dstbuf[dst++] = frm_a; 326 dstbuf[dst++] = frm_b; 327 328 dstsize -= sizeof(u32) * 2; 329 } 330 331 return 0; 332 } 333 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/ad1980 1/1] sound/soc/codecs/ad1980.c:243:9: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'?
by kbuild test robot 12 Feb '18

12 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/ad1980 head: db53c016644c0d33579c60cc10b5152bbf9706ff commit: db53c016644c0d33579c60cc10b5152bbf9706ff [1/1] ASoC: ad1980: replace codec to component config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout db53c016644c0d33579c60cc10b5152bbf9706ff # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): sound/soc/codecs/ad1980.c: In function 'ad1980_soc_probe': >> sound/soc/codecs/ad1980.c:243:9: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'? [-Werror=implicit-function-declaration] ac97 = snd_soc_new_ac97_component(component, 0, 0); ^~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_new_ac97_codec >> sound/soc/codecs/ad1980.c:243:7: warning: assignment makes pointer from integer without a cast [-Wint-conversion] ac97 = snd_soc_new_ac97_component(component, 0, 0); ^ >> sound/soc/codecs/ad1980.c:285:2: error: implicit declaration of function 'snd_soc_free_ac97_component'; did you mean 'snd_soc_free_ac97_codec'? [-Werror=implicit-function-declaration] snd_soc_free_ac97_component(ac97); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_free_ac97_codec cc1: some warnings being treated as errors vim +243 sound/soc/codecs/ad1980.c 234 235 static int ad1980_soc_probe(struct snd_soc_component *component) 236 { 237 struct snd_ac97 *ac97; 238 struct regmap *regmap; 239 int ret; 240 u16 vendor_id2; 241 u16 ext_status; 242 > 243 ac97 = snd_soc_new_ac97_component(component, 0, 0); 244 if (IS_ERR(ac97)) { 245 ret = PTR_ERR(ac97); 246 dev_err(component->dev, "Failed to register AC97 component: %d\n", ret); 247 return ret; 248 } 249 250 regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); 251 if (IS_ERR(regmap)) { 252 ret = PTR_ERR(regmap); 253 goto err_free_ac97; 254 } 255 256 snd_soc_component_init_regmap(component, regmap); 257 snd_soc_component_set_drvdata(component, ac97); 258 259 ret = ad1980_reset(component, 0); 260 if (ret < 0) 261 goto reset_err; 262 263 vendor_id2 = snd_soc_component_read32(component, AC97_VENDOR_ID2); 264 if (vendor_id2 == 0x5374) { 265 dev_warn(component->dev, 266 "Found AD1981 - only 2/2 IN/OUT Channels supported\n"); 267 } 268 269 /* unmute captures and playbacks volume */ 270 snd_soc_component_write(component, AC97_MASTER, 0x0000); 271 snd_soc_component_write(component, AC97_PCM, 0x0000); 272 snd_soc_component_write(component, AC97_REC_GAIN, 0x0000); 273 snd_soc_component_write(component, AC97_CENTER_LFE_MASTER, 0x0000); 274 snd_soc_component_write(component, AC97_SURROUND_MASTER, 0x0000); 275 276 /*power on LFE/CENTER/Surround DACs*/ 277 ext_status = snd_soc_component_read32(component, AC97_EXTENDED_STATUS); 278 snd_soc_component_write(component, AC97_EXTENDED_STATUS, ext_status&~0x3800); 279 280 return 0; 281 282 reset_err: 283 snd_soc_component_exit_regmap(component); 284 err_free_ac97: > 285 snd_soc_free_ac97_component(ac97); 286 return ret; 287 } 288 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/wm9081 1/1] sound/soc/codecs/wm9081.c:1161:7: warning: 'reg' may be used uninitialized in this function
by kbuild test robot 12 Feb '18

12 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm9081 head: 48c3387642967d62fc47aff4ad0e3816bf8292e3 commit: 48c3387642967d62fc47aff4ad0e3816bf8292e3 [1/1] ASoC: wm9081: replace codec to component config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 48c3387642967d62fc47aff4ad0e3816bf8292e3 # save the attached .config to linux build tree make.cross ARCH=arm Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/codecs/wm9081.c: In function 'wm9081_digital_mute': >> sound/soc/codecs/wm9081.c:1161:7: warning: 'reg' may be used uninitialized in this function [-Wmaybe-uninitialized] reg |= WM9081_DAC_MUTE; vim +/reg +1161 sound/soc/codecs/wm9081.c 86ed3669 Mark Brown 2009-05-22 1152 86ed3669 Mark Brown 2009-05-22 1153 static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) 86ed3669 Mark Brown 2009-05-22 1154 { 48c33876 Kuninori Morimoto 2018-01-29 1155 struct snd_soc_component *component = codec_dai->component; 86ed3669 Mark Brown 2009-05-22 1156 unsigned int reg; 86ed3669 Mark Brown 2009-05-22 1157 48c33876 Kuninori Morimoto 2018-01-29 1158 snd_soc_component_read32(component, WM9081_DAC_DIGITAL_2); 86ed3669 Mark Brown 2009-05-22 1159 86ed3669 Mark Brown 2009-05-22 1160 if (mute) 86ed3669 Mark Brown 2009-05-22 @1161 reg |= WM9081_DAC_MUTE; 86ed3669 Mark Brown 2009-05-22 1162 else 86ed3669 Mark Brown 2009-05-22 1163 reg &= ~WM9081_DAC_MUTE; 86ed3669 Mark Brown 2009-05-22 1164 48c33876 Kuninori Morimoto 2018-01-29 1165 snd_soc_component_write(component, WM9081_DAC_DIGITAL_2, reg); 86ed3669 Mark Brown 2009-05-22 1166 86ed3669 Mark Brown 2009-05-22 1167 return 0; 86ed3669 Mark Brown 2009-05-22 1168 } 86ed3669 Mark Brown 2009-05-22 1169 :::::: The code at line 1161 was first introduced by commit :::::: 86ed3669f068b514ab85ffd548456a342b3fb8d3 ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver :::::: TO: Mark Brown <broonie(a)opensource.wolfsonmicro.com> :::::: CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/stac9766 1/1] sound/soc/codecs/stac9766.c:281:9: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'?
by kbuild test robot 12 Feb '18

12 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/stac9766 head: 033bb535d0db54c7ae14f0dec14177cc424414a5 commit: 033bb535d0db54c7ae14f0dec14177cc424414a5 [1/1] ASoC: stac9766: replace codec to component config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 033bb535d0db54c7ae14f0dec14177cc424414a5 # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): sound/soc/codecs/stac9766.c: In function 'stac9766_component_probe': >> sound/soc/codecs/stac9766.c:281:9: error: implicit declaration of function 'snd_soc_new_ac97_component'; did you mean 'snd_soc_new_ac97_codec'? [-Werror=implicit-function-declaration] ac97 = snd_soc_new_ac97_component(component, STAC9766_VENDOR_ID, ^~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_new_ac97_codec >> sound/soc/codecs/stac9766.c:281:7: warning: assignment makes pointer from integer without a cast [-Wint-conversion] ac97 = snd_soc_new_ac97_component(component, STAC9766_VENDOR_ID, ^ >> sound/soc/codecs/stac9766.c:297:2: error: implicit declaration of function 'snd_soc_free_ac97_component'; did you mean 'snd_soc_free_ac97_codec'? [-Werror=implicit-function-declaration] snd_soc_free_ac97_component(ac97); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ snd_soc_free_ac97_codec cc1: some warnings being treated as errors vim +281 sound/soc/codecs/stac9766.c 274 275 static int stac9766_component_probe(struct snd_soc_component *component) 276 { 277 struct snd_ac97 *ac97; 278 struct regmap *regmap; 279 int ret; 280 > 281 ac97 = snd_soc_new_ac97_component(component, STAC9766_VENDOR_ID, 282 STAC9766_VENDOR_ID_MASK); 283 if (IS_ERR(ac97)) 284 return PTR_ERR(ac97); 285 286 regmap = regmap_init_ac97(ac97, &stac9766_regmap_config); 287 if (IS_ERR(regmap)) { 288 ret = PTR_ERR(regmap); 289 goto err_free_ac97; 290 } 291 292 snd_soc_component_init_regmap(component, regmap); 293 snd_soc_component_set_drvdata(component, ac97); 294 295 return 0; 296 err_free_ac97: > 297 snd_soc_free_ac97_component(ac97); 298 return ret; 299 } 300 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] ALSA: tons of false positives because of snd_ctl_find_id()
by Dan Carpenter 12 Feb '18

12 Feb '18

Hello Takashi Iwai, The patch 72cbfd45fac6: "ALSA: ice1724 - Add ESI Maya44 support" from May 6, 2009, leads to the following static checker warning: sound/pci/ice1712/maya44.c:204 maya_vol_put() error: buffer overflow 'chip->wm' 2 <= u32max. sound/pci/ice1712/maya44.c 199 static int maya_vol_put(struct snd_kcontrol *kcontrol, 200 struct snd_ctl_elem_value *ucontrol) 201 { 202 struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); 203 struct snd_wm8776 *wm = 204 &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)]; ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ This is a false positive obviously, but this code is crazy hard for static analysis to parse. snd_ctl_find_id() in sound/core/control.c 682 /** 683 * snd_ctl_find_id - find the control instance with the given id 684 * @card: the card instance 685 * @id: the id to search 686 * 687 * Finds the control instance with the given id from the card. 688 * 689 * The caller must down card->controls_rwsem before calling this function 690 * (if the race condition can happen). 691 * 692 * Return: The pointer of the instance if found, or %NULL if not. 693 * 694 */ 695 struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, 696 struct snd_ctl_elem_id *id) 697 { 698 struct snd_kcontrol *kctl; 699 700 if (snd_BUG_ON(!card || !id)) 701 return NULL; 702 if (id->numid != 0) 703 return snd_ctl_find_numid(card, id->numid); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ On this path, we don't check id->index. Which is fine because we don't use it. It would really make my life simpler if we could set the unchecked fields to zero. How Smatch works is that we merge all the success paths together so it would either be zero or checked which is easy to deal with. In the current code it's either checked or unchecked which is just unchecked when you merge the success paths together. I can probably figure out other ways to deal with this if that's not a good idea. 704 list_for_each_entry(kctl, &card->controls, list) { 705 if (kctl->id.iface != id->iface) 706 continue; 707 if (kctl->id.device != id->device) 708 continue; 709 if (kctl->id.subdevice != id->subdevice) 710 continue; 711 if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name))) 712 continue; 713 if (kctl->id.index > id->index) 714 continue; 715 if (kctl->id.index + kctl->count <= id->index) 716 continue; 717 return kctl; 718 } 719 return NULL; 720 } 721 EXPORT_SYMBOL(snd_ctl_find_id); regards, dan carpenter

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[alsa-devel] [PATCH 0/8] add UniPhier audio system support
by Katsuhiro Suzuki 12 Feb '18

12 Feb '18

This series adds support for Socionext audio system for UniPhier LD11/LD20 SoCs. This driver supports I2S output for Line-In, Line-Out and S/PDIF output. UniPhier AIO DAI driver provides sound devices such as I2S, S/PDIF. Since the AIO has mixed register map for those I/Os, it is hard to split register areas for each sound devices. UniPhier EVEA is a codec core for Line-In, Line-Out. Since this core is only in internal of UniPhier series SoC, this driver works independently but only UniPhier SoC can use it. Katsuhiro Suzuki (8): ASoC: spdif: Add S32_LE support for S/PDIF dummy codec drivers ASoC: uniphier: add DT bindings documentation for UniPhier EVEA ASoC: uniphier: add DT bindings documentation for UniPhier AIO ASoC: uniphier: add support for UniPhier EVEA codec ASoC: uniphier: add support for UniPhier AIO driver ASoC: uniphier: add support for UniPhier LD11/LD20 AIO driver MAINTAINERS: add entries for UniPhier ASoC sound drivers arm64: dts: uniphier: add sound node for UniPhier .../devicetree/bindings/sound/uniphier,aio.txt | 36 ++ .../devicetree/bindings/sound/uniphier,evea.txt | 26 + MAINTAINERS | 6 + .../boot/dts/socionext/uniphier-ld11-global.dts | 72 +++ arch/arm64/boot/dts/socionext/uniphier-ld11.dtsi | 31 + .../boot/dts/socionext/uniphier-ld20-global.dts | 72 +++ arch/arm64/boot/dts/socionext/uniphier-ld20.dtsi | 35 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/spdif_receiver.c | 5 +- sound/soc/codecs/spdif_transmitter.c | 5 +- sound/soc/uniphier/Kconfig | 34 + sound/soc/uniphier/Makefile | 8 + sound/soc/uniphier/aio-core.c | 368 +++++++++++ sound/soc/uniphier/aio-dma.c | 266 ++++++++ sound/soc/uniphier/aio-ld11.c | 343 ++++++++++ sound/soc/uniphier/aio-regctrl.c | 699 +++++++++++++++++++++ sound/soc/uniphier/aio-regctrl.h | 495 +++++++++++++++ sound/soc/uniphier/aio.h | 261 ++++++++ sound/soc/uniphier/evea.c | 567 +++++++++++++++++ 20 files changed, 3327 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt create mode 100644 Documentation/devicetree/bindings/sound/uniphier,evea.txt create mode 100644 sound/soc/uniphier/Kconfig create mode 100644 sound/soc/uniphier/Makefile create mode 100644 sound/soc/uniphier/aio-core.c create mode 100644 sound/soc/uniphier/aio-dma.c create mode 100644 sound/soc/uniphier/aio-ld11.c create mode 100644 sound/soc/uniphier/aio-regctrl.c create mode 100644 sound/soc/uniphier/aio-regctrl.h create mode 100644 sound/soc/uniphier/aio.h create mode 100644 sound/soc/uniphier/evea.c -- 2.15.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5659: Remove the routing path of the power widget "LDO2"
by Oder Chiou 12 Feb '18

12 Feb '18

This patch removes the routing path of the power widget "LDO2", and it should be depended on the hardware design. The power widget "LDO2" should add the routing path with the MICBIAS in the machine driver that is reasonable. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5659.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 07e7757..4ec1917 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2818,11 +2818,6 @@ static const struct snd_soc_dapm_route rt5659_dapm_routes[] = { { "I2S2", NULL, "I2S2 ASRC" }, { "I2S3", NULL, "I2S3 ASRC" }, - { "IN1P", NULL, "LDO2" }, - { "IN2P", NULL, "LDO2" }, - { "IN3P", NULL, "LDO2" }, - { "IN4P", NULL, "LDO2" }, - { "DMIC1", NULL, "DMIC L1" }, { "DMIC1", NULL, "DMIC R1" }, { "DMIC2", NULL, "DMIC L2" }, -- 2.7.4

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