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

[alsa-devel] [PATCH 1/5] ASoC: wm8994: Add digital loopback paths
by Mark Brown 30 May '13

30 May '13

There is loopback control within the audio interfaces, provide control of this as there are some obscure scenarios where this could be used in production. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 561c2f9..2081fa1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1499,6 +1499,24 @@ static const char *aif1dac_text[] = { "AIF1DACDAT", "AIF3DACDAT", }; +static const char *loopback_text[] = { + "None", "ADCDAT", +}; + +static const struct soc_enum aif1_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif1_loopback = + SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); + +static const struct soc_enum aif2_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif2_loopback = + SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); + static const struct soc_enum aif1dac_enum = SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); @@ -1745,6 +1763,9 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_MUX("AIF1 Loopback", SND_SOC_NOPM, 0, 0, &aif1_loopback), +SND_SOC_DAPM_MUX("AIF2 Loopback", SND_SOC_NOPM, 0, 0, &aif2_loopback), + SND_SOC_DAPM_POST("Debug log", post_ev), }; @@ -1876,9 +1897,9 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF1DAC2L", NULL, "AIF1DAC Mux" }, { "AIF1DAC2R", NULL, "AIF1DAC Mux" }, - { "AIF1DAC Mux", "AIF1DACDAT", "AIF1DACDAT" }, + { "AIF1DAC Mux", "AIF1DACDAT", "AIF1 Loopback" }, { "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, - { "AIF2DAC Mux", "AIF2DACDAT", "AIF2DACDAT" }, + { "AIF2DAC Mux", "AIF2DACDAT", "AIF2 Loopback" }, { "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, @@ -1929,6 +1950,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + /* Loopback */ + { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, + { "AIF1 Loopback", "None", "AIF1DACDAT" }, + { "AIF2 Loopback", "ADCDAT", "AIF2ADCDAT" }, + { "AIF2 Loopback", "None", "AIF2DACDAT" }, + /* Sidetone */ { "Left Sidetone", "ADC/DMIC1", "ADCL Mux" }, { "Left Sidetone", "DMIC2", "DMIC2L" }, -- 1.7.10.4

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[alsa-devel] [PATCH] hda_analyzer: don't fail to display audio selector if caps variables aren't available
by Anisse Astier 30 May '13

30 May '13

In particular, when an audio selector node has Amp-In caps: N/A, it allows to still interact with the selector, and not have TypeError traceback throwns because of NoneType variables. Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi Jared, I'm not sure if this is the proper way to fix this, so please consider this a formal report for the error thrown while parsing this node: Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1b 0x1d 0x0b 0x12 Regards, Anisse -- hda-analyzer/hda_guilib.py | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/hda-analyzer/hda_guilib.py b/hda-analyzer/hda_guilib.py index 93c85e1..62eddab 100644 --- a/hda-analyzer/hda_guilib.py +++ b/hda-analyzer/hda_guilib.py @@ -285,9 +285,9 @@ class NodeGui(gtk.ScrolledWindow): frame.set_border_width(4) vbox = gtk.VBox(False, 0) if caps: - str = 'Offset: %d\n' % caps.ofs - str += 'Number of steps: %d\n' % caps.nsteps - str += 'Step size: %d\n' % caps.stepsize + str = 'Offset: %s\n' % (caps.ofs if caps.ofs else "N/A") + str += 'Number of steps: %s\n' % (caps.nsteps if caps.nsteps else "N/A") + str += 'Step size: %s\n' % (caps.stepsize if caps.stepsize else "N/A") str += 'Mute: %s\n' % (caps.mute and "True" or "False") vbox.pack_start(self.__new_text_view(text=str), True, True, 0) idx = 0 -- 1.8.3.rc1

2 2

[alsa-devel] [PATCH 01/10] ASoC: blackfin: Remove unused bf5xx-{i2s, tdm, ac97}-pcm.h
by Lars-Peter Clausen 30 May '13

30 May '13

The structs defined in these files are completely unused, so remove both the structs and the files. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-ac97-pcm.h | 26 -------------------------- sound/soc/blackfin/bf5xx-ad1836.c | 1 - sound/soc/blackfin/bf5xx-ad193x.c | 1 - sound/soc/blackfin/bf5xx-ad1980.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.h | 26 -------------------------- sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.h | 18 ------------------ 11 files changed, 78 deletions(-) delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 7e2f360..53f8408 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.h b/sound/soc/blackfin/bf5xx-ac97-pcm.h deleted file mode 100644 index d324d58..0000000 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-ac97-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_AC97_PCM_H -#define _BF5XX_AC97_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d23f4b0..1528176 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -30,7 +30,6 @@ #include "../codecs/ad1836.h" -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad1836; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 0e55e9f..ce773b3 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -39,7 +39,6 @@ #include "../codecs/ad193x.h" -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad193x; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index b30f88b..3450e8f 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -48,7 +48,6 @@ #include "../codecs/ad1980.h" -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" static struct snd_soc_card bf5xx_board; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 61cc91d..786bbdd 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -45,7 +45,6 @@ #include "../codecs/ad73311.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" #if CONFIG_SND_BF5XX_SPORT_NUM == 0 #define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 262c1de..8726c3a 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-i2s-pcm.h" #include "bf5xx-sport.h" static void bf5xx_dma_irq(void *data) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h deleted file mode 100644 index 0c2c5a6..0000000 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-i2s-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_I2S_PCM_H -#define _BF5XX_I2S_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 7dbeef1..9c19ccc 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -40,7 +40,6 @@ #include <linux/gpio.h> #include "../codecs/ssm2602.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" static struct snd_soc_card bf5xx_ssm2602; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 19e855d..a6b5457 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h deleted file mode 100644 index 7f8cc01..0000000 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2009 Analog Device Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_TDM_PCM_H -#define _BF5XX_TDM_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -#endif -- 1.8.0

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[alsa-devel] [PATCH 0/4 V6] Power-well API implementation for Haswell
by Wang Xingchao 30 May '13

30 May '13

Hi all, This is V6 and here're some changes notes: change from V5-->V6: - Remove duplication code in new introduced probe work - move duplication code in azx_probe_continue - remove unused #ifdef - replace request_module with symbol_request - replace spin_lock_irq with spin_lock_irqsave in gfx side - other typo fixes (review by Takashi) change from V4-->V5: - fix reference count bug - new patch on general runtime pm support for audio pci device - new patch to avoid request_module() deadlock change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Takashi Iwai (1): ALSA: hda - Move azx_first_init() into azx_probe_continue() Wang Xingchao (3): ALSA: hda - Fix runtime PM check ALSA: hda - Add power-welll support for haswell HDA i915/drm: Add private api for power well usage drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 +++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 92 +++++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 +++++++++++++++ sound/pci/hda/Kconfig | 10 +++++ sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_i915.c | 75 +++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 87 ++++++++++++++++++++++++++++------- 10 files changed, 335 insertions(+), 24 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 6

[alsa-devel] [PATCH] ASoC: OMAP: Remove obsolete Makefile line
by Paul Bolle 30 May '13

30 May '13

Support for omap2evm was removed in v3.0. But only one of its two lines in this Makefile was removed. Remove the second line too. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Eyeball tested only. sound/soc/omap/Makefile | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 2b22594..a725905 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -26,7 +26,6 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o -- 1.7.11.7

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[alsa-devel] [PATCH] ASoC: wm0010: Set IRQ as a wake source
by Charles Keepax 30 May '13

30 May '13

The DSPs IRQ should be a wake source as several of the possible algorithms may run whilst the AP is asleepi and require to wake the AP to push or pull more data, such as compressed playback. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 370af0c..b6df319 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -972,6 +972,13 @@ static int wm0010_spi_probe(struct spi_device *spi) } wm0010->irq = irq; + ret = irq_set_irq_wake(irq, 1); + if (ret) { + dev_err(wm0010->dev, "Failed to set IRQ %d as wake source: %d\n", + irq, ret); + return ret; + } + if (spi->max_speed_hz) wm0010->board_max_spi_speed = spi->max_speed_hz; else -- 1.7.2.5

2 1

[alsa-devel] [PATCH v6 00/10] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 30 May '13

30 May '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Regards, Markus Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 441 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 12 +- sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 241 +++++++++-- 13 files changed, 643 insertions(+), 125 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

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[alsa-devel] [PATCH 0/2] ASoC: cs42l52: HP gain enum and PCM Mixer dB scale fixes.
by Nicolas Schichan 29 May '13

29 May '13

Hi, The two patches in the serie contains the following fixes: - Use the correct shift value in hp_gain_enum, effectively allowing the higher gain values to be effective and the correct gain value to be reported to userland wrt what is written to the register. - Fix the PCM Mixer dB scale so that userland can report the correct dB gain values. Note that in its current form, the PCM Mixer control will only allow the range [-0.5 dB (0x7f), +12 dB (0x18)] to be programmed to the hardware. I don't know if it is intentional or not. Regards, Nicolas Schichan (2): ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet. ASoC: cs42l52: fix hp_gain_enum shift value. sound/soc/codecs/cs42l52.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race
by Mark Brown 29 May '13

29 May '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1ddb067..4aaa8a0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3886,7 +3886,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) ret); } else if (!(ret & WM1811_JACKDET_LVL)) { dev_dbg(codec->dev, "Ignoring removed jack\n"); - return IRQ_HANDLED; + goto out; } } else if (!(reg & WM8958_MICD_STS)) { snd_soc_jack_report(wm8994->micdet[0].jack, 0, -- 1.7.10.4

1 0

[alsa-devel] UCM {Supported,Conflicting}Device confusion
by Juho Hämäläinen 29 May '13

29 May '13

Hello, Firstly as a sidenote, Supported/Conflicting is checked only one-way, if say devices A and B are enabled and say B has ConflictingDevice "C", and device C doesn't list anything, enabling of C is possible (and after that disabling C is not possible before B is disabled first). Now, looking at the way SupportedDevice and ConflictingDevice are implemented in ucm and I came to following conclusion: If no SupportedDevice or ConflictingDevice: All devices and combinations are compatible, so implicit "SupportedDevice ALL". If SupportedDevice is listed: Device can be enabled as long as at least one enabled device is in SupportedDevice list. If ConflictingDevice is listed: Device can be enabled if none of active devices are in ConflictingDevice list (noting aforementioned limit in checking this) This is at least how I figured all combinations of multiple devices can be defined with only zero or one either SupportedDevice or ConflictingDevice entry per device. Did I understand this right? -- juho

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