February 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/11] ASoC: Consolidate enum controls and DAPM MUXs
by Lars-Peter Clausen 28 Feb '14

28 Feb '14

We currently have 5 nearly identical implementations of enum controls. Two for normal controls and three for DAPM controls. The only difference between them is how and if the selected enum item is mapped to a register value. For normal enums there is a direct mapping of the selected item to the register value, for value enums there is an additional look-up table that maps the selected item to a register value. DAPM also as a virtual enum which is not backed by any hardware register and the enum is solely used to influence DAPM routing decisions. The different implementations can easily be consolidating by making a few small changes. The series first introduces two helper functions to translate between the selected item and the register value. These functions will do the right thing depending on whether the enum has a direct mapping or is a value enum. These functions will be used to consolidate enums and value enums as well as MUXs and value MUXs. The next step is consolidate MUXs and virtual MUXs. This requires enums that are only used as virtual enums to be marked as such. This is done the same way as for virtual mixer controls by setting reg to SND_SOC_NOPM. All users of virtual MUXs are updated accordingly and then virtual MUXs and MUXs are updated to share the same implementation. The last patch in the series does cleanup of dapm_set_path_status() on top of all this. Once this has been merged I'll send a second patch-set replacing and removing the now deprecated ENUM and MUX macros. The series depends on topic/dapm, topic/enum, topic/sign, topic/adau1373, topic/max98090, topic/mc13783, fix/wm8994, topic/wm8995 - Lars Lars-Peter Clausen (11): ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart include/sound/soc-dapm.h | 38 +---- include/sound/soc.h | 41 +++-- sound/soc/codecs/adau1373.c | 4 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/soc-core.c | 101 ++---------- sound/soc/soc-dapm.c | 369 +++++++++----------------------------------- 9 files changed, 136 insertions(+), 430 deletions(-) -- 1.8.0

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[alsa-devel] [RFC PATCH] ALSA: hda - Fix CORB reset to follow specification
by David Henningsson 28 Feb '14

28 Feb '14

According to the HDA spec, we must write 1 to bit 15 on a CORBRP reset, read back 1, then write 0, then read back 0. This must be done while the DMA is not running. We accidentaly ended up writing back the 0 by using a writel instead of a writew to CORBWP. This caused occasional controller failure on Bay Trail hardware. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) Hi Takashi, A while ago I researched a spurious response on some Bay Trail hardware, so I proofread some of our CORB/RIRB code and found the below. When applied, the spurious responses disappeared but other problems might have appeared instead. I don't have the hardware here so I'm not 100% sure. So I'm hesitant to actually apply it, especially as the existing non-compliant way seems to work for all other chips. Still wanted to show the issue to you. Also the timeout (1 ms) is completely arbitrarily as there are no time limits written in the HDA specification. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8d9bf3..59156f1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -775,6 +775,8 @@ static int azx_alloc_cmd_io(struct azx *chip) static void azx_init_cmd_io(struct azx *chip) { + int timeout; + spin_lock_irq(&chip->reg_lock); /* CORB set up */ chip->corb.addr = chip->rb.addr; @@ -786,8 +788,28 @@ static void azx_init_cmd_io(struct azx *chip) azx_writeb(chip, CORBSIZE, 0x02); /* set the corb write pointer to 0 */ azx_writew(chip, CORBWP, 0); + /* reset the corb hw read pointer */ azx_writew(chip, CORBRP, ICH6_CORBRP_RST); + for (timeout = 1000; timeout > 0; timeout--) { + if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST) + break; + udelay(1); + } + if (timeout <= 0) + snd_printk(KERN_ERR SFX "%s: CORB reset timeout - CORBRP = %d\n", + pci_name(chip->pci), (int) azx_readw(chip, CORBRP)); + + azx_writew(chip, CORBRP, 0); + for (timeout = 1000; timeout > 0; timeout--) { + if (azx_readw(chip, CORBRP) == 0) + break; + udelay(1); + } + if (timeout <= 0) + snd_printk(KERN_ERR SFX "%s: CORB reset timeout, CORBRP = %d\n", + pci_name(chip->pci), azx_readw(chip, CORBRP)); + /* enable corb dma */ azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN); @@ -862,7 +884,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) chip->rirb.cmds[addr]++; chip->corb.buf[wp] = cpu_to_le32(val); - azx_writel(chip, CORBWP, wp); + azx_writew(chip, CORBWP, wp); spin_unlock_irq(&chip->reg_lock); -- 1.7.9.5

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Re: [alsa-devel] [PATCH 00/48] Rewrite kernel messages with standard printk helpers
by Takashi Iwai 28 Feb '14

28 Feb '14

At Thu, 27 Feb 2014 15:32:53 +0800, Raymond Yau wrote: > > 2014-02-26 23:53 GMT+08:00 Takashi Iwai <tiwai(a)suse.de>: > > > Hi, > > > > this is another long series of patches for trivial cleanups. > > This time, they replace the printk and snd_print*() calls in ALSA PCI > > and USB drivers with dev_err() and co. This will give nicer and more > > consistent outputs. > > > > The only drawback by these replacements is that alsa-info.sh won't be > > able to catch the dmesg lines because it doesn't contain any longer > > the word "ALSA", but only show the module name like "snd_intel8x0". > > > > I'm going to fix alsa-info.sh on git soon later, but if you guys think > > "oh no it's disaster!", let me know. We need to reconsider a better > > way to put the known markers (like explicitly putting "ALSA" at the > > beginning.) > > > > > seem missing au88x0 Yes, because some driver codes are horrible to work on. Patches are welcome. > can the driver build as modules in kernel 3.11 (ubuntu 13.10) ? The dev_err() and co exist since ages ago, so it shouldn't be a problem. These patches, however, need the previous change of snd_card_new() to make the device pointer available in time. No idea about Ubuntu. Takashi

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[alsa-devel] [PATCH] ASoC: simple-card: cpu_dai_name creates confusion when DT case
by Kuninori Morimoto 28 Feb '14

28 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Basically, soc_bind_dai_link() checks cpu_dai->dev->of_node and dai_link->cpu_of_node in DT case. But after that it will check cpu_dai->name and dai_link->cpu_dai_name too. On the other hand, snd_soc_dai :: name is created by fmt_single_name() or fmt_multiple_name(). There is no confusion if dai name is created by fmt_multiple_name(), since cpu_dai->name is same as dai_link->cpu_dai_name. but, if dai name is created by fmt_single_name(), CPU DAI never match. Thus, simple-card not set dai_link->cpu_dai_name if DT case to skip naming match on soc_bind_dai_link() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..4a0e06e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -242,7 +242,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; + if (!of_cpu) + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; cinfo->snd_link.platform_name = cinfo->platform; cinfo->snd_link.codec_name = cinfo->codec; cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] ASoC: ARM Cirrus Logic CLPS711X DAI driver
by Alexander Shiyan 28 Feb '14

28 Feb '14

This series adds sound support for the Cirrus Logic CLPS711X CPUs. Driver is designed (and tested) for using with generic sound card. Alexander Shiyan (2): ASoC: Add driver for CLPS711X DAI interface ASoC: cirrus: clps711x: Add bindings documentation for the CLPS711X DAI .../bindings/sound/cirrus,clps711x-dai.txt | 20 + sound/soc/cirrus/Kconfig | 8 + sound/soc/cirrus/Makefile | 5 + sound/soc/cirrus/clps711x-dai.c | 522 +++++++++++++++++++++ sound/soc/cirrus/clps711x-dai.h | 61 +++ sound/soc/cirrus/clps711x-fiq.S | 56 +++ 6 files changed, 672 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt create mode 100644 sound/soc/cirrus/clps711x-dai.c create mode 100644 sound/soc/cirrus/clps711x-dai.h create mode 100644 sound/soc/cirrus/clps711x-fiq.S -- 1.8.3.2

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[alsa-devel] [PATCH v2] ASOC: dapm: add params_select callback
by nikesh＠opensource.wolfsonmicro.com 28 Feb '14

28 Feb '14

From: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional callback to select one of the param from a list of params. Signed-off-by: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> --- Change since v1: Used an index to select from a list of params, instead of overwriting the param. Thanks, Nikesh include/sound/soc-dapm.h | 3 ++- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 20 ++++++++++++++++++-- 4 files changed, 25 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 68d92e3..12b0d9e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -416,7 +416,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink); + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..20110aa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -876,6 +876,9 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + /* optional params selection for dai links */ + int (*params_select)(const struct snd_soc_pcm_stream *params, + unsigned int *param_index, int event); unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50..4f03e88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1469,7 +1469,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1481,7 +1481,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13..2e2f633 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3379,10 +3379,12 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; const struct snd_soc_pcm_stream *config = w->params; + struct snd_soc_dai_link *dai_link = w->priv; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; - int ret; + int ret = 0; + unsigned int param_index; if (WARN_ON(!config) || WARN_ON(list_empty(&w->sources) || list_empty(&w->sinks))) @@ -3402,6 +3404,18 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + if (dai_link && dai_link->params_select) { + ret = dai_link->params_select(config, &param_index, event); + if (ret < 0) { + dev_err(w->dapm->dev, + "ASoC: params_select for dai link widget failed %d\n", + ret); + goto out; + } + if (param_index > 0) + config += param_index; + } + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3485,7 +3499,8 @@ out: int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link) { struct snd_soc_dapm_route routes[2]; struct snd_soc_dapm_widget template; @@ -3517,6 +3532,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, } w->params = params; + w->priv = (void *)dai_link; memset(&routes, 0, sizeof(routes)); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: fsl: Separation of the main audio options and boards
by Alexander Shiyan 28 Feb '14

28 Feb '14

This patch provide separation of the main audio options and boards. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 36 ++++++++++++++++++++++-------------- 1 file changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 07f8f14..ecc1092 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,5 @@ +menu "SoC Audio for Freescale CPUs" + config SND_SOC_FSL_SAI tristate select SND_SOC_GENERIC_DMAENGINE_PCM @@ -14,13 +16,27 @@ config SND_SOC_FSL_ESAI config SND_SOC_FSL_UTILS tristate -menuconfig SND_POWERPC_SOC +config SND_SOC_IMX_PCM_DMA + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + +config SND_SOC_IMX_AUDMUX + tristate + +config SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. +config SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC || COMPILE_TEST + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + if SND_POWERPC_SOC config SND_MPC52xx_DMA @@ -29,6 +45,8 @@ config SND_MPC52xx_DMA config SND_SOC_POWERPC_DMA tristate +comment "SoC Audio support for Freescale PPC boards:" + config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" # I2C is necessary for the CS4270 driver @@ -106,13 +124,6 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC || COMPILE_TEST - help - Say Y or M if you want to add support for codecs attached to - the i.MX CPUs. - if SND_IMX_SOC config SND_SOC_IMX_SSI @@ -122,12 +133,7 @@ config SND_SOC_IMX_PCM_FIQ tristate select FIQ -config SND_SOC_IMX_PCM_DMA - tristate - select SND_SOC_GENERIC_DMAENGINE_PCM - -config SND_SOC_IMX_AUDMUX - tristate +comment "SoC Audio support for Freescale i.MX boards:" config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" @@ -219,3 +225,5 @@ config SND_SOC_IMX_MC13783 select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC + +endmenu -- 1.8.3.2

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[alsa-devel] [PATCH 1/3] ALSA: hda - Remove dependency on bus->pci in hda_beep.c
by Takashi Iwai 27 Feb '14

27 Feb '14

The default parent device can be obtained directly via card object, so we don't need to rely on pci->dev.parent. Since there is no access to pci_dev, we can reduce the inclusion of linux/pci.h, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0589b39cda6e..d8c437a94559 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -20,7 +20,6 @@ */ #include <linux/input.h> -#include <linux/pci.h> #include <linux/slab.h> #include <linux/workqueue.h> #include <linux/export.h> @@ -167,7 +166,7 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->evbit[0] = BIT_MASK(EV_SND); input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); input_dev->event = snd_hda_beep_event; - input_dev->dev.parent = &codec->bus->pci->dev; + input_dev->dev.parent = codec->bus->card->dev; input_set_drvdata(input_dev, beep); err = input_register_device(input_dev); -- 1.8.5.2

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[alsa-devel] NUC D34010WYK HDMI Audio does not work...
by Robert Krakora 27 Feb '14

27 Feb '14

Hi All, We have a NUC D34010WYK on which I have installed several distros/kernels in a, so far, vain attempt to get HDMI audio to function via ALSA. I upload and alsainfo snap shot here: http://www.alsa-project.org/db/?f=d5f75cecee1195591d85be99c62db09bcd2bbdaf The latest distro I tried was an Ubuntu 14.04 daily from Feburary 18th. I also found this link where someone had tried unsuccessfully to get HDMI audio from this same NUC model unsuccessfully with Ubuntu 13.10: http://ubuntuforums.org/showthread.php?t=2202727 Any and all help will be greatly appreciated! :-) Best Regards, Rob -- Rob Krakora, Senior Software Engineer MessageNet Systems 101 E Carmel Dr, Suite 105 Carmel, IN 46032 MessageNetSystems.com<http://www.messagenetcommunicationsystems.com/?utm_source=email+signature&u…> Rob.Krakora(a)MessageNetSystems.com <rob.krakora(a)messagenetsystems.com> P: 317.566.1677, 212 F: 317.663.0808 For the latest news, information, and blogs, please be sure to visit, follow, and like us... <http://www.messagenetcommunicationsystems.com/get-the-message-out-blog/?utm…> <http://www.youtube.com/user/MessageNetConnection/feed> <http://www.linkedin.com/company/messagenet-systems> <http://twitter.com/MessageNet> <http://www.facebook.com/MessageNetsystems>

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[alsa-devel] [PATCH] ALSA: hda/sigmatel - Allow auto-switching for dock line-in of HP laptops
by Takashi Iwai 27 Feb '14

27 Feb '14

Many HP laptops with STAC codecs have a docking station port and BIOS sets the pins for the input on the dock as a line in. Because the generic parser doesn't handle a line in pin as auto-switchable, this resulted in the manual capture source selection on these laptops. However, from the usability POV, the automatic switching is easier. This patch adds the line_in_auto_switch hint in the fixup function for these laptops. Even if no dock port is present, this should be harmless as the generic parser allows the auto-switching only in a limited situation (all three pins are located in different positions). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3784808758e..75515b494034 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2089,6 +2089,9 @@ static void stac92hd83xxx_fixup_hp(struct hda_codec *codec, codec_dbg(codec, "mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); + + /* allow auto-switching of dock line-in */ + spec->gen.line_in_auto_switch = true; } static void stac92hd83xxx_fixup_hp_zephyr(struct hda_codec *codec, -- 1.9.0

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