July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for 3.1
by Mark Brown 09 Jul '11

09 Jul '11

The following changes since commit f187700c2d769bd3160b52f99cc4e747d5c5014b: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-05 08:20:19 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Axel Lin (1): ASoC: pxa2xx-pcm: remove unused variable 'dai' Kuninori Morimoto (1): ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 Liam Girdwood (4): ASoC: core - Add API call to register platform kcontrols. ASoC: core - Add platform widget IO ASoC: core - Add convenience register for platform kcontrol and DAPM ASoC: core - Add platform IO tracing Mark Brown (3): ASoC: Manage WM8731 ACTIVE bit as a supply widget ASoC: Manage WM8731 ACTIVE bit as a supply widget Merge branch 'for-3.0' into for-3.1 Stephen Warren (1): ASoC: Tegra: Implement SPDIF CPU DAI include/sound/soc-dapm.h | 2 + include/sound/soc.h | 12 + include/trace/events/asoc.h | 45 ++++ sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/wm8731.c | 29 +--- sound/soc/pxa/pxa2xx-pcm.c | 1 - sound/soc/soc-core.c | 47 ++++ sound/soc/soc-dapm.c | 13 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 12 files changed, 976 insertions(+), 30 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h

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[alsa-devel] Jack event API - decision needed
by David Henningsson 09 Jul '11

09 Jul '11

Hi, I'm still new in the community in that sense that I'm not sure how decisions are made. But I could use the outcome of such a decision. Background: I'm trying to pull together the missing pieces of jack detection on both kernel / plumbing / application layers, so that when you plug something in (headset, microphone etc), userspace is notified and can take appropriate actions (e g routing decisions). As part of that I wrote a udev patch a few days ago, which nobody commented on in alsa-devel [1], but was somewhat disliked by at least Kay Sievers who maintains udev [2], who preferred we would rewrite our input layer to do something else within ALSA. So before I proceed further I'd like to know if 1) We're continuing the path with /dev/input devices 2a) We'll rewrite these devices to be read-only ALSA mixer controls 2b) We'll rewrite these devices to be something within ALSA, but not exactly mixer controls. For options 2a) and 2b) I guess the existing /dev/input thing should be deprecated and/or removed. So part of decision should maybe be based on information about how widespread the usage of these devices are currently...? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-June/040916.html [2] e g http://www.spinics.net/lists/hotplug/msg04949.html

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Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by Mark Brown 09 Jul '11

09 Jul '11

On Fri, Jul 08, 2011 at 12:36:34PM +0200, Takashi Iwai wrote: > just wondering how about holding a sound-BoF in LinuxCon Europe in > October? > I know ELCE will be held at the very same time, so it may need the > adjustment of schedule. But in general what do you think? I have no current plans to attend these conferences, I'm planning to go to LPC instead. Not sure if I'd be able to arrange it unless I get a KS invite. > An alternative is a workshop in KS2011 day 1, but I think KS suits > better for more broader topics, e.g. the discussions on the media- > notification support or the jack implementation like a thread in the > last month. Yes, I don't think it's good for covering at KS as it's far too detailed. If we're going to do something probably LinuxCon/ELC Europe make most sense.

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 09 Jul '11

09 Jul '11

The following changes since commit 873bd4cb4fbba6a3e99f750e17ef2ba6ef96e9d3: ASoC: Don't set invalid name string to snd_card->driver field (2011-07-05 14:39:27 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 The WM8731 patch fixes resume when we suspend with an active playback or record. Kuninori Morimoto (1): ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 Mark Brown (1): ASoC: Manage WM8731 ACTIVE bit as a supply widget sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 2 files changed, 4 insertions(+), 27 deletions(-)

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Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Takashi Iwai 09 Jul '11

09 Jul '11

At Thu, 7 Jul 2011 21:35:17 +0900, Toshiharu Okada wrote: > > Date: Thu, 07 Jul 2011 11:00:47 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > >> > >> Therefore, I am not considering about ASoC. > >> Do I need to fit this driver to an ASoC framework for an upstream? > > > >Yes, I guess this would be a better choice in your case. > > Thank you for your comment. > >From now on, We study about ASoC. > If there are a document, a website, a driver, etc. which are referred, > please tell us. Check Documentation/sound/alsa/soc/* files. > Why is choosing ASoS better? > Is it better to correspond to ASoC, when Audio Codec is unfixed? It's not clear what you mean exactly as "unfixed", but in general, the decision rather depends on the usage of the device. If it's designed for a use as a desktop PC component (such as a PCI card or onboard-audio on PC), it'd make sense to implement as a stand-alone, self-contained driver. OTOH, if it's targeted mainly for embedded area, ASoC is the right answer. It's more modularized, and can be more flexibly configured in the end. Takashi

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 08 Jul '11

08 Jul '11

On Thu, Jul 7, 2011 at 3:08 PM, Oliver Neukum <oliver(a)neukum.org> wrote: > Am Donnerstag, 7. Juli 2011, 14:38:51 schrieb Daniel Mack: >> On Thu, Jul 7, 2011 at 2:33 PM, Oliver Neukum <oliver(a)neukum.org> wrote: >> >> > PS: Do you still see this if you enable 64bit DMA for EHCI? >> >> The problem is that I personally don't see that issue at all. I even >> installed 4GB of RAM to my development machine last year to be able to >> reproduce this, but I can't, even when the memory allocator is under >> heavy load. The only people who see this effect are Pedro Ribeiro and >> William Light (both in Cc:), and both have been very helpful in trying >> patches and reporting back in detail. Which instructions could we >> probably give to these people to finally hunt this issue down? > > Do they have an IOMMU or are they using bounce buffers? > If the latter, you might be able to reproduce it if you disable your > IOMMU. Ok, Pedro and William - can I ask you to dump the following information to some paste bin: - your .config - the dmesg output of a booted kernel that shows the effect - the output of "sudo lspci -v" Also, the attached patch might give you at least one warning to your kernel logs once the audio stream starts and outputs noise. I'm curious whether it really does. Thanks, Daniel

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[alsa-devel] [RFC PATCH] ARM: imx27pdk (3ds): enable m13783 codec over SSI
by Jürgen Lambrecht 08 Jul '11

08 Jul '11

- add SSI in Kconfig - enable audio in MFD mc13783 - configure audmux-v1 - configure SSI4 Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- arch/arm/mach-imx/Kconfig | 1 + arch/arm/mach-imx/mach-mx27_3ds.c | 40 ++++++++++++++++++++++++++++++++++++- 2 files changed, 40 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-imx/Kconfig b/arch/arm/mach-imx/Kconfig index 0a3512b..2fe5e70 100644 --- a/arch/arm/mach-imx/Kconfig +++ b/arch/arm/mach-imx/Kconfig @@ -286,6 +286,7 @@ config MACH_MX27_3DS select IMX_HAVE_PLATFORM_MXC_MMC select IMX_HAVE_PLATFORM_MXC_NAND select IMX_HAVE_PLATFORM_SPI_IMX + select IMX_HAVE_PLATFORM_IMX_SSI select MXC_DEBUG_BOARD select MXC_ULPI if USB_ULPI help diff --git a/arch/arm/mach-imx/mach-mx27_3ds.c b/arch/arm/mach-imx/mach-mx27_3ds.c index 2787cf3..c0f7dcc 100644 --- a/arch/arm/mach-imx/mach-mx27_3ds.c +++ b/arch/arm/mach-imx/mach-mx27_3ds.c @@ -41,6 +41,7 @@ #include <mach/irqs.h> #include <mach/3ds_debugboard.h> #include <mach/mxc_nand.h> +#include <mach/audmux.h> #include "devices-imx27.h" @@ -101,6 +102,11 @@ static const int mx27pdk_pins[] __initconst = { PE2_PF_USBOTG_DIR, PE24_PF_USBOTG_CLK, PE25_PF_USBOTG_DATA7, + /* SSI4 for audio to MC13783 */ + PC16_PF_SSI4_FS, + PC17_PF_SSI4_RXD, + PC18_PF_SSI4_TXD, + PC19_PF_SSI4_CLK, /* CSPI1 */ PD31_PF_CSPI1_MOSI, PD30_PF_CSPI1_MISO, @@ -271,6 +277,13 @@ static struct regulator_init_data vgen_init = { .consumer_supplies = vgen_consumers, }; +static struct regulator_init_data vaudio_init = { + .constraints = { + .boot_on = 1, + .always_on = 1, + }, +}; + static struct mc13xxx_regulator_init_data mx27_3ds_regulators[] = { { .id = MC13783_REG_VMMC1, @@ -284,6 +297,9 @@ static struct mc13xxx_regulator_init_data mx27_3ds_regulators[] = { }, { .id = MC13783_REG_GPO3, /* Turn on 3.3V */ .init_data = &gpo_init, + }, { + .id = MC13783_REG_VAUDIO, /* Power codec */ + .init_data = &vaudio_init, }, }; @@ -295,7 +311,7 @@ static struct mc13xxx_platform_data mc13783_pdata = { }, .flags = MC13783_USE_REGULATOR | MC13783_USE_TOUCHSCREEN | - MC13783_USE_RTC, + MC13783_USE_RTC | MC13783_USE_CODEC, }; /* SPI */ @@ -385,6 +401,11 @@ static const struct imxi2c_platform_data mx27_3ds_i2c0_data __initconst = { .bitrate = 100000, }; +/* 4-wire is synchronous; SSI-slave and MC13783-master; DMA instead of FIQ */ +static const struct imx_ssi_platform_data mx27pdk_ssi4_pdata __initconst = { + .flags = IMX_SSI_SYN | IMX_SSI_DMA | IMX_SSI_USE_I2S_SLAVE, +}; + static void __init mx27pdk_init(void) { imx27_soc_init(); @@ -420,6 +441,23 @@ static void __init mx27pdk_init(void) pr_warn("Init of the debugboard failed, all devices on the debugboard are unusable.\n"); imx27_add_imx_i2c(0, &mx27_3ds_i2c0_data); imx27_add_imx_fb(&mx27_3ds_fb_data); + + /* SSI unit master I2S codec connected to SSI_PINS_4*/ + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V1_PCR_TFSDIR | /* Transmit Frame Sync is output */ + MXC_AUDMUX_V1_PCR_TCLKDIR | /* Transmit Clock is output */ + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | /*Transmit Frame Sync and Clock select source port to be HPCR3 or external SSI4*/ + MXC_AUDMUX_V1_PCR_RFSDIR | /* Receive Frame Sync is output */ + MXC_AUDMUX_V1_PCR_RCLKDIR | /* Receive Clock is output */ + MXC_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | /*Receive Frame Sync and Clock select source port to be HPCR3 or external SSI4*/ + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) /*Receive Data select source port to be HPCR3 or external SSI4*/ + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) /*Receive Data select source port to be HPCR1 or internal SSI1*/ + ); + imx27_add_imx_ssi(0, &mx27pdk_ssi4_pdata); } static void __init mx27pdk_timer_init(void) -- 1.7.1

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Re: [alsa-devel] [PATCH] ASoC: dapm - Add API calls to query valid DAPM paths.
by Mark Brown 07 Jul '11

07 Jul '11

Funny formatting due to phone, sorry... Indeed, but I'd have expected that anything that needed to do that would be able to (and want to) ask by name rather than type? Liam Girdwood <lrg(a)ti.com> wrote: >On 07/07/11 17:38, Mark Brown wrote: >> On Thu, Jul 07, 2011 at 05:18:47PM +0100, Liam Girdwood wrote: >> >>> The reason for using a separate walk here was because this code is >>> taken from my old auto-router and it was to avoid the extra logic >>> introduced by the auto-router. The auto-router did need to check for >>> loops and always select the shortest path when > 1 path was viable >>> (interesting as WM9713 has both loops and multiple paths). Let me see >>> how cleanly I can merge it into the current walk. >> >> Alternatively could we replace the current walk? It's O(n^2) or so in >> the number of widgets so it's not the greatest treasure ever. So long >> as we only have one walk and it works I'm happy. >> >>>> It does also occur to me that perhaps what we want to do here is allow >>>> widgets to find out about paths when they're being activated anyway? >> >>> For dynamic PCM, we need to also work out which DAIs are active based >>> on the graph. Hence for playback we can supply the DAC/AIF and find >>> out all the connected output AIF/Speakers/Pins/etc. This is required >>> at the start of open() in order that we can call the other PCM ops for >>> each DAI, codec and platform. >> >> Right, that's not really what I'm saying though - I'm saying that >> widgets in general could be interested in this information. For >> example, some charge pumps can save additional power with some input >> signal paths (which is currently handled but this would make that code >> more general). > >Ok, but I'm not stopping this sort of functionality from being added >later. Right now this is not a priority (for me anyway). > >> >>>>> + * snd_soc_dapm_get_connected_widgets_type - query audio path and it's widgets. >>>>> + * @dapm: the dapm context. >>>>> + * @stream_name: stream name. >>>>> + * @list: list of active widgets for this stream. >>>>> + * @stream: stream direction. >>>>> + * @type: Initial widget type. >>>>> + * >>>>> + * Queries DAPM graph as to whether an valid audio stream path exists for >>>>> + * the DAPM stream and initial widget type specified. This takes into account >>>>> + * current mixer and mux kcontrol settings. Creates list of valid widgets. >>>> >>>> Why would someone want to query by type? That seems surprising. Name >>>> and/or direction yes but type seems hard to find a use for. >> >>> We query by type so that we can qualify the root widget for a shared >>> common stream name. i.e. Dynamic DSP needs to find the AIF rather than >>> a DAC for stream X. >> >> It does? Hrm. Perhaps it would make life simpler if we just mandated >> the use of explicit AIF widgets? > >But we may also want to know what was after a DAC. I'm thinking back >from the conference discussions where we could use this information to >selectively enable/disable kcontrols. However, this is not going to be >used at the moment either so I'll stick by the AIF widgets. > >Liam > >

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[alsa-devel] [PATCH] ASoC: dapm - Add API calls to query valid DAPM paths.
by Liam Girdwood 07 Jul '11

07 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Add DAPM API calls to determine whether a DAPM audio path is valid between source and sink widgets and return a lits of active widgets in that path. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the DSP core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 14 ++ sound/soc/soc-dapm.c | 371 ++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 376 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..5a7dae6 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,13 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets_type(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream, enum snd_soc_dapm_type type); +int snd_soc_dapm_get_connected_widgets_name(struct snd_soc_dapm_context *dapm, + const char *name, struct snd_soc_dapm_widget_list **list, int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ @@ -432,6 +440,8 @@ struct snd_soc_dapm_path { u32 connect:1; /* source and sink widgets are connected */ u32 walked:1; /* path has been walked */ u32 weak:1; /* path ignored for power management */ + u32 length:6; /* path length - used by route checker */ + int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -456,6 +466,8 @@ struct snd_soc_dapm_widget { unsigned char shift; /* bits to shift */ unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ + unsigned int path_idx; + unsigned int hops; unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ unsigned int off_val; /* off state value */ @@ -518,6 +530,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int num_valid_paths; /* valid paths for route checker */ + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a78482..800af10 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -48,6 +48,8 @@ #include <trace/events/asoc.h> +#define DAPM_MAX_HOPS 16 + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -169,6 +171,366 @@ static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, return change; } +/* get snd_card from DAPM context */ +static inline struct snd_card *dapm_get_snd_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card->snd_card; + else if (dapm->platform) + return dapm->platform->card->snd_card; + + dev_err(dapm->dev, "no snd_card for dapm context\n"); + return NULL; +} + +/* get soc_card from DAPM context */ +static inline struct snd_soc_card *dapm_get_soc_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card; + else if (dapm->platform) + return dapm->platform->card; + + dev_err(dapm->dev, "no soc_card for dapm context\n"); + return NULL; +} + +/* reset 'walked' bit for each dapm path */ +static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_dapm_path *p; + + list_for_each_entry(p, &dapm->card->paths, list) + p->walked = 0; +} + +/* clear all 'walked' path data for each widget */ +static void dapm_clear_paths(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + struct list_head *l; + + list_for_each(l, &dapm->card->paths) { + p = list_entry(l, struct snd_soc_dapm_path, list); + p->length = 0; + } + list_for_each(l, &dapm->card->widgets) { + w = list_entry(l, struct snd_soc_dapm_widget, list); + w->hops = 0; + } + dapm_clear_walk(dapm); +} + +/* add widget to list if it's not already in the list */ +static int dapm_add_unique_widget(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + /* is the list empty ? */ + if (*list == NULL) { + + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + sizeof(struct snd_soc_dapm_widget *); + *list = kzalloc(wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + } else { + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + } + wlist = *list; + + /* insert the widget */ + dev_dbg(dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + +/* is widget an output endpoint */ +static int is_output_widget_ep(struct snd_soc_dapm_widget *widget) +{ + switch (widget->id) { + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + return 1; + case snd_soc_dapm_output: + if (widget->connected && !widget->ext) + return 1; + else + return 0; + case snd_soc_dapm_hp: + case snd_soc_dapm_spk: + case snd_soc_dapm_line: + return !list_empty(&widget->sources); + default: + return 0; + } +} + +/* is widget an input endpoint */ +static int is_input_widget_ep(struct snd_soc_dapm_widget *widget) +{ + switch (widget->id) { + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + return 1; + case snd_soc_dapm_input: + if (widget->connected && !widget->ext) + return 1; + else + return 0; + case snd_soc_dapm_mic: + return !list_empty(&widget->sources); + default: + return 0; + } +} + +/* + * Find all the playback paths between the root source widget and + * sink widgets. Store all path widgets in the list. + */ +static int dapm_find_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list, int hops) +{ + struct list_head *lp; + struct snd_soc_dapm_path *path; + int dist = 0; + + if (hops > DAPM_MAX_HOPS) + return 0; + + /* are we at an endpoint widget */ + if (is_output_widget_ep(root)) { + dev_dbg(dapm->dev," ! %d: valid playback route found\n", hops); + dapm->num_valid_paths++; + return 1; + } + + /* have we visited this widget before */ + if (root->hops && root->hops <= hops) + return 0; + root->hops = hops; + + /* check all the output paths on this source widget by walking + * from source to sink */ + list_for_each(lp, &root->sinks) { + path = list_entry(lp, struct snd_soc_dapm_path, list_source); + + dev_dbg(dapm->dev," %c %d: %s -> %s -> %s\n", + path->connect ? '*' : ' ', hops, + root->name, path->name, path->sink->name); + + /* have we been down this path before ? */ + if (path->length && path->length <= hops) + continue; + + /* check down the next path if connected */ + if (path->sink && path->connect && + dapm_find_playback_paths(dapm, path->sink, list, hops + 1)) { + path->length = hops; + + /* add widget to list */ + dapm_add_unique_widget(dapm, list, path->sink); + + if (!dist || dist > path->length) + dist = path->length; + } + } + + return dist; +} + +/* + * Find all the capture paths between the root sink widget and + * source widgets. Store all path widgets in the list. + */ +static int dapm_find_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list, int hops) +{ + struct list_head *lp; + struct snd_soc_dapm_path *path; + int dist = 0; + + if (hops > DAPM_MAX_HOPS) + return 0; + + /* are we at an endpoint widget */ + if (is_input_widget_ep(root)) { + dev_dbg(dapm->dev," ! %d: valid capture route found\n", hops); + dapm->num_valid_paths++; + return 1; + } + + /* have we visited this widget before */ + if (root->hops && root->hops <= hops) + return 0; + root->hops = hops; + + /* check all the output paths on this source widget by walking from + * sink to source */ + list_for_each(lp, &root->sources) { + path = list_entry(lp, struct snd_soc_dapm_path, list_sink); + + dev_dbg(dapm->dev," %c %d: %s <- %s <- %s\n", + path->connect ? '*' : ' ', hops, + root->name, path->name, path->source->name); + + /* have we been here before ? */ + if (path->length && path->length <= hops) + continue; + + /* check down the next path if connected */ + if (path->source && path->connect && + dapm_find_capture_paths(dapm, path->source, list, hops + 1)) { + path->length = hops; + + /* add widget to list */ + dapm_add_unique_widget(dapm, list, path->source); + + if (!dist || dist > path->length) + dist = path->length; + } + } + + return dist; +} + +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + dapm_find_playback_paths(dapm, root, list, 0); + return dapm->num_valid_paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + dev_dbg(dapm->dev, "Capture: checking paths to %s\n", root->name); + dapm_find_capture_paths(dapm, root, list, 0); + return dapm->num_valid_paths; +} + +/** + * snd_soc_dapm_get_connected_widgets_type - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * @type: Initial widget type. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the DAPM stream and initial widget type specified. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets_type( + struct snd_soc_dapm_context *dapm, const char *stream_name, + struct snd_soc_dapm_widget_list **list, int stream, + enum snd_soc_dapm_type type) +{ + struct snd_soc_dapm_widget *w; + int paths; + + /* get stream root widget AIF, DAC or ADC from stream string + * and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (!w->sname) + continue; + + if (w->id != type) + continue; + + if (strstr(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget not found\n"); + return 0; + +found: + dapm->num_valid_paths = 0; + *list = NULL; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_paths(dapm); + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets_name - query audio path and it's widgets. + * @dapm: the dapm context. + * @name: initial widget name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial widget specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets_name(struct snd_soc_dapm_context *dapm, + const char *name, struct snd_soc_dapm_widget_list **list, int stream) +{ + struct snd_soc_dapm_widget *w; + int paths; + + /* get stream root widget AIF, DAC or ADC from stream string + * and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (strstr(w->name, name)) + goto found; + } + dev_err(dapm->dev, "root widget %s not found\n", name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_paths(dapm); + return paths; +} + /** * snd_soc_dapm_set_bias_level - set the bias level for the system * @dapm: DAPM context @@ -576,15 +938,6 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } -/* reset 'walked' bit for each dapm path */ -static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) -{ - struct snd_soc_dapm_path *p; - - list_for_each_entry(p, &dapm->card->paths, list) - p->walked = 0; -} - /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
by Kuninori Morimoto 07 Jul '11

07 Jul '11

mask didn't cover update-data Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch based on asoc/for-3.1 sound/soc/codecs/ak4642.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4be0570..65f4604 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, PW_MGMT2, MS, data); + snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); /* format type */ -- 1.7.4.1

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Re: [alsa-devel] [PATCH] ASoC: pxa2xx-pcm: remove unused variable 'dai'
by Mark Brown 07 Jul '11

07 Jul '11

On Wed, Jul 06, 2011 at 09:20:42PM +0800, Axel Lin wrote: > Remove unused variable 'dai' to eliminate below warning. Applied, thanks.

1 0

[alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 07 Jul '11

07 Jul '11

I have to revisit an issue we've discussed in length around a year ago and which still remains unsolved. I've been getting feedback from more users of my driver snd-usb-caiaq who report the issue found in bug #15580 in the kernel bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=15580 Let me quickly summarize the current state of this issue as I see it. The problem seems to be that certain 64bit chipsets can't deal with the fact that an URB's transfer_buffer is allocated with kmalloc(GFP_KERNEL). The effect is that kmalloc() is very likely to hand out memory which is not addressable by devices that are connected via 32bit PCI busses, such as EHCI controllers. In theory, DMA bounce buffers should be installed in such cases, or the IOMMU would be in charge to re-map these buffers to suitable locations, but for at least two people who have reported the issue, this obviously fails. So my approach was to force the driver using memory that is DMA coherent, which leads to 32bit addressable memory if the device's DMA_MASK is set accordingly. Other drivers do the same, and the patch for doing this is found here: http://lkml.org/lkml/2010/5/7/61. Users that see the bug all report that this patch fixes their problem with my driver. However, it was rejected back then as other developers said it was fixing the wrong end, and that the driver doesn't necessarily need coherent memory. Takashi recently posted a patch to the bugzilla entry which uses a different approach: it introduces a function to determine suitable GFP flags for USB devices, and passes __DMA32 to kmalloc() eventually. However, using this flags directly with the SLUB allocator is illegal and causes a BUG() in mm/slub.c, cache_grow(). The question now is how to proceed. I see three possible ways. 1. Find a way to allocate 32bit-aware memory with kmalloc(), following Takashi's idea 2. Find out exactly why these machines fail to install bounce buffers or set up their IOMMU correctly 3. re-activate the currently disabled functions usb_buffer_{map,unmap,sync} functions and let the USB stack do the memory mapping I'm not really deep into all the memory management code and even less into how the IOMMU and bounce buffers work, so I need some help here. Can anyone elaborate what would be the best option? I really want to get this fixed now, and provide a driver that works for everyone. What really puzzles me is that this doesn't hit a lot more people with other drivers. Thanks, Daniel

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 07 Jul '11

07 Jul '11

On Thu, Jul 7, 2011 at 2:33 PM, Oliver Neukum <oliver(a)neukum.org> wrote: > PS: Do you still see this if you enable 64bit DMA for EHCI? The problem is that I personally don't see that issue at all. I even installed 4GB of RAM to my development machine last year to be able to reproduce this, but I can't, even when the memory allocator is under heavy load. The only people who see this effect are Pedro Ribeiro and William Light (both in Cc:), and both have been very helpful in trying patches and reporting back in detail. Which instructions could we probably give to these people to finally hunt this issue down? Daniel

1 0

Re: [alsa-devel] [PATCH] ASoC: pxa2xx-pcm: remove unused variable 'dai'
by Liam Girdwood 07 Jul '11

07 Jul '11

On 06/07/11 14:20, Axel Lin wrote: > Remove unused variable 'dai' to eliminate below warning. > > CC sound/soc/pxa/pxa2xx-pcm.o > sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new': > sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai' > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/pxa/pxa2xx-pcm.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c > index da28394..c430600 100644 > --- a/sound/soc/pxa/pxa2xx-pcm.c > +++ b/sound/soc/pxa/pxa2xx-pcm.c > @@ -88,7 +88,6 @@ static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); > static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > - struct snd_soc_dai *dai = rtd->cpu_dai; > struct snd_pcm *pcm = rtd->pcm; > int ret = 0; > Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Takashi Iwai 07 Jul '11

07 Jul '11

At Thu, 7 Jul 2011 16:54:18 +0900, Toshiharu Okada wrote: > > Date: Wed, 06 Jul 2011 13:06:15 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > > > >I just took a quick glance over the code, and wonder whether this kind > >of driver would fit better with ASoC framework. > >Have you considered the implementation on ASoC? > > Thank you for your comment. > > I referred to the following files and implemented this driver. > > http://www.mirrorservice.org/sites/ftp.sourceforge.net/pub/sourceforge/s/pr… > sound/drivers/sta2x11i2s.c > > Therefore, I am not considering about ASoC. > Do I need to fit this driver to an ASoC framework for an upstream? Yes, I guess this would be a better choice in your case. Takashi

1 0

[alsa-devel] hda-intel front headphone (8086:27d8 1043:83bc)
by Alex 07 Jul '11

07 Jul '11

Hi, I run a Debian/sid on an Asus P5KPL-SE mainboard, the snd-hda-intel seem to work fine, ever if it run with guess method: Unknown hardware: "HDA-Intel" "Realtek ALC887" "HDA:10ec0887,104383bc,00100202" "0x1043" "0x83bc" Hardware is initialized using a guess method The only annoying thing is about the front headphone, every mixer application show the front headphone item but doesn't have adjustable level and I hear nothing on headphone. Any hint? http://www.alsa-project.org/db/?f=d6bc3005fab4683f2606e48f2bc909258a4ee3e3 Thanks in advance Regards

2 3

[alsa-devel] [PATCH] ASoC: twl6040: Add back support for legacy mode
by Peter Ujfalusi 07 Jul '11

07 Jul '11

The legacy mode has been accidentaly removed by commit: ASoC: twl6040: add all ABE DAIs Add back the twl6040-hifi dai. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Tony, Mark, Liam, this patch fixes the accidental removal of the twl6040-hifi dai from the codec driver. The series, which this patch depends on not yet pulled (I have the branch ready), since I'm still waiting for few acks for it. The MFD series is mostly OK, but the threaded interrupt related followup waits for acks, and also the PLL restructuring series is in waiting state. Is it OK, if I queue up this patch as well for the same pull request, and goes with the rest through l-o (since it is on top of those changes)? I could try to squash it down, if you prefer to the offending commit - I hope it is not breaking the rebase too much... Thank you, Peter sound/soc/codecs/twl6040.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d4b944c..443032b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1464,6 +1464,24 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { static struct snd_soc_dai_driver twl6040_dai[] = { { + .name = "twl6040-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, +{ .name = "twl6040-ul", .capture = { .stream_name = "Capture", -- 1.7.6

2 1

[alsa-devel] [PATCH v3 0/5] MFD/input/ASoC: twl6040: irq registration changes
by Peter Ujfalusi 06 Jul '11

06 Jul '11

Hello, Changes since v2: - Fix the check in the slaves driver for the validity of the received irq - Mark Brown's ack added to patch 3 Intro mail froim v1: As Samuel, and Felipe requested for the twl6040 MFD series: http://marc.info/?l=linux-kernel&m=130978512225430&w=2 http://marc.info/?l=linux-omap&m=130978403224218&w=2 This is the update to remove the obfuscation around the threaded irq request in the twl6040 driver and it's slaves. With this change the slaves uses directly the threaded irq request, and the core driver passes the irq number for the slaves to use (configured runtime, so machine drivers do not need to be updated). Regards, Peter PS: for the remaining comments from Samuel I have changed the series, since the requested change was small. I'm not intended to send the series again, unless it is requested (again the change from v6 is minimal - few lines has been removed/replaced). --- Peter Ujfalusi (5): MFD: twl6040: Use resource to provide irq number for slaves input: twl6040-vibra: Do not use wrapper for irq request ASoC: twl6040: Do not use wrapper for irq request MFD: twl6040: Demand valid interrupt configuration MFD: twl6040: Remove wrapper for threaded irq request drivers/input/misc/twl6040-vibra.c | 17 ++++++--- drivers/mfd/twl6040-core.c | 69 ++++++++++++++++++++++++----------- drivers/mfd/twl6040-irq.c | 16 +-------- include/linux/mfd/twl6040.h | 22 ----------- sound/soc/codecs/twl6040.c | 19 +++++++--- 5 files changed, 74 insertions(+), 69 deletions(-) -- 1.7.6

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Re: [alsa-devel] [RFC] ASoC: Add machine driver for i.MX SGTL500 based boards
by Mark Brown 06 Jul '11

06 Jul '11

On Mon, Jul 04, 2011 at 03:41:55PM -0300, Fabio Estevam wrote: > sgtl5000 1-000a: sgtl5000 revision 17 > sgtl5000 1-000a: asoc: failed to probe CODEC sgtl5000.1-000a: -22 > asoc: failed to instantiate card sgtl5000: -22 Have you looked to see where the error is coming from? > + ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_LRCLK, > + params_rate(params), 0); > + > + if (ret) { > + pr_err("%s: failed setting codec sysclk\n", __func__); > + return ret; > + } Fix this in the CODEC driver, copying this code into the machine drivers is clearly not useful. > +static int __init imx_sgtl5000_init(void) > +{ > + int ret; > + > + imx_sgtl5000_snd_device = platform_device_alloc("soc-audio", -1); > + if (!imx_sgtl5000_snd_device) > + return -ENOMEM; We're not accepting new drivers using soc-audio, you should use a regular platform device and register the card with snd_soc_register_card().

2 4

[alsa-devel] [PATCH] ARM i.MX dma: Fix burstsize settings
by Sascha Hauer 06 Jul '11

06 Jul '11

dmaengine expects the maxburst parameter in words, not bytes. The imxdma driver and its users do this wrong. Fix this. As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work with imx-dma. This broke the driver with imx-sdma support which correctly takes the maxburst parameter in words. This patch puts the sdma based sound back to work. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- I made this as a cross subsystem patch to maintain bisectability. I suggest to push it either via my tree or the dmaengine tree. drivers/dma/imx-dma.c | 3 ++- drivers/mmc/host/mxcmmc.c | 8 ++++---- sound/soc/imx/imx-pcm-dma-mx2.c | 4 ++-- 3 files changed, 8 insertions(+), 7 deletions(-) diff --git a/drivers/dma/imx-dma.c b/drivers/dma/imx-dma.c index e18eaab..d99f71c 100644 --- a/drivers/dma/imx-dma.c +++ b/drivers/dma/imx-dma.c @@ -135,7 +135,8 @@ static int imxdma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, if (ret) return ret; - imx_dma_config_burstlen(imxdmac->imxdma_channel, imxdmac->watermark_level); + imx_dma_config_burstlen(imxdmac->imxdma_channel, + imxdmac->watermark_level * imxdmac->word_size); return 0; default: diff --git a/drivers/mmc/host/mxcmmc.c b/drivers/mmc/host/mxcmmc.c index cc20e02..14aa213 100644 --- a/drivers/mmc/host/mxcmmc.c +++ b/drivers/mmc/host/mxcmmc.c @@ -715,13 +715,13 @@ static void mxcmci_set_ios(struct mmc_host *mmc, struct mmc_ios *ios) int burstlen, ret; /* - * use burstlen of 64 in 4 bit mode (--> reg value 0) - * use burstlen of 16 in 1 bit mode (--> reg value 16) + * use burstlen of 64 (16 words) in 4 bit mode (--> reg value 0) + * use burstlen of 16 (4 words) in 1 bit mode (--> reg value 16) */ if (ios->bus_width == MMC_BUS_WIDTH_4) - burstlen = 64; - else burstlen = 16; + else + burstlen = 4; if (mxcmci_use_dma(host) && burstlen != host->burstlen) { host->burstlen = burstlen; diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 4173b3d..43fdc24f 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize * buswidth; + slave_config.dst_maxburst = dma_params->burstsize; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize * buswidth; + slave_config.src_maxburst = dma_params->burstsize; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); -- 1.7.5.3 -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

2 1

[alsa-devel] [PATCH] ASoC: Conditionalize the enable of WM8994 ADC TDM mode
by Mark Brown 06 Jul '11

06 Jul '11

Future devices will not benefit from this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 16 ++++++++++++---- 1 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c0d8686..299f6eb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3114,10 +3114,18 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); - /* Unconditionally enable AIF1 ADC TDM mode; it only affects - * behaviour on idle TDM clock cycles. */ - snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, - WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + /* Unconditionally enable AIF1 ADC TDM mode on chips which can + * use this; it only affects behaviour on idle TDM clock + * cycles. */ + switch (control->type) { + case WM8994: + case WM8958: + snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, + WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + break; + default: + break; + } wm8994_update_class_w(codec); -- 1.7.5.4

2 1

Re: [alsa-devel] Wrong bitrate from usb microphone
by Takashi Iwai 06 Jul '11

06 Jul '11

At Tue, 05 Jul 2011 15:34:06 +0200, Pali Rohár wrote: > > Hello, > > I have USB webcam with integrated USB microphone. Microphone does not working > correctly because kernel module snd-usb-audio set wrong bitrate. > > Problem appered after this commit: > http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.39.y.git;a=commi… > > When I reverted patch in above commit microphone works properly. > This commit set bitrate to 8000Hz for 041e:4068 but my webcam (propably) > produce 16000Hz. So what to do with my webcam? It needs to revert above patch! > > Here is output from lsusb of my webcam: > Bus 002 Device 003: ID 041e:4068 Creative Technology, Ltd Live! Cam Notebook > [VF0470] > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 1.10 > bDeviceClass 0 (Defined at Interface level) > bDeviceSubClass 0 > bDeviceProtocol 0 > bMaxPacketSize0 8 > idVendor 0x041e Creative Technology, Ltd > idProduct 0x4068 Live! Cam Notebook [VF0470] > bcdDevice 1.00 > iManufacturer 1 > iProduct 2 > iSerial 0 > bNumConfigurations 1 Could you give the lsusb -v output? Arseniy, also check whether the current code is really OK? thanks, Takashi

1 0

[alsa-devel] [PATCH] Cirrus Logic CS4210-rev-A1 support.
by Tim Howe 06 Jul '11

06 Jul '11

From: Tim Howe <timothyc.howe(a)gmail.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/pci/hda/patch_cirrus.c | 841 ++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 805 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c7b5ca2..7e09722 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -51,7 +51,7 @@ struct cs_spec { unsigned int cur_adc_format; hda_nid_t dig_in; - const struct hda_bind_ctls *capture_bind[2]; + struct hda_bind_ctls *capture_bind[2]; unsigned int gpio_mask; unsigned int gpio_dir; @@ -61,6 +61,12 @@ struct cs_spec { unsigned int hp_detect:1; unsigned int mic_detect:1; + /* CS421x */ + unsigned int spdif_detect:1; + unsigned int sense_b:1; + hda_nid_t vendor_nid; + struct hda_input_mux input_mux; + unsigned int last_input; }; /* available models */ @@ -114,18 +120,20 @@ enum { static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + return snd_hda_codec_read(codec, spec->vendor_nid, 0, AC_VERB_GET_PROC_COEF, 0); } static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, unsigned int coef) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_PROC_COEF, coef); } @@ -231,7 +239,7 @@ static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, /* */ -static const struct hda_pcm_stream cs_pcm_analog_playback = { +static struct hda_pcm_stream cs_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -242,7 +250,7 @@ static const struct hda_pcm_stream cs_pcm_analog_playback = { }, }; -static const struct hda_pcm_stream cs_pcm_analog_capture = { +static struct hda_pcm_stream cs_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -252,7 +260,7 @@ static const struct hda_pcm_stream cs_pcm_analog_capture = { }, }; -static const struct hda_pcm_stream cs_pcm_digital_playback = { +static struct hda_pcm_stream cs_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -264,7 +272,7 @@ static const struct hda_pcm_stream cs_pcm_digital_playback = { }, }; -static const struct hda_pcm_stream cs_pcm_digital_capture = { +static struct hda_pcm_stream cs_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -331,8 +339,8 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t pin = cfg->inputs[idx].pin; - unsigned int val; - if (!is_jack_detectable(codec, pin)) + unsigned int val = snd_hda_query_pin_caps(codec, pin); + if (!(val & AC_PINCAP_PRES_DETECT)) return 0; val = snd_hda_codec_get_pincfg(codec, pin); return (snd_hda_get_input_pin_attr(val) != INPUT_PIN_ATTR_INT); @@ -346,10 +354,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { - hda_nid_t pins[2]; + hda_nid_t pins[HDA_MAX_NUM_INPUTS]; unsigned int type; int j, nums; - type = get_wcaps_type(get_wcaps(codec, nid)); + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; if (type != AC_WID_AUD_IN) continue; nums = snd_hda_get_connections(codec, nid, pins, @@ -558,10 +567,10 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, const char *name; int err, index; struct snd_kcontrol *kctl; - static const char * const speakers[] = { + static char *speakers[] = { "Front Speaker", "Surround Speaker", "Bass Speaker" }; - static const char * const line_outs[] = { + static char *line_outs[] = { "Front Line-Out", "Surround Line-Out", "Bass Line-Out" }; @@ -641,7 +650,7 @@ static int build_output(struct hda_codec *codec) /* */ -static const struct snd_kcontrol_new cs_capture_ctls[] = { +static struct snd_kcontrol_new cs_capture_ctls[] = { HDA_BIND_SW("Capture Switch", 0), HDA_BIND_VOL("Capture Volume", 0), }; @@ -709,7 +718,7 @@ static int cs_capture_source_put(struct snd_kcontrol *kcontrol, return change_cur_input(codec, idx, 0); } -static const struct snd_kcontrol_new cs_capture_source = { +static struct snd_kcontrol_new cs_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -718,7 +727,7 @@ static const struct snd_kcontrol_new cs_capture_source = { .put = cs_capture_source_put, }; -static const struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, +static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, struct hda_ctl_ops *ops) { struct cs_spec *spec = codec->spec; @@ -821,8 +830,7 @@ static int build_digital_output(struct hda_codec *codec) if (!spec->multiout.dig_out_nid) return 0; - err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); @@ -847,14 +855,15 @@ static void cs_automute(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int hp_present; + unsigned int caps, hp_present; hda_nid_t nid; int i; hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; - if (!is_jack_detectable(codec, nid)) + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) continue; hp_present = snd_hda_jack_detect(codec, nid); if (hp_present) @@ -923,7 +932,7 @@ static void init_output(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (is_jack_detectable(codec, nid)) { + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); @@ -982,7 +991,7 @@ static void init_input(struct hda_codec *codec) cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); } -static const struct hda_verb cs_coef_init_verbs[] = { +static struct hda_verb cs_coef_init_verbs[] = { {0x11, AC_VERB_SET_PROC_STATE, 1}, {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, {0x11, AC_VERB_SET_PROC_COEF, @@ -1016,7 +1025,7 @@ static const struct hda_verb cs_coef_init_verbs[] = { * blocks, which will alleviate the issue. */ -static const struct hda_verb cs_errata_init_verbs[] = { +static struct hda_verb cs_errata_init_verbs[] = { {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ @@ -1125,7 +1134,7 @@ static void cs_unsol_event(struct hda_codec *codec, unsigned int res) } } -static const struct hda_codec_ops cs_patch_ops = { +static struct hda_codec_ops cs_patch_ops = { .build_controls = cs_build_controls, .build_pcms = cs_build_pcms, .init = cs_init, @@ -1165,7 +1174,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { }; -static const struct snd_pci_quirk cs420x_cfg_tbl[] = { +static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), @@ -1179,7 +1188,7 @@ struct cs_pincfg { u32 val; }; -static const struct cs_pincfg mbp53_pincfgs[] = { +static struct cs_pincfg mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1193,7 +1202,7 @@ static const struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg mbp55_pincfgs[] = { +static struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1207,7 +1216,7 @@ static const struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg imac27_pincfgs[] = { +static struct cs_pincfg imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1221,22 +1230,22 @@ static const struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { +static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, }; -static void fix_pincfg(struct hda_codec *codec, int model) +static void fix_pincfg(struct hda_codec *codec, int model, + struct cs_pincfg **pin_configs) { - const struct cs_pincfg *cfg = cs_pincfgs[model]; + const struct cs_pincfg *cfg = pin_configs[model]; if (!cfg) return; for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1247,11 +1256,13 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->vendor_nid = CS420X_VENDOR_NID; + spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config); + fix_pincfg(codec, spec->board_config, cs_pincfgs); switch (spec->board_config) { case CS420X_IMAC27: @@ -1278,18 +1289,776 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + Cirrus Logic CS4210 + + 1 DAC => HP(sense) / Speakers, + 1 ADC <= LineIn(sense) / MicIn / DMicIn, + 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ +#define CS4210_DAC_NID 0x02 +#define CS4210_ADC_NID 0x03 +#define CS421X_VENDOR_NID 0x0B +#define CS421X_DMIC_PIN_NID 0x09 /* Port E */ +#define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ + +#define CS421X_IDX_DEV_CFG 0x01 +#define CS421X_IDX_ADC_CFG 0x02 +#define CS421X_IDX_DAC_CFG 0x03 +#define CS421X_IDX_SPK_CTL 0x04 + +#define SPDIF_EVENT 0x04 + +/* CS4210 boards */ +enum { + CS421X_CDB4210, + CS421X_MODELS +}; + +/* CS4210 board names */ +static const char *cs421x_models[CS421X_MODELS] = { + [CS421X_CDB4210] = "cdb4210", +}; + +static struct snd_pci_quirk cs421x_cfg_tbl[] = { + /* Test Intel board + CDB2410 */ + SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), + {} /* terminator */ +}; + +/* CS4210 board pinconfigs */ +/* Default CS4210 (CDB4210)*/ +static struct cs_pincfg cdb4210_pincfgs[] = { + { 0x05, 0x0321401f }, + { 0x06, 0x90170010 }, + { 0x07, 0x03813031 }, + { 0x08, 0xb7a70037 }, + { 0x09, 0xb7a6003e }, + { 0x0a, 0x034510f0 }, + {} /* terminator */ +}; + +static struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { + [CS421X_CDB4210] = cdb4210_pincfgs, +}; + +static struct hda_verb cs421x_coef_init_verbs[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 1}, + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG}, + /* + Disable Coefficient Index Auto-Increment(DAI)=1, + PDREF=0 + */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0001 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG}, + /* ADC SZCMode = Digital Soft Ramp */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0002 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG}, + {0x0B, AC_VERB_SET_PROC_COEF, + (0x0002 /* DAC SZCMode = Digital Soft Ramp */ + | 0x0004 /* Mute DAC on FIFO error */ + | 0x0008 /* Enable DAC High Pass Filter */ + )}, + {} /* terminator */ +}; + +/* Errata: CS4210 rev A1 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ + * + * Description: 1. Performance degredation is present in the ADC. + * 2. Speaker output is not completely muted upon HP detect. + * 3. Noise is present when clipping occurs on the amplified + * speaker outputs. + * + * Workaround: The following verb sequence written to the registers during + * initialization will correct the issues listed above. + */ + +static struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0006}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x000A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0011}, + {0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001B}, + {0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */ + + {} /* terminator */ +}; + +#include <sound/tlv.h> +/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */ +static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0); + +static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 3; + return 0; +} + +static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003; + return 0; +} + +static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + unsigned int vol = ucontrol->value.integer.value[0]; + unsigned int coef = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL); + + coef &= ~0x0003; + coef |= (vol & 0x0003); + cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef); + + return 1; +} + +static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Speaker Boost Playback Volume", + .info = cs421x_boost_vol_info, + .get = cs421x_boost_vol_get, + .put = cs421x_boost_vol_put, + .tlv = { .p = cs421x_speaker_boost_db_scale }, +}; + +/* + CS4210 auto-output redirect + HP/SPK/SPDIF +*/ +static void cs421x_auto_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int caps, hp_present, spdif_present; + hda_nid_t nid; + int i; + + spdif_present = 0; + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + caps = snd_hda_query_pin_caps(codec, nid); + if (caps & AC_PINCAP_PRES_DETECT) { + /* + TODO: SPDIF output redirect when SENSE_B is enabled. + Shared (SENSE_A) jack (e.g HP/mini-TOSLINK) + assumed. + */ + if (snd_hda_jack_detect(codec, nid) + /* && spec->sense_b */) + spdif_present = 1; + } + } + + hp_present = 0; + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) + continue; + hp_present = snd_hda_jack_detect(codec, nid); + if (hp_present) + break; + } + + /* mute speakers is spdif or hp jack is plugged in */ + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (hp_present | spdif_present) ? 0 : PIN_OUT); + } + + /* mute HPs if spdif jack (SENSE_B) is present */ + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (spdif_present & spec->sense_b)? 0 : PIN_HP); + } + + /* SPDIF TX on/off */ + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? PIN_OUT:0); + + } + /* Update board GPIOs if neccessary ... */ +} + +/* + CS4210 auto-input redirect + Switch max 3 inputs of a single ADC (nid 3) +*/ +static void cs421x_auto_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + unsigned int present; + + nid = cfg->inputs[spec->automic_idx].pin; + present = snd_hda_jack_detect(codec, nid); + + if (present) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } else { + spec->cur_input = spec->last_input; + } + + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); +} + +static void cs421x_pinmux_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + unsigned int def_conf, coef; + + /* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */ + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + + if (spec->gpio_mask) + coef |= 0x0008; /* B1,B2 are GPIOs */ + else + coef &= ~0x0008; + + if (spec->sense_b) + coef |= 0x0010; /* B2 is SENSE_B, not inverted */ + else + coef &= ~0x0010; + + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + if ((spec->gpio_mask || spec->sense_b) && + is_active_pin(codec, CS421X_DMIC_PIN_NID)) + { + /* + GPIO or SENSE_B forced - disconnect the DMIC pin. + */ + def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID); + def_conf &= ~AC_DEFCFG_PORT_CONN; + def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT); + snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf); + } +} + +/* */ +static void init_cs421x_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + /* mute first */ + for (i = 0; i < spec->multiout.num_dacs; i++) + snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + if (spec->multiout.hp_nid) + snd_hda_codec_write(codec, spec->multiout.hp_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) { + if (!spec->multiout.extra_out_nid[i]) + break; + snd_hda_codec_write(codec, spec->multiout.extra_out_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + } + + /* set appropriate pin controls */ + for (i = 0; i < cfg->line_outs; i++) + snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* HP */ + for (i = 0; i < cfg->hp_outs; i++) { + hda_nid_t nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | HP_EVENT); + spec->hp_detect = 1; + } + } + + /* Speaker */ + for (i = 0; i < cfg->speaker_outs; i++) + snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + + /* SPDIF is enabled on presence detect */ + if (spec->hp_detect || spec->spdif_detect) + cs421x_auto_output(codec); +} + +static void init_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + unsigned int ctl; + hda_nid_t pin = cfg->inputs[i].pin; + + if (!spec->adc_nid[i]) + continue; + /* set appropriate pin control and mute first */ + ctl = PIN_IN; + if (cfg->inputs[i].type == AUTO_PIN_MIC) { + unsigned int caps = snd_hda_query_pin_caps(codec, pin); + caps >>= AC_PINCAP_VREF_SHIFT; + if (caps & AC_PINCAP_VREF_80) + ctl = PIN_VREF80; + } + + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + snd_hda_codec_write(codec, spec->adc_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(spec->adc_idx[i])); + + if (spec->mic_detect && spec->automic_idx == i) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | MIC_EVENT); + } + } + + if (spec->mic_detect) + cs421x_auto_input(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } + +} + +static void init_cs421x_digital(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + + for (i = 0; i < cfg->dig_outs; i++) { + hda_nid_t nid = cfg->dig_out_pins[i]; + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | SPDIF_EVENT); + spec->spdif_detect = 1; + } + } +} + +static int cs421x_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + + cs421x_pinmux_init(codec); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_cs421x_output(codec); + init_cs421x_input(codec); + init_cs421x_digital(codec); + + return 0; +} + +/* + * CS4210 Input MUX (1 ADC) + */ +static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_input; + return 0; +} + +static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_nid[0], &spec->cur_input); + +} + +static struct snd_kcontrol_new cs421x_capture_source = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs421x_mux_enum_info, + .get = cs421x_mux_enum_get, + .put = cs421x_mux_enum_put, +}; + +static int cs421x_add_input_volume_control(struct hda_codec *codec, int item) +{ + struct cs_spec* spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + const struct hda_input_mux *imux = &spec->input_mux; + hda_nid_t pin = cfg->inputs[item].pin; + struct snd_kcontrol *kctl; + u32 caps; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + + return add_volume(codec, imux->items[item].label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + +/* add a (input-boost) volume control to the given input pin */ +static int build_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->input_mux; + int i, err, type_idx; + const char *label; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + int n; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); + if (err < 0) + return err; + } + } + + /* Add Input MUX Items + Capture Volume/Switch */ + for (i = 0; i < spec->num_inputs; i++) { + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx); + + err = cs421x_add_input_volume_control(codec, i); + if (err < 0) + return err; + } + + /* + Add 'Capture Source' Switch if + * 2 inputs and no mic detec + * 3 inputs + */ + if ((spec->num_inputs == 2 && !spec->mic_detect) || + (spec->num_inputs == 3) ) + { + err = snd_hda_ctl_add(codec, spec->adc_nid[0], + snd_ctl_new1(&cs421x_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* Single DAC (Mute/Gain) */ +static int build_cs421x_output(struct hda_codec *codec) +{ + hda_nid_t dac = CS4210_DAC_NID; + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct snd_kcontrol *kctl; + int err; + char* name = "HP/Speakers"; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + err = add_mute(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + if (cfg->speaker_outs) { + err = snd_hda_ctl_add(codec, 0, + snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); + if (err < 0) + return err; + } + return err; +} + +static int cs421x_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_cs421x_output(codec); + if (err < 0) + return err; + err = build_cs421x_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + return cs421x_init(codec); +} + +static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x3f) { + case HP_EVENT: + case SPDIF_EVENT: + cs421x_auto_output(codec); + break; + + case MIC_EVENT: + cs421x_auto_input(codec); + break; + } +} + +static int parse_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + spec->cur_input = spec->last_input = i; + spec->num_inputs ++; + + /* check whether the automatic mic switch is available */ + if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) { + spec->mic_detect = 1; + spec->automic_idx = i; + } + } + return 0; +} + +static int cs421x_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = parse_output(codec); + if (err < 0) + return err; + err = parse_cs421x_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +/* + Manage PDREF, when transitioning to D3hot + (DAC,ADC) -> D3, PDREF=1, AFG->D3 +*/ +static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +{ + unsigned int coef; + + snd_hda_shutup_pins(codec); + + snd_hda_codec_write(codec, CS4210_DAC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, CS4210_ADC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + return 0; +} +#endif + +static struct hda_codec_ops cs4210_patch_ops = { + .build_controls = cs421x_build_controls, + .build_pcms = cs_build_pcms, + .init = cs421x_init, + .free = cs_free, + .unsol_event = cs421x_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = cs421x_suspend, +#endif +}; + +static int patch_cs421x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS421X_VENDOR_NID; + + spec->board_config = + snd_hda_check_board_config(codec, CS421X_MODELS, + cs421x_models, cs421x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config, cs421x_pincfgs); + /* + Setup GPIO/SENSE for each board (if used) + */ + switch (spec->board_config) { + case CS421X_CDB4210: + snd_printd("CS4210 board: %s\n", cs421x_models[spec->board_config]); + +/* spec->gpio_mask = 3; + spec->gpio_dir = 3; + spec->gpio_data = 3; +*/ + spec->sense_b = 1; + + break; + } + + /* + Update the GPIO/DMIC/SENSE_B pinmux before the configuration + is auto-parsed. If GPIO or SENSE_B is forced, DMIC input + is disabled. + */ + cs421x_pinmux_init(codec); + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs4210_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_cirrus[] = { +static struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -- 1.7.0.4

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[alsa-devel] [PATCH 0/6] ASoC/MFD: twl6040: PLL handling changes
by Peter Ujfalusi 06 Jul '11

06 Jul '11

Hello, The following series depends on the v6 of twl6040 MFD series. The aim is to simplify the PLL handling for twl6040: - Move the PLL selection control to the codec driver from machine driver - Sample rate constraint simplification for selected PLL - Avoid multiple, duplicated PLL configuration, which can slow down the audio startup. - Harmonize the PLL ID usage among the drivers, so we avoid redundancy. I have generated this series on top of l-o/devel-cleanup branch + my previous twl6040 MFD/Input/ASoC v6 series, since it depends on the changes introduced there. Is it OK, if Tony will be taking this as well to avoid cross tree merge issues? I can put the series on top of my previous series, and offer Tony a pull link. Regards, Peter --- Peter Ujfalusi (6): ASoC: twl6040: Use neutral name for power mode text/enum ASoC: twl6040: Move PLL selection to codec driver ASoC: twl6040: Simplify sample rate constraint handling ASoC: twl6040: Configure PLL only once MFD: twl6040: Remove enum for PLL tracking ASoC: twl6040: No need to convert the PLL ID drivers/mfd/twl6040-core.c | 26 +++--- include/linux/mfd/twl6040.h | 17 ++--- sound/soc/codecs/twl6040.c | 182 ++++++++++++++++++++++--------------------- sound/soc/codecs/twl6040.h | 1 + sound/soc/omap/sdp4430.c | 47 +---------- 5 files changed, 118 insertions(+), 155 deletions(-) -- 1.7.6

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[alsa-devel] [PATCH] ASoC: wm8731: set the ACTIVE bit if bias ON entered
by Nicolas Ferre 05 Jul '11

05 Jul '11

Resuming from a suspend/resume cycle, the ACTIVE bit was not set even if a stream had actually been processing. Setting this bit at entering SND_SOC_BIAS_ON level will ensure this. Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- Supersed the wrong patch: "[PATCH 1/5] ASoC: wm8731: rework power management" now based on broonie/sound-2.6 git tree branch: for-3.1. sound/soc/codecs/wm8731.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2dc964b..ee4ec31 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + snd_soc_write(codec, WM8731_ACTIVE, 0x0001); break; case SND_SOC_BIAS_PREPARE: break; -- 1.7.3

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