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

[alsa-devel] [PATCH v2] ALSA: sound/atmel/ac97c.c: Add device tree support
by Alexander Stein 19 Dec '14

19 Dec '14

This adds device tree support for the AC97 controller. It uses the soc-ac97link bindings, but actually only ac97-reset is used. Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- Changes in v2: * It now uses the soc-ac97link bindings (partly) instead of defining its own. This should ease the transition to ASoC once a new driver has been written. sound/atmel/ac97c.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 42 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index cb44c74..73b61b2 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -22,6 +22,9 @@ #include <linux/gpio.h> #include <linux/types.h> #include <linux/io.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/initval.h> @@ -902,6 +905,40 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) } } +#ifdef CONFIG_OF +static const struct of_device_id atmel_ac97c_dt_ids[] = { + { .compatible = "atmel,atmel_ac97c", }, + { } +}; +MODULE_DEVICE_TABLE(of, atmel_ac97c_dt_ids); + +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + struct ac97c_platform_data *pdata; + struct device_node *node = dev->of_node; + const struct of_device_id *match; + + if (!node) { + dev_err(dev, "Device does not have associated DT data\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + pdata->reset_pin = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); + + return pdata; +} +#else +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + dev_err(dev, "no platform data defined\n"); + return ERR_PTR(-ENXIO); +} +#endif + static int atmel_ac97c_probe(struct platform_device *pdev) { struct snd_card *card; @@ -922,10 +959,11 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pdata = pdev->dev.platform_data; + pdata = dev_get_platdata(&pdev->dev); if (!pdata) { - dev_dbg(&pdev->dev, "no platform data\n"); - return -ENXIO; + pdata = atmel_ac97c_probe_dt(&pdev->dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); } irq = platform_get_irq(pdev, 0); @@ -1205,6 +1243,7 @@ static struct platform_driver atmel_ac97c_driver = { .name = "atmel_ac97c", .owner = THIS_MODULE, .pm = ATMEL_AC97C_PM_OPS, + .of_match_table = of_match_ptr(atmel_ac97c_dt_ids), }, }; module_platform_driver(atmel_ac97c_driver); -- 2.2.0

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[alsa-devel] iMX6 ESAI TDM with underrun and overrun support
by Aurelien Bouin 19 Dec '14

19 Dec '14

Hello, I am currently working on an iMX6 board, I work with the master branch of git://git.kernel.org/pub/scm/linux/kernel/git/stable/linux-stable.git I have seen some problems in the freescale ESAI implementation : 1.The registers TCCR and RCCR have TFP and RFP bits in register of a width of 5, not 4 ! With a value of 5 we are able to use the full range of slots (32) Patch : http://pastebin.com/Ai1RZ9mE 2.The channels number are limited in Playback to 12, but actually we can rise 128, in Capture it is limited to 8 but actually we can rise 128 also Patch : http://pastebin.com/SHk5eHPa I use the iMX6 ESAI bus to send audio datas to other processors ... but actually I don't have any codec behind ... I am wondering if there is any solution with the devicetree to implement such configuration with "simple-audio-card" Because for the moment I have not find another solution than doing a module : http://pastebin.com/cnGwnEjn I also see that the chip errata of the iMX6 ERR008000 is not handle ...Which make channel swapping on my ESAI bus ... Is there someone that succeed in implementing the underrun overrun catch ? Thank you for your return, Regards Aurelien BOUIN

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[alsa-devel] [PATCH/RFC 00/14] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 19 Dec '14

19 Dec '14

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 reported by Daniel Drake [1], by exposing the CDCLK gate clock through clk API. The remaining clocks (mux and divider) are also exposed, so the clk API could be already used instead of the set_sysclk() calls in the machine drivers. I need some more thought about interaction between the clk API calls on the clocks being exposed and the ASoC calls into sound/soc/samsung/i2s.c. I'm sending teh patches for review though to avoid any waste of time should it turn out the direction taken is wrong. This whole series definitely needs more testing, so far I only tested it on Odroid X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (14): ASoC: samsung: i2s: Remove unused gpios field from struct i2s ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect access to more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card .../devicetree/bindings/sound/samsung-i2s.txt | 18 +- arch/arm/boot/dts/exynos4.dtsi | 9 + arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 26 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- sound/soc/samsung/i2s.c | 365 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 7 files changed, 282 insertions(+), 158 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH v2 1/3] snd_pcm_start_at: Add enum for timestamp class, and helper struct for start_at ioctl.
by Tim Cussins 19 Dec '14

19 Dec '14

As per Pierre's suggestion, timestamp types are of two types: System, and Audio. Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd6..bd1e9be 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -466,12 +466,30 @@ struct snd_xfern { }; enum { + SNDRV_PCM_TSTAMP_CLASS_SYSTEM = 0, + SNDRV_PCM_TSTAMP_CLASS_AUDIO, + SNDRV_PCM_TSTAMP_CLASS_LAST = SNDRV_PCM_TSTAMP_CLASS_AUDIO, +}; + +enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; +enum { + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 0, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, +}; + +struct snd_start_at { + int tstamp_class; + int tstamp_type; + struct timespec start_time; +}; + /* channel positions */ enum { SNDRV_CHMAP_UNKNOWN = 0, @@ -550,6 +568,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_start_at) + /***************************************************************************** * * -- 2.1.0

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[alsa-devel] [PATCH] ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe
by Mark Brown 19 Dec '14

19 Dec '14

This reverts commit f8781db8aeb18d (ASoC: dapm: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing) since it is broken for deferred probing as it ends up storing data allocated with devm_ over multiple instantiations of the device. Reported-by: Russell King - ARM Linux <linux(a)arm.linux.org.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 935721062c21..d7595465cfbc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3231,7 +3231,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes, old_routes; + int num_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -3249,9 +3249,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - old_routes = card->num_dapm_routes; - routes = devm_kzalloc(card->dev, - (old_routes + num_routes) * sizeof(*routes), + routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -3259,11 +3257,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); - for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[old_routes + i].sink); + 2 * i, &routes[i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -3271,7 +3267,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[old_routes + i].source); + (2 * i) + 1, &routes[i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -3280,7 +3276,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes += num_routes; + card->num_dapm_routes = num_routes; card->dapm_routes = routes; return 0; -- 2.1.3

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[alsa-devel] [PATCH 1/2] ALSA: hda_controller: Separate stream_tag for input and output streams.
by libin.yang＠intel.com 19 Dec '14

19 Dec '14

From: Rafal Redzimski <rafal.f.redzimski(a)intel.com> Implemented separate stream_tag assignment for input and output streams. According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. This change is necessary to support HW which provides a total of more than 15 stream DMA engines which with legacy implementation causes an overflow on SDxCTL.STRM field (and the whole SDxCTL register) and as a result usage of Reserved value 0 in the SDxCTL.STRM field which confuses HDA controller. Signed-off-by: Rafal Redzimski <rafal.f.redzimski(a)intel.com> Signed-off-by: Jayachandran B <jayachandran.b(a)intel.com> Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_controller.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_priv.h | 1 + 2 files changed, 23 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8337645..dcca1ef 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1922,10 +1922,18 @@ int azx_mixer_create(struct azx *chip) EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return (index >= chip->capture_index_offset && + index < chip->capture_index_offset + chip->capture_streams); +} + /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1946,21 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout + * the stream direction group, + * valid values 1...15 + * use separate stream tag if the flag + * AZX_DCAPS_SEPARATE_STREAM_TAG is used + */ + if (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG) + azx_dev->stream_tag = + is_input_stream(chip, i) ? + ++in_stream_tag : + ++out_stream_tag; + else + azx_dev->stream_tag = i + 1; } return 0; diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index aa484fd..166e3e8 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -171,6 +171,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ enum { AZX_SNOOP_TYPE_NONE , -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb-audio: extend KEF X300A FU 10 tweak to Arcam rPAC
by Jiri Jaburek 18 Dec '14

18 Dec '14

The Arcam rPAC seems to have the same problem - whenever anything (alsamixer, udevd, 3.9+ kernel from 60af3d037eb8c, ..) attempts to access mixer / control interface of the card, the firmware "locks up" the entire device, resulting in SNDRV_PCM_IOCTL_HW_PARAMS failed (-5): Input/output error from alsa-lib. Other operating systems can somehow read the mixer (there seems to be playback volume/mute), but any manipulation is ignored by the device (which has hardware volume controls). Cc: <stable(a)vger.kernel.org> Signed-off-by: Jiri Jaburek <jjaburek(a)redhat.com> --- sound/usb/mixer_maps.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1994d41..b703cb3 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -333,8 +333,11 @@ static struct usbmix_name_map gamecom780_map[] = { {} }; -static const struct usbmix_name_map kef_x300a_map[] = { - { 10, NULL }, /* firmware locks up (?) when we try to access this FU */ +/* some (all?) SCMS USB3318 devices are affected by a firmware lock up + * when anything attempts to access FU 10 (control) + */ +static const struct usbmix_name_map scms_usb3318_map[] = { + { 10, NULL }, { 0 } }; @@ -434,8 +437,14 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = ebox44_map, }, { + /* KEF X300A */ .id = USB_ID(0x27ac, 0x1000), - .map = kef_x300a_map, + .map = scms_usb3318_map, + }, + { + /* Arcam rPAC */ + .id = USB_ID(0x25c4, 0x0003), + .map = scms_usb3318_map, }, { 0 } /* terminator */ }; -- 2.1.0

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[alsa-devel] [PATCH v2] ASoC: fsl_esai: Fix ESAI_xCCR_xDC_WIDTH definition
by Aurelien Bouin 18 Dec '14

18 Dec '14

>From 9bcae891b8fe7a42cbc2cd88319f7bd4ab44d012 Mon Sep 17 00:00:00 2001 From: Aurelien BOUIN <a_bouin(a)yahoo.fr> Date: Thu, 18 Dec 2014 15:14:53 +0100 Subject: [PATCH 1/1] According to the mx6 reference manual the TDC and RDC field of registers ESAI_TCCR and ESAI_RCCR have a width of 5 bits, not 4 Signed-off-by: Aurelien BOUIN <a_bouin(a)yahoo.fr> diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f..5e793bb 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 -- 1.9.1

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[alsa-devel] [PATCH] sound/soc/adi/axi-spdif.c: Support programmable master clock
by Mike Looijmans 18 Dec '14

18 Dec '14

If the master clock supports programmable rates, program it to generate the desired frequency. Only apply constraints when the clock is fixed. This allows proper clock generation for both 44100 and 48000 Hz based sampling rates if the platform supports it. The clock frequency must be set before enabling it. Enabling the clock was done in "startup", but that occurs before "hw_params" where the rate is known. Move the clock start to the hw_params routine, and keep track of whether the clock has been started, because shutdown may be called without having called hw_params first. Starting the clock and enabling the SPDIF output AFTER programming the dividers is a more logical order anyway. Signed-off-by: Mike Looijmans <mike.looijmans(a)topic.nl> --- sound/soc/adi/axi-spdif.c | 56 ++++++++++++++++++++++++++++----------------- 1 file changed, 35 insertions(+), 21 deletions(-) diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 198e3a4..d67e010 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -4,7 +4,6 @@ * * Licensed under the GPL-2. */ - #include <linux/init.h> #include <linux/kernel.h> #include <linux/module.h> @@ -44,6 +43,8 @@ struct axi_spdif { struct snd_ratnum ratnum; struct snd_pcm_hw_constraint_ratnums rate_constraints; + + bool clk_ref_running; }; static int axi_spdif_trigger(struct snd_pcm_substream *substream, int cmd, @@ -79,6 +80,7 @@ static int axi_spdif_hw_params(struct snd_pcm_substream *substream, struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); unsigned int rate = params_rate(params); unsigned int clkdiv, stat; + int ret; switch (params_rate(params)) { case 32000: @@ -95,6 +97,9 @@ static int axi_spdif_hw_params(struct snd_pcm_substream *substream, break; } + /* Try to set the master clock */ + clk_set_rate(spdif->clk_ref, rate * 128); + clkdiv = DIV_ROUND_CLOSEST(clk_get_rate(spdif->clk_ref), rate * 64 * 2) - 1; clkdiv <<= AXI_SPDIF_CTRL_CLKDIV_OFFSET; @@ -103,6 +108,14 @@ static int axi_spdif_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, AXI_SPDIF_CTRL_CLKDIV_MASK, clkdiv); + ret = clk_prepare_enable(spdif->clk_ref); + if (ret) + return ret; + spdif->clk_ref_running = true; + + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_TXEN, AXI_SPDIF_CTRL_TXEN); + return 0; } @@ -121,18 +134,13 @@ static int axi_spdif_startup(struct snd_pcm_substream *substream, struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); int ret; - ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &spdif->rate_constraints); - if (ret) - return ret; - - ret = clk_prepare_enable(spdif->clk_ref); - if (ret) - return ret; - - regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, - AXI_SPDIF_CTRL_TXEN, AXI_SPDIF_CTRL_TXEN); + if (spdif->rate_constraints.nrats) { + ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &spdif->rate_constraints); + if (ret) + return ret; + } return 0; } @@ -145,7 +153,10 @@ static void axi_spdif_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, AXI_SPDIF_CTRL_TXEN, 0); - clk_disable_unprepare(spdif->clk_ref); + if (spdif->clk_ref_running) { + clk_disable_unprepare(spdif->clk_ref); + spdif->clk_ref_running = false; + } } static const struct snd_soc_dai_ops axi_spdif_dai_ops = { @@ -216,14 +227,17 @@ static int axi_spdif_probe(struct platform_device *pdev) spdif->dma_data.addr_width = 4; spdif->dma_data.maxburst = 1; - spdif->ratnum.num = clk_get_rate(spdif->clk_ref) / 128; - spdif->ratnum.den_step = 1; - spdif->ratnum.den_min = 1; - spdif->ratnum.den_max = 64; - - spdif->rate_constraints.rats = &spdif->ratnum; - spdif->rate_constraints.nrats = 1; + /* Determine if the clock rate is fixed. If it cannot change frequency, + * it returns an error here. */ + if (clk_round_rate(spdif->clk_ref, 128 * 44100) < 0) { + spdif->ratnum.num = clk_get_rate(spdif->clk_ref) / 128; + spdif->ratnum.den_step = 1; + spdif->ratnum.den_min = 1; + spdif->ratnum.den_max = 64; + spdif->rate_constraints.rats = &spdif->ratnum; + spdif->rate_constraints.nrats = 1; + } ret = devm_snd_soc_register_component(&pdev->dev, &axi_spdif_component, &axi_spdif_dai, 1); if (ret) -- 1.7.9.5

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[alsa-devel] [[PATCH] ALSA: In Freescale ESAI driver xDxDC bits in xCCR registers is 5 bits width not 4
by Aurelien BOUIN 18 Dec '14

18 Dec '14

Signed-off-by: Aurelien BOUIN <a_bouin(a)yahoo.fr> diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f..5e793bb 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 -- 1.9.1

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[alsa-devel] Device Tree simple-audio-card without codec HOWTO ?
by Aurelien Bouin 18 Dec '14

18 Dec '14

Hello, I use the iMX6 ESAI bus to send audio datas to other processors ... but actually I don't have any codec behind ... I am wondering if there is any solution with the devicetree to implement such configuration with "simple-audio-card" That will configure tdm slots and width. I tried without success that : sound { compatible = "simple-audio-card"; simple-audio-card,name = "Generic-ESAI-Sound-Card"; simple-audio-card,format = "left_j"; simple-audio-card,cpu { sound-dai = <&esai 0>; dai-tdm-slot-num = <32>; dai-tdm-slot-width = <8>; }; simple-audio-card,codec { sound-dai = <0 0>; }; }; Because for the moment I have not find another solution than doing a module : http://pastebin.com/cnGwnEjn Thank you for your help, Regards, Aurelien BOUIN

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[alsa-devel] [PATCH] ALSA: pcm: Don't leave PREPARED state after draining
by Takashi Iwai 18 Dec '14

18 Dec '14

When a PCM draining is performed to an empty stream that has been already in PREPARED state, the current code just ignores and leaves as it is, although the drain is supposed to set all such streams to SETUP state. This patch covers that overlooked case. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d9572ad2b..64d9863d6565 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1546,6 +1546,8 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) if (! snd_pcm_playback_empty(substream)) { snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING); snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING); + } else { + runtime->status->state = SNDRV_PCM_STATE_SETUP; } break; case SNDRV_PCM_STATE_RUNNING: -- 2.2.0

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[alsa-devel] [PATCH 1/2] ALSA: hda_controller: Separate stream_tag for input and output streams.
by libin.yang＠intel.com 18 Dec '14

18 Dec '14

From: rfredzim <rafal.f.redzimski(a)intel.com> Implemented separate stream_tag assignment for input and output streams. According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. This change is necessary to support HW which provides a total of more than 15 stream DMA engines which with legacy implementation causes an overflow on SDxCTL.STRM field (and the whole SDxCTL register) and as a result usage of Reserved value 0 in the SDxCTL.STRM field which confuses HDA controller. Signed-off-by: rfredzim <rafal.f.redzimski(a)intel.com> Signed-off-by: Jayachandran B <jayachandran.b(a)intel.com> Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_controller.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_priv.h | 1 + 2 files changed, 23 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8337645..dcca1ef 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1922,10 +1922,18 @@ int azx_mixer_create(struct azx *chip) EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return (index >= chip->capture_index_offset && + index < chip->capture_index_offset + chip->capture_streams); +} + /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1946,21 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout + * the stream direction group, + * valid values 1...15 + * use separate stream tag if the flag + * AZX_DCAPS_SEPARATE_STREAM_TAG is used + */ + if (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG) + azx_dev->stream_tag = + is_input_stream(chip, i) ? + ++in_stream_tag : + ++out_stream_tag; + else + azx_dev->stream_tag = i + 1; } return 0; diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index aa484fd..166e3e8 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -171,6 +171,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ enum { AZX_SNOOP_TYPE_NONE , -- 1.9.1

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[alsa-devel] [PATCH 1/3] ALSA: asihpi: relax firmware version check
by eliot＠blennerhassett.gen.nz 18 Dec '14

18 Dec '14

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Some products firmware is no longer being updated e.g. dsp5000, dsp8700 but it should continue to work with updated HPI versions. Avoid regression by allowing this firmware to be loaded as long as major version is the same. Warn about mismatching versions, as matching versions are preferred. Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> --- sound/pci/asihpi/hpidspcd.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c index ac91637..3603c24 100644 --- a/sound/pci/asihpi/hpidspcd.c +++ b/sound/pci/asihpi/hpidspcd.c @@ -1,8 +1,9 @@ -/***********************************************************************/ -/** +/*********************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support(a)audioscience.com> + Functions for reading DSP code using hotplug firmware loader + + Copyright (C) 1997-2014 AudioScience Inc. <support(a)audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -17,11 +18,7 @@ along with this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -\file -Functions for reading DSP code using -hotplug firmware loader from individual dsp code files -*/ -/***********************************************************************/ +***********************************************************************/ #define SOURCEFILE_NAME "hpidspcd.c" #include "hpidspcd.h" #include "hpidebug.h" @@ -68,17 +65,18 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, goto error2; } - if ((header.version >> 9) != (HPI_VER >> 9)) { - /* Consider even and subsequent odd minor versions to be compatible */ - dev_err(&dev->dev, "Incompatible firmware version DSP image %X != Driver %X\n", + if (HPI_VER_MAJOR(header.version) != HPI_VER_MAJOR(HPI_VER)) { + /* Major version change probably means Host-DSP protocol change */ + dev_err(&dev->dev, + "Incompatible firmware version DSP image %X != Driver %X\n", header.version, HPI_VER); goto error2; } if (header.version != HPI_VER) { - dev_info(&dev->dev, - "Firmware: release version mismatch DSP image %X != Driver %X\n", - header.version, HPI_VER); + dev_warn(&dev->dev, + "Firmware version mismatch: DSP image %X != Driver %X\n", + header.version, HPI_VER); } HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix Scarlett 6i6 initialization typo
by Chris J Arges 18 Dec '14

18 Dec '14

The num_controls field was incorrectly set to 0 causing 6i6 to not be initialized. Set this to 9. Reported-and-tested-by: Mark Roberts <sunifiram(a)gmail.com> Signed-off-by: Chris J Arges <chris.j.arges(a)canonical.com> --- sound/usb/mixer_scarlett.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 9109652..7438e7c 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -655,7 +655,7 @@ static struct scarlett_device_info s6i6_info = { .names = NULL }, - .num_controls = 0, + .num_controls = 9, .controls = { { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone" }, -- 1.9.1

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[alsa-devel] [PATCH 1/2] ALSA: hda - Create capture source ctls when stereo mix input is added
by Takashi Iwai 18 Dec '14

18 Dec '14

When the stereo mix input is explicitly enabled via a user hint, the driver should create always a capture source enum ctl and disable the auto-mic switch. Otherwise the behavior gets confused. For doing it, this patch just sets spec->suppress_auto_mic flag appropriately. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 63b69f750d8e..293af6b5ba3d 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3224,6 +3224,8 @@ static int create_input_ctls(struct hda_codec *codec) "Stereo Mix", 0); if (err < 0) return err; + else + spec->suppress_auto_mic = 1; } return 0; -- 2.2.0

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[alsa-devel] [PATCH] ALSA: hda - Add quirk for Packard Bell EasyNote MX65
by Takashi Iwai 17 Dec '14

17 Dec '14

Packard Bell EasyNote MX65 with AD1986A codec needs a few fixups, namely, the pin config overrides to set only the known I/O pins and the EAPD has to be turned on. In addition, add stereo mix input forcibly for avoiding the weird KDE behavior by this update. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=88251 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 39 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index eb54da757407..a9d78e275138 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -257,6 +257,18 @@ static void ad1986a_fixup_eapd(struct hda_codec *codec, } } +/* enable stereo-mix input for avoiding regression on KDE (bko#88251) */ +static void ad1986a_fixup_eapd_mix_in(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + ad1986a_fixup_eapd(codec, fix, action); + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_ENABLE; + } +} + enum { AD1986A_FIXUP_INV_JACK_DETECT, AD1986A_FIXUP_ULTRA, @@ -265,6 +277,8 @@ enum { AD1986A_FIXUP_LAPTOP, AD1986A_FIXUP_LAPTOP_IMIC, AD1986A_FIXUP_EAPD, + AD1986A_FIXUP_EAPD_MIX_IN, + AD1986A_FIXUP_EASYNOTE, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -329,6 +343,30 @@ static const struct hda_fixup ad1986a_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = ad1986a_fixup_eapd, }, + [AD1986A_FIXUP_EAPD_MIX_IN] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1986a_fixup_eapd_mix_in, + }, + [AD1986A_FIXUP_EASYNOTE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x0421402f }, /* headphone */ + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x90a70130 }, /* int mic */ + { 0x1e, 0x411111f0 }, /* N/A */ + { 0x1f, 0x04a19040 }, /* mic */ + { 0x20, 0x411111f0 }, /* N/A */ + { 0x21, 0x411111f0 }, /* N/A */ + { 0x22, 0x411111f0 }, /* N/A */ + { 0x23, 0x411111f0 }, /* N/A */ + { 0x24, 0x411111f0 }, /* N/A */ + { 0x25, 0x411111f0 }, /* N/A */ + {} + }, + .chained = true, + .chain_id = AD1986A_FIXUP_EAPD_MIX_IN, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { @@ -342,6 +380,7 @@ static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), + SND_PCI_QUIRK(0x1631, 0xc022, "PackardBell EasyNote MX65", AD1986A_FIXUP_EASYNOTE), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK), -- 2.2.0

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[alsa-devel] [PATCH] usb: add native DSD support for Matrix Audio DACs
by Jurgen Kramer 17 Dec '14

17 Dec '14

This patch adds native DSD support for two XMOS based DACs from Matrix Audio: - X-Sabre - Mini-i Pro Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4dbfb3d..a739841 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1245,8 +1245,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - /* iFi Audio micro/nano iDSD */ - case USB_ID(0x20b1, 0x3008): + case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ + case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ + case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 1.9.3

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[alsa-devel] Again: Alsa Driver Scarlett
by Orm Finnendahl 17 Dec '14

17 Dec '14

Hi, this is relating to my post from September: There doesn't seem to be any progress concerning the integration of the Scarlett alsa-driver patch into Alsa. Can someone on the list tell me how to invoke a process to get it integrated? The patch sources are here: https://github.com/smilingthax/alsa-driver_scarlett At the moment most of our students regularly have to reapply the patch as soon as they upgrade their systems and it's pretty time consuming and painful for the studio technicians and me to guide them through the process of patching kernel modules, so it would be great to simplify this process by integrating it into the official alsa distribution. I can confirm that the patch still works well on our systems. -- Orm ------------------------------------------------------------------ Prof. Orm Finnendahl Komposition Hochschule für Musik und Darstellende Kunst Eschersheimer Landstr. 29-39 60322 Frankfurt am Main ----- Forwarded message from Orm Finnendahl <orm.finnendahl(a)selma.hfmdk-frankfurt.de> ----- Subject: Re: [alsa-devel] Alsa Driver Scarlett From: Orm Finnendahl <orm.finnendahl(a)selma.hfmdk-frankfurt.de> To: alsa-devel(a)alsa-project.org Date: Tue, 16 Sep 2014 15:49:25 +0200 Am Dienstag, den 16. September 2014 um 15:22:27 Uhr (+0200) schrieb Johannes Kroll: > > I guess I'll need to find the time to port the patch to a recent kernel > when I upgrade, unless someone else does it first. The patch still works on our recent kernels (we are on debian testing with a 3.14-2 kernel), atm there is no porting necessary. I can also confirm that the driver has no known issues on our systems (we use a couple of Scarlett 18i20 in different configurations). It would be great, though if the patch could get integrated into alsa. At the moment it is quite painful for our students with linux laptops as they have to reapply the patch on every upgrade. -- Orm Finnendahl ------------------------------------------------------------------ Prof. Orm Finnendahl Komposition Hochschule für Musik und Darstellende Kunst Eschersheimer Landstr. 29-39 60322 Frankfurt am Main ----- End forwarded message -----

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Re: [alsa-devel] New codec support for ALC256
by Takashi Iwai 17 Dec '14

17 Dec '14

At Wed, 17 Dec 2014 07:56:56 +0000, Kailang wrote: > > Attach the patch. Don't we need the addition to alc_fill_eapd_coef()? thanks, Takashi

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Re: [alsa-devel] New Dell desktop support
by Takashi Iwai 17 Dec '14

17 Dec '14

At Wed, 17 Dec 2014 09:18:56 +0000, Kailang wrote: > > Hi Takashi, > > Attach is the new SSID for Dell desktop for ALC255. > I add it into fixup table first. > If it has more machine that verb table was the same, than let's move it to pin fixup table. OK, applied now with Cc to stable. thanks, Takashi

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[alsa-devel] New codec support for ALC256
by Kailang 17 Dec '14

17 Dec '14

Hi Takashi, Attach is the patch for new codec support. Thanks. BR, Kailang

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[alsa-devel] [GIT PULL] ASoC fixes for v3.19
by Mark Brown 16 Dec '14

16 Dec '14

The following changes since commit 1810afd3e1ded09c53d4e966dddce3c7d484521f: Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712' and 'asoc/topic/wm9713' into asoc-next (2014-12-08 13:12:36 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19-2 for you to fetch changes up to c52e5698efef1d9999d744be001be62d9fcabf55: Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/intel', 'asoc/fix/rt5645', 'asoc/fix/rt5677' and 'asoc/fix/samsung' into asoc-linus (2014-12-16 11:52:34 +0000) ---------------------------------------------------------------- ASoC: Merge window fixes for v3.19 A smattering of driver specific fixes here, some more cleanup of the Exynos7 and Intel ACPI support plus a fix for register cache sync on the pcm512x driver which has been lurking for a while. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt5645: Fix potential crash in jd function Ben Zhang (1): ASoC: rt5677: add REGMAP_I2C and REGMAP_IRQ dependency Kevin Strasser (1): ASoC: Intel: fix possible acpi enumeration panic Mark Brown (1): Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/intel', 'asoc/fix/rt5645', 'asoc/fix/rt5677' and 'asoc/fix/samsung' into asoc-linus Matthieu Crapet (1): ASoC: atmel_ssc_dai/trivial: typo fix Peter Rosin (1): ASoC: pcm512x: Trigger auto-increment of register addresses on i2c Sylwester Nawrocki (1): ASoC: samsung: i2s: Add missing assignment of variant_regs Wei Yongjun (1): ASoC: Intel: fix return value check in sst_acpi_probe() sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/pcm512x-i2c.c | 7 ++++++- sound/soc/codecs/rt5645.c | 4 ++++ sound/soc/intel/sst/sst_acpi.c | 10 +++++----- sound/soc/samsung/i2s.c | 2 ++ 6 files changed, 20 insertions(+), 7 deletions(-)

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[alsa-devel] [PATCH] ASoC: dapm: Don't use async I/O
by Mark Brown 16 Dec '14

16 Dec '14

The only user of the async I/O support in ASoC is SPI which was using it to avoid needless context thrashing and minimise controller runtime PM bounces. The SPI framework has now been enhanced so that even normal spi_sync() calls won't suffer these effects so we don't need to handle this in ASoC and in fact it can be more efficient not to since we don't need to set up and tear down the buffers needed to manage asynchronous I/O. The async completions that DAPM does are left in place so drivers can use them, they are very cheap if there is no asynchronous work queued. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..fe06b028130f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -517,8 +517,8 @@ static int soc_dapm_update_bits(struct snd_soc_dapm_context *dapm, { if (!dapm->component) return -EIO; - return snd_soc_component_update_bits_async(dapm->component, reg, - mask, value); + return snd_soc_component_update_bits(dapm->component, reg, + mask, value); } static int soc_dapm_test_bits(struct snd_soc_dapm_context *dapm, -- 2.1.3

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[alsa-devel] [PATCH RFC 0/9] audio timestamping evolutions
by Pierre-Louis Bossart 16 Dec '14

16 Dec '14

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new audio_timestamp definitions provided here. 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 5 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. TODO: for start_at, need to take audio tstamp snapshots even if stream is not running. Pierre-Louis Bossart (9): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info Documentation/sound/alsa/timestamping.txt | 171 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 52 ++++++++- include/uapi/sound/asound.h | 20 +++- sound/core/pcm_compat.c | 7 +- sound/core/pcm_lib.c | 81 +++++++++----- sound/core/pcm_native.c | 22 +++- sound/pci/hda/hda_controller.c | 41 +++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 355 insertions(+), 48 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 2.1.0

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[alsa-devel] [PATCH] ASoC: Intel: fix possible acpi enumeration panic
by Kevin Strasser 16 Dec '14

16 Dec '14

A crash can occur on some platforms where adsp is enumerated but codec is not matched. Check that the codec_id string is valid before attempting to match. Signed-off-by: Kevin Strasser <kevin.strasser(a)intel.com> --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 31124aa..dd72e58 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -236,7 +236,7 @@ static struct sst_machines *sst_acpi_find_machine( struct sst_machines *mach; bool found = false; - for (mach = machines; mach->codec_id; mach++) + for (mach = machines; mach->codec_id[0]; mach++) if (ACPI_SUCCESS(acpi_get_devices(mach->codec_id, sst_acpi_mach_match, &found, NULL)) && found) -- 1.9.1

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[alsa-devel] Streaming wake-up buffer from DSP to SoC over SPI
by Dylan Reid 16 Dec '14

16 Dec '14

What is the right way to model a bursty, SPI based audio transfer? There could be hours between samples, then several seconds of audio will be ready. The codec, an rt5677, is attached to the SoC with both an IIS bus and a SPI bus. The IIS bus is used for normal audio playback and record. The SPI bus is used for transferring audio that caused a voice wake, "OK Google" in this case. When the codec detects the wake up phrase, the SoC will read out two plus seconds of buffered data over the SPI bus and continue to stream audio data over SPI for the rest of the query. Nexus 9 has a custom ioctl to access rt5677-spi and some user-space smarts to poll for more data. I was thinking of making a separate PCM device. The odd things are that it would have to originate from soc/codecs, the codec would own the codec and cpu end of the link, and the hw_ptr position would need to be queried over SPI. But it would make userspace access to it more standard looking. Thanks for any opinions or thoughts. Dylan

3 4

[alsa-devel] Audio USB disappears suddenly
by mobidyc 16 Dec '14

16 Dec '14

Hello, don't know if I need to report my problem in alsa-dev list for the driver or linux-dev list for the usb interface, please correct me if I post in wrong mailing. sometimes, my USB audio device disappears from the system, the only way to make it appears again is to disconnect/reconnect the usb wire. Here are the traces I took : USB DAC suddenly disappears, from dmesg I can see: [13083.161157] usb 1-1.3: 1:2: usb_set_interface failed (-110) [13083.316344] usb 1-1.3: 1:2: usb_set_interface failed (-71) [13083.471927] usb 1-1.3: 1:2: usb_set_interface failed (-71) [13083.789810] usb 1-1.3: 1:2: usb_set_interface failed (-71) rmmod/modprobe snd-usb-audio: [13882.343950] usbcore: deregistering interface driver snd-usb-audio [13891.187513] usb 1-1.3: parse_audio_format_rates_v2(): unable to find clock source (clock -71) [13891.200234] usb 1-1.3: parse_audio_format_rates_v2(): unable to find clock source (clock -71) [13891.242722] usb 1-1.3: cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 [13891.242728] usb 1-1.3: 10:0: cannot get min/max values for control 2 (id 10) [13891.255466] usb 1-1.3: cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 [13891.255471] usb 1-1.3: 10:0: cannot get min/max values for control 2 (id 10) [13891.255737] usbcore: registered new interface driver snd-usb-audio [13891.276723] usb 1-1.3: cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0xa00, type = 1 [13891.284222] usb 1-1.3: cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0xa00, type = 1 [13891.290842] usb 1-1.3: cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0xa00, type = 1 [13891.297210] usb 1-1.3: cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0xa00, type = 1 from "aplay -l" the card is seen as : card 1: U0x249c0x9313 [USB Device 0x249c:0x9313], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 after disconnected/reconnected the usb wire: card 1: UAC2 [YoungDSD UAC2], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 [16005.411254] usb 1-1.3: USB disconnect, device number 4 [16007.132442] usb 1-1.3: new high-speed USB device number 6 using ehci-pci [16007.250087] input: M2Tech YoungDSD UAC2 as /devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.3/1-1.3:1.3/0003:249C:9313.0006/input/input22 [16007.250233] hid-generic 0003:249C:9313.0006: input,hidraw2: USB HID v1.10 Device [M2Tech YoungDSD UAC2] on usb-0000:00:1a.0-1.3/input3 Is there any debug i could turn on to get more informations? Additional informations: # ./aadebug ALSA Audio Debug v0.2.0 - Wed Dec 10 19:56:00 CET 2014 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/agpl-3.0.txt Kernel ---------------------------------------------------- Linux TEVERE 3.16.3 #1 SMP Sun Nov 23 16:17:25 CET 2014 x86_64 GNU/Linux Advanced Linux Sound Architecture Driver Version k3.16.3. Loaded Modules -------------------------------------------- snd_usb_audio 105962 1 snd_hda_codec_hdmi 31141 1 snd_hda_codec_realtek 47854 1 snd_hda_codec_generic 39920 1 snd_hda_codec_realtek snd_usbmidi_lib 16134 1 snd_usb_audio snd_rawmidi 14042 1 snd_usbmidi_lib snd_hda_intel 18271 0 snd_hda_controller 15146 1 snd_hda_intel snd_hda_codec 73127 5 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller snd_hwdep 4949 2 snd_usb_audio,snd_hda_codec snd_pcm 59473 6 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel,snd_hda_controller Proc Asound ----------------------------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7c10000 irq 45 1 [UAC2 ]: USB-Audio - YoungDSD UAC2 M2Tech YoungDSD UAC2 at usb-0000:00:1a.0-1.3, high speed 2: [ 1] : control 3: [ 1- 0]: digital audio playback 4: [ 0] : control 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0- 1]: digital audio playback 8: [ 0- 2]: digital audio capture 9: [ 0- 3]: digital audio playback 10: [ 0- 7]: digital audio playback 11: [ 0- 2]: hardware dependent 12: [ 0- 3]: hardware dependent 33: : timer 00-02: HDA Codec 2 00-03: HDA Codec 3 00-00: ALC892 Analog : ALC892 Analog : playback 1 : capture 1 00-01: ALC892 Digital : ALC892 Digital : playback 1 00-02: ALC892 Alt Analog : ALC892 Alt Analog : capture 1 00-03: HDMI 0 : HDMI 0 : playback 1 00-07: HDMI 1 : HDMI 1 : playback 1 01-00: USB Audio : USB Audio : playback 1 cat: can't open '/proc/asound/seq/clients': No such file or directory Dev Snd --------------------------------------------------- total 0 drwxr-xr-x 2 root root 60 Dec 10 18:28 by-id drwxr-xr-x 2 root root 80 Dec 10 18:28 by-path crw-rw---- 1 root audio 116, 4 Dec 10 14:01 controlC0 crw-rw---- 1 root audio 116, 2 Dec 10 18:28 controlC1 crw-rw---- 1 root audio 116, 11 Dec 10 14:01 hwC0D2 crw-rw---- 1 root audio 116, 12 Dec 10 14:01 hwC0D3 crw-rw---- 1 root audio 116, 6 Dec 10 14:01 pcmC0D0c crw-rw---- 1 root audio 116, 5 Dec 10 19:27 pcmC0D0p crw-rw---- 1 root audio 116, 7 Dec 10 14:01 pcmC0D1p crw-rw---- 1 root audio 116, 8 Dec 10 14:01 pcmC0D2c crw-rw---- 1 root audio 116, 9 Dec 10 14:01 pcmC0D3p crw-rw---- 1 root audio 116, 10 Dec 10 14:01 pcmC0D7p crw-rw---- 1 root audio 116, 3 Dec 10 19:29 pcmC1D0p crw-rw---- 1 root audio 116, 33 Dec 10 14:01 timer CPU ------------------------------------------------------- model name : Intel(R) Celeron(R) CPU G470 @ 2.00GHz cpu MHz : 1711.718 model name : Intel(R) Celeron(R) CPU G470 @ 2.00GHz cpu MHz : 1697.812 RAM ------------------------------------------------------- MemTotal: 1859156 kB SwapTotal: 0 kB Hardware -------------------------------------------------- 00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09) 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) Interupts ------------------------------------------------- CPU0 CPU1 0: 14 0 IO-APIC-edge timer 1: 2 0 IO-APIC-edge i8042 8: 1 0 IO-APIC-edge rtc0 9: 3 0 IO-APIC-fasteoi acpi 12: 4 0 IO-APIC-edge i8042 16: 15139141 0 IO-APIC-fasteoi ehci_hcd:usb1 23: 37 0 IO-APIC-fasteoi ehci_hcd:usb2 40: 386062 0 PCI-MSI-edge i915 41: 9653 0 PCI-MSI-edge ahci 42: 815667 0 PCI-MSI-edge eth0 43: 0 0 PCI-MSI-edge eth1 44: 0 0 PCI-MSI-edge xhci_hcd 45: 19140 0 PCI-MSI-edge snd_hda_intel NMI: 48 32 Non-maskable interrupts LOC: 4562425 1085862 Local timer interrupts SPU: 0 0 Spurious interrupts PMI: 48 32 Performance monitoring interrupts IWI: 0 0 IRQ work interrupts RTR: 1 0 APIC ICR read retries RES: 21236 17958 Rescheduling interrupts CAL: 136 121 Function call interrupts TLB: 2095 2460 TLB shootdowns ERR: 0 MIS: 0 Thanks for helping me understand this problem. -- Mobidyc

3 4

[alsa-devel] [RFC PATCH 0/6] Those patches is used for dw_hdmi audio.
by Yakir Yang 16 Dec '14

16 Dec '14

rk3288 use Designware HDMI IP, Designware HDMI supports four interfaces to config hdmi audio(I2S, S/PDIF, Generic Parallel Audio, AHB Audio DMA), but rk3288 only support two ways to config hdmi audio(I2S, S/PDIF), So we take I2S as hdmi audio operation interfaces, and then treat hdmi audio as an standard ALSA devices(only got platback function). Four things we have done: 1. modify the audio part of dw_hdmi drivers for rk3288 platform specific. 2. add codec driver for hdmi audio, and creat codec device dynamically in hdmi driver probe function, using platform_register_device_full way, so that hdmi codec driver can connected with hdmi driver. 3. add sound driver for hdmi audio, creat hdmi audio sound card. 4. add dt-bings file and add hdmi_audio node to corresponding dt file. [Explanation]: those patch is depend on patch by Andy Yan drm: bridge/dw_hdmi: add rockchip rk3288 support see https://patchwork.kernel.org/patch/5442361/ Yakir Yang (6): drm: bridge/dw_hdmi: add hdmi audio config interfaces drm: bridge/bridge: add support for rk3288 hdmi-audio ASoC: dw-hdmi-audio: add codec driver for dw hdmi audio ASoC: rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for Rockchip hdmi-audio ARM: dts: rockchip: add hdmi audio nodes .../sound/rockchip,rockchip-hdmi-audio.txt | 16 + arch/arm/boot/dts/rk3288-evb.dtsi | 11 + arch/arm/boot/dts/rk3288.dtsi | 8 + drivers/gpu/drm/bridge/dw_hdmi.c | 32 +- include/drm/bridge/dw_hdmi.h | 12 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 371 +++++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 82 +++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 208 ++++++++++++ 12 files changed, 756 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-hdmi-audio.txt create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

6 18

[alsa-devel] [PATCH] ASoC: Intel: Use proper print format for unsigned integers
by Rastislav Barlik 16 Dec '14

16 Dec '14

Use %u instead of %d format for printing unsigned integers. Also use unsigned integer for count and pages variables. Signed-off-by: Rastislav Barlik <barlik(a)zoho.com> --- sound/soc/intel/sst-haswell-dsp.c | 11 ++++++----- sound/soc/intel/sst-haswell-ipc.c | 30 +++++++++++++++--------------- sound/soc/intel/sst-haswell-pcm.c | 21 +++++++++++---------- 3 files changed, 32 insertions(+), 30 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4b6c163..96c7201 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -88,7 +88,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, struct sst_module *mod; struct sst_module_data block_data; struct sst_module_template template; - int count; + u32 count; void __iomem *ram; /* TODO: allowed module types need to be configurable */ @@ -127,7 +127,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, if (block->size <= 0) { dev_err(dsp->dev, - "error: block %d size invalid\n", count); + "error: block %u size invalid\n", count); sst_module_free(mod); return -EINVAL; } @@ -156,7 +156,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block_data.data = (void *)block + sizeof(*block); block_data.data_offset = block_data.data - fw->dma_buf; - dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + dev_dbg(dsp->dev, "copy firmware block %u type 0x%x " "size 0x%x ==> ram %p offset 0x%x\n", count, block->type, block->size, ram, block->ram_offset); @@ -175,7 +175,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) struct fw_module_header *module; struct sst_dsp *dsp = sst_fw->dsp; struct sst_hsw *hsw = sst_fw->private; - int ret, count; + int ret; + u32 count; /* Read the header information from the data pointer */ header = (struct fw_header *)sst_fw->dma_buf; @@ -198,7 +199,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* module */ ret = hsw_parse_module(dsp, sst_fw, module); if (ret < 0) { - dev_err(dsp->dev, "error: invalid module %d\n", count); + dev_err(dsp->dev, "error: invalid module %u\n", count); return ret; } module = (void *)module + sizeof(*module) + module->mod_size; diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index b629151..440cadd 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -577,7 +577,7 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) offset = (header & 0x1FFFFFFF) << 3; - dev_dbg(hsw->dev, "ipc: DSP is ready 0x%8.8x offset %d\n", + dev_dbg(hsw->dev, "ipc: DSP is ready 0x%8.8x offset %u\n", header, offset); /* copy data from the DSP FW ready offset */ @@ -626,7 +626,7 @@ static void hsw_notification_work(struct work_struct *work) stream->reply.stream_hw_id); sst_dsp_inbox_read(hsw->dsp, glitch, sizeof(*glitch)); - dev_err(hsw->dev, "glitch %d pos 0x%x write pos 0x%x\n", + dev_err(hsw->dev, "glitch %u pos 0x%x write pos 0x%x\n", glitch->glitch_type, glitch->present_pos, glitch->write_pos); break; @@ -878,7 +878,7 @@ static int hsw_process_notification(struct sst_hsw *hsw) case IPC_GLB_MAX_IPC_MESSAGE_TYPE: case IPC_GLB_RESTORE_CONTEXT: case IPC_GLB_SHORT_REPLY: - dev_err(hsw->dev, "error: message type %d header 0x%x\n", + dev_err(hsw->dev, "error: message type %u header 0x%x\n", type, header); break; case IPC_GLB_STREAM_MESSAGE: @@ -888,7 +888,7 @@ static int hsw_process_notification(struct sst_hsw *hsw) handled = hsw_log_message(hsw, header); break; default: - dev_err(hsw->dev, "error: unexpected type %d hdr 0x%8.8x\n", + dev_err(hsw->dev, "error: unexpected type %u hdr 0x%8.8x\n", type, header); break; } @@ -977,7 +977,7 @@ int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", + dev_err(hsw->dev, "error: can't unmute stream %u channel %u\n", stream->reply.stream_hw_id, channel); return ret; } @@ -996,7 +996,7 @@ int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, stream->mute_volume[channel]); if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", + dev_err(hsw->dev, "error: can't unmute stream %u channel %u\n", stream->reply.stream_hw_id, channel); return ret; } @@ -1076,7 +1076,7 @@ int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", + dev_err(hsw->dev, "error: failed to unmute mixer channel %u\n", channel); return ret; } @@ -1092,7 +1092,7 @@ int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, hsw->mixer_info.volume_register_address[channel]); if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", + dev_err(hsw->dev, "error: failed to unmute mixer channel %u\n", channel); return ret; } @@ -1219,7 +1219,7 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) ret = ipc_tx_message_wait(hsw, header, &stream->free_req, sizeof(stream->free_req), NULL, 0); if (ret < 0) { - dev_err(hsw->dev, "error: free stream %d failed\n", + dev_err(hsw->dev, "error: free stream %u failed\n", stream->free_req.stream_id); return -EAGAIN; } @@ -1517,7 +1517,7 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, stream->reply.stream_hw_id, wait); if (ret < 0) - dev_err(hsw->dev, "error: failed to pause stream %d\n", + dev_err(hsw->dev, "error: failed to pause stream %u\n", stream->reply.stream_hw_id); return ret; @@ -1533,7 +1533,7 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, stream->reply.stream_hw_id, wait); if (ret < 0) - dev_err(hsw->dev, "error: failed to resume stream %d\n", + dev_err(hsw->dev, "error: failed to resume stream %u\n", stream->reply.stream_hw_id); return ret; @@ -1551,7 +1551,7 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) while (stream->running && tries--) msleep(1); if (!tries) { - dev_err(hsw->dev, "error: reset stream %d still running\n", + dev_err(hsw->dev, "error: reset stream %u still running\n", stream->reply.stream_hw_id); return -EINVAL; } @@ -1561,7 +1561,7 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) ret = sst_hsw_stream_operations(hsw, IPC_STR_RESET, stream->reply.stream_hw_id, 1); if (ret < 0) - dev_err(hsw->dev, "error: failed to reset stream %d\n", + dev_err(hsw->dev, "error: failed to reset stream %u\n", stream->reply.stream_hw_id); return ret; } @@ -1609,7 +1609,7 @@ int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, sizeof(stream->wpos)); if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", + dev_err(hsw->dev, "error: stream %u set position %u failed\n", stream->reply.stream_hw_id, position); return ret; @@ -1670,7 +1670,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, dx->mem_info[item].size, dx->mem_info[item].source); } - dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", + dev_dbg(hsw->dev, "ipc: got %u entry numbers for state %d\n", dx->entries_no, state); memcpy(&hsw->dx, dx, sizeof(*dx)); diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 4df867c..7367326 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -278,11 +278,12 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, unsigned char *dma_area, size_t size, int pcm) { struct snd_dma_buffer *dmab = snd_pcm_get_dma_buf(substream); - int i, pages, stream = substream->stream; + int stream = substream->stream; + unsigned int i, pages; pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %u\n", dma_area, size, pages); for (i = 0; i < pages; i++) { @@ -290,7 +291,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, u32 pfn = snd_sgbuf_get_addr(dmab, i * PAGE_SIZE) >> PAGE_SHIFT; u32 *pg_table; - dev_dbg(rtd->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn); + dev_dbg(rtd->dev, "pfn i %u idx %u pfn %x\n", i, idx, pfn); pg_table = (u32 *)(pdata->dmab[pcm][stream].area + idx); @@ -388,7 +389,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); ret = sst_hsw_stream_set_rate(hsw, pcm_data->stream, rate); if (ret < 0) { - dev_err(rtd->dev, "error: could not set rate %d\n", rate); + dev_err(rtd->dev, "error: could not set rate %u\n", rate); return ret; } @@ -417,14 +418,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, ret = sst_hsw_stream_set_bits(hsw, pcm_data->stream, bits); if (ret < 0) { - dev_err(rtd->dev, "error: could not set bits %d\n", bits); + dev_err(rtd->dev, "error: could not set bits %u\n", bits); return ret; } /* we only support stereo atm */ channels = params_channels(params); if (channels != 2) { - dev_err(rtd->dev, "error: invalid channels %d\n", channels); + dev_err(rtd->dev, "error: invalid channels %u\n", channels); return -EINVAL; } @@ -434,14 +435,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, ret = sst_hsw_stream_set_channels(hsw, pcm_data->stream, channels); if (ret < 0) { - dev_err(rtd->dev, "error: could not set channels %d\n", + dev_err(rtd->dev, "error: could not set channels %u\n", channels); return ret; } ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (ret < 0) { - dev_err(rtd->dev, "error: could not allocate %d bytes for PCM %d\n", + dev_err(rtd->dev, "error: could not allocate %u bytes for PCM %d\n", params_buffer_bytes(params), ret); return ret; } @@ -558,7 +559,7 @@ static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) pos = frames_to_bytes(runtime, (runtime->control->appl_ptr % runtime->buffer_size)); - dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + dev_dbg(rtd->dev, "PCM: App pointer %u bytes\n", pos); /* let alsa know we have play a period */ snd_pcm_period_elapsed(substream); @@ -580,7 +581,7 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) offset = bytes_to_frames(runtime, position); ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream); - dev_dbg(rtd->dev, "PCM: DMA pointer %du bytes, pos %llu\n", + dev_dbg(rtd->dev, "PCM: DMA pointer %u bytes, pos %llu\n", position, ppos); return offset; } -- 2.1.3

1 0

[alsa-devel] [PATCH] ALSA: hda/hdmi - apply Haswell fix-ups to Skylake display codec
by libin.yang＠intel.com 16 Dec '14

16 Dec '14

From: Libin Yang <libin.yang(a)intel.com> Skylake and Haswell have the same behavior on display audio. So this patch applys Haswell fix-ups to Skylake. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 95129c2..5f13d2d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -47,7 +47,9 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) #define is_broadwell(codec) ((codec)->vendor_id == 0x80862808) -#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec)) +#define is_skylake(codec) ((codec)->vendor_id == 0x80862809) +#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ + || is_skylake(codec)) #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) -- 1.9.1

2 1

[alsa-devel] [PATCH -next] ASoC: Intel: fix return value check in sst_acpi_probe()
by weiyj_lk＠163.com 15 Dec '14

15 Dec '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function platform_device_register_data() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/intel/sst/sst_acpi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 31124aa..f59972a 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -277,16 +277,16 @@ int sst_acpi_probe(struct platform_device *pdev) dev_dbg(dev, "ACPI device id: %x\n", dev_id); plat_dev = platform_device_register_data(dev, mach->pdata->platform, -1, NULL, 0); - if (plat_dev == NULL) { + if (IS_ERR(plat_dev)) { dev_err(dev, "Failed to create machine device: %s\n", mach->pdata->platform); - return -ENODEV; + return PTR_ERR(plat_dev); } /* Create platform device for sst machine driver */ mdev = platform_device_register_data(dev, mach->machine, -1, NULL, 0); - if (mdev == NULL) { + if (IS_ERR(mdev)) { dev_err(dev, "Failed to create machine device: %s\n", mach->machine); - return -ENODEV; + return PTR_ERR(mdev); } ret = sst_alloc_drv_context(&ctx, dev, dev_id);

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt5677: add REGMAP_I2C and REGMAP_IRQ dependency
by Ben Zhang 15 Dec '14

15 Dec '14

The codec driver uses regmap to do i2c read/write. The codec driver started to use REGMAP_IRQ since: 5e3363ad1b7b2e1f197a3f56b01e21cb155ad454 ASoC: rt5677: add GPIO IRQ support Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6f21a76..60f9425 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -503,6 +503,8 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate + select REGMAP_I2C + select REGMAP_IRQ config SND_SOC_RT5677_SPI tristate -- 2.2.0.rc0.207.ga3a616c

3 7

[alsa-devel] [PATCH] ALSA: hda - add codec ID for Skylake display audio codec
by libin.yang＠intel.com 15 Dec '14

15 Dec '14

From: Libin Yang <libin.yang(a)intel.com> This patch adds codec ID (0x80862809) and module alias for Skylake display codec. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9dc9cf8..95129c2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3365,6 +3365,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862809, .name = "Skylake HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, @@ -3425,6 +3426,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862808"); +MODULE_ALIAS("snd-hda-codec-id:80862809"); MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:80862882"); MODULE_ALIAS("snd-hda-codec-id:80862883"); -- 1.9.1

2 1

[alsa-devel] [patch 1/2] ALSA: oxfw: some signedness bugs
by Dan Carpenter 15 Dec '14

15 Dec '14

This code tends to use unsigned variables by default and it causes signedness bugs when we use negative variables for error handling. The "i" and "j" variables are used to iterated over small positive values and so they should be type "int". The "len" variable doesn't *need* to be signed but it should be signed to make the code easier to read and audit. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c index 604808e..8ba4f9f2 100644 --- a/sound/firewire/oxfw/oxfw-proc.c +++ b/sound/firewire/oxfw/oxfw-proc.c @@ -15,7 +15,7 @@ static void proc_read_formation(struct snd_info_entry *entry, struct snd_oxfw_stream_formation formation, curr; u8 *format; char flag; - unsigned int i, err; + int i, err; /* Show input. */ err = snd_oxfw_stream_get_current_formation(oxfw, diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index b77cf80..bda845a 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -61,7 +61,8 @@ static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s, u8 **formats; struct snd_oxfw_stream_formation formation; enum avc_general_plug_dir dir; - unsigned int i, err, len; + unsigned int len; + int i, err; if (s == &oxfw->tx_stream) { formats = oxfw->tx_stream_formats; diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9bc556b..67ade07 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -19,7 +19,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, .min = UINT_MAX, .max = 0, .integer = 1 }; struct snd_oxfw_stream_formation formation; - unsigned int i, err; + int i, err; for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { if (formats[i] == NULL) @@ -47,7 +47,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, const struct snd_interval *r = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_oxfw_stream_formation formation; - unsigned int i, j, err; + int i, j, err; unsigned int count, list[SND_OXFW_STREAM_FORMAT_ENTRIES] = {0}; count = 0; @@ -80,7 +80,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats) { struct snd_oxfw_stream_formation formation; - unsigned int i, err; + int i, err; hw->channels_min = UINT_MAX; hw->channels_max = 0;

3 8

[alsa-devel] [PATCH v2 0/6] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 15 Dec '14

15 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries Andrew Jackson (6): ASoC: dwc: Remove unnecessary debug messages and tests ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Add documentation for I2S DT .../devicetree/bindings/sound/designware-i2s.txt | 32 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 294 ++++++++++++++------ 3 files changed, 248 insertions(+), 79 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

4 14

[alsa-devel] [PATCH v2] ASoC: rt5670: set platform data by dmi
by Bard Liao 15 Dec '14

15 Dec '14

This patch set specific data according to dmi data. Signed-off-by: Jin, Yao <yao.jin(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * rename dmi_platform_data to dmi_platform_intel_braswell --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8bf3a56..09cf100 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_intel_braswell[] __initconst = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_intel_braswell)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH 1/3] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 15 Dec '14

15 Dec '14

From: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..23831a5 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2255,10 +2255,19 @@ static const struct i2c_device_id wm8904_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); +static const struct of_device_id wm8904_of_match[] = { + { .compatible = "wlf,wm8904", }, + { .compatible = "wlf,wm8912", }, + { .compatible = "wlf,wm8918", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); + static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = wm8904_of_match, }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.1.0.24.g4109c28

2 4

[alsa-devel] [patch 2/2] ALSA: oxfw: fix detect_loud_models() return value
by Dan Carpenter 14 Dec '14

14 Dec '14

This code causes a static checker warning: sound/firewire/oxfw/oxfw.c:46 detect_loud_models() warn: signedness bug returning '(-2)' The detect_loud_models() function should return false on falure, so that we don't try to set up the loud code for hardware that doesn't support it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cf1d0b5..60e5cad 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -43,7 +43,7 @@ static bool detect_loud_models(struct fw_unit *unit) err = fw_csr_string(unit->directory, CSR_MODEL, model, sizeof(model)); if (err < 0) - return err; + return false; for (i = 0; i < ARRAY_SIZE(models); i++) { if (strcmp(models[i], model) == 0)

3 2

[alsa-devel] [PATCH] ASoC: rt5670: set platform data by dmi
by Bard Liao 12 Dec '14

12 Dec '14

This patch set specific data according to dmi data. Signed-off-by: Jin Yao <yao.jin(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8bf3a56..6074f04 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_data[] __initconst = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_data)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] linux-next: Tree for Dec 12 (sound/soc/codecs/rt5640)
by Randy Dunlap 12 Dec '14

12 Dec '14

On 12/11/14 23:35, Stephen Rothwell wrote: > Hi all, > > Please do not add any code destined for v3.20 to your linux-next included > trees/branches until after v3.19-rc1 is released. > > Changes since 20141211: on i386: When CONFIG_ACPI is not enabled & CONFIG_I2C is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH) selects SND_SST_IPC_ACPI which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) ../sound/soc/codecs/rt5640.c:2252:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(rt5640_i2c_driver); ^ ../sound/soc/codecs/rt5640.c:2252:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] ../sound/soc/codecs/rt5640.c:2252:1: warning: parameter names (without types) in function declaration [enabled by default] ../sound/soc/codecs/rt5640.c:2241:26: warning: 'rt5640_i2c_driver' defined but not used [-Wunused-variable] static struct i2c_driver rt5640_i2c_driver = { ^ cc1: some warnings being treated as errors Full randconfig file is attached. Can (should) SND_SOC_INTEL_BYTCR_RT5640_MACH depend on I2C? -- ~Randy

1 0

[alsa-devel] [PATCH] ASoC: soc-dapm: Fix comparison of pointers
by Rasmus Villemoes 12 Dec '14

12 Dec '14

The idiom 'return a - b;' often used in comparison functions is wrong unless one is certain the values being compared lie in a sufficiently small range. In this case dapm_seq_compare would also return 0 if the ->dapm pointers happened to differ by a multiple of 2^32. Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- Notes: I don't know if the other occurences of the idiom in this function are ok; that depends on whether the values always lie in [0, 2^31-1]. sound/soc/soc-dapm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..a894a7c71557 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1255,8 +1255,10 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, } if (a->reg != b->reg) return a->reg - b->reg; - if (a->dapm != b->dapm) - return (unsigned long)a->dapm - (unsigned long)b->dapm; + if (a->dapm < b->dapm) + return -1; + if (a->dapm > b->dapm) + return 1; return 0; } -- 2.1.3

2 3

[alsa-devel] [PATCH] ASoC: wm8750: Make VREF a VMID instead of OUTPUT widget
by Lars-Peter Clausen 12 Dec '14

12 Dec '14

Commit 04489eeb02a4 ("ALSA: wm8750: add missing VREF output") added the previously missing VREF widget. And while adding the widget as OUTPUT widget silences the error of the missing widget it should have rather been a VMID widget. The VREF widget is only used internally as a source for the VREF input of the Out3 Mux. Selecting that input means that the path should be kept powered up at the VREF/VMID voltage, but that there is no actual audio signal. Using a VMID widget will do this, whereas using a OUTPUT widget will simply cause the path to completely power down. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8750.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index f6847fd..eb0a164 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -323,7 +323,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT2"), SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), - SND_SOC_DAPM_OUTPUT("VREF"), + SND_SOC_DAPM_VMID("VREF"), SND_SOC_DAPM_INPUT("LINPUT1"), SND_SOC_DAPM_INPUT("LINPUT2"), -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: samsung: Fix non-DT use of I2S controller
by Mark Brown 12 Dec '14

12 Dec '14

The changes in commit a5a56871f804e (ASoC: samsung: add support for exynos7 I2S controller) introduce a new variant_regs structure in the driver data which is now mandatory for accessing registers. Unfortunately this is only hooked up for DT platforms so non-DT platforms like my primary development platform for audio are broken by this change and crash on boot. Since the only non-DT user of these device is s3c64xx fix this by making the standard samsung-i2s device be of type I2Sv3 and add a new I2Sv4 name to the platform data section, currently using the I2Sv5 information which should be about right. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index e1ace5270a8b..25919e090983 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1418,10 +1418,13 @@ static const struct samsung_i2s_dai_data samsung_dai_type_sec = { static struct platform_device_id samsung_i2s_driver_ids[] = { { .name = "samsung-i2s", - .driver_data = (kernel_ulong_t)&samsung_dai_type_pri, + .driver_data = (kernel_ulong_t)&i2sv3_dai_type, }, { .name = "samsung-i2s-sec", .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, + }, { + .name = "samsung-i2sv4", + .driver_data = (kernel_ulong_t)&i2sv5_dai_type_i2s1, }, {}, }; -- 2.1.3

2 6

[alsa-devel] [BUG] dmix reports channel layouts that are not supported
by wm4 11 Dec '14

11 Dec '14

Opening a dmix device (i.e. "default" on most distros), and calling snd_pcm_query_chmaps() on it, returns some surround audio channel maps, for example 5.1. Trying to set it with snd_pcm_set_chmap() will fail with ENXIO. Only setting the stereo channel map succeeds, and I suppose dmix supports stereo only. I think this should be considered a bug. dmix probably merely passes through the channel layout of the underlying device, even though it supports stereo only - this behavior makes no sense at all. Any device should strictly report supported channel maps only. To make matters worse, if you intend to play 5.1, and dmix fails in the manner described above, then it will report "FL FR NA NA NA NA" as channel map. This would require the application to pad stereo with 4 channels of silence, which requires extra code and is generally just stupid.

1 0

Re: [alsa-devel] [Patch v2 02/11] ASoC: qcom: Add device tree binding docs
by Kenneth Westfield 11 Dec '14

11 Dec '14

On Tue, December 9, 2014 7:37 am, Mark Brown wrote: > On Mon, Dec 08, 2014 at 02:01:04PM -0800, Kenneth Westfield wrote: >> From: Kenneth Westfield <kwestfie(a)codeaurora.org> >> Add documentation to the sound directory of the >> device-tree bindings for IPQ806x audio drivers. > Please don't send multiple separate changes in a single patch. One change per patch. In an effort to reduce the number of patches, would it be possible to package the DT doc with its respective driver in one patch? Or does the DT doc need to be in a patch by itself? I've seen examples of both. Also, I will combine the partial addition of each driver to its respective Kconfig/Makefile entries. For this, I believe I need to start with an empty Kconfig/Makefile in the first patch? -- Kenneth Westfield Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH 1/4] sound: oss: Remove SoftOSS documentation
by Jarkko Nikula 11 Dec '14

11 Dec '14

SoftOSS has been removed before kernel git history but not this documentation part. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Documentation/sound/oss/README.OSS | 46 -------------------------------------- 1 file changed, 46 deletions(-) diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS index 4be259428a1c..ad261dd602ce 100644 --- a/Documentation/sound/oss/README.OSS +++ b/Documentation/sound/oss/README.OSS @@ -516,52 +516,6 @@ Yamaha OPL3-SA1 use the standard MSS, MPU401 and OPL3 options with these chips and to activate the card using isapnptools. -4Front Technologies SoftOSS - - SoftOSS is a software based wave table emulation which works with - any 16 bit stereo sound card. Due to its nature a fast CPU is - required (P133 is minimum). Although SoftOSS does _not_ use MMX - instructions it has proven out that recent processors (which appear - to have MMX) perform significantly better with SoftOSS than earlier - ones. For example a P166MMX beats a PPro200. SoftOSS should not be used - on 486 or 386 machines. - - The amount of CPU load caused by SoftOSS can be controlled by - selecting the CONFIG_SOFTOSS_RATE and CONFIG_SOFTOSS_VOICES - parameters properly (they will be prompted by make config). It's - recommended to set CONFIG_SOFTOSS_VOICES to 32. If you have a - P166MMX or faster (PPro200 is not faster) you can set - CONFIG_SOFTOSS_RATE to 44100 (kHz). However with slower systems it - recommended to use sampling rates around 22050 or even 16000 kHz. - Selecting too high values for these parameters may hang your - system when playing MIDI files with hight degree of polyphony - (number of concurrently playing notes). It's also possible to - decrease CONFIG_SOFTOSS_VOICES. This makes it possible to use - higher sampling rates. However using fewer voices decreases - playback quality more than decreasing the sampling rate. - - SoftOSS keeps the samples loaded on the system's RAM so much RAM is - required. SoftOSS should never be used on machines with less than 16 MB - of RAM since this is potentially dangerous (you may accidentally run out - of memory which probably crashes the machine). - - SoftOSS implements the wave table API originally designed for GUS. For - this reason all applications designed for GUS should work (at least - after minor modifications). For example gmod/xgmod and playmidi -g are - known to work. - - To work SoftOSS will require GUS compatible - patch files to be installed on the system (in /dos/ultrasnd/midi). You - can use the public domain MIDIA patchset available from several ftp - sites. - - ********************************************************************* - IMPORTANT NOTICE! The original patch set distributed with the Gravis - Ultrasound card is not in public domain (even though it's available from - some FTP sites). You should contact Voice Crystal (www.voicecrystal.com) - if you like to use these patches with SoftOSS included in OSS/Free. - ********************************************************************* - PSS based cards (AD1848 + ADSP-2115 + Echo ESC614 ASIC) Analog Devices and Echo Speech have together defined a sound card architecture based on the above chips. The DSP chip is used -- 2.1.3

2 4

[alsa-devel] [PATCH v3 0/2] Disable the ThinkPad HW mute/level control when possible
by Andy Lutomirski 11 Dec '14

11 Dec '14

This series is being resurrected after several years, and it's quite different from before, so I'm restarting the version numbers :) ThinkPad volume and mute controls are a mess. For whatever reason, ThinkPads have mute buttons that send KEY_MUTE *and* control an invisible-to-ALSA mute switch. Some of them have volume controls that interact with this switch as well. This is a perennial source of problems. On most ThinkPads, if you press mute and then unmute using GUI controls, you have no sound, because userspace and ALSA state gets out of sync with the hardware switch. There's a separate "sound card" that exposes the hardware switch, but userspace code generally doesn't understand that. There are already a few _OSI(Linux) overrides to turn all the hardware buttons into regular buttons. Rather than quirking ACPI everywhere, just teach thinkpad-acpi to program the buttons for full software control and to disable hardware controls. That allows us to remove the ACPI quirks and have normal mute controls. This approach should be much simpler than adding even more kludgey ALSA integration for questionable gain. Tested on an X200s (with latching mute by default) and an X220 (with a mute light and toggle mute by default). Everything works as expected. Changes from v2: - Don't try to change the mute mode on IBM ThinkPads. - Further minor cleanups. - Try to restore the mode on module unload. Changes from v1: - Simplified the code a bit. - Improved suspend/hibernate behavior. Andy Lutomirski (2): thinkpad-acpi: Try to use full software mute control acpi: Remove _OSI(Linux) for ThinkPads drivers/acpi/blacklist.c | 54 ---------------- drivers/platform/x86/thinkpad_acpi.c | 116 ++++++++++++++++++++++++++++++++--- 2 files changed, 106 insertions(+), 64 deletions(-) -- 1.9.3

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Re: [alsa-devel] Laptop Asus Z99He internal speaker not working until I enable EAPD with HDA-Analizer
by Takashi Iwai 11 Dec '14

11 Dec '14

At Thu, 11 Dec 2014 11:53:16 +0300, Dmitry V. Zimin wrote: > > This patch will be applied only to the latest kernel version or to all versions of the long-term ? The commit contains a Cc to stable kernel, so that the stable kernel maintainers will pick it up. Which one to apply depends on the stable kernel maintainer. Takashi > > > Wed, 10 Dec 2014 14:45:58 +0100 от Takashi Iwai <tiwai(a)suse.de>: > >At Wed, 10 Dec 2014 15:28:09 +0300, > >Dmitry V. Zimin wrote: > >> > >> > >> Yes, it works. Thank you so much! > > > >Good to hear, the patch is queued now. > > > > > >Takashi > > > >> > >> > >> Tue, 09 Dec 2014 20:01:36 +0100 от Takashi Iwai < tiwai(a)suse.de >: > >> >At Tue, 09 Dec 2014 18:01:44 +0300, > >> >Dmitry V. Zimin wrote: > >> >> > >> >> Hello! > >> >> > >> >> I have laptop Asus Z99He and internal speaker doesn't work until I enable EAPD through HDA-Analyzer by hand. After connect and disconnect external headphones EAPD setting reset again. > >> >> How to enable it permanently? > >> >> model=eapd, laptop-eapd or many other does not change anything. > >> >> http://www.alsa-project.org/db/?f=df83e22e0c9a047db3778f394c1aa73376dfaf39 > >> >> > >> >> Also I try old Live CD with OpenELEC. He is working properly . http://www.alsa-project.org/db/?f=a5e75412c8230afe9516f0ebf6404afb18d9040c > >> > > >> >Could you try the patch below? > >> > > >> > > >> >Takashi > >> > > >> >-- 8< -- > >> >From: Takashi Iwai < tiwai(a)suse.de > > >> >Subject: [PATCH] ALSA: hda - Add EAPD fixup for ASUS Z99He laptop > >> > > >> >The same fixup to enable EAPD is needed for ASUS Z99He with AD1986A > >> >codec like another ASUS machine. > >> > > >> >Reported-by: Dmitry V. Zimin < pfzim(a)mail.ru > > >> >Cc: < stable(a)vger.kernel.org > > >> >Signed-off-by: Takashi Iwai < tiwai(a)suse.de > > >> >--- > >> > sound/pci/hda/patch_analog.c | 1 + > >> > 1 file changed, 1 insertion(+) > >> > > >> >diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c > >> >index 06275f8807a8..28815efeab06 100644 > >> >--- a/sound/pci/hda/patch_analog.c > >> >+++ b/sound/pci/hda/patch_analog.c > >> >@@ -332,6 +332,7 @@ static const struct hda_fixup ad1986a_fixups[] = { > >> > > >> > static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { > >> > SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), > >> >+ SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD), > >> > SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), > >> > SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), > >> > SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), > >> >-- > >> >2.2.0 > >> > > >> > >> > >> > >> [2 <text/html; utf-8 (base64)>] > >> > > >

1 0

[alsa-devel] About SNDRV_PCM_HW_PARAM_TICK_TIME
by Lars-Peter Clausen 10 Dec '14

10 Dec '14

Hi, I was trying to figure out what exactly SNDRV_PCM_HW_PARAM_TICK_TIME does, but couldn't find too much. Is it safe to say that nowadays this is unused and deprecated and the only reason it is kept around is for ABI reasons? Thanks, - Lars

2 1

Re: [alsa-devel] Laptop Asus Z99He internal speaker not working until I enable EAPD with HDA-Analizer
by Takashi Iwai 10 Dec '14

10 Dec '14

At Wed, 10 Dec 2014 15:28:09 +0300, Dmitry V. Zimin wrote: > > > Yes, it works. Thank you so much! Good to hear, the patch is queued now. Takashi > > > Tue, 09 Dec 2014 20:01:36 +0100 от Takashi Iwai <tiwai(a)suse.de>: > >At Tue, 09 Dec 2014 18:01:44 +0300, > >Dmitry V. Zimin wrote: > >> > >> Hello! > >> > >> I have laptop Asus Z99He and internal speaker doesn't work until I enable EAPD through HDA-Analyzer by hand. After connect and disconnect external headphones EAPD setting reset again. > >> How to enable it permanently? > >> model=eapd, laptop-eapd or many other does not change anything. > >> http://www.alsa-project.org/db/?f=df83e22e0c9a047db3778f394c1aa73376dfaf39 > >> > >> Also I try old Live CD with OpenELEC. He is working properly . http://www.alsa-project.org/db/?f=a5e75412c8230afe9516f0ebf6404afb18d9040c > > > >Could you try the patch below? > > > > > >Takashi > > > >-- 8< -- > >From: Takashi Iwai < tiwai(a)suse.de > > >Subject: [PATCH] ALSA: hda - Add EAPD fixup for ASUS Z99He laptop > > > >The same fixup to enable EAPD is needed for ASUS Z99He with AD1986A > >codec like another ASUS machine. > > > >Reported-by: Dmitry V. Zimin < pfzim(a)mail.ru > > >Cc: < stable(a)vger.kernel.org > > >Signed-off-by: Takashi Iwai < tiwai(a)suse.de > > >--- > > sound/pci/hda/patch_analog.c | 1 + > > 1 file changed, 1 insertion(+) > > > >diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c > >index 06275f8807a8..28815efeab06 100644 > >--- a/sound/pci/hda/patch_analog.c > >+++ b/sound/pci/hda/patch_analog.c > >@@ -332,6 +332,7 @@ static const struct hda_fixup ad1986a_fixups[] = { > > > > static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { > > SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), > >+ SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD), > > SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), > > SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), > > SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), > >-- > >2.2.0 > > > > > > [2 <text/html; utf-8 (base64)>] >

1 0

[alsa-devel] [patch] ALSA: pcxhr: NULL dereference on probe failure
by Dan Carpenter 10 Dec '14

10 Dec '14

"card" is NULL if snd_card_new() fails. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index a602930..c6092e4 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1638,7 +1638,7 @@ static int pcxhr_probe(struct pci_dev *pci, 0, &card); if (err < 0) { - dev_err(card->dev, "cannot allocate the card %d\n", i); + dev_err(&pci->dev, "cannot allocate the card %d\n", i); pcxhr_free(mgr); return err; }

2 1

[alsa-devel] [patch] ALSA: lola: NULL dereference on probe failure
by Dan Carpenter 10 Dec '14

10 Dec '14

"card" is NULL if snd_card_new() fails. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index a75c8dc..4cf4be5 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -719,7 +719,7 @@ static int lola_probe(struct pci_dev *pci, err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - dev_err(card->dev, "Error creating card!\n"); + dev_err(&pci->dev, "Error creating card!\n"); return err; }

2 1

[alsa-devel] [PATCH 00/15 v5] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 10 Dec '14

10 Dec '14

This patchset is the rest of patches to enable ALSA Dice and OXFW driver to handle more models. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/30 v4] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085092.h… - use the same functions for each of MIDI .open/.close callback (dice) - fix a comment for AV/C stream format information command (oxfw) - revert shortname template, just use model name (oxfw) I note that OXFW driver uses the shortname for MIDI substream name, which has 32 bytes, thus the string in shortname should not be so long. == Supplemental patch for userspace library This patchset adds Hwdep functionality for OXFW driver, thus userspace library needs a patch for entries of snd_hwdep_iface_t. Please apply below patch. [alsa-devel] [PATCH alsa-lib] hwdep: add OXFW driver support http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084824.h… Takashi Sakamoto (15): ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 10 +- sound/firewire/dice/Makefile | 4 +- sound/firewire/dice/dice-midi.c | 157 ++++++++ sound/firewire/dice/dice-pcm.c | 251 +++++++++---- sound/firewire/dice/dice-stream.c | 332 ++++++++++++---- sound/firewire/dice/dice-transaction.c | 5 - sound/firewire/dice/dice.c | 37 +- sound/firewire/dice/dice.h | 33 +- sound/firewire/oxfw/Makefile | 3 +- sound/firewire/oxfw/oxfw-command.c | 153 ++++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++++++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++++++++ sound/firewire/oxfw/oxfw-pcm.c | 425 +++++++++++++++------ sound/firewire/oxfw/oxfw-proc.c | 113 ++++++ sound/firewire/oxfw/oxfw-stream.c | 667 +++++++++++++++++++++++++++++++-- sound/firewire/oxfw/oxfw.c | 179 +++++++-- sound/firewire/oxfw/oxfw.h | 104 ++++- 19 files changed, 2499 insertions(+), 377 deletions(-) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c -- 2.1.0

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[alsa-devel] ALSA driver installation issue in Ubuntu 10.04 OS
by Bruce Teng 10 Dec '14

10 Dec '14

Hi Sir, I've encountered the following error when run the "alsa-utils-alsa-compile.sh" on Ubuntu 10.04 with 2.6.32 kernel OS: _____ /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 127: warning: initialization from incompatible pointer type /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 132: warning: initialization from incompatible pointer type /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: In function 'set_pages_array_wc': /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 700: error: implicit declaration of function 'set_pages_wc' /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: At top level: /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 703: error: conflicting types for 'set_pages_array_wb' /usr/src/linux-headers-2.6.32-21-generic/arch/x86/include/asm/cacheflush.h:1 44: note: previous declaration of 'set_pages_array_wb' was here /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: In function 'param_set_xint': /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 2937: warning: passing argument 2 of 'param_set_int' discards qualifiers from pointer target type include/linux/moduleparam.h:177: note: expected 'struct kernel_param *' but argument is of type 'const struct kernel_param *' make[4]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.o ] Error 1 make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda] Error 2 make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci] Error 2 make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-2.6.32-21-generic' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. _____ Could you help to look into this issue? Thanks! Bruce

1 0

[alsa-devel] Question on arm64 unaligned faults during playback
by Abhilash Kesavan 10 Dec '14

10 Dec '14

Hi, I am working on a 64-bit ARM SoC (Samsung's Exynos7) and have observed unaligned faults while testing certain sound streams with aplay. Playing a 44.1k sampling rate stream gives the following crash while 48k streams work fine. [root@(none) ~]# aplay test2_16bit_44k.wav Playing WAVE 'test2_16bit_44k.wav' : Signed 1[ 24.500724] sound: 2ch, 44100Hz, 88208bytes 6 bit Little Endian, Rate 44100 Hz, Stereo [ 24.520738] snd_pcm_lib_write_transfer:sound/core/pcm_lib.c hwbuf is ffffff8000080000, runtime->dma_area is ffffff8000080000, hwoff is 0, frames_to_bytes is 0, frames is 5513 [ 24.535661] snd_pcm_lib_write_transfer:sound/core/pcm_lib.c hwbuf is ffffff8000085624, runtime->dma_area is ffffff8000080000, hwoff is 5513, frames_to_bytes is 22052, frames is 5513 [ 24.551244] Unhandled fault: alignment fault (0x96000061) at 0xffffff8000085624 [ 24.558448] Internal error: : 96000061 [#1] PREEMPT SMP [ 24.563449] Modules linked in: [ 24.566489] CPU: 0 PID: 928 Comm: aplay Not tainted 3.16.0+ #232 [ 24.572473] task: ffffffc0a934c380 ti: ffffffc0038d4000 task.ti: ffffffc0038d4000 [ 24.579944] PC is at __copy_from_user+0x14/0x60 [ 24.584450] LR is at snd_pcm_lib_write_transfer+0xe4/0x104 [ 24.589912] pc : [<ffffffc0002d4784>] lr : [<ffffffc0004b58f0>] pstate: 20000145 [ 24.597285] sp : ffffffc0038d7c90 [ 24.600583] x29: ffffffc0038d7c90 x28: ffffffc0038c1c00 [ 24.605876] x27: ffffffc0a93eae00 x26: 0000000000001589 [ 24.611168] x25: 0000000000000000 x24: 0000000000000000 [ 24.616461] x23: 0000000000000008 x22: ffffff8000085624 [ 24.621753] x21: 0000000000005624 x20: ffffffc0038c1c00 [ 24.627046] x19: 0000000011d69620 x18: 0000007fd2e8b1c0 [ 24.632339] x17: 0000007fa8057370 x16: ffffffc000196108 [ 24.637632] x15: 232b226b00000000 x14: 617266202c333135 [ 24.642925] x13: 352073692066666f x12: 7768202c30303030 [ 24.648218] x11: 3830303030386666 x10: 6666666620736920 [ 24.653510] x9 : 616572615f616d64 x8 : 000000000000026f [ 24.658803] x7 : ffffffc000998790 x6 : 0000000000000001 [ 24.664095] x5 : 0000000000000000 x4 : 0000000011d6ec44 [ 24.669388] x3 : f459f780f2b3f6d4 x2 : 0000000000005614 [ 24.674681] x1 : 0000000011d69628 x0 : ffffff8000085624 [ 24.679973] [ 24.681451] Process aplay (pid: 928, stack limit = 0xffffffc0038d4058) [ 24.687959] Stack: (0xffffffc0038d7c90 to 0xffffffc0038d8000) [ 24.693688] 7c80: 038d7cd0 ffffffc0 004b7578 ffffffc0 [ 24.701846] 7ca0: 00001589 00000000 a93eaf50 ffffffc0 009ef000 ffffffc0 11d69620 00000000 [ 24.710001] 7cc0: 004b580c ffffffc0 004b755c ffffffc0 038d7d70 ffffffc0 004b7758 ffffffc0 [ 24.718158] 7ce0: a93eae00 ffffffc0 d2e8b438 0000007f a91eb858 ffffffc0 d2e8b438 0000007f [ 24.726313] 7d00: 40184150 00000000 d2e8b438 0000007f 00000115 00000000 0000001d 00000000 [ 24.734470] 7d20: 00960000 ffffffc0 038d4000 ffffffc0 0000000a 00000000 038d7d68 ffffffc0 [ 24.742626] 7d40: 00001589 00000000 00001589 00000000 00000000 00000000 009ef898 ffffffc0 [ 24.750782] 7d60: a8b6d800 ffffffc0 0000409b 00000000 038d7da0 ffffffc0 004b2dd0 ffffffc0 [ 24.758938] 7d80: a93eae00 ffffffc0 11d69620 00000000 11d69620 00000000 00001589 00000000 [ 24.767094] 7da0: 038d7df0 ffffffc0 004b2e10 ffffffc0 a8b6d900 ffffffc0 d2e8b438 0000007f [ 24.775249] 7dc0: a91eb858 ffffffc0 00000004 00000000 00000001 00000000 00000000 00000000 [ 24.783406] 7de0: 11d69620 00000000 00001589 00000000 038d7e00 ffffffc0 00195ecc ffffffc0 [ 24.791563] 7e00: 038d7e90 ffffffc0 0019618c ffffffc0 00000000 00000000 a8b6d900 ffffffc0 [ 24.799718] 7e20: a8b6d900 ffffffc0 00000004 00000000 40184150 00000000 00185888 ffffffc0 [ 24.807875] 7e40: a8b6d800 ffffffc0 a8b6d800 ffffffc0 ffffffff ffffffff a80f0588 0000007f [ 24.816030] 7e60: 80000000 00000000 00000015 00000000 038d7e80 ffffffc0 001a0340 ffffffc0 [ 24.824187] 7e80: 038d7e90 ffffffc0 00196148 ffffffc0 d2e8b410 0000007f 0008425c ffffffc0 [ 24.832343] 7ea0: 00000000 00000000 11d500e0 00000000 ffffffff ffffffff a805737c 0000007f [ 24.840499] 7ec0: 60000000 00000000 00000015 00000000 00000004 00000000 40184150 00000000 [ 24.848655] 7ee0: d2e8b438 0000007f a8223fbc 0000007f 00000003 00000000 00000000 00000000 [ 24.856811] 7f00: a82d46f0 0000007f 00000000 00000000 0000001d 00000000 00000120 00000000 [ 24.864967] 7f20: 01010101 01010101 00000016 00000000 00000003 00000000 11d510d0 00000000 [ 24.873123] 7f40: 00000000 00000000 00000000 232b226b a8299b80 0000007f a8057370 0000007f [ 24.881279] 7f60: d2e8b1c0 0000007f 11d691f0 00000000 11d500e0 00000000 00001589 00000000 [ 24.889435] 7f80: 11d69620 00000000 00000000 00000000 0041f7d8 00000000 00000003 00000000 [ 24.897591] 7fa0: 00000000 00000000 00000004 00000000 00000003 00000000 d2e8b410 0000007f [ 24.905748] 7fc0: a8223ff0 0000007f d2e8b410 0000007f a805737c 0000007f 60000000 00000000 [ 24.913904] 7fe0: 00000004 00000000 0000001d 00000000 d10d0579 f13ea086 786e62ff 1f2437e3 [ 24.922054] Call trace: [ 24.924488] [<ffffffc0002d4784>] __copy_from_user+0x14/0x60 [ 24.930040] [<ffffffc0004b7574>] snd_pcm_lib_write1+0x1fc/0x384 [ 24.935940] [<ffffffc0004b7754>] snd_pcm_lib_write+0x58/0x80 [ 24.941580] [<ffffffc0004b2dcc>] snd_pcm_playback_ioctl1+0x2e8/0x300 [ 24.947913] [<ffffffc0004b2e0c>] snd_pcm_playback_ioctl+0x28/0x44 [ 24.953990] [<ffffffc000195ec8>] do_vfs_ioctl+0x370/0x5b0 [ 24.959367] [<ffffffc000196188>] SyS_ioctl+0x80/0x98 [ 24.964314] Code: f1002042 540000a4 f8408423 f1002042 (f8008403) [ 24.976808] ---[ end trace 35415135d4341fc9 ]--- We are using the internal sram available for sound, for DMA buffer allocation, using the generic SRAM driver. From the above log, the buffer address offset is not 8-byte aligned and as we are using the SRAM driver which maps the memory as device memory we are getting an unaligned fault. Is it incorrect to use the generic sram driver for arm64 or am I missing something ? Regards, Abhilash

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[alsa-devel] Laptop Asus Z99He internal speaker not working until I enable EAPD with HDA-Analizer
by Dmitry V. Zimin 09 Dec '14

09 Dec '14

Hello! I have laptop Asus Z99He and internal speaker doesn't work until I enable EAPD through HDA-Analyzer by hand. After connect and disconnect external headphones EAPD setting reset again. How to enable it permanently? model=eapd, laptop-eapd or many other does not change anything. http://www.alsa-project.org/db/?f=df83e22e0c9a047db3778f394c1aa73376dfaf39 Also I try old Live CD with OpenELEC. He is working properly . http://www.alsa-project.org/db/?f=a5e75412c8230afe9516f0ebf6404afb18d9040c

2 1

[alsa-devel] [PATCH] sound / PM: Replace CONFIG_PM_RUNTIME with CONFIG_PM
by Rafael J. Wysocki 09 Dec '14

09 Dec '14

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> After commit b2b49ccbdd54 (PM: Kconfig: Set PM_RUNTIME if PM_SLEEP is selected) PM_RUNTIME is always set if PM is set, so #ifdef blocks depending on CONFIG_PM_RUNTIME may now be changed to depend on CONFIG_PM. Replace CONFIG_PM_RUNTIME with CONFIG_PM everywhere under sound/. Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- Note: This depends on commit b2b49ccbdd54 (PM: Kconfig: Set PM_RUNTIME if PM_SLEEP is selected) which is only in linux-next at the moment (via the linux-pm tree). Please let me know if it is OK to take this one into linux-pm. --- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_intel.c | 5 +---- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/pcm512x.c | 2 +- sound/soc/codecs/tas2552.c | 4 ++-- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/fsl/fsl_asrc.c | 4 ++-- sound/soc/samsung/i2s.c | 4 ++-- 12 files changed, 15 insertions(+), 18 deletions(-) Index: linux-pm/sound/pci/hda/hda_controller.c =================================================================== --- linux-pm.orig/sound/pci/hda/hda_controller.c +++ linux-pm/sound/pci/hda/hda_controller.c @@ -1676,7 +1676,7 @@ irqreturn_t azx_interrupt(int irq, void u8 sd_status; int i; -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) if (!pm_runtime_active(chip->card->dev)) return IRQ_NONE; Index: linux-pm/sound/pci/hda/hda_intel.c =================================================================== --- linux-pm.orig/sound/pci/hda/hda_intel.c +++ linux-pm/sound/pci/hda/hda_intel.c @@ -856,7 +856,7 @@ static int azx_resume(struct device *dev } #endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int azx_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -954,9 +954,6 @@ static int azx_runtime_idle(struct devic return 0; } -#endif /* CONFIG_PM_RUNTIME */ - -#ifdef CONFIG_PM static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) Index: linux-pm/sound/soc/codecs/cs35l32.c =================================================================== --- linux-pm.orig/sound/soc/codecs/cs35l32.c +++ linux-pm/sound/soc/codecs/cs35l32.c @@ -550,7 +550,7 @@ static int cs35l32_i2c_remove(struct i2c return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int cs35l32_runtime_suspend(struct device *dev) { struct cs35l32_private *cs35l32 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/cs42xx8.c =================================================================== --- linux-pm.orig/sound/soc/codecs/cs42xx8.c +++ linux-pm/sound/soc/codecs/cs42xx8.c @@ -537,7 +537,7 @@ err_enable: } EXPORT_SYMBOL_GPL(cs42xx8_probe); -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int cs42xx8_runtime_resume(struct device *dev) { struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/max98090.c =================================================================== --- linux-pm.orig/sound/soc/codecs/max98090.c +++ linux-pm/sound/soc/codecs/max98090.c @@ -2492,7 +2492,7 @@ static int max98090_i2c_remove(struct i2 return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/pcm512x.c =================================================================== --- linux-pm.orig/sound/soc/codecs/pcm512x.c +++ linux-pm/sound/soc/codecs/pcm512x.c @@ -517,7 +517,7 @@ void pcm512x_remove(struct device *dev) } EXPORT_SYMBOL_GPL(pcm512x_remove); -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int pcm512x_suspend(struct device *dev) { struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/tas2552.c =================================================================== --- linux-pm.orig/sound/soc/codecs/tas2552.c +++ linux-pm/sound/soc/codecs/tas2552.c @@ -115,7 +115,7 @@ static const struct snd_soc_dapm_route t {"ClassD", NULL, "PLL"}, }; -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) { u8 cfg1_reg; @@ -264,7 +264,7 @@ static int tas2552_mute(struct snd_soc_d return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int tas2552_runtime_suspend(struct device *dev) { struct tas2552_data *tas2552 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/wm2200.c =================================================================== --- linux-pm.orig/sound/soc/codecs/wm2200.c +++ linux-pm/sound/soc/codecs/wm2200.c @@ -2440,7 +2440,7 @@ static int wm2200_i2c_remove(struct i2c_ return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int wm2200_runtime_suspend(struct device *dev) { struct wm2200_priv *wm2200 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/wm5100.c =================================================================== --- linux-pm.orig/sound/soc/codecs/wm5100.c +++ linux-pm/sound/soc/codecs/wm5100.c @@ -2664,7 +2664,7 @@ static int wm5100_i2c_remove(struct i2c_ return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int wm5100_runtime_suspend(struct device *dev) { struct wm5100_priv *wm5100 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/wm8962.c =================================================================== --- linux-pm.orig/sound/soc/codecs/wm8962.c +++ linux-pm/sound/soc/codecs/wm8962.c @@ -3782,7 +3782,7 @@ static int wm8962_i2c_remove(struct i2c_ return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int wm8962_runtime_resume(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/fsl/fsl_asrc.c =================================================================== --- linux-pm.orig/sound/soc/fsl/fsl_asrc.c +++ linux-pm/sound/soc/fsl/fsl_asrc.c @@ -928,7 +928,7 @@ static int fsl_asrc_probe(struct platfor return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); @@ -954,7 +954,7 @@ static int fsl_asrc_runtime_suspend(stru return 0; } -#endif /* CONFIG_PM_RUNTIME */ +#endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP static int fsl_asrc_suspend(struct device *dev) Index: linux-pm/sound/soc/samsung/i2s.c =================================================================== --- linux-pm.orig/sound/soc/samsung/i2s.c +++ linux-pm/sound/soc/samsung/i2s.c @@ -1113,7 +1113,7 @@ static inline const struct samsung_i2s_d platform_get_device_id(pdev)->driver_data; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int i2s_runtime_suspend(struct device *dev) { struct i2s_dai *i2s = dev_get_drvdata(dev); @@ -1131,7 +1131,7 @@ static int i2s_runtime_resume(struct dev return 0; } -#endif /* CONFIG_PM_RUNTIME */ +#endif /* CONFIG_PM */ static int samsung_i2s_probe(struct platform_device *pdev) {

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[alsa-devel] [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 09 Dec '14

09 Dec '14

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This set of patches adds support for audio on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This encapsulates the MI2S port, which is what these drivers are configured to use. The I2S pins are connected to an external DAC/amp chip. In addition, a single GPIO is connected to the same DAC/amp, which gives the SOC enable/disable control. The specific drivers added are: - a Codec DAI driver that controls the SOC external pins - a CPU DAI driver for controlling the LPASS-LPAIF block - a PCM MI2S platform driver - a machine driver that ties the three drivers together Corresponding additions to the device tree for the ipq806x and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. == Updates from my previous post: [PATCH 00/9] ASoC: QCOM: Add support for ipq806x SOC http://thread.gmane.org/gmane.linux.ports.arm.msm/10701 - remove the native LPAIF driver, and move its functionality to the CPU DAI driver - add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver) - use devm_* and dev_* where possible - ISR only handles relevant DMA channel now - update device tree documentation to reflect changes - general code cleanup == Additional patch series upon which this patch series has device-tree dependencies (lcc) [PATCH v2 0/8] qcom audio clock control drivers http://thread.gmane.org/gmane.linux.ports.arm.msm/10793 Kenneth Westfield (11): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Add device tree binding docs ASoC: ipq806x: add LPAIF header file ASoC: codec: Add Maxim codec driver ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add I2S PCM platform driver ASoC: ipq806x: Add machine driver for IPQ806X SOC ASoC: codec: Add ability to build QCOM codec ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../bindings/sound/qcom,ipq806x-snd-card.txt | 25 ++ .../bindings/sound/qcom,lpass-cpu-dai.txt | 32 ++ .../bindings/sound/qcom,lpass-pcm-mi2s.txt | 12 + .../bindings/sound/qcom,max98357a-codec.txt | 23 ++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 33 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 267 ++++++++++++ sound/soc/qcom/Kconfig | 25 ++ sound/soc/qcom/Makefile | 11 + sound/soc/qcom/ipq806x.c | 127 ++++++ sound/soc/qcom/lpass-cpu-mi2s.c | 374 +++++++++++++++++ sound/soc/qcom/lpass-cpu-mi2s.h | 48 +++ sound/soc/qcom/lpass-lpaif.h | 154 +++++++ sound/soc/qcom/lpass-pcm-mi2s.c | 452 +++++++++++++++++++++ 18 files changed, 1598 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,ipq806x-snd-card.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu-dai.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-pcm-mi2s.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,max98357a-codec.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/ipq806x.c create mode 100644 sound/soc/qcom/lpass-cpu-mi2s.c create mode 100644 sound/soc/qcom/lpass-cpu-mi2s.h create mode 100644 sound/soc/qcom/lpass-lpaif.h create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.c -- 1.8.2.1

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[alsa-devel] I2S frame configuration
by thomas chen 09 Dec '14

09 Dec '14

I am working on a ALSA interface to a particular codec over I2S the audio stream format is a bit peculiar... (CODEC being MASTER) there are 24 BCLK cycle between transition of FSYNC... howver, there are only 16 bit that are valid bit 0: ignore bit 1-16: valid pcm data (MSB....LSB) bit 17-23: ignore Can anyone suggest how to configure the DAI format ??? Thanks

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[alsa-devel] [PATCH] ASoC: samsung: i2s: Add missing assignment of variant_regs
by Sylwester Nawrocki 09 Dec '14

09 Dec '14

Add assignment of the variant_regs field which is missing in commit a5a56871f804edac93a53b5e871c0e9818fb9033 ("ASoC: samsung: add support for exynos7 I2S controller"). Without this attempting to probe the secondary DAI fails with an error like: [ 1.763026] Unable to handle kernel NULL pointer dereference at virtual address 0000000c [ 1.780895] pgd = c0004000 [ 1.783606] [0000000c] *pgd=00000000 [ 1.838255] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 1.843514] Modules linked in: [ 1.846558] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.18.0-rc1-00009-g5dcb01e-dirty #1521 [ 1.854887] task: ee00a800 ti: ee088000 task.ti: ee088000 [ 1.860284] PC is at i2s_txctrl+0x40/0x2d4 [ 1.864350] LR is at i2s_txctrl+0x28/0x2d4 [ 1.868428] pc : [<c036ffd4>] lr : [<c036ffbc>] psr: 60000153 [ 1.868428] sp : ee089dc0 ip : 00000000 fp : ee21f000 [ 1.879883] r10: 00000000 r9 : ee21fb00 r8 : c06406c4 [ 1.885091] r7 : ee21fb00 r6 : 00000000 r5 : f00f6000 r4 : ed943410 [ 1.891601] r3 : 0000016c r2 : c0464550 r1 : c055cef8 r0 : ed943610 [ 1.898113] Flags: nZCv IRQs on FIQs off Mode SVC_32 ISA ARM Segment kernel [ 1.905490] Control: 10c5387d Table: 4000404a DAC: 00000015 [ 1.911218] Process swapper/0 (pid: 1, stack limit = 0xee088240) [ 1.917208] Stack: (0xee089dc0 to 0xee08a000) ... [ 2.068431] [<c036ffd4>] (i2s_txctrl) from [<c03719fc>] (samsung_i2s_dai_probe+0xb8/0x450) [ 2.076676] [<c03719fc>] (samsung_i2s_dai_probe) from [<c03607e0>] (snd_soc_register_card+0xd98/0x1348) [ 2.086044] [<c03607e0>] (snd_soc_register_card) from [<c03726e4>] (odroidx2_audio_probe+0xa8/0x11c) [ 2.095160] [<c03726e4>] (odroidx2_audio_probe) from [<c0249dd0>] (platform_drv_probe+0x48/0xa4) [ 2.103922] [<c0249dd0>] (platform_drv_probe) from [<c0248988>] (driver_probe_device+0x10c/0x22c) [ 2.112773] [<c0248988>] (driver_probe_device) from [<c0248b34>] (__driver_attach+0x8c/0x90) [ 2.121192] [<c0248b34>] (__driver_attach) from [<c02471c8>] (bus_for_each_dev+0x54/0x88) [ 2.129352] [<c02471c8>] (bus_for_each_dev) from [<c0248188>] (bus_add_driver+0xd4/0x1d0) [ 2.137510] [<c0248188>] (bus_add_driver) from [<c024915c>] (driver_register+0x78/0xf4) [ 2.145499] [<c024915c>] (driver_register) from [<c0008924>] (do_one_initcall+0x80/0x1b8) [ 2.153670] [<c0008924>] (do_one_initcall) from [<c05b7d40>] (kernel_init_freeable+0xfc/0x1c8) [ 2.162260] [<c05b7d40>] (kernel_init_freeable) from [<c04146c0>] (kernel_init+0x8/0xec) [ 2.170330] [<c04146c0>] (kernel_init) from [<c000e7f8>] (ret_from_fork+0x14/0x3c) [ 2.177873] Code: e5940000 e59f128c e59f228c e2800010 (e59c700c) Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index c7aafcd..eca17dc 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1261,6 +1261,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } + + sec_dai->variant_regs = pri_dai->variant_regs; sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.ch_name = "tx-sec"; -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: rt5645: Fix potential crash in jd function
by Bard Liao 09 Dec '14

09 Dec '14

If no one defined the rt5645->pdata.hp_det_gpio in coreboot/bios. It will cause kernel to reboot because rt5645->pdata.hp_det_gpio is 0. So it is worth to add a check in rt5645_jack_detect. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- * Return an error if gpio number is invalid --- sound/soc/codecs/rt5645.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a7789a8..27141e2 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2209,6 +2209,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) int gpio_state, jack_type = 0; unsigned int val; + if (!gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + dev_err(codec->dev, "invalid gpio\n"); + return -EINVAL; + } gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH RFC 0/4] alsa-lib: audio timestamping evolutions
by Pierre-Louis Bossart 09 Dec '14

09 Dec '14

alsa-lib patches corresponding to the kernel ones just posted. Tested with HDAudio and USB audio class. Pierre-Louis Bossart (4): sync asound.h with kernel version add support for get/set_audio_htstamp_config pcm: add helper functions to query timestamping capabilities test: fix audio_time with new get/set audio_tstamp_config include/pcm.h | 43 ++++- include/sound/asound.h | 20 ++- src/pcm/pcm.c | 82 ++++++++- test/audio_time.c | 473 +++++++++++++++++++++++++++++++------------------ 4 files changed, 436 insertions(+), 182 deletions(-) -- 2.1.0

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Re: [alsa-devel] [PATCH 9/9] ARM: dts: Model IPQ LPASS audio hardware
by Kenneth Westfield 09 Dec '14

09 Dec '14

On Wed, November 19, 2014 2:54 pm, Courtney Cavin wrote: > On Wed, Nov 19, 2014 at 07:52:49PM +0100, Kenneth Westfield wrote: >> --- a/arch/arm/boot/dts/qcom-ipq8064.dtsi >> +++ b/arch/arm/boot/dts/qcom-ipq8064.dtsi >> @@ -2,6 +2,7 @@ >> #include "skeleton.dtsi" >> #include <dt-bindings/clock/qcom,gcc-ipq806x.h> >> +#include <dt-bindings/clock/qcom,lcc-ipq806x.h> > Neither this file nor an associated clock controller driver exists in mainline. Is there some other series this depends on? > -Courtney This patch series has a dependency on the following patch series from linux-arm-msm: [PATCH v2 0/8] qcom audio clock control drivers http://thread.gmane.org/gmane.linux.ports.arm.msm/10793 -- Kenneth Westfield Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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Re: [alsa-devel] [PATCH 3/9] ASoC: ipq806x: add native LPAIF driver
by Kenneth Westfield 09 Dec '14

09 Dec '14

On Tue, November 25, 2014 1:44 pm, Mark Brown wrote: > On Wed, Nov 19, 2014 at 10:52:43AM -0800, Kenneth Westfield wrote: >> +static int lpaif_pcm_int_enable(uint8_t dma_ch) >> +{ >> + uint32_t intr_val; >> + uint32_t status_val; >> + unsigned long flags; >> + >> + if (dma_ch >= LPAIF_MAX_CHANNELS) { >> + pr_err("%s: invalid DMA channel given: %hhu\n", >> + __func__, dma_ch); > dev_err(). Now using it in most cases, except for two or three functions that would need to have a handle passed to it solely for the purpose of using dev_*. >> + while (intrsrc) { >> + dma_ch = lpaif_dai_find_dma_channel(intrsrc); >> + if (dma_ch != -1) { >> + if (lpaif_dai[dma_ch]->callback) { >> + >> + ret = lpaif_dai[dma_ch]->callback(intrsrc, >> + lpaif_dai[dma_ch]->private_data); >> + } >> + intrsrc &= ~LPAIF_PER_CH(dma_ch); >> + } else { >> + pr_err("%s: error getting channel\n", __func__); >> + break; >> + } >> + } >> + return ret; >> +} > This all looks like a very simple demux of a single register - don't we have a generic irqchip that can handle it? It looks like that's what you're trying to implement here, each DMA channel could be requesting the three interrupts it has separately rather than open coding an interrupt controller here. > I'm not actually immediately seeing one right now but it'd be simple to add (or regmap-irq could do it if we used a regmap, though it assumes threading and isn't a great fit). I will look into this. If one fits perfectly, great. If not, then I will add one. -- Kenneth Westfield Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

1 0

[alsa-devel] [PATCH] ASoC: pcm512x: Trigger auto-increment of register addresses on i2c
by Peter Rosin 08 Dec '14

08 Dec '14

From: Peter Rosin <peda(a)axentia.se> When the codec is connected using i2c, it will only auto-increment register addresses if msb (0x80) of the register address byte is set. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/codecs/pcm512x-i2c.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index 4d62230bd378..d0547fa275fc 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -24,8 +24,13 @@ static int pcm512x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct regmap *regmap; + struct regmap_config config = pcm512x_regmap; - regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + /* msb needs to be set to enable auto-increment of addresses */ + config.read_flag_mask = 0x80; + config.write_flag_mask = 0x80; + + regmap = devm_regmap_init_i2c(i2c, &config); if (IS_ERR(regmap)) return PTR_ERR(regmap); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: samsung: request memory region in driver probe()
by Sylwester Nawrocki 08 Dec '14

08 Dec '14

The memory mapped registers region is common for both DAIs so request it in the I2S platform device driver probe for the platform device corresponding to the primary DAI, rather than in the ASoC DAI probe callback. While at it switch to devm_ioremap_resource(). This also drops the hard coded (0x100) register region size in the driver. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Tested on Odroid X2. sound/soc/samsung/i2s.c | 44 +++++++------------------------------------- 1 file changed, 7 insertions(+), 37 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 26d1a1a..fff429da 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -59,10 +59,8 @@ struct samsung_i2s_dai_data { struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; - /* IOREMAP'd SFRs */ + /* Memory mapped SFR region */ void __iomem *addr; - /* Physical base address of SFRs */ - u32 base; /* Rate of RCLK source clock */ unsigned long rclk_srcrate; /* Frame Clock */ @@ -980,12 +978,6 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) goto probe_exit; } - i2s->addr = ioremap(i2s->base, 0x100); - if (i2s->addr == NULL) { - dev_err(&i2s->pdev->dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->clk = clk_get(&i2s->pdev->dev, "iis"); if (IS_ERR(i2s->clk)) { dev_err(&i2s->pdev->dev, "failed to get i2s_clock\n"); @@ -1002,7 +994,6 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { - other->addr = i2s->addr; other->clk = i2s->clk; } @@ -1044,8 +1035,6 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) clk_disable_unprepare(i2s->clk); clk_put(i2s->clk); - - iounmap(i2s->addr); } i2s->clk = NULL; @@ -1164,7 +1153,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) u32 regs_base, quirks = 0, idma_addr = 0; struct device_node *np = pdev->dev.of_node; const struct samsung_i2s_dai_data *i2s_dai_data; - int ret = 0; /* Call during Seconday interface registration */ i2s_dai_data = samsung_i2s_get_driver_data(pdev); @@ -1229,16 +1217,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "Unable to get I2S SFR address\n"); - return -ENXIO; - } + pri_dai->addr = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(pri_dai->addr)) + return PTR_ERR(pri_dai->addr); - if (!request_mem_region(res->start, resource_size(res), - "samsung-i2s")) { - dev_err(&pdev->dev, "Unable to request SFR region\n"); - return -EBUSY; - } regs_base = res->start; pri_dai->dma_playback.dma_addr = regs_base + I2STXD; @@ -1247,7 +1229,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; - pri_dai->base = regs_base; pri_dai->quirks = quirks; pri_dai->variant_regs = i2s_dai_data->i2s_variant_regs; @@ -1258,8 +1239,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.ch_name = "tx-sec"; @@ -1271,7 +1251,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_size = 4; - sec_dai->base = regs_base; + sec_dai->addr = pri_dai->addr; sec_dai->quirks = quirks; sec_dai->idma_playback.dma_addr = idma_addr; sec_dai->pri_dai = pri_dai; @@ -1280,8 +1260,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; + return -EINVAL; } devm_snd_soc_register_component(&pri_dai->pdev->dev, @@ -1293,17 +1272,11 @@ static int samsung_i2s_probe(struct platform_device *pdev) samsung_asoc_dma_platform_register(&pdev->dev); return 0; -err: - if (res) - release_mem_region(regs_base, resource_size(res)); - - return ret; } static int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; - struct resource *res; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; @@ -1313,9 +1286,6 @@ static int samsung_i2s_remove(struct platform_device *pdev) other->sec_dai = NULL; } else { pm_runtime_disable(&pdev->dev); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res) - release_mem_region(res->start, resource_size(res)); } i2s->pri_dai = NULL; -- 1.7.9.5

1 1

[alsa-devel] Nodes, Widgets, ELD and Digital Surround 5.1 (HDMI) Output?
by bencoxdev＠gmail.com 08 Dec '14

08 Dec '14

Hi All, I have an Intel HDA compatible device that supports 5.1 digital audio output over HDMI. It reports two nodes and two widgets:- Root(0) | AFG(1) / \ / \ / \ Audio output(2) Pin Complex(3) This setup produces a “Digital Stereo (HDMI) Output” device at user level in the operating system. The codec reports its self to the operating system as follows (from /proc/asound/card0/codec#1) -------------------------------------------------------------------- AFG Function Id: 0x1 (unsol 1) No Modem Function Group found Default PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 -------------------------------------------------------------------- Node 0x02 [Audio Output] wcaps 0x201: Stereo Digital Device: name="S3 HDMI 1", type="Audio", device=0 Converter: stream=0, channel=0 Digital: Digital category: 0x0 -------------------------------------------------------------------- Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000090: OUT HDMI Pin Default 0x185650f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Red DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x02 -------------------------------------------------------------------- the ELD from the receiving device is:- monitor_present 1 eld_valid 1 monitor_name EP-HDMI-RX connection_type HDMI eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0x1017 product_id 0x901 port_id 0x0 support_hdcp 0 support_ai 1 audio_sync_delay 0 speakers [0x4f] FL/FR LFE FC RL/RR RLC/RRC sad_count 2 sad0_coding_type [0x2] AC-3 sad0_channels 6 sad0_rates [0xe0] 32000 44100 48000 sad0_max_bitrate 640000 sad1_coding_type [0x7] DTS sad1_channels 7 sad1_rates [0xc0] 44100 48000 sad1_max_bitrate 1536000 Does anyone know what changes would be required to this setup in order for the device to be recognised as “Digital Surround 5.1 (HDMI) Output” by alsa? TIA ---------------------------------------------------------------------- Regards Ben

4 11

[alsa-devel] [GIT PULL] ASoC updates for v3.19
by Mark Brown 08 Dec '14

08 Dec '14

The following changes since commit b2776bf7149bddd1f4161f14f79520f17fc1d71d: Linux 3.18 (2014-12-07 14:21:05 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19 for you to fetch changes up to 1810afd3e1ded09c53d4e966dddce3c7d484521f: Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712' and 'asoc/topic/wm9713' into asoc-next (2014-12-08 13:12:36 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.19 Lots and lots of changes this time around, the usual set of driver updates and a huge bulk of cleanups from Lars-Peter. Probably the most interesting thing for most users is the Intel driver updates which will (with some more machine integration work) enable support for newer x86 laptops. - Conversion of AC'97 drivers to use regmap, bringing us closer to the removal of the ASoC level I/O code. - Clean up a lot of old drivers that were open coding things that have subsequently been implemented in the core. - Some DAPM performance improvements. - Removal of the now seldom used CODEC mutex. - Lots of updates for the newer Intel SoC support, including support for the DSP and some Cherrytrail and Braswell machine drivers. - Support for Samsung boards using rt5631 as the CODEC. - Removal of the obsolete AFEB9260 machine driver. - Driver support for the TI TS3A227E headset driver used in some Chrombeooks. ---------------------------------------------------------------- My plan for the next cycle is to try to send one of these each time I send fixes, I meant to do that this time around during the second half of the cycle but ended up not sending fixes (or perhaps sending one lot of fixes at the same time as I'd applied a bunch of changes I wanted to cook in -next, can't remember exactly). Alban Bedel (1): ASoC: tegra: Read and use the GPIO flags of the headphone detect Alexandre Courbot (1): ASoC: jack: update calls to gpiod_get*() Anatol Pomozov (4): ASoC: rt5677: Add option to configure gpio as floating/pullup/pulldown ASoC: rt5677: Print more information if setting DAI clock failed ASoC: rt5677: fix build when kernel compiled without GPIOLIB support ASoC: rt5677: add build dependency to spi Andreas Ruprecht (1): ASoC: rockchip: i2s: Fix Kconfig for I2S device driver Andrew Jackson (1): ASoC: dwc: Allocate resources with devm_ioremap_resource Anil Kumar (1): ASoC: davinvi-mcasp: Balance pm_runtime_enable() on probe failure Axel Lin (2): ASoC: cs4265: Remove unused *dev field from struct cs4265_private ASoC: cs4271: Split SPI and I2C code into different modules Bard Liao (14): ASoC: rt286: reduce power consumption ASoC: rt5645: make bias level more reasonale ASoC: rt286: set combo jack by dmi ASoC: rt286: fix comment style ASoC: rt5645: Add JD function support ASoC: rt5645: Add ASRC support ASoC: rt5670: add rt5672 codec support ASoC: rt5645: remove unused rt5645_clk_sel_put ASoC: rt5645: add register setting for TDM ASoC: rt5645: two jacks for hp and mic ASoC: rt5670: make bias level more reasonable ASoC: rt5645: multiple JD mode support ASoC: rt5670: check if asrc is useable ASoC: rt5670: add DMIC ASRC support Ben Zhang (2): ASoC: rt5677: Include gpio driver header ASoC: rt5677: fix rt5677 spi driver build Charles Keepax (3): ASoC: arizona: Add support for 768kHz DMIC operation ASoC: wm_adsp: Use vmalloc to allocate firmware download buffer ASoC: wm_adsp: Move core_ena to be co-located with start bit Dan Carpenter (1): ASoC: sigmadsp: uninitialized variable in sigmadsp_activate_ctrl() Daniel Mack (1): ASoC: soc-compress: consolidate two identical branches Dmitry Eremin-Solenikov (3): ARM: pxa: spitz: register spitz-audio device ASoC: pxa: Convert spitz to use snd_soc_register_card() ASoC: pxa: prepare/unprepare clocks in pxa-ssp Dylan Reid (3): ASoC: max98090: Different comp tables for different pclks ASoC: add TI ts3a227e headset chip driver ASoC: rt5677: make volume TLV closer to reality Fabio Estevam (15): ASoC: fsl: imx-wm8962: Delete unneeded test before of_node_put ASoC: fsl: imx-sgtl5000: Delete unneeded test before of_node_put ASoC: fsl: imx-spdif: Delete unneeded test before of_node_put ASoC: fsl: eukrea-tlv320: Delete unneeded test before of_node_put ASoC: sgtl5000: Cleanup the comments ASoC: simple-card: Delete unneeded test before of_node_put ASoC: sgtl5000: Use the preferred form for passing a size of a struct ASoC: wm8962: Use the preferred form for passing a size of a struct ASoC: wm8731: Use the preferred form for passing a size of a struct ASoC: mxs: mxs-saif: Register the irq with the device name ASoC: fsl_ssi: Remove comment about SSI running only in slave mode ASoC: sgtl5000: Remove MCLK restriction ASoC: sgtl5000: Allow 8kHz playback in codec slave mode ASoC: mxs-sgtl5000: Remove MCLK restriction ASoC: fsl_ssi: Fix module unbound Fang, Yang A (1): ASoC: Intel: use correct firmware name Fengguang Wu (1): ASoC: Intel: mrfld: fix semicolon.cocci warnings Geert Uytterhoeven (2): ASoC: sh: fsi: Document SoC-specific bindings ASoC: rsnd: Document SoC-specific bindings JS Park (1): ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs Jarkko Nikula (3): ASoC: max98090: Fix digital microphone ASoC: max98090: Fix ill-defined sidetone route ASoC: max98090: Fix right sidetone connection Jean-Francois Moine (4): ASoC: simple-card: Remove useless casts ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: Remove 'const' from the device_node pointers Jean-Michel Hautbois (2): ASoC: sgtl5000: Add MicBias resistor support in DT ASoC: sgtl5000: Add MicBias voltage support Jiada Wang (1): ASoC: fsl_ssi: fix error path in probe Jianqun (2): ASoC: rockchip: i2s: add text after tristate for SND_SOC_ROCKCHIP_I2S ASoC: simple-card: add "invert" property for detect GPIOs Jie Yang (9): ASoC: Intel: Fix block is enabled multiple times issue ASoC: Intel: Work around to fix HW D3 potential crash issue ASoC: Intel: Add jack detection for Broadwell ASoC: Intel: Correct a macro for FW message ASoC: Intel: Fix the driver data not set issue ASoC: Intel: Fix stream volume set no effect issue on Broadwell ASoC: Intel: Remove useless loopback volume control for Broadwell ASoC: Intel: Correct the xmax volume ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell/Haswell Jyri Sarha (3): ASoC: hdmi: Mark the maximum significant bits to HDMI codec ASoC: hdmi: HDMI codec doesn't benefit from pmdown delay ASoC: tlv320aic31xx: Fix off by one error in the loop stucture. Krishna Mohan Dani (5): ASoC: rt5631: Add device tree binding documentation ASoC: rt5631: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631/RT5631 codec driver ASoC: rt5631: Fixing compilation warning when DT is disabled ASoC: Samsung: Add arndale_rt5631 machine driver and binding Kuninori Morimoto (30): ASoC: rsnd: tidyup debug information when read/write ASoC: rsnd: tidyup RSND_DVC_VOLUME_NUM to RSND_DVC_CHANNELS ASoC: rsnd: add struct rsnd_dvc_cfg and control DVC settings ASoC: simple-card: add asoc_simple_card_parse_daifmt() ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() ASoC: rsnd: enable multiple DVC valume settings ASoC: rsnd: enable single DVC valume settings ASoC: rsnd: enable enumerated DVC valume settings ASoC: rsnd: tidyup debug message format and timing ASoC: rsnd: fallback to PIO mode if DMA mode was failed ASoC: rsnd: Add Volume Ramp support ASoC: rsnd: add .fallback callback ASoC: rsnd: add callback status check method ASoC: rsnd: rsnd_src_ssiu_stop() stops SSIU compulsorily ASoC: rsnd: tidyup PIO/DMA mode settings method ASoC: rsnd: tidyup SSI interrupt enable/disable method ASoC: rsnd: care SSIWSR register in rsnd_ssi_hw_start() ASoC: rsnd: clear status register when HW start ASoC: rsnd: synchronize SSI start/stop sequence between PIO/DMA mode ASoC: rsnd: show master clock rate when ADG probe ASoC: rsnd: move snd_kcontrol_new fucntions to core.c ASoC: rsnd: tidyup rsnd_io_to_runtime() macro ASoC: rsnd: use rsnd_src_convert_rate() once on rsnd_src_set_convert_rate_gen2() ASoC: rsnd: initialize SRC on rsnd_src_init() ASoC: rsnd: set SRC_ROUTE_MODE0 on each rsnd_src_set_convert_rate() ASoC: rsnd: share SSI starting method between PIO/DMA mode ASoC: rsnd: remove un-necessary parameter from rsnd_src_start/stop() ASoC: rsnd: add salvage support for under/over flow error on SSI ASoC: rsnd: rename SSI function name of PIO Lars-Peter Clausen (145): ASoC: ak4535: Cleanup manual bias level transitions ASoC: ak4535: Use table based setup for controls ASoC: ak4641: Cleanup manual bias level transitions ASoC: ak4642: Cleanup manual bias level transitions ASoC: ak4671: Cleanup manual bias level transitions ASoC: max98088: Cleanup manual bias level transitions ASoC: max98095: Cleanup manual bias level transitions ASoC: max9850: Cleanup manual bias level transitions ASoC: sta32x: Cleanup manual bias level transitions ASoC: sta350: Cleanup manual bias level transitions ASoC: sta529: Cleanup manual bias level transitions ASoC: stac9766: Cleanup manual bias level transitions ASoC: dapm: Reduce number of checked paths in dapm_widget_in_card_paths() ASoC: dapm: Remove always true path source/sink checks ASoC: dapm: Remove special DAI widget power check functions ASoC: dapm: Remove path 'walked' flag ASoC: dapm: Only mark paths dirty when the connection status changed ASoC: dapm: Do not add un-muxed paths to MUX control ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets ASoC: dapm: Introduce toplevel widget categories ASoC: dapm: Add a flag to mark paths connected to supply widgets ASoC: dapm: Mark endpoints instead of IO widgets dirty during suspend/resume ASoC: dapm: Add a few supply widget sanity checks ASoC: dapm: Use more aggressive caching ASoC: Use generic control handlers for S8 control ASoC: ad1980: Remove unused header ASoC: ad1980: Cleanup printk usage ASoC: ad1980: Use table based control setup ASoC: stac9766: Cleanup printk usage ASoC: stac9766: Use table based control setup ASoC: wm9705: Cleanup printk usage ASoC: wm9705: Use table based control setup ASoC: wm9712: Cleanup printk usage ASoC: wm9712: Use table based control setup ASoC: wm9713: Cleanup printk usage ASoC: wm9713: Use table based control setup ASoC: wm9713: Move driver state struct allocation to driver probe ASoC: wm9713: Use virtual control instead of virtual register ASoC: Remove snd_soc_platform_driver suspend/resume callbacks ASoC: Remove platform field from snd_soc_dai ASoC: mioa701_wm9713: Don't opencode CODEC register access ASoC: Remove card field from snd_soc_dai struct ASoC: Consolidate CPU and CODEC DAI probe ASoC: sn95031: Use table based control setup ASoC: tas2552: Use table based DAPM setup ASoC: wl1273: Use table based control setup ASoC: wm8737: Use table based DAPM and control setup ASoC: wm8961: Use table based DAPM and control setup ASoC: wm8995: Use table based DAPM and control setup ASoC: ad193x: Keep DAC output stage active in idle ASoC: ssm4567: Add DAC high-pass-filter control ASoC: ssm4567: Add support for setting the DAI format and TDM configuration ASoC: ssm4567: Add support for disabling the boost stage ASoC: wm9712/wm9713: Replace virtual registers with custom put/get callbacks ASoC: Forward calls to snd_soc_cache_sync() to regcache_sync() ASoC: Remove snd_soc_cache_sync() implementation ASoC: ab8500-codec: Move control lock to the driver level ASoC: max98095: Move mutex to the driver level ASoC: wm5102: Move ultrasonic response settings lock to the driver level ASoC: wm8731: Move the deemph lock to the driver level ASoC: wm8903: Move the deemph lock to the driver level ASoC: wm8958: Move DSP firmware lock to driver level ASoC: wm8962: Move DSP enable lock to the driver level ASoC: Remove CODEC mutex ASoC: adav80x: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: adau1373: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: adau17x1: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: mpc5200_psc_ac97: Remove unused on-stack snd_ac97 device ASoC: mpc5200_dma: Don't overwrite ac97 device private_data ASoC: Properly handle AC'97 device lifetime management ASoC: Move AC'97 support to its own file ASoC: ac97: Use static ac97_bus ASoC: ac97: Merge soc_ac97_dev_{un,}register()/soc_{un,}register_ac97_codec() ASoC: ac97: Drop support for setting platform data via the CPU DAI ASoC: ac97: Drop delayed device registration ASoC: Drop ac97_control initialization from CODEC driver DAIs ASoC: Rename snd_soc_dai_driver struct ac97_control field to bus_control ASoC: ac97: Push snd_ac97 pointer to the driver level ASoC: Add helper functions for deferred regmap setup ASoC: ad1980: Convert to regmap ASoC: ad1980: Remove ac97_read/ac97_write wrappers ASoC: sigmadsp: Refuse to load firmware files with a non-supported version ASoC: sigmadsp: Drop support support SIGMA_ACTION_DELAY ASoC: sigmadsp: Restructure in preparation for fw v2 support ASoC: sigmadsp: Add support for fw v2 ASoC: adau1701: Implement sigmadsp safeload ASoC: sirf-audio-codec: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98088: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98090: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98095: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: adau17x1: Mark DSP parameter memory as readable and precious ASoC: sigmadsp: Fix endianness conversion ASoC: cs42l51: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: cs42l73: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: uda134x: Remove is_powered_on_standby from platform data ASoC: uda134x: Cleanup manual bias level transitions ASoC: uda1380: Cleanup manual bias level transitions ASoC: alc5623: Cleanup bias level transitions ASoC: alc5632: Cleanup bias level transitions ASoC: rt5631: Cleanup bias level transitions ASoC: wm8350: Cleanup manual bias level transitions ASoC: wm8400: Cleanup manual bias level transitions ASoC: wm8510: Cleanup manual bias level transitions ASoC: wm8523: Cleanup manual bias level transitions ASoC: wm8580: Cleanup manual bias level transitions ASoC: wm8711: Cleanup manual bias level transitions ASoC: wm8728: Cleanup manual bias level transitions ASoC: wm8731: Cleanup manual bias level transitions ASoC: wm8737: Cleanup manual bias level transitions ASoC: wm8750: Cleanup manual bias level transitions ASoC: wm8776: Cleanup manual bias level transitions ASoC: wm8804: Cleanup manual bias level transitions ASoC: wm8900: Cleanup manual bias level transitions ASoC: wm8903: Cleanup manual bias level transitions ASoC: wm8940: Cleanup manual bias level transitions ASoC: wm8955: Cleanup manual bias level transitions ASoC: wm8960: Cleanup manual bias level transitions ASoC: wm8961: Cleanup manual bias level transitions ASoC: wm8983: Cleanup manual bias level transitions ASoC: wm8978: Cleanup manual bias level transitions ASoC: wm8974: Cleanup manual bias level transitions ASoC: wm8985: Cleanup manual bias level transitions ASoC: wm8988: Cleanup manual bias level transitions ASoC: wm8990: Cleanup manual bias level transitions ASoC: wm8991: Cleanup manual bias level transitions ASoC: wm8993: Cleanup manual bias level transitions ASoC: wm8994: Cleanup manual bias level transitions ASoC: wm8995: Cleanup manual bias level transitions ASoC: wm9081: Cleanup manual bias level transitions ASoC: wm9090: Cleanup manual bias level transitions ASoC: wm9712: Cleanup manual bias level transitions ASoC: wm9713: Cleanup manual bias level transitions ASoC: qi_lb60: Pass flags to gpiod_get() ASoC: Disable regmap helpers if regmap is disabled ASoC: cq93vc: Remove unused state struct ASoC: cq93vc: Cleanup manual bias level transitions ASoC: lm49453: Cleanup manual bias level transitions ASoC: sn95031: Cleanup bias level transitions ASoC: tlv320aic23: Cleanup manual bias level transitions ASoC: tlv320aix31xx: Cleanup manual bias level transitions ASoC: tlv320aic32x4: Cleanup manual bias level transitions ASoC: tlv320aic3x: Cleanup manual bias level transitions ASoC: tlv320dac33: Cleanup manual bias level transitions ASoC: twl4030: Cleanup manual bias level transitions ASoC: twl6040: Cleanup manual bias level transitions Liam Girdwood (11): ASoC: Intel: Add TDM support to HSW/BDW SSP port ASoC: Intel: Add 4 channel support to DSP. ASoC: Intel: Make HSW/BDW pointer debug verbose ASoC: Intel: Make ADSP memory block allocation more generic ASoC: Intel: Add generic support for DSP wake, sleep and stall ASoC: Intel: Add PM support to the HSW/BDW DSP core. ASoC: Intel: Add PM support to HSW/BDW IPC driver ASoC: Intel: Add PM support to HSW/BDW PCM driver ASoC: Intel: Add debug output when boot fails. ASoC: Intel: fix build with runtime PM disabled. ASoC: Intel: Fix build with CONFIG_SLEEP enabled. Mark Brown (46): Merge branch 'fix/sgtl5000' of git://git.kernel.org/.../broonie/sound into asoc-sgtl5000 Merge tag 'v3.18-rc2' into asoc-sgtl5000 ASoC: Intel: Add dependency on DesignWare DMA controller ASoC: core: Split ops out of soc-core.c Merge tag 'v3.18-rc3' into asoc-sgtl5000 Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'fix/max98090' of git://git.kernel.org/.../broonie/sound into asoc-max98090 Merge branches 'topic/wm9712' and 'topic/wm9713' of git://git.kernel.org/.../broonie/sound into asoc-wm971x Merge branch 'fix/davinci' of git://git.kernel.org/.../broonie/sound into asoc-davinci Merge branches 'topic/ad1980', 'topic/wm9705' and 'topic/wm971x' of git://git.kernel.org/.../broonie/sound into asoc-ac97 regmap: ac97: Add generic AC'97 callbacks Merge branch 'topic/ac97' of git://git.kernel.org/.../broonie/regmap into asoc-ac97 Merge branch 'topic/regmap' of git://git.kernel.org/.../broonie/sound into asoc-ac97 ASoC: samsung: Fix error handling for clock lookup ASoC: samsung: Fix non-DT use of I2S controller ASoC: wm5102: Initialize dac_comp_lock mutex Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/davinci', 'asoc/fix/max98090', 'asoc/fix/sigmadsp' and 'asoc/fix/tlv320aic31xx' into asoc-linus Merge remote-tracking branch 'asoc/topic/cache' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl-esai' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/ad193x', 'asoc/topic/adau1373' and 'asoc/topic/adau17x1' into asoc-next Merge remote-tracking branches 'asoc/topic/adav80x', 'asoc/topic/adsp', 'asoc/topic/ak4535', 'asoc/topic/ak4641' and 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/alc5623', 'asoc/topic/alc5632', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/codec-mutex', 'asoc/topic/compress' and 'asoc/topic/cq93vc' into asoc-next Merge remote-tracking branches 'asoc/topic/cs4265', 'asoc/topic/cs4271', 'asoc/topic/cs42l51' and 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/doc', 'asoc/topic/dpcm', 'asoc/topic/dwc' and 'asoc/topic/fsi' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-card', 'asoc/topic/fsl-dt' and 'asoc/topic/fsl-ssi' into asoc-next Merge remote-tracking branches 'asoc/topic/hdmi', 'asoc/topic/intel', 'asoc/topic/jack', 'asoc/topic/jz4740' and 'asoc/topic/lm49453' into asoc-next Merge remote-tracking branches 'asoc/topic/max98088', 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/max9850' and 'asoc/topic/mop500' into asoc-next Merge remote-tracking branches 'asoc/topic/multi-codec', 'asoc/topic/mxs-saif', 'asoc/topic/mxs-sgtl5000', 'asoc/topic/omap' and 'asoc/topic/pxa' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/rockchip', 'asoc/topic/rt286' and 'asoc/topic/rt5631' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/rt5670', 'asoc/topic/rt5677', 'asoc/topic/samsung' and 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/sigmadsp', 'asoc/topic/simple', 'asoc/topic/sirf' and 'asoc/topic/sn95031' into asoc-next Merge remote-tracking branches 'asoc/topic/ssm4567', 'asoc/topic/sta32x', 'asoc/topic/sta350', 'asoc/topic/sta529' and 'asoc/topic/stac9766' into asoc-next Merge remote-tracking branches 'asoc/topic/tas2552', 'asoc/topic/tegra', 'asoc/topic/tfa9879', 'asoc/topic/tlv320aic23' and 'asoc/topic/tlv320aic31xx' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic32x4', 'asoc/topic/tlv320aic3x', 'asoc/topic/tlv320dac33', 'asoc/topic/ts3a227e' and 'asoc/topic/twl4030' into asoc-next Merge remote-tracking branches 'asoc/topic/twl6040', 'asoc/topic/uda134x', 'asoc/topic/uda1380' and 'asoc/topic/wl1273' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8350', 'asoc/topic/wm8400', 'asoc/topic/wm8510', 'asoc/topic/wm8523' and 'asoc/topic/wm8580' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8711', 'asoc/topic/wm8728', 'asoc/topic/wm8731', 'asoc/topic/wm8737' and 'asoc/topic/wm8750' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8776', 'asoc/topic/wm8804', 'asoc/topic/wm8900', 'asoc/topic/wm8903' and 'asoc/topic/wm8940' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8955', 'asoc/topic/wm8960', 'asoc/topic/wm8961', 'asoc/topic/wm8962' and 'asoc/topic/wm8974' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8978', 'asoc/topic/wm8983', 'asoc/topic/wm8985', 'asoc/topic/wm8988' and 'asoc/topic/wm8990' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8991', 'asoc/topic/wm8993', 'asoc/topic/wm8994', 'asoc/topic/wm8995' and 'asoc/topic/wm9081' into asoc-next Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712' and 'asoc/topic/wm9713' into asoc-next Markus Elfring (3): ASoC: omap-mcbsp: Deletion of an unnecessary check before the function call "kfree" ASoC: fsi: Deletion of unnecessary checks before the function call "clk_disable" ASoC: mop500: Deletion of unnecessary checks before the function call "of_node_put" Masanari Iida (1): ASoC: jack: Fix warning while make htmldocs caused by soc-jack.c Max Filippov (2): ASoC: tlv320aic23: make codecs selectable in Kconfig ASoC: tlv320aic23: add dependencies on I2C/SPI_MASTER Mengdong Lin (4): ASoC: rt5670 : Add ACPI match ID for Intel CHT/BSW platforms ASoC: Intel: add missing ACPI device table ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672 ASoC: Intel: add support for Cherrytrail and Braswell in SST driver Misael Lopez Cruz (3): ASoC: davinci-mcasp: Active slots depend on active serializers ASoC: tlv320aic3x: Add output driver pop reduction controls ASoC: davinci-mcasp: Add overrun/underrun event handling Mythri P K (1): ASoC: Intel: use lock when changing SST state. Nicolas Ferre (2): ASoC: snd-soc-afeb9260: delete driver as board has just been removed ASoC: Kconfig: remove not used SND_AT91_SOC_AFEB9260 option Nicolin Chen (11): ASoC: fsl_esai: Add indentation for binding doc to increase readability ASoC: fsl_spdif: Add indentation for binding doc to increase readability ASoC: fsl_sai: Add indentation for binding doc to increase readability ASoC: eukrea-tlv320: Add indentation for binding doc to increase readability ASoC: imx-audmux: Add indentation for binding doc to increase readability ASoC: imx-sgtl5000: Add indentation for binding doc to increase readability ASoC: imx-spdif: Add indentation for binding doc to increase readability ASoC: imx-wm8962: Add indentation for binding doc to increase readability ASoC: fsl-asoc-card: Don't bypass settings if cpu-dai is Master ASoC: fsl_esai: Use dynamic slot width as default ASoC: fsl-asoc-card: Add slot_width setting for cpu-dai Oder Chiou (12): ASoC: rt5645: Add the workqueue of the jack detect function for the debouncing ASoC: rt5677: Support DSP function for VAD application ASoC: rt5677: add GPIO IRQ support ASoC: rt5677: Add TDM channel mapping function ASoC: rt5677: Minor coding style and typo fix ASoC: rt5677: Use specific r/w function for DSP mode ASoC: rt5677: Add TDM channel mux in DAC side of IF1 and IF2 ASoC: rt5677: Modify the default value of the MX-8E[4] for ASRC function ASoC: rt5677: Align the reg_default table with tab character ASoC: rt5677: Follow the gpio naming rule to rename the irq function ASoC: rt5677: Set the slow charge of the vref in the end of the power sequences ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed Padmavathi Venna (3): ASoC: Samsung: Add quirk for internal DMA ASoC: samsung: add support for exynos7 I2S controller ASoC: samsung: ASoC: samsung: Fix IISMOD setting in i2s_set_sysclk() Pavel Machek (1): ASoC: omap: enable sound support on n900 on devicetree-based boot Peter Rosin (3): ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex. ASoC: tfa9879: New driver for NXP Semiconductors TFA9879 amplifier. ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing Peter Ujfalusi (10): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence ASoC: davinci-mcasp: When stopping TX/RX stop the AFIFO as the last step ASoC: davinci-mcasp: Move the AFIFO related code under start_tx/rx functions ASoC: tlv320aic3x: Add TDM support ASoC: davinci-mcasp: Symmetric sample bits for IIS mode ASoC: davinci-mcasp: Fix rx format when more bclk is used on the bus ASoC: davinci-mcasp: Validate tdm_slots parameter at probe time ASoC: davinci-mcasp: Place constraint on number of channels ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to individual Qiao Zhou (2): ASoC: soc-pcm: skip dpcm path checking with incapable/unready FE ASoC: soc-pcm: do not hw_free BE if it's still used Ramesh Babu (1): ASoC: core: Call mute for cpu dais as well Randy Dunlap (1): ASoC: cs42l51: depends on I2C Rasmus Villemoes (1): ASoC: dapm: Remove redundant cast Ricardo Neri (2): ASoC: Intel: Add new dependency for Broadwell machine ASoC: Intel: Add new dependency for Haswell machine Richard Fitzgerald (1): ASoC: arizona: Add DSP_B and LEFT_J mode support Subhransu S. Prusty (12): ASoC: Intel: mrfld: Replace pci_id with unique device id ASoC: Intel: mrfld - Define ipc_info structure ASoC: Intel: mrfld: Define sst_res_info for acpi ASoC: Intel: mrfld: Fix runtime pm calls in sst_open_pcm_stream ASoC: mfld-compress: implement .power callback ASoC: Intel: mfld-pcm: Fix to Store device context in sst_data ASoC: Intel: move the driver wq init to a routine ASoC: Intel: move the lock and wq initialization to routine ASoC: Intel: move the driver context allocation to routine ASoC: Intel: modularize driver probe and remove ASoC: Intel: more probe modularization for sst ASoC: Intel: add BYTCR machine driver with RT5640 Sudip Mukherjee (2): ASoC: Intel: fix missing mutex ASoC: rt286: build warning of section mismatch Takashi Iwai (2): ASoC: eukrea-tlv320: Fix of_node_put() call with uninitialized object ASoC: core: Use snd_ctl_enum_info() Thomas Petazzoni (1): ASoC: cs42l51: make driver user-selectable Vinod Koul (27): ASoC: Intel: mrfld: add the gain controls ASoC: Intel: mrfld: add DSP core controls ASoC: Intel: mrfld: add the DSP DAPM widgets ASoC: Intel: mfld-pcm: add FE and BE ops ASoC: Intel: mrfld: add the DSP mixers ASoC: intel: turn off COMPILE_TEST for medfield ASoC: Intel: mrfld - add the dsp sst driver ASoC: Intel: mrfld - Add DSP load and management ASoC: Intel: sst - add pcm ops handling ASoC: Intel: sst: Add IPC handling ASoC: Intel: sst: add stream operations ASoC: Intel: sst: Add some helper functions ASoC: Intel: sst: Add makefile and kconfig changes ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: intel: log an error on double free ASoC: intel: fix the kernldoc comment ASoC: intel: explain why block not found isn't error always ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: Intel: sst: add runtime power management handling ASoC: Intel: sst: load firmware using async callback ASoC: Intel: sst - add compressed ops handling ASoC: Intel: mrfld - remove unnecessary check for pointer ASoC: Intel: mrfld - create separate module for pci part ASoC: Intel: mrfld - add shim save restore ASoC: Intel: mrfld - remove non static definition ASoC: Intel: mrfld- add ACPI module ASoC: Intel: cleanup runtime_pm initialization Zidan Wang (2): ASoC: wm8960: Add device tree support ASoC: wm8960: Move register initialisation to I2C driver probe() kbuild test robot (5): ASoC: rt5677: rt5677_irq_init() can be static ASoC: Intel: dw_pdata can be static ASoC: Intel: byt_rvp_platform_data can be static ASoC: Intel: chv_platform_data can be static ASoC: sigmadsp: fix simple_return.cocci warnings .../devicetree/bindings/sound/arndale.txt | 24 + .../bindings/sound/davinci-mcasp-audio.txt | 2 +- .../devicetree/bindings/sound/eukrea-tlv320.txt | 15 +- .../devicetree/bindings/sound/fsl,esai.txt | 44 +- .../devicetree/bindings/sound/fsl,spdif.txt | 37 +- .../devicetree/bindings/sound/fsl-sai.txt | 66 +- .../bindings/sound/imx-audio-sgtl5000.txt | 61 +- .../devicetree/bindings/sound/imx-audio-spdif.txt | 22 +- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 45 +- .../devicetree/bindings/sound/imx-audmux.txt | 22 +- .../devicetree/bindings/sound/max98090.txt | 2 + .../devicetree/bindings/sound/renesas,fsi.txt | 9 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 10 +- Documentation/devicetree/bindings/sound/rt5631.txt | 48 + Documentation/devicetree/bindings/sound/rt5677.txt | 17 + .../devicetree/bindings/sound/samsung-i2s.txt | 15 +- .../devicetree/bindings/sound/sgtl5000.txt | 13 + .../devicetree/bindings/sound/ts3a227e.txt | 26 + Documentation/devicetree/bindings/sound/wm8960.txt | 31 + MAINTAINERS | 6 + arch/arm/mach-pxa/spitz.c | 9 + arch/x86/include/asm/platform_sst_audio.h | 62 + drivers/base/regmap/Kconfig | 5 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-ac97.c | 114 ++ include/linux/mfd/arizona/core.h | 1 + include/linux/mfd/davinci_voicecodec.h | 7 - include/linux/platform_data/asoc-s3c.h | 1 + include/linux/regmap.h | 7 + include/sound/rcar_snd.h | 8 +- include/sound/rt5645.h | 4 + include/sound/rt5677.h | 10 + include/sound/soc-dai.h | 7 +- include/sound/soc-dapm.h | 9 +- include/sound/soc.h | 116 +- include/sound/uda134x.h | 12 - include/trace/events/asoc.h | 25 - sound/soc/Makefile | 6 +- sound/soc/atmel/Kconfig | 9 - sound/soc/atmel/Makefile | 1 - sound/soc/atmel/atmel_ssc_dai.c | 5 +- sound/soc/atmel/snd-soc-afeb9260.c | 151 -- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 - sound/soc/cirrus/Kconfig | 3 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/codecs/Kconfig | 47 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 32 +- sound/soc/codecs/ac97.c | 18 +- sound/soc/codecs/ad193x.c | 14 +- sound/soc/codecs/ad1980.c | 212 +-- sound/soc/codecs/ad1980.h | 26 - sound/soc/codecs/adau1373.c | 6 +- sound/soc/codecs/adau1701.c | 86 +- sound/soc/codecs/adau1761.c | 25 +- sound/soc/codecs/adau1781.c | 33 +- sound/soc/codecs/adau17x1.c | 71 +- sound/soc/codecs/adau17x1.h | 10 +- sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/ak4535.c | 31 +- sound/soc/codecs/ak4641.c | 33 +- sound/soc/codecs/ak4642.c | 16 - sound/soc/codecs/ak4671.c | 13 - sound/soc/codecs/alc5623.c | 22 +- sound/soc/codecs/alc5632.c | 22 +- sound/soc/codecs/arizona.c | 34 +- sound/soc/codecs/cq93vc.c | 33 - sound/soc/codecs/cs4265.c | 2 - sound/soc/codecs/cs4271-i2c.c | 62 + sound/soc/codecs/cs4271-spi.c | 55 + sound/soc/codecs/cs4271.c | 155 +- sound/soc/codecs/cs4271.h | 11 + sound/soc/codecs/cs42l51.c | 6 +- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/hdmi.c | 2 + sound/soc/codecs/lm49453.c | 8 - sound/soc/codecs/max98088.c | 31 +- sound/soc/codecs/max98090.c | 201 ++- sound/soc/codecs/max98090.h | 8 + sound/soc/codecs/max98095.c | 23 +- sound/soc/codecs/max9850.c | 22 +- sound/soc/codecs/rt286.c | 231 ++- sound/soc/codecs/rt5631.c | 38 +- sound/soc/codecs/rt5645.c | 200 ++- sound/soc/codecs/rt5645.h | 19 +- sound/soc/codecs/rt5670.c | 136 +- sound/soc/codecs/rt5670.h | 6 + sound/soc/codecs/rt5677-spi.c | 130 ++ sound/soc/codecs/rt5677-spi.h | 21 + sound/soc/codecs/rt5677.c | 1198 ++++++++++++- sound/soc/codecs/rt5677.h | 162 +- sound/soc/codecs/sgtl5000.c | 111 +- sound/soc/codecs/sigmadsp-i2c.c | 81 +- sound/soc/codecs/sigmadsp-regmap.c | 46 +- sound/soc/codecs/sigmadsp.c | 711 +++++++- sound/soc/codecs/sigmadsp.h | 59 +- sound/soc/codecs/sirf-audio-codec.c | 6 +- sound/soc/codecs/sn95031.c | 15 +- sound/soc/codecs/ssm4567.c | 128 ++ sound/soc/codecs/sta32x.c | 21 +- sound/soc/codecs/sta350.c | 21 +- sound/soc/codecs/sta529.c | 35 +- sound/soc/codecs/stac9766.c | 60 +- sound/soc/codecs/tas2552.c | 10 +- sound/soc/codecs/tfa9879.c | 328 ++++ sound/soc/codecs/tfa9879.h | 202 +++ sound/soc/codecs/tlv320aic23.c | 21 +- sound/soc/codecs/tlv320aic31xx.c | 31 +- sound/soc/codecs/tlv320aic32x4.c | 24 +- sound/soc/codecs/tlv320aic3x.c | 228 ++- sound/soc/codecs/tlv320aic3x.h | 1 + sound/soc/codecs/tlv320dac33.c | 2 - sound/soc/codecs/ts3a227e.c | 314 ++++ sound/soc/codecs/ts3a227e.h | 17 + sound/soc/codecs/twl4030.c | 2 - sound/soc/codecs/twl6040.c | 23 +- sound/soc/codecs/uda134x.c | 32 +- sound/soc/codecs/uda1380.c | 20 +- sound/soc/codecs/wl1273.c | 10 +- sound/soc/codecs/wm5102.c | 18 +- sound/soc/codecs/wm8350.c | 21 +- sound/soc/codecs/wm8400.c | 34 +- sound/soc/codecs/wm8510.c | 26 +- sound/soc/codecs/wm8523.c | 29 +- sound/soc/codecs/wm8580.c | 4 - sound/soc/codecs/wm8711.c | 27 +- sound/soc/codecs/wm8728.c | 34 +- sound/soc/codecs/wm8731.c | 37 +- sound/soc/codecs/wm8737.c | 49 +- sound/soc/codecs/wm8750.c | 25 +- sound/soc/codecs/wm8776.c | 31 +- sound/soc/codecs/wm8804.c | 3 - sound/soc/codecs/wm8900.c | 8 - sound/soc/codecs/wm8903.c | 43 +- sound/soc/codecs/wm8940.c | 22 +- sound/soc/codecs/wm8955.c | 33 +- sound/soc/codecs/wm8958-dsp2.c | 12 +- sound/soc/codecs/wm8960.c | 113 +- sound/soc/codecs/wm8961.c | 34 +- sound/soc/codecs/wm8962.c | 11 +- sound/soc/codecs/wm8974.c | 25 +- sound/soc/codecs/wm8978.c | 10 - sound/soc/codecs/wm8983.c | 27 +- sound/soc/codecs/wm8985.c | 28 +- sound/soc/codecs/wm8988.c | 27 +- sound/soc/codecs/wm8990.c | 24 +- sound/soc/codecs/wm8991.c | 32 +- sound/soc/codecs/wm8993.c | 12 - sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm8994.h | 2 + sound/soc/codecs/wm8995.c | 17 +- sound/soc/codecs/wm9081.c | 7 - sound/soc/codecs/wm9090.c | 32 +- sound/soc/codecs/wm9705.c | 46 +- sound/soc/codecs/wm9712.c | 219 ++- sound/soc/codecs/wm9713.c | 230 +-- sound/soc/codecs/wm_adsp.c | 97 +- sound/soc/davinci/davinci-mcasp.c | 339 +++- sound/soc/davinci/davinci-mcasp.h | 17 + sound/soc/dwc/designware_i2s.c | 46 +- sound/soc/fsl/eukrea-tlv320.c | 5 +- sound/soc/fsl/fsl-asoc-card.c | 19 +- sound/soc/fsl/fsl_esai.c | 12 +- sound/soc/fsl/fsl_ssi.c | 17 +- sound/soc/fsl/imx-sgtl5000.c | 6 +- sound/soc/fsl/imx-spdif.c | 3 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/fsl/imx-wm8962.c | 6 +- sound/soc/fsl/mpc5200_dma.c | 3 - sound/soc/fsl/mpc5200_psc_ac97.c | 6 +- sound/soc/generic/simple-card.c | 160 +- sound/soc/intel/Kconfig | 44 +- sound/soc/intel/Makefile | 7 + sound/soc/intel/broadwell.c | 64 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 230 +++ sound/soc/intel/cht_bsw_rt5672.c | 285 ++++ sound/soc/intel/haswell.c | 14 +- sound/soc/intel/sst-atom-controls.c | 1208 ++++++++++++- sound/soc/intel/sst-atom-controls.h | 428 +++++ sound/soc/intel/sst-baytrail-dsp.c | 24 +- sound/soc/intel/sst-dsp-priv.h | 136 +- sound/soc/intel/sst-dsp.c | 31 + sound/soc/intel/sst-dsp.h | 28 +- sound/soc/intel/sst-firmware.c | 937 +++++++++-- sound/soc/intel/sst-haswell-dsp.c | 295 +++- sound/soc/intel/sst-haswell-ipc.c | 413 ++++- sound/soc/intel/sst-haswell-ipc.h | 25 +- sound/soc/intel/sst-haswell-pcm.c | 419 ++++- sound/soc/intel/sst-mfld-platform-compress.c | 8 +- sound/soc/intel/sst-mfld-platform-pcm.c | 154 +- sound/soc/intel/sst-mfld-platform.h | 5 + sound/soc/intel/sst/Makefile | 7 + sound/soc/intel/sst/sst.c | 437 +++++ sound/soc/intel/sst/sst.h | 546 ++++++ sound/soc/intel/sst/sst_acpi.c | 383 +++++ sound/soc/intel/sst/sst_drv_interface.c | 686 ++++++++ sound/soc/intel/sst/sst_ipc.c | 373 +++++ sound/soc/intel/sst/sst_loader.c | 456 +++++ sound/soc/intel/sst/sst_pci.c | 209 +++ sound/soc/intel/sst/sst_pvt.c | 449 +++++ sound/soc/intel/sst/sst_stream.c | 437 +++++ sound/soc/jz4740/qi_lb60.c | 11 +- sound/soc/mxs/mxs-saif.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 7 - sound/soc/nuc900/nuc900-ac97.c | 2 +- sound/soc/omap/Kconfig | 8 +- sound/soc/omap/mcbsp.c | 3 +- sound/soc/pxa/mioa701_wm9713.c | 7 +- sound/soc/pxa/pxa-ssp.c | 16 +- sound/soc/pxa/pxa2xx-ac97.c | 6 +- sound/soc/pxa/spitz.c | 52 +- sound/soc/rockchip/Kconfig | 9 +- sound/soc/samsung/Kconfig | 8 +- sound/soc/samsung/Makefile | 2 + sound/soc/samsung/ac97.c | 4 +- sound/soc/samsung/arndale_rt5631.c | 150 ++ sound/soc/samsung/i2s-regs.h | 10 +- sound/soc/samsung/i2s.c | 244 ++- sound/soc/samsung/odroidx2_max98090.c | 4 +- sound/soc/sh/fsi.c | 9 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/core.c | 236 ++- sound/soc/sh/rcar/dvc.c | 215 +-- sound/soc/sh/rcar/gen.c | 30 +- sound/soc/sh/rcar/rsnd.h | 92 +- sound/soc/sh/rcar/src.c | 101 +- sound/soc/sh/rcar/ssi.c | 233 ++- sound/soc/soc-ac97.c | 256 +++ sound/soc/soc-cache.c | 149 -- sound/soc/soc-compress.c | 11 +- sound/soc/soc-core.c | 1768 +++----------------- sound/soc/soc-dapm.c | 755 +++++---- sound/soc/soc-jack.c | 11 +- sound/soc/soc-ops.c | 952 +++++++++++ sound/soc/soc-pcm.c | 23 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 6 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/ux500/mop500.c | 8 +- 244 files changed, 17569 insertions(+), 5861 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt create mode 100644 drivers/base/regmap/regmap-ac97.c delete mode 100644 sound/soc/atmel/snd-soc-afeb9260.c delete mode 100644 sound/soc/codecs/ad1980.h create mode 100644 sound/soc/codecs/cs4271-i2c.c create mode 100644 sound/soc/codecs/cs4271-spi.c create mode 100644 sound/soc/codecs/cs4271.h create mode 100644 sound/soc/codecs/rt5677-spi.c create mode 100644 sound/soc/codecs/rt5677-spi.h create mode 100644 sound/soc/codecs/tfa9879.c create mode 100644 sound/soc/codecs/tfa9879.h create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c create mode 100644 sound/soc/intel/cht_bsw_rt5672.c create mode 100644 sound/soc/intel/sst/Makefile create mode 100644 sound/soc/intel/sst/sst.c create mode 100644 sound/soc/intel/sst/sst.h create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_drv_interface.c create mode 100644 sound/soc/intel/sst/sst_ipc.c create mode 100644 sound/soc/intel/sst/sst_loader.c create mode 100644 sound/soc/intel/sst/sst_pci.c create mode 100644 sound/soc/intel/sst/sst_pvt.c create mode 100644 sound/soc/intel/sst/sst_stream.c create mode 100644 sound/soc/samsung/arndale_rt5631.c create mode 100644 sound/soc/soc-ac97.c create mode 100644 sound/soc/soc-ops.c

2 4

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai/trivial: typo fix
by Nicolas Ferre 08 Dec '14

08 Dec '14

From: Matthieu Crapet <mcrapet(a)gmail.com> Signed-off-by: Matthieu Crapet <mcrapet(a)gmail.com> Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f399808a..775a2badd6a4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -267,7 +267,7 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, if (!ssc_p->dir_mask) { if (ssc_p->initialized) { /* Shutdown the SSC clock. */ - pr_debug("atmel_ssc_dau: Stopping clock\n"); + pr_debug("atmel_ssc_dai: Stopping clock\n"); clk_disable(ssc_p->ssc->clk); free_irq(ssc_p->ssc->irq, ssc_p); -- 2.1.3

2 1

[alsa-devel] [PATCH v3] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 08 Dec '14

08 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. If platform data is not supplied, we will get MCLK from device tree and check the return code to handle probe deferral. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 56 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 54 insertions(+), 3 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..ea5f0b7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -978,7 +980,7 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; -static void wm8960_set_pdata_from_of(struct i2c_client *i2c, +static int wm8960_set_pdata_from_of(struct i2c_client *i2c, struct wm8960_data *pdata) { const struct device_node *np = i2c->dev.of_node; @@ -988,6 +990,15 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + + if (IS_ERR(pdata->mclk)) { + if (PTR_ERR(pdata->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + return 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1008,8 +1019,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); - else if (i2c->dev.of_node) - wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + else if (i2c->dev.of_node) { + ret = wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (ret != 0) + return ret; + } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { @@ -1041,6 +1055,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1070,38 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->pdata.mclk)) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + if (!IS_ERR(wm8960->pdata.mclk)) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1119,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: wm5102: Initialize dac_comp_lock mutex
by Mark Brown 08 Dec '14

08 Dec '14

Commit d74bcaaeb6682 (ASoC: wm5102: Move ultrasonic response settings lock to the driver level) created a driver local mutex for protecting the ultrasonic response settings but neglected to initialize that mutex, causing loud complaints from lockep and potential runtime failures. Fix this by initializing the mutex. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm5102.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8c99b7cf47e0..f439ae052128 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1900,6 +1900,8 @@ static int wm5102_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm5102); + mutex_init(&arizona->dac_comp_lock); + wm5102->core.arizona = arizona; wm5102->core.num_inputs = 6; -- 2.1.3

2 1

[alsa-devel] [PATCH 00/15 v4] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 08 Dec '14

08 Dec '14

This patchset is the rest of patches to enable ALSA Dice driver to handle more models and to improve OXFW driver. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/30 v3] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084791.h… - move mutex operations to a condition in pcm.hw_params (dice/oxfw) - fix naming regression (oxfw) == Supplemental patch for userspace library This patchset adds Hwdep functionality for OXFW driver, thus userspace library needs a patch for entries of snd_hwdep_iface_t. Please apply below patch. [alsa-devel] [PATCH alsa-lib] hwdep: add OXFW driver support http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084824.h… Takashi Sakamoto (15): ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 10 +- sound/firewire/dice/Makefile | 4 +- sound/firewire/dice/dice-midi.c | 194 ++++++++++ sound/firewire/dice/dice-pcm.c | 252 +++++++++---- sound/firewire/dice/dice-stream.c | 332 ++++++++++++---- sound/firewire/dice/dice-transaction.c | 5 - sound/firewire/dice/dice.c | 37 +- sound/firewire/dice/dice.h | 33 +- sound/firewire/oxfw/Makefile | 3 +- sound/firewire/oxfw/oxfw-command.c | 153 ++++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++++++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++++++++ sound/firewire/oxfw/oxfw-pcm.c | 425 +++++++++++++++------ sound/firewire/oxfw/oxfw-proc.c | 113 ++++++ sound/firewire/oxfw/oxfw-stream.c | 667 +++++++++++++++++++++++++++++++-- sound/firewire/oxfw/oxfw.c | 181 +++++++-- sound/firewire/oxfw/oxfw.h | 104 ++++- 19 files changed, 2539 insertions(+), 377 deletions(-) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c -- 2.1.0

2 19

[alsa-devel] [PATCH] sound: pci: asihpi: hpifunc.c: Remove some unused functions
by Rickard Strandqvist 07 Dec '14

07 Dec '14

Removes some functions that are not used anywhere: hpi_subsys_get_version_ex() hpi_instream_wait_start() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/asihpi/hpi.h | 4 ---- sound/pci/asihpi/hpifunc.c | 27 --------------------------- 2 files changed, 31 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 2088724..8338b6e 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1164,8 +1164,6 @@ u16 hpi_stream_estimate_buffer_size(struct hpi_format *pF, /* SubSystem */ /*************/ -u16 hpi_subsys_get_version_ex(u32 *pversion_ex); - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters); u16 hpi_subsys_get_adapter(int iterator, u32 *padapter_index, @@ -1289,8 +1287,6 @@ u16 hpi_instream_read_buf(u32 h_instream, u8 *pb_read_buf, u32 bytes_to_read); u16 hpi_instream_start(u32 h_instream); -u16 hpi_instream_wait_start(u32 h_instream); - u16 hpi_instream_stop(u32 h_instream); u16 hpi_instream_reset(u32 h_instream); diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 510e56c..8fd02dd 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -93,18 +93,6 @@ static inline void hpi_send_recvV1(struct hpi_message_header *m, hpi_send_recv((struct hpi_message *)m, (struct hpi_response *)r); } -u16 hpi_subsys_get_version_ex(u32 *pversion_ex) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_GET_VERSION); - hpi_send_recv(&hm, &hr); - *pversion_ex = hr.u.s.data; - return hr.error; -} - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters) { struct hpi_message hm; @@ -950,21 +938,6 @@ u16 hpi_instream_start(u32 h_instream) return hr.error; } -u16 hpi_instream_wait_start(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_WAIT_START); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - u16 hpi_instream_stop(u32 h_instream) { struct hpi_message hm; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 07 Dec '14

07 Dec '14

sound/soc/xtensa/xtfpga-i2s.c:666:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- xtfpga-i2s.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -663,7 +663,6 @@ static struct platform_driver xtfpga_i2s .remove = xtfpga_i2s_remove, .driver = { .name = "xtfpga-i2s", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(xtfpga_i2s_of_match), .pm = &xtfpga_i2s_pm_ops, },

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Don't resubmit pending URBs at MIDI error recovery
by Takashi Iwai 06 Dec '14

06 Dec '14

In snd_usbmidi_error_timer(), the driver tries to resubmit MIDI input URBs to reactivate the MIDI stream, but this causes the error when some of URBs are still pending like: WARNING: CPU: 0 PID: 0 at ../drivers/usb/core/urb.c:339 usb_submit_urb+0x5f/0x70() URB ef705c40 submitted while active CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.16.6-2-desktop #1 Hardware name: FOXCONN TPS01/TPS01, BIOS 080015 03/23/2010 c0984bfa f4009ed4 c078deaf f4009ee4 c024c884 c09a135c f4009f00 00000000 c0984bfa 00000153 c061ac4f c061ac4f 00000009 00000001 ef705c40 e854d1c0 f4009eec c024c8d3 00000009 f4009ee4 c09a135c f4009f00 f4009f04 c061ac4f Call Trace: [<c0205df6>] try_stack_unwind+0x156/0x170 [<c020482a>] dump_trace+0x5a/0x1b0 [<c0205e56>] show_trace_log_lvl+0x46/0x50 [<c02049d1>] show_stack_log_lvl+0x51/0xe0 [<c0205eb7>] show_stack+0x27/0x50 [<c078deaf>] dump_stack+0x45/0x65 [<c024c884>] warn_slowpath_common+0x84/0xa0 [<c024c8d3>] warn_slowpath_fmt+0x33/0x40 [<c061ac4f>] usb_submit_urb+0x5f/0x70 [<f7974104>] snd_usbmidi_submit_urb+0x14/0x60 [snd_usbmidi_lib] [<f797483a>] snd_usbmidi_error_timer+0x6a/0xa0 [snd_usbmidi_lib] [<c02570c0>] call_timer_fn+0x30/0x130 [<c0257442>] run_timer_softirq+0x1c2/0x260 [<c0251493>] __do_softirq+0xc3/0x270 [<c0204732>] do_softirq_own_stack+0x22/0x30 [<c025186d>] irq_exit+0x8d/0xa0 [<c0795228>] smp_apic_timer_interrupt+0x38/0x50 [<c0794a3c>] apic_timer_interrupt+0x34/0x3c [<c0673d9e>] cpuidle_enter_state+0x3e/0xd0 [<c028bb8d>] cpu_idle_loop+0x29d/0x3e0 [<c028bd23>] cpu_startup_entry+0x53/0x60 [<c0bfac1e>] start_kernel+0x415/0x41a For avoiding these errors, check the pending URBs and skip resubmitting such ones. Reported-and-tested-by: Stefan Seyfried <stefan.seyfried(a)googlemail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index d3d49525a16b..5bfb695547f8 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -365,6 +365,8 @@ static void snd_usbmidi_error_timer(unsigned long data) if (in && in->error_resubmit) { in->error_resubmit = 0; for (j = 0; j < INPUT_URBS; ++j) { + if (atomic_read(&in->urbs[j]->use_count)) + continue; in->urbs[j]->dev = umidi->dev; snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); } -- 2.2.0

2 2

[alsa-devel] [PATCH] ASoC: samsung: Fix error handling for clock lookup
by Mark Brown 05 Dec '14

05 Dec '14

Return the error code we got from clk_get() and check to make sure that clk_prepare_enable() worked. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 25919e090983..500b9ba5ebaa 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; + int ret; if (other && other->clk) { /* If this is probe on secondary */ samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, @@ -989,9 +990,14 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (IS_ERR(i2s->clk)) { dev_err(&i2s->pdev->dev, "failed to get i2s_clock\n"); iounmap(i2s->addr); - return -ENOENT; + return PTR_ERR(i2s->clk); + } + + ret = clk_prepare_enable(i2s->clk); + if (ret != 0) { + dev_err(&i2s->pdev->dev, "failed to enable clock: %d\n", ret); + return ret; } - clk_prepare_enable(i2s->clk); samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); -- 2.1.3

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: sound/atmel/ac97c.c: Add missing clock prepare
by Alexander Stein 05 Dec '14

05 Dec '14

Clocks must be prepared before enabling them. Do this in one step. Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. This fixes the following warning: ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1 at drivers/clk/clk.c:895 __clk_enable+0x24/0x9c() Modules linked in: CPU: 0 PID: 1 Comm: swapper Tainted: G W 3.18.0-rc7+ #245 [<c000dce8>] (unwind_backtrace) from [<c000bcf0>] (show_stack+0x10/0x14) [<c000bcf0>] (show_stack) from [<c001664c>] (warn_slowpath_common+0x60/0x80) [<c001664c>] (warn_slowpath_common) from [<c00166fc>] (warn_slowpath_null+0x18/0x20) [<c00166fc>] (warn_slowpath_null) from [<c02fd7ac>] (__clk_enable+0x24/0x9c) [<c02fd7ac>] (__clk_enable) from [<c02fdbb4>] (clk_enable+0x18/0x2c) [<c02fdbb4>] (clk_enable) from [<c0322688>] (atmel_ac97c_probe+0x154/0x694) [<c0322688>] (atmel_ac97c_probe) from [<c0235e08>] (platform_drv_probe+0x48/0x94) [<c0235e08>] (platform_drv_probe) from [<c02345f8>] (driver_probe_device+0x138/0x350) [<c02345f8>] (driver_probe_device) from [<c02348bc>] (__driver_attach+0x68/0x8c) [<c02348bc>] (__driver_attach) from [<c0232bd0>] (bus_for_each_dev+0x70/0x84) [<c0232bd0>] (bus_for_each_dev) from [<c0233cd8>] (bus_add_driver+0xfc/0x1f8) [<c0233cd8>] (bus_add_driver) from [<c0234f0c>] (driver_register+0x9c/0xe0) [<c0234f0c>] (driver_register) from [<c0008ac4>] (do_one_initcall+0x110/0x1c8) [<c0008ac4>] (do_one_initcall) from [<c053cd58>] (kernel_init_freeable+0xf8/0x1b8) [<c053cd58>] (kernel_init_freeable) from [<c03c0414>] (kernel_init+0x8/0xe4) [<c03c0414>] (kernel_init) from [<c00096d0>] (ret_from_fork+0x14/0x24) ---[ end trace cb88537fdc8fa201 ]--- atmel_ac97c fffa0000.sound: AC'97 0 does not respond - RESET atmel_ac97c fffa0000.sound: AC'97 0 access is not valid [0xffffffff], removing mixer. ------------[ cut here ]------------ Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 86f07e9..fa283d3 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -989,7 +989,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1167,7 +1167,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1184,7 +1184,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1194,7 +1194,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1222,7 +1222,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- 2.2.0

2 2

[alsa-devel] [PATCH v7] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 05 Dec '14

05 Dec '14

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 alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 154 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 157 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3a4d7da..9b62457 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); 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, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -533,6 +534,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..db60701 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -942,6 +942,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56..f4956e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1292,7 +1292,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, 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); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1304,7 +1305,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, 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); + dai_link->num_params, capture_w, + play_w); 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 c61cb9c..7e961ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -754,6 +754,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2721,6 +2751,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3206,6 +3239,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config = &config[w->params_select]; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3274,8 +3310,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3283,14 +3350,52 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3298,6 +3403,28 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3305,15 +3432,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] usb-audio: I'd like to change two things in usb audio, but I'm not sure if it's right
by Johan Aires Rastén 05 Dec '14

05 Dec '14

So I'll try again, this time with the proper cc:s and a second question. I've come across two issues in the usb audio module that are causing problems with my sound card, but I'm thinking that changing them might break something for some other card. In sound/usb/mixer.c function check_input_term(...), for the case UAC_INPUT_TERMINAL and UAC_VERSION_2: The recursive call to get clock selectors messes up the naming of my terminals. At this stage the terminals are already properly named Mic and Line but the call overwrites them both with the unnamed clock selector "Unit 22". Are other UAC2 cards different, so that the clock selector contains the correct terminal name, or is there some other reasoning here that I'm not seeing? Could the call be removed, or perhaps only done if no terminal name has been found yet? In sound/usb/stream.c function snd_usb_add_audio_stream(...), near the end of the function: A newly created pcm is added with "list_add(&as->list, &chip->pcm_list);" which causes devices to be listed in reverse order, i.e. device 0 will be at the end of the list, and this leads to that capture streams end up in the wrong device under certain circumstances. The problem arises when the code under "look for an empty stream" earlier in the same function executes. It will attach the capture stream to the first compatible existing stream, but since they are reversed it will find the highest numbered device. In my case analog input will be grouped with digital out (device 1) instead of analog out (device 0). This is assuming that playback streams are defined before capture streams but it could happen in the other direction as long as there are several streams of one type. I also suspect that this is what causes the M-Audio Audiophile USB to have swapped inputs, for which there's a fix in USB-Audio.conf ( http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/USB-Aud… ). Thanks to debianuser on IRC for helping me find it. I would like to change this call to use list_add_tail for adding new items instead, but I'm worried that there are quirks or other fixes that expect the current behaviour. There are very few other references to pcm_list and none of them seem to care about ordering. Sorry for being bad at referencing code, I'm new at this. //Johan Rastén

2 2

[alsa-devel] [PATCH 1/2] ALSA: sound/atmel/ac97c.c: Add missing clock prepare
by Alexander Stein 05 Dec '14

05 Dec '14

Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b2a3225..ff2f743 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -989,7 +989,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1167,7 +1167,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1184,7 +1184,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1194,7 +1194,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1222,7 +1222,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- 2.2.0

2 3

[alsa-devel] DMA buffer alignment
by Carlo Caione 05 Dec '14

05 Dec '14

Hi, I'm working with a SoC sound hw that requires that all the data written into the DMA memory is aligned a 64 byte boundary. Actually the alignment is required since the format of the data into the DMA memory is (for a 2 channel 16bit audio): 16 samples for the left channel followed by 16 samples for the right channel. If I try to write anything that is not aligned to the 64 bytes I got audio corruption (high pitched noise). Several user-space programs (like aplay) work well with my hw, since the buffer sent from the user-space is always (padded) of period-size bytes, that is aligned to my 64 bytes boundary. Others (like speaker-test) are not working because sometimes they snd_pcm_writei() a buffer that is not multiple of 64 bytes. In general every time I write into the DMA area a buffer that is not aligned, my hw pointer is not aligned anymore with the 64 bytes and the noise starts. IIUC it is not possible to force the user-space programs to send buffers with a fixed size so the problem should be solved in kernel-space (am I wrong on this point?). Any suggestion on how to tackle this problem? Thanks, -- Carlo Caione

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[alsa-devel] [PATCH v3] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 05 Dec '14

05 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Only use the resolution supported, ignoring the LSB of the volume register. This results in half the steps and 0.75dB per step, but reports accurate levels through TLV. Update the masks to reflect that these are registers have the LSB ignored. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 22 +++++++++++----------- sound/soc/codecs/rt5677.h | 24 ++++++++++++------------ 2 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 133010d..81fe146 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -763,9 +763,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); @@ -817,13 +817,13 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5677_DAC1_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5677_DAC2_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC3 Playback Volume", RT5677_DAC3_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC4 Playback Volume", RT5677_DAC4_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv), @@ -842,19 +842,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC1 Capture Volume", RT5677_STO1_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC2 Capture Volume", RT5677_STO2_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC3 Capture Volume", RT5677_STO3_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC4 Capture Volume", RT5677_STO4_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5677_MONO_ADC_DIG_VOL, - RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0, + RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), /* Sidetone Control */ diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index dbd9ffd..c0a625f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -306,10 +306,10 @@ #define RT5677_R_MUTE_SFT 7 #define RT5677_VOL_R_MUTE (0x1 << 6) #define RT5677_VOL_R_SFT 6 -#define RT5677_L_VOL_MASK (0x3f << 8) -#define RT5677_L_VOL_SFT 8 -#define RT5677_R_VOL_MASK (0x3f) -#define RT5677_R_VOL_SFT 0 +#define RT5677_L_VOL_MASK (0x7f << 9) +#define RT5677_L_VOL_SFT 9 +#define RT5677_R_VOL_MASK (0x7f << 1) +#define RT5677_R_VOL_SFT 1 /* LOUT1 Control (0x01) */ #define RT5677_LOUT1_L_MUTE (0x1 << 15) @@ -447,16 +447,16 @@ #define RT5677_SEL_DAC2_R_SRC_SFT 0 /* Stereo1 ADC Digital Volume Control (0x1c) */ -#define RT5677_STO1_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_STO1_ADC_L_VOL_SFT 8 -#define RT5677_STO1_ADC_R_VOL_MASK (0x7f) -#define RT5677_STO1_ADC_R_VOL_SFT 0 +#define RT5677_STO1_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_STO1_ADC_L_VOL_SFT 9 +#define RT5677_STO1_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_STO1_ADC_R_VOL_SFT 1 /* Mono ADC Digital Volume Control (0x1d) */ -#define RT5677_MONO_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_MONO_ADC_L_VOL_SFT 8 -#define RT5677_MONO_ADC_R_VOL_MASK (0x7f) -#define RT5677_MONO_ADC_R_VOL_SFT 0 +#define RT5677_MONO_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_MONO_ADC_L_VOL_SFT 9 +#define RT5677_MONO_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_MONO_ADC_R_VOL_SFT 1 /* Stereo 1/2 ADC Boost Gain Control (0x1e) */ #define RT5677_STO1_ADC_L_BST_MASK (0x3 << 14) -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] 'No audio' bug on lenovo ultra dock
by Alexander Lochmann 05 Dec '14

05 Dec '14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! The current linux kernel (3.17 as well as 3.18-rc7) does not support the audio output on a Lenovo Ultra Dock in conjunction with the X240. The issue is discussed in [1], which provides a patch as well. The patch is available at [2]. I'm asking you for integrating it into your linux tree. So it might be merged into the mainline kernel during the current or the next development phase. Thanks in advance! Regards, Alex [1] https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1385680 [2] https://launchpadlibrarian.net/188362011/lenovo-ultradock.patch - -- Technische Universität Dortmund Alexander Lochmann PGP key: 0xBC3EF6FD Otto-Hahn-Str. 16 phone: +49.231.7556141 D-44227 Dortmund fax: +49.231.7556116 http://ess.cs.tu-dortmund.de/Staff/al -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJUgXibAAoJEFk+7QW8Pvb9zSEP/0yezuJXBK9P8ejhfrmH7VxK PPNhsALQmgMa5vTPsepzBsbj5060yiyI8BTkt76h5d6QlLy3eKq8+AE4+hkGJNie sZtvbWPr8C0GP+8lnfRz6EFOdnWdXbEc4x1hSd7JVDgpNynzHvYZQgISVnxA1rFe NTqngYzFM8yBhUEKc0DYzvuCTw9J0VODWro9eD6tjuSI7/yyQmrLln7XihcLs3/3 VWcNYVBDzQOqjwOTJM7wPfprGgalr7vOBWo3WyshBnKcXmsnahswB+HSsiVWqtMd SZBoEfm811ocvtBFGfsuMkWJALtUfw3bBEPm2iV5jzKmK1CoCxCAuqTcUkiNEzCz m6f76LEtGzq10vpXsc9FmX1Mlbrt9mavE2a5cxoaxVEPJ2OVpwAU3ufgguwUkIrt L8E4RjBEHwttDbe4CPz1YSNz4j+OD4bxsRVjQGfpk8mVGTh0ljtno1chqxn0HZ5J +9Hto37DqDzywJnc4gYyhxTcAZ669qvdMow0avyPuEwGK///C9qgkGCgrm0W36Oq u/HU2lz9A6Vqxtzr8t4bVVEhiX+J2+HQAyf2OPx7o4iMyLAFoxvvWRZUJ3qvP1ZQ OlhXIHg22xTiy9WsVLiStSbT8xd6lV1UP/HotUwOjEoQLzv8pVh/5DH3fwTX5cIw jvRMnfYW+6Ai6Nn07Frj =Y9Kt -----END PGP SIGNATURE-----

1 0

[alsa-devel] [PATCH v2 0/2] fsl_ssi misc fix
by Jiada Wang 05 Dec '14

05 Dec '14

Hi, Changes in v2: - fix error path in probe - replace irq_of_parse_and_map with platform_get_irq Changes in v1: - free IRQ before irq_dispose_mapping Jiada Wang (2): ASoC: fsl_ssi: fix error path in probe ASoC: fsl_ssi: use platform_get_irq instead of irq_of_parse_and_map sound/soc/fsl/fsl_ssi.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix module unbound
by Fabio Estevam 05 Dec '14

05 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Trying to remove the snd-soc-fsl-ssi module leads to the following warning: [ 31.515336] ------------[ cut here ]------------ [ 31.520091] WARNING: CPU: 2 PID: 434 at fs/proc/generic.c:521 remove_proc_entry+0x14c/0x16c() [ 31.528708] remove_proc_entry: removing non-empty directory 'irq/79', leaking at least '202c000.ss' [ 31.537911] Modules linked in: snd_soc_wm8962 snd_soc_imx_wm8962 snd_soc_fsl_ssi(-) evbug [ 31.546249] CPU: 2 PID: 434 Comm: rmmod Not tainted 3.18.0-rc6-00028-g3314bf6-dirty #1 [ 31.554235] Backtrace: [ 31.556816] [<80011ea8>] (dump_backtrace) from [<80012044>] (show_stack+0x18/0x1c) [ 31.564416] r6:80142c88 r5:00000000 r4:00000000 r3:00000000 [ 31.570267] [<8001202c>] (show_stack) from [<806980ec>] (dump_stack+0x88/0xa4) [ 31.577588] [<80698064>] (dump_stack) from [<80029d78>] (warn_slowpath_common+0x70/0x94) [ 31.585711] r5:00000009 r4:bb61fd90 [ 31.589423] [<80029d08>] (warn_slowpath_common) from [<80029e40>] (warn_slowpath_fmt+0x38/0x40) [ 31.598187] r8:bb61fdfe r7:be05d76d r6:be05d9a8 r5:00000002 r4:be05d700 [ 31.605054] [<80029e0c>] (warn_slowpath_fmt) from [<80142c88>] (remove_proc_entry+0x14c/0x16c) [ 31.613709] r3:806a79c0 r2:808229a0 [ 31.617371] [<80142b3c>] (remove_proc_entry) from [<80070380>] (unregister_irq_proc+0x94/0xb8) [ 31.625989] r10:00000000 r8:8000ede4 r7:80955f2c r6:0000004f r5:8118e738 r4:be00af00 [ 31.633952] [<800702ec>] (unregister_irq_proc) from [<80069dac>] (free_desc+0x2c/0x64) [ 31.641898] r6:0000004f r5:80955f38 r4:be00af00 [ 31.646604] [<80069d80>] (free_desc) from [<80069e68>] (irq_free_descs+0x4c/0x8c) [ 31.654092] r7:00000081 r6:00000001 r5:0000004f r4:00000001 [ 31.659863] [<80069e1c>] (irq_free_descs) from [<8006fc3c>] (irq_dispose_mapping+0x40/0x5c) [ 31.668247] r6:be17b844 r5:be17b800 r4:0000004f r3:802c5ec0 [ 31.673998] [<8006fbfc>] (irq_dispose_mapping) from [<7f004ea4>] (fsl_ssi_remove+0x58/0x70 [snd_so) [ 31.683948] r4:bb5bba10 r3:00000001 [ 31.687618] [<7f004e4c>] (fsl_ssi_remove [snd_soc_fsl_ssi]) from [<803720a0>] (platform_drv_remove) [ 31.697564] r5:7f0064f8 r4:be17b810 [ 31.701195] [<80372080>] (platform_drv_remove) from [<80370494>] (__device_release_driver+0x78/0xc) [ 31.710361] r5:7f0064f8 r4:be17b810 [ 31.713987] [<8037041c>] (__device_release_driver) from [<80370d20>] (driver_detach+0xbc/0xc0) [ 31.722631] r5:7f0064f8 r4:be17b810 [ 31.726259] [<80370c64>] (driver_detach) from [<80370304>] (bus_remove_driver+0x54/0x98) [ 31.734382] r6:00000800 r5:00000000 r4:7f0064f8 r3:bb67f500 [ 31.740149] [<803702b0>] (bus_remove_driver) from [<80371398>] (driver_unregister+0x30/0x50) [ 31.748617] r4:7f0064f8 r3:bd9f7080 [ 31.752245] [<80371368>] (driver_unregister) from [<80371f3c>] (platform_driver_unregister+0x14/0x) [ 31.761498] r4:7f00655c r3:7f005a70 [ 31.765130] [<80371f28>] (platform_driver_unregister) from [<7f005a84>] (fsl_ssi_driver_exit+0x14/) [ 31.776147] [<7f005a70>] (fsl_ssi_driver_exit [snd_soc_fsl_ssi]) from [<8008ed80>] (SyS_delete_mod) [ 31.786553] [<8008ec64>] (SyS_delete_module) from [<8000ec20>] (ret_fast_syscall+0x0/0x48) [ 31.794824] r6:00c46d18 r5:00000800 r4:00c46d18 [ 31.799530] ---[ end trace 954e8a3a15379e52 ]--- The cause of problem and solution are well explained by Lars-Peter: "The driver creates the mapping by calling irq_of_parse_and_map(), so it also has to dispose the mapping. But the easy way out is to simply use platform_get_irq() instead of irq_of_parse_map(). In this case the mapping is not managed by the device but by the of core, so the device has not to dispose the mapping." Tested on a imx6q-sabresd board. Reported-by: Jiada Wang <jiada_wang(a)mentor.com> Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 32a31d9..c528f16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1361,7 +1361,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(ssi_private->regs); } - ssi_private->irq = irq_of_parse_and_map(np, 0); + ssi_private->irq = platform_get_irq(pdev, 0); if (!ssi_private->irq) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; @@ -1387,7 +1387,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ssi_private->soc->imx) { ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem); if (ret) - goto error_irqmap; + return ret; } ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, @@ -1458,10 +1458,6 @@ error_asoc_register: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -error_irqmap: - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return ret; } @@ -1478,9 +1474,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: soc-pcm: do not hw_free BE if it's still used
by Qiao Zhou 05 Dec '14

05 Dec '14

Do not free BE hw if it's still used by other FE during dpcm runtime shutdown. Otherwise the BE runtime state will be STATE_HW_FREE and won't be updated to STATE_CLOSE when shutdown ends, because BE dai shutdown function won't close pcm when detecting BE is still under use. With STATE_HW_FREE, BE can't be triggered start again. This corner case can easily appear when one BE is used by two FE, without this patch "ASoC: dpcm: Fix race between FE/BE updates and trigger"(ea9d0d771fcd32cd56070819749477d511ec9117). One FE tries to shutdown but it's raced against xrun on another FE. It improves the be dai hw_free logic. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 57277dd..70e8088 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1664,6 +1664,10 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + /* do not free hw if this BE is used by other FE */ + if (be->dpcm[stream].users > 1) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing
by Peter Rosin 05 Dec '14

05 Dec '14

From: Peter Rosin <peda(a)axentia.se> If a snd_soc_card has any DAPM routes when it calls snd_soc_of_parse_audio_routing, those are clobbered without this change. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-core.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56ebc0f..377db486852e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4585,7 +4585,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes; + int num_routes, old_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -4603,7 +4603,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), + old_routes = card->num_dapm_routes; + routes = devm_kzalloc(card->dev, + (old_routes + num_routes) * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -4611,9 +4613,11 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } + memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); + for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[i].sink); + 2 * i, &routes[old_routes + i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4621,7 +4625,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[i].source); + (2 * i) + 1, &routes[old_routes + i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4630,7 +4634,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; + card->num_dapm_routes += num_routes; card->dapm_routes = routes; return 0; -- 1.7.10.4

3 3

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Add ACPI device probing
by Ben Zhang 04 Dec '14

04 Dec '14

The rt5677 codec driver looks for ACPI device ID "RT5677CE", which is specified in coreboot. This patch allows platform data to be obtained via ACPI Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677.c | 52 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 50 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d317c68..384281d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -20,6 +20,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <linux/firmware.h> #include <linux/gpio.h> #include <sound/core.h> @@ -4525,6 +4526,43 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) return 0; } +#ifdef CONFIG_ACPI + +static unsigned long long rt5677_parse_acpi_entry(struct device *dev, + acpi_string name) +{ + acpi_handle handle = ACPI_HANDLE(dev); + unsigned long long val; + acpi_status status; + + status = acpi_evaluate_integer(handle, name, NULL, &val); + if (ACPI_FAILURE(status)) { + dev_err(dev, "Failed to parse ACPI entry %s, default to 0: %d\n", + name, status); + return 0; + } + return val; +} + +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ + rt5677->pdata.dmic2_clk_pin = (enum rt5677_dmic2_clk) + rt5677_parse_acpi_entry(dev, "DCLK"); + rt5677->pdata.in1_diff = (bool)rt5677_parse_acpi_entry(dev, "IN1"); + rt5677->pdata.in2_diff = (bool)rt5677_parse_acpi_entry(dev, "IN2"); + rt5677->pdata.lout1_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT1"); + rt5677->pdata.lout2_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT2"); + rt5677->pdata.lout3_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT3"); + rt5677->pdata.jd1_gpio = rt5677_parse_acpi_entry(dev, "JD1"); + rt5677->pdata.jd2_gpio = rt5677_parse_acpi_entry(dev, "JD2"); + rt5677->pdata.jd3_gpio = rt5677_parse_acpi_entry(dev, "JD3"); +} +#else +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ +} +#endif + static struct regmap_irq rt5677_irqs[] = { [RT5677_IRQ_JD1] = { .reg_offset = 0, @@ -4604,6 +4642,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (pdata) rt5677->pdata = *pdata; + rt5677->pow_ldo2 = -EINVAL; if (i2c->dev.of_node) { ret = rt5677_parse_dt(rt5677, i2c->dev.of_node); if (ret) { @@ -4611,8 +4650,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, ret); return ret; } - } else { - rt5677->pow_ldo2 = -EINVAL; + } else if (ACPI_HANDLE(&i2c->dev)) { + rt5677_parse_acpi(rt5677, &i2c->dev); } if (gpio_is_valid(rt5677->pow_ldo2)) { @@ -4708,10 +4747,19 @@ static int rt5677_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5677_acpi_id[] = { + { "RT5677CE", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_acpi_id); +#endif + static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5677_acpi_id), }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, -- 2.1.0.rc2.206.gedb03e5

6 35

[alsa-devel] [PATCH v2] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 04 Dec '14

04 Dec '14

XTFPGA boards provides an audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- Changes v1->v2: - fix module name in the sound/soc/xtensa/Makefile; - split PCM data transfer code and I2S interface setup code; - move clock enabling/disabling to runtime PM code; - document I2S interface RATIO field; - rewrite I2S interface LEVEL field calculation to avoid overflows; - drop select statements from Kconfig; - document RCU usage for tx_substream; - clean up code running in RCU read-side critical section; - use threaded IRQ handler instead of IRQ + tasklet; - remove empty PCM callback handlers; - add const to static structures where applicable; - use platform_set_drvdata instead of dev_set_drvdata; - drop redundant error reporting code; - use platform_get_irq instead of platform_get_resource; - clean up xtfpga_i2s_remove; - add comments. .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 676 +++++++++++++++++++++ 7 files changed, 707 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index acb4523..d7bfeec 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10204,6 +10204,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0e96233..9ca12e7 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -56,6 +56,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..99d7222 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -33,3 +33,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..c201beb --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,7 @@ +config SND_SOC_XTFPGA_I2S + tristate "XTFPGA I2S master" + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to the + I2S interface on XTFPGA daughter board. You will also need to select + the drivers for the rest of XTFPGA audio subsystem. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..15efbf9 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1,3 @@ +snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o + +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..bd66be2 --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,676 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + struct regmap *regmap; + void __iomem *regs; + + /* current playback substream. NULL if not playing. + * + * Access to that field is synchronized between the interrupt handler + * and userspace through RCU. + * + * Interrupt handler (threaded part) does PIO on substream data in RCU + * read-side critical section. Trigger callback sets and clears the + * pointer when the playback is started and stopped with + * rcu_assign_pointer. When userspace is about to free the playback + * stream in the pcm_close callback it synchronizes with the interrupt + * handler by means of synchronize_rcu call. + */ + struct snd_pcm_substream *tx_substream; + unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + struct snd_pcm_runtime *runtime, + unsigned tx_ptr); + unsigned tx_ptr; /* next frame index in the sample buffer */ + + /* current fifo level estimate. + * Doesn't have to be perfectly accurate, but must be not less than + * the actual FIFO level in order to avoid stall on push attempt. + */ + unsigned tx_fifo_level; + + /* FIFO level at which level interrupt occurs */ + unsigned tx_fifo_low; + + /* maximal FIFO level */ + unsigned tx_fifo_high; +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +/* Generate functions that do PIO from TX DMA area to FIFO for all supported + * stream formats. + * Functions will be called xtfpga_pcm_tx_<channels>x<sample bits>, e.g. + * xtfpga_pcm_tx_2x16 for 16-bit stereo. + * + * FIFO consists of 32-bit words, one word per channel, always 2 channels. + * If I2S interface is configured with smaller sample resolution, only + * the LSB of each word is used. + */ +#define xtfpga_pcm_tx_fn(channels, sample_bits) \ +static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ + struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ + unsigned tx_ptr) \ +{ \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ +\ + for (; i2s->tx_fifo_level < i2s->tx_fifo_high; \ + i2s->tx_fifo_level += 2) { \ + iowrite32(p[tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++tx_ptr >= runtime->buffer_size) \ + tx_ptr = 0; \ + } \ + return tx_ptr; \ +} + +xtfpga_pcm_tx_fn(1, 16) +xtfpga_pcm_tx_fn(2, 16) +xtfpga_pcm_tx_fn(1, 32) +xtfpga_pcm_tx_fn(2, 32) + +#undef xtfpga_pcm_tx_fn + +static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + bool tx_active; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + tx_ptr); + + cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); + } + rcu_read_unlock(); + + return tx_active; +} + +static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) +{ + unsigned int_status; + unsigned i; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + for (i = 0; i < 2; ++i) { + bool tx_active = xtfpga_pcm_push_tx(i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + if (tx_active) + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!tx_active || + !(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + + /* After the push the level IRQ is still asserted, + * means FIFO level is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + else + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); +} + +static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + unsigned config, int_status, int_mask; + + regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + + if (!(config & XTFPGA_I2S_CONFIG_INT_ENABLE) || + !(int_status & int_mask & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + /* Update FIFO level estimate in accordance with interrupt status + * register. + */ + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_level = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + /* The FIFO isn't empty, but is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + + if (tx_substream && snd_pcm_running(tx_substream)) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + } + rcu_read_unlock(); + + /* Refill FIFO, update allowed IRQ reasons, enable IRQ if FIFO is + * not empty. + */ + xtfpga_pcm_refill_fifo(i2s); + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + /* ratio field of the config register controls MCLK->I2S clock + * derivation: I2S clock = MCLK / (2 * (ratio + 2)). + * + * So with MCLK = 256 * sample rate ratio is 0 for 32 bit stereo + * and 2 for 16 bit stereo. + */ + ratio = (freq - (srate * sample_size * 8)) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + + /* period_size * 2: FIFO always gets 2 samples per frame */ + for (level = 1; + i2s->tx_fifo_low / 2 >= period_size * 2 && + level < (XTFPGA_I2S_CONFIG_LEVEL_MASK >> + XTFPGA_I2S_CONFIG_LEVEL_BASE); ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + unsigned channels = params_channels(hw_params); + + switch (channels) { + case 1: + case 2: + break; + + default: + return -EINVAL; + + } + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x16 : + xtfpga_pcm_tx_2x16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x32 : + xtfpga_pcm_tx_2x32; + break; + + default: + return -EINVAL; + } + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ACCESS_ONCE(i2s->tx_ptr) = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + xtfpga_pcm_refill_fifo(i2s); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + rcu_assign_pointer(i2s->tx_substream, NULL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = ACCESS_ONCE(i2s->tx_ptr); + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static const struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +static const struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .hw_params = xtfpga_i2s_hw_params, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_runtime_suspend(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + return 0; +} + +static int xtfpga_i2s_runtime_resume(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } + return 0; +} + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem; + int err, irq; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + platform_set_drvdata(pdev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = irq; + goto err; + } + err = devm_request_threaded_irq(&pdev->dev, irq, NULL, + xtfpga_i2s_threaded_irq_handler, + IRQF_SHARED | IRQF_ONESHOT, + pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register component\n"); + goto err_unregister_platform; + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + err = xtfpga_i2s_runtime_resume(&pdev->dev); + if (err) + goto err_pm_disable; + } + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_unregister_platform: + snd_soc_unregister_platform(&pdev->dev); +err: + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + xtfpga_i2s_runtime_suspend(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static const struct dev_pm_ops xtfpga_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(xtfpga_i2s_runtime_suspend, + xtfpga_i2s_runtime_resume, NULL) +}; + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + .pm = &xtfpga_i2s_pm_ops, + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

1 0

[alsa-devel] [RFC 0/2] get rid of remaining users of attach_adapter
by Wolfram Sang 04 Dec '14

04 Dec '14

The attach_adapter mechanism of the I2C framework is deprecated for years. There are two users left, drivers for old Macintosh computers. I recently got the idea of replacing this mechanism with a custom one with the help of deferred probing. Because I don't have the hardware, I worked on a seperate branch where I modified the windtunnel driver to be runnable on my Renesas Lager board (ARM). I first verified that the attach_adapter method was used, then the driver rightfully failed on detecting the i2c clients. Using my custom mechanism, the same happens: all busses get scanned, then the clients cannot be detected. Since I didn't change the actual detection code, I assume that it should be working on those Macintosh devices as well. The keywest driver is only compile-tested. That being said, I'd be more than happy, if we could find someone willing to test these patches. If they could be applied, we can _finally_ get rid of this legacy mechanism and clean up the i2c core. The branch I used for ARM compilation is here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/attach_adapter_removal_with_arm_compile-experimental Thanks, Wolfram Wolfram Sang (2): macintosh: therm_windtunnel: drop using attach adapter sound: ppc: keywest: drop using attach adapter drivers/macintosh/therm_windtunnel.c | 25 +++++++++++++++++++++++-- sound/ppc/keywest.c | 21 ++++++++++++++++++--- 2 files changed, 41 insertions(+), 5 deletions(-) -- 2.1.3

2 4

[alsa-devel] [PATCH v2] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 04 Dec '14

04 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. If platform data is not supplied, we will get MCLK from device tree and check the return code to handle probe deferral. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. If we use PLL to generate sysclk, wm8960_set_dai_pll() will enable PLL. But PLL doesn't power down after using codec. So power down PLL in suspend(). Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 59 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 57 insertions(+), 3 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..2c7a97a 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -978,7 +980,7 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; -static void wm8960_set_pdata_from_of(struct i2c_client *i2c, +static int wm8960_set_pdata_from_of(struct i2c_client *i2c, struct wm8960_data *pdata) { const struct device_node *np = i2c->dev.of_node; @@ -988,6 +990,15 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + + if (IS_ERR(pdata->mclk)) { + if (PTR_ERR(pdata->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + return 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1008,8 +1019,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); - else if (i2c->dev.of_node) - wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + else if (i2c->dev.of_node) { + ret = wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (ret != 0) + return ret; + } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { @@ -1041,6 +1055,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1070,41 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->pdata.mclk)) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + /* Power down PLL to save power */ + regmap_update_bits(wm8960->regmap, WM8960_POWER2, 0x1, 0); + + if (!IS_ERR(wm8960->pdata.mclk)) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1122,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

2 2

[alsa-devel] [PATCH 4/5] ASoC: dwc: Add devicetree support for Designware I2S
by Andrew Jackson 04 Dec '14

04 Dec '14

Convert to driver to use either platform_data or device-tree for configuration of the device. When using device-tree, the I2S block's configuration is read from the relevant registers: this reduces the amount of information required in the device tree. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- .../devicetree/bindings/sound/designware-i2s.txt | 32 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 238 ++++++++++++++++---- 3 files changed, 222 insertions(+), 49 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/designware-i2s.txt b/Documentation/devicetree/bindings/sound/designware-i2s.txt new file mode 100644 index 0000000..cdee591 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/designware-i2s.txt @@ -0,0 +1,32 @@ +DesignWare I2S controller + +Required properties: + - compatible : Must be "snps,designware-i2s" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the APB interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "apb_plck" for the clock to the APB interface, "i2sclk" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channels that are used by + the core. The core expects one or two dma channels, one for transmit and one for + receive. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + soc_i2s: i2s@7ff90000 { + compatible = "snps,designware-i2s"; + reg = <0x0 0x7ff90000 0x0 0x1000>; + clocks = <&scpi_i2sclk 0>, <&soc_refclk100mhz>; + clock-names = "i2sclk", "apb_pclk"; + #sound-dai-cells = <0>; + dmas = <&dma0 5>; + dma-names = "tx"; + }; diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index e334900..d50e085 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,6 +1,7 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" depends on CLKDEV_LOOKUP + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index c497ada..083779d 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -22,6 +22,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> /* common register for all channel */ #define IER 0x000 @@ -54,19 +55,46 @@ #define I2S_COMP_VERSION 0x01F8 #define I2S_COMP_TYPE 0x01FC +/* + * Component parameter register fields - define the I2S block's + * configuration. + */ +#define COMP1_TX_WORDSIZE_3(r) (((r) & GENMASK(27, 25)) >> 25) +#define COMP1_TX_WORDSIZE_2(r) (((r) & GENMASK(24, 22)) >> 22) +#define COMP1_TX_WORDSIZE_1(r) (((r) & GENMASK(21, 19)) >> 19) +#define COMP1_TX_WORDSIZE_0(r) (((r) & GENMASK(18, 16)) >> 16) +#define COMP1_TX_CHANNELS(r) (((r) & GENMASK(10, 9)) >> 9) +#define COMP1_RX_CHANNELS(r) (((r) & GENMASK(8, 7)) >> 7) +#define COMP1_RX_ENABLED(r) (((r) & BIT(6)) >> 6) +#define COMP1_TX_ENABLED(r) (((r) & BIT(5)) >> 5) +#define COMP1_MODE_EN(r) (((r) & BIT(4)) >> 4) +#define COMP1_FIFO_DEPTH_GLOBAL(r) (((r) & GENMASK(3, 2)) >> 2) +#define COMP1_APB_DATA_WIDTH(r) (((r) & GENMASK(1, 0)) >> 0) + +#define COMP2_RX_WORDSIZE_3(r) (((r) & GENMASK(12, 10)) >> 10) +#define COMP2_RX_WORDSIZE_2(r) (((r) & GENMASK(9, 7)) >> 7) +#define COMP2_RX_WORDSIZE_1(r) (((r) & GENMASK(5, 3)) >> 3) +#define COMP2_RX_WORDSIZE_0(r) (((r) & GENMASK(2, 0)) >> 0) + #define MAX_CHANNEL_NUM 8 #define MIN_CHANNEL_NUM 2 +union snd_dma_data { + struct i2s_dma_data pd; + struct snd_dmaengine_dai_dma_data dt; +}; + struct dw_i2s_dev { void __iomem *i2s_base; struct clk *clk; int active; unsigned int capability; struct device *dev; + bool using_pd; /* data related to DMA transfers b/w i2s and DMAC */ - struct i2s_dma_data play_dma_data; - struct i2s_dma_data capture_dma_data; + union snd_dma_data play_dma_data; + union snd_dma_data capture_dma_data; struct i2s_clk_config_data config; int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; @@ -153,7 +181,7 @@ static int dw_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - struct i2s_dma_data *dma_data = NULL; + union snd_dma_data *dma_data = NULL; if (!(dev->capability & DWC_I2S_RECORD) && (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) @@ -227,8 +255,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - /* Iterate over set of channels - independently controlled. - */ + /* Iterate over set of channels - independently controlled. */ do { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { i2s_write_reg(dev->i2s_base, TCR(ch_reg), @@ -251,13 +278,18 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, config->sample_rate = params_rate(params); - if (!dev->i2s_clk_cfg) - return -EINVAL; + if (dev->using_pd) { + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + } else { + u32 bitclk; - ret = dev->i2s_clk_cfg(config); - if (ret < 0) { - dev_err(dev->dev, "runtime audio clk config fail\n"); - return ret; + /* TODO: Validate sample rate against permissible set */ + bitclk = config->sample_rate * config->data_width * 2; + clk_set_rate(dev->clk, bitclk); } return 0; @@ -331,6 +363,31 @@ static int dw_i2s_resume(struct snd_soc_dai *dai) #define dw_i2s_resume NULL #endif +/* Maximum resolution of a channel - not uniformly spaced */ +static const u32 fifo_width[] = { + 12, 16, 20, 24, 32, 0, 0, 0 +}; + +/* Width of (DMA) bus */ +static const u32 bus_widths[] = { + DMA_SLAVE_BUSWIDTH_1_BYTE, + DMA_SLAVE_BUSWIDTH_2_BYTES, + DMA_SLAVE_BUSWIDTH_4_BYTES, + DMA_SLAVE_BUSWIDTH_UNDEFINED +}; + +/* PCM format to support channel resolution */ +static const u32 formats[] = { + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S32_LE, + 0, + 0, + 0 +}; + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -340,11 +397,6 @@ static int dw_i2s_probe(struct platform_device *pdev) unsigned int cap; struct snd_soc_dai_driver *dw_i2s_dai; - if (!pdata) { - dev_err(&pdev->dev, "Invalid platform data\n"); - return -EINVAL; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); @@ -373,47 +425,114 @@ static int dw_i2s_probe(struct platform_device *pdev) return PTR_ERR(dev->i2s_base); } - cap = pdata->cap; - dev->capability = cap; - dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (pdata) { + dev->using_pd = true; + cap = pdata->cap; + dev->capability = cap; + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (!dev->i2s_clk_cfg) { + dev_err(&pdev->dev, "no clock device\n"); + return -ENODEV; + } - /* Set DMA slaves info */ + /* Set DMA slaves info */ + + dev->play_dma_data.pd.data = pdata->play_dma_data; + dev->capture_dma_data.pd.data = pdata->capture_dma_data; + dev->play_dma_data.pd.addr = res->start + I2S_TXDMA; + dev->capture_dma_data.pd.addr = res->start + I2S_RXDMA; + dev->play_dma_data.pd.max_burst = 16; + dev->capture_dma_data.pd.max_burst = 16; + dev->play_dma_data.pd.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->capture_dma_data.pd.addr_width = + DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->play_dma_data.pd.filter = pdata->filter; + dev->capture_dma_data.pd.filter = pdata->filter; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + if (cap & DWC_I2S_PLAY) { + dev_dbg(&pdev->dev, " designware: play supported\n"); + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = pdata->channel; + dw_i2s_dai->playback.formats = pdata->snd_fmts; + dw_i2s_dai->playback.rates = pdata->snd_rates; + } - dev->play_dma_data.data = pdata->play_dma_data; - dev->capture_dma_data.data = pdata->capture_dma_data; - dev->play_dma_data.addr = res->start + I2S_TXDMA; - dev->capture_dma_data.addr = res->start + I2S_RXDMA; - dev->play_dma_data.max_burst = 16; - dev->capture_dma_data.max_burst = 16; - dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->play_dma_data.filter = pdata->filter; - dev->capture_dma_data.filter = pdata->filter; + if (cap & DWC_I2S_RECORD) { + dev_dbg(&pdev->dev, "designware: record supported\n"); + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = pdata->channel; + dw_i2s_dai->capture.formats = pdata->snd_fmts; + dw_i2s_dai->capture.rates = pdata->snd_rates; + } + } else { + /* + * Read component parameter registers to extract + * the I2S block's configuration. + */ + u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); + u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2); + u32 bus_width = bus_widths[COMP1_APB_DATA_WIDTH(comp1)]; + u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); + u32 max_size; + + dev->using_pd = false; + + dev->clk = devm_clk_get(&pdev->dev, "i2sclk"); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + /* + * Code presumes all channels are configured with the same + * word size. + */ + if (COMP1_TX_ENABLED(comp1)) { + dev_dbg(&pdev->dev, "playback capable\n"); + + dev->capability |= DWC_I2S_PLAY; + max_size = COMP1_TX_WORDSIZE_0(comp1); + dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; + dev->play_dma_data.dt.addr_width = bus_width; + dev->play_dma_data.dt.chan_name = "TX"; + dev->play_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[max_size]) >> 8; + dev->play_dma_data.dt.maxburst = 16; + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = + 1 << (COMP1_TX_CHANNELS(comp1) + 1); + dw_i2s_dai->playback.formats = formats[max_size]; + dw_i2s_dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + } + if (COMP1_RX_ENABLED(comp1)) { + dev_dbg(&pdev->dev, "record capable\n"); + + dev->capability |= DWC_I2S_RECORD; + max_size = COMP2_RX_WORDSIZE_0(comp2); + dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; + dev->capture_dma_data.dt.addr_width = bus_width; + dev->capture_dma_data.dt.chan_name = "RX"; + dev->capture_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[max_size] >> 8); + dev->capture_dma_data.dt.maxburst = 16; + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = + 1 << (COMP1_RX_CHANNELS(comp1) + 1); + dw_i2s_dai->capture.formats = formats[max_size]; + dw_i2s_dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + } + } - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return PTR_ERR(dev->clk); + ret = clk_prepare(dev->clk); + if (ret < 0) + goto err_clk_put; ret = clk_enable(dev->clk); if (ret < 0) goto err_clk_put; - if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " designware: play supported\n"); - dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->playback.channels_max = pdata->channel; - dw_i2s_dai->playback.formats = pdata->snd_fmts; - dw_i2s_dai->playback.rates = pdata->snd_rates; - } - - if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "designware: record supported\n"); - dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->capture.channels_max = pdata->channel; - dw_i2s_dai->capture.formats = pdata->snd_fmts; - dw_i2s_dai->capture.rates = pdata->snd_rates; - } - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, @@ -423,6 +542,15 @@ static int dw_i2s_probe(struct platform_device *pdev) goto err_clk_disable; } + if (!dev->using_pd) { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, + "Could not register PCM: %d\n", ret); + return ret; + } + } + return 0; err_clk_disable: @@ -443,12 +571,24 @@ static int dw_i2s_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id dw_i2s_of_match[] = { + { .compatible = "snps,designware-i2s", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, dw_i2s_of_match); +#endif + static struct platform_driver dw_i2s_driver = { .probe = dw_i2s_probe, .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = dw_i2s_of_match, +#endif }, }; -- 1.7.1

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[alsa-devel] [PATCH 2/5] ASoC: dwc: Don't allow negative use counts
by Andrew Jackson 04 Dec '14

04 Dec '14

Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 08f0229..f8946bd 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -280,7 +280,8 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev->active--; + if (dev->active > 0) + dev->active--; i2s_stop(dev, substream); break; default: -- 1.7.1

3 6

[alsa-devel] [PATCH 1/5] ASoC: dwc: Allocate resources with devm_ioremap_resource
by Andrew Jackson 04 Dec '14

04 Dec '14

Prepare for the introduction of device-tree support by re-ordering some of the allocations and using devm_iomap_resource to simplify IO mapping. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 46 ++++++++++++++++----------------------- 1 files changed, 19 insertions(+), 27 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e961388..08f0229 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -338,31 +338,34 @@ static int dw_i2s_probe(struct platform_device *pdev) return -EINVAL; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "no i2s resource defined\n"); - return -ENODEV; - } - - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(&pdev->dev, "i2s region already claimed\n"); - return -EBUSY; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); return -ENOMEM; } - dev->i2s_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!dev->i2s_base) { - dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); + if (!dw_i2s_dai) { + dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); return -ENOMEM; } + dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->suspend = dw_i2s_suspend; + dw_i2s_dai->resume = dw_i2s_resume; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no i2s resource defined\n"); + return -ENODEV; + } + + dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(dev->i2s_base)) { + dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + return PTR_ERR(dev->i2s_base); + } + cap = pdata->cap; dev->capability = cap; dev->i2s_clk_cfg = pdata->i2s_clk_cfg; @@ -388,13 +391,6 @@ static int dw_i2s_probe(struct platform_device *pdev) if (ret < 0) goto err_clk_put; - dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); - if (!dw_i2s_dai) { - dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); - ret = -ENOMEM; - goto err_clk_disable; - } - if (cap & DWC_I2S_PLAY) { dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; @@ -411,10 +407,6 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai->capture.rates = pdata->snd_rates; } - dw_i2s_dai->ops = &dw_i2s_dai_ops; - dw_i2s_dai->suspend = dw_i2s_suspend; - dw_i2s_dai->resume = dw_i2s_resume; - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, -- 1.7.1

3 5

[alsa-devel] [PATCH 3/5] ASoC: dwc: Iterate over all channels
by Andrew Jackson 04 Dec '14

04 Dec '14

On the Designware core, the channels are independent and not combined in higher registers. So as more channels are added, more registers need to be updated. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 32 +++++++++++++++++++------------- 1 files changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index f8946bd..c497ada 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -227,19 +227,25 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); - i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); - } else { - i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); - i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); - } + /* Iterate over set of channels - independently controlled. + */ + do { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } + } while (ch_reg-- > 0); i2s_write_reg(dev->i2s_base, CCR, ccr); -- 1.7.1

3 4

[alsa-devel] [PATCH 0/5] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 04 Dec '14

04 Dec '14

This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Andrew Jackson (5): ASoC: dwc: Allocate resources with devm_ioremap_resource ASoC: dwc: Don't allow negative use counts ASoC: dwc: Iterate over all channels ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap .../devicetree/bindings/sound/designware-i2s.txt | 32 ++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 313 ++++++++++++++------ 3 files changed, 260 insertions(+), 86 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

2 2

[alsa-devel] [PATCH 0/2] rockchip: i2s: patches to add optional clock output from
by Jianqun Xu 04 Dec '14

04 Dec '14

The clock output from rockchip SoCs' CRU to chip outside is from the same source as hclk which is for i2s controller. We make it optional because some audio codec outside not need the master clock from SoC but from other clock source. Jianqun Xu (2): ASoC: rockchip-i2s: dt: add an optional property "i2s_clk_out" ASoC: rockchip: i2s: add support for grabbing output clock to codec Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 8 ++++++-- sound/soc/rockchip/rockchip_i2s.c | 11 +++++++++++ 2 files changed, 17 insertions(+), 2 deletions(-) -- 1.9.1

6 16

[alsa-devel] [PATCH v2] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 03 Dec '14

03 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Switch to use TLV_DB_MINMAX as this allows the minimum to be off by only 0.005dB. This is an improvement from the ~500dB is was misreported by before. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..a4c71f9 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -538,9 +538,9 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); -- 2.1.2.330.g565301e

3 6

[alsa-devel] [PATCH 0/2] ALSA: ctxfi: Delete an unnecessary check before kfree()
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 20:35:41 +0100 Another update suggestion was taken into account after a patch was applied from static source code analysis. Markus Elfring (2): Deletion of an unnecessary check before the function call "kfree" One function call less in get_daio_rsc() after error detection sound/pci/ctxfi/ctdaio.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) -- 2.1.3

3 10