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

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[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)>] > >> > > >

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[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

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