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

[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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