February 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/6 v2] ALSA: dice: constrain PCM substreams to current sampling transfer frequency
by Takashi Sakamoto 11 Feb '16

11 Feb '16

Hi, This patchset is to update my former RFCv2 and for merge request to upstream. [alsa-devel][RFC][PATCH 00/10 v2] ALSA: dice: stabiliza packet streaming http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101897.h… Current ALSA Dice driver is written with somewhat ignoring actual hardware design to perform 'dynamic sample rate selection'. As a result, the driver includes some behaviors against users' expectation or functional misleading. This patchset purges such over-engineering. As a result, userspace applications can start PCM substreams just at current sampling transfer frequency. If users want to start at different sampling rate, they must set favorite rate in advance by tools. Currently, ffado-mixer with ffado-dbus-server[1] and hinawa-dice-common-cui[2] are available as such tool. For technical details, please refer to the post or our discussion in RFCv1: [alsa-devel] [RFC][PATCH 0/8] ALSA: dice: constrain PCM substreams to current sampling transfer frequency http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100525.h… [1] as a part of FFADO utility http://subversion.ffado.org/ [2] in hinawa-utils repository https://github.com/takaswie/hinawa-utils Regards Takashi Sakamoto (6): ALSA: dice: limit to current sampling transfer frequency ALSA: dice: limit stream to current sampling transfer frequency. ALSA: dice: add MIDI ports according to current number of MIDI substreams ALSA: dice: get the number of MBLA data channel at opening PCM substream ALSA: dice: purge generating channel cache ALSA: dice: ensure phase lock before starting streaming sound/firewire/dice/dice-midi.c | 23 +++- sound/firewire/dice/dice-pcm.c | 199 +++++++++------------------------ sound/firewire/dice/dice-stream.c | 69 +++++++----- sound/firewire/dice/dice-transaction.c | 54 --------- sound/firewire/dice/dice.c | 67 +---------- sound/firewire/dice/dice.h | 8 -- 6 files changed, 116 insertions(+), 304 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: sigmadsp: Fix missleading return value
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> Forwarding the return value of i2c_master_send, leads to errors later on, since i2c_master_send returns the number of bytes transmittet. Check for ret < 0 instead and return 0 otherwise. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/sigmadsp-i2c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 21ca3a5..d374c18 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data, kfree(buf); - return ret; + if (ret < 0) + return ret; + + return 0; } static int sigmadsp_read_i2c(void *control_data, -- 2.4.3

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[alsa-devel] [PATCH] ASoC: arizona: fref must be limited in pseudo-fractional mode
by Richard Fitzgerald 10 Feb '16

10 Feb '16

When the FLL is in pseudo-fractional mode there is an additional limit on fref based on the fratio, to prevent aliasing around the Nyquist frequency. If fref exceeds this limit the refclk divider must be increased and the calculation tried again until a suitable combination of fref and fratio is found or we have to fall back to integer mode. This patch also adds some debug log prints around this code. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a32cfb8..913cfa8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1899,6 +1899,25 @@ static struct { { 1000000, 13500000, 0, 1 }, }; +static const unsigned int pseudo_fref_max[ARIZONA_FLL_MAX_FRATIO] = { + 13500000, + 6144000, + 6144000, + 3072000, + 3072000, + 2822400, + 2822400, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 768000, +}; + static struct { unsigned int min; unsigned int max; @@ -2012,16 +2031,32 @@ static int arizona_calc_fratio(struct arizona_fll *fll, /* Adjust FRATIO/refdiv to avoid integer mode if possible */ refdiv = cfg->refdiv; + arizona_fll_dbg(fll, "pseudo: initial ratio=%u fref=%u refdiv=%u\n", + init_ratio, Fref, refdiv); + while (div <= ARIZONA_FLL_MAX_REFDIV) { for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / - (fll->vco_mult * ratio) < Fref) + (fll->vco_mult * ratio) < Fref) { + arizona_fll_dbg(fll, "pseudo: hit VCO corner\n"); break; + } + + if (Fref > pseudo_fref_max[ratio - 1]) { + arizona_fll_dbg(fll, + "pseudo: exceeded max fref(%u) for ratio=%u\n", + pseudo_fref_max[ratio - 1], + ratio); + break; + } if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); return ratio; } } @@ -2030,6 +2065,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll, if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); return ratio; } } @@ -2038,6 +2076,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll, Fref /= 2; refdiv++; init_ratio = arizona_find_fratio(Fref, NULL); + arizona_fll_dbg(fll, + "pseudo: change fref=%u refdiv=%d(%d) ratio=%u\n", + Fref, refdiv, div, init_ratio); } arizona_fll_warn(fll, "Falling back to integer mode operation\n"); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: sigmadsp: Fix missleading return value
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> Forwarding the return value of i2c_master_send, leads to errors later on, since i2c_master_send returns the number of bytes transmittet. Check for ret < 0 instead and return 0 otherwise. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/sigmadsp-i2c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 21ca3a5..d374c18 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data, kfree(buf); - return ret; + if (ret < 0) + return ret; + + return 0; } static int sigmadsp_read_i2c(void *control_data, -- 2.4.3

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[alsa-devel] topic/hda-mst branch merged to for-next
by Takashi Iwai 10 Feb '16

10 Feb '16

JFYI, I merged topic/hda-mst branch to for-next branch for wider testing. It means that this will be included in 4.6. I'll keep topic/hda-mst branch for a while, but for future patches, it might be better to based on for-next branch. Takashi

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[alsa-devel] [PATCH] ASoC: cs4271: add regulator consumer support
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> The cs4271 has three power domains: vd, vl and va. Enable them all, as long as the codec is in use. While at it, factored out the reset code into its own function. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 7 +++ sound/soc/codecs/cs4271.c | 69 +++++++++++++++++++--- 2 files changed, 68 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index e2cd1d7..6e699ce 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -33,12 +33,19 @@ Optional properties: Note that this is not needed in case the clocks are stable throughout the entire runtime of the codec. + - vd-supply: Digital power + - vl-supply: Logic power + - va-supply: Analog Power + Examples: codec_i2c: cs4271@10 { compatible = "cirrus,cs4271"; reg = <0x10>; reset-gpio = <&gpio 23 0>; + vd-supply = <&vdd_3v3_reg>; + vl-supply = <&vdd_3v3_reg>; + va-supply = <&vdd_3v3_reg>; }; codec_spi: cs4271@0 { diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e770ee6..0c0010b 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/tlv.h> @@ -157,6 +158,10 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) return reg == CS4271_CHIPID; } +static const char * const supply_names[] = { + "vd", "vl", "va" +}; + struct cs4271_private { unsigned int mclk; bool master; @@ -170,6 +175,7 @@ struct cs4271_private { int gpio_disable; /* enable soft reset workaround */ bool enable_soft_reset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = { @@ -487,6 +493,20 @@ static struct snd_soc_dai_driver cs4271_dai = { .symmetric_rates = 1, }; +static int cs4271_reset(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(cs4271->gpio_nreset)) { + gpio_set_value(cs4271->gpio_nreset, 0); + mdelay(1); + gpio_set_value(cs4271->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec) { @@ -499,6 +519,9 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; } @@ -507,6 +530,16 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) int ret; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + + /* Do a proper reset after power up */ + cs4271_reset(codec); + /* Restore codec state */ ret = regcache_sync(cs4271->regmap); if (ret < 0) @@ -553,19 +586,24 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) } #endif + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + if (cs4271plat) { amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_is_valid(cs4271->gpio_nreset)) { - /* Reset codec */ - gpio_direction_output(cs4271->gpio_nreset, 0); - mdelay(1); - gpio_set_value(cs4271->gpio_nreset, 1); - /* Give the codec time to wake up */ - mdelay(1); - } + /* Reset codec */ + cs4271_reset(codec); + + ret = regcache_sync(cs4271->regmap); + if (ret < 0) + return ret; ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, @@ -595,6 +633,9 @@ static int cs4271_codec_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(cs4271->gpio_nreset, 0); + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; }; @@ -617,6 +658,7 @@ static int cs4271_common_probe(struct device *dev, { struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; + int i, ret; cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) @@ -638,6 +680,17 @@ static int cs4271_common_probe(struct device *dev, return ret; } + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + cs4271->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + *c = cs4271; return 0; } -- 2.4.3

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[alsa-devel] [PATCH] MAINTAINERS: add msm mailing list to sound/soc/qcom
by Srinivas Kandagatla 10 Feb '16

10 Feb '16

This patch adds msm mailing list to sound/soc/qcom so that get_maintainers would return correct list to CC to. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 7f1fa4f..213be2a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5820,6 +5820,7 @@ F: include/uapi/linux/ipmi* QCOM AUDIO (ASoC) DRIVERS M: Patrick Lai <plai(a)codeaurora.org> M: Banajit Goswami <bgoswami(a)codeaurora.org> +L: linux-arm-msm(a)vger.kernel.org L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/qcom/ -- 1.9.1

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[alsa-devel] Should snd_card_free() check for null pointer?
by Jerome Marchand 10 Feb '16

10 Feb '16

Hi, Before commit f24640648186b (ALSA: Use standard device refcount for card accounting), snd_card_free() would return -EINVAL on a null pointer. Now it ends up in a null pointer dereference. There is at least one driver that can call snd_card_free() with null argument: saa7134_alsa. It can easily be triggered by just inserting and removing the module (no need to have the hardware). I don't think that is a rule, but it seems that the standard behavior of *_free() functions is to check for null pointer. What do you think? Thanks, Jerome

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[alsa-devel] [PATCH] ASoC: rt5645: add "coreboot" to dmi_system_id list
by Bard Liao 09 Feb '16

09 Feb '16

There are some Google projects with manufacturer name different from "GOOGLE". So we need to add DMI list for those projects. However some OEMs / vendors want to keep secret before product launch. So, we can't use DMI_SYS_VENDOR for them. Fortunately, the BIOS vendor field is the same for all chrome os projects. It allows us to match chrome projects by DMI_BIOS_VENDOR = "coreboot". Signed-off-by: John Lin <john.lin(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 93e8c90..8521b22 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3552,6 +3552,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, + { + .ident = "Google Chrome", + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "coreboot"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2 00/22] Sharing media resources across ALSA and au0828 drivers
by Shuah Khan 09 Feb '16

09 Feb '16

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API and Media Controller API to share media resource (tuner). This Patch v2 series is based on linux_media master. This work addresses Mauro and Takashi's comments. Composite or S-Video connector is the input case is still under discussion and isn't addressed in this series. Patches 3 and 4 need documentation updates and will be sent later this week. Shuah Khan (22): uapi/media.h: Declare interface types for ALSA media: Add ALSA Media Controller function entities media: Media Controller register/unregister entity_notify API media: Media Controller enable/disable source handler API media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add enable/disable source common interfaces media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: Change v4l-core to check if source is free media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_source() media: au0828 change to use Managed Media Controller API media: au0828 handle media_init and media_register window media: au0828 create tuner to decoder link in disabled state media: au0828 disable tuner to demod link media: au0828 Use au8522_media_pads enum for pad defines media: au0828-core register entity_notify hook media: au0828 add enable, disable source handlers sound/usb: Use Media Controller API to share media resources media: Ensure media device unregister is done only once drivers/media/dvb-core/dvb_frontend.c | 139 ++---------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 73 +++++- drivers/media/media-devnode.c | 15 +- drivers/media/media-entity.c | 51 ++++- drivers/media/usb/au0828/au0828-core.c | 272 +++++++++++++++++++++-- drivers/media/usb/au0828/au0828-video.c | 75 +------ drivers/media/usb/au0828/au0828.h | 4 + drivers/media/v4l2-core/Makefile | 2 +- drivers/media/v4l2-core/v4l2-fh.c | 2 + drivers/media/v4l2-core/v4l2-ioctl.c | 30 +++ drivers/media/v4l2-core/v4l2-mc.c | 60 +++++ drivers/media/v4l2-core/videobuf2-core.c | 4 + include/media/media-device.h | 44 ++++ include/media/media-devnode.h | 17 ++ include/media/media-entity.h | 12 + include/media/v4l2-dev.h | 1 + include/media/v4l2-mc.h | 52 +++++ include/uapi/linux/media.h | 33 +++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 1 + sound/usb/media.c | 319 +++++++++++++++++++++++++++ sound/usb/media.h | 72 ++++++ sound/usb/mixer.h | 1 + sound/usb/pcm.c | 28 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 3 + 33 files changed, 1117 insertions(+), 236 deletions(-) create mode 100644 drivers/media/v4l2-core/v4l2-mc.c create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.5.0

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