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

[alsa-devel] [PATCH 0/4] ASoC: sst-haswell-pcm: Move controls and DAPM elements to component
by Lars-Peter Clausen 06 Sep '14

06 Sep '14

The sst-haswell-pcm driver is the only one that registers controls and DAPM elements for a snd_soc_platform_driver. Moving them over to the component that is registered by the same driver make it possible to remove the steal_silbling_dai_hack in the core as well as the extra indirection that is used to register the controls and DAPM elements. Vinod, Subhransu, can you take a look at this and make sure it works as expected and also keep this in mind for the new DPCM drivers? Thanks, - Lars Lars-Peter Clausen (4): ASoC: Add snd_soc_component_{get,set}_drvdata() ASoC: sst-haswell-pcm: Alloc state struct in driver probe() ASoC: sst-haswell-pcm: Move controls and DAPM elements to component ASoC: Remove table based DAPM/control setup support from snd_soc_platform_driver include/sound/soc.h | 28 ++++++++++--------- sound/soc/intel/sst-haswell-pcm.c | 56 ++++++++++++++++++------------------- sound/soc/soc-core.c | 58 ++++----------------------------------- 3 files changed, 49 insertions(+), 93 deletions(-) -- 1.8.0

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[alsa-devel] Audio mini conference 2014?
by Takashi Iwai 03 Sep '14

03 Sep '14

Hi, did anyone think of a meeting in this year? AFAIK, a couple of good candidates to tie with are: - LinuxCon + KS will be at Chicago, USA (in August 18-22) - Plumbers will be at Duesseldorf, Germany (in October 15-17) Any other opportunities? I'm not sure whether I'll be in the former (due to private reason), but very likely I'll attend the latter. And, if we hold a meeting, in which format? I find the previous year's meeting was nice; i.e. a kind of mini-summit that doesn't conflict with other meetings is my preference. But I'm open for other options. thanks, Takashi

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[alsa-devel] [PATCH v2 0/4] ASoC: atmel_ssc: enable fslen extension feature
by Bo Shen 02 Sep '14

02 Sep '14

On at91sam9rl, at91sam9g10, at91sam9g20, at91sam9g45 and newer SoCs, the SSC peripheral supports frame sync length extension feature. When SSC works as master, it can generate frame sync lenght larger than 16 bits. After enable this feature, it can playback 24/32 bits audio clips. Changes in v2: - Using compatible string to distinguish whether SSC supports frame sync length extension to replace check IP version. - Add related modification for dts file. Bo Shen (4): ASoC: atmel-ssc: distinguish whether SSC supports fslen ext ASoC: atmel_ssc_dai: enable fslen extension feature dts: atmel: at91sam9rl: switch ssc compatible string dts: atmel: at91sam9g20: switch ssc compatible string arch/arm/boot/dts/at91sam9g20.dtsi | 4 ++++ arch/arm/boot/dts/at91sam9rl.dtsi | 4 ++-- drivers/misc/atmel-ssc.c | 13 +++++++++++++ include/linux/atmel-ssc.h | 13 +++++++++++++ sound/soc/atmel/atmel_ssc_dai.c | 34 ++++++++++++++++++---------------- 5 files changed, 50 insertions(+), 18 deletions(-) -- 1.8.5.2

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[alsa-devel] [RFC][PATCH] ASoC: simple-card: Merge single and muti DAI link code.
by Xiubo Li 02 Sep '14

02 Sep '14

This patch merge single DAI link and muti-DAI links code together, and simply the simple-card driver code. And also do some other improvement: Since from the DAI format micro SND_SOC_DAIFMT_CBx_CFx, the 'CBx' mean Codec's bit clock is as master/slave and the 'CFx' mean Codec's frame clock is as master/slave. So these same DAI formats should be informed to CPU and CODE DAIs at the same time. For the Codec driver will set the bit clock and frame clock as the DAI formats said, but for the CPU driver, if the the bit clock or frame clock is as Codec master, so it should be set CPU DAI device as bit clock or frame clock as slave, and vice versa. The old code will cause confusion, and we should be clear that the letter 'C' here mean to Codec. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Jean-Francois Moine <moinejf(a)free.fr> Cc: Jyri Sarha <jsarha(a)ti.com> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Hi, This patch will break the old DT, so i just send one RFC version, and will add the old DT patches in next version if this patch can work well. Any comments and advices are welcome. .../devicetree/bindings/sound/simple-card.txt | 98 ++++++++------- sound/soc/generic/simple-card.c | 135 ++++++--------------- 2 files changed, 88 insertions(+), 145 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c2e9841..457fd0b 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -1,13 +1,12 @@ Simple-Card: +==== Simple-Card specifies audio DAI connections of SoC <-> codec. Required properties: - - compatible : "simple-audio-card" Optional properties: - - simple-audio-card,name : User specified audio sound card name, one string property. - simple-audio-card,widgets : Please refer to widgets.txt. @@ -18,68 +17,57 @@ Optional properties: - simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec mclk. -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section bellow. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. +==== +DAI link node(s) and its or their subnodes: -Required dai-link subnodes: +There must be one DAI link node exsit at least, and each DAI link must +contain one CPU subnode and one CODEC subnode. -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: +Required DAI link node(s): +- simple-audio-card,dai-link : Container for DAI link level + properties and the CPU and CODEC + sub-nodes. +Optional DAI link node properties: - format : CPU/CODEC common audio format. "i2s", "right_j", "left_j" , "dsp_a" "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC +- frame-master : Boolean property. If present, for this + DAI link the Codec DAI's frame will be + as master and the CPU DAI's frame will + be as slave, or vice versa. +- bitclock-master : Boolean property. If present, for this + DAI link the Codec DAI's bit clock will + be as master and the CPU DAI's bit clock + will be as slave, or vice versa. +==== +DAI link's subnodes and their properties: + +Required DAI link subnodes: +- cpu : CPU sub-node +- codec : CODEC sub-node -Optional CPU/CODEC subnodes properties: +Required DAI link subnodes' properties: +- sound-dai : Phandle and port of CPU/CODEC +Optional DAI link subnodes' properties: +- bitclock-inversion : Boolean property. Add this if the DAI + device needs bit clock to be inversed. +- frame-inversion : Boolean property. Add this if the DAI + device needs frame clock to be inversed. - dai-tdm-slot-num : Please refer to tdm-slot.txt. - dai-tdm-slot-width : Please refer to tdm-slot.txt. - clocks / system-clock-frequency : specify subnode's clock if needed. it can be specified via "clocks" if system has clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) +==== Example 1 - single DAI link: sound { compatible = "simple-audio-card"; simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; simple-audio-card,widgets = "Microphone", "Microphone Jack", "Headphone", "Headphone Jack", @@ -89,13 +77,23 @@ sound { "Headphone Jack", "HP_OUT", "External Speaker", "LINE_OUT"; - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; + simple-audio-card,dai-link { + format = "left_j"; + cpu { + sound-dai = <&sh_fsi2 0>; + + bitclock-inversion; + frame-inversion; + }; + codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + + bitclock-inversion; + }; - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; + bitclock-master; + frame-master; }; }; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 21b0ea2..a0d3622 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -112,13 +112,20 @@ static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, struct device_node **p_node, - const char **name) + const char **name, + unsigned int *daifmt) { struct device_node *node; struct clk *clk; u32 val; int ret; + if (!daifmt) + return -EINVAL; + + *daifmt = snd_soc_of_parse_daifmt(np, NULL, NULL, NULL); + *daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + /* * get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() @@ -166,102 +173,55 @@ asoc_simple_card_sub_parse_of(struct device_node *np, static int simple_card_dai_link_of(struct device_node *node, struct device *dev, struct snd_soc_dai_link *dai_link, - struct simple_dai_props *dai_props, - bool is_top_level_node) + struct simple_dai_props *dai_props) { struct device_node *np = NULL; - struct device_node *bitclkmaster = NULL; - struct device_node *framemaster = NULL; unsigned int daifmt; char *name; - char prop[128]; - char *prefix = ""; int ret; - if (is_top_level_node) - prefix = "simple-audio-card,"; - - daifmt = snd_soc_of_parse_daifmt(node, prefix, - &bitclkmaster, &framemaster); - daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + daifmt = snd_soc_of_parse_daifmt(node, NULL, NULL, NULL); + daifmt &= SND_SOC_DAIFMT_MASTER_MASK; + dai_props->cpu_dai.fmt = dai_props->codec_dai.fmt = daifmt; - snprintf(prop, sizeof(prop), "%scpu", prefix); - np = of_get_child_by_name(node, prop); + np = of_get_child_by_name(node, "cpu"); if (!np) { ret = -EINVAL; - dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); + dev_err(dev, "%s: Can't find cpu DT node\n", __func__); goto dai_link_of_err; } ret = asoc_simple_card_sub_parse_of(np, &dai_props->cpu_dai, &dai_link->cpu_of_node, - &dai_link->cpu_dai_name); + &dai_link->cpu_dai_name, + &daifmt); if (ret < 0) goto dai_link_of_err; - dai_props->cpu_dai.fmt = daifmt; - switch (((np == bitclkmaster) << 4) | (np == framemaster)) { - case 0x11: - dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - case 0x10: - dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - case 0x01: - dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - default: - dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - } - + dai_props->cpu_dai.fmt |= daifmt; of_node_put(np); - snprintf(prop, sizeof(prop), "%scodec", prefix); - np = of_get_child_by_name(node, prop); + + np = of_get_child_by_name(node, "codec"); if (!np) { ret = -EINVAL; - dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); + dev_err(dev, "%s: Can't find codec DT node\n", __func__); goto dai_link_of_err; } ret = asoc_simple_card_sub_parse_of(np, &dai_props->codec_dai, &dai_link->codec_of_node, - &dai_link->codec_dai_name); + &dai_link->codec_dai_name, + &daifmt); if (ret < 0) goto dai_link_of_err; - if (strlen(prefix) && !bitclkmaster && !framemaster) { - /* No dai-link level and master setting was not found from - sound node level, revert back to legacy DT parsing and - take the settings from codec node. */ - dev_dbg(dev, "%s: Revert to legacy daifmt parsing\n", - __func__); - dai_props->cpu_dai.fmt = dai_props->codec_dai.fmt = - snd_soc_of_parse_daifmt(np, NULL, NULL, NULL) | - (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK); - } else { - dai_props->codec_dai.fmt = daifmt; - switch (((np == bitclkmaster) << 4) | (np == framemaster)) { - case 0x11: - dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - case 0x10: - dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - case 0x01: - dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - default: - dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - } - } - if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) { ret = -EINVAL; goto dai_link_of_err; } + dai_props->codec_dai.fmt |= daifmt; + /* simple-card assumes platform == cpu */ dai_link->platform_of_node = dai_link->cpu_of_node; @@ -288,22 +248,18 @@ static int simple_card_dai_link_of(struct device_node *node, dai_link_of_err: if (np) of_node_put(np); - if (bitclkmaster) - of_node_put(bitclkmaster); - if (framemaster) - of_node_put(framemaster); return ret; } static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv, - struct device *dev, - int multi) + struct device *dev) { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct simple_dai_props *dai_props = priv->dai_props; + struct device_node *np = NULL; u32 val; - int ret; + int ret, i; /* parsing the card name from DT */ snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); @@ -332,23 +288,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ? priv->snd_card.name : ""); - if (multi) { - struct device_node *np = NULL; - int i; - for (i = 0; (np = of_get_next_child(node, np)); i++) { - dev_dbg(dev, "\tlink %d:\n", i); - ret = simple_card_dai_link_of(np, dev, dai_link + i, - dai_props + i, false); - if (ret < 0) { - of_node_put(np); - return ret; - } - } - } else { - ret = simple_card_dai_link_of(node, dev, dai_link, dai_props, - true); - if (ret < 0) + /* Parse DAI link(s) */ + for (i = 0; (np = of_get_next_child(node, np)); i++) { + dev_dbg(dev, "\tlink %d:\n", i); + ret = simple_card_dai_link_of(np, dev, dai_link + i, + dai_props + i); + if (ret < 0) { + of_node_put(np); return ret; + } } if (!priv->snd_card.name) @@ -384,16 +332,13 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; - int num_links, multi, ret; + int num_links, ret; /* get the number of DAI links */ - if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) { + if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) num_links = of_get_child_count(np); - multi = 1; - } else { - num_links = 1; - multi = 0; - } + else + return -EINVAL; /* allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, @@ -420,7 +365,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, priv, dev, multi); + ret = asoc_simple_card_parse_of(np, priv, dev); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); -- 1.8.5

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[alsa-devel] ALC1150 broken on SuperMicro X10SAT
by Dana Goyette 02 Sep '14

02 Sep '14

On my SuperMicro X10SAT, the ALC1150 audio codec has horrific audio quality by default. The only output that seems to work is the front channel, and it's quiet, distorted, and unbalanced (the right is louder than the left). alsa-info.sh output: http://www.alsa-project.org/db/?f=8fbcab1d61590060ceddc1e63eb00d508e94d731 I've gotten the audio to work by guessing a model: options snd-hda-intel model=auto,acer-aspire,auto In tracing through the source: {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"} This name leads to: [ALC883_FIXUP_ACER_EAPD] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { /* eanable EAPD on Acer laptops */ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, { } } }, There's another override I did not try, since it has no model= name: [ALC883_FIXUP_EAPD] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { /* change to EAPD mode */ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x3070 }, { } } },

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[alsa-devel] [PATCH] ASoC: ab8500-codec: Revert back to regmap
by Lars-Peter Clausen 01 Sep '14

01 Sep '14

Commit ff795d614bfa ("ASoC: ab8500: Convert register I/O to regmap") initially converted the ab8500 CODEC driver to use regmap rather than legacy ASoC IO. This was reverted though in commit 63e6d43bf80d ("ASoC: ab8500: Revert to using custom I/O functions") since the inital conversion was not working properly. This was presumebly because the SOC_SINGLE_XR_SX controls, which are used by this driver, did not properly support regmap at that point. This has since been fixed in commit 6137a5ca326d ("ASoC: Prepare SOC_SINGLE_XR_SX controls for regmap"). So revert back to regmap again. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- I don't have the hardware, so I haven't been able to verify that it actually works now, but I think there is a very good chance. Lee, do you have access to a Snowball board and can give this a test? --- sound/soc/codecs/ab8500-codec.c | 62 +++++++++++++++++++---------------------- 1 file changed, 28 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 62cf231..fd43827 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -125,6 +125,8 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { + struct regmap *regmap; + /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -165,49 +167,35 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, - unsigned int reg) +static int ab8500_codec_read_reg(void *context, unsigned int reg, + unsigned int *value) { + struct device *dev = context; int status; - unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, - reg, &value8); - if (status < 0) { - dev_err(codec->dev, - "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - } else { - dev_dbg(codec->dev, - "%s: Read 0x%02x from register 0x%02x:0x%02x\n", - __func__, value8, (u8)AB8500_AUDIO, (u8)reg); - value = (unsigned int)value8; - } + status = abx500_get_register_interruptible(dev, AB8500_AUDIO, + reg, &value8); + *value = (unsigned int)value8; - return value; + return status; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int ab8500_codec_write_reg(void *context, unsigned int reg, + unsigned int value) { - int status; - - status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, - reg, value); - if (status < 0) - dev_err(codec->dev, - "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - else - dev_dbg(codec->dev, - "%s: Wrote 0x%02x into register %02x:%02x\n", - __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + struct device *dev = context; - return status; + return abx500_set_register_interruptible(dev, AB8500_AUDIO, + reg, value); } +static const struct regmap_config ab8500_codec_regmap = { + .reg_read = ab8500_codec_read_reg, + .reg_write = ab8500_codec_write_reg, +}; + /* * Controls - DAPM */ @@ -2564,9 +2552,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, - .read = ab8500_codec_read_reg, - .write = ab8500_codec_write_reg, - .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2591,6 +2576,15 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); + drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, + &ab8500_codec_regmap); + if (IS_ERR(drvdata->regmap)) { + status = PTR_ERR(drvdata->regmap); + dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", + __func__, status); + return status; + } + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, -- 1.8.0

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[alsa-devel] [PATCH v4 0/2] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 01 Sep '14

01 Sep '14

The NXP TDA998x HDMI transmitter may transmit audio to the HDMI link from 2 different sources, I2S and S/PDIF. This patch set adds a generic HDMI CODEC module which is used by the TDA998x driver to update the audio constraints from the display characteristics (EDID) and by the audio subsystem to connect the chosen audio source to the HDMI link. v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (2): ASoC:codecs: Add a generic HDMI audio CODEC drm/i2c:tda998x: Use the HDMI2 audio CODEC Documentation/devicetree/bindings/sound/hdmi2.txt | 32 ++++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 142 ++++++++++++++- include/sound/hdmi2.h | 24 +++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi2.c | 204 ++++++++++++++++++++++ 7 files changed, 400 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/hdmi2.txt create mode 100644 include/sound/hdmi2.h create mode 100644 sound/soc/codecs/hdmi2.c -- 2.1.0

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[alsa-devel] [PATCHv2] ASoC: fsl-sai: using 'lsb-first' property instead of 'big-endian-data'.
by Xiubo Li 01 Sep '14

01 Sep '14

The 'big-endian-data' property is originally used to indicate whether the LSB firstly or MSB firstly will be transmitted to the CODEC or received from the CODEC, and there has nothing relation to the memory data. Generally, if the audio data in big endian format, which will be using the bytes reversion, Here this can only be used to bits reversion. So using the 'lsb-first' instead of 'big-endian-data' can make the code to be readable easier and more easy to understand what this property is used to do. This property used for configuring whether the LSB or the MSB is transmitted first for the fifo data. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 8 ++++---- sound/soc/fsl/fsl_sai.c | 6 +++--- sound/soc/fsl/fsl_sai.h | 2 +- 3 files changed, 8 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 06a405e..4956b14 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -20,9 +20,9 @@ Required properties: See ../pinctrl/pinctrl-bindings.txt for details of the property values. - big-endian: Boolean property, required if all the FTM_PWM registers are big-endian rather than little-endian. -- big-endian-data: If this property is absent, the little endian mode will - be in use as default, or the big endian mode will be in use for all the - fifo data. +- lsb-first: Configures whether the LSB or the MSB is transmitted first for + the fifo data. If this property is absent, the MSB is transmitted first as + default, or the LSB is transmitted first. - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating that SAI will work in the synchronous mode (sync Tx with Rx) which means both the transimitter and receiver will send and receive data by following @@ -53,5 +53,5 @@ sai2: sai@40031000 { dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; big-endian; - big-endian-data; + lsb-first; }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a6eb784..7eeb1dd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, bool tx = fsl_dir == FSL_FMT_TRANSMITTER; u32 val_cr2 = 0, val_cr4 = 0; - if (!sai->big_endian_data) + if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - if (sai->big_endian_data) + if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); @@ -573,7 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) sai->sai_on_imx = true; - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 2cded44..3466720 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -132,7 +132,7 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool big_endian_data; + bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; -- 1.8.5

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[alsa-devel] [PATCH 1/3 v4 RESEND] mfd: arizona: Export function to control subsystem DVFS
by Richard Fitzgerald 01 Sep '14

01 Sep '14

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Moving this control from being a side-effect of the LDO1 regulator driver to a specific exported function. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/arizona-core.c | 89 ++++++++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/core.h | 12 +++++ 2 files changed, 101 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index cfc191a..04cde4b 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -94,6 +94,94 @@ int arizona_clk32k_disable(struct arizona *arizona) } EXPORT_SYMBOL_GPL(arizona_clk32k_disable); +int arizona_dvfs_up(struct arizona *arizona, unsigned int flags) +{ + unsigned int new_flags; + int ret = 0; + + mutex_lock(&arizona->subsys_max_lock); + + new_flags = arizona->subsys_max_rq | flags; + + if (arizona->subsys_max_rq != new_flags) { + switch (arizona->type) { + case WM5102: + case WM8997: + ret = regulator_set_voltage(arizona->dcvdd, + 1800000, 1800000); + if (ret != 0) { + dev_err(arizona->dev, + "Failed to set DCVDD (DVFS up): %d\n", + ret); + goto err; + } + + ret = regmap_update_bits(arizona->regmap, + ARIZONA_DYNAMIC_FREQUENCY_SCALING_1, + ARIZONA_SUBSYS_MAX_FREQ, 1); + if (ret != 0) { + dev_err(arizona->dev, + "Failed to enable subsys max: %d\n", + ret); + regulator_set_voltage(arizona->dcvdd, + 1200000, 1800000); + goto err; + } + break; + + default: + break; + } + + arizona->subsys_max_rq = new_flags; + } +err: + mutex_unlock(&arizona->subsys_max_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_up); + +int arizona_dvfs_down(struct arizona *arizona, unsigned int flags) +{ + int ret = 0; + + mutex_lock(&arizona->subsys_max_lock); + + if ((arizona->subsys_max_rq & flags) != flags) + dev_warn(arizona->dev, "Unbalanced DVFS down: %x\n", flags); + + arizona->subsys_max_rq &= ~flags; + + if (arizona->subsys_max_rq == 0) { + switch (arizona->type) { + case WM5102: + case WM8997: + ret = regmap_update_bits(arizona->regmap, + ARIZONA_DYNAMIC_FREQUENCY_SCALING_1, + ARIZONA_SUBSYS_MAX_FREQ, 0); + if (ret != 0) + dev_err(arizona->dev, + "Failed to disable subsys max: %d\n", + ret); + + ret = regulator_set_voltage(arizona->dcvdd, + 1200000, 1800000); + if (ret != 0) + dev_err(arizona->dev, + "Failed to set DCVDD (DVFS down): %d\n", + ret); + break; + + default: + break; + } + } + + mutex_unlock(&arizona->subsys_max_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_down); + static irqreturn_t arizona_clkgen_err(int irq, void *data) { struct arizona *arizona = data; @@ -641,6 +729,7 @@ int arizona_dev_init(struct arizona *arizona) dev_set_drvdata(arizona->dev, arizona); mutex_init(&arizona->clk_lock); + mutex_init(&arizona->subsys_max_lock); if (dev_get_platdata(arizona->dev)) memcpy(&arizona->pdata, dev_get_platdata(arizona->dev), diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index a614b33..4aec6a3 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -109,6 +109,9 @@ struct arizona { struct mutex clk_lock; int clk32k_ref; + struct mutex subsys_max_lock; + unsigned int subsys_max_rq; + struct snd_soc_dapm_context *dapm; int tdm_width[ARIZONA_MAX_AIF]; @@ -118,8 +121,17 @@ struct arizona { uint8_t dac_comp_enabled; }; +#define ARIZONA_DVFS_SR1_RQ 0x00000001 +#define ARIZONA_DVFS_SR2_RQ 0x00000002 +#define ARIZONA_DVFS_SR3_RQ 0x00000004 +#define ARIZONA_DVFS_ASR1_RQ 0x00000010 +#define ARIZONA_DVFS_ASR2_RQ 0x00000020 +#define ARIZONA_DVFS_ADSP1_RQ 0x00010000 + int arizona_clk32k_enable(struct arizona *arizona); int arizona_clk32k_disable(struct arizona *arizona); +int arizona_dvfs_up(struct arizona *arizona, unsigned int mask); +int arizona_dvfs_down(struct arizona *arizona, unsigned int mask); int arizona_request_irq(struct arizona *arizona, int irq, char *name, irq_handler_t handler, void *data); -- 1.7.2.5

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[alsa-devel] [PATCH alsa-lib] pcm: fix snd_pcm_mmap_hw_avail() near the boundary
by Alexander E. Patrakov 01 Sep '14

01 Sep '14

This function returned incorrect results when hw.ptr was near the boundary and hw.appl_ptr was near zero. Here "incorrect" means "greater than the boundary". The result was incorrect, because it was used as a return value of various *_rewindable() functions and also as the delay for ioplug. Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> --- src/pcm/pcm_local.h | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 80bbe59..74ebd60 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -461,13 +461,7 @@ static inline snd_pcm_sframes_t snd_pcm_mmap_capture_hw_avail(snd_pcm_t *pcm) static inline snd_pcm_sframes_t snd_pcm_mmap_hw_avail(snd_pcm_t *pcm) { - snd_pcm_sframes_t avail; - avail = *pcm->hw.ptr - *pcm->appl.ptr; - if (pcm->stream == SND_PCM_STREAM_PLAYBACK) - avail += pcm->buffer_size; - if (avail < 0) - avail += pcm->boundary; - return pcm->buffer_size - avail; + return pcm->buffer_size - snd_pcm_mmap_avail(pcm); } static inline const snd_pcm_channel_area_t *snd_pcm_mmap_areas(snd_pcm_t *pcm) -- 2.1.0

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