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

[alsa-devel] [PATCH v2 1/2] ASoC: cs4265: bindings: sound: Add binding for CS4265 CODEC.
by Paul Handrigan 24 Jun '14

24 Jun '14

This patch adds binding documentation for the Cirrus Logic CS4265 I2C CODEC. Changes for v2: * Change DT reset pin name to "reset-gpios" Signed-off-by: Paul Handrigan <paul.handrigan(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4265.txt | 29 ++++++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4265.txt diff --git a/Documentation/devicetree/bindings/sound/cs4265.txt b/Documentation/devicetree/bindings/sound/cs4265.txt new file mode 100644 index 0000000..380fff8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4265.txt @@ -0,0 +1,29 @@ +CS4265 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "cirrus,cs4265" + + - reg : the I2C address of the device for I2C. The I2C address depends on + the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f. + If it is low, the i2c address is 0x4e. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Examples: + +codec_ad0_high: cs4265@4f { /* AD0 Pin is high */ + compatible = "cirrus,cs4265"; + reg = <0x4f>; +}; + + +codec_ad0_low: cs4265@4e { /* AD0 Pin is low */ + compatible = "cirrus,cs4265"; + reg = <0x4e>; +}; -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Adjust speaker HPF and add LED support for HP Spectre 13
by Takashi Iwai 24 Jun '14

24 Jun '14

HP Spectre 13 has the IDT 92HD95 codec, and BIOS seems to set the default high-pass filter in some "safer" range, which results in the very soft tone from the built-in speakers in contrast to Windows. Also, the mute LED control is missing, since 92HD95 codec still has no HP-specific fixups for GPIO setups. This patch adds these missing features: the HPF is adjusted by the vendor-specific verb, and the LED is set up from a DMI string (but with the default polarity = 0 assumption due to the incomplete BIOS on the given machine). Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=74841 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 5 +++ sound/pci/hda/patch_sigmatel.c | 58 +++++++++++++++++++++++++++- 2 files changed, 62 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 85c362d8ea34..d1ab5e17eb13 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -286,6 +286,11 @@ STAC92HD83* hp-inv-led HP with broken BIOS for inverted mute LED auto BIOS setup (default) +STAC92HD95 +========== + hp-led LED support for HP laptops + hp-bass Bass HPF setup for HP Spectre 13 + STAC9872 ======== vaio VAIO laptop without SPDIF diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7f40a150899c..3744ea4e843d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -122,6 +122,12 @@ enum { }; enum { + STAC_92HD95_HP_LED, + STAC_92HD95_HP_BASS, + STAC_92HD95_MODELS +}; + +enum { STAC_925x_REF, STAC_M1, STAC_M1_2, @@ -4128,6 +4134,48 @@ static const struct snd_pci_quirk stac9205_fixup_tbl[] = { {} /* terminator */ }; +static void stac92hd95_fixup_hp_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + if (find_mute_led_cfg(codec, spec->default_polarity)) + codec_dbg(codec, "mute LED gpio %d polarity %d\n", + spec->gpio_led, + spec->gpio_led_polarity); +} + +static const struct hda_fixup stac92hd95_fixups[] = { + [STAC_92HD95_HP_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd95_fixup_hp_led, + }, + [STAC_92HD95_HP_BASS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x1a, 0x795, 0x00}, /* HPF to 100Hz */ + {} + }, + .chained = true, + .chain_id = STAC_92HD95_HP_LED, + }, +}; + +static const struct snd_pci_quirk stac92hd95_fixup_tbl[] = { + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1911, "HP Spectre 13", STAC_92HD95_HP_BASS), + {} /* terminator */ +}; + +static const struct hda_model_fixup stac92hd95_models[] = { + { .id = STAC_92HD95_HP_LED, .name = "hp-led" }, + { .id = STAC_92HD95_HP_BASS, .name = "hp-bass" }, + {} +}; + + static int stac_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -4580,10 +4628,16 @@ static int patch_stac92hd95(struct hda_codec *codec) spec->gen.beep_nid = 0x19; /* digital beep */ spec->pwr_nids = stac92hd95_pwr_nids; spec->num_pwrs = ARRAY_SIZE(stac92hd95_pwr_nids); - spec->default_polarity = -1; /* no default cfg */ + spec->default_polarity = 0; codec->patch_ops = stac_patch_ops; + snd_hda_pick_fixup(codec, stac92hd95_models, stac92hd95_fixup_tbl, + stac92hd95_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + + stac_setup_gpio(codec); + err = stac_parse_auto_config(codec); if (err < 0) { stac_free(codec); @@ -4592,6 +4646,8 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->proc_widget_hook = stac92hd_proc_hook; + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; } -- 2.0.0

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[alsa-devel] [PATCH 1/2] ALSA: hda - Make a SND_HDA_PIN_QUIRK macro
by David Henningsson 24 Jun '14

24 Jun '14

This is cosmetical - it makes the new pin quirk table look better. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_local.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) Hi Takashi, If I were you, I'd take these patches for 3.16 even though it's not strictly a regression - as it would make maintenance easier not to have any release with the quirk table in non-macro style. Sorry for not making this macro earlier. diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ebd1fa6..4e2d486 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -417,6 +417,27 @@ struct snd_hda_pin_quirk { int value; /* quirk value */ }; +#ifdef CONFIG_SND_DEBUG_VERBOSE + +#define SND_HDA_PIN_QUIRK(_codec, _subvendor, _name, _value, _pins...) \ + { .codec = _codec,\ + .subvendor = _subvendor,\ + .name = _name,\ + .value = _value,\ + .pins = (const struct hda_pintbl[]) { _pins } \ + } +#else + +#define SND_HDA_PIN_QUIRK(_codec, _subvendor, _name, _value, _pins...) \ + { .codec = _codec,\ + .subvendor = _subvendor,\ + .value = _value,\ + .pins = (const struct hda_pintbl[]) { _pins } \ + } + +#endif + + /* fixup types */ enum { HDA_FIXUP_INVALID, -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Add pin quirk for Dell XPS 15
by David Henningsson 24 Jun '14

24 Jun '14

Two bug reporters with Dell XPS 15 report that they need to use the dell-headset-multi model to get the headset mic working. The two bug reporters have different PCI SSID (1028:05fd and 1028:05fe) but this pin quirk matches both. BugLink: https://bugs.launchpad.net/bugs/1331915 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f52f096..229a189 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6258,6 +6258,27 @@ static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { }, .value = ALC668_FIXUP_AUTO_MUTE, }, + { + .codec = 0x10ec0668, + .subvendor = 0x1028, +#ifdef CONFIG_SND_DEBUG_VERBOSE + .name = "Dell XPS 15", +#endif + .pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x15, 0x0321101f}, + {0x16, 0x40000000}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x411111f0}, + {0x1d, 0x40d6832d}, + {0x1e, 0x411111f0}, + {0x1f, 0x411111f0}, + }, + .value = ALC668_FIXUP_AUTO_MUTE, + }, {} }; -- 1.9.1

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[alsa-devel] [PATCH] sound: oss: mpu401.c: Cleaning up missing break in a case
by Rickard Strandqvist 24 Jun '14

24 Jun '14

Added a missed break in a case statement This was found using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/oss/mpu401.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 25e4609..400896b 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -316,6 +316,7 @@ static int mpu_input_scanner(struct mpu_config *devc, unsigned char midic) case 0xf6: /* printk( "tune_request\n"); */ devc->m_state = ST_INIT; + break; /* * Real time messages -- 1.7.10.4

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[alsa-devel] [PATCH v3 0/4] ASoc: samsung: update s3c24xx to dmaengine and common clk framework
by Vasily Khoruzhick 24 Jun '14

24 Jun '14

This series updates s3c24{xx,12}-i2s driver to use newer DMA and CLK APIs. v2: use hardcoded dma channel number v3: rebase onto v3.16-rc1, add common clk framework-related changes Vasily Khoruzhick (4): ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API ASoC: samsung: drop support for legacy S3C24XX DMA API ASoC: samsung: s3c2412-i2s: Move to clk_prepare_enable/clk_disable_unprepare ASoC: samsung: s3c24xx-i2s: Move to clk_prepare_enable/clk_disable_unprepare sound/soc/samsung/Kconfig | 12 +- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/dma.c | 454 ---------------------------------------- sound/soc/samsung/dmaengine.c | 3 + sound/soc/samsung/s3c-i2s-v2.c | 17 +- sound/soc/samsung/s3c2412-i2s.c | 51 +++-- sound/soc/samsung/s3c24xx-i2s.c | 62 +++--- 7 files changed, 61 insertions(+), 540 deletions(-) delete mode 100644 sound/soc/samsung/dma.c -- 2.0.0

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[alsa-devel] [PATCH 0/3] More simple-card dai node clock parameters
by Jyri Sarha 24 Jun '14

24 Jun '14

These changes target to sii9022 HDMI audio support, but are independent fixes and additions. To cover more cases with simple-card it should be possible to select all parameters for set_sysclk() callback. The default behavior is not changed by these patches. Jyri Sarha (3): ASoC: simple-card: Make u32 DT parameter handling 64-bit proof ASoC: simple-card: Add system-clock-direction DT parameter to dai nodes ASoC: simple-card: Add system-clock-id DT parameter to dai nodes .../devicetree/bindings/sound/simple-card.txt | 3 ++ include/sound/simple_card.h | 2 ++ sound/soc/generic/simple-card.c | 31 +++++++++++++++----- 3 files changed, 29 insertions(+), 7 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Correct S24_3LE support
by Peter Ujfalusi 24 Jun '14

24 Jun '14

Correct the hw_params callback to configure the codec correctly in case of S24_3LE format. S24_LE is not defined as supported format for the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e12fafbb1e09..5360772bc1ad 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: -- 2.0.0

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Re: [alsa-devel] [PATCH] ASoC: cs42l56: Fix stereo channel register assignment for Headphone and LineOut volume mixers
by Brian Austin 23 Jun '14

23 Jun '14

On Mon, 23 Jun 2014, Ryan Harvey wrote: > Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- > sound/soc/codecs/cs42l56.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c > index fdc4bd2..8e68ef5 100644 > --- a/sound/soc/codecs/cs42l56.c > +++ b/sound/soc/codecs/cs42l56.c > @@ -445,9 +445,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { > SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), > > SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, > - CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv), > + CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv), > SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, > - CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv), > + CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv), > > SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, > 0, 0x00, 1, tone_tlv), > -- > 1.7.9.5 > > Acked-by: Brian Austin <brian.austin(a)cirrus.com> Thanks,

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[alsa-devel] [PATCH 0/4 v2] arizona: Improvements to codec DVFS control
by Charles Keepax 23 Jun '14

23 Jun '14

Hi, Respin of Richard's DVFS series, from me since he is on holiday. I have based these on Mark's ASoC for-next branch as there a few changes in there to the same parts of code. But let me know if you would rather have a series based on Lee's tree instead. This set of changes cleans up the way the drivers handle the control of internal subsystem clocks for codecs with DVFS. This previously relied on the clock control happening as a side-effect of changing the regulator voltage, rather than explicitly, which assumed that you were using the built-in regulator. Also DVFS control was being done from the ADSP driver despite that driver supposedly being codec- agnostic. There are other cases on some codecs where DVFS control is needed so this sequence of patches cleans up the way DVFS is implemented and adds the extra case of DVFS boost for higher sample rates on WM8997 and WM5102 Changes since v1: - Changed mask variable to flags to be more consistent with its usage - Fixed minor tabbing issue - Adding missing SND_SOC_DAPM_PRE_PMD event flag in WM_ADSP2_E Richard Fitzgerald (4): mfd: arizona: Export function to control subsystem DVFS ASoC: wm_adsp: Move DVFS control into codec driver ASoC: arizona: Add DVFS handling for sample rate control regulator: arizona-ldo1: Do not control clocking from regulator drivers/mfd/arizona-core.c | 84 ++++++++++++++++++++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 5 -- include/linux/mfd/arizona/core.h | 12 +++++ sound/soc/codecs/arizona.c | 22 +++++++++- sound/soc/codecs/wm5102.c | 47 ++++++++++++++++++++- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm_adsp.c | 73 +-------------------------------- sound/soc/codecs/wm_adsp.h | 13 +++--- 8 files changed, 171 insertions(+), 87 deletions(-) -- 1.7.2.5

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