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

[alsa-devel] alsa-tools 1.0.28 regression with sblive/emu10k1: init_live shell script broken.
by Roc Vallès 25 Jun '14

25 Jun '14

I've recently upgraded to alsa-tools 1.0.28. There's apparently a regression: The script that is bundled for sound blaster live (emu10k1 in my case) soundcard initialization has been broken. Without this, the soundcard can only be used at a very basic level (just 2 speakers, no equalization or other effects, most inputs unavailable). Reporting here as the bugtracker is down. Correct run (1.0.17): # sh /usr/bin/init_live No Emux synth hwdep device is found Found hardware: "EMU10K1" "SigmaTel STAC9721,23" "AC97a:83847609" "0x1102" "0x8040" Hardware is initialized using a generic method Broken run (1.0.28): # /usr/bin/init_live /usr/bin/init_live: line 49: log: command not found /usr/bin/init_live: line 51: local: can only be used in a function /usr/bin/init_live: line 55: [: 0: unary operator expected /usr/bin/init_live: line 60: runlo10k1: command not found /usr/bin/init_live: line 62: log: command not found rename: not enough arguments Usage: rename [options] expression replacement file... Options: -v, --verbose explain what is being done -s, --symlink act on symlink target -h, --help display this help and exit -V, --version output version information and exit For more details see rename(1). /usr/bin/init_live: line 79: log: command not found /usr/bin/init_live: line 81: runlo10k1: command not found /usr/bin/init_live: line 84: connect: command not found /usr/bin/init_live: line 90: log: command not found /usr/bin/init_live: line 92: runlo10k1: command not found /usr/bin/init_live: line 95: copy: command not found /usr/bin/init_live: line 97: local: can only be used in a function /usr/bin/init_live: line 98: local: can only be used in a function /usr/bin/init_live: line 100: log: command not found /usr/bin/init_live: line 103: runlo10k1: command not found /usr/bin/init_live: line 105: connect: command not found /usr/bin/init_live: line 108: volume: command not found /usr/bin/init_live: line 110: local: can only be used in a function /usr/bin/init_live: line 111: local: can only be used in a function /usr/bin/init_live: line 112: local: can only be used in a function /usr/bin/init_live: line 113: local: can only be used in a function /usr/bin/init_live: line 115: log: command not found /usr/bin/init_live: line 118: runlo10k1: command not found /usr/bin/init_live: line 120: connect: command not found /usr/bin/init_live: line 123: master_volume: command not found /usr/bin/init_live: line 125: local: can only be used in a function /usr/bin/init_live: line 126: local: can only be used in a function /usr/bin/init_live: line 127: local: can only be used in a function /usr/bin/init_live: line 128: local: can only be used in a function /usr/bin/init_live: line 130: log: command not found /usr/bin/init_live: line 133: runlo10k1: command not found /usr/bin/init_live: line 135: connect: command not found /usr/bin/init_live: line 138: switch: command not found /usr/bin/init_live: line 140: local: can only be used in a function /usr/bin/init_live: line 141: local: can only be used in a function /usr/bin/init_live: line 142: local: can only be used in a function /usr/bin/init_live: line 144: log: command not found /usr/bin/init_live: line 148: runlo10k1: command not found /usr/bin/init_live: line 150: connect: command not found /usr/bin/init_live: line 153: switch_2x2: command not found /usr/bin/init_live: line 155: local: can only be used in a function /usr/bin/init_live: line 156: local: can only be used in a function /usr/bin/init_live: line 157: local: can only be used in a function /usr/bin/init_live: line 158: local: can only be used in a function /usr/bin/init_live: line 160: log: command not found /usr/bin/init_live: line 163: runlo10k1: command not found /usr/bin/init_live: line 165: connect: command not found /usr/bin/init_live: line 166: connect: command not found /usr/bin/init_live: line 169: switch_5x2: command not found /usr/bin/init_live: line 171: local: can only be used in a function /usr/bin/init_live: line 172: local: can only be used in a function /usr/bin/init_live: line 173: local: can only be used in a function /usr/bin/init_live: line 174: local: can only be used in a function /usr/bin/init_live: line 176: log: command not found /usr/bin/init_live: line 179: runlo10k1: command not found /usr/bin/init_live: line 181: connect: command not found /usr/bin/init_live: line 182: connect: command not found /usr/bin/init_live: line 185: master_switch: command not found /usr/bin/init_live: line 187: local: can only be used in a function /usr/bin/init_live: line 188: local: can only be used in a function /usr/bin/init_live: line 189: local: can only be used in a function /usr/bin/init_live: line 190: local: can only be used in a function /usr/bin/init_live: line 192: log: command not found /usr/bin/init_live: line 195: runlo10k1: command not found /usr/bin/init_live: line 197: connect: command not found /usr/bin/init_live: line 200: playback: command not found /usr/bin/init_live: line 205: log: command not found /usr/bin/init_live: line 208: switch: command not found /usr/bin/init_live: line 209: volume: command not found /usr/bin/init_live: line 211: add_output: command not found /usr/bin/init_live: line 214: playback_noswitch: command not found /usr/bin/init_live: line 219: log: command not found /usr/bin/init_live: line 222: volume: command not found /usr/bin/init_live: line 224: add_output: command not found /usr/bin/init_live: line 227: capture: command not found /usr/bin/init_live: line 232: log: command not found /usr/bin/init_live: line 235: switch: command not found /usr/bin/init_live: line 236: volume: command not found /usr/bin/init_live: line 238: connect: command not found /usr/bin/init_live: line 241: capture_noswitch: command not found /usr/bin/init_live: line 246: log: command not found /usr/bin/init_live: line 249: volume: command not found /usr/bin/init_live: line 251: connect: command not found /usr/bin/init_live: line 254: master: command not found /usr/bin/init_live: line 256: local: can only be used in a function /usr/bin/init_live: line 257: local: can only be used in a function /usr/bin/init_live: line 258: local: can only be used in a function /usr/bin/init_live: line 260: log: command not found /usr/bin/init_live: line 263: master_volume: command not found /usr/bin/init_live: line 264: master_switch: command not found /usr/bin/init_live: line 267: dump_patches: command not found

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[alsa-devel] snd_usb_endpoint_free
by Julia Lawall 25 Jun '14

25 Jun '14

The function snd_usb_endpoint_free in sound/usb/endpoint.c is defined as follows: void snd_usb_endpoint_free(struct list_head *head) { struct snd_usb_endpoint *ep; ep = list_entry(head, struct snd_usb_endpoint, list); release_urbs(ep, 1); kfree(ep); } I wonder if the final kfree should be list_del? In practice, this function is only used from snd_usb_audio_disconnect in sound/usb/card.c where the entire list is destroyed, but it seems like quite a generic function that someone may someday want to use for just freeing one entry. julia

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[alsa-devel] [PATCH] sound: oss: mpu401.c: Cleaning up variable is set more than once
by Rickard Strandqvist 25 Jun '14

25 Jun '14

A struct member variable is set to the same value more than once 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 deletion(-) diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 3bbc3ec..31d7f72 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -972,7 +972,6 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner) devc->m_busy = 0; devc->m_state = ST_INIT; devc->shared_irq = hw_config->always_detect; - devc->irq = hw_config->irq; spin_lock_init(&devc->lock); if (devc->irq < 0) -- 1.7.10.4

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[alsa-devel] [PATCH v6 0/3] Add ES8328 audio codec
by Sean Cross 25 Jun '14

25 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks. Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 61 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 719 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 224 +++++++ 12 files changed, 1501 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: aic32x4_set_dai_fmt(): drop unneded register access
by Peter Korsgaard 24 Jun '14

24 Jun '14

There's no point in reading the IFACE2 register when the result isn't used, so drop it. Signed-off-by: Peter Korsgaard <peter(a)korsgaard.com> --- sound/soc/codecs/tlv320aic32x4.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 89e41d2..630367f 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -336,7 +336,6 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1); iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2); - iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2); iface_reg_2 = 0; iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3); iface_reg_3 = iface_reg_3 & ~(1 << 3); -- 2.0.0

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[alsa-devel] [PATCH 1/6] ASoC: Intel: byt-max98090: Move MICBIAS widget to supply of Headset Mic
by Jarkko Nikula 24 Jun '14

24 Jun '14

Move "MICBIAS" as a supply widget to "Headset Mic" instead of keeping it between input pin "IN34" and "Headset Mic". Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-max98090.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 5fc98c64a3f4..a3c5f02a60fc 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { {"IN34", NULL, "Headset Mic"}, - {"IN34", NULL, "MICBIAS"}, - {"MICBIAS", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS"}, {"DMICL", NULL, "Int Mic"}, {"Headphone", NULL, "HPL"}, {"Headphone", NULL, "HPR"}, -- 2.0.0

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[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

2 5

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,

2 4

[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

2 13

Re: [alsa-devel] [PATCH] ASoC: Fix SOC_DOUBLE_R_SX_TLV volume mixer arguments
by Brian Austin 23 Jun '14

23 Jun '14

On Mon, 23 Jun 2014, Ryan Harvey wrote: > Remove unnecessary bit shifts. Correct min value to match datasheet. #Steps equals number of steps between min and max. > > This patch depends on 8f0367b53d887726b1f4b0b39e0f09b7cca2f79f > ASoC: cs42l56: Fix stereo channel register assignment for Headphone and > LineOut volume mixers. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- > sound/soc/codecs/cs42l52.c | 14 +++++++------- > sound/soc/codecs/cs42l56.c | 12 ++++++------ > sound/soc/codecs/cs42l73.c | 2 +- > 3 files changed, 14 insertions(+), 14 deletions(-) > > diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c > index c0581c5..7aed07e 100644 > --- a/sound/soc/codecs/cs42l52.c > +++ b/sound/soc/codecs/cs42l52.c > @@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { > CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), > > SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, > - CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), > + CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv), > > SOC_ENUM("Headphone Analog Gain", hp_gain_enum), > > SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, > - CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), > + CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), > > SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, > - CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), > + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), > > SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), > > @@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { > SOC_ENUM("MIC Bias Level", mic_bias_level_enum), > > SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, > - CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), > + CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), > SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", > CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, > - 6, 0x7f, 0x19, ipd_tlv), > + 0, 0x19, 0x7F, ipd_tlv), > > SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), > > @@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { > CS42L52_ADCB_MIXER_VOL, 7, 1, 1), > > SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, > - CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), > + CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv), > > SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", > CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, > - 0, 0x7f, 0x19, mix_tlv), > + 0, 0x19, 0x7f, mix_tlv), > SOC_DOUBLE_R("PCM Mixer Switch", > CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), > > diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c > index 8e68ef5..24fbffe 100644 > --- a/sound/soc/codecs/cs42l56.c > +++ b/sound/soc/codecs/cs42l56.c > @@ -421,15 +421,15 @@ static const struct soc_enum ng_delay_enum = > static const struct snd_kcontrol_new cs42l56_snd_controls[] = { > > SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME, > - CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv), > + CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv), > SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1), > > SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME, > - CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), > + CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), > SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1), > > SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME, > - CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), > + CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), > SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1), > > SOC_SINGLE_TLV("Analog Advisory Volume", > @@ -438,16 +438,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { > CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv), > > SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME, > - CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv), > + CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv), > SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR, > CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv), > SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1), > SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), > > SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, > - CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv), > + CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), > SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, > - CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv), > + CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), > > SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, > 0, 0x00, 1, tone_tlv), > diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c > index 7f28c58..bdb8316 100644 > --- a/sound/soc/codecs/cs42l73.c > +++ b/sound/soc/codecs/cs42l73.c > @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { > CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), > > SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, > - CS42L73_MICBPREPGABVOL, 5, 0x34, > + CS42L73_MICBPREPGABVOL, 0, 0x34, > 0x24, micpga_tlv), > > SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, > -- > 1.7.9.5 > > Acked-by: Brian Austin <brian.austin(a)cirrus.com> Thanks,

1 0

Re: [alsa-devel] [PATCH] Sound: Core: fixed assignment in if condition style error in hwdep.c
by Takashi Iwai 23 Jun '14

23 Jun '14

At Fri, 20 Jun 2014 08:50:35 -0700, Vincent Heuken wrote: > > This is a patch to the hwdep.c file that fixes one instance of > the following error: > ERROR: do not use assignment in if condition > > It also, incidentally, fixes one instance of the following warning: > WARNING: line over 80 characters > > Signed-off-by: Vincent Heuken <me(a)vincentheuken.com> If this is a preliminary work for any real fix, I'd happily apply this patch. But, if it's only about a coding style fix, I won't buy it alone. Let me know if it's the former case. thanks, Takashi > --- > sound/core/hwdep.c | 4 +++- > 1 file changed, 3 insertions(+), 1 deletion(-) > > diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c > index 69459e5..42d8324 100644 > --- a/sound/core/hwdep.c > +++ b/sound/core/hwdep.c > @@ -527,7 +527,9 @@ static void __init snd_hwdep_proc_init(void) > { > struct snd_info_entry *entry; > > - if ((entry = snd_info_create_module_entry(THIS_MODULE, "hwdep", NULL)) != NULL) { > + entry = snd_info_create_module_entry(THIS_MODULE, "hwdep", NULL); > + > + if (entry != NULL) { > entry->c.text.read = snd_hwdep_proc_read; > if (snd_info_register(entry) < 0) { > snd_info_free_entry(entry); > -- > 2.0.0 >

1 0

[alsa-devel] [PATCH] trivial: sound/core/seq/seq_memory.c: Fix closing brace followed by if
by Rasmus Villemoes 23 Jun '14

23 Jun '14

Add a newline, and remove a space while at it. Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- sound/core/seq/seq_memory.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 1e206de..4854c21 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -101,7 +101,8 @@ int snd_seq_dump_var_event(const struct snd_seq_event *event, len -= size; } return 0; - } if (! (event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) { + } + if (!(event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) { return func(private_data, event->data.ext.ptr, len); } -- 1.9.2

2 3

[alsa-devel] [RFT v3 0/3] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 23 Jun '14

23 Jun '14

Hi Mark, Liam and Lars, Sorry for the delay. Here is the third version of the multicodec series. I changed the way to store the multiple codec_dai as suggested by Lars and added the missing iterations in all the places pointed by Lars in his comments. Please note that I have a very simple setup based on BBB to check if the series is working, so it will be good to test it on various platforms to ensure this is not generating any regression. Hence this version is still RFT. The series is based on asoc/for-next (v3.15-rc2). Test and comments are welcome. Thanks, Benoit v1: http://comments.gmane.org/gmane.linux.alsa.devel/120532 v2: http://comments.gmane.org/gmane.linux.alsa.devel/121024 - Split the first version in several patches. - Remove the TDM fixup v3: - Fixed all the places that were not iterating over all the codecs. - Change the structure and split it. Benoit Cousson (2): ASoC: core: Add one dai_get_widget helper instead of two rtd based ones ASoC: core: Add support for DAI multicodec Misael Lopez Cruz (1): ASoC: core: Add helpers for dai link and aux dev init include/sound/soc-dai.h | 5 + include/sound/soc.h | 13 ++ sound/soc/soc-compress.c | 83 +++++--- sound/soc/soc-core.c | 401 +++++++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 ++++--- sound/soc/soc-pcm.c | 529 +++++++++++++++++++++++++++++++---------------- 6 files changed, 763 insertions(+), 339 deletions(-) -- 1.8.3.2

4 16

[alsa-devel] [PATCH] ALSA: compress: fix the struct alignment to 4 bytes
by Vinod Koul 23 Jun '14

23 Jun '14

In 64bit systems the compiler can default align to 8bytes causing mis-match with 32bit usermode. Avoid this is future by ensuring all the structures shared with usermode are packed and aligned to 4 bytes irrespective of arch used Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/compress_offload.h | 14 +++++++------- include/uapi/sound/compress_params.h | 14 +++++++------- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 21eed48..bd06dd0 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -39,7 +39,7 @@ struct snd_compressed_buffer { __u32 fragment_size; __u32 fragments; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_compr_params: compressed stream params @@ -51,7 +51,7 @@ struct snd_compr_params { struct snd_compressed_buffer buffer; struct snd_codec codec; __u8 no_wake_mode; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_compr_tstamp: timestamp descriptor @@ -70,7 +70,7 @@ struct snd_compr_tstamp { __u32 pcm_frames; __u32 pcm_io_frames; __u32 sampling_rate; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_compr_avail: avail descriptor @@ -80,7 +80,7 @@ struct snd_compr_tstamp { struct snd_compr_avail { __u64 avail; struct snd_compr_tstamp tstamp; -} __attribute__((packed)); +}__attribute__((packed, aligned(4))); enum snd_compr_direction { SND_COMPRESS_PLAYBACK = 0, @@ -107,7 +107,7 @@ struct snd_compr_caps { __u32 max_fragments; __u32 codecs[MAX_NUM_CODECS]; __u32 reserved[11]; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_compr_codec_caps: query capability of codec @@ -119,7 +119,7 @@ struct snd_compr_codec_caps { __u32 codec; __u32 num_descriptors; struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; -}; +}__attribute__((packed, aligned(4))); /** * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the @@ -140,7 +140,7 @@ enum { struct snd_compr_metadata { __u32 key; __u32 value[8]; -}; +}__attribute__((packed, aligned(4))); /** * compress path ioctl definitions diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index 165e705..54299ba 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -268,7 +268,7 @@ struct snd_enc_vorbis { __u32 max_bit_rate; __u32 min_bit_rate; __u32 downmix; -}; +}__attribute__((packed, aligned(4))); /** @@ -284,7 +284,7 @@ struct snd_enc_real { __u32 quant_bits; __u32 start_region; __u32 num_regions; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_enc_flac @@ -308,12 +308,12 @@ struct snd_enc_real { struct snd_enc_flac { __u32 num; __u32 gain; -}; +}__attribute__((packed, aligned(4))); struct snd_enc_generic { __u32 bw; /* encoder bandwidth */ __s32 reserved[15]; -}; +}__attribute__((packed, aligned(4))); union snd_codec_options { struct snd_enc_wma wma; @@ -321,7 +321,7 @@ union snd_codec_options { struct snd_enc_real real; struct snd_enc_flac flac; struct snd_enc_generic generic; -}; +}__attribute__((packed, aligned(4))); /** struct snd_codec_desc - description of codec capabilities * @max_ch: Maximum number of audio channels @@ -358,7 +358,7 @@ struct snd_codec_desc { __u32 formats; __u32 min_buffer; __u32 reserved[15]; -}; +}__attribute__((packed, aligned(4))); /** struct snd_codec * @id: Identifies the supported audio encoder/decoder. @@ -399,6 +399,6 @@ struct snd_codec { __u32 align; union snd_codec_options options; __u32 reserved[3]; -}; +}__attribute__((packed, aligned(4))); #endif -- 1.7.0.4

2 1

[alsa-devel] [PATCH v2 00/11] ASoC: Add support for DAPM at the component level
by Lars-Peter Clausen 23 Jun '14

23 Jun '14

Changes since v1: * Rebased onto asoc/for-next and 3.16-rc1 * Added patch to remove platform field from widget and DAPM context to prevent new users from being added Original cover letter: Hi, This series is the next step in componentization. It moves the DAPM support to the component level. This will allow any component to have DAPM widgets and routes, which was previously only possible for CODECs, and will allow any component to have DAPM widgets with controls (i.e. Mixers and MUXs), which was previously only possible for CODECs. At the end of this patchset the DAPM code will have (almost) no dependency on snd_soc_codec or snd_soc_platform and the dependency on snd_soc_component is confined to 4 small helper functions. The only reason why there is still snd_soc_codec/snd_soc_platform dependent code is because there are still a lot of drivers which use widget->codec/platform and dapm->codec/platform. So those still need to be initialized. This will be cleaned up in a separate series and once that's done the codec and platform fields will be removed from the widget and DAPM context struct. One of the major problems of moving DAPM to the component level is that the DAPM context is directly embedded in the snd_soc_codec and snd_soc_platform structs and there are many drivers directly accessing that embedded field. This means it is not possible to move the field without causing a bit of code churn in the drivers. The patch series solves this by adding two fields to the component struct. One pointer to a DAPM context, which points to the DAPM context that is used by the component. And one DAPM context, which is used as the default if no other DAPM context was provided. Neither of the fields is accessed directly, instead a inline helper function is added that returns the DAPM context for a component. This will make it possible to eventually seamlessly switch over once all direct references to CODEC and platform DAPM contexts have been eliminated. - Lars Lars-Peter Clausen (11): ASoC: Move name_prefix from CODEC to component ASoC: Move name and id from CODEC/platform to component ASoC: Split component registration into two steps ASoC: Auto disconnect pins from all DAPM contexts ASoC: Add a set_bias_level() callback to the DAPM context struct ASoC: Add DAPM support at the component level ASoC: Use component DAPM context for platforms ASoC: Add component level stream_event() and seq_notifier() support ASoC: Add component level set_bias_level() support ASoC: dapm: Remove DAI DAPM context ASoC: dapm: Remove platform field from widget and dapm context struct include/sound/soc-dai.h | 1 - include/sound/soc-dapm.h | 8 +- include/sound/soc.h | 53 +++++-- include/trace/events/asoc.h | 6 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/soc-cache.c | 7 +- sound/soc/soc-compress.c | 8 +- sound/soc/soc-core.c | 330 +++++++++++++++++++++++------------------ sound/soc/soc-dapm.c | 183 +++++++++++++---------- sound/soc/soc-pcm.c | 4 +- 10 files changed, 344 insertions(+), 258 deletions(-) -- 1.8.0

3 27

[alsa-devel] [RESEND PATCH v2 1/2] ASoC: samsung: s3c24{xx, 12}-i2s: port to use generic dmaengine API
by Vasily Khoruzhick 23 Jun '14

23 Jun '14

Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412 Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v2: use hardcoded dma channel number sound/soc/samsung/Kconfig | 9 +++---- sound/soc/samsung/dmaengine.c | 3 +++ sound/soc/samsung/s3c-i2s-v2.c | 17 +------------ sound/soc/samsung/s3c2412-i2s.c | 47 +++++++++++++++++----------------- sound/soc/samsung/s3c24xx-i2s.c | 56 +++++++++++++++++++---------------------- 5 files changed, 56 insertions(+), 76 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3507574..022c7a4 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,11 +1,10 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX + select S3C24XX_DMAC if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA if !ARCH_S3C24XX - select SND_S3C_DMA_LEGACY if ARCH_S3C24XX - select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX + select SND_S3C_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to @@ -19,7 +18,6 @@ config SND_S3C_DMA_LEGACY config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +25,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index 750ce58..72f27d1 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/amba/pl08x.h> +#include <linux/platform_data/dma-s3c24xx.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,8 @@ #ifdef CONFIG_ARCH_S3C64XX #define filter_fn pl08x_filter_id +#elif defined(CONFIG_ARCH_S3C24XX) +#define filter_fn s3c24xx_dma_filter #else #define filter_fn NULL #endif diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 79e7efb..1a74051 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -392,8 +392,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -424,13 +422,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, local_irq_restore(irqs); - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; case SNDRV_PCM_TRIGGER_STOP: @@ -644,12 +635,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, /* record our i2s structure for later use in the callbacks */ snd_soc_dai_set_drvdata(dai, i2s); - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->iis_pclk = clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); @@ -729,7 +714,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id, struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) { - struct snd_soc_dai_ops *ops = dai_drv->ops; + struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops; ops->trigger = s3c2412_i2s_trigger; if (!ops->hw_params) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d079445..ba084e3 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,25 +33,19 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .ch_name = "tx", .dma_size = 4, }; static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .ch_name = "rx", .dma_size = 4, }; @@ -63,6 +57,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) pr_debug("Entered %s\n", __func__); + samsung_asoc_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, + &s3c2412_i2s_pcm_stereo_in); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); if (ret) return ret; @@ -70,10 +67,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); + s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); return PTR_ERR(s3c2412_i2s.iis_cclk); } @@ -94,8 +90,6 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { clk_disable(s3c2412_i2s.iis_cclk); - clk_put(s3c2412_i2s.iis_cclk); - iounmap(s3c2412_i2s.regs); return 0; } @@ -105,18 +99,10 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); - struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); @@ -169,6 +155,19 @@ static const struct snd_soc_component_driver s3c2412_i2s_component = { static int s3c2412_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c2412_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c2412_i2s.regs == NULL) + return -ENXIO; + + s3c2412_i2s_pcm_stereo_out.dma_addr = res->start + S3C2412_IISTXD; + s3c2412_i2s_pcm_stereo_in.dma_addr = res->start + S3C2412_IISRXD; ret = s3c_i2sv2_register_component(&pdev->dev, -1, &s3c2412_i2s_component, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916..97600df 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,25 +31,19 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "tx", .dma_size = 2, }; static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "rx", .dma_size = 2, }; @@ -231,18 +225,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -251,11 +239,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; + dma_data->addr_width = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; + dma_data->addr_width = 2; break; default: return -EINVAL; @@ -270,8 +258,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(dai, substream); pr_debug("Entered %s\n", __func__); @@ -290,7 +276,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -380,14 +365,12 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) { pr_debug("Entered %s\n", __func__); - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; + samsung_asoc_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out, + &s3c24xx_i2s_pcm_stereo_in); - s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); + s3c24xx_i2s.iis_clk = devm_clk_get(dai->dev, "iis"); if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); @@ -474,6 +457,19 @@ static const struct snd_soc_component_driver s3c24xx_i2s_component = { static int s3c24xx_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO; + s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO; ret = snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1); -- 1.9.3

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[alsa-devel] [PATCH 1/2] SOUND: TLV320AIC32x4: Fixed Coding style Issues
by Shahina Shaik 22 Jun '14

22 Jun '14

From: Shahina Shaik <sharab.shaik(a)gmail.com> Fixed a brace coding style issue in the tlv320aic32x4.c Signed-off-by: Shahina Shaik <sharab.shaik(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117..1e50f15 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -626,21 +626,18 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } - if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); - } tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? AIC32X4_LDOCTLEN : 0; snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) tmp_reg |= AIC32X4_LDOIN_18_36; - } - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) tmp_reg |= AIC32X4_LDOIN2HP; - } snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); /* Mic PGA routing */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: fsl: refine DMA/FIQ dependencies
by Arnd Bergmann 22 Jun '14

22 Jun '14

Commit 31ee2bfd724ab ("ASoC: fsl: select SND_SOC_IMX_PCM_DMA where needed") started selecting SND_SOC_IMX_PCM_DMA and SND_SOC_IMX_PCM_FIQ for two drivers when building for i.MX. This has turned out too aggressive, as FIQ is only available for i.mx2 through i.mx5, but not i.mx6 or vybrid. Further, two more drivers have become user-selectable in the meantime, and they both depend on DMA for the imx platform as well. This changes the selection of FIQ to depend on the TZIC or AVIC interrupt controllers that actually export the imx specific FIQ interfaces, and adds the missing select statements for SAI and ESAI. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3793362..52bbe9f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -5,6 +5,7 @@ comment "Common SoC Audio options for Freescale CPUs:" config SND_SOC_FSL_SAI tristate "Synchronous Audio Interface (SAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to add Synchronous Audio Interface (SAI) @@ -15,7 +16,7 @@ config SND_SOC_FSL_SAI config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO help Say Y if you want to add Synchronous Serial Interface (SSI) @@ -27,7 +28,7 @@ config SND_SOC_FSL_SPDIF tristate "Sony/Philips Digital Interface module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) help Say Y if you want to add Sony/Philips Digital Interface (SPDIF) support for the Freescale CPUs. @@ -37,6 +38,7 @@ config SND_SOC_FSL_SPDIF config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface

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