November 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] sound: codecs: pcm1792a: clean function exit
by Raphael Poggi 14 Dec '15

14 Dec '15

Signed-off-by: Raphael Poggi <poggi.raph(a)gmail.com> --- sound/soc/codecs/pcm1792a.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 08bb486..27d54bc 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -91,14 +91,9 @@ static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); - int ret; - ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + return regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, PCM1792A_MUTE_MASK, !!mute); - if (ret < 0) - return ret; - - return 0; } static int pcm1792a_hw_params(struct snd_pcm_substream *substream, @@ -107,7 +102,7 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); - int val = 0, ret; + int val = 0; priv->rate = params_rate(params); @@ -145,12 +140,8 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream, val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; - ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + return regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); - if (ret < 0) - return ret; - - return 0; } static const struct snd_soc_dai_ops pcm1792a_dai_ops = { -- 2.1.0

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[alsa-devel] [PATCH 1/2] ALSA: usb-audio: Add a volume scale quirk for AudioQuest DragonFly
by Anssi Hannula 14 Dec '15

14 Dec '15

AudioQuest DragonFly DAC reports a volume control range of 0..50 (0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which is obviously incorrect and causes software using the dB information in e.g. volume sliders to have a massive volume difference in 100..102% range. The actual volume mapping seems to be neither linear volume nor linear dB scale, but instead quite close to the cubic mapping e.g. alsamixer uses, with a range of -53...0 dB. Add a quirk for DragonFly to use a custom dB mapping, based on my measurements, using a 10-item range TLV (so it still fits in alsa-lib MAX_TLV_RANGE_SIZE). Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the range is 0..50, so if this gets fixed/changed in later HW revisions it will no longer be applied. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- sound/usb/mixer_quirks.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 337c317ead6f..39d7f34e44e6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -37,6 +37,7 @@ #include <sound/control.h> #include <sound/hwdep.h> #include <sound/info.h> +#include <sound/tlv.h> #include "usbaudio.h" #include "mixer.h" @@ -1733,6 +1734,38 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer) return 0; } +static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer) +{ + struct usb_mixer_elem_list *list; + struct usb_mixer_elem_info *cval; + static const int unit_id = 7; + + /* approximation using 10 ranges based on output measurement on hw v1.2 */ + static const DECLARE_TLV_DB_RANGE(scale, + 0, 1, TLV_DB_MINMAX_ITEM(-5300, -4970), + 2, 5, TLV_DB_MINMAX_ITEM(-4710, -4160), + 6, 7, TLV_DB_MINMAX_ITEM(-3884, -3710), + 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560), + 15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324), + 17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031), + 20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393), + 27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032), + 32, 40, TLV_DB_MINMAX_ITEM(-968, -490), + 41, 50, TLV_DB_MINMAX_ITEM(-441, 0), + ); + + for (list = mixer->id_elems[unit_id]; list; list = list->next_id_elem) { + cval = (struct usb_mixer_elem_info *)list; + if (cval->control == UAC_FU_VOLUME && + cval->min == 0 && cval->max == 50) { + usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n"); + list->kctl->tlv.p = scale; + list->kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + list->kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + } +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1810,6 +1843,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x800c): /* Focusrite Scarlett 18i20 */ err = snd_scarlett_controls_create(mixer); break; + + case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + snd_dragonfly_quirk_db_scale(mixer); + break; } return err; -- 2.3.8

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[alsa-devel] [PATCH] ASoC: es8328: Fix shifts for mixer switches
by John Keeping 13 Dec '15

13 Dec '15

These are all off by one; the playback and bypass switches are the top two bits of the registers, which are at shifts 7 and 6 not 8 and 7. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/codecs/es8328.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 969e337..84f5eb0 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -205,18 +205,18 @@ static const struct snd_kcontrol_new es8328_right_line_controls = /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0), - SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0), }; /* Right Mixer */ static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { - SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0), - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0), }; static const char * const es8328_pga_sel[] = { -- 2.6.3.462.gbe2c914

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[alsa-devel] [PATCH 0/2] Add support for Texas Instruments pcm3168a codec
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add driver and binding docuemnt for Texas Instruments pcm3168a codec Damien.Horsley (2): ASoC: pcm3168a: Add binding document for pcm3168a codec ASoC: pcm3168a: Add driver for pcm3168a codec .../devicetree/bindings/sound/ti,pcm3168a.txt | 48 ++ sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/pcm3168a-i2c.c | 66 ++ sound/soc/codecs/pcm3168a-spi.c | 65 ++ sound/soc/codecs/pcm3168a.c | 766 +++++++++++++++++++++ sound/soc/codecs/pcm3168a.h | 111 +++ 7 files changed, 1079 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 sound/soc/codecs/pcm3168a-i2c.c create mode 100644 sound/soc/codecs/pcm3168a-spi.c create mode 100644 sound/soc/codecs/pcm3168a.c create mode 100644 sound/soc/codecs/pcm3168a.h -- 2.1.4

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[alsa-devel] [PATCH 0/2] Add support for Pistachio internal DAC
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 350 +++++++++++++++++++++ 4 files changed, 378 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

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[alsa-devel] [PATCH 00/16] ASoC: Intel: Skylake: Add support for loadable modules
by Vinod Koul 08 Dec '15

08 Dec '15

This series adds first the infrastructure for loadable modules in SKL aDSP. The loadable modules are loaded using load/unload IPC to aDSP thru code load DMA and can be configured by user thru TLV byte kcontrols for these modules. This also adds the implementation of new DAPM sinks for phrase detection modules. Dharageswari R (2): ASoC: Intel: Skylake: Add support for Load/Unload IPCs ASoC: Intel: Skylake: Add support for Loadable modules Dharageswari.R (1): ASoC: Intel: Skylake: Add support for Mic Select module Jeeja KP (8): ASoC: Intel: Skylake: Update DMIC DAIs and capabilities ASoC: Intel: Skylake: Add helper routines to handle module params ASoC: Intel: Skylake: Add helper routine to handle Algo parameter ASoC: Intel: Skylake: Add support to configure module params ASoC: Intel: Skylake: Add tlv byte kcontrols ASoC: Intel: Skylake: Add memory pages to widget data. ASoC: Intel: Skylake: Fix module init data correctly ASoC: Intel: Skylake: Add dai link for DMIC capture Mousami Jana (1): ASoC: Intel: Skylake: add LARGE_CONFIG_GET IPC support Omair M Abdullah (2): ASoC: Intel: Skylake: update mailbox uplink window offset and size ASoC: Intel: Skylake: read params from DSP if module is on Vinod Koul (2): ASoC: Intel: Skylake: add wov as int sink ASoc: Intel: Skylake: Fix the dapm machine map sound/soc/intel/boards/skl_rt286.c | 58 ++++++- sound/soc/intel/skylake/skl-messages.c | 105 +++++++++++- sound/soc/intel/skylake/skl-pcm.c | 23 ++- sound/soc/intel/skylake/skl-sst-dsp.h | 18 +- sound/soc/intel/skylake/skl-sst-ipc.c | 100 +++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 9 + sound/soc/intel/skylake/skl-sst.c | 175 +++++++++++++++++++ sound/soc/intel/skylake/skl-topology.c | 242 ++++++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 32 +++- sound/soc/intel/skylake/skl-tplg-interface.h | 23 ++- 10 files changed, 761 insertions(+), 24 deletions(-) -- 2.4.3

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[alsa-devel] [PATCH v2 00/13] ASoC: topology: Add support for PCM, BE & CC links
by mengdong.lin＠linux.intel.com 08 Dec '15

08 Dec '15

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows the topology core to create PCM devices dynamically, as well as configure BE and CC DAI links. Mengdong Lin (12): ASoC: Vendor drivers get a link's runtime by snd_soc_get_pcm_runtime() ASoC: Change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list & define API to add a link ASoC: Add add_dai_link ops for a soc card ASoC: soc_bind_dai_link() directly returns success for a bound DAI link ASoC: Bind new DAI links after probing components ASoC: The soc card can have auxiliary components ASoC: Support adding a DAI dynamically ASoC: topology: Add PCM DAIs dynamically when loading them ASoC: topology: Add support for FE DAI links Vedang Patel (1): ASoC: topology: Add support for BE and CC DAI Links include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 22 +- include/sound/soc.h | 38 +- sound/soc/fsl/fsl-asoc-card.c | 5 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/rsrc-card.c | 6 +- sound/soc/soc-core.c | 629 ++++++++++++++++++--------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 284 ++++++++++-- sound/soc/tegra/tegra_wm8903.c | 3 +- 24 files changed, 879 insertions(+), 324 deletions(-) -- 1.9.1

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[alsa-devel] Sound UsbPre : minimal patch for support
by shal＠free.fr 07 Dec '15

07 Dec '15

Hello, I have a Sound UsbPre (the first not the UsbPRe2) sound card. There is no support for this card in alsa. I do a usb packet sniffing on Windows for detect problem. Some control quirk should be added at boot in order to use the card. It works to play and record in 44100 khz. It works only with 44100khz (for example, if you play a 88200 file, it will be played too slow). This patch is not well tested, work on my desktop : Linux Ubuntu with 4.2.19-lowlatency kernel. The button emulation is not done (by a control packet, you can change the card configuration, like phantom power activation). If somebody is interested, I can provide needed information. If Alsa's developer is interested, I can provide support for test patches or develop a second version of this patch in order to integrate the support for this card in the linux kernel Regards, Olivier linux-source-4.2.0/sound/usb$ diff -rub quirks.c.ORIG quirks.c --- quirks.c.ORIG 2015-11-24 22:43:16.878635155 +0100 +++ quirks.c 2015-11-26 20:56:09.492126460 +0100 @@ -601,6 +601,40 @@ return 0; } +/* Add some control messages needed by Sound Devices UsbPre. + * Values here are chosen based on sniffing USB traffic + * under Windows.*/ +static int snd_usb_usbpre_boot_quirk(struct usb_device *dev) +{ + u8 buf = 1; + u8 data1[1] = {0x81}; + u8 data2[1] = {0xc1}; + u8 data3[3] = {0x44,0xac,0x00}; + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 496, data1, 1); + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x81 , + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 0, &buf, 1); + if (buf == 0) { + return -ENODEV; + } + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 496, data2, 1); + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x81 , + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 0, &buf, 1); + if (buf == 0) { + return -ENODEV; + } + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_ENDPOINT, + 0x0100, 133, data3, 3); + return 0; +} + static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) { int err; @@ -985,6 +1019,10 @@ /* SB Audigy 2 NX needs its own boot-up magic, too */ return snd_usb_audigy2nx_boot_quirk(dev); + case USB_ID(0x0926, 0x0100): + /* Sound Devices UsbPre needs its own boot-up magic, too */ + return snd_usb_usbpre_boot_quirk(dev); + case USB_ID(0x10f5, 0x0200): /* C-Media CM106 / Turtle Beach Audio Advantage Roadie */ return snd_usb_cm106_boot_quirk(dev);

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[alsa-devel] M-audio 410 : complex procedure to get a sound
by ROUGE ETIENNE 06 Dec '15

06 Dec '15

Hello ! I've just bought a M-audio firewire 410 and I'm very thankfull to takashi's work that allows us to use the device almost like an USB one ! I can have sound and have access to the routing controls. However 2 bugs makes it a bit tedious : NOTE : I use an up to date freshly installed ubuntustudio 15.10. The only modifications I made was 1) deleting pulse 2) select the soundcard as default in asoundrc 1) I do not have sound out of the box, but I can manage to have sound by starting ffado-mixer then force restart alsa. After that everything works fine.... but I gotta do it at every restart which isn't nice 2) ffado-mixer suffers several bugs : - if I close the window I can't restart it again unless I kill ffado-dbus-server (it says : "another instance of ffado-mixer is already running). - I got few errors on startup "invalid samplerate for input"... don't know it it has any impact - when I tweak too much ffado-mixer (AFAIK, it's random, but happens within few minuts) the mouse shows a 'hand closed' cursor and I can't click anything. The system remains fully functionnal, but I can't close anything nor click... If I start an emergency terminal I can kill whatever I want, but the problem remains. I'll try with a fedora this evening to see if it's not related with ubuntustudio, but I'd be glad to have some help with those 2 issues (especially the stalling ffado-mixer) thanks ! Etienne

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Re: [alsa-devel] [bug] Volume at maximum when track with different frequency is played with my RME sound card
by Maeda 04 Dec '15

04 Dec '15

Hi ! Someone's here ? Thanks. Le 23/10/2015 18:30, Maeda a écrit : > Hi all ! > > I don't know if the bug is known from your team, but some people > advice me to send an email to the Alsa devel list if nobody answer in > the kernel's bugtracking. > > Here is the bug I have : > https://bugzilla.kernel.org/show_bug.cgi?id=105771 > > What do you think about it ? > > If I was wrong sending this mail to this list, I'm sorry :) > > Thanks. > Have a nice day.

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