April 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/8] ASoC: add snd_compress_ops and replace
by Kuninori Morimoto 20 Apr '20

20 Apr '20

Hi Mark Current snd_soc_component_driver has compr_ops, but it is created for ALSA, not for ALSA SoC. This means it doesn't know the callback is for which component, thus, each callback needs to get component via lookup function. It is not good design, and will not work in the future, because ASoC can have multipul same name component which current lookup function can't handle. This patch-set adds new snd_compress_ops and call callbacks with component. Kuninori Morimoto (8): ASoC: soc-compress: add snd_compress_ops ASoC: codec: wm_adsp: use snd_compress_ops ASoC: uniphier: use snd_compress_ops ASoC: qcom: q6sp6: use snd_compress_ops ASoC: intel: atom: use snd_compress_ops ASoC: sof: use snd_compress_ops ASoC: sprd: use snd_compress_ops ASoC: soc-compress: remove snd_compr_ops include/sound/soc-component.h | 40 ++++++++- sound/soc/codecs/cs47l15.c | 9 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 9 +- sound/soc/codecs/cs47l85.c | 9 +- sound/soc/codecs/cs47l90.c | 9 +- sound/soc/codecs/cs47l92.c | 9 +- sound/soc/codecs/wm5102.c | 9 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm_adsp.c | 18 ++-- sound/soc/codecs/wm_adsp.h | 18 ++-- .../intel/atom/sst-mfld-platform-compress.c | 43 +++++---- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 51 ++++++----- sound/soc/soc-compress.c | 89 ++++++++++--------- sound/soc/sof/compress.c | 34 ++++--- sound/soc/sof/compress.h | 34 ++++--- sound/soc/sof/pcm.c | 4 +- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 49 +++++----- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sprd/sprd-pcm-dma.h | 2 +- sound/soc/uniphier/aio-compress.c | 45 ++++++---- sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/uniphier/aio.h | 2 +- 26 files changed, 299 insertions(+), 210 deletions(-) -- 2.17.1

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[PATCH] ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos
by Takashi Iwai 20 Apr '20

20 Apr '20

TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need yet more quirks for the proper control names. This patch provides the mapping table for those boards, correcting the FU names for volume and mute controls as well as the terminal names for jack controls. It also improves build_connector_control() not to add the directional suffix blindly if the string is given from the mapping table. With this patch applied, the new UCM profiles will be effective. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 12 +++++++++--- sound/usb/mixer_maps.c | 24 +++++++++++++++++++++--- sound/usb/quirks-table.h | 14 ++++++++++++++ 3 files changed, 44 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e7b9040a54e6..ecd5036a0b44 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1776,8 +1776,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, { struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + const struct usbmix_name_map *map; - if (check_ignored_ctl(find_map(imap, term->id, 0))) + map = find_map(imap, term->id, 0); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1809,8 +1811,12 @@ static void build_connector_control(struct usb_mixer_interface *mixer, usb_mixer_elem_info_free(cval); return; } - get_connector_control_name(mixer, term, is_input, kctl->id.name, - sizeof(kctl->id.name)); + + if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) + strlcat(kctl->id.name, " Jack", sizeof(kctl->id.name)); + else + get_connector_control_name(mixer, term, is_input, kctl->id.name, + sizeof(kctl->id.name)); kctl->private_free = snd_usb_mixer_elem_free; snd_usb_mixer_add_control(&cval->head, kctl); } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index b4e77000f441..fb2c4a992951 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -369,6 +369,24 @@ static const struct usbmix_name_map asus_rog_map[] = { {} }; +/* TRX40 mobos with Realtek ALC1220-VB */ +static const struct usbmix_name_map trx40_mobo_map[] = { + { 18, NULL }, /* OT, IEC958 - broken response, disabled */ + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Speaker" }, /* OT */ + { 22, "Speaker Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 17, "Front Headphone" }, /* OT */ + { 23, "Front Headphone Playback" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + { 24, "IEC958 Playback" }, /* FU */ + {} +}; + /* * Control map entries */ @@ -500,7 +518,7 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), @@ -512,11 +530,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e009d584e7d0..7598d6d6740c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3635,4 +3635,18 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +#define ALC1220_VB_DESKTOP(vend, prod) { \ + USB_DEVICE(vend, prod), \ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + .vendor_name = "Realtek", \ + .product_name = "ALC1220-VB-DT", \ + .profile_name = "Realtek-ALC1220-VB-Desktop", \ + .ifnum = QUIRK_NO_INTERFACE \ + } \ +} +ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ +ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ +ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +#undef ALC1220_VB_DESKTOP + #undef USB_DEVICE_VENDOR_SPEC -- 2.16.4

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[PATCH] ALSA: usb-audio: Print more information in stream proc files
by Takashi Iwai 20 Apr '20

20 Apr '20

For more debug and usability information, add the entry showing the DSD raw states and the channel mapping in each stream proc file. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/proc.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 4174ad11fca6..5a36e192ebb0 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -54,6 +54,38 @@ void snd_usb_audio_create_proc(struct snd_usb_audio *chip) proc_audio_usbid_read); } +static const char * const channel_labels[] = { + [SNDRV_CHMAP_NA] = "N/A", + [SNDRV_CHMAP_MONO] = "MONO", + [SNDRV_CHMAP_FL] = "FL", + [SNDRV_CHMAP_FR] = "FR", + [SNDRV_CHMAP_FC] = "FC", + [SNDRV_CHMAP_LFE] = "LFE", + [SNDRV_CHMAP_RL] = "RL", + [SNDRV_CHMAP_RR] = "RR", + [SNDRV_CHMAP_FLC] = "FLC", + [SNDRV_CHMAP_FRC] = "FRC", + [SNDRV_CHMAP_RC] = "RC", + [SNDRV_CHMAP_SL] = "SL", + [SNDRV_CHMAP_SR] = "SR", + [SNDRV_CHMAP_TC] = "TC", + [SNDRV_CHMAP_TFL] = "TFL", + [SNDRV_CHMAP_TFC] = "TFC", + [SNDRV_CHMAP_TFR] = "TFR", + [SNDRV_CHMAP_TRL] = "TRL", + [SNDRV_CHMAP_TRC] = "TRC", + [SNDRV_CHMAP_TRR] = "TRR", + [SNDRV_CHMAP_TFLC] = "TFLC", + [SNDRV_CHMAP_TFRC] = "TFRC", + [SNDRV_CHMAP_LLFE] = "LLFE", + [SNDRV_CHMAP_RLFE] = "RLFE", + [SNDRV_CHMAP_TSL] = "TSL", + [SNDRV_CHMAP_TSR] = "TSR", + [SNDRV_CHMAP_BC] = "BC", + [SNDRV_CHMAP_RLC] = "RLC", + [SNDRV_CHMAP_RRC] = "RRC", +}; + /* * proc interface for list the supported pcm formats */ @@ -97,6 +129,27 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); snd_iprintf(buffer, " Bits: %d\n", fp->fmt_bits); + + if (fp->dsd_raw) + snd_iprintf(buffer, " DSD raw: DOP=%d, bitrev=%d\n", + fp->dsd_dop, fp->dsd_bitrev); + + if (fp->chmap) { + const struct snd_pcm_chmap_elem *map = fp->chmap; + int c; + + snd_iprintf(buffer, " Channel map:"); + for (c = 0; c < map->channels; c++) { + if (map->map[c] >= sizeof(channel_labels) || + !channel_labels[map->map[c]]) + snd_iprintf(buffer, " --"); + else + snd_iprintf(buffer, " %s", + channel_labels[map->map[c]]); + } + snd_iprintf(buffer, "\n"); + } + // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } -- 2.16.4

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No Sound Realtek ALC294ASUS UX534FT
by LACOUR François 19 Apr '20

19 Apr '20

Hi, I can't find a issue to this problem. Have you ? Thanks François Sent with [ProtonMail](https://protonmail.com) Secure Email.

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[PATCH] ALSA: hda: Remove ASUS ROG Zenith from the blacklist
by Takashi Iwai 19 Apr '20

19 Apr '20

The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. This patch reverts the corresponding entry as a temporary solution. Although Zenith II and co will see get the empty HD-audio bus again, it'd be merely resource wastes and won't affect the functionality, so it's no end of the world. We'll need to address this later, e.g. by either switching to DMI string matching or using PCI ID & SSID pairs. Fixes: 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") Reported-by: Johnathan Smithinovic <johnathan.smithinovic(a)gmx.at> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a5fab12defde..d7adae316c0d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2092,7 +2092,6 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * should be ignored from the beginning. */ static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), {} -- 2.16.4

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[PATCH 0/7] Add H6 I2S support
by Clément Péron 19 Apr '20

19 Apr '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 286 ++++++++++++++++-- 3 files changed, 295 insertions(+), 24 deletions(-) -- 2.20.1

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[PATCH v2] ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
by Alexander Tsoy 18 Apr '20

18 Apr '20

Many Focusrite devices supports a limited set of sample rates per altsetting. These includes audio interfaces with ADAT ports: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - Scarlett 18i8 3rd gen, 18i20 3rd gen; - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Maximum rate is exposed in the last 4 bytes of Format Type descriptor which has a non-standard bLength = 10. Tested-by: Alexey Skobkin <skobkin-ru(a)ya.ru> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- Notes: v2: updated description sound/usb/format.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50e1874c847c..5ffb457cc88c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -277,6 +277,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, return false; } +/* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) -- 2.25.3

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[PATCH] ALSA: hda/realtek - Fix unexpected init_amp override
by Takashi Iwai 18 Apr '20

18 Apr '20

The commit 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") changed the way to assign spec->init_amp field that specifies the way to initialize the amp. Along with the change, the commit also replaced a few fixups that set spec->init_amp in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather aligning to the other fixups, and not supposed to change the actual behavior. However, this change turned out to cause a regression on FSC S7020, which hit exactly the above. The reason was that there is still one place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE call, namely in alc_ssid_check(). This patch fixes the regression by adding the proper spec->init_amp override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED. Fixes: 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc5557d79c43..54e1c9c0a33f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -797,9 +797,11 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) { if (!alc_subsystem_id(codec, ports)) { struct alc_spec *spec = codec->spec; - codec_dbg(codec, - "realtek: Enable default setup for auto mode as fallback\n"); - spec->init_amp = ALC_INIT_DEFAULT; + if (spec->init_amp == ALC_INIT_UNDEFINED) { + codec_dbg(codec, + "realtek: Enable default setup for auto mode as fallback\n"); + spec->init_amp = ALC_INIT_DEFAULT; + } } } -- 2.25.0

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[PATCH] ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
by Alexander Tsoy 18 Apr '20

18 Apr '20

Many Focusrite devices supports a limited set of sample rates per altsetting. This includes interfaces with ADAT ports and interfaces with dedicated feedback channels: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - all Scarlett 3rd gen devices (feedback channels works only up to 96 kHz); - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Tested-by: Alexey Skobkin <skobkin-ru(a)ya.ru> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/format.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50e1874c847c..5ffb457cc88c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -277,6 +277,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, return false; } +/* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) -- 2.25.3

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[PATCH trivial 0/6] Fix misspellings of "Analog Devices"
by Geert Uytterhoeven 18 Apr '20

18 Apr '20

Hi all, In several files the company also known as ADI is spelled as "Analog Device". However, according to https://www.analog.com/, the company name is spelled "Analog Devices". Hence this patch series, one per subsystem, fixes these misspellings. Thanks for your comments! Geert Uytterhoeven (6): dt-bindings: Fix misspellings of "Analog Devices" dma: Fix misspelling of "Analog Devices" drm: Fix misspellings of "Analog Devices" iio: Fix misspellings of "Analog Devices" ALSA: Fix misspellings of "Analog Devices" ASoC: Fix misspellings of "Analog Devices" .../devicetree/bindings/display/bridge/adi,adv7123.txt | 4 ++-- .../devicetree/bindings/display/bridge/adi,adv7511.txt | 4 ++-- Documentation/devicetree/bindings/dma/adi,axi-dmac.txt | 2 +- Documentation/devicetree/bindings/iio/dac/ad5755.txt | 2 +- drivers/dma/Kconfig | 2 +- drivers/gpu/drm/bridge/adv7511/Kconfig | 2 +- drivers/gpu/drm/drm_fb_cma_helper.c | 2 +- drivers/gpu/drm/tegra/fb.c | 2 +- drivers/iio/adc/ad7791.c | 2 +- drivers/iio/trigger/iio-trig-hrtimer.c | 2 +- drivers/staging/iio/Documentation/overview.txt | 2 +- sound/isa/ad1816a/ad1816a.c | 2 +- sound/pci/ac97/ac97_patch.c | 2 +- sound/pci/hda/Kconfig | 4 ++-- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 2 +- sound/soc/codecs/wm8782.c | 2 +- 17 files changed, 20 insertions(+), 20 deletions(-) -- 2.17.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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