November 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [sound:for-linus 5/5] sound/pci//hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'?
by kbuild test robot 13 Nov '18

13 Nov '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: d99501b8575dc1248bacf1b58d2241cb4b265d49 commit: d99501b8575dc1248bacf1b58d2241cb4b265d49 [5/5] ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap() config: sh-allyesconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d99501b8575dc1248bacf1b58d2241cb4b265d49 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci//hda/patch_ca0132.c: In function 'ca0132_free': >> sound/pci//hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'? [-Werror=implicit-function-declaration] pci_iounmap(codec->bus->pci, spec->mem_base); ^~~~~~~~~~~ pcim_iounmap sound/pci//hda/patch_ca0132.c: In function 'patch_ca0132': sound/pci//hda/patch_ca0132.c:8799:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci//hda/patch_ca0132.c:8799:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +8416 sound/pci//hda/patch_ca0132.c 8386 8387 static void ca0132_free(struct hda_codec *codec) 8388 { 8389 struct ca0132_spec *spec = codec->spec; 8390 8391 cancel_delayed_work_sync(&spec->unsol_hp_work); 8392 snd_hda_power_up(codec); 8393 switch (spec->quirk) { 8394 case QUIRK_SBZ: 8395 sbz_exit_chip(codec); 8396 break; 8397 case QUIRK_ZXR: 8398 zxr_exit_chip(codec); 8399 break; 8400 case QUIRK_R3D: 8401 r3d_exit_chip(codec); 8402 break; 8403 case QUIRK_AE5: 8404 ae5_exit_chip(codec); 8405 break; 8406 case QUIRK_R3DI: 8407 r3di_gpio_shutdown(codec); 8408 break; 8409 } 8410 8411 snd_hda_sequence_write(codec, spec->base_exit_verbs); 8412 ca0132_exit_chip(codec); 8413 8414 snd_hda_power_down(codec); 8415 if (spec->mem_base) > 8416 pci_iounmap(codec->bus->pci, spec->mem_base); 8417 kfree(spec->spec_init_verbs); 8418 kfree(codec->spec); 8419 } 8420 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 17/17] ASoc: bcm2835-i2s: Switch to SPDX identifier
by Stefan Wahren 13 Nov '18

13 Nov '18

Adopt the SPDX license identifier headers to ease license compliance management. Cc: Florian Kauer <florian.kauer(a)koalo.de> Signed-off-by: Stefan Wahren <stefan.wahren(a)i2se.com> --- sound/soc/bcm/bcm2835-i2s.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index d5f73a8..0e4a0e9 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -1,3 +1,4 @@ +// SPDX-License-Identifier: GPL-2.0 /* * ALSA SoC I2S Audio Layer for Broadcom BCM2835 SoC * @@ -20,15 +21,6 @@ * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver * Author: Timur Tabi <timur(a)freescale.com> * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. */ #include <linux/bitops.h> -- 2.7.4

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[alsa-devel] snd_pcm_hw_params resets microphone boost
by sonntex 13 Nov '18

13 Nov '18

I have small program that captures audio from a microphone. As my microphone is very sensitive, I prefer to mute the microphone boost before capturing. I open alsamixer and set volume of the microphone boost to zero, save settings and exit. Then I execute my program. I see that it resets volume of the microphone boost in snd_pcm_hw_params. What should I do resolve this problem? Where can I find default value and change it?

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[alsa-devel] Realtek ALC294 Intel Sunrise Point-LP HD Audio driver - sound does not work
by Tomasz Przybył 13 Nov '18

13 Nov '18

Hello Devs Lots of users with Intel Sunrise Point-LP HD Audio driver (ALC294) (snd_hda_intel) have major issue. Not sound at all. Only crackling from external speakers OR silence from internal notebook speakers. Tried fix with extra boot flags does not work at all. Also affected Asus VivoBook R542UF ``` Machine: Type: Laptop System: ASUSTeK product: VivoBook 15_ASUS Laptop X542UF v: 1.0 serial: <filter> Mobo: ASUSTeK model: X542UF v: 1.0 serial: <filter> UEFI: American Megatrends v: X542UF.303 date: 03/15/2018 Audio: Device-1: Intel Sunrise Point-LP HD Audio driver: snd_hda_intel v: kernel bus ID: 00:1f.3 Sound Server: ALSA Kernel: 4.19 ``` ``` aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC294 Analog [ALC294 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` Dmesg parts ``` [ 12.342528] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC294: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker [ 12.342531] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 12.342533] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 12.342535] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 12.342536] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 12.342538] snd_hda_codec_realtek hdaudioC0D0: Mic=0x1b ``` ``` [ 12.634171] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input18 [ 12.634389] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input19 [ 12.634563] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 [ 12.634737] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 [ 12.634905] input: HDA Intel PCH HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input22 [ 12.635045] input: HDA Intel PCH HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input23 ``` Regards Tomasz

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[alsa-devel] [PATCH v5 2/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 13 Nov '18

13 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..135c0826859b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5764,6 +5765,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6594,6 +6611,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.17.2

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[alsa-devel] using the softvol plugin with min_dB > 0
by Guillermo Rodriguez Garcia 13 Nov '18

13 Nov '18

Hi all, I have a board with a tlv320aic3106 codec and a digital mic. The tlv320aic3106 does not support hardware gain for the digital mic, so I am using the softvol plugin to provide this via software. I would like to setup softvol so that it applies a gain in a range that goes from approx +10dB to approx +40dB. The problem is that softvol only accepts negative values for min_dB, i.e. assumes that you *always* want to do attenuation, and that you might or might not want to have gain (max_dB can be zero or negative). Is there any reason for this, or is it just historical ? (I am aware that the original softvol plugin only supported attenuation, and that support for gain was added later) Is it possible to setup softvol to accept a positive min_dB value ? Best regards, Guillermo Rodriguez Garcia guille.rodriguez(a)gmail.com

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[alsa-devel] Help with extplug transfer callback
by GitHub issues - opened 13 Nov '18

13 Nov '18

alsa-project/alsa-lib issue #2 was opened from dchabot: I've tried posting the following at alsa-users mailing list, but keep getting hit with "awaiting moderator approval" (I am registered to the list). I'm trying to create an extplug and having trouble deciphering how the source and destination buffer reading/writing is supposed to be done in the transfer callback. My goal is to create an extplug that takes a stereo data stream, divides that into 6 or 8 output streams, and applies FIR filters to each output stream. With the code I currently have (https://github.com/dchabot/extplug), trying to play a 1kHz tone results in a garbled, noisy mess. There is a tone in there, but it is a mess. I started from the skeleton code here - http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html. Note that the snd_pcm_extplug_create call here should be passing the "slave" variable, not "conf". That tripped me up for the better part of an hour. I based my initial transfer callback (snd_pcm_extplug_callback_t.transfer member) on that found in the speex plugin. Initially I didn't notice that speex operated with single in/out channels, so you can imagine how that went. Looking at another example of an extplug (https://github.com/dpapavas/alsaloudness), I noted that I should be iterating over each channel for src_areas and dst_areas. I just cannot see how this should be done properly. Is documentation for this critical step missing or simply hard to find. Seems kinda important to know how buffer operations should be done... At this point, all I want to do is copy input channels to output channels. I'm grappling with the fact that this is more complicated than I initially thought, or perhaps more difficult than it should be. My transfer callback currently looks like this: ``` /* copied from pcm-plugins/speex/pcm-speex.c */ static inline void* area_addr(const snd_pcm_channel_area_t *area, snd_pcm_uframes_t offset) { unsigned int bitofs = area->first + area->step * offset; return (char *) area->addr + bitofs / 8; } static snd_pcm_sframes_t transfer_callback(snd_pcm_extplug_t *ext, const snd_pcm_channel_area_t *dst_areas, snd_pcm_uframes_t dst_offset, const snd_pcm_channel_area_t *src_areas, snd_pcm_uframes_t src_offset, snd_pcm_uframes_t size) { int C = ext->channels; short *src, *dst; for(int i = 0; i < C; i++) { src = area_addr(&src_areas[i], src_offset); dst = area_addr(&dst_areas[i], dst_offset); memcpy(dst, src, size); } return size; } ``` Need a little help please. Issue URL : https://github.com/alsa-project/alsa-lib/issues/2 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH 1/4] README: update Vinod's email
by Vinod Koul 13 Nov '18

13 Nov '18

Update email address in README from old invalid id Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- README | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/README b/README index 689ca75e07fd..8ea6c26f5381 100644 --- a/README +++ b/README @@ -1,7 +1,7 @@ README for tinycompress ======================= - vinod.koul(a)linux.intel.com - ========================== + vkoul(a)kernel.org + ================ 1. WHAT tinycompress is a userspace library for anyone who wants to use the ALSA -- 2.14.5

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[alsa-devel] [PATCH] tinycompress: Update headers to include AUDIOCODEC_BESPOKE
by Charles Keepax 13 Nov '18

13 Nov '18

commit dbb6b94339e8 ("ALSA: compress: Add SND_AUDIOCODEC_BESPOKE") added an additional CODEC type for bespoke CODECs. Update the local headers to include this. Reported-by: Andrew Ford <andrew.ford(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/compress_params.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h index 9802292..7286556 100644 --- a/include/sound/compress_params.h +++ b/include/sound/compress_params.h @@ -38,8 +38,9 @@ #define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) +#define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE #define SND_AUDIOPROFILE_PCM ((__u32) 0x00000001) #define SND_AUDIOCHANMODE_MP3_MONO ((__u32) 0x00000001) -- 2.11.0

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[alsa-devel] [PATCH - dmix v5 0/1] pcm: dmix: Add option to allow alignment of slave pointers
by twischer＠de.adit-jv.com 13 Nov '18

13 Nov '18

On 11/12/18 17:20, Takashi Iwai wrote: > Please provide your own sign-off when you submitting a patch, too. > > The subject line should be adjusted better to match with the change; > it's about adding an option to allow alignment of hwptr reporting. > > About the changes: > the term "hw_ptr_filter" might sound confusing. IMO, this is about > the (hw)ptr alignment, not about filtering. > Also, a too long option string can easily lead to a misspell :) > > After all, we do either "no" alignment, "roundup" or "rounddown", or > "auto" choice. > > > thanks, > > Takashi Subject changed. Sign-off added. Option changed to hw_ptr_alignment. Option values changed to no, roundup, rounddown, auto Best regards Timo

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